BASELINE: Update Chromium to 52.0.2743.76 and Ninja to 1.7.1

Change-Id: I382f51b959689505a60f8b707255ecb344f7d8b4
Reviewed-by: Michael Brüning <michael.bruning@qt.io>

1493 files changed, 40677 insertions, 44538 deletions

diff --git a/chromium/third_party/webrtc/BUILD.gn b/chromium/third_party/webrtc/BUILD.gn
index 9f4fd5c7a66..01fa042af36 100644
--- a/chromium/third_party/webrtc/BUILD.gn
+++ b/chromium/third_party/webrtc/BUILD.gn
@@ -114,10 +114,18 @@ config("common_config") {
     }
 
     if (is_clang) {
-      cflags += [ "-Wthread-safety" ]
+      cflags += [
+        "-Wimplicit-fallthrough",
+        "-Wthread-safety",
+        "-Winconsistent-missing-override",
+      ]
     }
   }
 
+  if (rtc_enable_libevent) {
+    defines += [ "WEBRTC_BUILD_LIBEVENT" ]
+  }
+
   if (current_cpu == "arm64") {
     defines += [ "WEBRTC_ARCH_ARM64" ]
     defines += [ "WEBRTC_HAS_NEON" ]
@@ -129,8 +137,6 @@ config("common_config") {
       defines += [ "WEBRTC_ARCH_ARM_V7" ]
       if (arm_use_neon) {
         defines += [ "WEBRTC_HAS_NEON" ]
-      } else if (arm_optionally_use_neon) {
-        defines += [ "WEBRTC_DETECT_NEON" ]
       }
     }
   }
@@ -169,7 +175,6 @@ source_set("webrtc") {
   sources = [
     "call.h",
     "config.h",
-    "frame_callback.h",
     "transport.h",
   ]
 
@@ -202,10 +207,7 @@ source_set("webrtc") {
   ]
 
   if (build_with_chromium) {
-    deps += [
-      "modules/video_capture",
-      "modules/video_render",
-    ]
+    deps += [ "modules/video_capture" ]
   }
 
   if (rtc_enable_protobuf) {
@@ -220,7 +222,6 @@ if (!build_with_chromium) {
     deps = [
       ":webrtc",
       "modules/video_capture:video_capture_internal_impl",
-      "modules/video_render:video_render_internal_impl",
       "test",
     ]
   }
@@ -260,6 +261,8 @@ source_set("rtc_event_log") {
   sources = [
     "call/rtc_event_log.cc",
     "call/rtc_event_log.h",
+    "call/rtc_event_log_helper_thread.cc",
+    "call/rtc_event_log_helper_thread.h",
   ]
 
   defines = []
@@ -281,6 +284,29 @@ source_set("rtc_event_log") {
   }
 }
 
+if (rtc_enable_protobuf) {
+  source_set("rtc_event_log_parser") {
+    sources = [
+      "call/rtc_event_log_parser.cc",
+      "call/rtc_event_log_parser.h",
+    ]
+
+    configs += [ ":common_config" ]
+    public_configs = [ ":common_inherited_config" ]
+
+    deps = [
+      ":rtc_event_log_proto",
+      ":webrtc_common",
+    ]
+
+    if (is_clang && !is_nacl) {
+      # Suppress warnings from Chrome's Clang plugins.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}
+
 if (use_libfuzzer || use_drfuzz) {
   # This target is only here for gn to discover fuzzer build targets under
   # webrtc/test/fuzzers/.
diff --git a/chromium/third_party/webrtc/DEPS b/chromium/third_party/webrtc/DEPS
index 7f15817e370..40ec7a3d622 100644
--- a/chromium/third_party/webrtc/DEPS
+++ b/chromium/third_party/webrtc/DEPS
@@ -19,8 +19,6 @@ include_rules = [
   "+webrtc/common_types.h",
   "+webrtc/config.h",
   "+webrtc/engine_configurations.h",
-  "+webrtc/frame_callback.h",
-  "+webrtc/stream.h",
   "+webrtc/transport.h",
   "+webrtc/typedefs.h",
   "+webrtc/video_decoder.h",
@@ -30,6 +28,7 @@ include_rules = [
   "+webrtc/video_renderer.h",
   "+webrtc/video_send_stream.h",
 
+  "+WebRTC",
   "+webrtc/base",
   "+webrtc/modules/include",
   "+webrtc/test",
@@ -45,9 +44,11 @@ specific_include_rules = {
     "+webrtc/common_video",
   ],
   "video_receive_stream\.h": [
+    "+webrtc/common_video/include",
     "+webrtc/media/base",
   ],
   "video_send_stream\.h": [
+    "+webrtc/common_video/include",
     "+webrtc/media/base",
   ],
 }
diff --git a/chromium/third_party/webrtc/OWNERS b/chromium/third_party/webrtc/OWNERS
index 057b5636e86..613031f5cd4 100644
--- a/chromium/third_party/webrtc/OWNERS
+++ b/chromium/third_party/webrtc/OWNERS
@@ -1,3 +1,9 @@
+henrika@webrtc.org
+mflodman@webrtc.org
+niklas.enbom@webrtc.org
+tina.legrand@webrtc.org
+tommi@webrtc.org
+
 per-file *.isolate=kjellander@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
diff --git a/chromium/third_party/webrtc/api/BUILD.gn b/chromium/third_party/webrtc/api/BUILD.gn
index 784c048905d..6dc52174ece 100644
--- a/chromium/third_party/webrtc/api/BUILD.gn
+++ b/chromium/third_party/webrtc/api/BUILD.gn
@@ -7,104 +7,3 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../build/webrtc.gni")
-
-config("ios_config") {
-  libs = [
-    "CoreGraphics.framework",
-    "GLKit.framework",
-    "OpenGLES.framework",
-    "QuartzCore.framework",
-  ]
-}
-
-if (is_ios) {
-  source_set("rtc_api_objc") {
-    deps = [
-      "../base:rtc_base_objc",
-      #"../../talk/libjingle:libjingle_peerconnection",
-    ]
-    cflags = [
-      "-fobjc-arc",
-      "-Wobjc-missing-property-synthesis",
-    ]
-    sources = [
-      # Add these when there's a BUILD.gn for peer connection APIs
-      #"objc/RTCAVFoundationVideoSource+Private.h",
-      #"objc/RTCAVFoundationVideoSource.h",
-      #"objc/RTCAVFoundationVideoSource.mm",
-      #"objc/RTCAudioTrack+Private.h",
-      #"objc/RTCAudioTrack.h",
-      #"objc/RTCAudioTrack.mm",
-      #"objc/RTCConfiguration+Private.h",
-      #"objc/RTCConfiguration.h",
-      #"objc/RTCConfiguration.mm",
-      #"objc/RTCDataChannel+Private.h",
-      #"objc/RTCDataChannel.h",
-      #"objc/RTCDataChannel.mm",
-      #"objc/RTCDataChannelConfiguration+Private.h",
-      #"objc/RTCDataChannelConfiguration.h",
-      #"objc/RTCDataChannelConfiguration.mm",
-      #"objc/RTCIceCandidate+Private.h",
-      #"objc/RTCIceCandidate.h",
-      #"objc/RTCIceCandidate.mm",
-      #"objc/RTCMediaStream+Private.h",
-      #"objc/RTCMediaStream.h",
-      #"objc/RTCMediaStream.mm",
-      #"objc/RTCMediaStreamTrack+Private.h",
-      #"objc/RTCMediaStreamTrack.h",
-      #"objc/RTCMediaStreamTrack.mm",
-      #"objc/RTCPeerConnection+DataChannel.h",
-      #"objc/RTCPeerConnection+Private.h",
-      #"objc/RTCPeerConnection+Stats.h",
-      #"objc/RTCPeerConnection.h",
-      #"objc/RTCPeerConnection.mm",
-      #"objc/RTCPeerConnectionFactory+Private.h",
-      #"objc/RTCPeerConnectionFactory.h",
-      #"objc/RTCPeerConnectionFactory.mm",
-      #"objc/RTCVideoSource+Private.h",
-      #"objc/RTCVideoSource.h",
-      #"objc/RTCVideoSource.mm",
-      #"objc/RTCVideoTrack+Private.h",
-      #"objc/RTCVideoTrack.h",
-      #"objc/RTCVideoTrack.mm",
-      "objc/RTCIceServer+Private.h",
-      "objc/RTCIceServer.h",
-      "objc/RTCIceServer.mm",
-      "objc/RTCMediaConstraints+Private.h",
-      "objc/RTCMediaConstraints.h",
-      "objc/RTCMediaConstraints.mm",
-      "objc/RTCOpenGLVideoRenderer.h",
-      "objc/RTCOpenGLVideoRenderer.mm",
-      "objc/RTCSessionDescription+Private.h",
-      "objc/RTCSessionDescription.h",
-      "objc/RTCSessionDescription.mm",
-      "objc/RTCStatsReport+Private.h",
-      "objc/RTCStatsReport.h",
-      "objc/RTCStatsReport.mm",
-      "objc/RTCVideoFrame+Private.h",
-      "objc/RTCVideoFrame.h",
-      "objc/RTCVideoFrame.mm",
-      "objc/RTCVideoRenderer.h",
-      "objc/RTCVideoRendererAdapter+Private.h",
-      "objc/RTCVideoRendererAdapter.h",
-      "objc/RTCVideoRendererAdapter.mm",
-      "objc/WebRTC-Prefix.pch",
-      "objc/avfoundationvideocapturer.h",
-      "objc/avfoundationvideocapturer.mm",
-    ]
-
-    if (is_ios) {
-      sources += [
-        "objc/RTCEAGLVideoView.h",
-        "objc/RTCEAGLVideoView.m",
-      ]
-    }
-
-    if (is_mac) {
-      sources += [
-        "objc/RTCNSGLVideoView.h",
-        "objc/RTCNSGLVideoView.m",
-      ]
-    }
-  }
-}
diff --git a/chromium/third_party/webrtc/api/androidvideocapturer.cc b/chromium/third_party/webrtc/api/androidvideocapturer.cc
index e98a4be573d..71a94fedb06 100644
--- a/chromium/third_party/webrtc/api/androidvideocapturer.cc
+++ b/chromium/third_party/webrtc/api/androidvideocapturer.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/api/androidvideocapturer.h"
 
+#include <memory>
+
 #include "webrtc/api/java/jni/native_handle_impl.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/timeutils.h"
@@ -70,11 +72,11 @@ class AndroidVideoCapturer::FrameFactory : public cricket::VideoFrameFactory {
     RTC_CHECK(captured_frame == &captured_frame_);
     RTC_CHECK(buffer_->native_handle() == nullptr);
 
-    rtc::scoped_ptr<cricket::VideoFrame> frame(new cricket::WebRtcVideoFrame(
+    std::unique_ptr<cricket::VideoFrame> frame(new cricket::WebRtcVideoFrame(
         ShallowCenterCrop(buffer_, dst_width, dst_height),
         captured_frame->time_stamp, captured_frame->rotation));
     // Caller takes ownership.
-    // TODO(magjed): Change CreateAliasedFrame() to return a rtc::scoped_ptr.
+    // TODO(magjed): Change CreateAliasedFrame() to return a std::unique_ptr.
     return apply_rotation_ ? frame->GetCopyWithRotationApplied()->Copy()
                            : frame.release();
   }
@@ -86,14 +88,12 @@ class AndroidVideoCapturer::FrameFactory : public cricket::VideoFrameFactory {
       int output_width,
       int output_height) const override {
     if (buffer_->native_handle() != nullptr) {
-      // TODO(perkj) Implement cropping.
-      RTC_CHECK_EQ(cropped_input_width, buffer_->width());
-      RTC_CHECK_EQ(cropped_input_height, buffer_->height());
       rtc::scoped_refptr<webrtc::VideoFrameBuffer> scaled_buffer(
           static_cast<webrtc_jni::AndroidTextureBuffer*>(buffer_.get())
-              ->ScaleAndRotate(output_width, output_height,
-                               apply_rotation_ ? input_frame->rotation :
-                                   webrtc::kVideoRotation_0));
+              ->CropScaleAndRotate(cropped_input_width, cropped_input_height,
+                                   output_width, output_height,
+                                   apply_rotation_ ? input_frame->rotation
+                                                   : webrtc::kVideoRotation_0));
       return new cricket::WebRtcVideoFrame(
           scaled_buffer, input_frame->time_stamp,
           apply_rotation_ ? webrtc::kVideoRotation_0 : input_frame->rotation);
diff --git a/chromium/third_party/webrtc/api/api.gyp b/chromium/third_party/webrtc/api/api.gyp
index aa6ef38e5ef..377ef8f70be 100644
--- a/chromium/third_party/webrtc/api/api.gyp
+++ b/chromium/third_party/webrtc/api/api.gyp
@@ -66,7 +66,6 @@
             '-Wextra',
           ],
           'cflags_cc!': [
-            '-Wnon-virtual-dtor',
             '-Woverloaded-virtual',
           ],
           'msvs_disabled_warnings': [
@@ -115,139 +114,13 @@
               'java/android',
               '<(webrtc_base_dir)/java/src',
               '<(webrtc_modules_dir)/audio_device/android/java/src',
-              '<(webrtc_modules_dir)/video_render/android/java/src',
+
             ],
           },
           'includes': ['../../build/java.gypi'],
         }, # libjingle_peerconnection_java
       ]
     }],
-    ['OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")', {
-      'targets': [
-        {
-          'target_name': 'rtc_api_objc',
-          'type': 'static_library',
-          'includes': [
-            '../build/objc_common.gypi',
-          ],
-          'dependencies': [
-            '<(webrtc_root)/base/base.gyp:rtc_base_objc',
-            'libjingle_peerconnection',
-          ],
-          'sources': [
-            'objc/RTCAudioTrack+Private.h',
-            'objc/RTCAudioTrack.h',
-            'objc/RTCAudioTrack.mm',
-            'objc/RTCConfiguration+Private.h',
-            'objc/RTCConfiguration.h',
-            'objc/RTCConfiguration.mm',
-            'objc/RTCDataChannel+Private.h',
-            'objc/RTCDataChannel.h',
-            'objc/RTCDataChannel.mm',
-            'objc/RTCDataChannelConfiguration+Private.h',
-            'objc/RTCDataChannelConfiguration.h',
-            'objc/RTCDataChannelConfiguration.mm',
-            'objc/RTCIceCandidate+Private.h',
-            'objc/RTCIceCandidate.h',
-            'objc/RTCIceCandidate.mm',
-            'objc/RTCIceServer+Private.h',
-            'objc/RTCIceServer.h',
-            'objc/RTCIceServer.mm',
-            'objc/RTCMediaConstraints+Private.h',
-            'objc/RTCMediaConstraints.h',
-            'objc/RTCMediaConstraints.mm',
-            'objc/RTCMediaStream+Private.h',
-            'objc/RTCMediaStream.h',
-            'objc/RTCMediaStream.mm',
-            'objc/RTCMediaStreamTrack+Private.h',
-            'objc/RTCMediaStreamTrack.h',
-            'objc/RTCMediaStreamTrack.mm',
-            'objc/RTCOpenGLVideoRenderer.h',
-            'objc/RTCOpenGLVideoRenderer.mm',
-            'objc/RTCPeerConnection+DataChannel.mm',
-            'objc/RTCPeerConnection+Private.h',
-            'objc/RTCPeerConnection+Stats.mm',
-            'objc/RTCPeerConnection.h',
-            'objc/RTCPeerConnection.mm',
-            'objc/RTCPeerConnectionFactory+Private.h',
-            'objc/RTCPeerConnectionFactory.h',
-            'objc/RTCPeerConnectionFactory.mm',
-            'objc/RTCSessionDescription+Private.h',
-            'objc/RTCSessionDescription.h',
-            'objc/RTCSessionDescription.mm',
-            'objc/RTCStatsReport+Private.h',
-            'objc/RTCStatsReport.h',
-            'objc/RTCStatsReport.mm',
-            'objc/RTCVideoFrame+Private.h',
-            'objc/RTCVideoFrame.h',
-            'objc/RTCVideoFrame.mm',
-            'objc/RTCVideoRenderer.h',
-            'objc/RTCVideoRendererAdapter+Private.h',
-            'objc/RTCVideoRendererAdapter.h',
-            'objc/RTCVideoRendererAdapter.mm',
-            'objc/RTCVideoSource+Private.h',
-            'objc/RTCVideoSource.h',
-            'objc/RTCVideoSource.mm',
-            'objc/RTCVideoTrack+Private.h',
-            'objc/RTCVideoTrack.h',
-            'objc/RTCVideoTrack.mm',
-          ],
-          # TODO(hjon): Make this compile without linking to libstdc++
-          # See https://bugs.chromium.org/p/webrtc/issues/detail?id=5593
-          'link_settings': {
-            'libraries': [
-              '-lstdc++',
-            ],
-          },
-          'conditions': [
-            ['OS=="ios"', {
-              'sources': [
-                'objc/RTCAVFoundationVideoSource+Private.h',
-                'objc/RTCAVFoundationVideoSource.h',
-                'objc/RTCAVFoundationVideoSource.mm',
-                'objc/RTCEAGLVideoView.h',
-                'objc/RTCEAGLVideoView.m',
-                'objc/avfoundationvideocapturer.h',
-                'objc/avfoundationvideocapturer.mm',
-              ],
-              'all_dependent_settings': {
-                'xcode_settings': {
-                  'OTHER_LDFLAGS': [
-                    '-framework CoreGraphics',
-                    '-framework GLKit',
-                    '-framework OpenGLES',
-                    '-framework QuartzCore',
-                  ]
-                }
-              },
-              # TODO(kjellander): Make the code compile without disabling these.
-              # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
-              'cflags': [
-                '-Wno-return-type',
-              ],
-              'xcode_settings': {
-                'WARNING_CFLAGS': [
-                  '-Wno-return-type',
-                ],
-              },
-            }],
-            ['OS=="mac"', {
-              'sources': [
-                'objc/RTCNSGLVideoView.h',
-                'objc/RTCNSGLVideoView.m',
-              ],
-              'link_settings': {
-                'xcode_settings': {
-                  'OTHER_LDFLAGS': [
-                    '-framework OpenGL',
-                  ],
-                },
-              },
-            }],
-          ],
-        }
-      ],
-    }],  # OS=="ios"
   ],  # conditions
   'targets': [
     {
@@ -333,7 +206,6 @@
         '-Wno-sign-compare',
       ],
       'cflags_cc!': [
-        '-Wnon-virtual-dtor',
         '-Woverloaded-virtual',
       ],
       'conditions': [
@@ -366,6 +238,20 @@
             },
           },
         }],
+        ['use_quic==1', {
+          'dependencies': [
+            '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+          ],
+          'sources': [
+            'quicdatachannel.cc',
+            'quicdatachannel.h',
+            'quicdatatransport.cc',
+            'quicdatatransport.h',
+          ],
+          'export_dependent_settings': [
+            '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+          ],
+        }],
       ],
     },  # target libjingle_peerconnection
   ],  # targets
diff --git a/chromium/third_party/webrtc/api/api_tests.gyp b/chromium/third_party/webrtc/api/api_tests.gyp
index b7cbd687ad9..65bb4614744 100644
--- a/chromium/third_party/webrtc/api/api_tests.gyp
+++ b/chromium/third_party/webrtc/api/api_tests.gyp
@@ -43,7 +43,7 @@
         'peerconnectionendtoend_unittest.cc',
         'peerconnectionfactory_unittest.cc',
         'peerconnectioninterface_unittest.cc',
-        # 'peerconnectionproxy_unittest.cc',
+        'proxy_unittest.cc',
         'rtpsenderreceiver_unittest.cc',
         'statscollector_unittest.cc',
         'test/fakeaudiocapturemodule.cc',
@@ -72,7 +72,6 @@
         '-Wextra',
       ],
       'cflags_cc!': [
-        '-Wnon-virtual-dtor',
         '-Woverloaded-virtual',
       ],
       'msvs_disabled_warnings': [
@@ -113,6 +112,18 @@
             },
           },
         }],
+        ['use_quic==1', {
+          'dependencies': [
+            '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+          ],
+          'sources': [
+            'quicdatachannel_unittest.cc',
+            'quicdatatransport_unittest.cc',
+          ],
+          'export_dependent_settings': [
+            '<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
+          ],
+        }],
       ],  # conditions
     },  # target peerconnection_unittests
   ],  # targets
@@ -131,41 +142,17 @@
             'resource_dir': 'androidtests/res',
             'native_lib_target': 'libjingle_peerconnection_so',
             'is_test_apk': 1,
+            'test_type': 'instrumentation',
+            'tested_apk_path': '',
             'never_lint': 1,
           },
-          'includes': [ '../../build/java_apk.gypi' ],
-        },
-      ],  # targets
-    }],  # OS=="android"
-    ['OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")', {
-      'targets': [
-        {
-          'target_name': 'rtc_api_objc_tests',
-          'type': 'executable',
           'includes': [
-            '../build/objc_common.gypi',
+            '../../build/java_apk.gypi',
+            '../../build/android/test_runner.gypi',
           ],
-          'dependencies': [
-            '<(webrtc_root)/api/api.gyp:rtc_api_objc',
-            '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
-          ],
-          'sources': [
-            'objctests/RTCConfigurationTest.mm',
-            'objctests/RTCDataChannelConfigurationTest.mm',
-            'objctests/RTCIceCandidateTest.mm',
-            'objctests/RTCIceServerTest.mm',
-            'objctests/RTCMediaConstraintsTest.mm',
-            'objctests/RTCSessionDescriptionTest.mm',
-          ],
-          'xcode_settings': {
-            # |-ObjC| flag needed to make sure category method implementations
-            # are included:
-            # https://developer.apple.com/library/mac/qa/qa1490/_index.html
-            'OTHER_LDFLAGS': ['-ObjC'],
-          },
         },
-      ],
-    }],  # OS=="ios"
+      ],  # targets
+    }],  # OS=="android"
     ['OS=="android"', {
       'targets': [
         {
@@ -176,6 +163,27 @@
           ],
         },
       ],
+      'conditions': [
+        ['test_isolation_mode != "noop"',
+          {
+            'targets': [
+              {
+                'target_name': 'peerconnection_unittests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):peerconnection_unittests_apk',
+                ],
+                'includes': [
+                  '../build/isolate.gypi',
+                ],
+                'sources': [
+                  'peerconnection_unittests_apk.isolate',
+                ],
+              },
+            ]
+          }
+        ],
+      ],
     }],  # OS=="android"
     ['test_isolation_mode != "noop"', {
       'targets': [
diff --git a/chromium/third_party/webrtc/api/audiotrack.h b/chromium/third_party/webrtc/api/audiotrack.h
index e7ffee5d830..096caf9d0e0 100644
--- a/chromium/third_party/webrtc/api/audiotrack.h
+++ b/chromium/third_party/webrtc/api/audiotrack.h
@@ -16,7 +16,7 @@
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/mediastreamtrack.h"
 #include "webrtc/api/notifier.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread_checker.h"
 
diff --git a/chromium/third_party/webrtc/api/datachannel.cc b/chromium/third_party/webrtc/api/datachannel.cc
index 612d7e0b382..452e4b39611 100644
--- a/chromium/third_party/webrtc/api/datachannel.cc
+++ b/chromium/third_party/webrtc/api/datachannel.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/api/datachannel.h"
 
+#include <memory>
 #include <string>
 
 #include "webrtc/api/mediastreamprovider.h"
@@ -363,7 +364,7 @@ void DataChannel::OnDataReceived(cricket::DataChannel* channel,
   }
 
   bool binary = (params.type == cricket::DMT_BINARY);
-  rtc::scoped_ptr<DataBuffer> buffer(new DataBuffer(payload, binary));
+  std::unique_ptr<DataBuffer> buffer(new DataBuffer(payload, binary));
   if (state_ == kOpen && observer_) {
     observer_->OnMessage(*buffer.get());
   } else {
@@ -494,7 +495,7 @@ void DataChannel::DeliverQueuedReceivedData() {
   }
 
   while (!queued_received_data_.Empty()) {
-    rtc::scoped_ptr<DataBuffer> buffer(queued_received_data_.Front());
+    std::unique_ptr<DataBuffer> buffer(queued_received_data_.Front());
     observer_->OnMessage(*buffer);
     queued_received_data_.Pop();
   }
@@ -589,7 +590,7 @@ void DataChannel::SendQueuedControlMessages() {
   control_packets.Swap(&queued_control_data_);
 
   while (!control_packets.Empty()) {
-    rtc::scoped_ptr<DataBuffer> buf(control_packets.Front());
+    std::unique_ptr<DataBuffer> buf(control_packets.Front());
     SendControlMessage(buf->data);
     control_packets.Pop();
   }
diff --git a/chromium/third_party/webrtc/api/datachannel.h b/chromium/third_party/webrtc/api/datachannel.h
index b8830be300a..3fb400b7ec4 100644
--- a/chromium/third_party/webrtc/api/datachannel.h
+++ b/chromium/third_party/webrtc/api/datachannel.h
@@ -260,7 +260,7 @@ class DataChannel : public DataChannelInterface,
 };
 
 // Define proxy for DataChannelInterface.
-BEGIN_PROXY_MAP(DataChannel)
+BEGIN_SIGNALING_PROXY_MAP(DataChannel)
   PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
   PROXY_METHOD0(void, UnregisterObserver)
   PROXY_CONSTMETHOD0(std::string, label)
@@ -275,7 +275,7 @@ BEGIN_PROXY_MAP(DataChannel)
   PROXY_CONSTMETHOD0(uint64_t, buffered_amount)
   PROXY_METHOD0(void, Close)
   PROXY_METHOD1(bool, Send, const DataBuffer&)
-END_PROXY()
+END_SIGNALING_PROXY()
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/datachannel_unittest.cc b/chromium/third_party/webrtc/api/datachannel_unittest.cc
index 5958ec02f85..d55ab57b64d 100644
--- a/chromium/third_party/webrtc/api/datachannel_unittest.cc
+++ b/chromium/third_party/webrtc/api/datachannel_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/api/datachannel.h"
 #include "webrtc/api/sctputils.h"
 #include "webrtc/api/test/fakedatachannelprovider.h"
@@ -85,7 +87,7 @@ class SctpDataChannelTest : public testing::Test {
 
   webrtc::InternalDataChannelInit init_;
   FakeDataChannelProvider provider_;
-  rtc::scoped_ptr<FakeDataChannelObserver> observer_;
+  std::unique_ptr<FakeDataChannelObserver> observer_;
   rtc::scoped_refptr<DataChannel> webrtc_data_channel_;
 };
 
diff --git a/chromium/third_party/webrtc/api/dtlsidentitystore.cc b/chromium/third_party/webrtc/api/dtlsidentitystore.cc
index a4851880650..bdccc10c1b0 100644
--- a/chromium/third_party/webrtc/api/dtlsidentitystore.cc
+++ b/chromium/third_party/webrtc/api/dtlsidentitystore.cc
@@ -51,7 +51,7 @@ class DtlsIdentityStoreImpl::WorkerTask : public sigslot::has_slots<>,
  private:
   void GenerateIdentity_w() {
     LOG(LS_INFO) << "Generating identity, using keytype " << key_type_;
-    rtc::scoped_ptr<rtc::SSLIdentity> identity(
+    std::unique_ptr<rtc::SSLIdentity> identity(
         rtc::SSLIdentity::Generate(kIdentityName, key_type_));
 
     // Posting to |this| avoids touching |store_| on threads other than
@@ -74,7 +74,7 @@ class DtlsIdentityStoreImpl::WorkerTask : public sigslot::has_slots<>,
       case MSG_GENERATE_IDENTITY_RESULT:
         RTC_DCHECK(signaling_thread_->IsCurrent());
         {
-          rtc::scoped_ptr<IdentityResultMessageData> pdata(
+          std::unique_ptr<IdentityResultMessageData> pdata(
               static_cast<IdentityResultMessageData*>(msg->pdata));
           if (store_) {
             store_->OnIdentityGenerated(pdata->data()->key_type_,
@@ -108,12 +108,6 @@ DtlsIdentityStoreImpl::DtlsIdentityStoreImpl(rtc::Thread* signaling_thread,
       worker_thread_(worker_thread),
       request_info_() {
   RTC_DCHECK(signaling_thread_->IsCurrent());
-  // Preemptively generate identities unless the worker thread and signaling
-  // thread are the same (only do preemptive work in the background).
-  if (worker_thread_ != signaling_thread_) {
-    // Only necessary for RSA.
-    GenerateIdentity(rtc::KT_RSA, nullptr);
-  }
 }
 
 DtlsIdentityStoreImpl::~DtlsIdentityStoreImpl() {
@@ -137,7 +131,7 @@ void DtlsIdentityStoreImpl::OnMessage(rtc::Message* msg) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
   switch (msg->message_id) {
     case MSG_GENERATE_IDENTITY_RESULT: {
-      rtc::scoped_ptr<IdentityResultMessageData> pdata(
+      std::unique_ptr<IdentityResultMessageData> pdata(
           static_cast<IdentityResultMessageData*>(msg->pdata));
       OnIdentityGenerated(pdata->data()->key_type_,
                           std::move(pdata->data()->identity_));
@@ -192,7 +186,8 @@ void DtlsIdentityStoreImpl::GenerateIdentity(
 }
 
 void DtlsIdentityStoreImpl::OnIdentityGenerated(
-    rtc::KeyType key_type, rtc::scoped_ptr<rtc::SSLIdentity> identity) {
+    rtc::KeyType key_type,
+    std::unique_ptr<rtc::SSLIdentity> identity) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
 
   RTC_DCHECK(request_info_[key_type].gen_in_progress_counts_);
@@ -226,7 +221,7 @@ void DtlsIdentityStoreImpl::OnIdentityGenerated(
     if (worker_thread_ != signaling_thread_ && // Only do in background thread.
         key_type == rtc::KT_RSA &&             // Only necessary for RSA.
         !request_info_[key_type].free_identity_.get() &&
-        request_info_[key_type].request_observers_.size() <=
+        request_info_[key_type].request_observers_.size() ==
             request_info_[key_type].gen_in_progress_counts_) {
       GenerateIdentity(key_type, nullptr);
     }
diff --git a/chromium/third_party/webrtc/api/dtlsidentitystore.h b/chromium/third_party/webrtc/api/dtlsidentitystore.h
index af4229201cb..e25b79591a7 100644
--- a/chromium/third_party/webrtc/api/dtlsidentitystore.h
+++ b/chromium/third_party/webrtc/api/dtlsidentitystore.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_API_DTLSIDENTITYSTORE_H_
 #define WEBRTC_API_DTLSIDENTITYSTORE_H_
 
+#include <memory>
 #include <queue>
 #include <string>
 #include <utility>
@@ -19,7 +20,6 @@
 #include "webrtc/base/messagequeue.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
@@ -39,9 +39,9 @@ class DtlsIdentityRequestObserver : public rtc::RefCountInterface {
   // TODO(hbos): Unify the OnSuccess method once Chrome code is updated.
   virtual void OnSuccess(const std::string& der_cert,
                          const std::string& der_private_key) = 0;
-  // |identity| is a scoped_ptr because rtc::SSLIdentity is not copyable and the
+  // |identity| is a unique_ptr because rtc::SSLIdentity is not copyable and the
   // client has to get the ownership of the object to make use of it.
-  virtual void OnSuccess(rtc::scoped_ptr<rtc::SSLIdentity> identity) = 0;
+  virtual void OnSuccess(std::unique_ptr<rtc::SSLIdentity> identity) = 0;
 
  protected:
   virtual ~DtlsIdentityRequestObserver() {}
@@ -57,25 +57,10 @@ class DtlsIdentityStoreInterface {
 
   // The |observer| will be called when the requested identity is ready, or when
   // identity generation fails.
-  // TODO(torbjorng,hbos): There are currently two versions of RequestIdentity,
-  // with default implementation to call the other version of itself (so that a
-  // call can be made regardless of which version has been overridden). The 1st
-  // version exists because it is currently implemented in chromium. The 2nd
-  // version will become the one and only RequestIdentity as soon as chromium
-  // implements the correct version. crbug.com/544902, webrtc:5092.
-  virtual void RequestIdentity(
-      rtc::KeyParams key_params,
-      const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer) {
-    // Add default ("null") expiration.
-    RequestIdentity(key_params, rtc::Optional<uint64_t>(), observer);
-  }
   virtual void RequestIdentity(
       const rtc::KeyParams& key_params,
       const rtc::Optional<uint64_t>& expires_ms,
-      const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer) {
-    // Drop |expires|.
-    RequestIdentity(key_params, observer);
-  }
+      const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer) = 0;
 };
 
 // The WebRTC default implementation of DtlsIdentityStoreInterface.
@@ -106,7 +91,7 @@ class DtlsIdentityStoreImpl : public DtlsIdentityStoreInterface,
       rtc::KeyType key_type,
       const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);
   void OnIdentityGenerated(rtc::KeyType key_type,
-                           rtc::scoped_ptr<rtc::SSLIdentity> identity);
+                           std::unique_ptr<rtc::SSLIdentity> identity);
 
   class WorkerTask;
   typedef rtc::ScopedMessageData<DtlsIdentityStoreImpl::WorkerTask>
@@ -115,11 +100,11 @@ class DtlsIdentityStoreImpl : public DtlsIdentityStoreInterface,
   // A key type-identity pair.
   struct IdentityResult {
     IdentityResult(rtc::KeyType key_type,
-                   rtc::scoped_ptr<rtc::SSLIdentity> identity)
+                   std::unique_ptr<rtc::SSLIdentity> identity)
         : key_type_(key_type), identity_(std::move(identity)) {}
 
     rtc::KeyType key_type_;
-    rtc::scoped_ptr<rtc::SSLIdentity> identity_;
+    std::unique_ptr<rtc::SSLIdentity> identity_;
   };
 
   typedef rtc::ScopedMessageData<IdentityResult> IdentityResultMessageData;
@@ -139,7 +124,7 @@ class DtlsIdentityStoreImpl : public DtlsIdentityStoreInterface,
     std::queue<rtc::scoped_refptr<DtlsIdentityRequestObserver>>
         request_observers_;
     size_t gen_in_progress_counts_;
-    rtc::scoped_ptr<rtc::SSLIdentity> free_identity_;
+    std::unique_ptr<rtc::SSLIdentity> free_identity_;
   };
 
   // One RequestInfo per KeyType. Only touch on the |signaling_thread_|.
diff --git a/chromium/third_party/webrtc/api/dtlsidentitystore_unittest.cc b/chromium/third_party/webrtc/api/dtlsidentitystore_unittest.cc
index 809e885216c..31f0113f283 100644
--- a/chromium/third_party/webrtc/api/dtlsidentitystore_unittest.cc
+++ b/chromium/third_party/webrtc/api/dtlsidentitystore_unittest.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/api/dtlsidentitystore.h"
 
+#include <memory>
+
 #include "webrtc/api/webrtcsessiondescriptionfactory.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
@@ -34,7 +36,7 @@ class MockDtlsIdentityRequestObserver :
     LOG(LS_WARNING) << "The string version of OnSuccess is called unexpectedly";
     EXPECT_TRUE(false);
   }
-  void OnSuccess(rtc::scoped_ptr<rtc::SSLIdentity> identity) override {
+  void OnSuccess(std::unique_ptr<rtc::SSLIdentity> identity) override {
     EXPECT_FALSE(call_back_called_);
     call_back_called_ = true;
     last_request_success_ = true;
@@ -77,14 +79,12 @@ class DtlsIdentityStoreTest : public testing::Test {
     rtc::CleanupSSL();
   }
 
-  rtc::scoped_ptr<rtc::Thread> worker_thread_;
-  rtc::scoped_ptr<DtlsIdentityStoreImpl> store_;
+  std::unique_ptr<rtc::Thread> worker_thread_;
+  std::unique_ptr<DtlsIdentityStoreImpl> store_;
   rtc::scoped_refptr<MockDtlsIdentityRequestObserver> observer_;
 };
 
 TEST_F(DtlsIdentityStoreTest, RequestIdentitySuccessRSA) {
-  EXPECT_TRUE_WAIT(store_->HasFreeIdentityForTesting(rtc::KT_RSA), kTimeoutMs);
-
   store_->RequestIdentity(rtc::KeyParams(rtc::KT_RSA),
                           rtc::Optional<uint64_t>(),
                           observer_.get());
@@ -103,14 +103,14 @@ TEST_F(DtlsIdentityStoreTest, RequestIdentitySuccessRSA) {
 }
 
 TEST_F(DtlsIdentityStoreTest, RequestIdentitySuccessECDSA) {
-  // Since store currently does not preemptively generate free ECDSA identities
-  // we do not invoke HasFreeIdentityForTesting between requests.
-
   store_->RequestIdentity(rtc::KeyParams(rtc::KT_ECDSA),
                           rtc::Optional<uint64_t>(),
                           observer_.get());
   EXPECT_TRUE_WAIT(observer_->LastRequestSucceeded(), kTimeoutMs);
 
+  // Since store currently does not preemptively generate free ECDSA identities
+  // we do not invoke HasFreeIdentityForTesting between requests.
+
   observer_->Reset();
 
   // Verifies that the callback is async when a free identity is ready.
diff --git a/chromium/third_party/webrtc/api/dtmfsender.h b/chromium/third_party/webrtc/api/dtmfsender.h
index ae8aa445410..c85557f2e15 100644
--- a/chromium/third_party/webrtc/api/dtmfsender.h
+++ b/chromium/third_party/webrtc/api/dtmfsender.h
@@ -17,6 +17,7 @@
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/proxy.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/refcount.h"
 
@@ -82,7 +83,7 @@ class DtmfSender
   DtmfSender();
 
   // Implements MessageHandler.
-  virtual void OnMessage(rtc::Message* msg);
+  void OnMessage(rtc::Message* msg) override;
 
   // The DTMF sending task.
   void DoInsertDtmf();
@@ -103,7 +104,7 @@ class DtmfSender
 };
 
 // Define proxy for DtmfSenderInterface.
-BEGIN_PROXY_MAP(DtmfSender)
+BEGIN_SIGNALING_PROXY_MAP(DtmfSender)
   PROXY_METHOD1(void, RegisterObserver, DtmfSenderObserverInterface*)
   PROXY_METHOD0(void, UnregisterObserver)
   PROXY_METHOD0(bool, CanInsertDtmf)
@@ -112,7 +113,7 @@ BEGIN_PROXY_MAP(DtmfSender)
   PROXY_CONSTMETHOD0(std::string, tones)
   PROXY_CONSTMETHOD0(int, duration)
   PROXY_CONSTMETHOD0(int, inter_tone_gap)
-END_PROXY()
+END_SIGNALING_PROXY()
 
 // Get DTMF code from the DTMF event character.
 bool GetDtmfCode(char tone, int* code);
diff --git a/chromium/third_party/webrtc/api/dtmfsender_unittest.cc b/chromium/third_party/webrtc/api/dtmfsender_unittest.cc
index 0a944cb160d..e5fe26171b5 100644
--- a/chromium/third_party/webrtc/api/dtmfsender_unittest.cc
+++ b/chromium/third_party/webrtc/api/dtmfsender_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/api/dtmfsender.h"
 
+#include <memory>
 #include <set>
 #include <string>
 #include <vector>
@@ -84,9 +85,9 @@ class FakeDtmfProvider : public DtmfProviderInterface {
     // TODO(ronghuawu): Make the timer (basically the rtc::TimeNanos)
     // mockable and use a fake timer in the unit tests.
     if (last_insert_dtmf_call_ > 0) {
-      gap = static_cast<int>(rtc::Time() - last_insert_dtmf_call_);
+      gap = static_cast<int>(rtc::TimeMillis() - last_insert_dtmf_call_);
     }
-    last_insert_dtmf_call_ = rtc::Time();
+    last_insert_dtmf_call_ = rtc::TimeMillis();
 
     LOG(LS_VERBOSE) << "FakeDtmfProvider::InsertDtmf code=" << code
                     << " duration=" << duration
@@ -95,7 +96,7 @@ class FakeDtmfProvider : public DtmfProviderInterface {
     return true;
   }
 
-  virtual sigslot::signal0<>* GetOnDestroyedSignal() {
+  sigslot::signal0<>* GetOnDestroyedSignal() override {
     return &SignalDestroyed;
   }
 
@@ -214,8 +215,8 @@ class DtmfSenderTest : public testing::Test {
   }
 
   rtc::scoped_refptr<AudioTrackInterface> track_;
-  rtc::scoped_ptr<FakeDtmfObserver> observer_;
-  rtc::scoped_ptr<FakeDtmfProvider> provider_;
+  std::unique_ptr<FakeDtmfObserver> observer_;
+  std::unique_ptr<FakeDtmfProvider> provider_;
   rtc::scoped_refptr<DtmfSender> dtmf_;
 };
 
diff --git a/chromium/third_party/webrtc/api/java/jni/OWNERS b/chromium/third_party/webrtc/api/java/jni/OWNERS
index 4d31ffb663f..4178fd6920b 100644
--- a/chromium/third_party/webrtc/api/java/jni/OWNERS
+++ b/chromium/third_party/webrtc/api/java/jni/OWNERS
@@ -1 +1,8 @@
-magjed@webrtc.org
+per-file androidvideocapturer*=magjed@webrtc.org
+per-file androidmediaencoder*=magjed@webrtc.org
+per-file androidmediadecoder*=magjed@webrtc.org
+per-file androidmediacodeccommon.h=magjed@webrtc.org
+per-file surfacetexturehelper*=magjed@webrtc.org
+per-file native_handle_impl*=magjed@webrtc.org
+# Video related parts of peerconnection only.
+per-file peerconnection_jni.cc=magjed@webrtc.org
diff --git a/chromium/third_party/webrtc/api/java/jni/androidmediacodeccommon.h b/chromium/third_party/webrtc/api/java/jni/androidmediacodeccommon.h
index 2f9e05d1f54..db7260ade55 100644
--- a/chromium/third_party/webrtc/api/java/jni/androidmediacodeccommon.h
+++ b/chromium/third_party/webrtc/api/java/jni/androidmediacodeccommon.h
@@ -19,7 +19,6 @@
 #include "webrtc/api/java/jni/jni_helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc_jni {
 
@@ -59,10 +58,6 @@ enum { kMaxDecodedLogFrames = 10 };
 // Maximum amount of encoded frames for which per-frame logging is enabled.
 enum { kMaxEncodedLogFrames = 10 };
 
-static inline int64_t GetCurrentTimeMs() {
-  return webrtc::TickTime::Now().Ticks() / 1000000LL;
-}
-
 static inline void AllowBlockingCalls() {
   rtc::Thread* current_thread = rtc::Thread::Current();
   if (current_thread != NULL)
diff --git a/chromium/third_party/webrtc/api/java/jni/native_handle_impl.cc b/chromium/third_party/webrtc/api/java/jni/native_handle_impl.cc
index d52584acfe6..1f180ade9cc 100644
--- a/chromium/third_party/webrtc/api/java/jni/native_handle_impl.cc
+++ b/chromium/third_party/webrtc/api/java/jni/native_handle_impl.cc
@@ -10,12 +10,13 @@
 
 #include "webrtc/api/java/jni/native_handle_impl.h"
 
+#include <memory>
+
 #include "webrtc/api/java/jni/jni_helpers.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/keep_ref_until_done.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
 using webrtc::NativeHandleBuffer;
@@ -58,6 +59,37 @@ void RotateMatrix(float a[16], webrtc::VideoRotation rotation) {
   }
 }
 
+// Calculates result = a * b, in column-major order.
+void MultiplyMatrix(const float a[16], const float b[16], float result[16]) {
+  for (int i = 0; i < 4; ++i) {
+    for (int j = 0; j < 4; ++j) {
+      float sum = 0;
+      for (int k = 0; k < 4; ++k) {
+        sum += a[k * 4 + j] * b[i * 4 + k];
+      }
+      result[i * 4 + j] = sum;
+    }
+  }
+}
+
+// Center crop by keeping xFraction of the width and yFraction of the height,
+// so e.g. cropping from 640x480 to 640x360 would use
+// xFraction=1, yFraction=360/480.
+void CropMatrix(float a[16], float xFraction, float yFraction) {
+  // Move cropped area to the center of the frame by offsetting half the
+  // removed area.
+  const float xOffset = (1 - xFraction) / 2;
+  const float yOffset = (1 - yFraction) / 2;
+  const float crop_matrix[16] = {
+    xFraction, 0, 0, 0,
+    0, yFraction, 0, 0,
+    0, 0, 1, 0,
+    xOffset, yOffset, 0, 1};
+  float mul_result[16];
+  MultiplyMatrix(crop_matrix, a, mul_result);
+  memcpy(a, mul_result, sizeof(mul_result));
+}
+
 }  // anonymouse namespace
 
 namespace webrtc_jni {
@@ -104,7 +136,7 @@ AndroidTextureBuffer::NativeToI420Buffer() {
   //
   // TODO(nisse): Use an I420BufferPool. We then need to extend that
   // class, and I420Buffer, to support our memory layout.
-  rtc::scoped_ptr<uint8_t, webrtc::AlignedFreeDeleter> yuv_data(
+  std::unique_ptr<uint8_t, webrtc::AlignedFreeDeleter> yuv_data(
       static_cast<uint8_t*>(webrtc::AlignedMalloc(size, kBufferAlignment)));
   // See SurfaceTextureHelper.java for the required layout.
   uint8_t* y_data = yuv_data.get();
@@ -145,15 +177,18 @@ AndroidTextureBuffer::NativeToI420Buffer() {
 }
 
 rtc::scoped_refptr<AndroidTextureBuffer>
-AndroidTextureBuffer::ScaleAndRotate(int dst_widht,
-                                     int dst_height,
-                                     webrtc::VideoRotation rotation) {
-  if (width() == dst_widht && height() == dst_height &&
+AndroidTextureBuffer::CropScaleAndRotate(int cropped_width,
+                                         int cropped_height,
+                                         int dst_width,
+                                         int dst_height,
+                                         webrtc::VideoRotation rotation) {
+  if (cropped_width == dst_width && cropped_height == dst_height &&
+      width() == dst_width && height() == dst_height &&
       rotation == webrtc::kVideoRotation_0) {
     return this;
   }
-  int rotated_width = (rotation % 180 == 0) ? dst_widht : dst_height;
-  int rotated_height = (rotation % 180 == 0) ? dst_height : dst_widht;
+  int rotated_width = (rotation % 180 == 0) ? dst_width : dst_height;
+  int rotated_height = (rotation % 180 == 0) ? dst_height : dst_width;
 
   // Here we use Bind magic to add a reference count to |this| until the newly
   // created AndroidTextureBuffer is destructed
@@ -162,6 +197,11 @@ AndroidTextureBuffer::ScaleAndRotate(int dst_widht,
           rotated_width, rotated_height, native_handle_,
           surface_texture_helper_, rtc::KeepRefUntilDone(this)));
 
+  if (cropped_width != width() || cropped_height != height()) {
+    CropMatrix(buffer->native_handle_.sampling_matrix,
+               cropped_width / static_cast<float>(width()),
+               cropped_height / static_cast<float>(height()));
+  }
   RotateMatrix(buffer->native_handle_.sampling_matrix, rotation);
   return buffer;
 }
diff --git a/chromium/third_party/webrtc/api/java/jni/native_handle_impl.h b/chromium/third_party/webrtc/api/java/jni/native_handle_impl.h
index 859c3713b46..b781815718d 100644
--- a/chromium/third_party/webrtc/api/java/jni/native_handle_impl.h
+++ b/chromium/third_party/webrtc/api/java/jni/native_handle_impl.h
@@ -38,8 +38,11 @@ class AndroidTextureBuffer : public webrtc::NativeHandleBuffer {
   ~AndroidTextureBuffer();
   rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
 
-  rtc::scoped_refptr<AndroidTextureBuffer> ScaleAndRotate(
-      int dst_widht,
+  // First crop, then scale to dst resolution, and then rotate.
+  rtc::scoped_refptr<AndroidTextureBuffer> CropScaleAndRotate(
+      int cropped_width,
+      int cropped_height,
+      int dst_width,
       int dst_height,
       webrtc::VideoRotation rotation);
 
diff --git a/chromium/third_party/webrtc/api/jsepsessiondescription.cc b/chromium/third_party/webrtc/api/jsepsessiondescription.cc
index eb776c86fa7..547a60f1c3a 100644
--- a/chromium/third_party/webrtc/api/jsepsessiondescription.cc
+++ b/chromium/third_party/webrtc/api/jsepsessiondescription.cc
@@ -10,12 +10,13 @@
 
 #include "webrtc/api/jsepsessiondescription.h"
 
+#include <memory>
+
 #include "webrtc/api/webrtcsdp.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/pc/mediasession.h"
 
-using rtc::scoped_ptr;
 using cricket::SessionDescription;
 
 namespace webrtc {
@@ -57,7 +58,6 @@ const int JsepSessionDescription::kMaxVideoCodecHeight = 1280;
 const int JsepSessionDescription::kMaxVideoCodecWidth = 1920;
 const int JsepSessionDescription::kMaxVideoCodecHeight = 1080;
 #endif
-const int JsepSessionDescription::kDefaultVideoCodecPreference = 1;
 
 SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
                                                       const std::string& sdp,
@@ -125,7 +125,7 @@ bool JsepSessionDescription::AddCandidate(
     updated_candidate.set_password(transport_info->description.ice_pwd);
   }
 
-  scoped_ptr<JsepIceCandidate> updated_candidate_wrapper(
+  std::unique_ptr<JsepIceCandidate> updated_candidate_wrapper(
       new JsepIceCandidate(candidate->sdp_mid(),
                            static_cast<int>(mediasection_index),
                            updated_candidate));
diff --git a/chromium/third_party/webrtc/api/jsepsessiondescription.h b/chromium/third_party/webrtc/api/jsepsessiondescription.h
index 9a0d8732044..0248a07c726 100644
--- a/chromium/third_party/webrtc/api/jsepsessiondescription.h
+++ b/chromium/third_party/webrtc/api/jsepsessiondescription.h
@@ -13,12 +13,13 @@
 #ifndef WEBRTC_API_JSEPSESSIONDESCRIPTION_H_
 #define WEBRTC_API_JSEPSESSIONDESCRIPTION_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/jsepicecandidate.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/p2p/base/candidate.h"
 
 namespace cricket {
@@ -72,10 +73,9 @@ class JsepSessionDescription : public SessionDescriptionInterface {
   static const char kDefaultVideoCodecName[];
   static const int kMaxVideoCodecWidth;
   static const int kMaxVideoCodecHeight;
-  static const int kDefaultVideoCodecPreference;
 
  private:
-  rtc::scoped_ptr<cricket::SessionDescription> description_;
+  std::unique_ptr<cricket::SessionDescription> description_;
   std::string session_id_;
   std::string session_version_;
   std::string type_;
diff --git a/chromium/third_party/webrtc/api/jsepsessiondescription_unittest.cc b/chromium/third_party/webrtc/api/jsepsessiondescription_unittest.cc
index 3d875137311..6be590faf4d 100644
--- a/chromium/third_party/webrtc/api/jsepsessiondescription_unittest.cc
+++ b/chromium/third_party/webrtc/api/jsepsessiondescription_unittest.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/api/jsepicecandidate.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/p2p/base/candidate.h"
@@ -27,7 +27,6 @@ using webrtc::IceCandidateInterface;
 using webrtc::JsepIceCandidate;
 using webrtc::JsepSessionDescription;
 using webrtc::SessionDescriptionInterface;
-using rtc::scoped_ptr;
 
 static const char kCandidateUfrag[] = "ufrag";
 static const char kCandidatePwd[] = "pwd";
@@ -41,18 +40,18 @@ static const char kCandidatePwdVideo[] = "pwd_video";
 static cricket::SessionDescription* CreateCricketSessionDescription() {
   cricket::SessionDescription* desc(new cricket::SessionDescription());
   // AudioContentDescription
-  scoped_ptr<cricket::AudioContentDescription> audio(
+  std::unique_ptr<cricket::AudioContentDescription> audio(
       new cricket::AudioContentDescription());
 
   // VideoContentDescription
-  scoped_ptr<cricket::VideoContentDescription> video(
+  std::unique_ptr<cricket::VideoContentDescription> video(
       new cricket::VideoContentDescription());
 
-  audio->AddCodec(cricket::AudioCodec(103, "ISAC", 16000, 0, 0, 0));
+  audio->AddCodec(cricket::AudioCodec(103, "ISAC", 16000, 0, 0));
   desc->AddContent(cricket::CN_AUDIO, cricket::NS_JINGLE_RTP,
                    audio.release());
 
-  video->AddCodec(cricket::VideoCodec(120, "VP8", 640, 480, 30, 0));
+  video->AddCodec(cricket::VideoCodec(120, "VP8", 640, 480, 30));
   desc->AddContent(cricket::CN_VIDEO, cricket::NS_JINGLE_RTP,
                    video.release());
 
@@ -100,7 +99,7 @@ class JsepSessionDescriptionTest : public testing::Test {
   }
 
   cricket::Candidate candidate_;
-  rtc::scoped_ptr<JsepSessionDescription> jsep_desc_;
+  std::unique_ptr<JsepSessionDescription> jsep_desc_;
 };
 
 // Test that number_of_mediasections() returns the number of media contents in
@@ -204,7 +203,8 @@ TEST_F(JsepSessionDescriptionTest, AddCandidateDuplicates) {
 TEST_F(JsepSessionDescriptionTest, SerializeDeserialize) {
   std::string sdp = Serialize(jsep_desc_.get());
 
-  scoped_ptr<SessionDescriptionInterface> parsed_jsep_desc(DeSerialize(sdp));
+  std::unique_ptr<SessionDescriptionInterface> parsed_jsep_desc(
+      DeSerialize(sdp));
   EXPECT_EQ(2u, parsed_jsep_desc->number_of_mediasections());
 
   std::string parsed_sdp = Serialize(parsed_jsep_desc.get());
@@ -222,7 +222,7 @@ TEST_F(JsepSessionDescriptionTest, SerializeDeserializeWithCandidates) {
   std::string sdp_with_candidate = Serialize(jsep_desc_.get());
   EXPECT_NE(sdp, sdp_with_candidate);
 
-  scoped_ptr<SessionDescriptionInterface> parsed_jsep_desc(
+  std::unique_ptr<SessionDescriptionInterface> parsed_jsep_desc(
       DeSerialize(sdp_with_candidate));
   std::string parsed_sdp_with_candidate = Serialize(parsed_jsep_desc.get());
 
diff --git a/chromium/third_party/webrtc/api/localaudiosource.cc b/chromium/third_party/webrtc/api/localaudiosource.cc
index 3b22ad146ef..9da9fd26123 100644
--- a/chromium/third_party/webrtc/api/localaudiosource.cc
+++ b/chromium/third_party/webrtc/api/localaudiosource.cc
@@ -47,6 +47,8 @@ void FromConstraints(const MediaConstraintsInterface::Constraints& constraints,
        options->noise_suppression},
       {MediaConstraintsInterface::kExperimentalNoiseSuppression,
        options->experimental_ns},
+      {MediaConstraintsInterface::kIntelligibilityEnhancer,
+       options->intelligibility_enhancer},
       {MediaConstraintsInterface::kHighpassFilter, options->highpass_filter},
       {MediaConstraintsInterface::kTypingNoiseDetection,
        options->typing_detection},
diff --git a/chromium/third_party/webrtc/api/localaudiosource.h b/chromium/third_party/webrtc/api/localaudiosource.h
index e4de650537f..e1c023e5424 100644
--- a/chromium/third_party/webrtc/api/localaudiosource.h
+++ b/chromium/third_party/webrtc/api/localaudiosource.h
@@ -14,7 +14,6 @@
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/notifier.h"
 #include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/mediachannel.h"
 
 // LocalAudioSource implements AudioSourceInterface.
diff --git a/chromium/third_party/webrtc/api/mediaconstraintsinterface.cc b/chromium/third_party/webrtc/api/mediaconstraintsinterface.cc
index af258917e7a..6a014a2b039 100644
--- a/chromium/third_party/webrtc/api/mediaconstraintsinterface.cc
+++ b/chromium/third_party/webrtc/api/mediaconstraintsinterface.cc
@@ -46,6 +46,8 @@ const char MediaConstraintsInterface::kNoiseSuppression[] =
     "googNoiseSuppression";
 const char MediaConstraintsInterface::kExperimentalNoiseSuppression[] =
     "googNoiseSuppression2";
+const char MediaConstraintsInterface::kIntelligibilityEnhancer[] =
+    "intelligibilityEnhancer";
 const char MediaConstraintsInterface::kHighpassFilter[] =
     "googHighpassFilter";
 const char MediaConstraintsInterface::kTypingNoiseDetection[] =
@@ -179,29 +181,23 @@ void CopyConstraintsIntoRtcConfiguration(
     return;
   }
 
-  bool value;
+  bool enable_ipv6;
   if (FindConstraint(constraints, MediaConstraintsInterface::kEnableIPv6,
-                     &value, nullptr)) {
-    if (!value) {
-      configuration->disable_ipv6 = true;
-    }
-  }
-  ConstraintToOptionalBool(constraints, MediaConstraintsInterface::kEnableDscp,
-                           &configuration->enable_dscp);
-  ConstraintToOptionalBool(constraints,
-                           MediaConstraintsInterface::kCpuOveruseDetection,
-                           &configuration->cpu_overuse_detection);
-  if (FindConstraint(constraints,
-                     MediaConstraintsInterface::kEnableRtpDataChannels, &value,
-                     NULL) &&
-      value) {
-    configuration->enable_rtp_data_channel = true;
+                     &enable_ipv6, nullptr)) {
+    configuration->disable_ipv6 = !enable_ipv6;
   }
+  FindConstraint(constraints, MediaConstraintsInterface::kEnableDscp,
+                 &configuration->media_config.enable_dscp, nullptr);
+  FindConstraint(
+      constraints, MediaConstraintsInterface::kCpuOveruseDetection,
+      &configuration->media_config.video.enable_cpu_overuse_detection, nullptr);
+  FindConstraint(constraints, MediaConstraintsInterface::kEnableRtpDataChannels,
+                 &configuration->enable_rtp_data_channel, nullptr);
   // Find Suspend Below Min Bitrate constraint.
-  ConstraintToOptionalBool(
-      constraints,
-      MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
-      &configuration->suspend_below_min_bitrate);
+  FindConstraint(constraints,
+                 MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
+                 &configuration->media_config.video.suspend_below_min_bitrate,
+                 nullptr);
   ConstraintToOptionalInt(constraints,
                           MediaConstraintsInterface::kScreencastMinBitrate,
                           &configuration->screencast_min_bitrate);
diff --git a/chromium/third_party/webrtc/api/mediaconstraintsinterface.h b/chromium/third_party/webrtc/api/mediaconstraintsinterface.h
index 3db6e2672bd..13560dd35b9 100644
--- a/chromium/third_party/webrtc/api/mediaconstraintsinterface.h
+++ b/chromium/third_party/webrtc/api/mediaconstraintsinterface.h
@@ -73,6 +73,7 @@ class MediaConstraintsInterface {
   static const char kExperimentalAutoGainControl[];  // googAutoGainControl2
   static const char kNoiseSuppression[];  // googNoiseSuppression
   static const char kExperimentalNoiseSuppression[];  // googNoiseSuppression2
+  static const char kIntelligibilityEnhancer[];  // intelligibilityEnhancer
   static const char kHighpassFilter[];  // googHighpassFilter
   static const char kTypingNoiseDetection[];  // googTypingNoiseDetection
   static const char kAudioMirroring[];  // googAudioMirroring
diff --git a/chromium/third_party/webrtc/api/mediaconstraintsinterface_unittest.cc b/chromium/third_party/webrtc/api/mediaconstraintsinterface_unittest.cc
index 07338c15e82..dcf4bb7fde7 100644
--- a/chromium/third_party/webrtc/api/mediaconstraintsinterface_unittest.cc
+++ b/chromium/third_party/webrtc/api/mediaconstraintsinterface_unittest.cc
@@ -17,11 +17,24 @@ namespace webrtc {
 
 namespace {
 
+// Checks all settings touched by CopyConstraintsIntoRtcConfiguration,
+// plus audio_jitter_buffer_max_packets.
 bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
              const PeerConnectionInterface::RTCConfiguration& b) {
-  return a.audio_jitter_buffer_max_packets ==
+  return a.disable_ipv6 == b.disable_ipv6 &&
+         a.audio_jitter_buffer_max_packets ==
              b.audio_jitter_buffer_max_packets &&
-         a.disable_prerenderer_smoothing == b.disable_prerenderer_smoothing;
+         a.enable_rtp_data_channel == b.enable_rtp_data_channel &&
+         a.screencast_min_bitrate == b.screencast_min_bitrate &&
+         a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
+         a.enable_dtls_srtp == b.enable_dtls_srtp &&
+         a.media_config.enable_dscp == b.media_config.enable_dscp &&
+         a.media_config.video.enable_cpu_overuse_detection ==
+             b.media_config.video.enable_cpu_overuse_detection &&
+         a.media_config.video.disable_prerenderer_smoothing ==
+             b.media_config.video.disable_prerenderer_smoothing &&
+         a.media_config.video.suspend_below_min_bitrate ==
+             b.media_config.video.suspend_below_min_bitrate;
 }
 
 TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) {
diff --git a/chromium/third_party/webrtc/api/mediacontroller.cc b/chromium/third_party/webrtc/api/mediacontroller.cc
index 71964d52a7e..2e4501b3f1b 100644
--- a/chromium/third_party/webrtc/api/mediacontroller.cc
+++ b/chromium/third_party/webrtc/api/mediacontroller.cc
@@ -10,8 +10,11 @@
 
 #include "webrtc/api/mediacontroller.h"
 
+#include <memory>
+
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/call.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/media/base/mediachannel.h"
@@ -74,7 +77,7 @@ class MediaController : public webrtc::MediaControllerInterface,
   const cricket::MediaConfig media_config_;
   cricket::ChannelManager* const channel_manager_;
   webrtc::Call::Config call_config_;
-  rtc::scoped_ptr<webrtc::Call> call_;
+  std::unique_ptr<webrtc::Call> call_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController);
 };
diff --git a/chromium/third_party/webrtc/api/mediastream.h b/chromium/third_party/webrtc/api/mediastream.h
index 2a77f0ddc4b..1f80f25207a 100644
--- a/chromium/third_party/webrtc/api/mediastream.h
+++ b/chromium/third_party/webrtc/api/mediastream.h
@@ -31,10 +31,10 @@ class MediaStream : public Notifier<MediaStreamInterface> {
   bool AddTrack(VideoTrackInterface* track) override;
   bool RemoveTrack(AudioTrackInterface* track) override;
   bool RemoveTrack(VideoTrackInterface* track) override;
-  virtual rtc::scoped_refptr<AudioTrackInterface>
-      FindAudioTrack(const std::string& track_id);
-  virtual rtc::scoped_refptr<VideoTrackInterface>
-      FindVideoTrack(const std::string& track_id);
+  rtc::scoped_refptr<AudioTrackInterface>
+      FindAudioTrack(const std::string& track_id) override;
+  rtc::scoped_refptr<VideoTrackInterface>
+      FindVideoTrack(const std::string& track_id) override;
 
   AudioTrackVector GetAudioTracks() override { return audio_tracks_; }
   VideoTrackVector GetVideoTracks() override { return video_tracks_; }
diff --git a/chromium/third_party/webrtc/api/mediastream_unittest.cc b/chromium/third_party/webrtc/api/mediastream_unittest.cc
index dd63356e925..1881cceeb0e 100644
--- a/chromium/third_party/webrtc/api/mediastream_unittest.cc
+++ b/chromium/third_party/webrtc/api/mediastream_unittest.cc
@@ -18,7 +18,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
 
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kVideoTrackId[] = "dummy_video_cam_1";
diff --git a/chromium/third_party/webrtc/api/mediastreaminterface.h b/chromium/third_party/webrtc/api/mediastreaminterface.h
index 2a3f3473b02..7f563250249 100644
--- a/chromium/third_party/webrtc/api/mediastreaminterface.h
+++ b/chromium/third_party/webrtc/api/mediastreaminterface.h
@@ -31,7 +31,6 @@
 namespace cricket {
 
 class AudioRenderer;
-class VideoCapturer;
 class VideoRenderer;
 class VideoFrame;
 
@@ -113,14 +112,6 @@ class VideoTrackSourceInterface
     int input_width;
     int input_height;
   };
-  // Get access to the source implementation of cricket::VideoCapturer.
-  // This can be used for receiving frames and state notifications.
-  // But it should not be used for starting or stopping capturing.
-  // TODO(perkj): We are currently trying to replace all internal use of
-  // cricket::VideoCapturer with rtc::VideoSourceInterface. Once that
-  // refactoring is done,
-  // remove this method.
-  virtual cricket::VideoCapturer* GetVideoCapturer() = 0;
 
   virtual void Stop() = 0;
   virtual void Restart() = 0;
@@ -215,7 +206,8 @@ class AudioProcessorInterface : public rtc::RefCountInterface {
                             echo_return_loss_enhancement(0),
                             echo_delay_median_ms(0),
                             aec_quality_min(0.0),
-                            echo_delay_std_ms(0) {}
+                            echo_delay_std_ms(0),
+                            aec_divergent_filter_fraction(0.0) {}
     ~AudioProcessorStats() {}
 
     bool typing_noise_detected;
@@ -224,6 +216,7 @@ class AudioProcessorInterface : public rtc::RefCountInterface {
     int echo_delay_median_ms;
     float aec_quality_min;
     int echo_delay_std_ms;
+    float aec_divergent_filter_fraction;
   };
 
   // Get audio processor statistics.
diff --git a/chromium/third_party/webrtc/api/mediastreamprovider.h b/chromium/third_party/webrtc/api/mediastreamprovider.h
index 6814c416e20..b23e17bd4bb 100644
--- a/chromium/third_party/webrtc/api/mediastreamprovider.h
+++ b/chromium/third_party/webrtc/api/mediastreamprovider.h
@@ -11,17 +11,17 @@
 #ifndef WEBRTC_API_MEDIASTREAMPROVIDER_H_
 #define WEBRTC_API_MEDIASTREAMPROVIDER_H_
 
+#include <memory>
+
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/base/basictypes.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/videosinkinterface.h"
+#include "webrtc/media/base/videosourceinterface.h"
 
 namespace cricket {
 
 class AudioSource;
-class VideoCapturer;
 class VideoFrame;
-class VideoRenderer;
 struct AudioOptions;
 struct VideoOptions;
 
@@ -61,11 +61,16 @@ class AudioProviderInterface {
   // passed to the provider.
   virtual void SetRawAudioSink(
       uint32_t ssrc,
-      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
+      std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
+
+  virtual RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetAudioRtpSendParameters(uint32_t ssrc,
+                                         const RtpParameters& parameters) = 0;
 
-  virtual RtpParameters GetAudioRtpParameters(uint32_t ssrc) const = 0;
-  virtual bool SetAudioRtpParameters(uint32_t ssrc,
-                                     const RtpParameters& parameters) = 0;
+  virtual RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const = 0;
+  virtual bool SetAudioRtpReceiveParameters(
+      uint32_t ssrc,
+      const RtpParameters& parameters) = 0;
 
  protected:
   virtual ~AudioProviderInterface() {}
@@ -75,8 +80,9 @@ class AudioProviderInterface {
 // of a video track connected to a certain PeerConnection.
 class VideoProviderInterface {
  public:
-  virtual bool SetCaptureDevice(uint32_t ssrc,
-                                cricket::VideoCapturer* camera) = 0;
+  virtual bool SetSource(
+      uint32_t ssrc,
+      rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0;
   // Enable/disable the video playout of a remote video track with |ssrc|.
   virtual void SetVideoPlayout(
       uint32_t ssrc,
@@ -87,9 +93,14 @@ class VideoProviderInterface {
                             bool enable,
                             const cricket::VideoOptions* options) = 0;
 
-  virtual RtpParameters GetVideoRtpParameters(uint32_t ssrc) const = 0;
-  virtual bool SetVideoRtpParameters(uint32_t ssrc,
-                                     const RtpParameters& parameters) = 0;
+  virtual RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetVideoRtpSendParameters(uint32_t ssrc,
+                                         const RtpParameters& parameters) = 0;
+
+  virtual RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const = 0;
+  virtual bool SetVideoRtpReceiveParameters(
+      uint32_t ssrc,
+      const RtpParameters& parameters) = 0;
 
  protected:
   virtual ~VideoProviderInterface() {}
diff --git a/chromium/third_party/webrtc/api/mediastreamproxy.h b/chromium/third_party/webrtc/api/mediastreamproxy.h
index 645b28a3f79..06f8eb3b2cf 100644
--- a/chromium/third_party/webrtc/api/mediastreamproxy.h
+++ b/chromium/third_party/webrtc/api/mediastreamproxy.h
@@ -16,7 +16,7 @@
 
 namespace webrtc {
 
-BEGIN_PROXY_MAP(MediaStream)
+BEGIN_SIGNALING_PROXY_MAP(MediaStream)
   PROXY_CONSTMETHOD0(std::string, label)
   PROXY_METHOD0(AudioTrackVector, GetAudioTracks)
   PROXY_METHOD0(VideoTrackVector, GetVideoTracks)
@@ -30,7 +30,7 @@ BEGIN_PROXY_MAP(MediaStream)
   PROXY_METHOD1(bool, RemoveTrack, VideoTrackInterface*)
   PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
   PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
-END_PROXY()
+END_SIGNALING_PROXY()
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/mediastreamtrackproxy.h b/chromium/third_party/webrtc/api/mediastreamtrackproxy.h
index f68773223b9..12fdc36b368 100644
--- a/chromium/third_party/webrtc/api/mediastreamtrackproxy.h
+++ b/chromium/third_party/webrtc/api/mediastreamtrackproxy.h
@@ -19,7 +19,7 @@
 
 namespace webrtc {
 
-BEGIN_PROXY_MAP(AudioTrack)
+BEGIN_SIGNALING_PROXY_MAP(AudioTrack)
   PROXY_CONSTMETHOD0(std::string, kind)
   PROXY_CONSTMETHOD0(std::string, id)
   PROXY_CONSTMETHOD0(TrackState, state)
@@ -33,7 +33,7 @@ BEGIN_PROXY_MAP(AudioTrack)
   PROXY_METHOD1(bool, set_enabled, bool)
   PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
   PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
-END_PROXY()
+END_SIGNALING_PROXY()
 
 BEGIN_PROXY_MAP(VideoTrack)
   PROXY_CONSTMETHOD0(std::string, kind)
@@ -41,11 +41,13 @@ BEGIN_PROXY_MAP(VideoTrack)
   PROXY_CONSTMETHOD0(TrackState, state)
   PROXY_CONSTMETHOD0(bool, enabled)
   PROXY_METHOD1(bool, set_enabled, bool)
-  PROXY_METHOD2(void,
+  PROXY_WORKER_METHOD2(void,
                 AddOrUpdateSink,
                 rtc::VideoSinkInterface<cricket::VideoFrame>*,
                 const rtc::VideoSinkWants&)
-  PROXY_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<cricket::VideoFrame>*)
+  PROXY_WORKER_METHOD1(void,
+                       RemoveSink,
+                       rtc::VideoSinkInterface<cricket::VideoFrame>*)
   PROXY_CONSTMETHOD0(VideoTrackSourceInterface*, GetSource)
 
   PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
diff --git a/chromium/third_party/webrtc/api/objc/README b/chromium/third_party/webrtc/api/objc/README
deleted file mode 100644
index bd33e61921e..00000000000
--- a/chromium/third_party/webrtc/api/objc/README
+++ /dev/null
@@ -1,3 +0,0 @@
-This is a work-in-progress to update the Objective-C API according to the W3C
-specification. The Objective-C API located at talk/app/webrtc/objc is
-deprecated, but will remain for the time being.
diff --git a/chromium/third_party/webrtc/api/peerconnection.cc b/chromium/third_party/webrtc/api/peerconnection.cc
index b2b8062ca84..dda0eebb074 100644
--- a/chromium/third_party/webrtc/api/peerconnection.cc
+++ b/chromium/third_party/webrtc/api/peerconnection.cc
@@ -66,6 +66,9 @@ static const char kTransport[] = "transport";
 // NOTE: Must be in the same order as the ServiceType enum.
 static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
 
+// The length of RTCP CNAMEs.
+static const int kRtcpCnameLength = 16;
+
 // NOTE: A loop below assumes that the first value of this enum is 0 and all
 // other values are incremental.
 enum ServiceType {
@@ -373,10 +376,37 @@ void AddSendStreams(
   }
 }
 
+uint32_t ConvertIceTransportTypeToCandidateFilter(
+    PeerConnectionInterface::IceTransportsType type) {
+  switch (type) {
+    case PeerConnectionInterface::kNone:
+      return cricket::CF_NONE;
+    case PeerConnectionInterface::kRelay:
+      return cricket::CF_RELAY;
+    case PeerConnectionInterface::kNoHost:
+      return (cricket::CF_ALL & ~cricket::CF_HOST);
+    case PeerConnectionInterface::kAll:
+      return cricket::CF_ALL;
+    default:
+      ASSERT(false);
+  }
+  return cricket::CF_NONE;
+}
+
 }  // namespace
 
 namespace webrtc {
 
+// Generate a RTCP CNAME when a PeerConnection is created.
+std::string GenerateRtcpCname() {
+  std::string cname;
+  if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
+    LOG(LS_ERROR) << "Failed to generate CNAME.";
+    RTC_DCHECK(false);
+  }
+  return cname;
+}
+
 bool ExtractMediaSessionOptions(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
     bool is_offer,
@@ -508,6 +538,7 @@ PeerConnection::PeerConnection(PeerConnectionFactory* factory)
       ice_state_(kIceNew),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew),
+      rtcp_cname_(GenerateRtcpCname()),
       local_streams_(StreamCollection::Create()),
       remote_streams_(StreamCollection::Create()) {}
 
@@ -522,13 +553,20 @@ PeerConnection::~PeerConnection() {
   for (const auto& receiver : receivers_) {
     receiver->Stop();
   }
+  // Destroy stats_ because it depends on session_.
+  stats_.reset(nullptr);
+  // Now destroy session_ before destroying other members,
+  // because its destruction fires signals (such as VoiceChannelDestroyed)
+  // which will trigger some final actions in PeerConnection...
+  session_.reset(nullptr);
+  // port_allocator_ lives on the worker thread and should be destroyed there.
+  worker_thread()->Invoke<void>([this] { port_allocator_.reset(nullptr); });
 }
 
 bool PeerConnection::Initialize(
-    const cricket::MediaConfig& media_config,
     const PeerConnectionInterface::RTCConfiguration& configuration,
-    rtc::scoped_ptr<cricket::PortAllocator> allocator,
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+    std::unique_ptr<cricket::PortAllocator> allocator,
+    std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     PeerConnectionObserver* observer) {
   TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
   RTC_DCHECK(observer != nullptr);
@@ -539,41 +577,20 @@ bool PeerConnection::Initialize(
 
   port_allocator_ = std::move(allocator);
 
-  cricket::ServerAddresses stun_servers;
-  std::vector<cricket::RelayServerConfig> turn_servers;
-  if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
+  // The port allocator lives on the worker thread and should be initialized
+  // there.
+  if (!worker_thread()->Invoke<bool>(rtc::Bind(
+          &PeerConnection::InitializePortAllocator_w, this, configuration))) {
     return false;
   }
-  port_allocator_->SetIceServers(stun_servers, turn_servers);
-
-  // To handle both internal and externally created port allocator, we will
-  // enable BUNDLE here.
-  int portallocator_flags = port_allocator_->flags();
-  portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
-                         cricket::PORTALLOCATOR_ENABLE_IPV6;
-  // If the disable-IPv6 flag was specified, we'll not override it
-  // by experiment.
-  if (configuration.disable_ipv6) {
-    portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
-  } else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") ==
-             "Disabled") {
-    portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
-  }
-
-  if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
-    portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
-    LOG(LS_INFO) << "TCP candidates are disabled.";
-  }
 
-  port_allocator_->set_flags(portallocator_flags);
-  // No step delay is used while allocating ports.
-  port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
-
-  media_controller_.reset(factory_->CreateMediaController(media_config));
+  media_controller_.reset(
+      factory_->CreateMediaController(configuration.media_config));
 
   session_.reset(
-      new WebRtcSession(media_controller_.get(), factory_->signaling_thread(),
-                        factory_->worker_thread(), port_allocator_.get()));
+      new WebRtcSession(media_controller_.get(), factory_->network_thread(),
+                        factory_->worker_thread(), factory_->signaling_thread(),
+                        port_allocator_.get()));
   stats_.reset(new StatsCollector(this));
 
   // Initialize the WebRtcSession. It creates transport channels etc.
@@ -628,7 +645,7 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
                                           &PeerConnection::OnVideoTrackAdded);
   observer->SignalVideoTrackRemoved.connect(
       this, &PeerConnection::OnVideoTrackRemoved);
-  stream_observers_.push_back(rtc::scoped_ptr<MediaStreamObserver>(observer));
+  stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
 
   for (const auto& track : local_stream->GetAudioTracks()) {
     OnAudioTrackAdded(track.get(), local_stream);
@@ -655,7 +672,7 @@ void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
   stream_observers_.erase(
       std::remove_if(
           stream_observers_.begin(), stream_observers_.end(),
-          [local_stream](const rtc::scoped_ptr<MediaStreamObserver>& observer) {
+          [local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
             return observer->stream()->label().compare(local_stream->label()) ==
                    0;
           }),
@@ -835,7 +852,7 @@ PeerConnection::CreateDataChannel(
   TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
   bool first_datachannel = !HasDataChannels();
 
-  rtc::scoped_ptr<InternalDataChannelInit> internal_config;
+  std::unique_ptr<InternalDataChannelInit> internal_config;
   if (config) {
     internal_config.reset(new InternalDataChannelInit(*config));
   }
@@ -1144,18 +1161,19 @@ void PeerConnection::SetRemoteDescription(
   signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
 }
 
-bool PeerConnection::SetConfiguration(const RTCConfiguration& config) {
+bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration) {
   TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
   if (port_allocator_) {
-    cricket::ServerAddresses stun_servers;
-    std::vector<cricket::RelayServerConfig> turn_servers;
-    if (!ParseIceServers(config.servers, &stun_servers, &turn_servers)) {
+    if (!worker_thread()->Invoke<bool>(
+            rtc::Bind(&PeerConnection::ReconfigurePortAllocator_w, this,
+                      configuration))) {
       return false;
     }
-    port_allocator_->SetIceServers(stun_servers, turn_servers);
   }
-  session_->SetIceConfig(session_->ParseIceConfig(config));
-  return session_->SetIceTransports(config.type);
+
+  // TODO(deadbeef): Shouldn't have to hop to the worker thread twice...
+  session_->SetIceConfig(session_->ParseIceConfig(configuration));
+  return true;
 }
 
 bool PeerConnection::AddIceCandidate(
@@ -1503,6 +1521,8 @@ bool PeerConnection::GetOptionsForOffer(
   if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
     session_options->data_channel_type = cricket::DCT_SCTP;
   }
+
+  session_options->rtcp_cname = rtcp_cname_;
   return true;
 }
 
@@ -1540,6 +1560,8 @@ bool PeerConnection::GetOptionsForAnswer(
   if (!ParseConstraintsForAnswer(constraints, session_options)) {
     return false;
   }
+  session_options->rtcp_cname = rtcp_cname_;
+
   FinishOptionsForAnswer(session_options);
   return true;
 }
@@ -1552,6 +1574,8 @@ bool PeerConnection::GetOptionsForAnswer(
   if (!ExtractMediaSessionOptions(options, false, session_options)) {
     return false;
   }
+  session_options->rtcp_cname = rtcp_cname_;
+
   FinishOptionsForAnswer(session_options);
   return true;
 }
@@ -2064,4 +2088,60 @@ DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
   return nullptr;
 }
 
+bool PeerConnection::InitializePortAllocator_w(
+    const RTCConfiguration& configuration) {
+  cricket::ServerAddresses stun_servers;
+  std::vector<cricket::RelayServerConfig> turn_servers;
+  if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
+    return false;
+  }
+
+  // To handle both internal and externally created port allocator, we will
+  // enable BUNDLE here.
+  int portallocator_flags = port_allocator_->flags();
+  portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
+                         cricket::PORTALLOCATOR_ENABLE_IPV6;
+  // If the disable-IPv6 flag was specified, we'll not override it
+  // by experiment.
+  if (configuration.disable_ipv6) {
+    portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
+  } else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") ==
+             "Disabled") {
+    portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
+  }
+
+  if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
+    portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
+    LOG(LS_INFO) << "TCP candidates are disabled.";
+  }
+
+  port_allocator_->set_flags(portallocator_flags);
+  // No step delay is used while allocating ports.
+  port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
+  port_allocator_->set_candidate_filter(
+      ConvertIceTransportTypeToCandidateFilter(configuration.type));
+
+  // Call this last since it may create pooled allocator sessions using the
+  // properties set above.
+  port_allocator_->SetConfiguration(stun_servers, turn_servers,
+                                    configuration.ice_candidate_pool_size);
+  return true;
+}
+
+bool PeerConnection::ReconfigurePortAllocator_w(
+    const RTCConfiguration& configuration) {
+  cricket::ServerAddresses stun_servers;
+  std::vector<cricket::RelayServerConfig> turn_servers;
+  if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
+    return false;
+  }
+  port_allocator_->set_candidate_filter(
+      ConvertIceTransportTypeToCandidateFilter(configuration.type));
+  // Call this last since it may create pooled allocator sessions using the
+  // candidate filter set above.
+  port_allocator_->SetConfiguration(stun_servers, turn_servers,
+                                    configuration.ice_candidate_pool_size);
+  return true;
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/peerconnection.h b/chromium/third_party/webrtc/api/peerconnection.h
index d1676733d57..8ba7e583724 100644
--- a/chromium/third_party/webrtc/api/peerconnection.h
+++ b/chromium/third_party/webrtc/api/peerconnection.h
@@ -13,6 +13,7 @@
 
 #include <string>
 #include <map>
+#include <memory>
 #include <vector>
 
 #include "webrtc/api/dtlsidentitystore.h"
@@ -23,7 +24,6 @@
 #include "webrtc/api/statscollector.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/webrtcsession.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -68,10 +68,9 @@ class PeerConnection : public PeerConnectionInterface,
   explicit PeerConnection(PeerConnectionFactory* factory);
 
   bool Initialize(
-      const cricket::MediaConfig& media_config,
       const PeerConnectionInterface::RTCConfiguration& configuration,
-      rtc::scoped_ptr<cricket::PortAllocator> allocator,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<cricket::PortAllocator> allocator,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer);
 
   rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
@@ -131,7 +130,7 @@ class PeerConnection : public PeerConnectionInterface,
   void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                             SessionDescriptionInterface* desc) override;
   bool SetConfiguration(
-      const PeerConnectionInterface::RTCConfiguration& config) override;
+      const PeerConnectionInterface::RTCConfiguration& configuration) override;
   bool AddIceCandidate(const IceCandidateInterface* candidate) override;
   bool RemoveIceCandidates(
       const std::vector<cricket::Candidate>& candidates) override;
@@ -210,6 +209,8 @@ class PeerConnection : public PeerConnectionInterface,
     return factory_->signaling_thread();
   }
 
+  rtc::Thread* worker_thread() const { return factory_->worker_thread(); }
+
   void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
                                         const std::string& error);
   void PostCreateSessionDescriptionFailure(
@@ -351,6 +352,12 @@ class PeerConnection : public PeerConnectionInterface,
   // or nullptr if not found.
   DataChannel* FindDataChannelBySid(int sid) const;
 
+  // Called when first configuring the port allocator.
+  bool InitializePortAllocator_w(const RTCConfiguration& configuration);
+  // Called when SetConfiguration is called. Only a subset of the configuration
+  // is applied.
+  bool ReconfigurePortAllocator_w(const RTCConfiguration& configuration);
+
   // Storing the factory as a scoped reference pointer ensures that the memory
   // in the PeerConnectionFactoryImpl remains available as long as the
   // PeerConnection is running. It is passed to PeerConnection as a raw pointer.
@@ -366,15 +373,19 @@ class PeerConnection : public PeerConnectionInterface,
   IceConnectionState ice_connection_state_;
   IceGatheringState ice_gathering_state_;
 
-  rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
-  rtc::scoped_ptr<MediaControllerInterface> media_controller_;
+  std::unique_ptr<cricket::PortAllocator> port_allocator_;
+  std::unique_ptr<MediaControllerInterface> media_controller_;
+
+  // One PeerConnection has only one RTCP CNAME.
+  // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
+  std::string rtcp_cname_;
 
   // Streams added via AddStream.
   rtc::scoped_refptr<StreamCollection> local_streams_;
   // Streams created as a result of SetRemoteDescription.
   rtc::scoped_refptr<StreamCollection> remote_streams_;
 
-  std::vector<rtc::scoped_ptr<MediaStreamObserver>> stream_observers_;
+  std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_;
 
   // These lists store track info seen in local/remote descriptions.
   TrackInfos remote_audio_tracks_;
@@ -393,12 +404,8 @@ class PeerConnection : public PeerConnectionInterface,
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_;
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_;
 
-  // The session_ scoped_ptr is declared at the bottom of PeerConnection
-  // because its destruction fires signals (such as VoiceChannelDestroyed)
-  // which will trigger some final actions in PeerConnection...
-  rtc::scoped_ptr<WebRtcSession> session_;
-  // ... But stats_ depends on session_ so it should be destroyed even earlier.
-  rtc::scoped_ptr<StatsCollector> stats_;
+  std::unique_ptr<WebRtcSession> session_;
+  std::unique_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/peerconnection_unittest.cc b/chromium/third_party/webrtc/api/peerconnection_unittest.cc
index 67b4efbc73f..24411ab7290 100644
--- a/chromium/third_party/webrtc/api/peerconnection_unittest.cc
+++ b/chromium/third_party/webrtc/api/peerconnection_unittest.cc
@@ -13,6 +13,7 @@
 #include <algorithm>
 #include <list>
 #include <map>
+#include <memory>
 #include <utility>
 #include <vector>
 
@@ -31,15 +32,14 @@
 #include "webrtc/api/test/mockpeerconnectionobservers.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/media/engine/fakewebrtcvideoengine.h"
+#include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/sessiondescription.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
 #include "webrtc/pc/mediasession.h"
 
 #define MAYBE_SKIP_TEST(feature)                    \
@@ -154,11 +154,13 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
       const std::string& id,
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
-      rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
-      bool prefer_constraint_apis) {
+      std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
+      bool prefer_constraint_apis,
+      rtc::Thread* network_thread,
+      rtc::Thread* worker_thread) {
     PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
     if (!client->Init(constraints, options, std::move(dtls_identity_store),
-                      prefer_constraint_apis)) {
+                      prefer_constraint_apis, network_thread, worker_thread)) {
       delete client;
       return nullptr;
     }
@@ -168,24 +170,30 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   static PeerConnectionTestClient* CreateClient(
       const std::string& id,
       const MediaConstraintsInterface* constraints,
-      const PeerConnectionFactory::Options* options) {
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+      const PeerConnectionFactory::Options* options,
+      rtc::Thread* network_thread,
+      rtc::Thread* worker_thread) {
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
 
     return CreateClientWithDtlsIdentityStore(
-        id, constraints, options, std::move(dtls_identity_store), true);
+        id, constraints, options, std::move(dtls_identity_store), true,
+        network_thread, worker_thread);
   }
 
   static PeerConnectionTestClient* CreateClientPreferNoConstraints(
       const std::string& id,
-      const PeerConnectionFactory::Options* options) {
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+      const PeerConnectionFactory::Options* options,
+      rtc::Thread* network_thread,
+      rtc::Thread* worker_thread) {
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
 
     return CreateClientWithDtlsIdentityStore(
-        id, nullptr, options, std::move(dtls_identity_store), false);
+        id, nullptr, options, std::move(dtls_identity_store), false,
+        network_thread, worker_thread);
   }
 
   ~PeerConnectionTestClient() {
@@ -194,7 +202,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   void Negotiate() { Negotiate(true, true); }
 
   void Negotiate(bool audio, bool video) {
-    rtc::scoped_ptr<SessionDescriptionInterface> offer;
+    std::unique_ptr<SessionDescriptionInterface> offer;
     ASSERT_TRUE(DoCreateOffer(&offer));
 
     if (offer->description()->GetContentByName("audio")) {
@@ -226,7 +234,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
                          int sdp_mline_index,
                          const std::string& msg) override {
     LOG(INFO) << id_ << "ReceiveIceMessage";
-    rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
+    std::unique_ptr<webrtc::IceCandidateInterface> candidate(
         webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
     EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
   }
@@ -544,7 +552,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
 
   // Verify the CreateDtmfSender interface
   void VerifyDtmf() {
-    rtc::scoped_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
+    std::unique_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
     rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
 
     // We can't create a DTMF sender with an invalid audio track or a non local
@@ -799,8 +807,10 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   bool Init(
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
-      rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
-      bool prefer_constraint_apis) {
+      std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
+      bool prefer_constraint_apis,
+      rtc::Thread* network_thread,
+      rtc::Thread* worker_thread) {
     EXPECT_TRUE(!peer_connection_);
     EXPECT_TRUE(!peer_connection_factory_);
     if (!prefer_constraint_apis) {
@@ -808,8 +818,8 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     }
     prefer_constraint_apis_ = prefer_constraint_apis;
 
-    rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
-        new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
+    std::unique_ptr<cricket::PortAllocator> port_allocator(
+        new cricket::FakePortAllocator(network_thread, nullptr));
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
 
     if (fake_audio_capture_module_ == nullptr) {
@@ -817,8 +827,9 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     }
     fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
     fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
+    rtc::Thread* const signaling_thread = rtc::Thread::Current();
     peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
-        rtc::Thread::Current(), rtc::Thread::Current(),
+        network_thread, worker_thread, signaling_thread,
         fake_audio_capture_module_, fake_video_encoder_factory_,
         fake_video_decoder_factory_);
     if (!peer_connection_factory_) {
@@ -833,9 +844,9 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   }
 
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
-      rtc::scoped_ptr<cricket::PortAllocator> port_allocator,
+      std::unique_ptr<cricket::PortAllocator> port_allocator,
       const MediaConstraintsInterface* constraints,
-      rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
+      std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
     // CreatePeerConnection with RTCConfiguration.
     webrtc::PeerConnectionInterface::RTCConfiguration config;
     webrtc::PeerConnectionInterface::IceServer ice_server;
@@ -853,10 +864,10 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
       // If we are not sending any streams ourselves it is time to add some.
       AddMediaStream(true, true);
     }
-    rtc::scoped_ptr<SessionDescriptionInterface> desc(
+    std::unique_ptr<SessionDescriptionInterface> desc(
         webrtc::CreateSessionDescription("offer", msg, nullptr));
     EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
-    rtc::scoped_ptr<SessionDescriptionInterface> answer;
+    std::unique_ptr<SessionDescriptionInterface> answer;
     EXPECT_TRUE(DoCreateAnswer(&answer));
     std::string sdp;
     EXPECT_TRUE(answer->ToString(&sdp));
@@ -869,12 +880,12 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
 
   void HandleIncomingAnswer(const std::string& msg) {
     LOG(INFO) << id_ << "HandleIncomingAnswer";
-    rtc::scoped_ptr<SessionDescriptionInterface> desc(
+    std::unique_ptr<SessionDescriptionInterface> desc(
         webrtc::CreateSessionDescription("answer", msg, nullptr));
     EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
   }
 
-  bool DoCreateOfferAnswer(rtc::scoped_ptr<SessionDescriptionInterface>* desc,
+  bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                            bool offer) {
     rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
         observer(new rtc::RefCountedObject<
@@ -900,11 +911,11 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     return observer->result();
   }
 
-  bool DoCreateOffer(rtc::scoped_ptr<SessionDescriptionInterface>* desc) {
+  bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc) {
     return DoCreateOfferAnswer(desc, true);
   }
 
-  bool DoCreateAnswer(rtc::scoped_ptr<SessionDescriptionInterface>* desc) {
+  bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc) {
     return DoCreateOfferAnswer(desc, false);
   }
 
@@ -977,10 +988,10 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   // Needed to keep track of number of frames sent.
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
   // Needed to keep track of number of frames received.
-  std::map<std::string, rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
+  std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
       fake_video_renderers_;
   // Needed to ensure frames aren't received for removed tracks.
-  std::vector<rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
+  std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
       removed_fake_video_renderers_;
   // Needed to keep track of number of frames received when external decoder
   // used.
@@ -997,7 +1008,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   std::vector<cricket::FakeVideoCapturer*> video_capturers_;
   webrtc::VideoRotation capture_rotation_ = webrtc::kVideoRotation_0;
   // |local_video_renderer_| attached to the first created local video track.
-  rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
+  std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
 
   webrtc::FakeConstraints offer_answer_constraints_;
   PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
@@ -1011,15 +1022,20 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   bool remove_cvo_ = false;
 
   rtc::scoped_refptr<DataChannelInterface> data_channel_;
-  rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
+  std::unique_ptr<MockDataChannelObserver> data_observer_;
 };
 
 class P2PTestConductor : public testing::Test {
  public:
   P2PTestConductor()
-      : pss_(new rtc::PhysicalSocketServer),
+      : network_thread_(rtc::Thread::CreateWithSocketServer()),
+        worker_thread_(rtc::Thread::Create()),
+        pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
-        ss_scope_(ss_.get()) {}
+        ss_scope_(ss_.get()) {
+    RTC_CHECK(network_thread_->Start());
+    RTC_CHECK(worker_thread_->Start());
+  }
 
   bool SessionActive() {
     return initiating_client_->SessionActive() &&
@@ -1127,11 +1143,11 @@ class P2PTestConductor : public testing::Test {
 
   bool CreateTestClientsThatPreferNoConstraints() {
     initiating_client_.reset(
-        PeerConnectionTestClient::CreateClientPreferNoConstraints("Caller: ",
-                                                                  nullptr));
+        PeerConnectionTestClient::CreateClientPreferNoConstraints(
+            "Caller: ", nullptr, network_thread_.get(), worker_thread_.get()));
     receiving_client_.reset(
-        PeerConnectionTestClient::CreateClientPreferNoConstraints("Callee: ",
-                                                                  nullptr));
+        PeerConnectionTestClient::CreateClientPreferNoConstraints(
+            "Callee: ", nullptr, network_thread_.get(), worker_thread_.get()));
     if (!initiating_client_ || !receiving_client_) {
       return false;
     }
@@ -1151,9 +1167,11 @@ class P2PTestConductor : public testing::Test {
                          MediaConstraintsInterface* recv_constraints,
                          PeerConnectionFactory::Options* recv_options) {
     initiating_client_.reset(PeerConnectionTestClient::CreateClient(
-        "Caller: ", init_constraints, init_options));
+        "Caller: ", init_constraints, init_options, network_thread_.get(),
+        worker_thread_.get()));
     receiving_client_.reset(PeerConnectionTestClient::CreateClient(
-        "Callee: ", recv_constraints, recv_options));
+        "Callee: ", recv_constraints, recv_options, network_thread_.get(),
+        worker_thread_.get()));
     if (!initiating_client_ || !receiving_client_) {
       return false;
     }
@@ -1246,7 +1264,7 @@ class P2PTestConductor : public testing::Test {
     setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                    true);
 
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
     dtls_identity_store->use_alternate_key();
@@ -1254,7 +1272,8 @@ class P2PTestConductor : public testing::Test {
     // Make sure the new client is using a different certificate.
     return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
         "New Peer: ", &setup_constraints, nullptr,
-        std::move(dtls_identity_store), prefer_constraint_apis_);
+        std::move(dtls_identity_store), prefer_constraint_apis_,
+        network_thread_.get(), worker_thread_.get());
   }
 
   void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
@@ -1294,11 +1313,15 @@ class P2PTestConductor : public testing::Test {
   }
 
  private:
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  // |worker_thread_| is used by both |initiating_client_| and
+  // |receiving_client_|. Must be destroyed last.
+  std::unique_ptr<rtc::Thread> network_thread_;
+  std::unique_ptr<rtc::Thread> worker_thread_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
-  rtc::scoped_ptr<PeerConnectionTestClient> initiating_client_;
-  rtc::scoped_ptr<PeerConnectionTestClient> receiving_client_;
+  std::unique_ptr<PeerConnectionTestClient> initiating_client_;
+  std::unique_ptr<PeerConnectionTestClient> receiving_client_;
   bool prefer_constraint_apis_ = true;
 };
 
@@ -1394,7 +1417,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
 
   // Keeping the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
-  rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
+  std::unique_ptr<PeerConnectionTestClient> original_peer(
       set_initializing_client(CreateDtlsClientWithAlternateKey()));
   original_peer->pc()->Close();
 
@@ -1432,7 +1455,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
 
   // Keeping the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
-  rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
+  std::unique_ptr<PeerConnectionTestClient> original_peer(
       set_receiving_client(CreateDtlsClientWithAlternateKey()));
   original_peer->pc()->Close();
 
@@ -1970,6 +1993,39 @@ TEST_F(P2PTestConductor, EarlyWarmupTest) {
                    kMaxWaitForFramesMs);
 }
 
+TEST_F(P2PTestConductor, ForwardVideoOnlyStream) {
+  ASSERT_TRUE(CreateTestClients());
+  // One-way stream
+  receiving_client()->set_auto_add_stream(false);
+  // Video only, audio forwarding not expected to work.
+  initializing_client()->AddMediaStream(false, true);
+  initializing_client()->Negotiate();
+
+  ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
+  VerifySessionDescriptions();
+
+  ASSERT_TRUE(initializing_client()->can_receive_video());
+  ASSERT_TRUE(receiving_client()->can_receive_video());
+
+  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+                 initializing_client()->ice_connection_state(),
+                 kMaxWaitForFramesMs);
+  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+                 receiving_client()->ice_connection_state(),
+                 kMaxWaitForFramesMs);
+
+  ASSERT_TRUE(receiving_client()->remote_streams()->count() == 1);
+
+  // Echo the stream back.
+  receiving_client()->pc()->AddStream(
+      receiving_client()->remote_streams()->at(0));
+  receiving_client()->Negotiate();
+
+  EXPECT_TRUE_WAIT(
+      initializing_client()->VideoFramesReceivedCheck(kEndVideoFrameCount),
+      kMaxWaitForFramesMs);
+}
+
 class IceServerParsingTest : public testing::Test {
  public:
   // Convenience for parsing a single URL.
diff --git a/chromium/third_party/webrtc/api/peerconnection_unittests.isolate b/chromium/third_party/webrtc/api/peerconnection_unittests.isolate
index bc7db9084de..ad03ccf5f2f 100644
--- a/chromium/third_party/webrtc/api/peerconnection_unittests.isolate
+++ b/chromium/third_party/webrtc/api/peerconnection_unittests.isolate
@@ -1,28 +1,10 @@
+# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
 #
-# libjingle
-# Copyright 2013 Google Inc.
-#
-# Redistribution and use in source and binary forms, with or without
-# modification, are permitted provided that the following conditions are met:
-#
-#  1. Redistributions of source code must retain the above copyright notice,
-#     this list of conditions and the following disclaimer.
-#  2. Redistributions in binary form must reproduce the above copyright notice,
-#     this list of conditions and the following disclaimer in the documentation
-#     and/or other materials provided with the distribution.
-#  3. The name of the author may not be used to endorse or promote products
-#     derived from this software without specific prior written permission.
-#
-# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
-# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
-# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
-# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
-# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
-# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
-# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
-# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
 
 {
   'conditions': [
diff --git a/chromium/third_party/webrtc/api/peerconnection_unittests_apk.isolate b/chromium/third_party/webrtc/api/peerconnection_unittests_apk.isolate
new file mode 100644
index 00000000000..3b895e88e4e
--- /dev/null
+++ b/chromium/third_party/webrtc/api/peerconnection_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'peerconnection_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_peerconnection_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/peerconnection_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_peerconnection_unittests',
+      'peerconnection_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/api/peerconnectionendtoend_unittest.cc b/chromium/third_party/webrtc/api/peerconnectionendtoend_unittest.cc
index be662886d3f..4ca73babdb7 100644
--- a/chromium/third_party/webrtc/api/peerconnectionendtoend_unittest.cc
+++ b/chromium/third_party/webrtc/api/peerconnectionendtoend_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/api/test/peerconnectiontestwrapper.h"
 // Notice that mockpeerconnectionobservers.h must be included after the above!
 #include "webrtc/api/test/mockpeerconnectionobservers.h"
@@ -17,6 +19,7 @@
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/ssladapter.h"
+#include "webrtc/base/thread.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
@@ -35,7 +38,7 @@ using webrtc::PeerConnectionInterface;
 
 namespace {
 
-const size_t kMaxWait = 10000;
+const int kMaxWait = 10000;
 
 }  // namespace
 
@@ -46,11 +49,13 @@ class PeerConnectionEndToEndTest
   typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
       DataChannelList;
 
-  PeerConnectionEndToEndTest()
-      : caller_(new rtc::RefCountedObject<PeerConnectionTestWrapper>(
-                    "caller")),
-        callee_(new rtc::RefCountedObject<PeerConnectionTestWrapper>(
-                    "callee")) {
+  PeerConnectionEndToEndTest() {
+    RTC_CHECK(network_thread_.Start());
+    RTC_CHECK(worker_thread_.Start());
+    caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
+        "caller", &network_thread_, &worker_thread_);
+    callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
+        "callee", &network_thread_, &worker_thread_);
 #ifdef WEBRTC_ANDROID
     webrtc::InitializeAndroidObjects();
 #endif
@@ -110,10 +115,10 @@ class PeerConnectionEndToEndTest
   // Tests that |dc1| and |dc2| can send to and receive from each other.
   void TestDataChannelSendAndReceive(
       DataChannelInterface* dc1, DataChannelInterface* dc2) {
-    rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc1_observer(
+    std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer(
         new webrtc::MockDataChannelObserver(dc1));
 
-    rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc2_observer(
+    std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer(
         new webrtc::MockDataChannelObserver(dc2));
 
     static const std::string kDummyData = "abcdefg";
@@ -151,6 +156,8 @@ class PeerConnectionEndToEndTest
   }
 
  protected:
+  rtc::Thread network_thread_;
+  rtc::Thread worker_thread_;
   rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
   rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
   DataChannelList caller_signaled_data_channels_;
@@ -291,10 +298,10 @@ TEST_F(PeerConnectionEndToEndTest,
   WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0);
   WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
 
-  rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
       new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
 
-  rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
       new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
 
   const std::string message_1 = "hello 1";
diff --git a/chromium/third_party/webrtc/api/peerconnectionfactory.cc b/chromium/third_party/webrtc/api/peerconnectionfactory.cc
index 852b7a8cce8..9cb5b46785c 100644
--- a/chromium/third_party/webrtc/api/peerconnectionfactory.cc
+++ b/chromium/third_party/webrtc/api/peerconnectionfactory.cc
@@ -74,19 +74,17 @@ CreatePeerConnectionFactory() {
                                             pc_factory);
 }
 
-rtc::scoped_refptr<PeerConnectionFactoryInterface>
-CreatePeerConnectionFactory(
+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
+    rtc::Thread* network_thread,
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
   rtc::scoped_refptr<PeerConnectionFactory> pc_factory(
-      new rtc::RefCountedObject<PeerConnectionFactory>(worker_thread,
-                                                       signaling_thread,
-                                                       default_adm,
-                                                       encoder_factory,
-                                                       decoder_factory));
+      new rtc::RefCountedObject<PeerConnectionFactory>(
+          network_thread, worker_thread, signaling_thread, default_adm,
+          encoder_factory, decoder_factory));
 
   // Call Initialize synchronously but make sure its executed on
   // |signaling_thread|.
@@ -104,16 +102,19 @@ CreatePeerConnectionFactory(
 PeerConnectionFactory::PeerConnectionFactory()
     : owns_ptrs_(true),
       wraps_current_thread_(false),
-      signaling_thread_(rtc::ThreadManager::Instance()->CurrentThread()),
-      worker_thread_(new rtc::Thread) {
+      network_thread_(rtc::Thread::CreateWithSocketServer().release()),
+      worker_thread_(rtc::Thread::Create().release()),
+      signaling_thread_(rtc::Thread::Current()) {
   if (!signaling_thread_) {
     signaling_thread_ = rtc::ThreadManager::Instance()->WrapCurrentThread();
     wraps_current_thread_ = true;
   }
+  network_thread_->Start();
   worker_thread_->Start();
 }
 
 PeerConnectionFactory::PeerConnectionFactory(
+    rtc::Thread* network_thread,
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
@@ -121,13 +122,15 @@ PeerConnectionFactory::PeerConnectionFactory(
     cricket::WebRtcVideoDecoderFactory* video_decoder_factory)
     : owns_ptrs_(false),
       wraps_current_thread_(false),
-      signaling_thread_(signaling_thread),
+      network_thread_(network_thread),
       worker_thread_(worker_thread),
+      signaling_thread_(signaling_thread),
       default_adm_(default_adm),
       video_encoder_factory_(video_encoder_factory),
       video_decoder_factory_(video_decoder_factory) {
-  ASSERT(worker_thread != NULL);
-  ASSERT(signaling_thread != NULL);
+  RTC_DCHECK(network_thread);
+  RTC_DCHECK(worker_thread);
+  RTC_DCHECK(signaling_thread);
   // TODO: Currently there is no way creating an external adm in
   // libjingle source tree. So we can 't currently assert if this is NULL.
   // ASSERT(default_adm != NULL);
@@ -148,12 +151,13 @@ PeerConnectionFactory::~PeerConnectionFactory() {
     if (wraps_current_thread_)
       rtc::ThreadManager::Instance()->UnwrapCurrentThread();
     delete worker_thread_;
+    delete network_thread_;
   }
 }
 
 bool PeerConnectionFactory::Initialize() {
   RTC_DCHECK(signaling_thread_->IsCurrent());
-  rtc::InitRandom(rtc::Time());
+  rtc::InitRandom(rtc::Time32());
 
   default_network_manager_.reset(new rtc::BasicNetworkManager());
   if (!default_network_manager_) {
@@ -161,7 +165,7 @@ bool PeerConnectionFactory::Initialize() {
   }
 
   default_socket_factory_.reset(
-      new rtc::BasicPacketSocketFactory(worker_thread_));
+      new rtc::BasicPacketSocketFactory(network_thread_));
   if (!default_socket_factory_) {
     return false;
   }
@@ -172,16 +176,16 @@ bool PeerConnectionFactory::Initialize() {
       worker_thread_->Invoke<cricket::MediaEngineInterface*>(rtc::Bind(
       &PeerConnectionFactory::CreateMediaEngine_w, this));
 
-  channel_manager_.reset(
-      new cricket::ChannelManager(media_engine, worker_thread_));
+  channel_manager_.reset(new cricket::ChannelManager(
+      media_engine, worker_thread_, network_thread_));
 
   channel_manager_->SetVideoRtxEnabled(true);
   if (!channel_manager_->Init()) {
     return false;
   }
 
-  dtls_identity_store_ = new RefCountedDtlsIdentityStore(
-      signaling_thread_, worker_thread_);
+  dtls_identity_store_ =
+      new RefCountedDtlsIdentityStore(signaling_thread_, network_thread_);
 
   return true;
 }
@@ -211,7 +215,8 @@ PeerConnectionFactory::CreateVideoSource(
   rtc::scoped_refptr<VideoTrackSourceInterface> source(
       VideoCapturerTrackSource::Create(worker_thread_, capturer, constraints,
                                        false));
-  return VideoTrackSourceProxy::Create(signaling_thread_, source);
+  return VideoTrackSourceProxy::Create(signaling_thread_, worker_thread_,
+                                       source);
 }
 
 rtc::scoped_refptr<VideoTrackSourceInterface>
@@ -219,7 +224,8 @@ PeerConnectionFactory::CreateVideoSource(cricket::VideoCapturer* capturer) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
   rtc::scoped_refptr<VideoTrackSourceInterface> source(
       VideoCapturerTrackSource::Create(worker_thread_, capturer, false));
-  return VideoTrackSourceProxy::Create(signaling_thread_, source);
+  return VideoTrackSourceProxy::Create(signaling_thread_, worker_thread_,
+                                       source);
 }
 
 bool PeerConnectionFactory::StartAecDump(rtc::PlatformFile file,
@@ -233,9 +239,10 @@ void PeerConnectionFactory::StopAecDump() {
   channel_manager_->StopAecDump();
 }
 
-bool PeerConnectionFactory::StartRtcEventLog(rtc::PlatformFile file) {
+bool PeerConnectionFactory::StartRtcEventLog(rtc::PlatformFile file,
+                                             int64_t max_size_bytes) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
-  return channel_manager_->StartRtcEventLog(file);
+  return channel_manager_->StartRtcEventLog(file, max_size_bytes);
 }
 
 void PeerConnectionFactory::StopRtcEventLog() {
@@ -247,8 +254,8 @@ rtc::scoped_refptr<PeerConnectionInterface>
 PeerConnectionFactory::CreatePeerConnection(
     const PeerConnectionInterface::RTCConfiguration& configuration_in,
     const MediaConstraintsInterface* constraints,
-    rtc::scoped_ptr<cricket::PortAllocator> allocator,
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+    std::unique_ptr<cricket::PortAllocator> allocator,
+    std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     PeerConnectionObserver* observer) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
 
@@ -263,8 +270,8 @@ PeerConnectionFactory::CreatePeerConnection(
 rtc::scoped_refptr<PeerConnectionInterface>
 PeerConnectionFactory::CreatePeerConnection(
     const PeerConnectionInterface::RTCConfiguration& configuration,
-    rtc::scoped_ptr<cricket::PortAllocator> allocator,
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+    std::unique_ptr<cricket::PortAllocator> allocator,
+    std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     PeerConnectionObserver* observer) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
 
@@ -280,28 +287,14 @@ PeerConnectionFactory::CreatePeerConnection(
     allocator.reset(new cricket::BasicPortAllocator(
         default_network_manager_.get(), default_socket_factory_.get()));
   }
-  allocator->SetNetworkIgnoreMask(options_.network_ignore_mask);
+  worker_thread_->Invoke<void>(
+      rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask, allocator.get(),
+                options_.network_ignore_mask));
 
   rtc::scoped_refptr<PeerConnection> pc(
       new rtc::RefCountedObject<PeerConnection>(this));
-  // We rely on default values when constraints aren't found.
-  cricket::MediaConfig media_config;
 
-  media_config.video.disable_prerenderer_smoothing =
-      configuration.disable_prerenderer_smoothing;
-  if (configuration.enable_dscp) {
-    media_config.enable_dscp = *(configuration.enable_dscp);
-  }
-  if (configuration.cpu_overuse_detection) {
-    media_config.video.enable_cpu_overuse_detection =
-        *(configuration.cpu_overuse_detection);
-  }
-  if (configuration.suspend_below_min_bitrate) {
-    media_config.video.suspend_below_min_bitrate =
-        *(configuration.suspend_below_min_bitrate);
-  }
-
-  if (!pc->Initialize(media_config, configuration, std::move(allocator),
+  if (!pc->Initialize(configuration, std::move(allocator),
                       std::move(dtls_identity_store), observer)) {
     return nullptr;
   }
@@ -321,7 +314,7 @@ rtc::scoped_refptr<VideoTrackInterface> PeerConnectionFactory::CreateVideoTrack(
   RTC_DCHECK(signaling_thread_->IsCurrent());
   rtc::scoped_refptr<VideoTrackInterface> track(
       VideoTrack::Create(id, source));
-  return VideoTrackProxy::Create(signaling_thread_, track);
+  return VideoTrackProxy::Create(signaling_thread_, worker_thread_, track);
 }
 
 rtc::scoped_refptr<AudioTrackInterface>
@@ -350,6 +343,10 @@ rtc::Thread* PeerConnectionFactory::worker_thread() {
   return worker_thread_;
 }
 
+rtc::Thread* PeerConnectionFactory::network_thread() {
+  return network_thread_;
+}
+
 cricket::MediaEngineInterface* PeerConnectionFactory::CreateMediaEngine_w() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   return cricket::WebRtcMediaEngineFactory::Create(
diff --git a/chromium/third_party/webrtc/api/peerconnectionfactory.h b/chromium/third_party/webrtc/api/peerconnectionfactory.h
index b47f75a7ce9..21165cf3d28 100644
--- a/chromium/third_party/webrtc/api/peerconnectionfactory.h
+++ b/chromium/third_party/webrtc/api/peerconnectionfactory.h
@@ -11,13 +11,13 @@
 #ifndef WEBRTC_API_PEERCONNECTIONFACTORY_H_
 #define WEBRTC_API_PEERCONNECTIONFACTORY_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/api/dtlsidentitystore.h"
 #include "webrtc/api/mediacontroller.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/pc/channelmanager.h"
@@ -34,7 +34,7 @@ typedef rtc::RefCountedObject<DtlsIdentityStoreImpl>
 
 class PeerConnectionFactory : public PeerConnectionFactoryInterface {
  public:
-  virtual void SetOptions(const Options& options) {
+  void SetOptions(const Options& options) override {
     options_ = options;
   }
 
@@ -42,14 +42,14 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
   rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& configuration,
       const MediaConstraintsInterface* constraints,
-      rtc::scoped_ptr<cricket::PortAllocator> allocator,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<cricket::PortAllocator> allocator,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer) override;
 
   virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& configuration,
-      rtc::scoped_ptr<cricket::PortAllocator> allocator,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<cricket::PortAllocator> allocator,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer) override;
 
   bool Initialize();
@@ -83,18 +83,24 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
 
   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
   void StopAecDump() override;
-  bool StartRtcEventLog(rtc::PlatformFile file) override;
+  bool StartRtcEventLog(rtc::PlatformFile file) override {
+    return StartRtcEventLog(file, -1);
+  }
+  bool StartRtcEventLog(rtc::PlatformFile file,
+                        int64_t max_size_bytes) override;
   void StopRtcEventLog() override;
 
   virtual webrtc::MediaControllerInterface* CreateMediaController(
       const cricket::MediaConfig& config) const;
   virtual rtc::Thread* signaling_thread();
   virtual rtc::Thread* worker_thread();
+  virtual rtc::Thread* network_thread();
   const Options& options() const { return options_; }
 
  protected:
   PeerConnectionFactory();
   PeerConnectionFactory(
+      rtc::Thread* network_thread,
       rtc::Thread* worker_thread,
       rtc::Thread* signaling_thread,
       AudioDeviceModule* default_adm,
@@ -107,22 +113,21 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
 
   bool owns_ptrs_;
   bool wraps_current_thread_;
-  rtc::Thread* signaling_thread_;
+  rtc::Thread* network_thread_;
   rtc::Thread* worker_thread_;
+  rtc::Thread* signaling_thread_;
   Options options_;
   // External Audio device used for audio playback.
   rtc::scoped_refptr<AudioDeviceModule> default_adm_;
-  rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
+  std::unique_ptr<cricket::ChannelManager> channel_manager_;
   // External Video encoder factory. This can be NULL if the client has not
   // injected any. In that case, video engine will use the internal SW encoder.
-  rtc::scoped_ptr<cricket::WebRtcVideoEncoderFactory>
-      video_encoder_factory_;
+  std::unique_ptr<cricket::WebRtcVideoEncoderFactory> video_encoder_factory_;
   // External Video decoder factory. This can be NULL if the client has not
   // injected any. In that case, video engine will use the internal SW decoder.
-  rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
-      video_decoder_factory_;
-  rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
-  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
+  std::unique_ptr<cricket::WebRtcVideoDecoderFactory> video_decoder_factory_;
+  std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_;
+  std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
 
   rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
 };
diff --git a/chromium/third_party/webrtc/api/peerconnectionfactory_unittest.cc b/chromium/third_party/webrtc/api/peerconnectionfactory_unittest.cc
index 254e4e0a025..de21e806d55 100644
--- a/chromium/third_party/webrtc/api/peerconnectionfactory_unittest.cc
+++ b/chromium/third_party/webrtc/api/peerconnectionfactory_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 #include <utility>
 
@@ -19,12 +20,11 @@
 #include "webrtc/api/test/fakedtlsidentitystore.h"
 #include "webrtc/api/test/fakevideotrackrenderer.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/engine/webrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
+#include "webrtc/p2p/base/fakeportallocator.h"
 
 using webrtc::DataChannelInterface;
 using webrtc::DtlsIdentityStoreInterface;
@@ -64,6 +64,7 @@ static const char kTurnIceServerWithIPv6Address[] =
 
 class NullPeerConnectionObserver : public PeerConnectionObserver {
  public:
+  virtual ~NullPeerConnectionObserver() = default;
   virtual void OnMessage(const std::string& msg) {}
   virtual void OnSignalingMessage(const std::string& msg) {}
   virtual void OnSignalingChange(
@@ -86,11 +87,9 @@ class PeerConnectionFactoryTest : public testing::Test {
 #ifdef WEBRTC_ANDROID
     webrtc::InitializeAndroidObjects();
 #endif
-    factory_ = webrtc::CreatePeerConnectionFactory(rtc::Thread::Current(),
-                                                   rtc::Thread::Current(),
-                                                   NULL,
-                                                   NULL,
-                                                   NULL);
+    factory_ = webrtc::CreatePeerConnectionFactory(
+        rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
+        nullptr, nullptr, nullptr);
 
     ASSERT_TRUE(factory_.get() != NULL);
     port_allocator_.reset(
@@ -122,7 +121,7 @@ class PeerConnectionFactoryTest : public testing::Test {
 
   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_;
   NullPeerConnectionObserver observer_;
-  rtc::scoped_ptr<cricket::FakePortAllocator> port_allocator_;
+  std::unique_ptr<cricket::FakePortAllocator> port_allocator_;
   // Since the PC owns the port allocator after it's been initialized,
   // this should only be used when known to be safe.
   cricket::FakePortAllocator* raw_port_allocator_;
@@ -141,7 +140,7 @@ TEST(PeerConnectionFactoryTestInternal, CreatePCUsingInternalModules) {
   NullPeerConnectionObserver observer;
   webrtc::PeerConnectionInterface::RTCConfiguration config;
 
-  rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+  std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
       new FakeDtlsIdentityStore());
   rtc::scoped_refptr<PeerConnectionInterface> pc(factory->CreatePeerConnection(
       config, nullptr, nullptr, std::move(dtls_identity_store), &observer));
@@ -162,7 +161,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
   ice_server.uri = kTurnIceServerWithTransport;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
+  std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
       new FakeDtlsIdentityStore());
   rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
       config, nullptr, std::move(port_allocator_),
@@ -192,7 +191,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) {
   ice_server.urls.push_back(kTurnIceServerWithTransport);
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
+  std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
       new FakeDtlsIdentityStore());
   rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
       config, nullptr, std::move(port_allocator_),
@@ -221,7 +220,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) {
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
+  std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
       new FakeDtlsIdentityStore());
   rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
       config, nullptr, std::move(port_allocator_),
@@ -242,7 +241,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) {
   ice_server.uri = kTurnIceServerWithTransport;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
+  std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
       new FakeDtlsIdentityStore());
   rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
       config, nullptr, std::move(port_allocator_),
@@ -267,7 +266,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
   ice_server.uri = kSecureTurnIceServerWithoutTransportAndPortParam;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
+  std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
       new FakeDtlsIdentityStore());
   rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
       config, nullptr, std::move(port_allocator_),
@@ -302,7 +301,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
   ice_server.uri = kTurnIceServerWithIPv6Address;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
+  std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
       new FakeDtlsIdentityStore());
   rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
       config, nullptr, std::move(port_allocator_),
diff --git a/chromium/third_party/webrtc/api/peerconnectionfactoryproxy.h b/chromium/third_party/webrtc/api/peerconnectionfactoryproxy.h
index 829bf8177e1..c357de9b32b 100644
--- a/chromium/third_party/webrtc/api/peerconnectionfactoryproxy.h
+++ b/chromium/third_party/webrtc/api/peerconnectionfactoryproxy.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
 #define WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
 
+#include <memory>
 #include <string>
 #include <utility>
 
@@ -20,28 +21,30 @@
 
 namespace webrtc {
 
-BEGIN_PROXY_MAP(PeerConnectionFactory)
+BEGIN_SIGNALING_PROXY_MAP(PeerConnectionFactory)
   PROXY_METHOD1(void, SetOptions, const Options&)
-  // Can't use PROXY_METHOD5 because scoped_ptr must be moved.
-  // TODO(tommi,hbos): Use of templates to support scoped_ptr?
+  // Can't use PROXY_METHOD5 because unique_ptr must be moved.
+  // TODO(tommi,hbos): Use of templates to support unique_ptr?
   rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& a1,
       const MediaConstraintsInterface* a2,
-      rtc::scoped_ptr<cricket::PortAllocator> a3,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> a4,
+      std::unique_ptr<cricket::PortAllocator> a3,
+      std::unique_ptr<DtlsIdentityStoreInterface> a4,
       PeerConnectionObserver* a5) override {
-    return owner_thread_->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
-        rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot, this,
-                  a1, a2, a3.release(), a4.release(), a5));
+    return signaling_thread_
+        ->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
+            rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot,
+                      this, a1, a2, a3.release(), a4.release(), a5));
   }
   rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& a1,
-      rtc::scoped_ptr<cricket::PortAllocator> a3,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> a4,
+      std::unique_ptr<cricket::PortAllocator> a3,
+      std::unique_ptr<DtlsIdentityStoreInterface> a4,
       PeerConnectionObserver* a5) override {
-    return owner_thread_->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
-        rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot, this,
-                  a1, a3.release(), a4.release(), a5));
+    return signaling_thread_
+        ->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
+            rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot,
+                      this, a1, a3.release(), a4.release(), a5));
   }
   PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
                 CreateLocalMediaStream, const std::string&)
@@ -66,6 +69,7 @@ BEGIN_PROXY_MAP(PeerConnectionFactory)
   PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
   PROXY_METHOD0(void, StopAecDump)
   PROXY_METHOD1(bool, StartRtcEventLog, rtc::PlatformFile)
+  PROXY_METHOD2(bool, StartRtcEventLog, rtc::PlatformFile, int64_t)
   PROXY_METHOD0(void, StopRtcEventLog)
 
  private:
@@ -75,8 +79,8 @@ BEGIN_PROXY_MAP(PeerConnectionFactory)
       cricket::PortAllocator* a3,
       DtlsIdentityStoreInterface* a4,
       PeerConnectionObserver* a5) {
-    rtc::scoped_ptr<cricket::PortAllocator> ptr_a3(a3);
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
+    std::unique_ptr<cricket::PortAllocator> ptr_a3(a3);
+    std::unique_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
     return c_->CreatePeerConnection(a1, a2, std::move(ptr_a3),
                                     std::move(ptr_a4), a5);
   }
@@ -86,12 +90,12 @@ BEGIN_PROXY_MAP(PeerConnectionFactory)
       cricket::PortAllocator* a3,
       DtlsIdentityStoreInterface* a4,
       PeerConnectionObserver* a5) {
-    rtc::scoped_ptr<cricket::PortAllocator> ptr_a3(a3);
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
+    std::unique_ptr<cricket::PortAllocator> ptr_a3(a3);
+    std::unique_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
     return c_->CreatePeerConnection(a1, std::move(ptr_a3), std::move(ptr_a4),
                                     a5);
   }
-  END_PROXY()
+  END_SIGNALING_PROXY()
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/peerconnectioninterface.h b/chromium/third_party/webrtc/api/peerconnectioninterface.h
index 9259275b86a..4fa9bf2408d 100644
--- a/chromium/third_party/webrtc/api/peerconnectioninterface.h
+++ b/chromium/third_party/webrtc/api/peerconnectioninterface.h
@@ -51,13 +51,13 @@
 #ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_
 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
 
+#include <memory>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/datachannelinterface.h"
 #include "webrtc/api/dtlsidentitystore.h"
-#include "webrtc/api/dtlsidentitystore.h"
 #include "webrtc/api/dtmfsenderinterface.h"
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/mediastreaminterface.h"
@@ -70,6 +70,7 @@
 #include "webrtc/base/rtccertificate.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/sslstreamadapter.h"
+#include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/portallocator.h"
 
 namespace rtc {
@@ -222,6 +223,11 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
   };
 
   // TODO(hbos): Change into class with private data and public getters.
+  // TODO(nisse): In particular, accessing fields directly from an
+  // application is brittle, since the organization mirrors the
+  // organization of the implementation, which isn't stable. So we
+  // need getters and setters at least for fields which applications
+  // are interested in.
   struct RTCConfiguration {
     // This struct is subject to reorganization, both for naming
     // consistency, and to group settings to match where they are used
@@ -229,28 +235,33 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
     // methods for all settings which are of interest to applications,
     // Chrome in particular.
 
-    bool dscp() { return enable_dscp.value_or(false); }
-    void set_dscp(bool enable) { enable_dscp = rtc::Optional<bool>(enable); }
+    bool dscp() { return media_config.enable_dscp; }
+    void set_dscp(bool enable) { media_config.enable_dscp = enable; }
 
     // TODO(nisse): The corresponding flag in MediaConfig and
     // elsewhere should be renamed enable_cpu_adaptation.
-    bool cpu_adaptation() { return cpu_overuse_detection.value_or(true); }
+    bool cpu_adaptation() {
+      return media_config.video.enable_cpu_overuse_detection;
+    }
     void set_cpu_adaptation(bool enable) {
-      cpu_overuse_detection = rtc::Optional<bool>(enable);
+      media_config.video.enable_cpu_overuse_detection = enable;
     }
 
-    // TODO(nisse): Currently no getter method, since it collides with
-    // the flag itself. Add when the flag is moved to MediaConfig.
+    bool suspend_below_min_bitrate() {
+      return media_config.video.suspend_below_min_bitrate;
+    }
     void set_suspend_below_min_bitrate(bool enable) {
-      suspend_below_min_bitrate = rtc::Optional<bool>(enable);
+      media_config.video.suspend_below_min_bitrate = enable;
     }
 
     // TODO(nisse): The negation in the corresponding MediaConfig
     // attribute is inconsistent, and it should be renamed at some
     // point.
-    bool prerenderer_smoothing() { return !disable_prerenderer_smoothing; }
+    bool prerenderer_smoothing() {
+      return !media_config.video.disable_prerenderer_smoothing;
+    }
     void set_prerenderer_smoothing(bool enable) {
-      disable_prerenderer_smoothing = !enable;
+      media_config.video.disable_prerenderer_smoothing = !enable;
     }
 
     static const int kUndefined = -1;
@@ -258,46 +269,30 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
     static const int kAudioJitterBufferMaxPackets = 50;
     // TODO(pthatcher): Rename this ice_transport_type, but update
     // Chromium at the same time.
-    IceTransportsType type;
+    IceTransportsType type = kAll;
     // TODO(pthatcher): Rename this ice_servers, but update Chromium
     // at the same time.
     IceServers servers;
-    BundlePolicy bundle_policy;
-    RtcpMuxPolicy rtcp_mux_policy;
-    TcpCandidatePolicy tcp_candidate_policy;
-    int audio_jitter_buffer_max_packets;
-    bool audio_jitter_buffer_fast_accelerate;
-    int ice_connection_receiving_timeout;         // ms
-    int ice_backup_candidate_pair_ping_interval;  // ms
-    ContinualGatheringPolicy continual_gathering_policy;
+    BundlePolicy bundle_policy = kBundlePolicyBalanced;
+    RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyNegotiate;
+    TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
+    int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;
+    bool audio_jitter_buffer_fast_accelerate = false;
+    int ice_connection_receiving_timeout = kUndefined;         // ms
+    int ice_backup_candidate_pair_ping_interval = kUndefined;  // ms
+    ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
     std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
-    bool disable_prerenderer_smoothing;
-    bool prioritize_most_likely_ice_candidate_pairs;
+    bool prioritize_most_likely_ice_candidate_pairs = false;
+    struct cricket::MediaConfig media_config;
     // Flags corresponding to values set by constraint flags.
     // rtc::Optional flags can be "missing", in which case the webrtc
     // default applies.
-    bool disable_ipv6;
-    rtc::Optional<bool> enable_dscp;
-    bool enable_rtp_data_channel;
-    rtc::Optional<bool> cpu_overuse_detection;
-    rtc::Optional<bool> suspend_below_min_bitrate;
+    bool disable_ipv6 = false;
+    bool enable_rtp_data_channel = false;
     rtc::Optional<int> screencast_min_bitrate;
     rtc::Optional<bool> combined_audio_video_bwe;
     rtc::Optional<bool> enable_dtls_srtp;
-    RTCConfiguration()
-        : type(kAll),
-          bundle_policy(kBundlePolicyBalanced),
-          rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
-          tcp_candidate_policy(kTcpCandidatePolicyEnabled),
-          audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
-          audio_jitter_buffer_fast_accelerate(false),
-          ice_connection_receiving_timeout(kUndefined),
-          ice_backup_candidate_pair_ping_interval(kUndefined),
-          continual_gathering_policy(GATHER_ONCE),
-          disable_prerenderer_smoothing(false),
-          prioritize_most_likely_ice_candidate_pairs(false),
-          disable_ipv6(false),
-          enable_rtp_data_channel(false) {}
+    int ice_candidate_pool_size = 0;
   };
 
   struct RTCOfferAnswerOptions {
@@ -584,14 +579,14 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
   virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& configuration,
       const MediaConstraintsInterface* constraints,
-      rtc::scoped_ptr<cricket::PortAllocator> allocator,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<cricket::PortAllocator> allocator,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer) = 0;
 
   virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& configuration,
-      rtc::scoped_ptr<cricket::PortAllocator> allocator,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<cricket::PortAllocator> allocator,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer) = 0;
 
   virtual rtc::scoped_refptr<MediaStreamInterface>
@@ -644,13 +639,18 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
   // passes it on to VoiceEngine, which will take the ownership. If the
   // operation fails the file will be closed. The logging will stop
   // automatically after 10 minutes have passed, or when the StopRtcEventLog
-  // function is called.
+  // function is called. A maximum filesize in bytes can be set, the logging
+  // will be stopped before exceeding this limit. If max_size_bytes is set to a
+  // value <= 0, no limit will be used.
   // This function as well as the StopRtcEventLog don't really belong on this
   // interface, this is a temporary solution until we move the logging object
   // from inside voice engine to webrtc::Call, which will happen when the VoE
   // restructuring effort is further along.
   // TODO(ivoc): Move this into being:
   //             PeerConnection => MediaController => webrtc::Call.
+  virtual bool StartRtcEventLog(rtc::PlatformFile file,
+                                int64_t max_size_bytes) = 0;
+  // Deprecated, use the version above.
   virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
 
   // Stops logging the RtcEventLog.
@@ -678,19 +678,33 @@ CreatePeerConnectionFactory();
 
 // Create a new instance of PeerConnectionFactoryInterface.
 //
-// |worker_thread| and |signaling_thread| are the only mandatory
-// parameters.
+// |network_thread|, |worker_thread| and |signaling_thread| are
+// the only mandatory parameters.
 //
 // If non-null, ownership of |default_adm|, |encoder_factory| and
 // |decoder_factory| are transferred to the returned factory.
-rtc::scoped_refptr<PeerConnectionFactoryInterface>
-CreatePeerConnectionFactory(
+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
+    rtc::Thread* network_thread,
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
+// Create a new instance of PeerConnectionFactoryInterface.
+// Same thread is used as worker and network thread.
+inline rtc::scoped_refptr<PeerConnectionFactoryInterface>
+CreatePeerConnectionFactory(
+    rtc::Thread* worker_and_network_thread,
+    rtc::Thread* signaling_thread,
+    AudioDeviceModule* default_adm,
+    cricket::WebRtcVideoEncoderFactory* encoder_factory,
+    cricket::WebRtcVideoDecoderFactory* decoder_factory) {
+  return CreatePeerConnectionFactory(
+      worker_and_network_thread, worker_and_network_thread, signaling_thread,
+      default_adm, encoder_factory, decoder_factory);
+}
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
diff --git a/chromium/third_party/webrtc/api/peerconnectioninterface_unittest.cc b/chromium/third_party/webrtc/api/peerconnectioninterface_unittest.cc
index 14a067995b4..445f0257c12 100644
--- a/chromium/third_party/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/chromium/third_party/webrtc/api/peerconnectioninterface_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 #include <utility>
 
@@ -32,14 +33,13 @@
 #include "webrtc/api/videocapturertracksource.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/sctp/sctpdataengine.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
+#include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/pc/mediasession.h"
 
 static const char kStreamLabel1[] = "local_stream_1";
@@ -239,9 +239,9 @@ static const char kSdpStringMs1Video1[] =
     return;                                         \
   }
 
-using rtc::scoped_ptr;
-using rtc::scoped_refptr;
 using ::testing::Exactly;
+using cricket::StreamParams;
+using rtc::scoped_refptr;
 using webrtc::AudioSourceInterface;
 using webrtc::AudioTrack;
 using webrtc::AudioTrackInterface;
@@ -249,6 +249,7 @@ using webrtc::DataBuffer;
 using webrtc::DataChannelInterface;
 using webrtc::FakeConstraints;
 using webrtc::IceCandidateInterface;
+using webrtc::JsepSessionDescription;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStream;
 using webrtc::MediaStreamInterface;
@@ -326,12 +327,26 @@ bool ContainsSender(
   return false;
 }
 
+// Check if |senders| contains the specified sender, by id and stream id.
+bool ContainsSender(
+    const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
+    const std::string& id,
+    const std::string& stream_id) {
+  for (const auto& sender : senders) {
+    if (sender->id() == id && sender->stream_id() == stream_id) {
+      return true;
+    }
+  }
+  return false;
+}
+
 // Create a collection of streams.
 // CreateStreamCollection(1) creates a collection that
 // correspond to kSdpStringWithStream1.
 // CreateStreamCollection(2) correspond to kSdpStringWithStream1And2.
 rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
-    int number_of_streams) {
+    int number_of_streams,
+    int tracks_per_stream) {
   rtc::scoped_refptr<StreamCollection> local_collection(
       StreamCollection::Create());
 
@@ -339,16 +354,19 @@ rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
     rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
         webrtc::MediaStream::Create(kStreams[i]));
 
-    // Add a local audio track.
-    rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-        webrtc::AudioTrack::Create(kAudioTracks[i], nullptr));
-    stream->AddTrack(audio_track);
-
-    // Add a local video track.
-    rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
-        webrtc::VideoTrack::Create(kVideoTracks[i],
-                                   webrtc::FakeVideoTrackSource::Create()));
-    stream->AddTrack(video_track);
+    for (int j = 0; j < tracks_per_stream; ++j) {
+      // Add a local audio track.
+      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+          webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
+                                     nullptr));
+      stream->AddTrack(audio_track);
+
+      // Add a local video track.
+      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
+          webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
+                                     webrtc::FakeVideoTrackSource::Create()));
+      stream->AddTrack(video_track);
+    }
 
     local_collection->AddStream(stream);
   }
@@ -417,7 +435,7 @@ class MockTrackObserver : public ObserverInterface {
 class MockPeerConnectionObserver : public PeerConnectionObserver {
  public:
   MockPeerConnectionObserver() : remote_streams_(StreamCollection::Create()) {}
-  ~MockPeerConnectionObserver() {
+  virtual ~MockPeerConnectionObserver() {
   }
   void SetPeerConnectionInterface(PeerConnectionInterface* pc) {
     pc_ = pc;
@@ -425,8 +443,8 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
       state_ = pc_->signaling_state();
     }
   }
-  virtual void OnSignalingChange(
-      PeerConnectionInterface::SignalingState new_state) {
+  void OnSignalingChange(
+      PeerConnectionInterface::SignalingState new_state) override {
     EXPECT_EQ(pc_->signaling_state(), new_state);
     state_ = new_state;
   }
@@ -504,7 +522,7 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
 
   scoped_refptr<PeerConnectionInterface> pc_;
   PeerConnectionInterface::SignalingState state_;
-  scoped_ptr<IceCandidateInterface> last_candidate_;
+  std::unique_ptr<IceCandidateInterface> last_candidate_;
   scoped_refptr<DataChannelInterface> last_datachannel_;
   rtc::scoped_refptr<StreamCollection> remote_streams_;
   bool renegotiation_needed_ = false;
@@ -527,31 +545,40 @@ class PeerConnectionInterfaceTest : public testing::Test {
 
   virtual void SetUp() {
     pc_factory_ = webrtc::CreatePeerConnectionFactory(
-        rtc::Thread::Current(), rtc::Thread::Current(), NULL, NULL,
-        NULL);
-    ASSERT_TRUE(pc_factory_.get() != NULL);
+        rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
+        nullptr, nullptr, nullptr);
+    ASSERT_TRUE(pc_factory_);
   }
 
   void CreatePeerConnection() {
-    CreatePeerConnection("", "", NULL);
+    CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(), nullptr);
   }
 
   void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
-    CreatePeerConnection("", "", constraints);
+    CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
+                         constraints);
   }
 
-  void CreatePeerConnection(const std::string& uri,
-                            const std::string& password,
-                            webrtc::MediaConstraintsInterface* constraints) {
+  void CreatePeerConnectionWithIceTransportsType(
+      PeerConnectionInterface::IceTransportsType type) {
+    PeerConnectionInterface::RTCConfiguration config;
+    config.type = type;
+    return CreatePeerConnection(config, nullptr);
+  }
+
+  void CreatePeerConnectionWithIceServer(const std::string& uri,
+                                         const std::string& password) {
     PeerConnectionInterface::RTCConfiguration config;
     PeerConnectionInterface::IceServer server;
-    if (!uri.empty()) {
-      server.uri = uri;
-      server.password = password;
-      config.servers.push_back(server);
-    }
+    server.uri = uri;
+    server.password = password;
+    config.servers.push_back(server);
+    CreatePeerConnection(config, nullptr);
+  }
 
-    rtc::scoped_ptr<cricket::FakePortAllocator> port_allocator(
+  void CreatePeerConnection(PeerConnectionInterface::RTCConfiguration config,
+                            webrtc::MediaConstraintsInterface* constraints) {
+    std::unique_ptr<cricket::FakePortAllocator> port_allocator(
         new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
     port_allocator_ = port_allocator.get();
 
@@ -566,7 +593,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
           webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
     }
 
-    scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store;
+    std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store;
     bool dtls;
     if (FindConstraint(constraints,
                        webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -595,7 +622,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
   }
 
   void CreatePeerConnectionWithDifferentConfigurations() {
-    CreatePeerConnection(kStunAddressOnly, "", NULL);
+    CreatePeerConnectionWithIceServer(kStunAddressOnly, "");
     EXPECT_EQ(1u, port_allocator_->stun_servers().size());
     EXPECT_EQ(0u, port_allocator_->turn_servers().size());
     EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname());
@@ -606,7 +633,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
     CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
     CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
 
-    CreatePeerConnection(kTurnIceServerUri, kTurnPassword, NULL);
+    CreatePeerConnectionWithIceServer(kTurnIceServerUri, kTurnPassword);
     EXPECT_EQ(0u, port_allocator_->stun_servers().size());
     EXPECT_EQ(1u, port_allocator_->turn_servers().size());
     EXPECT_EQ(kTurnUsername,
@@ -668,7 +695,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
     observer_.renegotiation_needed_ = false;
   }
 
-  bool DoCreateOfferAnswer(rtc::scoped_ptr<SessionDescriptionInterface>* desc,
+  bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                            bool offer,
                            MediaConstraintsInterface* constraints) {
     rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
@@ -684,12 +711,12 @@ class PeerConnectionInterfaceTest : public testing::Test {
     return observer->result();
   }
 
-  bool DoCreateOffer(rtc::scoped_ptr<SessionDescriptionInterface>* desc,
+  bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc,
                      MediaConstraintsInterface* constraints) {
     return DoCreateOfferAnswer(desc, true, constraints);
   }
 
-  bool DoCreateAnswer(rtc::scoped_ptr<SessionDescriptionInterface>* desc,
+  bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                       MediaConstraintsInterface* constraints) {
     return DoCreateOfferAnswer(desc, false, constraints);
   }
@@ -750,7 +777,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
   }
 
   void CreateOfferAsRemoteDescription() {
-    rtc::scoped_ptr<SessionDescriptionInterface> offer;
+    std::unique_ptr<SessionDescriptionInterface> offer;
     ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
     std::string sdp;
     EXPECT_TRUE(offer->ToString(&sdp));
@@ -770,7 +797,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
   }
 
   void CreateAnswerAsLocalDescription() {
-    scoped_ptr<SessionDescriptionInterface> answer;
+    std::unique_ptr<SessionDescriptionInterface> answer;
     ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
 
     // TODO(perkj): Currently SetLocalDescription fails if any parameters in an
@@ -790,7 +817,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
   }
 
   void CreatePrAnswerAsLocalDescription() {
-    scoped_ptr<SessionDescriptionInterface> answer;
+    std::unique_ptr<SessionDescriptionInterface> answer;
     ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
 
     std::string sdp;
@@ -810,7 +837,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
   }
 
   void CreateOfferAsLocalDescription() {
-    rtc::scoped_ptr<SessionDescriptionInterface> offer;
+    std::unique_ptr<SessionDescriptionInterface> offer;
     ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
     // TODO(perkj): Currently SetLocalDescription fails if any parameters in an
     // audio codec change, even if the parameter has nothing to do with
@@ -880,7 +907,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
   // corresponding SessionDescriptionInterface. The SessionDescriptionInterface
   // is returned and the MediaStream is stored in
   // |reference_collection_|
-  rtc::scoped_ptr<SessionDescriptionInterface>
+  std::unique_ptr<SessionDescriptionInterface>
   CreateSessionDescriptionAndReference(size_t number_of_audio_tracks,
                                        size_t number_of_video_tracks) {
     EXPECT_LE(number_of_audio_tracks, 2u);
@@ -915,7 +942,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
       AddVideoTrack(kVideoTracks[1], stream);
     }
 
-    return rtc::scoped_ptr<SessionDescriptionInterface>(
+    return std::unique_ptr<SessionDescriptionInterface>(
         webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                          sdp_ms1, nullptr));
   }
@@ -935,6 +962,34 @@ class PeerConnectionInterfaceTest : public testing::Test {
     ASSERT_TRUE(stream->AddTrack(video_track));
   }
 
+  std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
+    CreatePeerConnection();
+    AddVoiceStream(kStreamLabel1);
+    std::unique_ptr<SessionDescriptionInterface> offer;
+    EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
+    return offer;
+  }
+
+  std::unique_ptr<SessionDescriptionInterface>
+  CreateAnswerWithOneAudioStream() {
+    std::unique_ptr<SessionDescriptionInterface> offer =
+        CreateOfferWithOneAudioStream();
+    EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
+    std::unique_ptr<SessionDescriptionInterface> answer;
+    EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
+    return answer;
+  }
+
+  const std::string& GetFirstAudioStreamCname(
+      const SessionDescriptionInterface* desc) {
+    const cricket::ContentInfo* audio_content =
+        cricket::GetFirstAudioContent(desc->description());
+    const cricket::AudioContentDescription* audio_desc =
+        static_cast<const cricket::AudioContentDescription*>(
+            audio_content->description);
+    return audio_desc->streams()[0].cname;
+  }
+
   cricket::FakePortAllocator* port_allocator_ = nullptr;
   scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
   scoped_refptr<PeerConnectionInterface> pc_;
@@ -942,11 +997,70 @@ class PeerConnectionInterfaceTest : public testing::Test {
   rtc::scoped_refptr<StreamCollection> reference_collection_;
 };
 
+// Generate different CNAMEs when PeerConnections are created.
+// The CNAMEs are expected to be generated randomly. It is possible
+// that the test fails, though the possibility is very low.
+TEST_F(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
+  std::unique_ptr<SessionDescriptionInterface> offer1 =
+      CreateOfferWithOneAudioStream();
+  std::unique_ptr<SessionDescriptionInterface> offer2 =
+      CreateOfferWithOneAudioStream();
+  EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
+            GetFirstAudioStreamCname(offer2.get()));
+}
+
+TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
+  std::unique_ptr<SessionDescriptionInterface> answer1 =
+      CreateAnswerWithOneAudioStream();
+  std::unique_ptr<SessionDescriptionInterface> answer2 =
+      CreateAnswerWithOneAudioStream();
+  EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
+            GetFirstAudioStreamCname(answer2.get()));
+}
+
 TEST_F(PeerConnectionInterfaceTest,
        CreatePeerConnectionWithDifferentConfigurations) {
   CreatePeerConnectionWithDifferentConfigurations();
 }
 
+TEST_F(PeerConnectionInterfaceTest,
+       CreatePeerConnectionWithDifferentIceTransportsTypes) {
+  CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNone);
+  EXPECT_EQ(cricket::CF_NONE, port_allocator_->candidate_filter());
+  CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kRelay);
+  EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
+  CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNoHost);
+  EXPECT_EQ(cricket::CF_ALL & ~cricket::CF_HOST,
+            port_allocator_->candidate_filter());
+  CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kAll);
+  EXPECT_EQ(cricket::CF_ALL, port_allocator_->candidate_filter());
+}
+
+// Test that when a PeerConnection is created with a nonzero candidate pool
+// size, the pooled PortAllocatorSession is created with all the attributes
+// in the RTCConfiguration.
+TEST_F(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
+  PeerConnectionInterface::RTCConfiguration config;
+  PeerConnectionInterface::IceServer server;
+  server.uri = kStunAddressOnly;
+  config.servers.push_back(server);
+  config.type = PeerConnectionInterface::kRelay;
+  config.disable_ipv6 = true;
+  config.tcp_candidate_policy =
+      PeerConnectionInterface::kTcpCandidatePolicyDisabled;
+  config.ice_candidate_pool_size = 1;
+  CreatePeerConnection(config, nullptr);
+
+  const cricket::FakePortAllocatorSession* session =
+      static_cast<const cricket::FakePortAllocatorSession*>(
+          port_allocator_->GetPooledSession());
+  ASSERT_NE(nullptr, session);
+  EXPECT_EQ(1UL, session->stun_servers().size());
+  EXPECT_EQ(0U, session->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6);
+  EXPECT_LT(0U, session->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
+  EXPECT_EQ(cricket::CF_RELAY, session->candidate_filter());
+}
+
 TEST_F(PeerConnectionInterfaceTest, AddStreams) {
   CreatePeerConnection();
   AddVideoStream(kStreamLabel1);
@@ -980,7 +1094,7 @@ TEST_F(PeerConnectionInterfaceTest, AddStreams) {
 TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
   CreatePeerConnection();
   AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
-  scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
 
   const cricket::ContentInfo* audio_content =
@@ -1055,7 +1169,7 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
   EXPECT_EQ(video_track, video_sender->track());
 
   // Now create an offer and check for the senders.
-  scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
 
   const cricket::ContentInfo* audio_content =
@@ -1193,13 +1307,13 @@ TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
 
   EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
   // SetRemoteDescription takes ownership of offer.
-  rtc::scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   AddVideoStream(kStreamLabel1);
   EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
   EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
 
   // SetLocalDescription takes ownership of answer.
-  rtc::scoped_ptr<SessionDescriptionInterface> answer;
+  std::unique_ptr<SessionDescriptionInterface> answer;
   EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
   EXPECT_TRUE(DoSetLocalDescription(answer.release()));
 
@@ -1214,7 +1328,7 @@ TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
 TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
   CreatePeerConnection();
   // Create a regular offer for the CreateAnswer test later.
-  rtc::scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
   EXPECT_TRUE(offer);
   offer.reset();
@@ -1226,7 +1340,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
   EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
 
   // Test CreateAnswer
-  rtc::scoped_ptr<SessionDescriptionInterface> answer;
+  std::unique_ptr<SessionDescriptionInterface> answer;
   EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
 }
 
@@ -1238,7 +1352,7 @@ TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
   AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
 
   // Test CreateOffer
-  scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
   int audio_ssrc = 0;
   int video_ssrc = 0;
@@ -1250,7 +1364,7 @@ TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
 
   // Test CreateAnswer
   EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
-  scoped_ptr<SessionDescriptionInterface> answer;
+  std::unique_ptr<SessionDescriptionInterface> answer;
   ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
   audio_ssrc = 0;
   video_ssrc = 0;
@@ -1276,7 +1390,7 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
       pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
   stream->AddTrack(video_track.get());
 
-  scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
 
   const cricket::MediaContentDescription* video_desc =
@@ -1296,7 +1410,7 @@ TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
   // Remove the video track.
   stream->RemoveTrack(stream->GetVideoTracks()[0]);
 
-  scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
 
   const cricket::MediaContentDescription* video_desc =
@@ -1310,7 +1424,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
   CreatePeerConnection();
   pc_->CreateSender("video", kStreamLabel1);
 
-  scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
 
   const cricket::MediaContentDescription* video_desc =
@@ -1373,9 +1487,9 @@ TEST_F(PeerConnectionInterfaceTest, TestDataChannel) {
   scoped_refptr<DataChannelInterface> data2  =
       pc_->CreateDataChannel("test2", NULL);
   ASSERT_TRUE(data1 != NULL);
-  rtc::scoped_ptr<MockDataChannelObserver> observer1(
+  std::unique_ptr<MockDataChannelObserver> observer1(
       new MockDataChannelObserver(data1));
-  rtc::scoped_ptr<MockDataChannelObserver> observer2(
+  std::unique_ptr<MockDataChannelObserver> observer2(
       new MockDataChannelObserver(data2));
 
   EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
@@ -1420,9 +1534,9 @@ TEST_F(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
   scoped_refptr<DataChannelInterface> data2  =
       pc_->CreateDataChannel("test2", NULL);
   ASSERT_TRUE(data1 != NULL);
-  rtc::scoped_ptr<MockDataChannelObserver> observer1(
+  std::unique_ptr<MockDataChannelObserver> observer1(
       new MockDataChannelObserver(data1));
-  rtc::scoped_ptr<MockDataChannelObserver> observer2(
+  std::unique_ptr<MockDataChannelObserver> observer2(
       new MockDataChannelObserver(data2));
 
   EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
@@ -1447,7 +1561,7 @@ TEST_F(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
   CreatePeerConnection(&constraints);
   scoped_refptr<DataChannelInterface> data1  =
       pc_->CreateDataChannel("test1", NULL);
-  rtc::scoped_ptr<MockDataChannelObserver> observer1(
+  std::unique_ptr<MockDataChannelObserver> observer1(
       new MockDataChannelObserver(data1));
 
   CreateOfferReceiveAnswerWithoutSsrc();
@@ -1662,9 +1776,9 @@ TEST_F(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
   scoped_refptr<DataChannelInterface> data2  =
       pc_->CreateDataChannel("test2", NULL);
   ASSERT_TRUE(data1 != NULL);
-  rtc::scoped_ptr<MockDataChannelObserver> observer1(
+  std::unique_ptr<MockDataChannelObserver> observer1(
       new MockDataChannelObserver(data1));
-  rtc::scoped_ptr<MockDataChannelObserver> observer2(
+  std::unique_ptr<MockDataChannelObserver> observer2(
       new MockDataChannelObserver(data2));
 
   CreateOfferReceiveAnswer();
@@ -1769,7 +1883,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
 
   // At this point we should be receiving stream 1, but not sending anything.
   // A new offer should be recvonly.
-  rtc::scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   DoCreateOffer(&offer, nullptr);
 
   const cricket::ContentInfo* video_content =
@@ -1801,7 +1915,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
   // At this point we should be receiving stream 1, but not sending anything.
   // A new offer would be recvonly, but we'll set the "no receive" constraints
   // to make it inactive.
-  rtc::scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   FakeConstraints offer_constraints;
   offer_constraints.AddMandatory(
       webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, false);
@@ -1840,6 +1954,35 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
             port_allocator_->stun_servers().begin()->hostname());
 }
 
+TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
+  CreatePeerConnection();
+  PeerConnectionInterface::RTCConfiguration config;
+  config.type = PeerConnectionInterface::kRelay;
+  EXPECT_TRUE(pc_->SetConfiguration(config));
+  EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
+}
+
+// Test that when SetConfiguration changes both the pool size and other
+// attributes, the pooled session is created with the updated attributes.
+TEST_F(PeerConnectionInterfaceTest,
+       SetConfigurationCreatesPooledSessionCorrectly) {
+  CreatePeerConnection();
+  PeerConnectionInterface::RTCConfiguration config;
+  config.ice_candidate_pool_size = 1;
+  PeerConnectionInterface::IceServer server;
+  server.uri = kStunAddressOnly;
+  config.servers.push_back(server);
+  config.type = PeerConnectionInterface::kRelay;
+  CreatePeerConnection(config, nullptr);
+
+  const cricket::FakePortAllocatorSession* session =
+      static_cast<const cricket::FakePortAllocatorSession*>(
+          port_allocator_->GetPooledSession());
+  ASSERT_NE(nullptr, session);
+  EXPECT_EQ(1UL, session->stun_servers().size());
+  EXPECT_EQ(cricket::CF_RELAY, session->candidate_filter());
+}
+
 // Test that PeerConnection::Close changes the states to closed and all remote
 // tracks change state to ended.
 TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
@@ -1896,9 +2039,9 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
   EXPECT_TRUE(pc_->local_description() != NULL);
   EXPECT_TRUE(pc_->remote_description() != NULL);
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer;
+  std::unique_ptr<SessionDescriptionInterface> offer;
   EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
-  rtc::scoped_ptr<SessionDescriptionInterface> answer;
+  std::unique_ptr<SessionDescriptionInterface> answer;
   EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
 
   std::string sdp;
@@ -1936,7 +2079,7 @@ TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithStream1);
 
-  rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1));
+  rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
   EXPECT_TRUE(
       CompareStreamCollections(observer_.remote_streams(), reference.get()));
   MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
@@ -1946,7 +2089,7 @@ TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
   // MediaStream.
   CreateAndSetRemoteOffer(kSdpStringWithStream1And2);
 
-  rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2));
+  rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1));
   EXPECT_TRUE(
       CompareStreamCollections(observer_.remote_streams(), reference2.get()));
 }
@@ -1959,14 +2102,14 @@ TEST_F(PeerConnectionInterfaceTest,
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1 =
+  std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
       CreateSessionDescriptionAndReference(1, 1);
   EXPECT_TRUE(DoSetRemoteDescription(desc_ms1.release()));
   EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
                                        reference_collection_));
 
   // Add extra audio and video tracks to the same MediaStream.
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1_two_tracks =
+  std::unique_ptr<SessionDescriptionInterface> desc_ms1_two_tracks =
       CreateSessionDescriptionAndReference(2, 2);
   EXPECT_TRUE(DoSetRemoteDescription(desc_ms1_two_tracks.release()));
   EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
@@ -1979,7 +2122,7 @@ TEST_F(PeerConnectionInterfaceTest,
   EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
 
   // Remove the extra audio and video tracks.
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_ms2 =
+  std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
       CreateSessionDescriptionAndReference(1, 1);
   MockTrackObserver audio_track_observer(audio_track2);
   MockTrackObserver video_track_observer(video_track2);
@@ -2018,7 +2161,7 @@ TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
       remote_stream->GetAudioTracks()[0];
   EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> local_answer;
+  std::unique_ptr<SessionDescriptionInterface> local_answer;
   EXPECT_TRUE(DoCreateAnswer(&local_answer, nullptr));
   cricket::ContentInfo* video_info =
       local_answer->description()->GetContentByName("video");
@@ -2028,7 +2171,7 @@ TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
   EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
 
   // Now create an offer where we reject both video and audio.
-  rtc::scoped_ptr<SessionDescriptionInterface> local_offer;
+  std::unique_ptr<SessionDescriptionInterface> local_offer;
   EXPECT_TRUE(DoCreateOffer(&local_offer, nullptr));
   video_info = local_offer->description()->GetContentByName("video");
   ASSERT_TRUE(video_info != nullptr);
@@ -2057,7 +2200,7 @@ TEST_F(PeerConnectionInterfaceTest, RemoveTrackThenRejectMediaContent) {
   remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
   remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
 
-  rtc::scoped_ptr<SessionDescriptionInterface> local_answer(
+  std::unique_ptr<SessionDescriptionInterface> local_answer(
       webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
                                        kSdpStringWithStream1, nullptr));
   cricket::ContentInfo* video_info =
@@ -2212,7 +2355,7 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
                            true);
   CreatePeerConnection(&constraints);
   CreateAndSetRemoteOffer(kSdpStringWithStream1);
-  rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1));
+  rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
   EXPECT_TRUE(
       CompareStreamCollections(observer_.remote_streams(), reference.get()));
 
@@ -2229,16 +2372,15 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
-  // Create an offer just to ensure we have an identity before we manually
-  // call SetLocalDescription.
-  rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
-  ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_1 =
-      CreateSessionDescriptionAndReference(2, 2);
+  // Create an offer with 1 stream with 2 tracks of each type.
+  rtc::scoped_refptr<StreamCollection> stream_collection =
+      CreateStreamCollection(1, 2);
+  pc_->AddStream(stream_collection->at(0));
+  std::unique_ptr<SessionDescriptionInterface> offer;
+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+  EXPECT_TRUE(DoSetLocalDescription(offer.release()));
 
-  pc_->AddStream(reference_collection_->at(0));
-  EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
   auto senders = pc_->GetSenders();
   EXPECT_EQ(4u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2247,11 +2389,12 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
 
   // Remove an audio and video track.
-  pc_->RemoveStream(reference_collection_->at(0));
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_2 =
-      CreateSessionDescriptionAndReference(1, 1);
-  pc_->AddStream(reference_collection_->at(0));
-  EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
+  pc_->RemoveStream(stream_collection->at(0));
+  stream_collection = CreateStreamCollection(1, 1);
+  pc_->AddStream(stream_collection->at(0));
+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+  EXPECT_TRUE(DoSetLocalDescription(offer.release()));
+
   senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2268,19 +2411,20 @@ TEST_F(PeerConnectionInterfaceTest,
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
-  // Create an offer just to ensure we have an identity before we manually
-  // call SetLocalDescription.
-  rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
-  ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_1 =
-      CreateSessionDescriptionAndReference(2, 2);
+  rtc::scoped_refptr<StreamCollection> stream_collection =
+      CreateStreamCollection(1, 2);
+  // Add a stream to create the offer, but remove it afterwards.
+  pc_->AddStream(stream_collection->at(0));
+  std::unique_ptr<SessionDescriptionInterface> offer;
+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+  pc_->RemoveStream(stream_collection->at(0));
 
-  EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
+  EXPECT_TRUE(DoSetLocalDescription(offer.release()));
   auto senders = pc_->GetSenders();
   EXPECT_EQ(0u, senders.size());
 
-  pc_->AddStream(reference_collection_->at(0));
+  pc_->AddStream(stream_collection->at(0));
   senders = pc_->GetSenders();
   EXPECT_EQ(4u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2297,37 +2441,44 @@ TEST_F(PeerConnectionInterfaceTest,
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
-  // Create an offer just to ensure we have an identity before we manually
-  // call SetLocalDescription.
-  rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
-  ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
-  rtc::scoped_ptr<SessionDescriptionInterface> desc =
-      CreateSessionDescriptionAndReference(1, 1);
-  std::string sdp;
-  desc->ToString(&sdp);
+  rtc::scoped_refptr<StreamCollection> stream_collection =
+      CreateStreamCollection(2, 1);
+  pc_->AddStream(stream_collection->at(0));
+  std::unique_ptr<SessionDescriptionInterface> offer;
+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+  // Grab a copy of the offer before it gets passed into the PC.
+  std::unique_ptr<JsepSessionDescription> modified_offer(
+      new JsepSessionDescription(JsepSessionDescription::kOffer));
+  modified_offer->Initialize(offer->description()->Copy(), offer->session_id(),
+                             offer->session_version());
+  EXPECT_TRUE(DoSetLocalDescription(offer.release()));
 
-  pc_->AddStream(reference_collection_->at(0));
-  EXPECT_TRUE(DoSetLocalDescription(desc.release()));
   auto senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
 
   // Change the ssrc of the audio and video track.
-  std::string ssrc_org = "a=ssrc:1";
-  std::string ssrc_to = "a=ssrc:97";
-  rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
-                       ssrc_to.length(), &sdp);
-  ssrc_org = "a=ssrc:2";
-  ssrc_to = "a=ssrc:98";
-  rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
-                       ssrc_to.length(), &sdp);
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
-      webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
-                                       nullptr));
-
-  EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
+  cricket::MediaContentDescription* desc =
+      cricket::GetFirstAudioContentDescription(modified_offer->description());
+  ASSERT_TRUE(desc != NULL);
+  for (StreamParams& stream : desc->mutable_streams()) {
+    for (unsigned int& ssrc : stream.ssrcs) {
+      ++ssrc;
+    }
+  }
+
+  desc =
+      cricket::GetFirstVideoContentDescription(modified_offer->description());
+  ASSERT_TRUE(desc != NULL);
+  for (StreamParams& stream : desc->mutable_streams()) {
+    for (unsigned int& ssrc : stream.ssrcs) {
+      ++ssrc;
+    }
+  }
+
+  EXPECT_TRUE(DoSetLocalDescription(modified_offer.release()));
   senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
   EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
@@ -2344,43 +2495,36 @@ TEST_F(PeerConnectionInterfaceTest,
   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                            true);
   CreatePeerConnection(&constraints);
-  // Create an offer just to ensure we have an identity before we manually
-  // call SetLocalDescription.
-  rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
-  ASSERT_TRUE(DoCreateOffer(&throwaway, nullptr));
 
-  rtc::scoped_ptr<SessionDescriptionInterface> desc =
-      CreateSessionDescriptionAndReference(1, 1);
-  std::string sdp;
-  desc->ToString(&sdp);
+  rtc::scoped_refptr<StreamCollection> stream_collection =
+      CreateStreamCollection(2, 1);
+  pc_->AddStream(stream_collection->at(0));
+  std::unique_ptr<SessionDescriptionInterface> offer;
+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+  EXPECT_TRUE(DoSetLocalDescription(offer.release()));
 
-  pc_->AddStream(reference_collection_->at(0));
-  EXPECT_TRUE(DoSetLocalDescription(desc.release()));
   auto senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
-  EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
-  EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
+  EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0], kStreams[0]));
+  EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
 
   // Add a new MediaStream but with the same tracks as in the first stream.
   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
       webrtc::MediaStream::Create(kStreams[1]));
-  stream_1->AddTrack(reference_collection_->at(0)->GetVideoTracks()[0]);
-  stream_1->AddTrack(reference_collection_->at(0)->GetAudioTracks()[0]);
+  stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
+  stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
   pc_->AddStream(stream_1);
 
-  // Replace msid in the original SDP.
-  rtc::replace_substrs(kStreams[0], strlen(kStreams[0]), kStreams[1],
-                       strlen(kStreams[1]), &sdp);
-
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
-      webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
-                                       nullptr));
+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
+  EXPECT_TRUE(DoSetLocalDescription(offer.release()));
 
-  EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
-  senders = pc_->GetSenders();
-  EXPECT_EQ(2u, senders.size());
-  EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
-  EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
+  auto new_senders = pc_->GetSenders();
+  // Should be the same senders as before, but with updated stream id.
+  // Note that this behavior is subject to change in the future.
+  // We may decide the PC should ignore existing tracks in AddStream.
+  EXPECT_EQ(senders, new_senders);
+  EXPECT_TRUE(ContainsSender(new_senders, kAudioTracks[0], kStreams[1]));
+  EXPECT_TRUE(ContainsSender(new_senders, kVideoTracks[0], kStreams[1]));
 }
 
 // The PeerConnectionMediaConfig tests below verify that configuration
diff --git a/chromium/third_party/webrtc/api/peerconnectionproxy.h b/chromium/third_party/webrtc/api/peerconnectionproxy.h
index 1183e619103..d35d5bacded 100644
--- a/chromium/third_party/webrtc/api/peerconnectionproxy.h
+++ b/chromium/third_party/webrtc/api/peerconnectionproxy.h
@@ -17,7 +17,7 @@
 namespace webrtc {
 
 // Define proxy for PeerConnectionInterface.
-BEGIN_PROXY_MAP(PeerConnection)
+BEGIN_SIGNALING_PROXY_MAP(PeerConnection)
   PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
                 local_streams)
   PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
@@ -75,7 +75,7 @@ BEGIN_PROXY_MAP(PeerConnection)
   PROXY_METHOD0(IceConnectionState, ice_connection_state)
   PROXY_METHOD0(IceGatheringState, ice_gathering_state)
   PROXY_METHOD0(void, Close)
-END_PROXY()
+END_SIGNALING_PROXY()
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/proxy.h b/chromium/third_party/webrtc/api/proxy.h
index 1351a0427e6..2df85c4bf46 100644
--- a/chromium/third_party/webrtc/api/proxy.h
+++ b/chromium/third_party/webrtc/api/proxy.h
@@ -18,7 +18,7 @@
 //  public:
 //   std::string FooA() = 0;
 //   std::string FooB(bool arg1) const = 0;
-//   std::string FooC(bool arg1)= 0;
+//   std::string FooC(bool arg1) = 0;
 //  };
 //
 // Note that return types can not be a const reference.
@@ -30,14 +30,25 @@
 // BEGIN_PROXY_MAP(Test)
 //   PROXY_METHOD0(std::string, FooA)
 //   PROXY_CONSTMETHOD1(std::string, FooB, arg1)
-//   PROXY_METHOD1(std::string, FooC, arg1)
+//   PROXY_WORKER_METHOD1(std::string, FooC, arg1)
 // END_PROXY()
 //
-// The proxy can be created using TestProxy::Create(Thread*, TestInterface*).
+// where the first two methods are invoked on the signaling thread,
+// and the third is invoked on the worker thread.
+//
+// The proxy can be created using
+//
+//   TestProxy::Create(Thread* signaling_thread, Thread* worker_thread,
+//                     TestInterface*).
+//
+// The variant defined with BEGIN_SIGNALING_PROXY_MAP is unaware of
+// the worker thread, and invokes all methods on the signaling thread.
 
 #ifndef WEBRTC_API_PROXY_H_
 #define WEBRTC_API_PROXY_H_
 
+#include <memory>
+
 #include "webrtc/base/event.h"
 #include "webrtc/base/thread.h"
 
@@ -108,7 +119,7 @@ class SynchronousMethodCall
 
  private:
   void OnMessage(rtc::Message*) { proxy_->OnMessage(NULL); e_->Set(); }
-  rtc::scoped_ptr<rtc::Event> e_;
+  std::unique_ptr<rtc::Event> e_;
   rtc::MessageHandler* proxy_;
 };
 
@@ -295,79 +306,125 @@ class MethodCall5 : public rtc::Message,
   T5 a5_;
 };
 
-#define BEGIN_PROXY_MAP(c)                                                \
+#define BEGIN_SIGNALING_PROXY_MAP(c)                                     \
   class c##Proxy : public c##Interface {                                  \
    protected:                                                             \
     typedef c##Interface C;                                               \
-    c##Proxy(rtc::Thread* thread, C* c) : owner_thread_(thread), c_(c) {} \
+    c##Proxy(rtc::Thread* signaling_thread, C* c)                         \
+      : signaling_thread_(signaling_thread), c_(c) {}                     \
     ~c##Proxy() {                                                         \
-      MethodCall0<c##Proxy, void> call(this, &c##Proxy::Release_s);       \
-      call.Marshal(owner_thread_);                                        \
+      MethodCall0<c##Proxy, void> call(                                   \
+          this, &c##Proxy::Release_s);                                    \
+      call.Marshal(signaling_thread_);                                    \
     }                                                                     \
                                                                           \
    public:                                                                \
-    static rtc::scoped_refptr<C> Create(rtc::Thread* thread, C* c) {      \
-      return new rtc::RefCountedObject<c##Proxy>(thread, c);              \
+    static rtc::scoped_refptr<C> Create(rtc::Thread* signaling_thread, C* c) { \
+      return new rtc::RefCountedObject<c##Proxy>(                         \
+          signaling_thread, c);                                           \
+    }
+
+#define BEGIN_PROXY_MAP(c)                                              \
+  class c##Proxy : public c##Interface {                                \
+   protected:                                                           \
+    typedef c##Interface C;                                             \
+    c##Proxy(rtc::Thread* signaling_thread, rtc::Thread* worker_thread, C* c) \
+      : signaling_thread_(signaling_thread),                            \
+        worker_thread_(worker_thread),                                  \
+        c_(c) {}                                                        \
+    ~c##Proxy() {                                                       \
+      MethodCall0<c##Proxy, void> call(this, &c##Proxy::Release_s);     \
+      call.Marshal(signaling_thread_);                                  \
+    }                                                                   \
+                                                                        \
+   public:                                                              \
+    static rtc::scoped_refptr<C> Create(                                \
+        rtc::Thread* signaling_thread, rtc::Thread* worker_thread, C* c) { \
+      return new rtc::RefCountedObject<c##Proxy>(                       \
+          signaling_thread, worker_thread, c);                          \
     }
 
 #define PROXY_METHOD0(r, method)                  \
   r method() override {                           \
     MethodCall0<C, r> call(c_.get(), &C::method); \
-    return call.Marshal(owner_thread_);           \
+    return call.Marshal(signaling_thread_);       \
   }
 
 #define PROXY_CONSTMETHOD0(r, method)                  \
   r method() const override {                          \
     ConstMethodCall0<C, r> call(c_.get(), &C::method); \
-    return call.Marshal(owner_thread_);                \
+    return call.Marshal(signaling_thread_);            \
   }
 
 #define PROXY_METHOD1(r, method, t1)                      \
   r method(t1 a1) override {                              \
     MethodCall1<C, r, t1> call(c_.get(), &C::method, a1); \
-    return call.Marshal(owner_thread_);                   \
+    return call.Marshal(signaling_thread_);               \
   }
 
 #define PROXY_CONSTMETHOD1(r, method, t1)                      \
   r method(t1 a1) const override {                             \
     ConstMethodCall1<C, r, t1> call(c_.get(), &C::method, a1); \
-    return call.Marshal(owner_thread_);                        \
+    return call.Marshal(signaling_thread_);                    \
   }
 
 #define PROXY_METHOD2(r, method, t1, t2)                          \
   r method(t1 a1, t2 a2) override {                               \
     MethodCall2<C, r, t1, t2> call(c_.get(), &C::method, a1, a2); \
-    return call.Marshal(owner_thread_);                           \
+    return call.Marshal(signaling_thread_);                       \
   }
 
 #define PROXY_METHOD3(r, method, t1, t2, t3)                              \
   r method(t1 a1, t2 a2, t3 a3) override {                                \
     MethodCall3<C, r, t1, t2, t3> call(c_.get(), &C::method, a1, a2, a3); \
-    return call.Marshal(owner_thread_);                                   \
+    return call.Marshal(signaling_thread_);                               \
   }
 
 #define PROXY_METHOD4(r, method, t1, t2, t3, t4)                             \
   r method(t1 a1, t2 a2, t3 a3, t4 a4) override {                            \
     MethodCall4<C, r, t1, t2, t3, t4> call(c_.get(), &C::method, a1, a2, a3, \
                                            a4);                              \
-    return call.Marshal(owner_thread_);                                      \
+    return call.Marshal(signaling_thread_);                                  \
   }
 
 #define PROXY_METHOD5(r, method, t1, t2, t3, t4, t5)                         \
   r method(t1 a1, t2 a2, t3 a3, t4 a4, t5 a5) override {                     \
     MethodCall5<C, r, t1, t2, t3, t4, t5> call(c_.get(), &C::method, a1, a2, \
                                                a3, a4, a5);                  \
-    return call.Marshal(owner_thread_);                                      \
+    return call.Marshal(signaling_thread_);                                  \
+  }
+
+// Define methods which should be invoked on the worker thread.
+#define PROXY_WORKER_METHOD1(r, method, t1)               \
+  r method(t1 a1) override {                              \
+    MethodCall1<C, r, t1> call(c_.get(), &C::method, a1); \
+    return call.Marshal(worker_thread_);                  \
+  }
+
+#define PROXY_WORKER_METHOD2(r, method, t1, t2)                   \
+  r method(t1 a1, t2 a2) override {                               \
+    MethodCall2<C, r, t1, t2> call(c_.get(), &C::method, a1, a2); \
+    return call.Marshal(worker_thread_);                          \
   }
 
-#define END_PROXY() \
+#define END_SIGNALING_PROXY() \
    private:\
     void Release_s() {\
       c_ = NULL;\
     }\
-    mutable rtc::Thread* owner_thread_;\
+    mutable rtc::Thread* signaling_thread_;\
     rtc::scoped_refptr<C> c_;\
-  };\
+  };
+
+#define END_PROXY()                                  \
+   private:                                          \
+    void Release_s() {                               \
+      c_ = NULL;                                     \
+    }                                                \
+    mutable rtc::Thread* signaling_thread_;          \
+    mutable rtc::Thread* worker_thread_;             \
+    rtc::scoped_refptr<C> c_;                        \
+  };                                                 \
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/proxy_unittest.cc b/chromium/third_party/webrtc/api/proxy_unittest.cc
index 85e54380768..931ba28eba8 100644
--- a/chromium/third_party/webrtc/api/proxy_unittest.cc
+++ b/chromium/third_party/webrtc/api/proxy_unittest.cc
@@ -10,12 +10,12 @@
 
 #include "webrtc/api/proxy.h"
 
+#include <memory>
 #include <string>
 
-#include "testing/base/public/gmock.h"
+#include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 
 using ::testing::_;
@@ -40,16 +40,6 @@ class FakeInterface : public rtc::RefCountInterface {
   ~FakeInterface() {}
 };
 
-// Proxy for the test interface.
-BEGIN_PROXY_MAP(Fake)
-  PROXY_METHOD0(void, VoidMethod0)
-  PROXY_METHOD0(std::string, Method0)
-  PROXY_CONSTMETHOD0(std::string, ConstMethod0)
-  PROXY_METHOD1(std::string, Method1, std::string)
-  PROXY_CONSTMETHOD1(std::string, ConstMethod1, std::string)
-  PROXY_METHOD2(std::string, Method2, std::string, std::string)
-END_PROXY()
-
 // Implementation of the test interface.
 class Fake : public FakeInterface {
  public:
@@ -71,60 +61,156 @@ class Fake : public FakeInterface {
   ~Fake() {}
 };
 
-class ProxyTest: public testing::Test {
+// Proxies for the test interface.
+BEGIN_PROXY_MAP(Fake)
+  PROXY_METHOD0(void, VoidMethod0)
+  PROXY_METHOD0(std::string, Method0)
+  PROXY_CONSTMETHOD0(std::string, ConstMethod0)
+  PROXY_WORKER_METHOD1(std::string, Method1, std::string)
+  PROXY_CONSTMETHOD1(std::string, ConstMethod1, std::string)
+  PROXY_WORKER_METHOD2(std::string, Method2, std::string, std::string)
+END_PROXY()
+
+// Preprocessor hack to get a proxy class a name different than FakeProxy.
+#define FakeProxy FakeSignalingProxy
+BEGIN_SIGNALING_PROXY_MAP(Fake)
+  PROXY_METHOD0(void, VoidMethod0)
+  PROXY_METHOD0(std::string, Method0)
+  PROXY_CONSTMETHOD0(std::string, ConstMethod0)
+  PROXY_METHOD1(std::string, Method1, std::string)
+  PROXY_CONSTMETHOD1(std::string, ConstMethod1, std::string)
+  PROXY_METHOD2(std::string, Method2, std::string, std::string)
+END_SIGNALING_PROXY()
+#undef FakeProxy
+
+class SignalingProxyTest : public testing::Test {
  public:
-  // Checks that the functions is called on the |signaling_thread_|.
-  void CheckThread() {
-    EXPECT_EQ(rtc::Thread::Current(), signaling_thread_.get());
-  }
+  // Checks that the functions are called on the right thread.
+  void CheckSignalingThread() { EXPECT_TRUE(signaling_thread_->IsCurrent()); }
 
  protected:
-  virtual void SetUp() {
+  void SetUp() override {
     signaling_thread_.reset(new rtc::Thread());
     ASSERT_TRUE(signaling_thread_->Start());
     fake_ = Fake::Create();
-    fake_proxy_ = FakeProxy::Create(signaling_thread_.get(), fake_.get());
+    fake_signaling_proxy_ =
+        FakeSignalingProxy::Create(signaling_thread_.get(), fake_.get());
   }
 
  protected:
-  rtc::scoped_ptr<rtc::Thread> signaling_thread_;
-  rtc::scoped_refptr<FakeInterface> fake_proxy_;
+  std::unique_ptr<rtc::Thread> signaling_thread_;
+  rtc::scoped_refptr<FakeInterface> fake_signaling_proxy_;
   rtc::scoped_refptr<Fake> fake_;
 };
 
+TEST_F(SignalingProxyTest, VoidMethod0) {
+  EXPECT_CALL(*fake_, VoidMethod0())
+      .Times(Exactly(1))
+      .WillOnce(
+          InvokeWithoutArgs(this, &SignalingProxyTest::CheckSignalingThread));
+  fake_signaling_proxy_->VoidMethod0();
+}
+
+TEST_F(SignalingProxyTest, Method0) {
+  EXPECT_CALL(*fake_, Method0())
+      .Times(Exactly(1))
+      .WillOnce(DoAll(
+          InvokeWithoutArgs(this, &SignalingProxyTest::CheckSignalingThread),
+          Return("Method0")));
+  EXPECT_EQ("Method0", fake_signaling_proxy_->Method0());
+}
+
+TEST_F(SignalingProxyTest, ConstMethod0) {
+  EXPECT_CALL(*fake_, ConstMethod0())
+      .Times(Exactly(1))
+      .WillOnce(DoAll(
+          InvokeWithoutArgs(this, &SignalingProxyTest::CheckSignalingThread),
+          Return("ConstMethod0")));
+  EXPECT_EQ("ConstMethod0", fake_signaling_proxy_->ConstMethod0());
+}
+
+TEST_F(SignalingProxyTest, Method1) {
+  const std::string arg1 = "arg1";
+  EXPECT_CALL(*fake_, Method1(arg1))
+      .Times(Exactly(1))
+      .WillOnce(DoAll(
+          InvokeWithoutArgs(this, &SignalingProxyTest::CheckSignalingThread),
+          Return("Method1")));
+  EXPECT_EQ("Method1", fake_signaling_proxy_->Method1(arg1));
+}
+
+TEST_F(SignalingProxyTest, ConstMethod1) {
+  const std::string arg1 = "arg1";
+  EXPECT_CALL(*fake_, ConstMethod1(arg1))
+      .Times(Exactly(1))
+      .WillOnce(DoAll(
+          InvokeWithoutArgs(this, &SignalingProxyTest::CheckSignalingThread),
+          Return("ConstMethod1")));
+  EXPECT_EQ("ConstMethod1", fake_signaling_proxy_->ConstMethod1(arg1));
+}
+
+TEST_F(SignalingProxyTest, Method2) {
+  const std::string arg1 = "arg1";
+  const std::string arg2 = "arg2";
+  EXPECT_CALL(*fake_, Method2(arg1, arg2))
+      .Times(Exactly(1))
+      .WillOnce(DoAll(
+          InvokeWithoutArgs(this, &SignalingProxyTest::CheckSignalingThread),
+          Return("Method2")));
+  EXPECT_EQ("Method2", fake_signaling_proxy_->Method2(arg1, arg2));
+}
+
+class ProxyTest : public SignalingProxyTest {
+ public:
+  // Checks that the functions are called on the right thread.
+  void CheckWorkerThread() { EXPECT_TRUE(worker_thread_->IsCurrent()); }
+
+ protected:
+  void SetUp() override {
+    SignalingProxyTest::SetUp();
+    worker_thread_.reset(new rtc::Thread());
+    ASSERT_TRUE(worker_thread_->Start());
+    fake_proxy_ = FakeProxy::Create(signaling_thread_.get(),
+                                    worker_thread_.get(), fake_.get());
+  }
+
+ protected:
+  std::unique_ptr<rtc::Thread> worker_thread_;
+  rtc::scoped_refptr<FakeInterface> fake_proxy_;
+};
+
 TEST_F(ProxyTest, VoidMethod0) {
   EXPECT_CALL(*fake_, VoidMethod0())
-            .Times(Exactly(1))
-            .WillOnce(InvokeWithoutArgs(this, &ProxyTest::CheckThread));
+      .Times(Exactly(1))
+      .WillOnce(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread));
   fake_proxy_->VoidMethod0();
 }
 
 TEST_F(ProxyTest, Method0) {
   EXPECT_CALL(*fake_, Method0())
-            .Times(Exactly(1))
-            .WillOnce(
-                DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckThread),
-                      Return("Method0")));
+      .Times(Exactly(1))
+      .WillOnce(
+          DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
+                Return("Method0")));
   EXPECT_EQ("Method0",
             fake_proxy_->Method0());
 }
 
 TEST_F(ProxyTest, ConstMethod0) {
   EXPECT_CALL(*fake_, ConstMethod0())
-            .Times(Exactly(1))
-            .WillOnce(
-                DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckThread),
-                      Return("ConstMethod0")));
+      .Times(Exactly(1))
+      .WillOnce(
+          DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
+                Return("ConstMethod0")));
   EXPECT_EQ("ConstMethod0",
             fake_proxy_->ConstMethod0());
 }
 
-TEST_F(ProxyTest, Method1) {
+TEST_F(ProxyTest, WorkerMethod1) {
   const std::string arg1 = "arg1";
   EXPECT_CALL(*fake_, Method1(arg1))
-            .Times(Exactly(1))
-            .WillOnce(
-                DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckThread),
+      .Times(Exactly(1))
+      .WillOnce(DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckWorkerThread),
                       Return("Method1")));
   EXPECT_EQ("Method1", fake_proxy_->Method1(arg1));
 }
@@ -132,20 +218,19 @@ TEST_F(ProxyTest, Method1) {
 TEST_F(ProxyTest, ConstMethod1) {
   const std::string arg1 = "arg1";
   EXPECT_CALL(*fake_, ConstMethod1(arg1))
-            .Times(Exactly(1))
-            .WillOnce(
-                DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckThread),
-                      Return("ConstMethod1")));
+      .Times(Exactly(1))
+      .WillOnce(
+          DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
+                Return("ConstMethod1")));
   EXPECT_EQ("ConstMethod1", fake_proxy_->ConstMethod1(arg1));
 }
 
-TEST_F(ProxyTest, Method2) {
+TEST_F(ProxyTest, WorkerMethod2) {
   const std::string arg1 = "arg1";
   const std::string arg2 = "arg2";
   EXPECT_CALL(*fake_, Method2(arg1, arg2))
-            .Times(Exactly(1))
-            .WillOnce(
-                DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckThread),
+      .Times(Exactly(1))
+      .WillOnce(DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckWorkerThread),
                       Return("Method2")));
   EXPECT_EQ("Method2", fake_proxy_->Method2(arg1, arg2));
 }
diff --git a/chromium/third_party/webrtc/api/quicdatachannel.cc b/chromium/third_party/webrtc/api/quicdatachannel.cc
new file mode 100644
index 00000000000..5420da15eab
--- /dev/null
+++ b/chromium/third_party/webrtc/api/quicdatachannel.cc
@@ -0,0 +1,391 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/quicdatachannel.h"
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/copyonwritebuffer.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/p2p/quic/quictransportchannel.h"
+#include "webrtc/p2p/quic/reliablequicstream.h"
+
+namespace webrtc {
+
+void WriteQuicDataChannelMessageHeader(int data_channel_id,
+                                       uint64_t message_id,
+                                       rtc::CopyOnWriteBuffer* header) {
+  RTC_DCHECK(header);
+  // 64-bit varints require at most 10 bytes (7*10 == 70), and 32-bit varints
+  // require at most 5 bytes (7*5 == 35).
+  size_t max_length = 15;
+  rtc::ByteBufferWriter byte_buffer(nullptr, max_length,
+                                    rtc::ByteBuffer::ByteOrder::ORDER_HOST);
+  byte_buffer.WriteUVarint(data_channel_id);
+  byte_buffer.WriteUVarint(message_id);
+  header->SetData(byte_buffer.Data(), byte_buffer.Length());
+}
+
+bool ParseQuicDataMessageHeader(const char* data,
+                                size_t len,
+                                int* data_channel_id,
+                                uint64_t* message_id,
+                                size_t* bytes_read) {
+  RTC_DCHECK(data_channel_id);
+  RTC_DCHECK(message_id);
+  RTC_DCHECK(bytes_read);
+
+  rtc::ByteBufferReader byte_buffer(data, len, rtc::ByteBuffer::ORDER_HOST);
+  uint64_t dcid;
+  if (!byte_buffer.ReadUVarint(&dcid)) {
+    LOG(LS_ERROR) << "Could not read the data channel ID";
+    return false;
+  }
+  *data_channel_id = dcid;
+  if (!byte_buffer.ReadUVarint(message_id)) {
+    LOG(LS_ERROR) << "Could not read message ID for data channel "
+                  << *data_channel_id;
+    return false;
+  }
+  size_t remaining_bytes = byte_buffer.Length();
+  *bytes_read = len - remaining_bytes;
+  return true;
+}
+
+QuicDataChannel::QuicDataChannel(rtc::Thread* signaling_thread,
+                                 rtc::Thread* worker_thread,
+                                 const std::string& label,
+                                 const DataChannelInit& config)
+    : signaling_thread_(signaling_thread),
+      worker_thread_(worker_thread),
+      id_(config.id),
+      state_(kConnecting),
+      buffered_amount_(0),
+      next_message_id_(0),
+      label_(label),
+      protocol_(config.protocol) {}
+
+QuicDataChannel::~QuicDataChannel() {}
+
+void QuicDataChannel::RegisterObserver(DataChannelObserver* observer) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  observer_ = observer;
+}
+
+void QuicDataChannel::UnregisterObserver() {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  observer_ = nullptr;
+}
+
+bool QuicDataChannel::Send(const DataBuffer& buffer) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  if (state_ != kOpen) {
+    LOG(LS_ERROR) << "QUIC data channel " << id_
+                  << " is not open so cannot send.";
+    return false;
+  }
+  return worker_thread_->Invoke<bool>(
+      rtc::Bind(&QuicDataChannel::Send_w, this, buffer));
+}
+
+bool QuicDataChannel::Send_w(const DataBuffer& buffer) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  // Encode and send the header containing the data channel ID and message ID.
+  rtc::CopyOnWriteBuffer header;
+  WriteQuicDataChannelMessageHeader(id_, ++next_message_id_, &header);
+  RTC_DCHECK(quic_transport_channel_);
+  cricket::ReliableQuicStream* stream =
+      quic_transport_channel_->CreateQuicStream();
+  RTC_DCHECK(stream);
+
+  // Send the header with a FIN if the message is empty.
+  bool header_fin = (buffer.size() == 0);
+  rtc::StreamResult header_result =
+      stream->Write(header.data<char>(), header.size(), header_fin);
+
+  if (header_result == rtc::SR_BLOCK) {
+    // The header is write blocked but we should try sending the message. Since
+    // the ReliableQuicStream queues data in order, if the header is write
+    // blocked then the message will be write blocked. Otherwise if the message
+    // is sent then the header is sent.
+    LOG(LS_INFO) << "Stream " << stream->id()
+                 << " header is write blocked for QUIC data channel " << id_;
+  } else if (header_result != rtc::SR_SUCCESS) {
+    LOG(LS_ERROR) << "Stream " << stream->id()
+                  << " failed to write header for QUIC data channel " << id_
+                  << ". Unexpected error " << header_result;
+    return false;
+  }
+
+  // If the message is not empty, then send the message with a FIN.
+  bool message_fin = true;
+  rtc::StreamResult message_result =
+      header_fin ? header_result : stream->Write(buffer.data.data<char>(),
+                                                 buffer.size(), message_fin);
+
+  if (message_result == rtc::SR_SUCCESS) {
+    // The message is sent and we don't need this QUIC stream.
+    LOG(LS_INFO) << "Stream " << stream->id()
+                 << " successfully wrote message for QUIC data channel " << id_;
+    stream->Close();
+    return true;
+  }
+  // TODO(mikescarlett): Register the ReliableQuicStream's priority to the
+  // QuicWriteBlockedList so that the QUIC session doesn't drop messages when
+  // the QUIC transport channel becomes unwritable.
+  if (message_result == rtc::SR_BLOCK) {
+    // The QUIC stream is write blocked, so the message is queued by the QUIC
+    // session. If this is due to the QUIC not being writable, it will be sent
+    // once QUIC becomes writable again. Otherwise it may be due to exceeding
+    // the QUIC flow control limit, in which case the remote peer's QUIC session
+    // will tell the QUIC stream to send more data.
+    LOG(LS_INFO) << "Stream " << stream->id()
+                 << " message is write blocked for QUIC data channel " << id_;
+    SetBufferedAmount_w(buffered_amount_ + stream->queued_data_bytes());
+    stream->SignalQueuedBytesWritten.connect(
+        this, &QuicDataChannel::OnQueuedBytesWritten);
+    write_blocked_quic_streams_[stream->id()] = stream;
+    // The QUIC stream will be removed from |write_blocked_quic_streams_| once
+    // it closes.
+    stream->SignalClosed.connect(this,
+                                 &QuicDataChannel::OnWriteBlockedStreamClosed);
+    return true;
+  }
+  LOG(LS_ERROR) << "Stream " << stream->id()
+                << " failed to write message for QUIC data channel " << id_
+                << ". Unexpected error: " << message_result;
+  return false;
+}
+
+void QuicDataChannel::OnQueuedBytesWritten(net::QuicStreamId stream_id,
+                                           uint64_t queued_bytes_written) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  SetBufferedAmount_w(buffered_amount_ - queued_bytes_written);
+  const auto& kv = write_blocked_quic_streams_.find(stream_id);
+  if (kv == write_blocked_quic_streams_.end()) {
+    RTC_DCHECK(false);
+    return;
+  }
+  cricket::ReliableQuicStream* stream = kv->second;
+  // True if the QUIC stream is done sending data.
+  if (stream->fin_sent()) {
+    LOG(LS_INFO) << "Stream " << stream->id()
+                 << " successfully wrote data for QUIC data channel " << id_;
+    stream->Close();
+  }
+}
+
+void QuicDataChannel::SetBufferedAmount_w(uint64_t buffered_amount) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  buffered_amount_ = buffered_amount;
+  invoker_.AsyncInvoke<void>(
+      signaling_thread_, rtc::Bind(&QuicDataChannel::OnBufferedAmountChange_s,
+                                   this, buffered_amount));
+}
+
+void QuicDataChannel::Close() {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  if (state_ == kClosed || state_ == kClosing) {
+    return;
+  }
+  LOG(LS_INFO) << "Closing QUIC data channel.";
+  SetState_s(kClosing);
+  worker_thread_->Invoke<void>(rtc::Bind(&QuicDataChannel::Close_w, this));
+  SetState_s(kClosed);
+}
+
+void QuicDataChannel::Close_w() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  for (auto& kv : incoming_quic_messages_) {
+    Message& message = kv.second;
+    cricket::ReliableQuicStream* stream = message.stream;
+    stream->Close();
+  }
+
+  for (auto& kv : write_blocked_quic_streams_) {
+    cricket::ReliableQuicStream* stream = kv.second;
+    stream->Close();
+  }
+}
+
+bool QuicDataChannel::SetTransportChannel(
+    cricket::QuicTransportChannel* channel) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+
+  if (!channel) {
+    LOG(LS_ERROR) << "|channel| is NULL. Cannot set transport channel.";
+    return false;
+  }
+  if (quic_transport_channel_) {
+    if (channel == quic_transport_channel_) {
+      LOG(LS_WARNING) << "Ignoring duplicate transport channel.";
+      return true;
+    }
+    LOG(LS_ERROR) << "|channel| does not match existing transport channel.";
+    return false;
+  }
+
+  quic_transport_channel_ = channel;
+  LOG(LS_INFO) << "Setting QuicTransportChannel for QUIC data channel " << id_;
+  DataState data_channel_state = worker_thread_->Invoke<DataState>(
+      rtc::Bind(&QuicDataChannel::SetTransportChannel_w, this));
+  SetState_s(data_channel_state);
+  return true;
+}
+
+DataChannelInterface::DataState QuicDataChannel::SetTransportChannel_w() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  quic_transport_channel_->SignalReadyToSend.connect(
+      this, &QuicDataChannel::OnReadyToSend);
+  quic_transport_channel_->SignalClosed.connect(
+      this, &QuicDataChannel::OnConnectionClosed);
+  if (quic_transport_channel_->writable()) {
+    return kOpen;
+  }
+  return kConnecting;
+}
+
+void QuicDataChannel::OnIncomingMessage(Message&& message) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(message.stream);
+  if (!observer_) {
+    LOG(LS_WARNING) << "QUIC data channel " << id_
+                    << " received a message but has no observer.";
+    message.stream->Close();
+    return;
+  }
+  // A FIN is received if the message fits into a single QUIC stream frame and
+  // the remote peer is done sending.
+  if (message.stream->fin_received()) {
+    LOG(LS_INFO) << "Stream " << message.stream->id()
+                 << " has finished receiving data for QUIC data channel "
+                 << id_;
+    DataBuffer final_message(message.buffer, false);
+    invoker_.AsyncInvoke<void>(signaling_thread_,
+                               rtc::Bind(&QuicDataChannel::OnMessage_s, this,
+                                         std::move(final_message)));
+    message.stream->Close();
+    return;
+  }
+  // Otherwise the message is divided across multiple QUIC stream frames, so
+  // queue the data. OnDataReceived() will be called each time the remaining
+  // QUIC stream frames arrive.
+  LOG(LS_INFO) << "QUIC data channel " << id_
+               << " is queuing incoming data for stream "
+               << message.stream->id();
+  incoming_quic_messages_[message.stream->id()] = std::move(message);
+  message.stream->SignalDataReceived.connect(this,
+                                             &QuicDataChannel::OnDataReceived);
+  // The QUIC stream will be removed from |incoming_quic_messages_| once it
+  // closes.
+  message.stream->SignalClosed.connect(
+      this, &QuicDataChannel::OnIncomingQueuedStreamClosed);
+}
+
+void QuicDataChannel::OnDataReceived(net::QuicStreamId stream_id,
+                                     const char* data,
+                                     size_t len) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(data);
+  const auto& kv = incoming_quic_messages_.find(stream_id);
+  if (kv == incoming_quic_messages_.end()) {
+    RTC_DCHECK(false);
+    return;
+  }
+  Message& message = kv->second;
+  cricket::ReliableQuicStream* stream = message.stream;
+  rtc::CopyOnWriteBuffer& received_data = message.buffer;
+  // If the QUIC stream has not received a FIN, then the remote peer is not
+  // finished sending data.
+  if (!stream->fin_received()) {
+    received_data.AppendData(data, len);
+    return;
+  }
+  // Otherwise we are done receiving and can provide the data channel observer
+  // with the message.
+  LOG(LS_INFO) << "Stream " << stream_id
+               << " has finished receiving data for QUIC data channel " << id_;
+  received_data.AppendData(data, len);
+  DataBuffer final_message(std::move(received_data), false);
+  invoker_.AsyncInvoke<void>(
+      signaling_thread_,
+      rtc::Bind(&QuicDataChannel::OnMessage_s, this, std::move(final_message)));
+  // Once the stream is closed, OnDataReceived will not fire for the stream.
+  stream->Close();
+}
+
+void QuicDataChannel::OnReadyToSend(cricket::TransportChannel* channel) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(channel == quic_transport_channel_);
+  LOG(LS_INFO) << "QuicTransportChannel is ready to send";
+  invoker_.AsyncInvoke<void>(
+      signaling_thread_, rtc::Bind(&QuicDataChannel::SetState_s, this, kOpen));
+}
+
+void QuicDataChannel::OnWriteBlockedStreamClosed(net::QuicStreamId stream_id,
+                                                 int error) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  LOG(LS_VERBOSE) << "Write blocked stream " << stream_id << " is closed.";
+  write_blocked_quic_streams_.erase(stream_id);
+}
+
+void QuicDataChannel::OnIncomingQueuedStreamClosed(net::QuicStreamId stream_id,
+                                                   int error) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  LOG(LS_VERBOSE) << "Incoming queued stream " << stream_id << " is closed.";
+  incoming_quic_messages_.erase(stream_id);
+}
+
+void QuicDataChannel::OnConnectionClosed() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  invoker_.AsyncInvoke<void>(signaling_thread_,
+                             rtc::Bind(&QuicDataChannel::Close, this));
+}
+
+void QuicDataChannel::OnMessage_s(const DataBuffer& received_data) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  if (observer_) {
+    observer_->OnMessage(received_data);
+  }
+}
+
+void QuicDataChannel::SetState_s(DataState state) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  if (state_ == state || state_ == kClosed) {
+    return;
+  }
+  if (state_ == kClosing && state != kClosed) {
+    return;
+  }
+  LOG(LS_INFO) << "Setting state to " << state << " for QUIC data channel "
+               << id_;
+  state_ = state;
+  if (observer_) {
+    observer_->OnStateChange();
+  }
+}
+
+void QuicDataChannel::OnBufferedAmountChange_s(uint64_t buffered_amount) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  if (observer_) {
+    observer_->OnBufferedAmountChange(buffered_amount);
+  }
+}
+
+size_t QuicDataChannel::GetNumWriteBlockedStreams() const {
+  return write_blocked_quic_streams_.size();
+}
+
+size_t QuicDataChannel::GetNumIncomingStreams() const {
+  return incoming_quic_messages_.size();
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/quicdatachannel.h b/chromium/third_party/webrtc/api/quicdatachannel.h
new file mode 100644
index 00000000000..a6b987b144d
--- /dev/null
+++ b/chromium/third_party/webrtc/api/quicdatachannel.h
@@ -0,0 +1,215 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_QUICDATACHANNEL_H_
+#define WEBRTC_API_QUICDATACHANNEL_H_
+
+#include <string>
+#include <unordered_map>
+#include <unordered_set>
+
+#include "webrtc/api/datachannelinterface.h"
+#include "webrtc/base/asyncinvoker.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/thread.h"
+
+namespace cricket {
+class QuicTransportChannel;
+class ReliableQuicStream;
+class TransportChannel;
+}  // namepsace cricket
+
+namespace net {
+// TODO(mikescarlett): Make this uint64_t once QUIC uses 64-bit ids.
+typedef uint32_t QuicStreamId;
+}  // namespace net
+
+namespace rtc {
+class CopyOnWriteBuffer;
+}  // namespace rtc
+
+namespace webrtc {
+
+// Encodes a QUIC message header with the data channel ID and message ID, then
+// stores the result in |header|.
+void WriteQuicDataChannelMessageHeader(int data_channel_id,
+                                       uint64_t message_id,
+                                       rtc::CopyOnWriteBuffer* header);
+
+// Decodes the data channel ID and message ID from the initial data received by
+// an incoming QUIC stream. The data channel ID is output to |data_channel_id|,
+// the message ID is output to |message_id|, and the number of bytes read is
+// output to |bytes_read|. Returns false if either ID cannot be read.
+bool ParseQuicDataMessageHeader(const char* data,
+                                size_t len,
+                                int* data_channel_id,
+                                uint64_t* message_id,
+                                size_t* bytes_read);
+
+// QuicDataChannel is an implementation of DataChannelInterface based on the
+// QUIC protocol. It uses a QuicTransportChannel to establish encryption and
+// transfer data, and a QuicDataTransport to receive incoming messages at
+// the correct data channel. Currently this class implements unordered, reliable
+// delivery and does not send an "OPEN" message.
+//
+// Each time a message is sent:
+//
+// - The QuicDataChannel prepends it with the data channel id and message id.
+//   The QuicTransportChannel creates a ReliableQuicStream, then the
+//   ReliableQuicStream sends the message with a FIN.
+//
+// - The remote QuicSession creates a ReliableQuicStream to receive the data.
+//   The remote QuicDataTransport dispatches the ReliableQuicStream to the
+//   QuicDataChannel with the same id as this data channel.
+//
+// - The remote QuicDataChannel queues data from the ReliableQuicStream. Once
+//   it receives a QUIC stream frame with a FIN, it provides the message to the
+//   DataChannelObserver.
+//
+// TODO(mikescarlett): Implement ordered delivery, unreliable delivery, and
+// an OPEN message similar to the one for SCTP.
+class QuicDataChannel : public rtc::RefCountedObject<DataChannelInterface>,
+                        public sigslot::has_slots<> {
+ public:
+  // Message stores buffered data from the incoming QUIC stream. The QUIC stream
+  // is provided so that remaining data can be received from the remote peer.
+  struct Message {
+    uint64_t id;
+    rtc::CopyOnWriteBuffer buffer;
+    cricket::ReliableQuicStream* stream;
+  };
+
+  QuicDataChannel(rtc::Thread* signaling_thread,
+                  rtc::Thread* worker_thread,
+                  const std::string& label,
+                  const DataChannelInit& config);
+  ~QuicDataChannel() override;
+
+  // DataChannelInterface overrides.
+  std::string label() const override { return label_; }
+  bool reliable() const override { return true; }
+  bool ordered() const override { return false; }
+  uint16_t maxRetransmitTime() const override { return -1; }
+  uint16_t maxRetransmits() const override { return -1; }
+  bool negotiated() const override { return false; }
+  int id() const override { return id_; }
+  DataState state() const override { return state_; }
+  uint64_t buffered_amount() const override { return buffered_amount_; }
+  std::string protocol() const override { return protocol_; }
+  void RegisterObserver(DataChannelObserver* observer) override;
+  void UnregisterObserver() override;
+  void Close() override;
+  bool Send(const DataBuffer& buffer) override;
+
+  // Called from QuicDataTransport to set the QUIC transport channel that the
+  // QuicDataChannel sends messages with. Returns false if a different QUIC
+  // transport channel is already set or |channel| is NULL.
+  //
+  // The QUIC transport channel is not set in the constructor to allow creating
+  // the QuicDataChannel before the PeerConnection has a QUIC transport channel,
+  // such as before the session description is not set.
+  bool SetTransportChannel(cricket::QuicTransportChannel* channel);
+
+  // Called from QuicDataTransport when an incoming ReliableQuicStream is
+  // receiving a message received for this data channel. Once this function is
+  // called, |message| is owned by the QuicDataChannel and should not be
+  // accessed by the QuicDataTransport.
+  void OnIncomingMessage(Message&& message);
+
+  // Methods for testing.
+  // Gets the number of outgoing QUIC streams with write blocked data that are
+  // currently open for this data channel and are not finished writing a
+  // message. This is equivalent to the size of |write_blocked_quic_streams_|.
+  size_t GetNumWriteBlockedStreams() const;
+  // Gets the number of incoming QUIC streams with buffered data that are
+  // currently open for this data channel and are not finished receiving a
+  // message. This is equivalent to the size of |incoming_quic_messages_|.
+  size_t GetNumIncomingStreams() const;
+
+ private:
+  // Callbacks from ReliableQuicStream.
+  // Called when an incoming QUIC stream in |incoming_quic_messages_| has
+  // received a QUIC stream frame.
+  void OnDataReceived(net::QuicStreamId stream_id,
+                      const char* data,
+                      size_t len);
+  // Called when a write blocked QUIC stream that has been added to
+  // |write_blocked_quic_streams_| is closed.
+  void OnWriteBlockedStreamClosed(net::QuicStreamId stream_id, int error);
+  // Called when an incoming QUIC stream that has been added to
+  // |incoming_quic_messages_| is closed.
+  void OnIncomingQueuedStreamClosed(net::QuicStreamId stream_id, int error);
+  // Called when a write blocked QUIC stream in |write_blocked_quic_streams_|
+  // has written previously queued data.
+  void OnQueuedBytesWritten(net::QuicStreamId stream_id,
+                            uint64_t queued_bytes_written);
+
+  // Callbacks from |quic_transport_channel_|.
+  void OnReadyToSend(cricket::TransportChannel* channel);
+  void OnConnectionClosed();
+
+  // Worker thread methods.
+  // Sends the data buffer to the remote peer using an outgoing QUIC stream.
+  // Returns true if the data buffer can be successfully sent, or if it is
+  // queued to be sent later.
+  bool Send_w(const DataBuffer& buffer);
+  // Connects the |quic_transport_channel_| signals to this QuicDataChannel,
+  // then returns the new QuicDataChannel state.
+  DataState SetTransportChannel_w();
+  // Closes the QUIC streams associated with this QuicDataChannel.
+  void Close_w();
+  // Sets |buffered_amount_|.
+  void SetBufferedAmount_w(uint64_t buffered_amount);
+
+  // Signaling thread methods.
+  // Triggers QuicDataChannelObserver::OnMessage when a message from the remote
+  // peer is ready to be read.
+  void OnMessage_s(const DataBuffer& received_data);
+  // Triggers QuicDataChannel::OnStateChange if the state change is valid.
+  // Otherwise does nothing if |state| == |state_| or |state| != kClosed when
+  // the data channel is closing.
+  void SetState_s(DataState state);
+  // Triggers QuicDataChannelObserver::OnBufferedAmountChange when the total
+  // buffered data changes for a QUIC stream.
+  void OnBufferedAmountChange_s(uint64_t buffered_amount);
+
+  // QUIC transport channel which owns the QUIC session. It is used to create
+  // a QUIC stream for sending outgoing messages.
+  cricket::QuicTransportChannel* quic_transport_channel_ = nullptr;
+  // Signaling thread for DataChannelInterface methods.
+  rtc::Thread* const signaling_thread_;
+  // Worker thread for sending data and |quic_transport_channel_| callbacks.
+  rtc::Thread* const worker_thread_;
+  rtc::AsyncInvoker invoker_;
+  // Map of QUIC stream ID => ReliableQuicStream* for write blocked QUIC
+  // streams.
+  std::unordered_map<net::QuicStreamId, cricket::ReliableQuicStream*>
+      write_blocked_quic_streams_;
+  // Map of QUIC stream ID => Message for each incoming QUIC stream.
+  std::unordered_map<net::QuicStreamId, Message> incoming_quic_messages_;
+  // Handles received data from the remote peer and data channel state changes.
+  DataChannelObserver* observer_ = nullptr;
+  // QuicDataChannel ID.
+  int id_;
+  // Connectivity state of the QuicDataChannel.
+  DataState state_;
+  // Total bytes that are buffered among the QUIC streams.
+  uint64_t buffered_amount_;
+  // Counter for number of sent messages that is used for message IDs.
+  uint64_t next_message_id_;
+
+  // Variables for application use.
+  const std::string& label_;
+  const std::string& protocol_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_QUICDATACHANNEL_H_
diff --git a/chromium/third_party/webrtc/api/quicdatachannel_unittest.cc b/chromium/third_party/webrtc/api/quicdatachannel_unittest.cc
new file mode 100644
index 00000000000..e701c29b4f3
--- /dev/null
+++ b/chromium/third_party/webrtc/api/quicdatachannel_unittest.cc
@@ -0,0 +1,659 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/quicdatachannel.h"
+
+#include <map>
+#include <memory>
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
+#include "webrtc/p2p/quic/quictransportchannel.h"
+#include "webrtc/p2p/quic/reliablequicstream.h"
+
+using cricket::FakeTransportChannel;
+using cricket::QuicTransportChannel;
+using cricket::ReliableQuicStream;
+
+using webrtc::DataBuffer;
+using webrtc::DataChannelObserver;
+using webrtc::DataChannelInit;
+using webrtc::QuicDataChannel;
+
+namespace {
+
+// Timeout for asynchronous operations.
+static const int kTimeoutMs = 1000;  // milliseconds
+
+// Small messages that can be sent within a single QUIC packet.
+static const std::string kSmallMessage1 = "Hello, world!";
+static const std::string kSmallMessage2 = "WebRTC";
+static const std::string kSmallMessage3 = "1";
+static const std::string kSmallMessage4 = "abcdefghijklmnopqrstuvwxyz";
+static const DataBuffer kSmallBuffer1(kSmallMessage1);
+static const DataBuffer kSmallBuffer2(kSmallMessage2);
+static const DataBuffer kSmallBuffer3(kSmallMessage3);
+static const DataBuffer kSmallBuffer4(kSmallMessage4);
+
+// Large messages (> 1350 bytes) that exceed the max size of a QUIC packet.
+// These are < 16 KB so they don't exceed the QUIC stream flow control limit.
+static const std::string kLargeMessage1 = std::string("a", 2000);
+static const std::string kLargeMessage2 = std::string("a", 4000);
+static const std::string kLargeMessage3 = std::string("a", 8000);
+static const std::string kLargeMessage4 = std::string("a", 12000);
+static const DataBuffer kLargeBuffer1(kLargeMessage1);
+static const DataBuffer kLargeBuffer2(kLargeMessage2);
+static const DataBuffer kLargeBuffer3(kLargeMessage3);
+static const DataBuffer kLargeBuffer4(kLargeMessage4);
+
+// Oversized message (> 16 KB) that violates the QUIC stream flow control limit.
+static const std::string kOversizedMessage = std::string("a", 20000);
+static const DataBuffer kOversizedBuffer(kOversizedMessage);
+
+// Creates a fingerprint from a certificate.
+static rtc::SSLFingerprint* CreateFingerprint(rtc::RTCCertificate* cert) {
+  std::string digest_algorithm;
+  cert->ssl_certificate().GetSignatureDigestAlgorithm(&digest_algorithm);
+  std::unique_ptr<rtc::SSLFingerprint> fingerprint(
+      rtc::SSLFingerprint::Create(digest_algorithm, cert->identity()));
+  return fingerprint.release();
+}
+
+// FakeObserver receives messages from the QuicDataChannel.
+class FakeObserver : public DataChannelObserver {
+ public:
+  FakeObserver()
+      : on_state_change_count_(0), on_buffered_amount_change_count_(0) {}
+
+  // DataChannelObserver overrides.
+  void OnStateChange() override { ++on_state_change_count_; }
+  void OnBufferedAmountChange(uint64_t previous_amount) override {
+    ++on_buffered_amount_change_count_;
+  }
+  void OnMessage(const webrtc::DataBuffer& buffer) override {
+    messages_.push_back(std::string(buffer.data.data<char>(), buffer.size()));
+  }
+
+  const std::vector<std::string>& messages() const { return messages_; }
+
+  size_t messages_received() const { return messages_.size(); }
+
+  size_t on_state_change_count() const { return on_state_change_count_; }
+
+  size_t on_buffered_amount_change_count() const {
+    return on_buffered_amount_change_count_;
+  }
+
+ private:
+  std::vector<std::string> messages_;
+  size_t on_state_change_count_;
+  size_t on_buffered_amount_change_count_;
+};
+
+// FakeQuicDataTransport simulates QuicDataTransport by dispatching QUIC
+// stream messages to data channels and encoding/decoding messages.
+class FakeQuicDataTransport : public sigslot::has_slots<> {
+ public:
+  FakeQuicDataTransport() {}
+
+  void ConnectToTransportChannel(QuicTransportChannel* quic_transport_channel) {
+    quic_transport_channel->SignalIncomingStream.connect(
+        this, &FakeQuicDataTransport::OnIncomingStream);
+  }
+
+  rtc::scoped_refptr<QuicDataChannel> CreateDataChannel(
+      int id,
+      const std::string& label,
+      const std::string& protocol) {
+    DataChannelInit config;
+    config.id = id;
+    config.protocol = protocol;
+    rtc::scoped_refptr<QuicDataChannel> data_channel(new QuicDataChannel(
+        rtc::Thread::Current(), rtc::Thread::Current(), label, config));
+    data_channel_by_id_[id] = data_channel;
+    return data_channel;
+  }
+
+ private:
+  void OnIncomingStream(cricket::ReliableQuicStream* stream) {
+    incoming_stream_ = stream;
+    incoming_stream_->SignalDataReceived.connect(
+        this, &FakeQuicDataTransport::OnDataReceived);
+  }
+
+  void OnDataReceived(net::QuicStreamId id, const char* data, size_t len) {
+    ASSERT_EQ(incoming_stream_->id(), id);
+    incoming_stream_->SignalDataReceived.disconnect(this);
+    // Retrieve the data channel ID and message ID.
+    int data_channel_id;
+    uint64_t message_id;
+    size_t bytes_read;
+    ASSERT_TRUE(webrtc::ParseQuicDataMessageHeader(data, len, &data_channel_id,
+                                                   &message_id, &bytes_read));
+    data += bytes_read;
+    len -= bytes_read;
+    // Dispatch the message to the matching QuicDataChannel.
+    const auto& kv = data_channel_by_id_.find(data_channel_id);
+    ASSERT_NE(kv, data_channel_by_id_.end());
+    QuicDataChannel* data_channel = kv->second;
+    QuicDataChannel::Message message;
+    message.id = message_id;
+    message.buffer = rtc::CopyOnWriteBuffer(data, len);
+    message.stream = incoming_stream_;
+    data_channel->OnIncomingMessage(std::move(message));
+    incoming_stream_ = nullptr;
+  }
+
+  // Map of data channel ID => QuicDataChannel.
+  std::map<int, rtc::scoped_refptr<QuicDataChannel>> data_channel_by_id_;
+  // Last incoming QUIC stream which has arrived.
+  cricket::ReliableQuicStream* incoming_stream_ = nullptr;
+};
+
+// A peer who creates a QuicDataChannel to transfer data, and simulates network
+// connectivity with a fake ICE channel wrapped by the QUIC transport channel.
+class QuicDataChannelPeer {
+ public:
+  QuicDataChannelPeer()
+      : ice_transport_channel_(new FakeTransportChannel("data", 0)),
+        quic_transport_channel_(ice_transport_channel_) {
+    ice_transport_channel_->SetAsync(true);
+    fake_quic_data_transport_.ConnectToTransportChannel(
+        &quic_transport_channel_);
+  }
+
+  void GenerateCertificateAndFingerprint() {
+    rtc::scoped_refptr<rtc::RTCCertificate> local_cert =
+        rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
+            rtc::SSLIdentity::Generate("cert_name", rtc::KT_DEFAULT)));
+    quic_transport_channel_.SetLocalCertificate(local_cert);
+    local_fingerprint_.reset(CreateFingerprint(local_cert.get()));
+  }
+
+  rtc::scoped_refptr<QuicDataChannel> CreateDataChannelWithTransportChannel(
+      int id,
+      const std::string& label,
+      const std::string& protocol) {
+    rtc::scoped_refptr<QuicDataChannel> data_channel =
+        fake_quic_data_transport_.CreateDataChannel(id, label, protocol);
+    data_channel->SetTransportChannel(&quic_transport_channel_);
+    return data_channel;
+  }
+
+  rtc::scoped_refptr<QuicDataChannel> CreateDataChannelWithoutTransportChannel(
+      int id,
+      const std::string& label,
+      const std::string& protocol) {
+    return fake_quic_data_transport_.CreateDataChannel(id, label, protocol);
+  }
+
+  // Connects |ice_transport_channel_| to that of the other peer.
+  void Connect(QuicDataChannelPeer* other_peer) {
+    ice_transport_channel_->Connect();
+    other_peer->ice_transport_channel_->Connect();
+    ice_transport_channel_->SetDestination(other_peer->ice_transport_channel_);
+  }
+
+  std::unique_ptr<rtc::SSLFingerprint>& local_fingerprint() {
+    return local_fingerprint_;
+  }
+
+  QuicTransportChannel* quic_transport_channel() {
+    return &quic_transport_channel_;
+  }
+
+  FakeTransportChannel* ice_transport_channel() {
+    return ice_transport_channel_;
+  }
+
+ private:
+  FakeTransportChannel* ice_transport_channel_;
+  QuicTransportChannel quic_transport_channel_;
+
+  std::unique_ptr<rtc::SSLFingerprint> local_fingerprint_;
+
+  FakeQuicDataTransport fake_quic_data_transport_;
+};
+
+class QuicDataChannelTest : public testing::Test {
+ public:
+  QuicDataChannelTest() {}
+
+  // Connect the QuicTransportChannels and complete the crypto handshake.
+  void ConnectTransportChannels() {
+    SetCryptoParameters();
+    peer1_.Connect(&peer2_);
+    ASSERT_TRUE_WAIT(peer1_.quic_transport_channel()->writable() &&
+                         peer2_.quic_transport_channel()->writable(),
+                     kTimeoutMs);
+  }
+
+  // Sets crypto parameters required for the QUIC handshake.
+  void SetCryptoParameters() {
+    peer1_.GenerateCertificateAndFingerprint();
+    peer2_.GenerateCertificateAndFingerprint();
+
+    peer1_.quic_transport_channel()->SetSslRole(rtc::SSL_CLIENT);
+    peer2_.quic_transport_channel()->SetSslRole(rtc::SSL_SERVER);
+
+    std::unique_ptr<rtc::SSLFingerprint>& peer1_fingerprint =
+        peer1_.local_fingerprint();
+    std::unique_ptr<rtc::SSLFingerprint>& peer2_fingerprint =
+        peer2_.local_fingerprint();
+
+    peer1_.quic_transport_channel()->SetRemoteFingerprint(
+        peer2_fingerprint->algorithm,
+        reinterpret_cast<const uint8_t*>(peer2_fingerprint->digest.data()),
+        peer2_fingerprint->digest.size());
+    peer2_.quic_transport_channel()->SetRemoteFingerprint(
+        peer1_fingerprint->algorithm,
+        reinterpret_cast<const uint8_t*>(peer1_fingerprint->digest.data()),
+        peer1_fingerprint->digest.size());
+  }
+
+ protected:
+  QuicDataChannelPeer peer1_;
+  QuicDataChannelPeer peer2_;
+};
+
+// Tests that a QuicDataChannel transitions from connecting to open when
+// the QuicTransportChannel becomes writable for the first time.
+TEST_F(QuicDataChannelTest, DataChannelOpensWhenTransportChannelConnects) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(4, "label", "protocol");
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel->state());
+  ConnectTransportChannels();
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, data_channel->state(),
+                 kTimeoutMs);
+}
+
+// Tests that a QuicDataChannel transitions from connecting to open when
+// SetTransportChannel is called with a QuicTransportChannel that is already
+// writable.
+TEST_F(QuicDataChannelTest, DataChannelOpensWhenTransportChannelWritable) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithoutTransportChannel(4, "label", "protocol");
+  ConnectTransportChannels();
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel->state());
+  data_channel->SetTransportChannel(peer1_.quic_transport_channel());
+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, data_channel->state());
+}
+
+// Tests that the QuicDataChannel transfers messages small enough to fit into a
+// single QUIC stream frame.
+TEST_F(QuicDataChannelTest, TransferSmallMessage) {
+  ConnectTransportChannels();
+  int data_channel_id = 2;
+  std::string label = "label";
+  std::string protocol = "protocol";
+  rtc::scoped_refptr<QuicDataChannel> peer1_data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer1_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+  rtc::scoped_refptr<QuicDataChannel> peer2_data_channel =
+      peer2_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer2_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+
+  FakeObserver peer1_observer;
+  peer1_data_channel->RegisterObserver(&peer1_observer);
+  FakeObserver peer2_observer;
+  peer2_data_channel->RegisterObserver(&peer2_observer);
+
+  // peer1 -> peer2
+  EXPECT_TRUE(peer1_data_channel->Send(kSmallBuffer1));
+  ASSERT_EQ_WAIT(1, peer2_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kSmallMessage1, peer2_observer.messages()[0]);
+  // peer2 -> peer1
+  EXPECT_TRUE(peer2_data_channel->Send(kSmallBuffer2));
+  ASSERT_EQ_WAIT(1, peer1_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kSmallMessage2, peer1_observer.messages()[0]);
+  // peer2 -> peer1
+  EXPECT_TRUE(peer2_data_channel->Send(kSmallBuffer3));
+  ASSERT_EQ_WAIT(2, peer1_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kSmallMessage3, peer1_observer.messages()[1]);
+  // peer1 -> peer2
+  EXPECT_TRUE(peer1_data_channel->Send(kSmallBuffer4));
+  ASSERT_EQ_WAIT(2, peer2_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kSmallMessage4, peer2_observer.messages()[1]);
+}
+
+// Tests that QuicDataChannel transfers messages large enough to fit into
+// multiple QUIC stream frames, which don't violate the QUIC flow control limit.
+// These require buffering by the QuicDataChannel.
+TEST_F(QuicDataChannelTest, TransferLargeMessage) {
+  ConnectTransportChannels();
+  int data_channel_id = 347;
+  std::string label = "label";
+  std::string protocol = "protocol";
+  rtc::scoped_refptr<QuicDataChannel> peer1_data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer1_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+  rtc::scoped_refptr<QuicDataChannel> peer2_data_channel =
+      peer2_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer2_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+
+  FakeObserver peer1_observer;
+  peer1_data_channel->RegisterObserver(&peer1_observer);
+  FakeObserver peer2_observer;
+  peer2_data_channel->RegisterObserver(&peer2_observer);
+
+  // peer1 -> peer2
+  EXPECT_TRUE(peer1_data_channel->Send(kLargeBuffer1));
+  ASSERT_TRUE_WAIT(peer2_observer.messages_received() == 1, kTimeoutMs);
+  EXPECT_EQ(kLargeMessage1, peer2_observer.messages()[0]);
+  // peer2 -> peer1
+  EXPECT_TRUE(peer2_data_channel->Send(kLargeBuffer2));
+  ASSERT_EQ_WAIT(1, peer1_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kLargeMessage2, peer1_observer.messages()[0]);
+  // peer2 -> peer1
+  EXPECT_TRUE(peer2_data_channel->Send(kLargeBuffer3));
+  ASSERT_EQ_WAIT(2, peer1_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kLargeMessage3, peer1_observer.messages()[1]);
+  // peer1 -> peer2
+  EXPECT_TRUE(peer1_data_channel->Send(kLargeBuffer4));
+  ASSERT_EQ_WAIT(2, peer2_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kLargeMessage4, peer2_observer.messages()[1]);
+}
+
+// Tests that when a message size exceeds the flow control limit (> 16KB), the
+// QuicDataChannel can queue the data and send it after receiving window update
+// frames from the remote peer.
+TEST_F(QuicDataChannelTest, TransferOversizedMessage) {
+  ConnectTransportChannels();
+  int data_channel_id = 189;
+  std::string label = "label";
+  std::string protocol = "protocol";
+  rtc::scoped_refptr<QuicDataChannel> peer1_data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  rtc::scoped_refptr<QuicDataChannel> peer2_data_channel =
+      peer2_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer2_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+
+  FakeObserver peer1_observer;
+  peer1_data_channel->RegisterObserver(&peer1_observer);
+  FakeObserver peer2_observer;
+  peer2_data_channel->RegisterObserver(&peer2_observer);
+
+  EXPECT_TRUE(peer1_data_channel->Send(kOversizedBuffer));
+  EXPECT_EQ(1, peer1_data_channel->GetNumWriteBlockedStreams());
+  EXPECT_EQ_WAIT(1, peer2_data_channel->GetNumIncomingStreams(), kTimeoutMs);
+  ASSERT_EQ_WAIT(1, peer2_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(kOversizedMessage, peer2_observer.messages()[0]);
+  EXPECT_EQ(0, peer1_data_channel->GetNumWriteBlockedStreams());
+  EXPECT_EQ(0, peer2_data_channel->GetNumIncomingStreams());
+}
+
+// Tests that empty messages can be sent.
+TEST_F(QuicDataChannelTest, TransferEmptyMessage) {
+  ConnectTransportChannels();
+  int data_channel_id = 69;
+  std::string label = "label";
+  std::string protocol = "protocol";
+  rtc::scoped_refptr<QuicDataChannel> peer1_data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  rtc::scoped_refptr<QuicDataChannel> peer2_data_channel =
+      peer2_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer2_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+
+  FakeObserver peer1_observer;
+  peer1_data_channel->RegisterObserver(&peer1_observer);
+  FakeObserver peer2_observer;
+  peer2_data_channel->RegisterObserver(&peer2_observer);
+
+  EXPECT_TRUE(peer1_data_channel->Send(DataBuffer("")));
+  ASSERT_EQ_WAIT(1, peer2_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ("", peer2_observer.messages()[0]);
+}
+
+// Tests that when the QuicDataChannel is open and sends a message while the
+// QuicTransportChannel is unwritable, it gets buffered then received once the
+// QuicTransportChannel becomes writable again.
+TEST_F(QuicDataChannelTest, MessagesReceivedWhenTransportChannelReconnects) {
+  ConnectTransportChannels();
+  int data_channel_id = 401;
+  std::string label = "label";
+  std::string protocol = "protocol";
+  rtc::scoped_refptr<QuicDataChannel> peer1_data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer1_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+  rtc::scoped_refptr<QuicDataChannel> peer2_data_channel =
+      peer2_.CreateDataChannelWithTransportChannel(data_channel_id, label,
+                                                   protocol);
+  ASSERT_TRUE(peer2_data_channel->state() ==
+              webrtc::DataChannelInterface::kOpen);
+
+  FakeObserver peer1_observer;
+  peer1_data_channel->RegisterObserver(&peer1_observer);
+  FakeObserver peer2_observer;
+  peer2_data_channel->RegisterObserver(&peer2_observer);
+  // writable => unwritable
+  peer1_.ice_transport_channel()->SetWritable(false);
+  ASSERT_FALSE(peer1_.quic_transport_channel()->writable());
+  // Verify that sent data is buffered.
+  EXPECT_TRUE(peer1_data_channel->Send(kSmallBuffer1));
+  EXPECT_EQ(1, peer1_data_channel->GetNumWriteBlockedStreams());
+  EXPECT_TRUE(peer1_data_channel->Send(kSmallBuffer2));
+  EXPECT_EQ(2, peer1_data_channel->GetNumWriteBlockedStreams());
+  EXPECT_TRUE(peer1_data_channel->Send(kSmallBuffer3));
+  EXPECT_EQ(3, peer1_data_channel->GetNumWriteBlockedStreams());
+  EXPECT_TRUE(peer1_data_channel->Send(kSmallBuffer4));
+  EXPECT_EQ(4, peer1_data_channel->GetNumWriteBlockedStreams());
+  // unwritable => writable
+  peer1_.ice_transport_channel()->SetWritable(true);
+  ASSERT_TRUE(peer1_.quic_transport_channel()->writable());
+  ASSERT_EQ_WAIT(4, peer2_observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ(0, peer1_data_channel->GetNumWriteBlockedStreams());
+  EXPECT_EQ(0, peer2_data_channel->GetNumIncomingStreams());
+}
+
+// Tests that the QuicDataChannel does not send before it is open.
+TEST_F(QuicDataChannelTest, TransferMessageBeforeChannelOpens) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(6, "label", "protocol");
+  ASSERT_TRUE(data_channel->state() ==
+              webrtc::DataChannelInterface::kConnecting);
+  EXPECT_FALSE(data_channel->Send(kSmallBuffer1));
+}
+
+// Tests that the QuicDataChannel does not send after it is closed.
+TEST_F(QuicDataChannelTest, TransferDataAfterChannelClosed) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(42, "label", "protocol");
+  data_channel->Close();
+  ASSERT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, data_channel->state(),
+                 kTimeoutMs);
+  EXPECT_FALSE(data_channel->Send(kSmallBuffer1));
+}
+
+// Tests that QuicDataChannel state changes fire OnStateChanged() for the
+// observer, with the correct data channel states, when the data channel
+// transitions from kConnecting => kOpen => kClosing => kClosed.
+TEST_F(QuicDataChannelTest, OnStateChangedFired) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(7, "label", "protocol");
+  FakeObserver observer;
+  data_channel->RegisterObserver(&observer);
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel->state());
+  EXPECT_EQ(0, observer.on_state_change_count());
+  ConnectTransportChannels();
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, data_channel->state(),
+                 kTimeoutMs);
+  EXPECT_EQ(1, observer.on_state_change_count());
+  data_channel->Close();
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, data_channel->state(),
+                 kTimeoutMs);
+  // 2 state changes due to kClosing and kClosed.
+  EXPECT_EQ(3, observer.on_state_change_count());
+}
+
+// Tests that a QuicTransportChannel can be closed without being opened when it
+// is connected to a transprot chanenl.
+TEST_F(QuicDataChannelTest, NeverOpenedWithTransportChannel) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(7, "label", "protocol");
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel->state());
+  data_channel->Close();
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, data_channel->state(),
+                 kTimeoutMs);
+}
+
+// Tests that a QuicTransportChannel can be closed without being opened or
+// connected to a transport channel.
+TEST_F(QuicDataChannelTest, NeverOpenedWithoutTransportChannel) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithoutTransportChannel(7, "label", "protocol");
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel->state());
+  data_channel->Close();
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, data_channel->state(),
+                 kTimeoutMs);
+}
+
+// Tests that the QuicDataChannel is closed when the QUIC connection closes.
+TEST_F(QuicDataChannelTest, ClosedOnTransportError) {
+  ConnectTransportChannels();
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(1, "label", "protocol");
+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, data_channel->state());
+  ReliableQuicStream* stream =
+      peer1_.quic_transport_channel()->CreateQuicStream();
+  ASSERT_NE(nullptr, stream);
+  stream->CloseConnectionWithDetails(net::QuicErrorCode::QUIC_NO_ERROR,
+                                     "Closing QUIC for testing");
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, data_channel->state(),
+                 kTimeoutMs);
+}
+
+// Tests that an already closed QuicDataChannel does not fire onStateChange and
+// remains closed.
+TEST_F(QuicDataChannelTest, DoesNotChangeStateWhenClosed) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(4, "label", "protocol");
+  FakeObserver observer;
+  data_channel->RegisterObserver(&observer);
+  data_channel->Close();
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, data_channel->state(),
+                 kTimeoutMs);
+  // OnStateChange called for kClosing and kClosed.
+  EXPECT_EQ(2, observer.on_state_change_count());
+  // Call Close() again to verify that the state cannot be kClosing.
+  data_channel->Close();
+  EXPECT_EQ(webrtc::DataChannelInterface::kClosed, data_channel->state());
+  EXPECT_EQ(2, observer.on_state_change_count());
+  ConnectTransportChannels();
+  EXPECT_EQ(webrtc::DataChannelInterface::kClosed, data_channel->state());
+  EXPECT_EQ(2, observer.on_state_change_count());
+  // writable => unwritable
+  peer1_.ice_transport_channel()->SetWritable(false);
+  ASSERT_FALSE(peer1_.quic_transport_channel()->writable());
+  EXPECT_EQ(webrtc::DataChannelInterface::kClosed, data_channel->state());
+  EXPECT_EQ(2, observer.on_state_change_count());
+  // unwritable => writable
+  peer1_.ice_transport_channel()->SetWritable(true);
+  ASSERT_TRUE(peer1_.quic_transport_channel()->writable());
+  EXPECT_EQ(webrtc::DataChannelInterface::kClosed, data_channel->state());
+  EXPECT_EQ(2, observer.on_state_change_count());
+}
+
+// Tests that when the QuicDataChannel is open and the QuicTransportChannel
+// transitions between writable and unwritable, it does not fire onStateChange
+// and remains open.
+TEST_F(QuicDataChannelTest, DoesNotChangeStateWhenTransportChannelReconnects) {
+  ConnectTransportChannels();
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(4, "label", "protocol");
+  FakeObserver observer;
+  data_channel->RegisterObserver(&observer);
+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, data_channel->state());
+  EXPECT_EQ(0, observer.on_state_change_count());
+  // writable => unwritable
+  peer1_.ice_transport_channel()->SetWritable(false);
+  ASSERT_FALSE(peer1_.quic_transport_channel()->writable());
+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, data_channel->state());
+  EXPECT_EQ(0, observer.on_state_change_count());
+  // unwritable => writable
+  peer1_.ice_transport_channel()->SetWritable(true);
+  ASSERT_TRUE(peer1_.quic_transport_channel()->writable());
+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, data_channel->state());
+  EXPECT_EQ(0, observer.on_state_change_count());
+}
+
+// Tests that SetTransportChannel returns false when setting a NULL transport
+// channel or a transport channel that is not equivalent to the one already set.
+TEST_F(QuicDataChannelTest, SetTransportChannelReturnValue) {
+  rtc::scoped_refptr<QuicDataChannel> data_channel =
+      peer1_.CreateDataChannelWithTransportChannel(4, "label", "protocol");
+  EXPECT_FALSE(data_channel->SetTransportChannel(nullptr));
+  QuicTransportChannel* transport_channel = peer1_.quic_transport_channel();
+  EXPECT_TRUE(data_channel->SetTransportChannel(transport_channel));
+  EXPECT_TRUE(data_channel->SetTransportChannel(transport_channel));
+  QuicTransportChannel* other_transport_channel =
+      peer2_.quic_transport_channel();
+  EXPECT_FALSE(data_channel->SetTransportChannel(other_transport_channel));
+}
+
+// Tests that the QUIC message header is encoded with the correct number of
+// bytes and is properly decoded.
+TEST_F(QuicDataChannelTest, EncodeParseQuicDataMessageHeader) {
+  int data_channel_id1 = 127;  // 1 byte
+  uint64_t message_id1 = 0;    // 1 byte
+  rtc::CopyOnWriteBuffer header1;
+  webrtc::WriteQuicDataChannelMessageHeader(data_channel_id1, message_id1,
+                                            &header1);
+  EXPECT_EQ(2u, header1.size());
+
+  int decoded_data_channel_id1;
+  uint64_t decoded_message_id1;
+  size_t bytes_read1;
+  ASSERT_TRUE(webrtc::ParseQuicDataMessageHeader(
+      header1.data<char>(), header1.size(), &decoded_data_channel_id1,
+      &decoded_message_id1, &bytes_read1));
+  EXPECT_EQ(data_channel_id1, decoded_data_channel_id1);
+  EXPECT_EQ(message_id1, decoded_message_id1);
+  EXPECT_EQ(2u, bytes_read1);
+
+  int data_channel_id2 = 4178;           // 2 bytes
+  uint64_t message_id2 = 1324921792003;  // 6 bytes
+  rtc::CopyOnWriteBuffer header2;
+  webrtc::WriteQuicDataChannelMessageHeader(data_channel_id2, message_id2,
+                                            &header2);
+  EXPECT_EQ(8u, header2.size());
+
+  int decoded_data_channel_id2;
+  uint64_t decoded_message_id2;
+  size_t bytes_read2;
+  ASSERT_TRUE(webrtc::ParseQuicDataMessageHeader(
+      header2.data<char>(), header2.size(), &decoded_data_channel_id2,
+      &decoded_message_id2, &bytes_read2));
+  EXPECT_EQ(data_channel_id2, decoded_data_channel_id2);
+  EXPECT_EQ(message_id2, decoded_message_id2);
+  EXPECT_EQ(8u, bytes_read2);
+}
+
+}  // namespace
diff --git a/chromium/third_party/webrtc/api/quicdatatransport.cc b/chromium/third_party/webrtc/api/quicdatatransport.cc
new file mode 100644
index 00000000000..70ad03dbfdd
--- /dev/null
+++ b/chromium/third_party/webrtc/api/quicdatatransport.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/quicdatatransport.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/p2p/quic/quictransportchannel.h"
+#include "webrtc/p2p/quic/reliablequicstream.h"
+
+namespace webrtc {
+
+QuicDataTransport::QuicDataTransport(rtc::Thread* signaling_thread,
+                                     rtc::Thread* worker_thread)
+    : signaling_thread_(signaling_thread), worker_thread_(worker_thread) {
+  RTC_DCHECK(signaling_thread_);
+  RTC_DCHECK(worker_thread_);
+}
+
+QuicDataTransport::~QuicDataTransport() {}
+
+bool QuicDataTransport::SetTransportChannel(
+    cricket::QuicTransportChannel* channel) {
+  if (!channel) {
+    LOG(LS_ERROR) << "|channel| is NULL. Cannot set transport channel.";
+    return false;
+  }
+  if (quic_transport_channel_) {
+    if (channel == quic_transport_channel_) {
+      LOG(LS_WARNING) << "Ignoring duplicate transport channel.";
+      return true;
+    }
+    LOG(LS_ERROR) << "|channel| does not match existing transport channel.";
+    return false;
+  }
+
+  LOG(LS_INFO) << "Setting QuicTransportChannel for QuicDataTransport";
+  quic_transport_channel_ = channel;
+  quic_transport_channel_->SignalIncomingStream.connect(
+      this, &QuicDataTransport::OnIncomingStream);
+
+  bool success = true;
+  for (const auto& kv : data_channel_by_id_) {
+    rtc::scoped_refptr<QuicDataChannel> data_channel = kv.second;
+    if (!data_channel->SetTransportChannel(quic_transport_channel_)) {
+      LOG(LS_ERROR)
+          << "Cannot set QUIC transport channel for QUIC data channel "
+          << kv.first;
+      success = false;
+    }
+  }
+  return success;
+}
+
+rtc::scoped_refptr<DataChannelInterface> QuicDataTransport::CreateDataChannel(
+    const std::string& label,
+    const DataChannelInit* config) {
+  if (config == nullptr) {
+    return nullptr;
+  }
+  if (data_channel_by_id_.find(config->id) != data_channel_by_id_.end()) {
+    LOG(LS_ERROR) << "QUIC data channel already exists with id " << config->id;
+    return nullptr;
+  }
+  rtc::scoped_refptr<QuicDataChannel> data_channel(
+      new QuicDataChannel(signaling_thread_, worker_thread_, label, *config));
+  if (quic_transport_channel_) {
+    if (!data_channel->SetTransportChannel(quic_transport_channel_)) {
+      LOG(LS_ERROR)
+          << "Cannot set QUIC transport channel for QUIC data channel "
+          << config->id;
+    }
+  }
+
+  data_channel_by_id_[data_channel->id()] = data_channel;
+  return data_channel;
+}
+
+void QuicDataTransport::DestroyDataChannel(int id) {
+  data_channel_by_id_.erase(id);
+}
+
+bool QuicDataTransport::HasDataChannel(int id) const {
+  return data_channel_by_id_.find(id) != data_channel_by_id_.end();
+}
+
+bool QuicDataTransport::HasDataChannels() const {
+  return !data_channel_by_id_.empty();
+}
+
+// Called when a QUIC stream is created for incoming data.
+void QuicDataTransport::OnIncomingStream(cricket::ReliableQuicStream* stream) {
+  RTC_DCHECK(stream != nullptr);
+  quic_stream_by_id_[stream->id()] = stream;
+  stream->SignalDataReceived.connect(this, &QuicDataTransport::OnDataReceived);
+}
+
+// Called when the first QUIC stream frame is received for incoming data.
+void QuicDataTransport::OnDataReceived(net::QuicStreamId id,
+                                       const char* data,
+                                       size_t len) {
+  const auto& quic_stream_kv = quic_stream_by_id_.find(id);
+  if (quic_stream_kv == quic_stream_by_id_.end()) {
+    RTC_DCHECK(false);
+    return;
+  }
+  cricket::ReliableQuicStream* stream = quic_stream_kv->second;
+  stream->SignalDataReceived.disconnect(this);
+  quic_stream_by_id_.erase(id);
+  // Read the data channel ID and message ID.
+  int data_channel_id;
+  uint64_t message_id;
+  size_t bytes_read;
+  if (!ParseQuicDataMessageHeader(data, len, &data_channel_id, &message_id,
+                                  &bytes_read)) {
+    LOG(LS_ERROR) << "Could not read QUIC message header from QUIC stream "
+                  << id;
+    return;
+  }
+  data += bytes_read;
+  len -= bytes_read;
+  // Retrieve the data channel which will handle the message.
+  const auto& data_channel_kv = data_channel_by_id_.find(data_channel_id);
+  if (data_channel_kv == data_channel_by_id_.end()) {
+    // TODO(mikescarlett): Implement OPEN message to create a new
+    // QuicDataChannel when messages are received for a nonexistent ID.
+    LOG(LS_ERROR) << "Data was received for QUIC data channel "
+                  << data_channel_id
+                  << " but it is not registered to the QuicDataTransport.";
+    return;
+  }
+  QuicDataChannel* data_channel = data_channel_kv->second;
+  QuicDataChannel::Message message;
+  message.id = message_id;
+  message.buffer = rtc::CopyOnWriteBuffer(data, len);
+  message.stream = stream;
+  data_channel->OnIncomingMessage(std::move(message));
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/quicdatatransport.h b/chromium/third_party/webrtc/api/quicdatatransport.h
new file mode 100644
index 00000000000..f0c427d1b53
--- /dev/null
+++ b/chromium/third_party/webrtc/api/quicdatatransport.h
@@ -0,0 +1,90 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_QUICDATATRANSPORT_H_
+#define WEBRTC_API_QUICDATATRANSPORT_H_
+
+#include <string>
+#include <unordered_map>
+
+#include "webrtc/api/datachannelinterface.h"
+#include "webrtc/api/quicdatachannel.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/thread.h"
+
+namespace cricket {
+class QuicTransportChannel;
+class ReliableQuicStream;
+}  // namepsace cricket
+
+namespace webrtc {
+
+// QuicDataTransport creates QuicDataChannels for the PeerConnection. It also
+// handles QUIC stream demuxing by distributing incoming QUIC streams from the
+// QuicTransportChannel among the QuicDataChannels that it has created.
+//
+// QuicDataTransport reads the data channel ID from the incoming QUIC stream,
+// then looks it up in a map of ID => QuicDataChannel. If the data channel
+// exists, it sends the QUIC stream to the QuicDataChannel.
+class QuicDataTransport : public sigslot::has_slots<> {
+ public:
+  QuicDataTransport(rtc::Thread* signaling_thread, rtc::Thread* worker_thread);
+  ~QuicDataTransport() override;
+
+  // Sets the QUIC transport channel for the QuicDataChannels and the
+  // QuicDataTransport. Returns false if a different QUIC transport channel is
+  // already set, the QUIC transport channel cannot be set for any of the
+  // QuicDataChannels, or |channel| is NULL.
+  bool SetTransportChannel(cricket::QuicTransportChannel* channel);
+
+  // Creates a QuicDataChannel that uses this QuicDataTransport.
+  rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
+      const std::string& label,
+      const DataChannelInit* config);
+
+  // Removes a QuicDataChannel with the given ID from the QuicDataTransport's
+  // data channel map.
+  void DestroyDataChannel(int id);
+
+  // True if the QuicDataTransport has a data channel with the given ID.
+  bool HasDataChannel(int id) const;
+
+  // True if the QuicDataTransport has data channels.
+  bool HasDataChannels() const;
+
+ private:
+  // Called from the QuicTransportChannel when a ReliableQuicStream is created
+  // to receive incoming data.
+  void OnIncomingStream(cricket::ReliableQuicStream* stream);
+  // Called from the ReliableQuicStream when the first QUIC stream frame is
+  // received for incoming data. The QuicDataTransport reads the data channel ID
+  // and message ID from the incoming data, then dispatches the
+  // ReliableQuicStream to the QuicDataChannel with the same data channel ID.
+  void OnDataReceived(net::QuicStreamId stream_id,
+                      const char* data,
+                      size_t len);
+
+  // Map of data channel ID => QUIC data channel values.
+  std::unordered_map<int, rtc::scoped_refptr<QuicDataChannel>>
+      data_channel_by_id_;
+  // Map of QUIC stream ID => ReliableQuicStream* values.
+  std::unordered_map<net::QuicStreamId, cricket::ReliableQuicStream*>
+      quic_stream_by_id_;
+  // QuicTransportChannel for sending/receiving data.
+  cricket::QuicTransportChannel* quic_transport_channel_ = nullptr;
+  // Signaling and worker threads for the QUIC data channel.
+  rtc::Thread* const signaling_thread_;
+  rtc::Thread* const worker_thread_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_QUICDATATRANSPORT_H_
diff --git a/chromium/third_party/webrtc/api/quicdatatransport_unittest.cc b/chromium/third_party/webrtc/api/quicdatatransport_unittest.cc
new file mode 100644
index 00000000000..d668c55b0bf
--- /dev/null
+++ b/chromium/third_party/webrtc/api/quicdatatransport_unittest.cc
@@ -0,0 +1,356 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/quicdatatransport.h"
+
+#include <memory>
+#include <set>
+#include <string>
+#include <unordered_map>
+#include <vector>
+
+#include "webrtc/api/quicdatachannel.h"
+#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
+#include "webrtc/p2p/quic/quictransportchannel.h"
+#include "webrtc/p2p/quic/reliablequicstream.h"
+
+using webrtc::DataBuffer;
+using webrtc::DataChannelInit;
+using webrtc::DataChannelInterface;
+using webrtc::DataChannelObserver;
+using webrtc::QuicDataChannel;
+using webrtc::QuicDataTransport;
+using cricket::FakeTransportChannel;
+using cricket::QuicTransportChannel;
+using cricket::ReliableQuicStream;
+
+namespace {
+
+// Timeout for asynchronous operations.
+static const int kTimeoutMs = 1000;  // milliseconds
+
+// FakeObserver receives messages from the data channel.
+class FakeObserver : public DataChannelObserver {
+ public:
+  FakeObserver() {}
+
+  void OnStateChange() override {}
+
+  void OnBufferedAmountChange(uint64_t previous_amount) override {}
+
+  void OnMessage(const webrtc::DataBuffer& buffer) override {
+    messages_.push_back(std::string(buffer.data.data<char>(), buffer.size()));
+  }
+
+  const std::vector<std::string>& messages() const { return messages_; }
+
+  size_t messages_received() const { return messages_.size(); }
+
+ private:
+  std::vector<std::string> messages_;
+};
+
+// A peer who uses a QUIC transport channel and fake ICE transport channel to
+// send or receive data.
+class QuicDataTransportPeer {
+ public:
+  QuicDataTransportPeer()
+      : quic_data_transport_(rtc::Thread::Current(), rtc::Thread::Current()),
+        ice_transport_channel_(new FakeTransportChannel("data", 0)),
+        quic_transport_channel_(ice_transport_channel_) {
+    ice_transport_channel_->SetAsync(true);
+  }
+
+  void GenerateCertificateAndFingerprint() {
+    rtc::scoped_refptr<rtc::RTCCertificate> local_cert =
+        rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
+            rtc::SSLIdentity::Generate("cert_name", rtc::KT_DEFAULT)));
+    quic_transport_channel_.SetLocalCertificate(local_cert);
+    local_fingerprint_.reset(CreateFingerprint(local_cert.get()));
+  }
+
+  // Connects |ice_transport_channel_| to that of the other peer.
+  void Connect(QuicDataTransportPeer* other_peer) {
+    ice_transport_channel_->Connect();
+    other_peer->ice_transport_channel_->Connect();
+    ice_transport_channel_->SetDestination(other_peer->ice_transport_channel_);
+  }
+
+  std::unique_ptr<rtc::SSLFingerprint>& local_fingerprint() {
+    return local_fingerprint_;
+  }
+
+  QuicTransportChannel* quic_transport_channel() {
+    return &quic_transport_channel_;
+  }
+
+  // Write a messge directly to the ReliableQuicStream.
+  void WriteMessage(int data_channel_id,
+                    uint64_t message_id,
+                    const std::string& message) {
+    ReliableQuicStream* stream = quic_transport_channel_.CreateQuicStream();
+    rtc::CopyOnWriteBuffer payload;
+    webrtc::WriteQuicDataChannelMessageHeader(data_channel_id, message_id,
+                                              &payload);
+    stream->Write(payload.data<char>(), payload.size(), false);
+    stream->Write(message.data(), message.size(), true);
+  }
+
+  rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
+      const DataChannelInit* config) {
+    return quic_data_transport_.CreateDataChannel("testing", config);
+  }
+
+  QuicDataTransport* quic_data_transport() { return &quic_data_transport_; }
+
+ private:
+  // Creates a fingerprint from a certificate.
+  rtc::SSLFingerprint* CreateFingerprint(rtc::RTCCertificate* cert) {
+    std::string digest_algorithm;
+    cert->ssl_certificate().GetSignatureDigestAlgorithm(&digest_algorithm);
+    std::unique_ptr<rtc::SSLFingerprint> fingerprint(
+        rtc::SSLFingerprint::Create(digest_algorithm, cert->identity()));
+    return fingerprint.release();
+  }
+
+  QuicDataTransport quic_data_transport_;
+  FakeTransportChannel* ice_transport_channel_;
+  QuicTransportChannel quic_transport_channel_;
+  std::unique_ptr<rtc::SSLFingerprint> local_fingerprint_;
+};
+
+class QuicDataTransportTest : public testing::Test {
+ public:
+  QuicDataTransportTest() {}
+
+  void ConnectTransportChannels() {
+    SetCryptoParameters();
+    peer1_.Connect(&peer2_);
+    ASSERT_TRUE_WAIT(peer1_.quic_transport_channel()->writable() &&
+                         peer2_.quic_transport_channel()->writable(),
+                     kTimeoutMs);
+  }
+
+  void SetTransportChannels() {
+    ASSERT_TRUE(peer1_.quic_data_transport()->SetTransportChannel(
+        peer1_.quic_transport_channel()));
+    ASSERT_TRUE(peer2_.quic_data_transport()->SetTransportChannel(
+        peer2_.quic_transport_channel()));
+  }
+
+  // Sets crypto parameters required for the QUIC handshake.
+  void SetCryptoParameters() {
+    peer1_.GenerateCertificateAndFingerprint();
+    peer2_.GenerateCertificateAndFingerprint();
+
+    peer1_.quic_transport_channel()->SetSslRole(rtc::SSL_CLIENT);
+    peer2_.quic_transport_channel()->SetSslRole(rtc::SSL_SERVER);
+
+    std::unique_ptr<rtc::SSLFingerprint>& peer1_fingerprint =
+        peer1_.local_fingerprint();
+    std::unique_ptr<rtc::SSLFingerprint>& peer2_fingerprint =
+        peer2_.local_fingerprint();
+
+    peer1_.quic_transport_channel()->SetRemoteFingerprint(
+        peer2_fingerprint->algorithm,
+        reinterpret_cast<const uint8_t*>(peer2_fingerprint->digest.data()),
+        peer2_fingerprint->digest.size());
+    peer2_.quic_transport_channel()->SetRemoteFingerprint(
+        peer1_fingerprint->algorithm,
+        reinterpret_cast<const uint8_t*>(peer1_fingerprint->digest.data()),
+        peer1_fingerprint->digest.size());
+  }
+
+ protected:
+  QuicDataTransportPeer peer1_;
+  QuicDataTransportPeer peer2_;
+};
+
+// Tests creation and destruction of data channels.
+TEST_F(QuicDataTransportTest, CreateAndDestroyDataChannels) {
+  QuicDataTransport* quic_data_transport = peer2_.quic_data_transport();
+  EXPECT_FALSE(quic_data_transport->HasDataChannels());
+  for (int data_channel_id = 0; data_channel_id < 5; ++data_channel_id) {
+    EXPECT_FALSE(quic_data_transport->HasDataChannel(data_channel_id));
+    webrtc::DataChannelInit config;
+    config.id = data_channel_id;
+    rtc::scoped_refptr<DataChannelInterface> data_channel =
+        peer2_.CreateDataChannel(&config);
+    EXPECT_NE(nullptr, data_channel);
+    EXPECT_EQ(data_channel_id, data_channel->id());
+    EXPECT_TRUE(quic_data_transport->HasDataChannel(data_channel_id));
+  }
+  EXPECT_TRUE(quic_data_transport->HasDataChannels());
+  for (int data_channel_id = 0; data_channel_id < 5; ++data_channel_id) {
+    quic_data_transport->DestroyDataChannel(data_channel_id);
+    EXPECT_FALSE(quic_data_transport->HasDataChannel(data_channel_id));
+  }
+  EXPECT_FALSE(quic_data_transport->HasDataChannels());
+}
+
+// Tests that the QuicDataTransport does not allow creating multiple
+// QuicDataChannels with the same id.
+TEST_F(QuicDataTransportTest, CannotCreateDataChannelsWithSameId) {
+  webrtc::DataChannelInit config;
+  config.id = 2;
+  EXPECT_NE(nullptr, peer2_.CreateDataChannel(&config));
+  EXPECT_EQ(nullptr, peer2_.CreateDataChannel(&config));
+}
+
+// Tests that any data channels created by the QuicDataTransport are in state
+// kConnecting before the QuicTransportChannel is set, then transiton to state
+// kOpen when the transport channel becomes writable.
+TEST_F(QuicDataTransportTest, DataChannelsOpenWhenTransportChannelWritable) {
+  webrtc::DataChannelInit config1;
+  config1.id = 7;
+  rtc::scoped_refptr<DataChannelInterface> data_channel1 =
+      peer2_.CreateDataChannel(&config1);
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel1->state());
+  SetTransportChannels();
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel1->state());
+  webrtc::DataChannelInit config2;
+  config2.id = 14;
+  rtc::scoped_refptr<DataChannelInterface> data_channel2 =
+      peer2_.CreateDataChannel(&config2);
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, data_channel2->state());
+  // Existing data channels should open once the transport channel is writable.
+  ConnectTransportChannels();
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, data_channel1->state(),
+                 kTimeoutMs);
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, data_channel2->state(),
+                 kTimeoutMs);
+  // Any data channels created afterwards should start in state kOpen.
+  webrtc::DataChannelInit config3;
+  config3.id = 21;
+  rtc::scoped_refptr<DataChannelInterface> data_channel3 =
+      peer2_.CreateDataChannel(&config3);
+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, data_channel3->state());
+}
+
+// Tests that the QuicTransport dispatches messages for one QuicDataChannel.
+TEST_F(QuicDataTransportTest, ReceiveMessagesForSingleDataChannel) {
+  ConnectTransportChannels();
+  SetTransportChannels();
+
+  int data_channel_id = 1337;
+  webrtc::DataChannelInit config;
+  config.id = data_channel_id;
+  rtc::scoped_refptr<DataChannelInterface> peer2_data_channel =
+      peer2_.CreateDataChannel(&config);
+  FakeObserver observer;
+  peer2_data_channel->RegisterObserver(&observer);
+
+  uint64_t message1_id = 26u;
+  peer1_.WriteMessage(data_channel_id, message1_id, "Testing");
+  ASSERT_EQ_WAIT(1, observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ("Testing", observer.messages()[0]);
+
+  uint64_t message2_id = 402u;
+  peer1_.WriteMessage(data_channel_id, message2_id, "Hello, World!");
+  ASSERT_EQ_WAIT(2, observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ("Hello, World!", observer.messages()[1]);
+
+  uint64_t message3_id = 100260415u;
+  peer1_.WriteMessage(data_channel_id, message3_id, "Third message");
+  ASSERT_EQ_WAIT(3, observer.messages_received(), kTimeoutMs);
+  EXPECT_EQ("Third message", observer.messages()[2]);
+}
+
+// Tests that the QuicTransport dispatches messages to the correct data channel
+// when multiple are in use.
+TEST_F(QuicDataTransportTest, ReceiveMessagesForMultipleDataChannels) {
+  ConnectTransportChannels();
+  SetTransportChannels();
+
+  std::vector<rtc::scoped_refptr<DataChannelInterface>> data_channels;
+  for (int data_channel_id = 0; data_channel_id < 5; ++data_channel_id) {
+    webrtc::DataChannelInit config;
+    config.id = data_channel_id;
+    data_channels.push_back(peer2_.CreateDataChannel(&config));
+  }
+
+  for (int data_channel_id = 0; data_channel_id < 5; ++data_channel_id) {
+    uint64_t message1_id = 48023u;
+    FakeObserver observer;
+    DataChannelInterface* peer2_data_channel =
+        data_channels[data_channel_id].get();
+    peer2_data_channel->RegisterObserver(&observer);
+    peer1_.WriteMessage(data_channel_id, message1_id, "Testing");
+    ASSERT_EQ_WAIT(1, observer.messages_received(), kTimeoutMs);
+    EXPECT_EQ("Testing", observer.messages()[0]);
+
+    uint64_t message2_id = 1372643095u;
+    peer1_.WriteMessage(data_channel_id, message2_id, "Hello, World!");
+    ASSERT_EQ_WAIT(2, observer.messages_received(), kTimeoutMs);
+    EXPECT_EQ("Hello, World!", observer.messages()[1]);
+  }
+}
+
+// Tests end-to-end that both peers can use multiple QuicDataChannels to
+// send/receive messages using a QuicDataTransport.
+TEST_F(QuicDataTransportTest, EndToEndSendReceiveMessages) {
+  ConnectTransportChannels();
+  SetTransportChannels();
+
+  std::vector<rtc::scoped_refptr<DataChannelInterface>> peer1_data_channels;
+  std::vector<rtc::scoped_refptr<DataChannelInterface>> peer2_data_channels;
+
+  for (int data_channel_id = 0; data_channel_id < 5; ++data_channel_id) {
+    webrtc::DataChannelInit config;
+    config.id = data_channel_id;
+    peer1_data_channels.push_back(peer1_.CreateDataChannel(&config));
+    peer2_data_channels.push_back(peer2_.CreateDataChannel(&config));
+  }
+
+  for (int data_channel_id = 0; data_channel_id < 5; ++data_channel_id) {
+    DataChannelInterface* peer1_data_channel =
+        peer1_data_channels[data_channel_id].get();
+    FakeObserver observer1;
+    peer1_data_channel->RegisterObserver(&observer1);
+    DataChannelInterface* peer2_data_channel =
+        peer2_data_channels[data_channel_id].get();
+    FakeObserver observer2;
+    peer2_data_channel->RegisterObserver(&observer2);
+
+    peer1_data_channel->Send(webrtc::DataBuffer("Peer 1 message 1"));
+    ASSERT_EQ_WAIT(1, observer2.messages_received(), kTimeoutMs);
+    EXPECT_EQ("Peer 1 message 1", observer2.messages()[0]);
+
+    peer1_data_channel->Send(webrtc::DataBuffer("Peer 1 message 2"));
+    ASSERT_EQ_WAIT(2, observer2.messages_received(), kTimeoutMs);
+    EXPECT_EQ("Peer 1 message 2", observer2.messages()[1]);
+
+    peer2_data_channel->Send(webrtc::DataBuffer("Peer 2 message 1"));
+    ASSERT_EQ_WAIT(1, observer1.messages_received(), kTimeoutMs);
+    EXPECT_EQ("Peer 2 message 1", observer1.messages()[0]);
+
+    peer2_data_channel->Send(webrtc::DataBuffer("Peer 2 message 2"));
+    ASSERT_EQ_WAIT(2, observer1.messages_received(), kTimeoutMs);
+    EXPECT_EQ("Peer 2 message 2", observer1.messages()[1]);
+  }
+}
+
+// Tests that SetTransportChannel returns false when setting a NULL transport
+// channel or a transport channel that is not equivalent to the one already set.
+TEST_F(QuicDataTransportTest, SetTransportChannelReturnValue) {
+  QuicDataTransport* quic_data_transport = peer1_.quic_data_transport();
+  EXPECT_FALSE(quic_data_transport->SetTransportChannel(nullptr));
+  QuicTransportChannel* transport_channel = peer1_.quic_transport_channel();
+  EXPECT_TRUE(quic_data_transport->SetTransportChannel(transport_channel));
+  EXPECT_TRUE(quic_data_transport->SetTransportChannel(transport_channel));
+  QuicTransportChannel* other_transport_channel =
+      peer2_.quic_transport_channel();
+  EXPECT_FALSE(
+      quic_data_transport->SetTransportChannel(other_transport_channel));
+}
+
+}  // namespace
diff --git a/chromium/third_party/webrtc/api/remoteaudiosource.cc b/chromium/third_party/webrtc/api/remoteaudiosource.cc
index 2fedc9d0f94..2d0785a08f1 100644
--- a/chromium/third_party/webrtc/api/remoteaudiosource.cc
+++ b/chromium/third_party/webrtc/api/remoteaudiosource.cc
@@ -12,10 +12,12 @@
 
 #include <algorithm>
 #include <functional>
+#include <memory>
 #include <utility>
 
 #include "webrtc/api/mediastreamprovider.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
 
@@ -80,7 +82,7 @@ void RemoteAudioSource::Initialize(uint32_t ssrc,
   // we register for callbacks here and not on demand in AddSink.
   if (provider) {  // May be null in tests.
     provider->SetRawAudioSink(
-        ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)));
+        ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this)));
   }
 }
 
diff --git a/chromium/third_party/webrtc/api/rtpparameters.h b/chromium/third_party/webrtc/api/rtpparameters.h
index 2c29d9843d9..5c79ab4e9df 100644
--- a/chromium/third_party/webrtc/api/rtpparameters.h
+++ b/chromium/third_party/webrtc/api/rtpparameters.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_API_RTPPARAMETERS_H_
 #define WEBRTC_API_RTPPARAMETERS_H_
 
+#include <string>
 #include <vector>
 
 namespace webrtc {
@@ -20,10 +21,37 @@ namespace webrtc {
 struct RtpEncodingParameters {
   bool active = true;
   int max_bitrate_bps = -1;
+
+  bool operator==(const RtpEncodingParameters& o) const {
+    return active == o.active && max_bitrate_bps == o.max_bitrate_bps;
+  }
+  bool operator!=(const RtpEncodingParameters& o) const {
+    return !(*this == o);
+  }
+};
+
+struct RtpCodecParameters {
+  int payload_type;
+  std::string mime_type;
+  int clock_rate;
+  int channels = 1;
+  // TODO(deadbeef): Add sdpFmtpLine field.
+
+  bool operator==(const RtpCodecParameters& o) const {
+    return payload_type == o.payload_type && mime_type == o.mime_type &&
+           clock_rate == o.clock_rate && channels == o.channels;
+  }
+  bool operator!=(const RtpCodecParameters& o) const { return !(*this == o); }
 };
 
 struct RtpParameters {
   std::vector<RtpEncodingParameters> encodings;
+  std::vector<RtpCodecParameters> codecs;
+
+  bool operator==(const RtpParameters& o) const {
+    return encodings == o.encodings && codecs == o.codecs;
+  }
+  bool operator!=(const RtpParameters& o) const { return !(*this == o); }
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/rtpreceiver.cc b/chromium/third_party/webrtc/api/rtpreceiver.cc
index 0150dfd546c..1b52ce27def 100644
--- a/chromium/third_party/webrtc/api/rtpreceiver.cc
+++ b/chromium/third_party/webrtc/api/rtpreceiver.cc
@@ -12,7 +12,9 @@
 
 #include "webrtc/api/mediastreamtrackproxy.h"
 #include "webrtc/api/audiotrack.h"
+#include "webrtc/api/videosourceproxy.h"
 #include "webrtc/api/videotrack.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
@@ -65,6 +67,15 @@ void AudioRtpReceiver::Stop() {
   provider_ = nullptr;
 }
 
+RtpParameters AudioRtpReceiver::GetParameters() const {
+  return provider_->GetAudioRtpReceiveParameters(ssrc_);
+}
+
+bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
+  return provider_->SetAudioRtpReceiveParameters(ssrc_, parameters);
+}
+
 void AudioRtpReceiver::Reconfigure() {
   if (!provider_) {
     return;
@@ -81,11 +92,15 @@ VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
       ssrc_(ssrc),
       provider_(provider),
       source_(new RefCountedObject<VideoTrackSource>(&broadcaster_,
-                                                     worker_thread,
                                                      true /* remote */)),
       track_(VideoTrackProxy::Create(
           rtc::Thread::Current(),
-          VideoTrack::Create(track_id, source_.get()))) {
+          worker_thread,
+          VideoTrack::Create(
+              track_id,
+              VideoTrackSourceProxy::Create(rtc::Thread::Current(),
+                                            worker_thread,
+                                            source_)))) {
   source_->SetState(MediaSourceInterface::kLive);
   provider_->SetVideoPlayout(ssrc_, true, &broadcaster_);
   stream->AddTrack(track_);
@@ -108,4 +123,13 @@ void VideoRtpReceiver::Stop() {
   provider_ = nullptr;
 }
 
+RtpParameters VideoRtpReceiver::GetParameters() const {
+  return provider_->GetVideoRtpReceiveParameters(ssrc_);
+}
+
+bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
+  return provider_->SetVideoRtpReceiveParameters(ssrc_, parameters);
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/rtpreceiver.h b/chromium/third_party/webrtc/api/rtpreceiver.h
index b5818573da1..2e7339d4da4 100644
--- a/chromium/third_party/webrtc/api/rtpreceiver.h
+++ b/chromium/third_party/webrtc/api/rtpreceiver.h
@@ -56,6 +56,9 @@ class AudioRtpReceiver : public ObserverInterface,
 
   void Stop() override;
 
+  RtpParameters GetParameters() const override;
+  bool SetParameters(const RtpParameters& parameters) override;
+
  private:
   void Reconfigure();
 
@@ -89,6 +92,9 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
 
   void Stop() override;
 
+  RtpParameters GetParameters() const override;
+  bool SetParameters(const RtpParameters& parameters) override;
+
  private:
   std::string id_;
   uint32_t ssrc_;
diff --git a/chromium/third_party/webrtc/api/rtpreceiverinterface.h b/chromium/third_party/webrtc/api/rtpreceiverinterface.h
index 5c7790db607..ef4f0e113f3 100644
--- a/chromium/third_party/webrtc/api/rtpreceiverinterface.h
+++ b/chromium/third_party/webrtc/api/rtpreceiverinterface.h
@@ -33,16 +33,24 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
 
   virtual void Stop() = 0;
 
+  // The WebRTC specification only defines RTCRtpParameters in terms of senders,
+  // but this API also applies them to receivers, similar to ORTC:
+  // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+  virtual RtpParameters GetParameters() const = 0;
+  virtual bool SetParameters(const RtpParameters& parameters) = 0;
+
  protected:
   virtual ~RtpReceiverInterface() {}
 };
 
 // Define proxy for RtpReceiverInterface.
-BEGIN_PROXY_MAP(RtpReceiver)
+BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
 PROXY_CONSTMETHOD0(std::string, id)
 PROXY_METHOD0(void, Stop)
-END_PROXY()
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
+PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
+END_SIGNALING_PROXY()
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/rtpsender.cc b/chromium/third_party/webrtc/api/rtpsender.cc
index 94cea6c2c99..5577b9c7500 100644
--- a/chromium/third_party/webrtc/api/rtpsender.cc
+++ b/chromium/third_party/webrtc/api/rtpsender.cc
@@ -13,6 +13,7 @@
 #include "webrtc/api/localaudiosource.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/base/helpers.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
@@ -86,6 +87,7 @@ AudioRtpSender::~AudioRtpSender() {
 }
 
 void AudioRtpSender::OnChanged() {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
@@ -96,6 +98,7 @@ void AudioRtpSender::OnChanged() {
 }
 
 bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetTrack");
   if (stopped_) {
     LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
     return false;
@@ -119,6 +122,9 @@ bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 
   // Attach to new track.
   bool prev_can_send_track = can_send_track();
+  // Keep a reference to the old track to keep it alive until we call
+  // SetAudioSend.
+  rtc::scoped_refptr<AudioTrackInterface> old_track = track_;
   track_ = audio_track;
   if (track_) {
     cached_track_enabled_ = track_->enabled();
@@ -140,6 +146,7 @@ bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 }
 
 void AudioRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
@@ -161,6 +168,7 @@ void AudioRtpSender::SetSsrc(uint32_t ssrc) {
 }
 
 void AudioRtpSender::Stop() {
+  TRACE_EVENT0("webrtc", "AudioRtpSender::Stop");
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
   if (stopped_) {
     return;
@@ -200,11 +208,12 @@ void AudioRtpSender::SetAudioSend() {
 }
 
 RtpParameters AudioRtpSender::GetParameters() const {
-  return provider_->GetAudioRtpParameters(ssrc_);
+  return provider_->GetAudioRtpSendParameters(ssrc_);
 }
 
 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
-  return provider_->SetAudioRtpParameters(ssrc_, parameters);
+  TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
+  return provider_->SetAudioRtpSendParameters(ssrc_, parameters);
 }
 
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
@@ -240,6 +249,7 @@ VideoRtpSender::~VideoRtpSender() {
 }
 
 void VideoRtpSender::OnChanged() {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
@@ -250,6 +260,7 @@ void VideoRtpSender::OnChanged() {
 }
 
 bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetTrack");
   if (stopped_) {
     LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
     return false;
@@ -268,6 +279,9 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 
   // Attach to new track.
   bool prev_can_send_track = can_send_track();
+  // Keep a reference to the old track to keep it alive until we call
+  // SetSource.
+  rtc::scoped_refptr<VideoTrackInterface> old_track = track_;
   track_ = video_track;
   if (track_) {
     cached_track_enabled_ = track_->enabled();
@@ -276,42 +290,40 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
 
   // Update video provider.
   if (can_send_track()) {
-    VideoTrackSourceInterface* source = track_->GetSource();
     // TODO(deadbeef): If SetTrack is called with a disabled track, and the
     // previous track was enabled, this could cause a frame from the new track
-    // to slip out. Really, what we need is for SetCaptureDevice and
-    // SetVideoSend
+    // to slip out. Really, what we need is for SetSource and SetVideoSend
     // to be combined into one atomic operation, all the way down to
     // WebRtcVideoSendStream.
-    provider_->SetCaptureDevice(ssrc_,
-                                source ? source->GetVideoCapturer() : nullptr);
+
+    provider_->SetSource(ssrc_, track_);
     SetVideoSend();
   } else if (prev_can_send_track) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
+    provider_->SetSource(ssrc_, nullptr);
     provider_->SetVideoSend(ssrc_, false, nullptr);
   }
   return true;
 }
 
 void VideoRtpSender::SetSsrc(uint32_t ssrc) {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
   // If we are already sending with a particular SSRC, stop sending.
   if (can_send_track()) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
+    provider_->SetSource(ssrc_, nullptr);
     provider_->SetVideoSend(ssrc_, false, nullptr);
   }
   ssrc_ = ssrc;
   if (can_send_track()) {
-    VideoTrackSourceInterface* source = track_->GetSource();
-    provider_->SetCaptureDevice(ssrc_,
-                                source ? source->GetVideoCapturer() : nullptr);
+    provider_->SetSource(ssrc_, track_);
     SetVideoSend();
   }
 }
 
 void VideoRtpSender::Stop() {
+  TRACE_EVENT0("webrtc", "VideoRtpSender::Stop");
   // TODO(deadbeef): Need to do more here to fully stop sending packets.
   if (stopped_) {
     return;
@@ -320,7 +332,7 @@ void VideoRtpSender::Stop() {
     track_->UnregisterObserver(this);
   }
   if (can_send_track()) {
-    provider_->SetCaptureDevice(ssrc_, nullptr);
+    provider_->SetSource(ssrc_, nullptr);
     provider_->SetVideoSend(ssrc_, false, nullptr);
   }
   stopped_ = true;
@@ -338,11 +350,12 @@ void VideoRtpSender::SetVideoSend() {
 }
 
 RtpParameters VideoRtpSender::GetParameters() const {
-  return provider_->GetVideoRtpParameters(ssrc_);
+  return provider_->GetVideoRtpSendParameters(ssrc_);
 }
 
 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
-  return provider_->SetVideoRtpParameters(ssrc_, parameters);
+  TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
+  return provider_->SetVideoRtpSendParameters(ssrc_, parameters);
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/rtpsender.h b/chromium/third_party/webrtc/api/rtpsender.h
index 3919e070dc5..ffe5daeb010 100644
--- a/chromium/third_party/webrtc/api/rtpsender.h
+++ b/chromium/third_party/webrtc/api/rtpsender.h
@@ -15,6 +15,7 @@
 #ifndef WEBRTC_API_RTPSENDER_H_
 #define WEBRTC_API_RTPSENDER_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/api/mediastreamprovider.h"
@@ -22,7 +23,6 @@
 #include "webrtc/api/statscollector.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/audiosource.h"
 
 namespace webrtc {
@@ -97,8 +97,8 @@ class AudioRtpSender : public ObserverInterface,
 
   void Stop() override;
 
-  RtpParameters GetParameters() const;
-  bool SetParameters(const RtpParameters& parameters);
+  RtpParameters GetParameters() const override;
+  bool SetParameters(const RtpParameters& parameters) override;
 
  private:
   // TODO(nisse): Since SSRC == 0 is technically valid, figure out
@@ -119,7 +119,7 @@ class AudioRtpSender : public ObserverInterface,
 
   // Used to pass the data callback from the |track_| to the other end of
   // cricket::AudioSource.
-  rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
+  std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
 };
 
 class VideoRtpSender : public ObserverInterface,
@@ -163,8 +163,8 @@ class VideoRtpSender : public ObserverInterface,
 
   void Stop() override;
 
-  RtpParameters GetParameters() const;
-  bool SetParameters(const RtpParameters& parameters);
+  RtpParameters GetParameters() const override;
+  bool SetParameters(const RtpParameters& parameters) override;
 
  private:
   bool can_send_track() const { return track_ && ssrc_; }
diff --git a/chromium/third_party/webrtc/api/rtpsenderinterface.h b/chromium/third_party/webrtc/api/rtpsenderinterface.h
index 776d01ace6b..2291bb4e214 100644
--- a/chromium/third_party/webrtc/api/rtpsenderinterface.h
+++ b/chromium/third_party/webrtc/api/rtpsenderinterface.h
@@ -60,7 +60,7 @@ class RtpSenderInterface : public rtc::RefCountInterface {
 };
 
 // Define proxy for RtpSenderInterface.
-BEGIN_PROXY_MAP(RtpSender)
+BEGIN_SIGNALING_PROXY_MAP(RtpSender)
 PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
 PROXY_METHOD1(void, SetSsrc, uint32_t)
@@ -72,7 +72,7 @@ PROXY_CONSTMETHOD0(std::string, stream_id)
 PROXY_METHOD0(void, Stop)
 PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
 PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
-END_PROXY()
+END_SIGNALING_PROXY()
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/rtpsenderreceiver_unittest.cc b/chromium/third_party/webrtc/api/rtpsenderreceiver_unittest.cc
index 7f32f29bc4b..3db9d5e00ee 100644
--- a/chromium/third_party/webrtc/api/rtpsenderreceiver_unittest.cc
+++ b/chromium/third_party/webrtc/api/rtpsenderreceiver_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 #include <utility>
 
@@ -27,6 +28,7 @@
 
 using ::testing::_;
 using ::testing::Exactly;
+using ::testing::InvokeWithoutArgs;
 using ::testing::Return;
 
 static const char kStreamLabel1[] = "local_stream_1";
@@ -42,7 +44,11 @@ namespace webrtc {
 // Helper class to test RtpSender/RtpReceiver.
 class MockAudioProvider : public AudioProviderInterface {
  public:
-  ~MockAudioProvider() override {}
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
+  ~MockAudioProvider() /* override */ {}
 
   MOCK_METHOD2(SetAudioPlayout,
                void(uint32_t ssrc,
@@ -53,25 +59,30 @@ class MockAudioProvider : public AudioProviderInterface {
                     const cricket::AudioOptions& options,
                     cricket::AudioSource* source));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
-  MOCK_CONST_METHOD1(GetAudioRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetAudioRtpParameters,
+  MOCK_CONST_METHOD1(GetAudioRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetAudioRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 
-  void SetRawAudioSink(uint32_t,
-                       rtc::scoped_ptr<AudioSinkInterface> sink) override {
+  void SetRawAudioSink(
+      uint32_t, std::unique_ptr<AudioSinkInterface> sink) /* override */ {
     sink_ = std::move(sink);
   }
 
  private:
-  rtc::scoped_ptr<AudioSinkInterface> sink_;
+  std::unique_ptr<AudioSinkInterface> sink_;
 };
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockVideoProvider : public VideoProviderInterface {
  public:
   virtual ~MockVideoProvider() {}
-  MOCK_METHOD2(SetCaptureDevice,
-               bool(uint32_t ssrc, cricket::VideoCapturer* camera));
+  MOCK_METHOD2(SetSource,
+               bool(uint32_t ssrc,
+                    rtc::VideoSourceInterface<cricket::VideoFrame>* source));
   MOCK_METHOD3(SetVideoPlayout,
                void(uint32_t ssrc,
                     bool enable,
@@ -81,8 +92,12 @@ class MockVideoProvider : public VideoProviderInterface {
                     bool enable,
                     const cricket::VideoOptions* options));
 
-  MOCK_CONST_METHOD1(GetVideoRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetVideoRtpParameters,
+  MOCK_CONST_METHOD1(GetVideoRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetVideoRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 };
 
@@ -111,9 +126,7 @@ class RtpSenderReceiverTest : public testing::Test {
 
   void CreateVideoRtpSender() {
     AddVideoTrack();
-    EXPECT_CALL(video_provider_,
-                SetCaptureDevice(
-                    kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
+    EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, video_track_.get()));
     EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
     video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
                                            stream_->label(), &video_provider_);
@@ -127,7 +140,7 @@ class RtpSenderReceiverTest : public testing::Test {
   }
 
   void DestroyVideoRtpSender() {
-    EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, NULL)).Times(1);
+    EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, NULL)).Times(1);
     EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
     video_rtp_sender_ = nullptr;
   }
@@ -345,14 +358,12 @@ TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
   rtc::scoped_refptr<VideoRtpSender> sender =
       new VideoRtpSender(&video_provider_);
   sender->SetSsrc(kVideoSsrc);
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, video_track_.get()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
   sender->SetTrack(video_track_);
 
   // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
 }
 
@@ -363,14 +374,12 @@ TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
   rtc::scoped_refptr<VideoRtpSender> sender =
       new VideoRtpSender(&video_provider_);
   sender->SetTrack(video_track_);
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, video_track_.get()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
   sender->SetSsrc(kVideoSsrc);
 
   // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
 }
 
@@ -396,21 +405,19 @@ TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
 // set to 0.
 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
   AddVideoTrack();
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, video_track_.get()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
   rtc::scoped_refptr<VideoRtpSender> sender =
       new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
   sender->SetSsrc(kVideoSsrc);
 
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
   sender->SetSsrc(0);
 
   // Make sure it's SetSsrc that called methods on the provider, and not the
   // destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
+  EXPECT_CALL(video_provider_, SetSource(_, _)).Times(0);
   EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
 }
 
@@ -422,7 +429,14 @@ TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
       new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
   sender->SetSsrc(kAudioSsrc);
 
-  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+  // Expect that SetAudioSend will be called before the reference to the track
+  // is released.
+  EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, nullptr))
+      .Times(1)
+      .WillOnce(InvokeWithoutArgs([&track] {
+        EXPECT_LT(2, track->AddRef());
+        track->Release();
+      }));
   EXPECT_TRUE(sender->SetTrack(nullptr));
 
   // Make sure it's SetTrack that called methods on the provider, and not the
@@ -431,22 +445,31 @@ TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
 }
 
 TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
-  AddVideoTrack();
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
+  rtc::scoped_refptr<VideoTrackSourceInterface> source(
+      FakeVideoTrackSource::Create());
+  rtc::scoped_refptr<VideoTrackInterface> track =
+      VideoTrack::Create(kVideoTrackId, source);
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, track.get()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
   rtc::scoped_refptr<VideoRtpSender> sender =
-      new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
+      new VideoRtpSender(track, kStreamLabel1, &video_provider_);
   sender->SetSsrc(kVideoSsrc);
 
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+  // Expect that SetSource will be called before the reference to the track
+  // is released.
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, nullptr))
+      .Times(1)
+      .WillOnce(InvokeWithoutArgs([&track] {
+        EXPECT_LT(2, track->AddRef());
+        track->Release();
+        return true;
+      }));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
   EXPECT_TRUE(sender->SetTrack(nullptr));
 
   // Make sure it's SetTrack that called methods on the provider, and not the
   // destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
+  EXPECT_CALL(video_provider_, SetSource(_, _)).Times(0);
   EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
 }
 
@@ -469,33 +492,29 @@ TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
 
 TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
   AddVideoTrack();
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc,
-                               video_track_->GetSource()->GetVideoCapturer()));
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, video_track_.get()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
   rtc::scoped_refptr<VideoRtpSender> sender =
       new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
   sender->SetSsrc(kVideoSsrc);
 
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-  EXPECT_CALL(video_provider_,
-              SetCaptureDevice(kVideoSsrc2,
-                               video_track_->GetSource()->GetVideoCapturer()));
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc2, video_track_.get()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
   sender->SetSsrc(kVideoSsrc2);
 
   // Calls expected from destructor.
-  EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
+  EXPECT_CALL(video_provider_, SetSource(kVideoSsrc2, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
 }
 
 TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
   CreateAudioRtpSender();
 
-  EXPECT_CALL(audio_provider_, GetAudioRtpParameters(kAudioSsrc))
+  EXPECT_CALL(audio_provider_, GetAudioRtpSendParameters(kAudioSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(audio_provider_, SetAudioRtpParameters(kAudioSsrc, _))
+  EXPECT_CALL(audio_provider_, SetAudioRtpSendParameters(kAudioSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
@@ -506,9 +525,9 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   CreateVideoRtpSender();
 
-  EXPECT_CALL(video_provider_, GetVideoRtpParameters(kVideoSsrc))
+  EXPECT_CALL(video_provider_, GetVideoRtpSendParameters(kVideoSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _))
+  EXPECT_CALL(video_provider_, SetVideoRtpSendParameters(kVideoSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
@@ -516,4 +535,30 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   DestroyVideoRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
+  CreateAudioRtpReceiver();
+
+  EXPECT_CALL(audio_provider_, GetAudioRtpReceiveParameters(kAudioSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(audio_provider_, SetAudioRtpReceiveParameters(kAudioSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = audio_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
+
+  DestroyAudioRtpReceiver();
+}
+
+TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
+  CreateVideoRtpReceiver();
+
+  EXPECT_CALL(video_provider_, GetVideoRtpReceiveParameters(kVideoSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(video_provider_, SetVideoRtpReceiveParameters(kVideoSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = video_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
+
+  DestroyVideoRtpReceiver();
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/statscollector.cc b/chromium/third_party/webrtc/api/statscollector.cc
index 0901fc61b1e..df1d0aa0d32 100644
--- a/chromium/third_party/webrtc/api/statscollector.cc
+++ b/chromium/third_party/webrtc/api/statscollector.cc
@@ -10,18 +10,16 @@
 
 #include "webrtc/api/statscollector.h"
 
+#include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/peerconnection.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/timing.h"
 #include "webrtc/pc/channel.h"
 
-using rtc::scoped_ptr;
-
 namespace webrtc {
 namespace {
 
@@ -90,7 +88,9 @@ void ExtractCommonSendProperties(const cricket::MediaSenderInfo& info,
                                  StatsReport* report) {
   report->AddString(StatsReport::kStatsValueNameCodecName, info.codec_name);
   report->AddInt64(StatsReport::kStatsValueNameBytesSent, info.bytes_sent);
-  report->AddInt64(StatsReport::kStatsValueNameRtt, info.rtt_ms);
+  if (info.rtt_ms >= 0) {
+    report->AddInt64(StatsReport::kStatsValueNameRtt, info.rtt_ms);
+  }
 }
 
 void ExtractCommonReceiveProperties(const cricket::MediaReceiverInfo& info,
@@ -107,17 +107,23 @@ void SetAudioProcessingStats(StatsReport* report,
                              int echo_delay_std_ms) {
   report->AddBoolean(StatsReport::kStatsValueNameTypingNoiseState,
                      typing_noise_detected);
-  report->AddFloat(StatsReport::kStatsValueNameEchoCancellationQualityMin,
-                   aec_quality_min);
+  if (aec_quality_min >= 0.0f) {
+    report->AddFloat(StatsReport::kStatsValueNameEchoCancellationQualityMin,
+                     aec_quality_min);
+  }
   const IntForAdd ints[] = {
-    { StatsReport::kStatsValueNameEchoReturnLoss, echo_return_loss },
-    { StatsReport::kStatsValueNameEchoReturnLossEnhancement,
-      echo_return_loss_enhancement },
     { StatsReport::kStatsValueNameEchoDelayMedian, echo_delay_median_ms },
     { StatsReport::kStatsValueNameEchoDelayStdDev, echo_delay_std_ms },
   };
-  for (const auto& i : ints)
-    report->AddInt(i.name, i.value);
+  for (const auto& i : ints) {
+    if (i.value >= 0) {
+      report->AddInt(i.name, i.value);
+    }
+  }
+  // These can take on valid negative values.
+  report->AddInt(StatsReport::kStatsValueNameEchoReturnLoss, echo_return_loss);
+  report->AddInt(StatsReport::kStatsValueNameEchoReturnLossEnhancement,
+                 echo_return_loss_enhancement);
 }
 
 void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
@@ -133,7 +139,6 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
   };
 
   const IntForAdd ints[] = {
-    { StatsReport::kStatsValueNameAudioOutputLevel, info.audio_level },
     { StatsReport::kStatsValueNameCurrentDelayMs, info.delay_estimate_ms },
     { StatsReport::kStatsValueNameDecodingCNG, info.decoding_cng },
     { StatsReport::kStatsValueNameDecodingCTN, info.decoding_calls_to_neteq },
@@ -155,11 +160,17 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
 
   for (const auto& i : ints)
     report->AddInt(i.name, i.value);
+  if (info.audio_level >= 0) {
+    report->AddInt(StatsReport::kStatsValueNameAudioOutputLevel,
+                   info.audio_level);
+  }
 
   report->AddInt64(StatsReport::kStatsValueNameBytesReceived,
                    info.bytes_rcvd);
-  report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
-                   info.capture_start_ntp_time_ms);
+  if (info.capture_start_ntp_time_ms >= 0) {
+    report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
+                     info.capture_start_ntp_time_ms);
+  }
   report->AddString(StatsReport::kStatsValueNameMediaType, "audio");
 }
 
@@ -179,8 +190,11 @@ void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
     { StatsReport::kStatsValueNamePacketsSent, info.packets_sent },
   };
 
-  for (const auto& i : ints)
-    report->AddInt(i.name, i.value);
+  for (const auto& i : ints) {
+    if (i.value >= 0) {
+      report->AddInt(i.name, i.value);
+    }
+  }
   report->AddString(StatsReport::kStatsValueNameMediaType, "audio");
 }
 
@@ -190,8 +204,10 @@ void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
                     info.decoder_implementation_name);
   report->AddInt64(StatsReport::kStatsValueNameBytesReceived,
                    info.bytes_rcvd);
-  report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
-                   info.capture_start_ntp_time_ms);
+  if (info.capture_start_ntp_time_ms >= 0) {
+    report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
+                     info.capture_start_ntp_time_ms);
+  }
   const IntForAdd ints[] = {
     { StatsReport::kStatsValueNameCurrentDelayMs, info.current_delay_ms },
     { StatsReport::kStatsValueNameDecodeMs, info.decode_ms },
@@ -532,7 +548,7 @@ StatsReport* StatsCollector::AddOneCertificateReport(
   if (!cert->GetSignatureDigestAlgorithm(&digest_algorithm))
     return nullptr;
 
-  rtc::scoped_ptr<rtc::SSLFingerprint> ssl_fingerprint(
+  std::unique_ptr<rtc::SSLFingerprint> ssl_fingerprint(
       rtc::SSLFingerprint::Create(digest_algorithm, cert));
 
   // SSLFingerprint::Create can fail if the algorithm returned by
@@ -574,7 +590,7 @@ StatsReport* StatsCollector::AddCertificateReports(
   RTC_DCHECK(cert != NULL);
 
   StatsReport* issuer = nullptr;
-  rtc::scoped_ptr<rtc::SSLCertChain> chain = cert->GetChain();
+  std::unique_ptr<rtc::SSLCertChain> chain = cert->GetChain();
   if (chain) {
     // This loop runs in reverse, i.e. from root to leaf, so that each
     // certificate's issuer's report ID is known before the child certificate's
@@ -702,7 +718,7 @@ void StatsCollector::ExtractSessionInfo() {
         local_cert_report_id = r->id();
     }
 
-    rtc::scoped_ptr<rtc::SSLCertificate> cert =
+    std::unique_ptr<rtc::SSLCertificate> cert =
         pc_->session()->GetRemoteSSLCertificate(
             transport_iter.second.transport_name);
     if (cert) {
@@ -872,7 +888,10 @@ void StatsCollector::ExtractDataInfo() {
     StatsReport* report = reports_.ReplaceOrAddNew(id);
     report->set_timestamp(stats_gathering_started_);
     report->AddString(StatsReport::kStatsValueNameLabel, dc->label());
-    report->AddInt(StatsReport::kStatsValueNameDataChannelId, dc->id());
+    // Filter out the initial id (-1).
+    if (dc->id() >= 0) {
+      report->AddInt(StatsReport::kStatsValueNameDataChannelId, dc->id());
+    }
     report->AddString(StatsReport::kStatsValueNameProtocol, dc->protocol());
     report->AddString(StatsReport::kStatsValueNameState,
                       DataChannelInterface::DataStateString(dc->state()));
@@ -937,6 +956,9 @@ void StatsCollector::UpdateReportFromAudioTrack(AudioTrackInterface* track,
         report, stats.typing_noise_detected, stats.echo_return_loss,
         stats.echo_return_loss_enhancement, stats.echo_delay_median_ms,
         stats.aec_quality_min, stats.echo_delay_std_ms);
+
+    report->AddFloat(StatsReport::kStatsValueNameAecDivergentFilterFraction,
+                     stats.aec_divergent_filter_fraction);
   }
 }
 
diff --git a/chromium/third_party/webrtc/api/statscollector_unittest.cc b/chromium/third_party/webrtc/api/statscollector_unittest.cc
index 5873e7382e8..7953515072b 100644
--- a/chromium/third_party/webrtc/api/statscollector_unittest.cc
+++ b/chromium/third_party/webrtc/api/statscollector_unittest.cc
@@ -11,6 +11,7 @@
 #include <stdio.h>
 
 #include <algorithm>
+#include <memory>
 
 #include "webrtc/api/statscollector.h"
 
@@ -32,7 +33,6 @@
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/channelmanager.h"
 
-using rtc::scoped_ptr;
 using testing::_;
 using testing::DoAll;
 using testing::Field;
@@ -67,10 +67,15 @@ const uint32_t kSsrcOfTrack = 1234;
 
 class MockWebRtcSession : public webrtc::WebRtcSession {
  public:
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
   explicit MockWebRtcSession(webrtc::MediaControllerInterface* media_controller)
       : WebRtcSession(media_controller,
                       rtc::Thread::Current(),
                       rtc::Thread::Current(),
+                      rtc::Thread::Current(),
                       nullptr) {}
   MOCK_METHOD0(voice_channel, cricket::VoiceChannel*());
   MOCK_METHOD0(video_channel, cricket::VideoChannel*());
@@ -84,9 +89,9 @@ class MockWebRtcSession : public webrtc::WebRtcSession {
                     rtc::scoped_refptr<rtc::RTCCertificate>* certificate));
 
   // Workaround for gmock's inability to cope with move-only return values.
-  rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
-      const std::string& transport_name) override {
-    return rtc::scoped_ptr<rtc::SSLCertificate>(
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
+      const std::string& transport_name) /* override */ {
+    return std::unique_ptr<rtc::SSLCertificate>(
         GetRemoteSSLCertificate_ReturnsRawPointer(transport_name));
   }
   MOCK_METHOD1(GetRemoteSSLCertificate_ReturnsRawPointer,
@@ -161,6 +166,48 @@ class FakeAudioTrack
   rtc::scoped_refptr<FakeAudioProcessor> processor_;
 };
 
+// This fake audio processor is used to verify that the undesired initial values
+// (-1) will be filtered out.
+class FakeAudioProcessorWithInitValue : public webrtc::AudioProcessorInterface {
+ public:
+  FakeAudioProcessorWithInitValue() {}
+  ~FakeAudioProcessorWithInitValue() {}
+
+ private:
+  void GetStats(AudioProcessorInterface::AudioProcessorStats* stats) override {
+    stats->typing_noise_detected = false;
+    stats->echo_return_loss = -100;
+    stats->echo_return_loss_enhancement = -100;
+    stats->echo_delay_median_ms = -1;
+    stats->aec_quality_min = -1.0f;
+    stats->echo_delay_std_ms = -1;
+  }
+};
+
+class FakeAudioTrackWithInitValue
+    : public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface> {
+ public:
+  explicit FakeAudioTrackWithInitValue(const std::string& id)
+      : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(id),
+        processor_(
+            new rtc::RefCountedObject<FakeAudioProcessorWithInitValue>()) {}
+  std::string kind() const override { return "audio"; }
+  webrtc::AudioSourceInterface* GetSource() const override { return NULL; }
+  void AddSink(webrtc::AudioTrackSinkInterface* sink) override {}
+  void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
+  bool GetSignalLevel(int* level) override {
+    *level = 1;
+    return true;
+  }
+  rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
+      override {
+    return processor_;
+  }
+
+ private:
+  rtc::scoped_refptr<FakeAudioProcessorWithInitValue> processor_;
+};
+
 bool GetValue(const StatsReport* report,
               StatsReport::StatsValueName name,
               std::string* value) {
@@ -440,7 +487,8 @@ void InitVoiceSenderInfo(cricket::VoiceSenderInfo* voice_sender_info) {
 }
 
 void UpdateVoiceSenderInfoFromAudioTrack(
-    FakeAudioTrack* audio_track, cricket::VoiceSenderInfo* voice_sender_info) {
+    AudioTrackInterface* audio_track,
+    cricket::VoiceSenderInfo* voice_sender_info) {
   audio_track->GetSignalLevel(&voice_sender_info->audio_level);
   webrtc::AudioProcessorInterface::AudioProcessorStats audio_processor_stats;
   audio_track->GetAudioProcessor()->GetStats(&audio_processor_stats);
@@ -492,12 +540,15 @@ class StatsCollectorForTest : public webrtc::StatsCollector {
 class StatsCollectorTest : public testing::Test {
  protected:
   StatsCollectorTest()
-      : media_engine_(new cricket::FakeMediaEngine()),
-        channel_manager_(
-            new cricket::ChannelManager(media_engine_, rtc::Thread::Current())),
+      : worker_thread_(rtc::Thread::Current()),
+        network_thread_(rtc::Thread::Current()),
+        media_engine_(new cricket::FakeMediaEngine()),
+        channel_manager_(new cricket::ChannelManager(media_engine_,
+                                                     worker_thread_,
+                                                     network_thread_)),
         media_controller_(
             webrtc::MediaControllerInterface::Create(cricket::MediaConfig(),
-                                                     rtc::Thread::Current(),
+                                                     worker_thread_,
                                                      channel_manager_.get())),
         session_(media_controller_.get()) {
     // By default, we ignore session GetStats calls.
@@ -671,7 +722,7 @@ class StatsCollectorTest : public testing::Test {
   void TestCertificateReports(
       const rtc::FakeSSLCertificate& local_cert,
       const std::vector<std::string>& local_ders,
-      rtc::scoped_ptr<rtc::FakeSSLCertificate> remote_cert,
+      std::unique_ptr<rtc::FakeSSLCertificate> remote_cert,
       const std::vector<std::string>& remote_ders) {
     StatsCollectorForTest stats(&pc_);
 
@@ -694,7 +745,7 @@ class StatsCollectorTest : public testing::Test {
 
     // Fake certificate to report
     rtc::scoped_refptr<rtc::RTCCertificate> local_certificate(
-        rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::FakeSSLIdentity>(
+        rtc::RTCCertificate::Create(std::unique_ptr<rtc::FakeSSLIdentity>(
             new rtc::FakeSSLIdentity(local_cert))));
 
     // Configure MockWebRtcSession
@@ -756,9 +807,11 @@ class StatsCollectorTest : public testing::Test {
               srtp_crypto_suite);
   }
 
+  rtc::Thread* const worker_thread_;
+  rtc::Thread* const network_thread_;
   cricket::FakeMediaEngine* media_engine_;
-  rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
-  rtc::scoped_ptr<webrtc::MediaControllerInterface> media_controller_;
+  std::unique_ptr<cricket::ChannelManager> channel_manager_;
+  std::unique_ptr<webrtc::MediaControllerInterface> media_controller_;
   MockWebRtcSession session_;
   MockPeerConnection pc_;
   FakeDataChannelProvider data_channel_provider_;
@@ -769,6 +822,29 @@ class StatsCollectorTest : public testing::Test {
   std::vector<rtc::scoped_refptr<DataChannel>> data_channels_;
 };
 
+TEST_F(StatsCollectorTest, FilterOutNegativeDataChannelId) {
+  const std::string label = "hacks";
+  // The data channel id is from the Config which is -1 initially.
+  const int id = -1;
+  const std::string state = DataChannelInterface::DataStateString(
+      DataChannelInterface::DataState::kConnecting);
+
+  AddDataChannel(cricket::DCT_SCTP, label, id);
+  StatsCollectorForTest stats(&pc_);
+
+  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
+
+  StatsReports reports;
+  stats.GetStats(NULL, &reports);
+
+  const StatsReport* report =
+      FindNthReportByType(reports, StatsReport::kStatsReportTypeDataChannel, 1);
+
+  std::string value_in_report;
+  EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameDataChannelId,
+                        &value_in_report));
+}
+
 // Verify that ExtractDataInfo populates reports.
 TEST_F(StatsCollectorTest, ExtractDataInfo) {
   const std::string label = "hacks";
@@ -824,8 +900,9 @@ TEST_F(StatsCollectorTest, BytesCounterHandles64Bits) {
                             Return(true)));
 
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, kVideoChannelName, false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, kVideoChannelName,
+                                      false);
   StatsReports reports;  // returned values.
   cricket::VideoSenderInfo video_sender_info;
   cricket::VideoMediaInfo stats_read;
@@ -870,8 +947,9 @@ TEST_F(StatsCollectorTest, BandwidthEstimationInfoIsReported) {
                             Return(true)));
 
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, kVideoChannelName, false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, kVideoChannelName,
+                                      false);
 
   StatsReports reports;  // returned values.
   cricket::VideoSenderInfo video_sender_info;
@@ -945,8 +1023,8 @@ TEST_F(StatsCollectorTest, TrackObjectExistsWithoutUpdateStats) {
   StatsCollectorForTest stats(&pc_);
 
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, "video", false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, "video", false);
   AddOutgoingVideoTrackStats();
   stats.AddStream(stream_);
 
@@ -981,8 +1059,9 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
                             Return(true)));
 
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, kVideoChannelName, false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, kVideoChannelName,
+                                      false);
   AddOutgoingVideoTrackStats();
   stats.AddStream(stream_);
 
@@ -1049,8 +1128,8 @@ TEST_F(StatsCollectorTest, TransportObjectLinkedFromSsrcObject) {
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
   // The transport_name known by the video channel.
   const std::string kVcName("vcname");
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, kVcName, false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, kVcName, false);
   AddOutgoingVideoTrackStats();
   stats.AddStream(stream_);
 
@@ -1107,8 +1186,8 @@ TEST_F(StatsCollectorTest, RemoteSsrcInfoIsAbsent) {
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
   // The transport_name known by the video channel.
   const std::string kVcName("vcname");
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, kVcName, false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, kVcName, false);
   AddOutgoingVideoTrackStats();
   stats.AddStream(stream_);
 
@@ -1133,8 +1212,8 @@ TEST_F(StatsCollectorTest, RemoteSsrcInfoIsPresent) {
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
   // The transport_name known by the video channel.
   const std::string kVcName("vcname");
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, kVcName, false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, kVcName, false);
   AddOutgoingVideoTrackStats();
   stats.AddStream(stream_);
 
@@ -1188,8 +1267,9 @@ TEST_F(StatsCollectorTest, ReportsFromRemoteTrack) {
                             Return(true)));
 
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel();
-  cricket::VideoChannel video_channel(rtc::Thread::Current(), media_channel,
-                                      nullptr, kVideoChannelName, false);
+  cricket::VideoChannel video_channel(worker_thread_, network_thread_,
+                                      media_channel, nullptr, kVideoChannelName,
+                                      false);
   AddIncomingVideoTrackStats();
   stats.AddStream(stream_);
 
@@ -1339,7 +1419,7 @@ TEST_F(StatsCollectorTest, ChainedCertificateReportsCreated) {
   remote_ders[1] = "non-";
   remote_ders[2] = "intersecting";
   remote_ders[3] = "set";
-  rtc::scoped_ptr<rtc::FakeSSLCertificate> remote_cert(
+  std::unique_ptr<rtc::FakeSSLCertificate> remote_cert(
       new rtc::FakeSSLCertificate(DersToPems(remote_ders)));
 
   TestCertificateReports(local_cert, local_ders, std::move(remote_cert),
@@ -1355,7 +1435,7 @@ TEST_F(StatsCollectorTest, ChainlessCertificateReportsCreated) {
 
   // Build remote certificate.
   std::string remote_der = "This is somebody else's der.";
-  rtc::scoped_ptr<rtc::FakeSSLCertificate> remote_cert(
+  std::unique_ptr<rtc::FakeSSLCertificate> remote_cert(
       new rtc::FakeSSLCertificate(DerToPem(remote_der)));
 
   TestCertificateReports(local_cert, std::vector<std::string>(1, local_der),
@@ -1445,7 +1525,7 @@ TEST_F(StatsCollectorTest, NoCertificates) {
       transport_stats;
 
   // Fake transport object.
-  rtc::scoped_ptr<cricket::FakeTransport> transport(
+  std::unique_ptr<cricket::FakeTransport> transport(
       new cricket::FakeTransport(transport_stats.transport_name));
 
   // Configure MockWebRtcSession
@@ -1479,7 +1559,7 @@ TEST_F(StatsCollectorTest, UnsupportedDigestIgnored) {
 
   // Build a remote certificate with an unsupported digest algorithm.
   std::string remote_der = "This is somebody else's der.";
-  rtc::scoped_ptr<rtc::FakeSSLCertificate> remote_cert(
+  std::unique_ptr<rtc::FakeSSLCertificate> remote_cert(
       new rtc::FakeSSLCertificate(DerToPem(remote_der)));
   remote_cert->set_digest_algorithm("foobar");
 
@@ -1487,6 +1567,113 @@ TEST_F(StatsCollectorTest, UnsupportedDigestIgnored) {
                          std::move(remote_cert), std::vector<std::string>());
 }
 
+// This test verifies that the audio/video related stats which are -1 initially
+// will be filtered out.
+TEST_F(StatsCollectorTest, FilterOutNegativeInitialValues) {
+  StatsCollectorForTest stats(&pc_);
+
+  EXPECT_CALL(session_, GetLocalCertificate(_, _))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(session_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
+      .WillRepeatedly(Return(nullptr));
+
+  MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
+  // The transport_name known by the voice channel.
+  const std::string kVcName("vcname");
+  cricket::VoiceChannel voice_channel(worker_thread_, network_thread_,
+                                        media_engine_, media_channel, nullptr,
+                                        kVcName, false);
+
+  // Create a local stream with a local audio track and adds it to the stats.
+  if (stream_ == NULL)
+    stream_ = webrtc::MediaStream::Create("streamlabel");
+
+  rtc::scoped_refptr<FakeAudioTrackWithInitValue> local_track(
+      new rtc::RefCountedObject<FakeAudioTrackWithInitValue>(kLocalTrackId));
+  stream_->AddTrack(local_track);
+  EXPECT_CALL(session_, GetLocalTrackIdBySsrc(kSsrcOfTrack, _))
+      .WillOnce(DoAll(SetArgPointee<1>(kLocalTrackId), Return(true)));
+  stats.AddStream(stream_);
+  stats.AddLocalAudioTrack(local_track.get(), kSsrcOfTrack);
+
+  // Create a remote stream with a remote audio track and adds it to the stats.
+  rtc::scoped_refptr<webrtc::MediaStream> remote_stream(
+      webrtc::MediaStream::Create("remotestreamlabel"));
+  rtc::scoped_refptr<FakeAudioTrackWithInitValue> remote_track(
+      new rtc::RefCountedObject<FakeAudioTrackWithInitValue>(kRemoteTrackId));
+  EXPECT_CALL(session_, GetRemoteTrackIdBySsrc(kSsrcOfTrack, _))
+      .WillOnce(DoAll(SetArgPointee<1>(kRemoteTrackId), Return(true)));
+  remote_stream->AddTrack(remote_track);
+  stats.AddStream(remote_stream);
+
+  // Instruct the session to return stats containing the transport channel.
+  InitSessionStats(kVcName);
+  EXPECT_CALL(session_, GetTransportStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_), Return(true)));
+
+  cricket::VoiceSenderInfo voice_sender_info;
+  voice_sender_info.add_ssrc(kSsrcOfTrack);
+  // These values are set to -1 initially in audio_send_stream.
+  // The voice_sender_info will read the values from audio_send_stream.
+  voice_sender_info.rtt_ms = -1;
+  voice_sender_info.packets_lost = -1;
+  voice_sender_info.jitter_ms = -1;
+
+  // Some of the contents in |voice_sender_info| needs to be updated from the
+  // |audio_track_|.
+  UpdateVoiceSenderInfoFromAudioTrack(local_track.get(), &voice_sender_info);
+
+  cricket::VoiceReceiverInfo voice_receiver_info;
+  voice_receiver_info.add_ssrc(kSsrcOfTrack);
+  voice_receiver_info.capture_start_ntp_time_ms = -1;
+  voice_receiver_info.audio_level = -1;
+
+  // Constructs an ssrc stats update.
+  cricket::VoiceMediaInfo stats_read;
+  stats_read.senders.push_back(voice_sender_info);
+  stats_read.receivers.push_back(voice_receiver_info);
+
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(Return(&voice_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(*media_channel, GetStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read), Return(true)));
+
+  StatsReports reports;
+  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
+
+  // Get stats for the local track.
+  stats.GetStats(local_track.get(), &reports);
+  const StatsReport* report =
+      FindNthReportByType(reports, StatsReport::kStatsReportTypeSsrc, 1);
+  EXPECT_TRUE(report);
+  // The -1 will not be added to the stats report.
+  std::string value_in_report;
+  EXPECT_FALSE(
+      GetValue(report, StatsReport::kStatsValueNameRtt, &value_in_report));
+  EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNamePacketsLost,
+                        &value_in_report));
+  EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameJitterReceived,
+                        &value_in_report));
+  EXPECT_FALSE(GetValue(report,
+                        StatsReport::kStatsValueNameEchoCancellationQualityMin,
+                        &value_in_report));
+  EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoDelayMedian,
+                        &value_in_report));
+  EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoDelayStdDev,
+                        &value_in_report));
+
+  // Get stats for the remote track.
+  reports.clear();
+  stats.GetStats(remote_track.get(), &reports);
+  report = FindNthReportByType(reports, StatsReport::kStatsReportTypeSsrc, 1);
+  EXPECT_TRUE(report);
+  EXPECT_FALSE(GetValue(report,
+                        StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
+                        &value_in_report));
+  EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameAudioInputLevel,
+                        &value_in_report));
+}
+
 // This test verifies that a local stats object can get statistics via
 // AudioTrackInterface::GetStats() method.
 TEST_F(StatsCollectorTest, GetStatsFromLocalAudioTrack) {
@@ -1500,8 +1687,9 @@ TEST_F(StatsCollectorTest, GetStatsFromLocalAudioTrack) {
   MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
   // The transport_name known by the voice channel.
   const std::string kVcName("vcname");
-  cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
-                                      media_channel, nullptr, kVcName, false);
+  cricket::VoiceChannel voice_channel(worker_thread_, network_thread_,
+                                      media_engine_, media_channel, nullptr,
+                                      kVcName, false);
   AddOutgoingAudioTrackStats();
   stats.AddStream(stream_);
   stats.AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
@@ -1535,8 +1723,9 @@ TEST_F(StatsCollectorTest, GetStatsFromRemoteStream) {
   MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
   // The transport_name known by the voice channel.
   const std::string kVcName("vcname");
-  cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
-                                      media_channel, nullptr, kVcName, false);
+  cricket::VoiceChannel voice_channel(worker_thread_, network_thread_,
+                                      media_engine_, media_channel, nullptr,
+                                      kVcName, false);
   AddIncomingAudioTrackStats();
   stats.AddStream(stream_);
 
@@ -1564,8 +1753,9 @@ TEST_F(StatsCollectorTest, GetStatsAfterRemoveAudioStream) {
   MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
   // The transport_name known by the voice channel.
   const std::string kVcName("vcname");
-  cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
-                                      media_channel, nullptr, kVcName, false);
+  cricket::VoiceChannel voice_channel(worker_thread_, network_thread_,
+                                      media_engine_, media_channel, nullptr,
+                                      kVcName, false);
   AddOutgoingAudioTrackStats();
   stats.AddStream(stream_);
   stats.AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
@@ -1625,8 +1815,9 @@ TEST_F(StatsCollectorTest, LocalAndRemoteTracksWithSameSsrc) {
   MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
   // The transport_name known by the voice channel.
   const std::string kVcName("vcname");
-  cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
-                                      media_channel, nullptr, kVcName, false);
+  cricket::VoiceChannel voice_channel(worker_thread_, network_thread_,
+                                      media_engine_, media_channel, nullptr,
+                                      kVcName, false);
 
   // Create a local stream with a local audio track and adds it to the stats.
   AddOutgoingAudioTrackStats();
@@ -1712,8 +1903,9 @@ TEST_F(StatsCollectorTest, TwoLocalTracksWithSameSsrc) {
   MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
   // The transport_name known by the voice channel.
   const std::string kVcName("vcname");
-  cricket::VoiceChannel voice_channel(rtc::Thread::Current(), media_engine_,
-                                      media_channel, nullptr, kVcName, false);
+  cricket::VoiceChannel voice_channel(worker_thread_, network_thread_,
+                                      media_engine_, media_channel, nullptr,
+                                      kVcName, false);
 
   // Create a local stream with a local audio track and adds it to the stats.
   AddOutgoingAudioTrackStats();
diff --git a/chromium/third_party/webrtc/api/statstypes.cc b/chromium/third_party/webrtc/api/statstypes.cc
index feeead43745..61af82467a7 100644
--- a/chromium/third_party/webrtc/api/statstypes.cc
+++ b/chromium/third_party/webrtc/api/statstypes.cc
@@ -363,6 +363,8 @@ bool StatsReport::Value::bool_val() const {
 
 const char* StatsReport::Value::display_name() const {
   switch (name) {
+    case kStatsValueNameAecDivergentFilterFraction:
+      return "aecDivergentFilterFraction";
     case kStatsValueNameAudioOutputLevel:
       return "audioOutputLevel";
     case kStatsValueNameAudioInputLevel:
diff --git a/chromium/third_party/webrtc/api/statstypes.h b/chromium/third_party/webrtc/api/statstypes.h
index 9a44724e8fd..4f58b975403 100644
--- a/chromium/third_party/webrtc/api/statstypes.h
+++ b/chromium/third_party/webrtc/api/statstypes.h
@@ -21,9 +21,9 @@
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/linked_ptr.h"
 #include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/thread_checker.h"
@@ -99,6 +99,7 @@ class StatsReport {
 
   enum StatsValueName {
     kStatsValueNameActiveConnection,
+    kStatsValueNameAecDivergentFilterFraction,
     kStatsValueNameAudioInputLevel,
     kStatsValueNameAudioOutputLevel,
     kStatsValueNameBytesReceived,
diff --git a/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.cc b/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.cc
index 0b6478ed2f6..a32ef64d037 100644
--- a/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.cc
+++ b/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.cc
@@ -23,7 +23,7 @@ static const int kHighSampleValue = 10000;
 
 // Same value as src/modules/audio_device/main/source/audio_device_config.h in
 // https://code.google.com/p/webrtc/
-static const uint32_t kAdmMaxIdleTimeProcess = 1000;
+static const int kAdmMaxIdleTimeProcess = 1000;
 
 // Constants here are derived by running VoE using a real ADM.
 // The constants correspond to 10ms of mono audio at 44kHz.
@@ -73,12 +73,12 @@ int FakeAudioCaptureModule::frames_received() const {
 }
 
 int64_t FakeAudioCaptureModule::TimeUntilNextProcess() {
-  const uint32_t current_time = rtc::Time();
+  const int64_t current_time = rtc::TimeMillis();
   if (current_time < last_process_time_ms_) {
     // TODO: wraparound could be handled more gracefully.
     return 0;
   }
-  const uint32_t elapsed_time = current_time - last_process_time_ms_;
+  const int64_t elapsed_time = current_time - last_process_time_ms_;
   if (kAdmMaxIdleTimeProcess < elapsed_time) {
     return 0;
   }
@@ -86,7 +86,7 @@ int64_t FakeAudioCaptureModule::TimeUntilNextProcess() {
 }
 
 void FakeAudioCaptureModule::Process() {
-  last_process_time_ms_ = rtc::Time();
+  last_process_time_ms_ = rtc::TimeMillis();
 }
 
 int32_t FakeAudioCaptureModule::ActiveAudioLayer(
@@ -205,8 +205,7 @@ int32_t FakeAudioCaptureModule::InitRecording() {
 }
 
 bool FakeAudioCaptureModule::RecordingIsInitialized() const {
-  ASSERT(false);
-  return 0;
+  return rec_is_initialized_;
 }
 
 int32_t FakeAudioCaptureModule::StartPlayout() {
@@ -591,7 +590,7 @@ bool FakeAudioCaptureModule::Initialize() {
   // sent to it. Note that the audio processing pipeline will likely distort the
   // original signal.
   SetSendBuffer(kHighSampleValue);
-  last_process_time_ms_ = rtc::Time();
+  last_process_time_ms_ = rtc::TimeMillis();
   return true;
 }
 
@@ -650,7 +649,7 @@ void FakeAudioCaptureModule::StartProcessP() {
 void FakeAudioCaptureModule::ProcessFrameP() {
   ASSERT(process_thread_->IsCurrent());
   if (!started_) {
-    next_frame_time_ = rtc::Time();
+    next_frame_time_ = rtc::TimeMillis();
     started_ = true;
   }
 
@@ -666,8 +665,8 @@ void FakeAudioCaptureModule::ProcessFrameP() {
   }
 
   next_frame_time_ += kTimePerFrameMs;
-  const uint32_t current_time = rtc::Time();
-  const uint32_t wait_time =
+  const int64_t current_time = rtc::TimeMillis();
+  const int64_t wait_time =
       (next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0;
   process_thread_->PostDelayed(wait_time, this, MSG_RUN_PROCESS);
 }
diff --git a/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.h b/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.h
index 9200bdf6f31..f89249ad2f9 100644
--- a/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.h
+++ b/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule.h
@@ -20,10 +20,11 @@
 #ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
 #define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
 
+#include <memory>
+
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagehandler.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
@@ -172,12 +173,12 @@ class FakeAudioCaptureModule
   int32_t ResetAudioDevice() override;
   int32_t SetLoudspeakerStatus(bool enable) override;
   int32_t GetLoudspeakerStatus(bool* enabled) const override;
-  virtual bool BuiltInAECIsAvailable() const { return false; }
-  virtual int32_t EnableBuiltInAEC(bool enable) { return -1; }
-  virtual bool BuiltInAGCIsAvailable() const { return false; }
-  virtual int32_t EnableBuiltInAGC(bool enable) { return -1; }
-  virtual bool BuiltInNSIsAvailable() const { return false; }
-  virtual int32_t EnableBuiltInNS(bool enable) { return -1; }
+  bool BuiltInAECIsAvailable() const override { return false; }
+  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
+  bool BuiltInAGCIsAvailable() const override { return false; }
+  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
+  bool BuiltInNSIsAvailable() const override { return false; }
+  int32_t EnableBuiltInNS(bool enable) override { return -1; }
   // End of functions inherited from webrtc::AudioDeviceModule.
 
   // The following function is inherited from rtc::MessageHandler.
@@ -225,7 +226,7 @@ class FakeAudioCaptureModule
 
   // The time in milliseconds when Process() was last called or 0 if no call
   // has been made.
-  uint32_t last_process_time_ms_;
+  int64_t last_process_time_ms_;
 
   // Callback for playout and recording.
   webrtc::AudioTransport* audio_callback_;
@@ -245,9 +246,9 @@ class FakeAudioCaptureModule
   // wall clock time the next frame should be generated and received. started_
   // ensures that next_frame_time_ can be initialized properly on first call.
   bool started_;
-  uint32_t next_frame_time_;
+  int64_t next_frame_time_;
 
-  rtc::scoped_ptr<rtc::Thread> process_thread_;
+  std::unique_ptr<rtc::Thread> process_thread_;
 
   // Buffer for storing samples received from the webrtc::AudioTransport.
   char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
diff --git a/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule_unittest.cc b/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule_unittest.cc
index 8ac1acce2b5..d0dcd85012b 100644
--- a/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule_unittest.cc
+++ b/chromium/third_party/webrtc/api/test/fakeaudiocapturemodule_unittest.cc
@@ -31,7 +31,7 @@ class FakeAdmTest : public testing::Test,
     memset(rec_buffer_, 0, sizeof(rec_buffer_));
   }
 
-  virtual void SetUp() {
+  void SetUp() override {
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
     EXPECT_TRUE(fake_audio_capture_module_.get() != NULL);
   }
diff --git a/chromium/third_party/webrtc/api/test/fakedtlsidentitystore.h b/chromium/third_party/webrtc/api/test/fakedtlsidentitystore.h
index 8bbffbf93e9..58de38fb326 100644
--- a/chromium/third_party/webrtc/api/test/fakedtlsidentitystore.h
+++ b/chromium/third_party/webrtc/api/test/fakedtlsidentitystore.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_API_TEST_FAKEDTLSIDENTITYSERVICE_H_
 #define WEBRTC_API_TEST_FAKEDTLSIDENTITYSERVICE_H_
 
+#include <memory>
 #include <string>
 #include <utility>
 
@@ -18,67 +19,104 @@
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/base/rtccertificate.h"
 
-static const struct {
-  const char* rsa_private_key_pem;
-  const char* cert_pem;
-} kKeysAndCerts[] = {
-    {"-----BEGIN RSA PRIVATE KEY-----\n"
-     "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
-     "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
-     "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
-     "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n"
-     "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n"
-     "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n"
-     "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n"
-     "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n"
-     "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n"
-     "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n"
-     "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n"
-     "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n"
-     "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n"
-     "UCXiYxSsu20QNVw=\n"
-     "-----END RSA PRIVATE KEY-----\n",
-     "-----BEGIN CERTIFICATE-----\n"
-     "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n"
-     "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n"
-     "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n"
-     "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
-     "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
-     "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n"
-     "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n"
-     "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
-     "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
-     "-----END CERTIFICATE-----\n"},
-    {"-----BEGIN RSA PRIVATE KEY-----\n"
-     "MIICXQIBAAKBgQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgj\n"
-     "Bl8CPZMvDh9EwB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQ\n"
-     "sOR/qPvviJx5I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQAB\n"
-     "AoGARni9eY8/hv+SX+I+05EdXt6MQXNUbQ+cSykBNCfVccLzIFEWUQMT2IHqwl6X\n"
-     "ShIXcq7/n1QzOAEiuzixauM3YHg4xZ1Um2Ha9a7ig5Xg4v6b43bmMkNE6LkoAtYs\n"
-     "qnQdfMh442b1liDud6IMb1Qk0amt3fSrgRMc547TZQVx4QECQQDxUeDm94r3p4ng\n"
-     "5rCLLC1K5/6HSTZsh7jatKPlz7GfP/IZlYV7iE5784/n0wRiCjZOS7hQRy/8m2Gp\n"
-     "pf4aZq+DAkEA6+np4d36FYikydvUrupLT3FkdRHGn/v83qOll/VmeNh+L1xMZlIP\n"
-     "tM26hAXCcQb7O5+J9y3cx2CAQsBS11ZXZQJAfGgTo76WG9p5UEJdXUInD2jOZPwv\n"
-     "XIATolxh6kXKcijLLLlSmT7KB0inNYIpzkkpee+7U1d/u6B3FriGaSHq9QJBAM/J\n"
-     "ICnDdLCgwNvWVraVQC3BpwSB2pswvCFwq7py94V60XFvbw80Ogc6qIv98qvQxVlX\n"
-     "hJIEgA/PjEi+0ng94Q0CQQDm8XSDby35gmjO+6eRmJtAjtB7nguLvrPXM6CPXRmD\n"
-     "sRoBocpHw6j9UdzZ6qYG0FkdXZghezXFY58ro2BYYRR3\n"
-     "-----END RSA PRIVATE KEY-----\n",
-     "-----BEGIN CERTIFICATE-----\n"
-     "MIICWDCCAcGgAwIBAgIJALgDjxMbBOhbMA0GCSqGSIb3DQEBCwUAMEUxCzAJBgNV\n"
-     "BAYTAkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRlcm5ldCBX\n"
-     "aWRnaXRzIFB0eSBMdGQwHhcNMTUxMTEzMjIzMjEzWhcNMTYxMTEyMjIzMjEzWjBF\n"
-     "MQswCQYDVQQGEwJBVTETMBEGA1UECAwKU29tZS1TdGF0ZTEhMB8GA1UECgwYSW50\n"
-     "ZXJuZXQgV2lkZ2l0cyBQdHkgTHRkMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKB\n"
-     "gQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgjBl8CPZMvDh9E\n"
-     "wB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQsOR/qPvviJx5\n"
-     "I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQABo1AwTjAdBgNV\n"
-     "HQ4EFgQUx2tbJdlcSTCepn09UdYORXKuSTAwHwYDVR0jBBgwFoAUx2tbJdlcSTCe\n"
-     "pn09UdYORXKuSTAwDAYDVR0TBAUwAwEB/zANBgkqhkiG9w0BAQsFAAOBgQAmp9Id\n"
-     "E716gHMqeBG4S2FCgVFCr0a0ugkaneQAN/c2L9CbMemEN9W6jvucUIVOtYd90dDW\n"
-     "lXuowWmT/JctPe3D2qt4yvYW3puECHk2tVQmrJOZiZiTRtWm6HxkmoUYHYp/DtaS\n"
-     "1Xe29gSTnZtI5sQCrGMzk3SGRSSs7ejLKiVDBQ==\n"
-     "-----END CERTIFICATE-----\n"}};
+// RSA with mod size 1024, pub exp 0x10001.
+static const rtc::RTCCertificatePEM kRsaPems[] = {
+    rtc::RTCCertificatePEM(
+        "-----BEGIN RSA PRIVATE KEY-----\n"
+         "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
+         "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
+         "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
+         "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n"
+         "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n"
+         "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n"
+         "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n"
+         "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n"
+         "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n"
+         "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n"
+         "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n"
+         "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n"
+         "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n"
+         "UCXiYxSsu20QNVw=\n"
+         "-----END RSA PRIVATE KEY-----\n",
+         "-----BEGIN CERTIFICATE-----\n"
+         "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n"
+         "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n"
+         "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n"
+         "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
+         "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
+         "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n"
+         "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n"
+         "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
+         "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
+         "-----END CERTIFICATE-----\n"),
+    rtc::RTCCertificatePEM(
+         "-----BEGIN RSA PRIVATE KEY-----\n"
+         "MIICXQIBAAKBgQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgj\n"
+         "Bl8CPZMvDh9EwB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQ\n"
+         "sOR/qPvviJx5I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQAB\n"
+         "AoGARni9eY8/hv+SX+I+05EdXt6MQXNUbQ+cSykBNCfVccLzIFEWUQMT2IHqwl6X\n"
+         "ShIXcq7/n1QzOAEiuzixauM3YHg4xZ1Um2Ha9a7ig5Xg4v6b43bmMkNE6LkoAtYs\n"
+         "qnQdfMh442b1liDud6IMb1Qk0amt3fSrgRMc547TZQVx4QECQQDxUeDm94r3p4ng\n"
+         "5rCLLC1K5/6HSTZsh7jatKPlz7GfP/IZlYV7iE5784/n0wRiCjZOS7hQRy/8m2Gp\n"
+         "pf4aZq+DAkEA6+np4d36FYikydvUrupLT3FkdRHGn/v83qOll/VmeNh+L1xMZlIP\n"
+         "tM26hAXCcQb7O5+J9y3cx2CAQsBS11ZXZQJAfGgTo76WG9p5UEJdXUInD2jOZPwv\n"
+         "XIATolxh6kXKcijLLLlSmT7KB0inNYIpzkkpee+7U1d/u6B3FriGaSHq9QJBAM/J\n"
+         "ICnDdLCgwNvWVraVQC3BpwSB2pswvCFwq7py94V60XFvbw80Ogc6qIv98qvQxVlX\n"
+         "hJIEgA/PjEi+0ng94Q0CQQDm8XSDby35gmjO+6eRmJtAjtB7nguLvrPXM6CPXRmD\n"
+         "sRoBocpHw6j9UdzZ6qYG0FkdXZghezXFY58ro2BYYRR3\n"
+         "-----END RSA PRIVATE KEY-----\n",
+         "-----BEGIN CERTIFICATE-----\n"
+         "MIICWDCCAcGgAwIBAgIJALgDjxMbBOhbMA0GCSqGSIb3DQEBCwUAMEUxCzAJBgNV\n"
+         "BAYTAkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRlcm5ldCBX\n"
+         "aWRnaXRzIFB0eSBMdGQwHhcNMTUxMTEzMjIzMjEzWhcNMTYxMTEyMjIzMjEzWjBF\n"
+         "MQswCQYDVQQGEwJBVTETMBEGA1UECAwKU29tZS1TdGF0ZTEhMB8GA1UECgwYSW50\n"
+         "ZXJuZXQgV2lkZ2l0cyBQdHkgTHRkMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKB\n"
+         "gQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgjBl8CPZMvDh9E\n"
+         "wB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQsOR/qPvviJx5\n"
+         "I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQABo1AwTjAdBgNV\n"
+         "HQ4EFgQUx2tbJdlcSTCepn09UdYORXKuSTAwHwYDVR0jBBgwFoAUx2tbJdlcSTCe\n"
+         "pn09UdYORXKuSTAwDAYDVR0TBAUwAwEB/zANBgkqhkiG9w0BAQsFAAOBgQAmp9Id\n"
+         "E716gHMqeBG4S2FCgVFCr0a0ugkaneQAN/c2L9CbMemEN9W6jvucUIVOtYd90dDW\n"
+         "lXuowWmT/JctPe3D2qt4yvYW3puECHk2tVQmrJOZiZiTRtWm6HxkmoUYHYp/DtaS\n"
+         "1Xe29gSTnZtI5sQCrGMzk3SGRSSs7ejLKiVDBQ==\n"
+         "-----END CERTIFICATE-----\n")
+};
+
+// ECDSA with EC_NIST_P256.
+// These PEM strings were created by generating an identity with
+// |SSLIdentity::Generate| and invoking |identity->PrivateKeyToPEMString()|,
+// |identity->PublicKeyToPEMString()| and
+// |identity->certificate().ToPEMString()|.
+static const rtc::RTCCertificatePEM kEcdsaPems[] = {
+    rtc::RTCCertificatePEM(
+        "-----BEGIN PRIVATE KEY-----\n"
+        "MIGHAgEAMBMGByqGSM49AgEGCCqGSM49AwEHBG0wawIBAQQg+qaRsR5uHtqG689M\n"
+        "A3PHSJNeVpyi5wUKCft62h0UWy+hRANCAAS5Mjc85q9fVq4ln+zOPlaEC/Rzj5Pb\n"
+        "MVZtf1x/8k2KsbmyZoAMDX2yer/atEuXmItMe3yd6/DXnvboU//D3Lyt\n"
+        "-----END PRIVATE KEY-----\n",
+        "-----BEGIN CERTIFICATE-----\n"
+        "MIIBFTCBu6ADAgECAgkA30tGY5XG7oowCgYIKoZIzj0EAwIwEDEOMAwGA1UEAwwF\n"
+        "dGVzdDMwHhcNMTYwNTA5MDkxODA4WhcNMTYwNjA5MDkxODA4WjAQMQ4wDAYDVQQD\n"
+        "DAV0ZXN0MzBZMBMGByqGSM49AgEGCCqGSM49AwEHA0IABLkyNzzmr19WriWf7M4+\n"
+        "VoQL9HOPk9sxVm1/XH/yTYqxubJmgAwNfbJ6v9q0S5eYi0x7fJ3r8Nee9uhT/8Pc\n"
+        "vK0wCgYIKoZIzj0EAwIDSQAwRgIhAIIc3+CqfkZ9lLwTj1PvUtt3KhnqF2kD0War\n"
+        "cCoTBbCxAiEAyp9Cn4vo2ZBhRIVDKyoxmwak8Z0PAVhJAQaWCgoY2D4=\n"
+        "-----END CERTIFICATE-----\n"),
+    rtc::RTCCertificatePEM(
+        "-----BEGIN PRIVATE KEY-----\n"
+        "MIGHAgEAMBMGByqGSM49AgEGCCqGSM49AwEHBG0wawIBAQQghL/G4JRYnuDNbQuh\n"
+        "LqkytcE39Alsq6FItDVFgOesfCmhRANCAATd53FjPLyVUcwYguEPbSJM03fP6Rx5\n"
+        "GY1dEZ00+ZykjJI83VfDAyvmpRuGahNtBH0hc+7xkDCbeo6TM0tN35xr\n"
+        "-----END PRIVATE KEY-----\n",
+        "-----BEGIN CERTIFICATE-----\n"
+        "MIIBFDCBu6ADAgECAgkArZYdXMyJ5rswCgYIKoZIzj0EAwIwEDEOMAwGA1UEAwwF\n"
+        "dGVzdDQwHhcNMTYwNTA5MDkxODA4WhcNMTYwNjA5MDkxODA4WjAQMQ4wDAYDVQQD\n"
+        "DAV0ZXN0NDBZMBMGByqGSM49AgEGCCqGSM49AwEHA0IABN3ncWM8vJVRzBiC4Q9t\n"
+        "IkzTd8/pHHkZjV0RnTT5nKSMkjzdV8MDK+alG4ZqE20EfSFz7vGQMJt6jpMzS03f\n"
+        "nGswCgYIKoZIzj0EAwIDSAAwRQIgb/LBc8OtsC5lEDyjCP6M9xt5mwzUNrQBOFWZ\n"
+        "1fE/g68CIQD7uoFfbiq6dTp8ZwzbwQ8jJf08KjriamqA9OW/4268Dw==\n"
+        "-----END CERTIFICATE-----\n")
+};
 
 class FakeDtlsIdentityStore : public webrtc::DtlsIdentityStoreInterface,
                               public rtc::MessageHandler {
@@ -100,62 +138,85 @@ class FakeDtlsIdentityStore : public webrtc::DtlsIdentityStoreInterface,
       const rtc::Optional<uint64_t>& expires_ms,
       const rtc::scoped_refptr<webrtc::DtlsIdentityRequestObserver>&
           observer) override {
-    // TODO(hbos): Should be able to generate KT_ECDSA too.
-    RTC_DCHECK((key_params.type() == rtc::KT_RSA &&
-                key_params.rsa_params().mod_size == 1024 &&
-                key_params.rsa_params().pub_exp == 0x10001 &&
-                !expires_ms) ||
-               should_fail_);
+    // The certificates are created from constant PEM strings and use its coded
+    // expiration time, we do not support modifying it.
+    RTC_DCHECK(!expires_ms);
     MessageData* msg = new MessageData(
         rtc::scoped_refptr<webrtc::DtlsIdentityRequestObserver>(observer));
-    rtc::Thread::Current()->Post(
-        this, should_fail_ ? MSG_FAILURE : MSG_SUCCESS, msg);
+    uint32_t msg_id;
+    // Only supports RSA-1024-0x10001 and ECDSA-P256.
+    if (should_fail_) {
+      msg_id = MSG_FAILURE;
+    } else if (key_params.type() == rtc::KT_RSA) {
+      RTC_DCHECK(key_params.rsa_params().mod_size == 1024 &&
+                 key_params.rsa_params().pub_exp == 0x10001);
+      msg_id = MSG_SUCCESS_RSA;
+    } else {
+      RTC_DCHECK(key_params.type() == rtc::KT_ECDSA &&
+                 key_params.ec_curve() == rtc::EC_NIST_P256);
+      msg_id = MSG_SUCCESS_ECDSA;
+    }
+    rtc::Thread::Current()->Post(this, msg_id, msg);
   }
 
   static rtc::scoped_refptr<rtc::RTCCertificate> GenerateCertificate() {
-    std::string cert;
-    std::string key;
-    rtc::SSLIdentity::PemToDer("CERTIFICATE", kKeysAndCerts[0].cert_pem, &cert);
-    rtc::SSLIdentity::PemToDer("RSA PRIVATE KEY",
-                               kKeysAndCerts[0].rsa_private_key_pem, &key);
-
-    std::string pem_cert = rtc::SSLIdentity::DerToPem(
-        rtc::kPemTypeCertificate,
-        reinterpret_cast<const unsigned char*>(cert.data()),
-        cert.length());
-    std::string pem_key = rtc::SSLIdentity::DerToPem(
-        rtc::kPemTypeRsaPrivateKey,
-        reinterpret_cast<const unsigned char*>(key.data()),
-        key.length());
-    rtc::scoped_ptr<rtc::SSLIdentity> identity(
-        rtc::SSLIdentity::FromPEMStrings(pem_key, pem_cert));
-
+    std::unique_ptr<rtc::SSLIdentity> identity;
+    switch (rtc::KT_DEFAULT) {
+    case rtc::KT_RSA:
+      identity.reset(
+          rtc::SSLIdentity::FromPEMStrings(kRsaPems[0].private_key(),
+                                           kRsaPems[0].certificate()));
+      break;
+    case rtc::KT_ECDSA:
+      identity.reset(
+          rtc::SSLIdentity::FromPEMStrings(kEcdsaPems[0].private_key(),
+                                           kEcdsaPems[0].certificate()));
+      break;
+    default:
+      RTC_NOTREACHED();
+    }
     return rtc::RTCCertificate::Create(std::move(identity));
   }
 
  private:
   enum {
-    MSG_SUCCESS,
+    MSG_SUCCESS_RSA,
+    MSG_SUCCESS_ECDSA,
     MSG_FAILURE,
   };
 
-  const char* get_key() {
-    return kKeysAndCerts[key_index_].rsa_private_key_pem;
+  const rtc::RTCCertificatePEM& get_pem(const rtc::KeyType& key_type) const {
+    switch (key_type) {
+    case rtc::KT_RSA:
+      return kRsaPems[key_index_];
+    case rtc::KT_ECDSA:
+      return kEcdsaPems[key_index_];
+    default:
+      RTC_NOTREACHED();
+      return kEcdsaPems[key_index_];
+    }
+  }
+  const std::string& get_key(const rtc::KeyType& key_type) const {
+    return get_pem(key_type).private_key();
+  }
+  const std::string& get_cert(const rtc::KeyType& key_type) const {
+    return get_pem(key_type).certificate();
   }
-  const char* get_cert() { return kKeysAndCerts[key_index_].cert_pem; }
 
   // rtc::MessageHandler implementation.
-  void OnMessage(rtc::Message* msg) {
+  void OnMessage(rtc::Message* msg) override {
     MessageData* message_data = static_cast<MessageData*>(msg->pdata);
     rtc::scoped_refptr<webrtc::DtlsIdentityRequestObserver> observer =
         message_data->data();
     switch (msg->message_id) {
-      case MSG_SUCCESS: {
-        std::string cert;
-        std::string key;
-        rtc::SSLIdentity::PemToDer("CERTIFICATE", get_cert(), &cert);
-        rtc::SSLIdentity::PemToDer("RSA PRIVATE KEY", get_key(), &key);
-        observer->OnSuccess(cert, key);
+      case MSG_SUCCESS_RSA:
+      case MSG_SUCCESS_ECDSA: {
+        rtc::KeyType key_type =
+            msg->message_id == MSG_SUCCESS_RSA ? rtc::KT_RSA : rtc::KT_ECDSA;
+        std::unique_ptr<rtc::SSLIdentity> identity(
+            rtc::SSLIdentity::FromPEMStrings(get_key(key_type),
+                                             get_cert(key_type)));
+        observer->OnSuccess(std::move(identity));
         break;
       }
       case MSG_FAILURE:
diff --git a/chromium/third_party/webrtc/api/test/fakevideotrackrenderer.h b/chromium/third_party/webrtc/api/test/fakevideotrackrenderer.h
index 3bd3d0b0005..0d9b2488ab5 100644
--- a/chromium/third_party/webrtc/api/test/fakevideotrackrenderer.h
+++ b/chromium/third_party/webrtc/api/test/fakevideotrackrenderer.h
@@ -16,8 +16,7 @@
 
 namespace webrtc {
 
-class FakeVideoTrackRenderer
-    : public rtc::VideoSinkInterface<cricket::VideoFrame> {
+class FakeVideoTrackRenderer : public cricket::FakeVideoRenderer {
  public:
   FakeVideoTrackRenderer(VideoTrackInterface* video_track)
       : video_track_(video_track) {
@@ -25,22 +24,7 @@ class FakeVideoTrackRenderer
   }
   ~FakeVideoTrackRenderer() { video_track_->RemoveSink(this); }
 
-  virtual void OnFrame(const cricket::VideoFrame& video_frame) override {
-    fake_renderer_.OnFrame(video_frame);
-  }
-
-  int errors() const { return fake_renderer_.errors(); }
-  int width() const { return fake_renderer_.width(); }
-  int height() const { return fake_renderer_.height(); }
-  webrtc::VideoRotation rotation() const { return fake_renderer_.rotation(); }
-  bool black_frame() const { return fake_renderer_.black_frame(); }
-
-  int num_rendered_frames() const {
-    return fake_renderer_.num_rendered_frames();
-  }
-
  private:
-  cricket::FakeVideoRenderer fake_renderer_;
   rtc::scoped_refptr<VideoTrackInterface> video_track_;
 };
 
diff --git a/chromium/third_party/webrtc/api/test/fakevideotracksource.h b/chromium/third_party/webrtc/api/test/fakevideotracksource.h
index 0b70a56b4e4..1cb264b736d 100644
--- a/chromium/third_party/webrtc/api/test/fakevideotracksource.h
+++ b/chromium/third_party/webrtc/api/test/fakevideotracksource.h
@@ -30,7 +30,6 @@ class FakeVideoTrackSource : public VideoTrackSource {
  protected:
   FakeVideoTrackSource()
       : VideoTrackSource(&fake_video_capturer_,
-                         rtc::Thread::Current(),
                          false /* remote */) {}
   virtual ~FakeVideoTrackSource() {}
 
diff --git a/chromium/third_party/webrtc/api/test/mockpeerconnectionobservers.h b/chromium/third_party/webrtc/api/test/mockpeerconnectionobservers.h
index bd593c2ab52..39a8f0134d6 100644
--- a/chromium/third_party/webrtc/api/test/mockpeerconnectionobservers.h
+++ b/chromium/third_party/webrtc/api/test/mockpeerconnectionobservers.h
@@ -13,6 +13,7 @@
 #ifndef WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
 #define WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/api/datachannelinterface.h"
@@ -44,7 +45,7 @@ class MockCreateSessionDescriptionObserver
  private:
   bool called_;
   bool result_;
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_;
+  std::unique_ptr<SessionDescriptionInterface> desc_;
 };
 
 class MockSetSessionDescriptionObserver
diff --git a/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.cc b/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.cc
index 2ff9b5d0092..ed8e031afc3 100644
--- a/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.cc
+++ b/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.cc
@@ -15,7 +15,7 @@
 #include "webrtc/api/test/mockpeerconnectionobservers.h"
 #include "webrtc/api/test/peerconnectiontestwrapper.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
+#include "webrtc/p2p/base/fakeportallocator.h"
 
 static const char kStreamLabelBase[] = "stream_label";
 static const char kVideoTrackLabelBase[] = "video_track";
@@ -47,15 +47,20 @@ void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
       caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
 }
 
-PeerConnectionTestWrapper::PeerConnectionTestWrapper(const std::string& name)
-    : name_(name) {}
+PeerConnectionTestWrapper::PeerConnectionTestWrapper(
+    const std::string& name,
+    rtc::Thread* network_thread,
+    rtc::Thread* worker_thread)
+    : name_(name),
+      network_thread_(network_thread),
+      worker_thread_(worker_thread) {}
 
 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
 
 bool PeerConnectionTestWrapper::CreatePc(
   const MediaConstraintsInterface* constraints) {
-  rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
-      new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
+  std::unique_ptr<cricket::PortAllocator> port_allocator(
+      new cricket::FakePortAllocator(network_thread_, nullptr));
 
   fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
   if (fake_audio_capture_module_ == NULL) {
@@ -63,7 +68,7 @@ bool PeerConnectionTestWrapper::CreatePc(
   }
 
   peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
-      rtc::Thread::Current(), rtc::Thread::Current(),
+      network_thread_, worker_thread_, rtc::Thread::Current(),
       fake_audio_capture_module_, NULL, NULL);
   if (!peer_connection_factory_) {
     return false;
@@ -74,9 +79,9 @@ bool PeerConnectionTestWrapper::CreatePc(
   webrtc::PeerConnectionInterface::IceServer ice_server;
   ice_server.uri = "stun:stun.l.google.com:19302";
   config.servers.push_back(ice_server);
-  rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
-      rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
-      new FakeDtlsIdentityStore() : nullptr);
+  std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
+      rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
+                                            : nullptr);
   peer_connection_ = peer_connection_factory_->CreatePeerConnection(
       config, constraints, std::move(port_allocator),
       std::move(dtls_identity_store), this);
@@ -117,7 +122,7 @@ void PeerConnectionTestWrapper::OnDataChannel(
 
 void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
   // This callback should take the ownership of |desc|.
-  rtc::scoped_ptr<SessionDescriptionInterface> owned_desc(desc);
+  std::unique_ptr<SessionDescriptionInterface> owned_desc(desc);
   std::string sdp;
   EXPECT_TRUE(desc->ToString(&sdp));
 
@@ -182,7 +187,7 @@ void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type,
 void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
                                                 int sdp_mline_index,
                                                 const std::string& candidate) {
-  rtc::scoped_ptr<webrtc::IceCandidateInterface> owned_candidate(
+  std::unique_ptr<webrtc::IceCandidateInterface> owned_candidate(
       webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
   EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
 }
diff --git a/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.h b/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.h
index 25510b2de07..3272366c762 100644
--- a/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.h
+++ b/chromium/third_party/webrtc/api/test/peerconnectiontestwrapper.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
 #define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
 
+#include <memory>
+
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/test/fakeaudiocapturemodule.h"
 #include "webrtc/api/test/fakeconstraints.h"
@@ -25,7 +27,9 @@ class PeerConnectionTestWrapper
   static void Connect(PeerConnectionTestWrapper* caller,
                       PeerConnectionTestWrapper* callee);
 
-  explicit PeerConnectionTestWrapper(const std::string& name);
+  PeerConnectionTestWrapper(const std::string& name,
+                            rtc::Thread* network_thread,
+                            rtc::Thread* worker_thread);
   virtual ~PeerConnectionTestWrapper();
 
   bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
@@ -88,11 +92,13 @@ class PeerConnectionTestWrapper
       bool video, const webrtc::FakeConstraints& video_constraints);
 
   std::string name_;
+  rtc::Thread* const network_thread_;
+  rtc::Thread* const worker_thread_;
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
       peer_connection_factory_;
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
-  rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
+  std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
 };
 
 #endif  // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
diff --git a/chromium/third_party/webrtc/api/videocapturertracksource.cc b/chromium/third_party/webrtc/api/videocapturertracksource.cc
index cb539614be7..b99a2d1edaf 100644
--- a/chromium/third_party/webrtc/api/videocapturertracksource.cc
+++ b/chromium/third_party/webrtc/api/videocapturertracksource.cc
@@ -286,8 +286,9 @@ VideoCapturerTrackSource::VideoCapturerTrackSource(
     rtc::Thread* worker_thread,
     cricket::VideoCapturer* capturer,
     bool remote)
-    : VideoTrackSource(capturer, worker_thread, remote),
+    : VideoTrackSource(capturer, remote),
       signaling_thread_(rtc::Thread::Current()),
+      worker_thread_(worker_thread),
       video_capturer_(capturer),
       started_(false) {
   video_capturer_->SignalStateChange.connect(
@@ -350,7 +351,7 @@ void VideoCapturerTrackSource::Initialize(
 
   format_ = GetBestCaptureFormat(formats);
   // Start the camera with our best guess.
-  if (!worker_thread()->Invoke<bool>(
+  if (!worker_thread_->Invoke<bool>(
           rtc::Bind(&cricket::VideoCapturer::StartCapturing,
                     video_capturer_.get(), format_))) {
     SetState(kEnded);
@@ -370,7 +371,7 @@ void VideoCapturerTrackSource::Stop() {
     return;
   }
   started_ = false;
-  worker_thread()->Invoke<void>(
+  worker_thread_->Invoke<void>(
       rtc::Bind(&cricket::VideoCapturer::Stop, video_capturer_.get()));
 }
 
@@ -378,7 +379,7 @@ void VideoCapturerTrackSource::Restart() {
   if (started_) {
     return;
   }
-  if (!worker_thread()->Invoke<bool>(
+  if (!worker_thread_->Invoke<bool>(
           rtc::Bind(&cricket::VideoCapturer::StartCapturing,
                     video_capturer_.get(), format_))) {
     SetState(kEnded);
diff --git a/chromium/third_party/webrtc/api/videocapturertracksource.h b/chromium/third_party/webrtc/api/videocapturertracksource.h
index 0d1142debad..92f00dc4e5a 100644
--- a/chromium/third_party/webrtc/api/videocapturertracksource.h
+++ b/chromium/third_party/webrtc/api/videocapturertracksource.h
@@ -11,10 +11,11 @@
 #ifndef WEBRTC_API_VIDEOCAPTURERTRACKSOURCE_H_
 #define WEBRTC_API_VIDEOCAPTURERTRACKSOURCE_H_
 
+#include <memory>
+
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/videotracksource.h"
 #include "webrtc/base/asyncinvoker.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/base/videocommon.h"
@@ -47,10 +48,6 @@ class VideoCapturerTrackSource : public VideoTrackSource,
       cricket::VideoCapturer* capturer,
       bool remote);
 
-  cricket::VideoCapturer* GetVideoCapturer() override {
-    return video_capturer_.get();
-  }
-
   bool is_screencast() const override {
     return video_capturer_->IsScreencast();
   }
@@ -75,8 +72,9 @@ class VideoCapturerTrackSource : public VideoTrackSource,
                      cricket::CaptureState capture_state);
 
   rtc::Thread* signaling_thread_;
+  rtc::Thread* worker_thread_;
   rtc::AsyncInvoker invoker_;
-  rtc::scoped_ptr<cricket::VideoCapturer> video_capturer_;
+  std::unique_ptr<cricket::VideoCapturer> video_capturer_;
   bool started_;
   cricket::VideoFormat format_;
   rtc::Optional<bool> needs_denoising_;
diff --git a/chromium/third_party/webrtc/api/videocapturertracksource_unittest.cc b/chromium/third_party/webrtc/api/videocapturertracksource_unittest.cc
index 39252934ebf..90d2cd2563d 100644
--- a/chromium/third_party/webrtc/api/videocapturertracksource_unittest.cc
+++ b/chromium/third_party/webrtc/api/videocapturertracksource_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -109,7 +110,7 @@ class VideoCapturerTrackSourceTest : public testing::Test {
  protected:
   VideoCapturerTrackSourceTest() { InitCapturer(false); }
   void InitCapturer(bool is_screencast) {
-    capturer_cleanup_ = rtc::scoped_ptr<TestVideoCapturer>(
+    capturer_cleanup_ = std::unique_ptr<TestVideoCapturer>(
         new TestVideoCapturer(is_screencast));
     capturer_ = capturer_cleanup_.get();
   }
@@ -126,17 +127,16 @@ class VideoCapturerTrackSourceTest : public testing::Test {
                                                constraints, false);
 
     ASSERT_TRUE(source_.get() != NULL);
-    EXPECT_EQ(capturer_, source_->GetVideoCapturer());
 
     state_observer_.reset(new StateObserver(source_));
     source_->RegisterObserver(state_observer_.get());
     source_->AddOrUpdateSink(&renderer_, rtc::VideoSinkWants());
   }
 
-  rtc::scoped_ptr<TestVideoCapturer> capturer_cleanup_;
+  std::unique_ptr<TestVideoCapturer> capturer_cleanup_;
   TestVideoCapturer* capturer_;
   cricket::FakeVideoRenderer renderer_;
-  rtc::scoped_ptr<StateObserver> state_observer_;
+  std::unique_ptr<StateObserver> state_observer_;
   rtc::scoped_refptr<VideoTrackSourceInterface> source_;
 };
 
diff --git a/chromium/third_party/webrtc/api/videosourceproxy.h b/chromium/third_party/webrtc/api/videosourceproxy.h
index f43c0db69f2..6d4dfcb8053 100644
--- a/chromium/third_party/webrtc/api/videosourceproxy.h
+++ b/chromium/third_party/webrtc/api/videosourceproxy.h
@@ -21,21 +21,22 @@ namespace webrtc {
 // destroyed on the signaling thread and marshals all method calls to the
 // signaling thread.
 BEGIN_PROXY_MAP(VideoTrackSource)
-PROXY_CONSTMETHOD0(SourceState, state)
-PROXY_CONSTMETHOD0(bool, remote)
-PROXY_METHOD0(cricket::VideoCapturer*, GetVideoCapturer)
-PROXY_METHOD0(void, Stop)
-PROXY_METHOD0(void, Restart)
-PROXY_CONSTMETHOD0(bool, is_screencast)
-PROXY_CONSTMETHOD0(rtc::Optional<bool>, needs_denoising)
-PROXY_METHOD1(bool, GetStats, Stats*)
-PROXY_METHOD2(void,
-              AddOrUpdateSink,
-              rtc::VideoSinkInterface<cricket::VideoFrame>*,
-              const rtc::VideoSinkWants&)
-PROXY_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<cricket::VideoFrame>*)
-PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
-PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
+  PROXY_CONSTMETHOD0(SourceState, state)
+  PROXY_CONSTMETHOD0(bool, remote)
+  PROXY_METHOD0(void, Stop)
+  PROXY_METHOD0(void, Restart)
+  PROXY_CONSTMETHOD0(bool, is_screencast)
+  PROXY_CONSTMETHOD0(rtc::Optional<bool>, needs_denoising)
+  PROXY_METHOD1(bool, GetStats, Stats*)
+  PROXY_WORKER_METHOD2(void,
+                       AddOrUpdateSink,
+                       rtc::VideoSinkInterface<cricket::VideoFrame>*,
+                       const rtc::VideoSinkWants&)
+  PROXY_WORKER_METHOD1(void,
+                       RemoveSink,
+                       rtc::VideoSinkInterface<cricket::VideoFrame>*)
+  PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
+  PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
 END_PROXY()
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/videotrack.cc b/chromium/third_party/webrtc/api/videotrack.cc
index bd380254b2d..234b4cf7ae9 100644
--- a/chromium/third_party/webrtc/api/videotrack.cc
+++ b/chromium/third_party/webrtc/api/videotrack.cc
@@ -20,6 +20,7 @@ VideoTrack::VideoTrack(const std::string& label,
                        VideoTrackSourceInterface* video_source)
     : MediaStreamTrack<VideoTrackInterface>(label),
       video_source_(video_source) {
+  worker_thread_checker_.DetachFromThread();
   video_source_->RegisterObserver(this);
 }
 
@@ -31,10 +32,12 @@ std::string VideoTrack::kind() const {
   return kVideoKind;
 }
 
+// AddOrUpdateSink and RemoveSink should be called on the worker
+// thread.
 void VideoTrack::AddOrUpdateSink(
     rtc::VideoSinkInterface<cricket::VideoFrame>* sink,
     const rtc::VideoSinkWants& wants) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   VideoSourceBase::AddOrUpdateSink(sink, wants);
   rtc::VideoSinkWants modified_wants = wants;
   modified_wants.black_frames = !enabled();
@@ -43,23 +46,25 @@ void VideoTrack::AddOrUpdateSink(
 
 void VideoTrack::RemoveSink(
     rtc::VideoSinkInterface<cricket::VideoFrame>* sink) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   VideoSourceBase::RemoveSink(sink);
   video_source_->RemoveSink(sink);
 }
 
 bool VideoTrack::set_enabled(bool enable) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(signaling_thread_checker_.CalledOnValidThread());
   for (auto& sink_pair : sink_pairs()) {
     rtc::VideoSinkWants modified_wants = sink_pair.wants;
     modified_wants.black_frames = !enable;
+    // video_source_ is a proxy object, marshalling the call to the
+    // worker thread.
     video_source_->AddOrUpdateSink(sink_pair.sink, modified_wants);
   }
   return MediaStreamTrack<VideoTrackInterface>::set_enabled(enable);
 }
 
 void VideoTrack::OnChanged() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(signaling_thread_checker_.CalledOnValidThread());
   if (video_source_->state() == MediaSourceInterface::kEnded) {
     set_state(kEnded);
   } else {
diff --git a/chromium/third_party/webrtc/api/videotrack.h b/chromium/third_party/webrtc/api/videotrack.h
index 3835d2c936e..60a0a64aec7 100644
--- a/chromium/third_party/webrtc/api/videotrack.h
+++ b/chromium/third_party/webrtc/api/videotrack.h
@@ -33,11 +33,11 @@ class VideoTrack : public MediaStreamTrack<VideoTrackInterface>,
                        const rtc::VideoSinkWants& wants) override;
   void RemoveSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
 
-  virtual VideoTrackSourceInterface* GetSource() const {
+  VideoTrackSourceInterface* GetSource() const override {
     return video_source_.get();
   }
-  virtual bool set_enabled(bool enable);
-  virtual std::string kind() const;
+  bool set_enabled(bool enable) override;
+  std::string kind() const override;
 
  protected:
   VideoTrack(const std::string& id, VideoTrackSourceInterface* video_source);
@@ -47,7 +47,8 @@ class VideoTrack : public MediaStreamTrack<VideoTrackInterface>,
   // Implements ObserverInterface. Observes |video_source_| state.
   void OnChanged() override;
 
-  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker signaling_thread_checker_;
+  rtc::ThreadChecker worker_thread_checker_;
   rtc::scoped_refptr<VideoTrackSourceInterface> video_source_;
 };
 
diff --git a/chromium/third_party/webrtc/api/videotrack_unittest.cc b/chromium/third_party/webrtc/api/videotrack_unittest.cc
index d35bcdb2a03..0b67c77829c 100644
--- a/chromium/third_party/webrtc/api/videotrack_unittest.cc
+++ b/chromium/third_party/webrtc/api/videotrack_unittest.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/api/test/fakevideotrackrenderer.h"
 #include "webrtc/api/videocapturertracksource.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/engine/webrtcvideoframe.h"
@@ -31,7 +31,7 @@ class VideoTrackTest : public testing::Test {
   VideoTrackTest() {
     static const char kVideoTrackId[] = "track_id";
     video_track_source_ = new rtc::RefCountedObject<VideoTrackSource>(
-        &capturer_, rtc::Thread::Current(), true /* remote */);
+        &capturer_, true /* remote */);
     video_track_ = VideoTrack::Create(kVideoTrackId, video_track_source_);
     capturer_.Start(
         cricket::VideoFormat(640, 480, cricket::VideoFormat::FpsToInterval(30),
@@ -55,14 +55,14 @@ TEST_F(VideoTrackTest, SourceStateChangeTrackState) {
 // frames to the source.
 TEST_F(VideoTrackTest, RenderVideo) {
   // FakeVideoTrackRenderer register itself to |video_track_|
-  rtc::scoped_ptr<FakeVideoTrackRenderer> renderer_1(
+  std::unique_ptr<FakeVideoTrackRenderer> renderer_1(
       new FakeVideoTrackRenderer(video_track_.get()));
 
   capturer_.CaptureFrame();
   EXPECT_EQ(1, renderer_1->num_rendered_frames());
 
   // FakeVideoTrackRenderer register itself to |video_track_|
-  rtc::scoped_ptr<FakeVideoTrackRenderer> renderer_2(
+  std::unique_ptr<FakeVideoTrackRenderer> renderer_2(
       new FakeVideoTrackRenderer(video_track_.get()));
   capturer_.CaptureFrame();
   EXPECT_EQ(2, renderer_1->num_rendered_frames());
@@ -75,7 +75,7 @@ TEST_F(VideoTrackTest, RenderVideo) {
 
 // Test that disabling the track results in blacked out frames.
 TEST_F(VideoTrackTest, DisableTrackBlackout) {
-  rtc::scoped_ptr<FakeVideoTrackRenderer> renderer(
+  std::unique_ptr<FakeVideoTrackRenderer> renderer(
       new FakeVideoTrackRenderer(video_track_.get()));
 
   capturer_.CaptureFrame();
diff --git a/chromium/third_party/webrtc/api/videotracksource.cc b/chromium/third_party/webrtc/api/videotracksource.cc
index f8212d7a70c..17d32fbc710 100644
--- a/chromium/third_party/webrtc/api/videotracksource.cc
+++ b/chromium/third_party/webrtc/api/videotracksource.cc
@@ -12,18 +12,14 @@
 
 #include <string>
 
-#include "webrtc/base/bind.h"
-
 namespace webrtc {
 
 VideoTrackSource::VideoTrackSource(
     rtc::VideoSourceInterface<cricket::VideoFrame>* source,
-    rtc::Thread* worker_thread,
     bool remote)
-    : source_(source),
-      worker_thread_(worker_thread),
-      state_(kInitializing),
-      remote_(remote) {}
+    : source_(source), state_(kInitializing), remote_(remote) {
+  worker_thread_checker_.DetachFromThread();
+}
 
 void VideoTrackSource::SetState(SourceState new_state) {
   if (state_ != new_state) {
@@ -39,22 +35,20 @@ void VideoTrackSource::OnSourceDestroyed() {
 void VideoTrackSource::AddOrUpdateSink(
     rtc::VideoSinkInterface<cricket::VideoFrame>* sink,
     const rtc::VideoSinkWants& wants) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (!source_) {
     return;
   }
-  worker_thread_->Invoke<void>(rtc::Bind(
-      &rtc::VideoSourceInterface<cricket::VideoFrame>::AddOrUpdateSink, source_,
-      sink, wants));
+  source_->AddOrUpdateSink(sink, wants);
 }
 
 void VideoTrackSource::RemoveSink(
     rtc::VideoSinkInterface<cricket::VideoFrame>* sink) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (!source_) {
     return;
   }
-  worker_thread_->Invoke<void>(
-      rtc::Bind(&rtc::VideoSourceInterface<cricket::VideoFrame>::RemoveSink,
-                source_, sink));
+  source_->RemoveSink(sink);
 }
 
 }  //  namespace webrtc
diff --git a/chromium/third_party/webrtc/api/videotracksource.h b/chromium/third_party/webrtc/api/videotracksource.h
index 108209dc2c1..10e24ab27ed 100644
--- a/chromium/third_party/webrtc/api/videotracksource.h
+++ b/chromium/third_party/webrtc/api/videotracksource.h
@@ -13,6 +13,7 @@
 
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/notifier.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/videosinkinterface.h"
 
@@ -22,7 +23,6 @@ namespace webrtc {
 class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
  public:
   VideoTrackSource(rtc::VideoSourceInterface<cricket::VideoFrame>* source,
-                   rtc::Thread* worker_thread,
                    bool remote);
   void SetState(SourceState new_state);
   // OnSourceDestroyed clears this instance pointer to |source_|. It is useful
@@ -36,8 +36,8 @@ class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
   void Stop() override{};
   void Restart() override{};
 
-  virtual bool is_screencast() const { return false; }
-  virtual rtc::Optional<bool> needs_denoising() const {
+  bool is_screencast() const override { return false; }
+  rtc::Optional<bool> needs_denoising() const override {
     return rtc::Optional<bool>(); }
 
   bool GetStats(Stats* stats) override { return false; }
@@ -46,14 +46,9 @@ class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
                        const rtc::VideoSinkWants& wants) override;
   void RemoveSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
 
-  cricket::VideoCapturer* GetVideoCapturer() override { return nullptr; }
-
- protected:
-  rtc::Thread* worker_thread() { return worker_thread_; }
-
  private:
+  rtc::ThreadChecker worker_thread_checker_;
   rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
-  rtc::Thread* worker_thread_;
   cricket::VideoOptions options_;
   SourceState state_;
   const bool remote_;
diff --git a/chromium/third_party/webrtc/api/webrtcsdp.cc b/chromium/third_party/webrtc/api/webrtcsdp.cc
index 85fecf8bd8f..b86d9038ddc 100644
--- a/chromium/third_party/webrtc/api/webrtcsdp.cc
+++ b/chromium/third_party/webrtc/api/webrtcsdp.cc
@@ -13,8 +13,11 @@
 #include <ctype.h>
 #include <limits.h>
 #include <stdio.h>
+
 #include <algorithm>
+#include <memory>
 #include <string>
+#include <unordered_map>
 #include <vector>
 
 #include "webrtc/api/jsepicecandidate.h"
@@ -269,7 +272,7 @@ static bool ParseContent(const std::string& message,
                          const MediaType media_type,
                          int mline_index,
                          const std::string& protocol,
-                         const std::vector<int>& codec_preference,
+                         const std::vector<int>& payload_types,
                          size_t* pos,
                          std::string* content_name,
                          MediaContentDescription* media_desc,
@@ -287,7 +290,7 @@ static bool ParseCryptoAttribute(const std::string& line,
                                  SdpParseError* error);
 static bool ParseRtpmapAttribute(const std::string& line,
                                  const MediaType media_type,
-                                 const std::vector<int>& codec_preference,
+                                 const std::vector<int>& payload_types,
                                  MediaContentDescription* media_desc,
                                  SdpParseError* error);
 static bool ParseFmtpAttributes(const std::string& line,
@@ -1653,9 +1656,8 @@ bool AddSctpDataCodec(DataContentDescription* media_desc,
                        NULL);
   }
   // Add the SCTP Port number as a pseudo-codec "port" parameter
-  cricket::DataCodec codec_port(
-      cricket::kGoogleSctpDataCodecId, cricket::kGoogleSctpDataCodecName,
-      0);
+  cricket::DataCodec codec_port(cricket::kGoogleSctpDataCodecId,
+                                cricket::kGoogleSctpDataCodecName);
   codec_port.SetParam(cricket::kCodecParamPort, sctp_port);
   LOG(INFO) << "AddSctpDataCodec: Got SCTP Port Number "
             << sctp_port;
@@ -2181,9 +2183,8 @@ void MaybeCreateStaticPayloadAudioCodecs(
   if (!media_desc) {
     return;
   }
-  int preference = static_cast<int>(fmts.size());
+  RTC_DCHECK(media_desc->codecs().empty());
   std::vector<int>::const_iterator it = fmts.begin();
-  bool add_new_codec = false;
   for (; it != fmts.end(); ++it) {
     int payload_type = *it;
     if (!media_desc->HasCodec(payload_type) &&
@@ -2193,14 +2194,8 @@ void MaybeCreateStaticPayloadAudioCodecs(
       int clock_rate = kStaticPayloadAudioCodecs[payload_type].clockrate;
       size_t channels = kStaticPayloadAudioCodecs[payload_type].channels;
       media_desc->AddCodec(cricket::AudioCodec(payload_type, encoding_name,
-                                               clock_rate, 0, channels,
-                                               preference));
-      add_new_codec = true;
+                                               clock_rate, 0, channels));
     }
-    --preference;
-  }
-  if (add_new_codec) {
-    media_desc->SortCodecs();
   }
 }
 
@@ -2209,7 +2204,7 @@ static C* ParseContentDescription(const std::string& message,
                                   const MediaType media_type,
                                   int mline_index,
                                   const std::string& protocol,
-                                  const std::vector<int>& codec_preference,
+                                  const std::vector<int>& payload_types,
                                   size_t* pos,
                                   std::string* content_name,
                                   TransportDescription* transport,
@@ -2230,14 +2225,28 @@ static C* ParseContentDescription(const std::string& message,
       ASSERT(false);
       break;
   }
-  if (!ParseContent(message, media_type, mline_index, protocol,
-                    codec_preference, pos, content_name,
-                    media_desc, transport, candidates, error)) {
+  if (!ParseContent(message, media_type, mline_index, protocol, payload_types,
+                    pos, content_name, media_desc, transport, candidates,
+                    error)) {
     delete media_desc;
     return NULL;
   }
   // Sort the codecs according to the m-line fmt list.
-  media_desc->SortCodecs();
+  std::unordered_map<int, int> payload_type_preferences;
+  // "size + 1" so that the lowest preference payload type has a preference of
+  // 1, which is greater than the default (0) for payload types not in the fmt
+  // list.
+  int preference = static_cast<int>(payload_types.size() + 1);
+  for (int pt : payload_types) {
+    payload_type_preferences[pt] = preference--;
+  }
+  std::vector<typename C::CodecType> codecs = media_desc->codecs();
+  std::sort(codecs.begin(), codecs.end(), [&payload_type_preferences](
+                                              const typename C::CodecType& a,
+                                              const typename C::CodecType& b) {
+    return payload_type_preferences[a.id] > payload_type_preferences[b.id];
+  });
+  media_desc->set_codecs(codecs);
   return media_desc;
 }
 
@@ -2276,7 +2285,7 @@ bool ParseMediaDescription(const std::string& message,
     std::string protocol = fields[2];
 
     // <fmt>
-    std::vector<int> codec_preference;
+    std::vector<int> payload_types;
     if (IsRtp(protocol)) {
       for (size_t j = 3 ; j < fields.size(); ++j) {
         // TODO(wu): Remove when below bug is fixed.
@@ -2289,7 +2298,7 @@ bool ParseMediaDescription(const std::string& message,
         if (!GetPayloadTypeFromString(line, fields[j], &pl, error)) {
           return false;
         }
-        codec_preference.push_back(pl);
+        payload_types.push_back(pl);
       }
     }
 
@@ -2300,24 +2309,21 @@ bool ParseMediaDescription(const std::string& message,
         session_td.ice_mode, session_td.connection_role,
         session_td.identity_fingerprint.get());
 
-    rtc::scoped_ptr<MediaContentDescription> content;
+    std::unique_ptr<MediaContentDescription> content;
     std::string content_name;
     if (HasAttribute(line, kMediaTypeVideo)) {
       content.reset(ParseContentDescription<VideoContentDescription>(
-                    message, cricket::MEDIA_TYPE_VIDEO, mline_index, protocol,
-                    codec_preference, pos, &content_name,
-                    &transport, candidates, error));
+          message, cricket::MEDIA_TYPE_VIDEO, mline_index, protocol,
+          payload_types, pos, &content_name, &transport, candidates, error));
     } else if (HasAttribute(line, kMediaTypeAudio)) {
       content.reset(ParseContentDescription<AudioContentDescription>(
-                    message, cricket::MEDIA_TYPE_AUDIO, mline_index, protocol,
-                    codec_preference, pos, &content_name,
-                    &transport, candidates, error));
+          message, cricket::MEDIA_TYPE_AUDIO, mline_index, protocol,
+          payload_types, pos, &content_name, &transport, candidates, error));
     } else if (HasAttribute(line, kMediaTypeData)) {
       DataContentDescription* data_desc =
           ParseContentDescription<DataContentDescription>(
-                    message, cricket::MEDIA_TYPE_DATA, mline_index, protocol,
-                    codec_preference, pos, &content_name,
-                    &transport, candidates, error);
+              message, cricket::MEDIA_TYPE_DATA, mline_index, protocol,
+              payload_types, pos, &content_name, &transport, candidates, error);
       content.reset(data_desc);
 
       int p;
@@ -2524,7 +2530,7 @@ bool ParseContent(const std::string& message,
                   const MediaType media_type,
                   int mline_index,
                   const std::string& protocol,
-                  const std::vector<int>& codec_preference,
+                  const std::vector<int>& payload_types,
                   size_t* pos,
                   std::string* content_name,
                   MediaContentDescription* media_desc,
@@ -2537,7 +2543,7 @@ bool ParseContent(const std::string& message,
 
   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
     MaybeCreateStaticPayloadAudioCodecs(
-        codec_preference, static_cast<AudioContentDescription*>(media_desc));
+        payload_types, static_cast<AudioContentDescription*>(media_desc));
   }
 
   // The media level "ice-ufrag" and "ice-pwd".
@@ -2672,8 +2678,8 @@ bool ParseContent(const std::string& message,
           return false;
         }
       } else if (HasAttribute(line, kAttributeRtpmap)) {
-        if (!ParseRtpmapAttribute(line, media_type, codec_preference,
-                                  media_desc, error)) {
+        if (!ParseRtpmapAttribute(line, media_type, payload_types, media_desc,
+                                  error)) {
           return false;
         }
       } else if (HasAttribute(line, kCodecParamMaxPTime)) {
@@ -2929,9 +2935,12 @@ bool ParseCryptoAttribute(const std::string& line,
 }
 
 // Updates or creates a new codec entry in the audio description with according
-// to |name|, |clockrate|, |bitrate|, |channels| and |preference|.
-void UpdateCodec(int payload_type, const std::string& name, int clockrate,
-                 int bitrate, size_t channels, int preference,
+// to |name|, |clockrate|, |bitrate|, and |channels|.
+void UpdateCodec(int payload_type,
+                 const std::string& name,
+                 int clockrate,
+                 int bitrate,
+                 size_t channels,
                  AudioContentDescription* audio_desc) {
   // Codec may already be populated with (only) optional parameters
   // (from an fmtp).
@@ -2941,15 +2950,17 @@ void UpdateCodec(int payload_type, const std::string& name, int clockrate,
   codec.clockrate = clockrate;
   codec.bitrate = bitrate;
   codec.channels = channels;
-  codec.preference = preference;
   AddOrReplaceCodec<AudioContentDescription, cricket::AudioCodec>(audio_desc,
                                                                   codec);
 }
 
 // Updates or creates a new codec entry in the video description according to
-// |name|, |width|, |height|, |framerate| and |preference|.
-void UpdateCodec(int payload_type, const std::string& name, int width,
-                 int height, int framerate, int preference,
+// |name|, |width|, |height|, and |framerate|.
+void UpdateCodec(int payload_type,
+                 const std::string& name,
+                 int width,
+                 int height,
+                 int framerate,
                  VideoContentDescription* video_desc) {
   // Codec may already be populated with (only) optional parameters
   // (from an fmtp).
@@ -2959,14 +2970,13 @@ void UpdateCodec(int payload_type, const std::string& name, int width,
   codec.width = width;
   codec.height = height;
   codec.framerate = framerate;
-  codec.preference = preference;
   AddOrReplaceCodec<VideoContentDescription, cricket::VideoCodec>(video_desc,
                                                                   codec);
 }
 
 bool ParseRtpmapAttribute(const std::string& line,
                           const MediaType media_type,
-                          const std::vector<int>& codec_preference,
+                          const std::vector<int>& payload_types,
                           MediaContentDescription* media_desc,
                           SdpParseError* error) {
   std::vector<std::string> fields;
@@ -2988,12 +2998,8 @@ bool ParseRtpmapAttribute(const std::string& line,
     return false;
   }
 
-  // Set the preference order depending on the order of the pl type in the
-  // <fmt> of the m-line.
-  const int preference = codec_preference.end() -
-      std::find(codec_preference.begin(), codec_preference.end(),
-                payload_type);
-  if (preference == 0) {
+  if (std::find(payload_types.begin(), payload_types.end(), payload_type) ==
+      payload_types.end()) {
     LOG(LS_WARNING) << "Ignore rtpmap line that did not appear in the "
                     << "<fmt> of the m-line: " << line;
     return true;
@@ -3023,7 +3029,7 @@ bool ParseRtpmapAttribute(const std::string& line,
                 JsepSessionDescription::kMaxVideoCodecWidth,
                 JsepSessionDescription::kMaxVideoCodecHeight,
                 JsepSessionDescription::kDefaultVideoCodecFramerate,
-                preference, video_desc);
+                video_desc);
   } else if (media_type == cricket::MEDIA_TYPE_AUDIO) {
     // RFC 4566
     // For audio streams, <encoding parameters> indicates the number
@@ -3051,12 +3057,11 @@ bool ParseRtpmapAttribute(const std::string& line,
     AudioContentDescription* audio_desc =
         static_cast<AudioContentDescription*>(media_desc);
     UpdateCodec(payload_type, encoding_name, clock_rate, bitrate, channels,
-                preference, audio_desc);
+                audio_desc);
   } else if (media_type == cricket::MEDIA_TYPE_DATA) {
     DataContentDescription* data_desc =
         static_cast<DataContentDescription*>(media_desc);
-    data_desc->AddCodec(cricket::DataCodec(payload_type, encoding_name,
-                                           preference));
+    data_desc->AddCodec(cricket::DataCodec(payload_type, encoding_name));
   }
   return true;
 }
diff --git a/chromium/third_party/webrtc/api/webrtcsdp_unittest.cc b/chromium/third_party/webrtc/api/webrtcsdp_unittest.cc
index 099c9c04c0d..4bc84fdc12d 100644
--- a/chromium/third_party/webrtc/api/webrtcsdp_unittest.cc
+++ b/chromium/third_party/webrtc/api/webrtcsdp_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <set>
 #include <string>
 #include <vector>
@@ -20,7 +21,6 @@
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagedigest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslfingerprint.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
@@ -1030,10 +1030,10 @@ class WebRtcSdpTest : public testing::Test {
         "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32",
         "dummy_session_params"));
     audio->set_protocol(cricket::kMediaProtocolSavpf);
-    AudioCodec opus(111, "opus", 48000, 0, 2, 3);
+    AudioCodec opus(111, "opus", 48000, 0, 2);
     audio->AddCodec(opus);
-    audio->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1, 2));
-    audio->AddCodec(AudioCodec(104, "ISAC", 32000, 56000, 1, 1));
+    audio->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1));
+    audio->AddCodec(AudioCodec(104, "ISAC", 32000, 56000, 1));
     return audio;
   }
 
@@ -1049,8 +1049,7 @@ class WebRtcSdpTest : public testing::Test {
         VideoCodec(120, JsepSessionDescription::kDefaultVideoCodecName,
                    JsepSessionDescription::kMaxVideoCodecWidth,
                    JsepSessionDescription::kMaxVideoCodecHeight,
-                   JsepSessionDescription::kDefaultVideoCodecFramerate,
-                   JsepSessionDescription::kDefaultVideoCodecPreference));
+                   JsepSessionDescription::kDefaultVideoCodecFramerate));
     return video;
   }
 
@@ -1395,12 +1394,11 @@ class WebRtcSdpTest : public testing::Test {
   }
 
   void AddSctpDataChannel() {
-    rtc::scoped_ptr<DataContentDescription> data(
-        new DataContentDescription());
+    std::unique_ptr<DataContentDescription> data(new DataContentDescription());
     data_desc_ = data.get();
     data_desc_->set_protocol(cricket::kMediaProtocolDtlsSctp);
     DataCodec codec(cricket::kGoogleSctpDataCodecId,
-                    cricket::kGoogleSctpDataCodecName, 0);
+                    cricket::kGoogleSctpDataCodecName);
     codec.SetParam(cricket::kCodecParamPort, kDefaultSctpPort);
     data_desc_->AddCodec(codec);
     desc_.AddContent(kDataContentName, NS_JINGLE_DRAFT_SCTP, data.release());
@@ -1409,11 +1407,10 @@ class WebRtcSdpTest : public testing::Test {
   }
 
   void AddRtpDataChannel() {
-    rtc::scoped_ptr<DataContentDescription> data(
-        new DataContentDescription());
+    std::unique_ptr<DataContentDescription> data(new DataContentDescription());
     data_desc_ = data.get();
 
-    data_desc_->AddCodec(DataCodec(101, "google-data", 1));
+    data_desc_->AddCodec(DataCodec(101, "google-data"));
     StreamParams data_stream;
     data_stream.id = kDataChannelMsid;
     data_stream.cname = kDataChannelCname;
@@ -1680,7 +1677,7 @@ class WebRtcSdpTest : public testing::Test {
   VideoContentDescription* video_desc_;
   DataContentDescription* data_desc_;
   Candidates candidates_;
-  rtc::scoped_ptr<IceCandidateInterface> jcandidate_;
+  std::unique_ptr<IceCandidateInterface> jcandidate_;
   JsepSessionDescription jdesc_;
 };
 
@@ -1995,8 +1992,8 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
       jsep_desc.description()->GetContentDescriptionByName(kDataContentName));
 
   const int kNewPort = 1234;
-  cricket::DataCodec codec(
-        cricket::kGoogleSctpDataCodecId, cricket::kGoogleSctpDataCodecName, 0);
+  cricket::DataCodec codec(cricket::kGoogleSctpDataCodecId,
+                           cricket::kGoogleSctpDataCodecName);
   codec.SetParam(cricket::kCodecParamPort, kNewPort);
   dcdesc->AddOrReplaceCodec(codec);
 
@@ -2083,8 +2080,8 @@ TEST_F(WebRtcSdpTest, SerializeTcpCandidates) {
                       "", "", LOCAL_PORT_TYPE, kCandidateGeneration,
                       kCandidateFoundation1);
   candidate.set_tcptype(cricket::TCPTYPE_ACTIVE_STR);
-  rtc::scoped_ptr<IceCandidateInterface> jcandidate(
-    new JsepIceCandidate(std::string("audio_content_name"), 0, candidate));
+  std::unique_ptr<IceCandidateInterface> jcandidate(
+      new JsepIceCandidate(std::string("audio_content_name"), 0, candidate));
 
   std::string message = webrtc::SdpSerializeCandidate(*jcandidate);
   EXPECT_EQ(std::string(kSdpTcpActiveCandidate), message);
@@ -2180,10 +2177,11 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmap) {
     static_cast<AudioContentDescription*>(
         jdesc.description()->GetContentDescriptionByName(cricket::CN_AUDIO));
   AudioCodecs ref_codecs;
-  // The codecs in the AudioContentDescription will be sorted by preference.
-  ref_codecs.push_back(AudioCodec(0, "PCMU", 8000, 0, 1, 3));
-  ref_codecs.push_back(AudioCodec(18, "G729", 16000, 0, 1, 2));
-  ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 32000, 1, 1));
+  // The codecs in the AudioContentDescription should be in the same order as
+  // the payload types (<fmt>s) on the m= line.
+  ref_codecs.push_back(AudioCodec(0, "PCMU", 8000, 0, 1));
+  ref_codecs.push_back(AudioCodec(18, "G729", 16000, 0, 1));
+  ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 32000, 1));
   EXPECT_EQ(ref_codecs, audio->codecs());
 }
 
@@ -2405,8 +2403,8 @@ TEST_F(WebRtcSdpTest, DeserializeCandidate) {
                       rtc::SocketAddress("192.168.1.5", 9), kCandidatePriority,
                       "", "", LOCAL_PORT_TYPE, kCandidateGeneration,
                       kCandidateFoundation1);
-  rtc::scoped_ptr<IceCandidateInterface> jcandidate_template(
-    new JsepIceCandidate(std::string("audio_content_name"), 0, candidate));
+  std::unique_ptr<IceCandidateInterface> jcandidate_template(
+      new JsepIceCandidate(std::string("audio_content_name"), 0, candidate));
   EXPECT_TRUE(jcandidate.candidate().IsEquivalent(
                     jcandidate_template->candidate()));
   sdp = kSdpTcpPassiveCandidate;
diff --git a/chromium/third_party/webrtc/api/webrtcsession.cc b/chromium/third_party/webrtc/api/webrtcsession.cc
index f5462855741..80352cf8324 100644
--- a/chromium/third_party/webrtc/api/webrtcsession.cc
+++ b/chromium/third_party/webrtc/api/webrtcsession.cc
@@ -421,22 +421,6 @@ static std::string MakeTdErrorString(const std::string& desc) {
   return MakeErrorString(kPushDownTDFailed, desc);
 }
 
-uint32_t ConvertIceTransportTypeToCandidateFilter(
-    PeerConnectionInterface::IceTransportsType type) {
-  switch (type) {
-    case PeerConnectionInterface::kNone:
-        return cricket::CF_NONE;
-    case PeerConnectionInterface::kRelay:
-        return cricket::CF_RELAY;
-    case PeerConnectionInterface::kNoHost:
-        return (cricket::CF_ALL & ~cricket::CF_HOST);
-    case PeerConnectionInterface::kAll:
-        return cricket::CF_ALL;
-    default: ASSERT(false);
-  }
-  return cricket::CF_NONE;
-}
-
 // Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
 bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
                               const SessionDescriptionInterface* new_desc,
@@ -470,18 +454,18 @@ bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
 }
 
 WebRtcSession::WebRtcSession(webrtc::MediaControllerInterface* media_controller,
-                             rtc::Thread* signaling_thread,
+                             rtc::Thread* network_thread,
                              rtc::Thread* worker_thread,
+                             rtc::Thread* signaling_thread,
                              cricket::PortAllocator* port_allocator)
-    : signaling_thread_(signaling_thread),
-      worker_thread_(worker_thread),
-      port_allocator_(port_allocator),
+    : worker_thread_(worker_thread),
+      signaling_thread_(signaling_thread),
       // RFC 3264: The numeric value of the session id and version in the
       // o line MUST be representable with a "64 bit signed integer".
       // Due to this constraint session id |sid_| is max limited to LLONG_MAX.
       sid_(rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX)),
       transport_controller_(new cricket::TransportController(signaling_thread,
-                                                             worker_thread,
+                                                             network_thread,
                                                              port_allocator)),
       media_controller_(media_controller),
       channel_manager_(media_controller_->channel_manager()),
@@ -528,7 +512,7 @@ WebRtcSession::~WebRtcSession() {
 
 bool WebRtcSession::Initialize(
     const PeerConnectionFactoryInterface::Options& options,
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+    std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
   bundle_policy_ = rtc_configuration.bundle_policy;
   rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
@@ -604,8 +588,6 @@ bool WebRtcSession::Initialize(
   if (options.disable_encryption) {
     webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
   }
-  port_allocator()->set_candidate_filter(
-      ConvertIceTransportTypeToCandidateFilter(rtc_configuration.type));
 
   return true;
 }
@@ -675,7 +657,7 @@ bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
   ASSERT(signaling_thread()->IsCurrent());
 
   // Takes the ownership of |desc| regardless of the result.
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
+  std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
 
   // Validate SDP.
   if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {
@@ -731,14 +713,14 @@ bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,
   ASSERT(signaling_thread()->IsCurrent());
 
   // Takes the ownership of |desc| regardless of the result.
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
+  std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
 
   // Validate SDP.
   if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {
     return false;
   }
 
-  rtc::scoped_ptr<SessionDescriptionInterface> old_remote_desc(
+  std::unique_ptr<SessionDescriptionInterface> old_remote_desc(
       remote_desc_.release());
   remote_desc_.reset(desc_temp.release());
 
@@ -1039,7 +1021,7 @@ bool WebRtcSession::GetLocalCertificate(
                                                     certificate);
 }
 
-rtc::scoped_ptr<rtc::SSLCertificate> WebRtcSession::GetRemoteSSLCertificate(
+std::unique_ptr<rtc::SSLCertificate> WebRtcSession::GetRemoteSSLCertificate(
     const std::string& transport_name) {
   ASSERT(signaling_thread()->IsCurrent());
   return transport_controller_->GetRemoteSSLCertificate(transport_name);
@@ -1145,12 +1127,6 @@ bool WebRtcSession::RemoveRemoteIceCandidates(
   return true;
 }
 
-bool WebRtcSession::SetIceTransports(
-    PeerConnectionInterface::IceTransportsType type) {
-  return port_allocator()->set_candidate_filter(
-        ConvertIceTransportTypeToCandidateFilter(type));
-}
-
 cricket::IceConfig WebRtcSession::ParseIceConfig(
     const PeerConnectionInterface::RTCConfiguration& config) const {
   cricket::IceConfig ice_config;
@@ -1236,33 +1212,52 @@ void WebRtcSession::SetAudioPlayoutVolume(uint32_t ssrc, double volume) {
 }
 
 void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
-                                    rtc::scoped_ptr<AudioSinkInterface> sink) {
+                                    std::unique_ptr<AudioSinkInterface> sink) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_)
     return;
 
-  voice_channel_->SetRawAudioSink(ssrc, rtc::ScopedToUnique(std::move(sink)));
+  voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
 }
 
-RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const {
+RtpParameters WebRtcSession::GetAudioRtpSendParameters(uint32_t ssrc) const {
   ASSERT(signaling_thread()->IsCurrent());
   if (voice_channel_) {
-    return voice_channel_->GetRtpParameters(ssrc);
+    return voice_channel_->GetRtpSendParameters(ssrc);
   }
   return RtpParameters();
 }
 
-bool WebRtcSession::SetAudioRtpParameters(uint32_t ssrc,
-                                          const RtpParameters& parameters) {
+bool WebRtcSession::SetAudioRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     return false;
   }
-  return voice_channel_->SetRtpParameters(ssrc, parameters);
+  return voice_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetAudioRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (voice_channel_) {
+    return voice_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
 }
 
-bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
-                                     cricket::VideoCapturer* camera) {
+bool WebRtcSession::SetAudioRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    return false;
+  }
+  return voice_channel_->SetRtpReceiveParameters(ssrc, parameters);
+}
+
+bool WebRtcSession::SetSource(
+    uint32_t ssrc,
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
   ASSERT(signaling_thread()->IsCurrent());
 
   if (!video_channel_) {
@@ -1271,13 +1266,7 @@ bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
     LOG(LS_WARNING) << "Video not used in this call.";
     return false;
   }
-  if (!video_channel_->SetCapturer(ssrc, camera)) {
-    // Allow that SetCapturer fail if |camera| is NULL but assert otherwise.
-    // This in the normal case when the underlying media channel has already
-    // been deleted.
-    ASSERT(camera == NULL);
-    return false;
-  }
+  video_channel_->SetSource(ssrc, source);
   return true;
 }
 
@@ -1314,21 +1303,39 @@ void WebRtcSession::SetVideoSend(uint32_t ssrc,
   }
 }
 
-RtpParameters WebRtcSession::GetVideoRtpParameters(uint32_t ssrc) const {
+RtpParameters WebRtcSession::GetVideoRtpSendParameters(uint32_t ssrc) const {
   ASSERT(signaling_thread()->IsCurrent());
   if (video_channel_) {
-    return video_channel_->GetRtpParameters(ssrc);
+    return video_channel_->GetRtpSendParameters(ssrc);
   }
   return RtpParameters();
 }
 
-bool WebRtcSession::SetVideoRtpParameters(uint32_t ssrc,
-                                          const RtpParameters& parameters) {
+bool WebRtcSession::SetVideoRtpSendParameters(uint32_t ssrc,
+                                              const RtpParameters& parameters) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!video_channel_) {
     return false;
   }
-  return video_channel_->SetRtpParameters(ssrc, parameters);
+  return video_channel_->SetRtpSendParameters(ssrc, parameters);
+}
+
+RtpParameters WebRtcSession::GetVideoRtpReceiveParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (video_channel_) {
+    return video_channel_->GetRtpReceiveParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetVideoRtpReceiveParameters(
+    uint32_t ssrc,
+    const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    return false;
+  }
+  return video_channel_->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
@@ -1732,13 +1739,41 @@ void WebRtcSession::RemoveUnusedChannels(const SessionDescription* desc) {
   }
 }
 
-// TODO(mallinath) - Add a correct error code if the channels are not created
-// due to BUNDLE is enabled but rtcp-mux is disabled.
+// Returns the name of the transport channel when BUNDLE is enabled, or nullptr
+// if the channel is not part of any bundle.
+const std::string* WebRtcSession::GetBundleTransportName(
+    const cricket::ContentInfo* content,
+    const cricket::ContentGroup* bundle) {
+  if (!bundle) {
+    return nullptr;
+  }
+  const std::string* first_content_name = bundle->FirstContentName();
+  if (!first_content_name) {
+    LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
+    return nullptr;
+  }
+  if (!bundle->HasContentName(content->name)) {
+    LOG(LS_WARNING) << content->name << " is not part of any bundle group";
+    return nullptr;
+  }
+  LOG(LS_INFO) << "Bundling " << content->name << " on " << *first_content_name;
+  return first_content_name;
+}
+
 bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
+  const cricket::ContentGroup* bundle_group = nullptr;
+  if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {
+    bundle_group = desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
+    if (!bundle_group) {
+      LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";
+      return false;
+    }
+  }
   // Creating the media channels and transport proxies.
   const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
   if (voice && !voice->rejected && !voice_channel_) {
-    if (!CreateVoiceChannel(voice)) {
+    if (!CreateVoiceChannel(voice,
+                            GetBundleTransportName(voice, bundle_group))) {
       LOG(LS_ERROR) << "Failed to create voice channel.";
       return false;
     }
@@ -1746,7 +1781,8 @@ bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
 
   const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
   if (video && !video->rejected && !video_channel_) {
-    if (!CreateVideoChannel(video)) {
+    if (!CreateVideoChannel(video,
+                            GetBundleTransportName(video, bundle_group))) {
       LOG(LS_ERROR) << "Failed to create video channel.";
       return false;
     }
@@ -1755,82 +1791,77 @@ bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
   const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);
   if (data_channel_type_ != cricket::DCT_NONE &&
       data && !data->rejected && !data_channel_) {
-    if (!CreateDataChannel(data)) {
+    if (!CreateDataChannel(data, GetBundleTransportName(data, bundle_group))) {
       LOG(LS_ERROR) << "Failed to create data channel.";
       return false;
     }
   }
 
-  if (rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire) {
-    if (voice_channel()) {
-      voice_channel()->ActivateRtcpMux();
-    }
-    if (video_channel()) {
-      video_channel()->ActivateRtcpMux();
-    }
-    if (data_channel()) {
-      data_channel()->ActivateRtcpMux();
-    }
-  }
-
-  // Enable BUNDLE immediately when kBundlePolicyMaxBundle is in effect.
-  if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {
-    const cricket::ContentGroup* bundle_group = desc->GetGroupByName(
-        cricket::GROUP_TYPE_BUNDLE);
-    if (!bundle_group) {
-      LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";
-      return false;
-    }
-    if (!EnableBundle(*bundle_group)) {
-      LOG(LS_WARNING) << "max-bundle failed to enable bundling.";
-      return false;
-    }
-  }
-
   return true;
 }
 
-bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content) {
+bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content,
+                                       const std::string* bundle_transport) {
+  bool require_rtcp_mux =
+      rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  bool create_rtcp_transport_channel = !require_rtcp_mux;
   voice_channel_.reset(channel_manager_->CreateVoiceChannel(
-      media_controller_, transport_controller_.get(), content->name, true,
-      audio_options_));
+      media_controller_, transport_controller_.get(), content->name,
+      bundle_transport, create_rtcp_transport_channel, audio_options_));
   if (!voice_channel_) {
     return false;
   }
+  if (require_rtcp_mux) {
+    voice_channel_->ActivateRtcpMux();
+  }
 
   voice_channel_->SignalDtlsSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSetupFailure);
 
   SignalVoiceChannelCreated();
-  voice_channel_->transport_channel()->SignalSentPacket.connect(
-      this, &WebRtcSession::OnSentPacket_w);
+  voice_channel_->SignalSentPacket.connect(this,
+                                           &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
-bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content) {
+bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
+                                       const std::string* bundle_transport) {
+  bool require_rtcp_mux =
+      rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  bool create_rtcp_transport_channel = !require_rtcp_mux;
   video_channel_.reset(channel_manager_->CreateVideoChannel(
-      media_controller_, transport_controller_.get(), content->name, true,
-      video_options_));
+      media_controller_, transport_controller_.get(), content->name,
+      bundle_transport, create_rtcp_transport_channel, video_options_));
   if (!video_channel_) {
     return false;
   }
-
+  if (require_rtcp_mux) {
+    video_channel_->ActivateRtcpMux();
+  }
   video_channel_->SignalDtlsSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSetupFailure);
 
   SignalVideoChannelCreated();
-  video_channel_->transport_channel()->SignalSentPacket.connect(
-      this, &WebRtcSession::OnSentPacket_w);
+  video_channel_->SignalSentPacket.connect(this,
+                                           &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
-bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content) {
+bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
+                                      const std::string* bundle_transport) {
   bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
+  bool require_rtcp_mux =
+      rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  bool create_rtcp_transport_channel = !sctp && !require_rtcp_mux;
   data_channel_.reset(channel_manager_->CreateDataChannel(
-      transport_controller_.get(), content->name, !sctp, data_channel_type_));
+      transport_controller_.get(), content->name, bundle_transport,
+      create_rtcp_transport_channel, data_channel_type_));
   if (!data_channel_) {
     return false;
   }
+  if (require_rtcp_mux) {
+    data_channel_->ActivateRtcpMux();
+  }
 
   if (sctp) {
     data_channel_->SignalDataReceived.connect(
@@ -1841,8 +1872,7 @@ bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content) {
       this, &WebRtcSession::OnDtlsSetupFailure);
 
   SignalDataChannelCreated();
-  data_channel_->transport_channel()->SignalSentPacket.connect(
-      this, &WebRtcSession::OnSentPacket_w);
+  data_channel_->SignalSentPacket.connect(this, &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
@@ -1944,6 +1974,9 @@ bool WebRtcSession::ValidateSessionDescription(
     return BadSdp(source, type, kBundleWithoutRtcpMux, err_desc);
   }
 
+  // TODO(skvlad): When the local rtcp-mux policy is Require, reject any
+  // m-lines that do not rtcp-mux enabled.
+
   // Verify m-lines in Answer when compared against Offer.
   if (action == kAnswer) {
     const cricket::SessionDescription* offer_desc =
@@ -2160,8 +2193,7 @@ void WebRtcSession::ReportNegotiatedCiphers(
   }
 }
 
-void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
-                                   const rtc::SentPacket& sent_packet) {
+void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
diff --git a/chromium/third_party/webrtc/api/webrtcsession.h b/chromium/third_party/webrtc/api/webrtcsession.h
index 01ec526501f..98217bff266 100644
--- a/chromium/third_party/webrtc/api/webrtcsession.h
+++ b/chromium/third_party/webrtc/api/webrtcsession.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_API_WEBRTCSESSION_H_
 #define WEBRTC_API_WEBRTCSESSION_H_
 
+#include <memory>
 #include <set>
 #include <string>
 #include <vector>
@@ -21,6 +22,7 @@
 #include "webrtc/api/mediastreamprovider.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/statstypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
@@ -34,7 +36,6 @@ namespace cricket {
 class ChannelManager;
 class DataChannel;
 class StatsReport;
-class VideoCapturer;
 class VideoChannel;
 class VoiceChannel;
 
@@ -137,22 +138,22 @@ class WebRtcSession : public AudioProviderInterface,
   };
 
   WebRtcSession(webrtc::MediaControllerInterface* media_controller,
-                rtc::Thread* signaling_thread,
+                rtc::Thread* network_thread,
                 rtc::Thread* worker_thread,
+                rtc::Thread* signaling_thread,
                 cricket::PortAllocator* port_allocator);
   virtual ~WebRtcSession();
 
   // These are const to allow them to be called from const methods.
-  rtc::Thread* signaling_thread() const { return signaling_thread_; }
   rtc::Thread* worker_thread() const { return worker_thread_; }
-  cricket::PortAllocator* port_allocator() const { return port_allocator_; }
+  rtc::Thread* signaling_thread() const { return signaling_thread_; }
 
   // The ID of this session.
   const std::string& id() const { return sid_; }
 
   bool Initialize(
       const PeerConnectionFactoryInterface::Options& options,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
   // Deletes the voice, video and data channel and changes the session state
   // to STATE_CLOSED.
@@ -213,8 +214,6 @@ class WebRtcSession : public AudioProviderInterface,
   bool RemoveRemoteIceCandidates(
       const std::vector<cricket::Candidate>& candidates);
 
-  bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
-
   cricket::IceConfig ParseIceConfig(
       const PeerConnectionInterface::RTCConfiguration& config) const;
 
@@ -243,14 +242,19 @@ class WebRtcSession : public AudioProviderInterface,
                     cricket::AudioSource* source) override;
   void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
   void SetRawAudioSink(uint32_t ssrc,
-                       rtc::scoped_ptr<AudioSinkInterface> sink) override;
+                       std::unique_ptr<AudioSinkInterface> sink) override;
 
-  RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
-  bool SetAudioRtpParameters(uint32_t ssrc,
-                             const RtpParameters& parameters) override;
+  RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const override;
+  bool SetAudioRtpSendParameters(uint32_t ssrc,
+                                 const RtpParameters& parameters) override;
+  RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const override;
+  bool SetAudioRtpReceiveParameters(uint32_t ssrc,
+                                    const RtpParameters& parameters) override;
 
   // Implements VideoMediaProviderInterface.
-  bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
+  bool SetSource(
+      uint32_t ssrc,
+      rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
   void SetVideoPlayout(
       uint32_t ssrc,
       bool enable,
@@ -259,15 +263,18 @@ class WebRtcSession : public AudioProviderInterface,
                     bool enable,
                     const cricket::VideoOptions* options) override;
 
-  RtpParameters GetVideoRtpParameters(uint32_t ssrc) const override;
-  bool SetVideoRtpParameters(uint32_t ssrc,
-                             const RtpParameters& parameters) override;
+  RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const override;
+  bool SetVideoRtpSendParameters(uint32_t ssrc,
+                                 const RtpParameters& parameters) override;
+  RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const override;
+  bool SetVideoRtpReceiveParameters(uint32_t ssrc,
+                                    const RtpParameters& parameters) override;
 
   // Implements DtmfProviderInterface.
-  virtual bool CanInsertDtmf(const std::string& track_id);
-  virtual bool InsertDtmf(const std::string& track_id,
-                          int code, int duration);
-  virtual sigslot::signal0<>* GetOnDestroyedSignal();
+  bool CanInsertDtmf(const std::string& track_id) override;
+  bool InsertDtmf(const std::string& track_id,
+                  int code, int duration) override;
+  sigslot::signal0<>* GetOnDestroyedSignal() override;
 
   // Implements DataChannelProviderInterface.
   bool SendData(const cricket::SendDataParams& params,
@@ -292,7 +299,7 @@ class WebRtcSession : public AudioProviderInterface,
       rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
 
   // Caller owns returned certificate
-  virtual rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
+  virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
       const std::string& transport_name);
 
   cricket::DataChannelType data_channel_type() const;
@@ -379,6 +386,12 @@ class WebRtcSession : public AudioProviderInterface,
       const std::string& content_name,
       cricket::TransportDescription* info);
 
+  // Returns the name of the transport channel when BUNDLE is enabled, or
+  // nullptr if the channel is not part of any bundle.
+  const std::string* GetBundleTransportName(
+      const cricket::ContentInfo* content,
+      const cricket::ContentGroup* bundle);
+
   // Cause all the BaseChannels in the bundle group to have the same
   // transport channel.
   bool EnableBundle(const cricket::ContentGroup& bundle);
@@ -405,9 +418,12 @@ class WebRtcSession : public AudioProviderInterface,
   bool CreateChannels(const cricket::SessionDescription* desc);
 
   // Helper methods to create media channels.
-  bool CreateVoiceChannel(const cricket::ContentInfo* content);
-  bool CreateVideoChannel(const cricket::ContentInfo* content);
-  bool CreateDataChannel(const cricket::ContentInfo* content);
+  bool CreateVoiceChannel(const cricket::ContentInfo* content,
+                          const std::string* bundle_transport);
+  bool CreateVideoChannel(const cricket::ContentInfo* content,
+                          const std::string* bundle_transport);
+  bool CreateDataChannel(const cricket::ContentInfo* content,
+                         const std::string* bundle_transport);
 
   // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
   // messages.
@@ -462,12 +478,10 @@ class WebRtcSession : public AudioProviderInterface,
 
   void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
 
-  void OnSentPacket_w(cricket::TransportChannel* channel,
-                      const rtc::SentPacket& sent_packet);
+  void OnSentPacket_w(const rtc::SentPacket& sent_packet);
 
-  rtc::Thread* const signaling_thread_;
   rtc::Thread* const worker_thread_;
-  cricket::PortAllocator* const port_allocator_;
+  rtc::Thread* const signaling_thread_;
 
   State state_ = STATE_INIT;
   Error error_ = ERROR_NONE;
@@ -476,17 +490,17 @@ class WebRtcSession : public AudioProviderInterface,
   const std::string sid_;
   bool initial_offerer_ = false;
 
-  rtc::scoped_ptr<cricket::TransportController> transport_controller_;
+  std::unique_ptr<cricket::TransportController> transport_controller_;
   MediaControllerInterface* media_controller_;
-  rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
-  rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
-  rtc::scoped_ptr<cricket::DataChannel> data_channel_;
+  std::unique_ptr<cricket::VoiceChannel> voice_channel_;
+  std::unique_ptr<cricket::VideoChannel> video_channel_;
+  std::unique_ptr<cricket::DataChannel> data_channel_;
   cricket::ChannelManager* channel_manager_;
   IceObserver* ice_observer_;
   PeerConnectionInterface::IceConnectionState ice_connection_state_;
   bool ice_connection_receiving_;
-  rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
-  rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
+  std::unique_ptr<SessionDescriptionInterface> local_desc_;
+  std::unique_ptr<SessionDescriptionInterface> remote_desc_;
   // If the remote peer is using a older version of implementation.
   bool older_version_remote_peer_;
   bool dtls_enabled_;
@@ -501,8 +515,7 @@ class WebRtcSession : public AudioProviderInterface,
   // List of content names for which the remote side triggered an ICE restart.
   std::set<std::string> pending_ice_restarts_;
 
-  rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
-      webrtc_session_desc_factory_;
+  std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
 
   // Member variables for caching global options.
   cricket::AudioOptions audio_options_;
diff --git a/chromium/third_party/webrtc/api/webrtcsession_unittest.cc b/chromium/third_party/webrtc/api/webrtcsession_unittest.cc
index 18c1a95116e..4207c24234c 100644
--- a/chromium/third_party/webrtc/api/webrtcsession_unittest.cc
+++ b/chromium/third_party/webrtc/api/webrtcsession_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <utility>
 #include <vector>
 
@@ -55,7 +56,6 @@
 using cricket::FakeVoiceMediaChannel;
 using cricket::TransportInfo;
 using rtc::SocketAddress;
-using rtc::scoped_ptr;
 using rtc::Thread;
 using webrtc::CreateSessionDescription;
 using webrtc::CreateSessionDescriptionObserver;
@@ -158,9 +158,12 @@ class MockIceObserver : public webrtc::IceObserver {
         ice_gathering_state_(PeerConnectionInterface::kIceGatheringNew) {
   }
 
+  virtual ~MockIceObserver() = default;
+
   void OnIceConnectionChange(
       PeerConnectionInterface::IceConnectionState new_state) override {
     ice_connection_state_ = new_state;
+    ice_connection_state_history_.push_back(new_state);
   }
   void OnIceGatheringChange(
       PeerConnectionInterface::IceGatheringState new_state) override {
@@ -191,7 +194,7 @@ class MockIceObserver : public webrtc::IceObserver {
 
   // Some local candidates are removed.
   void OnIceCandidatesRemoved(
-      const std::vector<cricket::Candidate>& candidates) {
+      const std::vector<cricket::Candidate>& candidates) override {
     num_candidates_removed_ += candidates.size();
   }
 
@@ -200,19 +203,23 @@ class MockIceObserver : public webrtc::IceObserver {
   std::vector<cricket::Candidate> mline_1_candidates_;
   PeerConnectionInterface::IceConnectionState ice_connection_state_;
   PeerConnectionInterface::IceGatheringState ice_gathering_state_;
+  std::vector<PeerConnectionInterface::IceConnectionState>
+      ice_connection_state_history_;
   size_t num_candidates_removed_ = 0;
 };
 
 class WebRtcSessionForTest : public webrtc::WebRtcSession {
  public:
   WebRtcSessionForTest(webrtc::MediaControllerInterface* media_controller,
-                       rtc::Thread* signaling_thread,
+                       rtc::Thread* network_thread,
                        rtc::Thread* worker_thread,
+                       rtc::Thread* signaling_thread,
                        cricket::PortAllocator* port_allocator,
                        webrtc::IceObserver* ice_observer)
       : WebRtcSession(media_controller,
-                      signaling_thread,
+                      network_thread,
                       worker_thread,
+                      signaling_thread,
                       port_allocator) {
     RegisterIceObserver(ice_observer);
   }
@@ -246,7 +253,7 @@ class WebRtcSessionForTest : public webrtc::WebRtcSession {
 
   using webrtc::WebRtcSession::SetAudioPlayout;
   using webrtc::WebRtcSession::SetAudioSend;
-  using webrtc::WebRtcSession::SetCaptureDevice;
+  using webrtc::WebRtcSession::SetSource;
   using webrtc::WebRtcSession::SetVideoPlayout;
   using webrtc::WebRtcSession::SetVideoSend;
 
@@ -297,7 +304,7 @@ class WebRtcSessionCreateSDPObserverForTest
   ~WebRtcSessionCreateSDPObserverForTest() {}
 
  private:
-  rtc::scoped_ptr<SessionDescriptionInterface> description_;
+  std::unique_ptr<SessionDescriptionInterface> description_;
   State state_;
 };
 
@@ -375,11 +382,11 @@ class WebRtcSessionTest
   // used if provided, otherwise one will be generated using the
   // |dtls_identity_store|.
   void Init(
-      rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
+      std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
     ASSERT_TRUE(session_.get() == NULL);
     session_.reset(new WebRtcSessionForTest(
         media_controller_.get(), rtc::Thread::Current(), rtc::Thread::Current(),
-        allocator_.get(), &observer_));
+        rtc::Thread::Current(), allocator_.get(), &observer_));
     session_->SignalDataChannelOpenMessage.connect(
         this, &WebRtcSessionTest::OnDataChannelOpenMessage);
     session_->GetOnDestroyedSignal()->connect(
@@ -405,12 +412,6 @@ class WebRtcSessionTest
 
   void Init() { Init(nullptr); }
 
-  void InitWithIceTransport(
-      PeerConnectionInterface::IceTransportsType ice_transport_type) {
-    configuration_.type = ice_transport_type;
-    Init();
-  }
-
   void InitWithBundlePolicy(
       PeerConnectionInterface::BundlePolicy bundle_policy) {
     configuration_.bundle_policy = bundle_policy;
@@ -427,7 +428,7 @@ class WebRtcSessionTest
   // Successfully init with DTLS; with a certificate generated and supplied or
   // with a store that generates it for us.
   void InitWithDtls(RTCCertificateGenerationMethod cert_gen_method) {
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store;
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store;
     if (cert_gen_method == ALREADY_GENERATED) {
       configuration_.certificates.push_back(
           FakeDtlsIdentityStore::GenerateCertificate());
@@ -442,7 +443,7 @@ class WebRtcSessionTest
 
   // Init with DTLS with a store that will fail to generate a certificate.
   void InitWithDtlsIdentityGenFail() {
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         new FakeDtlsIdentityStore());
     dtls_identity_store->set_should_fail(true);
     Init(std::move(dtls_identity_store));
@@ -450,8 +451,8 @@ class WebRtcSessionTest
 
   void InitWithDtmfCodec() {
     // Add kTelephoneEventCodec for dtmf test.
-    const cricket::AudioCodec kTelephoneEventCodec(
-        106, "telephone-event", 8000, 0, 1, 0);
+    const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000,
+                                                   0, 1);
     std::vector<cricket::AudioCodec> codecs;
     codecs.push_back(kTelephoneEventCodec);
     media_engine_->SetAudioCodecs(codecs);
@@ -692,7 +693,7 @@ class WebRtcSessionTest
         rtc::ToString(rtc::CreateRandomId());
     // Confirmed to work with KT_RSA and KT_ECDSA.
     tdesc_factory_->set_certificate(
-        rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+        rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
             rtc::SSLIdentity::Generate(identity_name, rtc::KT_DEFAULT))));
     tdesc_factory_->set_secure(cricket::SEC_REQUIRED);
   }
@@ -726,12 +727,12 @@ class WebRtcSessionTest
     cricket::MediaSessionOptions options;
     options.recv_video = true;
     options.bundle_enabled = true;
-    scoped_ptr<JsepSessionDescription> offer(
+    std::unique_ptr<JsepSessionDescription> offer(
         CreateRemoteOffer(options, cricket::SEC_REQUIRED));
     ASSERT_TRUE(offer.get() != NULL);
     VerifyCryptoParams(offer->description());
     SetRemoteDescriptionWithoutError(offer.release());
-    scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+    std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
     ASSERT_TRUE(answer.get() != NULL);
     VerifyCryptoParams(answer->description());
   }
@@ -930,7 +931,7 @@ class WebRtcSessionTest
     options.recv_video = true;
     options.bundle_enabled = true;
 
-    rtc::scoped_ptr<SessionDescriptionInterface> temp_offer(
+    std::unique_ptr<SessionDescriptionInterface> temp_offer(
         CreateRemoteOffer(options, cricket::SEC_ENABLED));
 
     *nodtls_answer =
@@ -1069,7 +1070,7 @@ class WebRtcSessionTest
     // and answer.
     SetLocalDescriptionWithoutError(offer);
 
-    rtc::scoped_ptr<SessionDescriptionInterface> answer(
+    std::unique_ptr<SessionDescriptionInterface> answer(
         CreateRemoteAnswer(session_->local_description()));
     std::string sdp;
     EXPECT_TRUE(answer->ToString(&sdp));
@@ -1113,6 +1114,22 @@ class WebRtcSessionTest
     EXPECT_EQ(can, session_->CanInsertDtmf(kAudioTrack1));
   }
 
+  bool ContainsVideoCodecWithName(const SessionDescriptionInterface* desc,
+                                  const std::string& codec_name) {
+    for (const auto& content : desc->description()->contents()) {
+      if (static_cast<cricket::MediaContentDescription*>(content.description)
+              ->type() == cricket::MEDIA_TYPE_VIDEO) {
+        const auto* mdesc =
+            static_cast<cricket::VideoContentDescription*>(content.description);
+        for (const auto& codec : mdesc->codecs()) {
+          if (codec.name == codec_name) {
+            return true;
+          }
+        }
+      }
+    }
+    return false;
+  }
   // Helper class to configure loopback network and verify Best
   // Connection using right IP protocol for TestLoopbackCall
   // method. LoopbackNetworkManager applies firewall rules to block
@@ -1300,6 +1317,8 @@ class WebRtcSessionTest
 
     SetupLoopbackCall();
 
+    // Wait for channel to be ready for sending.
+    EXPECT_TRUE_WAIT(media_engine_->GetVideoChannel(0)->sending(), 100);
     uint8_t test_packet[15] = {0};
     rtc::PacketOptions options;
     options.packet_id = 10;
@@ -1313,8 +1332,8 @@ class WebRtcSessionTest
 
   // Adds CN codecs to FakeMediaEngine and MediaDescriptionFactory.
   void AddCNCodecs() {
-    const cricket::AudioCodec kCNCodec1(102, "CN", 8000, 0, 1, 0);
-    const cricket::AudioCodec kCNCodec2(103, "CN", 16000, 0, 1, 0);
+    const cricket::AudioCodec kCNCodec1(102, "CN", 8000, 0, 1);
+    const cricket::AudioCodec kCNCodec2(103, "CN", 16000, 0, 1);
 
     // Add kCNCodec for dtmf test.
     std::vector<cricket::AudioCodec> codecs = media_engine_->audio_codecs();;
@@ -1370,8 +1389,8 @@ class WebRtcSessionTest
     SetFactoryDtlsSrtp();
     if (type == CreateSessionDescriptionRequest::kAnswer) {
       cricket::MediaSessionOptions options;
-      scoped_ptr<JsepSessionDescription> offer(
-            CreateRemoteOffer(options, cricket::SEC_DISABLED));
+      std::unique_ptr<JsepSessionDescription> offer(
+          CreateRemoteOffer(options, cricket::SEC_DISABLED));
       ASSERT_TRUE(offer.get() != NULL);
       SetRemoteDescriptionWithoutError(offer.release());
     }
@@ -1417,23 +1436,23 @@ class WebRtcSessionTest
 
   cricket::FakeMediaEngine* media_engine_;
   cricket::FakeDataEngine* data_engine_;
-  rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
+  std::unique_ptr<cricket::ChannelManager> channel_manager_;
   cricket::FakeCall fake_call_;
-  rtc::scoped_ptr<webrtc::MediaControllerInterface> media_controller_;
-  rtc::scoped_ptr<cricket::TransportDescriptionFactory> tdesc_factory_;
-  rtc::scoped_ptr<cricket::MediaSessionDescriptionFactory> desc_factory_;
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
-  rtc::scoped_ptr<rtc::FirewallSocketServer> fss_;
+  std::unique_ptr<webrtc::MediaControllerInterface> media_controller_;
+  std::unique_ptr<cricket::TransportDescriptionFactory> tdesc_factory_;
+  std::unique_ptr<cricket::MediaSessionDescriptionFactory> desc_factory_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
+  std::unique_ptr<rtc::FirewallSocketServer> fss_;
   rtc::SocketServerScope ss_scope_;
   rtc::SocketAddress stun_socket_addr_;
-  rtc::scoped_ptr<cricket::TestStunServer> stun_server_;
+  std::unique_ptr<cricket::TestStunServer> stun_server_;
   cricket::TestTurnServer turn_server_;
   rtc::FakeNetworkManager network_manager_;
-  rtc::scoped_ptr<cricket::BasicPortAllocator> allocator_;
+  std::unique_ptr<cricket::BasicPortAllocator> allocator_;
   PeerConnectionFactoryInterface::Options options_;
   PeerConnectionInterface::RTCConfiguration configuration_;
-  rtc::scoped_ptr<WebRtcSessionForTest> session_;
+  std::unique_ptr<WebRtcSessionForTest> session_;
   MockIceObserver observer_;
   cricket::FakeVideoMediaChannel* video_channel_;
   cricket::FakeVoiceMediaChannel* voice_channel_;
@@ -1509,50 +1528,6 @@ TEST_F(WebRtcSessionTest, MAYBE_TestStunError) {
   EXPECT_EQ(6u, observer_.mline_1_candidates_.size());
 }
 
-// Test session delivers no candidates gathered when constraint set to "none".
-TEST_F(WebRtcSessionTest, TestIceTransportsNone) {
-  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  InitWithIceTransport(PeerConnectionInterface::kNone);
-  SendAudioVideoStream1();
-  InitiateCall();
-  EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
-  EXPECT_EQ(0u, observer_.mline_0_candidates_.size());
-  EXPECT_EQ(0u, observer_.mline_1_candidates_.size());
-}
-
-// Test session delivers only relay candidates gathered when constaint set to
-// "relay".
-TEST_F(WebRtcSessionTest, TestIceTransportsRelay) {
-  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  ConfigureAllocatorWithTurn();
-  InitWithIceTransport(PeerConnectionInterface::kRelay);
-  SendAudioVideoStream1();
-  InitiateCall();
-  EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
-  EXPECT_EQ(2u, observer_.mline_0_candidates_.size());
-  EXPECT_EQ(2u, observer_.mline_1_candidates_.size());
-  for (size_t i = 0; i < observer_.mline_0_candidates_.size(); ++i) {
-    EXPECT_EQ(cricket::RELAY_PORT_TYPE,
-              observer_.mline_0_candidates_[i].type());
-  }
-  for (size_t i = 0; i < observer_.mline_1_candidates_.size(); ++i) {
-    EXPECT_EQ(cricket::RELAY_PORT_TYPE,
-              observer_.mline_1_candidates_[i].type());
-  }
-}
-
-// Test session delivers all candidates gathered when constaint set to "all".
-TEST_F(WebRtcSessionTest, TestIceTransportsAll) {
-  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  InitWithIceTransport(PeerConnectionInterface::kAll);
-  SendAudioVideoStream1();
-  InitiateCall();
-  EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
-  // Host + STUN. By default allocator is disabled to gather relay candidates.
-  EXPECT_EQ(4u, observer_.mline_0_candidates_.size());
-  EXPECT_EQ(4u, observer_.mline_1_candidates_.size());
-}
-
 TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
   Init();
   SessionDescriptionInterface* offer = NULL;
@@ -1871,7 +1846,7 @@ TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) {
   SessionDescriptionInterface* offer = CreateOffer();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
-  rtc::scoped_ptr<SessionDescriptionInterface> temp_offer(
+  std::unique_ptr<SessionDescriptionInterface> temp_offer(
       CreateRemoteOffer(options, cricket::SEC_ENABLED));
   JsepSessionDescription* answer =
       CreateRemoteAnswer(temp_offer.get(), options, cricket::SEC_ENABLED);
@@ -2097,7 +2072,7 @@ TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
 TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
   Init();
   SendNothing();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(offer.get());
@@ -2108,7 +2083,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
 TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
   Init();
   SendNothing();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   SessionDescriptionInterface* answer =
       CreateRemoteAnswer(offer.get());
@@ -2347,7 +2322,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
   EXPECT_TRUE_WAIT(0u < observer_.mline_0_candidates_.size(),
                    kIceCandidatesTimeout);
 
-  rtc::scoped_ptr<SessionDescriptionInterface> local_offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> local_offer(CreateOffer());
 
   ASSERT_TRUE(local_offer->candidates(kMediaContentIndex0) != NULL);
   EXPECT_LT(0u, local_offer->candidates(kMediaContentIndex0)->count());
@@ -2365,7 +2340,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
 TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
   Init();
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   // CreateOffer creates session description with the content names "audio" and
   // "video". Goal is to modify these content names and verify transport
@@ -2414,7 +2389,7 @@ TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
 // the send streams when no constraints have been set.
 TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
   Init();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   ASSERT_TRUE(offer != NULL);
   const cricket::ContentInfo* content =
@@ -2430,7 +2405,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
   Init();
   // Test Audio only offer.
   SendAudioOnlyStream2();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(offer->description());
@@ -2455,8 +2430,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
   options.offer_to_receive_audio = 0;
   options.offer_to_receive_video = 0;
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(
-      CreateOffer(options));
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
 
   ASSERT_TRUE(offer != NULL);
   const cricket::ContentInfo* content =
@@ -2474,8 +2448,7 @@ TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
   options.offer_to_receive_audio =
       RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(
-        CreateOffer(options));
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
 
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(offer->description());
@@ -2495,8 +2468,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
   options.offer_to_receive_video =
       RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(
-      CreateOffer(options));
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
 
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(offer->description());
@@ -2533,9 +2505,9 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
 TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraintsOrStreams) {
   Init();
   // Create a remote offer with audio and video content.
-  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(offer.release());
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(answer->description());
   ASSERT_TRUE(content != NULL);
@@ -2553,13 +2525,12 @@ TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
   // Create a remote offer with audio only.
   cricket::MediaSessionOptions options;
 
-  rtc::scoped_ptr<JsepSessionDescription> offer(
-      CreateRemoteOffer(options));
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
   ASSERT_TRUE(cricket::GetFirstVideoContent(offer->description()) == NULL);
   ASSERT_TRUE(cricket::GetFirstAudioContent(offer->description()) != NULL);
 
   SetRemoteDescriptionWithoutError(offer.release());
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(answer->description());
   ASSERT_TRUE(content != NULL);
@@ -2573,11 +2544,11 @@ TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
 TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
   Init();
   // Create a remote offer with audio and video content.
-  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(offer.release());
   // Test with a stream with tracks.
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(answer->description());
   ASSERT_TRUE(content != NULL);
@@ -2593,13 +2564,13 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
 TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
   Init();
   // Create a remote offer with audio and video content.
-  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(offer.release());
 
   cricket::MediaSessionOptions session_options;
   session_options.recv_audio = false;
   session_options.recv_video = false;
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(
+  std::unique_ptr<SessionDescriptionInterface> answer(
       CreateAnswer(session_options));
 
   const cricket::ContentInfo* content =
@@ -2617,7 +2588,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
 TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
   Init();
   // Create a remote offer with audio and video content.
-  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(offer.release());
 
   cricket::MediaSessionOptions options;
@@ -2626,7 +2597,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
 
   // Test with a stream with tracks.
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(options));
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer(options));
 
   // TODO(perkj): Should the direction be set to SEND_ONLY?
   const cricket::ContentInfo* content =
@@ -2648,8 +2619,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {
       RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
   options.voice_activity_detection = false;
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(
-      CreateOffer(options));
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer(options));
 
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(offer->description());
@@ -2661,12 +2631,12 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
   AddCNCodecs();
   Init();
   // Create a remote offer with audio and video content.
-  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
   SetRemoteDescriptionWithoutError(offer.release());
 
   cricket::MediaSessionOptions options;
   options.vad_enabled = false;
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(options));
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer(options));
   const cricket::ContentInfo* content =
       cricket::GetFirstAudioContent(answer->description());
   ASSERT_TRUE(content != NULL);
@@ -2788,10 +2758,10 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
 TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
   Init();
   SendAudioVideoStream1();
-  scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
   VerifyCryptoParams(offer->description());
   SetRemoteDescriptionWithoutError(offer.release());
-  scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   VerifyCryptoParams(answer->description());
 }
 
@@ -2799,7 +2769,7 @@ TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
   options_.disable_encryption = true;
   Init();
   SendAudioVideoStream1();
-  scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
   VerifyNoCryptoParams(offer->description(), false);
 }
 
@@ -2818,7 +2788,7 @@ TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
 TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
   Init();
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   std::string sdp;
   RemoveIceUfragPwdLines(offer.get(), &sdp);
@@ -2831,7 +2801,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
 // no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
 TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
   Init();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
   std::string sdp;
   RemoveIceUfragPwdLines(offer.get(), &sdp);
   SessionDescriptionInterface* modified_offer =
@@ -2844,7 +2814,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
 TEST_F(WebRtcSessionTest, TestSetLocalDescriptionInvalidIceCredentials) {
   Init();
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
   // Modifying ice ufrag and pwd in local offer with strings smaller than the
   // recommended values of 4 and 22 bytes respectively.
   SetIceUfragPwd(offer.get(), "ice", "icepwd");
@@ -2861,7 +2831,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalDescriptionInvalidIceCredentials) {
 // too short ice ufrag and pwd strings.
 TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionInvalidIceCredentials) {
   Init();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
   // Modifying ice ufrag and pwd in remote offer with strings smaller than the
   // recommended values of 4 and 22 bytes respectively.
   SetIceUfragPwd(offer.get(), "ice", "icepwd");
@@ -2879,7 +2849,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteOfferWithIceRestart) {
   Init();
 
   // Create the first offer.
-  scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
   SetIceUfragPwd(offer.get(), "0123456789012345", "abcdefghijklmnopqrstuvwx");
   cricket::Candidate candidate1(1, "udp", rtc::SocketAddress("1.1.1.1", 5000),
                                 0, "", "", "relay", 0, "");
@@ -2924,7 +2894,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithIceRestart) {
   SetLocalDescriptionWithoutError(offer);
 
   // Create the first answer.
-  scoped_ptr<JsepSessionDescription> answer(CreateRemoteAnswer(offer));
+  std::unique_ptr<JsepSessionDescription> answer(CreateRemoteAnswer(offer));
   answer->set_type(JsepSessionDescription::kPrAnswer);
   SetIceUfragPwd(answer.get(), "0123456789012345", "abcdefghijklmnopqrstuvwx");
   cricket::Candidate candidate1(1, "udp", rtc::SocketAddress("1.1.1.1", 5000),
@@ -3091,7 +3061,7 @@ TEST_F(WebRtcSessionTest, TestBalancedNoBundleInAnswer) {
   SendAudioVideoStream2();
 
   // Remove BUNDLE from the answer.
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(
+  std::unique_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
@@ -3177,7 +3147,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInAnswer) {
   SendAudioVideoStream2();
 
   // Remove BUNDLE from the answer.
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(
+  std::unique_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
@@ -3215,7 +3185,7 @@ TEST_F(WebRtcSessionTest, TestMaxBundleNoBundleInRemoteOffer) {
   SendAudioVideoStream1();
 
   // Remove BUNDLE from the offer.
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
   cricket::SessionDescription* offer_copy = offer->description()->Copy();
   offer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
   JsepSessionDescription* modified_offer =
@@ -3268,7 +3238,7 @@ TEST_F(WebRtcSessionTest, TestMaxCompatNoBundleInAnswer) {
   SendAudioVideoStream2();
 
   // Remove BUNDLE from the answer.
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(
+  std::unique_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
   answer_copy->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
@@ -3296,6 +3266,60 @@ TEST_F(WebRtcSessionTest, TestMaxBundleWithSetRemoteDescriptionFirst) {
             session_->video_rtp_transport_channel());
 }
 
+// Adding a new channel to a BUNDLE which is already connected should directly
+// assign the bundle transport to the channel, without first setting a
+// disconnected non-bundle transport and then replacing it. The application
+// should not receive any changes in the ICE state.
+TEST_F(WebRtcSessionTest, TestAddChannelToConnectedBundle) {
+  LoopbackNetworkConfiguration config;
+  LoopbackNetworkManager loopback_network_manager(this, config);
+  // Both BUNDLE and RTCP-mux need to be enabled for the ICE state to remain
+  // connected. Disabling either of these two means that we need to wait for the
+  // answer to find out if more transports are needed.
+  configuration_.bundle_policy =
+      PeerConnectionInterface::kBundlePolicyMaxBundle;
+  configuration_.rtcp_mux_policy =
+      PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  options_.disable_encryption = true;
+  Init();
+
+  // Negotiate an audio channel with MAX_BUNDLE enabled.
+  SendAudioOnlyStream2();
+  SessionDescriptionInterface* offer = CreateOffer();
+  SetLocalDescriptionWithoutError(offer);
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
+                 observer_.ice_gathering_state_, kIceCandidatesTimeout);
+  std::string sdp;
+  offer->ToString(&sdp);
+  SessionDescriptionInterface* answer = webrtc::CreateSessionDescription(
+      JsepSessionDescription::kAnswer, sdp, nullptr);
+  ASSERT_TRUE(answer != NULL);
+  SetRemoteDescriptionWithoutError(answer);
+
+  // Wait for the ICE state to stabilize.
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
+                 observer_.ice_connection_state_, kIceCandidatesTimeout);
+  observer_.ice_connection_state_history_.clear();
+
+  // Now add a video channel which should be using the same bundle transport.
+  SendAudioVideoStream2();
+  offer = CreateOffer();
+  offer->ToString(&sdp);
+  SetLocalDescriptionWithoutError(offer);
+  answer = webrtc::CreateSessionDescription(JsepSessionDescription::kAnswer,
+                                            sdp, nullptr);
+  ASSERT_TRUE(answer != NULL);
+  SetRemoteDescriptionWithoutError(answer);
+
+  // Wait for ICE state to stabilize
+  rtc::Thread::Current()->ProcessMessages(0);
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
+                 observer_.ice_connection_state_, kIceCandidatesTimeout);
+
+  // No ICE state changes are expected to happen.
+  EXPECT_EQ(0, observer_.ice_connection_state_history_.size());
+}
+
 TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
   InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyRequire);
   SendAudioVideoStream1();
@@ -3385,23 +3409,32 @@ TEST_F(WebRtcSessionTest, SetAudioPlayout) {
   EXPECT_EQ(1, volume);
 }
 
-TEST_F(WebRtcSessionTest, AudioMaxSendBitrateNotImplemented) {
-  // This test verifies that RtpParameters for audio RtpSenders cannot be
-  // changed.
-  // TODO(skvlad): Update the test after adding support for bitrate limiting in
-  // WebRtcAudioSendStream.
-
+TEST_F(WebRtcSessionTest, SetAudioMaxSendBitrate) {
   Init();
   SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
-  webrtc::RtpParameters params = session_->GetAudioRtpParameters(send_ssrc);
+  EXPECT_EQ(-1, channel->max_bps());
+  webrtc::RtpParameters params = session_->GetAudioRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetAudioRtpSendParameters(send_ssrc, params));
+
+  // Read back the parameters and verify they have been changed.
+  params = session_->GetAudioRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
-  EXPECT_EQ(0, params.encodings.size());
-  params.encodings.push_back(webrtc::RtpEncodingParameters());
-  EXPECT_FALSE(session_->SetAudioRtpParameters(send_ssrc, params));
+  // Verify that the audio channel received the new parameters.
+  params = channel->GetRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, channel->max_bps());
 }
 
 TEST_F(WebRtcSessionTest, SetAudioSend) {
@@ -3417,7 +3450,7 @@ TEST_F(WebRtcSessionTest, SetAudioSend) {
   cricket::AudioOptions options;
   options.echo_cancellation = rtc::Optional<bool>(true);
 
-  rtc::scoped_ptr<FakeAudioSource> source(new FakeAudioSource());
+  std::unique_ptr<FakeAudioSource> source(new FakeAudioSource());
   session_->SetAudioSend(send_ssrc, false, options, source.get());
   EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
   EXPECT_EQ(rtc::Optional<bool>(), channel->options().echo_cancellation);
@@ -3439,7 +3472,7 @@ TEST_F(WebRtcSessionTest, AudioSourceForLocalStream) {
   ASSERT_EQ(1u, channel->send_streams().size());
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
 
-  rtc::scoped_ptr<FakeAudioSource> source(new FakeAudioSource());
+  std::unique_ptr<FakeAudioSource> source(new FakeAudioSource());
   cricket::AudioOptions options;
   session_->SetAudioSend(send_ssrc, true, options, source.get());
   EXPECT_TRUE(source->sink() != nullptr);
@@ -3478,19 +3511,19 @@ TEST_F(WebRtcSessionTest, SetVideoMaxSendBitrate) {
   ASSERT_TRUE(channel != NULL);
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
   EXPECT_EQ(-1, channel->max_bps());
-  webrtc::RtpParameters params = session_->GetVideoRtpParameters(send_ssrc);
+  webrtc::RtpParameters params = session_->GetVideoRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
   params.encodings[0].max_bitrate_bps = 1000;
-  EXPECT_TRUE(session_->SetVideoRtpParameters(send_ssrc, params));
+  EXPECT_TRUE(session_->SetVideoRtpSendParameters(send_ssrc, params));
 
   // Read back the parameters and verify they have been changed.
-  params = session_->GetVideoRtpParameters(send_ssrc);
+  params = session_->GetVideoRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
   // Verify that the video channel received the new parameters.
-  params = channel->GetRtpParameters(send_ssrc);
+  params = channel->GetRtpSendParameters(send_ssrc);
   EXPECT_EQ(1, params.encodings.size());
   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
 
@@ -3579,7 +3612,7 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
   SendAudioVideoStream1();
   SessionDescriptionInterface* offer = CreateOffer();
   SetLocalDescriptionWithoutError(offer);
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(
+  std::unique_ptr<SessionDescriptionInterface> answer(
       CreateRemoteAnswer(session_->local_description()));
 
   cricket::SessionDescription* answer_copy = answer->description()->Copy();
@@ -3676,7 +3709,7 @@ TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
 TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
   Init();
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   // Making sure SetLocalDescription correctly sets crypto value in
   // SessionDescription object after de-serialization of sdp string. The value
@@ -3695,7 +3728,7 @@ TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
   options_.disable_encryption = true;
   Init();
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   // Making sure SetLocalDescription correctly sets crypto value in
   // SessionDescription object after de-serialization of sdp string. The value
@@ -3715,12 +3748,11 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
   Init();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
-  rtc::scoped_ptr<JsepSessionDescription> offer(
-      CreateRemoteOffer(options));
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
   SetRemoteDescriptionWithoutError(offer.release());
 
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   SetLocalDescriptionWithoutError(answer.release());
 
   // Receive an offer with new ufrag and password.
@@ -3728,18 +3760,18 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
        session_->local_description()->description()->contents()) {
     options.transport_options[content.name].ice_restart = true;
   }
-  rtc::scoped_ptr<JsepSessionDescription> updated_offer1(
+  std::unique_ptr<JsepSessionDescription> updated_offer1(
       CreateRemoteOffer(options, session_->remote_description()));
   SetRemoteDescriptionWithoutError(updated_offer1.release());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_answer1(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> updated_answer1(CreateAnswer());
 
   EXPECT_FALSE(IceUfragPwdEqual(updated_answer1->description(),
                                 session_->local_description()->description()));
 
   // Even a second answer (created before the description is set) should have
   // a new ufrag/password.
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_answer2(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> updated_answer2(CreateAnswer());
 
   EXPECT_FALSE(IceUfragPwdEqual(updated_answer2->description(),
                                 session_->local_description()->description()));
@@ -3758,34 +3790,34 @@ TEST_F(WebRtcSessionTest, TestOfferChangingOnlyUfragOrPassword) {
   options.recv_audio = true;
   options.recv_video = true;
   // Create an offer with audio and video.
-  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
   SetIceUfragPwd(offer.get(), "original_ufrag", "original_password12345");
   SetRemoteDescriptionWithoutError(offer.release());
 
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   SetLocalDescriptionWithoutError(answer.release());
 
   // Receive an offer with a new ufrag but stale password.
-  rtc::scoped_ptr<JsepSessionDescription> ufrag_changed_offer(
+  std::unique_ptr<JsepSessionDescription> ufrag_changed_offer(
       CreateRemoteOffer(options, session_->remote_description()));
   SetIceUfragPwd(ufrag_changed_offer.get(), "modified_ufrag",
                  "original_password12345");
   SetRemoteDescriptionWithoutError(ufrag_changed_offer.release());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_answer1(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> updated_answer1(CreateAnswer());
   EXPECT_FALSE(IceUfragPwdEqual(updated_answer1->description(),
                                 session_->local_description()->description()));
   SetLocalDescriptionWithoutError(updated_answer1.release());
 
   // Receive an offer with a new password but stale ufrag.
-  rtc::scoped_ptr<JsepSessionDescription> password_changed_offer(
+  std::unique_ptr<JsepSessionDescription> password_changed_offer(
       CreateRemoteOffer(options, session_->remote_description()));
   SetIceUfragPwd(password_changed_offer.get(), "modified_ufrag",
                  "modified_password12345");
   SetRemoteDescriptionWithoutError(password_changed_offer.release());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_answer2(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> updated_answer2(CreateAnswer());
   EXPECT_FALSE(IceUfragPwdEqual(updated_answer2->description(),
                                 session_->local_description()->description()));
   SetLocalDescriptionWithoutError(updated_answer2.release());
@@ -3797,20 +3829,19 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
   Init();
   cricket::MediaSessionOptions options;
   options.recv_video = true;
-  rtc::scoped_ptr<JsepSessionDescription> offer(
-      CreateRemoteOffer(options));
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
   SetRemoteDescriptionWithoutError(offer.release());
 
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   SetLocalDescriptionWithoutError(answer.release());
 
   // Receive an offer without changed ufrag or password.
-  rtc::scoped_ptr<JsepSessionDescription> updated_offer2(
+  std::unique_ptr<JsepSessionDescription> updated_offer2(
       CreateRemoteOffer(options, session_->remote_description()));
   SetRemoteDescriptionWithoutError(updated_offer2.release());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_answer2(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> updated_answer2(CreateAnswer());
 
   EXPECT_TRUE(IceUfragPwdEqual(updated_answer2->description(),
                                session_->local_description()->description()));
@@ -3827,7 +3858,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewAndOldUfragAndPassword) {
   options.recv_video = true;
   options.recv_audio = true;
   options.bundle_enabled = false;
-  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
+  std::unique_ptr<JsepSessionDescription> offer(CreateRemoteOffer(options));
 
   SetIceUfragPwd(offer.get(), cricket::MEDIA_TYPE_AUDIO, "aaaa",
                  "aaaaaaaaaaaaaaaaaaaaaa");
@@ -3836,18 +3867,18 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewAndOldUfragAndPassword) {
   SetRemoteDescriptionWithoutError(offer.release());
 
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   SetLocalDescriptionWithoutError(answer.release());
 
   // Receive an offer with new ufrag and password, but only for the video media
   // section.
-  rtc::scoped_ptr<JsepSessionDescription> updated_offer(
+  std::unique_ptr<JsepSessionDescription> updated_offer(
       CreateRemoteOffer(options, session_->remote_description()));
   SetIceUfragPwd(updated_offer.get(), cricket::MEDIA_TYPE_VIDEO, "cccc",
                  "cccccccccccccccccccccc");
   SetRemoteDescriptionWithoutError(updated_offer.release());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> updated_answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> updated_answer(CreateAnswer());
 
   EXPECT_TRUE(IceUfragPwdEqual(updated_answer->description(),
                                session_->local_description()->description(),
@@ -3940,7 +3971,7 @@ TEST_P(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
 
   InitWithDtls(GetParam());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
   EXPECT_TRUE(offer->description()->GetContentByName("data") == NULL);
   EXPECT_TRUE(offer->description()->GetTransportInfoByName("data") == NULL);
 }
@@ -3958,7 +3989,7 @@ TEST_P(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
   SetRemoteDescriptionWithoutError(offer);
 
   // Verifies the answer contains SCTP.
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   EXPECT_TRUE(answer != NULL);
   EXPECT_TRUE(answer->description()->GetContentByName("data") != NULL);
   EXPECT_TRUE(answer->description()->GetTransportInfoByName("data") != NULL);
@@ -4090,7 +4121,7 @@ TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
 
   EXPECT_TRUE(session_->waiting_for_certificate_for_testing());
   SendAudioVideoStream1();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
 
   EXPECT_TRUE(offer != NULL);
   VerifyNoCryptoParams(offer->description(), true);
@@ -4107,12 +4138,12 @@ TEST_P(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
 
   cricket::MediaSessionOptions options;
   options.recv_video = true;
-  scoped_ptr<JsepSessionDescription> offer(
-        CreateRemoteOffer(options, cricket::SEC_DISABLED));
+  std::unique_ptr<JsepSessionDescription> offer(
+      CreateRemoteOffer(options, cricket::SEC_DISABLED));
   ASSERT_TRUE(offer.get() != NULL);
   SetRemoteDescriptionWithoutError(offer.release());
 
-  rtc::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer());
+  std::unique_ptr<SessionDescriptionInterface> answer(CreateAnswer());
   EXPECT_TRUE(answer != NULL);
   VerifyNoCryptoParams(answer->description(), true);
   VerifyFingerprintStatus(answer->description(), true);
@@ -4127,7 +4158,7 @@ TEST_P(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
 
   EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000);
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
   EXPECT_TRUE(offer != NULL);
 }
 
@@ -4139,7 +4170,7 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
 
   EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000);
 
-  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
+  std::unique_ptr<SessionDescriptionInterface> offer(CreateOffer());
   EXPECT_TRUE(offer == NULL);
 }
 
@@ -4275,15 +4306,9 @@ TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) {
   SetLocalDescriptionWithoutError(answer);
 }
 
-// Flaky on Win and Mac only. See webrtc:4943
-#if defined(WEBRTC_WIN) || defined(WEBRTC_MAC)
-#define MAYBE_TestRtxRemovedByCreateAnswer DISABLED_TestRtxRemovedByCreateAnswer
-#else
-#define MAYBE_TestRtxRemovedByCreateAnswer TestRtxRemovedByCreateAnswer
-#endif
 // Tests that RTX codec is removed from the answer when it isn't supported
 // by local side.
-TEST_F(WebRtcSessionTest, MAYBE_TestRtxRemovedByCreateAnswer) {
+TEST_F(WebRtcSessionTest, TestRtxRemovedByCreateAnswer) {
   Init();
   SendAudioVideoStream1();
   std::string offer_sdp(kSdpWithRtx);
@@ -4293,14 +4318,12 @@ TEST_F(WebRtcSessionTest, MAYBE_TestRtxRemovedByCreateAnswer) {
   EXPECT_TRUE(offer->ToString(&offer_sdp));
 
   // Offer SDP contains the RTX codec.
-  EXPECT_TRUE(offer_sdp.find("rtx") != std::string::npos);
+  EXPECT_TRUE(ContainsVideoCodecWithName(offer, "rtx"));
   SetRemoteDescriptionWithoutError(offer);
 
   SessionDescriptionInterface* answer = CreateAnswer();
-  std::string answer_sdp;
-  answer->ToString(&answer_sdp);
-  // Answer SDP removes the unsupported RTX codec.
-  EXPECT_TRUE(answer_sdp.find("rtx") == std::string::npos);
+  // Answer SDP does not contain the RTX codec.
+  EXPECT_FALSE(ContainsVideoCodecWithName(answer, "rtx"));
   SetLocalDescriptionWithoutError(answer);
 }
 
diff --git a/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.cc b/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.cc
index 78840e2fac3..e88262fbdc5 100644
--- a/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.cc
+++ b/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.cc
@@ -64,7 +64,7 @@ struct CreateSessionDescriptionMsg : public rtc::MessageData {
 
   rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
   std::string error;
-  rtc::scoped_ptr<webrtc::SessionDescriptionInterface> description;
+  std::unique_ptr<webrtc::SessionDescriptionInterface> description;
 };
 }  // namespace
 
@@ -82,13 +82,13 @@ void WebRtcIdentityRequestObserver::OnSuccess(
       rtc::kPemTypeRsaPrivateKey,
       reinterpret_cast<const unsigned char*>(der_private_key.data()),
       der_private_key.length());
-  rtc::scoped_ptr<rtc::SSLIdentity> identity(
+  std::unique_ptr<rtc::SSLIdentity> identity(
       rtc::SSLIdentity::FromPEMStrings(pem_key, pem_cert));
   SignalCertificateReady(rtc::RTCCertificate::Create(std::move(identity)));
 }
 
 void WebRtcIdentityRequestObserver::OnSuccess(
-    rtc::scoped_ptr<rtc::SSLIdentity> identity) {
+    std::unique_ptr<rtc::SSLIdentity> identity) {
   SignalCertificateReady(rtc::RTCCertificate::Create(std::move(identity)));
 }
 
@@ -127,7 +127,7 @@ void WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
 WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
     rtc::Thread* signaling_thread,
     cricket::ChannelManager* channel_manager,
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+    std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     const rtc::scoped_refptr<WebRtcIdentityRequestObserver>&
         identity_request_observer,
     WebRtcSession* session,
@@ -168,7 +168,7 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
 WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
     rtc::Thread* signaling_thread,
     cricket::ChannelManager* channel_manager,
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+    std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     WebRtcSession* session,
     const std::string& session_id)
     : WebRtcSessionDescriptionFactory(
diff --git a/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.h b/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.h
index 71d083b20d9..17e2ddd3b06 100644
--- a/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.h
+++ b/chromium/third_party/webrtc/api/webrtcsessiondescriptionfactory.h
@@ -11,8 +11,11 @@
 #ifndef WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
 #define WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
 
+#include <memory>
+
 #include "webrtc/api/dtlsidentitystore.h"
 #include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/rtccertificate.h"
 #include "webrtc/p2p/base/transportdescriptionfactory.h"
@@ -37,7 +40,7 @@ class WebRtcIdentityRequestObserver : public DtlsIdentityRequestObserver,
   void OnFailure(int error) override;
   void OnSuccess(const std::string& der_cert,
                  const std::string& der_private_key) override;
-  void OnSuccess(rtc::scoped_ptr<rtc::SSLIdentity> identity) override;
+  void OnSuccess(std::unique_ptr<rtc::SSLIdentity> identity) override;
 
   sigslot::signal1<int> SignalRequestFailed;
   sigslot::signal1<const rtc::scoped_refptr<rtc::RTCCertificate>&>
@@ -82,7 +85,7 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
   WebRtcSessionDescriptionFactory(
       rtc::Thread* signaling_thread,
       cricket::ChannelManager* channel_manager,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       WebRtcSession* session,
       const std::string& session_id);
 
@@ -130,7 +133,7 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
   WebRtcSessionDescriptionFactory(
       rtc::Thread* signaling_thread,
       cricket::ChannelManager* channel_manager,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       const rtc::scoped_refptr<WebRtcIdentityRequestObserver>&
           identity_request_observer,
       WebRtcSession* session,
@@ -161,7 +164,7 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
   cricket::TransportDescriptionFactory transport_desc_factory_;
   cricket::MediaSessionDescriptionFactory session_desc_factory_;
   uint64_t session_version_;
-  const rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store_;
+  const std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store_;
   const rtc::scoped_refptr<WebRtcIdentityRequestObserver>
       identity_request_observer_;
   // TODO(jiayl): remove the dependency on session once bug 2264 is fixed.
diff --git a/chromium/third_party/webrtc/audio/audio_receive_stream.cc b/chromium/third_party/webrtc/audio/audio_receive_stream.cc
index 9c253894719..44b72d8a0ba 100644
--- a/chromium/third_party/webrtc/audio/audio_receive_stream.cc
+++ b/chromium/third_party/webrtc/audio/audio_receive_stream.cc
@@ -18,9 +18,9 @@
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
@@ -66,8 +66,6 @@ std::string AudioReceiveStream::Config::Rtp::ToString() const {
 std::string AudioReceiveStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
-  ss << ", receive_transport: "
-     << (receive_transport ? "(Transport)" : "nullptr");
   ss << ", rtcp_send_transport: "
      << (rtcp_send_transport ? "(Transport)" : "nullptr");
   ss << ", voe_channel_id: " << voe_channel_id;
@@ -95,6 +93,9 @@ AudioReceiveStream::AudioReceiveStream(
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
+
+  channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
+
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAudioLevel) {
       channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
@@ -127,6 +128,7 @@ AudioReceiveStream::AudioReceiveStream(
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
+  channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
@@ -141,45 +143,6 @@ void AudioReceiveStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
-void AudioReceiveStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-}
-
-bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  // TODO(solenberg): Tests call this function on a network thread, libjingle
-  // calls on the worker thread. We should move towards always using a network
-  // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  return false;
-}
-
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
-                                    size_t length,
-                                    const PacketTime& packet_time) {
-  // TODO(solenberg): Tests call this function on a network thread, libjingle
-  // calls on the worker thread. We should move towards always using a network
-  // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  RTPHeader header;
-  if (!rtp_header_parser_->Parse(packet, length, &header)) {
-    return false;
-  }
-
-  // Only forward if the parsed header has one of the headers necessary for
-  // bandwidth estimation. RTP timestamps has different rates for audio and
-  // video and shouldn't be mixed.
-  if (remote_bitrate_estimator_ &&
-      header.extension.hasTransportSequenceNumber) {
-    int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
-    if (packet_time.timestamp >= 0)
-      arrival_time_ms = (packet_time.timestamp + 500) / 1000;
-    size_t payload_size = length - header.headerLength;
-    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
-                                              header, false);
-  }
-  return true;
-}
-
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioReceiveStream::Stats stats;
@@ -238,6 +201,46 @@ const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
   return config_;
 }
 
+void AudioReceiveStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  return channel_proxy_->ReceivedRTCPPacket(packet, length);
+}
+
+bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                    size_t length,
+                                    const PacketTime& packet_time) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  RTPHeader header;
+  if (!rtp_header_parser_->Parse(packet, length, &header)) {
+    return false;
+  }
+
+  // Only forward if the parsed header has one of the headers necessary for
+  // bandwidth estimation. RTP timestamps has different rates for audio and
+  // video and shouldn't be mixed.
+  if (remote_bitrate_estimator_ &&
+      header.extension.hasTransportSequenceNumber) {
+    int64_t arrival_time_ms = rtc::TimeMillis();
+    if (packet_time.timestamp >= 0)
+      arrival_time_ms = (packet_time.timestamp + 500) / 1000;
+    size_t payload_size = length - header.headerLength;
+    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
+                                              header);
+  }
+
+  return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
+}
+
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
diff --git a/chromium/third_party/webrtc/audio/audio_receive_stream.h b/chromium/third_party/webrtc/audio/audio_receive_stream.h
index c9754afbf51..d99956c2ddd 100644
--- a/chromium/third_party/webrtc/audio/audio_receive_stream.h
+++ b/chromium/third_party/webrtc/audio/audio_receive_stream.h
@@ -15,6 +15,7 @@
 
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_state.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
@@ -35,20 +36,17 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
                      const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
   ~AudioReceiveStream() override;
 
-  // webrtc::ReceiveStream implementation.
+  // webrtc::AudioReceiveStream implementation.
   void Start() override;
   void Stop() override;
-  void SignalNetworkState(NetworkState state) override;
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
-  bool DeliverRtp(const uint8_t* packet,
-                  size_t length,
-                  const PacketTime& packet_time) override;
-
-  // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
-
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
 
+  void SignalNetworkState(NetworkState state);
+  bool DeliverRtcp(const uint8_t* packet, size_t length);
+  bool DeliverRtp(const uint8_t* packet,
+                  size_t length,
+                  const PacketTime& packet_time);
   const webrtc::AudioReceiveStream::Config& config() const;
 
  private:
diff --git a/chromium/third_party/webrtc/audio/audio_receive_stream_unittest.cc b/chromium/third_party/webrtc/audio/audio_receive_stream_unittest.cc
index 8703d6ed324..f6fe85850be 100644
--- a/chromium/third_party/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/chromium/third_party/webrtc/audio/audio_receive_stream_unittest.cc
@@ -98,6 +98,10 @@ struct ConfigHelper {
               .WillOnce(Return(&packet_router_));
           EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
+          EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
+              .Times(1);
+          EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
+              .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
@@ -120,6 +124,7 @@ struct ConfigHelper {
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
+  MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
 
   void SetupMockForBweFeedback(bool send_side_bwe) {
     EXPECT_CALL(congestion_controller_,
@@ -152,7 +157,7 @@ struct ConfigHelper {
  private:
   SimulatedClock simulated_clock_;
   PacketRouter packet_router_;
-  testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+  testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   MockCongestionController congestion_controller_;
   MockRemoteBitrateEstimator remote_bitrate_estimator_;
@@ -181,7 +186,7 @@ void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
                                                              shifted_value);
 }
 
-std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
+const std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
     int extension_id,
     uint32_t extension_value,
     size_t value_length) {
@@ -200,6 +205,18 @@ std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
                         extension_value, value_length);
   return header;
 }
+
+const std::vector<uint8_t> CreateRtcpSenderReport() {
+  std::vector<uint8_t> packet;
+  const size_t kRtcpSrLength = 28;  // In bytes.
+  packet.resize(kRtcpSrLength);
+  packet[0] = 0x80;  // Version 2.
+  packet[1] = 0xc8;  // PT = 200, SR.
+  // Length in number of 32-bit words - 1.
+  ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
+  return packet;
+}
 }  // namespace
 
 TEST(AudioReceiveStreamTest, ConfigToString) {
@@ -213,7 +230,7 @@ TEST(AudioReceiveStreamTest, ConfigToString) {
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}], "
       "transport_cc: off}, "
-      "receive_transport: nullptr, rtcp_send_transport: nullptr, "
+      "rtcp_send_transport: nullptr, "
       "voe_channel_id: 2}",
       config.ToString());
 }
@@ -235,7 +252,7 @@ MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
              expected_extension.transportSequenceNumber;
 }
 
-TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
+TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
   helper.SetupMockForBweFeedback(true);
@@ -252,12 +269,32 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
   EXPECT_CALL(*helper.remote_bitrate_estimator(),
               IncomingPacket(packet_time.timestamp / 1000,
                              rtp_packet.size() - kExpectedHeaderLength,
-                             VerifyHeaderExtension(expected_extension), false))
+                             VerifyHeaderExtension(expected_extension)))
       .Times(1);
+  EXPECT_CALL(*helper.channel_proxy(),
+              ReceivedRTPPacket(&rtp_packet[0],
+                                rtp_packet.size(),
+                                _))
+      .WillOnce(Return(true));
   EXPECT_TRUE(
       recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
 }
 
+TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
+  ConfigHelper helper;
+  helper.config().rtp.transport_cc = true;
+  helper.SetupMockForBweFeedback(true);
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+
+  std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
+  EXPECT_CALL(*helper.channel_proxy(),
+              ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
+      .WillOnce(Return(true));
+  EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size()));
+}
+
+
 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
diff --git a/chromium/third_party/webrtc/audio/audio_send_stream.cc b/chromium/third_party/webrtc/audio/audio_send_stream.cc
index 24afcbcf58e..c0a709e3c62 100644
--- a/chromium/third_party/webrtc/audio/audio_send_stream.cc
+++ b/chromium/third_party/webrtc/audio/audio_send_stream.cc
@@ -76,6 +76,8 @@ AudioSendStream::AudioSendStream(
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
 
+  channel_proxy_->RegisterExternalTransport(config.send_transport);
+
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAbsSendTime) {
       channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
@@ -92,6 +94,7 @@ AudioSendStream::AudioSendStream(
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
 }
 
@@ -113,18 +116,6 @@ void AudioSendStream::Stop() {
   }
 }
 
-void AudioSendStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-}
-
-bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  // TODO(solenberg): Tests call this function on a network thread, libjingle
-  // calls on the worker thread. We should move towards always using a network
-  // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  return false;
-}
-
 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
                                          int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ -215,6 +206,18 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   return stats;
 }
 
+void AudioSendStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  return channel_proxy_->ReceivedRTCPPacket(packet, length);
+}
+
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
diff --git a/chromium/third_party/webrtc/audio/audio_send_stream.h b/chromium/third_party/webrtc/audio/audio_send_stream.h
index d463b3da30f..61dd7f24b45 100644
--- a/chromium/third_party/webrtc/audio/audio_send_stream.h
+++ b/chromium/third_party/webrtc/audio/audio_send_stream.h
@@ -15,6 +15,7 @@
 
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/audio_state.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 
 namespace webrtc {
@@ -33,17 +34,15 @@ class AudioSendStream final : public webrtc::AudioSendStream {
                   CongestionController* congestion_controller);
   ~AudioSendStream() override;
 
-  // webrtc::SendStream implementation.
+  // webrtc::AudioSendStream implementation.
   void Start() override;
   void Stop() override;
-  void SignalNetworkState(NetworkState state) override;
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
-
-  // webrtc::AudioSendStream implementation.
   bool SendTelephoneEvent(int payload_type, int event,
                           int duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
+  void SignalNetworkState(NetworkState state);
+  bool DeliverRtcp(const uint8_t* packet, size_t length);
   const webrtc::AudioSendStream::Config& config() const;
 
  private:
diff --git a/chromium/third_party/webrtc/audio/audio_send_stream_unittest.cc b/chromium/third_party/webrtc/audio/audio_send_stream_unittest.cc
index c04a3de77c2..a94034c6496 100644
--- a/chromium/third_party/webrtc/audio/audio_send_stream_unittest.cc
+++ b/chromium/third_party/webrtc/audio/audio_send_stream_unittest.cc
@@ -16,7 +16,7 @@
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
-#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
@@ -89,6 +89,10 @@ struct ConfigHelper {
               .Times(1);
           EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
+          EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
+              .Times(1);
+          EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
+              .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
@@ -157,7 +161,7 @@ struct ConfigHelper {
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
-  testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+  testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   CongestionController congestion_controller_;
 };
diff --git a/chromium/third_party/webrtc/audio_receive_stream.h b/chromium/third_party/webrtc/audio_receive_stream.h
index 5254c41780a..6d72b4d3185 100644
--- a/chromium/third_party/webrtc/audio_receive_stream.h
+++ b/chromium/third_party/webrtc/audio_receive_stream.h
@@ -16,8 +16,8 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/common_types.h"
 #include "webrtc/config.h"
-#include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
@@ -31,7 +31,7 @@ class AudioSinkInterface;
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
-class AudioReceiveStream : public ReceiveStream {
+class AudioReceiveStream {
  public:
   struct Stats {
     uint32_t remote_ssrc = 0;
@@ -83,7 +83,6 @@ class AudioReceiveStream : public ReceiveStream {
       std::vector<RtpExtension> extensions;
     } rtp;
 
-    Transport* receive_transport = nullptr;
     Transport* rtcp_send_transport = nullptr;
 
     // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
@@ -104,6 +103,13 @@ class AudioReceiveStream : public ReceiveStream {
     std::map<uint8_t, AudioDecoder*> decoder_map;
   };
 
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
   virtual Stats GetStats() const = 0;
 
   // Sets an audio sink that receives unmixed audio from the receive stream.
@@ -115,6 +121,9 @@ class AudioReceiveStream : public ReceiveStream {
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
+
+ protected:
+  virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/audio_send_stream.h b/chromium/third_party/webrtc/audio_send_stream.h
index 24c3d77ab27..d8e98bb0ec9 100644
--- a/chromium/third_party/webrtc/audio_send_stream.h
+++ b/chromium/third_party/webrtc/audio_send_stream.h
@@ -11,13 +11,12 @@
 #ifndef WEBRTC_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_SEND_STREAM_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
@@ -28,7 +27,7 @@ namespace webrtc {
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
-class AudioSendStream : public SendStream {
+class AudioSendStream {
  public:
   struct Stats {
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
@@ -84,15 +83,25 @@ class AudioSendStream : public SendStream {
     // Ownership of the encoder object is transferred to Call when the config is
     // passed to Call::CreateAudioSendStream().
     // TODO(solenberg): Implement, once we configure codecs through the new API.
-    // rtc::scoped_ptr<AudioEncoder> encoder;
+    // std::unique_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
     int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
   };
 
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
   virtual Stats GetStats() const = 0;
+
+ protected:
+  virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/base/BUILD.gn b/chromium/third_party/webrtc/base/BUILD.gn
index bc6cb38023a..1e868b504e5 100644
--- a/chromium/third_party/webrtc/base/BUILD.gn
+++ b/chromium/third_party/webrtc/base/BUILD.gn
@@ -10,11 +10,12 @@ import("//build/config/crypto.gni")
 import("//build/config/ui.gni")
 import("../build/webrtc.gni")
 
-# Enable OpenSSL (BoringSSL) for iOS. This is covered in webrtc/supplement.gypi
-# for the GYP build.
 import("//build_overrides/webrtc.gni")
-if (is_ios && !build_with_chromium) {
-  use_openssl = true
+
+config("rtc_base_approved_all_dependent_config") {
+  if (is_mac && !build_with_chromium) {
+    libs = [ "Foundation.framework" ]  # needed for logging_mac.mm
+  }
 }
 
 config("rtc_base_config") {
@@ -27,12 +28,6 @@ config("rtc_base_config") {
     "FEATURE_ENABLE_SSL",
     "LOGGING=1",
   ]
-
-  if (is_posix) {
-    # TODO(henrike): issue 3307, make rtc_base build without disabling
-    # these flags.
-    cflags_cc = [ "-Wno-non-virtual-dtor" ]
-  }
 }
 
 config("rtc_base_chromium_config") {
@@ -46,45 +41,38 @@ config("openssl_config") {
   ]
 }
 
-config("ios_config") {
-  libs = [
-    "AVFoundation.framework",
-    "CFNetwork.framework",
-
-    #"Foundation.framework",  # Already included in //build/config:default_libs.
-    "Security.framework",
-    "SystemConfiguration.framework",
-
-    #"UIKit.framework",  # Already included in //build/config:default_libs.
-  ]
-}
-
-config("mac_config") {
-  libs = [
-    "Cocoa.framework",
+config("rtc_base_all_dependent_config") {
+  if (is_ios) {
+    libs = [
+      "CFNetwork.framework",
 
-    #"Foundation.framework",  # Already included in //build/config:default_libs.
-    #"IOKit.framework",  # Already included in //build/config:default_libs.
-    #"Security.framework",  # Already included in //build/config:default_libs.
-    "SystemConfiguration.framework",
-  ]
-}
-
-config("mac_x86_config") {
-  libs = [
-    #"Carbon.framework",  # Already included in //build/config:default_libs.
-  ]
+      #"Foundation.framework",  # Already in //build/config:default_libs.
+      "Security.framework",
+      "SystemConfiguration.framework",
+      "UIKit.framework",  # Already in //build/config:default_libs.
+    ]
+  }
+  if (is_mac) {
+    libs = [
+      "Cocoa.framework",
+      "Foundation.framework",
+      "IOKit.framework",
+      "Security.framework",
+      "SystemConfiguration.framework",
+    ]
+    if (current_cpu == "x86") {
+      libs = [ "Carbon.framework" ]  # Already in //build/config:default_libs.
+    }
+  }
 }
 
 if (is_linux && !build_with_chromium) {
   # Provides the same functionality as the //crypto:platform target, which
   # WebRTC cannot use as we don't sync src/crypto from Chromium.
   group("linux_system_ssl") {
-    if (use_openssl) {
-      deps = [
-        "//third_party/boringssl",
-      ]
-    }
+    deps = [
+      "//third_party/boringssl",
+    ]
   }
 }
 
@@ -98,16 +86,18 @@ if (rtc_build_ssl == 0) {
 
 # The subset of rtc_base approved for use outside of libjingle.
 static_library("rtc_base_approved") {
+  defines = []
   deps = []
   configs += [ "..:common_config" ]
   public_configs = [ "..:common_inherited_config" ]
+  all_dependent_configs = [ ":rtc_base_approved_all_dependent_config" ]
 
   sources = [
     "array_view.h",
     "atomicops.h",
+    "bind.h",
     "bitbuffer.cc",
     "bitbuffer.h",
-    "buffer.cc",
     "buffer.h",
     "bufferqueue.cc",
     "bufferqueue.h",
@@ -133,6 +123,7 @@ static_library("rtc_base_approved") {
     "md5digest.cc",
     "md5digest.h",
     "mod_ops.h",
+    "onetimeevent.h",
     "optional.h",
     "platform_file.cc",
     "platform_file.h",
@@ -148,7 +139,6 @@ static_library("rtc_base_approved") {
     "refcount.h",
     "safe_conversions.h",
     "safe_conversions_impl.h",
-    "scoped_ptr.h",
     "scoped_ref_ptr.h",
     "stringencode.cc",
     "stringencode.h",
@@ -157,6 +147,8 @@ static_library("rtc_base_approved") {
     "swap_queue.h",
     "systeminfo.cc",
     "systeminfo.h",
+    "task_queue.h",
+    "task_queue_posix.h",
     "template_util.h",
     "thread_annotations.h",
     "thread_checker.h",
@@ -179,8 +171,30 @@ static_library("rtc_base_approved") {
     sources += [
       "logging.cc",
       "logging.h",
+      "logging_mac.mm",
     ]
   }
+
+  if (rtc_build_libevent) {
+    deps += [ "//base/third_party/libevent" ]
+  }
+  if (rtc_enable_libevent) {
+    sources += [
+      "task_queue_libevent.cc",
+      "task_queue_posix.cc",
+    ]
+  } else {
+    # If not libevent, fall back to the other task queues.
+    if (is_mac || is_ios) {
+      sources += [
+        "task_queue_gcd.cc",
+        "task_queue_posix.cc",
+      ]
+    }
+    if (is_win) {
+      sources += [ "task_queue_win.cc" ]
+    }
+  }
 }
 
 static_library("rtc_base") {
@@ -206,6 +220,8 @@ static_library("rtc_base") {
     ":rtc_base_config",
   ]
 
+  all_dependent_configs = [ ":rtc_base_all_dependent_config" ]
+
   defines = [ "LOGGING=1" ]
 
   sources = [
@@ -274,7 +290,17 @@ static_library("rtc_base") {
     "network.h",
     "networkmonitor.cc",
     "networkmonitor.h",
+    "nullsocketserver.cc",
     "nullsocketserver.h",
+    "openssl.h",
+    "openssladapter.cc",
+    "openssladapter.h",
+    "openssldigest.cc",
+    "openssldigest.h",
+    "opensslidentity.cc",
+    "opensslidentity.h",
+    "opensslstreamadapter.cc",
+    "opensslstreamadapter.h",
     "pathutils.cc",
     "pathutils.h",
     "physicalsocketserver.cc",
@@ -287,6 +313,8 @@ static_library("rtc_base") {
     "ratelimiter.h",
     "rtccertificate.cc",
     "rtccertificate.h",
+    "rtccertificategenerator.cc",
+    "rtccertificategenerator.h",
     "sha1.cc",
     "sha1.h",
     "sha1digest.cc",
@@ -354,7 +382,6 @@ static_library("rtc_base") {
     sources += [
       "bandwidthsmoother.cc",
       "bandwidthsmoother.h",
-      "bind.h",
       "callback.h",
       "fileutils_mock.h",
       "httpserver.cc",
@@ -460,29 +487,12 @@ static_library("rtc_base") {
   configs += [ "//build/config/compiler:no_chromium_code" ]
   if (!is_win) {
     cflags += [ "-Wno-uninitialized" ]
-    cflags_cc += [ "-Wno-non-virtual-dtor" ]
   }
 
-  # TODO(kjellander): The use_openssl block should really go away in order for
-  # the GN build to be similar to the GYP build. See http://crbug.com/601042 for
-  # more details.
-  if (use_openssl) {
-    if (rtc_build_ssl) {
-      deps += [ "//third_party/boringssl" ]
-    } else {
-      configs += [ "external_ssl_library" ]
-    }
-    sources += [
-      "openssl.h",
-      "openssladapter.cc",
-      "openssladapter.h",
-      "openssldigest.cc",
-      "openssldigest.h",
-      "opensslidentity.cc",
-      "opensslidentity.h",
-      "opensslstreamadapter.cc",
-      "opensslstreamadapter.h",
-    ]
+  if (rtc_build_ssl) {
+    deps += [ "//third_party/boringssl" ]
+  } else {
+    configs += [ "external_ssl_library" ]
   }
 
   if (is_android) {
@@ -498,9 +508,6 @@ static_library("rtc_base") {
   }
 
   if (is_ios || is_mac) {
-    if (is_ios) {
-      all_dependent_configs = [ ":ios_config" ]
-    }
     sources += [
       "maccocoathreadhelper.h",
       "maccocoathreadhelper.mm",
@@ -539,12 +546,6 @@ static_library("rtc_base") {
       "macutils.cc",
       "macutils.h",
     ]
-
-    all_dependent_configs = [ ":mac_config" ]
-
-    if (current_cpu == "x86") {
-      all_dependent_configs += [ ":mac_x86_config" ]
-    }
   }
 
   if (is_win) {
@@ -612,39 +613,6 @@ static_library("rtc_base") {
   }
 }
 
-if (is_ios) {
-  source_set("rtc_base_objc") {
-    deps = [
-      ":rtc_base",
-    ]
-    cflags = [ "-fobjc-arc" ]
-    configs += [ "..:common_config" ]
-    public_configs = [ "..:common_inherited_config" ]
-
-    sources = [
-      "objc/NSString+StdString.h",
-      "objc/NSString+StdString.mm",
-      "objc/RTCCameraPreviewView.h",
-      "objc/RTCCameraPreviewView.m",
-      "objc/RTCDispatcher.h",
-      "objc/RTCDispatcher.m",
-      "objc/RTCFieldTrials.h",
-      "objc/RTCFieldTrials.mm",
-      "objc/RTCFileLogger.h",
-      "objc/RTCFileLogger.mm",
-      "objc/RTCLogging.h",
-      "objc/RTCLogging.mm",
-      "objc/RTCMacros.h",
-      "objc/RTCSSLAdapter.h",
-      "objc/RTCSSLAdapter.mm",
-      "objc/RTCTracing.h",
-      "objc/RTCTracing.mm",
-      "objc/RTCUIApplication.h",
-      "objc/RTCUIApplication.mm",
-    ]
-  }
-}
-
 source_set("gtest_prod") {
   sources = [
     "gtest_prod_util.h",
diff --git a/chromium/third_party/webrtc/base/DEPS b/chromium/third_party/webrtc/base/DEPS
index add7f38be50..bb76adfe31c 100644
--- a/chromium/third_party/webrtc/base/DEPS
+++ b/chromium/third_party/webrtc/base/DEPS
@@ -1,4 +1,5 @@
 include_rules = [
+  "+base/third_party/libevent",
   "+json",
   "+third_party/jsoncpp",
   "+webrtc/system_wrappers",
diff --git a/chromium/third_party/webrtc/base/OWNERS b/chromium/third_party/webrtc/base/OWNERS
index 2f400904c6c..b5d02af6b8c 100644
--- a/chromium/third_party/webrtc/base/OWNERS
+++ b/chromium/third_party/webrtc/base/OWNERS
@@ -15,4 +15,6 @@ per-file *.gyp=*
 per-file *.gypi=*
 
 per-file BUILD.gn=kjellander@webrtc.org
+per-file rate_statistics*=sprang@webrtc.org
+per-file rate_statistics*=stefan@webrtc.org
 
diff --git a/chromium/third_party/webrtc/base/array_view.h b/chromium/third_party/webrtc/base/array_view.h
index a7ca66cc95d..868009631f8 100644
--- a/chromium/third_party/webrtc/base/array_view.h
+++ b/chromium/third_party/webrtc/base/array_view.h
@@ -56,6 +56,7 @@ namespace rtc {
 //   Contains17(arr);                             // C array
 //   Contains17(arr);                             // std::vector
 //   Contains17(rtc::ArrayView<int>(arr, size));  // pointer + size
+//   Contains17(nullptr);                         // nullptr -> empty ArrayView
 //   ...
 //
 // One important point is that ArrayView<T> and ArrayView<const T> are
@@ -73,6 +74,7 @@ class ArrayView final {
  public:
   // Construct an empty ArrayView.
   ArrayView() : ArrayView(static_cast<T*>(nullptr), 0) {}
+  ArrayView(std::nullptr_t) : ArrayView() {}
 
   // Construct an ArrayView for a (pointer,size) pair.
   template <typename U>
diff --git a/chromium/third_party/webrtc/base/asyncinvoker.h b/chromium/third_party/webrtc/base/asyncinvoker.h
index a35133706a0..76e5d922e60 100644
--- a/chromium/third_party/webrtc/base/asyncinvoker.h
+++ b/chromium/third_party/webrtc/base/asyncinvoker.h
@@ -13,6 +13,7 @@
 
 #include "webrtc/base/asyncinvoker-inl.h"
 #include "webrtc/base/bind.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/scopedptrcollection.h"
 #include "webrtc/base/thread.h"
diff --git a/chromium/third_party/webrtc/base/asyncpacketsocket.h b/chromium/third_party/webrtc/base/asyncpacketsocket.h
index 949ec67c839..f18839ed373 100644
--- a/chromium/third_party/webrtc/base/asyncpacketsocket.h
+++ b/chromium/third_party/webrtc/base/asyncpacketsocket.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_
 #define WEBRTC_BASE_ASYNCPACKETSOCKET_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
diff --git a/chromium/third_party/webrtc/base/asynctcpsocket.cc b/chromium/third_party/webrtc/base/asynctcpsocket.cc
index 65ec0f287bb..9ba46d7abc9 100644
--- a/chromium/third_party/webrtc/base/asynctcpsocket.cc
+++ b/chromium/third_party/webrtc/base/asynctcpsocket.cc
@@ -13,6 +13,7 @@
 #include <string.h>
 
 #include <algorithm>
+#include <memory>
 
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/checks.h"
@@ -44,7 +45,7 @@ AsyncSocket* AsyncTCPSocketBase::ConnectSocket(
     rtc::AsyncSocket* socket,
     const rtc::SocketAddress& bind_address,
     const rtc::SocketAddress& remote_address) {
-  rtc::scoped_ptr<rtc::AsyncSocket> owned_socket(socket);
+  std::unique_ptr<rtc::AsyncSocket> owned_socket(socket);
   if (socket->Bind(bind_address) < 0) {
     LOG(LS_ERROR) << "Bind() failed with error " << socket->GetError();
     return NULL;
@@ -295,7 +296,7 @@ int AsyncTCPSocket::Send(const void *pv, size_t cb,
     return res;
   }
 
-  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
   SignalSentPacket(this, sent_packet);
 
   // We claim to have sent the whole thing, even if we only sent partial
diff --git a/chromium/third_party/webrtc/base/asynctcpsocket.h b/chromium/third_party/webrtc/base/asynctcpsocket.h
index ea314931d65..cd5c104d88e 100644
--- a/chromium/third_party/webrtc/base/asynctcpsocket.h
+++ b/chromium/third_party/webrtc/base/asynctcpsocket.h
@@ -11,9 +11,11 @@
 #ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_
 #define WEBRTC_BASE_ASYNCTCPSOCKET_H_
 
+#include <memory>
+
 #include "webrtc/base/asyncpacketsocket.h"
 #include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/socketfactory.h"
 
 namespace rtc {
@@ -70,7 +72,7 @@ class AsyncTCPSocketBase : public AsyncPacketSocket {
   void OnWriteEvent(AsyncSocket* socket);
   void OnCloseEvent(AsyncSocket* socket, int error);
 
-  scoped_ptr<AsyncSocket> socket_;
+  std::unique_ptr<AsyncSocket> socket_;
   bool listen_;
   Buffer inbuf_;
   Buffer outbuf_;
diff --git a/chromium/third_party/webrtc/base/asynctcpsocket_unittest.cc b/chromium/third_party/webrtc/base/asynctcpsocket_unittest.cc
index b9317586406..592b61d536e 100644
--- a/chromium/third_party/webrtc/base/asynctcpsocket_unittest.cc
+++ b/chromium/third_party/webrtc/base/asynctcpsocket_unittest.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/asynctcpsocket.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/virtualsocketserver.h"
 
 namespace rtc {
@@ -37,10 +37,10 @@ class AsyncTCPSocketTest
   }
 
  protected:
-  scoped_ptr<PhysicalSocketServer> pss_;
-  scoped_ptr<VirtualSocketServer> vss_;
+  std::unique_ptr<PhysicalSocketServer> pss_;
+  std::unique_ptr<VirtualSocketServer> vss_;
   AsyncSocket* socket_;
-  scoped_ptr<AsyncTCPSocket> tcp_socket_;
+  std::unique_ptr<AsyncTCPSocket> tcp_socket_;
   bool ready_to_send_;
 };
 
diff --git a/chromium/third_party/webrtc/base/asyncudpsocket.cc b/chromium/third_party/webrtc/base/asyncudpsocket.cc
index 4e807330843..fc7d88712cc 100644
--- a/chromium/third_party/webrtc/base/asyncudpsocket.cc
+++ b/chromium/third_party/webrtc/base/asyncudpsocket.cc
@@ -18,7 +18,7 @@ static const int BUF_SIZE = 64 * 1024;
 AsyncUDPSocket* AsyncUDPSocket::Create(
     AsyncSocket* socket,
     const SocketAddress& bind_address) {
-  scoped_ptr<AsyncSocket> owned_socket(socket);
+  std::unique_ptr<AsyncSocket> owned_socket(socket);
   if (socket->Bind(bind_address) < 0) {
     LOG(LS_ERROR) << "Bind() failed with error " << socket->GetError();
     return NULL;
@@ -59,7 +59,7 @@ SocketAddress AsyncUDPSocket::GetRemoteAddress() const {
 
 int AsyncUDPSocket::Send(const void *pv, size_t cb,
                          const rtc::PacketOptions& options) {
-  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
   int ret = socket_->Send(pv, cb);
   SignalSentPacket(this, sent_packet);
   return ret;
@@ -68,7 +68,7 @@ int AsyncUDPSocket::Send(const void *pv, size_t cb,
 int AsyncUDPSocket::SendTo(const void *pv, size_t cb,
                            const SocketAddress& addr,
                            const rtc::PacketOptions& options) {
-  rtc::SentPacket sent_packet(options.packet_id, rtc::Time());
+  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
   int ret = socket_->SendTo(pv, cb, addr);
   SignalSentPacket(this, sent_packet);
   return ret;
diff --git a/chromium/third_party/webrtc/base/asyncudpsocket.h b/chromium/third_party/webrtc/base/asyncudpsocket.h
index 4b47007ed79..aa6a9042b47 100644
--- a/chromium/third_party/webrtc/base/asyncudpsocket.h
+++ b/chromium/third_party/webrtc/base/asyncudpsocket.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_BASE_ASYNCUDPSOCKET_H_
 #define WEBRTC_BASE_ASYNCUDPSOCKET_H_
 
+#include <memory>
+
 #include "webrtc/base/asyncpacketsocket.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketfactory.h"
 
 namespace rtc {
@@ -56,7 +57,7 @@ class AsyncUDPSocket : public AsyncPacketSocket {
   // Called when the underlying socket is ready to send.
   void OnWriteEvent(AsyncSocket* socket);
 
-  scoped_ptr<AsyncSocket> socket_;
+  std::unique_ptr<AsyncSocket> socket_;
   char* buf_;
   size_t size_;
 };
diff --git a/chromium/third_party/webrtc/base/asyncudpsocket_unittest.cc b/chromium/third_party/webrtc/base/asyncudpsocket_unittest.cc
index bd65940fcb8..99220056b47 100644
--- a/chromium/third_party/webrtc/base/asyncudpsocket_unittest.cc
+++ b/chromium/third_party/webrtc/base/asyncudpsocket_unittest.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/virtualsocketserver.h"
 
 namespace rtc {
@@ -37,10 +37,10 @@ class AsyncUdpSocketTest
   }
 
  protected:
-  scoped_ptr<PhysicalSocketServer> pss_;
-  scoped_ptr<VirtualSocketServer> vss_;
+  std::unique_ptr<PhysicalSocketServer> pss_;
+  std::unique_ptr<VirtualSocketServer> vss_;
   AsyncSocket* socket_;
-  scoped_ptr<AsyncUDPSocket> udp_socket_;
+  std::unique_ptr<AsyncUDPSocket> udp_socket_;
   bool ready_to_send_;
 };
 
diff --git a/chromium/third_party/webrtc/base/base.gyp b/chromium/third_party/webrtc/base/base.gyp
index a5f0f7b44d4..9e55efc8073 100644
--- a/chromium/third_party/webrtc/base/base.gyp
+++ b/chromium/third_party/webrtc/base/base.gyp
@@ -22,52 +22,6 @@
         }],
       ],
     }],
-    ['OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")', {
-      'targets': [
-        {
-          'target_name': 'rtc_base_objc',
-          'type': 'static_library',
-          'includes': [ '../build/objc_common.gypi' ],
-          'dependencies': [
-            'rtc_base',
-          ],
-          'sources': [
-            'objc/NSString+StdString.h',
-            'objc/NSString+StdString.mm',
-            'objc/RTCDispatcher.h',
-            'objc/RTCDispatcher.m',
-            'objc/RTCFieldTrials.h',
-            'objc/RTCFieldTrials.mm',
-            'objc/RTCFileLogger.h',
-            'objc/RTCFileLogger.mm',
-            'objc/RTCLogging.h',
-            'objc/RTCLogging.mm',
-            'objc/RTCMacros.h',
-            'objc/RTCSSLAdapter.h',
-            'objc/RTCSSLAdapter.mm',
-            'objc/RTCTracing.h',
-            'objc/RTCTracing.mm',
-          ],
-          'conditions': [
-            ['OS=="ios"', {
-              'sources': [
-                'objc/RTCCameraPreviewView.h',
-                'objc/RTCCameraPreviewView.m',
-                'objc/RTCUIApplication.h',
-                'objc/RTCUIApplication.mm',
-              ],
-              'all_dependent_settings': {
-                'xcode_settings': {
-                  'OTHER_LDFLAGS': [
-                    '-framework AVFoundation',
-                  ],
-                },
-              },
-            }],
-          ],
-        }
-      ],
-    }], # OS=="ios"
   ],
   'targets': [
     {
@@ -77,9 +31,9 @@
       'sources': [
         'array_view.h',
         'atomicops.h',
+        'bind.h',
         'bitbuffer.cc',
         'bitbuffer.h',
-        'buffer.cc',
         'buffer.h',
         'bufferqueue.cc',
         'bufferqueue.h',
@@ -105,6 +59,7 @@
         'md5digest.cc',
         'md5digest.h',
         'mod_ops.h',
+        'onetimeevent.h',
         'optional.h',
         'platform_file.cc',
         'platform_file.h',
@@ -120,7 +75,6 @@
         'refcount.h',
         'safe_conversions.h',
         'safe_conversions_impl.h',
-        'scoped_ptr.h',
         'scoped_ref_ptr.h',
         'stringencode.cc',
         'stringencode.h',
@@ -129,6 +83,8 @@
         'swap_queue.h',
         'systeminfo.cc',
         'systeminfo.h',
+        'task_queue.h',
+        'task_queue_posix.h',
         'template_util.h',
         'thread_annotations.h',
         'thread_checker.h',
@@ -154,8 +110,43 @@
           'sources': [
             'logging.cc',
             'logging.h',
+            'logging_mac.mm',
+          ],
+        }],
+        ['build_libevent==1', {
+          'dependencies': [
+            '<(DEPTH)/base/third_party/libevent/libevent.gyp:libevent',
           ],
         }],
+        ['enable_libevent==1', {
+          'sources': [
+            'task_queue_libevent.cc',
+            'task_queue_posix.cc',
+          ],
+        }, {
+          # If not libevent, fall back to the other task queues.
+          'conditions': [
+            ['OS=="mac" or OS=="ios"', {
+             'sources': [
+               'task_queue_gcd.cc',
+               'task_queue_posix.cc',
+             ],
+            }],
+            ['OS=="win"', {
+              'sources': [ 'task_queue_win.cc' ],
+            }]
+          ],
+        }],
+        ['OS=="mac" and build_with_chromium==0', {
+          'all_dependent_settings': {
+            'xcode_settings': {
+              'OTHER_LDFLAGS': [
+                # needed for logging_mac.mm
+                '-framework Foundation',
+              ],
+            },
+          },
+        }], # OS=="mac" and build_with_chromium==0
       ],
     },
     {
@@ -240,6 +231,7 @@
         'network.h',
         'networkmonitor.cc',
         'networkmonitor.h',
+        'nullsocketserver.cc',
         'nullsocketserver.h',
         'openssl.h',
         'openssladapter.cc',
@@ -262,6 +254,8 @@
         'ratelimiter.h',
         'rtccertificate.cc',
         'rtccertificate.h',
+        'rtccertificategenerator.cc',
+        'rtccertificategenerator.h',
         'sha1.cc',
         'sha1.h',
         'sha1digest.cc',
@@ -319,13 +313,7 @@
         '-Wextra',
         '-Wall',
       ],
-      'cflags_cc!': [
-        '-Wnon-virtual-dtor',
-      ],
       'direct_dependent_settings': {
-        'cflags_cc!': [
-          '-Wnon-virtual-dtor',
-        ],
         'defines': [
           'FEATURE_ENABLE_SSL',
           'SSL_USE_OPENSSL',
@@ -361,7 +349,6 @@
           'sources': [
             'bandwidthsmoother.cc',
             'bandwidthsmoother.h',
-            'bind.h',
             'callback.h',
             'fileutils_mock.h',
             'httpserver.cc',
diff --git a/chromium/third_party/webrtc/base/base_tests.gyp b/chromium/third_party/webrtc/base/base_tests.gyp
index caef35c385a..063e8e164f4 100644
--- a/chromium/third_party/webrtc/base/base_tests.gyp
+++ b/chromium/third_party/webrtc/base/base_tests.gyp
@@ -79,6 +79,7 @@
           'multipart_unittest.cc',
           'nat_unittest.cc',
           'network_unittest.cc',
+          'onetimeevent_unittest.cc',
           'optional_unittest.cc',
           'optionsfile_unittest.cc',
           'pathutils_unittest.cc',
@@ -92,7 +93,8 @@
           'ratetracker_unittest.cc',
           'referencecountedsingletonfactory_unittest.cc',
           'rollingaccumulator_unittest.cc',
-          'rtccertificate_unittests.cc',
+          'rtccertificate_unittest.cc',
+          'rtccertificategenerator_unittest.cc',
           'scopedptrcollection_unittest.cc',
           'sha1digest_unittest.cc',
           'sharedexclusivelock_unittest.cc',
@@ -106,6 +108,7 @@
           'swap_queue_unittest.cc',
           # TODO(ronghuawu): Reenable this test.
           # 'systeminfo_unittest.cc',
+          'task_queue_unittest.cc',
           'task_unittest.cc',
           'testclient_unittest.cc',
           'thread_checker_unittest.cc',
diff --git a/chromium/third_party/webrtc/base/buffer.cc b/chromium/third_party/webrtc/base/buffer.cc
index 6051e6db959..79e48bd7a74 100644
--- a/chromium/third_party/webrtc/base/buffer.cc
+++ b/chromium/third_party/webrtc/base/buffer.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *  Copyright 2016 The WebRTC project authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -8,36 +8,5 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/base/buffer.h"
-
-#include <algorithm>
-#include <utility>
-
-namespace rtc {
-
-Buffer::Buffer() : size_(0), capacity_(0), data_(nullptr) {
-  RTC_DCHECK(IsConsistent());
-}
-
-Buffer::Buffer(Buffer&& buf)
-    : size_(buf.size()),
-      capacity_(buf.capacity()),
-      data_(std::move(buf.data_)) {
-  RTC_DCHECK(IsConsistent());
-  buf.OnMovedFrom();
-}
-
-Buffer::Buffer(size_t size) : Buffer(size, size) {
-}
-
-Buffer::Buffer(size_t size, size_t capacity)
-    : size_(size),
-      capacity_(std::max(size, capacity)),
-      data_(new uint8_t[capacity_]) {
-  RTC_DCHECK(IsConsistent());
-}
-
-// Note: The destructor works even if the buffer has been moved from.
-Buffer::~Buffer() = default;
-
-};  // namespace rtc
+// This is a dummy file, added because something in the Chromium build claims
+// to need it. We should fix that, and then remove it (bug 5845).
diff --git a/chromium/third_party/webrtc/base/buffer.h b/chromium/third_party/webrtc/base/buffer.h
index f007929a294..b8b8fc0e169 100644
--- a/chromium/third_party/webrtc/base/buffer.h
+++ b/chromium/third_party/webrtc/base/buffer.h
@@ -13,79 +13,114 @@
 
 #include <cstring>
 #include <memory>
+#include <type_traits>
 #include <utility>
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/constructormagic.h"
 
 namespace rtc {
 
 namespace internal {
 
-// (Internal; please don't use outside this file.) ByteType<T>::t is int if T
-// is uint8_t, int8_t, or char; otherwise, it's a compilation error. Use like
-// this:
-//
-//   template <typename T, typename ByteType<T>::t = 0>
-//   void foo(T* x);
-//
-// to let foo<T> be defined only for byte-sized integers.
-template <typename T>
-struct ByteType {
- private:
-  static int F(uint8_t*);
-  static int F(int8_t*);
-  static int F(char*);
-
- public:
-  using t = decltype(F(static_cast<T*>(nullptr)));
+// (Internal; please don't use outside this file.) Determines if elements of
+// type U are compatible with a BufferT<T>. For most types, we just ignore
+// top-level const and forbid top-level volatile and require T and U to be
+// otherwise equal, but all byte-sized integers (notably char, int8_t, and
+// uint8_t) are compatible with each other. (Note: We aim to get rid of this
+// behavior, and treat all types the same.)
+template <typename T, typename U>
+struct BufferCompat {
+  static constexpr bool value =
+      !std::is_volatile<U>::value &&
+      ((std::is_integral<T>::value && sizeof(T) == 1)
+           ? (std::is_integral<U>::value && sizeof(U) == 1)
+           : (std::is_same<T, typename std::remove_const<U>::type>::value));
 };
 
 }  // namespace internal
 
 // Basic buffer class, can be grown and shrunk dynamically.
-// Unlike std::string/vector, does not initialize data when expanding capacity.
-class Buffer {
+// Unlike std::string/vector, does not initialize data when increasing size.
+template <typename T>
+class BufferT {
+  // We want T's destructor and default constructor to be trivial, i.e. perform
+  // no action, so that we don't have to touch the memory we allocate and
+  // deallocate. And we want T to be trivially copyable, so that we can copy T
+  // instances with std::memcpy. This is precisely the definition of a trivial
+  // type.
+  static_assert(std::is_trivial<T>::value, "T must be a trivial type.");
+
+  // This class relies heavily on being able to mutate its data.
+  static_assert(!std::is_const<T>::value, "T may not be const");
+
  public:
-  Buffer();                   // An empty buffer.
-  Buffer(Buffer&& buf);       // Move contents from an existing buffer.
-
-  // Construct a buffer with the specified number of uninitialized bytes.
-  explicit Buffer(size_t size);
-  Buffer(size_t size, size_t capacity);
-
-  // Construct a buffer and copy the specified number of bytes into it. The
-  // source array may be (const) uint8_t*, int8_t*, or char*.
-  template <typename T, typename internal::ByteType<T>::t = 0>
-  Buffer(const T* data, size_t size)
-      : Buffer(data, size, size) {}
-
-  template <typename T, typename internal::ByteType<T>::t = 0>
-  Buffer(const T* data, size_t size, size_t capacity)
-      : Buffer(size, capacity) {
-    std::memcpy(data_.get(), data, size);
+  // An empty BufferT.
+  BufferT() : size_(0), capacity_(0), data_(nullptr) {
+    RTC_DCHECK(IsConsistent());
   }
 
-  // Construct a buffer from the contents of an array.
-  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
-  Buffer(const T(&array)[N])
-      : Buffer(array, N) {}
+  // Disable copy construction and copy assignment, since copying a buffer is
+  // expensive enough that we want to force the user to be explicit about it.
+  BufferT(const BufferT&) = delete;
+  BufferT& operator=(const BufferT&) = delete;
 
-  ~Buffer();
+  BufferT(BufferT&& buf)
+      : size_(buf.size()),
+        capacity_(buf.capacity()),
+        data_(std::move(buf.data_)) {
+    RTC_DCHECK(IsConsistent());
+    buf.OnMovedFrom();
+  }
+
+  // Construct a buffer with the specified number of uninitialized elements.
+  explicit BufferT(size_t size) : BufferT(size, size) {}
+
+  BufferT(size_t size, size_t capacity)
+      : size_(size),
+        capacity_(std::max(size, capacity)),
+        data_(new T[capacity_]) {
+    RTC_DCHECK(IsConsistent());
+  }
 
-  // Get a pointer to the data. Just .data() will give you a (const) uint8_t*,
-  // but you may also use .data<int8_t>() and .data<char>().
-  template <typename T = uint8_t, typename internal::ByteType<T>::t = 0>
-  const T* data() const {
+  // Construct a buffer and copy the specified number of elements into it.
+  template <typename U,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  BufferT(const U* data, size_t size) : BufferT(data, size, size) {}
+
+  template <typename U,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  BufferT(U* data, size_t size, size_t capacity) : BufferT(size, capacity) {
+    static_assert(sizeof(T) == sizeof(U), "");
+    std::memcpy(data_.get(), data, size * sizeof(U));
+  }
+
+  // Construct a buffer from the contents of an array.
+  template <typename U,
+            size_t N,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  BufferT(U (&array)[N]) : BufferT(array, N) {}
+
+  // Get a pointer to the data. Just .data() will give you a (const) T*, but if
+  // T is a byte-sized integer, you may also use .data<U>() for any other
+  // byte-sized integer U.
+  template <typename U = T,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  const U* data() const {
     RTC_DCHECK(IsConsistent());
-    return reinterpret_cast<T*>(data_.get());
+    return reinterpret_cast<U*>(data_.get());
   }
 
-  template <typename T = uint8_t, typename internal::ByteType<T>::t = 0>
-  T* data() {
+  template <typename U = T,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  U* data() {
     RTC_DCHECK(IsConsistent());
-    return reinterpret_cast<T*>(data_.get());
+    return reinterpret_cast<U*>(data_.get());
   }
 
   size_t size() const {
@@ -98,7 +133,7 @@ class Buffer {
     return capacity_;
   }
 
-  Buffer& operator=(Buffer&& buf) {
+  BufferT& operator=(BufferT&& buf) {
     RTC_DCHECK(IsConsistent());
     RTC_DCHECK(buf.IsConsistent());
     size_ = buf.size_;
@@ -108,94 +143,120 @@ class Buffer {
     return *this;
   }
 
-  bool operator==(const Buffer& buf) const {
+  bool operator==(const BufferT& buf) const {
     RTC_DCHECK(IsConsistent());
-    return size_ == buf.size() && memcmp(data_.get(), buf.data(), size_) == 0;
+    if (size_ != buf.size_) {
+      return false;
+    }
+    if (std::is_integral<T>::value) {
+      // Optimization.
+      return std::memcmp(data_.get(), buf.data_.get(), size_ * sizeof(T)) == 0;
+    }
+    for (size_t i = 0; i < size_; ++i) {
+      if (data_[i] != buf.data_[i]) {
+        return false;
+      }
+    }
+    return true;
   }
 
-  bool operator!=(const Buffer& buf) const { return !(*this == buf); }
+  bool operator!=(const BufferT& buf) const { return !(*this == buf); }
 
-  uint8_t& operator[](size_t index) {
+  T& operator[](size_t index) {
     RTC_DCHECK_LT(index, size_);
     return data()[index];
   }
 
-  uint8_t operator[](size_t index) const {
+  T operator[](size_t index) const {
     RTC_DCHECK_LT(index, size_);
     return data()[index];
   }
 
   // The SetData functions replace the contents of the buffer. They accept the
   // same input types as the constructors.
-  template <typename T, typename internal::ByteType<T>::t = 0>
-  void SetData(const T* data, size_t size) {
+  template <typename U,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  void SetData(const U* data, size_t size) {
     RTC_DCHECK(IsConsistent());
     size_ = 0;
     AppendData(data, size);
   }
 
-  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
-  void SetData(const T(&array)[N]) {
+  template <typename U,
+            size_t N,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  void SetData(const U (&array)[N]) {
     SetData(array, N);
   }
 
-  void SetData(const Buffer& buf) { SetData(buf.data(), buf.size()); }
+  void SetData(const BufferT& buf) { SetData(buf.data(), buf.size()); }
 
-  // Replace the data in the buffer with at most |max_bytes| of data, using the
-  // function |setter|, which should have the following signature:
-  //   size_t setter(ArrayView<T> view)
+  // Replace the data in the buffer with at most |max_elements| of data, using
+  // the function |setter|, which should have the following signature:
+  //   size_t setter(ArrayView<U> view)
   // |setter| is given an appropriately typed ArrayView of the area in which to
   // write the data (i.e. starting at the beginning of the buffer) and should
-  // return the number of bytes actually written. This number must be <=
-  // |max_bytes|.
-  template <typename T = uint8_t, typename F,
-            typename internal::ByteType<T>::t = 0>
-  size_t SetData(size_t max_bytes, F&& setter) {
+  // return the number of elements actually written. This number must be <=
+  // |max_elements|.
+  template <typename U = T,
+            typename F,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  size_t SetData(size_t max_elements, F&& setter) {
     RTC_DCHECK(IsConsistent());
     size_ = 0;
-    return AppendData<T>(max_bytes, std::forward<F>(setter));
+    return AppendData<U>(max_elements, std::forward<F>(setter));
   }
 
-  // The AppendData functions adds data to the end of the buffer. They accept
+  // The AppendData functions add data to the end of the buffer. They accept
   // the same input types as the constructors.
-  template <typename T, typename internal::ByteType<T>::t = 0>
-  void AppendData(const T* data, size_t size) {
+  template <typename U,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  void AppendData(const U* data, size_t size) {
     RTC_DCHECK(IsConsistent());
     const size_t new_size = size_ + size;
     EnsureCapacity(new_size);
-    std::memcpy(data_.get() + size_, data, size);
+    static_assert(sizeof(T) == sizeof(U), "");
+    std::memcpy(data_.get() + size_, data, size * sizeof(U));
     size_ = new_size;
     RTC_DCHECK(IsConsistent());
   }
 
-  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
-  void AppendData(const T(&array)[N]) {
+  template <typename U,
+            size_t N,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  void AppendData(const U (&array)[N]) {
     AppendData(array, N);
   }
 
-  void AppendData(const Buffer& buf) { AppendData(buf.data(), buf.size()); }
+  void AppendData(const BufferT& buf) { AppendData(buf.data(), buf.size()); }
 
-  // Append at most |max_bytes| of data to the end of the buffer, using the
-  // function |setter|, which should have the following signature:
-  //   size_t setter(ArrayView<T> view)
+  // Append at most |max_elements| to the end of the buffer, using the function
+  // |setter|, which should have the following signature:
+  //   size_t setter(ArrayView<U> view)
   // |setter| is given an appropriately typed ArrayView of the area in which to
   // write the data (i.e. starting at the former end of the buffer) and should
-  // return the number of bytes actually written. This number must be <=
-  // |max_bytes|.
-  template <typename T = uint8_t, typename F,
-            typename internal::ByteType<T>::t = 0>
-  size_t AppendData(size_t max_bytes, F&& setter) {
+  // return the number of elements actually written. This number must be <=
+  // |max_elements|.
+  template <typename U = T,
+            typename F,
+            typename std::enable_if<
+                internal::BufferCompat<T, U>::value>::type* = nullptr>
+  size_t AppendData(size_t max_elements, F&& setter) {
     RTC_DCHECK(IsConsistent());
     const size_t old_size = size_;
-    SetSize(old_size + max_bytes);
-    T *base_ptr = data<T>() + old_size;
-    size_t written_bytes =
-        setter(rtc::ArrayView<T>(base_ptr, max_bytes));
+    SetSize(old_size + max_elements);
+    U* base_ptr = data<U>() + old_size;
+    size_t written_elements = setter(rtc::ArrayView<U>(base_ptr, max_elements));
 
-    RTC_CHECK_LE(written_bytes, max_bytes);
-    size_ = old_size + written_bytes;
+    RTC_CHECK_LE(written_elements, max_elements);
+    size_ = old_size + written_elements;
     RTC_DCHECK(IsConsistent());
-    return written_bytes;
+    return written_elements;
   }
 
   // Sets the size of the buffer. If the new size is smaller than the old, the
@@ -214,8 +275,8 @@ class Buffer {
     RTC_DCHECK(IsConsistent());
     if (capacity <= capacity_)
       return;
-    std::unique_ptr<uint8_t[]> new_data(new uint8_t[capacity]);
-    std::memcpy(new_data.get(), data_.get(), size_);
+    std::unique_ptr<T[]> new_data(new T[capacity]);
+    std::memcpy(new_data.get(), data_.get(), size_ * sizeof(T));
     data_ = std::move(new_data);
     capacity_ = capacity;
     RTC_DCHECK(IsConsistent());
@@ -229,7 +290,7 @@ class Buffer {
   }
 
   // Swaps two buffers. Also works for buffers that have been moved from.
-  friend void swap(Buffer& a, Buffer& b) {
+  friend void swap(BufferT& a, BufferT& b) {
     using std::swap;
     swap(a.size_, b.size_);
     swap(a.capacity_, b.capacity_);
@@ -262,11 +323,12 @@ class Buffer {
 
   size_t size_;
   size_t capacity_;
-  std::unique_ptr<uint8_t[]> data_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(Buffer);
+  std::unique_ptr<T[]> data_;
 };
 
+// By far the most common sort of buffer.
+using Buffer = BufferT<uint8_t>;
+
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_BUFFER_H_
diff --git a/chromium/third_party/webrtc/base/buffer_unittest.cc b/chromium/third_party/webrtc/base/buffer_unittest.cc
index 2f3bcfd6060..e9a853c6153 100644
--- a/chromium/third_party/webrtc/base/buffer_unittest.cc
+++ b/chromium/third_party/webrtc/base/buffer_unittest.cc
@@ -11,8 +11,8 @@
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/gunit.h"
 
-#include <algorithm>  // std::swap (pre-C++11)
-#include <utility>  // std::swap (C++11 and later)
+#include <type_traits>
+#include <utility>
 
 namespace rtc {
 
@@ -301,4 +301,60 @@ TEST(BufferTest, TestBracketWrite) {
   }
 }
 
+TEST(BufferTest, TestInt16) {
+  static constexpr int16_t test_data[] = {14, 15, 16, 17, 18};
+  BufferT<int16_t> buf(test_data);
+  EXPECT_EQ(buf.size(), 5u);
+  EXPECT_EQ(buf.capacity(), 5u);
+  EXPECT_NE(buf.data(), nullptr);
+  for (size_t i = 0; i != buf.size(); ++i) {
+    EXPECT_EQ(test_data[i], buf[i]);
+  }
+  BufferT<int16_t> buf2(test_data);
+  EXPECT_EQ(buf, buf2);
+  buf2[0] = 9;
+  EXPECT_NE(buf, buf2);
+}
+
+TEST(BufferTest, TestFloat) {
+  static constexpr float test_data[] = {14, 15, 16, 17, 18};
+  BufferT<float> buf;
+  EXPECT_EQ(buf.size(), 0u);
+  EXPECT_EQ(buf.capacity(), 0u);
+  EXPECT_EQ(buf.data(), nullptr);
+  buf.SetData(test_data);
+  EXPECT_EQ(buf.size(), 5u);
+  EXPECT_EQ(buf.capacity(), 5u);
+  EXPECT_NE(buf.data(), nullptr);
+  float* p1 = buf.data();
+  while (buf.data() == p1) {
+    buf.AppendData(test_data);
+  }
+  EXPECT_EQ(buf.size(), buf.capacity());
+  EXPECT_GT(buf.size(), 5u);
+  EXPECT_EQ(buf.size() % 5, 0u);
+  EXPECT_NE(buf.data(), nullptr);
+  for (size_t i = 0; i != buf.size(); ++i) {
+    EXPECT_EQ(test_data[i % 5], buf[i]);
+  }
+}
+
+TEST(BufferTest, TestStruct) {
+  struct BloodStone {
+    bool blood;
+    const char* stone;
+  };
+  BufferT<BloodStone> buf(4);
+  EXPECT_EQ(buf.size(), 4u);
+  EXPECT_EQ(buf.capacity(), 4u);
+  EXPECT_NE(buf.data(), nullptr);
+  BufferT<BloodStone*> buf2(4);
+  for (size_t i = 0; i < buf2.size(); ++i) {
+    buf2[i] = &buf[i];
+  }
+  static const char kObsidian[] = "obsidian";
+  buf2[2]->stone = kObsidian;
+  EXPECT_EQ(kObsidian, buf[2].stone);
+}
+
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/bufferqueue.h b/chromium/third_party/webrtc/base/bufferqueue.h
index 623d85a3847..bc9fc842dbd 100644
--- a/chromium/third_party/webrtc/base/bufferqueue.h
+++ b/chromium/third_party/webrtc/base/bufferqueue.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 
 namespace rtc {
diff --git a/chromium/third_party/webrtc/base/bytebuffer.cc b/chromium/third_party/webrtc/base/bytebuffer.cc
index cf4ce42574e..9730ff23d67 100644
--- a/chromium/third_party/webrtc/base/bytebuffer.cc
+++ b/chromium/third_party/webrtc/base/bytebuffer.cc
@@ -88,6 +88,20 @@ void ByteBufferWriter::WriteUInt64(uint64_t val) {
   WriteBytes(reinterpret_cast<const char*>(&v), 8);
 }
 
+// Serializes an unsigned varint in the format described by
+// https://developers.google.com/protocol-buffers/docs/encoding#varints
+// with the caveat that integers are 64-bit, not 128-bit.
+void ByteBufferWriter::WriteUVarint(uint64_t val) {
+  while (val >= 0x80) {
+    // Write 7 bits at a time, then set the msb to a continuation byte (msb=1).
+    char byte = static_cast<char>(val) | 0x80;
+    WriteBytes(&byte, 1);
+    val >>= 7;
+  }
+  char last_byte = static_cast<char>(val);
+  WriteBytes(&last_byte, 1);
+}
+
 void ByteBufferWriter::WriteString(const std::string& val) {
   WriteBytes(val.c_str(), val.size());
 }
@@ -220,6 +234,29 @@ bool ByteBufferReader::ReadUInt64(uint64_t* val) {
   }
 }
 
+bool ByteBufferReader::ReadUVarint(uint64_t* val) {
+  if (!val) {
+    return false;
+  }
+  // Integers are deserialized 7 bits at a time, with each byte having a
+  // continuation byte (msb=1) if there are more bytes to be read.
+  uint64_t v = 0;
+  for (int i = 0; i < 64; i += 7) {
+    char byte;
+    if (!ReadBytes(&byte, 1)) {
+      return false;
+    }
+    // Read the first 7 bits of the byte, then offset by bits read so far.
+    v |= (static_cast<uint64_t>(byte) & 0x7F) << i;
+    // True if the msb is not a continuation byte.
+    if (static_cast<uint64_t>(byte) < 0x80) {
+      *val = v;
+      return true;
+    }
+  }
+  return false;
+}
+
 bool ByteBufferReader::ReadString(std::string* val, size_t len) {
   if (!val) return false;
 
diff --git a/chromium/third_party/webrtc/base/bytebuffer.h b/chromium/third_party/webrtc/base/bytebuffer.h
index 8fd086367db..cd7b2c6cea8 100644
--- a/chromium/third_party/webrtc/base/bytebuffer.h
+++ b/chromium/third_party/webrtc/base/bytebuffer.h
@@ -57,6 +57,7 @@ class ByteBufferWriter : public ByteBuffer {
   void WriteUInt24(uint32_t val);
   void WriteUInt32(uint32_t val);
   void WriteUInt64(uint64_t val);
+  void WriteUVarint(uint64_t val);
   void WriteString(const std::string& val);
   void WriteBytes(const char* val, size_t len);
 
@@ -110,6 +111,7 @@ class ByteBufferReader : public ByteBuffer {
   bool ReadUInt24(uint32_t* val);
   bool ReadUInt32(uint32_t* val);
   bool ReadUInt64(uint64_t* val);
+  bool ReadUVarint(uint64_t* val);
   bool ReadBytes(char* val, size_t len);
 
   // Appends next |len| bytes from the buffer to |val|. Returns false
diff --git a/chromium/third_party/webrtc/base/bytebuffer_unittest.cc b/chromium/third_party/webrtc/base/bytebuffer_unittest.cc
index 723641811f2..bdbb159b665 100644
--- a/chromium/third_party/webrtc/base/bytebuffer_unittest.cc
+++ b/chromium/third_party/webrtc/base/bytebuffer_unittest.cc
@@ -196,4 +196,64 @@ TEST(ByteBufferTest, TestReadWriteBuffer) {
   }
 }
 
+TEST(ByteBufferTest, TestReadWriteUVarint) {
+  ByteBufferWriter::ByteOrder orders[2] = {ByteBufferWriter::ORDER_HOST,
+                                           ByteBufferWriter::ORDER_NETWORK};
+  for (ByteBufferWriter::ByteOrder& order : orders) {
+    ByteBufferWriter write_buffer(order);
+    size_t size = 0;
+    EXPECT_EQ(size, write_buffer.Length());
+
+    write_buffer.WriteUVarint(1u);
+    ++size;
+    EXPECT_EQ(size, write_buffer.Length());
+
+    write_buffer.WriteUVarint(2u);
+    ++size;
+    EXPECT_EQ(size, write_buffer.Length());
+
+    write_buffer.WriteUVarint(27u);
+    ++size;
+    EXPECT_EQ(size, write_buffer.Length());
+
+    write_buffer.WriteUVarint(149u);
+    size += 2;
+    EXPECT_EQ(size, write_buffer.Length());
+
+    write_buffer.WriteUVarint(68719476736u);
+    size += 6;
+    EXPECT_EQ(size, write_buffer.Length());
+
+    ByteBufferReader read_buffer(write_buffer.Data(), write_buffer.Length(),
+                                 order);
+    EXPECT_EQ(size, read_buffer.Length());
+    uint64_t val1, val2, val3, val4, val5;
+
+    ASSERT_TRUE(read_buffer.ReadUVarint(&val1));
+    EXPECT_EQ(1u, val1);
+    --size;
+    EXPECT_EQ(size, read_buffer.Length());
+
+    ASSERT_TRUE(read_buffer.ReadUVarint(&val2));
+    EXPECT_EQ(2u, val2);
+    --size;
+    EXPECT_EQ(size, read_buffer.Length());
+
+    ASSERT_TRUE(read_buffer.ReadUVarint(&val3));
+    EXPECT_EQ(27u, val3);
+    --size;
+    EXPECT_EQ(size, read_buffer.Length());
+
+    ASSERT_TRUE(read_buffer.ReadUVarint(&val4));
+    EXPECT_EQ(149u, val4);
+    size -= 2;
+    EXPECT_EQ(size, read_buffer.Length());
+
+    ASSERT_TRUE(read_buffer.ReadUVarint(&val5));
+    EXPECT_EQ(68719476736u, val5);
+    size -= 6;
+    EXPECT_EQ(size, read_buffer.Length());
+  }
+}
+
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/callback_unittest.cc b/chromium/third_party/webrtc/base/callback_unittest.cc
index db294cd96eb..aba1e0ce28f 100644
--- a/chromium/third_party/webrtc/base/callback_unittest.cc
+++ b/chromium/third_party/webrtc/base/callback_unittest.cc
@@ -34,7 +34,7 @@ class RefCountedBindTester : public RefCountInterface {
   int AddRef() const override {
     return ++count_;
   }
-  int Release() const {
+  int Release() const override {
     return --count_;
   }
   int RefCount() const { return count_; }
diff --git a/chromium/third_party/webrtc/base/checks.cc b/chromium/third_party/webrtc/base/checks.cc
index 49a31f29b9d..d9dc8f2d63b 100644
--- a/chromium/third_party/webrtc/base/checks.cc
+++ b/chromium/third_party/webrtc/base/checks.cc
@@ -22,11 +22,16 @@
 #endif
 
 #if defined(WEBRTC_ANDROID)
-#define LOG_TAG "rtc"
+#define RTC_LOG_TAG "rtc"
 #include <android/log.h>  // NOLINT
 #endif
 
+#if defined(WEBRTC_WIN)
+#include <windows.h>
+#endif
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 
 #if defined(_MSC_VER)
 // Warning C4722: destructor never returns, potential memory leak.
@@ -38,7 +43,7 @@ namespace rtc {
 
 void VPrintError(const char* format, va_list args) {
 #if defined(WEBRTC_ANDROID)
-  __android_log_vprint(ANDROID_LOG_ERROR, LOG_TAG, format, args);
+  __android_log_vprint(ANDROID_LOG_ERROR, RTC_LOG_TAG, format, args);
 #else
   vfprintf(stderr, format, args);
 #endif
@@ -105,8 +110,11 @@ NO_RETURN FatalMessage::~FatalMessage() {
 }
 
 void FatalMessage::Init(const char* file, int line) {
-  stream_ << std::endl << std::endl << "#" << std::endl << "# Fatal error in "
-          << file << ", line " << line << std::endl << "# ";
+  stream_ << std::endl << std::endl
+          << "#" << std::endl
+          << "# Fatal error in " << file << ", line " << line << std::endl
+          << "# last system error: " << LAST_SYSTEM_ERROR << std::endl
+          << "# ";
 }
 
 // MSVC doesn't like complex extern templates and DLLs.
diff --git a/chromium/third_party/webrtc/base/copyonwritebuffer.h b/chromium/third_party/webrtc/base/copyonwritebuffer.h
index 87f24bf51d6..a7e52beea53 100644
--- a/chromium/third_party/webrtc/base/copyonwritebuffer.h
+++ b/chromium/third_party/webrtc/base/copyonwritebuffer.h
@@ -36,10 +36,14 @@ class CopyOnWriteBuffer {
 
   // Construct a buffer and copy the specified number of bytes into it. The
   // source array may be (const) uint8_t*, int8_t*, or char*.
-  template <typename T, typename internal::ByteType<T>::t = 0>
+  template <typename T,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   CopyOnWriteBuffer(const T* data, size_t size)
       : CopyOnWriteBuffer(data, size, size) {}
-  template <typename T, typename internal::ByteType<T>::t = 0>
+  template <typename T,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   CopyOnWriteBuffer(const T* data, size_t size, size_t capacity)
       : CopyOnWriteBuffer(size, capacity) {
     if (buffer_) {
@@ -48,22 +52,29 @@ class CopyOnWriteBuffer {
   }
 
   // Construct a buffer from the contents of an array.
-  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
-  CopyOnWriteBuffer(const T(&array)[N])  // NOLINT: runtime/explicit
+  template <typename T,
+            size_t N,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
+  CopyOnWriteBuffer(const T (&array)[N])  // NOLINT: runtime/explicit
       : CopyOnWriteBuffer(array, N) {}
 
   ~CopyOnWriteBuffer();
 
   // Get a pointer to the data. Just .data() will give you a (const) uint8_t*,
   // but you may also use .data<int8_t>() and .data<char>().
-  template <typename T = uint8_t, typename internal::ByteType<T>::t = 0>
+  template <typename T = uint8_t,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   const T* data() const {
     return cdata<T>();
   }
 
   // Get writable pointer to the data. This will create a copy of the underlying
   // data if it is shared with other buffers.
-  template <typename T = uint8_t, typename internal::ByteType<T>::t = 0>
+  template <typename T = uint8_t,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   T* data() {
     RTC_DCHECK(IsConsistent());
     if (!buffer_) {
@@ -75,7 +86,9 @@ class CopyOnWriteBuffer {
 
   // Get const pointer to the data. This will not create a copy of the
   // underlying data if it is shared with other buffers.
-  template <typename T = uint8_t, typename internal::ByteType<T>::t = 0>
+  template <typename T = uint8_t,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   T* cdata() const {
     RTC_DCHECK(IsConsistent());
     if (!buffer_) {
@@ -137,7 +150,9 @@ class CopyOnWriteBuffer {
 
   // Replace the contents of the buffer. Accepts the same types as the
   // constructors.
-  template <typename T, typename internal::ByteType<T>::t = 0>
+  template <typename T,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   void SetData(const T* data, size_t size) {
     RTC_DCHECK(IsConsistent());
     if (!buffer_ || !buffer_->HasOneRef()) {
@@ -149,8 +164,11 @@ class CopyOnWriteBuffer {
     RTC_DCHECK(IsConsistent());
   }
 
-  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
-  void SetData(const T(&array)[N]) {
+  template <typename T,
+            size_t N,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
+  void SetData(const T (&array)[N]) {
     SetData(array, N);
   }
 
@@ -163,7 +181,9 @@ class CopyOnWriteBuffer {
   }
 
   // Append data to the buffer. Accepts the same types as the constructors.
-  template <typename T, typename internal::ByteType<T>::t = 0>
+  template <typename T,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
   void AppendData(const T* data, size_t size) {
     RTC_DCHECK(IsConsistent());
     if (!buffer_) {
@@ -178,8 +198,11 @@ class CopyOnWriteBuffer {
     RTC_DCHECK(IsConsistent());
   }
 
-  template <typename T, size_t N, typename internal::ByteType<T>::t = 0>
-  void AppendData(const T(&array)[N]) {
+  template <typename T,
+            size_t N,
+            typename std::enable_if<
+                internal::BufferCompat<uint8_t, T>::value>::type* = nullptr>
+  void AppendData(const T (&array)[N]) {
     AppendData(array, N);
   }
 
diff --git a/chromium/third_party/webrtc/base/criticalsection_unittest.cc b/chromium/third_party/webrtc/base/criticalsection_unittest.cc
index a0e10338628..d33afacd8ce 100644
--- a/chromium/third_party/webrtc/base/criticalsection_unittest.cc
+++ b/chromium/third_party/webrtc/base/criticalsection_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <set>
 #include <vector>
 
@@ -17,7 +18,6 @@
 #include "webrtc/base/event.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scopedptrcollection.h"
 #include "webrtc/base/thread.h"
 
@@ -226,8 +226,8 @@ TEST(AtomicOpsTest, Simple) {
 TEST(AtomicOpsTest, SimplePtr) {
   class Foo {};
   Foo* volatile foo = nullptr;
-  scoped_ptr<Foo> a(new Foo());
-  scoped_ptr<Foo> b(new Foo());
+  std::unique_ptr<Foo> a(new Foo());
+  std::unique_ptr<Foo> b(new Foo());
   // Reading the initial value should work as expected.
   EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == nullptr);
   // Setting using compare and swap should work.
diff --git a/chromium/third_party/webrtc/base/cryptstring.h b/chromium/third_party/webrtc/base/cryptstring.h
index a6bae51f897..adaac2f36bc 100644
--- a/chromium/third_party/webrtc/base/cryptstring.h
+++ b/chromium/third_party/webrtc/base/cryptstring.h
@@ -13,11 +13,11 @@
 
 #include <string.h>
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/linked_ptr.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
@@ -42,8 +42,8 @@ public:
 };
 
 class CryptString {
-public:
- CryptString();
+ public:
+  CryptString();
   size_t GetLength() const { return impl_->GetLength(); }
   void CopyTo(char * dest, bool nullterminate) const { impl_->CopyTo(dest, nullterminate); }
   CryptString(const CryptString& other);
@@ -60,9 +60,9 @@ public:
   void CopyRawTo(std::vector<unsigned char> * dest) const {
     return impl_->CopyRawTo(dest);
   }
-  
-private:
-  scoped_ptr<const CryptStringImpl> impl_;
+
+ private:
+  std::unique_ptr<const CryptStringImpl> impl_;
 };
 
 
diff --git a/chromium/third_party/webrtc/base/dbus_unittest.cc b/chromium/third_party/webrtc/base/dbus_unittest.cc
index 17752f143f6..38c507d2893 100644
--- a/chromium/third_party/webrtc/base/dbus_unittest.cc
+++ b/chromium/third_party/webrtc/base/dbus_unittest.cc
@@ -10,6 +10,8 @@
 
 #ifdef HAVE_DBUS_GLIB
 
+#include <memory>
+
 #include "webrtc/base/dbus.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/thread.h"
@@ -51,7 +53,7 @@ class DBusSigFilterTest : public DBusSigFilter {
 
 TEST(DBusMonitorTest, StartStopStartStop) {
   DBusSigFilterTest filter;
-  rtc::scoped_ptr<rtc::DBusMonitor> monitor;
+  std::unique_ptr<rtc::DBusMonitor> monitor;
   monitor.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
   if (monitor) {
     EXPECT_TRUE(monitor->AddFilter(&filter));
@@ -83,7 +85,7 @@ TEST(DBusMonitorTest, StartStopStartStop) {
 // This test is to make sure that we capture the "NameAcquired" signal.
 TEST(DBusMonitorTest, ReceivedNameAcquiredSignal) {
   DBusSigFilterTest filter;
-  rtc::scoped_ptr<rtc::DBusMonitor> monitor;
+  std::unique_ptr<rtc::DBusMonitor> monitor;
   monitor.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
   if (monitor) {
     EXPECT_TRUE(monitor->AddFilter(&filter));
@@ -100,12 +102,12 @@ TEST(DBusMonitorTest, ReceivedNameAcquiredSignal) {
 
 TEST(DBusMonitorTest, ConcurrentMonitors) {
   DBusSigFilterTest filter1;
-  rtc::scoped_ptr<rtc::DBusMonitor> monitor1;
+  std::unique_ptr<rtc::DBusMonitor> monitor1;
   monitor1.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
   if (monitor1) {
     EXPECT_TRUE(monitor1->AddFilter(&filter1));
     DBusSigFilterTest filter2;
-    rtc::scoped_ptr<rtc::DBusMonitor> monitor2;
+    std::unique_ptr<rtc::DBusMonitor> monitor2;
     monitor2.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
     EXPECT_TRUE(monitor2->AddFilter(&filter2));
 
@@ -129,7 +131,7 @@ TEST(DBusMonitorTest, ConcurrentMonitors) {
 TEST(DBusMonitorTest, ConcurrentFilters) {
   DBusSigFilterTest filter1;
   DBusSigFilterTest filter2;
-  rtc::scoped_ptr<rtc::DBusMonitor> monitor;
+  std::unique_ptr<rtc::DBusMonitor> monitor;
   monitor.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
   if (monitor) {
     EXPECT_TRUE(monitor->AddFilter(&filter1));
@@ -151,7 +153,7 @@ TEST(DBusMonitorTest, ConcurrentFilters) {
 TEST(DBusMonitorTest, NoAddFilterIfRunning) {
   DBusSigFilterTest filter1;
   DBusSigFilterTest filter2;
-  rtc::scoped_ptr<rtc::DBusMonitor> monitor;
+  std::unique_ptr<rtc::DBusMonitor> monitor;
   monitor.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
   if (monitor) {
     EXPECT_TRUE(monitor->AddFilter(&filter1));
@@ -170,7 +172,7 @@ TEST(DBusMonitorTest, NoAddFilterIfRunning) {
 TEST(DBusMonitorTest, AddFilterAfterStop) {
   DBusSigFilterTest filter1;
   DBusSigFilterTest filter2;
-  rtc::scoped_ptr<rtc::DBusMonitor> monitor;
+  std::unique_ptr<rtc::DBusMonitor> monitor;
   monitor.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
   if (monitor) {
     EXPECT_TRUE(monitor->AddFilter(&filter1));
@@ -194,7 +196,7 @@ TEST(DBusMonitorTest, AddFilterAfterStop) {
 
 TEST(DBusMonitorTest, StopRightAfterStart) {
   DBusSigFilterTest filter;
-  rtc::scoped_ptr<rtc::DBusMonitor> monitor;
+  std::unique_ptr<rtc::DBusMonitor> monitor;
   monitor.reset(rtc::DBusMonitor::Create(DBUS_BUS_SYSTEM));
   if (monitor) {
     EXPECT_TRUE(monitor->AddFilter(&filter));
diff --git a/chromium/third_party/webrtc/base/diskcache.cc b/chromium/third_party/webrtc/base/diskcache.cc
index a1fba6af9ad..233d2ab4309 100644
--- a/chromium/third_party/webrtc/base/diskcache.cc
+++ b/chromium/third_party/webrtc/base/diskcache.cc
@@ -15,6 +15,8 @@
 #endif
 
 #include <algorithm>
+#include <memory>
+
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/diskcache.h"
@@ -123,7 +125,7 @@ StreamInterface* DiskCache::WriteResource(const std::string& id, size_t index) {
     previous_size = entry->size;
   }
 
-  scoped_ptr<FileStream> file(new FileStream);
+  std::unique_ptr<FileStream> file(new FileStream);
   if (!file->Open(filename, "wb", NULL)) {
     LOG_F(LS_ERROR) << "Couldn't create cache file";
     return NULL;
@@ -161,7 +163,7 @@ StreamInterface* DiskCache::ReadResource(const std::string& id,
   if (index >= entry->streams)
     return NULL;
 
-  scoped_ptr<FileStream> file(new FileStream);
+  std::unique_ptr<FileStream> file(new FileStream);
   if (!file->Open(IdToFilename(id, index), "rb", NULL))
     return NULL;
 
diff --git a/chromium/third_party/webrtc/base/fakenetwork.h b/chromium/third_party/webrtc/base/fakenetwork.h
index e3996e6649c..2dd2137aa81 100644
--- a/chromium/third_party/webrtc/base/fakenetwork.h
+++ b/chromium/third_party/webrtc/base/fakenetwork.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_BASE_FAKENETWORK_H_
 #define WEBRTC_BASE_FAKENETWORK_H_
 
+#include <memory>
 #include <string>
 #include <utility>
 #include <vector>
@@ -99,9 +100,9 @@ class FakeNetworkManager : public NetworkManagerBase,
         prefix_length = kFakeIPv6NetworkPrefixLength;
       }
       IPAddress prefix = TruncateIP(it->first.ipaddr(), prefix_length);
-      scoped_ptr<Network> net(new Network(it->first.hostname(),
-                                          it->first.hostname(), prefix,
-                                          prefix_length, it->second));
+      std::unique_ptr<Network> net(new Network(it->first.hostname(),
+                                               it->first.hostname(), prefix,
+                                               prefix_length, it->second));
       net->set_default_local_address_provider(this);
       net->AddIP(it->first.ipaddr());
       networks.push_back(net.release());
diff --git a/chromium/third_party/webrtc/base/fakesslidentity.h b/chromium/third_party/webrtc/base/fakesslidentity.h
index 47ff86d03a8..3b0df298412 100644
--- a/chromium/third_party/webrtc/base/fakesslidentity.h
+++ b/chromium/third_party/webrtc/base/fakesslidentity.h
@@ -12,6 +12,7 @@
 #define WEBRTC_BASE_FAKESSLIDENTITY_H_
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/common.h"
@@ -36,13 +37,13 @@ class FakeSSLCertificate : public rtc::SSLCertificate {
       certs_.push_back(FakeSSLCertificate(*it));
     }
   }
-  virtual FakeSSLCertificate* GetReference() const {
+  FakeSSLCertificate* GetReference() const override {
     return new FakeSSLCertificate(*this);
   }
-  virtual std::string ToPEMString() const {
+  std::string ToPEMString() const override {
     return data_;
   }
-  virtual void ToDER(Buffer* der_buffer) const {
+  void ToDER(Buffer* der_buffer) const override {
     std::string der_string;
     VERIFY(SSLIdentity::PemToDer(kPemTypeCertificate, data_, &der_string));
     der_buffer->SetData(der_string.c_str(), der_string.size());
@@ -56,24 +57,24 @@ class FakeSSLCertificate : public rtc::SSLCertificate {
   void set_digest_algorithm(const std::string& algorithm) {
     digest_algorithm_ = algorithm;
   }
-  virtual bool GetSignatureDigestAlgorithm(std::string* algorithm) const {
+  bool GetSignatureDigestAlgorithm(std::string* algorithm) const override {
     *algorithm = digest_algorithm_;
     return true;
   }
-  virtual bool ComputeDigest(const std::string& algorithm,
-                             unsigned char* digest,
-                             size_t size,
-                             size_t* length) const {
+  bool ComputeDigest(const std::string& algorithm,
+                     unsigned char* digest,
+                     size_t size,
+                     size_t* length) const override {
     *length = rtc::ComputeDigest(algorithm, data_.c_str(), data_.size(),
                                        digest, size);
     return (*length != 0);
   }
-  virtual rtc::scoped_ptr<SSLCertChain> GetChain() const {
+  std::unique_ptr<SSLCertChain> GetChain() const override {
     if (certs_.empty())
       return nullptr;
     std::vector<SSLCertificate*> new_certs(certs_.size());
     std::transform(certs_.begin(), certs_.end(), new_certs.begin(), DupCert);
-    rtc::scoped_ptr<SSLCertChain> chain(new SSLCertChain(new_certs));
+    std::unique_ptr<SSLCertChain> chain(new SSLCertChain(new_certs));
     std::for_each(new_certs.begin(), new_certs.end(), DeleteCert);
     return chain;
   }
@@ -98,6 +99,18 @@ class FakeSSLIdentity : public rtc::SSLIdentity {
     return new FakeSSLIdentity(*this);
   }
   virtual const FakeSSLCertificate& certificate() const { return cert_; }
+  virtual std::string PrivateKeyToPEMString() const {
+    RTC_NOTREACHED();  // Not implemented.
+    return "";
+  }
+  virtual std::string PublicKeyToPEMString() const {
+    RTC_NOTREACHED();  // Not implemented.
+    return "";
+  }
+  virtual bool operator==(const SSLIdentity& other) const {
+    RTC_NOTREACHED();  // Not implemented.
+    return false;
+  }
  private:
   FakeSSLCertificate cert_;
 };
diff --git a/chromium/third_party/webrtc/base/filerotatingstream.h b/chromium/third_party/webrtc/base/filerotatingstream.h
index 9e8e35ddd7f..a8522ff0c5f 100644
--- a/chromium/third_party/webrtc/base/filerotatingstream.h
+++ b/chromium/third_party/webrtc/base/filerotatingstream.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_BASE_FILEROTATINGSTREAM_H_
 #define WEBRTC_BASE_FILEROTATINGSTREAM_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -110,7 +111,7 @@ class FileRotatingStream : public StreamInterface {
   const Mode mode_;
 
   // FileStream is used to write to the current file.
-  scoped_ptr<FileStream> file_stream_;
+  std::unique_ptr<FileStream> file_stream_;
   // Convenience storage for file names so we don't generate them over and over.
   std::vector<std::string> file_names_;
   size_t max_file_size_;
diff --git a/chromium/third_party/webrtc/base/filerotatingstream_unittest.cc b/chromium/third_party/webrtc/base/filerotatingstream_unittest.cc
index 09438f870ee..bac2a3a3e32 100644
--- a/chromium/third_party/webrtc/base/filerotatingstream_unittest.cc
+++ b/chromium/third_party/webrtc/base/filerotatingstream_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/filerotatingstream.h"
@@ -57,13 +59,13 @@ class FileRotatingStreamTest : public ::testing::Test {
                         const size_t expected_length,
                         const std::string& dir_path,
                         const char* file_prefix) {
-    scoped_ptr<FileRotatingStream> stream;
+    std::unique_ptr<FileRotatingStream> stream;
     stream.reset(new FileRotatingStream(dir_path, file_prefix));
     ASSERT_TRUE(stream->Open());
     size_t read = 0;
     size_t stream_size = 0;
     EXPECT_TRUE(stream->GetSize(&stream_size));
-    scoped_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
+    std::unique_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
     EXPECT_EQ(SR_SUCCESS,
               stream->ReadAll(buffer.get(), expected_length, &read, nullptr));
     EXPECT_EQ(0, memcmp(expected_contents, buffer.get(), expected_length));
@@ -74,8 +76,8 @@ class FileRotatingStreamTest : public ::testing::Test {
   void VerifyFileContents(const char* expected_contents,
                           const size_t expected_length,
                           const std::string& file_path) {
-    scoped_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
-    scoped_ptr<FileStream> stream(Filesystem::OpenFile(file_path, "r"));
+    std::unique_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
+    std::unique_ptr<FileStream> stream(Filesystem::OpenFile(file_path, "r"));
     EXPECT_TRUE(stream);
     if (!stream) {
       return;
@@ -88,7 +90,7 @@ class FileRotatingStreamTest : public ::testing::Test {
     EXPECT_EQ(file_size, expected_length);
   }
 
-  scoped_ptr<FileRotatingStream> stream_;
+  std::unique_ptr<FileRotatingStream> stream_;
   std::string dir_path_;
 };
 
@@ -114,7 +116,7 @@ TEST_F(FileRotatingStreamTest, EmptyWrite) {
   WriteAndFlush("a", 0);
 
   std::string logfile_path = stream_->GetFilePath(0);
-  scoped_ptr<FileStream> stream(Filesystem::OpenFile(logfile_path, "r"));
+  std::unique_ptr<FileStream> stream(Filesystem::OpenFile(logfile_path, "r"));
   size_t file_size = 0;
   EXPECT_TRUE(stream->GetSize(&file_size));
   EXPECT_EQ(0u, file_size);
@@ -215,13 +217,13 @@ class CallSessionFileRotatingStreamTest : public ::testing::Test {
   void VerifyStreamRead(const char* expected_contents,
                         const size_t expected_length,
                         const std::string& dir_path) {
-    scoped_ptr<CallSessionFileRotatingStream> stream(
+    std::unique_ptr<CallSessionFileRotatingStream> stream(
         new CallSessionFileRotatingStream(dir_path));
     ASSERT_TRUE(stream->Open());
     size_t read = 0;
     size_t stream_size = 0;
     EXPECT_TRUE(stream->GetSize(&stream_size));
-    scoped_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
+    std::unique_ptr<uint8_t[]> buffer(new uint8_t[expected_length]);
     EXPECT_EQ(SR_SUCCESS,
               stream->ReadAll(buffer.get(), expected_length, &read, nullptr));
     EXPECT_EQ(0, memcmp(expected_contents, buffer.get(), expected_length));
@@ -229,7 +231,7 @@ class CallSessionFileRotatingStreamTest : public ::testing::Test {
     EXPECT_EQ(stream_size, read);
   }
 
-  scoped_ptr<CallSessionFileRotatingStream> stream_;
+  std::unique_ptr<CallSessionFileRotatingStream> stream_;
   std::string dir_path_;
 };
 
@@ -266,7 +268,7 @@ TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadLarge) {
 
   ASSERT_TRUE(stream_->Open());
   const size_t buffer_size = 1024 * 1024;
-  scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
   for (int i = 0; i < 8; i++) {
     memset(buffer.get(), i, buffer_size);
     EXPECT_EQ(SR_SUCCESS,
@@ -275,7 +277,7 @@ TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadLarge) {
 
   stream_.reset(new CallSessionFileRotatingStream(dir_path_));
   ASSERT_TRUE(stream_->Open());
-  scoped_ptr<uint8_t[]> expected_buffer(new uint8_t[buffer_size]);
+  std::unique_ptr<uint8_t[]> expected_buffer(new uint8_t[buffer_size]);
   int expected_vals[] = {0, 1, 2, 6, 7};
   for (size_t i = 0; i < arraysize(expected_vals); ++i) {
     memset(expected_buffer.get(), expected_vals[i], buffer_size);
@@ -293,7 +295,7 @@ TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadFirstHalf) {
        6 * 1024 * 1024);
   ASSERT_TRUE(stream_->Open());
   const size_t buffer_size = 1024 * 1024;
-  scoped_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
   for (int i = 0; i < 2; i++) {
     memset(buffer.get(), i, buffer_size);
     EXPECT_EQ(SR_SUCCESS,
@@ -302,7 +304,7 @@ TEST_F(CallSessionFileRotatingStreamTest, WriteAndReadFirstHalf) {
 
   stream_.reset(new CallSessionFileRotatingStream(dir_path_));
   ASSERT_TRUE(stream_->Open());
-  scoped_ptr<uint8_t[]> expected_buffer(new uint8_t[buffer_size]);
+  std::unique_ptr<uint8_t[]> expected_buffer(new uint8_t[buffer_size]);
   int expected_vals[] = {0, 1};
   for (size_t i = 0; i < arraysize(expected_vals); ++i) {
     memset(expected_buffer.get(), expected_vals[i], buffer_size);
diff --git a/chromium/third_party/webrtc/base/fileutils.cc b/chromium/third_party/webrtc/base/fileutils.cc
index cb23153de77..d354dd8eb56 100644
--- a/chromium/third_party/webrtc/base/fileutils.cc
+++ b/chromium/third_party/webrtc/base/fileutils.cc
@@ -133,7 +133,7 @@ bool DirectoryIterator::OlderThan(int seconds) const {
 #else
   file_modify_time = stat_.st_mtime;
 #endif
-  return TimeDiff(time(NULL), file_modify_time) >= seconds;
+  return time(NULL) - file_modify_time >= seconds;
 }
 
 FilesystemInterface* Filesystem::default_filesystem_ = NULL;
diff --git a/chromium/third_party/webrtc/base/fileutils.h b/chromium/third_party/webrtc/base/fileutils.h
index bf02571d936..23d36b15f47 100644
--- a/chromium/third_party/webrtc/base/fileutils.h
+++ b/chromium/third_party/webrtc/base/fileutils.h
@@ -23,8 +23,8 @@
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/platform_file.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
diff --git a/chromium/third_party/webrtc/base/fileutils_unittest.cc b/chromium/third_party/webrtc/base/fileutils_unittest.cc
index 6e98e145091..51396caefbb 100644
--- a/chromium/third_party/webrtc/base/fileutils_unittest.cc
+++ b/chromium/third_party/webrtc/base/fileutils_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/pathutils.h"
@@ -74,7 +76,7 @@ TEST(FilesystemTest, TestCreatePrivateFile) {
   EXPECT_FALSE(Filesystem::CreatePrivateFile(path));
 
   // Verify that we have permission to open the file for reading and writing.
-  scoped_ptr<FileStream> fs(Filesystem::OpenFile(path, "wb"));
+  std::unique_ptr<FileStream> fs(Filesystem::OpenFile(path, "wb"));
   EXPECT_TRUE(fs.get() != NULL);
   // Have to close the file on Windows before it will let us delete it.
   fs.reset();
diff --git a/chromium/third_party/webrtc/base/flags.h b/chromium/third_party/webrtc/base/flags.h
index 4ce857b74a7..d16f12bad63 100644
--- a/chromium/third_party/webrtc/base/flags.h
+++ b/chromium/third_party/webrtc/base/flags.h
@@ -27,6 +27,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 
 namespace rtc {
 
diff --git a/chromium/third_party/webrtc/base/gunit.h b/chromium/third_party/webrtc/base/gunit.h
index 1a6c36374e9..e705322e6f8 100644
--- a/chromium/third_party/webrtc/base/gunit.h
+++ b/chromium/third_party/webrtc/base/gunit.h
@@ -20,21 +20,22 @@
 #endif
 
 // Wait until "ex" is true, or "timeout" expires.
-#define WAIT(ex, timeout)                                                     \
-  for (uint32_t start = rtc::Time(); !(ex) && rtc::Time() < start + timeout;) \
+#define WAIT(ex, timeout)                             \
+  for (int64_t start = rtc::TimeMillis();             \
+       !(ex) && rtc::TimeMillis() < start + timeout;) \
     rtc::Thread::Current()->ProcessMessages(1);
 
 // This returns the result of the test in res, so that we don't re-evaluate
 // the expression in the XXXX_WAIT macros below, since that causes problems
 // when the expression is only true the first time you check it.
-#define WAIT_(ex, timeout, res)                     \
-  do {                                              \
-    uint32_t start = rtc::Time();                   \
-    res = (ex);                                     \
-    while (!res && rtc::Time() < start + timeout) { \
-      rtc::Thread::Current()->ProcessMessages(1);   \
-      res = (ex);                                   \
-    }                                               \
+#define WAIT_(ex, timeout, res)                           \
+  do {                                                    \
+    int64_t start = rtc::TimeMillis();                    \
+    res = (ex);                                           \
+    while (!res && rtc::TimeMillis() < start + timeout) { \
+      rtc::Thread::Current()->ProcessMessages(1);         \
+      res = (ex);                                         \
+    }                                                     \
   } while (0)
 
 // The typical EXPECT_XXXX and ASSERT_XXXXs, but done until true or a timeout.
diff --git a/chromium/third_party/webrtc/base/helpers.cc b/chromium/third_party/webrtc/base/helpers.cc
index 1ad5d0e12ba..0a39ee923e1 100644
--- a/chromium/third_party/webrtc/base/helpers.cc
+++ b/chromium/third_party/webrtc/base/helpers.cc
@@ -11,6 +11,7 @@
 #include "webrtc/base/helpers.h"
 
 #include <limits>
+#include <memory>
 
 #if defined(FEATURE_ENABLE_SSL)
 #include "webrtc/base/sslconfig.h"
@@ -28,7 +29,6 @@
 #include "webrtc/base/base64.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/timeutils.h"
 
 // Protect against max macro inclusion.
@@ -181,8 +181,8 @@ static const char kUuidDigit17[4] = {'8', '9', 'a', 'b'};
 
 // This round about way of creating a global RNG is to safe-guard against
 // indeterminant static initialization order.
-scoped_ptr<RandomGenerator>& GetGlobalRng() {
-  RTC_DEFINE_STATIC_LOCAL(scoped_ptr<RandomGenerator>, global_rng,
+std::unique_ptr<RandomGenerator>& GetGlobalRng() {
+  RTC_DEFINE_STATIC_LOCAL(std::unique_ptr<RandomGenerator>, global_rng,
                           (new SecureRandomGenerator()));
   return global_rng;
 }
@@ -223,7 +223,7 @@ bool CreateRandomString(size_t len,
                         const char* table, int table_size,
                         std::string* str) {
   str->clear();
-  scoped_ptr<uint8_t[]> bytes(new uint8_t[len]);
+  std::unique_ptr<uint8_t[]> bytes(new uint8_t[len]);
   if (!Rng().Generate(bytes.get(), len)) {
     LOG(LS_ERROR) << "Failed to generate random string!";
     return false;
@@ -250,7 +250,7 @@ bool CreateRandomString(size_t len, const std::string& table,
 // Where 'x' is a hex digit, and 'y' is 8, 9, a or b.
 std::string CreateRandomUuid() {
   std::string str;
-  scoped_ptr<uint8_t[]> bytes(new uint8_t[31]);
+  std::unique_ptr<uint8_t[]> bytes(new uint8_t[31]);
   if (!Rng().Generate(bytes.get(), 31)) {
     LOG(LS_ERROR) << "Failed to generate random string!";
     return str;
diff --git a/chromium/third_party/webrtc/base/httpbase.cc b/chromium/third_party/webrtc/base/httpbase.cc
index 81ca4cceeb1..efdc8af8a9e 100644
--- a/chromium/third_party/webrtc/base/httpbase.cc
+++ b/chromium/third_party/webrtc/base/httpbase.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 
 #if defined(WEBRTC_WIN)
 #include "webrtc/base/win32.h"
@@ -271,9 +272,8 @@ public:
     // When the method returns, we restore the old document.  Ideally, we would
     // pass our StreamInterface* to DoReceiveLoop, but due to the callbacks
     // of HttpParser, we would still need to store the pointer temporarily.
-    scoped_ptr<StreamInterface>
-        stream(new BlockingMemoryStream(reinterpret_cast<char*>(buffer),
-                                        buffer_len));
+    std::unique_ptr<StreamInterface> stream(
+        new BlockingMemoryStream(reinterpret_cast<char*>(buffer), buffer_len));
 
     // Replace the existing document with our wrapped buffer.
     base_->data_->document.swap(stream);
diff --git a/chromium/third_party/webrtc/base/httpclient.cc b/chromium/third_party/webrtc/base/httpclient.cc
index e078334094a..a458590bdb6 100644
--- a/chromium/third_party/webrtc/base/httpclient.cc
+++ b/chromium/third_party/webrtc/base/httpclient.cc
@@ -10,6 +10,7 @@
 
 #include <time.h>
 #include <algorithm>
+#include <memory>
 #include "webrtc/base/asyncsocket.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/diskcache.h"
@@ -17,7 +18,6 @@
 #include "webrtc/base/httpcommon-inl.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/pathutils.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketstream.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
@@ -466,7 +466,8 @@ bool HttpClient::BeginCacheFile() {
     return false;
   }
 
-  scoped_ptr<StreamInterface> stream(cache_->WriteResource(id, kCacheBody));
+  std::unique_ptr<StreamInterface> stream(
+      cache_->WriteResource(id, kCacheBody));
   if (!stream) {
     LOG_F(LS_ERROR) << "Couldn't open body cache";
     return false;
@@ -485,7 +486,8 @@ bool HttpClient::BeginCacheFile() {
 }
 
 HttpError HttpClient::WriteCacheHeaders(const std::string& id) {
-  scoped_ptr<StreamInterface> stream(cache_->WriteResource(id, kCacheHeader));
+  std::unique_ptr<StreamInterface> stream(
+      cache_->WriteResource(id, kCacheHeader));
   if (!stream) {
     LOG_F(LS_ERROR) << "Couldn't open header cache";
     return HE_CACHE;
@@ -563,7 +565,8 @@ bool HttpClient::CheckCache() {
 }
 
 HttpError HttpClient::ReadCacheHeaders(const std::string& id, bool override) {
-  scoped_ptr<StreamInterface> stream(cache_->ReadResource(id, kCacheHeader));
+  std::unique_ptr<StreamInterface> stream(
+      cache_->ReadResource(id, kCacheHeader));
   if (!stream) {
     return HE_CACHE;
   }
@@ -586,7 +589,7 @@ HttpError HttpClient::ReadCacheBody(const std::string& id) {
   HttpError error = HE_NONE;
 
   size_t data_size;
-  scoped_ptr<StreamInterface> stream(cache_->ReadResource(id, kCacheBody));
+  std::unique_ptr<StreamInterface> stream(cache_->ReadResource(id, kCacheBody));
   if (!stream || !stream->GetAvailable(&data_size)) {
     LOG_F(LS_ERROR) << "Unavailable cache body";
     error = HE_CACHE;
@@ -599,7 +602,7 @@ HttpError HttpClient::ReadCacheBody(const std::string& id) {
       && response().document) {
     // Allocate on heap to not explode the stack.
     const int array_size = 1024 * 64;
-    scoped_ptr<char[]> buffer(new char[array_size]);
+    std::unique_ptr<char[]> buffer(new char[array_size]);
     StreamResult result = Flow(stream.get(), buffer.get(), array_size,
                                response().document.get());
     if (SR_SUCCESS != result) {
diff --git a/chromium/third_party/webrtc/base/httpclient.h b/chromium/third_party/webrtc/base/httpclient.h
index e7d2c5ce7d5..0c19d2efb43 100644
--- a/chromium/third_party/webrtc/base/httpclient.h
+++ b/chromium/third_party/webrtc/base/httpclient.h
@@ -11,11 +11,12 @@
 #ifndef WEBRTC_BASE_HTTPCLIENT_H__
 #define WEBRTC_BASE_HTTPCLIENT_H__
 
+#include <memory>
+
 #include "webrtc/base/common.h"
 #include "webrtc/base/httpbase.h"
 #include "webrtc/base/nethelpers.h"
 #include "webrtc/base/proxyinfo.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/socketpool.h"
@@ -172,7 +173,7 @@ private:
   size_t retries_, attempt_, redirects_;
   RedirectAction redirect_action_;
   UriForm uri_form_;
-  scoped_ptr<HttpAuthContext> context_;
+  std::unique_ptr<HttpAuthContext> context_;
   DiskCache* cache_;
   CacheState cache_state_;
   AsyncResolverInterface* resolver_;
diff --git a/chromium/third_party/webrtc/base/httpcommon.h b/chromium/third_party/webrtc/base/httpcommon.h
index addc1bc30d2..3450b58b568 100644
--- a/chromium/third_party/webrtc/base/httpcommon.h
+++ b/chromium/third_party/webrtc/base/httpcommon.h
@@ -12,11 +12,11 @@
 #define WEBRTC_BASE_HTTPCOMMON_H__
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/common.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/stream.h"
 
@@ -292,7 +292,7 @@ struct HttpData {
   typedef HeaderMap::iterator iterator;
 
   HttpVersion version;
-  scoped_ptr<StreamInterface> document;
+  std::unique_ptr<StreamInterface> document;
 
   HttpData();
 
diff --git a/chromium/third_party/webrtc/base/httpserver.h b/chromium/third_party/webrtc/base/httpserver.h
index 30c8f4c5516..c322e81790b 100644
--- a/chromium/third_party/webrtc/base/httpserver.h
+++ b/chromium/third_party/webrtc/base/httpserver.h
@@ -12,6 +12,8 @@
 #define WEBRTC_BASE_HTTPSERVER_H__
 
 #include <map>
+#include <memory>
+
 #include "webrtc/base/httpbase.h"
 
 namespace rtc {
@@ -127,7 +129,7 @@ private:
   void OnConnectionClosed(HttpServer* server, int connection_id,
                           StreamInterface* stream);
 
-  scoped_ptr<AsyncSocket> listener_;
+  std::unique_ptr<AsyncSocket> listener_;
 };
 
 //////////////////////////////////////////////////////////////////////
diff --git a/chromium/third_party/webrtc/base/latebindingsymboltable.h b/chromium/third_party/webrtc/base/latebindingsymboltable.h
index 636e7d07072..794a4ab5a8f 100644
--- a/chromium/third_party/webrtc/base/latebindingsymboltable.h
+++ b/chromium/third_party/webrtc/base/latebindingsymboltable.h
@@ -14,6 +14,7 @@
 #include <string.h>
 
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 
 namespace rtc {
 
diff --git a/chromium/third_party/webrtc/base/linux.h b/chromium/third_party/webrtc/base/linux.h
index ba73b854ba8..b69de3b3701 100644
--- a/chromium/third_party/webrtc/base/linux.h
+++ b/chromium/third_party/webrtc/base/linux.h
@@ -14,9 +14,9 @@
 #if defined(WEBRTC_LINUX)
 #include <string>
 #include <map>
+#include <memory>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stream.h"
 
 namespace rtc {
@@ -51,7 +51,7 @@ class ConfigParser {
   virtual bool ParseLine(std::string* key, std::string* value);
 
  private:
-  scoped_ptr<StreamInterface> instream_;
+  std::unique_ptr<StreamInterface> instream_;
 };
 
 //////////////////////////////////////////////////////////////////////////////
diff --git a/chromium/third_party/webrtc/base/logging.cc b/chromium/third_party/webrtc/base/logging.cc
index 019a31623be..60603624a5e 100644
--- a/chromium/third_party/webrtc/base/logging.cc
+++ b/chromium/third_party/webrtc/base/logging.cc
@@ -42,7 +42,6 @@ static const char kLibjingle[] = "libjingle";
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
@@ -125,7 +124,7 @@ LogMessage::LogMessage(const char* file,
                        const char* module)
     : severity_(sev), tag_(kLibjingle) {
   if (timestamp_) {
-    uint32_t time = TimeSince(LogStartTime());
+    int64_t time = TimeSince(LogStartTime());
     // Also ensure WallClockStartTime is initialized, so that it matches
     // LogStartTime.
     WallClockStartTime();
@@ -171,10 +170,8 @@ LogMessage::LogMessage(const char* file,
 #endif  // WEBRTC_WIN
 #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
       case ERRCTX_OSSTATUS: {
-        tmp << " " << nonnull(GetMacOSStatusErrorString(err), "Unknown error");
-        if (const char* desc = GetMacOSStatusCommentString(err)) {
-          tmp << ": " << desc;
-        }
+        std::string desc(DescriptionFromOSStatus(err));
+        tmp << " " << (desc.empty() ? "Unknown error" : desc.c_str());
         break;
       }
 #endif  // WEBRTC_MAC && !defined(WEBRTC_IOS)
@@ -212,8 +209,8 @@ LogMessage::~LogMessage() {
   }
 }
 
-uint32_t LogMessage::LogStartTime() {
-  static const uint32_t g_start = Time();
+int64_t LogMessage::LogStartTime() {
+  static const int64_t g_start = TimeMillis();
   return g_start;
 }
 
diff --git a/chromium/third_party/webrtc/base/logging.h b/chromium/third_party/webrtc/base/logging.h
index 8c7b3f70e52..631c6384d35 100644
--- a/chromium/third_party/webrtc/base/logging.h
+++ b/chromium/third_party/webrtc/base/logging.h
@@ -46,11 +46,17 @@
 #ifndef WEBRTC_BASE_LOGGING_H_
 #define WEBRTC_BASE_LOGGING_H_
 
+#include <errno.h>
+
 #include <list>
 #include <sstream>
 #include <string>
 #include <utility>
 
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+#include <CoreServices/CoreServices.h>
+#endif
+
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
@@ -80,6 +86,11 @@ struct ConstantLabel { int value; const char * label; };
 const char* FindLabel(int value, const ConstantLabel entries[]);
 std::string ErrorName(int err, const ConstantLabel* err_table);
 
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+// Returns a UTF8 description from an OS X Status error.
+std::string DescriptionFromOSStatus(OSStatus err);
+#endif
+
 //////////////////////////////////////////////////////////////////////
 
 // Note that the non-standard LoggingSeverity aliases exist because they are
@@ -147,7 +158,7 @@ class LogMessage {
   // If this is not called externally, the LogMessage ctor also calls it, in
   // which case the logging start time will be the time of the first LogMessage
   // instance is created.
-  static uint32_t LogStartTime();
+  static int64_t LogStartTime();
 
   // Returns the wall clock equivalent of |LogStartTime|, in seconds from the
   // epoch.
diff --git a/chromium/third_party/webrtc/base/logging_mac.mm b/chromium/third_party/webrtc/base/logging_mac.mm
new file mode 100644
index 00000000000..ffee3541a48
--- /dev/null
+++ b/chromium/third_party/webrtc/base/logging_mac.mm
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/logging.h"
+
+#import <Foundation/Foundation.h>
+
+
+namespace rtc {
+std::string DescriptionFromOSStatus(OSStatus err) {
+  NSError* error =
+      [NSError errorWithDomain:NSOSStatusErrorDomain code:err userInfo:nil];
+  return error.description.UTF8String;
+}
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/logging_unittest.cc b/chromium/third_party/webrtc/base/logging_unittest.cc
index 6047361bf58..d5bd9d4a005 100644
--- a/chromium/third_party/webrtc/base/logging_unittest.cc
+++ b/chromium/third_party/webrtc/base/logging_unittest.cc
@@ -138,18 +138,18 @@ TEST(LogTest, Perf) {
   stream.DisableBuffering();
   LogMessage::AddLogToStream(&stream, LS_SENSITIVE);
 
-  uint32_t start = Time(), finish;
+  int64_t start = TimeMillis(), finish;
   std::string message('X', 80);
   for (int i = 0; i < 1000; ++i) {
     LOG(LS_SENSITIVE) << message;
   }
-  finish = Time();
+  finish = TimeMillis();
 
   LogMessage::RemoveLogToStream(&stream);
   stream.Close();
   Filesystem::DeleteFile(path);
 
-  LOG(LS_INFO) << "Average log time: " << TimeDiff(finish, start) << " us";
+  LOG(LS_INFO) << "Average log time: " << TimeDiff(finish, start) << " ms";
 }
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/logsinks.h b/chromium/third_party/webrtc/base/logsinks.h
index eabf0563986..e75120e3f54 100644
--- a/chromium/third_party/webrtc/base/logsinks.h
+++ b/chromium/third_party/webrtc/base/logsinks.h
@@ -11,12 +11,12 @@
 #ifndef WEBRTC_BASE_FILE_ROTATING_LOG_SINK_H_
 #define WEBRTC_BASE_FILE_ROTATING_LOG_SINK_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/filerotatingstream.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
@@ -46,7 +46,7 @@ class FileRotatingLogSink : public LogSink {
   explicit FileRotatingLogSink(FileRotatingStream* stream);
 
  private:
-  scoped_ptr<FileRotatingStream> stream_;
+  std::unique_ptr<FileRotatingStream> stream_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(FileRotatingLogSink);
 };
diff --git a/chromium/third_party/webrtc/base/macasyncsocket.h b/chromium/third_party/webrtc/base/macasyncsocket.h
index 5861ee3276a..c0f57b948c6 100644
--- a/chromium/third_party/webrtc/base/macasyncsocket.h
+++ b/chromium/third_party/webrtc/base/macasyncsocket.h
@@ -19,6 +19,7 @@
 #include <CoreFoundation/CoreFoundation.h>
 
 #include "webrtc/base/asyncsocket.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/nethelpers.h"
 
 namespace rtc {
diff --git a/chromium/third_party/webrtc/base/maccocoasocketserver.h b/chromium/third_party/webrtc/base/maccocoasocketserver.h
index 0acf8d757a5..3e07a452e67 100644
--- a/chromium/third_party/webrtc/base/maccocoasocketserver.h
+++ b/chromium/third_party/webrtc/base/maccocoasocketserver.h
@@ -13,6 +13,7 @@
 #ifndef WEBRTC_BASE_MACCOCOASOCKETSERVER_H_
 #define WEBRTC_BASE_MACCOCOASOCKETSERVER_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/macsocketserver.h"
 
 #ifdef __OBJC__
diff --git a/chromium/third_party/webrtc/base/maccocoasocketserver_unittest.mm b/chromium/third_party/webrtc/base/maccocoasocketserver_unittest.mm
index 5401ffb329a..3d9e4da6997 100644
--- a/chromium/third_party/webrtc/base/maccocoasocketserver_unittest.mm
+++ b/chromium/third_party/webrtc/base/maccocoasocketserver_unittest.mm
@@ -9,7 +9,6 @@
  */
 
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/maccocoasocketserver.h"
 
diff --git a/chromium/third_party/webrtc/base/macifaddrs_converter.cc b/chromium/third_party/webrtc/base/macifaddrs_converter.cc
index 0916cb5ba27..2ad070e8eab 100644
--- a/chromium/third_party/webrtc/base/macifaddrs_converter.cc
+++ b/chromium/third_party/webrtc/base/macifaddrs_converter.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include <net/if.h>
 #include <sys/ioctl.h>
 #include <unistd.h>
@@ -15,7 +17,6 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/ifaddrs_converter.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 #if !defined(WEBRTC_IOS)
 #include <net/if_media.h>
@@ -269,7 +270,7 @@ class MacIfAddrsConverter : public IfAddrsConverter {
   }
 
  private:
-  rtc::scoped_ptr<IPv6AttributesGetter> ip_attribute_getter_;
+  std::unique_ptr<IPv6AttributesGetter> ip_attribute_getter_;
 };
 
 }  // namespace
diff --git a/chromium/third_party/webrtc/base/macsocketserver_unittest.cc b/chromium/third_party/webrtc/base/macsocketserver_unittest.cc
index ecb9a706b71..87cfe07b2f1 100644
--- a/chromium/third_party/webrtc/base/macsocketserver_unittest.cc
+++ b/chromium/third_party/webrtc/base/macsocketserver_unittest.cc
@@ -8,8 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socket_unittest.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/macsocketserver.h"
@@ -98,7 +99,7 @@ class MacAsyncSocketTest : public SocketTest {
   virtual MacBaseSocketServer* CreateSocketServer() {
     return new MacCFSocketServer();
   };
-  rtc::scoped_ptr<MacBaseSocketServer> server_;
+  std::unique_ptr<MacBaseSocketServer> server_;
   SocketServerScope scope_;
 };
 
diff --git a/chromium/third_party/webrtc/base/macutils.cc b/chromium/third_party/webrtc/base/macutils.cc
index 7b1ff475368..74b49199647 100644
--- a/chromium/third_party/webrtc/base/macutils.cc
+++ b/chromium/third_party/webrtc/base/macutils.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <sstream>
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/macutils.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringutils.h"
 
 namespace rtc {
@@ -26,7 +26,7 @@ bool ToUtf8(const CFStringRef str16, std::string* str8) {
   }
   size_t maxlen = CFStringGetMaximumSizeForEncoding(CFStringGetLength(str16),
                                                     kCFStringEncodingUTF8) + 1;
-  scoped_ptr<char[]> buffer(new char[maxlen]);
+  std::unique_ptr<char[]> buffer(new char[maxlen]);
   if (!buffer || !CFStringGetCString(str16, buffer.get(), maxlen,
                                      kCFStringEncodingUTF8)) {
     return false;
diff --git a/chromium/third_party/webrtc/base/messagedigest.cc b/chromium/third_party/webrtc/base/messagedigest.cc
index 0c2b4a16ac9..c08cab4ea9d 100644
--- a/chromium/third_party/webrtc/base/messagedigest.cc
+++ b/chromium/third_party/webrtc/base/messagedigest.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/base/messagedigest.h"
 
+#include <memory>
+
 #include <string.h>
 
 #include "webrtc/base/basictypes.h"
@@ -20,7 +22,6 @@
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/sha1digest.h"
 #endif
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringencode.h"
 
 namespace rtc {
@@ -75,14 +76,14 @@ size_t ComputeDigest(MessageDigest* digest, const void* input, size_t in_len,
 
 size_t ComputeDigest(const std::string& alg, const void* input, size_t in_len,
                      void* output, size_t out_len) {
-  scoped_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
+  std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
   return (digest) ?
       ComputeDigest(digest.get(), input, in_len, output, out_len) :
       0;
 }
 
 std::string ComputeDigest(MessageDigest* digest, const std::string& input) {
-  scoped_ptr<char[]> output(new char[digest->Size()]);
+  std::unique_ptr<char[]> output(new char[digest->Size()]);
   ComputeDigest(digest, input.data(), input.size(),
                 output.get(), digest->Size());
   return hex_encode(output.get(), digest->Size());
@@ -90,7 +91,7 @@ std::string ComputeDigest(MessageDigest* digest, const std::string& input) {
 
 bool ComputeDigest(const std::string& alg, const std::string& input,
                    std::string* output) {
-  scoped_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
+  std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
   if (!digest) {
     return false;
   }
@@ -117,7 +118,7 @@ size_t ComputeHmac(MessageDigest* digest,
   }
   // Copy the key to a block-sized buffer to simplify padding.
   // If the key is longer than a block, hash it and use the result instead.
-  scoped_ptr<uint8_t[]> new_key(new uint8_t[block_len]);
+  std::unique_ptr<uint8_t[]> new_key(new uint8_t[block_len]);
   if (key_len > block_len) {
     ComputeDigest(digest, key, key_len, new_key.get(), block_len);
     memset(new_key.get() + digest->Size(), 0, block_len - digest->Size());
@@ -126,14 +127,14 @@ size_t ComputeHmac(MessageDigest* digest,
     memset(new_key.get() + key_len, 0, block_len - key_len);
   }
   // Set up the padding from the key, salting appropriately for each padding.
-  scoped_ptr<uint8_t[]> o_pad(new uint8_t[block_len]);
-  scoped_ptr<uint8_t[]> i_pad(new uint8_t[block_len]);
+  std::unique_ptr<uint8_t[]> o_pad(new uint8_t[block_len]);
+  std::unique_ptr<uint8_t[]> i_pad(new uint8_t[block_len]);
   for (size_t i = 0; i < block_len; ++i) {
     o_pad[i] = 0x5c ^ new_key[i];
     i_pad[i] = 0x36 ^ new_key[i];
   }
   // Inner hash; hash the inner padding, and then the input buffer.
-  scoped_ptr<uint8_t[]> inner(new uint8_t[digest->Size()]);
+  std::unique_ptr<uint8_t[]> inner(new uint8_t[digest->Size()]);
   digest->Update(i_pad.get(), block_len);
   digest->Update(input, in_len);
   digest->Finish(inner.get(), digest->Size());
@@ -146,7 +147,7 @@ size_t ComputeHmac(MessageDigest* digest,
 size_t ComputeHmac(const std::string& alg, const void* key, size_t key_len,
                    const void* input, size_t in_len,
                    void* output, size_t out_len) {
-  scoped_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
+  std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
   if (!digest) {
     return 0;
   }
@@ -156,7 +157,7 @@ size_t ComputeHmac(const std::string& alg, const void* key, size_t key_len,
 
 std::string ComputeHmac(MessageDigest* digest, const std::string& key,
                         const std::string& input) {
-  scoped_ptr<char[]> output(new char[digest->Size()]);
+  std::unique_ptr<char[]> output(new char[digest->Size()]);
   ComputeHmac(digest, key.data(), key.size(),
               input.data(), input.size(), output.get(), digest->Size());
   return hex_encode(output.get(), digest->Size());
@@ -164,7 +165,7 @@ std::string ComputeHmac(MessageDigest* digest, const std::string& key,
 
 bool ComputeHmac(const std::string& alg, const std::string& key,
                  const std::string& input, std::string* output) {
-  scoped_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
+  std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
   if (!digest) {
     return false;
   }
diff --git a/chromium/third_party/webrtc/base/messagehandler.h b/chromium/third_party/webrtc/base/messagehandler.h
index b55b229a6db..6a3c2ef7402 100644
--- a/chromium/third_party/webrtc/base/messagehandler.h
+++ b/chromium/third_party/webrtc/base/messagehandler.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_BASE_MESSAGEHANDLER_H_
 #define WEBRTC_BASE_MESSAGEHANDLER_H_
 
+#include <memory>
 #include <utility>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
@@ -50,18 +50,18 @@ class FunctorMessageHandler : public MessageHandler {
   ReturnT result_;
 };
 
-// Specialization for rtc::scoped_ptr<ReturnT>.
+// Specialization for std::unique_ptr<ReturnT>.
 template <class ReturnT, class FunctorT>
-class FunctorMessageHandler<class rtc::scoped_ptr<ReturnT>, FunctorT>
+class FunctorMessageHandler<class std::unique_ptr<ReturnT>, FunctorT>
     : public MessageHandler {
  public:
   explicit FunctorMessageHandler(const FunctorT& functor) : functor_(functor) {}
   virtual void OnMessage(Message* msg) { result_ = std::move(functor_()); }
-  rtc::scoped_ptr<ReturnT> result() { return std::move(result_); }
+  std::unique_ptr<ReturnT> result() { return std::move(result_); }
 
  private:
   FunctorT functor_;
-  rtc::scoped_ptr<ReturnT> result_;
+  std::unique_ptr<ReturnT> result_;
 };
 
 // Specialization for ReturnT of void.
diff --git a/chromium/third_party/webrtc/base/messagequeue.cc b/chromium/third_party/webrtc/base/messagequeue.cc
index 61aa61192bb..e8b5bf5bc1f 100644
--- a/chromium/third_party/webrtc/base/messagequeue.cc
+++ b/chromium/third_party/webrtc/base/messagequeue.cc
@@ -7,27 +7,17 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-
-#if defined(WEBRTC_POSIX)
-#include <sys/time.h>
-#endif
-
 #include <algorithm>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagequeue.h"
-#if defined(__native_client__)
-#include "webrtc/base/nullsocketserver.h"
-typedef rtc::NullSocketServer DefaultSocketServer;
-#else
-#include "webrtc/base/physicalsocketserver.h"
-typedef rtc::PhysicalSocketServer DefaultSocketServer;
-#endif
+#include "webrtc/base/trace_event.h"
 
 namespace rtc {
 
-const uint32_t kMaxMsgLatency = 150;  // 150 ms
+const int kMaxMsgLatency = 150;  // 150 ms
 
 //------------------------------------------------------------------
 // MessageQueueManager
@@ -115,25 +105,26 @@ void MessageQueueManager::ClearInternal(MessageHandler *handler) {
 
 //------------------------------------------------------------------
 // MessageQueue
-
 MessageQueue::MessageQueue(SocketServer* ss, bool init_queue)
     : fStop_(false), fPeekKeep_(false),
       dmsgq_next_num_(0), fInitialized_(false), fDestroyed_(false), ss_(ss) {
-  if (!ss_) {
-    // Currently, MessageQueue holds a socket server, and is the base class for
-    // Thread.  It seems like it makes more sense for Thread to hold the socket
-    // server, and provide it to the MessageQueue, since the Thread controls
-    // the I/O model, and MQ is agnostic to those details.  Anyway, this causes
-    // messagequeue_unittest to depend on network libraries... yuck.
-    default_ss_.reset(new DefaultSocketServer());
-    ss_ = default_ss_.get();
-  }
+  RTC_DCHECK(ss);
+  // Currently, MessageQueue holds a socket server, and is the base class for
+  // Thread.  It seems like it makes more sense for Thread to hold the socket
+  // server, and provide it to the MessageQueue, since the Thread controls
+  // the I/O model, and MQ is agnostic to those details.  Anyway, this causes
+  // messagequeue_unittest to depend on network libraries... yuck.
   ss_->SetMessageQueue(this);
   if (init_queue) {
     DoInit();
   }
 }
 
+MessageQueue::MessageQueue(std::unique_ptr<SocketServer> ss, bool init_queue)
+    : MessageQueue(ss.get(), init_queue) {
+  own_ss_ = std::move(ss);
+}
+
 MessageQueue::~MessageQueue() {
   DoDestroy();
 }
@@ -178,7 +169,7 @@ void MessageQueue::set_socketserver(SocketServer* ss) {
   // Other places that only read "ss_" can use a shared lock as simultaneous
   // read access is allowed.
   ExclusiveScope es(&ss_lock_);
-  ss_ = ss ? ss : default_ss_.get();
+  ss_ = ss ? ss : own_ss_.get();
   ss_->SetMessageQueue(this);
 }
 
@@ -224,16 +215,16 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
 
   // Get w/wait + timer scan / dispatch + socket / event multiplexer dispatch
 
-  int cmsTotal = cmsWait;
-  int cmsElapsed = 0;
-  uint32_t msStart = Time();
-  uint32_t msCurrent = msStart;
+  int64_t cmsTotal = cmsWait;
+  int64_t cmsElapsed = 0;
+  int64_t msStart = TimeMillis();
+  int64_t msCurrent = msStart;
   while (true) {
     // Check for sent messages
     ReceiveSends();
 
     // Check for posted events
-    int cmsDelayNext = kForever;
+    int64_t cmsDelayNext = kForever;
     bool first_pass = true;
     while (true) {
       // All queue operations need to be locked, but nothing else in this loop
@@ -246,7 +237,7 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
         if (first_pass) {
           first_pass = false;
           while (!dmsgq_.empty()) {
-            if (TimeIsLater(msCurrent, dmsgq_.top().msTrigger_)) {
+            if (msCurrent < dmsgq_.top().msTrigger_) {
               cmsDelayNext = TimeDiff(dmsgq_.top().msTrigger_, msCurrent);
               break;
             }
@@ -265,7 +256,7 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
 
       // Log a warning for time-sensitive messages that we're late to deliver.
       if (pmsg->ts_sensitive) {
-        int32_t delay = TimeDiff(msCurrent, pmsg->ts_sensitive);
+        int64_t delay = TimeDiff(msCurrent, pmsg->ts_sensitive);
         if (delay > 0) {
           LOG_F(LS_WARNING) << "id: " << pmsg->message_id << "  delay: "
                             << (delay + kMaxMsgLatency) << "ms";
@@ -286,11 +277,11 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
 
     // Which is shorter, the delay wait or the asked wait?
 
-    int cmsNext;
+    int64_t cmsNext;
     if (cmsWait == kForever) {
       cmsNext = cmsDelayNext;
     } else {
-      cmsNext = std::max(0, cmsTotal - cmsElapsed);
+      cmsNext = std::max<int64_t>(0, cmsTotal - cmsElapsed);
       if ((cmsDelayNext != kForever) && (cmsDelayNext < cmsNext))
         cmsNext = cmsDelayNext;
     }
@@ -298,13 +289,13 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
     {
       // Wait and multiplex in the meantime
       SharedScope ss(&ss_lock_);
-      if (!ss_->Wait(cmsNext, process_io))
+      if (!ss_->Wait(static_cast<int>(cmsNext), process_io))
         return false;
     }
 
     // If the specified timeout expired, return
 
-    msCurrent = Time();
+    msCurrent = TimeMillis();
     cmsElapsed = TimeDiff(msCurrent, msStart);
     if (cmsWait != kForever) {
       if (cmsElapsed >= cmsWait)
@@ -335,7 +326,7 @@ void MessageQueue::Post(MessageHandler* phandler,
     msg.message_id = id;
     msg.pdata = pdata;
     if (time_sensitive) {
-      msg.ts_sensitive = Time() + kMaxMsgLatency;
+      msg.ts_sensitive = TimeMillis() + kMaxMsgLatency;
     }
     msgq_.push_back(msg);
   }
@@ -353,11 +344,20 @@ void MessageQueue::PostAt(uint32_t tstamp,
                           MessageHandler* phandler,
                           uint32_t id,
                           MessageData* pdata) {
+  // This should work even if it is used (unexpectedly).
+  int delay = static_cast<uint32_t>(TimeMillis()) - tstamp;
+  return DoDelayPost(delay, tstamp, phandler, id, pdata);
+}
+
+void MessageQueue::PostAt(int64_t tstamp,
+                          MessageHandler* phandler,
+                          uint32_t id,
+                          MessageData* pdata) {
   return DoDelayPost(TimeUntil(tstamp), tstamp, phandler, id, pdata);
 }
 
 void MessageQueue::DoDelayPost(int cmsDelay,
-                               uint32_t tstamp,
+                               int64_t tstamp,
                                MessageHandler* phandler,
                                uint32_t id,
                                MessageData* pdata) {
@@ -451,6 +451,7 @@ void MessageQueue::Clear(MessageHandler* phandler,
 }
 
 void MessageQueue::Dispatch(Message *pmsg) {
+  TRACE_EVENT0("webrtc", "MessageQueue::Dispatch");
   pmsg->phandler->OnMessage(pmsg);
 }
 
diff --git a/chromium/third_party/webrtc/base/messagequeue.h b/chromium/third_party/webrtc/base/messagequeue.h
index efc479cf264..4aa96eb531f 100644
--- a/chromium/third_party/webrtc/base/messagequeue.h
+++ b/chromium/third_party/webrtc/base/messagequeue.h
@@ -15,6 +15,7 @@
 
 #include <algorithm>
 #include <list>
+#include <memory>
 #include <queue>
 #include <vector>
 
@@ -22,7 +23,6 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagehandler.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/sharedexclusivelock.h"
 #include "webrtc/base/sigslot.h"
@@ -89,10 +89,11 @@ template <class T>
 class ScopedMessageData : public MessageData {
  public:
   explicit ScopedMessageData(T* data) : data_(data) { }
-  const scoped_ptr<T>& data() const { return data_; }
-  scoped_ptr<T>& data() { return data_; }
+  const std::unique_ptr<T>& data() const { return data_; }
+  std::unique_ptr<T>& data() { return data_; }
+
  private:
-  scoped_ptr<T> data_;
+  std::unique_ptr<T> data_;
 };
 
 // Like ScopedMessageData, but for reference counted pointers.
@@ -141,7 +142,7 @@ struct Message {
   MessageHandler *phandler;
   uint32_t message_id;
   MessageData *pdata;
-  uint32_t ts_sensitive;
+  int64_t ts_sensitive;
 };
 
 typedef std::list<Message> MessageList;
@@ -151,7 +152,7 @@ typedef std::list<Message> MessageList;
 
 class DelayedMessage {
  public:
-  DelayedMessage(int delay, uint32_t trigger, uint32_t num, const Message& msg)
+  DelayedMessage(int delay, int64_t trigger, uint32_t num, const Message& msg)
       : cmsDelay_(delay), msTrigger_(trigger), num_(num), msg_(msg) {}
 
   bool operator< (const DelayedMessage& dmsg) const {
@@ -160,7 +161,7 @@ class DelayedMessage {
   }
 
   int cmsDelay_;  // for debugging
-  uint32_t msTrigger_;
+  int64_t msTrigger_;
   uint32_t num_;
   Message msg_;
 };
@@ -174,8 +175,8 @@ class MessageQueue {
   // init_queue and call DoInit() from their constructor to prevent races
   // with the MessageQueueManager using the object while the vtable is still
   // being created.
-  explicit MessageQueue(SocketServer* ss = NULL,
-                        bool init_queue = true);
+  MessageQueue(SocketServer* ss, bool init_queue);
+  MessageQueue(std::unique_ptr<SocketServer> ss, bool init_queue);
 
   // NOTE: SUBCLASSES OF MessageQueue THAT OVERRIDE Clear MUST CALL
   // DoDestroy() IN THEIR DESTRUCTORS! This is required to avoid a data race
@@ -211,6 +212,11 @@ class MessageQueue {
                            MessageHandler* phandler,
                            uint32_t id = 0,
                            MessageData* pdata = NULL);
+  virtual void PostAt(int64_t tstamp,
+                      MessageHandler* phandler,
+                      uint32_t id = 0,
+                      MessageData* pdata = NULL);
+  // TODO(honghaiz): Remove this when all the dependencies are removed.
   virtual void PostAt(uint32_t tstamp,
                       MessageHandler* phandler,
                       uint32_t id = 0,
@@ -249,7 +255,7 @@ class MessageQueue {
   };
 
   void DoDelayPost(int cmsDelay,
-                   uint32_t tstamp,
+                   int64_t tstamp,
                    MessageHandler* phandler,
                    uint32_t id,
                    MessageData* pdata);
@@ -275,13 +281,13 @@ class MessageQueue {
   bool fDestroyed_;
 
  private:
-  // The SocketServer is not owned by MessageQueue.
+  // The SocketServer might not be owned by MessageQueue.
   SocketServer* ss_ GUARDED_BY(ss_lock_);
-  // If a server isn't supplied in the constructor, use this one.
-  scoped_ptr<SocketServer> default_ss_;
+  // Used if SocketServer ownership lies with |this|.
+  std::unique_ptr<SocketServer> own_ss_;
   SharedExclusiveLock ss_lock_;
 
-  RTC_DISALLOW_COPY_AND_ASSIGN(MessageQueue);
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageQueue);
 };
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/messagequeue_unittest.cc b/chromium/third_party/webrtc/base/messagequeue_unittest.cc
index 78024e0b2d6..50c2ad0ba2e 100644
--- a/chromium/third_party/webrtc/base/messagequeue_unittest.cc
+++ b/chromium/third_party/webrtc/base/messagequeue_unittest.cc
@@ -21,6 +21,7 @@ using namespace rtc;
 
 class MessageQueueTest: public testing::Test, public MessageQueue {
  public:
+  MessageQueueTest() : MessageQueue(SocketServer::CreateDefault(), true) {}
   bool IsLocked_Worker() {
     if (!crit_.TryEnter()) {
       return true;
@@ -53,7 +54,7 @@ struct DeletedLockChecker {
 static void DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder(
     MessageQueue* q) {
   EXPECT_TRUE(q != NULL);
-  TimeStamp now = Time();
+  int64_t now = TimeMillis();
   q->PostAt(now, NULL, 3);
   q->PostAt(now - 2, NULL, 0);
   q->PostAt(now - 1, NULL, 1);
@@ -72,10 +73,11 @@ static void DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder(
 
 TEST_F(MessageQueueTest,
        DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder) {
-  MessageQueue q;
+  MessageQueue q(SocketServer::CreateDefault(), true);
   DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder(&q);
+
   NullSocketServer nullss;
-  MessageQueue q_nullss(&nullss);
+  MessageQueue q_nullss(&nullss, true);
   DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder(&q_nullss);
 }
 
diff --git a/chromium/third_party/webrtc/base/multipart.h b/chromium/third_party/webrtc/base/multipart.h
index a099230cc50..9ae3ca2adf0 100644
--- a/chromium/third_party/webrtc/base/multipart.h
+++ b/chromium/third_party/webrtc/base/multipart.h
@@ -14,6 +14,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/stream.h"
 
diff --git a/chromium/third_party/webrtc/base/multipart_unittest.cc b/chromium/third_party/webrtc/base/multipart_unittest.cc
index 9db316b15de..627d1c620ad 100644
--- a/chromium/third_party/webrtc/base/multipart_unittest.cc
+++ b/chromium/third_party/webrtc/base/multipart_unittest.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/pathutils.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/multipart.h"
 
 namespace rtc {
@@ -61,7 +61,7 @@ TEST(MultipartTest, TestAddAndRead) {
   EXPECT_TRUE(multipart.GetSize(&size));
   EXPECT_EQ(part_size, size);
 
-  rtc::scoped_ptr<rtc::MemoryStream> stream(
+  std::unique_ptr<rtc::MemoryStream> stream(
       new rtc::MemoryStream(kTestStreamContent));
   size_t stream_size = 0;
   EXPECT_TRUE(stream->GetSize(&stream_size));
diff --git a/chromium/third_party/webrtc/base/nat_unittest.cc b/chromium/third_party/webrtc/base/nat_unittest.cc
index 8be1be9f05a..ca72c9356ae 100644
--- a/chromium/third_party/webrtc/base/nat_unittest.cc
+++ b/chromium/third_party/webrtc/base/nat_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include <algorithm>
+#include <memory>
 #include <string>
 
 #include "webrtc/base/gunit.h"
@@ -178,11 +179,11 @@ bool TestConnectivity(const SocketAddress& src, const IPAddress& dst) {
   // The physical NAT tests require connectivity to the selected ip from the
   // internal address used for the NAT. Things like firewalls can break that, so
   // check to see if it's worth even trying with this ip.
-  scoped_ptr<PhysicalSocketServer> pss(new PhysicalSocketServer());
-  scoped_ptr<AsyncSocket> client(pss->CreateAsyncSocket(src.family(),
-                                                        SOCK_DGRAM));
-  scoped_ptr<AsyncSocket> server(pss->CreateAsyncSocket(src.family(),
-                                                        SOCK_DGRAM));
+  std::unique_ptr<PhysicalSocketServer> pss(new PhysicalSocketServer());
+  std::unique_ptr<AsyncSocket> client(
+      pss->CreateAsyncSocket(src.family(), SOCK_DGRAM));
+  std::unique_ptr<AsyncSocket> server(
+      pss->CreateAsyncSocket(src.family(), SOCK_DGRAM));
   if (client->Bind(SocketAddress(src.ipaddr(), 0)) != 0 ||
       server->Bind(SocketAddress(dst, 0)) != 0) {
     return false;
@@ -244,8 +245,8 @@ void TestPhysicalInternal(const SocketAddress& int_addr) {
       SocketAddress(ext_addr2)
   };
 
-  scoped_ptr<PhysicalSocketServer> int_pss(new PhysicalSocketServer());
-  scoped_ptr<PhysicalSocketServer> ext_pss(new PhysicalSocketServer());
+  std::unique_ptr<PhysicalSocketServer> int_pss(new PhysicalSocketServer());
+  std::unique_ptr<PhysicalSocketServer> ext_pss(new PhysicalSocketServer());
 
   TestBindings(int_pss.get(), int_addr, ext_pss.get(), ext_addrs);
   TestFilters(int_pss.get(), int_addr, ext_pss.get(), ext_addrs);
@@ -274,16 +275,16 @@ class TestVirtualSocketServer : public VirtualSocketServer {
   IPAddress GetNextIP(int af) { return VirtualSocketServer::GetNextIP(af); }
 
  private:
-  scoped_ptr<SocketServer> ss_;
+  std::unique_ptr<SocketServer> ss_;
 };
 
 }  // namespace
 
 void TestVirtualInternal(int family) {
-  scoped_ptr<TestVirtualSocketServer> int_vss(new TestVirtualSocketServer(
-      new PhysicalSocketServer()));
-  scoped_ptr<TestVirtualSocketServer> ext_vss(new TestVirtualSocketServer(
-      new PhysicalSocketServer()));
+  std::unique_ptr<TestVirtualSocketServer> int_vss(
+      new TestVirtualSocketServer(new PhysicalSocketServer()));
+  std::unique_ptr<TestVirtualSocketServer> ext_vss(
+      new TestVirtualSocketServer(new PhysicalSocketServer()));
 
   SocketAddress int_addr;
   SocketAddress ext_addrs[4];
@@ -351,15 +352,15 @@ class NatTcpTest : public testing::Test, public sigslot::has_slots<> {
   bool connected_;
   PhysicalSocketServer* int_pss_;
   PhysicalSocketServer* ext_pss_;
-  rtc::scoped_ptr<TestVirtualSocketServer> int_vss_;
-  rtc::scoped_ptr<TestVirtualSocketServer> ext_vss_;
-  rtc::scoped_ptr<Thread> int_thread_;
-  rtc::scoped_ptr<Thread> ext_thread_;
-  rtc::scoped_ptr<NATServer> nat_;
-  rtc::scoped_ptr<NATSocketFactory> natsf_;
-  rtc::scoped_ptr<AsyncSocket> client_;
-  rtc::scoped_ptr<AsyncSocket> server_;
-  rtc::scoped_ptr<AsyncSocket> accepted_;
+  std::unique_ptr<TestVirtualSocketServer> int_vss_;
+  std::unique_ptr<TestVirtualSocketServer> ext_vss_;
+  std::unique_ptr<Thread> int_thread_;
+  std::unique_ptr<Thread> ext_thread_;
+  std::unique_ptr<NATServer> nat_;
+  std::unique_ptr<NATSocketFactory> natsf_;
+  std::unique_ptr<AsyncSocket> client_;
+  std::unique_ptr<AsyncSocket> server_;
+  std::unique_ptr<AsyncSocket> accepted_;
 };
 
 TEST_F(NatTcpTest, DISABLED_TestConnectOut) {
@@ -377,8 +378,8 @@ TEST_F(NatTcpTest, DISABLED_TestConnectOut) {
   EXPECT_EQ(client_->GetRemoteAddress(), server_->GetLocalAddress());
   EXPECT_EQ(accepted_->GetRemoteAddress().ipaddr(), ext_addr_.ipaddr());
 
-  rtc::scoped_ptr<rtc::TestClient> in(CreateTCPTestClient(client_.release()));
-  rtc::scoped_ptr<rtc::TestClient> out(
+  std::unique_ptr<rtc::TestClient> in(CreateTCPTestClient(client_.release()));
+  std::unique_ptr<rtc::TestClient> out(
       CreateTCPTestClient(accepted_.release()));
 
   const char* buf = "test_packet";
diff --git a/chromium/third_party/webrtc/base/natserver.cc b/chromium/third_party/webrtc/base/natserver.cc
index b071e014dbd..222d2709e27 100644
--- a/chromium/third_party/webrtc/base/natserver.cc
+++ b/chromium/third_party/webrtc/base/natserver.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/natsocketfactory.h"
 #include "webrtc/base/natserver.h"
 #include "webrtc/base/logging.h"
@@ -195,7 +197,7 @@ void NATServer::OnExternalUDPPacket(
 
   // Forward this packet to the internal address.
   // First prepend the address in a quasi-STUN format.
-  scoped_ptr<char[]> real_buf(new char[size + kNATEncodedIPv6AddressSize]);
+  std::unique_ptr<char[]> real_buf(new char[size + kNATEncodedIPv6AddressSize]);
   size_t addrlength = PackAddressForNAT(real_buf.get(),
                                         size + kNATEncodedIPv6AddressSize,
                                         remote_addr);
diff --git a/chromium/third_party/webrtc/base/natserver.h b/chromium/third_party/webrtc/base/natserver.h
index b6a02feca38..460518bd2e4 100644
--- a/chromium/third_party/webrtc/base/natserver.h
+++ b/chromium/third_party/webrtc/base/natserver.h
@@ -15,6 +15,7 @@
 #include <set>
 
 #include "webrtc/base/asyncudpsocket.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/socketaddresspair.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/socketfactory.h"
diff --git a/chromium/third_party/webrtc/base/natsocketfactory.cc b/chromium/third_party/webrtc/base/natsocketfactory.cc
index 0abd2a1b056..985748cff90 100644
--- a/chromium/third_party/webrtc/base/natsocketfactory.cc
+++ b/chromium/third_party/webrtc/base/natsocketfactory.cc
@@ -141,7 +141,7 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
       return socket_->SendTo(data, size, addr);
     }
     // This array will be too large for IPv4 packets, but only by 12 bytes.
-    scoped_ptr<char[]> buf(new char[size + kNATEncodedIPv6AddressSize]);
+    std::unique_ptr<char[]> buf(new char[size + kNATEncodedIPv6AddressSize]);
     size_t addrlength = PackAddressForNAT(buf.get(),
                                           size + kNATEncodedIPv6AddressSize,
                                           addr);
diff --git a/chromium/third_party/webrtc/base/natsocketfactory.h b/chromium/third_party/webrtc/base/natsocketfactory.h
index 9ca0739440e..6fad30c5602 100644
--- a/chromium/third_party/webrtc/base/natsocketfactory.h
+++ b/chromium/third_party/webrtc/base/natsocketfactory.h
@@ -13,8 +13,10 @@
 
 #include <string>
 #include <map>
+#include <memory>
 #include <set>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/natserver.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/socketserver.h"
@@ -116,8 +118,8 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
 
    private:
     NATSocketServer* server_;
-    scoped_ptr<SocketFactory> internal_factory_;
-    scoped_ptr<NATServer> nat_server_;
+    std::unique_ptr<SocketFactory> internal_factory_;
+    std::unique_ptr<NATServer> nat_server_;
     TranslatorMap nats_;
     std::set<SocketAddress> clients_;
   };
diff --git a/chromium/third_party/webrtc/base/nethelpers.cc b/chromium/third_party/webrtc/base/nethelpers.cc
index 0c7cce6b7c3..d9015283b0b 100644
--- a/chromium/third_party/webrtc/base/nethelpers.cc
+++ b/chromium/third_party/webrtc/base/nethelpers.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/base/nethelpers.h"
 
+#include <memory>
+
 #if defined(WEBRTC_WIN)
 #include <ws2spi.h>
 #include <ws2tcpip.h>
@@ -127,7 +129,7 @@ bool HasIPv6Enabled() {
     return false;
   }
   DWORD protbuff_size = 4096;
-  scoped_ptr<char[]> protocols;
+  std::unique_ptr<char[]> protocols;
   LPWSAPROTOCOL_INFOW protocol_infos = NULL;
   int requested_protocols[2] = {AF_INET6, 0};
 
diff --git a/chromium/third_party/webrtc/base/network.cc b/chromium/third_party/webrtc/base/network.cc
index 00187794704..b6caaa8bbf0 100644
--- a/chromium/third_party/webrtc/base/network.cc
+++ b/chromium/third_party/webrtc/base/network.cc
@@ -32,10 +32,10 @@
 #include <stdio.h>
 
 #include <algorithm>
+#include <memory>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/networkmonitor.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socket.h"  // includes something that makes windows happy
 #include "webrtc/base/stream.h"
 #include "webrtc/base/stringencode.h"
@@ -357,12 +357,34 @@ bool NetworkManagerBase::GetDefaultLocalAddress(int family,
     *ipaddr = default_local_ipv4_address_;
     return true;
   } else if (family == AF_INET6 && !default_local_ipv6_address_.IsNil()) {
-    *ipaddr = default_local_ipv6_address_;
+    Network* ipv6_network = GetNetworkFromAddress(default_local_ipv6_address_);
+    if (ipv6_network) {
+      // If the default ipv6 network's BestIP is different than
+      // default_local_ipv6_address_, use it instead.
+      // This is to prevent potential IP address leakage. See WebRTC bug 5376.
+      *ipaddr = ipv6_network->GetBestIP();
+    } else {
+      *ipaddr = default_local_ipv6_address_;
+    }
     return true;
   }
   return false;
 }
 
+Network* NetworkManagerBase::GetNetworkFromAddress(
+    const rtc::IPAddress& ip) const {
+  for (Network* network : networks_) {
+    const auto& ips = network->GetIPs();
+    if (std::find_if(ips.begin(), ips.end(),
+                     [ip](const InterfaceAddress& existing_ip) {
+                       return ip == static_cast<rtc::IPAddress>(existing_ip);
+                     }) != ips.end()) {
+      return network;
+    }
+  }
+  return nullptr;
+}
+
 BasicNetworkManager::BasicNetworkManager()
     : thread_(NULL), sent_first_update_(false), start_count_(0),
       ignore_non_default_routes_(false) {
@@ -449,9 +471,9 @@ void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
       }
 #endif
       // TODO(phoglund): Need to recognize other types as well.
-      scoped_ptr<Network> network(new Network(cursor->ifa_name,
-                                              cursor->ifa_name, prefix,
-                                              prefix_length, adapter_type));
+      std::unique_ptr<Network> network(
+          new Network(cursor->ifa_name, cursor->ifa_name, prefix, prefix_length,
+                      adapter_type));
       network->set_default_local_address_provider(this);
       network->set_scope_id(scope_id);
       network->AddIP(ip);
@@ -475,7 +497,7 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
     return false;
   }
 
-  rtc::scoped_ptr<IfAddrsConverter> ifaddrs_converter(CreateIfAddrsConverter());
+  std::unique_ptr<IfAddrsConverter> ifaddrs_converter(CreateIfAddrsConverter());
   ConvertIfAddrs(interfaces, ifaddrs_converter.get(), include_ignored,
                  networks);
 
@@ -531,7 +553,7 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
   NetworkMap current_networks;
   // MSDN recommends a 15KB buffer for the first try at GetAdaptersAddresses.
   size_t buffer_size = 16384;
-  scoped_ptr<char[]> adapter_info(new char[buffer_size]);
+  std::unique_ptr<char[]> adapter_info(new char[buffer_size]);
   PIP_ADAPTER_ADDRESSES adapter_addrs =
       reinterpret_cast<PIP_ADAPTER_ADDRESSES>(adapter_info.get());
   int adapter_flags = (GAA_FLAG_SKIP_DNS_SERVER | GAA_FLAG_SKIP_ANYCAST |
@@ -567,7 +589,7 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
 
         IPAddress ip;
         int scope_id = 0;
-        scoped_ptr<Network> network;
+        std::unique_ptr<Network> network;
         switch (address->Address.lpSockaddr->sa_family) {
           case AF_INET: {
             sockaddr_in* v4_addr =
@@ -606,8 +628,8 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
             // TODO(phoglund): Need to recognize other types as well.
             adapter_type = ADAPTER_TYPE_LOOPBACK;
           }
-          scoped_ptr<Network> network(new Network(name, description, prefix,
-                                                  prefix_length, adapter_type));
+          std::unique_ptr<Network> network(new Network(
+              name, description, prefix, prefix_length, adapter_type));
           network->set_default_local_address_provider(this);
           network->set_scope_id(scope_id);
           network->AddIP(ip);
@@ -770,7 +792,7 @@ IPAddress BasicNetworkManager::QueryDefaultLocalAddress(int family) const {
   ASSERT(thread_->socketserver() != nullptr);
   ASSERT(family == AF_INET || family == AF_INET6);
 
-  scoped_ptr<AsyncSocket> socket(
+  std::unique_ptr<AsyncSocket> socket(
       thread_->socketserver()->CreateAsyncSocket(family, SOCK_DGRAM));
   if (!socket) {
     LOG_ERR(LERROR) << "Socket creation failed";
diff --git a/chromium/third_party/webrtc/base/network.h b/chromium/third_party/webrtc/base/network.h
index ee22d5e5730..a41da4a69a0 100644
--- a/chromium/third_party/webrtc/base/network.h
+++ b/chromium/third_party/webrtc/base/network.h
@@ -13,6 +13,7 @@
 
 #include <deque>
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -20,7 +21,6 @@
 #include "webrtc/base/ipaddress.h"
 #include "webrtc/base/networkmonitor.h"
 #include "webrtc/base/messagehandler.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 
 #if defined(WEBRTC_POSIX)
@@ -160,6 +160,8 @@ class NetworkManagerBase : public NetworkManager {
  private:
   friend class NetworkTest;
 
+  Network* GetNetworkFromAddress(const rtc::IPAddress& ip) const;
+
   EnumerationPermission enumeration_permission_;
 
   NetworkList networks_;
@@ -168,8 +170,8 @@ class NetworkManagerBase : public NetworkManager {
   NetworkMap networks_map_;
   bool ipv6_enabled_;
 
-  rtc::scoped_ptr<rtc::Network> ipv4_any_address_network_;
-  rtc::scoped_ptr<rtc::Network> ipv6_any_address_network_;
+  std::unique_ptr<rtc::Network> ipv4_any_address_network_;
+  std::unique_ptr<rtc::Network> ipv6_any_address_network_;
 
   IPAddress default_local_ipv4_address_;
   IPAddress default_local_ipv6_address_;
@@ -252,7 +254,7 @@ class BasicNetworkManager : public NetworkManagerBase,
   int start_count_;
   std::vector<std::string> network_ignore_list_;
   bool ignore_non_default_routes_;
-  scoped_ptr<NetworkMonitorInterface> network_monitor_;
+  std::unique_ptr<NetworkMonitorInterface> network_monitor_;
 };
 
 // Represents a Unix-type network interface, with a name and single address.
diff --git a/chromium/third_party/webrtc/base/network_unittest.cc b/chromium/third_party/webrtc/base/network_unittest.cc
index 7133d8b405d..f3193e2b0f3 100644
--- a/chromium/third_party/webrtc/base/network_unittest.cc
+++ b/chromium/third_party/webrtc/base/network_unittest.cc
@@ -12,6 +12,7 @@
 
 #include "webrtc/base/nethelpers.h"
 #include "webrtc/base/networkmonitor.h"
+#include <memory>
 #include <vector>
 #if defined(WEBRTC_POSIX)
 #include <sys/types.h>
@@ -108,7 +109,7 @@ class NetworkTest : public testing::Test, public sigslot::has_slots<>  {
                                  bool include_ignored,
                                  NetworkManager::NetworkList* networks) {
     // Use the base IfAddrsConverter for test cases.
-    rtc::scoped_ptr<IfAddrsConverter> ifaddrs_converter(new IfAddrsConverter());
+    std::unique_ptr<IfAddrsConverter> ifaddrs_converter(new IfAddrsConverter());
     network_manager.ConvertIfAddrs(interfaces, ifaddrs_converter.get(),
                                    include_ignored, networks);
   }
@@ -1006,6 +1007,35 @@ TEST_F(NetworkTest, DefaultLocalAddress) {
   EXPECT_EQ(ip, GetLoopbackIP(AF_INET));
   EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip));
   EXPECT_EQ(ip, GetLoopbackIP(AF_INET6));
+
+  // More tests on GetDefaultLocalAddress with ipv6 addresses where the set
+  // default address may be different from the best IP address of any network.
+  InterfaceAddress ip1;
+  EXPECT_TRUE(IPFromString("abcd::1234:5678:abcd:1111",
+                           IPV6_ADDRESS_FLAG_TEMPORARY, &ip1));
+  // Create a network with a prefix of ip1.
+  Network ipv6_network("test_eth0", "Test NetworkAdapter", TruncateIP(ip1, 64),
+                       64);
+  IPAddress ip2;
+  EXPECT_TRUE(IPFromString("abcd::1234:5678:abcd:2222", &ip2));
+  ipv6_network.AddIP(ip1);
+  ipv6_network.AddIP(ip2);
+  BasicNetworkManager::NetworkList list(1, new Network(ipv6_network));
+  bool changed;
+  MergeNetworkList(manager, list, &changed);
+  // If the set default address is not in any network, GetDefaultLocalAddress
+  // should return it.
+  IPAddress ip3;
+  EXPECT_TRUE(IPFromString("abcd::1234:5678:abcd:3333", &ip3));
+  manager.set_default_local_addresses(GetLoopbackIP(AF_INET), ip3);
+  EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip));
+  EXPECT_EQ(ip3, ip);
+  // If the set default address is in a network, GetDefaultLocalAddress will
+  // return the best IP in that network.
+  manager.set_default_local_addresses(GetLoopbackIP(AF_INET), ip2);
+  EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip));
+  EXPECT_EQ(static_cast<IPAddress>(ip1), ip);
+
   manager.StopUpdating();
 }
 
diff --git a/chromium/third_party/webrtc/base/networkmonitor.h b/chromium/third_party/webrtc/base/networkmonitor.h
index 35ab2b120e3..5459cd63e95 100644
--- a/chromium/third_party/webrtc/base/networkmonitor.h
+++ b/chromium/third_party/webrtc/base/networkmonitor.h
@@ -12,7 +12,6 @@
 #define WEBRTC_BASE_NETWORKMONITOR_H_
 
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
 
diff --git a/chromium/third_party/webrtc/base/networkroute.h b/chromium/third_party/webrtc/base/networkroute.h
index 95c1c088ac1..52fa6d612f4 100644
--- a/chromium/third_party/webrtc/base/networkroute.h
+++ b/chromium/third_party/webrtc/base/networkroute.h
@@ -15,7 +15,7 @@
 // the media code can rely on and the network code can implement, and both can
 // depend on that, but not depend on each other. Then, move this file to that
 // directory.
-namespace cricket {
+namespace rtc {
 
 struct NetworkRoute {
   bool connected;
@@ -47,6 +47,6 @@ struct NetworkRoute {
 
   bool operator!=(const NetworkRoute& nr) const { return !(*this == nr); }
 };
-}  // namespace cricket
+}  // namespace rtc
 
 #endif  // WEBRTC_BASE_NETWORKROUTE_H_
diff --git a/chromium/third_party/webrtc/base/nullsocketserver.cc b/chromium/third_party/webrtc/base/nullsocketserver.cc
new file mode 100644
index 00000000000..5dfd49030d9
--- /dev/null
+++ b/chromium/third_party/webrtc/base/nullsocketserver.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/nullsocketserver.h"
+
+namespace rtc {
+
+NullSocketServer::NullSocketServer() : event_(false, false) {}
+NullSocketServer::~NullSocketServer() {}
+
+bool NullSocketServer::Wait(int cms, bool process_io) {
+  event_.Wait(cms);
+  return true;
+}
+
+void NullSocketServer::WakeUp() {
+  event_.Set();
+}
+
+rtc::Socket* NullSocketServer::CreateSocket(int /* type */) {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+rtc::Socket* NullSocketServer::CreateSocket(int /* family */, int /* type */) {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+rtc::AsyncSocket* NullSocketServer::CreateAsyncSocket(int /* type */) {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+rtc::AsyncSocket* NullSocketServer::CreateAsyncSocket(int /* family */,
+                                                      int /* type */) {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/nullsocketserver.h b/chromium/third_party/webrtc/base/nullsocketserver.h
index 5378e43158c..e59f2fafe58 100644
--- a/chromium/third_party/webrtc/base/nullsocketserver.h
+++ b/chromium/third_party/webrtc/base/nullsocketserver.h
@@ -12,48 +12,25 @@
 #define WEBRTC_BASE_NULLSOCKETSERVER_H_
 
 #include "webrtc/base/event.h"
-#include "webrtc/base/physicalsocketserver.h"
+#include "webrtc/base/socketserver.h"
 
 namespace rtc {
 
-// NullSocketServer
-
-class NullSocketServer : public rtc::SocketServer {
+class NullSocketServer : public SocketServer {
  public:
-  NullSocketServer() : event_(false, false) {}
-
-  virtual bool Wait(int cms, bool process_io) {
-    event_.Wait(cms);
-    return true;
-  }
-
-  virtual void WakeUp() {
-    event_.Set();
-  }
-
-  virtual rtc::Socket* CreateSocket(int type) {
-    ASSERT(false);
-    return NULL;
-  }
-
-  virtual rtc::Socket* CreateSocket(int family, int type) {
-    ASSERT(false);
-    return NULL;
-  }
-
-  virtual rtc::AsyncSocket* CreateAsyncSocket(int type) {
-    ASSERT(false);
-    return NULL;
-  }
+  NullSocketServer();
+  ~NullSocketServer() override;
 
-  virtual rtc::AsyncSocket* CreateAsyncSocket(int family, int type) {
-    ASSERT(false);
-    return NULL;
-  }
+  bool Wait(int cms, bool process_io) override;
+  void WakeUp() override;
 
+  Socket* CreateSocket(int type) override;
+  Socket* CreateSocket(int family, int type) override;
+  AsyncSocket* CreateAsyncSocket(int type) override;
+  AsyncSocket* CreateAsyncSocket(int family, int type) override;
 
  private:
-  rtc::Event event_;
+  Event event_;
 };
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/nullsocketserver_unittest.cc b/chromium/third_party/webrtc/base/nullsocketserver_unittest.cc
index 4f22c382d80..e18afb2e943 100644
--- a/chromium/third_party/webrtc/base/nullsocketserver_unittest.cc
+++ b/chromium/third_party/webrtc/base/nullsocketserver_unittest.cc
@@ -37,7 +37,7 @@ TEST_F(NullSocketServerTest, WaitAndSet) {
 }
 
 TEST_F(NullSocketServerTest, TestWait) {
-  uint32_t start = Time();
+  int64_t start = TimeMillis();
   ss_.Wait(200, true);
   // The actual wait time is dependent on the resolution of the timer used by
   // the Event class. Allow for the event to signal ~20ms early.
diff --git a/chromium/third_party/webrtc/base/objc/OWNERS b/chromium/third_party/webrtc/base/objc/OWNERS
deleted file mode 100644
index cd06158b7fc..00000000000
--- a/chromium/third_party/webrtc/base/objc/OWNERS
+++ /dev/null
@@ -1 +0,0 @@
-tkchin@webrtc.org
diff --git a/chromium/third_party/webrtc/base/onetimeevent.h b/chromium/third_party/webrtc/base/onetimeevent.h
new file mode 100644
index 00000000000..240cf14c584
--- /dev/null
+++ b/chromium/third_party/webrtc/base/onetimeevent.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_ONETIMEEVENT_H_
+#define WEBRTC_BASE_ONETIMEEVENT_H_
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+// Provides a simple way to perform an operation (such as logging) one
+// time in a certain scope.
+// Example:
+//   OneTimeEvent firstFrame;
+//   ...
+//   if (firstFrame()) {
+//     LOG(LS_INFO) << "This is the first frame".
+//   }
+class OneTimeEvent {
+ public:
+  OneTimeEvent() {}
+  bool operator()() {
+    rtc::CritScope cs(&critsect_);
+    if (happened_) {
+      return false;
+    }
+    happened_ = true;
+    return true;
+  }
+
+ private:
+  bool happened_ = false;
+  rtc::CriticalSection critsect_;
+};
+
+// A non-thread-safe, ligher-weight version of the OneTimeEvent class.
+class ThreadUnsafeOneTimeEvent {
+ public:
+  ThreadUnsafeOneTimeEvent() {}
+  bool operator()() {
+    if (happened_) {
+      return false;
+    }
+    happened_ = true;
+    return true;
+  }
+
+ private:
+  bool happened_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_BASE_ONETIMEEVENT_H_
diff --git a/chromium/third_party/webrtc/base/onetimeevent_unittest.cc b/chromium/third_party/webrtc/base/onetimeevent_unittest.cc
new file mode 100644
index 00000000000..4ebc139b2a8
--- /dev/null
+++ b/chromium/third_party/webrtc/base/onetimeevent_unittest.cc
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/onetimeevent.h"
+
+namespace webrtc {
+
+TEST(OneTimeEventTest, ThreadSafe) {
+  OneTimeEvent ot;
+
+  // The one time event is expected to evaluate to true only the first time.
+  EXPECT_TRUE(ot());
+  EXPECT_FALSE(ot());
+  EXPECT_FALSE(ot());
+}
+
+TEST(OneTimeEventTest, ThreadUnsafe) {
+  ThreadUnsafeOneTimeEvent ot;
+
+  EXPECT_TRUE(ot());
+  EXPECT_FALSE(ot());
+  EXPECT_FALSE(ot());
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/base/opensslidentity.cc b/chromium/third_party/webrtc/base/opensslidentity.cc
index 9c2112e157c..58a0cd8adeb 100644
--- a/chromium/third_party/webrtc/base/opensslidentity.cc
+++ b/chromium/third_party/webrtc/base/opensslidentity.cc
@@ -12,6 +12,8 @@
 
 #include "webrtc/base/opensslidentity.h"
 
+#include <memory>
+
 // Must be included first before openssl headers.
 #include "webrtc/base/win32.h"  // NOLINT
 
@@ -164,6 +166,29 @@ OpenSSLKeyPair* OpenSSLKeyPair::Generate(const KeyParams& key_params) {
   return new OpenSSLKeyPair(pkey);
 }
 
+OpenSSLKeyPair* OpenSSLKeyPair::FromPrivateKeyPEMString(
+    const std::string& pem_string) {
+  BIO* bio = BIO_new_mem_buf(const_cast<char*>(pem_string.c_str()), -1);
+  if (!bio) {
+    LOG(LS_ERROR) << "Failed to create a new BIO buffer.";
+    return nullptr;
+  }
+  BIO_set_mem_eof_return(bio, 0);
+  EVP_PKEY* pkey =
+      PEM_read_bio_PrivateKey(bio, nullptr, nullptr, const_cast<char*>("\0"));
+  BIO_free(bio);  // Frees the BIO, but not the pointed-to string.
+  if (!pkey) {
+    LOG(LS_ERROR) << "Failed to create the private key from PEM string.";
+    return nullptr;
+  }
+  if (EVP_PKEY_missing_parameters(pkey) != 0) {
+    LOG(LS_ERROR) << "The resulting key pair is missing public key parameters.";
+    EVP_PKEY_free(pkey);
+    return nullptr;
+  }
+  return new OpenSSLKeyPair(pkey);
+}
+
 OpenSSLKeyPair::~OpenSSLKeyPair() {
   EVP_PKEY_free(pkey_);
 }
@@ -181,6 +206,57 @@ void OpenSSLKeyPair::AddReference() {
 #endif
 }
 
+std::string OpenSSLKeyPair::PrivateKeyToPEMString() const {
+  BIO* temp_memory_bio = BIO_new(BIO_s_mem());
+  if (!temp_memory_bio) {
+    LOG_F(LS_ERROR) << "Failed to allocate temporary memory bio";
+    RTC_NOTREACHED();
+    return "";
+  }
+  if (!PEM_write_bio_PrivateKey(
+      temp_memory_bio, pkey_, nullptr, nullptr, 0, nullptr, nullptr)) {
+    LOG_F(LS_ERROR) << "Failed to write private key";
+    BIO_free(temp_memory_bio);
+    RTC_NOTREACHED();
+    return "";
+  }
+  BIO_write(temp_memory_bio, "\0", 1);
+  char* buffer;
+  BIO_get_mem_data(temp_memory_bio, &buffer);
+  std::string priv_key_str = buffer;
+  BIO_free(temp_memory_bio);
+  return priv_key_str;
+}
+
+std::string OpenSSLKeyPair::PublicKeyToPEMString() const {
+  BIO* temp_memory_bio = BIO_new(BIO_s_mem());
+  if (!temp_memory_bio) {
+    LOG_F(LS_ERROR) << "Failed to allocate temporary memory bio";
+    RTC_NOTREACHED();
+    return "";
+  }
+  if (!PEM_write_bio_PUBKEY(temp_memory_bio, pkey_)) {
+    LOG_F(LS_ERROR) << "Failed to write public key";
+    BIO_free(temp_memory_bio);
+    RTC_NOTREACHED();
+    return "";
+  }
+  BIO_write(temp_memory_bio, "\0", 1);
+  char* buffer;
+  BIO_get_mem_data(temp_memory_bio, &buffer);
+  std::string pub_key_str = buffer;
+  BIO_free(temp_memory_bio);
+  return pub_key_str;
+}
+
+bool OpenSSLKeyPair::operator==(const OpenSSLKeyPair& other) const {
+  return EVP_PKEY_cmp(this->pkey_, other.pkey_) == 1;
+}
+
+bool OpenSSLKeyPair::operator!=(const OpenSSLKeyPair& other) const {
+  return !(*this == other);
+}
+
 #if !defined(NDEBUG)
 // Print a certificate to the log, for debugging.
 static void PrintCert(X509* x509) {
@@ -280,7 +356,7 @@ bool OpenSSLCertificate::GetSignatureDigestAlgorithm(
   return true;
 }
 
-rtc::scoped_ptr<SSLCertChain> OpenSSLCertificate::GetChain() const {
+std::unique_ptr<SSLCertChain> OpenSSLCertificate::GetChain() const {
   // Chains are not yet supported when using OpenSSL.
   // OpenSSLStreamAdapter::SSLVerifyCallback currently requires the remote
   // certificate to be self-signed.
@@ -368,6 +444,14 @@ void OpenSSLCertificate::AddReference() const {
 #endif
 }
 
+bool OpenSSLCertificate::operator==(const OpenSSLCertificate& other) const {
+  return X509_cmp(this->x509_, other.x509_) == 0;
+}
+
+bool OpenSSLCertificate::operator!=(const OpenSSLCertificate& other) const {
+  return !(*this == other);
+}
+
 // Documented in sslidentity.h.
 int64_t OpenSSLCertificate::CertificateExpirationTime() const {
   ASN1_TIME* expire_time = X509_get_notAfter(x509_);
@@ -430,29 +514,21 @@ OpenSSLIdentity* OpenSSLIdentity::GenerateForTest(
 SSLIdentity* OpenSSLIdentity::FromPEMStrings(
     const std::string& private_key,
     const std::string& certificate) {
-  scoped_ptr<OpenSSLCertificate> cert(
+  std::unique_ptr<OpenSSLCertificate> cert(
       OpenSSLCertificate::FromPEMString(certificate));
   if (!cert) {
     LOG(LS_ERROR) << "Failed to create OpenSSLCertificate from PEM string.";
-    return NULL;
-  }
-
-  BIO* bio = BIO_new_mem_buf(const_cast<char*>(private_key.c_str()), -1);
-  if (!bio) {
-    LOG(LS_ERROR) << "Failed to create a new BIO buffer.";
-    return NULL;
+    return nullptr;
   }
-  BIO_set_mem_eof_return(bio, 0);
-  EVP_PKEY* pkey =
-      PEM_read_bio_PrivateKey(bio, NULL, NULL, const_cast<char*>("\0"));
-  BIO_free(bio);  // Frees the BIO, but not the pointed-to string.
 
-  if (!pkey) {
-    LOG(LS_ERROR) << "Failed to create the private key from PEM string.";
-    return NULL;
+  OpenSSLKeyPair* key_pair =
+      OpenSSLKeyPair::FromPrivateKeyPEMString(private_key);
+  if (!key_pair) {
+    LOG(LS_ERROR) << "Failed to create key pair from PEM string.";
+    return nullptr;
   }
 
-  return new OpenSSLIdentity(new OpenSSLKeyPair(pkey),
+  return new OpenSSLIdentity(key_pair,
                              cert.release());
 }
 
@@ -475,6 +551,23 @@ bool OpenSSLIdentity::ConfigureIdentity(SSL_CTX* ctx) {
   return true;
 }
 
+std::string OpenSSLIdentity::PrivateKeyToPEMString() const {
+  return key_pair_->PrivateKeyToPEMString();
+}
+
+std::string OpenSSLIdentity::PublicKeyToPEMString() const {
+  return key_pair_->PublicKeyToPEMString();
+}
+
+bool OpenSSLIdentity::operator==(const OpenSSLIdentity& other) const {
+  return *this->key_pair_ == *other.key_pair_ &&
+         *this->certificate_ == *other.certificate_;
+}
+
+bool OpenSSLIdentity::operator!=(const OpenSSLIdentity& other) const {
+  return !(*this == other);
+}
+
 }  // namespace rtc
 
 #endif  // HAVE_OPENSSL_SSL_H
diff --git a/chromium/third_party/webrtc/base/opensslidentity.h b/chromium/third_party/webrtc/base/opensslidentity.h
index df495087e35..316572c48bb 100644
--- a/chromium/third_party/webrtc/base/opensslidentity.h
+++ b/chromium/third_party/webrtc/base/opensslidentity.h
@@ -14,10 +14,11 @@
 #include <openssl/evp.h>
 #include <openssl/x509.h>
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/common.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sslidentity.h"
 
 typedef struct ssl_ctx_st SSL_CTX;
@@ -33,12 +34,20 @@ class OpenSSLKeyPair {
   }
 
   static OpenSSLKeyPair* Generate(const KeyParams& key_params);
+  // Constructs a key pair from the private key PEM string. This must not result
+  // in missing public key parameters. Returns null on error.
+  static OpenSSLKeyPair* FromPrivateKeyPEMString(
+      const std::string& pem_string);
 
   virtual ~OpenSSLKeyPair();
 
   virtual OpenSSLKeyPair* GetReference();
 
   EVP_PKEY* pkey() const { return pkey_; }
+  std::string PrivateKeyToPEMString() const;
+  std::string PublicKeyToPEMString() const;
+  bool operator==(const OpenSSLKeyPair& other) const;
+  bool operator!=(const OpenSSLKeyPair& other) const;
 
  private:
   void AddReference();
@@ -68,8 +77,9 @@ class OpenSSLCertificate : public SSLCertificate {
   X509* x509() const { return x509_; }
 
   std::string ToPEMString() const override;
-
   void ToDER(Buffer* der_buffer) const override;
+  bool operator==(const OpenSSLCertificate& other) const;
+  bool operator!=(const OpenSSLCertificate& other) const;
 
   // Compute the digest of the certificate given algorithm
   bool ComputeDigest(const std::string& algorithm,
@@ -85,7 +95,7 @@ class OpenSSLCertificate : public SSLCertificate {
                             size_t* length);
 
   bool GetSignatureDigestAlgorithm(std::string* algorithm) const override;
-  rtc::scoped_ptr<SSLCertChain> GetChain() const override;
+  std::unique_ptr<SSLCertChain> GetChain() const override;
 
   int64_t CertificateExpirationTime() const override;
 
@@ -115,13 +125,18 @@ class OpenSSLIdentity : public SSLIdentity {
   // Configure an SSL context object to use our key and certificate.
   bool ConfigureIdentity(SSL_CTX* ctx);
 
+  std::string PrivateKeyToPEMString() const override;
+  std::string PublicKeyToPEMString() const override;
+  bool operator==(const OpenSSLIdentity& other) const;
+  bool operator!=(const OpenSSLIdentity& other) const;
+
  private:
   OpenSSLIdentity(OpenSSLKeyPair* key_pair, OpenSSLCertificate* certificate);
 
   static OpenSSLIdentity* GenerateInternal(const SSLIdentityParams& params);
 
-  scoped_ptr<OpenSSLKeyPair> key_pair_;
-  scoped_ptr<OpenSSLCertificate> certificate_;
+  std::unique_ptr<OpenSSLKeyPair> key_pair_;
+  std::unique_ptr<OpenSSLCertificate> certificate_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLIdentity);
 };
diff --git a/chromium/third_party/webrtc/base/opensslstreamadapter.cc b/chromium/third_party/webrtc/base/opensslstreamadapter.cc
index 16dd9803b40..abdf5e4834c 100644
--- a/chromium/third_party/webrtc/base/opensslstreamadapter.cc
+++ b/chromium/third_party/webrtc/base/opensslstreamadapter.cc
@@ -18,7 +18,11 @@
 #include <openssl/rand.h>
 #include <openssl/tls1.h>
 #include <openssl/x509v3.h>
+#ifndef OPENSSL_IS_BORINGSSL
+#include <openssl/dtls1.h>
+#endif
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/common.h"
@@ -290,9 +294,9 @@ void OpenSSLStreamAdapter::SetServerRole(SSLRole role) {
   role_ = role;
 }
 
-rtc::scoped_ptr<SSLCertificate> OpenSSLStreamAdapter::GetPeerCertificate()
+std::unique_ptr<SSLCertificate> OpenSSLStreamAdapter::GetPeerCertificate()
     const {
-  return peer_certificate_ ? rtc::scoped_ptr<SSLCertificate>(
+  return peer_certificate_ ? std::unique_ptr<SSLCertificate>(
                                  peer_certificate_->GetReference())
                            : nullptr;
 }
@@ -1138,7 +1142,9 @@ static const cipher_list OK_RSA_ciphers[] = {
 #ifdef TLS1_CK_ECDHE_RSA_WITH_AES_256_GCM_SHA256
   CDEF(ECDHE_RSA_WITH_AES_256_GCM_SHA256),
 #endif
+#ifdef TLS1_CK_ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256
   CDEF(ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256),
+#endif
 };
 
 static const cipher_list OK_ECDSA_ciphers[] = {
@@ -1148,7 +1154,9 @@ static const cipher_list OK_ECDSA_ciphers[] = {
 #ifdef TLS1_CK_ECDHE_ECDSA_WITH_AES_256_GCM_SHA256
   CDEF(ECDHE_ECDSA_WITH_AES_256_GCM_SHA256),
 #endif
+#ifdef TLS1_CK_ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256
   CDEF(ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256),
+#endif
 };
 #undef CDEF
 
diff --git a/chromium/third_party/webrtc/base/opensslstreamadapter.h b/chromium/third_party/webrtc/base/opensslstreamadapter.h
index 463c8f2280b..1e90bacc0e4 100644
--- a/chromium/third_party/webrtc/base/opensslstreamadapter.h
+++ b/chromium/third_party/webrtc/base/opensslstreamadapter.h
@@ -12,6 +12,7 @@
 #define WEBRTC_BASE_OPENSSLSTREAMADAPTER_H__
 
 #include <string>
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/buffer.h"
@@ -69,7 +70,7 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
                                 const unsigned char* digest_val,
                                 size_t digest_len) override;
 
-  rtc::scoped_ptr<SSLCertificate> GetPeerCertificate() const override;
+  std::unique_ptr<SSLCertificate> GetPeerCertificate() const override;
 
   int StartSSLWithServer(const char* server_name) override;
   int StartSSLWithPeer() override;
@@ -184,13 +185,13 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
   SSL_CTX* ssl_ctx_;
 
   // Our key and certificate, mostly useful in peer-to-peer mode.
-  scoped_ptr<OpenSSLIdentity> identity_;
+  std::unique_ptr<OpenSSLIdentity> identity_;
   // in traditional mode, the server name that the server's certificate
   // must specify. Empty in peer-to-peer mode.
   std::string ssl_server_name_;
   // The certificate that the peer must present or did present. Initially
   // null in traditional mode, until the connection is established.
-  scoped_ptr<OpenSSLCertificate> peer_certificate_;
+  std::unique_ptr<OpenSSLCertificate> peer_certificate_;
   // In peer-to-peer mode, the digest of the certificate that
   // the peer must present.
   Buffer peer_certificate_digest_value_;
diff --git a/chromium/third_party/webrtc/base/optional.h b/chromium/third_party/webrtc/base/optional.h
index b8071e63587..25cfbfe4175 100644
--- a/chromium/third_party/webrtc/base/optional.h
+++ b/chromium/third_party/webrtc/base/optional.h
@@ -12,17 +12,14 @@
 #define WEBRTC_BASE_OPTIONAL_H_
 
 #include <algorithm>
+#include <memory>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 
 namespace rtc {
 
-// Simple std::experimental::optional-wannabe. It either contains a T or not.
-// In order to keep the implementation simple and portable, this implementation
-// actually contains a (default-constructed) T even when it supposedly doesn't
-// contain a value; use e.g. rtc::scoped_ptr<T> instead if that's too
-// expensive.
+// Simple std::optional-wannabe. It either contains a T or not.
 //
 // A moved-from Optional<T> may only be destroyed, and assigned to if T allows
 // being assigned to after having been moved from. Specifically, you may not
@@ -65,28 +62,90 @@ class Optional final {
   Optional() : has_value_(false) {}
 
   // Construct an Optional that contains a value.
-  explicit Optional(const T& val) : value_(val), has_value_(true) {}
-  explicit Optional(T&& val) : value_(std::move(val)), has_value_(true) {}
-
-  // Copy and move constructors.
-  // TODO(kwiberg): =default the move constructor when MSVC supports it.
-  Optional(const Optional&) = default;
-  Optional(Optional&& m)
-      : value_(std::move(m.value_)), has_value_(m.has_value_) {}
-
-  // Assignment.
-  // TODO(kwiberg): =default the move assignment op when MSVC supports it.
-  Optional& operator=(const Optional&) = default;
+  explicit Optional(const T& value) : has_value_(true) {
+    new (&value_) T(value);
+  }
+  explicit Optional(T&& value) : has_value_(true) {
+    new (&value_) T(std::move(value));
+  }
+
+  // Copy constructor: copies the value from m if it has one.
+  Optional(const Optional& m) : has_value_(m.has_value_) {
+    if (has_value_)
+      new (&value_) T(m.value_);
+  }
+
+  // Move constructor: if m has a value, moves the value from m, leaving m
+  // still in a state where it has a value, but a moved-from one (the
+  // properties of which depends on T; the only general guarantee is that we
+  // can destroy m).
+  Optional(Optional&& m) : has_value_(m.has_value_) {
+    if (has_value_)
+      new (&value_) T(std::move(m.value_));
+  }
+
+  ~Optional() {
+    if (has_value_)
+      value_.~T();
+  }
+
+  // Copy assignment. Uses T's copy assignment if both sides have a value, T's
+  // copy constructor if only the right-hand side has a value.
+  Optional& operator=(const Optional& m) {
+    if (m.has_value_) {
+      if (has_value_) {
+        value_ = m.value_;  // T's copy assignment.
+      } else {
+        new (&value_) T(m.value_);  // T's copy constructor.
+        has_value_ = true;
+      }
+    } else if (has_value_) {
+      value_.~T();
+      has_value_ = false;
+    }
+    return *this;
+  }
+
+  // Move assignment. Uses T's move assignment if both sides have a value, T's
+  // move constructor if only the right-hand side has a value. The state of m
+  // after it's been moved from is as for the move constructor.
   Optional& operator=(Optional&& m) {
-    value_ = std::move(m.value_);
-    has_value_ = m.has_value_;
+    if (m.has_value_) {
+      if (has_value_) {
+        value_ = std::move(m.value_);  // T's move assignment.
+      } else {
+        new (&value_) T(std::move(m.value_));  // T's move constructor.
+        has_value_ = true;
+      }
+    } else if (has_value_) {
+      value_.~T();
+      has_value_ = false;
+    }
     return *this;
   }
 
+  // Swap the values if both m1 and m2 have values; move the value if only one
+  // of them has one.
   friend void swap(Optional& m1, Optional& m2) {
-    using std::swap;
-    swap(m1.value_, m2.value_);
-    swap(m1.has_value_, m2.has_value_);
+    if (m1.has_value_) {
+      if (m2.has_value_) {
+        // Both have values: swap.
+        using std::swap;
+        swap(m1.value_, m2.value_);
+      } else {
+        // Only m1 has a value: move it to m2.
+        new (&m2.value_) T(std::move(m1.value_));
+        m1.value_.~T();  // Destroy the moved-from value.
+        m1.has_value_ = false;
+        m2.has_value_ = true;
+      }
+    } else if (m2.has_value_) {
+      // Only m2 has a value: move it to m1.
+      new (&m1.value_) T(std::move(m2.value_));
+      m2.value_.~T();  // Destroy the moved-from value.
+      m1.has_value_ = true;
+      m2.has_value_ = false;
+    }
   }
 
   // Conversion to bool to test if we have a value.
@@ -128,10 +187,15 @@ class Optional final {
   }
 
  private:
-  // Invariant: Unless *this has been moved from, value_ is default-initialized
-  // (or copied or moved from a default-initialized T) if !has_value_.
-  T value_;
-  bool has_value_;
+  bool has_value_;  // True iff value_ contains a live value.
+  union {
+    // By placing value_ in a union, we get to manage its construction and
+    // destruction manually: the Optional constructors won't automatically
+    // construct it, and the Optional destructor won't automatically destroy
+    // it. Basically, this just allocates a properly sized and aligned block of
+    // memory in which we can manually put a T with placement new.
+    T value_;
+  };
 };
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/optional_unittest.cc b/chromium/third_party/webrtc/base/optional_unittest.cc
index 8ddbebadb9a..b51701f6b26 100644
--- a/chromium/third_party/webrtc/base/optional_unittest.cc
+++ b/chromium/third_party/webrtc/base/optional_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <sstream>
 #include <string>
 #include <utility>
@@ -64,8 +65,8 @@ class Logger {
   }
   void Foo() { Log("Foo()"); }
   void Foo() const { Log("Foo() const"); }
-  static rtc::scoped_ptr<std::vector<std::string>> Setup() {
-    rtc::scoped_ptr<std::vector<std::string>> s(new std::vector<std::string>);
+  static std::unique_ptr<std::vector<std::string>> Setup() {
+    std::unique_ptr<std::vector<std::string>> s(new std::vector<std::string>);
     g_log = s.get();
     g_next_id = 0;
     return s;
@@ -124,7 +125,7 @@ TEST(OptionalTest, TestConstructDefault) {
     Optional<Logger> x;
     EXPECT_FALSE(x);
   }
-  EXPECT_EQ(V("0:0. default constructor", "0:0. destructor"), *log);
+  EXPECT_EQ(V(), *log);
 }
 
 TEST(OptionalTest, TestConstructCopyEmpty) {
@@ -135,9 +136,7 @@ TEST(OptionalTest, TestConstructCopyEmpty) {
     auto y = x;
     EXPECT_FALSE(y);
   }
-  EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
-              "1:0. destructor", "0:0. destructor"),
-            *log);
+  EXPECT_EQ(V(), *log);
 }
 
 TEST(OptionalTest, TestConstructCopyFull) {
@@ -165,9 +164,7 @@ TEST(OptionalTest, TestConstructMoveEmpty) {
     auto y = std::move(x);
     EXPECT_FALSE(y);
   }
-  EXPECT_EQ(V("0:0. default constructor", "1:0. move constructor (from 0:0)",
-              "1:0. destructor", "0:0. destructor"),
-            *log);
+  EXPECT_EQ(V(), *log);
 }
 
 TEST(OptionalTest, TestConstructMoveFull) {
@@ -194,10 +191,7 @@ TEST(OptionalTest, TestCopyAssignToEmptyFromEmpty) {
     Optional<Logger> x, y;
     x = y;
   }
-  EXPECT_EQ(
-      V("0:0. default constructor", "1:1. default constructor",
-        "0:1. operator= copy (from 1:1)", "1:1. destructor", "0:1. destructor"),
-      *log);
+  EXPECT_EQ(V(), *log);
 }
 
 TEST(OptionalTest, TestCopyAssignToFullFromEmpty) {
@@ -211,9 +205,7 @@ TEST(OptionalTest, TestCopyAssignToFullFromEmpty) {
   }
   EXPECT_EQ(
       V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
-        "0:17. destructor", "2:2. default constructor", "---",
-        "1:2. operator= copy (from 2:2)", "---", "2:2. destructor",
-        "1:2. destructor"),
+        "0:17. destructor", "---", "1:17. destructor", "---"),
       *log);
 }
 
@@ -226,11 +218,11 @@ TEST(OptionalTest, TestCopyAssignToEmptyFromFull) {
     x = y;
     log->push_back("---");
   }
-  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
-              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
-              "0:17. operator= copy (from 2:17)", "---", "2:17. destructor",
-              "0:17. destructor"),
-            *log);
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "---", "2:17. copy constructor (from 1:17)", "---",
+        "1:17. destructor", "2:17. destructor"),
+      *log);
 }
 
 TEST(OptionalTest, TestCopyAssignToFullFromFull) {
@@ -260,10 +252,10 @@ TEST(OptionalTest, TestCopyAssignToEmptyFromT) {
     x = Optional<Logger>(y);
     log->push_back("---");
   }
-  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
-              "2:17. copy constructor (from 1:17)",
-              "0:17. operator= move (from 2:17)", "2:17. destructor", "---",
-              "1:17. destructor", "0:17. destructor"),
+  EXPECT_EQ(V("0:17. explicit constructor", "---",
+              "1:17. copy constructor (from 0:17)",
+              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
+              "0:17. destructor", "2:17. destructor"),
             *log);
 }
 
@@ -291,10 +283,7 @@ TEST(OptionalTest, TestMoveAssignToEmptyFromEmpty) {
     Optional<Logger> x, y;
     x = std::move(y);
   }
-  EXPECT_EQ(
-      V("0:0. default constructor", "1:1. default constructor",
-        "0:1. operator= move (from 1:1)", "1:1. destructor", "0:1. destructor"),
-      *log);
+  EXPECT_EQ(V(), *log);
 }
 
 TEST(OptionalTest, TestMoveAssignToFullFromEmpty) {
@@ -308,9 +297,7 @@ TEST(OptionalTest, TestMoveAssignToFullFromEmpty) {
   }
   EXPECT_EQ(
       V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
-        "0:17. destructor", "2:2. default constructor", "---",
-        "1:2. operator= move (from 2:2)", "---", "2:2. destructor",
-        "1:2. destructor"),
+        "0:17. destructor", "---", "1:17. destructor", "---"),
       *log);
 }
 
@@ -323,11 +310,11 @@ TEST(OptionalTest, TestMoveAssignToEmptyFromFull) {
     x = std::move(y);
     log->push_back("---");
   }
-  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
-              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
-              "0:17. operator= move (from 2:17)", "---", "2:17. destructor",
-              "0:17. destructor"),
-            *log);
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+        "0:17. destructor", "---", "2:17. move constructor (from 1:17)", "---",
+        "1:17. destructor", "2:17. destructor"),
+      *log);
 }
 
 TEST(OptionalTest, TestMoveAssignToFullFromFull) {
@@ -357,10 +344,10 @@ TEST(OptionalTest, TestMoveAssignToEmptyFromT) {
     x = Optional<Logger>(std::move(y));
     log->push_back("---");
   }
-  EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
-              "2:17. move constructor (from 1:17)",
-              "0:17. operator= move (from 2:17)", "2:17. destructor", "---",
-              "1:17. destructor", "0:17. destructor"),
+  EXPECT_EQ(V("0:17. explicit constructor", "---",
+              "1:17. move constructor (from 0:17)",
+              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
+              "0:17. destructor", "2:17. destructor"),
             *log);
 }
 
@@ -425,14 +412,13 @@ TEST(OptionalTest, TestDereferenceWithDefault) {
   }
   EXPECT_EQ(
       V("0:17. explicit constructor", "1:42. explicit constructor",
-        "2:17. copy constructor (from 0:17)", "3:3. default constructor", "-1-",
-        "4:42. explicit constructor", "operator== 0:17, 2:17",
-        "4:42. destructor", "-2-", "5:42. explicit constructor",
-        "operator== 1:42, 5:42", "5:42. destructor", "-3-",
-        "6:17. explicit constructor", "7:17. move constructor (from 6:17)",
-        "operator== 0:17, 7:17", "7:17. destructor", "6:17. destructor", "-4-",
-        "8:8. default constructor", "operator== 1:42, 1:42", "8:8. destructor",
-        "-5-", "3:3. destructor", "2:17. destructor", "1:42. destructor",
+        "2:17. copy constructor (from 0:17)", "-1-",
+        "3:42. explicit constructor", "operator== 0:17, 2:17",
+        "3:42. destructor", "-2-", "4:42. explicit constructor",
+        "operator== 1:42, 4:42", "4:42. destructor", "-3-",
+        "5:17. explicit constructor", "6:17. move constructor (from 5:17)",
+        "operator== 0:17, 6:17", "6:17. destructor", "5:17. destructor", "-4-",
+        "operator== 1:42, 1:42", "-5-", "2:17. destructor", "1:42. destructor",
         "0:17. destructor"),
       *log);
 }
@@ -451,16 +437,15 @@ TEST(OptionalTest, TestEquality) {
     EXPECT_EQ(me1, me2);
     log->push_back("---");
   }
-  EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
-              "2:17. copy constructor (from 0:17)",
-              "3:17. copy constructor (from 0:17)",
-              "4:42. copy constructor (from 1:42)", "5:5. default constructor",
-              "6:6. default constructor", "---", "operator== 2:17, 2:17",
-              "operator== 2:17, 3:17", "operator!= 2:17, 4:42", "---",
-              "6:6. destructor", "5:5. destructor", "4:42. destructor",
-              "3:17. destructor", "2:17. destructor", "1:42. destructor",
-              "0:17. destructor"),
-            *log);
+  EXPECT_EQ(
+      V("0:17. explicit constructor", "1:42. explicit constructor",
+        "2:17. copy constructor (from 0:17)",
+        "3:17. copy constructor (from 0:17)",
+        "4:42. copy constructor (from 1:42)", "---", "operator== 2:17, 2:17",
+        "operator== 2:17, 3:17", "operator!= 2:17, 4:42", "---",
+        "4:42. destructor", "3:17. destructor", "2:17. destructor",
+        "1:42. destructor", "0:17. destructor"),
+      *log);
 }
 
 TEST(OptionalTest, TestSwap) {
@@ -477,11 +462,9 @@ TEST(OptionalTest, TestSwap) {
   EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
               "2:17. copy constructor (from 0:17)",
               "3:42. copy constructor (from 1:42)",
-              "4:17. copy constructor (from 0:17)", "5:5. default constructor",
-              "6:6. default constructor", "7:7. default constructor", "---",
-              "swap 2:42, 3:17", "swap 4:5, 5:17", "swap 6:7, 7:6", "---",
-              "7:6. destructor", "6:7. destructor", "5:17. destructor",
-              "4:5. destructor", "3:17. destructor", "2:42. destructor",
+              "4:17. copy constructor (from 0:17)", "---", "swap 2:42, 3:17",
+              "5:17. move constructor (from 4:17)", "4:17. destructor", "---",
+              "5:17. destructor", "3:17. destructor", "2:42. destructor",
               "1:42. destructor", "0:17. destructor"),
             *log);
 }
diff --git a/chromium/third_party/webrtc/base/optionsfile_unittest.cc b/chromium/third_party/webrtc/base/optionsfile_unittest.cc
index bcfb3efcf6f..f22a2f1f07e 100644
--- a/chromium/third_party/webrtc/base/optionsfile_unittest.cc
+++ b/chromium/third_party/webrtc/base/optionsfile_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/optionsfile.h"
@@ -51,7 +53,7 @@ class OptionsFileTest : public testing::Test {
     store_.reset(new OptionsFile(test_file_));
   }
 
-  rtc::scoped_ptr<OptionsFile> store_;
+  std::unique_ptr<OptionsFile> store_;
 
  private:
   std::string test_file_;
diff --git a/chromium/third_party/webrtc/base/pathutils.cc b/chromium/third_party/webrtc/base/pathutils.cc
index b5227ecb100..a420eb5136a 100644
--- a/chromium/third_party/webrtc/base/pathutils.cc
+++ b/chromium/third_party/webrtc/base/pathutils.cc
@@ -13,7 +13,7 @@
 #include <shellapi.h>
 #include <shlobj.h>
 #include <tchar.h>
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/fileutils.h"
@@ -35,9 +35,9 @@ const char* const FOLDER_DELIMS = "/\\";
 // DEFAULT_FOLDER_DELIM is the preferred delimiter for this platform
 #if WEBRTC_WIN
 const char DEFAULT_FOLDER_DELIM = '\\';
-#else  // !WEBRTC_WIN 
+#else  // !WEBRTC_WIN
 const char DEFAULT_FOLDER_DELIM = '/';
-#endif  // !WEBRTC_WIN 
+#endif  // !WEBRTC_WIN
 
 ///////////////////////////////////////////////////////////////////////////////
 // Pathname - parsing of pathnames into components, and vice versa
@@ -55,6 +55,9 @@ Pathname::Pathname()
     : folder_delimiter_(DEFAULT_FOLDER_DELIM) {
 }
 
+Pathname::Pathname(const Pathname&) = default;
+Pathname::Pathname(Pathname&&) = default;
+
 Pathname::Pathname(const std::string& pathname)
     : folder_delimiter_(DEFAULT_FOLDER_DELIM) {
   SetPathname(pathname);
@@ -65,6 +68,9 @@ Pathname::Pathname(const std::string& folder, const std::string& filename)
   SetPathname(folder, filename);
 }
 
+Pathname& Pathname::operator=(const Pathname&) = default;
+Pathname& Pathname::operator=(Pathname&&) = default;
+
 void Pathname::SetFolderDelimiter(char delimiter) {
   ASSERT(IsFolderDelimiter(delimiter));
   folder_delimiter_ = delimiter;
diff --git a/chromium/third_party/webrtc/base/pathutils.h b/chromium/third_party/webrtc/base/pathutils.h
index 2d5819f1b61..2a0efa9763f 100644
--- a/chromium/third_party/webrtc/base/pathutils.h
+++ b/chromium/third_party/webrtc/base/pathutils.h
@@ -44,9 +44,14 @@ public:
   static char DefaultFolderDelimiter();
 
   Pathname();
+  Pathname(const Pathname&);
+  Pathname(Pathname&&);
   Pathname(const std::string& pathname);
   Pathname(const std::string& folder, const std::string& filename);
 
+  Pathname& operator=(const Pathname&);
+  Pathname& operator=(Pathname&&);
+
   // Set's the default folder delimiter for this Pathname
   char folder_delimiter() const { return folder_delimiter_; }
   void SetFolderDelimiter(char delimiter);
diff --git a/chromium/third_party/webrtc/base/physicalsocketserver.cc b/chromium/third_party/webrtc/base/physicalsocketserver.cc
index 708499df531..0230077a52c 100644
--- a/chromium/third_party/webrtc/base/physicalsocketserver.cc
+++ b/chromium/third_party/webrtc/base/physicalsocketserver.cc
@@ -46,6 +46,7 @@
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/networkmonitor.h"
+#include "webrtc/base/nullsocketserver.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/winping.h"
 #include "webrtc/base/win32socketinit.h"
@@ -62,6 +63,14 @@ typedef char* SockOptArg;
 
 namespace rtc {
 
+std::unique_ptr<SocketServer> SocketServer::CreateDefault() {
+#if defined(__native_client__)
+  return std::unique_ptr<SocketServer>(new rtc::NullSocketServer);
+#else
+  return std::unique_ptr<SocketServer>(new rtc::PhysicalSocketServer);
+#endif
+}
+
 #if defined(WEBRTC_WIN)
 // Standard MTUs, from RFC 1191
 const uint16_t PACKET_MAXIMUMS[] = {
@@ -1461,9 +1470,9 @@ bool PhysicalSocketServer::InstallSignal(int signum, void (*handler)(int)) {
 
 #if defined(WEBRTC_WIN)
 bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
-  int cmsTotal = cmsWait;
-  int cmsElapsed = 0;
-  uint32_t msStart = Time();
+  int64_t cmsTotal = cmsWait;
+  int64_t cmsElapsed = 0;
+  int64_t msStart = Time();
 
   fWait_ = true;
   while (fWait_) {
@@ -1500,18 +1509,18 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
 
     // Which is shorter, the delay wait or the asked wait?
 
-    int cmsNext;
+    int64_t cmsNext;
     if (cmsWait == kForever) {
       cmsNext = cmsWait;
     } else {
-      cmsNext = std::max(0, cmsTotal - cmsElapsed);
+      cmsNext = std::max<int64_t>(0, cmsTotal - cmsElapsed);
     }
 
     // Wait for one of the events to signal
     DWORD dw = WSAWaitForMultipleEvents(static_cast<DWORD>(events.size()),
                                         &events[0],
                                         false,
-                                        cmsNext,
+                                        static_cast<DWORD>(cmsNext),
                                         false);
 
     if (dw == WSA_WAIT_FAILED) {
diff --git a/chromium/third_party/webrtc/base/physicalsocketserver.h b/chromium/third_party/webrtc/base/physicalsocketserver.h
index 583306c31d7..f5867d25f92 100644
--- a/chromium/third_party/webrtc/base/physicalsocketserver.h
+++ b/chromium/third_party/webrtc/base/physicalsocketserver.h
@@ -11,11 +11,11 @@
 #ifndef WEBRTC_BASE_PHYSICALSOCKETSERVER_H__
 #define WEBRTC_BASE_PHYSICALSOCKETSERVER_H__
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/asyncfile.h"
 #include "webrtc/base/nethelpers.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketserver.h"
 #include "webrtc/base/criticalsection.h"
 
@@ -104,7 +104,7 @@ class PhysicalSocketServer : public SocketServer {
 #if defined(WEBRTC_POSIX)
   static bool InstallSignal(int signum, void (*handler)(int));
 
-  scoped_ptr<PosixSignalDispatcher> signal_dispatcher_;
+  std::unique_ptr<PosixSignalDispatcher> signal_dispatcher_;
 #endif
   DispatcherList dispatchers_;
   IteratorList iterators_;
diff --git a/chromium/third_party/webrtc/base/physicalsocketserver_unittest.cc b/chromium/third_party/webrtc/base/physicalsocketserver_unittest.cc
index c53441d1a05..a04362d919d 100644
--- a/chromium/third_party/webrtc/base/physicalsocketserver_unittest.cc
+++ b/chromium/third_party/webrtc/base/physicalsocketserver_unittest.cc
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <signal.h>
 #include <stdarg.h>
 
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socket_unittest.h"
 #include "webrtc/base/testutils.h"
 #include "webrtc/base/thread.h"
@@ -100,7 +100,7 @@ class PhysicalSocketTest : public SocketTest {
   void ConnectInternalAcceptError(const IPAddress& loopback);
   void WritableAfterPartialWrite(const IPAddress& loopback);
 
-  rtc::scoped_ptr<FakePhysicalSocketServer> server_;
+  std::unique_ptr<FakePhysicalSocketServer> server_;
   SocketServerScope scope_;
   bool fail_accept_;
   int max_send_size_;
@@ -172,20 +172,20 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create two clients.
-  scoped_ptr<AsyncSocket> client1(server_->CreateAsyncSocket(loopback.family(),
-                                                             SOCK_STREAM));
+  std::unique_ptr<AsyncSocket> client1(
+      server_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client1.get());
   EXPECT_EQ(AsyncSocket::CS_CLOSED, client1->GetState());
   EXPECT_PRED1(IsUnspecOrEmptyIP, client1->GetLocalAddress().ipaddr());
 
-  scoped_ptr<AsyncSocket> client2(server_->CreateAsyncSocket(loopback.family(),
-                                                             SOCK_STREAM));
+  std::unique_ptr<AsyncSocket> client2(
+      server_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client2.get());
   EXPECT_EQ(AsyncSocket::CS_CLOSED, client2->GetState());
   EXPECT_PRED1(IsUnspecOrEmptyIP, client2->GetLocalAddress().ipaddr());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       server_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -211,7 +211,7 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) {
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
   // Simulate "::accept" returning an error.
   SetFailAccept(true);
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   EXPECT_FALSE(accepted);
   ASSERT_TRUE(accept_addr.IsNil());
 
@@ -233,7 +233,7 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) {
   // Server has pending connection, try to accept it (will succeed).
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
   SetFailAccept(false);
-  scoped_ptr<AsyncSocket> accepted2(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted2(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted2);
   EXPECT_FALSE(accept_addr.IsNil());
   EXPECT_EQ(accepted2->GetRemoteAddress(), accept_addr);
@@ -515,7 +515,7 @@ class PosixSignalDeliveryTest : public testing::Test {
   static std::vector<int> signals_received_;
   static Thread *signaled_thread_;
 
-  scoped_ptr<PhysicalSocketServer> ss_;
+  std::unique_ptr<PhysicalSocketServer> ss_;
 };
 
 std::vector<int> PosixSignalDeliveryTest::signals_received_;
@@ -583,8 +583,8 @@ TEST_F(PosixSignalDeliveryTest, SignalOnDifferentThread) {
   // Start a new thread that raises it. It will have to be delivered to that
   // thread. Our implementation should safely handle it and dispatch
   // RecordSignal() on this thread.
-  scoped_ptr<Thread> thread(new Thread());
-  scoped_ptr<RaiseSigTermRunnable> runnable(new RaiseSigTermRunnable());
+  std::unique_ptr<Thread> thread(new Thread());
+  std::unique_ptr<RaiseSigTermRunnable> runnable(new RaiseSigTermRunnable());
   thread->Start(runnable.get());
   EXPECT_TRUE(ss_->Wait(1500, true));
   EXPECT_TRUE(ExpectSignal(SIGTERM));
diff --git a/chromium/third_party/webrtc/base/platform_thread.cc b/chromium/third_party/webrtc/base/platform_thread.cc
index b6fd8732aaa..286bee95f12 100644
--- a/chromium/third_party/webrtc/base/platform_thread.cc
+++ b/chromium/third_party/webrtc/base/platform_thread.cc
@@ -99,7 +99,8 @@ PlatformThread::PlatformThread(ThreadRunFunction func,
       name_(thread_name ? thread_name : "webrtc"),
 #if defined(WEBRTC_WIN)
       stop_(false),
-      thread_(NULL) {
+      thread_(NULL),
+      thread_id_(0) {
 #else
       stop_event_(false, false),
       thread_(0) {
@@ -112,6 +113,7 @@ PlatformThread::~PlatformThread() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 #if defined(WEBRTC_WIN)
   RTC_DCHECK(!thread_);
+  RTC_DCHECK(!thread_id_);
 #endif  // defined(WEBRTC_WIN)
 }
 
@@ -136,10 +138,10 @@ void PlatformThread::Start() {
   // See bug 2902 for background on STACK_SIZE_PARAM_IS_A_RESERVATION.
   // Set the reserved stack stack size to 1M, which is the default on Windows
   // and Linux.
-  DWORD thread_id;
   thread_ = ::CreateThread(NULL, 1024 * 1024, &StartThread, this,
-                           STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id);
+                           STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id_);
   RTC_CHECK(thread_) << "CreateThread failed";
+  RTC_DCHECK(thread_id_);
 #else
   ThreadAttributes attr;
   // Set the stack stack size to 1M.
@@ -157,6 +159,14 @@ bool PlatformThread::IsRunning() const {
 #endif  // defined(WEBRTC_WIN)
 }
 
+PlatformThreadRef PlatformThread::GetThreadRef() const {
+#if defined(WEBRTC_WIN)
+  return thread_id_;
+#else
+  return thread_;
+#endif  // defined(WEBRTC_WIN)
+}
+
 void PlatformThread::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (!IsRunning())
@@ -164,10 +174,13 @@ void PlatformThread::Stop() {
 
 #if defined(WEBRTC_WIN)
   // Set stop_ to |true| on the worker thread.
-  QueueUserAPC(&RaiseFlag, thread_, reinterpret_cast<ULONG_PTR>(&stop_));
+  bool queued = QueueAPC(&RaiseFlag, reinterpret_cast<ULONG_PTR>(&stop_));
+  // Queuing the APC can fail if the thread is being terminated.
+  RTC_CHECK(queued || GetLastError() == ERROR_GEN_FAILURE);
   WaitForSingleObject(thread_, INFINITE);
   CloseHandle(thread_);
   thread_ = nullptr;
+  thread_id_ = 0;
 #else
   stop_event_.Set();
   RTC_CHECK_EQ(0, pthread_join(thread_, nullptr));
@@ -247,4 +260,13 @@ bool PlatformThread::SetPriority(ThreadPriority priority) {
 #endif  // defined(WEBRTC_WIN)
 }
 
+#if defined(WEBRTC_WIN)
+bool PlatformThread::QueueAPC(PAPCFUNC function, ULONG_PTR data) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(IsRunning());
+
+  return QueueUserAPC(function, thread_, data) != FALSE;
+}
+#endif
+
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/platform_thread.h b/chromium/third_party/webrtc/base/platform_thread.h
index 53465e4b17a..d74aec28114 100644
--- a/chromium/third_party/webrtc/base/platform_thread.h
+++ b/chromium/third_party/webrtc/base/platform_thread.h
@@ -16,7 +16,6 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/platform_thread_types.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 
 namespace rtc {
@@ -59,18 +58,30 @@ class PlatformThread {
   PlatformThread(ThreadRunFunction func, void* obj, const char* thread_name);
   virtual ~PlatformThread();
 
+  const std::string& name() const { return name_; }
+
   // Spawns a thread and tries to set thread priority according to the priority
   // from when CreateThread was called.
   void Start();
 
   bool IsRunning() const;
 
+  // Returns an identifier for the worker thread that can be used to do
+  // thread checks.
+  PlatformThreadRef GetThreadRef() const;
+
   // Stops (joins) the spawned thread.
   void Stop();
 
   // Set the priority of the thread. Must be called when thread is running.
   bool SetPriority(ThreadPriority priority);
 
+ protected:
+#if defined(WEBRTC_WIN)
+  // Exposed to derived classes to allow for special cases specific to Windows.
+  bool QueueAPC(PAPCFUNC apc_function, ULONG_PTR data);
+#endif
+
  private:
   void Run();
 
@@ -85,6 +96,7 @@ class PlatformThread {
 
   bool stop_;
   HANDLE thread_;
+  DWORD thread_id_;
 #else
   static void* StartThread(void* param);
 
diff --git a/chromium/third_party/webrtc/base/platform_thread_unittest.cc b/chromium/third_party/webrtc/base/platform_thread_unittest.cc
index f9db8e34a30..847946aaca4 100644
--- a/chromium/third_party/webrtc/base/platform_thread_unittest.cc
+++ b/chromium/third_party/webrtc/base/platform_thread_unittest.cc
@@ -11,34 +11,49 @@
 #include "webrtc/base/platform_thread.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 
-namespace webrtc {
-
+namespace rtc {
+namespace {
 // Function that does nothing, and reports success.
 bool NullRunFunction(void* obj) {
-  SleepMs(0);  // Hand over timeslice, prevents busy looping.
+  webrtc::SleepMs(0);  // Hand over timeslice, prevents busy looping.
   return true;
 }
 
-TEST(PlatformThreadTest, StartStop) {
-  rtc::PlatformThread thread(&NullRunFunction, nullptr, "PlatformThreadTest");
-  thread.Start();
-  thread.Stop();
-}
-
 // Function that sets a boolean.
 bool SetFlagRunFunction(void* obj) {
   bool* obj_as_bool = static_cast<bool*>(obj);
   *obj_as_bool = true;
-  SleepMs(0);  // Hand over timeslice, prevents busy looping.
+  webrtc::SleepMs(0);  // Hand over timeslice, prevents busy looping.
   return true;
 }
+}  // namespace
+
+TEST(PlatformThreadTest, StartStop) {
+  PlatformThread thread(&NullRunFunction, nullptr, "PlatformThreadTest");
+  EXPECT_TRUE(thread.name() == "PlatformThreadTest");
+  EXPECT_TRUE(thread.GetThreadRef() == 0);
+  thread.Start();
+  EXPECT_TRUE(thread.GetThreadRef() != 0);
+  thread.Stop();
+  EXPECT_TRUE(thread.GetThreadRef() == 0);
+}
+
+TEST(PlatformThreadTest, StartStop2) {
+  PlatformThread thread1(&NullRunFunction, nullptr, "PlatformThreadTest1");
+  PlatformThread thread2(&NullRunFunction, nullptr, "PlatformThreadTest2");
+  EXPECT_TRUE(thread1.GetThreadRef() == thread2.GetThreadRef());
+  thread1.Start();
+  thread2.Start();
+  EXPECT_TRUE(thread1.GetThreadRef() != thread2.GetThreadRef());
+  thread2.Stop();
+  thread1.Stop();
+}
 
 TEST(PlatformThreadTest, RunFunctionIsCalled) {
   bool flag = false;
-  rtc::PlatformThread thread(&SetFlagRunFunction, &flag, "RunFunctionIsCalled");
+  PlatformThread thread(&SetFlagRunFunction, &flag, "RunFunctionIsCalled");
   thread.Start();
 
   // At this point, the flag may be either true or false.
@@ -47,5 +62,4 @@ TEST(PlatformThreadTest, RunFunctionIsCalled) {
   // We expect the thread to have run at least once.
   EXPECT_TRUE(flag);
 }
-
-}  // namespace webrtc
+}  // rtc
diff --git a/chromium/third_party/webrtc/base/profiler.h b/chromium/third_party/webrtc/base/profiler.h
index 419763fc8a1..4dd35f5baca 100644
--- a/chromium/third_party/webrtc/base/profiler.h
+++ b/chromium/third_party/webrtc/base/profiler.h
@@ -36,6 +36,7 @@
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/sharedexclusivelock.h"
 
diff --git a/chromium/third_party/webrtc/base/proxy_unittest.cc b/chromium/third_party/webrtc/base/proxy_unittest.cc
index d8a523fe17e..4dba0dd8f34 100644
--- a/chromium/third_party/webrtc/base/proxy_unittest.cc
+++ b/chromium/third_party/webrtc/base/proxy_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 #include "webrtc/base/autodetectproxy.h"
 #include "webrtc/base/gunit.h"
@@ -67,10 +68,10 @@ class ProxyTest : public testing::Test {
   }
 
  private:
-  rtc::scoped_ptr<rtc::SocketServer> ss_;
-  rtc::scoped_ptr<rtc::SocksProxyServer> socks_;
+  std::unique_ptr<rtc::SocketServer> ss_;
+  std::unique_ptr<rtc::SocksProxyServer> socks_;
   // TODO: Make this a real HTTPS proxy server.
-  rtc::scoped_ptr<rtc::HttpListenServer> https_;
+  std::unique_ptr<rtc::HttpListenServer> https_;
 };
 
 // Tests whether we can use a SOCKS5 proxy to connect to a server.
diff --git a/chromium/third_party/webrtc/base/proxydetect.cc b/chromium/third_party/webrtc/base/proxydetect.cc
index abb8f0a9279..10e7a02241c 100644
--- a/chromium/third_party/webrtc/base/proxydetect.cc
+++ b/chromium/third_party/webrtc/base/proxydetect.cc
@@ -29,6 +29,7 @@
 #endif
 
 #include <map>
+#include <memory>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/fileutils.h"
@@ -430,7 +431,7 @@ bool GetDefaultFirefoxProfile(Pathname* profile_path) {
   // Note: we are looking for the first entry with "Default=1", or the last
   // entry in the file
   path.SetFilename("profiles.ini");
-  scoped_ptr<FileStream> fs(Filesystem::OpenFile(path, "r"));
+  std::unique_ptr<FileStream> fs(Filesystem::OpenFile(path, "r"));
   if (!fs) {
     return false;
   }
@@ -495,7 +496,7 @@ bool GetDefaultFirefoxProfile(Pathname* profile_path) {
 bool ReadFirefoxPrefs(const Pathname& filename,
                       const char * prefix,
                       StringMap* settings) {
-  scoped_ptr<FileStream> fs(Filesystem::OpenFile(filename, "r"));
+  std::unique_ptr<FileStream> fs(Filesystem::OpenFile(filename, "r"));
   if (!fs) {
     LOG(LS_ERROR) << "Failed to open file: " << filename.pathname();
     return false;
diff --git a/chromium/third_party/webrtc/base/proxyserver.h b/chromium/third_party/webrtc/base/proxyserver.h
index adb26ae9d04..86007c3606f 100644
--- a/chromium/third_party/webrtc/base/proxyserver.h
+++ b/chromium/third_party/webrtc/base/proxyserver.h
@@ -12,7 +12,9 @@
 #define WEBRTC_BASE_PROXYSERVER_H_
 
 #include <list>
+#include <memory>
 #include "webrtc/base/asyncsocket.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/socketadapters.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/stream.h"
@@ -50,8 +52,8 @@ class ProxyBinding : public sigslot::has_slots<> {
   void Destroy();
 
   static const int kBufferSize = 4096;
-  scoped_ptr<AsyncProxyServerSocket> int_socket_;
-  scoped_ptr<AsyncSocket> ext_socket_;
+  std::unique_ptr<AsyncProxyServerSocket> int_socket_;
+  std::unique_ptr<AsyncSocket> ext_socket_;
   bool connected_;
   FifoBuffer out_buffer_;
   FifoBuffer in_buffer_;
@@ -76,7 +78,7 @@ class ProxyServer : public sigslot::has_slots<> {
   typedef std::list<ProxyBinding*> BindingList;
   SocketFactory* ext_factory_;
   SocketAddress ext_ip_;
-  scoped_ptr<AsyncSocket> server_socket_;
+  std::unique_ptr<AsyncSocket> server_socket_;
   BindingList bindings_;
   RTC_DISALLOW_COPY_AND_ASSIGN(ProxyServer);
 };
diff --git a/chromium/third_party/webrtc/base/rate_statistics.cc b/chromium/third_party/webrtc/base/rate_statistics.cc
index 8db2851e682..6529aa1f7a6 100644
--- a/chromium/third_party/webrtc/base/rate_statistics.cc
+++ b/chromium/third_party/webrtc/base/rate_statistics.cc
@@ -10,7 +10,9 @@
 
 #include "webrtc/base/rate_statistics.h"
 
-#include <assert.h>
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 
@@ -20,7 +22,7 @@ RateStatistics::RateStatistics(uint32_t window_size_ms, float scale)
       accumulated_count_(0),
       oldest_time_(0),
       oldest_index_(0),
-      scale_(scale / (num_buckets_ - 1)) {}
+      scale_(scale) {}
 
 RateStatistics::~RateStatistics() {}
 
@@ -42,7 +44,7 @@ void RateStatistics::Update(size_t count, int64_t now_ms) {
   EraseOld(now_ms);
 
   int now_offset = static_cast<int>(now_ms - oldest_time_);
-  assert(now_offset < num_buckets_);
+  RTC_DCHECK_LT(now_offset, num_buckets_);
   int index = oldest_index_ + now_offset;
   if (index >= num_buckets_) {
     index -= num_buckets_;
@@ -53,18 +55,20 @@ void RateStatistics::Update(size_t count, int64_t now_ms) {
 
 uint32_t RateStatistics::Rate(int64_t now_ms) {
   EraseOld(now_ms);
-  return static_cast<uint32_t>(accumulated_count_ * scale_ + 0.5f);
+  float scale =  scale_ / (now_ms - oldest_time_ + 1);
+  return static_cast<uint32_t>(accumulated_count_ * scale + 0.5f);
 }
 
 void RateStatistics::EraseOld(int64_t now_ms) {
   int64_t new_oldest_time = now_ms - num_buckets_ + 1;
   if (new_oldest_time <= oldest_time_) {
+    if (accumulated_count_ == 0)
+      oldest_time_ = now_ms;
     return;
   }
-
   while (oldest_time_ < new_oldest_time) {
     size_t count_in_oldest_bucket = buckets_[oldest_index_];
-    assert(accumulated_count_ >= count_in_oldest_bucket);
+    RTC_DCHECK_GE(accumulated_count_, count_in_oldest_bucket);
     accumulated_count_ -= count_in_oldest_bucket;
     buckets_[oldest_index_] = 0;
     if (++oldest_index_ >= num_buckets_) {
@@ -74,6 +78,7 @@ void RateStatistics::EraseOld(int64_t now_ms) {
     if (accumulated_count_ == 0) {
       // This guarantees we go through all the buckets at most once, even if
       // |new_oldest_time| is far greater than |oldest_time_|.
+      new_oldest_time = now_ms;
       break;
     }
   }
diff --git a/chromium/third_party/webrtc/base/rate_statistics.h b/chromium/third_party/webrtc/base/rate_statistics.h
index 21f6ce61607..aea8d793fed 100644
--- a/chromium/third_party/webrtc/base/rate_statistics.h
+++ b/chromium/third_party/webrtc/base/rate_statistics.h
@@ -11,7 +11,8 @@
 #ifndef WEBRTC_BASE_RATE_STATISTICS_H_
 #define WEBRTC_BASE_RATE_STATISTICS_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -34,7 +35,7 @@ class RateStatistics {
   // Counters are kept in buckets (circular buffer), with one bucket
   // per millisecond.
   const int num_buckets_;
-  rtc::scoped_ptr<size_t[]> buckets_;
+  std::unique_ptr<size_t[]> buckets_;
 
   // Total count recorded in buckets.
   size_t accumulated_count_;
diff --git a/chromium/third_party/webrtc/base/rate_statistics_unittest.cc b/chromium/third_party/webrtc/base/rate_statistics_unittest.cc
index 0270253d5e2..9702da0699d 100644
--- a/chromium/third_party/webrtc/base/rate_statistics_unittest.cc
+++ b/chromium/third_party/webrtc/base/rate_statistics_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/rate_statistics.h"
 
@@ -15,9 +17,11 @@ namespace {
 
 using webrtc::RateStatistics;
 
+const int64_t kWindowMs = 500;
+
 class RateStatisticsTest : public ::testing::Test {
  protected:
-  RateStatisticsTest() : stats_(500, 8000) {}
+  RateStatisticsTest() : stats_(kWindowMs, 8000) {}
   RateStatistics stats_;
 };
 
@@ -26,8 +30,9 @@ TEST_F(RateStatisticsTest, TestStrictMode) {
   // Should be initialized to 0.
   EXPECT_EQ(0u, stats_.Rate(now_ms));
   stats_.Update(1500, now_ms);
-  // Expecting 24 kbps given a 500 ms window with one 1500 bytes packet.
-  EXPECT_EQ(24000u, stats_.Rate(now_ms));
+  // Expecting 1200 kbps since the window is initially kept small and grows as
+  // we have more data.
+  EXPECT_EQ(12000000u, stats_.Rate(now_ms));
   stats_.Reset();
   // Expecting 0 after init.
   EXPECT_EQ(0u, stats_.Rate(now_ms));
@@ -37,12 +42,12 @@ TEST_F(RateStatisticsTest, TestStrictMode) {
     }
     // Approximately 1200 kbps expected. Not exact since when packets
     // are removed we will jump 10 ms to the next packet.
-    if (now_ms > 0 && now_ms % 500 == 0) {
-      EXPECT_NEAR(1200000u, stats_.Rate(now_ms), 24000u);
+    if (now_ms > 0 && now_ms % kWindowMs == 0) {
+      EXPECT_NEAR(1200000u, stats_.Rate(now_ms), 22000u);
     }
     now_ms += 1;
   }
-  now_ms += 500;
+  now_ms += kWindowMs;
   // The window is 2 seconds. If nothing has been received for that time
   // the estimate should be 0.
   EXPECT_EQ(0u, stats_.Rate(now_ms));
@@ -54,25 +59,26 @@ TEST_F(RateStatisticsTest, IncreasingThenDecreasingBitrate) {
   // Expecting 0 after init.
   uint32_t bitrate = stats_.Rate(now_ms);
   EXPECT_EQ(0u, bitrate);
+  const uint32_t kExpectedBitrate = 8000000;
   // 1000 bytes per millisecond until plateau is reached.
+  int prev_error = kExpectedBitrate;
   while (++now_ms < 10000) {
     stats_.Update(1000, now_ms);
-    uint32_t new_bitrate = stats_.Rate(now_ms);
-    if (new_bitrate != bitrate) {
-      // New bitrate must be higher than previous one.
-      EXPECT_GT(new_bitrate, bitrate);
-    } else {
-      // Plateau reached, 8000 kbps expected.
-      EXPECT_NEAR(8000000u, bitrate, 80000u);
-      break;
-    }
-    bitrate = new_bitrate;
+    bitrate = stats_.Rate(now_ms);
+    int error = kExpectedBitrate - bitrate;
+    error = std::abs(error);
+    // Expect the estimation error to decrease as the window is extended.
+    EXPECT_LE(error, prev_error + 1);
+    prev_error = error;
   }
+  // Window filled, expect to be close to 8000000.
+  EXPECT_EQ(kExpectedBitrate, bitrate);
+
   // 1000 bytes per millisecond until 10-second mark, 8000 kbps expected.
   while (++now_ms < 10000) {
     stats_.Update(1000, now_ms);
     bitrate = stats_.Rate(now_ms);
-    EXPECT_NEAR(8000000u, bitrate, 80000u);
+    EXPECT_EQ(kExpectedBitrate, bitrate);
   }
   // Zero bytes per millisecond until 0 is reached.
   while (++now_ms < 20000) {
@@ -94,4 +100,33 @@ TEST_F(RateStatisticsTest, IncreasingThenDecreasingBitrate) {
     EXPECT_EQ(0u, stats_.Rate(now_ms));
   }
 }
+
+TEST_F(RateStatisticsTest, ResetAfterSilence) {
+  int64_t now_ms = 0;
+  stats_.Reset();
+  // Expecting 0 after init.
+  uint32_t bitrate = stats_.Rate(now_ms);
+  EXPECT_EQ(0u, bitrate);
+  const uint32_t kExpectedBitrate = 8000000;
+  // 1000 bytes per millisecond until the window has been filled.
+  int prev_error = kExpectedBitrate;
+  while (++now_ms < 10000) {
+    stats_.Update(1000, now_ms);
+    bitrate = stats_.Rate(now_ms);
+    int error = kExpectedBitrate - bitrate;
+    error = std::abs(error);
+    // Expect the estimation error to decrease as the window is extended.
+    EXPECT_LE(error, prev_error + 1);
+    prev_error = error;
+  }
+  // Window filled, expect to be close to 8000000.
+  EXPECT_EQ(kExpectedBitrate, bitrate);
+
+  now_ms += kWindowMs + 1;
+  EXPECT_EQ(0u, stats_.Rate(now_ms));
+  stats_.Update(1000, now_ms);
+  // We expect one sample of 1000 bytes, and that the bitrate is measured over
+  // 1 ms, i.e., 8 * 1000 / 0.001 = 8000000.
+  EXPECT_EQ(kExpectedBitrate, stats_.Rate(now_ms));
+}
 }  // namespace
diff --git a/chromium/third_party/webrtc/base/ratetracker.cc b/chromium/third_party/webrtc/base/ratetracker.cc
index c1ad2d5e844..a59ec2fc3f4 100644
--- a/chromium/third_party/webrtc/base/ratetracker.cc
+++ b/chromium/third_party/webrtc/base/ratetracker.cc
@@ -19,14 +19,16 @@
 
 namespace rtc {
 
-RateTracker::RateTracker(uint32_t bucket_milliseconds, size_t bucket_count)
+static const int64_t kTimeUnset = -1;
+
+RateTracker::RateTracker(int64_t bucket_milliseconds, size_t bucket_count)
     : bucket_milliseconds_(bucket_milliseconds),
       bucket_count_(bucket_count),
       sample_buckets_(new size_t[bucket_count + 1]),
       total_sample_count_(0u),
-      bucket_start_time_milliseconds_(~0u) {
-  RTC_CHECK(bucket_milliseconds > 0u);
-  RTC_CHECK(bucket_count > 0u);
+      bucket_start_time_milliseconds_(kTimeUnset) {
+  RTC_CHECK(bucket_milliseconds > 0);
+  RTC_CHECK(bucket_count > 0);
 }
 
 RateTracker::~RateTracker() {
@@ -34,33 +36,33 @@ RateTracker::~RateTracker() {
 }
 
 double RateTracker::ComputeRateForInterval(
-    uint32_t interval_milliseconds) const {
-  if (bucket_start_time_milliseconds_ == ~0u) {
+    int64_t interval_milliseconds) const {
+  if (bucket_start_time_milliseconds_ == kTimeUnset) {
     return 0.0;
   }
-  uint32_t current_time = Time();
+  int64_t current_time = Time();
   // Calculate which buckets to sum up given the current time.  If the time
   // has passed to a new bucket then we have to skip some of the oldest buckets.
-  uint32_t available_interval_milliseconds = std::min<uint32_t>(
-      interval_milliseconds,
-      bucket_milliseconds_ * static_cast<uint32_t>(bucket_count_));
+  int64_t available_interval_milliseconds =
+      std::min(interval_milliseconds,
+               bucket_milliseconds_ * static_cast<int64_t>(bucket_count_));
   // number of old buckets (i.e. after the current bucket in the ring buffer)
   // that are expired given our current time interval.
   size_t buckets_to_skip;
   // Number of milliseconds of the first bucket that are not a portion of the
   // current interval.
-  uint32_t milliseconds_to_skip;
+  int64_t milliseconds_to_skip;
   if (current_time >
       initialization_time_milliseconds_ + available_interval_milliseconds) {
-    uint32_t time_to_skip =
+    int64_t time_to_skip =
         current_time - bucket_start_time_milliseconds_ +
-        static_cast<uint32_t>(bucket_count_) * bucket_milliseconds_ -
+        static_cast<int64_t>(bucket_count_) * bucket_milliseconds_ -
         available_interval_milliseconds;
     buckets_to_skip = time_to_skip / bucket_milliseconds_;
     milliseconds_to_skip = time_to_skip % bucket_milliseconds_;
   } else {
     buckets_to_skip = bucket_count_ - current_bucket_;
-    milliseconds_to_skip = 0u;
+    milliseconds_to_skip = 0;
     available_interval_milliseconds =
         TimeDiff(current_time, initialization_time_milliseconds_);
     // Let one bucket interval pass after initialization before reporting.
@@ -70,8 +72,7 @@ double RateTracker::ComputeRateForInterval(
   }
   // If we're skipping all buckets that means that there have been no samples
   // within the sampling interval so report 0.
-  if (buckets_to_skip > bucket_count_ ||
-      available_interval_milliseconds == 0u) {
+  if (buckets_to_skip > bucket_count_ || available_interval_milliseconds == 0) {
     return 0.0;
   }
   size_t start_bucket = NextBucketIndex(current_bucket_ + buckets_to_skip);
@@ -88,21 +89,21 @@ double RateTracker::ComputeRateForInterval(
     total_samples += sample_buckets_[i];
   }
   // Convert to samples per second.
-  return static_cast<double>(total_samples * 1000u) /
-      static_cast<double>(available_interval_milliseconds);
+  return static_cast<double>(total_samples * 1000) /
+         static_cast<double>(available_interval_milliseconds);
 }
 
 double RateTracker::ComputeTotalRate() const {
-  if (bucket_start_time_milliseconds_ == ~0u) {
+  if (bucket_start_time_milliseconds_ == kTimeUnset) {
     return 0.0;
   }
-  uint32_t current_time = Time();
-  if (TimeIsLaterOrEqual(current_time, initialization_time_milliseconds_)) {
+  int64_t current_time = Time();
+  if (current_time <= initialization_time_milliseconds_) {
     return 0.0;
   }
-  return static_cast<double>(total_sample_count_ * 1000u) /
-      static_cast<double>(
-          TimeDiff(current_time, initialization_time_milliseconds_));
+  return static_cast<double>(total_sample_count_ * 1000) /
+         static_cast<double>(
+             TimeDiff(current_time, initialization_time_milliseconds_));
 }
 
 size_t RateTracker::TotalSampleCount() const {
@@ -111,15 +112,16 @@ size_t RateTracker::TotalSampleCount() const {
 
 void RateTracker::AddSamples(size_t sample_count) {
   EnsureInitialized();
-  uint32_t current_time = Time();
+  int64_t current_time = Time();
   // Advance the current bucket as needed for the current time, and reset
   // bucket counts as we advance.
-  for (size_t i = 0u; i <= bucket_count_ &&
-      current_time >= bucket_start_time_milliseconds_ + bucket_milliseconds_;
-      ++i) {
+  for (size_t i = 0;
+       i <= bucket_count_ &&
+       current_time >= bucket_start_time_milliseconds_ + bucket_milliseconds_;
+       ++i) {
     bucket_start_time_milliseconds_ += bucket_milliseconds_;
     current_bucket_ = NextBucketIndex(current_bucket_);
-    sample_buckets_[current_bucket_] = 0u;
+    sample_buckets_[current_bucket_] = 0;
   }
   // Ensure that bucket_start_time_milliseconds_ is updated appropriately if
   // the entire buffer of samples has been expired.
@@ -130,18 +132,18 @@ void RateTracker::AddSamples(size_t sample_count) {
   total_sample_count_ += sample_count;
 }
 
-uint32_t RateTracker::Time() const {
-  return rtc::Time();
+int64_t RateTracker::Time() const {
+  return rtc::TimeMillis();
 }
 
 void RateTracker::EnsureInitialized() {
-  if (bucket_start_time_milliseconds_ == ~0u) {
+  if (bucket_start_time_milliseconds_ == kTimeUnset) {
     initialization_time_milliseconds_ = Time();
     bucket_start_time_milliseconds_ = initialization_time_milliseconds_;
-    current_bucket_ = 0u;
+    current_bucket_ = 0;
     // We only need to initialize the first bucket because we reset buckets when
     // current_bucket_ increments.
-    sample_buckets_[current_bucket_] = 0u;
+    sample_buckets_[current_bucket_] = 0;
   }
 }
 
diff --git a/chromium/third_party/webrtc/base/ratetracker.h b/chromium/third_party/webrtc/base/ratetracker.h
index d49d7cacdd5..6ae9bec119d 100644
--- a/chromium/third_party/webrtc/base/ratetracker.h
+++ b/chromium/third_party/webrtc/base/ratetracker.h
@@ -21,19 +21,19 @@ namespace rtc {
 // that over each bucket the rate was constant.
 class RateTracker {
  public:
-  RateTracker(uint32_t bucket_milliseconds, size_t bucket_count);
+  RateTracker(int64_t bucket_milliseconds, size_t bucket_count);
   virtual ~RateTracker();
 
   // Computes the average rate over the most recent interval_milliseconds,
   // or if the first sample was added within this period, computes the rate
   // since the first sample was added.
-  double ComputeRateForInterval(uint32_t interval_milliseconds) const;
+  double ComputeRateForInterval(int64_t interval_milliseconds) const;
 
   // Computes the average rate over the rate tracker's recording interval
   // of bucket_milliseconds * bucket_count.
   double ComputeRate() const {
     return ComputeRateForInterval(bucket_milliseconds_ *
-                                  static_cast<uint32_t>(bucket_count_));
+                                  static_cast<int64_t>(bucket_count_));
   }
 
   // Computes the average rate since the first sample was added to the
@@ -49,19 +49,19 @@ class RateTracker {
 
  protected:
   // overrideable for tests
-  virtual uint32_t Time() const;
+  virtual int64_t Time() const;
 
  private:
   void EnsureInitialized();
   size_t NextBucketIndex(size_t bucket_index) const;
 
-  const uint32_t bucket_milliseconds_;
+  const int64_t bucket_milliseconds_;
   const size_t bucket_count_;
   size_t* sample_buckets_;
   size_t total_sample_count_;
   size_t current_bucket_;
-  uint32_t bucket_start_time_milliseconds_;
-  uint32_t initialization_time_milliseconds_;
+  int64_t bucket_start_time_milliseconds_;
+  int64_t initialization_time_milliseconds_;
 };
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/ratetracker_unittest.cc b/chromium/third_party/webrtc/base/ratetracker_unittest.cc
index 75fec558188..136934fea34 100644
--- a/chromium/third_party/webrtc/base/ratetracker_unittest.cc
+++ b/chromium/third_party/webrtc/base/ratetracker_unittest.cc
@@ -19,11 +19,11 @@ namespace {
 class RateTrackerForTest : public RateTracker {
  public:
   RateTrackerForTest() : RateTracker(kBucketIntervalMs, 10u), time_(0) {}
-  virtual uint32_t Time() const { return time_; }
-  void AdvanceTime(uint32_t delta) { time_ += delta; }
+  virtual int64_t Time() const { return time_; }
+  void AdvanceTime(int delta) { time_ += delta; }
 
  private:
-  uint32_t time_;
+  int64_t time_;
 };
 
 TEST(RateTrackerTest, Test30FPS) {
@@ -36,7 +36,7 @@ TEST(RateTrackerTest, Test30FPS) {
       tracker.AdvanceTime(1);
     }
   }
-  EXPECT_DOUBLE_EQ(30.0, tracker.ComputeRateForInterval(50000u));
+  EXPECT_DOUBLE_EQ(30.0, tracker.ComputeRateForInterval(50000));
 }
 
 TEST(RateTrackerTest, Test60FPS) {
@@ -49,12 +49,12 @@ TEST(RateTrackerTest, Test60FPS) {
       tracker.AdvanceTime(1);
     }
   }
-  EXPECT_DOUBLE_EQ(60.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(60.0, tracker.ComputeRateForInterval(1000));
 }
 
 TEST(RateTrackerTest, TestRateTrackerBasics) {
   RateTrackerForTest tracker;
-  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(1000));
 
   // Add a sample.
   tracker.AddSamples(1234);
@@ -63,7 +63,7 @@ TEST(RateTrackerTest, TestRateTrackerBasics) {
   EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRate());
   // Advance the clock by 100 ms (one bucket interval).
   tracker.AdvanceTime(1);
-  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRateForInterval(1000));
   EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRate());
   EXPECT_EQ(1234U, tracker.TotalSampleCount());
   EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeTotalRate());
@@ -71,7 +71,7 @@ TEST(RateTrackerTest, TestRateTrackerBasics) {
   // Repeat.
   tracker.AddSamples(1234);
   tracker.AdvanceTime(100);
-  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRateForInterval(1000));
   EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeRate());
   EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
   EXPECT_DOUBLE_EQ(12340.0, tracker.ComputeTotalRate());
@@ -79,20 +79,20 @@ TEST(RateTrackerTest, TestRateTrackerBasics) {
   // Advance the clock by 800 ms, so we've elapsed a full second.
   // units_second should now be filled in properly.
   tracker.AdvanceTime(800);
-  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRateForInterval(1000));
   EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRate());
   EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
   EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeTotalRate());
 
   // Poll the tracker again immediately. The reported rate should stay the same.
-  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRateForInterval(1000));
   EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeRate());
   EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
   EXPECT_DOUBLE_EQ(1234.0 * 2.0, tracker.ComputeTotalRate());
 
   // Do nothing and advance by a second. We should drop down to zero.
   tracker.AdvanceTime(1000);
-  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(1000));
   EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRate());
   EXPECT_EQ(1234U * 2, tracker.TotalSampleCount());
   EXPECT_DOUBLE_EQ(1234.0, tracker.ComputeTotalRate());
@@ -103,7 +103,7 @@ TEST(RateTrackerTest, TestRateTrackerBasics) {
     tracker.AddSamples(9876U);
     tracker.AdvanceTime(100);
   }
-  EXPECT_DOUBLE_EQ(9876.0 * 10.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(9876.0 * 10.0, tracker.ComputeRateForInterval(1000));
   EXPECT_DOUBLE_EQ(9876.0 * 10.0, tracker.ComputeRate());
   EXPECT_EQ(1234U * 2 + 9876U * 55, tracker.TotalSampleCount());
   EXPECT_DOUBLE_EQ((1234.0 * 2.0 + 9876.0 * 55.0) / 7.5,
@@ -112,14 +112,14 @@ TEST(RateTrackerTest, TestRateTrackerBasics) {
   // Advance the clock by 500 ms. Since we sent nothing over this half-second,
   // the reported rate should be reduced by half.
   tracker.AdvanceTime(500);
-  EXPECT_DOUBLE_EQ(9876.0 * 5.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(9876.0 * 5.0, tracker.ComputeRateForInterval(1000));
   EXPECT_DOUBLE_EQ(9876.0 * 5.0, tracker.ComputeRate());
   EXPECT_EQ(1234U * 2 + 9876U * 55, tracker.TotalSampleCount());
   EXPECT_DOUBLE_EQ((1234.0 * 2.0 + 9876.0 * 55.0) / 8.0,
       tracker.ComputeTotalRate());
 
   // Rate over the last half second should be zero.
-  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(500u));
+  EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(500));
 }
 
 TEST(RateTrackerTest, TestLongPeriodBetweenSamples) {
@@ -149,7 +149,7 @@ TEST(RateTrackerTest, TestRolloff) {
     tracker.AdvanceTime(50);
   }
   EXPECT_DOUBLE_EQ(15.0, tracker.ComputeRate());
-  EXPECT_DOUBLE_EQ(20.0, tracker.ComputeRateForInterval(500u));
+  EXPECT_DOUBLE_EQ(20.0, tracker.ComputeRateForInterval(500));
 
   for (int i = 0; i < 10; ++i) {
     tracker.AddSamples(1U);
@@ -162,7 +162,7 @@ TEST(RateTrackerTest, TestGetUnitSecondsAfterInitialValue) {
   RateTrackerForTest tracker;
   tracker.AddSamples(1234);
   tracker.AdvanceTime(1000);
-  EXPECT_DOUBLE_EQ(1234.0, tracker.ComputeRateForInterval(1000u));
+  EXPECT_DOUBLE_EQ(1234.0, tracker.ComputeRateForInterval(1000));
 }
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/referencecountedsingletonfactory.h b/chromium/third_party/webrtc/base/referencecountedsingletonfactory.h
index f9559868274..500150f6fd5 100644
--- a/chromium/third_party/webrtc/base/referencecountedsingletonfactory.h
+++ b/chromium/third_party/webrtc/base/referencecountedsingletonfactory.h
@@ -11,10 +11,12 @@
 #ifndef WEBRTC_BASE_REFERENCECOUNTEDSINGLETONFACTORY_H_
 #define WEBRTC_BASE_REFERENCECOUNTEDSINGLETONFACTORY_H_
 
+#include <memory>
+
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
@@ -41,11 +43,11 @@ class ReferenceCountedSingletonFactory {
  protected:
   // Must be implemented in a sub-class. The sub-class may choose whether or not
   // to cache the instance across lifetimes by either reset()'ing or not
-  // reset()'ing the scoped_ptr in CleanupInstance().
+  // reset()'ing the unique_ptr in CleanupInstance().
   virtual bool SetupInstance() = 0;
   virtual void CleanupInstance() = 0;
 
-  scoped_ptr<Interface> instance_;
+  std::unique_ptr<Interface> instance_;
 
  private:
   Interface* GetInstance() {
diff --git a/chromium/third_party/webrtc/base/rollingaccumulator.h b/chromium/third_party/webrtc/base/rollingaccumulator.h
index 0c8e5fb8bdc..72415ad758d 100644
--- a/chromium/third_party/webrtc/base/rollingaccumulator.h
+++ b/chromium/third_party/webrtc/base/rollingaccumulator.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 
 namespace rtc {
 
diff --git a/chromium/third_party/webrtc/base/rtccertificate.cc b/chromium/third_party/webrtc/base/rtccertificate.cc
index 7b764bd72e6..574bf75bf29 100644
--- a/chromium/third_party/webrtc/base/rtccertificate.cc
+++ b/chromium/third_party/webrtc/base/rtccertificate.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/rtccertificate.h"
 
 #include "webrtc/base/checks.h"
@@ -15,7 +17,7 @@
 namespace rtc {
 
 scoped_refptr<RTCCertificate> RTCCertificate::Create(
-    scoped_ptr<SSLIdentity> identity) {
+    std::unique_ptr<SSLIdentity> identity) {
   return new RefCountedObject<RTCCertificate>(identity.release());
 }
 
@@ -43,4 +45,24 @@ const SSLCertificate& RTCCertificate::ssl_certificate() const {
   return identity_->certificate();
 }
 
+RTCCertificatePEM RTCCertificate::ToPEM() const {
+  return RTCCertificatePEM(identity_->PrivateKeyToPEMString(),
+                           ssl_certificate().ToPEMString());
+}
+
+scoped_refptr<RTCCertificate> RTCCertificate::FromPEM(
+    const RTCCertificatePEM& pem) {
+  std::unique_ptr<SSLIdentity> identity(SSLIdentity::FromPEMStrings(
+      pem.private_key(), pem.certificate()));
+  return new RefCountedObject<RTCCertificate>(identity.release());
+}
+
+bool RTCCertificate::operator==(const RTCCertificate& certificate) const {
+  return *this->identity_ == *certificate.identity_;
+}
+
+bool RTCCertificate::operator!=(const RTCCertificate& certificate) const {
+  return !(*this == certificate);
+}
+
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/rtccertificate.h b/chromium/third_party/webrtc/base/rtccertificate.h
index 600739bc86a..46d6fd427cf 100644
--- a/chromium/third_party/webrtc/base/rtccertificate.h
+++ b/chromium/third_party/webrtc/base/rtccertificate.h
@@ -11,21 +11,45 @@
 #ifndef WEBRTC_BASE_RTCCERTIFICATE_H_
 #define WEBRTC_BASE_RTCCERTIFICATE_H_
 
+#include <memory>
+
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/sslidentity.h"
 
 namespace rtc {
 
+// This class contains PEM strings of an RTCCertificate's private key and
+// certificate and acts as a text representation of RTCCertificate. Certificates
+// can be serialized and deserialized to and from this format, which allows for
+// cloning and storing of certificates to disk. The PEM format is that of
+// |SSLIdentity::PrivateKeyToPEMString| and |SSLCertificate::ToPEMString|, e.g.
+// the string representations used by OpenSSL.
+class RTCCertificatePEM {
+ public:
+  RTCCertificatePEM(
+      const std::string& private_key,
+      const std::string& certificate)
+      : private_key_(private_key),
+        certificate_(certificate) {}
+
+  const std::string& private_key() const { return private_key_; }
+  const std::string& certificate() const { return certificate_; }
+
+ private:
+  std::string private_key_;
+  std::string certificate_;
+};
+
 // A thin abstraction layer between "lower level crypto stuff" like
 // SSLCertificate and WebRTC usage. Takes ownership of some lower level objects,
 // reference counting protects these from premature destruction.
 class RTCCertificate : public RefCountInterface {
  public:
   // Takes ownership of |identity|.
-  static scoped_refptr<RTCCertificate> Create(scoped_ptr<SSLIdentity> identity);
+  static scoped_refptr<RTCCertificate> Create(
+      std::unique_ptr<SSLIdentity> identity);
 
   // Returns the expiration time in ms relative to epoch, 1970-01-01T00:00:00Z.
   uint64_t Expires() const;
@@ -40,6 +64,12 @@ class RTCCertificate : public RefCountInterface {
   // However, some places might need SSLIdentity* for its public/private key...
   SSLIdentity* identity() const { return identity_.get(); }
 
+  // To/from PEM, a text representation of the RTCCertificate.
+  RTCCertificatePEM ToPEM() const;
+  static scoped_refptr<RTCCertificate> FromPEM(const RTCCertificatePEM& pem);
+  bool operator==(const RTCCertificate& certificate) const;
+  bool operator!=(const RTCCertificate& certificate) const;
+
  protected:
   explicit RTCCertificate(SSLIdentity* identity);
   ~RTCCertificate() override;
@@ -47,7 +77,7 @@ class RTCCertificate : public RefCountInterface {
  private:
   // The SSLIdentity is the owner of the SSLCertificate. To protect our
   // ssl_certificate() we take ownership of |identity_|.
-  scoped_ptr<SSLIdentity> identity_;
+  std::unique_ptr<SSLIdentity> identity_;
 };
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/rtccertificate_unittests.cc b/chromium/third_party/webrtc/base/rtccertificate_unittest.cc
index 84c854478b2..f5df7f1130b 100644
--- a/chromium/third_party/webrtc/base/rtccertificate_unittests.cc
+++ b/chromium/third_party/webrtc/base/rtccertificate_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <utility>
 
 #include "webrtc/base/checks.h"
@@ -16,7 +17,6 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rtccertificate.h"
 #include "webrtc/base/safe_conversions.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
@@ -35,6 +35,13 @@ class RTCCertificateTest : public testing::Test {
   ~RTCCertificateTest() {}
 
  protected:
+  scoped_refptr<RTCCertificate> GenerateECDSA() {
+    std::unique_ptr<SSLIdentity> identity(
+        SSLIdentity::Generate(kTestCertCommonName, KeyParams::ECDSA()));
+    RTC_CHECK(identity);
+    return RTCCertificate::Create(std::move(identity));
+  }
+
   // Timestamp note:
   //   All timestamps in this unittest are expressed in number of seconds since
   // epoch, 1970-01-01T00:00:00Z (UTC). The RTCCertificate interface uses ms,
@@ -77,7 +84,7 @@ class RTCCertificateTest : public testing::Test {
     // is fast to generate.
     params.key_params = KeyParams::ECDSA();
 
-    scoped_ptr<SSLIdentity> identity(SSLIdentity::GenerateForTest(params));
+    std::unique_ptr<SSLIdentity> identity(SSLIdentity::GenerateForTest(params));
     return RTCCertificate::Create(std::move(identity));
   }
 };
@@ -85,10 +92,7 @@ class RTCCertificateTest : public testing::Test {
 TEST_F(RTCCertificateTest, NewCertificateNotExpired) {
   // Generate a real certificate without specifying the expiration time.
   // Certificate type doesn't matter, using ECDSA because it's fast to generate.
-  scoped_ptr<SSLIdentity> identity(
-      SSLIdentity::Generate(kTestCertCommonName, KeyParams::ECDSA()));
-  scoped_refptr<RTCCertificate> certificate =
-      RTCCertificate::Create(std::move(identity));
+  scoped_refptr<RTCCertificate> certificate = GenerateECDSA();
 
   uint64_t now = NowSeconds();
   EXPECT_FALSE(HasExpiredSeconds(certificate, now));
@@ -115,4 +119,22 @@ TEST_F(RTCCertificateTest, ExpiresInOneSecond) {
   EXPECT_TRUE(HasExpiredSeconds(certificate, now + 2));
 }
 
+TEST_F(RTCCertificateTest, DifferentCertificatesNotEqual) {
+  scoped_refptr<RTCCertificate> a = GenerateECDSA();
+  scoped_refptr<RTCCertificate> b = GenerateECDSA();
+  EXPECT_TRUE(*a != *b);
+}
+
+TEST_F(RTCCertificateTest, CloneWithPEMSerialization) {
+  scoped_refptr<RTCCertificate> orig = GenerateECDSA();
+
+  // To PEM.
+  RTCCertificatePEM orig_pem = orig->ToPEM();
+  // Clone from PEM.
+  scoped_refptr<RTCCertificate> clone = RTCCertificate::FromPEM(orig_pem);
+  EXPECT_TRUE(clone);
+  EXPECT_TRUE(*orig == *clone);
+  EXPECT_EQ(orig->Expires(), clone->Expires());
+}
+
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/rtccertificategenerator.cc b/chromium/third_party/webrtc/base/rtccertificategenerator.cc
new file mode 100644
index 00000000000..f0d72e7fe4d
--- /dev/null
+++ b/chromium/third_party/webrtc/base/rtccertificategenerator.cc
@@ -0,0 +1,158 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/rtccertificategenerator.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/sslidentity.h"
+
+namespace rtc {
+
+namespace {
+
+// A certificates' subject and issuer name.
+const char kIdentityName[] = "WebRTC";
+
+uint64_t kYearInSeconds = 365 * 24 * 60 * 60;
+
+enum {
+  MSG_GENERATE,
+  MSG_GENERATE_DONE,
+};
+
+// Helper class for generating certificates asynchronously; a single task
+// instance is responsible for a single asynchronous certificate generation
+// request. We are using a separate helper class so that a generation request
+// can outlive the |RTCCertificateGenerator| that spawned it.
+class RTCCertificateGenerationTask : public RefCountInterface,
+                                     public MessageHandler {
+ public:
+  RTCCertificateGenerationTask(
+      Thread* signaling_thread,
+      Thread* worker_thread,
+      const KeyParams& key_params,
+      const Optional<uint64_t>& expires_ms,
+      const scoped_refptr<RTCCertificateGeneratorCallback>& callback)
+      : signaling_thread_(signaling_thread),
+        worker_thread_(worker_thread),
+        key_params_(key_params),
+        expires_ms_(expires_ms),
+        callback_(callback) {
+    RTC_DCHECK(signaling_thread_);
+    RTC_DCHECK(worker_thread_);
+    RTC_DCHECK(callback_);
+  }
+  ~RTCCertificateGenerationTask() override {}
+
+  // Handles |MSG_GENERATE| and its follow-up |MSG_GENERATE_DONE|.
+  void OnMessage(Message* msg) override {
+    switch (msg->message_id) {
+      case MSG_GENERATE:
+        RTC_DCHECK(worker_thread_->IsCurrent());
+
+        // Perform the certificate generation work here on the worker thread.
+        certificate_ = RTCCertificateGenerator::GenerateCertificate(
+            key_params_, expires_ms_);
+
+        // Handle callbacks on signaling thread. Pass on the |msg->pdata|
+        // (which references |this| with ref counting) to that thread.
+        signaling_thread_->Post(this, MSG_GENERATE_DONE, msg->pdata);
+        break;
+      case MSG_GENERATE_DONE:
+        RTC_DCHECK(signaling_thread_->IsCurrent());
+
+        // Perform callback with result here on the signaling thread.
+        if (certificate_) {
+          callback_->OnSuccess(certificate_);
+        } else {
+          callback_->OnFailure();
+        }
+
+        // Destroy |msg->pdata| which references |this| with ref counting. This
+        // may result in |this| being deleted - do not touch member variables
+        // after this line.
+        delete msg->pdata;
+        return;
+      default:
+        RTC_NOTREACHED();
+    }
+  }
+
+ private:
+  Thread* const signaling_thread_;
+  Thread* const worker_thread_;
+  const KeyParams key_params_;
+  const Optional<uint64_t> expires_ms_;
+  const scoped_refptr<RTCCertificateGeneratorCallback> callback_;
+  scoped_refptr<RTCCertificate> certificate_;
+};
+
+}  // namespace
+
+// static
+scoped_refptr<RTCCertificate>
+RTCCertificateGenerator::GenerateCertificate(
+    const KeyParams& key_params,
+    const Optional<uint64_t>& expires_ms) {
+  if (!key_params.IsValid())
+    return nullptr;
+  SSLIdentity* identity;
+  if (!expires_ms) {
+    identity = SSLIdentity::Generate(kIdentityName, key_params);
+  } else {
+    uint64_t expires_s = *expires_ms / 1000;
+    // Limit the expiration time to something reasonable (a year). This was
+    // somewhat arbitrarily chosen. It also ensures that the value is not too
+    // large for the unspecified |time_t|.
+    expires_s = std::min(expires_s, kYearInSeconds);
+    // TODO(torbjorng): Stop using |time_t|, its type is unspecified. It it safe
+    // to assume it can hold up to a year's worth of seconds (and more), but
+    // |SSLIdentity::Generate| should stop relying on |time_t|.
+    // See bugs.webrtc.org/5720.
+    time_t cert_lifetime_s = static_cast<time_t>(expires_s);
+    identity = SSLIdentity::GenerateWithExpiration(
+        kIdentityName, key_params, cert_lifetime_s);
+  }
+  if (!identity)
+    return nullptr;
+  std::unique_ptr<SSLIdentity> identity_sptr(identity);
+  return RTCCertificate::Create(std::move(identity_sptr));
+}
+
+RTCCertificateGenerator::RTCCertificateGenerator(
+    Thread* signaling_thread, Thread* worker_thread)
+    : signaling_thread_(signaling_thread),
+      worker_thread_(worker_thread) {
+  RTC_DCHECK(signaling_thread_);
+  RTC_DCHECK(worker_thread_);
+}
+
+void RTCCertificateGenerator::GenerateCertificateAsync(
+    const KeyParams& key_params,
+    const Optional<uint64_t>& expires_ms,
+    const scoped_refptr<RTCCertificateGeneratorCallback>& callback) {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  RTC_DCHECK(callback);
+
+  // Create a new |RTCCertificateGenerationTask| for this generation request. It
+  // is reference counted and referenced by the message data, ensuring it lives
+  // until the task has completed (independent of |RTCCertificateGenerator|).
+  ScopedRefMessageData<RTCCertificateGenerationTask>* msg_data =
+      new ScopedRefMessageData<RTCCertificateGenerationTask>(
+          new RefCountedObject<RTCCertificateGenerationTask>(
+              signaling_thread_, worker_thread_, key_params, expires_ms,
+              callback));
+  worker_thread_->Post(msg_data->data().get(), MSG_GENERATE, msg_data);
+}
+
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/rtccertificategenerator.h b/chromium/third_party/webrtc/base/rtccertificategenerator.h
new file mode 100644
index 00000000000..08fe67108d4
--- /dev/null
+++ b/chromium/third_party/webrtc/base/rtccertificategenerator.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_
+#define WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_
+
+#include "webrtc/base/optional.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/rtccertificate.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/sslidentity.h"
+#include "webrtc/base/thread.h"
+
+namespace rtc {
+
+class RTCCertificateGeneratorCallback : public RefCountInterface {
+ public:
+  virtual void OnSuccess(
+    const scoped_refptr<RTCCertificate>& certificate) = 0;
+  virtual void OnFailure() = 0;
+
+ protected:
+  ~RTCCertificateGeneratorCallback() override {}
+};
+
+// Generates |RTCCertificate|s.
+// The static function |GenerateCertificate| generates a certificate on the
+// current thread. The |RTCCertificateGenerator| instance generates certificates
+// asynchronously on the worker thread with |GenerateCertificateAsync|.
+class RTCCertificateGenerator {
+ public:
+  // Generates a certificate on the current thread. Returns null on failure.
+  // If |expires_ms| is specified, the certificate will expire in approximately
+  // that many milliseconds from now. |expires_ms| is limited to a year, a
+  // larger value than that is clamped down to a year. If |expires_ms| is not
+  // specified, a default expiration time is used.
+  static scoped_refptr<RTCCertificate> GenerateCertificate(
+      const KeyParams& key_params,
+      const Optional<uint64_t>& expires_ms);
+
+  RTCCertificateGenerator(Thread* signaling_thread, Thread* worker_thread);
+
+  // Generates a certificate asynchronously on the worker thread.
+  // Must be called on the signaling thread. The |callback| is invoked with the
+  // result on the signaling thread. If |expires_ms| is specified, the
+  // certificate will expire in approximately that many milliseconds from now.
+  // |expires_ms| is limited to a year, a larger value than that is clamped down
+  // to a year. If |expires_ms| is not  specified, a default expiration time is
+  // used.
+  void GenerateCertificateAsync(
+      const KeyParams& key_params,
+      const Optional<uint64_t>& expires_ms,
+      const scoped_refptr<RTCCertificateGeneratorCallback>& callback);
+
+ private:
+  Thread* const signaling_thread_;
+  Thread* const worker_thread_;
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_RTCCERTIFICATEGENERATOR_H_
diff --git a/chromium/third_party/webrtc/base/rtccertificategenerator_unittest.cc b/chromium/third_party/webrtc/base/rtccertificategenerator_unittest.cc
new file mode 100644
index 00000000000..a6e88a1a866
--- /dev/null
+++ b/chromium/third_party/webrtc/base/rtccertificategenerator_unittest.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/rtccertificategenerator.h"
+
+#include <memory>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/thread.h"
+
+namespace rtc {
+
+class RTCCertificateGeneratorFixture : public RTCCertificateGeneratorCallback {
+ public:
+  RTCCertificateGeneratorFixture()
+      : signaling_thread_(Thread::Current()),
+        worker_thread_(new Thread()),
+        generate_async_completed_(false) {
+    RTC_CHECK(signaling_thread_);
+    RTC_CHECK(worker_thread_->Start());
+    generator_.reset(
+      new RTCCertificateGenerator(signaling_thread_, worker_thread_.get()));
+  }
+  ~RTCCertificateGeneratorFixture() override {}
+
+  RTCCertificateGenerator* generator() const { return generator_.get(); }
+  RTCCertificate* certificate() const { return certificate_.get(); }
+
+  void OnSuccess(const scoped_refptr<RTCCertificate>& certificate) override {
+    RTC_CHECK(signaling_thread_->IsCurrent());
+    RTC_CHECK(certificate);
+    certificate_ = certificate;
+    generate_async_completed_ = true;
+  }
+  void OnFailure() override {
+    RTC_CHECK(signaling_thread_->IsCurrent());
+    certificate_ = nullptr;
+    generate_async_completed_ = true;
+  }
+
+  bool GenerateAsyncCompleted() {
+    RTC_CHECK(signaling_thread_->IsCurrent());
+    if (generate_async_completed_) {
+      // Reset flag so that future generation requests are not considered done.
+      generate_async_completed_ = false;
+      return true;
+    }
+    return false;
+  }
+
+ protected:
+  Thread* const signaling_thread_;
+  std::unique_ptr<Thread> worker_thread_;
+  std::unique_ptr<RTCCertificateGenerator> generator_;
+  scoped_refptr<RTCCertificate> certificate_;
+  bool generate_async_completed_;
+};
+
+class RTCCertificateGeneratorTest
+    : public testing::Test {
+ public:
+  RTCCertificateGeneratorTest()
+      : fixture_(new RefCountedObject<RTCCertificateGeneratorFixture>()) {}
+  ~RTCCertificateGeneratorTest() {}
+
+ protected:
+  static const int kGenerationTimeoutMs = 1000;
+
+  scoped_refptr<RTCCertificateGeneratorFixture> fixture_;
+};
+
+TEST_F(RTCCertificateGeneratorTest, GenerateECDSA) {
+  EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(
+      KeyParams::ECDSA(),
+      Optional<uint64_t>()));
+}
+
+TEST_F(RTCCertificateGeneratorTest, GenerateRSA) {
+  EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(
+      KeyParams::RSA(),
+      Optional<uint64_t>()));
+}
+
+TEST_F(RTCCertificateGeneratorTest, GenerateAsyncECDSA) {
+  EXPECT_FALSE(fixture_->certificate());
+  fixture_->generator()->GenerateCertificateAsync(
+      KeyParams::ECDSA(),
+      Optional<uint64_t>(),
+      fixture_);
+  // Until generation has completed, the certificate is null. Since this is an
+  // async call, generation must not have completed until we process messages
+  // posted to this thread (which is done by |EXPECT_TRUE_WAIT|).
+  EXPECT_FALSE(fixture_->GenerateAsyncCompleted());
+  EXPECT_FALSE(fixture_->certificate());
+  EXPECT_TRUE_WAIT(fixture_->GenerateAsyncCompleted(), kGenerationTimeoutMs);
+  EXPECT_TRUE(fixture_->certificate());
+}
+
+TEST_F(RTCCertificateGeneratorTest, GenerateWithExpires) {
+  // By generating two certificates with different expiration we can compare the
+  // two expiration times relative to each other without knowing the current
+  // time relative to epoch, 1970-01-01T00:00:00Z. This verifies that the
+  // expiration parameter is correctly used relative to the generator's clock,
+  // but does not verify that this clock is relative to epoch.
+
+  // Generate a certificate that expires immediately.
+  scoped_refptr<RTCCertificate> cert_a =
+      RTCCertificateGenerator::GenerateCertificate(
+          KeyParams::ECDSA(), Optional<uint64_t>(0));
+  EXPECT_TRUE(cert_a);
+
+  // Generate a certificate that expires in one minute.
+  const uint64_t kExpiresMs = 60000;
+  scoped_refptr<RTCCertificate> cert_b =
+      RTCCertificateGenerator::GenerateCertificate(
+          KeyParams::ECDSA(), Optional<uint64_t>(kExpiresMs));
+  EXPECT_TRUE(cert_b);
+
+  // Verify that |cert_b| expires approximately |kExpiresMs| after |cert_a|
+  // (allowing a +/- 1 second plus maximum generation time difference).
+  EXPECT_GT(cert_b->Expires(), cert_a->Expires());
+  uint64_t expires_diff = cert_b->Expires() - cert_a->Expires();
+  EXPECT_GE(expires_diff, kExpiresMs);
+  EXPECT_LE(expires_diff, kExpiresMs + 2*kGenerationTimeoutMs + 1000);
+}
+
+TEST_F(RTCCertificateGeneratorTest, GenerateWithInvalidParamsShouldFail) {
+  KeyParams invalid_params = KeyParams::RSA(0, 0);
+  EXPECT_FALSE(invalid_params.IsValid());
+
+  EXPECT_FALSE(RTCCertificateGenerator::GenerateCertificate(
+      invalid_params, Optional<uint64_t>()));
+
+  fixture_->generator()->GenerateCertificateAsync(
+      invalid_params,
+      Optional<uint64_t>(),
+      fixture_);
+  EXPECT_TRUE_WAIT(fixture_->GenerateAsyncCompleted(), kGenerationTimeoutMs);
+  EXPECT_FALSE(fixture_->certificate());
+}
+
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/scoped_autorelease_pool.h b/chromium/third_party/webrtc/base/scoped_autorelease_pool.h
index 9aac112793e..808231b5d25 100644
--- a/chromium/third_party/webrtc/base/scoped_autorelease_pool.h
+++ b/chromium/third_party/webrtc/base/scoped_autorelease_pool.h
@@ -23,6 +23,7 @@
 #if defined(WEBRTC_MAC)
 
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 
 // This header may be included from Obj-C files or C++ files.
 #ifdef __OBJC__
diff --git a/chromium/third_party/webrtc/base/scoped_ptr.h b/chromium/third_party/webrtc/base/scoped_ptr.h
deleted file mode 100644
index b7e94989e57..00000000000
--- a/chromium/third_party/webrtc/base/scoped_ptr.h
+++ /dev/null
@@ -1,626 +0,0 @@
-/*
- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Borrowed from Chromium's src/base/memory/scoped_ptr.h.
-
-// Scopers help you manage ownership of a pointer, helping you easily manage a
-// pointer within a scope, and automatically destroying the pointer at the end
-// of a scope.  There are two main classes you will use, which correspond to the
-// operators new/delete and new[]/delete[].
-//
-// Example usage (scoped_ptr<T>):
-//   {
-//     scoped_ptr<Foo> foo(new Foo("wee"));
-//   }  // foo goes out of scope, releasing the pointer with it.
-//
-//   {
-//     scoped_ptr<Foo> foo;          // No pointer managed.
-//     foo.reset(new Foo("wee"));    // Now a pointer is managed.
-//     foo.reset(new Foo("wee2"));   // Foo("wee") was destroyed.
-//     foo.reset(new Foo("wee3"));   // Foo("wee2") was destroyed.
-//     foo->Method();                // Foo::Method() called.
-//     foo.get()->Method();          // Foo::Method() called.
-//     SomeFunc(foo.release());      // SomeFunc takes ownership, foo no longer
-//                                   // manages a pointer.
-//     foo.reset(new Foo("wee4"));   // foo manages a pointer again.
-//     foo.reset();                  // Foo("wee4") destroyed, foo no longer
-//                                   // manages a pointer.
-//   }  // foo wasn't managing a pointer, so nothing was destroyed.
-//
-// Example usage (scoped_ptr<T[]>):
-//   {
-//     scoped_ptr<Foo[]> foo(new Foo[100]);
-//     foo.get()->Method();  // Foo::Method on the 0th element.
-//     foo[10].Method();     // Foo::Method on the 10th element.
-//   }
-//
-// These scopers also implement part of the functionality of C++11 unique_ptr
-// in that they are "movable but not copyable." You can use the scopers in the
-// parameter and return types of functions to signify ownership transfer in to
-// and out of a function. When calling a function that has a scoper as the
-// argument type, it must be called with the result of calling std::move on an
-// analogous scoper, or another function that generates a temporary; passing by
-// copy will NOT work. Here is an example using scoped_ptr:
-//
-//   void TakesOwnership(scoped_ptr<Foo> arg) {
-//     // Do something with arg
-//   }
-//   scoped_ptr<Foo> CreateFoo() {
-//     // No need for calling std::move because we are constructing a temporary
-//     // for the return value.
-//     return scoped_ptr<Foo>(new Foo("new"));
-//   }
-//   scoped_ptr<Foo> PassThru(scoped_ptr<Foo> arg) {
-//     return std::move(arg);
-//   }
-//
-//   {
-//     scoped_ptr<Foo> ptr(new Foo("yay"));  // ptr manages Foo("yay").
-//     TakesOwnership(std::move(ptr));       // ptr no longer owns Foo("yay").
-//     scoped_ptr<Foo> ptr2 = CreateFoo();   // ptr2 owns the return Foo.
-//     scoped_ptr<Foo> ptr3 =                // ptr3 now owns what was in ptr2.
-//         PassThru(std::move(ptr2));        // ptr2 is correspondingly nullptr.
-//   }
-//
-// Notice that if you do not call std::move when returning from PassThru(), or
-// when invoking TakesOwnership(), the code will not compile because scopers
-// are not copyable; they only implement move semantics which require calling
-// std::move to signify a destructive transfer of state. CreateFoo() is
-// different though because we are constructing a temporary on the return line
-// and thus can avoid needing to call std::move.
-
-#ifndef WEBRTC_BASE_SCOPED_PTR_H__
-#define WEBRTC_BASE_SCOPED_PTR_H__
-
-// This is an implementation designed to match the anticipated future TR2
-// implementation of the scoped_ptr class.
-
-#include <assert.h>
-#include <stddef.h>
-#include <stdlib.h>
-
-#include <algorithm>  // For std::swap().
-#include <cstddef>
-#include <memory>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/template_util.h"
-#include "webrtc/typedefs.h"
-
-namespace rtc {
-
-// Function object which deletes its parameter, which must be a pointer.
-// If C is an array type, invokes 'delete[]' on the parameter; otherwise,
-// invokes 'delete'. The default deleter for scoped_ptr<T>.
-template <class T>
-struct DefaultDeleter {
-  DefaultDeleter() {}
-  template <typename U> DefaultDeleter(const DefaultDeleter<U>& other) {
-    // IMPLEMENTATION NOTE: C++11 20.7.1.1.2p2 only provides this constructor
-    // if U* is implicitly convertible to T* and U is not an array type.
-    //
-    // Correct implementation should use SFINAE to disable this
-    // constructor. However, since there are no other 1-argument constructors,
-    // using a static_assert based on is_convertible<> and requiring
-    // complete types is simpler and will cause compile failures for equivalent
-    // misuses.
-    //
-    // Note, the is_convertible<U*, T*> check also ensures that U is not an
-    // array. T is guaranteed to be a non-array, so any U* where U is an array
-    // cannot convert to T*.
-    enum { T_must_be_complete = sizeof(T) };
-    enum { U_must_be_complete = sizeof(U) };
-    static_assert(rtc::is_convertible<U*, T*>::value,
-                  "U* must implicitly convert to T*");
-  }
-  inline void operator()(T* ptr) const {
-    enum { type_must_be_complete = sizeof(T) };
-    delete ptr;
-  }
-};
-
-// Specialization of DefaultDeleter for array types.
-template <class T>
-struct DefaultDeleter<T[]> {
-  inline void operator()(T* ptr) const {
-    enum { type_must_be_complete = sizeof(T) };
-    delete[] ptr;
-  }
-
- private:
-  // Disable this operator for any U != T because it is undefined to execute
-  // an array delete when the static type of the array mismatches the dynamic
-  // type.
-  //
-  // References:
-  //   C++98 [expr.delete]p3
-  //   http://cplusplus.github.com/LWG/lwg-defects.html#938
-  template <typename U> void operator()(U* array) const;
-};
-
-template <class T, int n>
-struct DefaultDeleter<T[n]> {
-  // Never allow someone to declare something like scoped_ptr<int[10]>.
-  static_assert(sizeof(T) == -1, "do not use array with size as type");
-};
-
-// Function object which invokes 'free' on its parameter, which must be
-// a pointer. Can be used to store malloc-allocated pointers in scoped_ptr:
-//
-// scoped_ptr<int, rtc::FreeDeleter> foo_ptr(
-//     static_cast<int*>(malloc(sizeof(int))));
-struct FreeDeleter {
-  inline void operator()(void* ptr) const {
-    free(ptr);
-  }
-};
-
-namespace internal {
-
-template <typename T>
-struct ShouldAbortOnSelfReset {
-  template <typename U>
-  static rtc::internal::NoType Test(const typename U::AllowSelfReset*);
-
-  template <typename U>
-  static rtc::internal::YesType Test(...);
-
-  static const bool value =
-      sizeof(Test<T>(0)) == sizeof(rtc::internal::YesType);
-};
-
-// Minimal implementation of the core logic of scoped_ptr, suitable for
-// reuse in both scoped_ptr and its specializations.
-template <class T, class D>
-class scoped_ptr_impl {
- public:
-  explicit scoped_ptr_impl(T* p) : data_(p) {}
-
-  // Initializer for deleters that have data parameters.
-  scoped_ptr_impl(T* p, const D& d) : data_(p, d) {}
-
-  // Templated constructor that destructively takes the value from another
-  // scoped_ptr_impl.
-  template <typename U, typename V>
-  scoped_ptr_impl(scoped_ptr_impl<U, V>* other)
-      : data_(other->release(), other->get_deleter()) {
-    // We do not support move-only deleters.  We could modify our move
-    // emulation to have rtc::subtle::move() and rtc::subtle::forward()
-    // functions that are imperfect emulations of their C++11 equivalents,
-    // but until there's a requirement, just assume deleters are copyable.
-  }
-
-  template <typename U, typename V>
-  void TakeState(scoped_ptr_impl<U, V>* other) {
-    // See comment in templated constructor above regarding lack of support
-    // for move-only deleters.
-    reset(other->release());
-    get_deleter() = other->get_deleter();
-  }
-
-  ~scoped_ptr_impl() {
-    if (data_.ptr != nullptr) {
-      // Not using get_deleter() saves one function call in non-optimized
-      // builds.
-      static_cast<D&>(data_)(data_.ptr);
-    }
-  }
-
-  void reset(T* p) {
-    // This is a self-reset, which is no longer allowed for default deleters:
-    // https://crbug.com/162971
-    assert(!ShouldAbortOnSelfReset<D>::value || p == nullptr || p != data_.ptr);
-
-    // Note that running data_.ptr = p can lead to undefined behavior if
-    // get_deleter()(get()) deletes this. In order to prevent this, reset()
-    // should update the stored pointer before deleting its old value.
-    //
-    // However, changing reset() to use that behavior may cause current code to
-    // break in unexpected ways. If the destruction of the owned object
-    // dereferences the scoped_ptr when it is destroyed by a call to reset(),
-    // then it will incorrectly dispatch calls to |p| rather than the original
-    // value of |data_.ptr|.
-    //
-    // During the transition period, set the stored pointer to nullptr while
-    // deleting the object. Eventually, this safety check will be removed to
-    // prevent the scenario initially described from occurring and
-    // http://crbug.com/176091 can be closed.
-    T* old = data_.ptr;
-    data_.ptr = nullptr;
-    if (old != nullptr)
-      static_cast<D&>(data_)(old);
-    data_.ptr = p;
-  }
-
-  T* get() const { return data_.ptr; }
-
-  D& get_deleter() { return data_; }
-  const D& get_deleter() const { return data_; }
-
-  void swap(scoped_ptr_impl& p2) {
-    // Standard swap idiom: 'using std::swap' ensures that std::swap is
-    // present in the overload set, but we call swap unqualified so that
-    // any more-specific overloads can be used, if available.
-    using std::swap;
-    swap(static_cast<D&>(data_), static_cast<D&>(p2.data_));
-    swap(data_.ptr, p2.data_.ptr);
-  }
-
-  T* release() {
-    T* old_ptr = data_.ptr;
-    data_.ptr = nullptr;
-    return old_ptr;
-  }
-
-  T** accept() {
-    reset(nullptr);
-    return &(data_.ptr);
-  }
-
-  T** use() {
-    return &(data_.ptr);
-  }
-
- private:
-  // Needed to allow type-converting constructor.
-  template <typename U, typename V> friend class scoped_ptr_impl;
-
-  // Use the empty base class optimization to allow us to have a D
-  // member, while avoiding any space overhead for it when D is an
-  // empty class.  See e.g. http://www.cantrip.org/emptyopt.html for a good
-  // discussion of this technique.
-  struct Data : public D {
-    explicit Data(T* ptr_in) : ptr(ptr_in) {}
-    Data(T* ptr_in, const D& other) : D(other), ptr(ptr_in) {}
-    T* ptr;
-  };
-
-  Data data_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(scoped_ptr_impl);
-};
-
-}  // namespace internal
-
-// A scoped_ptr<T> is like a T*, except that the destructor of scoped_ptr<T>
-// automatically deletes the pointer it holds (if any).
-// That is, scoped_ptr<T> owns the T object that it points to.
-// Like a T*, a scoped_ptr<T> may hold either nullptr or a pointer to a T
-// object. Also like T*, scoped_ptr<T> is thread-compatible, and once you
-// dereference it, you get the thread safety guarantees of T.
-//
-// The size of scoped_ptr is small. On most compilers, when using the
-// DefaultDeleter, sizeof(scoped_ptr<T>) == sizeof(T*). Custom deleters will
-// increase the size proportional to whatever state they need to have. See
-// comments inside scoped_ptr_impl<> for details.
-//
-// Current implementation targets having a strict subset of  C++11's
-// unique_ptr<> features. Known deficiencies include not supporting move-only
-// deleters, function pointers as deleters, and deleters with reference
-// types.
-template <class T, class D = rtc::DefaultDeleter<T> >
-class scoped_ptr {
-
-  // TODO(ajm): If we ever import RefCountedBase, this check needs to be
-  // enabled.
-  //static_assert(rtc::internal::IsNotRefCounted<T>::value,
-  //              "T is refcounted type and needs scoped refptr");
-
- public:
-  // The element and deleter types.
-  typedef T element_type;
-  typedef D deleter_type;
-
-  // Constructor.  Defaults to initializing with nullptr.
-  scoped_ptr() : impl_(nullptr) {}
-
-  // Constructor.  Takes ownership of p.
-  explicit scoped_ptr(element_type* p) : impl_(p) {}
-
-  // Constructor.  Allows initialization of a stateful deleter.
-  scoped_ptr(element_type* p, const D& d) : impl_(p, d) {}
-
-  // Constructor.  Allows construction from a nullptr.
-  scoped_ptr(std::nullptr_t) : impl_(nullptr) {}
-
-  // Constructor.  Allows construction from a scoped_ptr rvalue for a
-  // convertible type and deleter.
-  //
-  // IMPLEMENTATION NOTE: C++11 unique_ptr<> keeps this constructor distinct
-  // from the normal move constructor. By C++11 20.7.1.2.1.21, this constructor
-  // has different post-conditions if D is a reference type. Since this
-  // implementation does not support deleters with reference type,
-  // we do not need a separate move constructor allowing us to avoid one
-  // use of SFINAE. You only need to care about this if you modify the
-  // implementation of scoped_ptr.
-  template <typename U, typename V>
-  scoped_ptr(scoped_ptr<U, V>&& other)
-      : impl_(&other.impl_) {
-    static_assert(!rtc::is_array<U>::value, "U cannot be an array");
-  }
-
-  // operator=.  Allows assignment from a scoped_ptr rvalue for a convertible
-  // type and deleter.
-  //
-  // IMPLEMENTATION NOTE: C++11 unique_ptr<> keeps this operator= distinct from
-  // the normal move assignment operator. By C++11 20.7.1.2.3.4, this templated
-  // form has different requirements on for move-only Deleters. Since this
-  // implementation does not support move-only Deleters, we do not need a
-  // separate move assignment operator allowing us to avoid one use of SFINAE.
-  // You only need to care about this if you modify the implementation of
-  // scoped_ptr.
-  template <typename U, typename V>
-  scoped_ptr& operator=(scoped_ptr<U, V>&& rhs) {
-    static_assert(!rtc::is_array<U>::value, "U cannot be an array");
-    impl_.TakeState(&rhs.impl_);
-    return *this;
-  }
-
-  // operator=.  Allows assignment from a nullptr. Deletes the currently owned
-  // object, if any.
-  scoped_ptr& operator=(std::nullptr_t) {
-    reset();
-    return *this;
-  }
-
-  // Deleted copy constructor and copy assignment, to make the type move-only.
-  scoped_ptr(const scoped_ptr& other) = delete;
-  scoped_ptr& operator=(const scoped_ptr& other) = delete;
-
-  // Reset.  Deletes the currently owned object, if any.
-  // Then takes ownership of a new object, if given.
-  void reset(element_type* p = nullptr) { impl_.reset(p); }
-
-  // Accessors to get the owned object.
-  // operator* and operator-> will assert() if there is no current object.
-  element_type& operator*() const {
-    assert(impl_.get() != nullptr);
-    return *impl_.get();
-  }
-  element_type* operator->() const  {
-    assert(impl_.get() != nullptr);
-    return impl_.get();
-  }
-  element_type* get() const { return impl_.get(); }
-
-  // Access to the deleter.
-  deleter_type& get_deleter() { return impl_.get_deleter(); }
-  const deleter_type& get_deleter() const { return impl_.get_deleter(); }
-
-  // Allow scoped_ptr<element_type> to be used in boolean expressions, but not
-  // implicitly convertible to a real bool (which is dangerous).
-  //
-  // Note that this trick is only safe when the == and != operators
-  // are declared explicitly, as otherwise "scoped_ptr1 ==
-  // scoped_ptr2" will compile but do the wrong thing (i.e., convert
-  // to Testable and then do the comparison).
- private:
-  typedef rtc::internal::scoped_ptr_impl<element_type, deleter_type>
-      scoped_ptr::*Testable;
-
- public:
-  operator Testable() const {
-    return impl_.get() ? &scoped_ptr::impl_ : nullptr;
-  }
-
-  // Comparison operators.
-  // These return whether two scoped_ptr refer to the same object, not just to
-  // two different but equal objects.
-  bool operator==(const element_type* p) const { return impl_.get() == p; }
-  bool operator!=(const element_type* p) const { return impl_.get() != p; }
-
-  // Swap two scoped pointers.
-  void swap(scoped_ptr& p2) {
-    impl_.swap(p2.impl_);
-  }
-
-  // Release a pointer.
-  // The return value is the current pointer held by this object. If this object
-  // holds a nullptr, the return value is nullptr. After this operation, this
-  // object will hold a nullptr, and will not own the object any more.
-  element_type* release() WARN_UNUSED_RESULT {
-    return impl_.release();
-  }
-
-  // Delete the currently held pointer and return a pointer
-  // to allow overwriting of the current pointer address.
-  element_type** accept() WARN_UNUSED_RESULT {
-    return impl_.accept();
-  }
-
-  // Return a pointer to the current pointer address.
-  element_type** use() WARN_UNUSED_RESULT {
-    return impl_.use();
-  }
-
- private:
-  // Needed to reach into |impl_| in the constructor.
-  template <typename U, typename V> friend class scoped_ptr;
-  rtc::internal::scoped_ptr_impl<element_type, deleter_type> impl_;
-
-  // Forbidden for API compatibility with std::unique_ptr.
-  explicit scoped_ptr(int disallow_construction_from_null);
-
-  // Forbid comparison of scoped_ptr types.  If U != T, it totally
-  // doesn't make sense, and if U == T, it still doesn't make sense
-  // because you should never have the same object owned by two different
-  // scoped_ptrs.
-  template <class U> bool operator==(scoped_ptr<U> const& p2) const;
-  template <class U> bool operator!=(scoped_ptr<U> const& p2) const;
-};
-
-template <class T, class D>
-class scoped_ptr<T[], D> {
- public:
-  // The element and deleter types.
-  typedef T element_type;
-  typedef D deleter_type;
-
-  // Constructor.  Defaults to initializing with nullptr.
-  scoped_ptr() : impl_(nullptr) {}
-
-  // Constructor. Stores the given array. Note that the argument's type
-  // must exactly match T*. In particular:
-  // - it cannot be a pointer to a type derived from T, because it is
-  //   inherently unsafe in the general case to access an array through a
-  //   pointer whose dynamic type does not match its static type (eg., if
-  //   T and the derived types had different sizes access would be
-  //   incorrectly calculated). Deletion is also always undefined
-  //   (C++98 [expr.delete]p3). If you're doing this, fix your code.
-  // - it cannot be const-qualified differently from T per unique_ptr spec
-  //   (http://cplusplus.github.com/LWG/lwg-active.html#2118). Users wanting
-  //   to work around this may use implicit_cast<const T*>().
-  //   However, because of the first bullet in this comment, users MUST
-  //   NOT use implicit_cast<Base*>() to upcast the static type of the array.
-  explicit scoped_ptr(element_type* array) : impl_(array) {}
-
-  // Constructor.  Allows construction from a nullptr.
-  scoped_ptr(std::nullptr_t) : impl_(nullptr) {}
-
-  // Constructor.  Allows construction from a scoped_ptr rvalue.
-  scoped_ptr(scoped_ptr&& other) : impl_(&other.impl_) {}
-
-  // operator=.  Allows assignment from a scoped_ptr rvalue.
-  scoped_ptr& operator=(scoped_ptr&& rhs) {
-    impl_.TakeState(&rhs.impl_);
-    return *this;
-  }
-
-  // operator=.  Allows assignment from a nullptr. Deletes the currently owned
-  // array, if any.
-  scoped_ptr& operator=(std::nullptr_t) {
-    reset();
-    return *this;
-  }
-
-  // Deleted copy constructor and copy assignment, to make the type move-only.
-  scoped_ptr(const scoped_ptr& other) = delete;
-  scoped_ptr& operator=(const scoped_ptr& other) = delete;
-
-  // Reset.  Deletes the currently owned array, if any.
-  // Then takes ownership of a new object, if given.
-  void reset(element_type* array = nullptr) { impl_.reset(array); }
-
-  // Accessors to get the owned array.
-  element_type& operator[](size_t i) const {
-    assert(impl_.get() != nullptr);
-    return impl_.get()[i];
-  }
-  element_type* get() const { return impl_.get(); }
-
-  // Access to the deleter.
-  deleter_type& get_deleter() { return impl_.get_deleter(); }
-  const deleter_type& get_deleter() const { return impl_.get_deleter(); }
-
-  // Allow scoped_ptr<element_type> to be used in boolean expressions, but not
-  // implicitly convertible to a real bool (which is dangerous).
- private:
-  typedef rtc::internal::scoped_ptr_impl<element_type, deleter_type>
-      scoped_ptr::*Testable;
-
- public:
-  operator Testable() const {
-    return impl_.get() ? &scoped_ptr::impl_ : nullptr;
-  }
-
-  // Comparison operators.
-  // These return whether two scoped_ptr refer to the same object, not just to
-  // two different but equal objects.
-  bool operator==(element_type* array) const { return impl_.get() == array; }
-  bool operator!=(element_type* array) const { return impl_.get() != array; }
-
-  // Swap two scoped pointers.
-  void swap(scoped_ptr& p2) {
-    impl_.swap(p2.impl_);
-  }
-
-  // Release a pointer.
-  // The return value is the current pointer held by this object. If this object
-  // holds a nullptr, the return value is nullptr. After this operation, this
-  // object will hold a nullptr, and will not own the object any more.
-  element_type* release() WARN_UNUSED_RESULT {
-    return impl_.release();
-  }
-
-  // Delete the currently held pointer and return a pointer
-  // to allow overwriting of the current pointer address.
-  element_type** accept() WARN_UNUSED_RESULT {
-    return impl_.accept();
-  }
-
-  // Return a pointer to the current pointer address.
-  element_type** use() WARN_UNUSED_RESULT {
-    return impl_.use();
-  }
-
- private:
-  // Force element_type to be a complete type.
-  enum { type_must_be_complete = sizeof(element_type) };
-
-  // Actually hold the data.
-  rtc::internal::scoped_ptr_impl<element_type, deleter_type> impl_;
-
-  // Disable initialization from any type other than element_type*, by
-  // providing a constructor that matches such an initialization, but is
-  // private and has no definition. This is disabled because it is not safe to
-  // call delete[] on an array whose static type does not match its dynamic
-  // type.
-  template <typename U> explicit scoped_ptr(U* array);
-  explicit scoped_ptr(int disallow_construction_from_null);
-
-  // Disable reset() from any type other than element_type*, for the same
-  // reasons as the constructor above.
-  template <typename U> void reset(U* array);
-  void reset(int disallow_reset_from_null);
-
-  // Forbid comparison of scoped_ptr types.  If U != T, it totally
-  // doesn't make sense, and if U == T, it still doesn't make sense
-  // because you should never have the same object owned by two different
-  // scoped_ptrs.
-  template <class U> bool operator==(scoped_ptr<U> const& p2) const;
-  template <class U> bool operator!=(scoped_ptr<U> const& p2) const;
-};
-
-template <class T, class D>
-void swap(rtc::scoped_ptr<T, D>& p1, rtc::scoped_ptr<T, D>& p2) {
-  p1.swap(p2);
-}
-
-// Convert between the most common kinds of scoped_ptr and unique_ptr.
-template <typename T>
-std::unique_ptr<T> ScopedToUnique(scoped_ptr<T> sp) {
-  return std::unique_ptr<T>(sp.release());
-}
-template <typename T>
-scoped_ptr<T> UniqueToScoped(std::unique_ptr<T> up) {
-  return scoped_ptr<T>(up.release());
-}
-
-}  // namespace rtc
-
-template <class T, class D>
-bool operator==(T* p1, const rtc::scoped_ptr<T, D>& p2) {
-  return p1 == p2.get();
-}
-
-template <class T, class D>
-bool operator!=(T* p1, const rtc::scoped_ptr<T, D>& p2) {
-  return p1 != p2.get();
-}
-
-// A function to convert T* into scoped_ptr<T>
-// Doing e.g. make_scoped_ptr(new FooBarBaz<type>(arg)) is a shorter notation
-// for scoped_ptr<FooBarBaz<type> >(new FooBarBaz<type>(arg))
-template <typename T>
-rtc::scoped_ptr<T> rtc_make_scoped_ptr(T* ptr) {
-  return rtc::scoped_ptr<T>(ptr);
-}
-
-#endif  // #ifndef WEBRTC_BASE_SCOPED_PTR_H__
diff --git a/chromium/third_party/webrtc/base/scopedptrcollection_unittest.cc b/chromium/third_party/webrtc/base/scopedptrcollection_unittest.cc
index 933173e3faf..929fd0f324c 100644
--- a/chromium/third_party/webrtc/base/scopedptrcollection_unittest.cc
+++ b/chromium/third_party/webrtc/base/scopedptrcollection_unittest.cc
@@ -8,6 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scopedptrcollection.h"
 #include "webrtc/base/gunit.h"
 
@@ -41,7 +44,7 @@ class ScopedPtrCollectionTest : public testing::Test {
   }
 
   int num_instances_;
-  scoped_ptr<ScopedPtrCollection<InstanceCounter> > collection_;
+  std::unique_ptr<ScopedPtrCollection<InstanceCounter> > collection_;
 };
 
 TEST_F(ScopedPtrCollectionTest, PushBack) {
diff --git a/chromium/third_party/webrtc/base/sharedexclusivelock_unittest.cc b/chromium/third_party/webrtc/base/sharedexclusivelock_unittest.cc
index 9b64ed760af..45902af39e6 100644
--- a/chromium/third_party/webrtc/base/sharedexclusivelock_unittest.cc
+++ b/chromium/third_party/webrtc/base/sharedexclusivelock_unittest.cc
@@ -8,15 +8,26 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/common.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/messagequeue.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sharedexclusivelock.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/timeutils.h"
 
+#if defined(MEMORY_SANITIZER)
+// Flaky under MemorySanitizer, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5824
+#define MAYBE_TestSharedExclusive DISABLED_TestSharedExclusive
+#define MAYBE_TestExclusiveExclusive DISABLED_TestExclusiveExclusive
+#else
+#define MAYBE_TestSharedExclusive TestSharedExclusive
+#define MAYBE_TestExclusiveExclusive TestExclusiveExclusive
+#endif
+
 namespace rtc {
 
 static const uint32_t kMsgRead = 0;
@@ -39,12 +50,12 @@ class SharedExclusiveTask : public MessageHandler {
     worker_thread_->Start();
   }
 
-  int waiting_time_in_ms() const { return waiting_time_in_ms_; }
+  int64_t waiting_time_in_ms() const { return waiting_time_in_ms_; }
 
  protected:
-  scoped_ptr<Thread> worker_thread_;
+  std::unique_ptr<Thread> worker_thread_;
   SharedExclusiveLock* shared_exclusive_lock_;
-  int waiting_time_in_ms_;
+  int64_t waiting_time_in_ms_;
   int* value_;
   bool* done_;
 };
@@ -68,10 +79,10 @@ class ReadTask : public SharedExclusiveTask {
     TypedMessageData<int*>* message_data =
         static_cast<TypedMessageData<int*>*>(message->pdata);
 
-    uint32_t start_time = Time();
+    int64_t start_time = TimeMillis();
     {
       SharedScope ss(shared_exclusive_lock_);
-      waiting_time_in_ms_ = TimeDiff(Time(), start_time);
+      waiting_time_in_ms_ = TimeDiff(TimeMillis(), start_time);
 
       Thread::SleepMs(kProcessTimeInMs);
       *message_data->data() = *value_;
@@ -101,10 +112,10 @@ class WriteTask : public SharedExclusiveTask {
     TypedMessageData<int>* message_data =
         static_cast<TypedMessageData<int>*>(message->pdata);
 
-    uint32_t start_time = Time();
+    int64_t start_time = TimeMillis();
     {
       ExclusiveScope es(shared_exclusive_lock_);
-      waiting_time_in_ms_ = TimeDiff(Time(), start_time);
+      waiting_time_in_ms_ = TimeDiff(TimeMillis(), start_time);
 
       Thread::SleepMs(kProcessTimeInMs);
       *value_ = message_data->data();
@@ -127,7 +138,7 @@ class SharedExclusiveLockTest
   }
 
  protected:
-  scoped_ptr<SharedExclusiveLock> shared_exclusive_lock_;
+  std::unique_ptr<SharedExclusiveLock> shared_exclusive_lock_;
   int value_;
 };
 
@@ -157,7 +168,7 @@ TEST_F(SharedExclusiveLockTest, TestSharedShared) {
   EXPECT_LE(reader1.waiting_time_in_ms(), kNoWaitThresholdInMs);
 }
 
-TEST_F(SharedExclusiveLockTest, TestSharedExclusive) {
+TEST_F(SharedExclusiveLockTest, MAYBE_TestSharedExclusive) {
   bool done;
   WriteTask writer(shared_exclusive_lock_.get(), &value_, &done);
 
@@ -196,7 +207,7 @@ TEST_F(SharedExclusiveLockTest, TestExclusiveShared) {
   EXPECT_GE(reader.waiting_time_in_ms(), kWaitThresholdInMs);
 }
 
-TEST_F(SharedExclusiveLockTest, TestExclusiveExclusive) {
+TEST_F(SharedExclusiveLockTest, MAYBE_TestExclusiveExclusive) {
   bool done;
   WriteTask writer(shared_exclusive_lock_.get(), &value_, &done);
 
diff --git a/chromium/third_party/webrtc/base/signalthread_unittest.cc b/chromium/third_party/webrtc/base/signalthread_unittest.cc
index a583aefcb5c..57190128c53 100644
--- a/chromium/third_party/webrtc/base/signalthread_unittest.cc
+++ b/chromium/third_party/webrtc/base/signalthread_unittest.cc
@@ -8,6 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/signalthread.h"
 #include "webrtc/base/thread.h"
@@ -135,7 +138,7 @@ class OwnerThread : public Thread, public sigslot::has_slots<> {
 // when shutting down the process.
 TEST_F(SignalThreadTest, OwnerThreadGoesAway) {
   {
-    scoped_ptr<OwnerThread> owner(new OwnerThread(this));
+    std::unique_ptr<OwnerThread> owner(new OwnerThread(this));
     main_thread_ = owner.get();
     owner->Start();
     while (!owner->has_run()) {
diff --git a/chromium/third_party/webrtc/base/socket_unittest.cc b/chromium/third_party/webrtc/base/socket_unittest.cc
index d1369e2f78c..5f6de42a2f1 100644
--- a/chromium/third_party/webrtc/base/socket_unittest.cc
+++ b/chromium/third_party/webrtc/base/socket_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/socket_unittest.h"
 
 #include "webrtc/base/arraysize.h"
@@ -198,14 +200,14 @@ void SocketTest::ConnectInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(ss_->CreateAsyncSocket(loopback.family(),
-                                                        SOCK_STREAM));
+  std::unique_ptr<AsyncSocket> client(
+      ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
   EXPECT_EQ(AsyncSocket::CS_CLOSED, client->GetState());
   EXPECT_PRED1(IsUnspecOrEmptyIP, client->GetLocalAddress().ipaddr());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -229,7 +231,7 @@ void SocketTest::ConnectInternal(const IPAddress& loopback) {
 
   // Server has pending connection, accept it.
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   EXPECT_FALSE(accept_addr.IsNil());
   EXPECT_EQ(accepted->GetRemoteAddress(), accept_addr);
@@ -253,12 +255,12 @@ void SocketTest::ConnectWithDnsLookupInternal(const IPAddress& loopback,
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -278,7 +280,7 @@ void SocketTest::ConnectWithDnsLookupInternal(const IPAddress& loopback,
 
   // Server has pending connection, accept it.
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   EXPECT_FALSE(accept_addr.IsNil());
   EXPECT_EQ(accepted->GetRemoteAddress(), accept_addr);
@@ -301,12 +303,12 @@ void SocketTest::ConnectFailInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
 
   // Create server, but don't listen yet.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -334,12 +336,12 @@ void SocketTest::ConnectWithDnsLookupFailInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
 
   // Create server, but don't listen yet.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -372,13 +374,13 @@ void SocketTest::ConnectWithDnsLookupFailInternal(const IPAddress& loopback) {
 
 void SocketTest::ConnectWithClosedSocketInternal(const IPAddress& loopback) {
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
   EXPECT_EQ(0, server->Listen(5));
 
   // Create a client and put in to CS_CLOSED state.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   EXPECT_EQ(0, client->Close());
   EXPECT_EQ(AsyncSocket::CS_CLOSED, client->GetState());
@@ -391,13 +393,13 @@ void SocketTest::ConnectWithClosedSocketInternal(const IPAddress& loopback) {
 void SocketTest::ConnectWhileNotClosedInternal(const IPAddress& loopback) {
   // Create server and listen.
   testing::StreamSink sink;
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
   EXPECT_EQ(0, server->Listen(5));
   // Create client, connect.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   EXPECT_EQ(0, client->Connect(SocketAddress(server->GetLocalAddress())));
   EXPECT_EQ(AsyncSocket::CS_CONNECTING, client->GetState());
@@ -408,7 +410,7 @@ void SocketTest::ConnectWhileNotClosedInternal(const IPAddress& loopback) {
   // Accept the original connection.
   SocketAddress accept_addr;
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   EXPECT_FALSE(accept_addr.IsNil());
 
@@ -435,12 +437,12 @@ void SocketTest::ServerCloseDuringConnectInternal(const IPAddress& loopback) {
   testing::StreamSink sink;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -464,12 +466,12 @@ void SocketTest::ClientCloseDuringConnectInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -483,7 +485,7 @@ void SocketTest::ClientCloseDuringConnectInternal(const IPAddress& loopback) {
   client->Close();
 
   // The connection should still be able to be accepted.
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   sink.Monitor(accepted.get());
   EXPECT_EQ(AsyncSocket::CS_CONNECTED, accepted->GetState());
@@ -502,12 +504,12 @@ void SocketTest::ServerCloseInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -518,7 +520,7 @@ void SocketTest::ServerCloseInternal(const IPAddress& loopback) {
 
   // Accept connection.
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   sink.Monitor(accepted.get());
 
@@ -576,13 +578,13 @@ void SocketTest::CloseInClosedCallbackInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
   client->SignalCloseEvent.connect(&closer, &SocketCloser::OnClose);
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -593,7 +595,7 @@ void SocketTest::CloseInClosedCallbackInternal(const IPAddress& loopback) {
 
   // Accept connection.
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   sink.Monitor(accepted.get());
 
@@ -630,9 +632,9 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create & connect server and client sockets.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
   sink.Monitor(server.get());
@@ -642,7 +644,7 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) {
   EXPECT_EQ(0, client->Connect(server->GetLocalAddress()));
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
 
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   sink.Monitor(accepted.get());
   EXPECT_EQ(AsyncSocket::CS_CONNECTED, accepted->GetState());
@@ -663,7 +665,7 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) {
   EXPECT_FALSE(sink.Check(accepted.get(), testing::SSE_READ));
 
   // Shouldn't signal when blocked in a thread Send, where process_io is false.
-  scoped_ptr<Thread> thread(new Thread());
+  std::unique_ptr<Thread> thread(new Thread());
   thread->Start();
   Sleeper sleeper;
   TypedMessageData<AsyncSocket*> data(client.get());
@@ -681,12 +683,12 @@ void SocketTest::TcpInternal(const IPAddress& loopback, size_t data_size,
   SocketAddress accept_addr;
 
   // Create receiving client.
-  scoped_ptr<AsyncSocket> receiver(
+  std::unique_ptr<AsyncSocket> receiver(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(receiver.get());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -697,7 +699,7 @@ void SocketTest::TcpInternal(const IPAddress& loopback, size_t data_size,
 
   // Accept connection which will be used for sending.
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
-  scoped_ptr<AsyncSocket> sender(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> sender(server->Accept(&accept_addr));
   ASSERT_TRUE(sender);
   sink.Monitor(sender.get());
 
@@ -809,12 +811,12 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) {
   SocketAddress accept_addr;
 
   // Create client.
-  scoped_ptr<AsyncSocket> client(
+  std::unique_ptr<AsyncSocket> client(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(client.get());
 
   // Create server and listen.
-  scoped_ptr<AsyncSocket> server(
+  std::unique_ptr<AsyncSocket> server(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
   sink.Monitor(server.get());
   EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
@@ -825,7 +827,7 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) {
 
   // Accept connection.
   EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
-  scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+  std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
   ASSERT_TRUE(accepted);
   sink.Monitor(accepted.get());
 
@@ -887,9 +889,9 @@ void SocketTest::UdpInternal(const IPAddress& loopback) {
   delete socket;
 
   // Test send/receive behavior.
-  scoped_ptr<TestClient> client1(
+  std::unique_ptr<TestClient> client1(
       new TestClient(AsyncUDPSocket::Create(ss_, addr1)));
-  scoped_ptr<TestClient> client2(
+  std::unique_ptr<TestClient> client2(
       new TestClient(AsyncUDPSocket::Create(ss_, empty)));
 
   SocketAddress addr2;
@@ -928,10 +930,10 @@ void SocketTest::UdpReadyToSend(const IPAddress& loopback) {
   SocketAddress test_addr(dest, 2345);
 
   // Test send
-  scoped_ptr<TestClient> client(
+  std::unique_ptr<TestClient> client(
       new TestClient(AsyncUDPSocket::Create(ss_, empty)));
   int test_packet_size = 1200;
-  rtc::scoped_ptr<char[]> test_packet(new char[test_packet_size]);
+  std::unique_ptr<char[]> test_packet(new char[test_packet_size]);
   // Init the test packet just to avoid memcheck warning.
   memset(test_packet.get(), 0, test_packet_size);
   // Set the send buffer size to the same size as the test packet to have a
@@ -965,7 +967,7 @@ void SocketTest::UdpReadyToSend(const IPAddress& loopback) {
 }
 
 void SocketTest::GetSetOptionsInternal(const IPAddress& loopback) {
-  rtc::scoped_ptr<AsyncSocket> socket(
+  std::unique_ptr<AsyncSocket> socket(
       ss_->CreateAsyncSocket(loopback.family(), SOCK_DGRAM));
   socket->Bind(SocketAddress(loopback, 0));
 
@@ -1000,9 +1002,8 @@ void SocketTest::GetSetOptionsInternal(const IPAddress& loopback) {
   // Skip the esimate MTU test for IPv6 for now.
   if (loopback.family() != AF_INET6) {
     // Try estimating MTU.
-    rtc::scoped_ptr<AsyncSocket>
-        mtu_socket(
-            ss_->CreateAsyncSocket(loopback.family(), SOCK_DGRAM));
+    std::unique_ptr<AsyncSocket> mtu_socket(
+        ss_->CreateAsyncSocket(loopback.family(), SOCK_DGRAM));
     mtu_socket->Bind(SocketAddress(loopback, 0));
     uint16_t mtu;
     // should fail until we connect
diff --git a/chromium/third_party/webrtc/base/socketadapters.h b/chromium/third_party/webrtc/base/socketadapters.h
index ae0ed9ac392..970a3b5eb6c 100644
--- a/chromium/third_party/webrtc/base/socketadapters.h
+++ b/chromium/third_party/webrtc/base/socketadapters.h
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "webrtc/base/asyncsocket.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/cryptstring.h"
 #include "webrtc/base/logging.h"
 
diff --git a/chromium/third_party/webrtc/base/socketserver.h b/chromium/third_party/webrtc/base/socketserver.h
index 1b4136d74e3..7071f225268 100644
--- a/chromium/third_party/webrtc/base/socketserver.h
+++ b/chromium/third_party/webrtc/base/socketserver.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_BASE_SOCKETSERVER_H_
 #define WEBRTC_BASE_SOCKETSERVER_H_
 
+#include <memory>
 #include "webrtc/base/socketfactory.h"
 
 namespace rtc {
@@ -27,6 +28,7 @@ class SocketServer : public SocketFactory {
  public:
   static const int kForever = -1;
 
+  static std::unique_ptr<SocketServer> CreateDefault();
   // When the socket server is installed into a Thread, this function is
   // called to allow the socket server to use the thread's message queue for
   // any messaging that it might need to perform.
diff --git a/chromium/third_party/webrtc/base/socketstream.h b/chromium/third_party/webrtc/base/socketstream.h
index fd8b559007b..c64656345d2 100644
--- a/chromium/third_party/webrtc/base/socketstream.h
+++ b/chromium/third_party/webrtc/base/socketstream.h
@@ -13,6 +13,7 @@
 
 #include "webrtc/base/asyncsocket.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/stream.h"
 
 namespace rtc {
diff --git a/chromium/third_party/webrtc/base/ssladapter_unittest.cc b/chromium/third_party/webrtc/base/ssladapter_unittest.cc
index 7869b6eb63b..16e5c2ea322 100644
--- a/chromium/third_party/webrtc/base/ssladapter_unittest.cc
+++ b/chromium/third_party/webrtc/base/ssladapter_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/gunit.h"
@@ -123,7 +124,7 @@ class SSLAdapterTestDummyClient : public sigslot::has_slots<> {
  private:
   const rtc::SSLMode ssl_mode_;
 
-  rtc::scoped_ptr<rtc::SSLAdapter> ssl_adapter_;
+  std::unique_ptr<rtc::SSLAdapter> ssl_adapter_;
 
   std::string data_;
 };
@@ -259,10 +260,10 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
 
   const rtc::SSLMode ssl_mode_;
 
-  rtc::scoped_ptr<rtc::AsyncSocket> server_socket_;
-  rtc::scoped_ptr<rtc::SSLStreamAdapter> ssl_stream_adapter_;
+  std::unique_ptr<rtc::AsyncSocket> server_socket_;
+  std::unique_ptr<rtc::SSLStreamAdapter> ssl_stream_adapter_;
 
-  rtc::scoped_ptr<rtc::SSLIdentity> ssl_identity_;
+  std::unique_ptr<rtc::SSLIdentity> ssl_identity_;
 
   std::string data_;
 };
@@ -339,8 +340,8 @@ class SSLAdapterTestBase : public testing::Test,
 
   const rtc::SocketServerScope ss_scope_;
 
-  rtc::scoped_ptr<SSLAdapterTestDummyServer> server_;
-  rtc::scoped_ptr<SSLAdapterTestDummyClient> client_;
+  std::unique_ptr<SSLAdapterTestDummyServer> server_;
+  std::unique_ptr<SSLAdapterTestDummyClient> client_;
 
   int handshake_wait_;
 };
diff --git a/chromium/third_party/webrtc/base/sslfingerprint.cc b/chromium/third_party/webrtc/base/sslfingerprint.cc
index 1939b4fd0bd..2c3e1e974be 100644
--- a/chromium/third_party/webrtc/base/sslfingerprint.cc
+++ b/chromium/third_party/webrtc/base/sslfingerprint.cc
@@ -85,7 +85,7 @@ std::string SSLFingerprint::GetRfc4572Fingerprint() const {
   return fingerprint;
 }
 
-std::string SSLFingerprint::ToString() {
+std::string SSLFingerprint::ToString() const {
   std::string fp_str = algorithm;
   fp_str.append(" ");
   fp_str.append(GetRfc4572Fingerprint());
diff --git a/chromium/third_party/webrtc/base/sslfingerprint.h b/chromium/third_party/webrtc/base/sslfingerprint.h
index 1413a4cd272..4ffb2b0524e 100644
--- a/chromium/third_party/webrtc/base/sslfingerprint.h
+++ b/chromium/third_party/webrtc/base/sslfingerprint.h
@@ -41,7 +41,7 @@ struct SSLFingerprint {
 
   std::string GetRfc4572Fingerprint() const;
 
-  std::string ToString();
+  std::string ToString() const;
 
   std::string algorithm;
   rtc::CopyOnWriteBuffer digest;
diff --git a/chromium/third_party/webrtc/base/sslidentity.cc b/chromium/third_party/webrtc/base/sslidentity.cc
index 5fa8bbf6b43..5f3a73fad17 100644
--- a/chromium/third_party/webrtc/base/sslidentity.cc
+++ b/chromium/third_party/webrtc/base/sslidentity.cc
@@ -187,6 +187,14 @@ SSLIdentity* SSLIdentity::FromPEMStrings(const std::string& private_key,
   return OpenSSLIdentity::FromPEMStrings(private_key, certificate);
 }
 
+bool operator==(const SSLIdentity& a, const SSLIdentity& b) {
+  return static_cast<const OpenSSLIdentity&>(a) ==
+         static_cast<const OpenSSLIdentity&>(b);
+}
+bool operator!=(const SSLIdentity& a, const SSLIdentity& b) {
+  return !(a == b);
+}
+
 #else  // !SSL_USE_OPENSSL
 
 #error "No SSL implementation"
diff --git a/chromium/third_party/webrtc/base/sslidentity.h b/chromium/third_party/webrtc/base/sslidentity.h
index 77c9e186c1c..8e7e7861308 100644
--- a/chromium/third_party/webrtc/base/sslidentity.h
+++ b/chromium/third_party/webrtc/base/sslidentity.h
@@ -14,12 +14,13 @@
 #define WEBRTC_BASE_SSLIDENTITY_H_
 
 #include <algorithm>
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/messagedigest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/timeutils.h"
 
 namespace rtc {
@@ -53,7 +54,7 @@ class SSLCertificate {
 
   // Provides the cert chain, or null. The chain includes a copy of each
   // certificate, excluding the leaf.
-  virtual rtc::scoped_ptr<SSLCertChain> GetChain() const = 0;
+  virtual std::unique_ptr<SSLCertChain> GetChain() const = 0;
 
   // Returns a PEM encoded string representation of the certificate.
   virtual std::string ToPEMString() const = 0;
@@ -113,13 +114,23 @@ class SSLCertChain {
   RTC_DISALLOW_COPY_AND_ASSIGN(SSLCertChain);
 };
 
-// KT_DEFAULT is currently an alias for KT_RSA.  This is likely to change.
 // KT_LAST is intended for vector declarations and loops over all key types;
 // it does not represent any key type in itself.
-// TODO(hbos,torbjorng): Don't change KT_DEFAULT without first updating
-// PeerConnectionFactory_nativeCreatePeerConnection's certificate generation
-// code.
-enum KeyType { KT_RSA, KT_ECDSA, KT_LAST, KT_DEFAULT = KT_RSA };
+// KT_DEFAULT is used as the default KeyType for KeyParams.
+enum KeyType {
+  KT_RSA, KT_ECDSA, KT_LAST,
+#if defined(WEBRTC_CHROMIUM_BUILD)
+  // TODO(hbos): Because of an experiment running in Chromium which relies on
+  // RSA being the default (for performance reasons) we have this #if. ECDSA
+  // launches in Chromium by flipping a flag which overrides the default. As
+  // soon as the experiment has ended and there is no risk of RSA being the
+  // default we should make KT_DEFAULT = KT_ECDSA unconditionally.
+  // crbug.com/611698
+  KT_DEFAULT = KT_RSA
+#else
+  KT_DEFAULT = KT_ECDSA
+#endif
+};
 
 static const int kRsaDefaultModSize = 1024;
 static const int kRsaDefaultExponent = 0x10001;  // = 2^16+1 = 65537
@@ -226,6 +237,8 @@ class SSLIdentity {
 
   // Returns a temporary reference to the certificate.
   virtual const SSLCertificate& certificate() const = 0;
+  virtual std::string PrivateKeyToPEMString() const = 0;
+  virtual std::string PublicKeyToPEMString() const = 0;
 
   // Helpers for parsing converting between PEM and DER format.
   static bool PemToDer(const std::string& pem_type,
@@ -236,6 +249,9 @@ class SSLIdentity {
                               size_t length);
 };
 
+bool operator==(const SSLIdentity& a, const SSLIdentity& b);
+bool operator!=(const SSLIdentity& a, const SSLIdentity& b);
+
 // Convert from ASN1 time as restricted by RFC 5280 to seconds from 1970-01-01
 // 00.00 ("epoch").  If the ASN1 time cannot be read, return -1.  The data at
 // |s| is not 0-terminated; its char count is defined by |length|.
diff --git a/chromium/third_party/webrtc/base/sslidentity_unittest.cc b/chromium/third_party/webrtc/base/sslidentity_unittest.cc
index f110f76861e..399fe9d4745 100644
--- a/chromium/third_party/webrtc/base/sslidentity_unittest.cc
+++ b/chromium/third_party/webrtc/base/sslidentity_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/gunit.h"
@@ -173,12 +174,43 @@ class SSLIdentityTest : public testing::Test {
     EXPECT_EQ(0, memcmp(digest, expected_digest, expected_len));
   }
 
- private:
-  rtc::scoped_ptr<SSLIdentity> identity_rsa1_;
-  rtc::scoped_ptr<SSLIdentity> identity_rsa2_;
-  rtc::scoped_ptr<SSLIdentity> identity_ecdsa1_;
-  rtc::scoped_ptr<SSLIdentity> identity_ecdsa2_;
-  rtc::scoped_ptr<rtc::SSLCertificate> test_cert_;
+  void TestCloningIdentity(const SSLIdentity& identity) {
+    // Convert |identity| to PEM strings and create a new identity by converting
+    // back from the string format.
+    std::string priv_pem = identity.PrivateKeyToPEMString();
+    std::string publ_pem = identity.PublicKeyToPEMString();
+    std::string cert_pem = identity.certificate().ToPEMString();
+    std::unique_ptr<SSLIdentity> clone(
+        SSLIdentity::FromPEMStrings(priv_pem, cert_pem));
+    EXPECT_TRUE(clone);
+
+    // Make sure the clone is identical to the original.
+    EXPECT_TRUE(identity == *clone);
+    ASSERT_EQ(identity.certificate().CertificateExpirationTime(),
+              clone->certificate().CertificateExpirationTime());
+
+    // At this point we are confident that the identities are identical. To be
+    // extra sure, we compare PEM strings of the clone with the original. Note
+    // that the PEM strings of two identities are not strictly guaranteed to be
+    // equal (they describe structs whose members could be listed in a different
+    // order, for example). But because the same function is used to produce
+    // both PEMs, its a good enough bet that this comparison will work. If the
+    // assumption stops holding in the future we can always remove this from the
+    // unittest.
+    std::string clone_priv_pem = clone->PrivateKeyToPEMString();
+    std::string clone_publ_pem = clone->PublicKeyToPEMString();
+    std::string clone_cert_pem = clone->certificate().ToPEMString();
+    ASSERT_EQ(priv_pem, clone_priv_pem);
+    ASSERT_EQ(publ_pem, clone_publ_pem);
+    ASSERT_EQ(cert_pem, clone_cert_pem);
+  }
+
+ protected:
+  std::unique_ptr<SSLIdentity> identity_rsa1_;
+  std::unique_ptr<SSLIdentity> identity_rsa2_;
+  std::unique_ptr<SSLIdentity> identity_ecdsa1_;
+  std::unique_ptr<SSLIdentity> identity_ecdsa2_;
+  std::unique_ptr<rtc::SSLCertificate> test_cert_;
 };
 
 TEST_F(SSLIdentityTest, FixedDigestSHA1) {
@@ -219,71 +251,118 @@ TEST_F(SSLIdentityTest, DigestSHA512) {
   TestDigestForGeneratedCert(rtc::DIGEST_SHA_512, 64);
 }
 
+TEST_F(SSLIdentityTest, IdentityComparison) {
+  EXPECT_TRUE(*identity_rsa1_ == *identity_rsa1_);
+  EXPECT_FALSE(*identity_rsa1_ == *identity_rsa2_);
+  EXPECT_FALSE(*identity_rsa1_ == *identity_ecdsa1_);
+  EXPECT_FALSE(*identity_rsa1_ == *identity_ecdsa2_);
+
+  EXPECT_TRUE(*identity_rsa2_ == *identity_rsa2_);
+  EXPECT_FALSE(*identity_rsa2_ == *identity_ecdsa1_);
+  EXPECT_FALSE(*identity_rsa2_ == *identity_ecdsa2_);
+
+  EXPECT_TRUE(*identity_ecdsa1_ == *identity_ecdsa1_);
+  EXPECT_FALSE(*identity_ecdsa1_ == *identity_ecdsa2_);
+}
+
 TEST_F(SSLIdentityTest, FromPEMStringsRSA) {
+  // These PEM strings were created by generating an identity with
+  // |SSLIdentity::Generate| and invoking |identity->PrivateKeyToPEMString()|,
+  // |identity->PublicKeyToPEMString()| and
+  // |identity->certificate().ToPEMString()|. If the crypto library is updated,
+  // and the update changes the string form of the keys, these will have to be
+  // updated too.
   static const char kRSA_PRIVATE_KEY_PEM[] =
-      "-----BEGIN RSA PRIVATE KEY-----\n"
-      "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
-      "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
-      "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
-      "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n"
-      "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n"
-      "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n"
-      "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n"
-      "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n"
-      "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n"
-      "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n"
-      "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n"
-      "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n"
-      "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n"
-      "UCXiYxSsu20QNVw=\n"
-      "-----END RSA PRIVATE KEY-----\n";
-
+      "-----BEGIN PRIVATE KEY-----\n"
+      "MIICdQIBADANBgkqhkiG9w0BAQEFAASCAl8wggJbAgEAAoGBAMQPqDStRlYeDpkX\n"
+      "erRmv+a1naM8vSVSY0gG2plnrnofViWRW3MRqWC+020MsIj3hPZeSAnt/y/FL/nr\n"
+      "4Ea7NXcwdRo1/1xEK7U/f/cjSg1aunyvHCHwcFcMr31HLFvHr0ZgcFwbgIuFLNEl\n"
+      "7kK5HMO9APz1ntUjek8BmBj8yMl9AgMBAAECgYA8FWBC5GcNtSBcIinkZyigF0A7\n"
+      "6j081sa+J/uNz4xUuI257ZXM6biygUhhvuXK06/XoIULJfhyN0fAm1yb0HtNhiUs\n"
+      "kMOYeon6b8FqFaPjrQf7Gr9FMiIHXNK19uegTMKztXyPZoUWlX84X0iawY95x0Y3\n"
+      "73f6P2rN2UOjlVVjAQJBAOKy3l2w3Zj2w0oAJox0eMwl+RxBNt1C42SHrob2mFUT\n"
+      "rytpVVYOasr8CoDI0kjacjI94sLum+buJoXXX6YTGO0CQQDdZwlYIEkoS3ftfxPa\n"
+      "Ai0YTBzAWvHJg0r8Gk/TkHo6IM+LSsZ9ZYUv/vBe4BKLw1I4hZ+bQvBiq+f8ROtk\n"
+      "+TDRAkAPL3ghwoU1h+IRBO2QHwUwd6K2N9AbBi4BP+168O3HVSg4ujeTKigRLMzv\n"
+      "T4R2iNt5bhfQgvdCgtVlxcWMdF8JAkBwDCg3eEdt5BuyjwBt8XH+/O4ED0KUWCTH\n"
+      "x00k5dZlupsuhE5Fwe4QpzXg3gekwdnHjyCCQ/NCDHvgOMTkmhQxAkA9V03KRX9b\n"
+      "bhvEzY/fu8gEp+EzsER96/D79az5z1BaMGL5OPM2xHBPJATKlswnAa7Lp3QKGZGk\n"
+      "TxslfL18J71s\n"
+      "-----END PRIVATE KEY-----\n";
+  static const char kRSA_PUBLIC_KEY_PEM[] =
+      "-----BEGIN PUBLIC KEY-----\n"
+      "MIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDED6g0rUZWHg6ZF3q0Zr/mtZ2j\n"
+      "PL0lUmNIBtqZZ656H1YlkVtzEalgvtNtDLCI94T2XkgJ7f8vxS/56+BGuzV3MHUa\n"
+      "Nf9cRCu1P3/3I0oNWrp8rxwh8HBXDK99Ryxbx69GYHBcG4CLhSzRJe5CuRzDvQD8\n"
+      "9Z7VI3pPAZgY/MjJfQIDAQAB\n"
+      "-----END PUBLIC KEY-----\n";
   static const char kCERT_PEM[] =
       "-----BEGIN CERTIFICATE-----\n"
-      "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n"
-      "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n"
-      "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n"
-      "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
-      "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
-      "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n"
-      "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n"
-      "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
-      "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
+      "MIIBnDCCAQWgAwIBAgIJAOEHLgeWYwrpMA0GCSqGSIb3DQEBCwUAMBAxDjAMBgNV\n"
+      "BAMMBXRlc3QxMB4XDTE2MDQyNDE4MTAyMloXDTE2MDUyNTE4MTAyMlowEDEOMAwG\n"
+      "A1UEAwwFdGVzdDEwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMQPqDStRlYe\n"
+      "DpkXerRmv+a1naM8vSVSY0gG2plnrnofViWRW3MRqWC+020MsIj3hPZeSAnt/y/F\n"
+      "L/nr4Ea7NXcwdRo1/1xEK7U/f/cjSg1aunyvHCHwcFcMr31HLFvHr0ZgcFwbgIuF\n"
+      "LNEl7kK5HMO9APz1ntUjek8BmBj8yMl9AgMBAAEwDQYJKoZIhvcNAQELBQADgYEA\n"
+      "C3ehaZFl+oEYN069C2ht/gMzuC77L854RF/x7xRtNZzkcg9TVgXXdM3auUvJi8dx\n"
+      "yTpU3ixErjQvoZew5ngXTEvTY8BSQUijJEaLWh8n6NDKRbEGTdAk8nPAmq9hdCFq\n"
+      "e3UkexqNHm3g/VxG4NUC1Y+w29ai0/Rgh+VvgbDwK+Q=\n"
       "-----END CERTIFICATE-----\n";
 
-  rtc::scoped_ptr<SSLIdentity> identity(
+  std::unique_ptr<SSLIdentity> identity(
       SSLIdentity::FromPEMStrings(kRSA_PRIVATE_KEY_PEM, kCERT_PEM));
   EXPECT_TRUE(identity);
+  EXPECT_EQ(kRSA_PRIVATE_KEY_PEM, identity->PrivateKeyToPEMString());
+  EXPECT_EQ(kRSA_PUBLIC_KEY_PEM, identity->PublicKeyToPEMString());
   EXPECT_EQ(kCERT_PEM, identity->certificate().ToPEMString());
 }
 
 TEST_F(SSLIdentityTest, FromPEMStringsEC) {
-  static const char kRSA_PRIVATE_KEY_PEM[] =
-      "-----BEGIN EC PRIVATE KEY-----\n"
-      "MHcCAQEEIKkIztWLPbs4Y2zWv7VW2Ov4is2ifleCuPgRB8fRv3IkoAoGCCqGSM49\n"
-      "AwEHoUQDQgAEDPV33NrhSdhg9cBRkUWUXnVMXc3h17i9ARbSmNgminKcBXb8/y8L\n"
-      "A76cMWQPPM0ybHO8OS7ZVg2U/m+TwE1M2g==\n"
-      "-----END EC PRIVATE KEY-----\n";
+  // These PEM strings were created by generating an identity with
+  // |SSLIdentity::Generate| and invoking |identity->PrivateKeyToPEMString()|,
+  // |identity->PublicKeyToPEMString()| and
+  // |identity->certificate().ToPEMString()|. If the crypto library is updated,
+  // and the update changes the string form of the keys, these will have to be
+  // updated too.
+  static const char kECDSA_PRIVATE_KEY_PEM[] =
+      "-----BEGIN PRIVATE KEY-----\n"
+      "MIGHAgEAMBMGByqGSM49AgEGCCqGSM49AwEHBG0wawIBAQQg/AkEA2hklq7dQ2rN\n"
+      "ZxYL6hOUACL4pn7P4FYlA3ZQhIChRANCAAR7YgdO3utP/8IqVRq8G4VZKreMAxeN\n"
+      "rUa12twthv4uFjuHAHa9D9oyAjncmn+xvZZRyVmKrA56jRzENcEEHoAg\n"
+      "-----END PRIVATE KEY-----\n";
+  static const char kECDSA_PUBLIC_KEY_PEM[] =
+      "-----BEGIN PUBLIC KEY-----\n"
+      "MFkwEwYHKoZIzj0CAQYIKoZIzj0DAQcDQgAEe2IHTt7rT//CKlUavBuFWSq3jAMX\n"
+      "ja1GtdrcLYb+LhY7hwB2vQ/aMgI53Jp/sb2WUclZiqwOeo0cxDXBBB6AIA==\n"
+      "-----END PUBLIC KEY-----\n";
   static const char kCERT_PEM[] =
       "-----BEGIN CERTIFICATE-----\n"
-      "MIIB0jCCAXmgAwIBAgIJAMCjpFt9t6LMMAoGCCqGSM49BAMCMEUxCzAJBgNVBAYT\n"
-      "AkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRlcm5ldCBXaWRn\n"
-      "aXRzIFB0eSBMdGQwIBcNMTUwNjMwMTMwMTIyWhgPMjI4OTA0MTMxMzAxMjJaMEUx\n"
-      "CzAJBgNVBAYTAkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRl\n"
-      "cm5ldCBXaWRnaXRzIFB0eSBMdGQwWTATBgcqhkjOPQIBBggqhkjOPQMBBwNCAAQM\n"
-      "9Xfc2uFJ2GD1wFGRRZRedUxdzeHXuL0BFtKY2CaKcpwFdvz/LwsDvpwxZA88zTJs\n"
-      "c7w5LtlWDZT+b5PATUzao1AwTjAdBgNVHQ4EFgQUYHq6nxNNIE832ZmaHc/noODO\n"
-      "rtAwHwYDVR0jBBgwFoAUYHq6nxNNIE832ZmaHc/noODOrtAwDAYDVR0TBAUwAwEB\n"
-      "/zAKBggqhkjOPQQDAgNHADBEAiAQRojsTyZG0BlKoU7gOt5h+yAMLl2cxmDtOIQr\n"
-      "GWP/PwIgJynB4AUDsPT0DWmethOXYijB5sY5UPd9DvgmiS/Mr6s=\n"
+      "MIIBFDCBu6ADAgECAgkArpkxjw62sW4wCgYIKoZIzj0EAwIwEDEOMAwGA1UEAwwF\n"
+      "dGVzdDMwHhcNMTYwNDI0MTgxNDM4WhcNMTYwNTI1MTgxNDM4WjAQMQ4wDAYDVQQD\n"
+      "DAV0ZXN0MzBZMBMGByqGSM49AgEGCCqGSM49AwEHA0IABHtiB07e60//wipVGrwb\n"
+      "hVkqt4wDF42tRrXa3C2G/i4WO4cAdr0P2jICOdyaf7G9llHJWYqsDnqNHMQ1wQQe\n"
+      "gCAwCgYIKoZIzj0EAwIDSAAwRQIhANyreQ/K5yuPPpirsd0e/4WGLHou6bIOSQks\n"
+      "DYzo56NmAiAKOr3u8ol3LmygbUCwEvtWrS8QcJDygxHPACo99hkekw==\n"
       "-----END CERTIFICATE-----\n";
 
-  rtc::scoped_ptr<SSLIdentity> identity(
-      SSLIdentity::FromPEMStrings(kRSA_PRIVATE_KEY_PEM, kCERT_PEM));
+  std::unique_ptr<SSLIdentity> identity(
+      SSLIdentity::FromPEMStrings(kECDSA_PRIVATE_KEY_PEM, kCERT_PEM));
   EXPECT_TRUE(identity);
+  EXPECT_EQ(kECDSA_PRIVATE_KEY_PEM, identity->PrivateKeyToPEMString());
+  EXPECT_EQ(kECDSA_PUBLIC_KEY_PEM, identity->PublicKeyToPEMString());
   EXPECT_EQ(kCERT_PEM, identity->certificate().ToPEMString());
 }
 
+TEST_F(SSLIdentityTest, CloneIdentityRSA) {
+  TestCloningIdentity(*identity_rsa1_);
+  TestCloningIdentity(*identity_rsa2_);
+}
+
+TEST_F(SSLIdentityTest, CloneIdentityECDSA) {
+  TestCloningIdentity(*identity_ecdsa1_);
+  TestCloningIdentity(*identity_ecdsa2_);
+}
+
 TEST_F(SSLIdentityTest, PemDerConversion) {
   std::string der;
   EXPECT_TRUE(SSLIdentity::PemToDer("CERTIFICATE", kTestCertificate, &der));
diff --git a/chromium/third_party/webrtc/base/sslsocketfactory.cc b/chromium/third_party/webrtc/base/sslsocketfactory.cc
index d6ec56fd406..7ab58fd04c6 100644
--- a/chromium/third_party/webrtc/base/sslsocketfactory.cc
+++ b/chromium/third_party/webrtc/base/sslsocketfactory.cc
@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/autodetectproxy.h"
 #include "webrtc/base/httpcommon.h"
 #include "webrtc/base/httpcommon-inl.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketadapters.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslsocketfactory.h"
@@ -167,7 +168,7 @@ AsyncSocket* SslSocketFactory::CreateProxySocket(const ProxyInfo& proxy,
   }
 
   if (!hostname_.empty()) {
-    rtc::scoped_ptr<SSLAdapter> ssl_adapter(SSLAdapter::Create(socket));
+    std::unique_ptr<SSLAdapter> ssl_adapter(SSLAdapter::Create(socket));
     if (!ssl_adapter) {
       LOG_F(LS_ERROR) << "SSL unavailable";
       delete socket;
diff --git a/chromium/third_party/webrtc/base/sslstreamadapter.h b/chromium/third_party/webrtc/base/sslstreamadapter.h
index f6f0befa05a..c5045f184fb 100644
--- a/chromium/third_party/webrtc/base/sslstreamadapter.h
+++ b/chromium/third_party/webrtc/base/sslstreamadapter.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_
 #define WEBRTC_BASE_SSLSTREAMADAPTER_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -24,8 +25,12 @@ const int TLS_NULL_WITH_NULL_NULL = 0;
 
 // Constants for SRTP profiles.
 const int SRTP_INVALID_CRYPTO_SUITE = 0;
+#ifndef SRTP_AES128_CM_SHA1_80
 const int SRTP_AES128_CM_SHA1_80 = 0x0001;
+#endif
+#ifndef SRTP_AES128_CM_SHA1_32
 const int SRTP_AES128_CM_SHA1_32 = 0x0002;
+#endif
 
 // Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
 // in applications (voice) where the additional bandwidth may be significant.
@@ -155,7 +160,7 @@ class SSLStreamAdapter : public StreamAdapterInterface {
   // Retrieves the peer's X.509 certificate, if a connection has been
   // established. It returns the transmitted over SSL, including the entire
   // chain.
-  virtual rtc::scoped_ptr<SSLCertificate> GetPeerCertificate() const = 0;
+  virtual std::unique_ptr<SSLCertificate> GetPeerCertificate() const = 0;
 
   // Retrieves the IANA registration id of the cipher suite used for the
   // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
diff --git a/chromium/third_party/webrtc/base/sslstreamadapter_unittest.cc b/chromium/third_party/webrtc/base/sslstreamadapter_unittest.cc
index 8d5b275db8c..dc62ac081b7 100644
--- a/chromium/third_party/webrtc/base/sslstreamadapter_unittest.cc
+++ b/chromium/third_party/webrtc/base/sslstreamadapter_unittest.cc
@@ -10,13 +10,13 @@
 
 
 #include <algorithm>
+#include <memory>
 #include <set>
 #include <string>
 
 #include "webrtc/base/bufferqueue.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslconfig.h"
 #include "webrtc/base/sslidentity.h"
@@ -474,7 +474,7 @@ class SSLStreamAdapterTestBase : public testing::Test,
       return server_ssl_->GetDtlsSrtpCryptoSuite(retval);
   }
 
-  rtc::scoped_ptr<rtc::SSLCertificate> GetPeerCertificate(bool client) {
+  std::unique_ptr<rtc::SSLCertificate> GetPeerCertificate(bool client) {
     if (client)
       return client_ssl_->GetPeerCertificate();
     else
@@ -526,8 +526,8 @@ class SSLStreamAdapterTestBase : public testing::Test,
   rtc::KeyParams server_key_type_;
   SSLDummyStreamBase *client_stream_;  // freed by client_ssl_ destructor
   SSLDummyStreamBase *server_stream_;  // freed by server_ssl_ destructor
-  rtc::scoped_ptr<rtc::SSLStreamAdapter> client_ssl_;
-  rtc::scoped_ptr<rtc::SSLStreamAdapter> server_ssl_;
+  std::unique_ptr<rtc::SSLStreamAdapter> client_ssl_;
+  std::unique_ptr<rtc::SSLStreamAdapter> server_ssl_;
   rtc::SSLIdentity *client_identity_;  // freed by client_ssl_ destructor
   rtc::SSLIdentity *server_identity_;  // freed by server_ssl_ destructor
   int delay_;
@@ -562,7 +562,7 @@ class SSLStreamAdapterTestTLS
   }
 
   // Test data transfer for TLS
-  virtual void TestTransfer(int size) {
+  void TestTransfer(int size) override {
     LOG(LS_INFO) << "Starting transfer test with " << size << " bytes";
     // Create some dummy data to send.
     size_t received;
@@ -591,7 +591,7 @@ class SSLStreamAdapterTestTLS
                         recv_stream_.GetBuffer(), size));
   }
 
-  void WriteData() {
+  void WriteData() override {
     size_t position, tosend, size;
     rtc::StreamResult rv;
     size_t sent;
@@ -627,7 +627,7 @@ class SSLStreamAdapterTestTLS
     }
   };
 
-  virtual void ReadData(rtc::StreamInterface *stream) {
+  void ReadData(rtc::StreamInterface *stream) override {
     char buffer[1600];
     size_t bread;
     int err2;
@@ -691,7 +691,7 @@ class SSLStreamAdapterTestDTLS
         new SSLDummyStreamDTLS(this, "s2c", &server_buffer_, &client_buffer_);
   }
 
-  virtual void WriteData() {
+  void WriteData() override {
     unsigned char *packet = new unsigned char[1600];
 
     while (sent_ < count_) {
@@ -720,7 +720,7 @@ class SSLStreamAdapterTestDTLS
     delete [] packet;
   }
 
-  virtual void ReadData(rtc::StreamInterface *stream) {
+  void ReadData(rtc::StreamInterface *stream) override {
     unsigned char buffer[2000];
     size_t bread;
     int err2;
@@ -756,7 +756,7 @@ class SSLStreamAdapterTestDTLS
     }
   }
 
-  virtual void TestTransfer(int count) {
+  void TestTransfer(int count) override {
     count_ = count;
 
     WriteData();
@@ -1043,7 +1043,7 @@ TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
   TestHandshake();
 
   // The client should have a peer certificate after the handshake.
-  rtc::scoped_ptr<rtc::SSLCertificate> client_peer_cert =
+  std::unique_ptr<rtc::SSLCertificate> client_peer_cert =
       GetPeerCertificate(true);
   ASSERT_TRUE(client_peer_cert);
 
@@ -1055,7 +1055,7 @@ TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
   ASSERT_FALSE(client_peer_cert->GetChain());
 
   // The server should have a peer certificate after the handshake.
-  rtc::scoped_ptr<rtc::SSLCertificate> server_peer_cert =
+  std::unique_ptr<rtc::SSLCertificate> server_peer_cert =
       GetPeerCertificate(false);
   ASSERT_TRUE(server_peer_cert);
 
diff --git a/chromium/third_party/webrtc/base/stream.h b/chromium/third_party/webrtc/base/stream.h
index e624df55044..bc949065c8f 100644
--- a/chromium/third_party/webrtc/base/stream.h
+++ b/chromium/third_party/webrtc/base/stream.h
@@ -11,15 +11,16 @@
 #ifndef WEBRTC_BASE_STREAM_H_
 #define WEBRTC_BASE_STREAM_H_
 
+#include <memory>
 #include <stdio.h>
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/messagequeue.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 
 namespace rtc {
@@ -334,7 +335,7 @@ class StreamTap : public StreamAdapterInterface {
                      int* error) override;
 
  private:
-  scoped_ptr<StreamInterface> tap_;
+  std::unique_ptr<StreamInterface> tap_;
   StreamResult tap_result_;
   int tap_error_;
   RTC_DISALLOW_COPY_AND_ASSIGN(StreamTap);
@@ -554,7 +555,7 @@ class FifoBuffer : public StreamInterface {
   // keeps the opened/closed state of the stream
   StreamState state_ GUARDED_BY(crit_);
   // the allocated buffer
-  scoped_ptr<char[]> buffer_ GUARDED_BY(crit_);
+  std::unique_ptr<char[]> buffer_ GUARDED_BY(crit_);
   // size of the allocated buffer
   size_t buffer_length_ GUARDED_BY(crit_);
   // amount of readable data in the buffer
diff --git a/chromium/third_party/webrtc/base/swap_queue.h b/chromium/third_party/webrtc/base/swap_queue.h
index d3af225eb17..1851309bd73 100644
--- a/chromium/third_party/webrtc/base/swap_queue.h
+++ b/chromium/third_party/webrtc/base/swap_queue.h
@@ -16,6 +16,7 @@
 #include <vector>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/base/task.h b/chromium/third_party/webrtc/base/task.h
index 28702e49a7a..b2a60671534 100644
--- a/chromium/third_party/webrtc/base/task.h
+++ b/chromium/third_party/webrtc/base/task.h
@@ -13,7 +13,6 @@
 
 #include <string>
 #include "webrtc/base/basictypes.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/taskparent.h"
 
diff --git a/chromium/third_party/webrtc/base/task_queue.h b/chromium/third_party/webrtc/base/task_queue.h
new file mode 100644
index 00000000000..dad4f431b75
--- /dev/null
+++ b/chromium/third_party/webrtc/base/task_queue.h
@@ -0,0 +1,277 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TASK_QUEUE_H_
+#define WEBRTC_BASE_TASK_QUEUE_H_
+
+#include <list>
+#include <memory>
+
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_BUILD_LIBEVENT)
+#include <dispatch/dispatch.h>
+#endif
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+
+#if defined(WEBRTC_WIN) || defined(WEBRTC_BUILD_LIBEVENT)
+#include "webrtc/base/platform_thread.h"
+#endif
+
+#if defined(WEBRTC_BUILD_LIBEVENT)
+struct event_base;
+struct event;
+#endif
+
+namespace rtc {
+
+// Base interface for asynchronously executed tasks.
+// The interface basically consists of a single function, Run(), that executes
+// on the target queue.  For more details see the Run() method and TaskQueue.
+class QueuedTask {
+ public:
+  QueuedTask() {}
+  virtual ~QueuedTask() {}
+
+  // Main routine that will run when the task is executed on the desired queue.
+  // The task should return |true| to indicate that it should be deleted or
+  // |false| to indicate that the queue should consider ownership of the task
+  // having been transferred.  Returning |false| can be useful if a task has
+  // re-posted itself to a different queue or is otherwise being re-used.
+  virtual bool Run() = 0;
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(QueuedTask);
+};
+
+// Simple implementation of QueuedTask for use with rtc::Bind and lambdas.
+template <class Closure>
+class ClosureTask : public QueuedTask {
+ public:
+  explicit ClosureTask(const Closure& closure) : closure_(closure) {}
+
+ private:
+  bool Run() override {
+    closure_();
+    return true;
+  }
+
+  Closure closure_;
+};
+
+// Extends ClosureTask to also allow specifying cleanup code.
+// This is useful when using lambdas if guaranteeing cleanup, even if a task
+// was dropped (queue is too full), is required.
+template <class Closure, class Cleanup>
+class ClosureTaskWithCleanup : public ClosureTask<Closure> {
+ public:
+  ClosureTaskWithCleanup(const Closure& closure, Cleanup cleanup)
+      : ClosureTask<Closure>(closure), cleanup_(cleanup) {}
+  ~ClosureTaskWithCleanup() { cleanup_(); }
+
+ private:
+  Cleanup cleanup_;
+};
+
+// Convenience function to construct closures that can be passed directly
+// to methods that support std::unique_ptr<QueuedTask> but not template
+// based parameters.
+template <class Closure>
+static std::unique_ptr<QueuedTask> NewClosure(const Closure& closure) {
+  return std::unique_ptr<QueuedTask>(new ClosureTask<Closure>(closure));
+}
+
+template <class Closure, class Cleanup>
+static std::unique_ptr<QueuedTask> NewClosure(const Closure& closure,
+                                              const Cleanup& cleanup) {
+  return std::unique_ptr<QueuedTask>(
+      new ClosureTaskWithCleanup<Closure, Cleanup>(closure, cleanup));
+}
+
+// Implements a task queue that asynchronously executes tasks in a way that
+// guarantees that they're executed in FIFO order and that tasks never overlap.
+// Tasks may always execute on the same worker thread and they may not.
+// To DCHECK that tasks are executing on a known task queue, use IsCurrent().
+//
+// Here are some usage examples:
+//
+//   1) Asynchronously running a lambda:
+//
+//     class MyClass {
+//       ...
+//       TaskQueue queue_("MyQueue");
+//     };
+//
+//     void MyClass::StartWork() {
+//       queue_.PostTask([]() { Work(); });
+//     ...
+//
+//   2) Doing work asynchronously on a worker queue and providing a notification
+//      callback on the current queue, when the work has been done:
+//
+//     void MyClass::StartWorkAndLetMeKnowWhenDone(
+//         std::unique_ptr<QueuedTask> callback) {
+//       DCHECK(TaskQueue::Current()) << "Need to be running on a queue";
+//       queue_.PostTaskAndReply([]() { Work(); }, std::move(callback));
+//     }
+//     ...
+//     my_class->StartWorkAndLetMeKnowWhenDone(
+//         NewClosure([]() { LOG(INFO) << "The work is done!";}));
+//
+//   3) Posting a custom task on a timer.  The task posts itself again after
+//      every running:
+//
+//     class TimerTask : public QueuedTask {
+//      public:
+//       TimerTask() {}
+//      private:
+//       bool Run() override {
+//         ++count_;
+//         TaskQueue::Current()->PostDelayedTask(
+//             std::unique_ptr<QueuedTask>(this), 1000);
+//         // Ownership has been transferred to the next occurance,
+//         // so return false to prevent from being deleted now.
+//         return false;
+//       }
+//       int count_ = 0;
+//     };
+//     ...
+//     queue_.PostDelayedTask(
+//         std::unique_ptr<QueuedTask>(new TimerTask()), 1000);
+//
+// For more examples, see task_queue_unittests.cc.
+//
+// A note on destruction:
+//
+// When a TaskQueue is deleted, pending tasks will not be executed but they will
+// be deleted.  The deletion of tasks may happen asynchronously after the
+// TaskQueue itself has been deleted or it may happen synchronously while the
+// TaskQueue instance is being deleted.  This may vary from one OS to the next
+// so assumptions about lifetimes of pending tasks should not be made.
+class TaskQueue {
+ public:
+  explicit TaskQueue(const char* queue_name);
+  // TODO(tommi): Implement move semantics?
+  ~TaskQueue();
+
+  static TaskQueue* Current();
+
+  // Used for DCHECKing the current queue.
+  static bool IsCurrent(const char* queue_name);
+  bool IsCurrent() const;
+
+  // TODO(tommi): For better debuggability, implement FROM_HERE.
+
+  // Ownership of the task is passed to PostTask.
+  void PostTask(std::unique_ptr<QueuedTask> task);
+  void PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                        std::unique_ptr<QueuedTask> reply,
+                        TaskQueue* reply_queue);
+  void PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                        std::unique_ptr<QueuedTask> reply);
+
+  void PostDelayedTask(std::unique_ptr<QueuedTask> task, uint32_t milliseconds);
+
+  template <class Closure>
+  void PostTask(const Closure& closure) {
+    PostTask(std::unique_ptr<QueuedTask>(new ClosureTask<Closure>(closure)));
+  }
+
+  template <class Closure>
+  void PostDelayedTask(const Closure& closure, uint32_t milliseconds) {
+    PostDelayedTask(
+        std::unique_ptr<QueuedTask>(new ClosureTask<Closure>(closure)),
+        milliseconds);
+  }
+
+  template <class Closure1, class Closure2>
+  void PostTaskAndReply(const Closure1& task,
+                        const Closure2& reply,
+                        TaskQueue* reply_queue) {
+    PostTaskAndReply(
+        std::unique_ptr<QueuedTask>(new ClosureTask<Closure1>(task)),
+        std::unique_ptr<QueuedTask>(new ClosureTask<Closure2>(reply)),
+        reply_queue);
+  }
+
+  template <class Closure>
+  void PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                        const Closure& reply) {
+    PostTaskAndReply(std::move(task), std::unique_ptr<QueuedTask>(
+                                          new ClosureTask<Closure>(reply)));
+  }
+
+  template <class Closure>
+  void PostTaskAndReply(const Closure& task,
+                        std::unique_ptr<QueuedTask> reply) {
+    PostTaskAndReply(
+        std::unique_ptr<QueuedTask>(new ClosureTask<Closure>(task)),
+        std::move(reply));
+  }
+
+  template <class Closure1, class Closure2>
+  void PostTaskAndReply(const Closure1& task, const Closure2& reply) {
+    PostTaskAndReply(
+        std::unique_ptr<QueuedTask>(new ClosureTask<Closure1>(task)),
+        std::unique_ptr<QueuedTask>(new ClosureTask<Closure2>(reply)));
+  }
+
+ private:
+#if defined(WEBRTC_BUILD_LIBEVENT)
+  static bool ThreadMain(void* context);
+  static void OnWakeup(int socket, short flags, void* context);  // NOLINT
+  static void RunTask(int fd, short flags, void* context);       // NOLINT
+  static void RunTimer(int fd, short flags, void* context);      // NOLINT
+
+  class PostAndReplyTask;
+  class SetTimerTask;
+
+  void PrepareReplyTask(PostAndReplyTask* reply_task);
+  void ReplyTaskDone(PostAndReplyTask* reply_task);
+
+  struct QueueContext;
+
+  int wakeup_pipe_in_ = -1;
+  int wakeup_pipe_out_ = -1;
+  event_base* event_base_;
+  std::unique_ptr<event> wakeup_event_;
+  PlatformThread thread_;
+  rtc::CriticalSection pending_lock_;
+  std::list<std::unique_ptr<QueuedTask>> pending_ GUARDED_BY(pending_lock_);
+  std::list<PostAndReplyTask*> pending_replies_ GUARDED_BY(pending_lock_);
+#elif defined(WEBRTC_MAC)
+  struct QueueContext;
+  struct TaskContext;
+  struct PostTaskAndReplyContext;
+  dispatch_queue_t queue_;
+  QueueContext* const context_;
+#elif defined(WEBRTC_WIN)
+  static bool ThreadMain(void* context);
+
+  class WorkerThread : public PlatformThread {
+   public:
+    WorkerThread(ThreadRunFunction func, void* obj, const char* thread_name)
+        : PlatformThread(func, obj, thread_name) {}
+
+    bool QueueAPC(PAPCFUNC apc_function, ULONG_PTR data) {
+      return PlatformThread::QueueAPC(apc_function, data);
+    }
+  };
+  WorkerThread thread_;
+#else
+#error not supported.
+#endif
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(TaskQueue);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_TASK_QUEUE_H_
diff --git a/chromium/third_party/webrtc/base/task_queue_gcd.cc b/chromium/third_party/webrtc/base/task_queue_gcd.cc
new file mode 100644
index 00000000000..2c7d649fc95
--- /dev/null
+++ b/chromium/third_party/webrtc/base/task_queue_gcd.cc
@@ -0,0 +1,167 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains the implementation of TaskQueue for Mac and iOS.
+// The implementation uses Grand Central Dispatch queues (GCD) to
+// do the actual task queuing.
+
+#include "webrtc/base/task_queue.h"
+
+#include <string.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/task_queue_posix.h"
+
+namespace rtc {
+using internal::GetQueuePtrTls;
+using internal::AutoSetCurrentQueuePtr;
+
+struct TaskQueue::QueueContext {
+  explicit QueueContext(TaskQueue* q) : queue(q), is_active(true) {}
+
+  static void SetNotActive(void* context) {
+    QueueContext* qc = static_cast<QueueContext*>(context);
+    qc->is_active = false;
+  }
+
+  static void DeleteContext(void* context) {
+    QueueContext* qc = static_cast<QueueContext*>(context);
+    delete qc;
+  }
+
+  TaskQueue* const queue;
+  bool is_active;
+};
+
+struct TaskQueue::TaskContext {
+  TaskContext(QueueContext* queue_ctx, std::unique_ptr<QueuedTask> task)
+      : queue_ctx(queue_ctx), task(std::move(task)) {}
+  virtual ~TaskContext() {}
+
+  static void RunTask(void* context) {
+    std::unique_ptr<TaskContext> tc(static_cast<TaskContext*>(context));
+    if (tc->queue_ctx->is_active) {
+      AutoSetCurrentQueuePtr set_current(tc->queue_ctx->queue);
+      if (!tc->task->Run())
+        tc->task.release();
+    }
+  }
+
+  QueueContext* const queue_ctx;
+  std::unique_ptr<QueuedTask> task;
+};
+
+// Special case context for holding two tasks, a |first_task| + the task
+// that's owned by the parent struct, TaskContext, that then becomes the
+// second (i.e. 'reply') task.
+struct TaskQueue::PostTaskAndReplyContext : public TaskQueue::TaskContext {
+  explicit PostTaskAndReplyContext(QueueContext* first_queue_ctx,
+                                   std::unique_ptr<QueuedTask> first_task,
+                                   QueueContext* second_queue_ctx,
+                                   std::unique_ptr<QueuedTask> second_task)
+      : TaskContext(second_queue_ctx, std::move(second_task)),
+        first_queue_ctx(first_queue_ctx),
+        first_task(std::move(first_task)) {
+    // Retain the reply queue for as long as this object lives.
+    // If we don't, we may have memory leaks and/or failures.
+    dispatch_retain(first_queue_ctx->queue->queue_);
+  }
+  ~PostTaskAndReplyContext() override {
+    dispatch_release(first_queue_ctx->queue->queue_);
+  }
+
+  static void RunTask(void* context) {
+    auto* rc = static_cast<PostTaskAndReplyContext*>(context);
+    if (rc->first_queue_ctx->is_active) {
+      AutoSetCurrentQueuePtr set_current(rc->first_queue_ctx->queue);
+      if (!rc->first_task->Run())
+        rc->first_task.release();
+    }
+    // Post the reply task.  This hands the work over to the parent struct.
+    // This task will eventually delete |this|.
+    dispatch_async_f(rc->queue_ctx->queue->queue_, rc, &TaskContext::RunTask);
+  }
+
+  QueueContext* const first_queue_ctx;
+  std::unique_ptr<QueuedTask> first_task;
+};
+
+TaskQueue::TaskQueue(const char* queue_name)
+    : queue_(dispatch_queue_create(queue_name, DISPATCH_QUEUE_SERIAL)),
+      context_(new QueueContext(this)) {
+  RTC_DCHECK(queue_name);
+  RTC_CHECK(queue_);
+  dispatch_set_context(queue_, context_);
+  // Assign a finalizer that will delete the context when the last reference
+  // to the queue is released.  This may run after the TaskQueue object has
+  // been deleted.
+  dispatch_set_finalizer_f(queue_, &QueueContext::DeleteContext);
+}
+
+TaskQueue::~TaskQueue() {
+  RTC_DCHECK(!IsCurrent());
+  // Implementation/behavioral note:
+  // Dispatch queues are reference counted via calls to dispatch_retain and
+  // dispatch_release. Pending blocks submitted to a queue also hold a
+  // reference to the queue until they have finished. Once all references to a
+  // queue have been released, the queue will be deallocated by the system.
+  // This is why we check the context before running tasks.
+
+  // Use dispatch_sync to set the context to null to guarantee that there's not
+  // a race between checking the context and using it from a task.
+  dispatch_sync_f(queue_, context_, &QueueContext::SetNotActive);
+  dispatch_release(queue_);
+}
+
+// static
+TaskQueue* TaskQueue::Current() {
+  return static_cast<TaskQueue*>(pthread_getspecific(GetQueuePtrTls()));
+}
+
+// static
+bool TaskQueue::IsCurrent(const char* queue_name) {
+  TaskQueue* current = Current();
+  return current &&
+         strcmp(queue_name, dispatch_queue_get_label(current->queue_)) == 0;
+}
+
+bool TaskQueue::IsCurrent() const {
+  RTC_DCHECK(queue_);
+  return this == Current();
+}
+
+void TaskQueue::PostTask(std::unique_ptr<QueuedTask> task) {
+  auto* context = new TaskContext(context_, std::move(task));
+  dispatch_async_f(queue_, context, &TaskContext::RunTask);
+}
+
+void TaskQueue::PostDelayedTask(std::unique_ptr<QueuedTask> task,
+                                uint32_t milliseconds) {
+  auto* context = new TaskContext(context_, std::move(task));
+  dispatch_after_f(
+      dispatch_time(DISPATCH_TIME_NOW, milliseconds * NSEC_PER_MSEC), queue_,
+      context, &TaskContext::RunTask);
+}
+
+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                                 std::unique_ptr<QueuedTask> reply,
+                                 TaskQueue* reply_queue) {
+  auto* context = new PostTaskAndReplyContext(
+      context_, std::move(task), reply_queue->context_, std::move(reply));
+  dispatch_async_f(queue_, context, &PostTaskAndReplyContext::RunTask);
+}
+
+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                                 std::unique_ptr<QueuedTask> reply) {
+  return PostTaskAndReply(std::move(task), std::move(reply), Current());
+}
+
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/task_queue_libevent.cc b/chromium/third_party/webrtc/base/task_queue_libevent.cc
new file mode 100644
index 00000000000..a59b450828c
--- /dev/null
+++ b/chromium/third_party/webrtc/base/task_queue_libevent.cc
@@ -0,0 +1,318 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/task_queue.h"
+
+#include <fcntl.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "base/third_party/libevent/event.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/task_queue_posix.h"
+#include "webrtc/base/timeutils.h"
+
+namespace rtc {
+using internal::GetQueuePtrTls;
+using internal::AutoSetCurrentQueuePtr;
+
+namespace {
+static const char kQuit = 1;
+static const char kRunTask = 2;
+
+struct TimerEvent {
+  explicit TimerEvent(std::unique_ptr<QueuedTask> task)
+      : task(std::move(task)) {}
+  ~TimerEvent() { event_del(&ev); }
+  event ev;
+  std::unique_ptr<QueuedTask> task;
+};
+
+bool SetNonBlocking(int fd) {
+  const int flags = fcntl(fd, F_GETFL);
+  RTC_CHECK(flags != -1);
+  return (flags & O_NONBLOCK) || fcntl(fd, F_SETFL, flags | O_NONBLOCK) != -1;
+}
+}  // namespace
+
+struct TaskQueue::QueueContext {
+  explicit QueueContext(TaskQueue* q) : queue(q), is_active(true) {}
+  TaskQueue* queue;
+  bool is_active;
+  // Holds a list of events pending timers for cleanup when the loop exits.
+  std::list<TimerEvent*> pending_timers_;
+};
+
+class TaskQueue::PostAndReplyTask : public QueuedTask {
+ public:
+  PostAndReplyTask(std::unique_ptr<QueuedTask> task,
+                   std::unique_ptr<QueuedTask> reply,
+                   TaskQueue* reply_queue)
+      : task_(std::move(task)),
+        reply_(std::move(reply)),
+        reply_queue_(reply_queue) {
+    reply_queue->PrepareReplyTask(this);
+  }
+
+  ~PostAndReplyTask() override {
+    CritScope lock(&lock_);
+    if (reply_queue_)
+      reply_queue_->ReplyTaskDone(this);
+  }
+
+  void OnReplyQueueGone() {
+    CritScope lock(&lock_);
+    reply_queue_ = nullptr;
+  }
+
+ private:
+  bool Run() override {
+    if (!task_->Run())
+      task_.release();
+
+    CritScope lock(&lock_);
+    if (reply_queue_)
+      reply_queue_->PostTask(std::move(reply_));
+    return true;
+  }
+
+  CriticalSection lock_;
+  std::unique_ptr<QueuedTask> task_;
+  std::unique_ptr<QueuedTask> reply_;
+  TaskQueue* reply_queue_ GUARDED_BY(lock_);
+};
+
+class TaskQueue::SetTimerTask : public QueuedTask {
+ public:
+  SetTimerTask(std::unique_ptr<QueuedTask> task, uint32_t milliseconds)
+      : task_(std::move(task)),
+        milliseconds_(milliseconds),
+        posted_(Time32()) {}
+
+ private:
+  bool Run() override {
+    // Compensate for the time that has passed since construction
+    // and until we got here.
+    uint32_t post_time = Time32() - posted_;
+    TaskQueue::Current()->PostDelayedTask(
+        std::move(task_),
+        post_time > milliseconds_ ? 0 : milliseconds_ - post_time);
+    return true;
+  }
+
+  std::unique_ptr<QueuedTask> task_;
+  const uint32_t milliseconds_;
+  const uint32_t posted_;
+};
+
+TaskQueue::TaskQueue(const char* queue_name)
+    : event_base_(event_base_new()),
+      wakeup_event_(new event()),
+      thread_(&TaskQueue::ThreadMain, this, queue_name) {
+  RTC_DCHECK(queue_name);
+  int fds[2];
+  RTC_CHECK(pipe(fds) == 0);
+  SetNonBlocking(fds[0]);
+  SetNonBlocking(fds[1]);
+  wakeup_pipe_out_ = fds[0];
+  wakeup_pipe_in_ = fds[1];
+  event_set(wakeup_event_.get(), wakeup_pipe_out_, EV_READ | EV_PERSIST,
+            OnWakeup, this);
+  event_base_set(event_base_, wakeup_event_.get());
+  event_add(wakeup_event_.get(), 0);
+  thread_.Start();
+}
+
+TaskQueue::~TaskQueue() {
+  RTC_DCHECK(!IsCurrent());
+  struct timespec ts;
+  char message = kQuit;
+  while (write(wakeup_pipe_in_, &message, sizeof(message)) != sizeof(message)) {
+    // The queue is full, so we have no choice but to wait and retry.
+    RTC_CHECK_EQ(EAGAIN, errno);
+    ts.tv_sec = 0;
+    ts.tv_nsec = 1000000;
+    nanosleep(&ts, nullptr);
+  }
+
+  thread_.Stop();
+
+  event_del(wakeup_event_.get());
+  close(wakeup_pipe_in_);
+  close(wakeup_pipe_out_);
+  wakeup_pipe_in_ = -1;
+  wakeup_pipe_out_ = -1;
+
+  {
+    // Synchronize against any pending reply tasks that might be running on
+    // other queues.
+    CritScope lock(&pending_lock_);
+    for (auto* reply : pending_replies_)
+      reply->OnReplyQueueGone();
+    pending_replies_.clear();
+  }
+
+  event_base_free(event_base_);
+}
+
+// static
+TaskQueue* TaskQueue::Current() {
+  QueueContext* ctx =
+      static_cast<QueueContext*>(pthread_getspecific(GetQueuePtrTls()));
+  return ctx ? ctx->queue : nullptr;
+}
+
+// static
+bool TaskQueue::IsCurrent(const char* queue_name) {
+  TaskQueue* current = Current();
+  return current && current->thread_.name().compare(queue_name) == 0;
+}
+
+bool TaskQueue::IsCurrent() const {
+  return IsThreadRefEqual(thread_.GetThreadRef(), CurrentThreadRef());
+}
+
+void TaskQueue::PostTask(std::unique_ptr<QueuedTask> task) {
+  RTC_DCHECK(task.get());
+  // libevent isn't thread safe.  This means that we can't use methods such
+  // as event_base_once to post tasks to the worker thread from a different
+  // thread.  However, we can use it when posting from the worker thread itself.
+  if (IsCurrent()) {
+    if (event_base_once(event_base_, -1, EV_TIMEOUT, &TaskQueue::RunTask,
+                        task.get(), nullptr) == 0) {
+      task.release();
+    }
+  } else {
+    QueuedTask* task_id = task.get();  // Only used for comparison.
+    {
+      CritScope lock(&pending_lock_);
+      pending_.push_back(std::move(task));
+    }
+    char message = kRunTask;
+    if (write(wakeup_pipe_in_, &message, sizeof(message)) != sizeof(message)) {
+      LOG(WARNING) << "Failed to queue task.";
+      CritScope lock(&pending_lock_);
+      pending_.remove_if([task_id](std::unique_ptr<QueuedTask>& t) {
+        return t.get() == task_id;
+      });
+    }
+  }
+}
+
+void TaskQueue::PostDelayedTask(std::unique_ptr<QueuedTask> task,
+                                uint32_t milliseconds) {
+  if (IsCurrent()) {
+    TimerEvent* timer = new TimerEvent(std::move(task));
+    evtimer_set(&timer->ev, &TaskQueue::RunTimer, timer);
+    event_base_set(event_base_, &timer->ev);
+    QueueContext* ctx =
+        static_cast<QueueContext*>(pthread_getspecific(GetQueuePtrTls()));
+    ctx->pending_timers_.push_back(timer);
+    timeval tv = {milliseconds / 1000, (milliseconds % 1000) * 1000};
+    event_add(&timer->ev, &tv);
+  } else {
+    PostTask(std::unique_ptr<QueuedTask>(
+        new SetTimerTask(std::move(task), milliseconds)));
+  }
+}
+
+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                                 std::unique_ptr<QueuedTask> reply,
+                                 TaskQueue* reply_queue) {
+  std::unique_ptr<QueuedTask> wrapper_task(
+      new PostAndReplyTask(std::move(task), std::move(reply), reply_queue));
+  PostTask(std::move(wrapper_task));
+}
+
+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                                 std::unique_ptr<QueuedTask> reply) {
+  return PostTaskAndReply(std::move(task), std::move(reply), Current());
+}
+
+// static
+bool TaskQueue::ThreadMain(void* context) {
+  TaskQueue* me = static_cast<TaskQueue*>(context);
+
+  QueueContext queue_context(me);
+  pthread_setspecific(GetQueuePtrTls(), &queue_context);
+
+  while (queue_context.is_active)
+    event_base_loop(me->event_base_, 0);
+
+  pthread_setspecific(GetQueuePtrTls(), nullptr);
+
+  for (TimerEvent* timer : queue_context.pending_timers_)
+    delete timer;
+
+  return false;
+}
+
+// static
+void TaskQueue::OnWakeup(int socket, short flags, void* context) {  // NOLINT
+  QueueContext* ctx =
+      static_cast<QueueContext*>(pthread_getspecific(GetQueuePtrTls()));
+  RTC_DCHECK(ctx->queue->wakeup_pipe_out_ == socket);
+  char buf;
+  RTC_CHECK(sizeof(buf) == read(socket, &buf, sizeof(buf)));
+  switch (buf) {
+    case kQuit:
+      ctx->is_active = false;
+      event_base_loopbreak(ctx->queue->event_base_);
+      break;
+    case kRunTask: {
+      std::unique_ptr<QueuedTask> task;
+      {
+        CritScope lock(&ctx->queue->pending_lock_);
+        RTC_DCHECK(!ctx->queue->pending_.empty());
+        task = std::move(ctx->queue->pending_.front());
+        ctx->queue->pending_.pop_front();
+        RTC_DCHECK(task.get());
+      }
+      if (!task->Run())
+        task.release();
+      break;
+    }
+    default:
+      RTC_NOTREACHED();
+      break;
+  }
+}
+
+// static
+void TaskQueue::RunTask(int fd, short flags, void* context) {  // NOLINT
+  auto* task = static_cast<QueuedTask*>(context);
+  if (task->Run())
+    delete task;
+}
+
+// static
+void TaskQueue::RunTimer(int fd, short flags, void* context) {  // NOLINT
+  TimerEvent* timer = static_cast<TimerEvent*>(context);
+  if (!timer->task->Run())
+    timer->task.release();
+  QueueContext* ctx =
+      static_cast<QueueContext*>(pthread_getspecific(GetQueuePtrTls()));
+  ctx->pending_timers_.remove(timer);
+  delete timer;
+}
+
+void TaskQueue::PrepareReplyTask(PostAndReplyTask* reply_task) {
+  RTC_DCHECK(reply_task);
+  CritScope lock(&pending_lock_);
+  pending_replies_.push_back(reply_task);
+}
+
+void TaskQueue::ReplyTaskDone(PostAndReplyTask* reply_task) {
+  CritScope lock(&pending_lock_);
+  pending_replies_.remove(reply_task);
+}
+
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/task_queue_posix.cc b/chromium/third_party/webrtc/base/task_queue_posix.cc
new file mode 100644
index 00000000000..3b00ac8e12d
--- /dev/null
+++ b/chromium/third_party/webrtc/base/task_queue_posix.cc
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/task_queue_posix.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/task_queue.h"
+
+namespace rtc {
+namespace internal {
+pthread_key_t g_queue_ptr_tls = 0;
+
+void InitializeTls() {
+  RTC_CHECK(pthread_key_create(&g_queue_ptr_tls, nullptr) == 0);
+}
+
+pthread_key_t GetQueuePtrTls() {
+  static pthread_once_t init_once = PTHREAD_ONCE_INIT;
+  RTC_CHECK(pthread_once(&init_once, &InitializeTls) == 0);
+  return g_queue_ptr_tls;
+}
+
+AutoSetCurrentQueuePtr::AutoSetCurrentQueuePtr(TaskQueue* q)
+    : prev_(TaskQueue::Current()) {
+  pthread_setspecific(GetQueuePtrTls(), q);
+}
+
+AutoSetCurrentQueuePtr::~AutoSetCurrentQueuePtr() {
+  pthread_setspecific(GetQueuePtrTls(), prev_);
+}
+
+}  // namespace internal
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/task_queue_posix.h b/chromium/third_party/webrtc/base/task_queue_posix.h
new file mode 100644
index 00000000000..b677b78a38c
--- /dev/null
+++ b/chromium/third_party/webrtc/base/task_queue_posix.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_TASK_QUEUE_POSIX_H_
+#define WEBRTC_BASE_TASK_QUEUE_POSIX_H_
+
+#include <pthread.h>
+
+namespace rtc {
+
+class TaskQueue;
+
+namespace internal {
+
+class AutoSetCurrentQueuePtr {
+ public:
+  explicit AutoSetCurrentQueuePtr(TaskQueue* q);
+  ~AutoSetCurrentQueuePtr();
+
+ private:
+  TaskQueue* const prev_;
+};
+
+pthread_key_t GetQueuePtrTls();
+
+}  // namespace internal
+}  // namespace rtc
+
+#endif  // WEBRTC_BASE_TASK_QUEUE_POSIX_H_
diff --git a/chromium/third_party/webrtc/base/task_queue_unittest.cc b/chromium/third_party/webrtc/base/task_queue_unittest.cc
new file mode 100644
index 00000000000..db4e6c2f7ed
--- /dev/null
+++ b/chromium/third_party/webrtc/base/task_queue_unittest.cc
@@ -0,0 +1,261 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/timeutils.h"
+
+namespace rtc {
+
+namespace {
+void CheckCurrent(const char* expected_queue, Event* signal, TaskQueue* queue) {
+  EXPECT_TRUE(TaskQueue::IsCurrent(expected_queue));
+  EXPECT_TRUE(queue->IsCurrent());
+  if (signal)
+    signal->Set();
+}
+
+}  // namespace
+
+TEST(TaskQueueTest, Construct) {
+  static const char kQueueName[] = "Construct";
+  TaskQueue queue(kQueueName);
+  EXPECT_FALSE(queue.IsCurrent());
+}
+
+TEST(TaskQueueTest, PostAndCheckCurrent) {
+  static const char kQueueName[] = "PostAndCheckCurrent";
+  TaskQueue queue(kQueueName);
+
+  // We're not running a task, so there shouldn't be a current queue.
+  EXPECT_FALSE(queue.IsCurrent());
+  EXPECT_FALSE(TaskQueue::Current());
+
+  Event event(false, false);
+  queue.PostTask(Bind(&CheckCurrent, kQueueName, &event, &queue));
+  EXPECT_TRUE(event.Wait(1000));
+}
+
+TEST(TaskQueueTest, PostCustomTask) {
+  static const char kQueueName[] = "PostCustomImplementation";
+  TaskQueue queue(kQueueName);
+
+  Event event(false, false);
+
+  class CustomTask : public QueuedTask {
+   public:
+    explicit CustomTask(Event* event) : event_(event) {}
+
+   private:
+    bool Run() override {
+      event_->Set();
+      return false;  // Never allows the task to be deleted by the queue.
+    }
+
+    Event* const event_;
+  } my_task(&event);
+
+  // Please don't do this in production code! :)
+  queue.PostTask(std::unique_ptr<QueuedTask>(&my_task));
+  EXPECT_TRUE(event.Wait(1000));
+}
+
+TEST(TaskQueueTest, PostLambda) {
+  static const char kQueueName[] = "PostLambda";
+  TaskQueue queue(kQueueName);
+
+  Event event(false, false);
+  queue.PostTask([&event]() { event.Set(); });
+  EXPECT_TRUE(event.Wait(1000));
+}
+
+TEST(TaskQueueTest, PostFromQueue) {
+  static const char kQueueName[] = "PostFromQueue";
+  TaskQueue queue(kQueueName);
+
+  Event event(false, false);
+  queue.PostTask(
+      [&event, &queue]() { queue.PostTask([&event]() { event.Set(); }); });
+  EXPECT_TRUE(event.Wait(1000));
+}
+
+TEST(TaskQueueTest, PostDelayed) {
+  static const char kQueueName[] = "PostDelayed";
+  TaskQueue queue(kQueueName);
+
+  Event event(false, false);
+  uint32_t start = Time();
+  queue.PostDelayedTask(Bind(&CheckCurrent, kQueueName, &event, &queue), 100);
+  EXPECT_TRUE(event.Wait(1000));
+  uint32_t end = Time();
+  EXPECT_GE(end - start, 100u);
+  EXPECT_NEAR(end - start, 200u, 100u);  // Accept 100-300.
+}
+
+TEST(TaskQueueTest, PostMultipleDelayed) {
+  static const char kQueueName[] = "PostMultipleDelayed";
+  TaskQueue queue(kQueueName);
+
+  std::vector<std::unique_ptr<Event>> events;
+  for (int i = 0; i < 10; ++i) {
+    events.push_back(std::unique_ptr<Event>(new Event(false, false)));
+    queue.PostDelayedTask(
+        Bind(&CheckCurrent, kQueueName, events.back().get(), &queue), 10);
+  }
+
+  for (const auto& e : events)
+    EXPECT_TRUE(e->Wait(100));
+}
+
+TEST(TaskQueueTest, PostDelayedAfterDestruct) {
+  static const char kQueueName[] = "PostDelayedAfterDestruct";
+  Event event(false, false);
+  {
+    TaskQueue queue(kQueueName);
+    queue.PostDelayedTask(Bind(&CheckCurrent, kQueueName, &event, &queue), 100);
+  }
+  EXPECT_FALSE(event.Wait(200));  // Task should not run.
+}
+
+TEST(TaskQueueTest, PostAndReply) {
+  static const char kPostQueue[] = "PostQueue";
+  static const char kReplyQueue[] = "ReplyQueue";
+  TaskQueue post_queue(kPostQueue);
+  TaskQueue reply_queue(kReplyQueue);
+
+  Event event(false, false);
+  post_queue.PostTaskAndReply(
+      Bind(&CheckCurrent, kPostQueue, nullptr, &post_queue),
+      Bind(&CheckCurrent, kReplyQueue, &event, &reply_queue), &reply_queue);
+  EXPECT_TRUE(event.Wait(1000));
+}
+
+TEST(TaskQueueTest, PostAndReuse) {
+  static const char kPostQueue[] = "PostQueue";
+  static const char kReplyQueue[] = "ReplyQueue";
+  TaskQueue post_queue(kPostQueue);
+  TaskQueue reply_queue(kReplyQueue);
+
+  int call_count = 0;
+
+  class ReusedTask : public QueuedTask {
+   public:
+    ReusedTask(int* counter, TaskQueue* reply_queue, Event* event)
+        : counter_(counter), reply_queue_(reply_queue), event_(event) {
+      EXPECT_EQ(0, *counter_);
+    }
+
+   private:
+    bool Run() override {
+      if (++(*counter_) == 1) {
+        std::unique_ptr<QueuedTask> myself(this);
+        reply_queue_->PostTask(std::move(myself));
+        // At this point, the object is owned by reply_queue_ and it's
+        // theoratically possible that the object has been deleted (e.g. if
+        // posting wasn't possible).  So, don't touch any member variables here.
+
+        // Indicate to the current queue that ownership has been transferred.
+        return false;
+      } else {
+        EXPECT_EQ(2, *counter_);
+        EXPECT_TRUE(reply_queue_->IsCurrent());
+        event_->Set();
+        return true;  // Indicate that the object should be deleted.
+      }
+    }
+
+    int* const counter_;
+    TaskQueue* const reply_queue_;
+    Event* const event_;
+  };
+
+  Event event(false, false);
+  std::unique_ptr<QueuedTask> task(
+      new ReusedTask(&call_count, &reply_queue, &event));
+
+  post_queue.PostTask(std::move(task));
+  EXPECT_TRUE(event.Wait(1000));
+}
+
+TEST(TaskQueueTest, PostAndReplyLambda) {
+  static const char kPostQueue[] = "PostQueue";
+  static const char kReplyQueue[] = "ReplyQueue";
+  TaskQueue post_queue(kPostQueue);
+  TaskQueue reply_queue(kReplyQueue);
+
+  Event event(false, false);
+  bool my_flag = false;
+  post_queue.PostTaskAndReply([&my_flag]() { my_flag = true; },
+                              [&event]() { event.Set(); }, &reply_queue);
+  EXPECT_TRUE(event.Wait(1000));
+  EXPECT_TRUE(my_flag);
+}
+
+void TestPostTaskAndReply(TaskQueue* work_queue,
+                          const char* work_queue_name,
+                          Event* event) {
+  ASSERT_FALSE(work_queue->IsCurrent());
+  work_queue->PostTaskAndReply(
+      Bind(&CheckCurrent, work_queue_name, nullptr, work_queue),
+      NewClosure([event]() { event->Set(); }));
+}
+
+// Does a PostTaskAndReply from within a task to post and reply to the current
+// queue.  All in all there will be 3 tasks posted and run.
+TEST(TaskQueueTest, PostAndReply2) {
+  static const char kQueueName[] = "PostAndReply2";
+  static const char kWorkQueueName[] = "PostAndReply2_Worker";
+  TaskQueue queue(kQueueName);
+  TaskQueue work_queue(kWorkQueueName);
+
+  Event event(false, false);
+  queue.PostTask(
+      Bind(&TestPostTaskAndReply, &work_queue, kWorkQueueName, &event));
+  EXPECT_TRUE(event.Wait(1000));
+}
+
+// Tests posting more messages than a queue can queue up.
+// In situations like that, tasks will get dropped.
+TEST(TaskQueueTest, PostALot) {
+  // To destruct the event after the queue has gone out of scope.
+  Event event(false, false);
+
+  int tasks_executed = 0;
+  int tasks_cleaned_up = 0;
+  static const int kTaskCount = 0xffff;
+
+  {
+    static const char kQueueName[] = "PostALot";
+    TaskQueue queue(kQueueName);
+
+    // On linux, the limit of pending bytes in the pipe buffer is 0xffff.
+    // So here we post a total of 0xffff+1 messages, which triggers a failure
+    // case inside of the libevent queue implementation.
+
+    queue.PostTask([&event]() { event.Wait(Event::kForever); });
+    for (int i = 0; i < kTaskCount; ++i)
+      queue.PostTask(NewClosure([&tasks_executed]() { ++tasks_executed; },
+                                [&tasks_cleaned_up]() { ++tasks_cleaned_up; }));
+    event.Set();  // Unblock the first task.
+  }
+
+  EXPECT_GE(tasks_cleaned_up, tasks_executed);
+  EXPECT_EQ(kTaskCount, tasks_cleaned_up);
+
+  LOG(INFO) << "tasks executed: " << tasks_executed
+            << ", tasks cleaned up: " << tasks_cleaned_up;
+}
+
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/task_queue_win.cc b/chromium/third_party/webrtc/base/task_queue_win.cc
new file mode 100644
index 00000000000..5ae6d9275ba
--- /dev/null
+++ b/chromium/third_party/webrtc/base/task_queue_win.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/task_queue.h"
+
+#include <string.h>
+#include <unordered_map>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
+namespace rtc {
+namespace {
+#define WM_RUN_TASK WM_USER + 1
+#define WM_QUEUE_DELAYED_TASK WM_USER + 2
+
+DWORD g_queue_ptr_tls = 0;
+
+BOOL CALLBACK InitializeTls(PINIT_ONCE init_once, void* param, void** context) {
+  g_queue_ptr_tls = TlsAlloc();
+  return TRUE;
+}
+
+DWORD GetQueuePtrTls() {
+  static INIT_ONCE init_once = INIT_ONCE_STATIC_INIT;
+  InitOnceExecuteOnce(&init_once, InitializeTls, nullptr, nullptr);
+  return g_queue_ptr_tls;
+}
+
+struct ThreadStartupData {
+  Event* started;
+  void* thread_context;
+};
+
+void CALLBACK InitializeQueueThread(ULONG_PTR param) {
+  MSG msg;
+  PeekMessage(&msg, NULL, WM_USER, WM_USER, PM_NOREMOVE);
+  ThreadStartupData* data = reinterpret_cast<ThreadStartupData*>(param);
+  TlsSetValue(GetQueuePtrTls(), data->thread_context);
+  data->started->Set();
+}
+}  // namespace
+
+TaskQueue::TaskQueue(const char* queue_name)
+    : thread_(&TaskQueue::ThreadMain, this, queue_name) {
+  RTC_DCHECK(queue_name);
+  thread_.Start();
+  Event event(false, false);
+  ThreadStartupData startup = {&event, this};
+  RTC_CHECK(thread_.QueueAPC(&InitializeQueueThread,
+                             reinterpret_cast<ULONG_PTR>(&startup)));
+  event.Wait(Event::kForever);
+}
+
+TaskQueue::~TaskQueue() {
+  RTC_DCHECK(!IsCurrent());
+  while (!PostThreadMessage(thread_.GetThreadRef(), WM_QUIT, 0, 0)) {
+    RTC_CHECK(ERROR_NOT_ENOUGH_QUOTA == ::GetLastError());
+    Sleep(1);
+  }
+  thread_.Stop();
+}
+
+// static
+TaskQueue* TaskQueue::Current() {
+  return static_cast<TaskQueue*>(TlsGetValue(GetQueuePtrTls()));
+}
+
+// static
+bool TaskQueue::IsCurrent(const char* queue_name) {
+  TaskQueue* current = Current();
+  return current && current->thread_.name().compare(queue_name) == 0;
+}
+
+bool TaskQueue::IsCurrent() const {
+  return IsThreadRefEqual(thread_.GetThreadRef(), CurrentThreadRef());
+}
+
+void TaskQueue::PostTask(std::unique_ptr<QueuedTask> task) {
+  if (PostThreadMessage(thread_.GetThreadRef(), WM_RUN_TASK, 0,
+                        reinterpret_cast<LPARAM>(task.get()))) {
+    task.release();
+  }
+}
+
+void TaskQueue::PostDelayedTask(std::unique_ptr<QueuedTask> task,
+                                uint32_t milliseconds) {
+  WPARAM wparam;
+#if defined(_WIN64)
+  // GetTickCount() returns a fairly coarse tick count (resolution or about 8ms)
+  // so this compensation isn't that accurate, but since we have unused 32 bits
+  // on Win64, we might as well use them.
+  wparam = (static_cast<WPARAM>(::GetTickCount()) << 32) | milliseconds;
+#else
+  wparam = milliseconds;
+#endif
+  if (PostThreadMessage(thread_.GetThreadRef(), WM_QUEUE_DELAYED_TASK, wparam,
+                        reinterpret_cast<LPARAM>(task.get()))) {
+    task.release();
+  }
+}
+
+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                                 std::unique_ptr<QueuedTask> reply,
+                                 TaskQueue* reply_queue) {
+  QueuedTask* task_ptr = task.release();
+  QueuedTask* reply_task_ptr = reply.release();
+  DWORD reply_thread_id = reply_queue->thread_.GetThreadRef();
+  PostTask([task_ptr, reply_task_ptr, reply_thread_id]() {
+    if (task_ptr->Run())
+      delete task_ptr;
+    // If the thread's message queue is full, we can't queue the task and will
+    // have to drop it (i.e. delete).
+    if (!PostThreadMessage(reply_thread_id, WM_RUN_TASK, 0,
+                           reinterpret_cast<LPARAM>(reply_task_ptr))) {
+      delete reply_task_ptr;
+    }
+  });
+}
+
+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
+                                 std::unique_ptr<QueuedTask> reply) {
+  return PostTaskAndReply(std::move(task), std::move(reply), Current());
+}
+
+// static
+bool TaskQueue::ThreadMain(void* context) {
+  std::unordered_map<UINT_PTR, std::unique_ptr<QueuedTask>> delayed_tasks;
+
+  BOOL ret;
+  MSG msg;
+
+  while ((ret = GetMessage(&msg, nullptr, 0, 0)) != 0 && ret != -1) {
+    if (!msg.hwnd) {
+      switch (msg.message) {
+        case WM_RUN_TASK: {
+          QueuedTask* task = reinterpret_cast<QueuedTask*>(msg.lParam);
+          if (task->Run())
+            delete task;
+          break;
+        }
+        case WM_QUEUE_DELAYED_TASK: {
+          QueuedTask* task = reinterpret_cast<QueuedTask*>(msg.lParam);
+          uint32_t milliseconds = msg.wParam & 0xFFFFFFFF;
+#if defined(_WIN64)
+          // Subtract the time it took to queue the timer.
+          const DWORD now = GetTickCount();
+          DWORD post_time = now - (msg.wParam >> 32);
+          milliseconds =
+              post_time > milliseconds ? 0 : milliseconds - post_time;
+#endif
+          UINT_PTR timer_id = SetTimer(nullptr, 0, milliseconds, nullptr);
+          delayed_tasks.insert(std::make_pair(timer_id, task));
+          break;
+        }
+        case WM_TIMER: {
+          KillTimer(nullptr, msg.wParam);
+          auto found = delayed_tasks.find(msg.wParam);
+          RTC_DCHECK(found != delayed_tasks.end());
+          if (!found->second->Run())
+            found->second.release();
+          delayed_tasks.erase(found);
+          break;
+        }
+        default:
+          RTC_NOTREACHED();
+          break;
+      }
+    } else {
+      TranslateMessage(&msg);
+      DispatchMessage(&msg);
+    }
+  }
+
+  return false;
+}
+}  // namespace rtc
diff --git a/chromium/third_party/webrtc/base/task_unittest.cc b/chromium/third_party/webrtc/base/task_unittest.cc
index 7492436a5d1..f3ccc24a8df 100644
--- a/chromium/third_party/webrtc/base/task_unittest.cc
+++ b/chromium/third_party/webrtc/base/task_unittest.cc
@@ -22,6 +22,7 @@
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task.h"
@@ -32,7 +33,7 @@
 namespace rtc {
 
 static int64_t GetCurrentTime() {
-  return static_cast<int64_t>(Time()) * 10000;
+  return TimeMillis() * 10000;
 }
 
 // feel free to change these numbers.  Note that '0' won't work, though
diff --git a/chromium/third_party/webrtc/base/taskparent.h b/chromium/third_party/webrtc/base/taskparent.h
index 41008fa98ee..3c0b81cf6a9 100644
--- a/chromium/third_party/webrtc/base/taskparent.h
+++ b/chromium/third_party/webrtc/base/taskparent.h
@@ -11,10 +11,11 @@
 #ifndef WEBRTC_BASE_TASKPARENT_H__
 #define WEBRTC_BASE_TASKPARENT_H__
 
+#include <memory>
 #include <set>
 
 #include "webrtc/base/basictypes.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 
 namespace rtc {
 
@@ -52,7 +53,7 @@ class TaskParent {
   TaskRunner *runner_;
   bool child_error_;
   typedef std::set<Task *> ChildSet;
-  scoped_ptr<ChildSet> children_;
+  std::unique_ptr<ChildSet> children_;
   RTC_DISALLOW_COPY_AND_ASSIGN(TaskParent);
 };
 
diff --git a/chromium/third_party/webrtc/base/taskrunner.cc b/chromium/third_party/webrtc/base/taskrunner.cc
index c50c9f833ec..73916a07194 100644
--- a/chromium/third_party/webrtc/base/taskrunner.cc
+++ b/chromium/third_party/webrtc/base/taskrunner.cc
@@ -13,7 +13,6 @@
 #include "webrtc/base/taskrunner.h"
 
 #include "webrtc/base/common.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/task.h"
 #include "webrtc/base/logging.h"
 
diff --git a/chromium/third_party/webrtc/base/testclient.h b/chromium/third_party/webrtc/base/testclient.h
index 5d8ee98a6fb..df831fefb75 100644
--- a/chromium/third_party/webrtc/base/testclient.h
+++ b/chromium/third_party/webrtc/base/testclient.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 #include "webrtc/base/asyncudpsocket.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 
 namespace rtc {
diff --git a/chromium/third_party/webrtc/base/testechoserver.h b/chromium/third_party/webrtc/base/testechoserver.h
index 51d7d539e4f..76d714b4f4a 100644
--- a/chromium/third_party/webrtc/base/testechoserver.h
+++ b/chromium/third_party/webrtc/base/testechoserver.h
@@ -12,7 +12,9 @@
 #define WEBRTC_BASE_TESTECHOSERVER_H_
 
 #include <list>
+#include <memory>
 #include "webrtc/base/asynctcpsocket.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
@@ -63,7 +65,7 @@ class TestEchoServer : public sigslot::has_slots<> {
   }
 
   typedef std::list<AsyncTCPSocket*> ClientList;
-  scoped_ptr<AsyncSocket> server_socket_;
+  std::unique_ptr<AsyncSocket> server_socket_;
   ClientList client_sockets_;
   RTC_DISALLOW_COPY_AND_ASSIGN(TestEchoServer);
 };
diff --git a/chromium/third_party/webrtc/base/testutils.h b/chromium/third_party/webrtc/base/testutils.h
index 6e7e22a928c..e5e571b9bf9 100644
--- a/chromium/third_party/webrtc/base/testutils.h
+++ b/chromium/third_party/webrtc/base/testutils.h
@@ -24,6 +24,7 @@
 
 #include <algorithm>
 #include <map>
+#include <memory>
 #include <vector>
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/asyncsocket.h"
@@ -371,7 +372,7 @@ private:
   void OnCloseEvent(AsyncSocket* socket, int error) {
   }
 
-  scoped_ptr<AsyncSocket> socket_;
+  std::unique_ptr<AsyncSocket> socket_;
   Buffer send_buffer_, recv_buffer_;
 };
 
@@ -414,7 +415,7 @@ class SocketTestServer : public sigslot::has_slots<> {
     clients_.push_back(new SocketTestClient(accepted));
   }
 
-  scoped_ptr<AsyncSocket> socket_;
+  std::unique_ptr<AsyncSocket> socket_;
   std::vector<SocketTestClient*> clients_;
 };
 
diff --git a/chromium/third_party/webrtc/base/thread.cc b/chromium/third_party/webrtc/base/thread.cc
index dc8ccdfd4dd..3971169ae7d 100644
--- a/chromium/third_party/webrtc/base/thread.cc
+++ b/chromium/third_party/webrtc/base/thread.cc
@@ -22,6 +22,7 @@
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/nullsocketserver.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
@@ -138,7 +139,9 @@ Thread::ScopedDisallowBlockingCalls::~ScopedDisallowBlockingCalls() {
   thread_->SetAllowBlockingCalls(previous_state_);
 }
 
-Thread::Thread(SocketServer* ss, bool init_queue)
+Thread::Thread() : Thread(SocketServer::CreateDefault()) {}
+
+Thread::Thread(SocketServer* ss)
     : MessageQueue(ss, false),
       running_(true, false),
 #if defined(WEBRTC_WIN)
@@ -148,9 +151,20 @@ Thread::Thread(SocketServer* ss, bool init_queue)
       owned_(true),
       blocking_calls_allowed_(true) {
   SetName("Thread", this);  // default name
-  if (init_queue) {
-    DoInit();
-  }
+  DoInit();
+}
+
+Thread::Thread(std::unique_ptr<SocketServer> ss)
+    : MessageQueue(std::move(ss), false),
+      running_(true, false),
+#if defined(WEBRTC_WIN)
+      thread_(NULL),
+      thread_id_(0),
+#endif
+      owned_(true),
+      blocking_calls_allowed_(true) {
+  SetName("Thread", this);  // default name
+  DoInit();
 }
 
 Thread::~Thread() {
@@ -158,6 +172,15 @@ Thread::~Thread() {
   DoDestroy();
 }
 
+std::unique_ptr<Thread> Thread::CreateWithSocketServer() {
+  return std::unique_ptr<Thread>(new Thread(SocketServer::CreateDefault()));
+}
+
+std::unique_ptr<Thread> Thread::Create() {
+  return std::unique_ptr<Thread>(
+      new Thread(std::unique_ptr<SocketServer>(new NullSocketServer())));
+}
+
 bool Thread::SleepMs(int milliseconds) {
   AssertBlockingIsAllowedOnCurrentThread();
 
@@ -459,7 +482,7 @@ void Thread::Clear(MessageHandler* phandler,
 }
 
 bool Thread::ProcessMessages(int cmsLoop) {
-  uint32_t msEnd = (kForever == cmsLoop) ? 0 : TimeAfter(cmsLoop);
+  int64_t msEnd = (kForever == cmsLoop) ? 0 : TimeAfter(cmsLoop);
   int cmsNext = cmsLoop;
 
   while (true) {
@@ -513,7 +536,7 @@ bool Thread::WrapCurrentWithThreadManager(ThreadManager* thread_manager,
   return true;
 }
 
-AutoThread::AutoThread(SocketServer* ss) : Thread(ss) {
+AutoThread::AutoThread() {
   if (!ThreadManager::Instance()->CurrentThread()) {
     ThreadManager::Instance()->SetCurrentThread(this);
   }
diff --git a/chromium/third_party/webrtc/base/thread.h b/chromium/third_party/webrtc/base/thread.h
index df5a686952f..1a0c447264e 100644
--- a/chromium/third_party/webrtc/base/thread.h
+++ b/chromium/third_party/webrtc/base/thread.h
@@ -13,6 +13,7 @@
 
 #include <algorithm>
 #include <list>
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -95,11 +96,9 @@ class Runnable {
 class Thread : public MessageQueue {
  public:
   // Create a new Thread and optionally assign it to the passed SocketServer.
-  // Subclasses that override Clear should pass false for init_queue and call
-  // DoInit() from their constructor to prevent races with the
-  // MessageQueueManager already using the object while the vtable is still
-  // being created.
-  explicit Thread(SocketServer* ss = nullptr, bool init_queue = true);
+  Thread();
+  explicit Thread(SocketServer* ss);
+  explicit Thread(std::unique_ptr<SocketServer> ss);
 
   // NOTE: ALL SUBCLASSES OF Thread MUST CALL Stop() IN THEIR DESTRUCTORS (or
   // guarantee Stop() is explicitly called before the subclass is destroyed).
@@ -107,6 +106,8 @@ class Thread : public MessageQueue {
   // vtable, and the Thread::PreRun calling the virtual method Run().
   ~Thread() override;
 
+  static std::unique_ptr<Thread> CreateWithSocketServer();
+  static std::unique_ptr<Thread> Create();
   static Thread* Current();
 
   // Used to catch performance regressions. Use this to disallow blocking calls
@@ -291,7 +292,7 @@ class Thread : public MessageQueue {
 
 class AutoThread : public Thread {
  public:
-  explicit AutoThread(SocketServer* ss = nullptr);
+  AutoThread();
   ~AutoThread() override;
 
  private:
diff --git a/chromium/third_party/webrtc/base/thread_checker_unittest.cc b/chromium/third_party/webrtc/base/thread_checker_unittest.cc
index 338190093d0..372f6f4a77f 100644
--- a/chromium/third_party/webrtc/base/thread_checker_unittest.cc
+++ b/chromium/third_party/webrtc/base/thread_checker_unittest.cc
@@ -10,11 +10,13 @@
 
 // Borrowed from Chromium's src/base/threading/thread_checker_unittest.cc.
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/base/scoped_ptr.h"
 
 // Duplicated from base/threading/thread_checker.h so that we can be
 // good citizens there and undef the macro.
@@ -91,7 +93,7 @@ class DeleteThreadCheckerClassOnThread : public Thread {
   }
 
  private:
-  scoped_ptr<ThreadCheckerClass> thread_checker_class_;
+  std::unique_ptr<ThreadCheckerClass> thread_checker_class_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DeleteThreadCheckerClassOnThread);
 };
@@ -99,7 +101,7 @@ class DeleteThreadCheckerClassOnThread : public Thread {
 }  // namespace
 
 TEST(ThreadCheckerTest, CallsAllowedOnSameThread) {
-  scoped_ptr<ThreadCheckerClass> thread_checker_class(
+  std::unique_ptr<ThreadCheckerClass> thread_checker_class(
       new ThreadCheckerClass);
 
   // Verify that DoStuff doesn't assert.
@@ -110,7 +112,7 @@ TEST(ThreadCheckerTest, CallsAllowedOnSameThread) {
 }
 
 TEST(ThreadCheckerTest, DestructorAllowedOnDifferentThread) {
-  scoped_ptr<ThreadCheckerClass> thread_checker_class(
+  std::unique_ptr<ThreadCheckerClass> thread_checker_class(
       new ThreadCheckerClass);
 
   // Verify that the destructor doesn't assert
@@ -123,7 +125,7 @@ TEST(ThreadCheckerTest, DestructorAllowedOnDifferentThread) {
 }
 
 TEST(ThreadCheckerTest, DetachFromThread) {
-  scoped_ptr<ThreadCheckerClass> thread_checker_class(
+  std::unique_ptr<ThreadCheckerClass> thread_checker_class(
       new ThreadCheckerClass);
 
   // Verify that DoStuff doesn't assert when called on a different thread after
@@ -138,7 +140,7 @@ TEST(ThreadCheckerTest, DetachFromThread) {
 #if GTEST_HAS_DEATH_TEST || !ENABLE_THREAD_CHECKER
 
 void ThreadCheckerClass::MethodOnDifferentThreadImpl() {
-  scoped_ptr<ThreadCheckerClass> thread_checker_class(
+  std::unique_ptr<ThreadCheckerClass> thread_checker_class(
       new ThreadCheckerClass);
 
   // DoStuff should assert in debug builds only when called on a
@@ -162,7 +164,7 @@ TEST(ThreadCheckerTest, MethodAllowedOnDifferentThreadInRelease) {
 #endif  // ENABLE_THREAD_CHECKER
 
 void ThreadCheckerClass::DetachThenCallFromDifferentThreadImpl() {
-  scoped_ptr<ThreadCheckerClass> thread_checker_class(
+  std::unique_ptr<ThreadCheckerClass> thread_checker_class(
       new ThreadCheckerClass);
 
   // DoStuff doesn't assert when called on a different thread
diff --git a/chromium/third_party/webrtc/base/thread_unittest.cc b/chromium/third_party/webrtc/base/thread_unittest.cc
index 7889e29da8d..bf3cbd08963 100644
--- a/chromium/third_party/webrtc/base/thread_unittest.cc
+++ b/chromium/third_party/webrtc/base/thread_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/event.h"
@@ -589,17 +591,18 @@ class GuardedAsyncInvokeTest : public testing::Test {
 
 // Functor for creating an invoker.
 struct CreateInvoker {
-  CreateInvoker(scoped_ptr<GuardedAsyncInvoker>* invoker) : invoker_(invoker) {}
+  CreateInvoker(std::unique_ptr<GuardedAsyncInvoker>* invoker)
+      : invoker_(invoker) {}
   void operator()() { invoker_->reset(new GuardedAsyncInvoker()); }
-  scoped_ptr<GuardedAsyncInvoker>* invoker_;
+  std::unique_ptr<GuardedAsyncInvoker>* invoker_;
 };
 
 // Test that we can call AsyncInvoke<void>() after the thread died.
 TEST_F(GuardedAsyncInvokeTest, KillThreadFireAndForget) {
   // Create and start the thread.
-  scoped_ptr<Thread> thread(new Thread());
+  std::unique_ptr<Thread> thread(new Thread());
   thread->Start();
-  scoped_ptr<GuardedAsyncInvoker> invoker;
+  std::unique_ptr<GuardedAsyncInvoker> invoker;
   // Create the invoker on |thread|.
   thread->Invoke<void>(CreateInvoker(&invoker));
   // Kill |thread|.
@@ -615,9 +618,9 @@ TEST_F(GuardedAsyncInvokeTest, KillThreadFireAndForget) {
 // Test that we can call AsyncInvoke with callback after the thread died.
 TEST_F(GuardedAsyncInvokeTest, KillThreadWithCallback) {
   // Create and start the thread.
-  scoped_ptr<Thread> thread(new Thread());
+  std::unique_ptr<Thread> thread(new Thread());
   thread->Start();
-  scoped_ptr<GuardedAsyncInvoker> invoker;
+  std::unique_ptr<GuardedAsyncInvoker> invoker;
   // Create the invoker on |thread|.
   thread->Invoke<void>(CreateInvoker(&invoker));
   // Kill |thread|.
diff --git a/chromium/third_party/webrtc/base/timeutils.cc b/chromium/third_party/webrtc/base/timeutils.cc
index de3e6afb28d..a9fe49d6075 100644
--- a/chromium/third_party/webrtc/base/timeutils.cc
+++ b/chromium/third_party/webrtc/base/timeutils.cc
@@ -30,8 +30,6 @@
 
 namespace rtc {
 
-const uint32_t HALF = 0x80000000;
-
 uint64_t TimeNanos() {
   int64_t ticks = 0;
 #if defined(WEBRTC_MAC)
@@ -82,7 +80,7 @@ uint32_t Time32() {
   return static_cast<uint32_t>(TimeNanos() / kNumNanosecsPerMillisec);
 }
 
-int64_t Time64() {
+int64_t TimeMillis() {
   return static_cast<int64_t>(TimeNanos() / kNumNanosecsPerMillisec);
 }
 
@@ -90,27 +88,16 @@ uint64_t TimeMicros() {
   return static_cast<uint64_t>(TimeNanos() / kNumNanosecsPerMicrosec);
 }
 
-uint32_t TimeAfter(int32_t elapsed) {
+int64_t TimeAfter(int64_t elapsed) {
   RTC_DCHECK_GE(elapsed, 0);
-  RTC_DCHECK_LT(static_cast<uint32_t>(elapsed), HALF);
-  return Time() + elapsed;
-}
-
-bool TimeIsLaterOrEqual(uint32_t earlier, uint32_t later) {
-  int32_t diff = later - earlier;
-  return (diff >= 0 && static_cast<uint32_t>(diff) < HALF);
-}
-
-bool TimeIsLater(uint32_t earlier, uint32_t later) {
-  int32_t diff = later - earlier;
-  return (diff > 0 && static_cast<uint32_t>(diff) < HALF);
+  return TimeMillis() + elapsed;
 }
 
-int32_t TimeDiff(uint32_t later, uint32_t earlier) {
+int32_t TimeDiff32(uint32_t later, uint32_t earlier) {
   return later - earlier;
 }
 
-int64_t TimeDiff64(int64_t later, int64_t earlier) {
+int64_t TimeDiff(int64_t later, int64_t earlier) {
   return later - earlier;
 }
 
diff --git a/chromium/third_party/webrtc/base/timeutils.h b/chromium/third_party/webrtc/base/timeutils.h
index 4af0d9538f0..222d5c26e47 100644
--- a/chromium/third_party/webrtc/base/timeutils.h
+++ b/chromium/third_party/webrtc/base/timeutils.h
@@ -29,58 +29,40 @@ static const int64_t kNumNanosecsPerMillisec =
 static const int64_t kNumNanosecsPerMicrosec =
     kNumNanosecsPerSec / kNumMicrosecsPerSec;
 
-typedef uint32_t TimeStamp;
+// TODO(honghaiz): Define a type for the time value specifically.
 
 // Returns the current time in milliseconds in 32 bits.
 uint32_t Time32();
 
 // Returns the current time in milliseconds in 64 bits.
-int64_t Time64();
-
-// Returns the current time in milliseconds.
-// TODO(honghaiz): Returns Time64 once majority of the webrtc code migrates to
-// 64-bit timestamp.
-inline uint32_t Time() {
-  return Time32();
+int64_t TimeMillis();
+// Deprecated. Do not use this in any new code.
+inline int64_t Time() {
+  return TimeMillis();
 }
 
 // Returns the current time in microseconds.
 uint64_t TimeMicros();
+
 // Returns the current time in nanoseconds.
 uint64_t TimeNanos();
 
 // Returns a future timestamp, 'elapsed' milliseconds from now.
-uint32_t TimeAfter(int32_t elapsed);
-
-bool TimeIsLaterOrEqual(uint32_t earlier, uint32_t later);  // Inclusive
-bool TimeIsLater(uint32_t earlier, uint32_t later);         // Exclusive
-
-// Returns the later of two timestamps.
-inline uint32_t TimeMax(uint32_t ts1, uint32_t ts2) {
-  return TimeIsLaterOrEqual(ts1, ts2) ? ts2 : ts1;
-}
-
-// Returns the earlier of two timestamps.
-inline uint32_t TimeMin(uint32_t ts1, uint32_t ts2) {
-  return TimeIsLaterOrEqual(ts1, ts2) ? ts1 : ts2;
-}
-
-// Number of milliseconds that would elapse between 'earlier' and 'later'
-// timestamps.  The value is negative if 'later' occurs before 'earlier'.
-int32_t TimeDiff(uint32_t later, uint32_t earlier);
+int64_t TimeAfter(int64_t elapsed);
 
 // Number of milliseconds that would elapse between 'earlier' and 'later'
 // timestamps.  The value is negative if 'later' occurs before 'earlier'.
-int64_t TimeDiff64(int64_t later, int64_t earlier);
+int64_t TimeDiff(int64_t later, int64_t earlier);
+int32_t TimeDiff32(uint32_t later, uint32_t earlier);
 
 // The number of milliseconds that have elapsed since 'earlier'.
-inline int32_t TimeSince(uint32_t earlier) {
-  return TimeDiff(Time(), earlier);
+inline int64_t TimeSince(int64_t earlier) {
+  return TimeMillis() - earlier;
 }
 
 // The number of milliseconds that will elapse between now and 'later'.
-inline int32_t TimeUntil(uint32_t later) {
-  return TimeDiff(later, Time());
+inline int64_t TimeUntil(uint64_t later) {
+  return later - TimeMillis();
 }
 
 class TimestampWrapAroundHandler {
diff --git a/chromium/third_party/webrtc/base/timeutils_unittest.cc b/chromium/third_party/webrtc/base/timeutils_unittest.cc
index 61e41b7c644..0971c037fa5 100644
--- a/chromium/third_party/webrtc/base/timeutils_unittest.cc
+++ b/chromium/third_party/webrtc/base/timeutils_unittest.cc
@@ -17,59 +17,21 @@
 namespace rtc {
 
 TEST(TimeTest, TimeInMs) {
-  uint32_t ts_earlier = Time();
+  int64_t ts_earlier = TimeMillis();
   Thread::SleepMs(100);
-  uint32_t ts_now = Time();
+  int64_t ts_now = TimeMillis();
   // Allow for the thread to wakeup ~20ms early.
   EXPECT_GE(ts_now, ts_earlier + 80);
   // Make sure the Time is not returning in smaller unit like microseconds.
   EXPECT_LT(ts_now, ts_earlier + 1000);
 }
 
-TEST(TimeTest, Comparison) {
-  // Obtain two different times, in known order
-  TimeStamp ts_earlier = Time();
-  Thread::SleepMs(100);
-  TimeStamp ts_now = Time();
-  EXPECT_NE(ts_earlier, ts_now);
-
-  // Common comparisons
-  EXPECT_TRUE( TimeIsLaterOrEqual(ts_earlier, ts_now));
-  EXPECT_TRUE( TimeIsLater(       ts_earlier, ts_now));
-  EXPECT_FALSE(TimeIsLaterOrEqual(ts_now,     ts_earlier));
-  EXPECT_FALSE(TimeIsLater(       ts_now,     ts_earlier));
-
-  // Edge cases
-  EXPECT_TRUE( TimeIsLaterOrEqual(ts_earlier, ts_earlier));
-  EXPECT_FALSE(TimeIsLater(       ts_earlier, ts_earlier));
-
-  // Obtain a third time
-  TimeStamp ts_later = TimeAfter(100);
-  EXPECT_NE(ts_now, ts_later);
-  EXPECT_TRUE( TimeIsLater(ts_now,     ts_later));
-  EXPECT_TRUE( TimeIsLater(ts_earlier, ts_later));
-
-  // Earlier of two times
-  EXPECT_EQ(ts_earlier, TimeMin(ts_earlier, ts_earlier));
-  EXPECT_EQ(ts_earlier, TimeMin(ts_earlier, ts_now));
-  EXPECT_EQ(ts_earlier, TimeMin(ts_earlier, ts_later));
-  EXPECT_EQ(ts_earlier, TimeMin(ts_now,     ts_earlier));
-  EXPECT_EQ(ts_earlier, TimeMin(ts_later,   ts_earlier));
-
-  // Later of two times
-  EXPECT_EQ(ts_earlier, TimeMax(ts_earlier, ts_earlier));
-  EXPECT_EQ(ts_now,     TimeMax(ts_earlier, ts_now));
-  EXPECT_EQ(ts_later,   TimeMax(ts_earlier, ts_later));
-  EXPECT_EQ(ts_now,     TimeMax(ts_now,     ts_earlier));
-  EXPECT_EQ(ts_later,   TimeMax(ts_later,   ts_earlier));
-}
-
 TEST(TimeTest, Intervals) {
-  TimeStamp ts_earlier = Time();
-  TimeStamp ts_later = TimeAfter(500);
+  int64_t ts_earlier = TimeMillis();
+  int64_t ts_later = TimeAfter(500);
 
   // We can't depend on ts_later and ts_earlier to be exactly 500 apart
-  // since time elapses between the calls to Time() and TimeAfter(500)
+  // since time elapses between the calls to TimeMillis() and TimeAfter(500)
   EXPECT_LE(500,  TimeDiff(ts_later, ts_earlier));
   EXPECT_GE(-500, TimeDiff(ts_earlier, ts_later));
 
@@ -87,36 +49,9 @@ TEST(TimeTest, Intervals) {
   EXPECT_LE(TimeUntil(ts_later), 500);
 }
 
-TEST(TimeTest, BoundaryComparison) {
-  // Obtain two different times, in known order
-  TimeStamp ts_earlier = static_cast<TimeStamp>(-50);
-  TimeStamp ts_later = ts_earlier + 100;
-  EXPECT_NE(ts_earlier, ts_later);
-
-  // Common comparisons
-  EXPECT_TRUE( TimeIsLaterOrEqual(ts_earlier, ts_later));
-  EXPECT_TRUE( TimeIsLater(       ts_earlier, ts_later));
-  EXPECT_FALSE(TimeIsLaterOrEqual(ts_later,   ts_earlier));
-  EXPECT_FALSE(TimeIsLater(       ts_later,   ts_earlier));
-
-  // Earlier of two times
-  EXPECT_EQ(ts_earlier, TimeMin(ts_earlier, ts_earlier));
-  EXPECT_EQ(ts_earlier, TimeMin(ts_earlier, ts_later));
-  EXPECT_EQ(ts_earlier, TimeMin(ts_later,   ts_earlier));
-
-  // Later of two times
-  EXPECT_EQ(ts_earlier, TimeMax(ts_earlier, ts_earlier));
-  EXPECT_EQ(ts_later,   TimeMax(ts_earlier, ts_later));
-  EXPECT_EQ(ts_later,   TimeMax(ts_later,   ts_earlier));
-
-  // Interval
-  EXPECT_EQ(100,  TimeDiff(ts_later, ts_earlier));
-  EXPECT_EQ(-100, TimeDiff(ts_earlier, ts_later));
-}
-
 TEST(TimeTest, TestTimeDiff64) {
   int64_t ts_diff = 100;
-  int64_t ts_earlier = rtc::Time64();
+  int64_t ts_earlier = rtc::TimeMillis();
   int64_t ts_later = ts_earlier + ts_diff;
   EXPECT_EQ(ts_diff, rtc::TimeDiff(ts_later, ts_earlier));
   EXPECT_EQ(-ts_diff, rtc::TimeDiff(ts_earlier, ts_later));
diff --git a/chromium/third_party/webrtc/base/virtualsocket_unittest.cc b/chromium/third_party/webrtc/base/virtualsocket_unittest.cc
index 2cd2b5e4de6..e63310423f0 100644
--- a/chromium/third_party/webrtc/base/virtualsocket_unittest.cc
+++ b/chromium/third_party/webrtc/base/virtualsocket_unittest.cc
@@ -14,6 +14,8 @@
 #include <netinet/in.h>
 #endif
 
+#include <memory>
+
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/gunit.h"
@@ -33,7 +35,7 @@ struct Sender : public MessageHandler {
         done(false),
         rate(rt),
         count(0) {
-    last_send = rtc::Time();
+    last_send = rtc::TimeMillis();
     thread->PostDelayed(NextDelay(), this, 1);
   }
 
@@ -48,9 +50,9 @@ struct Sender : public MessageHandler {
     if (done)
       return;
 
-    uint32_t cur_time = rtc::Time();
-    uint32_t delay = cur_time - last_send;
-    uint32_t size = rate * delay / 1000;
+    int64_t cur_time = rtc::TimeMillis();
+    int64_t delay = cur_time - last_send;
+    uint32_t size = static_cast<uint32_t>(rate * delay / 1000);
     size = std::min<uint32_t>(size, 4096);
     size = std::max<uint32_t>(size, sizeof(uint32_t));
 
@@ -63,12 +65,12 @@ struct Sender : public MessageHandler {
   }
 
   Thread* thread;
-  scoped_ptr<AsyncUDPSocket> socket;
+  std::unique_ptr<AsyncUDPSocket> socket;
   rtc::PacketOptions options;
   bool done;
   uint32_t rate;  // bytes per second
   uint32_t count;
-  uint32_t last_send;
+  int64_t last_send;
   char dummy[4096];
 };
 
@@ -101,7 +103,7 @@ struct Receiver : public MessageHandler, public sigslot::has_slots<> {
     sec_count += size;
 
     uint32_t send_time = *reinterpret_cast<const uint32_t*>(data);
-    uint32_t recv_time = rtc::Time();
+    uint32_t recv_time = rtc::TimeMillis();
     uint32_t delay = recv_time - send_time;
     sum += delay;
     sum_sq += delay * delay;
@@ -123,7 +125,7 @@ struct Receiver : public MessageHandler, public sigslot::has_slots<> {
   }
 
   Thread* thread;
-  scoped_ptr<AsyncUDPSocket> socket;
+  std::unique_ptr<AsyncUDPSocket> socket;
   uint32_t bandwidth;
   bool done;
   size_t count;
@@ -345,11 +347,11 @@ class VirtualSocketServerTest : public testing::Test {
         EmptySocketAddressWithFamily(initial_addr.family());
 
     // Create client and server
-    scoped_ptr<AsyncSocket> client(ss_->CreateAsyncSocket(initial_addr.family(),
-                                                          SOCK_STREAM));
+    std::unique_ptr<AsyncSocket> client(
+        ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
     sink.Monitor(client.get());
-    scoped_ptr<AsyncSocket> server(ss_->CreateAsyncSocket(initial_addr.family(),
-                                                          SOCK_STREAM));
+    std::unique_ptr<AsyncSocket> server(
+        ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
     sink.Monitor(server.get());
 
     // Initiate connect
@@ -406,7 +408,7 @@ class VirtualSocketServerTest : public testing::Test {
 
     // Server accepts connection
     EXPECT_TRUE(sink.Check(server.get(), testing::SSE_READ));
-    scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+    std::unique_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
     ASSERT_TRUE(NULL != accepted.get());
     sink.Monitor(accepted.get());
 
@@ -435,9 +437,8 @@ class VirtualSocketServerTest : public testing::Test {
     a->Bind(initial_addr);
     EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family());
 
-
-    scoped_ptr<AsyncSocket> b(ss_->CreateAsyncSocket(initial_addr.family(),
-                                                     SOCK_STREAM));
+    std::unique_ptr<AsyncSocket> b(
+        ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
     sink.Monitor(b.get());
     b->Bind(initial_addr);
     EXPECT_EQ(b->GetLocalAddress().family(), initial_addr.family());
diff --git a/chromium/third_party/webrtc/base/virtualsocketserver.cc b/chromium/third_party/webrtc/base/virtualsocketserver.cc
index c6d402f1f3f..c76fe42f1ee 100644
--- a/chromium/third_party/webrtc/base/virtualsocketserver.cc
+++ b/chromium/third_party/webrtc/base/virtualsocketserver.cc
@@ -15,6 +15,7 @@
 
 #include <algorithm>
 #include <map>
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/checks.h"
@@ -500,15 +501,25 @@ int VirtualSocket::SendTcp(const void* pv, size_t cb) {
 }
 
 VirtualSocketServer::VirtualSocketServer(SocketServer* ss)
-    : server_(ss), server_owned_(false), msg_queue_(NULL), stop_on_idle_(false),
-      network_delay_(Time()), next_ipv4_(kInitialNextIPv4),
-      next_ipv6_(kInitialNextIPv6), next_port_(kFirstEphemeralPort),
-      bindings_(new AddressMap()), connections_(new ConnectionMap()),
-      bandwidth_(0), network_capacity_(kDefaultNetworkCapacity),
+    : server_(ss),
+      server_owned_(false),
+      msg_queue_(NULL),
+      stop_on_idle_(false),
+      network_delay_(TimeMillis()),
+      next_ipv4_(kInitialNextIPv4),
+      next_ipv6_(kInitialNextIPv6),
+      next_port_(kFirstEphemeralPort),
+      bindings_(new AddressMap()),
+      connections_(new ConnectionMap()),
+      bandwidth_(0),
+      network_capacity_(kDefaultNetworkCapacity),
       send_buffer_capacity_(kDefaultTcpBufferSize),
       recv_buffer_capacity_(kDefaultTcpBufferSize),
-      delay_mean_(0), delay_stddev_(0), delay_samples_(NUM_SAMPLES),
-      delay_dist_(NULL), drop_prob_(0.0) {
+      delay_mean_(0),
+      delay_stddev_(0),
+      delay_samples_(NUM_SAMPLES),
+      delay_dist_(NULL),
+      drop_prob_(0.0) {
   if (!server_) {
     server_ = new PhysicalSocketServer();
     server_owned_ = true;
@@ -567,7 +578,9 @@ AsyncSocket* VirtualSocketServer::CreateAsyncSocket(int family, int type) {
 }
 
 VirtualSocket* VirtualSocketServer::CreateSocketInternal(int family, int type) {
-  return new VirtualSocket(this, family, type, true);
+  VirtualSocket* socket = new VirtualSocket(this, family, type, true);
+  SignalSocketCreated(socket);
+  return socket;
 }
 
 void VirtualSocketServer::SetMessageQueue(MessageQueue* msg_queue) {
@@ -771,7 +784,7 @@ int VirtualSocketServer::SendUdp(VirtualSocket* socket,
   VirtualSocket* recipient = LookupBinding(remote_addr);
   if (!recipient) {
     // Make a fake recipient for address family checking.
-    scoped_ptr<VirtualSocket> dummy_socket(
+    std::unique_ptr<VirtualSocket> dummy_socket(
         CreateSocketInternal(AF_INET, SOCK_DGRAM));
     dummy_socket->SetLocalAddress(remote_addr);
     if (!CanInteractWith(socket, dummy_socket.get())) {
@@ -791,7 +804,7 @@ int VirtualSocketServer::SendUdp(VirtualSocket* socket,
 
   CritScope cs(&socket->crit_);
 
-  uint32_t cur_time = Time();
+  int64_t cur_time = TimeMillis();
   PurgeNetworkPackets(socket, cur_time);
 
   // Determine whether we have enough bandwidth to accept this packet.  To do
@@ -831,7 +844,7 @@ void VirtualSocketServer::SendTcp(VirtualSocket* socket) {
 
   CritScope cs(&socket->crit_);
 
-  uint32_t cur_time = Time();
+  int64_t cur_time = TimeMillis();
   PurgeNetworkPackets(socket, cur_time);
 
   while (true) {
@@ -866,7 +879,7 @@ void VirtualSocketServer::SendTcp(VirtualSocket* socket) {
 
 void VirtualSocketServer::AddPacketToNetwork(VirtualSocket* sender,
                                              VirtualSocket* recipient,
-                                             uint32_t cur_time,
+                                             int64_t cur_time,
                                              const char* data,
                                              size_t data_size,
                                              size_t header_size,
@@ -894,19 +907,19 @@ void VirtualSocketServer::AddPacketToNetwork(VirtualSocket* sender,
   // Post the packet as a message to be delivered (on our own thread)
   Packet* p = new Packet(data, data_size, sender_addr);
 
-  uint32_t ts = TimeAfter(send_delay + transit_delay);
+  int64_t ts = TimeAfter(send_delay + transit_delay);
   if (ordered) {
     // Ensure that new packets arrive after previous ones
     // TODO: consider ordering on a per-socket basis, since this
-    // introduces artifical delay.
-    ts = TimeMax(ts, network_delay_);
+    // introduces artificial delay.
+    ts = std::max(ts, network_delay_);
   }
   msg_queue_->PostAt(ts, recipient, MSG_ID_PACKET, p);
-  network_delay_ = TimeMax(ts, network_delay_);
+  network_delay_ = std::max(ts, network_delay_);
 }
 
 void VirtualSocketServer::PurgeNetworkPackets(VirtualSocket* socket,
-                                              uint32_t cur_time) {
+                                              int64_t cur_time) {
   while (!socket->network_.empty() &&
          (socket->network_.front().done_time <= cur_time)) {
     ASSERT(socket->network_size_ >= socket->network_.front().size);
diff --git a/chromium/third_party/webrtc/base/virtualsocketserver.h b/chromium/third_party/webrtc/base/virtualsocketserver.h
index daf0145a26c..897ba9e5ebc 100644
--- a/chromium/third_party/webrtc/base/virtualsocketserver.h
+++ b/chromium/third_party/webrtc/base/virtualsocketserver.h
@@ -16,6 +16,7 @@
 #include <deque>
 #include <map>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/messagequeue.h"
 #include "webrtc/base/socketserver.h"
 
@@ -121,6 +122,9 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
   bool CloseTcpConnections(const SocketAddress& addr_local,
                            const SocketAddress& addr_remote);
 
+  // For testing purpose only. Fired when a client socket is created.
+  sigslot::signal1<VirtualSocket*> SignalSocketCreated;
+
  protected:
   // Returns a new IP not used before in this network.
   IPAddress GetNextIP(int family);
@@ -168,14 +172,14 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
   // Places a packet on the network.
   void AddPacketToNetwork(VirtualSocket* socket,
                           VirtualSocket* recipient,
-                          uint32_t cur_time,
+                          int64_t cur_time,
                           const char* data,
                           size_t data_size,
                           size_t header_size,
                           bool ordered);
 
   // Removes stale packets from the network
-  void PurgeNetworkPackets(VirtualSocket* socket, uint32_t cur_time);
+  void PurgeNetworkPackets(VirtualSocket* socket, int64_t cur_time);
 
   // Computes the number of milliseconds required to send a packet of this size.
   uint32_t SendDelay(uint32_t size);
@@ -226,7 +230,7 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
   bool server_owned_;
   MessageQueue* msg_queue_;
   bool stop_on_idle_;
-  uint32_t network_delay_;
+  int64_t network_delay_;
   in_addr next_ipv4_;
   in6_addr next_ipv6_;
   uint16_t next_port_;
@@ -292,7 +296,7 @@ class VirtualSocket : public AsyncSocket, public MessageHandler {
  private:
   struct NetworkEntry {
     size_t size;
-    uint32_t done_time;
+    int64_t done_time;
   };
 
   typedef std::deque<SocketAddress> ListenQueue;
diff --git a/chromium/third_party/webrtc/base/win32filesystem.cc b/chromium/third_party/webrtc/base/win32filesystem.cc
index b731974bac4..84574032e68 100644
--- a/chromium/third_party/webrtc/base/win32filesystem.cc
+++ b/chromium/third_party/webrtc/base/win32filesystem.cc
@@ -15,10 +15,11 @@
 #include <shlobj.h>
 #include <tchar.h>
 
+#include <memory>
+
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/pathutils.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stream.h"
 #include "webrtc/base/stringutils.h"
 
@@ -95,7 +96,7 @@ bool Win32Filesystem::CreatePrivateFile(const Pathname &filename) {
                               &token_user_size);
 
   // Get the TOKEN_USER structure.
-  scoped_ptr<char[]> token_user_bytes(new char[token_user_size]);
+  std::unique_ptr<char[]> token_user_bytes(new char[token_user_size]);
   PTOKEN_USER token_user = reinterpret_cast<PTOKEN_USER>(
       token_user_bytes.get());
   memset(token_user, 0, token_user_size);
@@ -121,7 +122,7 @@ bool Win32Filesystem::CreatePrivateFile(const Pathname &filename) {
       GetLengthSid(token_user->User.Sid);
 
   // Allocate it.
-  scoped_ptr<char[]> acl_bytes(new char[acl_size]);
+  std::unique_ptr<char[]> acl_bytes(new char[acl_size]);
   PACL acl = reinterpret_cast<PACL>(acl_bytes.get());
   memset(acl, 0, acl_size);
   if (!::InitializeAcl(acl, acl_size, ACL_REVISION)) {
@@ -425,7 +426,7 @@ bool Win32Filesystem::GetDiskFreeSpace(const Pathname& path,
 Pathname Win32Filesystem::GetCurrentDirectory() {
   Pathname cwd;
   int path_len = 0;
-  scoped_ptr<wchar_t[]> path;
+  std::unique_ptr<wchar_t[]> path;
   do {
     int needed = ::GetCurrentDirectory(path_len, path.get());
     if (needed == 0) {
diff --git a/chromium/third_party/webrtc/base/win32regkey.cc b/chromium/third_party/webrtc/base/win32regkey.cc
index c53386fffe4..447086aff5f 100644
--- a/chromium/third_party/webrtc/base/win32regkey.cc
+++ b/chromium/third_party/webrtc/base/win32regkey.cc
@@ -21,9 +21,10 @@
 
 #include <shlwapi.h>
 
+#include <memory>
+
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 
@@ -146,7 +147,7 @@ HRESULT RegKey::GetValue(const wchar_t* full_key_name,
   byte* buffer_raw = nullptr;
   HRESULT hr = GetValueStaticHelper(full_key_name, value_name,
                                     REG_BINARY, &buffer_raw, &byte_count);
-  scoped_ptr<byte[]> buffer(buffer_raw);
+  std::unique_ptr<byte[]> buffer(buffer_raw);
   if (SUCCEEDED(hr)) {
     ASSERT(byte_count == sizeof(*value));
     if (byte_count == sizeof(*value)) {
@@ -166,7 +167,7 @@ HRESULT RegKey::GetValue(const wchar_t* full_key_name,
   byte* buffer_raw = nullptr;
   HRESULT hr = GetValueStaticHelper(full_key_name, value_name,
                                     REG_BINARY, &buffer_raw, &byte_count);
-  scoped_ptr<byte[]> buffer(buffer_raw);
+  std::unique_ptr<byte[]> buffer(buffer_raw);
   if (SUCCEEDED(hr)) {
     ASSERT(byte_count == sizeof(*value));
     if (byte_count == sizeof(*value)) {
@@ -193,7 +194,7 @@ HRESULT RegKey::GetValue(const wchar_t* full_key_name,
 
   wchar_t* buffer_raw = nullptr;
   HRESULT hr = RegKey::GetValue(full_key_name, value_name, &buffer_raw);
-  scoped_ptr<wchar_t[]> buffer(buffer_raw);
+  std::unique_ptr<wchar_t[]> buffer(buffer_raw);
   if (SUCCEEDED(hr)) {
     value->assign(buffer.get());
   }
diff --git a/chromium/third_party/webrtc/base/win32socketserver.cc b/chromium/third_party/webrtc/base/win32socketserver.cc
index 72ce4ebb7c9..5423eed9bf0 100644
--- a/chromium/third_party/webrtc/base/win32socketserver.cc
+++ b/chromium/third_party/webrtc/base/win32socketserver.cc
@@ -627,7 +627,7 @@ void Win32Socket::OnSocketNotify(SOCKET socket, int event, int error) {
       if (error != ERROR_SUCCESS) {
         ReportWSAError("WSAAsync:connect notify", error, addr_);
 #if !defined(NDEBUG)
-        int32_t duration = TimeSince(connect_time_);
+        int64_t duration = TimeSince(connect_time_);
         LOG(LS_INFO) << "WSAAsync:connect error (" << duration
                      << " ms), faking close";
 #endif
@@ -640,7 +640,7 @@ void Win32Socket::OnSocketNotify(SOCKET socket, int event, int error) {
         SignalCloseEvent(this, error);
       } else {
 #if !defined(NDEBUG)
-        int32_t duration = TimeSince(connect_time_);
+        int64_t duration = TimeSince(connect_time_);
         LOG(LS_INFO) << "WSAAsync:connect (" << duration << " ms)";
 #endif
         state_ = CS_CONNECTED;
diff --git a/chromium/third_party/webrtc/base/win32socketserver.h b/chromium/third_party/webrtc/base/win32socketserver.h
index b468cfd9e3d..f47ed756964 100644
--- a/chromium/third_party/webrtc/base/win32socketserver.h
+++ b/chromium/third_party/webrtc/base/win32socketserver.h
@@ -159,6 +159,6 @@ class Win32Thread : public Thread {
 
 }  // namespace rtc
 
-#endif  // WEBRTC_WIN 
+#endif  // WEBRTC_WIN
 
 #endif  // WEBRTC_BASE_WIN32SOCKETSERVER_H_
diff --git a/chromium/third_party/webrtc/base/windowpicker_unittest.cc b/chromium/third_party/webrtc/base/windowpicker_unittest.cc
index edd01bc0b2c..a1258327083 100644
--- a/chromium/third_party/webrtc/base/windowpicker_unittest.cc
+++ b/chromium/third_party/webrtc/base/windowpicker_unittest.cc
@@ -7,6 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#include <memory>
+
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/testutils.h"
 #include "webrtc/base/window.h"
@@ -25,7 +27,7 @@ TEST(WindowPickerTest, GetWindowList) {
     LOG(LS_INFO) << "skipping test: window capturing is not supported with "
                  << "current configuration.";
   }
-  rtc::scoped_ptr<rtc::WindowPicker> picker(
+  std::unique_ptr<rtc::WindowPicker> picker(
       rtc::WindowPickerFactory::CreateWindowPicker());
   EXPECT_TRUE(picker->Init());
   rtc::WindowDescriptionList descriptions;
@@ -40,7 +42,7 @@ TEST(WindowPickerTest, DISABLE_ON_MAC(GetDesktopList)) {
     LOG(LS_INFO) << "skipping test: window capturing is not supported with "
                  << "current configuration.";
   }
-  rtc::scoped_ptr<rtc::WindowPicker> picker(
+  std::unique_ptr<rtc::WindowPicker> picker(
       rtc::WindowPickerFactory::CreateWindowPicker());
   EXPECT_TRUE(picker->Init());
   rtc::DesktopDescriptionList descriptions;
diff --git a/chromium/third_party/webrtc/base/x11windowpicker.cc b/chromium/third_party/webrtc/base/x11windowpicker.cc
index 21f71c61e3e..e588304d035 100644
--- a/chromium/third_party/webrtc/base/x11windowpicker.cc
+++ b/chromium/third_party/webrtc/base/x11windowpicker.cc
@@ -21,6 +21,7 @@
 #include <X11/extensions/Xrender.h>
 #include <X11/Xutil.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 
 namespace rtc {
diff --git a/chromium/third_party/webrtc/base/x11windowpicker.h b/chromium/third_party/webrtc/base/x11windowpicker.h
index 501adf5820b..d741759368b 100644
--- a/chromium/third_party/webrtc/base/x11windowpicker.h
+++ b/chromium/third_party/webrtc/base/x11windowpicker.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_BASE_LINUXWINDOWPICKER_H_
 #define WEBRTC_BASE_LINUXWINDOWPICKER_H_
 
+#include <memory>
+
 #include "webrtc/base/basictypes.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/windowpicker.h"
 
 // Avoid include <X11/Xlib.h>.
@@ -44,7 +45,7 @@ class X11WindowPicker : public WindowPicker {
   uint8_t* GetDesktopThumbnail(const DesktopId& id, int width, int height);
 
  private:
-  scoped_ptr<XWindowEnumerator> enumerator_;
+  std::unique_ptr<XWindowEnumerator> enumerator_;
 };
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/build/apk_tests.gyp b/chromium/third_party/webrtc/build/apk_tests.gyp
index f7e9a902205..ba83acc14bb 100644
--- a/chromium/third_party/webrtc/build/apk_tests.gyp
+++ b/chromium/third_party/webrtc/build/apk_tests.gyp
@@ -17,13 +17,32 @@
   'includes': [
     'common.gypi',
   ],
+  'variables': {
+    'shard_timeout': 900,
+  },
   'targets': [
     {
+      'target_name': 'audio_codec_speed_tests_apk',
+      'type': 'none',
+      'variables': {
+        'test_suite_name': 'audio_codec_speed_tests',
+        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)audio_codec_speed_tests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../modules/audio_codec_speed_tests.isolate',
+      },
+      'dependencies': [
+        '<(webrtc_root)/modules/modules.gyp:audio_codec_speed_tests',
+      ],
+      'includes': [
+        '../../build/apk_test.gypi',
+      ],
+    },
+    {
       'target_name': 'audio_decoder_unittests_apk',
       'type': 'none',
       'variables': {
         'test_suite_name': 'audio_decoder_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)audio_decoder_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../modules/audio_decoder_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/modules/modules.gyp:audio_decoder_unittests',
@@ -38,6 +57,7 @@
       'variables': {
         'test_suite_name': 'common_audio_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)common_audio_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../common_audio/common_audio_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio_unittests',
@@ -52,6 +72,7 @@
       'variables': {
         'test_suite_name': 'common_video_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)common_video_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../common_video/common_video_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/common_video/common_video_unittests.gyp:common_video_unittests',
@@ -61,44 +82,47 @@
       ],
     },
     {
-      'target_name': 'peerconnection_unittests_apk',
+      'target_name': 'modules_tests_apk',
       'type': 'none',
       'variables': {
-        'test_suite_name': 'peerconnection_unittests',
-        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)peerconnection_unittests<(SHARED_LIB_SUFFIX)',
+        'test_suite_name': 'modules_tests',
+        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)modules_tests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../modules/modules_tests.isolate',
       },
       'dependencies': [
-        '<(webrtc_root)/api/api_tests.gyp:peerconnection_unittests',
-        '<(webrtc_root)/api/api.gyp:libjingle_peerconnection_java',
+        '<(webrtc_root)/modules/modules.gyp:modules_tests',
       ],
       'includes': [
         '../../build/apk_test.gypi',
       ],
     },
     {
-      'target_name': 'modules_tests_apk',
+      'target_name': 'modules_unittests_apk',
       'type': 'none',
       'variables': {
-        'test_suite_name': 'modules_tests',
-        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)modules_tests<(SHARED_LIB_SUFFIX)',
+        'test_suite_name': 'modules_unittests',
+        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)modules_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../modules/modules_unittests.isolate',
       },
       'dependencies': [
-        '<(webrtc_root)/modules/modules.gyp:modules_tests',
+        '<(webrtc_root)/modules/modules.gyp:modules_unittests',
+        'audio_device_java',
       ],
       'includes': [
         '../../build/apk_test.gypi',
       ],
     },
     {
-      'target_name': 'modules_unittests_apk',
+      'target_name': 'peerconnection_unittests_apk',
       'type': 'none',
       'variables': {
-        'test_suite_name': 'modules_unittests',
-        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)modules_unittests<(SHARED_LIB_SUFFIX)',
+        'test_suite_name': 'peerconnection_unittests',
+        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)peerconnection_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../api/peerconnection_unittests.isolate',
       },
       'dependencies': [
-        '<(webrtc_root)/modules/modules.gyp:modules_unittests',
-        'audio_device_java',
+        '<(webrtc_root)/api/api_tests.gyp:peerconnection_unittests',
+        '<(webrtc_root)/api/api.gyp:libjingle_peerconnection_java',
       ],
       'includes': [
         '../../build/apk_test.gypi',
@@ -110,6 +134,7 @@
       'variables': {
         'test_suite_name': 'rtc_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)rtc_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../rtc_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/webrtc.gyp:rtc_unittests',
@@ -124,6 +149,7 @@
       'variables': {
         'test_suite_name': 'system_wrappers_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)system_wrappers_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../system_wrappers/system_wrappers_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/system_wrappers/system_wrappers_tests.gyp:system_wrappers_unittests',
@@ -138,6 +164,7 @@
       'variables': {
         'test_suite_name': 'test_support_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)test_support_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../test/test_support_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/test/test.gyp:test_support_unittests',
@@ -152,6 +179,7 @@
       'variables': {
         'test_suite_name': 'tools_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)tools_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../tools/tools_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/tools/tools.gyp:tools_unittests',
@@ -166,6 +194,7 @@
       'variables': {
         'test_suite_name': 'video_engine_tests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)video_engine_tests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../video_engine_tests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/webrtc.gyp:video_engine_tests',
@@ -180,6 +209,7 @@
       'variables': {
         'test_suite_name': 'voice_engine_unittests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)voice_engine_unittests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../voice_engine/voice_engine_unittests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_unittests',
@@ -194,6 +224,8 @@
       'variables': {
         'test_suite_name': 'webrtc_perf_tests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)webrtc_perf_tests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../webrtc_perf_tests.isolate',
+        'shard_timeout': 2700,
       },
       'dependencies': [
         '<(webrtc_root)/webrtc.gyp:webrtc_perf_tests',
@@ -208,6 +240,7 @@
       'variables': {
         'test_suite_name': 'webrtc_nonparallel_tests',
         'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)webrtc_nonparallel_tests<(SHARED_LIB_SUFFIX)',
+        'isolate_file': '../webrtc_nonparallel_tests.isolate',
       },
       'dependencies': [
         '<(webrtc_root)/webrtc.gyp:webrtc_nonparallel_tests',
@@ -217,20 +250,6 @@
       ],
     },
     {
-      'target_name': 'audio_codec_speed_tests_apk',
-      'type': 'none',
-      'variables': {
-        'test_suite_name': 'audio_codec_speed_tests',
-        'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)audio_codec_speed_tests<(SHARED_LIB_SUFFIX)',
-      },
-      'dependencies': [
-        '<(webrtc_root)/modules/modules.gyp:audio_codec_speed_tests',
-      ],
-      'includes': [
-        '../../build/apk_test.gypi',
-      ],
-    },
-    {
       'target_name': 'audio_device_java',
       'type': 'none',
       'variables': {
diff --git a/chromium/third_party/webrtc/build/common.gypi b/chromium/third_party/webrtc/build/common.gypi
index 6ec0f1c9426..26acf4b5e27 100644
--- a/chromium/third_party/webrtc/build/common.gypi
+++ b/chromium/third_party/webrtc/build/common.gypi
@@ -41,10 +41,23 @@
             'apk_tests_path%': '<(DEPTH)/webrtc/build/apk_tests.gyp',
             'modules_java_gyp_path%': '<(DEPTH)/webrtc/modules/modules_java.gyp',
           }],
+
+          # Controls whether we use libevent on posix platforms.
+          # TODO(phoglund): should arguably be controlled by platform #ifdefs
+          # in the code instead.
+          ['OS=="win" or OS=="mac" or OS=="ios"', {
+            'build_libevent%': 0,
+            'enable_libevent%': 0,
+          }, {
+            'build_libevent%': 1,
+            'enable_libevent%': 1,
+          }],
         ],
       },
       'build_with_chromium%': '<(build_with_chromium)',
       'build_with_mozilla%': '<(build_with_mozilla)',
+      'build_libevent%': '<(build_libevent)',
+      'enable_libevent%': '<(enable_libevent)',
       'webrtc_root%': '<(webrtc_root)',
       'apk_tests_path%': '<(apk_tests_path)',
       'modules_java_gyp_path%': '<(modules_java_gyp_path)',
@@ -56,8 +69,11 @@
     },
     'build_with_chromium%': '<(build_with_chromium)',
     'build_with_mozilla%': '<(build_with_mozilla)',
+    'build_libevent%': '<(build_libevent)',
+    'enable_libevent%': '<(enable_libevent)',
     'webrtc_root%': '<(webrtc_root)',
     'apk_tests_path%': '<(apk_tests_path)',
+    'test_runner_path': '<(DEPTH)/webrtc/build/android/test_runner.py',
     'modules_java_gyp_path%': '<(modules_java_gyp_path)',
     'webrtc_vp8_dir%': '<(webrtc_vp8_dir)',
     'webrtc_vp9_dir%': '<(webrtc_vp9_dir)',
@@ -98,9 +114,9 @@
     # Disable these to not build components which can be externally provided.
     'build_expat%': 1,
     'build_json%': 1,
-    'build_libjpeg%': 1,
     'build_libsrtp%': 1,
     'build_libvpx%': 1,
+    'libvpx_build_vp9%': 1,
     'build_libyuv%': 1,
     'build_openmax_dl%': 1,
     'build_opus%': 1,
@@ -202,13 +218,10 @@
         'include_tests%': 1,
         'restrict_webrtc_logging%': 0,
       }],
-      ['OS=="ios"', {
-        'build_libjpeg%': 0,
-      }],
       ['target_arch=="arm" or target_arch=="arm64" or target_arch=="mipsel"', {
         'prefer_fixed_point%': 1,
       }],
-      ['(target_arch=="arm" and (arm_neon==1 or arm_neon_optional==1)) or target_arch=="arm64"', {
+      ['(target_arch=="arm" and arm_neon==1) or target_arch=="arm64"', {
         'build_with_neon%': 1,
       }],
       ['OS!="ios" and (target_arch!="arm" or arm_version>=7) and target_arch!="mips64el"', {
@@ -312,10 +325,16 @@
             'cflags': [
               '-Wimplicit-fallthrough',
               '-Wthread-safety',
+              '-Winconsistent-missing-override',
             ],
           }],
         ],
       }],
+      ['enable_libevent==1', {
+        'defines': [
+          'WEBRTC_BUILD_LIBEVENT',
+        ],
+      }],
       ['target_arch=="arm64"', {
         'defines': [
           'WEBRTC_ARCH_ARM64',
@@ -333,9 +352,6 @@
               ['arm_neon==1', {
                 'defines': ['WEBRTC_HAS_NEON',],
               }],
-              ['arm_neon==0 and arm_neon_optional==1', {
-                'defines': ['WEBRTC_DETECT_NEON',],
-              }],
             ],
           }],
         ],
@@ -371,6 +387,7 @@
       ['coverage==1 and OS=="linux"', {
         'cflags': [ '-ftest-coverage',
                     '-fprofile-arcs' ],
+        'ldflags': [ '--coverage' ],
         'link_settings': { 'libraries': [ '-lgcov' ] },
       }],
       ['os_posix==1', {
@@ -444,6 +461,11 @@
           'WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE',
         ],
       }],
+      ['libvpx_build_vp9==0', {
+        'defines': [
+          'RTC_DISABLE_VP9',
+        ],
+      }],
     ], # conditions
     'direct_dependent_settings': {
       'conditions': [
diff --git a/chromium/third_party/webrtc/build/gyp_webrtc b/chromium/third_party/webrtc/build/gyp_webrtc
index 2376cfbb85c..c0a9ed9da8e 100755..100644
--- a/chromium/third_party/webrtc/build/gyp_webrtc
+++ b/chromium/third_party/webrtc/build/gyp_webrtc
@@ -8,108 +8,10 @@
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
-# This script is used to run GYP for WebRTC. It contains selected parts of the
-# main function from the src/build/gyp_chromium file.
+# Simple launcher script for gyp_webrtc.py.
+# TODO(kjellander): This should probably be shell script but for historical
+# reasons (all the python code used to live in this script without a
+# .py extension, and was often run as 'python gyp_webrtc') it is
+# currently still python.
 
-import glob
-import os
-import shlex
-import sys
-
-script_dir = os.path.dirname(os.path.realpath(__file__))
-checkout_root = os.path.abspath(os.path.join(script_dir, os.pardir, os.pardir))
-
-sys.path.insert(0, os.path.join(checkout_root, 'build'))
-import gyp_chromium
-import gyp_helper
-import vs_toolchain
-
-sys.path.insert(0, os.path.join(checkout_root, 'tools', 'gyp', 'pylib'))
-import gyp
-
-def GetSupplementalFiles():
-  """Returns a list of the supplemental files that are included in all GYP
-  sources."""
-  # Can't use the one in gyp_chromium since the directory location of the root
-  # is different.
-  return glob.glob(os.path.join(checkout_root, '*', 'supplement.gypi'))
-
-
-if __name__ == '__main__':
-  args = sys.argv[1:]
-
-  if int(os.environ.get('GYP_CHROMIUM_NO_ACTION', 0)):
-    print 'Skipping gyp_webrtc due to GYP_CHROMIUM_NO_ACTION env var.'
-    sys.exit(0)
-
-  if 'SKIP_WEBRTC_GYP_ENV' not in os.environ:
-    # Update the environment based on webrtc.gyp_env
-    gyp_env_path = os.path.join(os.path.dirname(checkout_root),
-                                'webrtc.gyp_env')
-    gyp_helper.apply_gyp_environment_from_file(gyp_env_path)
-
-  # This could give false positives since it doesn't actually do real option
-  # parsing.  Oh well.
-  gyp_file_specified = False
-  for arg in args:
-    if arg.endswith('.gyp'):
-      gyp_file_specified = True
-      break
-
-  # If we didn't get a file, assume 'all.gyp' in the root of the checkout.
-  if not gyp_file_specified:
-    # Because of a bug in gyp, simply adding the abspath to all.gyp doesn't
-    # work, but chdir'ing and adding the relative path does. Spooky :/
-    os.chdir(checkout_root)
-    args.append('all.gyp')
-
-  # There shouldn't be a circular dependency relationship between .gyp files,
-  args.append('--no-circular-check')
-
-  # Default to ninja unless GYP_GENERATORS is set.
-  if not os.environ.get('GYP_GENERATORS'):
-    os.environ['GYP_GENERATORS'] = 'ninja'
-
-  # Enable check for missing sources in GYP files on Windows.
-  if sys.platform.startswith('win'):
-    gyp_generator_flags = os.getenv('GYP_GENERATOR_FLAGS', '')
-    if not 'msvs_error_on_missing_sources' in gyp_generator_flags:
-      os.environ['GYP_GENERATOR_FLAGS'] = (
-          gyp_generator_flags + ' msvs_error_on_missing_sources=1')
-
-  vs2013_runtime_dll_dirs = None
-  if int(os.environ.get('DEPOT_TOOLS_WIN_TOOLCHAIN', '1')):
-    vs2013_runtime_dll_dirs = vs_toolchain.SetEnvironmentAndGetRuntimeDllDirs()
-
-  # Enforce gyp syntax checking. This adds about 20% execution time.
-  args.append('--check')
-
-  supplemental_includes = GetSupplementalFiles()
-  gyp_vars = gyp_chromium.GetGypVars(supplemental_includes)
-
-  # Automatically turn on crosscompile support for platforms that need it.
-  if all(('ninja' in os.environ.get('GYP_GENERATORS', ''),
-          gyp_vars.get('OS') in ['android', 'ios'],
-          'GYP_CROSSCOMPILE' not in os.environ)):
-    os.environ['GYP_CROSSCOMPILE'] = '1'
-
-  args.extend(['-I' + i for i in
-               gyp_chromium.additional_include_files(supplemental_includes,
-                                                     args)])
-
-  # Set the gyp depth variable to the root of the checkout.
-  args.append('--depth=' + os.path.relpath(checkout_root))
-
-  print 'Updating projects from gyp files...'
-  sys.stdout.flush()
-
-  # Off we go...
-  gyp_rc = gyp.main(args)
-
-  if vs2013_runtime_dll_dirs:
-    x64_runtime, x86_runtime = vs2013_runtime_dll_dirs
-    vs_toolchain.CopyVsRuntimeDlls(
-        os.path.join(checkout_root, gyp_chromium.GetOutputDirectory()),
-        (x86_runtime, x64_runtime))
-
-  sys.exit(gyp_rc)
+execfile(__file__ + '.py')
diff --git a/chromium/third_party/webrtc/build/gyp_webrtc.py b/chromium/third_party/webrtc/build/gyp_webrtc.py
index 87d8a57ddf0..05729489952 100644..100755
--- a/chromium/third_party/webrtc/build/gyp_webrtc.py
+++ b/chromium/third_party/webrtc/build/gyp_webrtc.py
@@ -8,17 +8,123 @@
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
-# This file is (possibly, depending on python version) imported by
-# gyp_webrtc when GYP_PARALLEL=1 and it creates sub-processes
-# through the multiprocessing library.
-
-# Importing in Python 2.6 (fixed in 2.7) on Windows doesn't search for
-# imports that don't end in .py (and aren't directories with an
-# __init__.py). This wrapper makes "import gyp_webrtc" work with
-# those old versions and makes it possible to execute gyp_webrtc.py
-# directly on Windows where the extension is useful.
+# This script is used to run GYP for WebRTC. It contains selected parts of the
+# main function from the src/build/gyp_chromium.py file while other parts are
+# reused to minimize code duplication.
 
+import gc
+import glob
 import os
+import sys
+
+script_dir = os.path.dirname(os.path.realpath(__file__))
+checkout_root = os.path.abspath(os.path.join(script_dir, os.pardir, os.pardir))
+
+sys.path.insert(0, os.path.join(checkout_root, 'build'))
+import gyp_chromium
+import gyp_helper
+import vs_toolchain
+
+sys.path.insert(0, os.path.join(checkout_root, 'tools', 'gyp', 'pylib'))
+import gyp
+
+
+def GetSupplementalFiles():
+  """Returns a list of the supplemental files.
+
+  A supplemental file is included in all GYP sources. Such files can be used to
+  override default values.
+  """
+  # Can't use the one in gyp_chromium since the directory location of the root
+  # is different.
+  return glob.glob(os.path.join(checkout_root, '*', 'supplement.gypi'))
+
+
+def main():
+  # Disabling garbage collection saves about 5% processing time. Since this is a
+  # short-lived process it's not a problem.
+  gc.disable()
+
+  args = sys.argv[1:]
+
+  if int(os.environ.get('GYP_CHROMIUM_NO_ACTION', 0)):
+    print 'Skipping gyp_webrtc.py due to GYP_CHROMIUM_NO_ACTION env var.'
+    sys.exit(0)
+
+  if 'SKIP_WEBRTC_GYP_ENV' not in os.environ:
+    # Update the environment based on webrtc.gyp_env.
+    gyp_env_path = os.path.join(os.path.dirname(checkout_root),
+                                'webrtc.gyp_env')
+    gyp_helper.apply_gyp_environment_from_file(gyp_env_path)
+
+  # This could give false positives since it doesn't actually do real option
+  # parsing.  Oh well.
+  gyp_file_specified = False
+  for arg in args:
+    if arg.endswith('.gyp'):
+      gyp_file_specified = True
+      break
+
+  # If we didn't get a file, assume 'all.gyp' in the root of the checkout.
+  if not gyp_file_specified:
+    # Because of a bug in gyp, simply adding the abspath to all.gyp doesn't
+    # work, but chdir'ing and adding the relative path does. Spooky :/
+    os.chdir(checkout_root)
+    args.append('all.gyp')
+
+  # There shouldn't be a circular dependency relationship between .gyp files,
+  args.append('--no-circular-check')
+
+  # Default to ninja unless GYP_GENERATORS is set.
+  if not os.environ.get('GYP_GENERATORS'):
+    os.environ['GYP_GENERATORS'] = 'ninja'
+
+  # Enable check for missing sources in GYP files on Windows.
+  if sys.platform.startswith('win'):
+    gyp_generator_flags = os.getenv('GYP_GENERATOR_FLAGS', '')
+    if not 'msvs_error_on_missing_sources' in gyp_generator_flags:
+      os.environ['GYP_GENERATOR_FLAGS'] = (
+          gyp_generator_flags + ' msvs_error_on_missing_sources=1')
+
+  vs2013_runtime_dll_dirs = None
+  if int(os.environ.get('DEPOT_TOOLS_WIN_TOOLCHAIN', '1')):
+    vs2013_runtime_dll_dirs = vs_toolchain.SetEnvironmentAndGetRuntimeDllDirs()
+
+  # Enforce gyp syntax checking. This adds about 20% execution time.
+  args.append('--check')
+
+  supplemental_includes = GetSupplementalFiles()
+  gyp_vars = gyp_chromium.GetGypVars(supplemental_includes)
+
+  # Automatically turn on crosscompile support for platforms that need it.
+  if all(('ninja' in os.environ.get('GYP_GENERATORS', ''),
+          gyp_vars.get('OS') in ['android', 'ios'],
+          'GYP_CROSSCOMPILE' not in os.environ)):
+    os.environ['GYP_CROSSCOMPILE'] = '1'
+
+  args.extend(['-I' + i for i in
+               gyp_chromium.additional_include_files(supplemental_includes,
+                                                     args)])
+
+  # Set the gyp depth variable to the root of the checkout.
+  args.append('--depth=' + os.path.relpath(checkout_root))
+
+  print 'Updating projects from gyp files...'
+  sys.stdout.flush()
+
+  # Off we go...
+  gyp_rc = gyp.main(args)
+
+  if vs2013_runtime_dll_dirs:
+    # pylint: disable=unpacking-non-sequence
+    x64_runtime, x86_runtime = vs2013_runtime_dll_dirs
+    vs_toolchain.CopyVsRuntimeDlls(
+        os.path.join(checkout_root, gyp_chromium.GetOutputDirectory()),
+        (x86_runtime, x64_runtime))
+
+  sys.exit(gyp_rc)
+
+
+if __name__ == '__main__':
+  sys.exit(main())
 
-path = os.path.abspath(os.path.split(__file__)[0])
-execfile(os.path.join(path, 'gyp_webrtc'))
diff --git a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC.xcodeproj/project.pbxproj b/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC.xcodeproj/project.pbxproj
deleted file mode 100644
index 209dbe5be89..00000000000
--- a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC.xcodeproj/project.pbxproj
+++ /dev/null
@@ -1,910 +0,0 @@
-// !$*UTF8*$!
-{
-	archiveVersion = 1;
-	classes = {
-	};
-	objectVersion = 46;
-	objects = {
-
-/* Begin PBXBuildFile section */
-		980224981CA243DE00295D57 /* libaudio_coding_module.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224561CA243DE00295D57 /* libaudio_coding_module.a */; };
-		980224991CA243DE00295D57 /* libaudio_conference_mixer.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224571CA243DE00295D57 /* libaudio_conference_mixer.a */; };
-		9802249A1CA243DE00295D57 /* libaudio_decoder_interface.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224581CA243DE00295D57 /* libaudio_decoder_interface.a */; };
-		9802249B1CA243DE00295D57 /* libaudio_device.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224591CA243DE00295D57 /* libaudio_device.a */; };
-		9802249C1CA243DE00295D57 /* libaudio_encoder_interface.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802245A1CA243DE00295D57 /* libaudio_encoder_interface.a */; };
-		9802249D1CA243DE00295D57 /* libaudio_processing.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802245B1CA243DE00295D57 /* libaudio_processing.a */; };
-		9802249E1CA243DE00295D57 /* libaudioproc_debug_proto.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802245C1CA243DE00295D57 /* libaudioproc_debug_proto.a */; };
-		9802249F1CA243DE00295D57 /* libbitrate_controller.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802245D1CA243DE00295D57 /* libbitrate_controller.a */; };
-		980224A01CA243DE00295D57 /* libboringssl.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802245E1CA243DE00295D57 /* libboringssl.a */; };
-		980224A11CA243DE00295D57 /* libcng.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802245F1CA243DE00295D57 /* libcng.a */; };
-		980224A21CA243DE00295D57 /* libcommon_audio.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224601CA243DE00295D57 /* libcommon_audio.a */; };
-		980224A31CA243DE00295D57 /* libcommon_video.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224611CA243DE00295D57 /* libcommon_video.a */; };
-		980224A41CA243DE00295D57 /* libcongestion_controller.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224621CA243DE00295D57 /* libcongestion_controller.a */; };
-		980224A51CA243DE00295D57 /* libexpat.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224631CA243DE00295D57 /* libexpat.a */; };
-		980224A61CA243DE00295D57 /* libfield_trial_default.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224641CA243DE00295D57 /* libfield_trial_default.a */; };
-		980224A71CA243DE00295D57 /* libg711.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224651CA243DE00295D57 /* libg711.a */; };
-		980224A81CA243DE00295D57 /* libg722.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224661CA243DE00295D57 /* libg722.a */; };
-		980224A91CA243DE00295D57 /* libilbc.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224671CA243DE00295D57 /* libilbc.a */; };
-		980224AA1CA243DE00295D57 /* libisac_common.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224681CA243DE00295D57 /* libisac_common.a */; };
-		980224AB1CA243DE00295D57 /* libisac.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224691CA243DE00295D57 /* libisac.a */; };
-		980224AC1CA243DE00295D57 /* libjingle_peerconnection.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802246A1CA243DE00295D57 /* libjingle_peerconnection.a */; };
-		980224AD1CA243DE00295D57 /* libjsoncpp.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802246B1CA243DE00295D57 /* libjsoncpp.a */; };
-		980224AE1CA243DE00295D57 /* libmedia_file.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802246C1CA243DE00295D57 /* libmedia_file.a */; };
-		980224AF1CA243DE00295D57 /* libmetrics_default.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802246D1CA243DE00295D57 /* libmetrics_default.a */; };
-		980224B01CA243DE00295D57 /* libneteq.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802246E1CA243DE00295D57 /* libneteq.a */; };
-		980224B11CA243DE00295D57 /* libopus.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 9802246F1CA243DE00295D57 /* libopus.a */; };
-		980224B21CA243DE00295D57 /* libpaced_sender.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224701CA243DE00295D57 /* libpaced_sender.a */; };
-		980224B31CA243DE00295D57 /* libpcm16b.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224711CA243DE00295D57 /* libpcm16b.a */; };
-		980224B41CA243DE00295D57 /* libprotobuf_lite.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224721CA243DE00295D57 /* libprotobuf_lite.a */; };
-		980224B51CA243DE00295D57 /* libred.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 980224731CA243DE00295D57 /* libred.a */; };
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-				986649D71C986B7D008EC831 /* RTCDataChannelConfiguration.mm in Sources */,
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-				986649E71C986B7D008EC831 /* RTCVideoRendererAdapter.mm in Sources */,
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-				986649D61C986B7D008EC831 /* RTCDataChannel.mm in Sources */,
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-		};
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-
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-				GCC_NO_COMMON_BLOCKS = YES;
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-				GCC_WARN_64_TO_32_BIT_CONVERSION = YES;
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-				IPHONEOS_DEPLOYMENT_TARGET = 9.2;
-				MTL_ENABLE_DEBUG_INFO = YES;
-				ONLY_ACTIVE_ARCH = YES;
-				SDKROOT = iphoneos;
-				TARGETED_DEVICE_FAMILY = "1,2";
-				VERSIONING_SYSTEM = "apple-generic";
-				VERSION_INFO_PREFIX = "";
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-		9820AACA1C977D11001E5793 /* Release */ = {
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-				CLANG_WARN_BOOL_CONVERSION = YES;
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-				CLANG_WARN_EMPTY_BODY = YES;
-				CLANG_WARN_ENUM_CONVERSION = YES;
-				CLANG_WARN_INT_CONVERSION = YES;
-				CLANG_WARN_OBJC_ROOT_CLASS = YES_ERROR;
-				CLANG_WARN_UNREACHABLE_CODE = YES;
-				CLANG_WARN__DUPLICATE_METHOD_MATCH = YES;
-				"CODE_SIGN_IDENTITY[sdk=iphoneos*]" = "iPhone Developer";
-				COPY_PHASE_STRIP = NO;
-				CURRENT_PROJECT_VERSION = 1;
-				DEBUG_INFORMATION_FORMAT = "dwarf-with-dsym";
-				DEPLOYMENT_POSTPROCESSING = YES;
-				ENABLE_NS_ASSERTIONS = NO;
-				ENABLE_STRICT_OBJC_MSGSEND = YES;
-				GCC_C_LANGUAGE_STANDARD = gnu99;
-				GCC_NO_COMMON_BLOCKS = YES;
-				GCC_WARN_64_TO_32_BIT_CONVERSION = YES;
-				GCC_WARN_ABOUT_RETURN_TYPE = YES_ERROR;
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-				GCC_WARN_UNINITIALIZED_AUTOS = YES_AGGRESSIVE;
-				GCC_WARN_UNUSED_FUNCTION = YES;
-				GCC_WARN_UNUSED_VARIABLE = YES;
-				IPHONEOS_DEPLOYMENT_TARGET = 9.2;
-				MTL_ENABLE_DEBUG_INFO = NO;
-				SDKROOT = iphoneos;
-				TARGETED_DEVICE_FAMILY = "1,2";
-				VALIDATE_PRODUCT = YES;
-				VERSIONING_SYSTEM = "apple-generic";
-				VERSION_INFO_PREFIX = "";
-			};
-			name = Release;
-		};
-		9820AACC1C977D11001E5793 /* Debug */ = {
-			isa = XCBuildConfiguration;
-			buildSettings = {
-				ALTERNATE_GROUP = "";
-				ALTERNATE_OWNER = "";
-				DEFINES_MODULE = YES;
-				DYLIB_COMPATIBILITY_VERSION = 1;
-				DYLIB_CURRENT_VERSION = 1;
-				DYLIB_INSTALL_NAME_BASE = "@rpath";
-				ENABLE_BITCODE = NO;
-				GCC_ENABLE_CPP_RTTI = NO;
-				GCC_PREPROCESSOR_DEFINITIONS = (
-					"DEBUG=1",
-					"$(inherited)",
-					V8_DEPRECATION_WARNINGS,
-					"CLD_VERSION=2",
-					DISABLE_NACL,
-					CHROMIUM_BUILD,
-					"CR_CLANG_REVISION=263324-1",
-					"USE_LIBJPEG_TURBO=1",
-					FIELDTRIAL_TESTING_ENABLED,
-					"DISABLE_FTP_SUPPORT=1",
-					V8_USE_EXTERNAL_STARTUP_DATA,
-					"GTEST_HAS_POSIX_RE=0",
-					"GTEST_LANG_CXX11=0",
-					WEBRTC_MAC,
-					WEBRTC_IOS,
-					WEBRTC_POSIX,
-					PROTOBUF_USE_DLLS,
-					GOOGLE_PROTOBUF_NO_RTTI,
-					GOOGLE_PROTOBUF_NO_STATIC_INITIALIZER,
-					"USE_LIBPCI=1",
-					"USE_OPENSSL=1",
-					NDEBUG,
-					NVALGRIND,
-					"DYNAMIC_ANNOTATIONS_ENABLED=0",
-					"NS_BLOCK_ASSERTIONS=1",
-				);
-				HEADER_SEARCH_PATHS = "$(SRCROOT)/../../../../..";
-				INFOPLIST_FILE = WebRTC/Info.plist;
-				INSTALL_PATH = "$(LOCAL_LIBRARY_DIR)/Frameworks";
-				LD_RUNPATH_SEARCH_PATHS = "$(inherited) @executable_path/Frameworks @loader_path/Frameworks";
-				LIBRARY_SEARCH_PATHS = "$(SRCROOT)/../../../../../out_ios_libs/lib";
-				OTHER_LDFLAGS = "-ObjC";
-				PRODUCT_BUNDLE_IDENTIFIER = com.google.WebRTC;
-				PRODUCT_NAME = "$(TARGET_NAME)";
-				SKIP_INSTALL = YES;
-				SYMROOT = $SRCROOT/build;
-			};
-			name = Debug;
-		};
-		9820AACD1C977D11001E5793 /* Release */ = {
-			isa = XCBuildConfiguration;
-			buildSettings = {
-				ALTERNATE_GROUP = "";
-				ALTERNATE_OWNER = "";
-				DEFINES_MODULE = YES;
-				DYLIB_COMPATIBILITY_VERSION = 1;
-				DYLIB_CURRENT_VERSION = 1;
-				DYLIB_INSTALL_NAME_BASE = "@rpath";
-				ENABLE_BITCODE = NO;
-				GCC_ENABLE_CPP_RTTI = NO;
-				GCC_PREPROCESSOR_DEFINITIONS = (
-					V8_DEPRECATION_WARNINGS,
-					"CLD_VERSION=2",
-					DISABLE_NACL,
-					CHROMIUM_BUILD,
-					"CR_CLANG_REVISION=263324-1",
-					"USE_LIBJPEG_TURBO=1",
-					FIELDTRIAL_TESTING_ENABLED,
-					"DISABLE_FTP_SUPPORT=1",
-					V8_USE_EXTERNAL_STARTUP_DATA,
-					"GTEST_HAS_POSIX_RE=0",
-					"GTEST_LANG_CXX11=0",
-					WEBRTC_MAC,
-					WEBRTC_IOS,
-					WEBRTC_POSIX,
-					PROTOBUF_USE_DLLS,
-					GOOGLE_PROTOBUF_NO_RTTI,
-					GOOGLE_PROTOBUF_NO_STATIC_INITIALIZER,
-					"USE_LIBPCI=1",
-					"USE_OPENSSL=1",
-					NDEBUG,
-					NVALGRIND,
-					"DYNAMIC_ANNOTATIONS_ENABLED=0",
-					"NS_BLOCK_ASSERTIONS=1",
-				);
-				HEADER_SEARCH_PATHS = "$(SRCROOT)/../../../../..";
-				INFOPLIST_FILE = WebRTC/Info.plist;
-				INSTALL_PATH = "$(LOCAL_LIBRARY_DIR)/Frameworks";
-				LD_RUNPATH_SEARCH_PATHS = "$(inherited) @executable_path/Frameworks @loader_path/Frameworks";
-				LIBRARY_SEARCH_PATHS = "$(SRCROOT)/../../../../../out_ios_libs/lib";
-				OTHER_LDFLAGS = "-ObjC";
-				PRODUCT_BUNDLE_IDENTIFIER = com.google.WebRTC;
-				PRODUCT_NAME = "$(TARGET_NAME)";
-				SKIP_INSTALL = YES;
-				SYMROOT = $SRCROOT/build;
-			};
-			name = Release;
-		};
-/* End XCBuildConfiguration section */
-
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-			defaultConfigurationIsVisible = 0;
-			defaultConfigurationName = Release;
-		};
-		9820AACB1C977D11001E5793 /* Build configuration list for PBXNativeTarget "WebRTC" */ = {
-			isa = XCConfigurationList;
-			buildConfigurations = (
-				9820AACC1C977D11001E5793 /* Debug */,
-				9820AACD1C977D11001E5793 /* Release */,
-			);
-			defaultConfigurationIsVisible = 0;
-			defaultConfigurationName = Release;
-		};
-/* End XCConfigurationList section */
-	};
-	rootObject = 9820AABA1C977D11001E5793 /* Project object */;
-}
diff --git a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC.xcodeproj/xcshareddata/xcschemes/WebRTC.xcscheme b/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC.xcodeproj/xcshareddata/xcschemes/WebRTC.xcscheme
deleted file mode 100644
index 2447c448df3..00000000000
--- a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC.xcodeproj/xcshareddata/xcschemes/WebRTC.xcscheme
+++ /dev/null
@@ -1,80 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<Scheme
-   LastUpgradeVersion = "0720"
-   version = "1.3">
-   <BuildAction
-      parallelizeBuildables = "YES"
-      buildImplicitDependencies = "YES">
-      <BuildActionEntries>
-         <BuildActionEntry
-            buildForTesting = "YES"
-            buildForRunning = "YES"
-            buildForProfiling = "YES"
-            buildForArchiving = "YES"
-            buildForAnalyzing = "YES">
-            <BuildableReference
-               BuildableIdentifier = "primary"
-               BlueprintIdentifier = "9820AAC21C977D11001E5793"
-               BuildableName = "WebRTC.framework"
-               BlueprintName = "WebRTC"
-               ReferencedContainer = "container:WebRTC.xcodeproj">
-            </BuildableReference>
-         </BuildActionEntry>
-      </BuildActionEntries>
-   </BuildAction>
-   <TestAction
-      buildConfiguration = "Debug"
-      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.LLDB"
-      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.LLDB"
-      shouldUseLaunchSchemeArgsEnv = "YES">
-      <Testables>
-      </Testables>
-      <AdditionalOptions>
-      </AdditionalOptions>
-   </TestAction>
-   <LaunchAction
-      buildConfiguration = "Release"
-      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.LLDB"
-      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.LLDB"
-      launchStyle = "0"
-      useCustomWorkingDirectory = "NO"
-      ignoresPersistentStateOnLaunch = "NO"
-      debugDocumentVersioning = "YES"
-      debugServiceExtension = "internal"
-      allowLocationSimulation = "YES">
-      <MacroExpansion>
-         <BuildableReference
-            BuildableIdentifier = "primary"
-            BlueprintIdentifier = "9820AAC21C977D11001E5793"
-            BuildableName = "WebRTC.framework"
-            BlueprintName = "WebRTC"
-            ReferencedContainer = "container:WebRTC.xcodeproj">
-         </BuildableReference>
-      </MacroExpansion>
-      <AdditionalOptions>
-      </AdditionalOptions>
-   </LaunchAction>
-   <ProfileAction
-      buildConfiguration = "Release"
-      shouldUseLaunchSchemeArgsEnv = "YES"
-      savedToolIdentifier = ""
-      useCustomWorkingDirectory = "NO"
-      debugDocumentVersioning = "YES">
-      <MacroExpansion>
-         <BuildableReference
-            BuildableIdentifier = "primary"
-            BlueprintIdentifier = "9820AAC21C977D11001E5793"
-            BuildableName = "WebRTC.framework"
-            BlueprintName = "WebRTC"
-            ReferencedContainer = "container:WebRTC.xcodeproj">
-         </BuildableReference>
-      </MacroExpansion>
-   </ProfileAction>
-   <AnalyzeAction
-      buildConfiguration = "Debug">
-   </AnalyzeAction>
-   <ArchiveAction
-      buildConfiguration = "Release"
-      revealArchiveInOrganizer = "YES">
-   </ArchiveAction>
-</Scheme>
diff --git a/chromium/third_party/webrtc/build/ios/SDK/PodTest/Podfile b/chromium/third_party/webrtc/build/ios/SDK/PodTest/Podfile
index 730afcfa185..fec22d917fa 100644
--- a/chromium/third_party/webrtc/build/ios/SDK/PodTest/Podfile
+++ b/chromium/third_party/webrtc/build/ios/SDK/PodTest/Podfile
@@ -4,6 +4,6 @@
 # use_frameworks!
 
 target 'PodTest' do
-pod "WebRTC", :path => "../../../../../out_ios_framework"
+pod "WebRTC", :path => "../../../../../out_ios_libs"
 end
 
diff --git a/chromium/third_party/webrtc/build/ios/SDK/README b/chromium/third_party/webrtc/build/ios/SDK/README
deleted file mode 100644
index c234617bcad..00000000000
--- a/chromium/third_party/webrtc/build/ios/SDK/README
+++ /dev/null
@@ -1,50 +0,0 @@
-We want to generate a dynamic framework for a CocoaPod. Unfortunately, using gyp-xcode generation
-for a framework currently presents some issues. To get around those issues, we chose to create a
-project with Xcode for building the framework directly.
-
-The Framework directory contains the Xcode project used to build the framework. The PodTest
-directory contains a project used to confirm that the built framework installs correctly with
-CocoaPods. The build_ios_framework.sh script needs to be run first, then `pod install` should be run
-in the same directory as the Xcode project. After installing, the PodTest.xcworkspace should be used
-instead of PodTest.xcodeproj.
-
-USAGE
-To build the framework, run the build_ios_framework.sh script. This will build static libraries for
-iOS (via the build_ios_libs.sh script), collect header files and adjust import/include statements
-for use inside the framework, build the framework using the Xcode project, merge multiple
-architectures together, and collect the framework files (the .framework itself and the .dSYM) with
-the Podspec into a common directory.
-
-DETAILS OF THE XCODE PROJECT
-The Xcode project contains relative references to the built static libraries (from
-build_ios_libs.sh), with the exception of librtc_api_objc and librtc_base_objc. The iOS source files
-from webrtc/api/objc and webrtc/base/objc are also included. NOTE: This will require updating
-if/when the built static libraries change.
-
-The flattened header files for webrtc/api/objc and webrtc/base/objc were also added to the Public
-Headers of the framework target. NOTE: This will require updating as the Obj-C API changes.
-
-Preprocessor definitions were copied from a gyp-xcode generated project.
-
-RTTI was disabled due to compiler errors and based on the setting in build/common.gypi
-(https://code.google.com/p/chromium/codesearch#chromium/src/build/common.gypi&q=rtti&sq=package:chromium&type=cs&l=5069).
-
-Bitcode is disabled for the time being.
-
-The minimum number of system frameworks were linked against based on build errors (currently
-AVFoundation, AudioToolbox, CoreMedia, VideoToolbox).
-
-The Build Products Path (SYMROOT) was changed to $SRCROOT/build so the build products are in a known
-location.
-
-The created WebRTC scheme was shared so the build_ios_framework.sh script will work on any machine
-that runs it.
-
-DEPLOYMENT_POSTPROCESSING is set to "Yes" so debug symbols will be stripped (the iOS Default for
-STRIP_INSTALLED_PRODUCT is already set to "Yes").
-
-"-ObjC" is added to OTHER_LDFLAGS to ensure category methods (in particular, those from
-RTCAudioSession+Configuration.mm) are included in the framework binary.
-
-During the build process, dSYMs will be generated for each architecture alongside each framework
-bundle. These are merged together and placed alongside the final framework product.
diff --git a/chromium/third_party/webrtc/build/ios/build_ios_framework.sh b/chromium/third_party/webrtc/build/ios/build_ios_framework.sh
deleted file mode 100755
index 35ec552798c..00000000000
--- a/chromium/third_party/webrtc/build/ios/build_ios_framework.sh
+++ /dev/null
@@ -1,85 +0,0 @@
-#!/bin/bash
-
-#  Copyright 2015 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-
-# Generates dynamic FAT framework for iOS in out_ios_framework.
-
-# Check for Darwin.
-if [[ ! $(uname) = "Darwin" ]]; then
-  echo "OS X required." >&2
-fi
-
-# Check for iOS library build script.
-SCRIPT_DIR=$(dirname $0)
-WEBRTC_BASE_DIR=${SCRIPT_DIR}/../../..
-BUILD_WEBRTC_SCRIPT=${WEBRTC_BASE_DIR}/webrtc/build/ios/build_ios_libs.sh
-if [[ ! -x ${BUILD_WEBRTC_SCRIPT} ]]; then
-  echo "Failed to find iOS library build script." >&2
-  exit 1
-fi
-# Check for flatten iOS headers script.
-FLATTEN_HEADERS_SCRIPT=${WEBRTC_BASE_DIR}/webrtc/build/ios/flatten_ios_headers
-if [[ ! -x ${FLATTEN_HEADERS_SCRIPT} ]]; then
-  echo "Failed to find flatten iOS headers script." >&2
-  exit 1
-fi
-
-pushd ${WEBRTC_BASE_DIR}
-LIB_BASE_DIR=out_ios_libs
-FRAMEWORK_BASE_DIR=out_ios_framework
-
-# Build static libraries for iOS.
-${BUILD_WEBRTC_SCRIPT}
-if [ $? -ne 0 ]; then
-  echo "Failed to build iOS static libraries." >&2
-  exit 1
-fi
-
-# Flatten the directory structure for iOS headers.
-${FLATTEN_HEADERS_SCRIPT} ${LIB_BASE_DIR} ${FRAMEWORK_BASE_DIR}
-if [ $? -ne 0 ]; then
-  echo "Failed to flatten iOS headers." >&2
-  exit 1
-fi
-
-# Replace full paths for headers with framework paths.
-SED_PATTERN='
-  s/(\#import )\"webrtc\/api\/objc\/(.*)\"/\1<WebRTC\/\2>/g;
-  s/(\#import )\"webrtc\/base\/objc\/(.*)\"/\1<WebRTC\/\2>/g;
-  s/(\#include )\"webrtc\/base\/objc\/(.*)\"/\1<WebRTC\/\2>/g;
-'
-sed -E -i '' "$SED_PATTERN" ${FRAMEWORK_BASE_DIR}/include/*.h
-
-SDK_DIR=webrtc/build/ios/SDK
-PROJECT_DIR=${SDK_DIR}/Framework
-# Build the framework.
-pushd ${PROJECT_DIR}
-xcodebuild -project WebRTC.xcodeproj -scheme WebRTC -configuration Release \
-    build CODE_SIGN_IDENTITY="" CODE_SIGNING_REQUIRED=NO
-xcodebuild -project WebRTC.xcodeproj -scheme WebRTC -configuration Release \
-    build -destination 'platform=iOS Simulator,name=iPhone 6'
-popd
-
-# Copy podspec, framework, dSYM and LICENSE to FRAMEWORK_BASE_DIR
-DEVICE_BUILD_DIR=${PROJECT_DIR}/build/Release-iphoneos
-cp ${SDK_DIR}/WebRTC.podspec ${FRAMEWORK_BASE_DIR}/
-cp -R ${DEVICE_BUILD_DIR}/WebRTC.framework ${FRAMEWORK_BASE_DIR}/
-cp -R ${DEVICE_BUILD_DIR}/WebRTC.framework.dSYM ${FRAMEWORK_BASE_DIR}/
-cp -R webrtc/LICENSE ${FRAMEWORK_BASE_DIR}/
-
-# Combine multiple architectures
-SIMULATOR_BUILD_DIR=${PROJECT_DIR}/build/Release-iphonesimulator
-DYLIB_PATH=WebRTC.framework/WebRTC
-DWARF_PATH=WebRTC.framework.dSYM/Contents/Resources/DWARF/WebRTC
-lipo ${FRAMEWORK_BASE_DIR}/${DYLIB_PATH} ${SIMULATOR_BUILD_DIR}/${DYLIB_PATH} \
-    -create -output ${FRAMEWORK_BASE_DIR}/${DYLIB_PATH}
-lipo ${FRAMEWORK_BASE_DIR}/${DWARF_PATH} ${SIMULATOR_BUILD_DIR}/${DWARF_PATH} \
-    -create -output ${FRAMEWORK_BASE_DIR}/${DWARF_PATH}
-
-popd
diff --git a/chromium/third_party/webrtc/build/ios/build_ios_libs.sh b/chromium/third_party/webrtc/build/ios/build_ios_libs.sh
index 5d6157aab2f..772fc6824e4 100755
--- a/chromium/third_party/webrtc/build/ios/build_ios_libs.sh
+++ b/chromium/third_party/webrtc/build/ios/build_ios_libs.sh
@@ -8,79 +8,262 @@
 #  in the file PATENTS.  All contributing project authors may
 #  be found in the AUTHORS file in the root of the source tree.
 
-# Generates static FAT libraries for ios in out_ios_libs.
+# Generates static or dynamic FAT libraries for ios in out_ios_libs.
 
-# Flag to build the new or legacy version of the API.
-USE_LEGACY_API=0
+# Exit on errors.
+set -e
 
-# Check for Darwin.
-if [[ ! $(uname) = "Darwin" ]]; then
-  echo "OS/X required." >&2
-fi
+# Globals.
+SCRIPT_DIR=$(cd $(dirname $0) && pwd)
+WEBRTC_BASE_DIR=${SCRIPT_DIR}/../../..
+GYP_WEBRTC_SCRIPT=${WEBRTC_BASE_DIR}/webrtc/build/gyp_webrtc.py
+MERGE_SCRIPT=${SCRIPT_DIR}/merge_ios_libs.py
+LICENSE_SCRIPT=${SCRIPT_DIR}/generate_licenses.py
 
-# Check for libtool.
-if [[ -z $(which libtool) ]]; then
-  echo "Missing libtool binary." >&2
-fi
+function check_preconditions {
+  # Check for Darwin.
+  if [[ ! $(uname) = "Darwin" ]]; then
+    echo "OS/X required." >&2
+    exit 1
+  fi
 
-# Check for GYP generator.
-SCRIPT_DIR=$(dirname $0)
-WEBRTC_BASE_DIR=${SCRIPT_DIR}/../../..
-GYP_WEBRTC_SCRIPT=${WEBRTC_BASE_DIR}/webrtc/build/gyp_webrtc
-if [[ ! -x ${GYP_WEBRTC_SCRIPT} ]]; then
-  echo "Failed to find gyp generator." >&2
-  exit 1
-fi
-# Check for export headers script.
-EXPORT_HEADERS_SCRIPT=${SCRIPT_DIR}/export_headers
-if [[ ! -x ${EXPORT_HEADERS_SCRIPT} ]]; then
-  echo "Failed to find export headers script." >&2
-  exit 1
-fi
-# Check for merge script.
-MERGE_SCRIPT=${SCRIPT_DIR}/merge_ios_libs
-if [[ ! -x ${MERGE_SCRIPT} ]]; then
-  echo "Failed to find library merging script." >&2
-  exit 1
-fi
+  # Check for libtool.
+  if [[ -z $(which libtool) ]]; then
+    echo "Missing libtool binary." >&2
+    exit 1
+  fi
 
-pushd ${WEBRTC_BASE_DIR}
-LIBRARY_BASE_DIR="out_ios_libs"
+  # Check for GYP generator.
+  if [[ ! -x ${GYP_WEBRTC_SCRIPT} ]]; then
+    echo "Failed to find gyp generator." >&2
+    exit 1
+  fi
+
+  # Check for merge script.
+  if [[ ! -x ${MERGE_SCRIPT} ]]; then
+    echo "Failed to find library merging script." >&2
+    exit 1
+  fi
+}
 
 function build_webrtc {
-  OUTPUT_DIR=$1
-  FLAVOR=$2
-  TARGET_ARCH=$3
-  if [[ ${TARGET_ARCH} = 'arm' || ${TARGET_ARCH} = 'arm64' ]]; then
-    FLAVOR="${FLAVOR}-iphoneos"
+  local base_output_dir=$1
+  local flavor=$2
+  local target_arch=$3
+  local build_type=$4
+
+  local ninja_output_dir=${base_output_dir}/${target_arch}_ninja
+  local library_output_dir=${base_output_dir}/${target_arch}_libs
+  if [[ ${target_arch} = 'arm' || ${target_arch} = 'arm64' ]]; then
+    flavor="${flavor}-iphoneos"
   else
-    FLAVOR="${FLAVOR}-iphonesimulator"
+    flavor="${flavor}-iphonesimulator"
   fi
-  export GYP_DEFINES="OS=ios target_arch=${TARGET_ARCH} use_objc_h264=1 \
-clang_xcode=1 ios_override_visibility=1"
-  export GYP_GENERATORS="ninja"
-  export GYP_GENERATOR_FLAGS="output_dir=${OUTPUT_DIR}"
-  webrtc/build/gyp_webrtc webrtc/build/ios/merge_ios_libs.gyp
-  if [[ ${USE_LEGACY_API} -eq 1 ]]; then
-    ninja -C ${OUTPUT_DIR}/${FLAVOR} libjingle_peerconnection_objc_no_op
+  local ninja_flavor_dir=${ninja_output_dir}/${flavor}
+
+  # Compile framework by default.
+  local gyp_file=webrtc/sdk/sdk.gyp
+  local gyp_target=rtc_sdk_framework_objc
+  # Set to 1 to explicitly not hide symbols. We'll want this if we're just
+  # generating static libs.
+  local override_visibility=0
+  if [[ ${build_type} = "legacy" ]]; then
+    echo "Building legacy."
+    gyp_file=webrtc/build/ios/merge_ios_libs.gyp
+    gyp_target=libjingle_peerconnection_objc_no_op
+    override_visibility=1
+  elif [[ ${build_type} = "static_only" ]]; then
+    echo "Building static only."
+    gyp_file=webrtc/build/ios/merge_ios_libs.gyp
+    gyp_target=rtc_sdk_peerconnection_objc_no_op
+    override_visibility=1
+  elif [[ ${build_type} == "framework" ]]; then
+    echo "Building framework."
   else
-    ninja -C ${OUTPUT_DIR}/${FLAVOR} webrtc_api_objc_no_op
+    echo "Unexpected build type: ${build_type}"
+    exit 1
   fi
-  mkdir -p ${LIBRARY_BASE_DIR}/${TARGET_ARCH}
-  mv ${OUTPUT_DIR}/${FLAVOR}/*.a ${LIBRARY_BASE_DIR}/${TARGET_ARCH}
+
+  export GYP_DEFINES="OS=ios target_arch=${target_arch} use_objc_h264=1 \
+clang_xcode=1 ios_deployment_target=8.0 \
+ios_override_visibility=${override_visibility}"
+  export GYP_GENERATORS="ninja"
+  export GYP_GENERATOR_FLAGS="output_dir=${ninja_output_dir}"
+
+  # GYP generation requires relative path for some reason.
+  pushd ${WEBRTC_BASE_DIR}
+  webrtc/build/gyp_webrtc.py ${gyp_file}
+  popd
+  # Compile the target we're interested in.
+  ninja -C ${ninja_flavor_dir} ${gyp_target}
+
+  if [[ ${build_type} = "framework" ]]; then
+    # Manually generate the dSYM files before stripping them. GYP does not seem
+    # to instruct ninja to generate dSYM files.
+    dsymutil --out=${ninja_flavor_dir}/WebRTC.framework.dSYM \
+        ${ninja_flavor_dir}/WebRTC.framework/WebRTC
+  fi
+
+  # Make links to the generated static archives.
+  mkdir -p ${library_output_dir}
+  for f in ${ninja_flavor_dir}/*.a
+  do
+    ln -sf "${f}" "${library_output_dir}/$(basename ${f})"
+  done
 }
 
+function clean_artifacts {
+  local output_dir=$1
+  if [[ -d ${output_dir} ]]; then
+    rm -r ${output_dir}
+  fi
+}
+
+function usage {
+  echo "WebRTC iOS FAT libraries build script."
+  echo "Each architecture is compiled separately before being merged together."
+  echo "By default, the fat libraries will be created in out_ios_libs/fat_libs."
+  echo "The headers will be copied to out_ios_libs/include."
+  echo "Usage: $0 [-h] [-b build_type] [-c] [-o output_dir]"
+  echo "  -h Print this help."
+  echo "  -b The build type. Can be framework, static_only or legacy."
+  echo "     Defaults to framework."
+  echo "  -c Removes generated build output."
+  echo "  -o Specifies a directory to output build artifacts to."
+  echo "     If specified together with -c, deletes the dir."
+  echo "  -r Specifies a revision number to embed if building the framework."
+  exit 0
+}
+
+check_preconditions
+
+# Set default arguments.
+# Output directory for build artifacts.
+OUTPUT_DIR=${WEBRTC_BASE_DIR}/out_ios_libs
+# The type of build to perform. Valid arguments are framework, static_only and
+# legacy.
+BUILD_TYPE="framework"
+PERFORM_CLEAN=0
+FLAVOR="Profile"
+POINT_VERSION="0"
+
+# Parse arguments.
+while getopts "hb:co:r:" opt; do
+  case "${opt}" in
+    h) usage;;
+    b) BUILD_TYPE="${OPTARG}";;
+    c) PERFORM_CLEAN=1;;
+    o) OUTPUT_DIR="${OPTARG}";;
+    r) POINT_VERSION="${OPTARG}";;
+    *)
+      usage
+      exit 1
+      ;;
+  esac
+done
+
+if [[ ${PERFORM_CLEAN} -ne 0 ]]; then
+  clean_artifacts ${OUTPUT_DIR}
+  exit 0
+fi
+
 # Build all the common architectures.
-build_webrtc "out_ios_arm" "Release" "arm"
-build_webrtc "out_ios_arm64" "Release" "arm64"
-build_webrtc "out_ios_ia32" "Release" "ia32"
-build_webrtc "out_ios_x86_64" "Release" "x64"
+ARCHS=( "arm" "arm64" "ia32" "x64" )
+for ARCH in "${ARCHS[@]}"
+do
+  echo "Building WebRTC arch: ${ARCH}"
+  build_webrtc ${OUTPUT_DIR} ${FLAVOR} $ARCH ${BUILD_TYPE}
+done
+
+ARM_NINJA_DIR=${OUTPUT_DIR}/arm_ninja/${FLAVOR}-iphoneos
+ARM64_NINJA_DIR=${OUTPUT_DIR}/arm64_ninja/${FLAVOR}-iphoneos
+IA32_NINJA_DIR=${OUTPUT_DIR}/ia32_ninja/${FLAVOR}-iphonesimulator
+X64_NINJA_DIR=${OUTPUT_DIR}/x64_ninja/${FLAVOR}-iphonesimulator
+
+if [[ ${BUILD_TYPE} = "framework" ]]; then
+  # Merge the framework slices together into a FAT library by copying one arch
+  # output and merging the rest in.
+  DYLIB_PATH="WebRTC.framework/WebRTC"
+  cp -R ${ARM_NINJA_DIR}/WebRTC.framework ${OUTPUT_DIR}
+  rm ${OUTPUT_DIR}/${DYLIB_PATH}
+  echo "Merging framework slices."
+  lipo ${ARM_NINJA_DIR}/${DYLIB_PATH} \
+      ${ARM64_NINJA_DIR}/${DYLIB_PATH} \
+      ${IA32_NINJA_DIR}/${DYLIB_PATH} \
+      ${X64_NINJA_DIR}/${DYLIB_PATH} \
+      -create -output ${OUTPUT_DIR}/${DYLIB_PATH}
 
-popd
+  # Merge the dSYM files together in a similar fashion.
+  DSYM_PATH="WebRTC.framework.dSYM/Contents/Resources/DWARF/WebRTC"
+  cp -R ${ARM_NINJA_DIR}/WebRTC.framework.dSYM ${OUTPUT_DIR}
+  rm ${OUTPUT_DIR}/${DSYM_PATH}
+  echo "Merging dSYM slices."
+  lipo ${ARM_NINJA_DIR}/${DSYM_PATH} \
+      ${ARM64_NINJA_DIR}/${DSYM_PATH} \
+      ${IA32_NINJA_DIR}/${DSYM_PATH} \
+      ${X64_NINJA_DIR}/${DSYM_PATH} \
+      -create -output ${OUTPUT_DIR}/${DSYM_PATH}
+
+  # Strip the dynamic framework of non-global symbols.
+  # TODO(tkchin): Override chromium strip settings in supplement.gypi instead.
+  echo "Stripping non-global symbols."
+  strip -x ${OUTPUT_DIR}/${DYLIB_PATH}
+
+  # Modify the version number.
+  INFOPLIST_PATH=${OUTPUT_DIR}/WebRTC.framework/Info.plist
+  MAJOR_MINOR=$(plistbuddy -c "Print :CFBundleShortVersionString" \
+                ${INFOPLIST_PATH})
+  VERSION_NUMBER="${MAJOR_MINOR}.${POINT_VERSION}"
+  echo "Substituting revision number: ${VERSION_NUMBER}"
+  plistbuddy -c "Set :CFBundleVersion ${VERSION_NUMBER}" ${INFOPLIST_PATH}
+  plutil -convert binary1 ${INFOPLIST_PATH}
+
+  # Copy pod file.
+  FORMAT_STRING=s/\${FRAMEWORK_VERSION_NUMBER}/${VERSION_NUMBER}/g
+  sed -e ${FORMAT_STRING} ${WEBRTC_BASE_DIR}/webrtc/sdk/objc/WebRTC.podspec > \
+      ${OUTPUT_DIR}/WebRTC.podspec
+else
+  echo "Merging static library slices."
+  # Merge the static libraries together into individual FAT archives.
+  ${MERGE_SCRIPT} ${OUTPUT_DIR}
+
+  # Merge the dSYM files together.
+  TARGET_NAME="rtc_sdk_peerconnection_objc_no_op"
+  if [[ ${BUILD_TYPE} = "legacy" ]]; then
+    TARGET_NAME="libjingle_peerconnection_objc_no_op"
+  fi
+  DSYM_PATH="${TARGET_NAME}.app.dSYM/Contents/Resources/DWARF/${TARGET_NAME}"
+  cp -R ${ARM_NINJA_DIR}/${TARGET_NAME}.app.dSYM ${OUTPUT_DIR}
+  echo "Merging dSYM slices."
+  lipo ${ARM_NINJA_DIR}/${DSYM_PATH} \
+      ${ARM64_NINJA_DIR}/${DSYM_PATH} \
+      ${IA32_NINJA_DIR}/${DSYM_PATH} \
+      ${X64_NINJA_DIR}/${DSYM_PATH} \
+      -create -output ${OUTPUT_DIR}/${DSYM_PATH}
+
+  # Strip debugging symbols.
+  # TODO(tkchin): Override chromium settings in supplement.gypi instead to do
+  # stripping at build time.
+  echo "Stripping debug symbols."
+  strip -S ${OUTPUT_DIR}/fat_libs/*.a
+
+  # Symlink the headers.
+  echo "Symlinking headers."
+  INPUT_HEADER_DIR="${WEBRTC_BASE_DIR}/webrtc/sdk/objc/Framework/Headers/WebRTC"
+  OUTPUT_HEADER_DIR="${OUTPUT_DIR}/include"
+  if [[ -d ${OUTPUT_HEADER_DIR} ]]; then
+    rm -rf ${OUTPUT_HEADER_DIR}
+  fi
+  if [[ ${BUILD_TYPE} = "legacy" ]]; then
+    INPUT_HEADER_DIR="${WEBRTC_BASE_DIR}/talk/app/webrtc/objc/public"
+    ln -sf ${INPUT_HEADER_DIR} ${OUTPUT_HEADER_DIR}
+  else
+    mkdir -p ${OUTPUT_HEADER_DIR}
+    ln -sf ${INPUT_HEADER_DIR} ${OUTPUT_HEADER_DIR}/WebRTC
+  fi
+fi
 
-# Export header files.
-${EXPORT_HEADERS_SCRIPT} ${WEBRTC_BASE_DIR}/${LIBRARY_BASE_DIR} \
-    ${USE_LEGACY_API}
+echo "Generating LICENSE.html."
+${LICENSE_SCRIPT} ${OUTPUT_DIR}/arm64_libs ${OUTPUT_DIR}
 
-# Merge the libraries together.
-${MERGE_SCRIPT} ${WEBRTC_BASE_DIR}/${LIBRARY_BASE_DIR}
+echo "Done!"
diff --git a/chromium/third_party/webrtc/build/ios/export_headers b/chromium/third_party/webrtc/build/ios/export_headers
deleted file mode 100755
index 84e738c0504..00000000000
--- a/chromium/third_party/webrtc/build/ios/export_headers
+++ /dev/null
@@ -1,95 +0,0 @@
-#!/usr/bin/python
-
-#  Copyright 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-
-"""Script for exporting iOS header files."""
-
-import errno
-import optparse
-import os
-import re
-import shutil
-import sys
-
-LEGACY_HEADER_DIRS = ['talk/app/webrtc/objc/public', 'webrtc/base/objc/']
-HEADER_DIRS = ['webrtc/api/objc/', 'webrtc/base/objc/',
-               'webrtc/modules/audio_device/ios/objc']
-# Individual header files that should also be exported.
-LEGACY_HEADER_INCLUDES = []
-HEADER_INCLUDES = []
-# Individual header files that should not be exported.
-LEGACY_HEADER_EXCLUDES = ['talk/app/webrtc/objc/public/RTCNSGLVideoView.h']
-HEADER_EXCLUDES = [
-  'webrtc/api/objc/avfoundationvideocapturer.h',
-  'webrtc/api/objc/RTCNSGLVideoView.h',
-  'webrtc/base/objc/NSString+StdString.h',
-  'webrtc/base/objc/RTCUIApplication.h',
-  'webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h',
-]
-
-def ExportHeaders(include_base_dir, use_legacy_headers):
-  """Exports iOS header files.
-
-  Creates an include directory and recreates the hierarchy for the header files
-  within the include directory.
-
-  Args:
-    include_base_dir: directory where the include directory should be created
-  """
-
-  include_dir_name = 'include'
-  include_path = os.path.join(include_base_dir, include_dir_name)
-  # Remove existing directory first in case files change.
-  if (os.path.exists(include_path)):
-    shutil.rmtree(include_path)
-
-  script_path = sys.path[0]
-  webrtc_base_path = os.path.join(script_path, '../../..')
-
-  header_dirs = HEADER_DIRS
-  include_headers = HEADER_INCLUDES
-  exclude_headers = HEADER_EXCLUDES
-  if use_legacy_headers:
-    header_dirs = LEGACY_HEADER_DIRS
-    include_headers = LEGACY_HEADER_INCLUDES
-    exclude_headers = LEGACY_HEADER_EXCLUDES
-
-  for directory in header_dirs:
-    full_dir_path = os.path.join(webrtc_base_path, directory)
-    filenames = os.listdir(full_dir_path)
-    for filename in filenames:
-      if filename.endswith('.h') and not filename.endswith('+Private.h'):
-        include_headers.append(os.path.join(directory, filename))
-
-  for header in exclude_headers:
-    include_headers.remove(header)
-
-  for header_path in include_headers:
-    output_dir = os.path.join(include_path, os.path.dirname(header_path))
-    # Create hierarchy for the header file within the include directory.
-    try:
-      os.makedirs(output_dir)
-    except OSError as exc:
-      if exc.errno != errno.EEXIST:
-        raise exc
-    current_path = os.path.join(webrtc_base_path, header_path)
-    new_path = os.path.join(include_path, header_path)
-    shutil.copy(current_path, new_path)
-
-def Main():
-  parser = optparse.OptionParser()
-  _, args = parser.parse_args()
-  if len(args) != 2:
-    parser.error('Error: Exactly 2 arguments required.')
-  include_base_dir = args[0]
-  use_legacy_headers = False if int(args[1]) == 0 else True
-  ExportHeaders(include_base_dir, use_legacy_headers)
-
-if __name__ == '__main__':
-  sys.exit(Main())
diff --git a/chromium/third_party/webrtc/build/ios/flatten_ios_headers b/chromium/third_party/webrtc/build/ios/flatten_ios_headers
deleted file mode 100755
index 67c06acc91a..00000000000
--- a/chromium/third_party/webrtc/build/ios/flatten_ios_headers
+++ /dev/null
@@ -1,46 +0,0 @@
-#!/usr/bin/python
-
-#  Copyright 2016 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-
-"""Script for flattening iOS header structure."""
-
-import optparse
-import os
-import shutil
-import sys
-
-def FlattenHeaders(lib_base_dir, framework_base_dir):
-  """Flattens iOS header file directory structure."""
-  include_dir = 'include'
-  unflattened_include_dir_path = os.path.join(lib_base_dir, include_dir)
-  flattened_include_dir_path = os.path.join(framework_base_dir, include_dir)
-
-  # Create output directories.
-  if not os.path.exists(framework_base_dir):
-    os.mkdir(framework_base_dir)
-  if not os.path.exists(flattened_include_dir_path):
-    os.mkdir(flattened_include_dir_path)
-
-  for dirpath, _, filenames in os.walk(unflattened_include_dir_path):
-    for filename in filenames:
-      current_path = os.path.join(dirpath, filename)
-      new_path = os.path.join(flattened_include_dir_path, filename)
-      shutil.copy(current_path, new_path)
-
-def Main():
-  parser = optparse.OptionParser()
-  _, args = parser.parse_args()
-  if len(args) != 2:
-    parser.error('Error: Exactly 2 arguments required.')
-  lib_base_dir = args[0]
-  framework_base_dir = args[1]
-  FlattenHeaders(lib_base_dir, framework_base_dir)
-
-if __name__ == '__main__':
-  sys.exit(Main())
diff --git a/chromium/third_party/webrtc/build/ios/generate_licenses.py b/chromium/third_party/webrtc/build/ios/generate_licenses.py
new file mode 100755
index 00000000000..7a2db5382e3
--- /dev/null
+++ b/chromium/third_party/webrtc/build/ios/generate_licenses.py
@@ -0,0 +1,145 @@
+#!/usr/bin/python
+
+#  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+#
+#  Use of this source code is governed by a BSD-style license
+#  that can be found in the LICENSE file in the root of the source
+#  tree. An additional intellectual property rights grant can be found
+#  in the file PATENTS.  All contributing project authors may
+#  be found in the AUTHORS file in the root of the source tree.
+
+"""Generates license HTML for a prebuilt version of WebRTC for iOS."""
+
+import sys
+
+import argparse
+import cgi
+import fnmatch
+import os
+import re
+import textwrap
+
+
+LIB_TO_LICENSES_DICT = {
+    'boringssl': ['third_party/boringssl/src/LICENSE'],
+    'expat': ['third_party/expat/files/COPYING'],
+    'jsoncpp': ['third_party/jsoncpp/LICENSE'],
+    'opus': ['third_party/opus/src/COPYING'],
+    'protobuf_lite': ['third_party/protobuf/LICENSE'],
+    'srtp': ['third_party/libsrtp/srtp/LICENSE'],
+    'usrsctplib': ['third_party/usrsctp/LICENSE'],
+    'webrtc': ['webrtc/LICENSE', 'webrtc/LICENSE_THIRD_PARTY'],
+    'vpx': ['third_party/libvpx/source/libvpx/LICENSE'],
+    'yuv': ['third_party/libyuv/LICENSE'],
+}
+
+SCRIPT_DIR = os.path.dirname(os.path.realpath(sys.argv[0]))
+CHECKOUT_ROOT = os.path.abspath(os.path.join(SCRIPT_DIR, os.pardir, os.pardir,
+                                             os.pardir))
+TALK_ROOT = os.path.join(CHECKOUT_ROOT, 'talk')
+WEBRTC_ROOT = os.path.join(CHECKOUT_ROOT, 'webrtc')
+
+
+def GetWebRTCGypFilePaths():
+  gyp_filepaths = []
+  search_roots = [TALK_ROOT, WEBRTC_ROOT]
+  for search_root in search_roots:
+    for root, _, filenames in os.walk(search_root):
+      for filename in fnmatch.filter(filenames, '*.gyp*'):
+        gyp_filepaths.append(os.path.join(root, filename))
+  return gyp_filepaths
+
+
+def GetWebRTCTargetNames():
+  gyp_filepaths = GetWebRTCGypFilePaths()
+  target_names = []
+  for gyp_filepath in gyp_filepaths:
+    with open(gyp_filepath, 'r') as gyp_file:
+      for line in gyp_file:
+        match = re.search(r'\'target_name\'.*\'(\w+)\'', line)
+        if match:
+          target_name = match.group(1)
+          target_names.append(target_name)
+  return target_names
+
+
+class LicenseBuilder(object):
+
+  def __init__(self):
+    self.webrtc_target_names = GetWebRTCTargetNames()
+
+  def IsWebRTCLib(self, lib_name):
+    alternate_lib_name = 'lib' + lib_name
+    return (lib_name in self.webrtc_target_names or
+            alternate_lib_name in self.webrtc_target_names)
+
+  def GenerateLicenseText(self, static_lib_dir, output_dir):
+    # Get a list of libs from the files without their prefix and extension.
+    static_libs = []
+    for static_lib in os.listdir(static_lib_dir):
+      # Skip non libraries.
+      if not (static_lib.endswith('.a') and static_lib.startswith('lib')):
+        continue
+      # Extract library name.
+      static_libs.append(static_lib[3:-2])
+
+    # Generate amalgamated list of libraries. Mostly this just collapses the
+    # various WebRTC libs names into just 'webrtc'. Will exit with error if a
+    # lib is unrecognized.
+    license_libs = set()
+    for static_lib in static_libs:
+      license_lib = 'webrtc' if self.IsWebRTCLib(static_lib) else static_lib
+      license_path = LIB_TO_LICENSES_DICT.get(license_lib)
+      if license_path is None:
+        print 'Missing license path for lib: %s' % license_lib
+        return 1
+      license_libs.add(license_lib)
+
+    # Put webrtc at the front of the list.
+    assert 'webrtc' in license_libs
+    license_libs.remove('webrtc')
+    license_libs = sorted(license_libs)
+    license_libs.insert(0, 'webrtc')
+
+    # Generate HTML.
+    output_license_file = open(os.path.join(output_dir, 'LICENSE.html'), 'w+')
+    output_license_file.write('<!DOCTYPE html>\n')
+    output_license_file.write('<html>\n<head>\n')
+    output_license_file.write('<meta charset="UTF-8">\n')
+    output_license_file.write('<title>Licenses</title>\n')
+    style_tag = textwrap.dedent('''\
+    <style>
+      body { margin: 0; font-family: sans-serif; }
+      pre { background-color: #eeeeee; padding: 1em; white-space: pre-wrap; }
+      p { margin: 1em; white-space: nowrap; }
+    </style>
+    ''')
+    output_license_file.write(style_tag)
+    output_license_file.write('</head>\n')
+
+    for license_lib in license_libs:
+      output_license_file.write('<p>%s<br/></p>\n' % license_lib)
+      output_license_file.write('<pre>\n')
+      for path in LIB_TO_LICENSES_DICT[license_lib]:
+        license_path = os.path.join(CHECKOUT_ROOT, path)
+        with open(license_path, 'r') as license_file:
+          license_text = cgi.escape(license_file.read(), quote=True)
+          output_license_file.write(license_text)
+          output_license_file.write('\n')
+      output_license_file.write('</pre>\n')
+
+    output_license_file.write('</body>\n')
+    output_license_file.write('</html>')
+    output_license_file.close()
+    return 0
+
+
+if __name__ == '__main__':
+  parser = argparse.ArgumentParser(description='Generate WebRTC LICENSE.html')
+  parser.add_argument('static_lib_dir',
+                      help='Directory with built static libraries.')
+  parser.add_argument('output_dir',
+                      help='Directory to output LICENSE.html to.')
+  args = parser.parse_args()
+  builder = LicenseBuilder()
+  sys.exit(builder.GenerateLicenseText(args.static_lib_dir, args.output_dir))
diff --git a/chromium/third_party/webrtc/build/ios/merge_ios_libs.gyp b/chromium/third_party/webrtc/build/ios/merge_ios_libs.gyp
index cc24d6530ef..5b0a9cbbb09 100644
--- a/chromium/third_party/webrtc/build/ios/merge_ios_libs.gyp
+++ b/chromium/third_party/webrtc/build/ios/merge_ios_libs.gyp
@@ -21,11 +21,11 @@
           'sources': ['no_op.cc',],
         },
         {
-          'target_name': 'webrtc_api_objc_no_op',
+          'target_name': 'rtc_sdk_peerconnection_objc_no_op',
           'includes': [ 'objc_app.gypi' ],
           'type': 'executable',
           'dependencies': [
-            '<(webrtc_root)/api/api.gyp:rtc_api_objc',
+            '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_peerconnection_objc',
           ],
           'sources': ['no_op.cc',],
         },
diff --git a/chromium/third_party/webrtc/build/ios/merge_ios_libs b/chromium/third_party/webrtc/build/ios/merge_ios_libs.py
index d96d5e3c0c6..651024eb6ac 100755
--- a/chromium/third_party/webrtc/build/ios/merge_ios_libs
+++ b/chromium/third_party/webrtc/build/ios/merge_ios_libs.py
@@ -10,11 +10,15 @@
 
 """Script for merging generated iOS libraries."""
 
-import optparse
+import sys
+
+import argparse
 import os
 import re
 import subprocess
-import sys
+
+# Valid arch subdir names.
+VALID_ARCHS = ['arm_libs', 'arm64_libs', 'ia32_libs', 'x64_libs']
 
 
 def MergeLibs(lib_base_dir):
@@ -29,24 +33,22 @@ def MergeLibs(lib_base_dir):
   Returns:
     Exit code of libtool.
   """
-  include_dir_name = 'include'
-  output_dir_name = 'lib'
+  output_dir_name = 'fat_libs'
   archs = [arch for arch in os.listdir(lib_base_dir)
-           if arch[:1] != '.' and arch != output_dir_name
-           and arch != include_dir_name]
+           if arch in VALID_ARCHS]
   # For each arch, find (library name, libary path) for arch. We will merge
   # all libraries with the same name.
   libs = {}
-  for dirpath, _, filenames in os.walk(lib_base_dir):
-    if dirpath.endswith(output_dir_name):
+  for lib_dir in [os.path.join(lib_base_dir, arch) for arch in VALID_ARCHS]:
+    if not os.path.exists(lib_dir):
       continue
-    for filename in filenames:
-      if not filename.endswith('.a'):
-        continue
-      entry = libs.get(filename, [])
-      entry.append(os.path.join(dirpath, filename))
-      libs[filename] = entry
-
+    for dirpath, _, filenames in os.walk(lib_dir):
+      for filename in filenames:
+        if not filename.endswith('.a'):
+          continue
+        entry = libs.get(filename, [])
+        entry.append(os.path.join(dirpath, filename))
+        libs[filename] = entry
   orphaned_libs = {}
   valid_libs = {}
   for library, paths in libs.items():
@@ -69,7 +71,6 @@ def MergeLibs(lib_base_dir):
     if not found:
       base_prefix = library[:-2].split('_')[0]
       for valid_lib, valid_paths in valid_libs.items():
-        prefix = '_'.join(components)
         if valid_lib[:len(base_prefix)] == base_prefix:
           valid_paths.extend(paths)
           found = True
@@ -89,6 +90,7 @@ def MergeLibs(lib_base_dir):
   libtool_re = re.compile(r'^.*libtool:.*file: .* has no symbols$')
 
   # Merge libraries using libtool.
+  libtool_returncode = 0
   for library, paths in valid_libs.items():
     cmd_list = ['libtool', '-static', '-v', '-o',
                 os.path.join(output_dir_path, library)] + paths
@@ -99,22 +101,23 @@ def MergeLibs(lib_base_dir):
         print >>sys.stderr, line
     # Unconditionally touch the output .a file on the command line if present
     # and the command succeeded. A bit hacky.
-    if not libtoolout.returncode:
+    libtool_returncode = libtoolout.returncode
+    if not libtool_returncode:
       for i in range(len(cmd_list) - 1):
         if cmd_list[i] == '-o' and cmd_list[i+1].endswith('.a'):
           os.utime(cmd_list[i+1], None)
           break
-    else:
-      return libtoolout.returncode
-  return libtoolout.returncode
+  return libtool_returncode
 
 
 def Main():
-  parser = optparse.OptionParser()
-  _, args = parser.parse_args()
-  if len(args) != 1:
-    parser.error('Error: Exactly 1 argument required.')
-  lib_base_dir = args[0]
+  parser_description = 'Merge WebRTC libraries.'
+  parser = argparse.ArgumentParser(description=parser_description)
+  parser.add_argument('lib_base_dir',
+                      help='Directory with built libraries. ',
+                      type=str)
+  args = parser.parse_args()
+  lib_base_dir = args.lib_base_dir
   MergeLibs(lib_base_dir)
 
 if __name__ == '__main__':
diff --git a/chromium/third_party/webrtc/build/isolate.gypi b/chromium/third_party/webrtc/build/isolate.gypi
index ea44e2cc7f3..2d4ea778697 100644
--- a/chromium/third_party/webrtc/build/isolate.gypi
+++ b/chromium/third_party/webrtc/build/isolate.gypi
@@ -107,7 +107,6 @@
         '--config-variable', 'use_instrumented_libraries=<(use_instrumented_libraries)',
         '--config-variable',
         'use_prebuilt_instrumented_libraries=<(use_prebuilt_instrumented_libraries)',
-        '--config-variable', 'use_openssl=<(use_openssl)',
         '--config-variable', 'use_ozone=<(use_ozone)',
         '--config-variable', 'use_x11=<(use_x11)',
         '--config-variable', 'v8_use_external_startup_data=<(v8_use_external_startup_data)',
diff --git a/chromium/third_party/webrtc/build/objc_common.gypi b/chromium/third_party/webrtc/build/objc_common.gypi
index 086210ee757..ff908cfe629 100644
--- a/chromium/third_party/webrtc/build/objc_common.gypi
+++ b/chromium/third_party/webrtc/build/objc_common.gypi
@@ -10,7 +10,7 @@
 
 {
   'variables': {
-    'objc_prefix_file': './WebRTC-Prefix.pch',
+    'objc_prefix_file': '../sdk/objc/WebRTC-Prefix.pch',
   },
   'xcode_settings': {
     'CLANG_ENABLE_OBJC_ARC': 'YES',
diff --git a/chromium/third_party/webrtc/build/protoc.gypi b/chromium/third_party/webrtc/build/protoc.gypi
index 682bc22cc5c..6e9af457aaf 100644
--- a/chromium/third_party/webrtc/build/protoc.gypi
+++ b/chromium/third_party/webrtc/build/protoc.gypi
@@ -12,6 +12,11 @@
 # build/common.gypi is different for the standalone and Chromium builds. Gyp
 # doesn't permit conditional inclusion or variable expansion in include paths.
 # http://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
+#
+# Local changes:
+# * Removed <(DEPTH) from include_dir due to difficulties with generated
+#   downstream code.
+
 
 # This file is meant to be included into a target to provide a rule
 # to invoke protoc in a consistent manner. For Java-targets, see
@@ -111,12 +116,10 @@
   ],
   'include_dirs': [
     '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
-    '<(DEPTH)',
   ],
   'direct_dependent_settings': {
     'include_dirs': [
       '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
-      '<(DEPTH)',
     ]
   },
   # This target exports a hard dependency because it generates header
diff --git a/chromium/third_party/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc b/chromium/third_party/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
index 30a9a4572eb..fc7ddbf23b7 100644
--- a/chromium/third_party/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
+++ b/chromium/third_party/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
@@ -76,9 +76,7 @@ char kTSanDefaultSuppressions[] =
 
 // Potential deadlocks detected after roll in r6516.
 // https://code.google.com/p/webrtc/issues/detail?id=3509
-"deadlock:webrtc::RTCPReceiver::SetSsrcs\n"
 "deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n"
-"deadlock:webrtc::vcm::VideoReceiver::RegisterPacketRequestCallback\n"
 
 // TODO(pbos): Trace events are racy due to lack of proper POD atomics.
 // https://code.google.com/p/webrtc/issues/detail?id=2497
diff --git a/chromium/third_party/webrtc/build/webrtc.gni b/chromium/third_party/webrtc/build/webrtc.gni
index 8e1b9521f74..788e60824c5 100644
--- a/chromium/third_party/webrtc/build/webrtc.gni
+++ b/chromium/third_party/webrtc/build/webrtc.gni
@@ -36,7 +36,6 @@ declare_args() {
   # Disable these to not build components which can be externally provided.
   rtc_build_expat = true
   rtc_build_json = true
-  rtc_build_libjpeg = true
   rtc_build_libvpx = true
   rtc_build_libyuv = true
   rtc_build_openmax_dl = true
@@ -59,8 +58,16 @@ declare_args() {
   rtc_include_tests = false
   rtc_restrict_logging = true
 
+  # Enable libevent task queues on platforms that support it.
+  if (is_win || is_mac || is_ios || is_nacl) {
+    rtc_enable_libevent = false
+    rtc_build_libevent = false
+  } else {
+    rtc_enable_libevent = true
+    rtc_build_libevent = true
+  }
+
   if (is_ios) {
-    rtc_build_libjpeg = false
     rtc_enable_protobuf = false
   }
 
@@ -78,8 +85,7 @@ declare_args() {
 
   # Determines whether NEON code will be built.
   rtc_build_with_neon =
-      (current_cpu == "arm" && (arm_use_neon || arm_optionally_use_neon)) ||
-      current_cpu == "arm64"
+      (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64"
 
   # Enable this to use HW H.264 encoder/decoder on iOS PeerConnections.
   # Enabling this may break interop with Android clients that support H264.
diff --git a/chromium/third_party/webrtc/call.h b/chromium/third_party/webrtc/call.h
index 3ba473fec07..80134fa27d4 100644
--- a/chromium/third_party/webrtc/call.h
+++ b/chromium/third_party/webrtc/call.h
@@ -17,6 +17,7 @@
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/audio_state.h"
+#include "webrtc/base/networkroute.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
@@ -140,6 +141,10 @@ class Call {
   virtual void SignalChannelNetworkState(MediaType media,
                                          NetworkState state) = 0;
 
+  virtual void OnNetworkRouteChanged(
+      const std::string& transport_name,
+      const rtc::NetworkRoute& network_route) = 0;
+
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
diff --git a/chromium/third_party/webrtc/call/bitrate_allocator.cc b/chromium/third_party/webrtc/call/bitrate_allocator.cc
index 097378f02a1..3672ef520ca 100644
--- a/chromium/third_party/webrtc/call/bitrate_allocator.cc
+++ b/chromium/third_party/webrtc/call/bitrate_allocator.cc
@@ -14,6 +14,7 @@
 #include <algorithm>
 #include <utility>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 
 namespace webrtc {
@@ -24,10 +25,10 @@ const int kTransmissionMaxBitrateMultiplier = 2;
 const int kDefaultBitrateBps = 300000;
 
 BitrateAllocator::BitrateAllocator()
-    : bitrate_observers_(),
-      bitrate_observers_modified_(false),
+    : bitrate_observer_configs_(),
       enforce_min_bitrate_(true),
       last_bitrate_bps_(kDefaultBitrateBps),
+      last_non_zero_bitrate_bps_(kDefaultBitrateBps),
       last_fraction_loss_(0),
       last_rtt_(0) {}
 
@@ -36,10 +37,13 @@ uint32_t BitrateAllocator::OnNetworkChanged(uint32_t bitrate,
                                             int64_t rtt) {
   rtc::CritScope lock(&crit_sect_);
   last_bitrate_bps_ = bitrate;
+  last_non_zero_bitrate_bps_ =
+      bitrate > 0 ? bitrate : last_non_zero_bitrate_bps_;
   last_fraction_loss_ = fraction_loss;
   last_rtt_ = rtt;
+
   uint32_t allocated_bitrate_bps = 0;
-  ObserverBitrateMap allocation = AllocateBitrates();
+  ObserverAllocation allocation = AllocateBitrates(bitrate);
   for (const auto& kv : allocation) {
     kv.first->OnBitrateUpdated(kv.second, last_fraction_loss_, last_rtt_);
     allocated_bitrate_bps += kv.second;
@@ -47,121 +51,123 @@ uint32_t BitrateAllocator::OnNetworkChanged(uint32_t bitrate,
   return allocated_bitrate_bps;
 }
 
-BitrateAllocator::ObserverBitrateMap BitrateAllocator::AllocateBitrates() {
-  if (bitrate_observers_.empty())
-    return ObserverBitrateMap();
-
-  uint32_t sum_min_bitrates = 0;
-  for (const auto& observer : bitrate_observers_)
-    sum_min_bitrates += observer.second.min_bitrate;
-  if (last_bitrate_bps_ <= sum_min_bitrates)
-    return LowRateAllocation(last_bitrate_bps_);
-  else
-    return NormalRateAllocation(last_bitrate_bps_, sum_min_bitrates);
-}
-
 int BitrateAllocator::AddObserver(BitrateAllocatorObserver* observer,
                                   uint32_t min_bitrate_bps,
-                                  uint32_t max_bitrate_bps) {
+                                  uint32_t max_bitrate_bps,
+                                  bool enforce_min_bitrate) {
   rtc::CritScope lock(&crit_sect_);
+  // TODO(mflodman): Enforce this per observer.
+  EnforceMinBitrate(enforce_min_bitrate);
 
-  BitrateObserverConfList::iterator it =
-      FindObserverConfigurationPair(observer);
+  auto it = FindObserverConfig(observer);
 
   // Allow the max bitrate to be exceeded for FEC and retransmissions.
   // TODO(holmer): We have to get rid of this hack as it makes it difficult to
   // properly allocate bitrate. The allocator should instead distribute any
   // extra bitrate after all streams have maxed out.
   max_bitrate_bps *= kTransmissionMaxBitrateMultiplier;
-  if (it != bitrate_observers_.end()) {
+  if (it != bitrate_observer_configs_.end()) {
     // Update current configuration.
-    it->second.min_bitrate = min_bitrate_bps;
-    it->second.max_bitrate = max_bitrate_bps;
+    it->min_bitrate_bps = min_bitrate_bps;
+    it->max_bitrate_bps = max_bitrate_bps;
   } else {
     // Add new settings.
-    bitrate_observers_.push_back(BitrateObserverConfiguration(
-        observer, BitrateConfiguration(min_bitrate_bps, max_bitrate_bps)));
-    bitrate_observers_modified_ = true;
+    bitrate_observer_configs_.push_back(ObserverConfig(
+        observer, min_bitrate_bps, max_bitrate_bps, enforce_min_bitrate));
   }
 
-  ObserverBitrateMap allocation = AllocateBitrates();
   int new_observer_bitrate_bps = 0;
-  for (auto& kv : allocation) {
-    kv.first->OnBitrateUpdated(kv.second, last_fraction_loss_, last_rtt_);
-    if (kv.first == observer)
-      new_observer_bitrate_bps = kv.second;
+  if (last_bitrate_bps_ > 0) {  // We have a bitrate to allocate.
+    ObserverAllocation allocation = AllocateBitrates(last_bitrate_bps_);
+    for (auto& kv : allocation) {
+      // Update all observers with the new allocation.
+      kv.first->OnBitrateUpdated(kv.second, last_fraction_loss_, last_rtt_);
+      if (kv.first == observer)
+        new_observer_bitrate_bps = kv.second;
+    }
+  } else {
+    // Currently, an encoder is not allowed to produce frames.
+    // But we still have to return the initial config bitrate + let the
+    // observer know that it can not produce frames.
+    ObserverAllocation allocation =
+        AllocateBitrates(last_non_zero_bitrate_bps_);
+    observer->OnBitrateUpdated(0, last_fraction_loss_, last_rtt_);
+    new_observer_bitrate_bps = allocation[observer];
   }
   return new_observer_bitrate_bps;
 }
 
 void BitrateAllocator::RemoveObserver(BitrateAllocatorObserver* observer) {
   rtc::CritScope lock(&crit_sect_);
-  BitrateObserverConfList::iterator it =
-      FindObserverConfigurationPair(observer);
-  if (it != bitrate_observers_.end()) {
-    bitrate_observers_.erase(it);
-    bitrate_observers_modified_ = true;
+  auto it = FindObserverConfig(observer);
+  if (it != bitrate_observer_configs_.end()) {
+    bitrate_observer_configs_.erase(it);
   }
 }
 
-void BitrateAllocator::GetMinMaxBitrateSumBps(int* min_bitrate_sum_bps,
-                                              int* max_bitrate_sum_bps) const {
-  *min_bitrate_sum_bps = 0;
-  *max_bitrate_sum_bps = 0;
-
-  rtc::CritScope lock(&crit_sect_);
-  for (const auto& observer : bitrate_observers_) {
-    *min_bitrate_sum_bps += observer.second.min_bitrate;
-    *max_bitrate_sum_bps += observer.second.max_bitrate;
-  }
+void BitrateAllocator::EnforceMinBitrate(bool enforce_min_bitrate) {
+  enforce_min_bitrate_ = enforce_min_bitrate;
 }
 
-BitrateAllocator::BitrateObserverConfList::iterator
-BitrateAllocator::FindObserverConfigurationPair(
+BitrateAllocator::ObserverConfigList::iterator
+BitrateAllocator::FindObserverConfig(
     const BitrateAllocatorObserver* observer) {
-  for (auto it = bitrate_observers_.begin(); it != bitrate_observers_.end();
-       ++it) {
-    if (it->first == observer)
+  for (auto it = bitrate_observer_configs_.begin();
+       it != bitrate_observer_configs_.end(); ++it) {
+    if (it->observer == observer)
       return it;
   }
-  return bitrate_observers_.end();
+  return bitrate_observer_configs_.end();
 }
 
-void BitrateAllocator::EnforceMinBitrate(bool enforce_min_bitrate) {
-  rtc::CritScope lock(&crit_sect_);
-  enforce_min_bitrate_ = enforce_min_bitrate;
+BitrateAllocator::ObserverAllocation BitrateAllocator::AllocateBitrates(
+    uint32_t bitrate) {
+  if (bitrate_observer_configs_.empty())
+    return ObserverAllocation();
+
+  if (bitrate == 0)
+    return ZeroRateAllocation();
+
+  uint32_t sum_min_bitrates = 0;
+  for (const auto& observer_config : bitrate_observer_configs_)
+    sum_min_bitrates += observer_config.min_bitrate_bps;
+  if (bitrate <= sum_min_bitrates)
+    return LowRateAllocation(bitrate);
+
+  return NormalRateAllocation(bitrate, sum_min_bitrates);
 }
 
-BitrateAllocator::ObserverBitrateMap BitrateAllocator::NormalRateAllocation(
+BitrateAllocator::ObserverAllocation BitrateAllocator::NormalRateAllocation(
     uint32_t bitrate,
     uint32_t sum_min_bitrates) {
-  uint32_t number_of_observers =
-      static_cast<uint32_t>(bitrate_observers_.size());
+  uint32_t num_remaining_observers =
+      static_cast<uint32_t>(bitrate_observer_configs_.size());
+  RTC_DCHECK_GT(num_remaining_observers, 0u);
+
   uint32_t bitrate_per_observer =
-      (bitrate - sum_min_bitrates) / number_of_observers;
+      (bitrate - sum_min_bitrates) / num_remaining_observers;
   // Use map to sort list based on max bitrate.
   ObserverSortingMap list_max_bitrates;
-  for (const auto& observer : bitrate_observers_) {
-    list_max_bitrates.insert(std::pair<uint32_t, ObserverConfiguration>(
-        observer.second.max_bitrate,
-        ObserverConfiguration(observer.first, observer.second.min_bitrate)));
+  for (const auto& config : bitrate_observer_configs_) {
+    list_max_bitrates.insert(std::pair<uint32_t, const ObserverConfig*>(
+        config.max_bitrate_bps, &config));
   }
-  ObserverBitrateMap allocation;
+
+  ObserverAllocation allocation;
   ObserverSortingMap::iterator max_it = list_max_bitrates.begin();
   while (max_it != list_max_bitrates.end()) {
-    number_of_observers--;
+    num_remaining_observers--;
     uint32_t observer_allowance =
-        max_it->second.min_bitrate + bitrate_per_observer;
+        max_it->second->min_bitrate_bps + bitrate_per_observer;
     if (max_it->first < observer_allowance) {
       // We have more than enough for this observer.
       // Carry the remainder forward.
       uint32_t remainder = observer_allowance - max_it->first;
-      if (number_of_observers != 0) {
-        bitrate_per_observer += remainder / number_of_observers;
-      }
-      allocation[max_it->second.observer] = max_it->first;
+      if (num_remaining_observers != 0)
+        bitrate_per_observer += remainder / num_remaining_observers;
+      allocation[max_it->second->observer] = max_it->first;
     } else {
-      allocation[max_it->second.observer] = observer_allowance;
+      allocation[max_it->second->observer] = observer_allowance;
     }
     list_max_bitrates.erase(max_it);
     // Prepare next iteration.
@@ -170,21 +176,29 @@ BitrateAllocator::ObserverBitrateMap BitrateAllocator::NormalRateAllocation(
   return allocation;
 }
 
-BitrateAllocator::ObserverBitrateMap BitrateAllocator::LowRateAllocation(
+BitrateAllocator::ObserverAllocation BitrateAllocator::ZeroRateAllocation() {
+  ObserverAllocation allocation;
+  // Zero bitrate to all observers.
+  for (const auto& observer_config : bitrate_observer_configs_)
+    allocation[observer_config.observer] = 0;
+  return allocation;
+}
+
+BitrateAllocator::ObserverAllocation BitrateAllocator::LowRateAllocation(
     uint32_t bitrate) {
-  ObserverBitrateMap allocation;
+  ObserverAllocation allocation;
   if (enforce_min_bitrate_) {
     // Min bitrate to all observers.
-    for (const auto& observer : bitrate_observers_)
-      allocation[observer.first] = observer.second.min_bitrate;
+    for (const auto& observer_config : bitrate_observer_configs_)
+      allocation[observer_config.observer] = observer_config.min_bitrate_bps;
   } else {
-    // Allocate up to |min_bitrate| to one observer at a time, until
+    // Allocate up to |min_bitrate_bps| to one observer at a time, until
     // |bitrate| is depleted.
     uint32_t remainder = bitrate;
-    for (const auto& observer : bitrate_observers_) {
+    for (const auto& observer_config : bitrate_observer_configs_) {
       uint32_t allocated_bitrate =
-          std::min(remainder, observer.second.min_bitrate);
-      allocation[observer.first] = allocated_bitrate;
+          std::min(remainder, observer_config.min_bitrate_bps);
+      allocation[observer_config.observer] = allocated_bitrate;
       remainder -= allocated_bitrate;
     }
   }
diff --git a/chromium/third_party/webrtc/call/bitrate_allocator.h b/chromium/third_party/webrtc/call/bitrate_allocator.h
index 404a312dad3..3bafcd756be 100644
--- a/chromium/third_party/webrtc/call/bitrate_allocator.h
+++ b/chromium/third_party/webrtc/call/bitrate_allocator.h
@@ -53,59 +53,69 @@ class BitrateAllocator {
   // |observer| updates bitrates if already in use.
   // |min_bitrate_bps| = 0 equals no min bitrate.
   // |max_bitrate_bps| = 0 equals no max bitrate.
-  // Returns bitrate allocated for the bitrate observer.
+  // |enforce_min_bitrate| = 'true' will allocate at least |min_bitrate_bps| for
+  //    this observer, even if the BWE is too low, 'false' will allocate 0 to
+  //    the observer if BWE doesn't allow |min_bitrate_bps|.
+  // Returns initial bitrate allocated for |observer|.
+  // Note that |observer|->OnBitrateUpdated() will be called within the scope of
+  // this method with the current rtt, fraction_loss and available bitrate and
+  // that the bitrate in OnBitrateUpdated will be zero if the |observer| is
+  // currently not allowed to send data.
   int AddObserver(BitrateAllocatorObserver* observer,
                   uint32_t min_bitrate_bps,
-                  uint32_t max_bitrate_bps);
+                  uint32_t max_bitrate_bps,
+                  bool enforce_min_bitrate);
 
   void RemoveObserver(BitrateAllocatorObserver* observer);
 
-  void GetMinMaxBitrateSumBps(int* min_bitrate_sum_bps,
-                              int* max_bitrate_sum_bps) const;
+ private:
+  struct ObserverConfig {
+    ObserverConfig(BitrateAllocatorObserver* observer,
+                   uint32_t min_bitrate_bps,
+                   uint32_t max_bitrate_bps,
+                   bool enforce_min_bitrate)
+        : observer(observer),
+          min_bitrate_bps(min_bitrate_bps),
+          max_bitrate_bps(max_bitrate_bps),
+          enforce_min_bitrate(enforce_min_bitrate) {}
+    BitrateAllocatorObserver* const observer;
+    uint32_t min_bitrate_bps;
+    uint32_t max_bitrate_bps;
+    bool enforce_min_bitrate;
+  };
 
   // This method controls the behavior when the available bitrate is lower than
   // the minimum bitrate, or the sum of minimum bitrates.
   // When true, the bitrate will never be set lower than the minimum bitrate(s).
   // When false, the bitrate observers will be allocated rates up to their
   // respective minimum bitrate, satisfying one observer after the other.
-  void EnforceMinBitrate(bool enforce_min_bitrate);
-
- private:
-  struct BitrateConfiguration {
-    BitrateConfiguration(uint32_t min_bitrate, uint32_t max_bitrate)
-        : min_bitrate(min_bitrate), max_bitrate(max_bitrate) {}
-    uint32_t min_bitrate;
-    uint32_t max_bitrate;
-  };
-  struct ObserverConfiguration {
-    ObserverConfiguration(BitrateAllocatorObserver* observer, uint32_t bitrate)
-        : observer(observer), min_bitrate(bitrate) {}
-    BitrateAllocatorObserver* const observer;
-    uint32_t min_bitrate;
-  };
-  typedef std::pair<BitrateAllocatorObserver*, BitrateConfiguration>
-      BitrateObserverConfiguration;
-  typedef std::list<BitrateObserverConfiguration> BitrateObserverConfList;
-  typedef std::multimap<uint32_t, ObserverConfiguration> ObserverSortingMap;
-  typedef std::map<BitrateAllocatorObserver*, int> ObserverBitrateMap;
+  void EnforceMinBitrate(bool enforce_min_bitrate)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
-  BitrateObserverConfList::iterator FindObserverConfigurationPair(
+  typedef std::list<ObserverConfig> ObserverConfigList;
+  ObserverConfigList::iterator FindObserverConfig(
       const BitrateAllocatorObserver* observer)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-  ObserverBitrateMap AllocateBitrates() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-  ObserverBitrateMap NormalRateAllocation(uint32_t bitrate,
+
+  typedef std::multimap<uint32_t, const ObserverConfig*> ObserverSortingMap;
+  typedef std::map<BitrateAllocatorObserver*, int> ObserverAllocation;
+
+  ObserverAllocation AllocateBitrates(uint32_t bitrate)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  ObserverAllocation NormalRateAllocation(uint32_t bitrate,
                                           uint32_t sum_min_bitrates)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
-  ObserverBitrateMap LowRateAllocation(uint32_t bitrate)
+  ObserverAllocation ZeroRateAllocation() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  ObserverAllocation LowRateAllocation(uint32_t bitrate)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   rtc::CriticalSection crit_sect_;
   // Stored in a list to keep track of the insertion order.
-  BitrateObserverConfList bitrate_observers_ GUARDED_BY(crit_sect_);
-  bool bitrate_observers_modified_ GUARDED_BY(crit_sect_);
+  ObserverConfigList bitrate_observer_configs_;
   bool enforce_min_bitrate_ GUARDED_BY(crit_sect_);
   uint32_t last_bitrate_bps_ GUARDED_BY(crit_sect_);
+  uint32_t last_non_zero_bitrate_bps_ GUARDED_BY(crit_sect_);
   uint8_t last_fraction_loss_ GUARDED_BY(crit_sect_);
   int64_t last_rtt_ GUARDED_BY(crit_sect_);
 };
diff --git a/chromium/third_party/webrtc/call/bitrate_allocator_unittest.cc b/chromium/third_party/webrtc/call/bitrate_allocator_unittest.cc
index 6e0cdd4d781..4017645c1d6 100644
--- a/chromium/third_party/webrtc/call/bitrate_allocator_unittest.cc
+++ b/chromium/third_party/webrtc/call/bitrate_allocator_unittest.cc
@@ -48,7 +48,7 @@ class BitrateAllocatorTest : public ::testing::Test {
 TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
   TestBitrateObserver bitrate_observer;
   int start_bitrate =
-      allocator_->AddObserver(&bitrate_observer, 100000, 1500000);
+      allocator_->AddObserver(&bitrate_observer, 100000, 1500000, true);
   EXPECT_EQ(300000, start_bitrate);
   allocator_->OnNetworkChanged(200000, 0, 0);
   EXPECT_EQ(200000u, bitrate_observer.last_bitrate_);
@@ -57,10 +57,12 @@ TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
   // bitrate for FEC/retransmissions (see todo in BitrateAllocator).
   allocator_->OnNetworkChanged(4000000, 0, 0);
   EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
-  start_bitrate = allocator_->AddObserver(&bitrate_observer, 100000, 4000000);
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer, 100000, 4000000, true);
   EXPECT_EQ(4000000, start_bitrate);
 
-  start_bitrate = allocator_->AddObserver(&bitrate_observer, 100000, 1500000);
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer, 100000, 1500000, true);
   EXPECT_EQ(3000000, start_bitrate);
   EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
   allocator_->OnNetworkChanged(1500000, 0, 0);
@@ -71,9 +73,10 @@ TEST_F(BitrateAllocatorTest, TwoBitrateObserversOneRtcpObserver) {
   TestBitrateObserver bitrate_observer_1;
   TestBitrateObserver bitrate_observer_2;
   int start_bitrate =
-      allocator_->AddObserver(&bitrate_observer_1, 100000, 300000);
+      allocator_->AddObserver(&bitrate_observer_1, 100000, 300000, true);
   EXPECT_EQ(300000, start_bitrate);
-  start_bitrate = allocator_->AddObserver(&bitrate_observer_2, 200000, 300000);
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer_2, 200000, 300000, true);
   EXPECT_EQ(200000, start_bitrate);
 
   // Test too low start bitrate, hence lower than sum of min. Min bitrates will
@@ -96,12 +99,17 @@ TEST_F(BitrateAllocatorTest, TwoBitrateObserversOneRtcpObserver) {
   allocator_->OnNetworkChanged(1500000, 0, 50);
   EXPECT_EQ(600000u, bitrate_observer_1.last_bitrate_);
   EXPECT_EQ(600000u, bitrate_observer_2.last_bitrate_);
+
+  // Verify that if the bandwidth estimate is set to zero, the allocated rate is
+  // zero.
+  allocator_->OnNetworkChanged(0, 0, 50);
+  EXPECT_EQ(0u, bitrate_observer_1.last_bitrate_);
+  EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_);
 }
 
 class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
  protected:
   BitrateAllocatorTestNoEnforceMin() : allocator_(new BitrateAllocator()) {
-    allocator_->EnforceMinBitrate(false);
     allocator_->OnNetworkChanged(300000u, 0, 0);
   }
   ~BitrateAllocatorTestNoEnforceMin() {}
@@ -114,7 +122,7 @@ class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
 TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserver) {
   TestBitrateObserver bitrate_observer_1;
   int start_bitrate =
-      allocator_->AddObserver(&bitrate_observer_1, 100000, 400000);
+      allocator_->AddObserver(&bitrate_observer_1, 100000, 400000, false);
   EXPECT_EQ(300000, start_bitrate);
 
   // High REMB.
@@ -134,14 +142,16 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, ThreeBitrateObservers) {
   TestBitrateObserver bitrate_observer_3;
   // Set up the observers with min bitrates at 100000, 200000, and 300000.
   int start_bitrate =
-      allocator_->AddObserver(&bitrate_observer_1, 100000, 400000);
+      allocator_->AddObserver(&bitrate_observer_1, 100000, 400000, false);
   EXPECT_EQ(300000, start_bitrate);
 
-  start_bitrate = allocator_->AddObserver(&bitrate_observer_2, 200000, 400000);
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer_2, 200000, 400000, false);
   EXPECT_EQ(200000, start_bitrate);
   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
 
-  start_bitrate = allocator_->AddObserver(&bitrate_observer_3, 300000, 400000);
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer_3, 300000, 400000, false);
   EXPECT_EQ(0, start_bitrate);
   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
   EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);
@@ -171,6 +181,13 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, ThreeBitrateObservers) {
   EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_);
   EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_);
 
+  allocator_->OnNetworkChanged(0, 0, 0);
+  // Verify that zero estimated bandwidth, means that that all gets zero,
+  // regardless of set min bitrate.
+  EXPECT_EQ(0u, bitrate_observer_1.last_bitrate_);
+  EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_);
+  EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_);
+
   allocator_->RemoveObserver(&bitrate_observer_1);
   allocator_->RemoveObserver(&bitrate_observer_2);
   allocator_->RemoveObserver(&bitrate_observer_3);
@@ -181,14 +198,16 @@ TEST_F(BitrateAllocatorTest, ThreeBitrateObserversLowRembEnforceMin) {
   TestBitrateObserver bitrate_observer_2;
   TestBitrateObserver bitrate_observer_3;
   int start_bitrate =
-      allocator_->AddObserver(&bitrate_observer_1, 100000, 400000);
+      allocator_->AddObserver(&bitrate_observer_1, 100000, 400000, true);
   EXPECT_EQ(300000, start_bitrate);
 
-  start_bitrate = allocator_->AddObserver(&bitrate_observer_2, 200000, 400000);
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer_2, 200000, 400000, true);
   EXPECT_EQ(200000, start_bitrate);
   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
 
-  start_bitrate = allocator_->AddObserver(&bitrate_observer_3, 300000, 400000);
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer_3, 300000, 400000, true);
   EXPECT_EQ(300000, start_bitrate);
   EXPECT_EQ(100000, static_cast<int>(bitrate_observer_1.last_bitrate_));
   EXPECT_EQ(200000, static_cast<int>(bitrate_observer_2.last_bitrate_));
@@ -203,4 +222,32 @@ TEST_F(BitrateAllocatorTest, ThreeBitrateObserversLowRembEnforceMin) {
   allocator_->RemoveObserver(&bitrate_observer_2);
   allocator_->RemoveObserver(&bitrate_observer_3);
 }
+
+TEST_F(BitrateAllocatorTest, AddObserverWhileNetworkDown) {
+  TestBitrateObserver bitrate_observer_1;
+  int start_bitrate =
+      allocator_->AddObserver(&bitrate_observer_1, 50000, 400000, true);
+  EXPECT_EQ(300000, start_bitrate);
+
+  // Set network down, ie, no available bitrate.
+  allocator_->OnNetworkChanged(0, 0, 0);
+
+  EXPECT_EQ(0u, bitrate_observer_1.last_bitrate_);
+
+  TestBitrateObserver bitrate_observer_2;
+  start_bitrate =
+      allocator_->AddObserver(&bitrate_observer_2, 50000, 400000, true);
+
+  // Expect the start_bitrate to be set as if the network was still up but that
+  // the new observer have been notified that the network is down.
+  EXPECT_EQ(300000 / 2, start_bitrate);
+  EXPECT_EQ(0u, bitrate_observer_1.last_bitrate_);
+  EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_);
+
+  // Set network back up.
+  allocator_->OnNetworkChanged(1500000, 0, 50);
+  EXPECT_EQ(750000u, bitrate_observer_1.last_bitrate_);
+  EXPECT_EQ(750000u, bitrate_observer_2.last_bitrate_);
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/call/bitrate_estimator_tests.cc b/chromium/third_party/webrtc/call/bitrate_estimator_tests.cc
index c63d45dc6e6..122e5bb19b6 100644
--- a/chromium/third_party/webrtc/call/bitrate_estimator_tests.cc
+++ b/chromium/third_party/webrtc/call/bitrate_estimator_tests.cc
@@ -302,7 +302,8 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
-TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
+// This test is flaky. See webrtc:5790.
+TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
   video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
diff --git a/chromium/third_party/webrtc/call/call.cc b/chromium/third_party/webrtc/call/call.cc
index 91c27c80ad9..f7c66db4b21 100644
--- a/chromium/third_party/webrtc/call/call.cc
+++ b/chromium/third_party/webrtc/call/call.cc
@@ -19,6 +19,7 @@
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
@@ -39,6 +40,7 @@
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/video/call_stats.h"
+#include "webrtc/video/send_delay_stats.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video/vie_remb.h"
@@ -50,8 +52,9 @@ const int Call::Config::kDefaultStartBitrateBps = 300000;
 
 namespace internal {
 
-class Call : public webrtc::Call, public PacketReceiver,
-             public BitrateObserver {
+class Call : public webrtc::Call,
+             public PacketReceiver,
+             public CongestionController::Observer {
  public:
   explicit Call(const Call::Config& config);
   virtual ~Call();
@@ -89,6 +92,9 @@ class Call : public webrtc::Call, public PacketReceiver,
 
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
+  void OnNetworkRouteChanged(const std::string& transport_name,
+                             const rtc::NetworkRoute& network_route) override;
+
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   // Implements BitrateObserver.
@@ -102,7 +108,6 @@ class Call : public webrtc::Call, public PacketReceiver,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
-
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
@@ -170,8 +175,11 @@ class Call : public webrtc::Call, public PacketReceiver,
   int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
 
+  std::map<std::string, rtc::NetworkRoute> network_routes_;
+
   VieRemb remb_;
   const std::unique_ptr<CongestionController> congestion_controller_;
+  const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
@@ -186,9 +194,8 @@ namespace internal {
 Call::Call(const Call::Config& config)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
-      module_process_thread_(
-          rtc::ScopedToUnique(ProcessThread::Create("ModuleProcessThread"))),
-      pacer_thread_(rtc::ScopedToUnique(ProcessThread::Create("PacerThread"))),
+      module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
+      pacer_thread_(ProcessThread::Create("PacerThread")),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator()),
       config_(config),
@@ -206,7 +213,8 @@ Call::Call(const Call::Config& config)
       pacer_bitrate_sum_kbits_(0),
       num_bitrate_updates_(0),
       remb_(clock_),
-      congestion_controller_(new CongestionController(clock_, this, &remb_)) {
+      congestion_controller_(new CongestionController(clock_, this, &remb_)),
+      video_send_delay_stats_(new SendDelayStats(clock_)) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
@@ -399,12 +407,14 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
+  video_send_delay_stats_->AddSsrcs(config);
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
-      congestion_controller_.get(), bitrate_allocator_.get(), &remb_, config,
-      encoder_config, suspended_video_send_ssrcs_);
+      congestion_controller_.get(), bitrate_allocator_.get(),
+      video_send_delay_stats_.get(), &remb_, config, encoder_config,
+      suspended_video_send_ssrcs_);
   {
     WriteLockScoped write_lock(*send_crit_);
     for (uint32_t ssrc : config.rtp.ssrcs) {
@@ -591,6 +601,37 @@ void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
   }
 }
 
+// TODO(honghaiz): Add tests for this method.
+void Call::OnNetworkRouteChanged(const std::string& transport_name,
+                                 const rtc::NetworkRoute& network_route) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  // Check if the network route is connected.
+  if (!network_route.connected) {
+    LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
+    // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
+    // consider merging these two methods.
+    return;
+  }
+
+  // Check whether the network route has changed on each transport.
+  auto result =
+      network_routes_.insert(std::make_pair(transport_name, network_route));
+  auto kv = result.first;
+  bool inserted = result.second;
+  if (inserted) {
+    // No need to reset BWE if this is the first time the network connects.
+    return;
+  }
+  if (kv->second != network_route) {
+    kv->second = network_route;
+    LOG(LS_INFO) << "Network route changed on transport " << transport_name
+                 << ": new local network id " << network_route.local_network_id
+                 << " new remote network id "
+                 << network_route.remote_network_id;
+    // TODO(holmer): Update the BWE bitrates.
+  }
+}
+
 void Call::UpdateAggregateNetworkState() {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
@@ -626,6 +667,8 @@ void Call::UpdateAggregateNetworkState() {
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   if (first_packet_sent_ms_ == -1)
     first_packet_sent_ms_ = clock_->TimeInMilliseconds();
+  video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
+                                        clock_->TimeInMilliseconds());
   congestion_controller_->OnSentPacket(sent_packet);
 }
 
@@ -657,10 +700,8 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
     pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
     ++num_bitrate_updates_;
   }
-  congestion_controller_->UpdatePacerBitrate(
-      target_bitrate_bps / 1000,
-      PacedSender::kDefaultPaceMultiplier * pacer_bitrate_bps / 1000,
-      pad_up_to_bitrate_bps / 1000);
+  congestion_controller_->SetAllocatedSendBitrate(allocated_bitrate_bps,
+                                                  pad_up_to_bitrate_bps);
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
@@ -715,8 +756,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
                                                  const uint8_t* packet,
                                                  size_t length) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
-  // TODO(pbos): Figure out what channel needs it actually.
-  //             Do NOT broadcast! Also make sure it's a valid packet.
+  // TODO(pbos): Make sure it's a valid packet.
   //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
   //             there's no receiver of the packet.
   received_rtcp_bytes_ += length;
@@ -724,25 +764,35 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*receive_crit_);
     for (VideoReceiveStream* stream : video_receive_streams_) {
-      if (stream->DeliverRtcp(packet, length)) {
+      if (stream->DeliverRtcp(packet, length))
+        rtcp_delivered = true;
+    }
+  }
+  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
+    ReadLockScoped read_lock(*receive_crit_);
+    for (auto& kv : audio_receive_ssrcs_) {
+      if (kv.second->DeliverRtcp(packet, length))
         rtcp_delivered = true;
-        if (event_log_)
-          event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet,
-                                    length);
-      }
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*send_crit_);
     for (VideoSendStream* stream : video_send_streams_) {
-      if (stream->DeliverRtcp(packet, length)) {
+      if (stream->DeliverRtcp(packet, length))
+        rtcp_delivered = true;
+    }
+  }
+  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
+    ReadLockScoped read_lock(*send_crit_);
+    for (auto& kv : audio_send_ssrcs_) {
+      if (kv.second->DeliverRtcp(packet, length))
         rtcp_delivered = true;
-        if (event_log_)
-          event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet,
-                                    length);
-      }
     }
   }
+
+  if (event_log_ && rtcp_delivered)
+    event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
+
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
diff --git a/chromium/third_party/webrtc/call/call_perf_tests.cc b/chromium/third_party/webrtc/call/call_perf_tests.cc
index 9aa50d0ad97..329c1f25b61 100644
--- a/chromium/third_party/webrtc/call/call_perf_tests.cc
+++ b/chromium/third_party/webrtc/call/call_perf_tests.cc
@@ -17,6 +17,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
@@ -41,7 +42,6 @@
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
@@ -149,39 +149,11 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
-  class AudioPacketReceiver : public PacketReceiver {
-   public:
-    AudioPacketReceiver(int channel, VoENetwork* voe_network)
-        : channel_(channel),
-          voe_network_(voe_network),
-          parser_(RtpHeaderParser::Create()) {}
-    DeliveryStatus DeliverPacket(MediaType media_type,
-                                 const uint8_t* packet,
-                                 size_t length,
-                                 const PacketTime& packet_time) override {
-      EXPECT_TRUE(media_type == MediaType::ANY ||
-                  media_type == MediaType::AUDIO);
-      int ret;
-      if (parser_->IsRtcp(packet, length)) {
-        ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length);
-      } else {
-        ret = voe_network_->ReceivedRTPPacket(channel_, packet, length,
-                                              PacketTime());
-      }
-      return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
-    }
-
-   private:
-    int channel_;
-    VoENetwork* voe_network_;
-    std::unique_ptr<RtpHeaderParser> parser_;
-  };
 
   test::ClearHistograms();
   VoiceEngine* voice_engine = VoiceEngine::Create();
   VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
   VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
-  VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine);
   const std::string audio_filename =
       test::ResourcePath("voice_engine/audio_long16", "pcm");
   ASSERT_STRNE("", audio_filename.c_str());
@@ -201,44 +173,56 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   receiver_config.audio_state = sender_config.audio_state;
   CreateCalls(sender_config, receiver_config);
 
-  AudioPacketReceiver voe_send_packet_receiver(send_channel_id, voe_network);
-  AudioPacketReceiver voe_recv_packet_receiver(recv_channel_id, voe_network);
 
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
 
-  FakeNetworkPipe::Config net_config;
-  net_config.queue_delay_ms = 500;
-  net_config.loss_percent = 5;
-  test::PacketTransport audio_send_transport(
-      nullptr, &observer, test::PacketTransport::kSender, net_config);
-  audio_send_transport.SetReceiver(&voe_recv_packet_receiver);
-  test::PacketTransport audio_receive_transport(
-      nullptr, &observer, test::PacketTransport::kReceiver, net_config);
-  audio_receive_transport.SetReceiver(&voe_send_packet_receiver);
-
-  internal::TransportAdapter send_transport_adapter(&audio_send_transport);
-  send_transport_adapter.Enable();
-  EXPECT_EQ(0, voe_network->RegisterExternalTransport(send_channel_id,
-                                                      send_transport_adapter));
-
-  internal::TransportAdapter recv_transport_adapter(&audio_receive_transport);
-  recv_transport_adapter.Enable();
-  EXPECT_EQ(0, voe_network->RegisterExternalTransport(recv_channel_id,
-                                                      recv_transport_adapter));
-
-  test::PacketTransport sync_send_transport(sender_call_.get(), &observer,
-                                            test::PacketTransport::kSender,
-                                            FakeNetworkPipe::Config());
-  sync_send_transport.SetReceiver(receiver_call_->Receiver());
-  test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer,
-                                               test::PacketTransport::kReceiver,
-                                               FakeNetworkPipe::Config());
-  sync_receive_transport.SetReceiver(sender_call_->Receiver());
+  // Helper class to ensure we deliver correct media_type to the receiving call.
+  class MediaTypePacketReceiver : public PacketReceiver {
+   public:
+    MediaTypePacketReceiver(PacketReceiver* packet_receiver,
+                            MediaType media_type)
+        : packet_receiver_(packet_receiver), media_type_(media_type) {}
+
+    DeliveryStatus DeliverPacket(MediaType media_type,
+                                 const uint8_t* packet,
+                                 size_t length,
+                                 const PacketTime& packet_time) override {
+      return packet_receiver_->DeliverPacket(media_type_, packet, length,
+                                             packet_time);
+    }
+   private:
+    PacketReceiver* packet_receiver_;
+    const MediaType media_type_;
+
+    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaTypePacketReceiver);
+  };
+
+  FakeNetworkPipe::Config audio_net_config;
+  audio_net_config.queue_delay_ms = 500;
+  audio_net_config.loss_percent = 5;
+  test::PacketTransport audio_send_transport(sender_call_.get(), &observer,
+                                             test::PacketTransport::kSender,
+                                             audio_net_config);
+  MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(),
+                                         MediaType::AUDIO);
+  audio_send_transport.SetReceiver(&audio_receiver);
+
+  test::PacketTransport video_send_transport(sender_call_.get(), &observer,
+                                             test::PacketTransport::kSender,
+                                             FakeNetworkPipe::Config());
+  MediaTypePacketReceiver video_receiver(receiver_call_->Receiver(),
+                                         MediaType::VIDEO);
+  video_send_transport.SetReceiver(&video_receiver);
+
+  test::PacketTransport receive_transport(
+      receiver_call_.get(), &observer, test::PacketTransport::kReceiver,
+      FakeNetworkPipe::Config());
+  receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
-  CreateSendConfig(1, 0, &sync_send_transport);
-  CreateMatchingReceiveConfigs(&sync_receive_transport);
+  CreateSendConfig(1, 0, &video_send_transport);
+  CreateMatchingReceiveConfigs(&receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
@@ -298,10 +282,9 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   fake_audio_device.Stop();
 
   Stop();
-  sync_send_transport.StopSending();
-  sync_receive_transport.StopSending();
+  video_send_transport.StopSending();
   audio_send_transport.StopSending();
-  audio_receive_transport.StopSending();
+  receive_transport.StopSending();
 
   DestroyStreams();
 
@@ -312,7 +295,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   voe_base->DeleteChannel(recv_channel_id);
   voe_base->Release();
   voe_codec->Release();
-  voe_network->Release();
 
   DestroyCalls();
 
@@ -413,7 +395,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
       EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
     }
 
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
diff --git a/chromium/third_party/webrtc/call/mock/mock_rtc_event_log.h b/chromium/third_party/webrtc/call/mock/mock_rtc_event_log.h
index f523105d0e4..8ca73a3dc7d 100644
--- a/chromium/third_party/webrtc/call/mock/mock_rtc_event_log.h
+++ b/chromium/third_party/webrtc/call/mock/mock_rtc_event_log.h
@@ -21,12 +21,11 @@ namespace webrtc {
 
 class MockRtcEventLog : public RtcEventLog {
  public:
-  MOCK_METHOD1(SetBufferDuration, void(int64_t buffer_duration_us));
-
   MOCK_METHOD2(StartLogging,
-               void(const std::string& file_name, int duration_ms));
+               bool(const std::string& file_name, int64_t max_size_bytes));
 
-  MOCK_METHOD1(StartLogging, bool(rtc::PlatformFile log_file));
+  MOCK_METHOD2(StartLogging,
+               bool(rtc::PlatformFile log_file, int64_t max_size_bytes));
 
   MOCK_METHOD0(StopLogging, void());
 
diff --git a/chromium/third_party/webrtc/call/ringbuffer.h b/chromium/third_party/webrtc/call/ringbuffer.h
new file mode 100644
index 00000000000..fa5e4227ff7
--- /dev/null
+++ b/chromium/third_party/webrtc/call/ringbuffer.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef WEBRTC_CALL_RINGBUFFER_H_
+#define WEBRTC_CALL_RINGBUFFER_H_
+
+#include <memory>
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
+
+namespace webrtc {
+
+// A RingBuffer works like a fixed size queue which starts discarding
+// the oldest elements when it becomes full.
+template <typename T>
+class RingBuffer {
+ public:
+  // Creates a RingBuffer with space for |capacity| elements.
+  explicit RingBuffer(size_t capacity)
+      :  // We allocate space for one extra sentinel element.
+        data_(new T[capacity + 1]) {
+    RTC_DCHECK(capacity > 0);
+    end_ = data_.get() + (capacity + 1);
+    front_ = data_.get();
+    back_ = data_.get();
+  }
+
+  ~RingBuffer() {
+    // The unique_ptr will free the memory.
+  }
+
+  // Removes an element from the front of the queue.
+  void pop_front() {
+    RTC_DCHECK(!empty());
+    ++front_;
+    if (front_ == end_) {
+      front_ = data_.get();
+    }
+  }
+
+  // Appends an element to the back of the queue (and removes an
+  // element from the front if there is no space at the back of the queue).
+  void push_back(const T& elem) {
+    *back_ = elem;
+    ++back_;
+    if (back_ == end_) {
+      back_ = data_.get();
+    }
+    if (back_ == front_) {
+      ++front_;
+    }
+    if (front_ == end_) {
+      front_ = data_.get();
+    }
+  }
+
+  // Appends an element to the back of the queue (and removes an
+  // element from the front if there is no space at the back of the queue).
+  void push_back(T&& elem) {
+    *back_ = std::move(elem);
+    ++back_;
+    if (back_ == end_) {
+      back_ = data_.get();
+    }
+    if (back_ == front_) {
+      ++front_;
+    }
+    if (front_ == end_) {
+      front_ = data_.get();
+    }
+  }
+
+  T& front() { return *front_; }
+
+  const T& front() const { return *front_; }
+
+  bool empty() const { return (front_ == back_); }
+
+ private:
+  std::unique_ptr<T[]> data_;
+  T* end_;
+  T* front_;
+  T* back_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RingBuffer);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RINGBUFFER_H_
diff --git a/chromium/third_party/webrtc/call/ringbuffer_unittest.cc b/chromium/third_party/webrtc/call/ringbuffer_unittest.cc
new file mode 100644
index 00000000000..e2b5a41461a
--- /dev/null
+++ b/chromium/third_party/webrtc/call/ringbuffer_unittest.cc
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include <list>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+#include "webrtc/call/ringbuffer.h"
+
+namespace {
+template <typename T>
+class MovableType {
+ public:
+  MovableType() : value_(), moved_from_(false), moved_to_(false) {}
+  explicit MovableType(T value)
+      : value_(value), moved_from_(false), moved_to_(false) {}
+  MovableType(const MovableType<T>& other)
+      : value_(other.value_), moved_from_(false), moved_to_(false) {}
+  MovableType(MovableType<T>&& other)
+      : value_(other.value_), moved_from_(false), moved_to_(true) {
+    other.moved_from_ = true;
+  }
+
+  MovableType& operator=(const MovableType<T>& other) {
+    value_ = other.value_;
+    moved_from_ = false;
+    moved_to_ = false;
+    return *this;
+  }
+
+  MovableType& operator=(MovableType<T>&& other) {
+    value_ = other.value_;
+    moved_from_ = false;
+    moved_to_ = true;
+    other.moved_from_ = true;
+    return *this;
+  }
+
+  T Value() { return value_; }
+  bool IsMovedFrom() { return moved_from_; }
+  bool IsMovedTo() { return moved_to_; }
+
+ private:
+  T value_;
+  bool moved_from_;
+  bool moved_to_;
+};
+
+}  // namespace
+
+namespace webrtc {
+
+// Verify that the ring buffer works as a simple queue.
+TEST(RingBufferTest, SimpleQueue) {
+  size_t capacity = 100;
+  RingBuffer<size_t> q(capacity);
+  EXPECT_TRUE(q.empty());
+  for (size_t i = 0; i < capacity; i++) {
+    q.push_back(i);
+    EXPECT_FALSE(q.empty());
+  }
+
+  for (size_t i = 0; i < capacity; i++) {
+    EXPECT_FALSE(q.empty());
+    EXPECT_EQ(i, q.front());
+    q.pop_front();
+  }
+  EXPECT_TRUE(q.empty());
+}
+
+// Do a "random" sequence of queue operations and verify that the
+// result is consistent with the same operation performed on a std::list.
+TEST(RingBufferTest, ConsistentWithStdList) {
+  Random prng(987654321ull);
+  size_t capacity = 10;
+  RingBuffer<int> q(capacity);
+  std::list<int> l;
+  EXPECT_TRUE(q.empty());
+  for (size_t i = 0; i < 100 * capacity; i++) {
+    bool insert = prng.Rand<bool>();
+    if ((insert && l.size() < capacity) || l.size() == 0) {
+      int x = prng.Rand<int>();
+      l.push_back(x);
+      q.push_back(x);
+      EXPECT_FALSE(q.empty());
+    } else {
+      EXPECT_FALSE(q.empty());
+      EXPECT_EQ(l.front(), q.front());
+      l.pop_front();
+      q.pop_front();
+    }
+  }
+  while (!l.empty()) {
+    EXPECT_FALSE(q.empty());
+    EXPECT_EQ(l.front(), q.front());
+    l.pop_front();
+    q.pop_front();
+  }
+  EXPECT_TRUE(q.empty());
+}
+
+// Test that the ringbuffer starts reusing elements from the front
+// when the queue becomes full.
+TEST(RingBufferTest, OverwriteOldElements) {
+  size_t capacity = 100;
+  size_t insertions = 3 * capacity + 25;
+  RingBuffer<size_t> q(capacity);
+  EXPECT_TRUE(q.empty());
+  for (size_t i = 0; i < insertions; i++) {
+    q.push_back(i);
+    EXPECT_FALSE(q.empty());
+  }
+
+  for (size_t i = insertions - capacity; i < insertions; i++) {
+    EXPECT_FALSE(q.empty());
+    EXPECT_EQ(i, q.front());
+    q.pop_front();
+  }
+  EXPECT_TRUE(q.empty());
+}
+
+// Test that the ringbuffer uses std::move when pushing an rvalue reference.
+TEST(RingBufferTest, MoveSemanticsForPushBack) {
+  size_t capacity = 100;
+  size_t insertions = 3 * capacity + 25;
+  RingBuffer<MovableType<size_t>> q(capacity);
+  EXPECT_TRUE(q.empty());
+  for (size_t i = 0; i < insertions; i++) {
+    MovableType<size_t> tmp(i);
+    EXPECT_FALSE(tmp.IsMovedFrom());
+    EXPECT_FALSE(tmp.IsMovedTo());
+    q.push_back(std::move(tmp));
+    EXPECT_TRUE(tmp.IsMovedFrom());
+    EXPECT_FALSE(tmp.IsMovedTo());
+    EXPECT_FALSE(q.empty());
+  }
+
+  for (size_t i = insertions - capacity; i < insertions; i++) {
+    EXPECT_FALSE(q.empty());
+    EXPECT_EQ(i, q.front().Value());
+    EXPECT_FALSE(q.front().IsMovedFrom());
+    EXPECT_TRUE(q.front().IsMovedTo());
+    q.pop_front();
+  }
+  EXPECT_TRUE(q.empty());
+}
+
+TEST(RingBufferTest, SmallCapacity) {
+  size_t capacity = 1;
+  RingBuffer<int> q(capacity);
+  EXPECT_TRUE(q.empty());
+  q.push_back(4711);
+  EXPECT_FALSE(q.empty());
+  EXPECT_EQ(4711, q.front());
+  q.push_back(1024);
+  EXPECT_FALSE(q.empty());
+  EXPECT_EQ(1024, q.front());
+  q.pop_front();
+  EXPECT_TRUE(q.empty());
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/call/rtc_event_log.cc b/chromium/third_party/webrtc/call/rtc_event_log.cc
index ce4d6ef39f4..6dd02c808eb 100644
--- a/chromium/third_party/webrtc/call/rtc_event_log.cc
+++ b/chromium/third_party/webrtc/call/rtc_event_log.cc
@@ -10,18 +10,22 @@
 
 #include "webrtc/call/rtc_event_log.h"
 
-#include <deque>
+#include <limits>
 #include <vector>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/swap_queue.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
+#include "webrtc/call/rtc_event_log_helper_thread.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
 // Files generated at build-time by the protobuf compiler.
@@ -37,12 +41,17 @@ namespace webrtc {
 #ifndef ENABLE_RTC_EVENT_LOG
 
 // No-op implementation if flag is not set.
-class RtcEventLogImpl final : public RtcEventLog {
+class RtcEventLogNullImpl final : public RtcEventLog {
  public:
-  void SetBufferDuration(int64_t buffer_duration_us) override {}
-  void StartLogging(const std::string& file_name, int duration_ms) override {}
-  bool StartLogging(rtc::PlatformFile log_file) override { return false; }
-  void StopLogging(void) override {}
+  bool StartLogging(const std::string& file_name,
+                    int64_t max_size_bytes) override {
+    return false;
+  }
+  bool StartLogging(rtc::PlatformFile platform_file,
+                    int64_t max_size_bytes) override {
+    return false;
+  }
+  void StopLogging() override {}
   void LogVideoReceiveStreamConfig(
       const VideoReceiveStream::Config& config) override {}
   void LogVideoSendStreamConfig(
@@ -65,11 +74,13 @@ class RtcEventLogImpl final : public RtcEventLog {
 
 class RtcEventLogImpl final : public RtcEventLog {
  public:
-  RtcEventLogImpl();
+  explicit RtcEventLogImpl(const Clock* clock);
+  ~RtcEventLogImpl() override;
 
-  void SetBufferDuration(int64_t buffer_duration_us) override;
-  void StartLogging(const std::string& file_name, int duration_ms) override;
-  bool StartLogging(rtc::PlatformFile log_file) override;
+  bool StartLogging(const std::string& file_name,
+                    int64_t max_size_bytes) override;
+  bool StartLogging(rtc::PlatformFile platform_file,
+                    int64_t max_size_bytes) override;
   void StopLogging() override;
   void LogVideoReceiveStreamConfig(
       const VideoReceiveStream::Config& config) override;
@@ -88,37 +99,21 @@ class RtcEventLogImpl final : public RtcEventLog {
                              int32_t total_packets) override;
 
  private:
-  // Starts logging. This function assumes the file_ has been opened succesfully
-  // and that the start_time_us_ and _duration_us_ have been set.
-  void StartLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  // Stops logging and clears the stored data and buffers.
-  void StopLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  // Adds a new event to the logfile if logging is active, or adds it to the
-  // list of recent log events otherwise.
-  void HandleEvent(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  // Writes the event to the file. Note that this will destroy the state of the
-  // input argument.
-  void StoreToFile(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  // Adds the event to the list of recent events, and removes any events that
-  // are too old and no longer fall in the time window.
-  void AddRecentEvent(const rtclog::Event& event)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_);
-
-  rtc::CriticalSection crit_;
-  std::unique_ptr<FileWrapper> file_ GUARDED_BY(crit_) =
-      std::unique_ptr<FileWrapper>(FileWrapper::Create());
-  rtc::PlatformFile platform_file_ GUARDED_BY(crit_) =
-      rtc::kInvalidPlatformFileValue;
-  rtclog::EventStream stream_ GUARDED_BY(crit_);
-  std::deque<rtclog::Event> recent_log_events_ GUARDED_BY(crit_);
-  std::vector<rtclog::Event> config_events_ GUARDED_BY(crit_);
-
-  // Microseconds to record log events, before starting the actual log.
-  int64_t buffer_duration_us_ GUARDED_BY(crit_);
-  bool currently_logging_ GUARDED_BY(crit_);
-  int64_t start_time_us_ GUARDED_BY(crit_);
-  int64_t duration_us_ GUARDED_BY(crit_);
+  // Message queue for passing control messages to the logging thread.
+  SwapQueue<RtcEventLogHelperThread::ControlMessage> message_queue_;
+
+  // Message queue for passing events to the logging thread.
+  SwapQueue<std::unique_ptr<rtclog::Event> > event_queue_;
+
+  rtc::Event wake_up_;
+  rtc::Event stopped_;
+
   const Clock* const clock_;
+
+  RtcEventLogHelperThread helper_thread_;
+  rtc::ThreadChecker thread_checker_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtcEventLogImpl);
 };
 
 namespace {
@@ -126,10 +121,6 @@ namespace {
 // that the rest of the WebRtc project can use, to the corresponding
 // serialized enum which is defined by the protobuf.
 
-// Do not add default return values to the conversion functions in this
-// unnamed namespace. The intention is to make the compiler warn if anyone
-// adds unhandled new events/modes/etc.
-
 rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
   switch (rtcp_mode) {
     case RtcpMode::kCompound:
@@ -159,114 +150,108 @@ rtclog::MediaType ConvertMediaType(MediaType media_type) {
   return rtclog::ANY;
 }
 
-}  // namespace
-
-namespace {
-bool IsConfigEvent(const rtclog::Event& event) {
-  rtclog::Event_EventType event_type = event.type();
-  return event_type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT ||
-         event_type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT ||
-         event_type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT ||
-         event_type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT;
-}
+// The RTP and RTCP buffers reserve space for twice the expected number of
+// sent packets because they also contain received packets.
+static const int kEventsPerSecond = 1000;
+static const int kControlMessagesPerSecond = 10;
 }  // namespace
 
 // RtcEventLogImpl member functions.
-RtcEventLogImpl::RtcEventLogImpl()
-    : file_(FileWrapper::Create()),
-      stream_(),
-      buffer_duration_us_(10000000),
-      currently_logging_(false),
-      start_time_us_(0),
-      duration_us_(0),
-      clock_(Clock::GetRealTimeClock()) {
+RtcEventLogImpl::RtcEventLogImpl(const Clock* clock)
+    // Allocate buffers for roughly one second of history.
+    : message_queue_(kControlMessagesPerSecond),
+      event_queue_(kEventsPerSecond),
+      wake_up_(false, false),
+      stopped_(false, false),
+      clock_(clock),
+      helper_thread_(&message_queue_,
+                     &event_queue_,
+                     &wake_up_,
+                     &stopped_,
+                     clock),
+      thread_checker_() {
+  thread_checker_.DetachFromThread();
 }
 
-void RtcEventLogImpl::SetBufferDuration(int64_t buffer_duration_us) {
-  rtc::CritScope lock(&crit_);
-  buffer_duration_us_ = buffer_duration_us;
+RtcEventLogImpl::~RtcEventLogImpl() {
+  // The RtcEventLogHelperThread destructor closes the file
+  // and waits for the thread to terminate.
 }
 
-void RtcEventLogImpl::StartLogging(const std::string& file_name,
-                                   int duration_ms) {
-  rtc::CritScope lock(&crit_);
-  if (currently_logging_) {
-    StopLoggingLocked();
+bool RtcEventLogImpl::StartLogging(const std::string& file_name,
+                                   int64_t max_size_bytes) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RtcEventLogHelperThread::ControlMessage message;
+  message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE;
+  message.max_size_bytes = max_size_bytes <= 0
+                               ? std::numeric_limits<int64_t>::max()
+                               : max_size_bytes;
+  message.start_time = clock_->TimeInMicroseconds();
+  message.stop_time = std::numeric_limits<int64_t>::max();
+  message.file.reset(FileWrapper::Create());
+  if (message.file->OpenFile(file_name.c_str(), false) != 0) {
+    return false;
   }
-  if (file_->OpenFile(file_name.c_str(), false) != 0) {
-    return;
+  if (!message_queue_.Insert(&message)) {
+    LOG(LS_WARNING) << "Message queue full. Can't start logging.";
+    return false;
   }
-  start_time_us_ = clock_->TimeInMicroseconds();
-  duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
-  StartLoggingLocked();
+  return true;
 }
 
-bool RtcEventLogImpl::StartLogging(rtc::PlatformFile log_file) {
-  rtc::CritScope lock(&crit_);
-
-  if (currently_logging_) {
-    StopLoggingLocked();
+bool RtcEventLogImpl::StartLogging(rtc::PlatformFile platform_file,
+                                   int64_t max_size_bytes) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RtcEventLogHelperThread::ControlMessage message;
+  message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE;
+  message.max_size_bytes = max_size_bytes <= 0
+                               ? std::numeric_limits<int64_t>::max()
+                               : max_size_bytes;
+  message.start_time = clock_->TimeInMicroseconds();
+  message.stop_time = std::numeric_limits<int64_t>::max();
+  message.file.reset(FileWrapper::Create());
+  FILE* file_handle = rtc::FdopenPlatformFileForWriting(platform_file);
+  if (!file_handle) {
+    return false;
   }
-  RTC_DCHECK(platform_file_ == rtc::kInvalidPlatformFileValue);
-
-  FILE* file_stream = rtc::FdopenPlatformFileForWriting(log_file);
-  if (!file_stream) {
-    rtc::ClosePlatformFile(log_file);
+  if (message.file->OpenFromFileHandle(file_handle, true, false) != 0) {
     return false;
   }
-
-  if (file_->OpenFromFileHandle(file_stream, true, false) != 0) {
-    rtc::ClosePlatformFile(log_file);
+  if (!message_queue_.Insert(&message)) {
+    LOG(LS_WARNING) << "Message queue full. Can't start logging.";
     return false;
   }
-  platform_file_ = log_file;
-  // Set the start time and duration to keep logging for 10 minutes.
-  start_time_us_ = clock_->TimeInMicroseconds();
-  duration_us_ = 10 * 60 * 1000000;
-  StartLoggingLocked();
   return true;
 }
 
-void RtcEventLogImpl::StartLoggingLocked() {
-  currently_logging_ = true;
-
-  // Write all old configuration events to the log file.
-  for (auto& event : config_events_) {
-    StoreToFile(&event);
-  }
-  // Write all recent configuration events to the log file, and
-  // write all other recent events to the log file, ignoring any old events.
-  for (auto& event : recent_log_events_) {
-    if (IsConfigEvent(event)) {
-      StoreToFile(&event);
-      config_events_.push_back(event);
-    } else if (event.timestamp_us() >= start_time_us_ - buffer_duration_us_) {
-      StoreToFile(&event);
-    }
-  }
-  recent_log_events_.clear();
-  // Write a LOG_START event to the file.
-  rtclog::Event start_event;
-  start_event.set_timestamp_us(start_time_us_);
-  start_event.set_type(rtclog::Event::LOG_START);
-  StoreToFile(&start_event);
-}
-
 void RtcEventLogImpl::StopLogging() {
-  rtc::CritScope lock(&crit_);
-  StopLoggingLocked();
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RtcEventLogHelperThread::ControlMessage message;
+  message.message_type = RtcEventLogHelperThread::ControlMessage::STOP_FILE;
+  message.stop_time = clock_->TimeInMicroseconds();
+  while (!message_queue_.Insert(&message)) {
+    // TODO(terelius): We would like to have a blocking Insert function in the
+    // SwapQueue, but for the time being we will just clear any previous
+    // messages.
+    // Since StopLogging waits for the thread, it is essential that we don't
+    // clear any STOP_FILE messages. To ensure that there is only one call at a
+    // time, we require that all calls to StopLogging are made on the same
+    // thread.
+    LOG(LS_WARNING) << "Message queue full. Clearing queue to stop logging.";
+    message_queue_.Clear();
+  }
+  wake_up_.Set();                       // Request the output thread to wake up.
+  stopped_.Wait(rtc::Event::kForever);  // Wait for the log to stop.
 }
 
 void RtcEventLogImpl::LogVideoReceiveStreamConfig(
     const VideoReceiveStream::Config& config) {
-  rtc::CritScope lock(&crit_);
-
-  rtclog::Event event;
-  event.set_timestamp_us(clock_->TimeInMicroseconds());
-  event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
 
   rtclog::VideoReceiveConfig* receiver_config =
-      event.mutable_video_receiver_config();
+      event->mutable_video_receiver_config();
   receiver_config->set_remote_ssrc(config.rtp.remote_ssrc);
   receiver_config->set_local_ssrc(config.rtp.local_ssrc);
 
@@ -292,18 +277,18 @@ void RtcEventLogImpl::LogVideoReceiveStreamConfig(
     decoder->set_name(d.payload_name);
     decoder->set_payload_type(d.payload_type);
   }
-  HandleEvent(&event);
+  if (!event_queue_.Insert(&event)) {
+    LOG(LS_WARNING) << "Config queue full. Not logging config event.";
+  }
 }
 
 void RtcEventLogImpl::LogVideoSendStreamConfig(
     const VideoSendStream::Config& config) {
-  rtc::CritScope lock(&crit_);
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
 
-  rtclog::Event event;
-  event.set_timestamp_us(clock_->TimeInMicroseconds());
-  event.set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
-
-  rtclog::VideoSendConfig* sender_config = event.mutable_video_sender_config();
+  rtclog::VideoSendConfig* sender_config = event->mutable_video_sender_config();
 
   for (const auto& ssrc : config.rtp.ssrcs) {
     sender_config->add_ssrcs(ssrc);
@@ -324,7 +309,9 @@ void RtcEventLogImpl::LogVideoSendStreamConfig(
   rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
   encoder->set_name(config.encoder_settings.payload_name);
   encoder->set_payload_type(config.encoder_settings.payload_type);
-  HandleEvent(&event);
+  if (!event_queue_.Insert(&event)) {
+    LOG(LS_WARNING) << "Config queue full. Not logging config event.";
+  }
 }
 
 void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
@@ -347,27 +334,27 @@ void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
     header_length += (x_len + 1) * 4;
   }
 
-  rtc::CritScope lock(&crit_);
-  rtclog::Event rtp_event;
-  rtp_event.set_timestamp_us(clock_->TimeInMicroseconds());
-  rtp_event.set_type(rtclog::Event::RTP_EVENT);
-  rtp_event.mutable_rtp_packet()->set_incoming(direction == kIncomingPacket);
-  rtp_event.mutable_rtp_packet()->set_type(ConvertMediaType(media_type));
-  rtp_event.mutable_rtp_packet()->set_packet_length(packet_length);
-  rtp_event.mutable_rtp_packet()->set_header(header, header_length);
-  HandleEvent(&rtp_event);
+  std::unique_ptr<rtclog::Event> rtp_event(new rtclog::Event());
+  rtp_event->set_timestamp_us(clock_->TimeInMicroseconds());
+  rtp_event->set_type(rtclog::Event::RTP_EVENT);
+  rtp_event->mutable_rtp_packet()->set_incoming(direction == kIncomingPacket);
+  rtp_event->mutable_rtp_packet()->set_type(ConvertMediaType(media_type));
+  rtp_event->mutable_rtp_packet()->set_packet_length(packet_length);
+  rtp_event->mutable_rtp_packet()->set_header(header, header_length);
+  if (!event_queue_.Insert(&rtp_event)) {
+    LOG(LS_WARNING) << "RTP queue full. Not logging RTP packet.";
+  }
 }
 
 void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
                                     MediaType media_type,
                                     const uint8_t* packet,
                                     size_t length) {
-  rtc::CritScope lock(&crit_);
-  rtclog::Event rtcp_event;
-  rtcp_event.set_timestamp_us(clock_->TimeInMicroseconds());
-  rtcp_event.set_type(rtclog::Event::RTCP_EVENT);
-  rtcp_event.mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket);
-  rtcp_event.mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
+  std::unique_ptr<rtclog::Event> rtcp_event(new rtclog::Event());
+  rtcp_event->set_timestamp_us(clock_->TimeInMicroseconds());
+  rtcp_event->set_type(rtclog::Event::RTCP_EVENT);
+  rtcp_event->mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket);
+  rtcp_event->mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
 
   RTCPUtility::RtcpCommonHeader header;
   const uint8_t* block_begin = packet;
@@ -413,87 +400,35 @@ void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
 
     block_begin += block_size;
   }
-  rtcp_event.mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
-  HandleEvent(&rtcp_event);
+  rtcp_event->mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
+  if (!event_queue_.Insert(&rtcp_event)) {
+    LOG(LS_WARNING) << "RTCP queue full. Not logging RTCP packet.";
+  }
 }
 
 void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
-  rtc::CritScope lock(&crit_);
-  rtclog::Event event;
-  event.set_timestamp_us(clock_->TimeInMicroseconds());
-  event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
-  auto playout_event = event.mutable_audio_playout_event();
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
+  auto playout_event = event->mutable_audio_playout_event();
   playout_event->set_local_ssrc(ssrc);
-  HandleEvent(&event);
+  if (!event_queue_.Insert(&event)) {
+    LOG(LS_WARNING) << "Playout queue full. Not logging ACM playout.";
+  }
 }
 
 void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
                                             uint8_t fraction_loss,
                                             int32_t total_packets) {
-  rtc::CritScope lock(&crit_);
-  rtclog::Event event;
-  event.set_timestamp_us(clock_->TimeInMicroseconds());
-  event.set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
-  auto bwe_event = event.mutable_bwe_packet_loss_event();
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
+  auto bwe_event = event->mutable_bwe_packet_loss_event();
   bwe_event->set_bitrate(bitrate);
   bwe_event->set_fraction_loss(fraction_loss);
   bwe_event->set_total_packets(total_packets);
-  HandleEvent(&event);
-}
-
-void RtcEventLogImpl::StopLoggingLocked() {
-  if (currently_logging_) {
-    currently_logging_ = false;
-    // Create a LogEnd event
-    rtclog::Event event;
-    event.set_timestamp_us(clock_->TimeInMicroseconds());
-    event.set_type(rtclog::Event::LOG_END);
-    // Store the event and close the file
-    RTC_DCHECK(file_->Open());
-    StoreToFile(&event);
-    file_->CloseFile();
-    if (platform_file_ != rtc::kInvalidPlatformFileValue) {
-      rtc::ClosePlatformFile(platform_file_);
-      platform_file_ = rtc::kInvalidPlatformFileValue;
-    }
-  }
-  RTC_DCHECK(!file_->Open());
-  stream_.Clear();
-}
-
-void RtcEventLogImpl::HandleEvent(rtclog::Event* event) {
-  if (currently_logging_) {
-    if (clock_->TimeInMicroseconds() < start_time_us_ + duration_us_) {
-      StoreToFile(event);
-      return;
-    }
-    StopLoggingLocked();
-  }
-  AddRecentEvent(*event);
-}
-
-void RtcEventLogImpl::StoreToFile(rtclog::Event* event) {
-  // Reuse the same object at every log event.
-  if (stream_.stream_size() < 1) {
-    stream_.add_stream();
-  }
-  RTC_DCHECK_EQ(stream_.stream_size(), 1);
-  stream_.mutable_stream(0)->Swap(event);
-  // TODO(terelius): Doesn't this create a new EventStream per event?
-  // Is this guaranteed to work e.g. in future versions of protobuf?
-  std::string dump_buffer;
-  stream_.SerializeToString(&dump_buffer);
-  file_->Write(dump_buffer.data(), dump_buffer.size());
-}
-
-void RtcEventLogImpl::AddRecentEvent(const rtclog::Event& event) {
-  recent_log_events_.push_back(event);
-  while (recent_log_events_.front().timestamp_us() <
-         event.timestamp_us() - buffer_duration_us_) {
-    if (IsConfigEvent(recent_log_events_.front())) {
-      config_events_.push_back(recent_log_events_.front());
-    }
-    recent_log_events_.pop_front();
+  if (!event_queue_.Insert(&event)) {
+    LOG(LS_WARNING) << "BWE loss queue full. Not logging BWE update.";
   }
 }
 
@@ -516,8 +451,12 @@ bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
 #endif  // ENABLE_RTC_EVENT_LOG
 
 // RtcEventLog member functions.
-std::unique_ptr<RtcEventLog> RtcEventLog::Create() {
-  return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl());
+std::unique_ptr<RtcEventLog> RtcEventLog::Create(const Clock* clock) {
+#ifdef ENABLE_RTC_EVENT_LOG
+  return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl(clock));
+#else
+  return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
+#endif  // ENABLE_RTC_EVENT_LOG
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/call/rtc_event_log.h b/chromium/third_party/webrtc/call/rtc_event_log.h
index 518308bf2d2..bea57b01cd1 100644
--- a/chromium/third_party/webrtc/call/rtc_event_log.h
+++ b/chromium/third_party/webrtc/call/rtc_event_log.h
@@ -26,6 +26,7 @@ namespace rtclog {
 class EventStream;
 }  // namespace rtclog
 
+class Clock;
 class RtcEventLogImpl;
 
 enum class MediaType;
@@ -36,30 +37,40 @@ class RtcEventLog {
  public:
   virtual ~RtcEventLog() {}
 
-  static std::unique_ptr<RtcEventLog> Create();
+  // Factory method to create an RtcEventLog object.
+  static std::unique_ptr<RtcEventLog> Create(const Clock* clock);
 
-  // Sets the time that events are stored in the internal event buffer
-  // before the user calls StartLogging.  The default is 10 000 000 us = 10 s
-  virtual void SetBufferDuration(int64_t buffer_duration_us) = 0;
-
-  // Starts logging for the specified duration to the specified file.
-  // The logging will stop automatically after the specified duration.
+  // Starts logging a maximum of max_size_bytes bytes to the specified file.
   // If the file already exists it will be overwritten.
-  // If the file cannot be opened, the RtcEventLog will not start logging.
-  virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
-
-  // Starts logging until either the 10 minute timer runs out or the StopLogging
-  // function is called. The RtcEventLog takes ownership of the supplied
-  // rtc::PlatformFile.
-  virtual bool StartLogging(rtc::PlatformFile log_file) = 0;
-
+  // If max_size_bytes <= 0, logging will be active until StopLogging is called.
+  // The function has no effect and returns false if we can't start a new log
+  // e.g. because we are already logging or the file cannot be opened.
+  virtual bool StartLogging(const std::string& file_name,
+                            int64_t max_size_bytes) = 0;
+
+  // Same as above. The RtcEventLog takes ownership of the file if the call
+  // is successful, i.e. if it returns true.
+  virtual bool StartLogging(rtc::PlatformFile platform_file,
+                            int64_t max_size_bytes) = 0;
+
+  // Deprecated. Pass an explicit file size limit.
+  bool StartLogging(const std::string& file_name) {
+    return StartLogging(file_name, 10000000);
+  }
+
+  // Deprecated. Pass an explicit file size limit.
+  bool StartLogging(rtc::PlatformFile platform_file) {
+    return StartLogging(platform_file, 10000000);
+  }
+
+  // Stops logging to file and waits until the thread has finished.
   virtual void StopLogging() = 0;
 
-  // Logs configuration information for webrtc::VideoReceiveStream
+  // Logs configuration information for webrtc::VideoReceiveStream.
   virtual void LogVideoReceiveStreamConfig(
       const webrtc::VideoReceiveStream::Config& config) = 0;
 
-  // Logs configuration information for webrtc::VideoSendStream
+  // Logs configuration information for webrtc::VideoSendStream.
   virtual void LogVideoSendStreamConfig(
       const webrtc::VideoSendStream::Config& config) = 0;
 
@@ -76,7 +87,7 @@ class RtcEventLog {
                              const uint8_t* packet,
                              size_t length) = 0;
 
-  // Logs an audio playout event
+  // Logs an audio playout event.
   virtual void LogAudioPlayout(uint32_t ssrc) = 0;
 
   // Logs a bitrate update from the bandwidth estimator based on packet loss.
@@ -86,6 +97,11 @@ class RtcEventLog {
 
   // Reads an RtcEventLog file and returns true when reading was successful.
   // The result is stored in the given EventStream object.
+  // The order of the events in the EventStream is implementation defined.
+  // The current implementation writes a LOG_START event, then the old
+  // configurations, then the remaining events in timestamp order and finally
+  // a LOG_END event. However, this might change without further notice.
+  // TODO(terelius): Change result type to a vector?
   static bool ParseRtcEventLog(const std::string& file_name,
                                rtclog::EventStream* result);
 };
diff --git a/chromium/third_party/webrtc/call/rtc_event_log2rtp_dump.cc b/chromium/third_party/webrtc/call/rtc_event_log2rtp_dump.cc
index ef0be9a1b73..5733cfa31d2 100644
--- a/chromium/third_party/webrtc/call/rtc_event_log2rtp_dump.cc
+++ b/chromium/third_party/webrtc/call/rtc_event_log2rtp_dump.cc
@@ -15,17 +15,12 @@
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
+#include "webrtc/call/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/rtp_file_writer.h"
 
-// Files generated at build-time by the protobuf compiler.
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
-#else
-#include "webrtc/call/rtc_event_log.pb.h"
-#endif
-
 namespace {
 
 DEFINE_bool(noaudio,
@@ -94,8 +89,8 @@ int main(int argc, char* argv[]) {
     RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
         << "Flag verification has failed.";
 
-  webrtc::rtclog::EventStream event_stream;
-  if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) {
+  webrtc::ParsedRtcEventLog parsed_stream;
+  if (!parsed_stream.ParseFile(input_file)) {
     std::cerr << "Error while parsing input file: " << input_file << std::endl;
     return -1;
   }
@@ -110,94 +105,78 @@ int main(int argc, char* argv[]) {
     return -1;
   }
 
-  std::cout << "Found " << event_stream.stream_size()
+  std::cout << "Found " << parsed_stream.GetNumberOfEvents()
             << " events in the input file." << std::endl;
   int rtp_counter = 0, rtcp_counter = 0;
   bool header_only = false;
-  // TODO(ivoc): This can be refactored once the packet interpretation
-  //             functions are finished.
-  for (int i = 0; i < event_stream.stream_size(); i++) {
-    const webrtc::rtclog::Event& event = event_stream.stream(i);
-    if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) {
-      if (event.has_timestamp_us() && event.has_rtp_packet() &&
-          event.rtp_packet().has_header() &&
-          event.rtp_packet().header().size() >= 12 &&
-          event.rtp_packet().has_packet_length() &&
-          event.rtp_packet().has_type()) {
-        const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet();
-        if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
-          continue;
-        if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
-          continue;
-        if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
+  for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
+    // The parsed_stream will assert if the protobuf event is missing
+    // some required fields and we attempt to access them. We could consider
+    // a softer failure option, but it does not seem useful to generate
+    // RTP dumps based on broken event logs.
+    if (!FLAGS_nortp &&
+        parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
+      webrtc::test::RtpPacket packet;
+      webrtc::PacketDirection direction;
+      webrtc::MediaType media_type;
+      parsed_stream.GetRtpHeader(i, &direction, &media_type, packet.data,
+                                 &packet.length, &packet.original_length);
+      if (packet.original_length > packet.length)
+        header_only = true;
+      packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
+
+      // TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
+      if (direction == webrtc::kOutgoingPacket)
+        continue;
+      if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
+        continue;
+      if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+        continue;
+      if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+        continue;
+      if (!FLAGS_ssrc.empty()) {
+        const uint32_t packet_ssrc =
+            webrtc::ByteReader<uint32_t>::ReadBigEndian(
+                reinterpret_cast<const uint8_t*>(packet.data + 8));
+        if (packet_ssrc != ssrc_filter)
           continue;
-        if (!FLAGS_ssrc.empty()) {
-          const uint32_t packet_ssrc =
-              webrtc::ByteReader<uint32_t>::ReadBigEndian(
-                  reinterpret_cast<const uint8_t*>(rtp_packet.header().data() +
-                                                   8));
-          if (packet_ssrc != ssrc_filter)
-            continue;
-        }
-
-        webrtc::test::RtpPacket packet;
-        packet.length = rtp_packet.header().size();
-        if (packet.length > packet.kMaxPacketBufferSize) {
-          std::cout << "Skipping packet with size " << packet.length
-                    << ", the maximum supported size is "
-                    << packet.kMaxPacketBufferSize << std::endl;
-          continue;
-        }
-        packet.original_length = rtp_packet.packet_length();
-        if (packet.original_length > packet.length)
-          header_only = true;
-        packet.time_ms = event.timestamp_us() / 1000;
-        memcpy(packet.data, rtp_packet.header().data(), packet.length);
-        rtp_writer->WritePacket(&packet);
-        rtp_counter++;
-      } else {
-        std::cout << "Skipping malformed event." << std::endl;
       }
+
+      rtp_writer->WritePacket(&packet);
+      rtp_counter++;
     }
-    if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) {
-      if (event.has_timestamp_us() && event.has_rtcp_packet() &&
-          event.rtcp_packet().has_type() &&
-          event.rtcp_packet().has_packet_data() &&
-          event.rtcp_packet().packet_data().size() > 0) {
-        const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
-        if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
+    if (!FLAGS_nortcp &&
+        parsed_stream.GetEventType(i) ==
+            webrtc::ParsedRtcEventLog::RTCP_EVENT) {
+      webrtc::test::RtpPacket packet;
+      webrtc::PacketDirection direction;
+      webrtc::MediaType media_type;
+      parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet.data,
+                                  &packet.length);
+      // For RTCP packets the original_length should be set to 0 in the
+      // RTPdump format.
+      packet.original_length = 0;
+      packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
+
+      // TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
+      if (direction == webrtc::kOutgoingPacket)
+        continue;
+      if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
+        continue;
+      if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+        continue;
+      if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+        continue;
+      if (!FLAGS_ssrc.empty()) {
+        const uint32_t packet_ssrc =
+            webrtc::ByteReader<uint32_t>::ReadBigEndian(
+                reinterpret_cast<const uint8_t*>(packet.data + 4));
+        if (packet_ssrc != ssrc_filter)
           continue;
-        if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
-          continue;
-        if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA)
-          continue;
-        if (!FLAGS_ssrc.empty()) {
-          const uint32_t packet_ssrc =
-              webrtc::ByteReader<uint32_t>::ReadBigEndian(
-                  reinterpret_cast<const uint8_t*>(
-                      rtcp_packet.packet_data().data() + 4));
-          if (packet_ssrc != ssrc_filter)
-            continue;
-        }
-
-        webrtc::test::RtpPacket packet;
-        packet.length = rtcp_packet.packet_data().size();
-        if (packet.length > packet.kMaxPacketBufferSize) {
-          std::cout << "Skipping packet with size " << packet.length
-                    << ", the maximum supported size is "
-                    << packet.kMaxPacketBufferSize << std::endl;
-          continue;
-        }
-        // For RTCP packets the original_length should be set to 0 in the
-        // RTPdump format.
-        packet.original_length = 0;
-        packet.time_ms = event.timestamp_us() / 1000;
-        memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length);
-        rtp_writer->WritePacket(&packet);
-        rtcp_counter++;
-      } else {
-        std::cout << "Skipping malformed event." << std::endl;
       }
+
+      rtp_writer->WritePacket(&packet);
+      rtcp_counter++;
     }
   }
   std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
diff --git a/chromium/third_party/webrtc/call/rtc_event_log_helper_thread.cc b/chromium/third_party/webrtc/call/rtc_event_log_helper_thread.cc
new file mode 100644
index 00000000000..a9aa85144f0
--- /dev/null
+++ b/chromium/third_party/webrtc/call/rtc_event_log_helper_thread.cc
@@ -0,0 +1,285 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/call/rtc_event_log_helper_thread.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/system_wrappers/include/logging.h"
+
+#ifdef ENABLE_RTC_EVENT_LOG
+
+namespace webrtc {
+
+namespace {
+const int kEventsInHistory = 10000;
+
+bool IsConfigEvent(const rtclog::Event& event) {
+  rtclog::Event_EventType event_type = event.type();
+  return event_type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT ||
+         event_type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT ||
+         event_type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT ||
+         event_type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT;
+}
+}  // namespace
+
+// RtcEventLogImpl member functions.
+RtcEventLogHelperThread::RtcEventLogHelperThread(
+    SwapQueue<ControlMessage>* message_queue,
+    SwapQueue<std::unique_ptr<rtclog::Event>>* event_queue,
+    rtc::Event* wake_up,
+    rtc::Event* stopped,
+    const Clock* const clock)
+    : message_queue_(message_queue),
+      event_queue_(event_queue),
+      history_(kEventsInHistory),
+      config_history_(),
+      file_(FileWrapper::Create()),
+      thread_(&ThreadOutputFunction, this, "RtcEventLog thread"),
+      max_size_bytes_(std::numeric_limits<int64_t>::max()),
+      written_bytes_(0),
+      start_time_(0),
+      stop_time_(std::numeric_limits<int64_t>::max()),
+      has_recent_event_(false),
+      most_recent_event_(),
+      output_string_(),
+      wake_up_(wake_up),
+      stopped_(stopped),
+      clock_(clock) {
+  RTC_DCHECK(message_queue_);
+  RTC_DCHECK(event_queue_);
+  RTC_DCHECK(wake_up_);
+  RTC_DCHECK(stopped_);
+  RTC_DCHECK(clock_);
+  thread_.Start();
+}
+
+RtcEventLogHelperThread::~RtcEventLogHelperThread() {
+  ControlMessage message;
+  message.message_type = ControlMessage::TERMINATE_THREAD;
+  message.stop_time = clock_->TimeInMicroseconds();
+  while (!message_queue_->Insert(&message)) {
+    // We can't destroy the event log until we have stopped the thread,
+    // so clear the message queue and try again. Note that if we clear
+    // any STOP_FILE events, then the threads calling StopLogging would likely
+    // wait indefinitely. However, there should not be any such calls as we
+    // are executing the destructor.
+    LOG(LS_WARNING) << "Clearing message queue to terminate thread.";
+    message_queue_->Clear();
+  }
+  wake_up_->Set();  // Wake up the output thread.
+  thread_.Stop();   // Wait for the thread to terminate.
+}
+
+bool RtcEventLogHelperThread::AppendEventToString(rtclog::Event* event) {
+  rtclog::EventStream event_stream;
+  event_stream.add_stream();
+  event_stream.mutable_stream(0)->Swap(event);
+  // We create a new event stream per event but because of the way protobufs
+  // are encoded, events can be merged by concatenating them. Therefore,
+  // it will look like a single stream when we read it back from file.
+  bool stop = true;
+  if (written_bytes_ + static_cast<int64_t>(output_string_.size()) +
+          event_stream.ByteSize() <=
+      max_size_bytes_) {
+    event_stream.AppendToString(&output_string_);
+    stop = false;
+  }
+  // Swap the event back so that we don't mix event types in the queues.
+  event_stream.mutable_stream(0)->Swap(event);
+  return stop;
+}
+
+void RtcEventLogHelperThread::LogToMemory() {
+  RTC_DCHECK(!file_->Open());
+
+  // Process each event earlier than the current time and append it to the
+  // appropriate history_.
+  int64_t current_time = clock_->TimeInMicroseconds();
+  if (!has_recent_event_) {
+    has_recent_event_ = event_queue_->Remove(&most_recent_event_);
+  }
+  while (has_recent_event_ &&
+         most_recent_event_->timestamp_us() <= current_time) {
+    if (IsConfigEvent(*most_recent_event_)) {
+      config_history_.push_back(std::move(most_recent_event_));
+    } else {
+      history_.push_back(std::move(most_recent_event_));
+    }
+    has_recent_event_ = event_queue_->Remove(&most_recent_event_);
+  }
+}
+
+void RtcEventLogHelperThread::StartLogFile() {
+  RTC_DCHECK(file_->Open());
+  bool stop = false;
+  output_string_.clear();
+
+  // Create and serialize the LOG_START event.
+  rtclog::Event start_event;
+  start_event.set_timestamp_us(start_time_);
+  start_event.set_type(rtclog::Event::LOG_START);
+  AppendEventToString(&start_event);
+
+  // Serialize the config information for all old streams.
+  for (auto& event : config_history_) {
+    AppendEventToString(event.get());
+  }
+
+  // Serialize the events in the event queue.
+  while (!history_.empty() && !stop) {
+    stop = AppendEventToString(history_.front().get());
+    if (!stop) {
+      history_.pop_front();
+    }
+  }
+
+  // Write to file.
+  file_->Write(output_string_.data(), output_string_.size());
+  written_bytes_ += output_string_.size();
+
+  // Free the allocated memory since we probably won't need this amount of
+  // space again.
+  output_string_.clear();
+  output_string_.shrink_to_fit();
+
+  if (stop) {
+    RTC_DCHECK(file_->Open());
+    StopLogFile();
+  }
+}
+
+void RtcEventLogHelperThread::LogToFile() {
+  RTC_DCHECK(file_->Open());
+  output_string_.clear();
+
+  // Append each event older than both the current time and the stop time
+  // to the output_string_.
+  int64_t current_time = clock_->TimeInMicroseconds();
+  int64_t time_limit = std::min(current_time, stop_time_);
+  if (!has_recent_event_) {
+    has_recent_event_ = event_queue_->Remove(&most_recent_event_);
+  }
+  bool stop = false;
+  while (!stop && has_recent_event_ &&
+         most_recent_event_->timestamp_us() <= time_limit) {
+    stop = AppendEventToString(most_recent_event_.get());
+    if (!stop) {
+      if (IsConfigEvent(*most_recent_event_)) {
+        config_history_.push_back(std::move(most_recent_event_));
+      }
+      has_recent_event_ = event_queue_->Remove(&most_recent_event_);
+    }
+  }
+
+  // Write string to file.
+  file_->Write(output_string_.data(), output_string_.size());
+  written_bytes_ += output_string_.size();
+
+  if (!file_->Open()) {
+    LOG(LS_WARNING) << "WebRTC event log file closed by FileWrapper.";
+  }
+
+  // We want to stop logging if we have reached the file size limit. We also
+  // want to stop logging if the remaining events are more recent than the
+  // time limit, or in other words if we have terminated the loop despite
+  // having more events in the queue.
+  if ((has_recent_event_ && most_recent_event_->timestamp_us() > stop_time_) ||
+      stop) {
+    RTC_DCHECK(file_->Open());
+    StopLogFile();
+  }
+}
+
+void RtcEventLogHelperThread::StopLogFile() {
+  RTC_DCHECK(file_->Open());
+  output_string_.clear();
+
+  rtclog::Event end_event;
+  end_event.set_timestamp_us(stop_time_);
+  end_event.set_type(rtclog::Event::LOG_END);
+  AppendEventToString(&end_event);
+
+  if (written_bytes_ + static_cast<int64_t>(output_string_.size()) <=
+      max_size_bytes_) {
+    file_->Write(output_string_.data(), output_string_.size());
+    written_bytes_ += output_string_.size();
+  }
+
+  max_size_bytes_ = std::numeric_limits<int64_t>::max();
+  written_bytes_ = 0;
+  start_time_ = 0;
+  stop_time_ = std::numeric_limits<int64_t>::max();
+  output_string_.clear();
+  file_->CloseFile();
+  RTC_DCHECK(!file_->Open());
+}
+
+void RtcEventLogHelperThread::WriteLog() {
+  ControlMessage message;
+
+  while (true) {
+    // Process control messages.
+    while (message_queue_->Remove(&message)) {
+      switch (message.message_type) {
+        case ControlMessage::START_FILE:
+          if (!file_->Open()) {
+            max_size_bytes_ = message.max_size_bytes;
+            start_time_ = message.start_time;
+            stop_time_ = message.stop_time;
+            file_.swap(message.file);
+            StartLogFile();
+          } else {
+            // Already started. Ignore message and close file handle.
+            message.file->CloseFile();
+          }
+          break;
+        case ControlMessage::STOP_FILE:
+          if (file_->Open()) {
+            stop_time_ = message.stop_time;
+            LogToFile();  // Log remaining events from message queues.
+          }
+          // LogToFile might stop on it's own so we need to recheck the state.
+          if (file_->Open()) {
+            StopLogFile();
+          }
+          stopped_->Set();
+          break;
+        case ControlMessage::TERMINATE_THREAD:
+          if (file_->Open()) {
+            StopLogFile();
+          }
+          return;
+      }
+    }
+
+    // Write events to file or memory
+    if (file_->Open()) {
+      LogToFile();
+    } else {
+      LogToMemory();
+    }
+
+    // Accumulate a new batch of events instead of processing them one at a
+    // time.
+    wake_up_->Wait(50);
+  }
+}
+
+bool RtcEventLogHelperThread::ThreadOutputFunction(void* obj) {
+  RtcEventLogHelperThread* helper = static_cast<RtcEventLogHelperThread*>(obj);
+  helper->WriteLog();
+  return false;
+}
+
+}  // namespace webrtc
+
+#endif  // ENABLE_RTC_EVENT_LOG
diff --git a/chromium/third_party/webrtc/call/rtc_event_log_helper_thread.h b/chromium/third_party/webrtc/call/rtc_event_log_helper_thread.h
new file mode 100644
index 00000000000..60ed912b651
--- /dev/null
+++ b/chromium/third_party/webrtc/call/rtc_event_log_helper_thread.h
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_RTC_EVENT_LOG_HELPER_THREAD_H_
+#define WEBRTC_CALL_RTC_EVENT_LOG_HELPER_THREAD_H_
+
+#include <limits>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/swap_queue.h"
+#include "webrtc/call/ringbuffer.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+
+#ifdef ENABLE_RTC_EVENT_LOG
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+#endif
+
+#ifdef ENABLE_RTC_EVENT_LOG
+
+namespace webrtc {
+
+class RtcEventLogHelperThread final {
+ public:
+  struct ControlMessage {
+    ControlMessage()
+        : message_type(STOP_FILE),
+          file(nullptr),
+          max_size_bytes(0),
+          start_time(0),
+          stop_time(0) {}
+    enum { START_FILE, STOP_FILE, TERMINATE_THREAD } message_type;
+
+    std::unique_ptr<FileWrapper> file;  // Only used with START_FILE.
+    int64_t max_size_bytes;             // Only used with START_FILE.
+    int64_t start_time;                 // Only used with START_FILE.
+    int64_t stop_time;                  // Used with all 3 message types.
+
+    friend void swap(ControlMessage& lhs, ControlMessage& rhs) {
+      using std::swap;
+      swap(lhs.message_type, rhs.message_type);
+      lhs.file.swap(rhs.file);
+      swap(lhs.max_size_bytes, rhs.max_size_bytes);
+      swap(lhs.start_time, rhs.start_time);
+      swap(lhs.stop_time, rhs.stop_time);
+    }
+  };
+
+  RtcEventLogHelperThread(
+      SwapQueue<ControlMessage>* message_queue,
+      SwapQueue<std::unique_ptr<rtclog::Event>>* event_queue,
+      rtc::Event* wake_up,
+      rtc::Event* file_finished,
+      const Clock* const clock);
+  ~RtcEventLogHelperThread();
+
+ private:
+  static bool ThreadOutputFunction(void* obj);
+
+  void TerminateThread();
+  bool AppendEventToString(rtclog::Event* event);
+  void AppendEventToHistory(const rtclog::Event& event);
+  void LogToMemory();
+  void StartLogFile();
+  void LogToFile();
+  void StopLogFile();
+  void WriteLog();
+
+  // Message queues for passing events to the logging thread.
+  SwapQueue<ControlMessage>* message_queue_;
+  SwapQueue<std::unique_ptr<rtclog::Event>>* event_queue_;
+
+  // History containing the most recent events (~ 10 s).
+  RingBuffer<std::unique_ptr<rtclog::Event>> history_;
+
+  // History containing all past configuration events.
+  std::vector<std::unique_ptr<rtclog::Event>> config_history_;
+
+  std::unique_ptr<FileWrapper> file_;
+  rtc::PlatformThread thread_;
+
+  int64_t max_size_bytes_;
+  int64_t written_bytes_;
+  int64_t start_time_;
+  int64_t stop_time_;
+
+  bool has_recent_event_;
+  std::unique_ptr<rtclog::Event> most_recent_event_;
+
+  // Temporary space for serializing profobuf data.
+  std::string output_string_;
+
+  rtc::Event* wake_up_;
+  rtc::Event* stopped_;
+
+  const Clock* const clock_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtcEventLogHelperThread);
+};
+
+}  // namespace webrtc
+
+#endif  // ENABLE_RTC_EVENT_LOG
+
+#endif  // WEBRTC_CALL_RTC_EVENT_LOG_HELPER_THREAD_H_
diff --git a/chromium/third_party/webrtc/call/rtc_event_log_parser.cc b/chromium/third_party/webrtc/call/rtc_event_log_parser.cc
new file mode 100644
index 00000000000..c49b9b47b43
--- /dev/null
+++ b/chromium/third_party/webrtc/call/rtc_event_log_parser.cc
@@ -0,0 +1,394 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/call/rtc_event_log_parser.h"
+
+#include <string.h>
+
+#include <fstream>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/call.h"
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+
+namespace webrtc {
+
+namespace {
+MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
+  switch (media_type) {
+    case rtclog::MediaType::ANY:
+      return MediaType::ANY;
+    case rtclog::MediaType::AUDIO:
+      return MediaType::AUDIO;
+    case rtclog::MediaType::VIDEO:
+      return MediaType::VIDEO;
+    case rtclog::MediaType::DATA:
+      return MediaType::DATA;
+  }
+  RTC_NOTREACHED();
+  return MediaType::ANY;
+}
+
+RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
+  switch (rtcp_mode) {
+    case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
+      return RtcpMode::kCompound;
+    case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
+      return RtcpMode::kReducedSize;
+  }
+  RTC_NOTREACHED();
+  return RtcpMode::kOff;
+}
+
+ParsedRtcEventLog::EventType GetRuntimeEventType(
+    rtclog::Event::EventType event_type) {
+  switch (event_type) {
+    case rtclog::Event::UNKNOWN_EVENT:
+      return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
+    case rtclog::Event::LOG_START:
+      return ParsedRtcEventLog::EventType::LOG_START;
+    case rtclog::Event::LOG_END:
+      return ParsedRtcEventLog::EventType::LOG_END;
+    case rtclog::Event::RTP_EVENT:
+      return ParsedRtcEventLog::EventType::RTP_EVENT;
+    case rtclog::Event::RTCP_EVENT:
+      return ParsedRtcEventLog::EventType::RTCP_EVENT;
+    case rtclog::Event::AUDIO_PLAYOUT_EVENT:
+      return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
+    case rtclog::Event::BWE_PACKET_LOSS_EVENT:
+      return ParsedRtcEventLog::EventType::BWE_PACKET_LOSS_EVENT;
+    case rtclog::Event::BWE_PACKET_DELAY_EVENT:
+      return ParsedRtcEventLog::EventType::BWE_PACKET_DELAY_EVENT;
+    case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
+    case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
+    case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
+    case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
+  }
+  RTC_NOTREACHED();
+  return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
+}
+
+bool ParseVarInt(std::FILE* file, uint64_t* varint, size_t* bytes_read) {
+  uint8_t byte;
+  *varint = 0;
+  for (*bytes_read = 0; *bytes_read < 10 && fread(&byte, 1, 1, file) == 1;
+       ++(*bytes_read)) {
+    // The most significant bit of each byte is 0 if it is the last byte in
+    // the varint and 1 otherwise. Thus, we take the 7 least significant bits
+    // of each byte and shift them 7 bits for each byte read previously to get
+    // the (unsigned) integer.
+    *varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * *bytes_read);
+    if ((byte & 0x80) == 0) {
+      return true;
+    }
+  }
+  return false;
+}
+
+}  // namespace
+
+bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
+  stream_.clear();
+  const size_t kMaxEventSize = (1u << 16) - 1;
+  char tmp_buffer[kMaxEventSize];
+
+  std::FILE* file = fopen(filename.c_str(), "rb");
+  if (!file) {
+    LOG(LS_WARNING) << "Could not open file for reading.";
+    return false;
+  }
+
+  while (1) {
+    // Peek at the next message tag. The tag number is defined as
+    // (fieldnumber << 3) | wire_type. In our case, the field number is
+    // supposed to be 1 and the wire type for an length-delimited field is 2.
+    const uint64_t kExpectedTag = (1 << 3) | 2;
+    uint64_t tag;
+    size_t bytes_read;
+    if (!ParseVarInt(file, &tag, &bytes_read) || tag != kExpectedTag) {
+      fclose(file);
+      if (bytes_read == 0) {
+        return true;  // Reached end of file.
+      }
+      LOG(LS_WARNING) << "Missing field tag from beginning of protobuf event.";
+      return false;
+    }
+
+    // Peek at the length field.
+    uint64_t message_length;
+    if (!ParseVarInt(file, &message_length, &bytes_read)) {
+      LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
+      fclose(file);
+      return false;
+    } else if (message_length > kMaxEventSize) {
+      LOG(LS_WARNING) << "Protobuf message length is too large.";
+      fclose(file);
+      return false;
+    }
+
+    if (fread(tmp_buffer, 1, message_length, file) != message_length) {
+      LOG(LS_WARNING) << "Failed to read protobuf message from file.";
+      fclose(file);
+      return false;
+    }
+
+    rtclog::Event event;
+    if (!event.ParseFromArray(tmp_buffer, message_length)) {
+      LOG(LS_WARNING) << "Failed to parse protobuf message.";
+      fclose(file);
+      return false;
+    }
+    stream_.push_back(event);
+  }
+}
+
+size_t ParsedRtcEventLog::GetNumberOfEvents() const {
+  return stream_.size();
+}
+
+int64_t ParsedRtcEventLog::GetTimestamp(size_t index) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(event.has_timestamp_us());
+  return event.timestamp_us();
+}
+
+ParsedRtcEventLog::EventType ParsedRtcEventLog::GetEventType(
+    size_t index) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(event.has_type());
+  return GetRuntimeEventType(event.type());
+}
+
+// The header must have space for at least IP_PACKET_SIZE bytes.
+void ParsedRtcEventLog::GetRtpHeader(size_t index,
+                                     PacketDirection* incoming,
+                                     MediaType* media_type,
+                                     uint8_t* header,
+                                     size_t* header_length,
+                                     size_t* total_length) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
+  RTC_CHECK(event.has_rtp_packet());
+  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+  // Get direction of packet.
+  RTC_CHECK(rtp_packet.has_incoming());
+  if (incoming != nullptr) {
+    *incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
+  }
+  // Get media type.
+  RTC_CHECK(rtp_packet.has_type());
+  if (media_type != nullptr) {
+    *media_type = GetRuntimeMediaType(rtp_packet.type());
+  }
+  // Get packet length.
+  RTC_CHECK(rtp_packet.has_packet_length());
+  if (total_length != nullptr) {
+    *total_length = rtp_packet.packet_length();
+  }
+  // Get header length.
+  RTC_CHECK(rtp_packet.has_header());
+  if (header_length != nullptr) {
+    *header_length = rtp_packet.header().size();
+  }
+  // Get header contents.
+  if (header != nullptr) {
+    const size_t kMinRtpHeaderSize = 12;
+    RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
+    RTC_CHECK_LE(rtp_packet.header().size(),
+                 static_cast<size_t>(IP_PACKET_SIZE));
+    memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
+  }
+}
+
+// The packet must have space for at least IP_PACKET_SIZE bytes.
+void ParsedRtcEventLog::GetRtcpPacket(size_t index,
+                                      PacketDirection* incoming,
+                                      MediaType* media_type,
+                                      uint8_t* packet,
+                                      size_t* length) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
+  RTC_CHECK(event.has_rtcp_packet());
+  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
+  // Get direction of packet.
+  RTC_CHECK(rtcp_packet.has_incoming());
+  if (incoming != nullptr) {
+    *incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
+  }
+  // Get media type.
+  RTC_CHECK(rtcp_packet.has_type());
+  if (media_type != nullptr) {
+    *media_type = GetRuntimeMediaType(rtcp_packet.type());
+  }
+  // Get packet length.
+  RTC_CHECK(rtcp_packet.has_packet_data());
+  if (length != nullptr) {
+    *length = rtcp_packet.packet_data().size();
+  }
+  // Get packet contents.
+  if (packet != nullptr) {
+    RTC_CHECK_LE(rtcp_packet.packet_data().size(),
+                 static_cast<unsigned>(IP_PACKET_SIZE));
+    memcpy(packet, rtcp_packet.packet_data().data(),
+           rtcp_packet.packet_data().size());
+  }
+}
+
+void ParsedRtcEventLog::GetVideoReceiveConfig(
+    size_t index,
+    VideoReceiveStream::Config* config) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(config != nullptr);
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
+  RTC_CHECK(event.has_video_receiver_config());
+  const rtclog::VideoReceiveConfig& receiver_config =
+      event.video_receiver_config();
+  // Get SSRCs.
+  RTC_CHECK(receiver_config.has_remote_ssrc());
+  config->rtp.remote_ssrc = receiver_config.remote_ssrc();
+  RTC_CHECK(receiver_config.has_local_ssrc());
+  config->rtp.local_ssrc = receiver_config.local_ssrc();
+  // Get RTCP settings.
+  RTC_CHECK(receiver_config.has_rtcp_mode());
+  config->rtp.rtcp_mode = GetRuntimeRtcpMode(receiver_config.rtcp_mode());
+  RTC_CHECK(receiver_config.has_remb());
+  config->rtp.remb = receiver_config.remb();
+  // Get RTX map.
+  config->rtp.rtx.clear();
+  for (int i = 0; i < receiver_config.rtx_map_size(); i++) {
+    const rtclog::RtxMap& map = receiver_config.rtx_map(i);
+    RTC_CHECK(map.has_payload_type());
+    RTC_CHECK(map.has_config());
+    RTC_CHECK(map.config().has_rtx_ssrc());
+    RTC_CHECK(map.config().has_rtx_payload_type());
+    webrtc::VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
+    rtx_pair.ssrc = map.config().rtx_ssrc();
+    rtx_pair.payload_type = map.config().rtx_payload_type();
+    config->rtp.rtx.insert(std::make_pair(map.payload_type(), rtx_pair));
+  }
+  // Get header extensions.
+  config->rtp.extensions.clear();
+  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
+    RTC_CHECK(receiver_config.header_extensions(i).has_name());
+    RTC_CHECK(receiver_config.header_extensions(i).has_id());
+    const std::string& name = receiver_config.header_extensions(i).name();
+    int id = receiver_config.header_extensions(i).id();
+    config->rtp.extensions.push_back(RtpExtension(name, id));
+  }
+  // Get decoders.
+  config->decoders.clear();
+  for (int i = 0; i < receiver_config.decoders_size(); i++) {
+    RTC_CHECK(receiver_config.decoders(i).has_name());
+    RTC_CHECK(receiver_config.decoders(i).has_payload_type());
+    VideoReceiveStream::Decoder decoder;
+    decoder.payload_name = receiver_config.decoders(i).name();
+    decoder.payload_type = receiver_config.decoders(i).payload_type();
+    config->decoders.push_back(decoder);
+  }
+}
+
+void ParsedRtcEventLog::GetVideoSendConfig(
+    size_t index,
+    VideoSendStream::Config* config) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(config != nullptr);
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
+  RTC_CHECK(event.has_video_sender_config());
+  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
+  // Get SSRCs.
+  config->rtp.ssrcs.clear();
+  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
+    config->rtp.ssrcs.push_back(sender_config.ssrcs(i));
+  }
+  // Get header extensions.
+  config->rtp.extensions.clear();
+  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
+    RTC_CHECK(sender_config.header_extensions(i).has_name());
+    RTC_CHECK(sender_config.header_extensions(i).has_id());
+    const std::string& name = sender_config.header_extensions(i).name();
+    int id = sender_config.header_extensions(i).id();
+    config->rtp.extensions.push_back(RtpExtension(name, id));
+  }
+  // Get RTX settings.
+  config->rtp.rtx.ssrcs.clear();
+  for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
+    config->rtp.rtx.ssrcs.push_back(sender_config.rtx_ssrcs(i));
+  }
+  if (sender_config.rtx_ssrcs_size() > 0) {
+    RTC_CHECK(sender_config.has_rtx_payload_type());
+    config->rtp.rtx.payload_type = sender_config.rtx_payload_type();
+  } else {
+    // Reset RTX payload type default value if no RTX SSRCs are used.
+    config->rtp.rtx.payload_type = -1;
+  }
+  // Get encoder.
+  RTC_CHECK(sender_config.has_encoder());
+  RTC_CHECK(sender_config.encoder().has_name());
+  RTC_CHECK(sender_config.encoder().has_payload_type());
+  config->encoder_settings.payload_name = sender_config.encoder().name();
+  config->encoder_settings.payload_type =
+      sender_config.encoder().payload_type();
+}
+
+void ParsedRtcEventLog::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
+  RTC_CHECK(event.has_audio_playout_event());
+  const rtclog::AudioPlayoutEvent& loss_event = event.audio_playout_event();
+  RTC_CHECK(loss_event.has_local_ssrc());
+  if (ssrc != nullptr) {
+    *ssrc = loss_event.local_ssrc();
+  }
+}
+
+void ParsedRtcEventLog::GetBwePacketLossEvent(size_t index,
+                                              int32_t* bitrate,
+                                              uint8_t* fraction_loss,
+                                              int32_t* total_packets) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PACKET_LOSS_EVENT);
+  RTC_CHECK(event.has_bwe_packet_loss_event());
+  const rtclog::BwePacketLossEvent& loss_event = event.bwe_packet_loss_event();
+  RTC_CHECK(loss_event.has_bitrate());
+  if (bitrate != nullptr) {
+    *bitrate = loss_event.bitrate();
+  }
+  RTC_CHECK(loss_event.has_fraction_loss());
+  if (fraction_loss != nullptr) {
+    *fraction_loss = loss_event.fraction_loss();
+  }
+  RTC_CHECK(loss_event.has_total_packets());
+  if (total_packets != nullptr) {
+    *total_packets = loss_event.total_packets();
+  }
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/call/rtc_event_log_parser.h b/chromium/third_party/webrtc/call/rtc_event_log_parser.h
new file mode 100644
index 00000000000..acdfa77da17
--- /dev/null
+++ b/chromium/third_party/webrtc/call/rtc_event_log_parser.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
+#define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/video_receive_stream.h"
+#include "webrtc/video_send_stream.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+
+enum class MediaType;
+
+class ParsedRtcEventLog {
+  friend class RtcEventLogTestHelper;
+
+ public:
+  enum EventType {
+    UNKNOWN_EVENT = 0,
+    LOG_START = 1,
+    LOG_END = 2,
+    RTP_EVENT = 3,
+    RTCP_EVENT = 4,
+    AUDIO_PLAYOUT_EVENT = 5,
+    BWE_PACKET_LOSS_EVENT = 6,
+    BWE_PACKET_DELAY_EVENT = 7,
+    VIDEO_RECEIVER_CONFIG_EVENT = 8,
+    VIDEO_SENDER_CONFIG_EVENT = 9,
+    AUDIO_RECEIVER_CONFIG_EVENT = 10,
+    AUDIO_SENDER_CONFIG_EVENT = 11
+  };
+
+  // Reads an RtcEventLog file and returns true if parsing was successful.
+  bool ParseFile(const std::string& file_name);
+
+  // Returns the number of events in an EventStream.
+  size_t GetNumberOfEvents() const;
+
+  // Reads the arrival timestamp (in microseconds) from a rtclog::Event.
+  int64_t GetTimestamp(size_t index) const;
+
+  // Reads the event type of the rtclog::Event at |index|.
+  EventType GetEventType(size_t index) const;
+
+  // Reads the header, direction, media type, header length and packet length
+  // from the RTP event at |index|, and stores the values in the corresponding
+  // output parameters. The output parameters can be set to nullptr if those
+  // values aren't needed.
+  // NB: The header must have space for at least IP_PACKET_SIZE bytes.
+  void GetRtpHeader(size_t index,
+                    PacketDirection* incoming,
+                    MediaType* media_type,
+                    uint8_t* header,
+                    size_t* header_length,
+                    size_t* total_length) const;
+
+  // Reads packet, direction, media type and packet length from the RTCP event
+  // at |index|, and stores the values in the corresponding output parameters.
+  // The output parameters can be set to nullptr if those values aren't needed.
+  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
+  void GetRtcpPacket(size_t index,
+                     PacketDirection* incoming,
+                     MediaType* media_type,
+                     uint8_t* packet,
+                     size_t* length) const;
+
+  // Reads a config event to a (non-NULL) VideoReceiveStream::Config struct.
+  // Only the fields that are stored in the protobuf will be written.
+  void GetVideoReceiveConfig(size_t index,
+                             VideoReceiveStream::Config* config) const;
+
+  // Reads a config event to a (non-NULL) VideoSendStream::Config struct.
+  // Only the fields that are stored in the protobuf will be written.
+  void GetVideoSendConfig(size_t index, VideoSendStream::Config* config) const;
+
+  // Reads the SSRC from the audio playout event at |index|. The SSRC is stored
+  // in the output parameter ssrc. The output parameter can be set to nullptr
+  // and in that case the function only asserts that the event is well formed.
+  void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
+
+  // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
+  // expected packets from the BWE event at |index| and stores the values in
+  // the corresponding output parameters. The output parameters can be set to
+  // nullptr if those values aren't needed.
+  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
+  void GetBwePacketLossEvent(size_t index,
+                             int32_t* bitrate,
+                             uint8_t* fraction_loss,
+                             int32_t* total_packets) const;
+
+ private:
+  std::vector<rtclog::Event> stream_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
diff --git a/chromium/third_party/webrtc/call/rtc_event_log_unittest.cc b/chromium/third_party/webrtc/call/rtc_event_log_unittest.cc
index e3104591d9a..067c44123b9 100644
--- a/chromium/third_party/webrtc/call/rtc_event_log_unittest.cc
+++ b/chromium/third_party/webrtc/call/rtc_event_log_unittest.cc
@@ -10,6 +10,7 @@
 
 #ifdef ENABLE_RTC_EVENT_LOG
 
+#include <map>
 #include <memory>
 #include <string>
 #include <utility>
@@ -19,9 +20,10 @@
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"
-#include "webrtc/base/thread.h"
 #include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
+#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/call/rtc_event_log_unittest_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
@@ -53,244 +55,50 @@ const char* kExtensionNames[] = {RtpExtension::kTOffset,
                                  RtpExtension::kTransportSequenceNumber};
 const size_t kNumExtensions = 5;
 
-}  // namespace
-
-// TODO(terelius): Place this definition with other parsing functions?
-MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
-  switch (media_type) {
-    case rtclog::MediaType::ANY:
-      return MediaType::ANY;
-    case rtclog::MediaType::AUDIO:
-      return MediaType::AUDIO;
-    case rtclog::MediaType::VIDEO:
-      return MediaType::VIDEO;
-    case rtclog::MediaType::DATA:
-      return MediaType::DATA;
-  }
-  RTC_NOTREACHED();
-  return MediaType::ANY;
-}
-
-// Checks that the event has a timestamp, a type and exactly the data field
-// corresponding to the type.
-::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
-  if (!event.has_timestamp_us())
-    return ::testing::AssertionFailure() << "Event has no timestamp";
-  if (!event.has_type())
-    return ::testing::AssertionFailure() << "Event has no event type";
-  rtclog::Event_EventType type = event.type();
-  if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet())
-    return ::testing::AssertionFailure()
-           << "Event of type " << type << " has "
-           << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
-  if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet())
-    return ::testing::AssertionFailure()
-           << "Event of type " << type << " has "
-           << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
-  if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
-      event.has_audio_playout_event())
-    return ::testing::AssertionFailure()
-           << "Event of type " << type << " has "
-           << (event.has_audio_playout_event() ? "" : "no ")
-           << "audio_playout event";
-  if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) !=
-      event.has_video_receiver_config())
-    return ::testing::AssertionFailure()
-           << "Event of type " << type << " has "
-           << (event.has_video_receiver_config() ? "" : "no ")
-           << "receiver config";
-  if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
-      event.has_video_sender_config())
-    return ::testing::AssertionFailure()
-           << "Event of type " << type << " has "
-           << (event.has_video_sender_config() ? "" : "no ") << "sender config";
-  if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
-      event.has_audio_receiver_config()) {
-    return ::testing::AssertionFailure()
-           << "Event of type " << type << " has "
-           << (event.has_audio_receiver_config() ? "" : "no ")
-           << "audio receiver config";
-  }
-  if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) !=
-      event.has_audio_sender_config()) {
-    return ::testing::AssertionFailure()
-           << "Event of type " << type << " has "
-           << (event.has_audio_sender_config() ? "" : "no ")
-           << "audio sender config";
-  }
-  return ::testing::AssertionSuccess();
-}
-
-void VerifyReceiveStreamConfig(const rtclog::Event& event,
-                               const VideoReceiveStream::Config& config) {
-  ASSERT_TRUE(IsValidBasicEvent(event));
-  ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
-  const rtclog::VideoReceiveConfig& receiver_config =
-      event.video_receiver_config();
-  // Check SSRCs.
-  ASSERT_TRUE(receiver_config.has_remote_ssrc());
-  EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
-  ASSERT_TRUE(receiver_config.has_local_ssrc());
-  EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
-  // Check RTCP settings.
-  ASSERT_TRUE(receiver_config.has_rtcp_mode());
-  if (config.rtp.rtcp_mode == RtcpMode::kCompound)
-    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
-              receiver_config.rtcp_mode());
-  else
-    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
-              receiver_config.rtcp_mode());
-  ASSERT_TRUE(receiver_config.has_remb());
-  EXPECT_EQ(config.rtp.remb, receiver_config.remb());
-  // Check RTX map.
-  ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
-            receiver_config.rtx_map_size());
-  for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
-    ASSERT_TRUE(rtx_map.has_payload_type());
-    ASSERT_TRUE(rtx_map.has_config());
-    EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
-    const rtclog::RtxConfig& rtx_config = rtx_map.config();
-    const VideoReceiveStream::Config::Rtp::Rtx& rtx =
-        config.rtp.rtx.at(rtx_map.payload_type());
-    ASSERT_TRUE(rtx_config.has_rtx_ssrc());
-    ASSERT_TRUE(rtx_config.has_rtx_payload_type());
-    EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
-    EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
-  }
-  // Check header extensions.
-  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
-            receiver_config.header_extensions_size());
-  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
-    ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
-    ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
-    const std::string& name = receiver_config.header_extensions(i).name();
-    int id = receiver_config.header_extensions(i).id();
-    EXPECT_EQ(config.rtp.extensions[i].id, id);
-    EXPECT_EQ(config.rtp.extensions[i].name, name);
-  }
-  // Check decoders.
-  ASSERT_EQ(static_cast<int>(config.decoders.size()),
-            receiver_config.decoders_size());
-  for (int i = 0; i < receiver_config.decoders_size(); i++) {
-    ASSERT_TRUE(receiver_config.decoders(i).has_name());
-    ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
-    const std::string& decoder_name = receiver_config.decoders(i).name();
-    int decoder_type = receiver_config.decoders(i).payload_type();
-    EXPECT_EQ(config.decoders[i].payload_name, decoder_name);
-    EXPECT_EQ(config.decoders[i].payload_type, decoder_type);
-  }
-}
-
-void VerifySendStreamConfig(const rtclog::Event& event,
-                            const VideoSendStream::Config& config) {
-  ASSERT_TRUE(IsValidBasicEvent(event));
-  ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
-  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
-  // Check SSRCs.
-  ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()),
-            sender_config.ssrcs_size());
-  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
-    EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
+void PrintActualEvents(const ParsedRtcEventLog& parsed_log) {
+  std::map<int, size_t> actual_event_counts;
+  for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
+    actual_event_counts[parsed_log.GetEventType(i)]++;
   }
-  // Check header extensions.
-  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
-            sender_config.header_extensions_size());
-  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
-    ASSERT_TRUE(sender_config.header_extensions(i).has_name());
-    ASSERT_TRUE(sender_config.header_extensions(i).has_id());
-    const std::string& name = sender_config.header_extensions(i).name();
-    int id = sender_config.header_extensions(i).id();
-    EXPECT_EQ(config.rtp.extensions[i].id, id);
-    EXPECT_EQ(config.rtp.extensions[i].name, name);
+  printf("Actual events: ");
+  for (auto kv : actual_event_counts) {
+    printf("%d_count = %zu, ", kv.first, kv.second);
   }
-  // Check RTX settings.
-  ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
-            sender_config.rtx_ssrcs_size());
-  for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
-    EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
+  printf("\n");
+  for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
+    printf("%4d ", parsed_log.GetEventType(i));
   }
-  if (sender_config.rtx_ssrcs_size() > 0) {
-    ASSERT_TRUE(sender_config.has_rtx_payload_type());
-    EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
-  }
-  // Check encoder.
-  ASSERT_TRUE(sender_config.has_encoder());
-  ASSERT_TRUE(sender_config.encoder().has_name());
-  ASSERT_TRUE(sender_config.encoder().has_payload_type());
-  EXPECT_EQ(config.encoder_settings.payload_name,
-            sender_config.encoder().name());
-  EXPECT_EQ(config.encoder_settings.payload_type,
-            sender_config.encoder().payload_type());
+  printf("\n");
 }
 
-void VerifyRtpEvent(const rtclog::Event& event,
-                    bool incoming,
-                    MediaType media_type,
-                    const uint8_t* header,
-                    size_t header_size,
-                    size_t total_size) {
-  ASSERT_TRUE(IsValidBasicEvent(event));
-  ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
-  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
-  ASSERT_TRUE(rtp_packet.has_incoming());
-  EXPECT_EQ(incoming, rtp_packet.incoming());
-  ASSERT_TRUE(rtp_packet.has_type());
-  EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
-  ASSERT_TRUE(rtp_packet.has_packet_length());
-  EXPECT_EQ(total_size, rtp_packet.packet_length());
-  ASSERT_TRUE(rtp_packet.has_header());
-  ASSERT_EQ(header_size, rtp_packet.header().size());
-  for (size_t i = 0; i < header_size; i++) {
-    EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
-  }
-}
-
-void VerifyRtcpEvent(const rtclog::Event& event,
-                     bool incoming,
-                     MediaType media_type,
-                     const uint8_t* packet,
-                     size_t total_size) {
-  ASSERT_TRUE(IsValidBasicEvent(event));
-  ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
-  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
-  ASSERT_TRUE(rtcp_packet.has_incoming());
-  EXPECT_EQ(incoming, rtcp_packet.incoming());
-  ASSERT_TRUE(rtcp_packet.has_type());
-  EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
-  ASSERT_TRUE(rtcp_packet.has_packet_data());
-  ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
-  for (size_t i = 0; i < total_size; i++) {
-    EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
+void PrintExpectedEvents(size_t rtp_count,
+                         size_t rtcp_count,
+                         size_t playout_count,
+                         size_t bwe_loss_count) {
+  printf(
+      "Expected events: rtp_count = %zu, rtcp_count = %zu,"
+      "playout_count = %zu, bwe_loss_count = %zu\n",
+      rtp_count, rtcp_count, playout_count, bwe_loss_count);
+  size_t rtcp_index = 1, playout_index = 1, bwe_loss_index = 1;
+  printf("strt  cfg  cfg ");
+  for (size_t i = 1; i <= rtp_count; i++) {
+    printf(" rtp ");
+    if (i * rtcp_count >= rtcp_index * rtp_count) {
+      printf("rtcp ");
+      rtcp_index++;
+    }
+    if (i * playout_count >= playout_index * rtp_count) {
+      printf("play ");
+      playout_index++;
+    }
+    if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
+      printf("loss ");
+      bwe_loss_index++;
+    }
   }
+  printf("end \n");
 }
-
-void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) {
-  ASSERT_TRUE(IsValidBasicEvent(event));
-  ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
-  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
-  ASSERT_TRUE(playout_event.has_local_ssrc());
-  EXPECT_EQ(ssrc, playout_event.local_ssrc());
-}
-
-void VerifyBweLossEvent(const rtclog::Event& event,
-                        int32_t bitrate,
-                        uint8_t fraction_loss,
-                        int32_t total_packets) {
-  ASSERT_TRUE(IsValidBasicEvent(event));
-  ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
-  const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
-  ASSERT_TRUE(bwe_event.has_bitrate());
-  EXPECT_EQ(bitrate, bwe_event.bitrate());
-  ASSERT_TRUE(bwe_event.has_fraction_loss());
-  EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
-  ASSERT_TRUE(bwe_event.has_total_packets());
-  EXPECT_EQ(total_packets, bwe_event.total_packets());
-}
-
-void VerifyLogStartEvent(const rtclog::Event& event) {
-  ASSERT_TRUE(IsValidBasicEvent(event));
-  EXPECT_EQ(rtclog::Event::LOG_START, event.type());
-}
+}  // namespace
 
 /*
  * Bit number i of extension_bitvector is set to indicate the
@@ -314,7 +122,8 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
                        nullptr,   // BitrateStatisticsObserver*
                        nullptr,   // FrameCountObserver*
                        nullptr,   // SendSideDelayObserver*
-                       nullptr);  // RtcEventLog*
+                       nullptr,   // RtcEventLog*
+                       nullptr);  // SendPacketObserver*
 
   std::vector<uint32_t> csrcs;
   for (unsigned i = 0; i < csrcs_count; i++) {
@@ -472,9 +281,12 @@ void LogSessionAndReadBack(size_t rtp_count,
   // When log_dumper goes out of scope, it causes the log file to be flushed
   // to disk.
   {
-    std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    SimulatedClock fake_clock(prng.Rand<uint32_t>());
+    std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create(&fake_clock));
     log_dumper->LogVideoReceiveStreamConfig(receiver_config);
+    fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
     log_dumper->LogVideoSendStreamConfig(sender_config);
+    fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
     size_t rtcp_index = 1;
     size_t playout_index = 1;
     size_t bwe_loss_index = 1;
@@ -483,6 +295,7 @@ void LogSessionAndReadBack(size_t rtp_count,
           (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
           (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
           rtp_packets[i - 1].data(), rtp_packets[i - 1].size());
+      fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
       if (i * rtcp_count >= rtcp_index * rtp_count) {
         log_dumper->LogRtcpPacket(
             (rtcp_index % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
@@ -490,73 +303,84 @@ void LogSessionAndReadBack(size_t rtp_count,
             rtcp_packets[rtcp_index - 1].data(),
             rtcp_packets[rtcp_index - 1].size());
         rtcp_index++;
+        fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
       }
       if (i * playout_count >= playout_index * rtp_count) {
         log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
         playout_index++;
+        fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
       }
       if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
         log_dumper->LogBwePacketLossEvent(
             bwe_loss_updates[bwe_loss_index - 1].first,
             bwe_loss_updates[bwe_loss_index - 1].second, i);
         bwe_loss_index++;
+        fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
       }
       if (i == rtp_count / 2) {
         log_dumper->StartLogging(temp_filename, 10000000);
+        fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
       }
     }
+    log_dumper->StopLogging();
   }
 
   // Read the generated file from disk.
-  rtclog::EventStream parsed_stream;
+  ParsedRtcEventLog parsed_log;
 
-  ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
+  ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
 
   // Verify that what we read back from the event log is the same as
   // what we wrote down. For RTCP we log the full packets, but for
   // RTP we should only log the header.
-  const int event_count = config_count + playout_count + bwe_loss_count +
-                          rtcp_count + rtp_count + 1;
-  EXPECT_EQ(event_count, parsed_stream.stream_size());
-  VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
-  VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
-  size_t event_index = config_count;
+  const size_t event_count = config_count + playout_count + bwe_loss_count +
+                             rtcp_count + rtp_count + 2;
+  EXPECT_GE(1000u, event_count);  // The events must fit in the message queue.
+  EXPECT_EQ(event_count, parsed_log.GetNumberOfEvents());
+  if (event_count != parsed_log.GetNumberOfEvents()) {
+    // Print the expected and actual event types for easier debugging.
+    PrintActualEvents(parsed_log);
+    PrintExpectedEvents(rtp_count, rtcp_count, playout_count, bwe_loss_count);
+  }
+  RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
+  RtcEventLogTestHelper::VerifyReceiveStreamConfig(parsed_log, 1,
+                                                   receiver_config);
+  RtcEventLogTestHelper::VerifySendStreamConfig(parsed_log, 2, sender_config);
+  size_t event_index = config_count + 1;
   size_t rtcp_index = 1;
   size_t playout_index = 1;
   size_t bwe_loss_index = 1;
   for (size_t i = 1; i <= rtp_count; i++) {
-    VerifyRtpEvent(parsed_stream.stream(event_index),
-                   (i % 2 == 0),  // Every second packet is incoming.
-                   (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
-                   rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
-                   rtp_packets[i - 1].size());
+    RtcEventLogTestHelper::VerifyRtpEvent(
+        parsed_log, event_index,
+        (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
+        (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+        rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
+        rtp_packets[i - 1].size());
     event_index++;
     if (i * rtcp_count >= rtcp_index * rtp_count) {
-      VerifyRtcpEvent(parsed_stream.stream(event_index),
-                      rtcp_index % 2 == 0,  // Every second packet is incoming.
-                      rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
-                      rtcp_packets[rtcp_index - 1].data(),
-                      rtcp_packets[rtcp_index - 1].size());
+      RtcEventLogTestHelper::VerifyRtcpEvent(
+          parsed_log, event_index,
+          rtcp_index % 2 == 0 ? kIncomingPacket : kOutgoingPacket,
+          rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
+          rtcp_packets[rtcp_index - 1].data(),
+          rtcp_packets[rtcp_index - 1].size());
       event_index++;
       rtcp_index++;
     }
     if (i * playout_count >= playout_index * rtp_count) {
-      VerifyPlayoutEvent(parsed_stream.stream(event_index),
-                         playout_ssrcs[playout_index - 1]);
+      RtcEventLogTestHelper::VerifyPlayoutEvent(
+          parsed_log, event_index, playout_ssrcs[playout_index - 1]);
       event_index++;
       playout_index++;
     }
     if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
-      VerifyBweLossEvent(parsed_stream.stream(event_index),
-                         bwe_loss_updates[bwe_loss_index - 1].first,
-                         bwe_loss_updates[bwe_loss_index - 1].second, i);
+      RtcEventLogTestHelper::VerifyBweLossEvent(
+          parsed_log, event_index, bwe_loss_updates[bwe_loss_index - 1].first,
+          bwe_loss_updates[bwe_loss_index - 1].second, i);
       event_index++;
       bwe_loss_index++;
     }
-    if (i == rtp_count / 2) {
-      VerifyLogStartEvent(parsed_stream.stream(event_index));
-      event_index++;
-    }
   }
 
   // Clean up temporary file - can be pretty slow.
@@ -596,39 +420,15 @@ TEST(RtcEventLogTest, LogSessionAndReadBack) {
   }
 }
 
-// Tests that the event queue works correctly, i.e. drops old RTP, RTCP and
-// debug events, but keeps config events even if they are older than the limit.
-void DropOldEvents(uint32_t extensions_bitvector,
-                   uint32_t csrcs_count,
-                   unsigned int random_seed) {
-  rtc::Buffer old_rtp_packet;
-  rtc::Buffer recent_rtp_packet;
-  rtc::Buffer old_rtcp_packet;
-  rtc::Buffer recent_rtcp_packet;
-
-  VideoReceiveStream::Config receiver_config(nullptr);
-  VideoSendStream::Config sender_config(nullptr);
-
-  Random prng(random_seed);
+TEST(RtcEventLogTest, LogEventAndReadBack) {
+  Random prng(987654321);
 
-  // Create two RTP packets containing random data.
+  // Create one RTP and one RTCP packet containing random data.
   size_t packet_size = prng.Rand(1000, 1100);
-  old_rtp_packet.SetSize(packet_size);
-  GenerateRtpPacket(extensions_bitvector, csrcs_count, old_rtp_packet.data(),
-                    packet_size, &prng);
-  packet_size = prng.Rand(1000, 1100);
-  recent_rtp_packet.SetSize(packet_size);
-  size_t recent_header_size =
-      GenerateRtpPacket(extensions_bitvector, csrcs_count,
-                        recent_rtp_packet.data(), packet_size, &prng);
-
-  // Create two RTCP packets containing random data.
-  old_rtcp_packet = GenerateRtcpPacket(&prng);
-  recent_rtcp_packet = GenerateRtcpPacket(&prng);
-
-  // Create configurations for the video streams.
-  GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
-  GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
+  rtc::Buffer rtp_packet(packet_size);
+  size_t header_size =
+      GenerateRtpPacket(0, 0, rtp_packet.data(), packet_size, &prng);
+  rtc::Buffer rtcp_packet = GenerateRtcpPacket(&prng);
 
   // Find the name of the current test, in order to use it as a temporary
   // filename.
@@ -636,58 +436,46 @@ void DropOldEvents(uint32_t extensions_bitvector,
   const std::string temp_filename =
       test::OutputPath() + test_info->test_case_name() + test_info->name();
 
-  // The log file will be flushed to disk when the log_dumper goes out of scope.
-  {
-    std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
-    // Reduce the time old events are stored to 50 ms.
-    log_dumper->SetBufferDuration(50000);
-    log_dumper->LogVideoReceiveStreamConfig(receiver_config);
-    log_dumper->LogVideoSendStreamConfig(sender_config);
-    log_dumper->LogRtpHeader(kOutgoingPacket, MediaType::AUDIO,
-                             old_rtp_packet.data(), old_rtp_packet.size());
-    log_dumper->LogRtcpPacket(kIncomingPacket, MediaType::AUDIO,
-                              old_rtcp_packet.data(),
-                              old_rtcp_packet.size());
-    // Sleep 55 ms to let old events be removed from the queue.
-    rtc::Thread::SleepMs(55);
-    log_dumper->StartLogging(temp_filename, 10000000);
-    log_dumper->LogRtpHeader(kIncomingPacket, MediaType::VIDEO,
-                             recent_rtp_packet.data(),
-                             recent_rtp_packet.size());
-    log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO,
-                              recent_rtcp_packet.data(),
-                              recent_rtcp_packet.size());
-  }
+  // Add RTP, start logging, add RTCP and then stop logging
+  SimulatedClock fake_clock(prng.Rand<uint32_t>());
+  std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create(&fake_clock));
+
+  log_dumper->LogRtpHeader(kIncomingPacket, MediaType::VIDEO, rtp_packet.data(),
+                           rtp_packet.size());
+  fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
+
+  log_dumper->StartLogging(temp_filename, 10000000);
+  fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
+
+  log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO,
+                            rtcp_packet.data(), rtcp_packet.size());
+  fake_clock.AdvanceTimeMicroseconds(prng.Rand(1, 1000));
+
+  log_dumper->StopLogging();
 
   // Read the generated file from disk.
-  rtclog::EventStream parsed_stream;
-  ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
+  ParsedRtcEventLog parsed_log;
+  ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
 
   // Verify that what we read back from the event log is the same as
-  // what we wrote. Old RTP and RTCP events should have been discarded,
-  // but old configuration events should still be available.
-  EXPECT_EQ(5, parsed_stream.stream_size());
-  VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
-  VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
-  VerifyLogStartEvent(parsed_stream.stream(2));
-  VerifyRtpEvent(parsed_stream.stream(3), true, MediaType::VIDEO,
-                 recent_rtp_packet.data(), recent_header_size,
-                 recent_rtp_packet.size());
-  VerifyRtcpEvent(parsed_stream.stream(4), false, MediaType::VIDEO,
-                  recent_rtcp_packet.data(), recent_rtcp_packet.size());
+  // what we wrote down.
+  EXPECT_EQ(4u, parsed_log.GetNumberOfEvents());
+
+  RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
+
+  RtcEventLogTestHelper::VerifyRtpEvent(parsed_log, 1, kIncomingPacket,
+                                        MediaType::VIDEO, rtp_packet.data(),
+                                        header_size, rtp_packet.size());
+
+  RtcEventLogTestHelper::VerifyRtcpEvent(parsed_log, 2, kOutgoingPacket,
+                                         MediaType::VIDEO, rtcp_packet.data(),
+                                         rtcp_packet.size());
+
+  RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3);
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
-
-TEST(RtcEventLogTest, DropOldEvents) {
-  // Enable all header extensions
-  uint32_t extensions = (1u << kNumExtensions) - 1;
-  uint32_t csrcs_count = 2;
-  DropOldEvents(extensions, csrcs_count, 141421356);
-  DropOldEvents(extensions, csrcs_count, 173205080);
-}
-
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG
diff --git a/chromium/third_party/webrtc/call/rtc_event_log_unittest_helper.cc b/chromium/third_party/webrtc/call/rtc_event_log_unittest_helper.cc
new file mode 100644
index 00000000000..5a06d97170e
--- /dev/null
+++ b/chromium/third_party/webrtc/call/rtc_event_log_unittest_helper.cc
@@ -0,0 +1,409 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifdef ENABLE_RTC_EVENT_LOG
+
+#include "webrtc/call/rtc_event_log_unittest_helper.h"
+
+#include <string.h>
+
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/test/test_suite.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+
+namespace {
+MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
+  switch (media_type) {
+    case rtclog::MediaType::ANY:
+      return MediaType::ANY;
+    case rtclog::MediaType::AUDIO:
+      return MediaType::AUDIO;
+    case rtclog::MediaType::VIDEO:
+      return MediaType::VIDEO;
+    case rtclog::MediaType::DATA:
+      return MediaType::DATA;
+  }
+  RTC_NOTREACHED();
+  return MediaType::ANY;
+}
+}  // namespace
+
+// Checks that the event has a timestamp, a type and exactly the data field
+// corresponding to the type.
+::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
+  if (!event.has_timestamp_us()) {
+    return ::testing::AssertionFailure() << "Event has no timestamp";
+  }
+  if (!event.has_type()) {
+    return ::testing::AssertionFailure() << "Event has no event type";
+  }
+  rtclog::Event_EventType type = event.type();
+  if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
+  }
+  if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
+  }
+  if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
+      event.has_audio_playout_event()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_audio_playout_event() ? "" : "no ")
+           << "audio_playout event";
+  }
+  if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) !=
+      event.has_video_receiver_config()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_video_receiver_config() ? "" : "no ")
+           << "receiver config";
+  }
+  if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
+      event.has_video_sender_config()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_video_sender_config() ? "" : "no ") << "sender config";
+  }
+  if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
+      event.has_audio_receiver_config()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_audio_receiver_config() ? "" : "no ")
+           << "audio receiver config";
+  }
+  if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) !=
+      event.has_audio_sender_config()) {
+    return ::testing::AssertionFailure()
+           << "Event of type " << type << " has "
+           << (event.has_audio_sender_config() ? "" : "no ")
+           << "audio sender config";
+  }
+  return ::testing::AssertionSuccess();
+}
+
+void RtcEventLogTestHelper::VerifyReceiveStreamConfig(
+    const ParsedRtcEventLog& parsed_log,
+    size_t index,
+    const VideoReceiveStream::Config& config) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
+  const rtclog::VideoReceiveConfig& receiver_config =
+      event.video_receiver_config();
+  // Check SSRCs.
+  ASSERT_TRUE(receiver_config.has_remote_ssrc());
+  EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
+  ASSERT_TRUE(receiver_config.has_local_ssrc());
+  EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
+  // Check RTCP settings.
+  ASSERT_TRUE(receiver_config.has_rtcp_mode());
+  if (config.rtp.rtcp_mode == RtcpMode::kCompound) {
+    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
+              receiver_config.rtcp_mode());
+  } else {
+    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
+              receiver_config.rtcp_mode());
+  }
+  ASSERT_TRUE(receiver_config.has_remb());
+  EXPECT_EQ(config.rtp.remb, receiver_config.remb());
+  // Check RTX map.
+  ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
+            receiver_config.rtx_map_size());
+  for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
+    ASSERT_TRUE(rtx_map.has_payload_type());
+    ASSERT_TRUE(rtx_map.has_config());
+    EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
+    const rtclog::RtxConfig& rtx_config = rtx_map.config();
+    const VideoReceiveStream::Config::Rtp::Rtx& rtx =
+        config.rtp.rtx.at(rtx_map.payload_type());
+    ASSERT_TRUE(rtx_config.has_rtx_ssrc());
+    ASSERT_TRUE(rtx_config.has_rtx_payload_type());
+    EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
+    EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
+  }
+  // Check header extensions.
+  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
+            receiver_config.header_extensions_size());
+  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
+    ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
+    ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
+    const std::string& name = receiver_config.header_extensions(i).name();
+    int id = receiver_config.header_extensions(i).id();
+    EXPECT_EQ(config.rtp.extensions[i].id, id);
+    EXPECT_EQ(config.rtp.extensions[i].name, name);
+  }
+  // Check decoders.
+  ASSERT_EQ(static_cast<int>(config.decoders.size()),
+            receiver_config.decoders_size());
+  for (int i = 0; i < receiver_config.decoders_size(); i++) {
+    ASSERT_TRUE(receiver_config.decoders(i).has_name());
+    ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
+    const std::string& decoder_name = receiver_config.decoders(i).name();
+    int decoder_type = receiver_config.decoders(i).payload_type();
+    EXPECT_EQ(config.decoders[i].payload_name, decoder_name);
+    EXPECT_EQ(config.decoders[i].payload_type, decoder_type);
+  }
+
+  // Check consistency of the parser.
+  VideoReceiveStream::Config parsed_config(nullptr);
+  parsed_log.GetVideoReceiveConfig(index, &parsed_config);
+  EXPECT_EQ(config.rtp.remote_ssrc, parsed_config.rtp.remote_ssrc);
+  EXPECT_EQ(config.rtp.local_ssrc, parsed_config.rtp.local_ssrc);
+  // Check RTCP settings.
+  EXPECT_EQ(config.rtp.rtcp_mode, parsed_config.rtp.rtcp_mode);
+  EXPECT_EQ(config.rtp.remb, parsed_config.rtp.remb);
+  // Check RTX map.
+  EXPECT_EQ(config.rtp.rtx.size(), parsed_config.rtp.rtx.size());
+  for (const auto& kv : config.rtp.rtx) {
+    auto parsed_kv = parsed_config.rtp.rtx.find(kv.first);
+    EXPECT_EQ(kv.first, parsed_kv->first);
+    EXPECT_EQ(kv.second.ssrc, parsed_kv->second.ssrc);
+    EXPECT_EQ(kv.second.payload_type, parsed_kv->second.payload_type);
+  }
+  // Check header extensions.
+  EXPECT_EQ(config.rtp.extensions.size(), parsed_config.rtp.extensions.size());
+  for (size_t i = 0; i < parsed_config.rtp.extensions.size(); i++) {
+    EXPECT_EQ(config.rtp.extensions[i].name,
+              parsed_config.rtp.extensions[i].name);
+    EXPECT_EQ(config.rtp.extensions[i].id, parsed_config.rtp.extensions[i].id);
+  }
+  // Check decoders.
+  EXPECT_EQ(config.decoders.size(), parsed_config.decoders.size());
+  for (size_t i = 0; i < parsed_config.decoders.size(); i++) {
+    EXPECT_EQ(config.decoders[i].payload_name,
+              parsed_config.decoders[i].payload_name);
+    EXPECT_EQ(config.decoders[i].payload_type,
+              parsed_config.decoders[i].payload_type);
+  }
+}
+
+void RtcEventLogTestHelper::VerifySendStreamConfig(
+    const ParsedRtcEventLog& parsed_log,
+    size_t index,
+    const VideoSendStream::Config& config) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
+  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
+  // Check SSRCs.
+  ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()),
+            sender_config.ssrcs_size());
+  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
+    EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
+  }
+  // Check header extensions.
+  ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
+            sender_config.header_extensions_size());
+  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
+    ASSERT_TRUE(sender_config.header_extensions(i).has_name());
+    ASSERT_TRUE(sender_config.header_extensions(i).has_id());
+    const std::string& name = sender_config.header_extensions(i).name();
+    int id = sender_config.header_extensions(i).id();
+    EXPECT_EQ(config.rtp.extensions[i].id, id);
+    EXPECT_EQ(config.rtp.extensions[i].name, name);
+  }
+  // Check RTX settings.
+  ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
+            sender_config.rtx_ssrcs_size());
+  for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
+    EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
+  }
+  if (sender_config.rtx_ssrcs_size() > 0) {
+    ASSERT_TRUE(sender_config.has_rtx_payload_type());
+    EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
+  }
+  // Check encoder.
+  ASSERT_TRUE(sender_config.has_encoder());
+  ASSERT_TRUE(sender_config.encoder().has_name());
+  ASSERT_TRUE(sender_config.encoder().has_payload_type());
+  EXPECT_EQ(config.encoder_settings.payload_name,
+            sender_config.encoder().name());
+  EXPECT_EQ(config.encoder_settings.payload_type,
+            sender_config.encoder().payload_type());
+
+  // Check consistency of the parser.
+  VideoSendStream::Config parsed_config(nullptr);
+  parsed_log.GetVideoSendConfig(index, &parsed_config);
+  // Check SSRCs
+  EXPECT_EQ(config.rtp.ssrcs.size(), parsed_config.rtp.ssrcs.size());
+  for (size_t i = 0; i < config.rtp.ssrcs.size(); i++) {
+    EXPECT_EQ(config.rtp.ssrcs[i], parsed_config.rtp.ssrcs[i]);
+  }
+  // Check header extensions.
+  EXPECT_EQ(config.rtp.extensions.size(), parsed_config.rtp.extensions.size());
+  for (size_t i = 0; i < parsed_config.rtp.extensions.size(); i++) {
+    EXPECT_EQ(config.rtp.extensions[i].name,
+              parsed_config.rtp.extensions[i].name);
+    EXPECT_EQ(config.rtp.extensions[i].id, parsed_config.rtp.extensions[i].id);
+  }
+  // Check RTX settings.
+  EXPECT_EQ(config.rtp.rtx.ssrcs.size(), parsed_config.rtp.rtx.ssrcs.size());
+  for (size_t i = 0; i < config.rtp.rtx.ssrcs.size(); i++) {
+    EXPECT_EQ(config.rtp.rtx.ssrcs[i], parsed_config.rtp.rtx.ssrcs[i]);
+  }
+  EXPECT_EQ(config.rtp.rtx.payload_type, parsed_config.rtp.rtx.payload_type);
+  // Check encoder.
+  EXPECT_EQ(config.encoder_settings.payload_name,
+            parsed_config.encoder_settings.payload_name);
+  EXPECT_EQ(config.encoder_settings.payload_type,
+            parsed_config.encoder_settings.payload_type);
+}
+
+void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
+                                           size_t index,
+                                           PacketDirection direction,
+                                           MediaType media_type,
+                                           const uint8_t* header,
+                                           size_t header_size,
+                                           size_t total_size) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
+  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+  ASSERT_TRUE(rtp_packet.has_incoming());
+  EXPECT_EQ(direction == kIncomingPacket, rtp_packet.incoming());
+  ASSERT_TRUE(rtp_packet.has_type());
+  EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
+  ASSERT_TRUE(rtp_packet.has_packet_length());
+  EXPECT_EQ(total_size, rtp_packet.packet_length());
+  ASSERT_TRUE(rtp_packet.has_header());
+  ASSERT_EQ(header_size, rtp_packet.header().size());
+  for (size_t i = 0; i < header_size; i++) {
+    EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
+  }
+
+  // Check consistency of the parser.
+  PacketDirection parsed_direction;
+  MediaType parsed_media_type;
+  uint8_t parsed_header[1500];
+  size_t parsed_header_size, parsed_total_size;
+  parsed_log.GetRtpHeader(index, &parsed_direction, &parsed_media_type,
+                          parsed_header, &parsed_header_size,
+                          &parsed_total_size);
+  EXPECT_EQ(direction, parsed_direction);
+  EXPECT_EQ(media_type, parsed_media_type);
+  ASSERT_EQ(header_size, parsed_header_size);
+  EXPECT_EQ(0, std::memcmp(header, parsed_header, header_size));
+  EXPECT_EQ(total_size, parsed_total_size);
+}
+
+void RtcEventLogTestHelper::VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
+                                            size_t index,
+                                            PacketDirection direction,
+                                            MediaType media_type,
+                                            const uint8_t* packet,
+                                            size_t total_size) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
+  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
+  ASSERT_TRUE(rtcp_packet.has_incoming());
+  EXPECT_EQ(direction == kIncomingPacket, rtcp_packet.incoming());
+  ASSERT_TRUE(rtcp_packet.has_type());
+  EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
+  ASSERT_TRUE(rtcp_packet.has_packet_data());
+  ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
+  for (size_t i = 0; i < total_size; i++) {
+    EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
+  }
+
+  // Check consistency of the parser.
+  PacketDirection parsed_direction;
+  MediaType parsed_media_type;
+  uint8_t parsed_packet[1500];
+  size_t parsed_total_size;
+  parsed_log.GetRtcpPacket(index, &parsed_direction, &parsed_media_type,
+                           parsed_packet, &parsed_total_size);
+  EXPECT_EQ(direction, parsed_direction);
+  EXPECT_EQ(media_type, parsed_media_type);
+  ASSERT_EQ(total_size, parsed_total_size);
+  EXPECT_EQ(0, std::memcmp(packet, parsed_packet, total_size));
+}
+
+void RtcEventLogTestHelper::VerifyPlayoutEvent(
+    const ParsedRtcEventLog& parsed_log,
+    size_t index,
+    uint32_t ssrc) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
+  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
+  ASSERT_TRUE(playout_event.has_local_ssrc());
+  EXPECT_EQ(ssrc, playout_event.local_ssrc());
+
+  // Check consistency of the parser.
+  uint32_t parsed_ssrc;
+  parsed_log.GetAudioPlayout(index, &parsed_ssrc);
+  EXPECT_EQ(ssrc, parsed_ssrc);
+}
+
+void RtcEventLogTestHelper::VerifyBweLossEvent(
+    const ParsedRtcEventLog& parsed_log,
+    size_t index,
+    int32_t bitrate,
+    uint8_t fraction_loss,
+    int32_t total_packets) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
+  const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
+  ASSERT_TRUE(bwe_event.has_bitrate());
+  EXPECT_EQ(bitrate, bwe_event.bitrate());
+  ASSERT_TRUE(bwe_event.has_fraction_loss());
+  EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
+  ASSERT_TRUE(bwe_event.has_total_packets());
+  EXPECT_EQ(total_packets, bwe_event.total_packets());
+
+  // Check consistency of the parser.
+  int32_t parsed_bitrate;
+  uint8_t parsed_fraction_loss;
+  int32_t parsed_total_packets;
+  parsed_log.GetBwePacketLossEvent(
+      index, &parsed_bitrate, &parsed_fraction_loss, &parsed_total_packets);
+  EXPECT_EQ(bitrate, parsed_bitrate);
+  EXPECT_EQ(fraction_loss, parsed_fraction_loss);
+  EXPECT_EQ(total_packets, parsed_total_packets);
+}
+
+void RtcEventLogTestHelper::VerifyLogStartEvent(
+    const ParsedRtcEventLog& parsed_log,
+    size_t index) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  EXPECT_EQ(rtclog::Event::LOG_START, event.type());
+}
+
+void RtcEventLogTestHelper::VerifyLogEndEvent(
+    const ParsedRtcEventLog& parsed_log,
+    size_t index) {
+  const rtclog::Event& event = parsed_log.stream_[index];
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  EXPECT_EQ(rtclog::Event::LOG_END, event.type());
+}
+
+}  // namespace webrtc
+
+#endif  // ENABLE_RTC_EVENT_LOG
diff --git a/chromium/third_party/webrtc/call/rtc_event_log_unittest_helper.h b/chromium/third_party/webrtc/call/rtc_event_log_unittest_helper.h
new file mode 100644
index 00000000000..b662c3ccc36
--- /dev/null
+++ b/chromium/third_party/webrtc/call/rtc_event_log_unittest_helper.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
+#define WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
+
+#include "webrtc/call.h"
+#include "webrtc/call/rtc_event_log_parser.h"
+
+namespace webrtc {
+
+class RtcEventLogTestHelper {
+ public:
+  static void VerifyReceiveStreamConfig(
+      const ParsedRtcEventLog& parsed_log,
+      size_t index,
+      const VideoReceiveStream::Config& config);
+  static void VerifySendStreamConfig(const ParsedRtcEventLog& parsed_log,
+                                     size_t index,
+                                     const VideoSendStream::Config& config);
+  static void VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
+                             size_t index,
+                             PacketDirection direction,
+                             MediaType media_type,
+                             const uint8_t* header,
+                             size_t header_size,
+                             size_t total_size);
+  static void VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
+                              size_t index,
+                              PacketDirection direction,
+                              MediaType media_type,
+                              const uint8_t* packet,
+                              size_t total_size);
+  static void VerifyPlayoutEvent(const ParsedRtcEventLog& parsed_log,
+                                 size_t index,
+                                 uint32_t ssrc);
+  static void VerifyBweLossEvent(const ParsedRtcEventLog& parsed_log,
+                                 size_t index,
+                                 int32_t bitrate,
+                                 uint8_t fraction_loss,
+                                 int32_t total_packets);
+
+  static void VerifyLogStartEvent(const ParsedRtcEventLog& parsed_log,
+                                  size_t index);
+  static void VerifyLogEndEvent(const ParsedRtcEventLog& parsed_log,
+                                size_t index);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
diff --git a/chromium/third_party/webrtc/common.h b/chromium/third_party/webrtc/common.h
index f2a868241d2..3aeea814c6b 100644
--- a/chromium/third_party/webrtc/common.h
+++ b/chromium/third_party/webrtc/common.h
@@ -14,6 +14,7 @@
 #include <map>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 
 namespace webrtc {
 
diff --git a/chromium/third_party/webrtc/common_audio/audio_ring_buffer.h b/chromium/third_party/webrtc/common_audio/audio_ring_buffer.h
index ae825a3cd0b..6bf3a195314 100644
--- a/chromium/third_party/webrtc/common_audio/audio_ring_buffer.h
+++ b/chromium/third_party/webrtc/common_audio/audio_ring_buffer.h
@@ -11,6 +11,8 @@
 #define WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
 
 #include <stddef.h>
+
+#include <memory>
 #include <vector>
 
 struct RingBuffer;
diff --git a/chromium/third_party/webrtc/common_audio/common_audio.gyp b/chromium/third_party/webrtc/common_audio/common_audio.gyp
index 57d9f1ca645..30ba322dfe8 100644
--- a/chromium/third_party/webrtc/common_audio/common_audio.gyp
+++ b/chromium/third_party/webrtc/common_audio/common_audio.gyp
@@ -294,7 +294,28 @@
               ],
             },
           ],
-        }],
+          'conditions': [
+            ['test_isolation_mode != "noop"',
+              {
+                'targets': [
+                  {
+                    'target_name': 'common_audio_unittests_apk_run',
+                    'type': 'none',
+                    'dependencies': [
+                      '<(apk_tests_path):common_audio_unittests_apk',
+                    ],
+                    'includes': [
+                      '../build/isolate.gypi',
+                    ],
+                    'sources': [
+                      'common_audio_unittests_apk.isolate',
+                    ],
+                  },
+                ],
+              },
+            ],
+          ],
+        }],  # OS=="android"
         ['test_isolation_mode != "noop"', {
           'targets': [
             {
diff --git a/chromium/third_party/webrtc/common_audio/common_audio_unittests_apk.isolate b/chromium/third_party/webrtc/common_audio/common_audio_unittests_apk.isolate
new file mode 100644
index 00000000000..d426732fe9e
--- /dev/null
+++ b/chromium/third_party/webrtc/common_audio/common_audio_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'common_audio_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_common_audio_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/common_audio_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_common_audio_unittests',
+      'common_audio_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/common_audio/fir_filter.cc b/chromium/third_party/webrtc/common_audio/fir_filter.cc
index 13a2237dae1..ed1f1bbfb81 100644
--- a/chromium/third_party/webrtc/common_audio/fir_filter.cc
+++ b/chromium/third_party/webrtc/common_audio/fir_filter.cc
@@ -61,13 +61,6 @@ FIRFilter* FIRFilter::Create(const float* coefficients,
 #elif defined(WEBRTC_HAS_NEON)
   filter =
       new FIRFilterNEON(coefficients, coefficients_length, max_input_length);
-#elif defined(WEBRTC_DETECT_NEON)
-  if (WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) {
-    filter =
-        new FIRFilterNEON(coefficients, coefficients_length, max_input_length);
-  } else {
-    filter = new FIRFilterC(coefficients, coefficients_length);
-  }
 #else
   filter = new FIRFilterC(coefficients, coefficients_length);
 #endif
diff --git a/chromium/third_party/webrtc/common_audio/lapped_transform.cc b/chromium/third_party/webrtc/common_audio/lapped_transform.cc
index 0edf586d783..5ab1db1b258 100644
--- a/chromium/third_party/webrtc/common_audio/lapped_transform.cc
+++ b/chromium/third_party/webrtc/common_audio/lapped_transform.cc
@@ -72,8 +72,7 @@ LappedTransform::LappedTransform(size_t num_in_channels,
                window,
                shift_amount,
                &blocker_callback_),
-      fft_(rtc::ScopedToUnique(
-          RealFourier::Create(RealFourier::FftOrder(block_length_)))),
+      fft_(RealFourier::Create(RealFourier::FftOrder(block_length_))),
       cplx_length_(RealFourier::ComplexLength(fft_->order())),
       real_buf_(num_in_channels,
                 block_length_,
diff --git a/chromium/third_party/webrtc/common_audio/real_fourier.cc b/chromium/third_party/webrtc/common_audio/real_fourier.cc
index 55ec49cba2c..67f942d560a 100644
--- a/chromium/third_party/webrtc/common_audio/real_fourier.cc
+++ b/chromium/third_party/webrtc/common_audio/real_fourier.cc
@@ -21,11 +21,11 @@ using std::complex;
 
 const size_t RealFourier::kFftBufferAlignment = 32;
 
-rtc::scoped_ptr<RealFourier> RealFourier::Create(int fft_order) {
+std::unique_ptr<RealFourier> RealFourier::Create(int fft_order) {
 #if defined(RTC_USE_OPENMAX_DL)
-  return rtc::scoped_ptr<RealFourier>(new RealFourierOpenmax(fft_order));
+  return std::unique_ptr<RealFourier>(new RealFourierOpenmax(fft_order));
 #else
-  return rtc::scoped_ptr<RealFourier>(new RealFourierOoura(fft_order));
+  return std::unique_ptr<RealFourier>(new RealFourierOoura(fft_order));
 #endif
 }
 
diff --git a/chromium/third_party/webrtc/common_audio/real_fourier.h b/chromium/third_party/webrtc/common_audio/real_fourier.h
index 0be56a58b0e..5e83e37f70e 100644
--- a/chromium/third_party/webrtc/common_audio/real_fourier.h
+++ b/chromium/third_party/webrtc/common_audio/real_fourier.h
@@ -12,8 +12,8 @@
 #define WEBRTC_COMMON_AUDIO_REAL_FOURIER_H_
 
 #include <complex>
+#include <memory>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 // Uniform interface class for the real DFT and its inverse, for power-of-2
@@ -25,8 +25,8 @@ namespace webrtc {
 class RealFourier {
  public:
   // Shorthand typenames for the scopers used by the buffer allocation helpers.
-  typedef rtc::scoped_ptr<float[], AlignedFreeDeleter> fft_real_scoper;
-  typedef rtc::scoped_ptr<std::complex<float>[], AlignedFreeDeleter>
+  typedef std::unique_ptr<float[], AlignedFreeDeleter> fft_real_scoper;
+  typedef std::unique_ptr<std::complex<float>[], AlignedFreeDeleter>
       fft_cplx_scoper;
 
   // The alignment required for all input and output buffers, in bytes.
@@ -34,7 +34,7 @@ class RealFourier {
 
   // Construct a wrapper instance for the given input order, which must be
   // between 1 and kMaxFftOrder, inclusively.
-  static rtc::scoped_ptr<RealFourier> Create(int fft_order);
+  static std::unique_ptr<RealFourier> Create(int fft_order);
   virtual ~RealFourier() {};
 
   // Helper to compute the smallest FFT order (a power of 2) which will contain
diff --git a/chromium/third_party/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc b/chromium/third_party/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
index afb0963c3fd..aca73b2d4bc 100644
--- a/chromium/third_party/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
+++ b/chromium/third_party/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
@@ -14,10 +14,10 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -86,16 +86,17 @@ void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
   const double io_ratio = input_rate_ / static_cast<double>(output_rate_);
   SincResampler sinc_resampler(io_ratio, SincResampler::kDefaultRequestSize,
                                &resampler_source);
-  TickTime start = TickTime::Now();
+  int64_t start = rtc::TimeNanos();
   for (int i = 0; i < kResampleIterations; ++i) {
     sinc_resampler.Resample(output_samples, resampled_destination.get());
   }
-  double total_time_sinc_us = (TickTime::Now() - start).Microseconds();
+  double total_time_sinc_us =
+      (rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
   printf("SincResampler took %.2f us per frame.\n",
          total_time_sinc_us / kResampleIterations);
 
   PushSincResampler resampler(input_samples, output_samples);
-  start = TickTime::Now();
+  start = rtc::TimeNanos();
   if (int_format) {
     for (int i = 0; i < kResampleIterations; ++i) {
       EXPECT_EQ(output_samples,
@@ -113,7 +114,8 @@ void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
                                    output_samples));
     }
   }
-  double total_time_us = (TickTime::Now() - start).Microseconds();
+  double total_time_us =
+      (rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
   printf("PushSincResampler took %.2f us per frame; which is a %.1f%% overhead "
          "on SincResampler.\n\n", total_time_us / kResampleIterations,
          (total_time_us - total_time_sinc_us) / total_time_sinc_us * 100);
diff --git a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.cc b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.cc
index 69ac2208cfb..c8bc15a362f 100644
--- a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.cc
+++ b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.cc
@@ -136,12 +136,6 @@ void SincResampler::InitializeCPUSpecificFeatures() {
 #elif defined(WEBRTC_HAS_NEON)
 #define CONVOLVE_FUNC Convolve_NEON
 void SincResampler::InitializeCPUSpecificFeatures() {}
-#elif defined(WEBRTC_DETECT_NEON)
-#define CONVOLVE_FUNC convolve_proc_
-void SincResampler::InitializeCPUSpecificFeatures() {
-  convolve_proc_ = WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON ?
-      Convolve_NEON : Convolve_C;
-}
 #else
 // Unknown architecture.
 #define CONVOLVE_FUNC Convolve_C
diff --git a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.h b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.h
index d8ea6df72f7..ecfd64c0a64 100644
--- a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.h
+++ b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.h
@@ -107,7 +107,7 @@ class SincResampler {
   static float Convolve_SSE(const float* input_ptr, const float* k1,
                             const float* k2,
                             double kernel_interpolation_factor);
-#elif defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
+#elif defined(WEBRTC_HAS_NEON)
   static float Convolve_NEON(const float* input_ptr, const float* k1,
                              const float* k2,
                              double kernel_interpolation_factor);
diff --git a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
index 42172ebdda7..d3b0dcd428a 100644
--- a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
+++ b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
@@ -20,11 +20,11 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/resampler/sinc_resampler.h"
 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/system_wrappers/include/stringize_macros.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/test_suite.h"
 
 using testing::_;
@@ -107,10 +107,11 @@ TEST(SincResamplerTest, DISABLED_SetRatioBench) {
   SincResampler resampler(kSampleRateRatio, SincResampler::kDefaultRequestSize,
                           &mock_source);
 
-  TickTime start = TickTime::Now();
+  int64_t start = rtc::TimeNanos();
   for (int i = 1; i < 10000; ++i)
     resampler.SetRatio(1.0 / i);
-  double total_time_c_us = (TickTime::Now() - start).Microseconds();
+  double total_time_c_us =
+      (rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
   printf("SetRatio() took %.2fms.\n", total_time_c_us / 1000);
 }
 
@@ -179,13 +180,14 @@ TEST(SincResamplerTest, ConvolveBenchmark) {
   printf("Benchmarking %d iterations:\n", kConvolveIterations);
 
   // Benchmark Convolve_C().
-  TickTime start = TickTime::Now();
+  int64_t start = rtc::TimeNanos();
   for (int i = 0; i < kConvolveIterations; ++i) {
     resampler.Convolve_C(
         resampler.kernel_storage_.get(), resampler.kernel_storage_.get(),
         resampler.kernel_storage_.get(), kKernelInterpolationFactor);
   }
-  double total_time_c_us = (TickTime::Now() - start).Microseconds();
+  double total_time_c_us =
+      (rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
   printf("Convolve_C took %.2fms.\n", total_time_c_us / 1000);
 
 #if defined(CONVOLVE_FUNC)
@@ -196,27 +198,27 @@ TEST(SincResamplerTest, ConvolveBenchmark) {
 #endif
 
   // Benchmark with unaligned input pointer.
-  start = TickTime::Now();
+  start = rtc::TimeNanos();
   for (int j = 0; j < kConvolveIterations; ++j) {
     resampler.CONVOLVE_FUNC(
         resampler.kernel_storage_.get() + 1, resampler.kernel_storage_.get(),
         resampler.kernel_storage_.get(), kKernelInterpolationFactor);
   }
   double total_time_optimized_unaligned_us =
-      (TickTime::Now() - start).Microseconds();
+      (rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
   printf(STRINGIZE(CONVOLVE_FUNC) "(unaligned) took %.2fms; which is %.2fx "
          "faster than Convolve_C.\n", total_time_optimized_unaligned_us / 1000,
          total_time_c_us / total_time_optimized_unaligned_us);
 
   // Benchmark with aligned input pointer.
-  start = TickTime::Now();
+  start = rtc::TimeNanos();
   for (int j = 0; j < kConvolveIterations; ++j) {
     resampler.CONVOLVE_FUNC(
         resampler.kernel_storage_.get(), resampler.kernel_storage_.get(),
         resampler.kernel_storage_.get(), kKernelInterpolationFactor);
   }
   double total_time_optimized_aligned_us =
-      (TickTime::Now() - start).Microseconds();
+      (rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
   printf(STRINGIZE(CONVOLVE_FUNC) " (aligned) took %.2fms; which is %.2fx "
          "faster than Convolve_C and %.2fx faster than "
          STRINGIZE(CONVOLVE_FUNC) " (unaligned).\n",
diff --git a/chromium/third_party/webrtc/common_audio/ring_buffer.c b/chromium/third_party/webrtc/common_audio/ring_buffer.c
index 60fb5dff20d..5fc653bd57e 100644
--- a/chromium/third_party/webrtc/common_audio/ring_buffer.c
+++ b/chromium/third_party/webrtc/common_audio/ring_buffer.c
@@ -17,20 +17,6 @@
 #include <stdlib.h>
 #include <string.h>
 
-enum Wrap {
-  SAME_WRAP,
-  DIFF_WRAP
-};
-
-struct RingBuffer {
-  size_t read_pos;
-  size_t write_pos;
-  size_t element_count;
-  size_t element_size;
-  enum Wrap rw_wrap;
-  char* data;
-};
-
 // Get address of region(s) from which we can read data.
 // If the region is contiguous, |data_ptr_bytes_2| will be zero.
 // If non-contiguous, |data_ptr_bytes_2| will be the size in bytes of the second
diff --git a/chromium/third_party/webrtc/common_audio/ring_buffer.h b/chromium/third_party/webrtc/common_audio/ring_buffer.h
index 4125c48d011..74951a8b2da 100644
--- a/chromium/third_party/webrtc/common_audio/ring_buffer.h
+++ b/chromium/third_party/webrtc/common_audio/ring_buffer.h
@@ -20,7 +20,16 @@ extern "C" {
 
 #include <stddef.h>  // size_t
 
-typedef struct RingBuffer RingBuffer;
+enum Wrap { SAME_WRAP, DIFF_WRAP };
+
+typedef struct RingBuffer {
+  size_t read_pos;
+  size_t write_pos;
+  size_t element_count;
+  size_t element_size;
+  enum Wrap rw_wrap;
+  char* data;
+} RingBuffer;
 
 // Creates and initializes the buffer. Returns NULL on failure.
 RingBuffer* WebRtc_CreateBuffer(size_t element_count, size_t element_size);
diff --git a/chromium/third_party/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/chromium/third_party/webrtc/common_audio/signal_processing/include/signal_processing_library.h
index 2e96883e6de..19379bf084f 100644
--- a/chromium/third_party/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/chromium/third_party/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -106,10 +106,8 @@ extern "C" {
 
 // Initialize SPL. Currently it contains only function pointer initialization.
 // If the underlying platform is known to be ARM-Neon (WEBRTC_HAS_NEON defined),
-// the pointers will be assigned to code optimized for Neon; otherwise
-// if run-time Neon detection (WEBRTC_DETECT_NEON) is enabled, the pointers
-// will be assigned to either Neon code or generic C code; otherwise, generic C
-// code will be assigned.
+// the pointers will be assigned to code optimized for Neon; otherwise, generic
+// C code will be assigned.
 // Note that this function MUST be called in any application that uses SPL
 // functions.
 void WebRtcSpl_Init();
@@ -153,7 +151,7 @@ void WebRtcSpl_ZerosArrayW32(int32_t* vector,
 typedef int16_t (*MaxAbsValueW16)(const int16_t* vector, size_t length);
 extern MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
 int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, size_t length);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
@@ -170,7 +168,7 @@ int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, size_t length);
 typedef int32_t (*MaxAbsValueW32)(const int32_t* vector, size_t length);
 extern MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32;
 int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, size_t length);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, size_t length);
 #endif
 #if defined(MIPS_DSP_R1_LE)
@@ -187,7 +185,7 @@ int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, size_t length);
 typedef int16_t (*MaxValueW16)(const int16_t* vector, size_t length);
 extern MaxValueW16 WebRtcSpl_MaxValueW16;
 int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
@@ -204,7 +202,7 @@ int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, size_t length);
 typedef int32_t (*MaxValueW32)(const int32_t* vector, size_t length);
 extern MaxValueW32 WebRtcSpl_MaxValueW32;
 int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
@@ -221,7 +219,7 @@ int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, size_t length);
 typedef int16_t (*MinValueW16)(const int16_t* vector, size_t length);
 extern MinValueW16 WebRtcSpl_MinValueW16;
 int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
@@ -238,7 +236,7 @@ int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, size_t length);
 typedef int32_t (*MinValueW32)(const int32_t* vector, size_t length);
 extern MinValueW32 WebRtcSpl_MinValueW32;
 int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, size_t length);
 #endif
 #if defined(MIPS32_LE)
@@ -531,7 +529,7 @@ void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
                                  size_t dim_cross_correlation,
                                  int right_shifts,
                                  int step_seq2);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 void WebRtcSpl_CrossCorrelationNeon(int32_t* cross_correlation,
                                     const int16_t* seq1,
                                     const int16_t* seq2,
@@ -698,7 +696,7 @@ int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
                               size_t coefficients_length,
                               int factor,
                               size_t delay);
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int WebRtcSpl_DownsampleFastNeon(const int16_t* data_in,
                                  size_t data_in_length,
                                  int16_t* data_out,
diff --git a/chromium/third_party/webrtc/common_audio/signal_processing/spl_init.c b/chromium/third_party/webrtc/common_audio/signal_processing/spl_init.c
index fdab0383997..c9c4e659cf3 100644
--- a/chromium/third_party/webrtc/common_audio/signal_processing/spl_init.c
+++ b/chromium/third_party/webrtc/common_audio/signal_processing/spl_init.c
@@ -28,8 +28,7 @@ CrossCorrelation WebRtcSpl_CrossCorrelation;
 DownsampleFast WebRtcSpl_DownsampleFast;
 ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
 
-#if (defined(WEBRTC_DETECT_NEON) || !defined(WEBRTC_HAS_NEON)) && \
-    !defined(MIPS32_LE)
+#if (!defined(WEBRTC_HAS_NEON)) && !defined(MIPS32_LE)
 /* Initialize function pointers to the generic C version. */
 static void InitPointersToC() {
   WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16C;
@@ -45,7 +44,7 @@ static void InitPointersToC() {
 }
 #endif
 
-#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 /* Initialize function pointers to the Neon version. */
 static void InitPointersToNeon() {
   WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16Neon;
@@ -84,19 +83,13 @@ static void InitPointersToMIPS() {
 #endif
 
 static void InitFunctionPointers(void) {
-#if defined(WEBRTC_DETECT_NEON)
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    InitPointersToNeon();
-  } else {
-    InitPointersToC();
-  }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
   InitPointersToNeon();
 #elif defined(MIPS32_LE)
   InitPointersToMIPS();
 #else
   InitPointersToC();
-#endif  /* WEBRTC_DETECT_NEON */
+#endif  /* WEBRTC_HAS_NEON */
 }
 
 #if defined(WEBRTC_POSIX)
diff --git a/chromium/third_party/webrtc/common_audio/vad/vad.cc b/chromium/third_party/webrtc/common_audio/vad/vad.cc
index 99d6ffeee66..77de5166db3 100644
--- a/chromium/third_party/webrtc/common_audio/vad/vad.cc
+++ b/chromium/third_party/webrtc/common_audio/vad/vad.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/common_audio/vad/include/vad.h"
 
+#include <memory>
+
 #include "webrtc/base/checks.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/common_types.h b/chromium/third_party/webrtc/common_types.h
index e171e1bd8af..587b90d753c 100644
--- a/chromium/third_party/webrtc/common_types.h
+++ b/chromium/third_party/webrtc/common_types.h
@@ -296,6 +296,18 @@ class SendSideDelayObserver {
                                     uint32_t ssrc) = 0;
 };
 
+// Callback, used to notify an observer whenever a packet is sent to the
+// transport.
+// TODO(asapersson): This class will remove the need for SendSideDelayObserver.
+// Remove SendSideDelayObserver once possible.
+class SendPacketObserver {
+ public:
+  virtual ~SendPacketObserver() {}
+  virtual void OnSendPacket(uint16_t packet_id,
+                            int64_t capture_time_ms,
+                            uint32_t ssrc) = 0;
+};
+
 // ==================================================================
 // Voice specific types
 // ==================================================================
@@ -880,6 +892,11 @@ class StreamDataCountersCallback {
 // RTCP mode is described by RFC 5506.
 enum class RtcpMode { kOff, kCompound, kReducedSize };
 
+enum NetworkState {
+  kNetworkUp,
+  kNetworkDown,
+};
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_
diff --git a/chromium/third_party/webrtc/common_video/BUILD.gn b/chromium/third_party/webrtc/common_video/BUILD.gn
index ed877bd344a..7474974c528 100644
--- a/chromium/third_party/webrtc/common_video/BUILD.gn
+++ b/chromium/third_party/webrtc/common_video/BUILD.gn
@@ -17,7 +17,10 @@ config("common_video_config") {
 
 source_set("common_video") {
   sources = [
+    "bitrate_adjuster.cc",
     "i420_buffer_pool.cc",
+    "include/bitrate_adjuster.h",
+    "include/frame_callback.h",
     "include/i420_buffer_pool.h",
     "include/incoming_video_stream.h",
     "include/video_frame_buffer.h",
diff --git a/chromium/third_party/webrtc/common_video/DEPS b/chromium/third_party/webrtc/common_video/DEPS
index 28059580703..ad03e661693 100644
--- a/chromium/third_party/webrtc/common_video/DEPS
+++ b/chromium/third_party/webrtc/common_video/DEPS
@@ -1,4 +1,5 @@
 include_rules = [
   "+webrtc/base",
+  "+webrtc/media/base",
   "+webrtc/system_wrappers",
 ]
diff --git a/chromium/third_party/webrtc/modules/video_coding/bitrate_adjuster.cc b/chromium/third_party/webrtc/common_video/bitrate_adjuster.cc
index b6828ee6e1e..ada6c5db4b7 100644
--- a/chromium/third_party/webrtc/modules/video_coding/bitrate_adjuster.cc
+++ b/chromium/third_party/webrtc/common_video/bitrate_adjuster.cc
@@ -8,8 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/video_coding/include/bitrate_adjuster.h"
+#include "webrtc/common_video/include/bitrate_adjuster.h"
 
+#include <algorithm>
 #include <cmath>
 
 #include "webrtc/base/checks.h"
diff --git a/chromium/third_party/webrtc/modules/video_coding/bitrate_adjuster_unittest.cc b/chromium/third_party/webrtc/common_video/bitrate_adjuster_unittest.cc
index 1d14ee31606..23b278731ef 100644
--- a/chromium/third_party/webrtc/modules/video_coding/bitrate_adjuster_unittest.cc
+++ b/chromium/third_party/webrtc/common_video/bitrate_adjuster_unittest.cc
@@ -10,7 +10,7 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/modules/video_coding/include/bitrate_adjuster.h"
+#include "webrtc/common_video/include/bitrate_adjuster.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -86,7 +86,7 @@ TEST_F(BitrateAdjusterTest, VaryingBitrates) {
   SimulateBitrateBps(actual_bitrate_bps);
   VerifyAdjustment();
   adjusted_bitrate_bps = adjuster_.GetAdjustedBitrateBps();
-  EXPECT_LT(adjusted_bitrate_bps, last_adjusted_bitrate_bps);
+  EXPECT_LE(adjusted_bitrate_bps, last_adjusted_bitrate_bps);
   last_adjusted_bitrate_bps = adjusted_bitrate_bps;
   // After two cycles we should've stabilized and hit the lower bound.
   EXPECT_EQ(GetTargetBitrateBpsPct(kMinAdjustedBitratePct),
diff --git a/chromium/third_party/webrtc/common_video/common_video.gyp b/chromium/third_party/webrtc/common_video/common_video.gyp
index b392bd2aa4d..47ba3e59d71 100644
--- a/chromium/third_party/webrtc/common_video/common_video.gyp
+++ b/chromium/third_party/webrtc/common_video/common_video.gyp
@@ -52,9 +52,12 @@
         }],
       ],
       'sources': [
+        'bitrate_adjuster.cc',
         'i420_buffer_pool.cc',
         'video_frame.cc',
         'incoming_video_stream.cc',
+        'include/bitrate_adjuster.h',
+        'include/frame_callback.h',
         'include/i420_buffer_pool.h',
         'include/incoming_video_stream.h',
         'include/video_frame_buffer.h',
diff --git a/chromium/third_party/webrtc/common_video/common_video_unittests.gyp b/chromium/third_party/webrtc/common_video/common_video_unittests.gyp
index 545bfa46fce..8372c0b4a97 100644
--- a/chromium/third_party/webrtc/common_video/common_video_unittests.gyp
+++ b/chromium/third_party/webrtc/common_video/common_video_unittests.gyp
@@ -20,6 +20,7 @@
          '<(webrtc_root)/test/test.gyp:video_test_common',
       ],
       'sources': [
+        'bitrate_adjuster_unittest.cc',
         'i420_buffer_pool_unittest.cc',
         'i420_video_frame_unittest.cc',
         'libyuv/libyuv_unittest.cc',
@@ -54,7 +55,28 @@
           ],
         },
       ],
-    }],
+      'conditions': [
+        ['test_isolation_mode != "noop"',
+          {
+            'targets': [
+              {
+                'target_name': 'common_video_unittests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):common_video_unittests_apk',
+                ],
+                'includes': [
+                  '../build/isolate.gypi',
+                ],
+                'sources': [
+                  'common_video_unittests_apk.isolate',
+                ],
+              },
+            ],
+          },
+        ],
+      ],
+    }],  # OS=="android"
     ['test_isolation_mode != "noop"', {
       'targets': [
         {
diff --git a/chromium/third_party/webrtc/common_video/common_video_unittests_apk.isolate b/chromium/third_party/webrtc/common_video/common_video_unittests_apk.isolate
new file mode 100644
index 00000000000..36514cdb61c
--- /dev/null
+++ b/chromium/third_party/webrtc/common_video/common_video_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'common_video_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_common_video_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/common_video_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_common_video_unittests',
+      'common_video_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/common_video/i420_buffer_pool.cc b/chromium/third_party/webrtc/common_video/i420_buffer_pool.cc
index 82a10797b35..8896260fc0e 100644
--- a/chromium/third_party/webrtc/common_video/i420_buffer_pool.cc
+++ b/chromium/third_party/webrtc/common_video/i420_buffer_pool.cc
@@ -26,18 +26,28 @@ class PooledI420Buffer : public webrtc::VideoFrameBuffer {
 
   int width() const override { return buffer_->width(); }
   int height() const override { return buffer_->height(); }
-  const uint8_t* data(webrtc::PlaneType type) const override {
-    return buffer_->data(type);
+  const uint8_t* DataY() const override { return buffer_->DataY(); }
+  const uint8_t* DataU() const override { return buffer_->DataU(); }
+  const uint8_t* DataV() const override { return buffer_->DataV(); }
+
+  bool IsMutable() override { return HasOneRef(); }
+  // Make the IsMutable() check here instead of in |buffer_|, because the pool
+  // also has a reference to |buffer_|.
+  uint8_t* MutableDataY() override {
+    RTC_DCHECK(IsMutable());
+    return const_cast<uint8_t*>(buffer_->DataY());
   }
-  uint8_t* MutableData(webrtc::PlaneType type) override {
-    // Make the HasOneRef() check here instead of in |buffer_|, because the pool
-    // also has a reference to |buffer_|.
-    RTC_DCHECK(HasOneRef());
-    return const_cast<uint8_t*>(buffer_->data(type));
+  uint8_t* MutableDataU() override {
+    RTC_DCHECK(IsMutable());
+    return const_cast<uint8_t*>(buffer_->DataU());
   }
-  int stride(webrtc::PlaneType type) const override {
-    return buffer_->stride(type);
+  uint8_t* MutableDataV() override {
+    RTC_DCHECK(IsMutable());
+    return const_cast<uint8_t*>(buffer_->DataV());
   }
+  int StrideY() const override { return buffer_->StrideY(); }
+  int StrideU() const override { return buffer_->StrideU(); }
+  int StrideV() const override { return buffer_->StrideV(); }
   void* native_handle() const override { return nullptr; }
 
   rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override {
@@ -80,7 +90,7 @@ rtc::scoped_refptr<VideoFrameBuffer> I420BufferPool::CreateBuffer(int width,
     // CreateBuffer that has not been released yet. If the ref count is 1
     // (HasOneRef), then the list we are looping over holds the only reference
     // and it's safe to reuse.
-    if (buffer->HasOneRef())
+    if (buffer->IsMutable())
       return new rtc::RefCountedObject<PooledI420Buffer>(buffer);
   }
   // Allocate new buffer.
diff --git a/chromium/third_party/webrtc/common_video/i420_buffer_pool_unittest.cc b/chromium/third_party/webrtc/common_video/i420_buffer_pool_unittest.cc
index b030ee774aa..2110066a928 100644
--- a/chromium/third_party/webrtc/common_video/i420_buffer_pool_unittest.cc
+++ b/chromium/third_party/webrtc/common_video/i420_buffer_pool_unittest.cc
@@ -21,16 +21,16 @@ TEST(TestI420BufferPool, SimpleFrameReuse) {
   EXPECT_EQ(16, buffer->width());
   EXPECT_EQ(16, buffer->height());
   // Extract non-refcounted pointers for testing.
-  const uint8_t* y_ptr = buffer->data(kYPlane);
-  const uint8_t* u_ptr = buffer->data(kUPlane);
-  const uint8_t* v_ptr = buffer->data(kVPlane);
+  const uint8_t* y_ptr = buffer->DataY();
+  const uint8_t* u_ptr = buffer->DataU();
+  const uint8_t* v_ptr = buffer->DataV();
   // Release buffer so that it is returned to the pool.
   buffer = nullptr;
   // Check that the memory is resued.
   buffer = pool.CreateBuffer(16, 16);
-  EXPECT_EQ(y_ptr, buffer->data(kYPlane));
-  EXPECT_EQ(u_ptr, buffer->data(kUPlane));
-  EXPECT_EQ(v_ptr, buffer->data(kVPlane));
+  EXPECT_EQ(y_ptr, buffer->DataY());
+  EXPECT_EQ(u_ptr, buffer->DataU());
+  EXPECT_EQ(v_ptr, buffer->DataV());
   EXPECT_EQ(16, buffer->width());
   EXPECT_EQ(16, buffer->height());
 }
@@ -39,23 +39,23 @@ TEST(TestI420BufferPool, FailToReuse) {
   I420BufferPool pool;
   rtc::scoped_refptr<VideoFrameBuffer> buffer = pool.CreateBuffer(16, 16);
   // Extract non-refcounted pointers for testing.
-  const uint8_t* u_ptr = buffer->data(kUPlane);
-  const uint8_t* v_ptr = buffer->data(kVPlane);
+  const uint8_t* u_ptr = buffer->DataU();
+  const uint8_t* v_ptr = buffer->DataV();
   // Release buffer so that it is returned to the pool.
   buffer = nullptr;
   // Check that the pool doesn't try to reuse buffers of incorrect size.
   buffer = pool.CreateBuffer(32, 16);
   EXPECT_EQ(32, buffer->width());
   EXPECT_EQ(16, buffer->height());
-  EXPECT_NE(u_ptr, buffer->data(kUPlane));
-  EXPECT_NE(v_ptr, buffer->data(kVPlane));
+  EXPECT_NE(u_ptr, buffer->DataU());
+  EXPECT_NE(v_ptr, buffer->DataV());
 }
 
 TEST(TestI420BufferPool, ExclusiveOwner) {
   // Check that created buffers are exclusive so that they can be written to.
   I420BufferPool pool;
   rtc::scoped_refptr<VideoFrameBuffer> buffer = pool.CreateBuffer(16, 16);
-  EXPECT_TRUE(buffer->HasOneRef());
+  EXPECT_TRUE(buffer->IsMutable());
 }
 
 TEST(TestI420BufferPool, FrameValidAfterPoolDestruction) {
@@ -64,11 +64,11 @@ TEST(TestI420BufferPool, FrameValidAfterPoolDestruction) {
     I420BufferPool pool;
     buffer = pool.CreateBuffer(16, 16);
   }
-  EXPECT_TRUE(buffer->HasOneRef());
+  EXPECT_TRUE(buffer->IsMutable());
   EXPECT_EQ(16, buffer->width());
   EXPECT_EQ(16, buffer->height());
   // Try to trigger use-after-free errors by writing to y-plane.
-  memset(buffer->MutableData(kYPlane), 0xA5, 16 * buffer->stride(kYPlane));
+  memset(buffer->MutableDataY(), 0xA5, 16 * buffer->StrideY());
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/common_video/i420_video_frame_unittest.cc b/chromium/third_party/webrtc/common_video/i420_video_frame_unittest.cc
index 9b2bdd762dc..c942e4a8009 100644
--- a/chromium/third_party/webrtc/common_video/i420_video_frame_unittest.cc
+++ b/chromium/third_party/webrtc/common_video/i420_video_frame_unittest.cc
@@ -51,9 +51,9 @@ TEST(TestVideoFrame, SizeAllocation) {
   VideoFrame frame;
   frame. CreateEmptyFrame(10, 10, 12, 14, 220);
   int height = frame.height();
-  int stride_y = frame.stride(kYPlane);
-  int stride_u = frame.stride(kUPlane);
-  int stride_v = frame.stride(kVPlane);
+  int stride_y = frame.video_frame_buffer()->StrideY();
+  int stride_u = frame.video_frame_buffer()->StrideU();
+  int stride_v = frame.video_frame_buffer()->StrideV();
   // Verify that allocated size was computed correctly.
   EXPECT_EQ(ExpectedSize(stride_y, height, kYPlane),
             frame.allocated_size(kYPlane));
@@ -101,9 +101,12 @@ TEST(TestVideoFrame, CopyFrame) {
   // Frame of larger dimensions.
   small_frame.CreateEmptyFrame(width, height,
                                stride_y, stride_u, stride_v);
-  memset(small_frame.buffer(kYPlane), 1, small_frame.allocated_size(kYPlane));
-  memset(small_frame.buffer(kUPlane), 2, small_frame.allocated_size(kUPlane));
-  memset(small_frame.buffer(kVPlane), 3, small_frame.allocated_size(kVPlane));
+  memset(small_frame.video_frame_buffer()->MutableDataY(), 1,
+         small_frame.allocated_size(kYPlane));
+  memset(small_frame.video_frame_buffer()->MutableDataU(), 2,
+         small_frame.allocated_size(kUPlane));
+  memset(small_frame.video_frame_buffer()->MutableDataV(), 3,
+         small_frame.allocated_size(kVPlane));
   big_frame.CopyFrame(small_frame);
   EXPECT_TRUE(test::FramesEqual(small_frame, big_frame));
 }
@@ -141,12 +144,12 @@ TEST(TestVideoFrame, ShallowCopy) {
   const VideoFrame* const_frame1_ptr = &frame1;
   const VideoFrame* const_frame2_ptr = &frame2;
 
-  EXPECT_TRUE(const_frame1_ptr->buffer(kYPlane) ==
-              const_frame2_ptr->buffer(kYPlane));
-  EXPECT_TRUE(const_frame1_ptr->buffer(kUPlane) ==
-              const_frame2_ptr->buffer(kUPlane));
-  EXPECT_TRUE(const_frame1_ptr->buffer(kVPlane) ==
-              const_frame2_ptr->buffer(kVPlane));
+  EXPECT_TRUE(const_frame1_ptr->video_frame_buffer()->DataY() ==
+              const_frame2_ptr->video_frame_buffer()->DataY());
+  EXPECT_TRUE(const_frame1_ptr->video_frame_buffer()->DataU() ==
+              const_frame2_ptr->video_frame_buffer()->DataU());
+  EXPECT_TRUE(const_frame1_ptr->video_frame_buffer()->DataV() ==
+              const_frame2_ptr->video_frame_buffer()->DataV());
 
   EXPECT_EQ(frame2.timestamp(), frame1.timestamp());
   EXPECT_EQ(frame2.ntp_time_ms(), frame1.ntp_time_ms());
@@ -164,21 +167,6 @@ TEST(TestVideoFrame, ShallowCopy) {
   EXPECT_NE(frame2.rotation(), frame1.rotation());
 }
 
-TEST(TestVideoFrame, Reset) {
-  VideoFrame frame;
-  frame.CreateEmptyFrame(5, 5, 5, 5, 5);
-  frame.set_ntp_time_ms(1);
-  frame.set_timestamp(2);
-  frame.set_render_time_ms(3);
-  ASSERT_TRUE(frame.video_frame_buffer() != NULL);
-
-  frame.Reset();
-  EXPECT_EQ(0u, frame.ntp_time_ms());
-  EXPECT_EQ(0u, frame.render_time_ms());
-  EXPECT_EQ(0u, frame.timestamp());
-  EXPECT_TRUE(frame.video_frame_buffer() == NULL);
-}
-
 TEST(TestVideoFrame, CopyBuffer) {
   VideoFrame frame1, frame2;
   int width = 15;
@@ -199,12 +187,12 @@ TEST(TestVideoFrame, CopyBuffer) {
                      width, height, stride_y, stride_uv, stride_uv,
                      kVideoRotation_0);
   // Expect exactly the same pixel data.
-  EXPECT_TRUE(
-      test::EqualPlane(buffer_y, frame2.buffer(kYPlane), stride_y, 15, 15));
-  EXPECT_TRUE(
-      test::EqualPlane(buffer_u, frame2.buffer(kUPlane), stride_uv, 8, 8));
-  EXPECT_TRUE(
-      test::EqualPlane(buffer_v, frame2.buffer(kVPlane), stride_uv, 8, 8));
+  EXPECT_TRUE(test::EqualPlane(buffer_y, frame2.video_frame_buffer()->DataY(),
+                               stride_y, 15, 15));
+  EXPECT_TRUE(test::EqualPlane(buffer_u, frame2.video_frame_buffer()->DataU(),
+                               stride_uv, 8, 8));
+  EXPECT_TRUE(test::EqualPlane(buffer_v, frame2.video_frame_buffer()->DataV(),
+                               stride_uv, 8, 8));
 
   // Compare size.
   EXPECT_LE(kSizeY, frame2.allocated_size(kYPlane));
@@ -215,27 +203,27 @@ TEST(TestVideoFrame, CopyBuffer) {
 TEST(TestVideoFrame, ReuseAllocation) {
   VideoFrame frame;
   frame.CreateEmptyFrame(640, 320, 640, 320, 320);
-  const uint8_t* y = frame.buffer(kYPlane);
-  const uint8_t* u = frame.buffer(kUPlane);
-  const uint8_t* v = frame.buffer(kVPlane);
+  const uint8_t* y = frame.video_frame_buffer()->DataY();
+  const uint8_t* u = frame.video_frame_buffer()->DataU();
+  const uint8_t* v = frame.video_frame_buffer()->DataV();
   frame.CreateEmptyFrame(640, 320, 640, 320, 320);
-  EXPECT_EQ(y, frame.buffer(kYPlane));
-  EXPECT_EQ(u, frame.buffer(kUPlane));
-  EXPECT_EQ(v, frame.buffer(kVPlane));
+  EXPECT_EQ(y, frame.video_frame_buffer()->DataY());
+  EXPECT_EQ(u, frame.video_frame_buffer()->DataU());
+  EXPECT_EQ(v, frame.video_frame_buffer()->DataV());
 }
 
 TEST(TestVideoFrame, FailToReuseAllocation) {
   VideoFrame frame1;
   frame1.CreateEmptyFrame(640, 320, 640, 320, 320);
-  const uint8_t* y = frame1.buffer(kYPlane);
-  const uint8_t* u = frame1.buffer(kUPlane);
-  const uint8_t* v = frame1.buffer(kVPlane);
+  const uint8_t* y = frame1.video_frame_buffer()->DataY();
+  const uint8_t* u = frame1.video_frame_buffer()->DataU();
+  const uint8_t* v = frame1.video_frame_buffer()->DataV();
   // Make a shallow copy of |frame1|.
   VideoFrame frame2(frame1.video_frame_buffer(), 0, 0, kVideoRotation_0);
   frame1.CreateEmptyFrame(640, 320, 640, 320, 320);
-  EXPECT_NE(y, frame1.buffer(kYPlane));
-  EXPECT_NE(u, frame1.buffer(kUPlane));
-  EXPECT_NE(v, frame1.buffer(kVPlane));
+  EXPECT_NE(y, frame1.video_frame_buffer()->DataY());
+  EXPECT_NE(u, frame1.video_frame_buffer()->DataU());
+  EXPECT_NE(v, frame1.video_frame_buffer()->DataV());
 }
 
 TEST(TestVideoFrame, TextureInitialValues) {
@@ -246,7 +234,8 @@ TEST(TestVideoFrame, TextureInitialValues) {
   EXPECT_EQ(480, frame.height());
   EXPECT_EQ(100u, frame.timestamp());
   EXPECT_EQ(10, frame.render_time_ms());
-  EXPECT_EQ(handle, frame.native_handle());
+  ASSERT_TRUE(frame.video_frame_buffer() != nullptr);
+  EXPECT_EQ(handle, frame.video_frame_buffer()->native_handle());
 
   frame.set_timestamp(200);
   EXPECT_EQ(200u, frame.timestamp());
@@ -257,9 +246,9 @@ TEST(TestVideoFrame, TextureInitialValues) {
 TEST(TestI420FrameBuffer, Copy) {
   rtc::scoped_refptr<I420Buffer> buf1(
       new rtc::RefCountedObject<I420Buffer>(20, 10));
-  memset(buf1->MutableData(kYPlane), 1, 200);
-  memset(buf1->MutableData(kUPlane), 2, 50);
-  memset(buf1->MutableData(kVPlane), 3, 50);
+  memset(buf1->MutableDataY(), 1, 200);
+  memset(buf1->MutableDataU(), 2, 50);
+  memset(buf1->MutableDataV(), 3, 50);
   rtc::scoped_refptr<I420Buffer> buf2 = I420Buffer::Copy(buf1);
   EXPECT_TRUE(test::FrameBufsEqual(buf1, buf2));
 }
diff --git a/chromium/third_party/webrtc/modules/video_coding/include/bitrate_adjuster.h b/chromium/third_party/webrtc/common_video/include/bitrate_adjuster.h
index ec58445ae27..1f2474f88b1 100644
--- a/chromium/third_party/webrtc/modules/video_coding/include/bitrate_adjuster.h
+++ b/chromium/third_party/webrtc/common_video/include/bitrate_adjuster.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_BITRATE_ADJUSTER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_BITRATE_ADJUSTER_H_
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
 
 #include <functional>
 
@@ -86,4 +86,4 @@ class BitrateAdjuster {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_BITRATE_ADJUSTER_H_
+#endif  // WEBRTC_COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
diff --git a/chromium/third_party/webrtc/frame_callback.h b/chromium/third_party/webrtc/common_video/include/frame_callback.h
index 2bae25036d9..81737c6b962 100644
--- a/chromium/third_party/webrtc/frame_callback.h
+++ b/chromium/third_party/webrtc/common_video/include/frame_callback.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_FRAME_CALLBACK_H_
-#define WEBRTC_FRAME_CALLBACK_H_
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_FRAME_CALLBACK_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_FRAME_CALLBACK_H_
 
 #include <stddef.h>
 #include <stdint.h>
@@ -52,4 +52,4 @@ class EncodedFrameObserver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_FRAME_CALLBACK_H_
+#endif  // WEBRTC_COMMON_VIDEO_INCLUDE_FRAME_CALLBACK_H_
diff --git a/chromium/third_party/webrtc/common_video/include/incoming_video_stream.h b/chromium/third_party/webrtc/common_video/include/incoming_video_stream.h
index b6ab917bc18..32e3184ce97 100644
--- a/chromium/third_party/webrtc/common_video/include/incoming_video_stream.h
+++ b/chromium/third_party/webrtc/common_video/include/incoming_video_stream.h
@@ -17,34 +17,25 @@
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_video/video_render_frames.h"
+#include "webrtc/media/base/videosinkinterface.h"
 
 namespace webrtc {
 class EventTimerWrapper;
 
-class VideoRenderCallback {
- public:
-  virtual int32_t RenderFrame(const uint32_t streamId,
-                              const VideoFrame& videoFrame) = 0;
-
- protected:
-  virtual ~VideoRenderCallback() {}
-};
 
-class IncomingVideoStream : public VideoRenderCallback {
+class IncomingVideoStream : public rtc::VideoSinkInterface<VideoFrame> {
  public:
-  IncomingVideoStream(uint32_t stream_id, bool disable_prerenderer_smoothing);
+  explicit IncomingVideoStream(bool disable_prerenderer_smoothing);
   ~IncomingVideoStream();
 
-  // Get callback to deliver frames to the module.
-  VideoRenderCallback* ModuleCallback();
-  virtual int32_t RenderFrame(const uint32_t stream_id,
-                              const VideoFrame& video_frame);
+  // Overrides VideoSinkInterface
+  void OnFrame(const VideoFrame& video_frame) override;
 
   // Set callback to the platform dependent code.
-  void SetRenderCallback(VideoRenderCallback* render_callback);
+  void SetRenderCallback(rtc::VideoSinkInterface<VideoFrame>* render_callback);
 
   // Callback for file recording, snapshot, ...
-  void SetExternalCallback(VideoRenderCallback* render_object);
+  void SetExternalCallback(rtc::VideoSinkInterface<VideoFrame>* render_object);
 
   // Start/Stop.
   int32_t Start();
@@ -54,14 +45,8 @@ class IncomingVideoStream : public VideoRenderCallback {
   int32_t Reset();
 
   // Properties.
-  uint32_t StreamId() const;
   uint32_t IncomingRate() const;
 
-  void SetStartImage(const VideoFrame& video_frame);
-
-  void SetTimeoutImage(const VideoFrame& video_frame,
-                       const uint32_t timeout);
-
   int32_t SetExpectedRenderDelay(int32_t delay_ms);
 
  protected:
@@ -75,7 +60,6 @@ class IncomingVideoStream : public VideoRenderCallback {
 
   void DeliverFrame(const VideoFrame& video_frame);
 
-  uint32_t const stream_id_;
   const bool disable_prerenderer_smoothing_;
   // Critsects in allowed to enter order.
   rtc::CriticalSection stream_critsect_;
@@ -88,19 +72,16 @@ class IncomingVideoStream : public VideoRenderCallback {
   std::unique_ptr<EventTimerWrapper> deliver_buffer_event_;
 
   bool running_ GUARDED_BY(stream_critsect_);
-  VideoRenderCallback* external_callback_ GUARDED_BY(thread_critsect_);
-  VideoRenderCallback* render_callback_ GUARDED_BY(thread_critsect_);
+  rtc::VideoSinkInterface<VideoFrame>* external_callback_
+      GUARDED_BY(thread_critsect_);
+  rtc::VideoSinkInterface<VideoFrame>* render_callback_
+      GUARDED_BY(thread_critsect_);
   const std::unique_ptr<VideoRenderFrames> render_buffers_
       GUARDED_BY(buffer_critsect_);
 
   uint32_t incoming_rate_ GUARDED_BY(stream_critsect_);
   int64_t last_rate_calculation_time_ms_ GUARDED_BY(stream_critsect_);
   uint16_t num_frames_since_last_calculation_ GUARDED_BY(stream_critsect_);
-  int64_t last_render_time_ms_ GUARDED_BY(thread_critsect_);
-  VideoFrame temp_frame_ GUARDED_BY(thread_critsect_);
-  VideoFrame start_image_ GUARDED_BY(thread_critsect_);
-  VideoFrame timeout_image_ GUARDED_BY(thread_critsect_);
-  uint32_t timeout_time_ GUARDED_BY(thread_critsect_);
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/common_video/include/video_frame_buffer.h b/chromium/third_party/webrtc/common_video/include/video_frame_buffer.h
index 9cf57a43591..6f082dee9ba 100644
--- a/chromium/third_party/webrtc/common_video/include/video_frame_buffer.h
+++ b/chromium/third_party/webrtc/common_video/include/video_frame_buffer.h
@@ -33,7 +33,16 @@ enum PlaneType {
 // not contain any frame metadata such as rotation, timestamp, pixel_width, etc.
 class VideoFrameBuffer : public rtc::RefCountInterface {
  public:
-  // Returns true if this buffer has a single exclusive owner.
+  // Returns true if the caller is exclusive owner, and allowed to
+  // call MutableData.
+
+  // TODO(nisse): Delete default implementation when subclasses in
+  // Chrome are updated.
+  virtual bool IsMutable() { return false; }
+
+  // Underlying refcount access, used to implement IsMutable.
+  // TODO(nisse): Demote to protected, as soon as Chrome is changed to
+  // use IsMutable.
   virtual bool HasOneRef() const = 0;
 
   // The resolution of the frame in pixels. For formats where some planes are
@@ -41,16 +50,36 @@ class VideoFrameBuffer : public rtc::RefCountInterface {
   virtual int width() const = 0;
   virtual int height() const = 0;
 
+  // TODO(nisse): For the transition, we use default implementations
+  // of the stride and data methods where the new methods calls the
+  // old method, and the old method calls the new methods. Subclasses
+  // must override either the new methods or the old method, to break
+  // infinite recursion. And similarly for the strides. When
+  // applications, in particular Chrome, are updated, delete the old
+  // method and delete the default implementation of the new methods.
+
   // Returns pointer to the pixel data for a given plane. The memory is owned by
   // the VideoFrameBuffer object and must not be freed by the caller.
-  virtual const uint8_t* data(PlaneType type) const = 0;
-
-  // Non-const data access is disallowed by default. You need to make sure you
-  // have exclusive access and a writable buffer before calling this function.
+  virtual const uint8_t* DataY() const;
+  virtual const uint8_t* DataU() const;
+  virtual const uint8_t* DataV() const;
+  // Deprecated method.
+  // TODO(nisse): Delete after all users are updated.
+  virtual const uint8_t* data(PlaneType type) const;
+
+  // Non-const data access is allowed only if HasOneRef() is true.
+  virtual uint8_t* MutableDataY();
+  virtual uint8_t* MutableDataU();
+  virtual uint8_t* MutableDataV();
+  // Deprecated method. TODO(nisse): Delete after all users are updated.
   virtual uint8_t* MutableData(PlaneType type);
 
   // Returns the number of bytes between successive rows for a given plane.
-  virtual int stride(PlaneType type) const = 0;
+  virtual int StrideY() const;
+  virtual int StrideU() const;
+  virtual int StrideV() const;
+  // Deprecated method. TODO(nisse): Delete after all users are updated.
+  virtual int stride(PlaneType type) const;
 
   // Return the handle of the underlying video frame. This is used when the
   // frame is backed by a texture.
@@ -73,11 +102,19 @@ class I420Buffer : public VideoFrameBuffer {
 
   int width() const override;
   int height() const override;
-  const uint8_t* data(PlaneType type) const override;
-  // Non-const data access is only allowed if HasOneRef() is true to protect
+  const uint8_t* DataY() const override;
+  const uint8_t* DataU() const override;
+  const uint8_t* DataV() const override;
+  // Non-const data access is only allowed if IsMutable() is true, to protect
   // against unexpected overwrites.
-  uint8_t* MutableData(PlaneType type) override;
-  int stride(PlaneType type) const override;
+  bool IsMutable() override;
+  uint8_t* MutableDataY() override;
+  uint8_t* MutableDataU() override;
+  uint8_t* MutableDataV() override;
+  int StrideY() const override;
+  int StrideU() const override;
+  int StrideV() const override;
+
   void* native_handle() const override;
   rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
 
@@ -107,9 +144,15 @@ class NativeHandleBuffer : public VideoFrameBuffer {
 
   int width() const override;
   int height() const override;
-  const uint8_t* data(PlaneType type) const override;
-  int stride(PlaneType type) const override;
+  const uint8_t* DataY() const override;
+  const uint8_t* DataU() const override;
+  const uint8_t* DataV() const override;
+  int StrideY() const override;
+  int StrideU() const override;
+  int StrideV() const override;
+
   void* native_handle() const override;
+  bool IsMutable() override;
 
  protected:
   void* native_handle_;
@@ -131,9 +174,15 @@ class WrappedI420Buffer : public webrtc::VideoFrameBuffer {
   int width() const override;
   int height() const override;
 
-  const uint8_t* data(PlaneType type) const override;
+  bool IsMutable() override;
+
+  const uint8_t* DataY() const override;
+  const uint8_t* DataU() const override;
+  const uint8_t* DataV() const override;
+  int StrideY() const override;
+  int StrideU() const override;
+  int StrideV() const override;
 
-  int stride(PlaneType type) const override;
   void* native_handle() const override;
 
   rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
diff --git a/chromium/third_party/webrtc/common_video/incoming_video_stream.cc b/chromium/third_party/webrtc/common_video/incoming_video_stream.cc
index 0e048a6cc00..5fc8eae0bd5 100644
--- a/chromium/third_party/webrtc/common_video/incoming_video_stream.cc
+++ b/chromium/third_party/webrtc/common_video/incoming_video_stream.cc
@@ -22,19 +22,17 @@
 #endif
 
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/common_video/video_render_frames.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
-IncomingVideoStream::IncomingVideoStream(uint32_t stream_id,
-                                         bool disable_prerenderer_smoothing)
-    : stream_id_(stream_id),
-      disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
+IncomingVideoStream::IncomingVideoStream(bool disable_prerenderer_smoothing)
+    : disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
       incoming_render_thread_(),
       deliver_buffer_event_(EventTimerWrapper::Create()),
       running_(false),
@@ -43,32 +41,22 @@ IncomingVideoStream::IncomingVideoStream(uint32_t stream_id,
       render_buffers_(new VideoRenderFrames()),
       incoming_rate_(0),
       last_rate_calculation_time_ms_(0),
-      num_frames_since_last_calculation_(0),
-      last_render_time_ms_(0),
-      temp_frame_(),
-      start_image_(),
-      timeout_image_(),
-      timeout_time_() {}
+      num_frames_since_last_calculation_(0) {}
 
 IncomingVideoStream::~IncomingVideoStream() {
   Stop();
 }
 
-VideoRenderCallback* IncomingVideoStream::ModuleCallback() {
-  return this;
-}
-
-int32_t IncomingVideoStream::RenderFrame(const uint32_t stream_id,
-                                         const VideoFrame& video_frame) {
+void IncomingVideoStream::OnFrame(const VideoFrame& video_frame) {
   rtc::CritScope csS(&stream_critsect_);
 
   if (!running_) {
-    return -1;
+    return;
   }
 
   // Rate statistics.
   num_frames_since_last_calculation_++;
-  int64_t now_ms = TickTime::MillisecondTimestamp();
+  int64_t now_ms = rtc::TimeMillis();
   if (now_ms >= last_rate_calculation_time_ms_ + kFrameRatePeriodMs) {
     incoming_rate_ =
         static_cast<uint32_t>(1000 * num_frames_since_last_calculation_ /
@@ -86,23 +74,10 @@ int32_t IncomingVideoStream::RenderFrame(const uint32_t stream_id,
       deliver_buffer_event_->Set();
     }
   }
-  return 0;
-}
-
-void IncomingVideoStream::SetStartImage(const VideoFrame& video_frame) {
-  rtc::CritScope csS(&thread_critsect_);
-  start_image_.CopyFrame(video_frame);
-}
-
-void IncomingVideoStream::SetTimeoutImage(const VideoFrame& video_frame,
-                                          const uint32_t timeout) {
-  rtc::CritScope csS(&thread_critsect_);
-  timeout_time_ = timeout;
-  timeout_image_.CopyFrame(video_frame);
 }
 
 void IncomingVideoStream::SetRenderCallback(
-    VideoRenderCallback* render_callback) {
+    rtc::VideoSinkInterface<VideoFrame>* render_callback) {
   rtc::CritScope cs(&thread_critsect_);
   render_callback_ = render_callback;
 }
@@ -118,7 +93,7 @@ int32_t IncomingVideoStream::SetExpectedRenderDelay(
 }
 
 void IncomingVideoStream::SetExternalCallback(
-    VideoRenderCallback* external_callback) {
+    rtc::VideoSinkInterface<VideoFrame>* external_callback) {
   rtc::CritScope cs(&thread_critsect_);
   external_callback_ = external_callback;
 }
@@ -182,10 +157,6 @@ int32_t IncomingVideoStream::Reset() {
   return 0;
 }
 
-uint32_t IncomingVideoStream::StreamId() const {
-  return stream_id_;
-}
-
 uint32_t IncomingVideoStream::IncomingRate() const {
   rtc::CritScope cs(&stream_critsect_);
   return incoming_rate_;
@@ -226,33 +197,16 @@ bool IncomingVideoStream::IncomingVideoStreamProcess() {
 void IncomingVideoStream::DeliverFrame(const VideoFrame& video_frame) {
   rtc::CritScope cs(&thread_critsect_);
   if (video_frame.IsZeroSize()) {
-    if (render_callback_) {
-      if (last_render_time_ms_ == 0 && !start_image_.IsZeroSize()) {
-        // We have not rendered anything and have a start image.
-        temp_frame_.CopyFrame(start_image_);
-        render_callback_->RenderFrame(stream_id_, temp_frame_);
-      } else if (!timeout_image_.IsZeroSize() &&
-                 last_render_time_ms_ + timeout_time_ <
-                     TickTime::MillisecondTimestamp()) {
-        // Render a timeout image.
-        temp_frame_.CopyFrame(timeout_image_);
-        render_callback_->RenderFrame(stream_id_, temp_frame_);
-      }
-    }
-
     // No frame.
     return;
   }
 
   // Send frame for rendering.
   if (external_callback_) {
-    external_callback_->RenderFrame(stream_id_, video_frame);
+    external_callback_->OnFrame(video_frame);
   } else if (render_callback_) {
-    render_callback_->RenderFrame(stream_id_, video_frame);
+    render_callback_->OnFrame(video_frame);
   }
-
-  // We're done with this frame.
-  last_render_time_ms_ = video_frame.render_time_ms();
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/common_video/libyuv/libyuv_unittest.cc b/chromium/third_party/webrtc/common_video/libyuv/libyuv_unittest.cc
index 9f92b8bb15c..e45b5334704 100644
--- a/chromium/third_party/webrtc/common_video/libyuv/libyuv_unittest.cc
+++ b/chromium/third_party/webrtc/common_video/libyuv/libyuv_unittest.cc
@@ -15,67 +15,11 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/video_frame.h"
 
 namespace webrtc {
 
-int PrintBuffer(const uint8_t* buffer, int width, int height, int stride) {
-  if (buffer == NULL)
-    return -1;
-  int k;
-  const uint8_t* tmp_buffer = buffer;
-  for (int i = 0; i < height; i++) {
-    k = 0;
-    for (int j = 0; j < width; j++) {
-      printf("%d ", tmp_buffer[k++]);
-    }
-    tmp_buffer += stride;
-    printf(" \n");
-  }
-  printf(" \n");
-  return 0;
-}
-
-int PrintFrame(const VideoFrame* frame, const char* str) {
-  if (frame == NULL)
-     return -1;
-  printf("%s %dx%d \n", str, frame->width(), frame->height());
-
-  int ret = 0;
-  for (int plane_num = 0; plane_num < kNumOfPlanes; ++plane_num) {
-    PlaneType plane_type = static_cast<PlaneType>(plane_num);
-    int width = (plane_num ? (frame->width() + 1) / 2 : frame->width());
-    int height = (plane_num ? (frame->height() + 1) / 2 : frame->height());
-    ret += PrintBuffer(frame->buffer(plane_type), width, height,
-                       frame->stride(plane_type));
-  }
-  return ret;
-}
-
-
-// Create an image from on a YUV frame. Every plane value starts with a start
-// value, and will be set to increasing values.
-void CreateImage(VideoFrame* frame, int plane_offset[kNumOfPlanes]) {
-  if (frame == NULL)
-    return;
-  for (int plane_num = 0; plane_num < kNumOfPlanes; ++plane_num) {
-    int width = (plane_num != kYPlane ? (frame->width() + 1) / 2 :
-      frame->width());
-    int height = (plane_num != kYPlane ? (frame->height() + 1) / 2 :
-      frame->height());
-    PlaneType plane_type = static_cast<PlaneType>(plane_num);
-    uint8_t *data = frame->buffer(plane_type);
-    for (int i = 0; i < height; i++) {
-      for (int j = 0; j < width; j++) {
-        data[j] = static_cast<uint8_t>(i + plane_offset[plane_num] + j);
-      }
-      data += frame->stride(plane_type);
-    }
-  }
-}
-
 class TestLibYuv : public ::testing::Test {
  protected:
   TestLibYuv();
diff --git a/chromium/third_party/webrtc/common_video/libyuv/scaler.cc b/chromium/third_party/webrtc/common_video/libyuv/scaler.cc
index c6adbf95078..6e683c003b4 100644
--- a/chromium/third_party/webrtc/common_video/libyuv/scaler.cc
+++ b/chromium/third_party/webrtc/common_video/libyuv/scaler.cc
@@ -47,6 +47,7 @@ int Scaler::Set(int src_width, int src_height,
   return 0;
 }
 
+// TODO(nisse): Should work with VideoFrameBuffer instead.
 int Scaler::Scale(const VideoFrame& src_frame, VideoFrame* dst_frame) {
   assert(dst_frame);
   if (src_frame.IsZeroSize())
@@ -69,31 +70,35 @@ int Scaler::Scale(const VideoFrame& src_frame, VideoFrame* dst_frame) {
   const int src_offset_x = ((src_width_ - cropped_src_width) / 2) & ~1;
   const int src_offset_y = ((src_height_ - cropped_src_height) / 2) & ~1;
 
-  const uint8_t* y_ptr = src_frame.buffer(kYPlane) +
-                         src_offset_y * src_frame.stride(kYPlane) +
-                         src_offset_x;
-  const uint8_t* u_ptr = src_frame.buffer(kUPlane) +
-                         src_offset_y / 2 * src_frame.stride(kUPlane) +
-                         src_offset_x / 2;
-  const uint8_t* v_ptr = src_frame.buffer(kVPlane) +
-                         src_offset_y / 2 * src_frame.stride(kVPlane) +
-                         src_offset_x / 2;
-
-  return libyuv::I420Scale(y_ptr,
-                           src_frame.stride(kYPlane),
-                           u_ptr,
-                           src_frame.stride(kUPlane),
-                           v_ptr,
-                           src_frame.stride(kVPlane),
-                           cropped_src_width, cropped_src_height,
-                           dst_frame->buffer(kYPlane),
-                           dst_frame->stride(kYPlane),
-                           dst_frame->buffer(kUPlane),
-                           dst_frame->stride(kUPlane),
-                           dst_frame->buffer(kVPlane),
-                           dst_frame->stride(kVPlane),
-                           dst_width_, dst_height_,
-                           libyuv::FilterMode(method_));
+  const uint8_t* y_ptr =
+      src_frame.video_frame_buffer()->DataY() +
+      src_offset_y * src_frame.video_frame_buffer()->StrideY() +
+      src_offset_x;
+  const uint8_t* u_ptr =
+      src_frame.video_frame_buffer()->DataU() +
+      src_offset_y / 2 * src_frame.video_frame_buffer()->StrideU() +
+      src_offset_x / 2;
+  const uint8_t* v_ptr =
+      src_frame.video_frame_buffer()->DataV() +
+      src_offset_y / 2 * src_frame.video_frame_buffer()->StrideV() +
+      src_offset_x / 2;
+
+  return libyuv::I420Scale(
+      y_ptr,
+      src_frame.video_frame_buffer()->StrideY(),
+      u_ptr,
+      src_frame.video_frame_buffer()->StrideU(),
+      v_ptr,
+      src_frame.video_frame_buffer()->StrideV(),
+      cropped_src_width, cropped_src_height,
+      dst_frame->video_frame_buffer()->MutableDataY(),
+      dst_frame->video_frame_buffer()->StrideY(),
+      dst_frame->video_frame_buffer()->MutableDataU(),
+      dst_frame->video_frame_buffer()->StrideU(),
+      dst_frame->video_frame_buffer()->MutableDataV(),
+      dst_frame->video_frame_buffer()->StrideV(),
+      dst_width_, dst_height_,
+      libyuv::FilterMode(method_));
 }
 
 bool Scaler::SupportedVideoType(VideoType src_video_type,
diff --git a/chromium/third_party/webrtc/common_video/libyuv/scaler_unittest.cc b/chromium/third_party/webrtc/common_video/libyuv/scaler_unittest.cc
index 9ba1b9d94f3..29a0a7403d6 100644
--- a/chromium/third_party/webrtc/common_video/libyuv/scaler_unittest.cc
+++ b/chromium/third_party/webrtc/common_video/libyuv/scaler_unittest.cc
@@ -14,8 +14,8 @@
 #include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/libyuv/include/scaler.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -378,9 +378,9 @@ void TestScaler::ScaleSequence(ScaleMethod method,
                             (src_width + 1) / 2,
                             kVideoRotation_0);
 
-    start_clock = TickTime::MillisecondTimestamp();
+    start_clock = rtc::TimeMillis();
     EXPECT_EQ(0, test_scaler_.Scale(input_frame, &output_frame));
-    total_clock += TickTime::MillisecondTimestamp() - start_clock;
+    total_clock += rtc::TimeMillis() - start_clock;
     if (PrintVideoFrame(output_frame, output_file) < 0) {
         return;
     }
diff --git a/chromium/third_party/webrtc/common_video/libyuv/webrtc_libyuv.cc b/chromium/third_party/webrtc/common_video/libyuv/webrtc_libyuv.cc
index 48f5c2036ba..7f5e3300c13 100644
--- a/chromium/third_party/webrtc/common_video/libyuv/webrtc_libyuv.cc
+++ b/chromium/third_party/webrtc/common_video/libyuv/webrtc_libyuv.cc
@@ -102,23 +102,42 @@ size_t CalcBufferSize(VideoType type, int width, int height) {
   return buffer_size;
 }
 
+static int PrintPlane(const uint8_t* buf,
+                      int width,
+                      int height,
+                      int stride,
+                      FILE* file) {
+  for (int i = 0; i < height; i++, buf += stride) {
+    if (fwrite(buf, 1, width, file) != static_cast<unsigned int>(width))
+      return -1;
+  }
+  return 0;
+}
+
+// TODO(nisse): Belongs with the test code?
 int PrintVideoFrame(const VideoFrame& frame, FILE* file) {
   if (file == NULL)
     return -1;
   if (frame.IsZeroSize())
     return -1;
-  for (int planeNum = 0; planeNum < kNumOfPlanes; ++planeNum) {
-    int width = (planeNum ? (frame.width() + 1) / 2 : frame.width());
-    int height = (planeNum ? (frame.height() + 1) / 2 : frame.height());
-    PlaneType plane_type = static_cast<PlaneType>(planeNum);
-    const uint8_t* plane_buffer = frame.buffer(plane_type);
-    for (int y = 0; y < height; y++) {
-     if (fwrite(plane_buffer, 1, width, file) !=
-         static_cast<unsigned int>(width)) {
-       return -1;
-       }
-       plane_buffer += frame.stride(plane_type);
-    }
+  int width = frame.video_frame_buffer()->width();
+  int height = frame.video_frame_buffer()->height();
+  int chroma_width = (width + 1) / 2;
+  int chroma_height = (height + 1) / 2;
+
+  if (PrintPlane(frame.video_frame_buffer()->DataY(), width, height,
+                 frame.video_frame_buffer()->StrideY(), file) < 0) {
+    return -1;
+  }
+  if (PrintPlane(frame.video_frame_buffer()->DataU(),
+                 chroma_width, chroma_height,
+                 frame.video_frame_buffer()->StrideU(), file) < 0) {
+    return -1;
+  }
+  if (PrintPlane(frame.video_frame_buffer()->DataV(),
+                 chroma_width, chroma_height,
+                 frame.video_frame_buffer()->StrideV(), file) < 0) {
+    return -1;
   }
   return 0;
 }
@@ -133,22 +152,23 @@ int ExtractBuffer(const VideoFrame& input_frame, size_t size, uint8_t* buffer) {
      return -1;
   }
 
-  int pos = 0;
-  uint8_t* buffer_ptr = buffer;
+  int width = input_frame.video_frame_buffer()->width();
+  int height = input_frame.video_frame_buffer()->height();
+  int chroma_width = (width + 1) / 2;
+  int chroma_height = (height + 1) / 2;
+
+  libyuv::I420Copy(input_frame.video_frame_buffer()->DataY(),
+                   input_frame.video_frame_buffer()->StrideY(),
+                   input_frame.video_frame_buffer()->DataU(),
+                   input_frame.video_frame_buffer()->StrideU(),
+                   input_frame.video_frame_buffer()->DataV(),
+                   input_frame.video_frame_buffer()->StrideV(),
+                   buffer, width,
+                   buffer + width*height, chroma_width,
+                   buffer + width*height + chroma_width*chroma_height,
+                   chroma_width,
+                   width, height);
 
-  for (int plane = 0; plane < kNumOfPlanes; ++plane) {
-    int width = (plane ? (input_frame.width() + 1) / 2 :
-      input_frame.width());
-    int height = (plane ? (input_frame.height() + 1) / 2 :
-      input_frame.height());
-    const uint8_t* plane_ptr = input_frame.buffer(
-        static_cast<PlaneType>(plane));
-    for (int y = 0; y < height; y++) {
-      memcpy(&buffer_ptr[pos], plane_ptr, width);
-      pos += width;
-      plane_ptr += input_frame.stride(static_cast<PlaneType>(plane));
-    }
-  }
   return static_cast<int>(length);
 }
 
@@ -228,6 +248,7 @@ int ConvertVideoType(VideoType video_type) {
   return libyuv::FOURCC_ANY;
 }
 
+// TODO(nisse): Delete this wrapper, let callers use libyuv directly.
 int ConvertToI420(VideoType src_video_type,
                   const uint8_t* src_frame,
                   int crop_x,
@@ -245,33 +266,35 @@ int ConvertToI420(VideoType src_video_type,
     dst_width = dst_frame->height();
     dst_height = dst_frame->width();
   }
-  return libyuv::ConvertToI420(src_frame, sample_size,
-                               dst_frame->buffer(kYPlane),
-                               dst_frame->stride(kYPlane),
-                               dst_frame->buffer(kUPlane),
-                               dst_frame->stride(kUPlane),
-                               dst_frame->buffer(kVPlane),
-                               dst_frame->stride(kVPlane),
-                               crop_x, crop_y,
-                               src_width, src_height,
-                               dst_width, dst_height,
-                               ConvertRotationMode(rotation),
-                               ConvertVideoType(src_video_type));
+  return libyuv::ConvertToI420(
+      src_frame, sample_size,
+      dst_frame->video_frame_buffer()->MutableDataY(),
+      dst_frame->video_frame_buffer()->StrideY(),
+      dst_frame->video_frame_buffer()->MutableDataU(),
+      dst_frame->video_frame_buffer()->StrideU(),
+      dst_frame->video_frame_buffer()->MutableDataV(),
+      dst_frame->video_frame_buffer()->StrideV(),
+      crop_x, crop_y,
+      src_width, src_height,
+      dst_width, dst_height,
+      ConvertRotationMode(rotation),
+      ConvertVideoType(src_video_type));
 }
 
 int ConvertFromI420(const VideoFrame& src_frame,
                     VideoType dst_video_type,
                     int dst_sample_size,
                     uint8_t* dst_frame) {
-  return libyuv::ConvertFromI420(src_frame.buffer(kYPlane),
-                                 src_frame.stride(kYPlane),
-                                 src_frame.buffer(kUPlane),
-                                 src_frame.stride(kUPlane),
-                                 src_frame.buffer(kVPlane),
-                                 src_frame.stride(kVPlane),
-                                 dst_frame, dst_sample_size,
-                                 src_frame.width(), src_frame.height(),
-                                 ConvertVideoType(dst_video_type));
+  return libyuv::ConvertFromI420(
+      src_frame.video_frame_buffer()->DataY(),
+      src_frame.video_frame_buffer()->StrideY(),
+      src_frame.video_frame_buffer()->DataU(),
+      src_frame.video_frame_buffer()->StrideU(),
+      src_frame.video_frame_buffer()->DataV(),
+      src_frame.video_frame_buffer()->StrideV(),
+      dst_frame, dst_sample_size,
+      src_frame.width(), src_frame.height(),
+      ConvertVideoType(dst_video_type));
 }
 
 // TODO(mikhal): Create a designated VideoFrame for non I420.
@@ -280,15 +303,16 @@ int ConvertFromYV12(const VideoFrame& src_frame,
                     int dst_sample_size,
                     uint8_t* dst_frame) {
   // YV12 = Y, V, U
-  return libyuv::ConvertFromI420(src_frame.buffer(kYPlane),
-                                 src_frame.stride(kYPlane),
-                                 src_frame.buffer(kVPlane),
-                                 src_frame.stride(kVPlane),
-                                 src_frame.buffer(kUPlane),
-                                 src_frame.stride(kUPlane),
-                                 dst_frame, dst_sample_size,
-                                 src_frame.width(), src_frame.height(),
-                                 ConvertVideoType(dst_video_type));
+  return libyuv::ConvertFromI420(
+      src_frame.video_frame_buffer()->DataY(),
+      src_frame.video_frame_buffer()->StrideY(),
+      src_frame.video_frame_buffer()->DataV(),
+      src_frame.video_frame_buffer()->StrideV(),
+      src_frame.video_frame_buffer()->DataU(),
+      src_frame.video_frame_buffer()->StrideU(),
+      dst_frame, dst_sample_size,
+      src_frame.width(), src_frame.height(),
+      ConvertVideoType(dst_video_type));
 }
 
 // Compute PSNR for an I420 frame (all planes)
@@ -301,18 +325,18 @@ double I420PSNR(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
   else if (ref_frame->width() < 0 || ref_frame->height() < 0)
     return -1;
 
-  double psnr = libyuv::I420Psnr(ref_frame->buffer(kYPlane),
-                                 ref_frame->stride(kYPlane),
-                                 ref_frame->buffer(kUPlane),
-                                 ref_frame->stride(kUPlane),
-                                 ref_frame->buffer(kVPlane),
-                                 ref_frame->stride(kVPlane),
-                                 test_frame->buffer(kYPlane),
-                                 test_frame->stride(kYPlane),
-                                 test_frame->buffer(kUPlane),
-                                 test_frame->stride(kUPlane),
-                                 test_frame->buffer(kVPlane),
-                                 test_frame->stride(kVPlane),
+  double psnr = libyuv::I420Psnr(ref_frame->video_frame_buffer()->DataY(),
+                                 ref_frame->video_frame_buffer()->StrideY(),
+                                 ref_frame->video_frame_buffer()->DataU(),
+                                 ref_frame->video_frame_buffer()->StrideU(),
+                                 ref_frame->video_frame_buffer()->DataV(),
+                                 ref_frame->video_frame_buffer()->StrideV(),
+                                 test_frame->video_frame_buffer()->DataY(),
+                                 test_frame->video_frame_buffer()->StrideY(),
+                                 test_frame->video_frame_buffer()->DataU(),
+                                 test_frame->video_frame_buffer()->StrideU(),
+                                 test_frame->video_frame_buffer()->DataV(),
+                                 test_frame->video_frame_buffer()->StrideV(),
                                  test_frame->width(), test_frame->height());
   // LibYuv sets the max psnr value to 128, we restrict it here.
   // In case of 0 mse in one frame, 128 can skew the results significantly.
@@ -329,18 +353,18 @@ double I420SSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
   else if (ref_frame->width() < 0 || ref_frame->height()  < 0)
     return -1;
 
-  return libyuv::I420Ssim(ref_frame->buffer(kYPlane),
-                          ref_frame->stride(kYPlane),
-                          ref_frame->buffer(kUPlane),
-                          ref_frame->stride(kUPlane),
-                          ref_frame->buffer(kVPlane),
-                          ref_frame->stride(kVPlane),
-                          test_frame->buffer(kYPlane),
-                          test_frame->stride(kYPlane),
-                          test_frame->buffer(kUPlane),
-                          test_frame->stride(kUPlane),
-                          test_frame->buffer(kVPlane),
-                          test_frame->stride(kVPlane),
+  return libyuv::I420Ssim(ref_frame->video_frame_buffer()->DataY(),
+                          ref_frame->video_frame_buffer()->StrideY(),
+                          ref_frame->video_frame_buffer()->DataU(),
+                          ref_frame->video_frame_buffer()->StrideU(),
+                          ref_frame->video_frame_buffer()->DataV(),
+                          ref_frame->video_frame_buffer()->StrideV(),
+                          test_frame->video_frame_buffer()->DataY(),
+                          test_frame->video_frame_buffer()->StrideY(),
+                          test_frame->video_frame_buffer()->DataU(),
+                          test_frame->video_frame_buffer()->StrideU(),
+                          test_frame->video_frame_buffer()->DataV(),
+                          test_frame->video_frame_buffer()->StrideV(),
                           test_frame->width(), test_frame->height());
 }
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/common_video/video_frame.cc b/chromium/third_party/webrtc/common_video/video_frame.cc
index a30f658ea0e..cf6b9c82af3 100644
--- a/chromium/third_party/webrtc/common_video/video_frame.cc
+++ b/chromium/third_party/webrtc/common_video/video_frame.cc
@@ -29,11 +29,12 @@ int ExpectedSize(int plane_stride, int image_height, PlaneType type) {
   return plane_stride * ((image_height + 1) / 2);
 }
 
-VideoFrame::VideoFrame() {
-  // Intentionally using Reset instead of initializer list so that any missed
-  // fields in Reset will be caught by memory checkers.
-  Reset();
-}
+VideoFrame::VideoFrame()
+    : video_frame_buffer_(nullptr),
+      timestamp_(0),
+      ntp_time_ms_(0),
+      render_time_ms_(0),
+      rotation_(kVideoRotation_0) {}
 
 VideoFrame::VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
                        uint32_t timestamp,
@@ -65,11 +66,13 @@ void VideoFrame::CreateEmptyFrame(int width,
   rotation_ = kVideoRotation_0;
 
   // Check if it's safe to reuse allocation.
-  if (video_frame_buffer_ && video_frame_buffer_->HasOneRef() &&
+  if (video_frame_buffer_ && video_frame_buffer_->IsMutable() &&
       !video_frame_buffer_->native_handle() &&
       width == video_frame_buffer_->width() &&
-      height == video_frame_buffer_->height() && stride_y == stride(kYPlane) &&
-      stride_u == stride(kUPlane) && stride_v == stride(kVPlane)) {
+      height == video_frame_buffer_->height() &&
+      stride_y == video_frame_buffer_->StrideY() &&
+      stride_u == video_frame_buffer_->StrideU() &&
+      stride_v == video_frame_buffer_->StrideV()) {
     return;
   }
 
@@ -92,9 +95,9 @@ void VideoFrame::CreateFrame(const uint8_t* buffer_y,
   const int expected_size_u = half_height * stride_u;
   const int expected_size_v = half_height * stride_v;
   CreateEmptyFrame(width, height, stride_y, stride_u, stride_v);
-  memcpy(buffer(kYPlane), buffer_y, expected_size_y);
-  memcpy(buffer(kUPlane), buffer_u, expected_size_u);
-  memcpy(buffer(kVPlane), buffer_v, expected_size_v);
+  memcpy(video_frame_buffer_->MutableDataY(), buffer_y, expected_size_y);
+  memcpy(video_frame_buffer_->MutableDataU(), buffer_u, expected_size_u);
+  memcpy(video_frame_buffer_->MutableDataV(), buffer_v, expected_size_v);
   rotation_ = rotation;
 }
 
@@ -113,22 +116,12 @@ void VideoFrame::CreateFrame(const uint8_t* buffer,
 }
 
 void VideoFrame::CopyFrame(const VideoFrame& videoFrame) {
-  if (videoFrame.IsZeroSize()) {
-    video_frame_buffer_ = nullptr;
-  } else if (videoFrame.native_handle()) {
-    video_frame_buffer_ = videoFrame.video_frame_buffer();
-  } else {
-    CreateFrame(videoFrame.buffer(kYPlane), videoFrame.buffer(kUPlane),
-                videoFrame.buffer(kVPlane), videoFrame.width(),
-                videoFrame.height(), videoFrame.stride(kYPlane),
-                videoFrame.stride(kUPlane), videoFrame.stride(kVPlane),
-                kVideoRotation_0);
-  }
+  ShallowCopy(videoFrame);
 
-  timestamp_ = videoFrame.timestamp_;
-  ntp_time_ms_ = videoFrame.ntp_time_ms_;
-  render_time_ms_ = videoFrame.render_time_ms_;
-  rotation_ = videoFrame.rotation_;
+  // If backed by a plain memory buffer, create a new, non-shared, copy.
+  if (video_frame_buffer_ && !video_frame_buffer_->native_handle()) {
+    video_frame_buffer_ = I420Buffer::Copy(video_frame_buffer_);
+  }
 }
 
 void VideoFrame::ShallowCopy(const VideoFrame& videoFrame) {
@@ -139,30 +132,26 @@ void VideoFrame::ShallowCopy(const VideoFrame& videoFrame) {
   rotation_ = videoFrame.rotation_;
 }
 
-void VideoFrame::Reset() {
-  video_frame_buffer_ = nullptr;
-  timestamp_ = 0;
-  ntp_time_ms_ = 0;
-  render_time_ms_ = 0;
-  rotation_ = kVideoRotation_0;
-}
-
-uint8_t* VideoFrame::buffer(PlaneType type) {
-  return video_frame_buffer_ ? video_frame_buffer_->MutableData(type)
-                             : nullptr;
-}
-
-const uint8_t* VideoFrame::buffer(PlaneType type) const {
-  return video_frame_buffer_ ? video_frame_buffer_->data(type) : nullptr;
-}
-
+// TODO(nisse): Delete. Besides test code, only one use, in
+// webrtcvideoengine2.cc:CreateBlackFrame.
 int VideoFrame::allocated_size(PlaneType type) const {
   const int plane_height = (type == kYPlane) ? height() : (height() + 1) / 2;
-  return plane_height * stride(type);
-}
-
-int VideoFrame::stride(PlaneType type) const {
-  return video_frame_buffer_ ? video_frame_buffer_->stride(type) : 0;
+  int stride;
+  switch (type) {
+    case kYPlane:
+      stride = video_frame_buffer_->StrideY();
+      break;
+    case kUPlane:
+      stride = video_frame_buffer_->StrideU();
+      break;
+    case kVPlane:
+      stride = video_frame_buffer_->StrideV();
+      break;
+    default:
+      RTC_NOTREACHED();
+      return 0;
+  }
+  return plane_height * stride;
 }
 
 int VideoFrame::width() const {
@@ -177,11 +166,8 @@ bool VideoFrame::IsZeroSize() const {
   return !video_frame_buffer_;
 }
 
-void* VideoFrame::native_handle() const {
-  return video_frame_buffer_ ? video_frame_buffer_->native_handle() : nullptr;
-}
-
-rtc::scoped_refptr<VideoFrameBuffer> VideoFrame::video_frame_buffer() const {
+const rtc::scoped_refptr<VideoFrameBuffer>& VideoFrame::video_frame_buffer()
+    const {
   return video_frame_buffer_;
 }
 
@@ -191,7 +177,7 @@ void VideoFrame::set_video_frame_buffer(
 }
 
 VideoFrame VideoFrame::ConvertNativeToI420Frame() const {
-  RTC_DCHECK(native_handle());
+  RTC_DCHECK(video_frame_buffer_->native_handle());
   VideoFrame frame;
   frame.ShallowCopy(*this);
   frame.set_video_frame_buffer(video_frame_buffer_->NativeToI420Buffer());
diff --git a/chromium/third_party/webrtc/common_video/video_frame_buffer.cc b/chromium/third_party/webrtc/common_video/video_frame_buffer.cc
index 6f49e8aef9d..700dcaf02b7 100644
--- a/chromium/third_party/webrtc/common_video/video_frame_buffer.cc
+++ b/chromium/third_party/webrtc/common_video/video_frame_buffer.cc
@@ -27,10 +27,80 @@ int I420DataSize(int height, int stride_y, int stride_u, int stride_v) {
 
 }  // namespace
 
-uint8_t* VideoFrameBuffer::MutableData(PlaneType type) {
+const uint8_t* VideoFrameBuffer::data(PlaneType type) const {
+  switch (type) {
+    case kYPlane:
+      return DataY();
+    case kUPlane:
+      return DataU();
+    case kVPlane:
+      return DataV();
+    default:
+      RTC_NOTREACHED();
+      return nullptr;
+  }
+}
+
+const uint8_t* VideoFrameBuffer::DataY() const {
+  return data(kYPlane);
+}
+const uint8_t* VideoFrameBuffer::DataU() const {
+  return data(kUPlane);
+}
+const uint8_t* VideoFrameBuffer::DataV() const {
+  return data(kVPlane);
+}
+
+int VideoFrameBuffer::stride(PlaneType type) const {
+  switch (type) {
+    case kYPlane:
+      return StrideY();
+    case kUPlane:
+      return StrideU();
+    case kVPlane:
+      return StrideV();
+    default:
+      RTC_NOTREACHED();
+      return 0;
+  }
+}
+
+int VideoFrameBuffer::StrideY() const {
+  return stride(kYPlane);
+}
+int VideoFrameBuffer::StrideU() const {
+  return stride(kUPlane);
+}
+int VideoFrameBuffer::StrideV() const {
+  return stride(kVPlane);
+}
+
+uint8_t* VideoFrameBuffer::MutableDataY() {
   RTC_NOTREACHED();
   return nullptr;
 }
+uint8_t* VideoFrameBuffer::MutableDataU() {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+uint8_t* VideoFrameBuffer::MutableDataV() {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+uint8_t* VideoFrameBuffer::MutableData(PlaneType type) {
+  switch (type) {
+    case kYPlane:
+      return MutableDataY();
+    case kUPlane:
+      return MutableDataU();
+    case kVPlane:
+      return MutableDataV();
+    default:
+      RTC_NOTREACHED();
+      return nullptr;
+  }
+}
 
 VideoFrameBuffer::~VideoFrameBuffer() {}
 
@@ -74,39 +144,41 @@ int I420Buffer::height() const {
   return height_;
 }
 
-const uint8_t* I420Buffer::data(PlaneType type) const {
-  switch (type) {
-    case kYPlane:
-      return data_.get();
-    case kUPlane:
-      return data_.get() + stride_y_ * height_;
-    case kVPlane:
-      return data_.get() + stride_y_ * height_ +
-             stride_u_ * ((height_ + 1) / 2);
-    default:
-      RTC_NOTREACHED();
-      return nullptr;
-  }
+const uint8_t* I420Buffer::DataY() const {
+  return data_.get();
+}
+const uint8_t* I420Buffer::DataU() const {
+  return data_.get() + stride_y_ * height_;
+}
+const uint8_t* I420Buffer::DataV() const {
+  return data_.get() + stride_y_ * height_ + stride_u_ * ((height_ + 1) / 2);
 }
 
-uint8_t* I420Buffer::MutableData(PlaneType type) {
-  RTC_DCHECK(HasOneRef());
-  return const_cast<uint8_t*>(
-      static_cast<const VideoFrameBuffer*>(this)->data(type));
+bool I420Buffer::IsMutable() {
+  return HasOneRef();
 }
 
-int I420Buffer::stride(PlaneType type) const {
-  switch (type) {
-    case kYPlane:
-      return stride_y_;
-    case kUPlane:
-      return stride_u_;
-    case kVPlane:
-      return stride_v_;
-    default:
-      RTC_NOTREACHED();
-      return 0;
-  }
+uint8_t* I420Buffer::MutableDataY() {
+  RTC_DCHECK(IsMutable());
+  return const_cast<uint8_t*>(DataY());
+}
+uint8_t* I420Buffer::MutableDataU() {
+  RTC_DCHECK(IsMutable());
+  return const_cast<uint8_t*>(DataU());
+}
+uint8_t* I420Buffer::MutableDataV() {
+  RTC_DCHECK(IsMutable());
+  return const_cast<uint8_t*>(DataV());
+}
+
+int I420Buffer::StrideY() const {
+  return stride_y_;
+}
+int I420Buffer::StrideU() const {
+  return stride_u_;
+}
+int I420Buffer::StrideV() const {
+  return stride_v_;
 }
 
 void* I420Buffer::native_handle() const {
@@ -124,12 +196,12 @@ rtc::scoped_refptr<I420Buffer> I420Buffer::Copy(
   int height = buffer->height();
   rtc::scoped_refptr<I420Buffer> copy =
       new rtc::RefCountedObject<I420Buffer>(width, height);
-  RTC_CHECK(libyuv::I420Copy(buffer->data(kYPlane), buffer->stride(kYPlane),
-                             buffer->data(kUPlane), buffer->stride(kUPlane),
-                             buffer->data(kVPlane), buffer->stride(kVPlane),
-                             copy->MutableData(kYPlane), copy->stride(kYPlane),
-                             copy->MutableData(kUPlane), copy->stride(kUPlane),
-                             copy->MutableData(kVPlane), copy->stride(kVPlane),
+  RTC_CHECK(libyuv::I420Copy(buffer->DataY(), buffer->StrideY(),
+                             buffer->DataU(), buffer->StrideU(),
+                             buffer->DataV(), buffer->StrideV(),
+                             copy->MutableDataY(), copy->StrideY(),
+                             copy->MutableDataU(), copy->StrideU(),
+                             copy->MutableDataV(), copy->StrideV(),
                              width, height) == 0);
 
   return copy;
@@ -144,6 +216,10 @@ NativeHandleBuffer::NativeHandleBuffer(void* native_handle,
   RTC_DCHECK_GT(height, 0);
 }
 
+bool NativeHandleBuffer::IsMutable() {
+  return false;
+}
+
 int NativeHandleBuffer::width() const {
   return width_;
 }
@@ -152,12 +228,28 @@ int NativeHandleBuffer::height() const {
   return height_;
 }
 
-const uint8_t* NativeHandleBuffer::data(PlaneType type) const {
+const uint8_t* NativeHandleBuffer::DataY() const {
+  RTC_NOTREACHED();  // Should not be called.
+  return nullptr;
+}
+const uint8_t* NativeHandleBuffer::DataU() const {
+  RTC_NOTREACHED();  // Should not be called.
+  return nullptr;
+}
+const uint8_t* NativeHandleBuffer::DataV() const {
   RTC_NOTREACHED();  // Should not be called.
   return nullptr;
 }
 
-int NativeHandleBuffer::stride(PlaneType type) const {
+int NativeHandleBuffer::StrideY() const {
+  RTC_NOTREACHED();  // Should not be called.
+  return 0;
+}
+int NativeHandleBuffer::StrideU() const {
+  RTC_NOTREACHED();  // Should not be called.
+  return 0;
+}
+int NativeHandleBuffer::StrideV() const {
   RTC_NOTREACHED();  // Should not be called.
   return 0;
 }
@@ -190,6 +282,11 @@ WrappedI420Buffer::~WrappedI420Buffer() {
   no_longer_used_cb_();
 }
 
+// Data owned by creator; never mutable.
+bool WrappedI420Buffer::IsMutable() {
+  return false;
+}
+
 int WrappedI420Buffer::width() const {
   return width_;
 }
@@ -198,32 +295,24 @@ int WrappedI420Buffer::height() const {
   return height_;
 }
 
-const uint8_t* WrappedI420Buffer::data(PlaneType type) const {
-  switch (type) {
-    case kYPlane:
-      return y_plane_;
-    case kUPlane:
-      return u_plane_;
-    case kVPlane:
-      return v_plane_;
-    default:
-      RTC_NOTREACHED();
-      return nullptr;
-  }
+const uint8_t* WrappedI420Buffer::DataY() const {
+  return y_plane_;
+}
+const uint8_t* WrappedI420Buffer::DataU() const {
+  return u_plane_;
+}
+const uint8_t* WrappedI420Buffer::DataV() const {
+  return v_plane_;
 }
 
-int WrappedI420Buffer::stride(PlaneType type) const {
-  switch (type) {
-    case kYPlane:
-      return y_stride_;
-    case kUPlane:
-      return u_stride_;
-    case kVPlane:
-      return v_stride_;
-    default:
-      RTC_NOTREACHED();
-      return 0;
-  }
+int WrappedI420Buffer::StrideY() const {
+  return y_stride_;
+}
+int WrappedI420Buffer::StrideU() const {
+  return u_stride_;
+}
+int WrappedI420Buffer::StrideV() const {
+  return v_stride_;
 }
 
 void* WrappedI420Buffer::native_handle() const {
@@ -252,17 +341,17 @@ rtc::scoped_refptr<VideoFrameBuffer> ShallowCenterCrop(
   const int offset_x = uv_offset_x * 2;
   const int offset_y = uv_offset_y * 2;
 
-  const uint8_t* y_plane = buffer->data(kYPlane) +
-                           buffer->stride(kYPlane) * offset_y + offset_x;
-  const uint8_t* u_plane = buffer->data(kUPlane) +
-                           buffer->stride(kUPlane) * uv_offset_y + uv_offset_x;
-  const uint8_t* v_plane = buffer->data(kVPlane) +
-                           buffer->stride(kVPlane) * uv_offset_y + uv_offset_x;
+  const uint8_t* y_plane = buffer->DataY() +
+                           buffer->StrideY() * offset_y + offset_x;
+  const uint8_t* u_plane = buffer->DataU() +
+                           buffer->StrideU() * uv_offset_y + uv_offset_x;
+  const uint8_t* v_plane = buffer->DataV() +
+                           buffer->StrideV() * uv_offset_y + uv_offset_x;
   return new rtc::RefCountedObject<WrappedI420Buffer>(
       cropped_width, cropped_height,
-      y_plane, buffer->stride(kYPlane),
-      u_plane, buffer->stride(kUPlane),
-      v_plane, buffer->stride(kVPlane),
+      y_plane, buffer->StrideY(),
+      u_plane, buffer->StrideU(),
+      v_plane, buffer->StrideV(),
       rtc::KeepRefUntilDone(buffer));
 }
 
diff --git a/chromium/third_party/webrtc/common_video/video_render_frames.cc b/chromium/third_party/webrtc/common_video/video_render_frames.cc
index 8b447cb10fa..3e5dfe9d10f 100644
--- a/chromium/third_party/webrtc/common_video/video_render_frames.cc
+++ b/chromium/third_party/webrtc/common_video/video_render_frames.cc
@@ -12,8 +12,8 @@
 
 #include <assert.h>
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -27,7 +27,7 @@ VideoRenderFrames::VideoRenderFrames()
 }
 
 int32_t VideoRenderFrames::AddFrame(const VideoFrame& new_frame) {
-  const int64_t time_now = TickTime::MillisecondTimestamp();
+  const int64_t time_now = rtc::TimeMillis();
 
   // Drop old frames only when there are other frames in the queue, otherwise, a
   // really slow system never renders any frames.
@@ -74,7 +74,7 @@ uint32_t VideoRenderFrames::TimeToNextFrameRelease() {
   }
   const int64_t time_to_release = incoming_frames_.front().render_time_ms() -
                                   render_delay_ms_ -
-                                  TickTime::MillisecondTimestamp();
+                                  rtc::TimeMillis();
   return time_to_release < 0 ? 0u : static_cast<uint32_t>(time_to_release);
 }
 
diff --git a/chromium/third_party/webrtc/examples/DEPS b/chromium/third_party/webrtc/examples/DEPS
index df9e0406b79..f065c89458b 100644
--- a/chromium/third_party/webrtc/examples/DEPS
+++ b/chromium/third_party/webrtc/examples/DEPS
@@ -1,4 +1,5 @@
 include_rules = [
+  "+WebRTC",
   "+webrtc/api",
   "+webrtc/base",
   "+webrtc/media",
diff --git a/chromium/third_party/webrtc/examples/OWNERS b/chromium/third_party/webrtc/examples/OWNERS
index f489e6b0b0b..d0955711f5f 100644
--- a/chromium/third_party/webrtc/examples/OWNERS
+++ b/chromium/third_party/webrtc/examples/OWNERS
@@ -1,2 +1,4 @@
 glaznev@webrtc.org
+magjed@webrtc.org
+perkj@webrtc.org
 tkchin@webrtc.org
diff --git a/chromium/third_party/webrtc/examples/androidapp/AndroidManifest.xml b/chromium/third_party/webrtc/examples/androidapp/AndroidManifest.xml
index 1e9cbf17d9a..dfe8975b314 100644
--- a/chromium/third_party/webrtc/examples/androidapp/AndroidManifest.xml
+++ b/chromium/third_party/webrtc/examples/androidapp/AndroidManifest.xml
@@ -22,7 +22,9 @@
                  android:allowBackup="false">
 
         <activity android:name="ConnectActivity"
-                  android:label="@string/app_name">
+                  android:label="@string/app_name"
+                  android:theme="@style/Theme.AppCompat"
+                  android:windowSoftInputMode="adjustPan">
             <intent-filter>
                 <action android:name="android.intent.action.MAIN"/>
                 <category android:name="android.intent.category.LAUNCHER"/>
@@ -32,8 +34,8 @@
                 <action android:name="android.intent.action.VIEW"/>
                 <category android:name="android.intent.category.DEFAULT"/>
                 <category android:name="android.intent.category.BROWSABLE"/>
-                <data android:scheme="https" android:host="apprtc.appspot.com"/>
-                <data android:scheme="http" android:host="apprtc.appspot.com"/>
+                <data android:scheme="https" android:host="appr.tc"/>
+                <data android:scheme="http" android:host="appr.tc"/>
             </intent-filter>
         </activity>
 
diff --git a/chromium/third_party/webrtc/examples/androidapp/README b/chromium/third_party/webrtc/examples/androidapp/README
index 856879e4d77..c8705a8bbaa 100644
--- a/chromium/third_party/webrtc/examples/androidapp/README
+++ b/chromium/third_party/webrtc/examples/androidapp/README
@@ -1,4 +1,4 @@
-This directory contains an example Android client for https://apprtc.appspot.com
+This directory contains an example Android client for https://appr.tc
 
 Prerequisites:
 - "Getting the code" on http://www.webrtc.org/native-code/android
@@ -18,8 +18,8 @@ cd <path/to/webrtc>/src
 ninja -C out/Debug AppRTCDemo
 adb install -r out/Debug/apks/AppRTCDemo.apk
 
-In desktop chrome, navigate to https://apprtc.appspot.com and note the r=<NNN> room
-this redirects to or navigate directly to https://apprtc.appspot.com/r/<NNN> with
+In desktop chrome, navigate to https://appr.tc and note the r=<NNN> room
+this redirects to or navigate directly to https://appr.tc/r/<NNN> with
 your own room number. Launch AppRTC on the device and add same <NNN> into the room name list.
 
 You can also run application from a command line to connect to the first room in a list:
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/drawable-hdpi/ic_add_white_24dp.png b/chromium/third_party/webrtc/examples/androidapp/res/drawable-hdpi/ic_add_white_24dp.png
new file mode 100644
index 00000000000..694179bd46b
--- /dev/null
+++ b/chromium/third_party/webrtc/examples/androidapp/res/drawable-hdpi/ic_add_white_24dp.png
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/drawable-mdpi/ic_add_white_24dp.png b/chromium/third_party/webrtc/examples/androidapp/res/drawable-mdpi/ic_add_white_24dp.png
new file mode 100644
index 00000000000..3856041d702
--- /dev/null
+++ b/chromium/third_party/webrtc/examples/androidapp/res/drawable-mdpi/ic_add_white_24dp.png
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/drawable-xhdpi/ic_add_white_24dp.png b/chromium/third_party/webrtc/examples/androidapp/res/drawable-xhdpi/ic_add_white_24dp.png
new file mode 100644
index 00000000000..67bb598e52a
--- /dev/null
+++ b/chromium/third_party/webrtc/examples/androidapp/res/drawable-xhdpi/ic_add_white_24dp.png
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/drawable-xxhdpi/ic_add_white_24dp.png b/chromium/third_party/webrtc/examples/androidapp/res/drawable-xxhdpi/ic_add_white_24dp.png
new file mode 100644
index 00000000000..0fdced8fce7
--- /dev/null
+++ b/chromium/third_party/webrtc/examples/androidapp/res/drawable-xxhdpi/ic_add_white_24dp.png
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/drawable-xxxhdpi/ic_add_white_24dp.png b/chromium/third_party/webrtc/examples/androidapp/res/drawable-xxxhdpi/ic_add_white_24dp.png
new file mode 100644
index 00000000000..d64c22e9edf
--- /dev/null
+++ b/chromium/third_party/webrtc/examples/androidapp/res/drawable-xxxhdpi/ic_add_white_24dp.png
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/layout/activity_connect.xml b/chromium/third_party/webrtc/examples/androidapp/res/layout/activity_connect.xml
index 5b807715883..acc20dc6a18 100644
--- a/chromium/third_party/webrtc/examples/androidapp/res/layout/activity_connect.xml
+++ b/chromium/third_party/webrtc/examples/androidapp/res/layout/activity_connect.xml
@@ -1,72 +1,82 @@
 <?xml version="1.0" encoding="utf-8"?>
 <LinearLayout
-        xmlns:android="http://schemas.android.com/apk/res/android"
+    xmlns:android="http://schemas.android.com/apk/res/android"
+    xmlns:app="http://schemas.android.com/apk/res-auto"
+    android:layout_margin="16dp"
+    android:layout_width="match_parent"
+    android:layout_height="match_parent"
+    android:orientation="vertical"
+    android:weightSum="1"
+    android:layout_centerHorizontal="true">
+
+    <TextView
+        android:id="@+id/room_edittext_description"
         android:layout_width="match_parent"
-        android:layout_height="match_parent"
-        android:orientation="vertical"
-        android:weightSum="1"
-        android:layout_margin="8dp"
-        android:layout_centerHorizontal="true">
+        android:layout_height="wrap_content"
+        android:text="@string/room_description"/>
 
     <LinearLayout
+        android:orientation="horizontal"
         android:layout_width="match_parent"
         android:layout_height="wrap_content"
-        android:orientation="horizontal" >
-            <ImageButton
-                android:id="@+id/add_room_button"
-                android:background="@android:drawable/ic_menu_add"
-                android:contentDescription="@string/add_room_description"
-                android:layout_marginEnd="20dp"
-                android:layout_width="48dp"
-                android:layout_height="48dp"/>
-            <ImageButton
-                android:id="@+id/remove_room_button"
-                android:background="@android:drawable/ic_delete"
-                android:contentDescription="@string/remove_room_description"
-                android:layout_marginEnd="20dp"
-                android:layout_width="48dp"
-                android:layout_height="48dp"/>
-            <ImageButton
-                android:id="@+id/connect_button"
-                android:background="@android:drawable/sym_action_call"
-                android:contentDescription="@string/connect_description"
-                android:layout_marginEnd="20dp"
-                android:layout_width="48dp"
-                android:layout_height="48dp"/>
-            <ImageButton
-                android:id="@+id/connect_loopback_button"
-                android:background="@drawable/ic_loopback_call"
-                android:contentDescription="@string/connect_loopback_description"
-                android:layout_width="48dp"
-                android:layout_height="48dp"/>
-    </LinearLayout>
-    <TextView
-            android:id="@+id/room_edittext_description"
-            android:layout_width="fill_parent"
-            android:layout_height="wrap_content"
-            android:layout_margin="5dp"
-            android:text="@string/room_description"/>
-    <EditText
+        android:gravity="center"
+        android:layout_marginBottom="8dp">
+
+        <EditText
             android:id="@+id/room_edittext"
-            android:layout_width="match_parent"
+            android:layout_width="0dp"
             android:layout_height="wrap_content"
+            android:layout_weight="1"
             android:singleLine="true"
-            android:imeOptions="actionDone"/>
+            android:imeOptions="actionDone"
+            android:inputType="text"/>
+
+        <ImageButton
+            android:id="@+id/connect_button"
+            android:layout_width="48dp"
+            android:layout_height="48dp"
+            android:contentDescription="@string/connect_description"
+            android:background="@android:drawable/sym_action_call" />
+    </LinearLayout>
+
     <TextView
-            android:id="@+id/room_listview_description"
-            android:layout_width="fill_parent"
-            android:layout_height="wrap_content"
-            android:layout_marginTop="5dp"
-            android:lines="1"
-            android:maxLines="1"
-            android:textAppearance="?android:attr/textAppearanceMedium"
-            android:text="@string/room_names"/>
-    <ListView
+        android:id="@+id/room_listview_description"
+        android:layout_width="match_parent"
+        android:layout_height="48dp"
+        android:layout_marginTop="8dp"
+        android:lines="1"
+        android:maxLines="1"
+        android:textAppearance="?android:attr/textAppearanceMedium"
+        android:text="@string/favorites"
+        android:gravity="center_vertical"/>
+
+    <FrameLayout
+        android:layout_width="match_parent"
+        android:layout_height="0dp"
+        android:layout_weight="1">
+
+        <ListView
             android:id="@+id/room_listview"
-            android:layout_width="fill_parent"
-            android:layout_height="wrap_content"
-            android:choiceMode="singleChoice"
-            android:listSelector="@android:color/darker_gray"
+            android:layout_width="match_parent"
+            android:layout_height="match_parent"
             android:drawSelectorOnTop="false" />
 
+        <TextView
+            android:id="@android:id/empty"
+            android:layout_width="match_parent"
+            android:layout_height="match_parent"
+            android:gravity="center"
+            android:text="@string/no_favorites" />
+
+        <android.support.design.widget.FloatingActionButton
+            android:id="@+id/add_favorite_button"
+            android:layout_width="56dp"
+            android:layout_height="56dp"
+            android:layout_margin="8dp"
+            android:layout_gravity="bottom|end"
+            android:src="@drawable/ic_add_white_24dp"
+            android:contentDescription="@string/add_favorite_description"
+            app:backgroundTint="@android:color/holo_green_dark"
+            app:elevation="8dp"/>
+    </FrameLayout>
 </LinearLayout>
\ No newline at end of file
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/layout/dialog_add_favorite.xml b/chromium/third_party/webrtc/examples/androidapp/res/layout/dialog_add_favorite.xml
new file mode 100644
index 00000000000..28b533aa0c6
--- /dev/null
+++ b/chromium/third_party/webrtc/examples/androidapp/res/layout/dialog_add_favorite.xml
@@ -0,0 +1,14 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+              android:orientation="vertical"
+              android:layout_width="match_parent"
+              android:layout_height="match_parent"
+              android:padding="24dp">
+
+    <EditText
+        android:layout_width="match_parent"
+        android:layout_height="wrap_content"
+        android:id="@+id/favorite_edittext"
+        android:layout_gravity="center_horizontal"
+        android:inputType="text"/>
+</LinearLayout>
\ No newline at end of file
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/menu/connect_menu.xml b/chromium/third_party/webrtc/examples/androidapp/res/menu/connect_menu.xml
index d9f94867b97..8beddddd45f 100644
--- a/chromium/third_party/webrtc/examples/androidapp/res/menu/connect_menu.xml
+++ b/chromium/third_party/webrtc/examples/androidapp/res/menu/connect_menu.xml
@@ -1,8 +1,14 @@
-<menu xmlns:android="http://schemas.android.com/apk/res/android" >
+<menu xmlns:android="http://schemas.android.com/apk/res/android"
+      xmlns:app="http://schemas.android.com/apk/res-auto">
+    <item
+        android:id="@+id/action_loopback"
+        android:icon="@drawable/ic_loopback_call"
+        app:showAsAction="always"
+        android:title="@string/action_loopback"/>
     <item
         android:id="@+id/action_settings"
         android:orderInCategory="100"
         android:icon="@android:drawable/ic_menu_preferences"
-        android:showAsAction="ifRoom"
+        app:showAsAction="ifRoom"
         android:title="@string/action_settings"/>
 </menu>
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/values/arrays.xml b/chromium/third_party/webrtc/examples/androidapp/res/values/arrays.xml
index ba8c8918ce0..dd06c46ed16 100644
--- a/chromium/third_party/webrtc/examples/androidapp/res/values/arrays.xml
+++ b/chromium/third_party/webrtc/examples/androidapp/res/values/arrays.xml
@@ -36,4 +36,8 @@
         <item>ISAC</item>
     </string-array>
 
+    <string-array name="roomListContextMenu">
+        <item>Remove favorite</item>
+    </string-array>
+
 </resources>
diff --git a/chromium/third_party/webrtc/examples/androidapp/res/values/strings.xml b/chromium/third_party/webrtc/examples/androidapp/res/values/strings.xml
index 492c34b6840..24244238e07 100644
--- a/chromium/third_party/webrtc/examples/androidapp/res/values/strings.xml
+++ b/chromium/third_party/webrtc/examples/androidapp/res/values/strings.xml
@@ -3,12 +3,13 @@
     <string name="app_name" translatable="no">AppRTC</string>
     <string name="settings_name" translatable="no">AppRTC Settings</string>
     <string name="disconnect_call">Disconnect Call</string>
-    <string name="room_names">Room names:</string>
     <string name="room_description">
         Please enter a room name. Room names are shared with everyone, so think
         of something unique and send it to a friend.
     </string>
-    <string name="connect_text">Connect</string>
+    <string name="favorites">Favorites</string>
+    <string name="no_favorites">No favorites</string>
+    <string name="add_favorite_title">Add favorite</string>
     <string name="invalid_url_title">Invalid URL</string>
     <string name="invalid_url_text">The URL or room name you entered resulted in an invalid URL: %1$s
     </string>
@@ -16,14 +17,15 @@
     <string name="connecting_to">Connecting to: %1$s</string>
     <string name="missing_url">FATAL ERROR: Missing URL to connect to.</string>
     <string name="ok">OK</string>
+    <string name="add">Add</string>
+    <string name="cancel">Cancel</string>
     <string name="switch_camera">Switch front/back camera</string>
     <string name="toggle_debug">Toggle debug view</string>
     <string name="toggle_mic">Toggle microphone on/off</string>
     <string name="action_settings">Settings</string>
-    <string name="add_room_description">Add new room to the list</string>
-    <string name="remove_room_description">Remove room from the list</string>
+    <string name="action_loopback">Loopback connection</string>
     <string name="connect_description">Connect to the room</string>
-    <string name="connect_loopback_description">Loopback connection</string>
+    <string name="add_favorite_description">Add favorite</string>
 
     <!-- Settings strings. -->
     <string name="pref_room_key">room_preference</string>
@@ -119,7 +121,7 @@
     <string name="pref_room_server_url_key">room_server_url_preference</string>
     <string name="pref_room_server_url_title">Room server URL.</string>
     <string name="pref_room_server_url_dlg">Enter a room server URL.</string>
-    <string name="pref_room_server_url_default" translatable="false">https://apprtc.appspot.com</string>
+    <string name="pref_room_server_url_default" translatable="false">https://appr.tc</string>
 
     <string name="pref_displayhud_key">displayhud_preference</string>
     <string name="pref_displayhud_title">Display call statistics.</string>
diff --git a/chromium/third_party/webrtc/examples/androidjunit/README b/chromium/third_party/webrtc/examples/androidjunit/README
new file mode 100644
index 00000000000..502d71460f2
--- /dev/null
+++ b/chromium/third_party/webrtc/examples/androidjunit/README
@@ -0,0 +1,8 @@
+This directory contains example JUnit tests for Android AppRTCDemo.
+Many of these test utilize Robolectric to mock Android classes.
+
+To compile:
+ninja -C out/Debug AppRTCDemoJUnitTest
+
+To run:
+out/Debug/bin/run_AppRTCDemoJUnitTest
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient+Internal.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient+Internal.h
index 3fd218c2fd8..f2da4136e79 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient+Internal.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient+Internal.h
@@ -10,7 +10,7 @@
 
 #import "ARDAppClient.h"
 
-#import "webrtc/api/objc/RTCPeerConnection.h"
+#import "WebRTC/RTCPeerConnection.h"
 
 #import "ARDRoomServerClient.h"
 #import "ARDSignalingChannel.h"
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.h
index b59838e3cc6..b7f75462671 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.h
@@ -9,8 +9,9 @@
  */
 
 #import <Foundation/Foundation.h>
-#import "webrtc/api/objc/RTCPeerConnection.h"
-#import "webrtc/api/objc/RTCVideoTrack.h"
+
+#import "WebRTC/RTCPeerConnection.h"
+#import "WebRTC/RTCVideoTrack.h"
 
 typedef NS_ENUM(NSInteger, ARDAppClientState) {
   // Disconnected from servers.
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.m
index d8dc7714248..86d18d1333c 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppClient.m
@@ -11,17 +11,18 @@
 #import "ARDAppClient+Internal.h"
 
 #if defined(WEBRTC_IOS)
-#import "webrtc/base/objc/RTCTracing.h"
-#import "webrtc/api/objc/RTCAVFoundationVideoSource.h"
+#import "WebRTC/RTCAVFoundationVideoSource.h"
+#import "WebRTC/RTCTracing.h"
 #endif
-#import "webrtc/api/objc/RTCAudioTrack.h"
-#import "webrtc/api/objc/RTCConfiguration.h"
-#import "webrtc/api/objc/RTCIceServer.h"
-#import "webrtc/api/objc/RTCMediaConstraints.h"
-#import "webrtc/api/objc/RTCMediaStream.h"
-#import "webrtc/api/objc/RTCPeerConnectionFactory.h"
-#import "webrtc/base/objc/RTCFileLogger.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCAudioTrack.h"
+#import "WebRTC/RTCConfiguration.h"
+#import "WebRTC/RTCFileLogger.h"
+#import "WebRTC/RTCIceServer.h"
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCMediaConstraints.h"
+#import "WebRTC/RTCMediaStream.h"
+#import "WebRTC/RTCPeerConnectionFactory.h"
+#import "WebRTC/RTCRtpSender.h"
 
 #import "ARDAppEngineClient.h"
 #import "ARDCEODTURNClient.h"
@@ -48,8 +49,11 @@ static NSInteger const kARDAppClientErrorCreateSDP = -3;
 static NSInteger const kARDAppClientErrorSetSDP = -4;
 static NSInteger const kARDAppClientErrorInvalidClient = -5;
 static NSInteger const kARDAppClientErrorInvalidRoom = -6;
+static NSString * const kARDMediaStreamId = @"ARDAMS";
+static NSString * const kARDAudioTrackId = @"ARDAMSa0";
+static NSString * const kARDVideoTrackId = @"ARDAMSv0";
 
-// TODO(tkchin): Remove guard once rtc_base_objc compiles on Mac.
+// TODO(tkchin): Remove guard once rtc_sdk_common_objc compiles on Mac.
 #if defined(WEBRTC_IOS)
 // TODO(tkchin): Add this as a UI option.
 static BOOL const kARDAppClientEnableTracing = NO;
@@ -505,9 +509,9 @@ static BOOL const kARDAppClientEnableTracing = NO;
   _peerConnection = [_factory peerConnectionWithConfiguration:config
                                                   constraints:constraints
                                                      delegate:self];
-  // Create AV media stream and add it to the peer connection.
-  RTCMediaStream *localStream = [self createLocalMediaStream];
-  [_peerConnection addStream:localStream];
+  // Create AV senders.
+  [self createAudioSender];
+  [self createVideoSender];
   if (_isInitiator) {
     // Send offer.
     __weak ARDAppClient *weakSelf = self;
@@ -606,17 +610,25 @@ static BOOL const kARDAppClientEnableTracing = NO;
   }
 }
 
-- (RTCMediaStream *)createLocalMediaStream {
-  RTCMediaStream *localStream = [_factory mediaStreamWithStreamId:@"ARDAMS"];
-  RTCVideoTrack *localVideoTrack = [self createLocalVideoTrack];
-  if (localVideoTrack) {
-    [localStream addVideoTrack:localVideoTrack];
-    [_delegate appClient:self didReceiveLocalVideoTrack:localVideoTrack];
+- (RTCRtpSender *)createVideoSender {
+  RTCRtpSender *sender =
+      [_peerConnection senderWithKind:kRTCMediaStreamTrackKindVideo
+                             streamId:kARDMediaStreamId];
+  RTCVideoTrack *track = [self createLocalVideoTrack];
+  if (track) {
+    sender.track = track;
+    [_delegate appClient:self didReceiveLocalVideoTrack:track];
   }
-  RTCAudioTrack *localAudioTrack =
-      [_factory audioTrackWithTrackId:@"ARDAMSa0"];
-  [localStream addAudioTrack:localAudioTrack];
-  return localStream;
+  return sender;
+}
+
+- (RTCRtpSender *)createAudioSender {
+  RTCRtpSender *sender =
+      [_peerConnection senderWithKind:kRTCMediaStreamTrackKindAudio
+                             streamId:kARDMediaStreamId];
+  RTCAudioTrack *track = [_factory audioTrackWithTrackId:kARDAudioTrackId];
+  sender.track = track;
+  return sender;
 }
 
 - (RTCVideoTrack *)createLocalVideoTrack {
@@ -634,7 +646,7 @@ static BOOL const kARDAppClientEnableTracing = NO;
         [_factory avFoundationVideoSourceWithConstraints:mediaConstraints];
     localVideoTrack =
         [_factory videoTrackWithSource:source
-                               trackId:@"ARDAMSv0"];
+                               trackId:kARDVideoTrackId];
   }
 #endif
   return localVideoTrack;
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppEngineClient.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppEngineClient.m
index 7d8a185f6d7..0ba32c448a2 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppEngineClient.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDAppEngineClient.m
@@ -10,7 +10,7 @@
 
 #import "ARDAppEngineClient.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 
 #import "ARDJoinResponse.h"
 #import "ARDMessageResponse.h"
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSDPUtils.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSDPUtils.m
index be93ae26f81..3a8a578dc68 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSDPUtils.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSDPUtils.m
@@ -10,8 +10,8 @@
 
 #import "ARDSDPUtils.h"
 
-#import "webrtc/api/objc/RTCSessionDescription.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCSessionDescription.h"
 
 @implementation ARDSDPUtils
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.h
index 5cba49559d8..18eafe26736 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.h
@@ -9,8 +9,9 @@
  */
 
 #import <Foundation/Foundation.h>
-#import "webrtc/api/objc/RTCIceCandidate.h"
-#import "webrtc/api/objc/RTCSessionDescription.h"
+
+#import "WebRTC/RTCIceCandidate.h"
+#import "WebRTC/RTCSessionDescription.h"
 
 typedef enum {
   kARDSignalingMessageTypeCandidate,
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.m
index 6dd04cc6e02..a5ca29576a0 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDSignalingMessage.m
@@ -10,7 +10,7 @@
 
 #import "ARDSignalingMessage.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 
 #import "ARDUtilities.h"
 #import "RTCIceCandidate+JSON.h"
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDStatsBuilder.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDStatsBuilder.m
index 60cf5516488..1f4ada6237d 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDStatsBuilder.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDStatsBuilder.m
@@ -10,7 +10,7 @@
 
 #import "ARDStatsBuilder.h"
 
-#import "webrtc/api/objc/RTCStatsReport.h"
+#import "WebRTC/RTCStatsReport.h"
 
 #import "ARDBitrateTracker.h"
 #import "ARDUtilities.h"
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDWebSocketChannel.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDWebSocketChannel.m
index 0e753e845b9..87a690ebeb0 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDWebSocketChannel.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ARDWebSocketChannel.m
@@ -10,7 +10,7 @@
 
 #import "ARDWebSocketChannel.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 #import "SRWebSocket.h"
 
 #import "ARDSignalingMessage.h"
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.h
index 0ee6a7872f2..1051f8ee48f 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCIceCandidate.h"
+#import "WebRTC/RTCIceCandidate.h"
 
 @interface RTCIceCandidate (JSON)
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.m
index 773c346c6b6..f20c490682a 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICECandidate+JSON.m
@@ -10,7 +10,7 @@
 
 #import "RTCIceCandidate+JSON.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 
 static NSString const *kRTCICECandidateTypeKey = @"type";
 static NSString const *kRTCICECandidateTypeValue = @"candidate";
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICEServer+JSON.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICEServer+JSON.h
index a6d63e41823..69fb432b446 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICEServer+JSON.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCICEServer+JSON.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCIceServer.h"
+#import "WebRTC/RTCIceServer.h"
 
 @interface RTCIceServer (JSON)
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h
index 88a4c3576cf..74f89a953fb 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCMediaConstraints.h"
+#import "WebRTC/RTCMediaConstraints.h"
 
 @interface RTCMediaConstraints (JSON)
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCSessionDescription+JSON.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCSessionDescription+JSON.h
index 2f952eec261..cccff9ad74c 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCSessionDescription+JSON.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/RTCSessionDescription+JSON.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCSessionDescription.h"
+#import "WebRTC/RTCSessionDescription.h"
 
 @interface RTCSessionDescription (JSON)
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/common/ARDUtilities.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/common/ARDUtilities.m
index ad5768d1f0c..c9d029f629d 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/common/ARDUtilities.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/common/ARDUtilities.m
@@ -12,7 +12,7 @@
 
 #import <mach/mach.h>
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 
 @implementation NSDictionary (ARDUtilites)
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDAppDelegate.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDAppDelegate.m
index c18c81827cf..8f192623058 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDAppDelegate.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDAppDelegate.m
@@ -10,10 +10,10 @@
 
 #import "ARDAppDelegate.h"
 
-#import "webrtc/base/objc/RTCFieldTrials.h"
-#import "webrtc/base/objc/RTCLogging.h"
-#import "webrtc/base/objc/RTCSSLAdapter.h"
-#import "webrtc/base/objc/RTCTracing.h"
+#import "WebRTC/RTCFieldTrials.h"
+#import "WebRTC/RTCLogging.h"
+#import "WebRTC/RTCSSLAdapter.h"
+#import "WebRTC/RTCTracing.h"
 
 #import "ARDMainViewController.h"
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.h
index d3a583dfa9a..a783ca1a1c4 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.h
@@ -15,10 +15,10 @@
 @protocol ARDMainViewDelegate <NSObject>
 
 - (void)mainView:(ARDMainView *)mainView
-             didInputRoom:(NSString *)room
-               isLoopback:(BOOL)isLoopback
-              isAudioOnly:(BOOL)isAudioOnly
-   shouldDelayAudioConfig:(BOOL)shouldDelayAudioConfig;
+      didInputRoom:(NSString *)room
+        isLoopback:(BOOL)isLoopback
+       isAudioOnly:(BOOL)isAudioOnly
+    useManualAudio:(BOOL)useManualAudio;
 
 - (void)mainViewDidToggleAudioLoop:(ARDMainView *)mainView;
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m
index b2647630e1f..e7e0f940d15 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainView.m
@@ -122,8 +122,8 @@ static CGFloat const kAppLabelHeight = 20;
   UILabel *_audioOnlyLabel;
   UISwitch *_loopbackSwitch;
   UILabel *_loopbackLabel;
-  UISwitch *_audioConfigDelaySwitch;
-  UILabel *_audioConfigDelayLabel;
+  UISwitch *_useManualAudioSwitch;
+  UILabel *_useManualAudioLabel;
   UIButton *_startCallButton;
   UIButton *_audioLoopButton;
 }
@@ -175,17 +175,17 @@ static CGFloat const kAppLabelHeight = 20;
     [_loopbackLabel sizeToFit];
     [self addSubview:_loopbackLabel];
 
-    _audioConfigDelaySwitch = [[UISwitch alloc] initWithFrame:CGRectZero];
-    [_audioConfigDelaySwitch sizeToFit];
-    _audioConfigDelaySwitch.on = YES;
-    [self addSubview:_audioConfigDelaySwitch];
+    _useManualAudioSwitch = [[UISwitch alloc] initWithFrame:CGRectZero];
+    [_useManualAudioSwitch sizeToFit];
+    _useManualAudioSwitch.on = YES;
+    [self addSubview:_useManualAudioSwitch];
 
-    _audioConfigDelayLabel = [[UILabel alloc] initWithFrame:CGRectZero];
-    _audioConfigDelayLabel.text = @"Delay audio config";
-    _audioConfigDelayLabel.font = controlFont;
-    _audioConfigDelayLabel.textColor = controlFontColor;
-    [_audioConfigDelayLabel sizeToFit];
-    [self addSubview:_audioConfigDelayLabel];
+    _useManualAudioLabel = [[UILabel alloc] initWithFrame:CGRectZero];
+    _useManualAudioLabel.text = @"Use manual audio config";
+    _useManualAudioLabel.font = controlFont;
+    _useManualAudioLabel.textColor = controlFontColor;
+    [_useManualAudioLabel sizeToFit];
+    [self addSubview:_useManualAudioLabel];
 
     _startCallButton = [UIButton buttonWithType:UIButtonTypeSystem];
     _startCallButton.backgroundColor = [UIColor blueColor];
@@ -275,22 +275,22 @@ static CGFloat const kAppLabelHeight = 20;
   _loopbackLabel.center = CGPointMake(loopbackModeLabelCenterX,
                                       CGRectGetMidY(loopbackModeRect));
 
-  CGFloat audioConfigDelayTop =
+  CGFloat useManualAudioTop =
       CGRectGetMaxY(_loopbackSwitch.frame) + kCallControlMargin;
-  CGRect audioConfigDelayRect =
+  CGRect useManualAudioRect =
       CGRectMake(kCallControlMargin * 3,
-                 audioConfigDelayTop,
-                 _audioConfigDelaySwitch.frame.size.width,
-                 _audioConfigDelaySwitch.frame.size.height);
-  _audioConfigDelaySwitch.frame = audioConfigDelayRect;
-  CGFloat audioConfigDelayLabelCenterX = CGRectGetMaxX(audioConfigDelayRect) +
-      kCallControlMargin + _audioConfigDelayLabel.frame.size.width / 2;
-  _audioConfigDelayLabel.center =
-      CGPointMake(audioConfigDelayLabelCenterX,
-                  CGRectGetMidY(audioConfigDelayRect));
+                 useManualAudioTop,
+                 _useManualAudioSwitch.frame.size.width,
+                 _useManualAudioSwitch.frame.size.height);
+  _useManualAudioSwitch.frame = useManualAudioRect;
+  CGFloat useManualAudioLabelCenterX = CGRectGetMaxX(useManualAudioRect) +
+      kCallControlMargin + _useManualAudioLabel.frame.size.width / 2;
+  _useManualAudioLabel.center =
+      CGPointMake(useManualAudioLabelCenterX,
+                  CGRectGetMidY(useManualAudioRect));
 
   CGFloat audioLoopTop =
-     CGRectGetMaxY(audioConfigDelayRect) + kCallControlMargin * 3;
+     CGRectGetMaxY(useManualAudioRect) + kCallControlMargin * 3;
   _audioLoopButton.frame = CGRectMake(kCallControlMargin,
                                       audioLoopTop,
                                       _audioLoopButton.frame.size.width,
@@ -335,7 +335,7 @@ static CGFloat const kAppLabelHeight = 20;
                 didInputRoom:room
                   isLoopback:_loopbackSwitch.isOn
                  isAudioOnly:_audioOnlySwitch.isOn
-      shouldDelayAudioConfig:_audioConfigDelaySwitch.isOn];
+              useManualAudio:_useManualAudioSwitch.isOn];
 }
 
 @end
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m
index a634952c297..a684ba32b06 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDMainViewController.m
@@ -12,8 +12,8 @@
 
 #import <AVFoundation/AVFoundation.h>
 
-#import "webrtc/base/objc/RTCDispatcher.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCDispatcher.h"
+#import "WebRTC/RTCLogging.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
@@ -23,13 +23,14 @@
 
 @interface ARDMainViewController () <
     ARDMainViewDelegate,
+    ARDVideoCallViewControllerDelegate,
     RTCAudioSessionDelegate>
 @end
 
 @implementation ARDMainViewController {
   ARDMainView *_mainView;
   AVAudioPlayer *_audioPlayer;
-  BOOL _shouldDelayAudioConfig;
+  BOOL _useManualAudio;
 }
 
 - (void)loadView {
@@ -37,17 +38,26 @@
   _mainView.delegate = self;
   self.view = _mainView;
 
-  [self setupAudioSession];
+  RTCAudioSessionConfiguration *webRTCConfig =
+      [RTCAudioSessionConfiguration webRTCConfiguration];
+  webRTCConfig.categoryOptions = webRTCConfig.categoryOptions |
+      AVAudioSessionCategoryOptionDefaultToSpeaker;
+  [RTCAudioSessionConfiguration setWebRTCConfiguration:webRTCConfig];
+
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  [session addDelegate:self];
+
+  [self configureAudioSession];
   [self setupAudioPlayer];
 }
 
 #pragma mark - ARDMainViewDelegate
 
 - (void)mainView:(ARDMainView *)mainView
-              didInputRoom:(NSString *)room
-                isLoopback:(BOOL)isLoopback
-               isAudioOnly:(BOOL)isAudioOnly
-    shouldDelayAudioConfig:(BOOL)shouldDelayAudioConfig {
+      didInputRoom:(NSString *)room
+        isLoopback:(BOOL)isLoopback
+       isAudioOnly:(BOOL)isAudioOnly
+    useManualAudio:(BOOL)useManualAudio {
   if (!room.length) {
     [self showAlertWithMessage:@"Missing room name."];
     return;
@@ -77,15 +87,16 @@
     return;
   }
 
-  _shouldDelayAudioConfig = shouldDelayAudioConfig;
   RTCAudioSession *session = [RTCAudioSession sharedInstance];
-  session.shouldDelayAudioConfiguration = _shouldDelayAudioConfig;
+  session.useManualAudio = useManualAudio;
+  session.isAudioEnabled = NO;
 
   // Kick off the video call.
   ARDVideoCallViewController *videoCallViewController =
       [[ARDVideoCallViewController alloc] initForRoom:trimmedRoom
                                            isLoopback:isLoopback
-                                          isAudioOnly:isAudioOnly];
+                                          isAudioOnly:isAudioOnly
+                                             delegate:self];
   videoCallViewController.modalTransitionStyle =
       UIModalTransitionStyleCrossDissolve;
   [self presentViewController:videoCallViewController
@@ -102,10 +113,22 @@
   mainView.isAudioLoopPlaying = _audioPlayer.playing;
 }
 
+#pragma mark - ARDVideoCallViewControllerDelegate
+
+- (void)viewControllerDidFinish:(ARDVideoCallViewController *)viewController {
+  if (![viewController isBeingDismissed]) {
+    RTCLog(@"Dismissing VC");
+    [self dismissViewControllerAnimated:YES completion:^{
+      [self restartAudioPlayerIfNeeded];
+    }];
+  }
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  session.isAudioEnabled = NO;
+}
+
 #pragma mark - RTCAudioSessionDelegate
 
-- (void)audioSessionShouldConfigure:(RTCAudioSession *)session {
-  // Won't get called unless audio config is delayed.
+- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session {
   // Stop playback on main queue and then configure WebRTC.
   [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeMain
                                block:^{
@@ -113,35 +136,23 @@
       RTCLog(@"Stopping audio loop due to WebRTC start.");
       [_audioPlayer stop];
     }
-    // TODO(tkchin): Shouldn't lock on main queue. Figure out better way to
-    // check audio loop state.
-    [session lockForConfiguration];
-    [session configureWebRTCSession:nil];
-    [session unlockForConfiguration];
+    RTCLog(@"Setting isAudioEnabled to YES.");
+    session.isAudioEnabled = YES;
   }];
 }
 
-- (void)audioSessionShouldUnconfigure:(RTCAudioSession *)session {
-  // Won't get called unless audio config is delayed.
-  [session lockForConfiguration];
-  [session unconfigureWebRTCSession:nil];
-  [session unlockForConfiguration];
-}
-
-- (void)audioSessionDidUnconfigure:(RTCAudioSession *)session {
+- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session {
   // WebRTC is done with the audio session. Restart playback.
   [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeMain
                                block:^{
-    if (_mainView.isAudioLoopPlaying) {
-      RTCLog(@"Starting audio loop due to WebRTC end.");
-      [_audioPlayer play];
-    }
+    RTCLog(@"audioSessionDidStopPlayOrRecord");
+    [self restartAudioPlayerIfNeeded];
   }];
 }
 
 #pragma mark - Private
 
-- (void)setupAudioSession {
+- (void)configureAudioSession {
   RTCAudioSessionConfiguration *configuration =
       [[RTCAudioSessionConfiguration alloc] init];
   configuration.category = AVAudioSessionCategoryAmbient;
@@ -149,10 +160,17 @@
   configuration.mode = AVAudioSessionModeDefault;
 
   RTCAudioSession *session = [RTCAudioSession sharedInstance];
-  [session addDelegate:self];
   [session lockForConfiguration];
+  BOOL hasSucceeded = NO;
   NSError *error = nil;
-  if (![session setConfiguration:configuration active:YES error:&error]) {
+  if (session.isActive) {
+    hasSucceeded = [session setConfiguration:configuration error:&error];
+  } else {
+    hasSucceeded = [session setConfiguration:configuration
+                                      active:YES
+                                       error:&error];
+  }
+  if (!hasSucceeded) {
     RTCLogError(@"Error setting configuration: %@", error.localizedDescription);
   }
   [session unlockForConfiguration];
@@ -169,6 +187,14 @@
   [_audioPlayer prepareToPlay];
 }
 
+- (void)restartAudioPlayerIfNeeded {
+  if (_mainView.isAudioLoopPlaying && !self.presentedViewController) {
+    RTCLog(@"Starting audio loop due to WebRTC end.");
+    [self configureAudioSession];
+    [_audioPlayer play];
+  }
+}
+
 - (void)showAlertWithMessage:(NSString*)message {
   UIAlertView* alertView = [[UIAlertView alloc] initWithTitle:nil
                                                       message:message
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDStatsView.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDStatsView.m
index be0bdd5b3c4..f4522423b9f 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDStatsView.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDStatsView.m
@@ -10,7 +10,7 @@
 
 #import "ARDStatsView.h"
 
-#import "webrtc/api/objc/RTCStatsReport.h"
+#import "WebRTC/RTCStatsReport.h"
 
 #import "ARDStatsBuilder.h"
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallView.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallView.h
index 980abd4520f..dec1bfcba06 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallView.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallView.h
@@ -10,8 +10,8 @@
 
 #import <UIKit/UIKit.h>
 
-#import "webrtc/api/objc/RTCEAGLVideoView.h"
-#import "webrtc/base/objc/RTCCameraPreviewView.h"
+#import "WebRTC/RTCCameraPreviewView.h"
+#import "WebRTC/RTCEAGLVideoView.h"
 
 #import "ARDStatsView.h"
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h
index 7dee2d8537b..f3a95548209 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h
@@ -10,10 +10,20 @@
 
 #import <UIKit/UIKit.h>
 
+@class ARDVideoCallViewController;
+@protocol ARDVideoCallViewControllerDelegate <NSObject>
+
+- (void)viewControllerDidFinish:(ARDVideoCallViewController *)viewController;
+
+@end
+
 @interface ARDVideoCallViewController : UIViewController
 
+@property(nonatomic, weak) id<ARDVideoCallViewControllerDelegate> delegate;
+
 - (instancetype)initForRoom:(NSString *)room
                  isLoopback:(BOOL)isLoopback
-                isAudioOnly:(BOOL)isAudioOnly;
+                isAudioOnly:(BOOL)isAudioOnly
+                   delegate:(id<ARDVideoCallViewControllerDelegate>)delegate;
 
 @end
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
index b8786773038..d35e1e6b2d0 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
@@ -10,11 +10,11 @@
 
 #import "ARDVideoCallViewController.h"
 
-#import "webrtc/base/objc/RTCDispatcher.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 
-#import "webrtc/api/objc/RTCAVFoundationVideoSource.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCAVFoundationVideoSource.h"
+#import "WebRTC/RTCDispatcher.h"
+#import "WebRTC/RTCLogging.h"
 
 #import "ARDAppClient.h"
 #import "ARDVideoCallView.h"
@@ -34,11 +34,14 @@
 }
 
 @synthesize videoCallView = _videoCallView;
+@synthesize delegate = _delegate;
 
 - (instancetype)initForRoom:(NSString *)room
                  isLoopback:(BOOL)isLoopback
-                isAudioOnly:(BOOL)isAudioOnly {
+                isAudioOnly:(BOOL)isAudioOnly
+                   delegate:(id<ARDVideoCallViewControllerDelegate>)delegate {
   if (self = [super init]) {
+    _delegate = delegate;
     _client = [[ARDAppClient alloc] initWithDelegate:self];
     [_client connectToRoomWithId:room
                       isLoopback:isLoopback
@@ -177,10 +180,7 @@
   self.remoteVideoTrack = nil;
   self.localVideoTrack = nil;
   [_client disconnect];
-  if (![self isBeingDismissed]) {
-    [self.presentingViewController dismissViewControllerAnimated:YES
-                                                      completion:nil];
-  }
+  [_delegate viewControllerDidFinish:self];
 }
 
 - (void)switchCamera {
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m
index f0b2a04b5f3..e79519acc29 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m
@@ -14,7 +14,7 @@
 
 #import "APPRTCAppDelegate.h"
 
-#import "webrtc/base/objc/RTCSSLAdapter.h"
+#import "WebRTC/RTCSSLAdapter.h"
 
 #import "APPRTCViewController.h"
 
diff --git a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCViewController.m b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCViewController.m
index db7b19c0e52..4d58d2ad347 100644
--- a/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCViewController.m
+++ b/chromium/third_party/webrtc/examples/objc/AppRTCDemo/mac/APPRTCViewController.m
@@ -11,8 +11,9 @@
 #import "APPRTCViewController.h"
 
 #import <AVFoundation/AVFoundation.h>
-#import "webrtc/api/objc/RTCNSGLVideoView.h"
-#import "webrtc/api/objc/RTCVideoTrack.h"
+
+#import "WebRTC/RTCNSGLVideoView.h"
+#import "WebRTC/RTCVideoTrack.h"
 
 #import "ARDAppClient.h"
 
diff --git a/chromium/third_party/webrtc/examples/peerconnection/client/conductor.cc b/chromium/third_party/webrtc/examples/peerconnection/client/conductor.cc
index e26d403248d..8ec6ed9c102 100644
--- a/chromium/third_party/webrtc/examples/peerconnection/client/conductor.cc
+++ b/chromium/third_party/webrtc/examples/peerconnection/client/conductor.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/examples/peerconnection/client/conductor.h"
 
+#include <memory>
 #include <utility>
 #include <vector>
 
@@ -308,7 +309,7 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
       return;
     }
     webrtc::SdpParseError error;
-    rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
+    std::unique_ptr<webrtc::IceCandidateInterface> candidate(
         webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, sdp, &error));
     if (!candidate.get()) {
       LOG(WARNING) << "Can't parse received candidate message. "
diff --git a/chromium/third_party/webrtc/examples/peerconnection/client/conductor.h b/chromium/third_party/webrtc/examples/peerconnection/client/conductor.h
index db2f77b6464..02351b78be1 100644
--- a/chromium/third_party/webrtc/examples/peerconnection/client/conductor.h
+++ b/chromium/third_party/webrtc/examples/peerconnection/client/conductor.h
@@ -21,7 +21,6 @@
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/examples/peerconnection/client/main_wnd.h"
 #include "webrtc/examples/peerconnection/client/peer_connection_client.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 class VideoCaptureModule;
diff --git a/chromium/third_party/webrtc/examples/peerconnection/client/linux/main_wnd.h b/chromium/third_party/webrtc/examples/peerconnection/client/linux/main_wnd.h
index e756fc98c4a..3c718577d10 100644
--- a/chromium/third_party/webrtc/examples/peerconnection/client/linux/main_wnd.h
+++ b/chromium/third_party/webrtc/examples/peerconnection/client/linux/main_wnd.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_
 #define WEBRTC_EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/examples/peerconnection/client/main_wnd.h"
@@ -92,7 +93,7 @@ class GtkMainWnd : public MainWindow {
 
    protected:
     void SetSize(int width, int height);
-    rtc::scoped_ptr<uint8_t[]> image_;
+    std::unique_ptr<uint8_t[]> image_;
     int width_;
     int height_;
     GtkMainWnd* main_wnd_;
@@ -111,9 +112,9 @@ class GtkMainWnd : public MainWindow {
   std::string port_;
   bool autoconnect_;
   bool autocall_;
-  rtc::scoped_ptr<VideoRenderer> local_renderer_;
-  rtc::scoped_ptr<VideoRenderer> remote_renderer_;
-  rtc::scoped_ptr<uint8_t[]> draw_buffer_;
+  std::unique_ptr<VideoRenderer> local_renderer_;
+  std::unique_ptr<VideoRenderer> remote_renderer_;
+  std::unique_ptr<uint8_t[]> draw_buffer_;
   int draw_buffer_size_;
 };
 
diff --git a/chromium/third_party/webrtc/examples/peerconnection/client/main_wnd.h b/chromium/third_party/webrtc/examples/peerconnection/client/main_wnd.h
index 80db2a5adcf..03db80d0b6d 100644
--- a/chromium/third_party/webrtc/examples/peerconnection/client/main_wnd.h
+++ b/chromium/third_party/webrtc/examples/peerconnection/client/main_wnd.h
@@ -13,6 +13,7 @@
 #pragma once
 
 #include <map>
+#include <memory>
 #include <string>
 
 #include "webrtc/api/mediastreaminterface.h"
@@ -131,7 +132,7 @@ class MainWnd : public MainWindow {
 
     HWND wnd_;
     BITMAPINFO bmi_;
-    rtc::scoped_ptr<uint8_t[]> image_;
+    std::unique_ptr<uint8_t[]> image_;
     CRITICAL_SECTION buffer_lock_;
     rtc::scoped_refptr<webrtc::VideoTrackInterface> rendered_track_;
   };
@@ -176,8 +177,8 @@ class MainWnd : public MainWindow {
   void HandleTabbing();
 
  private:
-  rtc::scoped_ptr<VideoRenderer> local_renderer_;
-  rtc::scoped_ptr<VideoRenderer> remote_renderer_;
+  std::unique_ptr<VideoRenderer> local_renderer_;
+  std::unique_ptr<VideoRenderer> remote_renderer_;
   UI ui_;
   HWND wnd_;
   DWORD ui_thread_id_;
diff --git a/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.h b/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.h
index b7abfdfe18e..dbf2d8ff3d2 100644
--- a/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.h
+++ b/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.h
@@ -13,11 +13,11 @@
 #pragma once
 
 #include <map>
+#include <memory>
 #include <string>
 
 #include "webrtc/base/nethelpers.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/signalthread.h"
 #include "webrtc/base/sigslot.h"
 
@@ -109,8 +109,8 @@ class PeerConnectionClient : public sigslot::has_slots<>,
   PeerConnectionClientObserver* callback_;
   rtc::SocketAddress server_address_;
   rtc::AsyncResolver* resolver_;
-  rtc::scoped_ptr<rtc::AsyncSocket> control_socket_;
-  rtc::scoped_ptr<rtc::AsyncSocket> hanging_get_;
+  std::unique_ptr<rtc::AsyncSocket> control_socket_;
+  std::unique_ptr<rtc::AsyncSocket> hanging_get_;
   std::string onconnect_data_;
   std::string control_data_;
   std::string notification_data_;
diff --git a/chromium/third_party/webrtc/examples/relayserver/relayserver_main.cc b/chromium/third_party/webrtc/examples/relayserver/relayserver_main.cc
index 31f43c4d15a..c14ea459c86 100644
--- a/chromium/third_party/webrtc/examples/relayserver/relayserver_main.cc
+++ b/chromium/third_party/webrtc/examples/relayserver/relayserver_main.cc
@@ -9,9 +9,9 @@
  */
 
 #include <iostream>  // NOLINT
+#include <memory>
 
 #include "webrtc/p2p/base/relayserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 
 int main(int argc, char **argv) {
@@ -35,7 +35,7 @@ int main(int argc, char **argv) {
 
   rtc::Thread *pthMain = rtc::Thread::Current();
 
-  rtc::scoped_ptr<rtc::AsyncUDPSocket> int_socket(
+  std::unique_ptr<rtc::AsyncUDPSocket> int_socket(
       rtc::AsyncUDPSocket::Create(pthMain->socketserver(), int_addr));
   if (!int_socket) {
     std::cerr << "Failed to create a UDP socket bound at"
@@ -43,7 +43,7 @@ int main(int argc, char **argv) {
     return 1;
   }
 
-  rtc::scoped_ptr<rtc::AsyncUDPSocket> ext_socket(
+  std::unique_ptr<rtc::AsyncUDPSocket> ext_socket(
       rtc::AsyncUDPSocket::Create(pthMain->socketserver(), ext_addr));
   if (!ext_socket) {
     std::cerr << "Failed to create a UDP socket bound at"
diff --git a/chromium/third_party/webrtc/libjingle/xmllite/xmlbuilder.h b/chromium/third_party/webrtc/libjingle/xmllite/xmlbuilder.h
index 354bf0b70a5..3b48f282479 100644
--- a/chromium/third_party/webrtc/libjingle/xmllite/xmlbuilder.h
+++ b/chromium/third_party/webrtc/libjingle/xmllite/xmlbuilder.h
@@ -11,10 +11,10 @@
 #ifndef _xmlbuilder_h_
 #define _xmlbuilder_h_
 
+#include <memory>
 #include <string>
 #include <vector>
 #include "webrtc/libjingle/xmllite/xmlparser.h"
-#include "webrtc/base/scoped_ptr.h"
 
 #ifdef EXPAT_RELATIVE_PATH
 #include "expat.h"
@@ -52,8 +52,8 @@ public:
 
 private:
   XmlElement * pelCurrent_;
-  rtc::scoped_ptr<XmlElement> pelRoot_;
-  rtc::scoped_ptr<std::vector<XmlElement*> > pvParents_;
+  std::unique_ptr<XmlElement> pelRoot_;
+  std::unique_ptr<std::vector<XmlElement*> > pvParents_;
 };
 
 }
diff --git a/chromium/third_party/webrtc/libjingle/xmllite/xmlelement.h b/chromium/third_party/webrtc/libjingle/xmllite/xmlelement.h
index 70c6f799230..37adfc01707 100644
--- a/chromium/third_party/webrtc/libjingle/xmllite/xmlelement.h
+++ b/chromium/third_party/webrtc/libjingle/xmllite/xmlelement.h
@@ -15,7 +15,6 @@
 #include <string>
 
 #include "webrtc/libjingle/xmllite/qname.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 
diff --git a/chromium/third_party/webrtc/libjingle/xmllite/xmlnsstack.h b/chromium/third_party/webrtc/libjingle/xmllite/xmlnsstack.h
index 174f8aefc05..64e2f6ee66f 100644
--- a/chromium/third_party/webrtc/libjingle/xmllite/xmlnsstack.h
+++ b/chromium/third_party/webrtc/libjingle/xmllite/xmlnsstack.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_LIBJINGLE_XMLLITE_XMLNSSTACK_H_
 #define WEBRTC_LIBJINGLE_XMLLITE_XMLNSSTACK_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 #include "webrtc/libjingle/xmllite/qname.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 
@@ -37,8 +37,8 @@ public:
 
 private:
 
-  rtc::scoped_ptr<std::vector<std::string> > pxmlnsStack_;
-  rtc::scoped_ptr<std::vector<size_t> > pxmlnsDepthStack_;
+  std::unique_ptr<std::vector<std::string> > pxmlnsStack_;
+  std::unique_ptr<std::vector<size_t> > pxmlnsDepthStack_;
 };
 }
 
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/chatroommoduleimpl.cc b/chromium/third_party/webrtc/libjingle/xmpp/chatroommoduleimpl.cc
index 52fba4cbae5..4feb5bc9b8c 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/chatroommoduleimpl.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/chatroommoduleimpl.cc
@@ -11,6 +11,7 @@
 #include <algorithm>
 #include <iostream>
 #include <map>
+#include <memory>
 #include <sstream>
 #include <string>
 #include <vector>
@@ -105,7 +106,7 @@ public:
   const XmppPresence* presence() const;
 
 private:
-  rtc::scoped_ptr<XmppPresence>  presence_;
+  std::unique_ptr<XmppPresence>  presence_;
 };
 
 class XmppChatroomMemberEnumeratorImpl :
@@ -430,7 +431,7 @@ void
 XmppChatroomModuleImpl::FireEnteredStatus(const XmlElement* presence,
                                           XmppChatroomEnteredStatus status) {
   if (chatroom_handler_) {
-    rtc::scoped_ptr<XmppPresence> xmpp_presence(XmppPresence::Create());
+    std::unique_ptr<XmppPresence> xmpp_presence(XmppPresence::Create());
     xmpp_presence->set_raw_xml(presence);
     chatroom_handler_->ChatroomEnteredStatus(this, xmpp_presence.get(), status);
   }
@@ -472,7 +473,7 @@ XmppReturnStatus
 XmppChatroomModuleImpl::ServerChangedOtherPresence(const XmlElement&
                                                    presence_element) {
   XmppReturnStatus xmpp_status = XMPP_RETURN_OK;
-  rtc::scoped_ptr<XmppPresence> presence(XmppPresence::Create());
+  std::unique_ptr<XmppPresence> presence(XmppPresence::Create());
   IFR(presence->set_raw_xml(&presence_element));
 
   JidMemberMap::iterator pos = chatroom_jid_members_.find(presence->jid());
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/discoitemsquerytask.cc b/chromium/third_party/webrtc/libjingle/xmpp/discoitemsquerytask.cc
index 765ee143978..10f6e1d7452 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/discoitemsquerytask.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/discoitemsquerytask.cc
@@ -11,7 +11,6 @@
 #include "webrtc/libjingle/xmpp/constants.h"
 #include "webrtc/libjingle/xmpp/discoitemsquerytask.h"
 #include "webrtc/libjingle/xmpp/xmpptask.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/fakexmppclient.h b/chromium/third_party/webrtc/libjingle/xmpp/fakexmppclient.h
index 453a7c86f16..63c216caf21 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/fakexmppclient.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/fakexmppclient.h
@@ -13,6 +13,7 @@
 #ifndef WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_
 #define WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_
 
+#include <algorithm>
 #include <string>
 #include <vector>
 
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient.h b/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient.h
index 2586768e2bd..fecc727604f 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient.h
@@ -12,13 +12,13 @@
 #define WEBRTC_LIBJINGLE_XMPP_HANGOUTPUBSUBCLIENT_H_
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/libjingle/xmpp/jid.h"
 #include "webrtc/libjingle/xmpp/pubsubclient.h"
 #include "webrtc/libjingle/xmpp/pubsubstateclient.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sigslotrepeater.h"
 
@@ -163,14 +163,14 @@ class HangoutPubSubClient : public sigslot::has_slots<> {
                                 const XmlElement* stanza);
   Jid mucjid_;
   std::string nick_;
-  rtc::scoped_ptr<PubSubClient> media_client_;
-  rtc::scoped_ptr<PubSubClient> presenter_client_;
-  rtc::scoped_ptr<PubSubStateClient<bool> > presenter_state_client_;
-  rtc::scoped_ptr<PubSubStateClient<bool> > audio_mute_state_client_;
-  rtc::scoped_ptr<PubSubStateClient<bool> > video_mute_state_client_;
-  rtc::scoped_ptr<PubSubStateClient<bool> > video_pause_state_client_;
-  rtc::scoped_ptr<PubSubStateClient<bool> > recording_state_client_;
-  rtc::scoped_ptr<PubSubStateClient<bool> > media_block_state_client_;
+  std::unique_ptr<PubSubClient> media_client_;
+  std::unique_ptr<PubSubClient> presenter_client_;
+  std::unique_ptr<PubSubStateClient<bool> > presenter_state_client_;
+  std::unique_ptr<PubSubStateClient<bool> > audio_mute_state_client_;
+  std::unique_ptr<PubSubStateClient<bool> > video_mute_state_client_;
+  std::unique_ptr<PubSubStateClient<bool> > video_pause_state_client_;
+  std::unique_ptr<PubSubStateClient<bool> > recording_state_client_;
+  std::unique_ptr<PubSubStateClient<bool> > media_block_state_client_;
 };
 
 }  // namespace buzz
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient_unittest.cc b/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient_unittest.cc
index 7c6ea58f2d2..dd706b6a9a8 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient_unittest.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/hangoutpubsubclient_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/libjingle/xmllite/qname.h"
@@ -228,11 +229,11 @@ class HangoutPubSubClientTest : public testing::Test {
         listener.get(), &TestHangoutPubSubListener::OnMediaBlockError);
   }
 
-  rtc::scoped_ptr<rtc::FakeTaskRunner> runner;
+  std::unique_ptr<rtc::FakeTaskRunner> runner;
   // xmpp_client deleted by deleting runner.
   buzz::FakeXmppClient* xmpp_client;
-  rtc::scoped_ptr<buzz::HangoutPubSubClient> client;
-  rtc::scoped_ptr<TestHangoutPubSubListener> listener;
+  std::unique_ptr<buzz::HangoutPubSubClient> client;
+  std::unique_ptr<TestHangoutPubSubListener> listener;
   buzz::Jid pubsubjid;
   std::string nick;
 };
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/iqtask.h b/chromium/third_party/webrtc/libjingle/xmpp/iqtask.h
index 1d50c383947..0db00711e86 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/iqtask.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/iqtask.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_LIBJINGLE_XMPP_IQTASK_H_
 #define WEBRTC_LIBJINGLE_XMPP_IQTASK_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/libjingle/xmpp/xmppengine.h"
@@ -40,7 +41,7 @@ class IqTask : public XmppTask {
   virtual int OnTimeout();
 
   Jid to_;
-  rtc::scoped_ptr<XmlElement> stanza_;
+  std::unique_ptr<XmlElement> stanza_;
 };
 
 }  // namespace buzz
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/jingleinfotask.cc b/chromium/third_party/webrtc/libjingle/xmpp/jingleinfotask.cc
index a5a07121bd1..5599edc0eb3 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/jingleinfotask.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/jingleinfotask.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/libjingle/xmpp/jingleinfotask.h"
 
+#include <memory>
+
 #include "webrtc/libjingle/xmpp/constants.h"
 #include "webrtc/libjingle/xmpp/xmppclient.h"
 #include "webrtc/libjingle/xmpp/xmpptask.h"
@@ -24,7 +26,7 @@ class JingleInfoTask::JingleInfoGetTask : public XmppTask {
         done_(false) {}
 
   virtual int ProcessStart() {
-    rtc::scoped_ptr<XmlElement> get(
+    std::unique_ptr<XmlElement> get(
         MakeIq(STR_GET, Jid(), task_id()));
     get->AddElement(new XmlElement(QN_JINGLE_INFO_QUERY, true));
     if (SendStanza(get.get()) != XMPP_RETURN_OK) {
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/mucroomconfigtask.cc b/chromium/third_party/webrtc/libjingle/xmpp/mucroomconfigtask.cc
index 08b10650a96..5938a2ca722 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/mucroomconfigtask.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/mucroomconfigtask.cc
@@ -14,7 +14,6 @@
 #include "webrtc/libjingle/xmpp/mucroomconfigtask.h"
 
 #include "webrtc/libjingle/xmpp/constants.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/mucroomlookuptask.cc b/chromium/third_party/webrtc/libjingle/xmpp/mucroomlookuptask.cc
index 8c0a4d78561..a52491c9184 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/mucroomlookuptask.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/mucroomlookuptask.cc
@@ -12,7 +12,6 @@
 
 #include "webrtc/libjingle/xmpp/constants.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 
 namespace buzz {
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/pingtask.cc b/chromium/third_party/webrtc/libjingle/xmpp/pingtask.cc
index 479dc23ff5c..ad4566a51c3 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/pingtask.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/pingtask.cc
@@ -10,9 +10,10 @@
 
 #include "webrtc/libjingle/xmpp/pingtask.h"
 
+#include <memory>
+
 #include "webrtc/libjingle/xmpp/constants.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 
@@ -56,7 +57,7 @@ int PingTask::ProcessStart() {
     ping_response_deadline_ = 0;
   }
 
-  uint32_t now = rtc::Time();
+  int64_t now = rtc::TimeMillis();
 
   // If the ping timed out, signal.
   if (ping_response_deadline_ != 0 && now >= ping_response_deadline_) {
@@ -66,7 +67,7 @@ int PingTask::ProcessStart() {
 
   // Send a ping if it's time.
   if (now >= next_ping_time_) {
-    rtc::scoped_ptr<buzz::XmlElement> stanza(
+    std::unique_ptr<buzz::XmlElement> stanza(
         MakeIq(buzz::STR_GET, Jid(STR_EMPTY), task_id()));
     stanza->AddElement(new buzz::XmlElement(QN_PING));
     SendStanza(stanza.get());
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/pingtask.h b/chromium/third_party/webrtc/libjingle/xmpp/pingtask.h
index 22fd94d7217..b070a1ccf1e 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/pingtask.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/pingtask.h
@@ -46,8 +46,8 @@ class PingTask : public buzz::XmppTask, private rtc::MessageHandler {
   rtc::MessageQueue* message_queue_;
   uint32_t ping_period_millis_;
   uint32_t ping_timeout_millis_;
-  uint32_t next_ping_time_;
-  uint32_t ping_response_deadline_;  // 0 if the response has been received
+  int64_t next_ping_time_;
+  int64_t ping_response_deadline_;  // 0 if the response has been received
 };
 
 } // namespace buzz
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/pubsub_task.cc b/chromium/third_party/webrtc/libjingle/xmpp/pubsub_task.cc
index f30c0518144..812cb95832c 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/pubsub_task.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/pubsub_task.cc
@@ -11,6 +11,7 @@
 #include "webrtc/libjingle/xmpp/pubsub_task.h"
 
 #include <map>
+#include <memory>
 #include <string>
 
 #include "webrtc/libjingle/xmpp/constants.h"
@@ -82,7 +83,7 @@ int PubsubTask::ProcessResponse() {
 bool PubsubTask::SubscribeToNode(const std::string& pubsub_node,
                                  NodeHandler handler) {
   subscribed_nodes_[pubsub_node] = handler;
-  rtc::scoped_ptr<buzz::XmlElement> get_iq_request(
+  std::unique_ptr<buzz::XmlElement> get_iq_request(
       MakeIq(buzz::STR_GET, pubsub_node_jid_, task_id()));
   if (!get_iq_request) {
     return false;
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/pubsubclient_unittest.cc b/chromium/third_party/webrtc/libjingle/xmpp/pubsubclient_unittest.cc
index 3815ef8a509..ab4c853e565 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/pubsubclient_unittest.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/pubsubclient_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/libjingle/xmllite/qname.h"
@@ -103,11 +104,11 @@ class PubSubClientTest : public testing::Test {
         listener.get(), &TestPubSubItemsListener::OnRetractError);
   }
 
-  rtc::scoped_ptr<rtc::FakeTaskRunner> runner;
+  std::unique_ptr<rtc::FakeTaskRunner> runner;
   // xmpp_client deleted by deleting runner.
   buzz::FakeXmppClient* xmpp_client;
-  rtc::scoped_ptr<buzz::PubSubClient> client;
-  rtc::scoped_ptr<TestPubSubItemsListener> listener;
+  std::unique_ptr<buzz::PubSubClient> client;
+  std::unique_ptr<TestPubSubItemsListener> listener;
   buzz::Jid pubsubjid;
   std::string node;
   std::string itemid;
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/pubsubstateclient.h b/chromium/third_party/webrtc/libjingle/xmpp/pubsubstateclient.h
index ffb794af4c0..07aa26dbad5 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/pubsubstateclient.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/pubsubstateclient.h
@@ -12,6 +12,7 @@
 #define WEBRTC_LIBJINGLE_XMPP_PUBSUBSTATECLIENT_H_
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -20,7 +21,7 @@
 #include "webrtc/libjingle/xmpp/constants.h"
 #include "webrtc/libjingle/xmpp/jid.h"
 #include "webrtc/libjingle/xmpp/pubsubclient.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sigslotrepeater.h"
 
@@ -256,8 +257,8 @@ class PubSubStateClient : public sigslot::has_slots<> {
   PubSubClient* client_;
   const QName state_name_;
   C default_state_;
-  rtc::scoped_ptr<PubSubStateKeySerializer> key_serializer_;
-  rtc::scoped_ptr<PubSubStateSerializer<C> > state_serializer_;
+  std::unique_ptr<PubSubStateKeySerializer> key_serializer_;
+  std::unique_ptr<PubSubStateSerializer<C> > state_serializer_;
   // key => state
   std::map<std::string, C> state_by_key_;
   // itemid => StateItemInfo
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/pubsubtasks_unittest.cc b/chromium/third_party/webrtc/libjingle/xmpp/pubsubtasks_unittest.cc
index 8062e58e18f..79e656bb0bb 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/pubsubtasks_unittest.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/pubsubtasks_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/libjingle/xmllite/qname.h"
@@ -80,10 +81,10 @@ class PubSubTasksTest : public testing::Test {
     listener.reset(new TestPubSubTasksListener());
   }
 
-  rtc::scoped_ptr<rtc::FakeTaskRunner> runner;
+  std::unique_ptr<rtc::FakeTaskRunner> runner;
   // Client deleted by deleting runner.
   buzz::FakeXmppClient* client;
-  rtc::scoped_ptr<TestPubSubTasksListener> listener;
+  std::unique_ptr<TestPubSubTasksListener> listener;
   buzz::Jid pubsubjid;
   std::string node;
   std::string itemid;
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/rostermodule_unittest.cc b/chromium/third_party/webrtc/libjingle/xmpp/rostermodule_unittest.cc
index 1ae6c226a57..18f20253572 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/rostermodule_unittest.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/rostermodule_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include <iostream>
+#include <memory>
 #include <sstream>
 #include <string>
 
@@ -18,7 +19,6 @@
 #include "webrtc/libjingle/xmpp/util_unittest.h"
 #include "webrtc/libjingle/xmpp/xmppengine.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 
 #define TEST_OK(x) EXPECT_EQ((x),XMPP_RETURN_OK)
 #define TEST_BADARGUMENT(x) EXPECT_EQ((x),XMPP_RETURN_BADARGUMENT)
@@ -250,7 +250,7 @@ TEST_F(RosterModuleTest, TestPresence) {
   status->AddAttr(QN_STATUS, STR_PSTN_CONFERENCE_STATUS_CONNECTING);
   XmlElement presence_xml(QN_PRESENCE);
   presence_xml.AddElement(status);
-  rtc::scoped_ptr<XmppPresence> presence(XmppPresence::Create());
+  std::unique_ptr<XmppPresence> presence(XmppPresence::Create());
   presence->set_raw_xml(&presence_xml);
   EXPECT_EQ(presence->connection_status(), XMPP_CONNECTION_STATUS_CONNECTING);
 }
@@ -258,11 +258,11 @@ TEST_F(RosterModuleTest, TestPresence) {
 TEST_F(RosterModuleTest, TestOutgoingPresence) {
   std::stringstream dump;
 
-  rtc::scoped_ptr<XmppEngine> engine(XmppEngine::Create());
+  std::unique_ptr<XmppEngine> engine(XmppEngine::Create());
   XmppTestHandler handler(engine.get());
   XmppTestRosterHandler roster_handler;
 
-  rtc::scoped_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
+  std::unique_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
   roster->set_roster_handler(&roster_handler);
 
   // Configure the roster module
@@ -364,7 +364,7 @@ TEST_F(RosterModuleTest, TestOutgoingPresence) {
   EXPECT_EQ(handler.SessionActivity(), "");
 
   // Construct a directed presence
-  rtc::scoped_ptr<XmppPresence> directed_presence(XmppPresence::Create());
+  std::unique_ptr<XmppPresence> directed_presence(XmppPresence::Create());
   TEST_OK(directed_presence->set_available(XMPP_PRESENCE_AVAILABLE));
   TEST_OK(directed_presence->set_priority(120));
   TEST_OK(directed_presence->set_status("*very* available"));
@@ -381,11 +381,11 @@ TEST_F(RosterModuleTest, TestOutgoingPresence) {
 }
 
 TEST_F(RosterModuleTest, TestIncomingPresence) {
-  rtc::scoped_ptr<XmppEngine> engine(XmppEngine::Create());
+  std::unique_ptr<XmppEngine> engine(XmppEngine::Create());
   XmppTestHandler handler(engine.get());
   XmppTestRosterHandler roster_handler;
 
-  rtc::scoped_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
+  std::unique_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
   roster->set_roster_handler(&roster_handler);
 
   // Configure the roster module
@@ -513,11 +513,11 @@ TEST_F(RosterModuleTest, TestIncomingPresence) {
 }
 
 TEST_F(RosterModuleTest, TestPresenceSubscription) {
-  rtc::scoped_ptr<XmppEngine> engine(XmppEngine::Create());
+  std::unique_ptr<XmppEngine> engine(XmppEngine::Create());
   XmppTestHandler handler(engine.get());
   XmppTestRosterHandler roster_handler;
 
-  rtc::scoped_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
+  std::unique_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
   roster->set_roster_handler(&roster_handler);
 
   // Configure the roster module
@@ -576,11 +576,11 @@ TEST_F(RosterModuleTest, TestPresenceSubscription) {
 }
 
 TEST_F(RosterModuleTest, TestRosterReceive) {
-  rtc::scoped_ptr<XmppEngine> engine(XmppEngine::Create());
+  std::unique_ptr<XmppEngine> engine(XmppEngine::Create());
   XmppTestHandler handler(engine.get());
   XmppTestRosterHandler roster_handler;
 
-  rtc::scoped_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
+  std::unique_ptr<XmppRosterModule> roster(XmppRosterModule::Create());
   roster->set_roster_handler(&roster_handler);
 
   // Configure the roster module
@@ -696,7 +696,7 @@ TEST_F(RosterModuleTest, TestRosterReceive) {
   EXPECT_EQ(handler.SessionActivity(), "");
 
   // Request that someone be added
-  rtc::scoped_ptr<XmppRosterContact> contact(XmppRosterContact::Create());
+  std::unique_ptr<XmppRosterContact> contact(XmppRosterContact::Create());
   TEST_OK(contact->set_jid(Jid("brandt@example.net")));
   TEST_OK(contact->set_name("Brandt"));
   TEST_OK(contact->AddGroup("Business Partners"));
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/rostermoduleimpl.h b/chromium/third_party/webrtc/libjingle/xmpp/rostermoduleimpl.h
index 6e3bd91c8a2..772692bd65b 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/rostermoduleimpl.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/rostermoduleimpl.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_LIBJINGLE_XMPP_XMPPTHREAD_H_
 #define WEBRTC_LIBJINGLE_XMPP_XMPPTHREAD_H_
 
+#include <memory>
+
 #include "webrtc/libjingle/xmpp/moduleimpl.h"
 #include "webrtc/libjingle/xmpp/rostermodule.h"
 
@@ -86,7 +88,7 @@ private:
 
   // Store everything in the XML element. If this becomes a perf issue we can
   // cache the data.
-  rtc::scoped_ptr<XmlElement> raw_xml_;
+  std::unique_ptr<XmlElement> raw_xml_;
 };
 
 //! A contact as given by the server
@@ -151,7 +153,7 @@ private:
   int group_count_;
   int group_index_returned_;
   XmlElement * group_returned_;
-  rtc::scoped_ptr<XmlElement> raw_xml_;
+  std::unique_ptr<XmlElement> raw_xml_;
 };
 
 //! An XmppModule for handle roster and presence functionality
@@ -273,11 +275,11 @@ private:
 
   typedef std::vector<XmppPresenceImpl*> PresenceVector;
   typedef std::map<Jid, PresenceVector*> JidPresenceVectorMap;
-  rtc::scoped_ptr<JidPresenceVectorMap> incoming_presence_map_;
-  rtc::scoped_ptr<PresenceVector> incoming_presence_vector_;
+  std::unique_ptr<JidPresenceVectorMap> incoming_presence_map_;
+  std::unique_ptr<PresenceVector> incoming_presence_vector_;
 
   typedef std::vector<XmppRosterContactImpl*> ContactVector;
-  rtc::scoped_ptr<ContactVector> contacts_;
+  std::unique_ptr<ContactVector> contacts_;
 };
 
 }
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmpp.gyp b/chromium/third_party/webrtc/libjingle/xmpp/xmpp.gyp
index 4dc75f7b52e..37b75122b4d 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmpp.gyp
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmpp.gyp
@@ -19,9 +19,6 @@
       'defines': [
         'FEATURE_ENABLE_SSL',
       ],
-      'cflags_cc!': [
-        '-Wnon-virtual-dtor',
-      ],
       'sources': [
         'asyncsocket.h',
         'chatroommodule.h',
@@ -98,9 +95,6 @@
         'xmppthread.h',
       ],
       'direct_dependent_settings': {
-        'cflags_cc!': [
-          '-Wnon-virtual-dtor',
-        ],
         'defines': [
           'FEATURE_ENABLE_SSL',
           'FEATURE_ENABLE_VOICEMAIL',
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.cc b/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.cc
index f7e88c39278..a87659824bb 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.cc
@@ -15,7 +15,6 @@
 #include "webrtc/libjingle/xmpp/prexmppauth.h"
 #include "webrtc/libjingle/xmpp/saslplainmechanism.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/stringutils.h"
 #include "xmpptask.h"
@@ -48,9 +47,9 @@ public:
   XmppClient* const client_;
 
   // the two main objects
-  rtc::scoped_ptr<AsyncSocket> socket_;
-  rtc::scoped_ptr<XmppEngine> engine_;
-  rtc::scoped_ptr<PreXmppAuth> pre_auth_;
+  std::unique_ptr<AsyncSocket> socket_;
+  std::unique_ptr<XmppEngine> engine_;
+  std::unique_ptr<PreXmppAuth> pre_auth_;
   rtc::CryptString pass_;
   std::string auth_mechanism_;
   std::string auth_token_;
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.h b/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.h
index 84ca6c1418d..dba591cb44a 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmppclient.h
@@ -11,7 +11,9 @@
 #ifndef WEBRTC_LIBJINGLE_XMPP_XMPPCLIENT_H_
 #define WEBRTC_LIBJINGLE_XMPP_XMPPCLIENT_H_
 
+#include <memory>
 #include <string>
+
 #include "webrtc/libjingle/xmpp/asyncsocket.h"
 #include "webrtc/libjingle/xmpp/xmppclientsettings.h"
 #include "webrtc/libjingle/xmpp/xmppengine.h"
@@ -136,7 +138,7 @@ public:
 
   class Private;
   friend class Private;
-  rtc::scoped_ptr<Private> d_;
+  std::unique_ptr<Private> d_;
 
   bool delivering_signal_;
   bool valid_;
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmppengine_unittest.cc b/chromium/third_party/webrtc/libjingle/xmpp/xmppengine_unittest.cc
index 7af151ba48d..d5afead660e 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmppengine_unittest.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmppengine_unittest.cc
@@ -9,8 +9,10 @@
  */
 
 #include <iostream>
+#include <memory>
 #include <sstream>
 #include <string>
+
 #include "webrtc/libjingle/xmllite/xmlelement.h"
 #include "webrtc/libjingle/xmpp/constants.h"
 #include "webrtc/libjingle/xmpp/plainsaslhandler.h"
@@ -77,8 +79,8 @@ class XmppEngineTest : public testing::Test {
   void RunLogin();
 
  private:
-  rtc::scoped_ptr<XmppEngine> engine_;
-  rtc::scoped_ptr<XmppTestHandler> handler_;
+  std::unique_ptr<XmppEngine> engine_;
+  std::unique_ptr<XmppTestHandler> handler_;
 };
 
 void XmppEngineTest::RunLogin() {
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmppengineimpl.h b/chromium/third_party/webrtc/libjingle/xmpp/xmppengineimpl.h
index c322596c46f..e1f3061f90d 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmppengineimpl.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmppengineimpl.h
@@ -11,8 +11,10 @@
 #ifndef WEBRTC_LIBJINGLE_XMPP_XMPPENGINEIMPL_H_
 #define WEBRTC_LIBJINGLE_XMPP_XMPPENGINEIMPL_H_
 
+#include <memory>
 #include <sstream>
 #include <vector>
+
 #include "webrtc/libjingle/xmpp/xmppengine.h"
 #include "webrtc/libjingle/xmpp/xmppstanzaparser.h"
 
@@ -233,7 +235,7 @@ class XmppEngineImpl : public XmppEngine {
   TlsOptions tls_option_;
   std::string tls_server_hostname_;
   std::string tls_server_domain_;
-  rtc::scoped_ptr<XmppLoginTask> login_task_;
+  std::unique_ptr<XmppLoginTask> login_task_;
   std::string lang_;
 
   int next_id_;
@@ -242,7 +244,7 @@ class XmppEngineImpl : public XmppEngine {
   bool encrypted_;
   Error error_code_;
   int subcode_;
-  rtc::scoped_ptr<XmlElement> stream_error_;
+  std::unique_ptr<XmlElement> stream_error_;
   bool raised_reset_;
   XmppOutputHandler* output_handler_;
   XmppSessionHandler* session_handler_;
@@ -250,14 +252,14 @@ class XmppEngineImpl : public XmppEngine {
   XmlnsStack xmlns_stack_;
 
   typedef std::vector<XmppStanzaHandler*> StanzaHandlerVector;
-  rtc::scoped_ptr<StanzaHandlerVector> stanza_handlers_[HL_COUNT];
+  std::unique_ptr<StanzaHandlerVector> stanza_handlers_[HL_COUNT];
 
   typedef std::vector<XmppIqEntry*> IqEntryVector;
-  rtc::scoped_ptr<IqEntryVector> iq_entries_;
+  std::unique_ptr<IqEntryVector> iq_entries_;
 
-  rtc::scoped_ptr<SaslHandler> sasl_handler_;
+  std::unique_ptr<SaslHandler> sasl_handler_;
 
-  rtc::scoped_ptr<std::stringstream> output_;
+  std::unique_ptr<std::stringstream> output_;
 };
 
 }  // namespace buzz
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask.h b/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask.h
index f69a648394e..c015c2e7a1c 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask.h
@@ -11,13 +11,13 @@
 #ifndef WEBRTC_LIBJINGLE_XMPP_LOGINTASK_H_
 #define WEBRTC_LIBJINGLE_XMPP_LOGINTASK_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/libjingle/xmpp/jid.h"
 #include "webrtc/libjingle/xmpp/xmppengine.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 
@@ -70,12 +70,12 @@ private:
   const XmlElement * pelStanza_;
   bool isStart_;
   std::string iqId_;
-  rtc::scoped_ptr<XmlElement> pelFeatures_;
+  std::unique_ptr<XmlElement> pelFeatures_;
   Jid fullJid_;
   std::string streamId_;
-  rtc::scoped_ptr<std::vector<XmlElement *> > pvecQueuedStanzas_;
+  std::unique_ptr<std::vector<XmlElement *> > pvecQueuedStanzas_;
 
-  rtc::scoped_ptr<SaslMechanism> sasl_mech_;
+  std::unique_ptr<SaslMechanism> sasl_mech_;
 
 #if !defined(NDEBUG)
   static const rtc::ConstantLabel LOGINTASK_STATES[];
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask_unittest.cc b/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask_unittest.cc
index 221cbdeaf5e..18fce97645b 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask_unittest.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmpplogintask_unittest.cc
@@ -9,8 +9,10 @@
  */
 
 #include <iostream>
+#include <memory>
 #include <sstream>
 #include <string>
+
 #include "webrtc/libjingle/xmllite/xmlelement.h"
 #include "webrtc/libjingle/xmpp/constants.h"
 #include "webrtc/libjingle/xmpp/plainsaslhandler.h"
@@ -20,6 +22,7 @@
 #include "webrtc/base/common.h"
 #include "webrtc/base/cryptstring.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/typedefs.h"
 
 using buzz::Jid;
 using buzz::QName;
@@ -67,8 +70,8 @@ class XmppLoginTaskTest : public testing::Test {
   void SetTlsOptions(buzz::TlsOptions option);
 
  private:
-  rtc::scoped_ptr<XmppEngine> engine_;
-  rtc::scoped_ptr<XmppTestHandler> handler_;
+  std::unique_ptr<XmppEngine> engine_;
+  std::unique_ptr<XmppTestHandler> handler_;
 };
 
 void XmppLoginTaskTest::SetTlsOptions(buzz::TlsOptions option) {
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmpppump.cc b/chromium/third_party/webrtc/libjingle/xmpp/xmpppump.cc
index a428ffa4dc4..4412483cb83 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmpppump.cc
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmpppump.cc
@@ -50,7 +50,7 @@ void XmppPump::WakeTasks() {
 }
 
 int64_t XmppPump::CurrentTime() {
-  return (int64_t)rtc::Time();
+  return (int64_t)rtc::TimeMillis();
 }
 
 void XmppPump::OnMessage(rtc::Message *pmsg) {
diff --git a/chromium/third_party/webrtc/libjingle/xmpp/xmpptask.h b/chromium/third_party/webrtc/libjingle/xmpp/xmpptask.h
index 36351b7fa8f..3f3a39c791c 100644
--- a/chromium/third_party/webrtc/libjingle/xmpp/xmpptask.h
+++ b/chromium/third_party/webrtc/libjingle/xmpp/xmpptask.h
@@ -12,8 +12,11 @@
 #define WEBRTC_LIBJINGLE_XMPP_XMPPTASK_H_
 
 #include <deque>
+#include <memory>
 #include <string>
+
 #include "webrtc/libjingle/xmpp/xmppengine.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/task.h"
 #include "webrtc/base/taskparent.h"
@@ -159,7 +162,7 @@ private:
 
   bool stopped_;
   std::deque<XmlElement*> stanza_queue_;
-  rtc::scoped_ptr<XmlElement> next_stanza_;
+  std::unique_ptr<XmlElement> next_stanza_;
   std::string id_;
 
 #if !defined(NDEBUG)
diff --git a/chromium/third_party/webrtc/media/base/audioframe.h b/chromium/third_party/webrtc/media/base/audioframe.h
deleted file mode 100644
index 15553c47520..00000000000
--- a/chromium/third_party/webrtc/media/base/audioframe.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MEDIA_BASE_AUDIOFRAME_H_
-#define WEBRTC_MEDIA_BASE_AUDIOFRAME_H_
-
-namespace cricket {
-
-class AudioFrame {
- public:
-  AudioFrame()
-      : audio10ms_(NULL),
-        length_(0),
-        sampling_frequency_(8000),
-        stereo_(false) {
-  }
-  AudioFrame(int16_t* audio, size_t audio_length, int sample_freq, bool stereo)
-      : audio10ms_(audio),
-        length_(audio_length),
-        sampling_frequency_(sample_freq),
-        stereo_(stereo) {}
-
-  int16_t* GetData() { return audio10ms_; }
-  size_t GetSize() const { return length_; }
-  int GetSamplingFrequency() const { return sampling_frequency_; }
-  bool GetStereo() const { return stereo_; }
-
- private:
-  // TODO(janahan): currently the data is not owned by this class.
-  // add ownership when we come up with the first use case that requires it.
-  int16_t* audio10ms_;
-  size_t length_;
-  int sampling_frequency_;
-  bool stereo_;
-};
-
-}  // namespace cricket
-#endif  // WEBRTC_MEDIA_BASE_AUDIOFRAME_H_
diff --git a/chromium/third_party/webrtc/media/base/codec.cc b/chromium/third_party/webrtc/media/base/codec.cc
index 287de0cdbf4..01350f71236 100644
--- a/chromium/third_party/webrtc/media/base/codec.cc
+++ b/chromium/third_party/webrtc/media/base/codec.cc
@@ -71,12 +71,10 @@ bool FeedbackParams::HasDuplicateEntries() const {
   return false;
 }
 
-Codec::Codec(int id, const std::string& name, int clockrate, int preference)
-    : id(id), name(name), clockrate(clockrate), preference(preference) {
-}
+Codec::Codec(int id, const std::string& name, int clockrate)
+    : id(id), name(name), clockrate(clockrate) {}
 
-Codec::Codec() : id(0), clockrate(0), preference(0) {
-}
+Codec::Codec() : id(0), clockrate(0) {}
 
 Codec::Codec(const Codec& c) = default;
 
@@ -86,7 +84,6 @@ Codec& Codec::operator=(const Codec& c) {
   this->id = c.id;  // id is reserved in objective-c
   name = c.name;
   clockrate = c.clockrate;
-  preference = c.preference;
   params = c.params;
   feedback_params = c.feedback_params;
   return *this;
@@ -94,8 +91,7 @@ Codec& Codec::operator=(const Codec& c) {
 
 bool Codec::operator==(const Codec& c) const {
   return this->id == c.id &&  // id is reserved in objective-c
-         name == c.name && clockrate == c.clockrate &&
-         preference == c.preference && params == c.params &&
+         name == c.name && clockrate == c.clockrate && params == c.params &&
          feedback_params == c.feedback_params;
 }
 
@@ -146,16 +142,20 @@ void Codec::IntersectFeedbackParams(const Codec& other) {
   feedback_params.Intersect(other.feedback_params);
 }
 
+webrtc::RtpCodecParameters Codec::ToCodecParameters() const {
+  webrtc::RtpCodecParameters codec_params;
+  codec_params.payload_type = id;
+  codec_params.mime_type = name;
+  codec_params.clock_rate = clockrate;
+  return codec_params;
+}
+
 AudioCodec::AudioCodec(int id,
                        const std::string& name,
                        int clockrate,
                        int bitrate,
-                       size_t channels,
-                       int preference)
-    : Codec(id, name, clockrate, preference),
-      bitrate(bitrate),
-      channels(channels) {
-}
+                       size_t channels)
+    : Codec(id, name, clockrate), bitrate(bitrate), channels(channels) {}
 
 AudioCodec::AudioCodec() : Codec(), bitrate(0), channels(0) {
 }
@@ -190,17 +190,23 @@ bool AudioCodec::Matches(const AudioCodec& codec) const {
       ((codec.channels < 2 && channels < 2) || channels == codec.channels);
 }
 
+webrtc::RtpCodecParameters AudioCodec::ToCodecParameters() const {
+  webrtc::RtpCodecParameters codec_params = Codec::ToCodecParameters();
+  codec_params.channels = channels;
+  return codec_params;
+}
+
 std::string AudioCodec::ToString() const {
   std::ostringstream os;
   os << "AudioCodec[" << id << ":" << name << ":" << clockrate << ":" << bitrate
-     << ":" << channels << ":" << preference << "]";
+     << ":" << channels << "]";
   return os.str();
 }
 
 std::string VideoCodec::ToString() const {
   std::ostringstream os;
   os << "VideoCodec[" << id << ":" << name << ":" << width << ":" << height
-     << ":" << framerate << ":" << preference << "]";
+     << ":" << framerate << "]";
   return os.str();
 }
 
@@ -208,20 +214,17 @@ VideoCodec::VideoCodec(int id,
                        const std::string& name,
                        int width,
                        int height,
-                       int framerate,
-                       int preference)
-    : Codec(id, name, kVideoCodecClockrate, preference),
+                       int framerate)
+    : Codec(id, name, kVideoCodecClockrate),
       width(width),
       height(height),
-      framerate(framerate) {
-}
+      framerate(framerate) {}
 
 VideoCodec::VideoCodec(int id, const std::string& name)
-    : Codec(id, name, kVideoCodecClockrate, 0),
+    : Codec(id, name, kVideoCodecClockrate),
       width(0),
       height(0),
-      framerate(0) {
-}
+      framerate(0) {}
 
 VideoCodec::VideoCodec() : Codec(), width(0), height(0), framerate(0) {
   clockrate = kVideoCodecClockrate;
@@ -244,7 +247,7 @@ bool VideoCodec::operator==(const VideoCodec& c) const {
 
 VideoCodec VideoCodec::CreateRtxCodec(int rtx_payload_type,
                                       int associated_payload_type) {
-  VideoCodec rtx_codec(rtx_payload_type, kRtxCodecName, 0, 0, 0, 0);
+  VideoCodec rtx_codec(rtx_payload_type, kRtxCodecName, 0, 0, 0);
   rtx_codec.SetParam(kCodecParamAssociatedPayloadType, associated_payload_type);
   return rtx_codec;
 }
@@ -291,9 +294,8 @@ bool VideoCodec::ValidateCodecFormat() const {
   return true;
 }
 
-DataCodec::DataCodec(int id, const std::string& name, int preference)
-    : Codec(id, name, kDataCodecClockrate, preference) {
-}
+DataCodec::DataCodec(int id, const std::string& name)
+    : Codec(id, name, kDataCodecClockrate) {}
 
 DataCodec::DataCodec() : Codec() {
   clockrate = kDataCodecClockrate;
diff --git a/chromium/third_party/webrtc/media/base/codec.h b/chromium/third_party/webrtc/media/base/codec.h
index 7476c026942..76a623a0de8 100644
--- a/chromium/third_party/webrtc/media/base/codec.h
+++ b/chromium/third_party/webrtc/media/base/codec.h
@@ -16,6 +16,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/rtpparameters.h"
 #include "webrtc/media/base/mediaconstants.h"
 
 namespace cricket {
@@ -64,16 +65,15 @@ struct Codec {
   int id;
   std::string name;
   int clockrate;
-  int preference;
   CodecParameterMap params;
   FeedbackParams feedback_params;
 
   // Creates a codec with the given parameters.
-  Codec(int id, const std::string& name, int clockrate, int preference);
+  Codec(int id, const std::string& name, int clockrate);
   // Creates an empty codec.
   Codec();
   Codec(const Codec& c);
-  ~Codec();
+  virtual ~Codec();
 
   // Indicates if this codec is compatible with the specified codec.
   bool Matches(const Codec& codec) const;
@@ -92,14 +92,12 @@ struct Codec {
   bool HasFeedbackParam(const FeedbackParam& param) const;
   void AddFeedbackParam(const FeedbackParam& param);
 
-  static bool Preferable(const Codec& first, const Codec& other) {
-    return first.preference > other.preference;
-  }
-
   // Filter |this| feedbacks params such that only those shared by both |this|
   // and |other| are kept.
   void IntersectFeedbackParams(const Codec& other);
 
+  virtual webrtc::RtpCodecParameters ToCodecParameters() const;
+
   Codec& operator=(const Codec& c);
 
   bool operator==(const Codec& c) const;
@@ -118,22 +116,19 @@ struct AudioCodec : public Codec {
              const std::string& name,
              int clockrate,
              int bitrate,
-             size_t channels,
-             int preference);
+             size_t channels);
   // Creates an empty codec.
   AudioCodec();
   AudioCodec(const AudioCodec& c);
-  ~AudioCodec() = default;
+  virtual ~AudioCodec() = default;
 
   // Indicates if this codec is compatible with the specified codec.
   bool Matches(const AudioCodec& codec) const;
 
-  static bool Preferable(const AudioCodec& first, const AudioCodec& other) {
-    return first.preference > other.preference;
-  }
-
   std::string ToString() const;
 
+  webrtc::RtpCodecParameters ToCodecParameters() const override;
+
   AudioCodec& operator=(const AudioCodec& c);
 
   bool operator==(const AudioCodec& c) const;
@@ -153,17 +148,12 @@ struct VideoCodec : public Codec {
              const std::string& name,
              int width,
              int height,
-             int framerate,
-             int preference);
+             int framerate);
   VideoCodec(int id, const std::string& name);
   // Creates an empty codec.
   VideoCodec();
   VideoCodec(const VideoCodec& c);
-  ~VideoCodec() = default;
-
-  static bool Preferable(const VideoCodec& first, const VideoCodec& other) {
-    return first.preference > other.preference;
-  }
+  virtual ~VideoCodec() = default;
 
   std::string ToString() const;
 
@@ -193,9 +183,10 @@ struct VideoCodec : public Codec {
 };
 
 struct DataCodec : public Codec {
-  DataCodec(int id, const std::string& name, int preference);
+  DataCodec(int id, const std::string& name);
   DataCodec();
   DataCodec(const DataCodec& c);
+  virtual ~DataCodec() = default;
 
   DataCodec& operator=(const DataCodec& c);
 
diff --git a/chromium/third_party/webrtc/media/base/codec_unittest.cc b/chromium/third_party/webrtc/media/base/codec_unittest.cc
index 88d49632d00..a3468803973 100644
--- a/chromium/third_party/webrtc/media/base/codec_unittest.cc
+++ b/chromium/third_party/webrtc/media/base/codec_unittest.cc
@@ -26,7 +26,7 @@ class CodecTest : public testing::Test {
 };
 
 TEST_F(CodecTest, TestCodecOperators) {
-  Codec c0(96, "D", 1000, 0);
+  Codec c0(96, "D", 1000);
   c0.SetParam("a", 1);
 
   Codec c1 = c0;
@@ -50,35 +50,29 @@ TEST_F(CodecTest, TestCodecOperators) {
   EXPECT_TRUE(c0 != c1);
 
   c1 = c0;
-  c1.preference = 1;
-  EXPECT_TRUE(c0 != c1);
-
-  c1 = c0;
   c1.SetParam("a", 2);
   EXPECT_TRUE(c0 != c1);
 
   Codec c5;
-  Codec c6(0, "", 0, 0);
+  Codec c6(0, "", 0);
   EXPECT_TRUE(c5 == c6);
 }
 
 TEST_F(CodecTest, TestAudioCodecOperators) {
-  AudioCodec c0(96, "A", 44100, 20000, 2, 3);
-  AudioCodec c1(95, "A", 44100, 20000, 2, 3);
-  AudioCodec c2(96, "x", 44100, 20000, 2, 3);
-  AudioCodec c3(96, "A", 48000, 20000, 2, 3);
-  AudioCodec c4(96, "A", 44100, 10000, 2, 3);
-  AudioCodec c5(96, "A", 44100, 20000, 1, 3);
-  AudioCodec c6(96, "A", 44100, 20000, 2, 1);
+  AudioCodec c0(96, "A", 44100, 20000, 2);
+  AudioCodec c1(95, "A", 44100, 20000, 2);
+  AudioCodec c2(96, "x", 44100, 20000, 2);
+  AudioCodec c3(96, "A", 48000, 20000, 2);
+  AudioCodec c4(96, "A", 44100, 10000, 2);
+  AudioCodec c5(96, "A", 44100, 20000, 1);
   EXPECT_TRUE(c0 != c1);
   EXPECT_TRUE(c0 != c2);
   EXPECT_TRUE(c0 != c3);
   EXPECT_TRUE(c0 != c4);
   EXPECT_TRUE(c0 != c5);
-  EXPECT_TRUE(c0 != c6);
 
   AudioCodec c7;
-  AudioCodec c8(0, "", 0, 0, 0, 0);
+  AudioCodec c8(0, "", 0, 0, 0);
   AudioCodec c9 = c0;
   EXPECT_TRUE(c8 == c7);
   EXPECT_TRUE(c9 != c7);
@@ -103,61 +97,59 @@ TEST_F(CodecTest, TestAudioCodecOperators) {
 
 TEST_F(CodecTest, TestAudioCodecMatches) {
   // Test a codec with a static payload type.
-  AudioCodec c0(95, "A", 44100, 20000, 1, 3);
-  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 44100, 20000, 1, 0)));
-  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 44100, 20000, 0, 0)));
-  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 44100, 0, 0, 0)));
-  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 0, 0, 0, 0)));
-  EXPECT_FALSE(c0.Matches(AudioCodec(96, "", 44100, 20000, 1, 0)));
-  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 55100, 20000, 1, 0)));
-  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 44100, 30000, 1, 0)));
-  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 44100, 20000, 2, 0)));
-  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 55100, 30000, 2, 0)));
+  AudioCodec c0(95, "A", 44100, 20000, 1);
+  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 44100, 20000, 1)));
+  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 44100, 20000, 0)));
+  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 44100, 0, 0)));
+  EXPECT_TRUE(c0.Matches(AudioCodec(95, "", 0, 0, 0)));
+  EXPECT_FALSE(c0.Matches(AudioCodec(96, "", 44100, 20000, 1)));
+  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 55100, 20000, 1)));
+  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 44100, 30000, 1)));
+  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 44100, 20000, 2)));
+  EXPECT_FALSE(c0.Matches(AudioCodec(95, "", 55100, 30000, 2)));
 
   // Test a codec with a dynamic payload type.
-  AudioCodec c1(96, "A", 44100, 20000, 1, 3);
-  EXPECT_TRUE(c1.Matches(AudioCodec(96, "A", 0, 0, 0, 0)));
-  EXPECT_TRUE(c1.Matches(AudioCodec(97, "A", 0, 0, 0, 0)));
-  EXPECT_TRUE(c1.Matches(AudioCodec(96, "a", 0, 0, 0, 0)));
-  EXPECT_TRUE(c1.Matches(AudioCodec(97, "a", 0, 0, 0, 0)));
-  EXPECT_FALSE(c1.Matches(AudioCodec(95, "A", 0, 0, 0, 0)));
-  EXPECT_FALSE(c1.Matches(AudioCodec(96, "", 44100, 20000, 2, 0)));
-  EXPECT_FALSE(c1.Matches(AudioCodec(96, "A", 55100, 30000, 1, 0)));
+  AudioCodec c1(96, "A", 44100, 20000, 1);
+  EXPECT_TRUE(c1.Matches(AudioCodec(96, "A", 0, 0, 0)));
+  EXPECT_TRUE(c1.Matches(AudioCodec(97, "A", 0, 0, 0)));
+  EXPECT_TRUE(c1.Matches(AudioCodec(96, "a", 0, 0, 0)));
+  EXPECT_TRUE(c1.Matches(AudioCodec(97, "a", 0, 0, 0)));
+  EXPECT_FALSE(c1.Matches(AudioCodec(95, "A", 0, 0, 0)));
+  EXPECT_FALSE(c1.Matches(AudioCodec(96, "", 44100, 20000, 2)));
+  EXPECT_FALSE(c1.Matches(AudioCodec(96, "A", 55100, 30000, 1)));
 
   // Test a codec with a dynamic payload type, and auto bitrate.
-  AudioCodec c2(97, "A", 16000, 0, 1, 3);
+  AudioCodec c2(97, "A", 16000, 0, 1);
   // Use default bitrate.
-  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, 0, 1, 0)));
-  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, 0, 0, 0)));
+  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, 0, 1)));
+  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, 0, 0)));
   // Use explicit bitrate.
-  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, 32000, 1, 0)));
+  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, 32000, 1)));
   // Backward compatibility with clients that might send "-1" (for default).
-  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, -1, 1, 0)));
+  EXPECT_TRUE(c2.Matches(AudioCodec(97, "A", 16000, -1, 1)));
 
   // Stereo doesn't match channels = 0.
-  AudioCodec c3(96, "A", 44100, 20000, 2, 3);
-  EXPECT_TRUE(c3.Matches(AudioCodec(96, "A", 44100, 20000, 2, 3)));
-  EXPECT_FALSE(c3.Matches(AudioCodec(96, "A", 44100, 20000, 1, 3)));
-  EXPECT_FALSE(c3.Matches(AudioCodec(96, "A", 44100, 20000, 0, 3)));
+  AudioCodec c3(96, "A", 44100, 20000, 2);
+  EXPECT_TRUE(c3.Matches(AudioCodec(96, "A", 44100, 20000, 2)));
+  EXPECT_FALSE(c3.Matches(AudioCodec(96, "A", 44100, 20000, 1)));
+  EXPECT_FALSE(c3.Matches(AudioCodec(96, "A", 44100, 20000, 0)));
 }
 
 TEST_F(CodecTest, TestVideoCodecOperators) {
-  VideoCodec c0(96, "V", 320, 200, 30, 3);
-  VideoCodec c1(95, "V", 320, 200, 30, 3);
-  VideoCodec c2(96, "x", 320, 200, 30, 3);
-  VideoCodec c3(96, "V", 120, 200, 30, 3);
-  VideoCodec c4(96, "V", 320, 100, 30, 3);
-  VideoCodec c5(96, "V", 320, 200, 10, 3);
-  VideoCodec c6(96, "V", 320, 200, 30, 1);
+  VideoCodec c0(96, "V", 320, 200, 30);
+  VideoCodec c1(95, "V", 320, 200, 30);
+  VideoCodec c2(96, "x", 320, 200, 30);
+  VideoCodec c3(96, "V", 120, 200, 30);
+  VideoCodec c4(96, "V", 320, 100, 30);
+  VideoCodec c5(96, "V", 320, 200, 10);
   EXPECT_TRUE(c0 != c1);
   EXPECT_TRUE(c0 != c2);
   EXPECT_TRUE(c0 != c3);
   EXPECT_TRUE(c0 != c4);
   EXPECT_TRUE(c0 != c5);
-  EXPECT_TRUE(c0 != c6);
 
   VideoCodec c7;
-  VideoCodec c8(0, "", 0, 0, 0, 0);
+  VideoCodec c8(0, "", 0, 0, 0);
   VideoCodec c9 = c0;
   EXPECT_TRUE(c8 == c7);
   EXPECT_TRUE(c9 != c7);
@@ -182,34 +174,34 @@ TEST_F(CodecTest, TestVideoCodecOperators) {
 
 TEST_F(CodecTest, TestVideoCodecMatches) {
   // Test a codec with a static payload type.
-  VideoCodec c0(95, "V", 320, 200, 30, 3);
-  EXPECT_TRUE(c0.Matches(VideoCodec(95, "", 640, 400, 15, 0)));
-  EXPECT_FALSE(c0.Matches(VideoCodec(96, "", 320, 200, 30, 0)));
+  VideoCodec c0(95, "V", 320, 200, 30);
+  EXPECT_TRUE(c0.Matches(VideoCodec(95, "", 640, 400, 15)));
+  EXPECT_FALSE(c0.Matches(VideoCodec(96, "", 320, 200, 30)));
 
   // Test a codec with a dynamic payload type.
-  VideoCodec c1(96, "V", 320, 200, 30, 3);
-  EXPECT_TRUE(c1.Matches(VideoCodec(96, "V", 640, 400, 15, 0)));
-  EXPECT_TRUE(c1.Matches(VideoCodec(97, "V", 640, 400, 15, 0)));
-  EXPECT_TRUE(c1.Matches(VideoCodec(96, "v", 640, 400, 15, 0)));
-  EXPECT_TRUE(c1.Matches(VideoCodec(97, "v", 640, 400, 15, 0)));
-  EXPECT_FALSE(c1.Matches(VideoCodec(96, "", 320, 200, 30, 0)));
-  EXPECT_FALSE(c1.Matches(VideoCodec(95, "V", 640, 400, 15, 0)));
+  VideoCodec c1(96, "V", 320, 200, 30);
+  EXPECT_TRUE(c1.Matches(VideoCodec(96, "V", 640, 400, 15)));
+  EXPECT_TRUE(c1.Matches(VideoCodec(97, "V", 640, 400, 15)));
+  EXPECT_TRUE(c1.Matches(VideoCodec(96, "v", 640, 400, 15)));
+  EXPECT_TRUE(c1.Matches(VideoCodec(97, "v", 640, 400, 15)));
+  EXPECT_FALSE(c1.Matches(VideoCodec(96, "", 320, 200, 30)));
+  EXPECT_FALSE(c1.Matches(VideoCodec(95, "V", 640, 400, 15)));
 }
 
 TEST_F(CodecTest, TestDataCodecMatches) {
   // Test a codec with a static payload type.
-  DataCodec c0(95, "D", 0);
-  EXPECT_TRUE(c0.Matches(DataCodec(95, "", 0)));
-  EXPECT_FALSE(c0.Matches(DataCodec(96, "", 0)));
+  DataCodec c0(95, "D");
+  EXPECT_TRUE(c0.Matches(DataCodec(95, "")));
+  EXPECT_FALSE(c0.Matches(DataCodec(96, "")));
 
   // Test a codec with a dynamic payload type.
-  DataCodec c1(96, "D", 3);
-  EXPECT_TRUE(c1.Matches(DataCodec(96, "D", 0)));
-  EXPECT_TRUE(c1.Matches(DataCodec(97, "D", 0)));
-  EXPECT_TRUE(c1.Matches(DataCodec(96, "d", 0)));
-  EXPECT_TRUE(c1.Matches(DataCodec(97, "d", 0)));
-  EXPECT_FALSE(c1.Matches(DataCodec(96, "", 0)));
-  EXPECT_FALSE(c1.Matches(DataCodec(95, "D", 0)));
+  DataCodec c1(96, "D");
+  EXPECT_TRUE(c1.Matches(DataCodec(96, "D")));
+  EXPECT_TRUE(c1.Matches(DataCodec(97, "D")));
+  EXPECT_TRUE(c1.Matches(DataCodec(96, "d")));
+  EXPECT_TRUE(c1.Matches(DataCodec(97, "d")));
+  EXPECT_FALSE(c1.Matches(DataCodec(96, "")));
+  EXPECT_FALSE(c1.Matches(DataCodec(95, "D")));
 }
 
 TEST_F(CodecTest, TestSetParamGetParamAndRemoveParam) {
@@ -254,10 +246,10 @@ TEST_F(CodecTest, TestIntersectFeedbackParams) {
 
 TEST_F(CodecTest, TestGetCodecType) {
   // Codec type comparison should be case insenstive on names.
-  const VideoCodec codec(96, "V", 320, 200, 30, 3);
-  const VideoCodec rtx_codec(96, "rTx", 320, 200, 30, 3);
-  const VideoCodec ulpfec_codec(96, "ulpFeC", 320, 200, 30, 3);
-  const VideoCodec red_codec(96, "ReD", 320, 200, 30, 3);
+  const VideoCodec codec(96, "V", 320, 200, 30);
+  const VideoCodec rtx_codec(96, "rTx", 320, 200, 30);
+  const VideoCodec ulpfec_codec(96, "ulpFeC", 320, 200, 30);
+  const VideoCodec red_codec(96, "ReD", 320, 200, 30);
   EXPECT_EQ(VideoCodec::CODEC_VIDEO, codec.GetCodecType());
   EXPECT_EQ(VideoCodec::CODEC_RTX, rtx_codec.GetCodecType());
   EXPECT_EQ(VideoCodec::CODEC_ULPFEC, ulpfec_codec.GetCodecType());
@@ -275,7 +267,7 @@ TEST_F(CodecTest, TestCreateRtxCodec) {
 }
 
 TEST_F(CodecTest, TestValidateCodecFormat) {
-  const VideoCodec codec(96, "V", 320, 200, 30, 3);
+  const VideoCodec codec(96, "V", 320, 200, 30);
   ASSERT_TRUE(codec.ValidateCodecFormat());
 
   // Accept 0-127 as payload types.
@@ -329,3 +321,19 @@ TEST_F(CodecTest, TestValidateCodecFormat) {
   different_bitrates.params[kCodecParamMaxBitrate] = "100";
   EXPECT_TRUE(different_bitrates.ValidateCodecFormat());
 }
+
+TEST_F(CodecTest, TestToCodecParameters) {
+  const VideoCodec v(96, "V", 320, 200, 30);
+  webrtc::RtpCodecParameters codec_params_1 = v.ToCodecParameters();
+  EXPECT_EQ(96, codec_params_1.payload_type);
+  EXPECT_EQ("V", codec_params_1.mime_type);
+  EXPECT_EQ(cricket::kVideoCodecClockrate, codec_params_1.clock_rate);
+  EXPECT_EQ(1, codec_params_1.channels);
+
+  const AudioCodec a(97, "A", 44100, 20000, 2);
+  webrtc::RtpCodecParameters codec_params_2 = a.ToCodecParameters();
+  EXPECT_EQ(97, codec_params_2.payload_type);
+  EXPECT_EQ("A", codec_params_2.mime_type);
+  EXPECT_EQ(44100, codec_params_2.clock_rate);
+  EXPECT_EQ(2, codec_params_2.channels);
+}
diff --git a/chromium/third_party/webrtc/media/base/fakemediaengine.h b/chromium/third_party/webrtc/media/base/fakemediaengine.h
index d6d653991c6..bde584386f9 100644
--- a/chromium/third_party/webrtc/media/base/fakemediaengine.h
+++ b/chromium/third_party/webrtc/media/base/fakemediaengine.h
@@ -55,7 +55,9 @@ template <class Base> class RtpHelper : public Base {
   const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
   const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
 
-  bool SendRtp(const void* data, int len, const rtc::PacketOptions& options) {
+  bool SendRtp(const void* data,
+               size_t len,
+               const rtc::PacketOptions& options) {
     if (!sending_) {
       return false;
     }
@@ -63,13 +65,13 @@ template <class Base> class RtpHelper : public Base {
                                   kMaxRtpPacketLen);
     return Base::SendPacket(&packet, options);
   }
-  bool SendRtcp(const void* data, int len) {
+  bool SendRtcp(const void* data, size_t len) {
     rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
                                   kMaxRtpPacketLen);
     return Base::SendRtcp(&packet, rtc::PacketOptions());
   }
 
-  bool CheckRtp(const void* data, int len) {
+  bool CheckRtp(const void* data, size_t len) {
     bool success = !rtp_packets_.empty();
     if (success) {
       std::string packet = rtp_packets_.front();
@@ -78,7 +80,7 @@ template <class Base> class RtpHelper : public Base {
     }
     return success;
   }
-  bool CheckRtcp(const void* data, int len) {
+  bool CheckRtcp(const void* data, size_t len) {
     bool success = !rtcp_packets_.empty();
     if (success) {
       std::string packet = rtcp_packets_.front();
@@ -97,13 +99,14 @@ template <class Base> class RtpHelper : public Base {
       return false;
     }
     send_streams_.push_back(sp);
-    rtp_parameters_[sp.first_ssrc()] = CreateRtpParametersWithOneEncoding();
+    rtp_send_parameters_[sp.first_ssrc()] =
+        CreateRtpParametersWithOneEncoding();
     return true;
   }
   virtual bool RemoveSendStream(uint32_t ssrc) {
-    auto parameters_iterator = rtp_parameters_.find(ssrc);
-    if (parameters_iterator != rtp_parameters_.end()) {
-      rtp_parameters_.erase(parameters_iterator);
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_parameters_.end()) {
+      rtp_send_parameters_.erase(parameters_iterator);
     }
     return RemoveStreamBySsrc(&send_streams_, ssrc);
   }
@@ -113,23 +116,49 @@ template <class Base> class RtpHelper : public Base {
       return false;
     }
     receive_streams_.push_back(sp);
+    rtp_receive_parameters_[sp.first_ssrc()] =
+        CreateRtpParametersWithOneEncoding();
     return true;
   }
   virtual bool RemoveRecvStream(uint32_t ssrc) {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_parameters_.end()) {
+      rtp_receive_parameters_.erase(parameters_iterator);
+    }
     return RemoveStreamBySsrc(&receive_streams_, ssrc);
   }
 
-  virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const {
-    auto parameters_iterator = rtp_parameters_.find(ssrc);
-    if (parameters_iterator != rtp_parameters_.end()) {
+  virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const {
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_parameters_.end()) {
+      return parameters_iterator->second;
+    }
+    return webrtc::RtpParameters();
+  }
+  virtual bool SetRtpSendParameters(uint32_t ssrc,
+                                    const webrtc::RtpParameters& parameters) {
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_parameters_.end()) {
+      parameters_iterator->second = parameters;
+      return true;
+    }
+    // Replicate the behavior of the real media channel: return false
+    // when setting parameters for unknown SSRCs.
+    return false;
+  }
+
+  virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_parameters_.end()) {
       return parameters_iterator->second;
     }
     return webrtc::RtpParameters();
   }
-  virtual bool SetRtpParameters(uint32_t ssrc,
-                                const webrtc::RtpParameters& parameters) {
-    auto parameters_iterator = rtp_parameters_.find(ssrc);
-    if (parameters_iterator != rtp_parameters_.end()) {
+  virtual bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_parameters_.end()) {
       parameters_iterator->second = parameters;
       return true;
     }
@@ -179,7 +208,7 @@ template <class Base> class RtpHelper : public Base {
     return ready_to_send_;
   }
 
-  NetworkRoute last_network_route() const { return last_network_route_; }
+  rtc::NetworkRoute last_network_route() const { return last_network_route_; }
   int num_network_route_changes() const { return num_network_route_changes_; }
   void set_num_network_route_changes(int changes) {
     num_network_route_changes_ = changes;
@@ -224,7 +253,7 @@ template <class Base> class RtpHelper : public Base {
     ready_to_send_ = ready;
   }
   virtual void OnNetworkRouteChanged(const std::string& transport_name,
-                                     const NetworkRoute& network_route) {
+                                     const rtc::NetworkRoute& network_route) {
     last_network_route_ = network_route;
     ++num_network_route_changes_;
   }
@@ -241,13 +270,14 @@ template <class Base> class RtpHelper : public Base {
   std::vector<StreamParams> send_streams_;
   std::vector<StreamParams> receive_streams_;
   std::set<uint32_t> muted_streams_;
-  std::map<uint32_t, webrtc::RtpParameters> rtp_parameters_;
+  std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
+  std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
   bool fail_set_send_codecs_;
   bool fail_set_recv_codecs_;
   uint32_t send_ssrc_;
   std::string rtcp_cname_;
   bool ready_to_send_;
-  NetworkRoute last_network_route_;
+  rtc::NetworkRoute last_network_route_;
   int num_network_route_changes_ = 0;
 };
 
@@ -482,23 +512,23 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
     return sinks_;
   }
   int max_bps() const { return max_bps_; }
-  virtual bool SetSendParameters(const VideoSendParameters& params) {
+  bool SetSendParameters(const VideoSendParameters& params) override {
     return (SetSendCodecs(params.codecs) &&
             SetSendRtpHeaderExtensions(params.extensions) &&
             SetMaxSendBandwidth(params.max_bandwidth_bps));
   }
-  virtual bool SetRecvParameters(const VideoRecvParameters& params) {
+  bool SetRecvParameters(const VideoRecvParameters& params) override {
     return (SetRecvCodecs(params.codecs) &&
             SetRecvRtpHeaderExtensions(params.extensions));
   }
-  virtual bool AddSendStream(const StreamParams& sp) {
+  bool AddSendStream(const StreamParams& sp) override {
     return RtpHelper<VideoMediaChannel>::AddSendStream(sp);
   }
-  virtual bool RemoveSendStream(uint32_t ssrc) {
+  bool RemoveSendStream(uint32_t ssrc) override {
     return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
   }
 
-  virtual bool GetSendCodec(VideoCodec* send_codec) {
+  bool GetSendCodec(VideoCodec* send_codec) override {
     if (send_codecs_.empty()) {
       return false;
     }
@@ -516,9 +546,9 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
     return true;
   }
 
-  virtual bool SetSend(bool send) { return set_sending(send); }
-  virtual bool SetVideoSend(uint32_t ssrc, bool enable,
-                            const VideoOptions* options) {
+  bool SetSend(bool send) override { return set_sending(send); }
+  bool SetVideoSend(uint32_t ssrc, bool enable,
+                    const VideoOptions* options) override {
     if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) {
       return false;
     }
@@ -527,27 +557,29 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
     }
     return true;
   }
-  virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
-    capturers_[ssrc] = capturer;
-    return true;
+  void SetSource(
+      uint32_t ssrc,
+      rtc::VideoSourceInterface<cricket::VideoFrame>* source) override {
+    sources_[ssrc] = source;
   }
-  bool HasCapturer(uint32_t ssrc) const {
-    return capturers_.find(ssrc) != capturers_.end();
+
+  bool HasSource(uint32_t ssrc) const {
+    return sources_.find(ssrc) != sources_.end();
   }
-  virtual bool AddRecvStream(const StreamParams& sp) {
+  bool AddRecvStream(const StreamParams& sp) override {
     if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
       return false;
     sinks_[sp.first_ssrc()] = NULL;
     return true;
   }
-  virtual bool RemoveRecvStream(uint32_t ssrc) {
+  bool RemoveRecvStream(uint32_t ssrc) override {
     if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
       return false;
     sinks_.erase(ssrc);
     return true;
   }
 
-  virtual bool GetStats(VideoMediaInfo* info) { return false; }
+  bool GetStats(VideoMediaInfo* info) override { return false; }
 
  private:
   bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
@@ -580,7 +612,7 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
   std::vector<VideoCodec> recv_codecs_;
   std::vector<VideoCodec> send_codecs_;
   std::map<uint32_t, rtc::VideoSinkInterface<VideoFrame>*> sinks_;
-  std::map<uint32_t, VideoCapturer*> capturers_;
+  std::map<uint32_t, rtc::VideoSourceInterface<VideoFrame>*> sources_;
   VideoOptions options_;
   int max_bps_;
 };
@@ -700,7 +732,7 @@ class FakeVoiceEngine : public FakeBaseEngine {
       : output_volume_(-1) {
     // Add a fake audio codec. Note that the name must not be "" as there are
     // sanity checks against that.
-    codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
+    codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
   }
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
     return rtc::scoped_refptr<webrtc::AudioState>();
@@ -744,7 +776,9 @@ class FakeVoiceEngine : public FakeBaseEngine {
 
   void StopAecDump() {}
 
-  bool StartRtcEventLog(rtc::PlatformFile file) { return false; }
+  bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes) {
+    return false;
+  }
 
   void StopRtcEventLog() {}
 
@@ -761,7 +795,7 @@ class FakeVideoEngine : public FakeBaseEngine {
   FakeVideoEngine() : capture_(false) {
     // Add a fake video codec. Note that the name must not be "" as there are
     // sanity checks against that.
-    codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
+    codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0));
   }
   void Init() {}
   bool SetOptions(const VideoOptions& options) {
diff --git a/chromium/third_party/webrtc/media/base/fakevideocapturer.h b/chromium/third_party/webrtc/media/base/fakevideocapturer.h
index 89dcf652f14..026bf80a7b1 100644
--- a/chromium/third_party/webrtc/media/base/fakevideocapturer.h
+++ b/chromium/third_party/webrtc/media/base/fakevideocapturer.h
@@ -31,7 +31,7 @@ class FakeVideoCapturer : public cricket::VideoCapturer {
  public:
   FakeVideoCapturer(bool is_screencast)
       : running_(false),
-        initial_unix_timestamp_(time(NULL) * rtc::kNumNanosecsPerSec),
+        initial_timestamp_(rtc::TimeNanos()),
         next_timestamp_(rtc::kNumNanosecsPerMillisec),
         is_screencast_(is_screencast),
         rotation_(webrtc::kVideoRotation_0) {
@@ -99,7 +99,7 @@ class FakeVideoCapturer : public cricket::VideoCapturer {
     frame.height = height;
     frame.fourcc = fourcc;
     frame.data_size = size;
-    frame.time_stamp = initial_unix_timestamp_ + next_timestamp_;
+    frame.time_stamp = initial_timestamp_ + next_timestamp_;
     next_timestamp_ += timestamp_interval;
 
     std::unique_ptr<char[]> data(new char[size]);
@@ -153,7 +153,7 @@ class FakeVideoCapturer : public cricket::VideoCapturer {
 
  private:
   bool running_;
-  int64_t initial_unix_timestamp_;
+  int64_t initial_timestamp_;
   int64_t next_timestamp_;
   const bool is_screencast_;
   webrtc::VideoRotation rotation_;
diff --git a/chromium/third_party/webrtc/media/base/fakevideorenderer.h b/chromium/third_party/webrtc/media/base/fakevideorenderer.h
index f75888e1ee7..7398bbaa8bb 100644
--- a/chromium/third_party/webrtc/media/base/fakevideorenderer.h
+++ b/chromium/third_party/webrtc/media/base/fakevideorenderer.h
@@ -39,7 +39,7 @@ class FakeVideoRenderer : public rtc::VideoSinkInterface<cricket::VideoFrame> {
     ++num_rendered_frames_;
     width_ = frame.width();
     height_ = frame.height();
-    rotation_ = frame.GetVideoRotation();
+    rotation_ = frame.rotation();
     timestamp_ = frame.GetTimeStamp();
     SignalRenderFrame(&frame);
   }
@@ -82,15 +82,15 @@ class FakeVideoRenderer : public rtc::VideoSinkInterface<cricket::VideoFrame> {
                                  uint8_t v_min,
                                  uint8_t v_max,
                                  const cricket::VideoFrame* frame) {
-    if (!frame) {
+    if (!frame || !frame->video_frame_buffer()) {
       return false;
     }
     // Y
     int y_width = frame->width();
     int y_height = frame->height();
-    const uint8_t* y_plane = frame->GetYPlane();
+    const uint8_t* y_plane = frame->video_frame_buffer()->DataY();
     const uint8_t* y_pos = y_plane;
-    int32_t y_pitch = frame->GetYPitch();
+    int32_t y_pitch = frame->video_frame_buffer()->StrideY();
     for (int i = 0; i < y_height; ++i) {
       for (int j = 0; j < y_width; ++j) {
         uint8_t y_value = *(y_pos + j);
@@ -103,12 +103,12 @@ class FakeVideoRenderer : public rtc::VideoSinkInterface<cricket::VideoFrame> {
     // U and V
     int chroma_width = (frame->width() + 1)/2;
     int chroma_height = (frame->height() + 1)/2;
-    const uint8_t* u_plane = frame->GetUPlane();
-    const uint8_t* v_plane = frame->GetVPlane();
+    const uint8_t* u_plane = frame->video_frame_buffer()->DataU();
+    const uint8_t* v_plane = frame->video_frame_buffer()->DataV();
     const uint8_t* u_pos = u_plane;
     const uint8_t* v_pos = v_plane;
-    int32_t u_pitch = frame->GetUPitch();
-    int32_t v_pitch = frame->GetVPitch();
+    int32_t u_pitch = frame->video_frame_buffer()->StrideU();
+    int32_t v_pitch = frame->video_frame_buffer()->StrideV();
     for (int i = 0; i < chroma_height; ++i) {
       for (int j = 0; j < chroma_width; ++j) {
         uint8_t u_value = *(u_pos + j);
diff --git a/chromium/third_party/webrtc/media/base/mediachannel.h b/chromium/third_party/webrtc/media/base/mediachannel.h
index 424572de2f0..54347098542 100644
--- a/chromium/third_party/webrtc/media/base/mediachannel.h
+++ b/chromium/third_party/webrtc/media/base/mediachannel.h
@@ -17,6 +17,7 @@
 
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
@@ -29,11 +30,11 @@
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
+#include "webrtc/media/base/videosourceinterface.h"
 // TODO(juberti): re-evaluate this include
 #include "webrtc/pc/audiomonitor.h"
 
 namespace rtc {
-class Buffer;
 class RateLimiter;
 class Timing;
 }
@@ -156,6 +157,7 @@ struct AudioOptions {
     SetFrom(&extended_filter_aec, change.extended_filter_aec);
     SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
     SetFrom(&experimental_ns, change.experimental_ns);
+    SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
     SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
     SetFrom(&tx_agc_digital_compression_gain,
             change.tx_agc_digital_compression_gain);
@@ -180,6 +182,7 @@ struct AudioOptions {
         extended_filter_aec == o.extended_filter_aec &&
         delay_agnostic_aec == o.delay_agnostic_aec &&
         experimental_ns == o.experimental_ns &&
+        intelligibility_enhancer == o.intelligibility_enhancer &&
         adjust_agc_delta == o.adjust_agc_delta &&
         tx_agc_target_dbov == o.tx_agc_target_dbov &&
         tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
@@ -209,6 +212,7 @@ struct AudioOptions {
     ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
     ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
     ost << ToStringIfSet("experimental_ns", experimental_ns);
+    ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
     ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
     ost << ToStringIfSet("tx_agc_digital_compression_gain",
         tx_agc_digital_compression_gain);
@@ -243,6 +247,7 @@ struct AudioOptions {
   rtc::Optional<bool> extended_filter_aec;
   rtc::Optional<bool> delay_agnostic_aec;
   rtc::Optional<bool> experimental_ns;
+  rtc::Optional<bool> intelligibility_enhancer;
   // Note that tx_agc_* only applies to non-experimental AGC.
   rtc::Optional<uint16_t> tx_agc_target_dbov;
   rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
@@ -391,8 +396,9 @@ class MediaChannel : public sigslot::has_slots<> {
   // Called when the socket's ability to send has changed.
   virtual void OnReadyToSend(bool ready) = 0;
   // Called when the network route used for sending packets changed.
-  virtual void OnNetworkRouteChanged(const std::string& transport_name,
-                                     const NetworkRoute& network_route) = 0;
+  virtual void OnNetworkRouteChanged(
+      const std::string& transport_name,
+      const rtc::NetworkRoute& network_route) = 0;
   // Creates a new outgoing media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddSendStream(const StreamParams& sp) = 0;
@@ -839,6 +845,7 @@ struct RtpParameters {
   std::vector<RtpHeaderExtension> extensions;
   // TODO(pthatcher): Add streams.
   RtcpParameters rtcp;
+  virtual ~RtpParameters() = default;
 };
 
 // TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
@@ -904,6 +911,15 @@ class VoiceMediaChannel : public MediaChannel {
   virtual ~VoiceMediaChannel() {}
   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
+  virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
+  virtual webrtc::RtpParameters GetRtpReceiveParameters(
+      uint32_t ssrc) const = 0;
+  virtual bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
   // Starts or stops playout of received audio.
   virtual bool SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
@@ -980,9 +996,15 @@ class VideoMediaChannel : public MediaChannel {
 
   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
   virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
-  virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0;
-  virtual bool SetRtpParameters(uint32_t ssrc,
-                                const webrtc::RtpParameters& parameters) = 0;
+  virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
+  virtual bool SetRtpSendParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
+  virtual webrtc::RtpParameters GetRtpReceiveParameters(
+      uint32_t ssrc) const = 0;
+  virtual bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) = 0;
   // Gets the currently set codecs/payload types to be used for outgoing media.
   virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
   // Starts or stops transmission (and potentially capture) of local video.
@@ -995,9 +1017,10 @@ class VideoMediaChannel : public MediaChannel {
   // If SSRC is 0, the renderer is used for the 'default' stream.
   virtual bool SetSink(uint32_t ssrc,
                        rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0;
-  // If |ssrc| is 0, replace the default capturer (engine capturer) with
-  // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC.
-  virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0;
+  // Register a source. The |ssrc| must correspond to a registered send stream.
+  virtual void SetSource(
+      uint32_t ssrc,
+      rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0;
   // Gets quality stats for the channel.
   virtual bool GetStats(VideoMediaInfo* info) = 0;
 };
@@ -1107,7 +1130,7 @@ class DataMediaChannel : public MediaChannel {
   virtual bool SetReceive(bool receive) = 0;
 
   virtual void OnNetworkRouteChanged(const std::string& transport_name,
-                                     const NetworkRoute& network_route) {}
+                                     const rtc::NetworkRoute& network_route) {}
 
   virtual bool SendData(
       const SendDataParams& params,
diff --git a/chromium/third_party/webrtc/media/base/mediaconstants.cc b/chromium/third_party/webrtc/media/base/mediaconstants.cc
index d6bb7e3d48a..a8e2b12599b 100644
--- a/chromium/third_party/webrtc/media/base/mediaconstants.cc
+++ b/chromium/third_party/webrtc/media/base/mediaconstants.cc
@@ -64,7 +64,7 @@ const int kOpusDefaultUseInbandFec = 0;
 const int kOpusDefaultUseDtx = 0;
 const int kOpusDefaultMaxPlaybackRate = 48000;
 
-const int kPreferredMaxPTime = 60;
+const int kPreferredMaxPTime = 120;
 const int kPreferredMinPTime = 10;
 const int kPreferredSPropStereo = 0;
 const int kPreferredStereo = 0;
diff --git a/chromium/third_party/webrtc/media/base/mediaengine.h b/chromium/third_party/webrtc/media/base/mediaengine.h
index 4c7d62a1e4b..4446bd6b0e4 100644
--- a/chromium/third_party/webrtc/media/base/mediaengine.h
+++ b/chromium/third_party/webrtc/media/base/mediaengine.h
@@ -93,8 +93,11 @@ class MediaEngineInterface {
   // Stops recording AEC dump.
   virtual void StopAecDump() = 0;
 
-  // Starts RtcEventLog using existing file.
-  virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
+  // Starts RtcEventLog using existing file. A maximum file size in bytes can be
+  // specified. Logging is stopped just before the size limit is exceeded.
+  // If max_size_bytes is set to a value <= 0, no limit will be used.
+  virtual bool StartRtcEventLog(rtc::PlatformFile file,
+                                int64_t max_size_bytes) = 0;
 
   // Stops recording an RtcEventLog.
   virtual void StopRtcEventLog() = 0;
@@ -176,8 +179,9 @@ class CompositeMediaEngine : public MediaEngineInterface {
     voice_.StopAecDump();
   }
 
-  virtual bool StartRtcEventLog(rtc::PlatformFile file) {
-    return voice_.StartRtcEventLog(file);
+  virtual bool StartRtcEventLog(rtc::PlatformFile file,
+                                int64_t max_size_bytes) {
+    return voice_.StartRtcEventLog(file, max_size_bytes);
   }
 
   virtual void StopRtcEventLog() { voice_.StopRtcEventLog(); }
diff --git a/chromium/third_party/webrtc/media/base/rtpdataengine.cc b/chromium/third_party/webrtc/media/base/rtpdataengine.cc
index ae361745557..4b6647c649a 100644
--- a/chromium/third_party/webrtc/media/base/rtpdataengine.cc
+++ b/chromium/third_party/webrtc/media/base/rtpdataengine.cc
@@ -36,8 +36,7 @@ static const size_t kMaxSrtpHmacOverhead = 16;
 
 RtpDataEngine::RtpDataEngine() {
   data_codecs_.push_back(
-      DataCodec(kGoogleRtpDataCodecId,
-                kGoogleRtpDataCodecName, 0));
+      DataCodec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName));
   SetTiming(new rtc::Timing());
 }
 
@@ -92,7 +91,7 @@ void RtpClock::Tick(double now, int* seq_num, uint32_t* timestamp) {
 }
 
 const DataCodec* FindUnknownCodec(const std::vector<DataCodec>& codecs) {
-  DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
+  DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName);
   std::vector<DataCodec>::const_iterator iter;
   for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
     if (!iter->Matches(data_codec)) {
@@ -103,7 +102,7 @@ const DataCodec* FindUnknownCodec(const std::vector<DataCodec>& codecs) {
 }
 
 const DataCodec* FindKnownCodec(const std::vector<DataCodec>& codecs) {
-  DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
+  DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName);
   std::vector<DataCodec>::const_iterator iter;
   for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
     if (iter->Matches(data_codec)) {
diff --git a/chromium/third_party/webrtc/media/base/rtpdump.cc b/chromium/third_party/webrtc/media/base/rtpdump.cc
index a109f2d8e26..246085913e1 100644
--- a/chromium/third_party/webrtc/media/base/rtpdump.cc
+++ b/chromium/third_party/webrtc/media/base/rtpdump.cc
@@ -28,13 +28,12 @@ namespace cricket {
 
 const char RtpDumpFileHeader::kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n";
 
-RtpDumpFileHeader::RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p)
-    : start_sec(start_ms / 1000),
-      start_usec(start_ms % 1000 * 1000),
+RtpDumpFileHeader::RtpDumpFileHeader(int64_t start_ms, uint32_t s, uint16_t p)
+    : start_sec(static_cast<uint32_t>(start_ms / 1000)),
+      start_usec(static_cast<uint32_t>(start_ms % 1000 * 1000)),
       source(s),
       port(p),
-      padding(0) {
-}
+      padding(0) {}
 
 void RtpDumpFileHeader::WriteToByteBuffer(rtc::ByteBufferWriter* buf) {
   buf->WriteUInt32(start_sec);
@@ -44,7 +43,7 @@ void RtpDumpFileHeader::WriteToByteBuffer(rtc::ByteBufferWriter* buf) {
   buf->WriteUInt16(padding);
 }
 
-static const uint32_t kDefaultTimeIncrease = 30;
+static const int kDefaultTimeIncrease = 30;
 
 bool RtpDumpPacket::IsValidRtpPacket() const {
   return original_data_len >= data.size() &&
@@ -162,7 +161,7 @@ rtc::StreamResult RtpDumpReader::ReadFileHeader() {
     uint32_t start_usec;
     buf.ReadUInt32(&start_sec);
     buf.ReadUInt32(&start_usec);
-    start_time_ms_ = start_sec * 1000 + start_usec / 1000;
+    start_time_ms_ = static_cast<int64_t>(start_sec * 1000 + start_usec / 1000);
     // Increase the length by 1 since first_line does not contain the ending \n.
     first_line_and_file_header_len_ = first_line.size() + 1 + sizeof(header);
   }
@@ -305,9 +304,8 @@ RtpDumpWriter::RtpDumpWriter(rtc::StreamInterface* stream)
     : stream_(stream),
       packet_filter_(PF_ALL),
       file_header_written_(false),
-      start_time_ms_(rtc::Time()),
-      warn_slow_writes_delay_(kWarnSlowWritesDelayMs) {
-}
+      start_time_ms_(rtc::TimeMillis()),
+      warn_slow_writes_delay_(kWarnSlowWritesDelayMs) {}
 
 void RtpDumpWriter::set_packet_filter(int filter) {
   packet_filter_ = filter;
@@ -315,7 +313,7 @@ void RtpDumpWriter::set_packet_filter(int filter) {
 }
 
 uint32_t RtpDumpWriter::GetElapsedTime() const {
-  return rtc::TimeSince(start_time_ms_);
+  return static_cast<uint32_t>(rtc::TimeSince(start_time_ms_));
 }
 
 rtc::StreamResult RtpDumpWriter::WriteFileHeader() {
@@ -327,7 +325,7 @@ rtc::StreamResult RtpDumpWriter::WriteFileHeader() {
   }
 
   rtc::ByteBufferWriter buf;
-  RtpDumpFileHeader file_header(rtc::Time(), 0, 0);
+  RtpDumpFileHeader file_header(rtc::TimeMillis(), 0, 0);
   file_header.WriteToByteBuffer(&buf);
   return WriteToStream(buf.Data(), buf.Length());
 }
@@ -395,10 +393,10 @@ size_t RtpDumpWriter::FilterPacket(const void* data, size_t data_len,
 
 rtc::StreamResult RtpDumpWriter::WriteToStream(
     const void* data, size_t data_len) {
-  uint32_t before = rtc::Time();
+  int64_t before = rtc::TimeMillis();
   rtc::StreamResult result =
       stream_->WriteAll(data, data_len, NULL, NULL);
-  uint32_t delay = rtc::TimeSince(before);
+  int64_t delay = rtc::TimeSince(before);
   if (delay >= warn_slow_writes_delay_) {
     LOG(LS_WARNING) << "Slow RtpDump: took " << delay << "ms to write "
                     << data_len << " bytes.";
diff --git a/chromium/third_party/webrtc/media/base/rtpdump.h b/chromium/third_party/webrtc/media/base/rtpdump.h
index 4ce479277af..8ea7800e9c7 100644
--- a/chromium/third_party/webrtc/media/base/rtpdump.h
+++ b/chromium/third_party/webrtc/media/base/rtpdump.h
@@ -18,6 +18,7 @@
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/stream.h"
 
 namespace cricket {
@@ -39,7 +40,7 @@ enum RtpDumpPacketFilter {
 };
 
 struct RtpDumpFileHeader {
-  RtpDumpFileHeader(uint32_t start_ms, uint32_t s, uint16_t p);
+  RtpDumpFileHeader(int64_t start_ms, uint32_t s, uint16_t p);
   void WriteToByteBuffer(rtc::ByteBufferWriter* buf);
 
   static const char kFirstLine[];
@@ -112,7 +113,7 @@ class RtpDumpReader {
   rtc::StreamInterface* stream_;
   bool file_header_read_;
   size_t first_line_and_file_header_len_;
-  uint32_t start_time_ms_;
+  int64_t start_time_ms_;
   uint32_t ssrc_override_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
@@ -158,10 +159,10 @@ class RtpDumpLoopReader : public RtpDumpReader {
   // the previous dump packets in the input stream.
   uint32_t first_elapsed_time_;
   int first_rtp_seq_num_;
-  uint32_t first_rtp_timestamp_;
+  int64_t first_rtp_timestamp_;
   uint32_t prev_elapsed_time_;
   int prev_rtp_seq_num_;
-  uint32_t prev_rtp_timestamp_;
+  int64_t prev_rtp_timestamp_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpLoopReader);
 };
@@ -206,9 +207,9 @@ class RtpDumpWriter {
   rtc::StreamInterface* stream_;
   int packet_filter_;
   bool file_header_written_;
-  uint32_t start_time_ms_;  // Time when the record starts.
+  int64_t start_time_ms_;  // Time when the record starts.
   // If writing to the stream takes longer than this many ms, log a warning.
-  uint32_t warn_slow_writes_delay_;
+  int64_t warn_slow_writes_delay_;
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
 };
 
diff --git a/chromium/third_party/webrtc/media/base/videoadapter.cc b/chromium/third_party/webrtc/media/base/videoadapter.cc
index 797a876f2b4..9c3837c5729 100644
--- a/chromium/third_party/webrtc/media/base/videoadapter.cc
+++ b/chromium/third_party/webrtc/media/base/videoadapter.cc
@@ -13,69 +13,77 @@
 #include <algorithm>
 #include <limits>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/videocommon.h"
 
 namespace {
 
+struct Fraction {
+  int numerator;
+  int denominator;
+};
+
 // Scale factors optimized for in libYUV that we accept.
 // Must be sorted in decreasing scale factors for FindScaleLargerThan to work.
-const float kScaleFactors[] = {
-    1.f / 1.f,   // Full size.
-    3.f / 4.f,   // 3/4 scale.
-    1.f / 2.f,   // 1/2 scale.
-    3.f / 8.f,   // 3/8 scale.
-    1.f / 4.f,   // 1/4 scale.
-    3.f / 16.f,  // 3/16 scale.
+const Fraction kScaleFractions[] = {
+  {1, 1},
+  {3, 4},
+  {1, 2},
+  {3, 8},
+  {1, 4},
+  {3, 16},
 };
 
-float FindScaleLessThanOrEqual(int width,
-                               int height,
-                               int target_num_pixels,
-                               int* resulting_number_of_pixels) {
+// Round |valueToRound| to a multiple of |multiple|. Prefer rounding upwards,
+// but never more than |maxValue|.
+int roundUp(int valueToRound, int multiple, int maxValue) {
+  const int roundedValue = (valueToRound + multiple - 1) / multiple * multiple;
+  return roundedValue <= maxValue ? roundedValue
+                                  : (maxValue / multiple * multiple);
+}
+
+Fraction FindScaleLessThanOrEqual(int input_num_pixels, int target_num_pixels) {
   float best_distance = std::numeric_limits<float>::max();
-  float best_scale = 0.0f;  // Default to 0 if nothing matches.
-  float pixels = width * height;
-  float best_number_of_pixels = 0.0f;
-  for (const auto& scale : kScaleFactors) {
-    float test_num_pixels = pixels * scale * scale;
+  Fraction best_scale = {0, 1};  // Default to 0 if nothing matches.
+  for (const auto& fraction : kScaleFractions) {
+    const float scale =
+        fraction.numerator / static_cast<float>(fraction.denominator);
+    float test_num_pixels = input_num_pixels * scale * scale;
     float diff = target_num_pixels - test_num_pixels;
     if (diff < 0) {
       continue;
     }
     if (diff < best_distance) {
       best_distance = diff;
-      best_scale = scale;
-      best_number_of_pixels = test_num_pixels;
+      best_scale = fraction;
       if (best_distance == 0) {  // Found exact match.
         break;
       }
     }
   }
-  if (resulting_number_of_pixels) {
-    *resulting_number_of_pixels = static_cast<int>(best_number_of_pixels + .5f);
-  }
   return best_scale;
 }
 
-float FindScaleLargerThan(int width,
-                          int height,
-                          int target_num_pixels,
-                          int* resulting_number_of_pixels) {
+Fraction FindScaleLargerThan(int input_num_pixels,
+                             int target_num_pixels,
+                             int* resulting_number_of_pixels) {
   float best_distance = std::numeric_limits<float>::max();
-  float best_scale = 1.f;  // Default to unscaled if nothing matches.
-  float pixels = width * height;
-  float best_number_of_pixels = pixels;  // Default to input number of pixels.
-  for (const auto& scale : kScaleFactors) {
-    float test_num_pixels = pixels * scale * scale;
+  Fraction best_scale = {1, 1};  // Default to unscaled if nothing matches.
+  // Default to input number of pixels.
+  float best_number_of_pixels = input_num_pixels;
+  for (const auto& fraction : kScaleFractions) {
+    const float scale =
+        fraction.numerator / static_cast<float>(fraction.denominator);
+    float test_num_pixels = input_num_pixels * scale * scale;
     float diff = test_num_pixels - target_num_pixels;
     if (diff <= 0) {
       break;
     }
     if (diff < best_distance) {
       best_distance = diff;
-      best_scale = scale;
+      best_scale = fraction;
       best_number_of_pixels = test_num_pixels;
     }
   }
@@ -84,21 +92,36 @@ float FindScaleLargerThan(int width,
   return best_scale;
 }
 
+Fraction FindScale(int input_num_pixels,
+                   int max_pixel_count_step_up,
+                   int max_pixel_count) {
+  // Try scale just above |max_pixel_count_step_up_|.
+  if (max_pixel_count_step_up > 0) {
+    int resulting_pixel_count;
+    const Fraction scale = FindScaleLargerThan(
+        input_num_pixels, max_pixel_count_step_up, &resulting_pixel_count);
+    if (resulting_pixel_count <= max_pixel_count)
+      return scale;
+  }
+  // Return largest scale below |max_pixel_count|.
+  return FindScaleLessThanOrEqual(input_num_pixels, max_pixel_count);
+}
+
 }  // namespace
 
 namespace cricket {
 
 VideoAdapter::VideoAdapter()
-    : output_num_pixels_(std::numeric_limits<int>::max()),
-      frames_in_(0),
+    : frames_in_(0),
       frames_out_(0),
       frames_scaled_(0),
       adaption_changes_(0),
       previous_width_(0),
       previous_height_(0),
+      input_interval_(0),
       interval_next_frame_(0),
-      format_request_max_pixel_count_(std::numeric_limits<int>::max()),
-      resolution_request_max_pixel_count_(std::numeric_limits<int>::max()) {}
+      resolution_request_max_pixel_count_(std::numeric_limits<int>::max()),
+      resolution_request_max_pixel_count_step_up_(0) {}
 
 VideoAdapter::~VideoAdapter() {}
 
@@ -106,57 +129,43 @@ void VideoAdapter::SetExpectedInputFrameInterval(int64_t interval) {
   // TODO(perkj): Consider measuring input frame rate instead.
   // Frame rate typically varies depending on lighting.
   rtc::CritScope cs(&critical_section_);
-  input_format_.interval = interval;
-}
-
-void VideoAdapter::SetInputFormat(const VideoFormat& format) {
-  bool is_resolution_change = (input_format().width != format.width ||
-                               input_format().height != format.height);
-  int64_t old_input_interval = input_format_.interval;
-  input_format_ = format;
-  output_format_.interval =
-      std::max(output_format_.interval, input_format_.interval);
-  if (old_input_interval != input_format_.interval) {
-    LOG(LS_INFO) << "VAdapt input interval changed from "
-      << old_input_interval << " to " << input_format_.interval;
-  }
-  if (is_resolution_change) {
-    // Trigger the adaptation logic again, to potentially reset the adaptation
-    // state for things like view requests that may not longer be capping
-    // output (or may now cap output).
-    Adapt(std::min(format_request_max_pixel_count_,
-                   resolution_request_max_pixel_count_),
-          0);
-  }
-}
-
-const VideoFormat& VideoAdapter::input_format() const {
-  rtc::CritScope cs(&critical_section_);
-  return input_format_;
+  input_interval_ = interval;
 }
 
-VideoFormat VideoAdapter::AdaptFrameResolution(int in_width, int in_height) {
+void VideoAdapter::AdaptFrameResolution(int in_width,
+                                        int in_height,
+                                        int* cropped_width,
+                                        int* cropped_height,
+                                        int* out_width,
+                                        int* out_height) {
   rtc::CritScope cs(&critical_section_);
   ++frames_in_;
 
-  SetInputFormat(VideoFormat(
-      in_width, in_height, input_format_.interval, input_format_.fourcc));
+  // The max output pixel count is the minimum of the requests from
+  // OnOutputFormatRequest and OnResolutionRequest.
+  int max_pixel_count = resolution_request_max_pixel_count_;
+  if (requested_format_) {
+    max_pixel_count = std::min(
+        max_pixel_count, requested_format_->width * requested_format_->height);
+  }
 
   // Drop the input frame if necessary.
   bool should_drop = false;
-  if (!output_num_pixels_) {
+  if (max_pixel_count == 0) {
     // Drop all frames as the output format is 0x0.
     should_drop = true;
-  } else {
+  } else if (requested_format_ && requested_format_->interval > 0) {
     // Drop some frames based on input fps and output fps.
     // Normally output fps is less than input fps.
-    interval_next_frame_ += input_format_.interval;
-    if (output_format_.interval > 0) {
-      if (interval_next_frame_ >= output_format_.interval) {
-        interval_next_frame_ %= output_format_.interval;
-      } else {
-        should_drop = true;
-      }
+    interval_next_frame_ += input_interval_;
+    if (interval_next_frame_ >= requested_format_->interval) {
+      interval_next_frame_ -= requested_format_->interval;
+      // Reset |interval_next_frame_| if it accumulates too much to avoid
+      // "catching up" behaviour.
+      if (interval_next_frame_ >= requested_format_->interval)
+        interval_next_frame_ = 0;
+    } else {
+      should_drop = true;
     }
   }
   if (should_drop) {
@@ -170,48 +179,79 @@ VideoFormat VideoAdapter::AdaptFrameResolution(int in_width, int in_height) {
                    << " Changes: " << adaption_changes_
                    << " Input: " << in_width
                    << "x" << in_height
-                   << " i" << input_format_.interval
-                   << " Output: i" << output_format_.interval;
+                   << " i" << input_interval_
+                   << " Output: i"
+                   << (requested_format_ ? requested_format_->interval : 0);
     }
 
-    return VideoFormat();  // Drop frame.
+    // Drop frame.
+    *cropped_width = 0;
+    *cropped_height = 0;
+    *out_width = 0;
+    *out_height = 0;
+    return;
+  }
+
+  // Calculate how the input should be cropped.
+  if (!requested_format_ ||
+      requested_format_->width == 0 || requested_format_->height == 0) {
+    *cropped_width = in_width;
+    *cropped_height = in_height;
+  } else {
+    // Adjust |requested_format_| orientation to match input.
+    if ((in_width > in_height) !=
+        (requested_format_->width > requested_format_->height)) {
+      std::swap(requested_format_->width, requested_format_->height);
+    }
+    const float requested_aspect =
+        requested_format_->width /
+        static_cast<float>(requested_format_->height);
+    *cropped_width =
+        std::min(in_width, static_cast<int>(in_height * requested_aspect));
+    *cropped_height =
+        std::min(in_height, static_cast<int>(in_width / requested_aspect));
   }
 
-  const float scale = FindScaleLessThanOrEqual(in_width, in_height,
-                                               output_num_pixels_, nullptr);
-  const int output_width = static_cast<int>(in_width * scale + .5f);
-  const int output_height = static_cast<int>(in_height * scale + .5f);
+  // Find best scale factor.
+  const Fraction scale =
+      FindScale(*cropped_width * *cropped_height,
+                resolution_request_max_pixel_count_step_up_, max_pixel_count);
+
+  // Adjust cropping slightly to get even integer output size and a perfect
+  // scale factor.
+  *cropped_width = roundUp(*cropped_width, scale.denominator, in_width);
+  *cropped_height = roundUp(*cropped_height, scale.denominator, in_height);
+  RTC_DCHECK_EQ(0, *cropped_width % scale.denominator);
+  RTC_DCHECK_EQ(0, *cropped_height % scale.denominator);
+
+  // Calculate final output size.
+  *out_width = *cropped_width / scale.denominator * scale.numerator;
+  *out_height = *cropped_height / scale.denominator * scale.numerator;
 
   ++frames_out_;
-  if (scale != 1)
+  if (scale.numerator != scale.denominator)
     ++frames_scaled_;
 
-  if (previous_width_ && (previous_width_ != output_width ||
-                          previous_height_ != output_height)) {
+  if (previous_width_ && (previous_width_ != *out_width ||
+                          previous_height_ != *out_height)) {
     ++adaption_changes_;
     LOG(LS_INFO) << "Frame size changed: scaled " << frames_scaled_ << " / out "
                  << frames_out_ << " / in " << frames_in_
                  << " Changes: " << adaption_changes_ << " Input: " << in_width
-                 << "x" << in_height << " i" << input_format_.interval
-                 << " Scale: " << scale << " Output: " << output_width << "x"
-                 << output_height << " i" << output_format_.interval;
+                 << "x" << in_height << " i" << input_interval_
+                 << " Scale: " << scale.numerator << "/" << scale.denominator
+                 << " Output: " << *out_width << "x" << *out_height << " i"
+                 << (requested_format_ ? requested_format_->interval : 0);
   }
 
-  output_format_.width = output_width;
-  output_format_.height = output_height;
-  previous_width_ = output_width;
-  previous_height_ = output_height;
-
-  return output_format_;
+  previous_width_ = *out_width;
+  previous_height_ = *out_height;
 }
 
 void VideoAdapter::OnOutputFormatRequest(const VideoFormat& format) {
   rtc::CritScope cs(&critical_section_);
-  format_request_max_pixel_count_ = format.width * format.height;
-  output_format_.interval = format.interval;
-  Adapt(std::min(format_request_max_pixel_count_,
-                 resolution_request_max_pixel_count_),
-        0);
+  requested_format_ = rtc::Optional<VideoFormat>(format);
+  interval_next_frame_ = 0;
 }
 
 void VideoAdapter::OnResolutionRequest(
@@ -220,44 +260,8 @@ void VideoAdapter::OnResolutionRequest(
   rtc::CritScope cs(&critical_section_);
   resolution_request_max_pixel_count_ =
       max_pixel_count.value_or(std::numeric_limits<int>::max());
-  Adapt(std::min(format_request_max_pixel_count_,
-                 resolution_request_max_pixel_count_),
-        max_pixel_count_step_up.value_or(0));
-}
-
-bool VideoAdapter::Adapt(int max_num_pixels, int max_pixel_count_step_up) {
-  float scale_lower =
-      FindScaleLessThanOrEqual(input_format_.width, input_format_.height,
-                               max_num_pixels, &max_num_pixels);
-  float scale_upper =
-      max_pixel_count_step_up > 0
-          ? FindScaleLargerThan(input_format_.width, input_format_.height,
-                                max_pixel_count_step_up,
-                                &max_pixel_count_step_up)
-          : 1.f;
-
-  bool use_max_pixel_count_step_up =
-      max_pixel_count_step_up > 0 && max_num_pixels > max_pixel_count_step_up;
-
-  int old_num_pixels = output_num_pixels_;
-  output_num_pixels_ =
-      use_max_pixel_count_step_up ? max_pixel_count_step_up : max_num_pixels;
-  // Log the new size.
-  float scale = use_max_pixel_count_step_up ? scale_upper : scale_lower;
-  int new_width = static_cast<int>(input_format_.width * scale + .5f);
-  int new_height = static_cast<int>(input_format_.height * scale + .5f);
-
-  bool changed = output_num_pixels_ != old_num_pixels;
-  LOG(LS_INFO) << "OnResolutionRequest: "
-               << " Max pixels: " << max_num_pixels
-               << " Max pixels step up: " << max_pixel_count_step_up
-               << " Output Pixels: " << output_num_pixels_
-               << " Input: " << input_format_.width << "x"
-               << input_format_.height << " Scale: " << scale
-               << " Resolution: " << new_width << "x" << new_height
-               << " Changed: " << (changed ? "true" : "false");
-
-  return changed;
+  resolution_request_max_pixel_count_step_up_ =
+      max_pixel_count_step_up.value_or(0);
 }
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/media/base/videoadapter.h b/chromium/third_party/webrtc/media/base/videoadapter.h
index b7aba1406a4..2db0ada2713 100644
--- a/chromium/third_party/webrtc/media/base/videoadapter.h
+++ b/chromium/third_party/webrtc/media/base/videoadapter.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
 #define WEBRTC_MEDIA_BASE_VIDEOADAPTER_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/media/base/videocommon.h"
@@ -31,12 +32,23 @@ class VideoAdapter {
   // interval.
   void SetExpectedInputFrameInterval(int64_t interval);
 
-  // Return the adapted resolution given the input resolution. The returned
-  // resolution will be 0x0 if the frame should be dropped.
-  VideoFormat AdaptFrameResolution(int in_width, int in_height);
+  // Return the adapted resolution given the input resolution. The input
+  // resolution should first be cropped to the specified resolution, and then
+  // scaled to the final output resolution. The output resolution will be 0x0 if
+  // the frame should be dropped.
+  void AdaptFrameResolution(int in_width,
+                            int in_height,
+                            int* cropped_width,
+                            int* cropped_height,
+                            int* out_width,
+                            int* out_height);
 
   // Requests the output frame size and frame interval from
-  // |AdaptFrameResolution| to not be larger than |format|.
+  // |AdaptFrameResolution| to not be larger than |format|. Also, the input
+  // frame size will be cropped to match the requested aspect ratio. The
+  // requested aspect ratio is orientation agnostic and will be adjusted to
+  // maintain the input orientation, so it doesn't matter if e.g. 1280x720 or
+  // 720x1280 is requested.
   void OnOutputFormatRequest(const VideoFormat& format);
 
   // Requests the output frame size from |AdaptFrameResolution| to not have
@@ -45,28 +57,22 @@ class VideoAdapter {
   void OnResolutionRequest(rtc::Optional<int> max_pixel_count,
                            rtc::Optional<int> max_pixel_count_step_up);
 
-  const VideoFormat& input_format() const;
-
  private:
-  void SetInputFormat(const VideoFormat& format);
-  bool Adapt(int max_num_pixels, int max_pixel_count_step_up);
-
-  VideoFormat input_format_;
-  VideoFormat output_format_;
-  int output_num_pixels_;
   int frames_in_;         // Number of input frames.
   int frames_out_;        // Number of output frames.
   int frames_scaled_;     // Number of frames scaled.
   int adaption_changes_;  // Number of changes in scale factor.
   int previous_width_;    // Previous adapter output width.
   int previous_height_;   // Previous adapter output height.
-  int64_t interval_next_frame_;
+  int input_interval_ GUARDED_BY(critical_section_);
+  int64_t interval_next_frame_ GUARDED_BY(critical_section_);
 
   // Max number of pixels requested via calls to OnOutputFormatRequest,
   // OnResolutionRequest respectively.
   // The adapted output format is the minimum of these.
-  int format_request_max_pixel_count_;
-  int resolution_request_max_pixel_count_;
+  rtc::Optional<VideoFormat> requested_format_ GUARDED_BY(critical_section_);
+  int resolution_request_max_pixel_count_ GUARDED_BY(critical_section_);
+  int resolution_request_max_pixel_count_step_up_ GUARDED_BY(critical_section_);
 
   // The critical section to protect the above variables.
   rtc::CriticalSection critical_section_;
diff --git a/chromium/third_party/webrtc/media/base/videoadapter_unittest.cc b/chromium/third_party/webrtc/media/base/videoadapter_unittest.cc
index 1ad58d165c0..92b4a558b9b 100644
--- a/chromium/third_party/webrtc/media/base/videoadapter_unittest.cc
+++ b/chromium/third_party/webrtc/media/base/videoadapter_unittest.cc
@@ -10,6 +10,7 @@
 
 #include <limits.h>  // For INT_MAX
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -49,8 +50,10 @@ class VideoAdapterTest : public testing::Test {
       int dropped_frames;
       bool last_adapt_was_no_op;
 
-      int adapted_width;
-      int adapted_height;
+      int cropped_width;
+      int cropped_height;
+      int out_width;
+      int out_height;
     };
 
     explicit VideoCapturerListener(VideoAdapter* adapter)
@@ -65,12 +68,21 @@ class VideoAdapterTest : public testing::Test {
       rtc::CritScope lock(&crit_);
       const int in_width = captured_frame->width;
       const int in_height = abs(captured_frame->height);
-      const VideoFormat adapted_format =
-          video_adapter_->AdaptFrameResolution(in_width, in_height);
-      if (!adapted_format.IsSize0x0()) {
-        adapted_format_ = adapted_format;
-        last_adapt_was_no_op_ = (in_width == adapted_format.width &&
-                                 in_height == adapted_format.height);
+      int cropped_width;
+      int cropped_height;
+      int out_width;
+      int out_height;
+      video_adapter_->AdaptFrameResolution(in_width, in_height,
+                                           &cropped_width, &cropped_height,
+                                           &out_width, &out_height);
+      if (out_width != 0 && out_height != 0) {
+        cropped_width_ = cropped_width;
+        cropped_height_ = cropped_height;
+        out_width_ = out_width;
+        out_height_ = out_height;
+        last_adapt_was_no_op_ =
+            (in_width == cropped_width && in_height == cropped_height &&
+             in_width == out_width && in_height == out_height);
       } else {
         ++dropped_frames_;
       }
@@ -83,20 +95,20 @@ class VideoAdapterTest : public testing::Test {
       stats.captured_frames = captured_frames_;
       stats.dropped_frames = dropped_frames_;
       stats.last_adapt_was_no_op = last_adapt_was_no_op_;
-      if (!adapted_format_.IsSize0x0()) {
-        stats.adapted_width = adapted_format_.width;
-        stats.adapted_height = adapted_format_.height;
-      } else {
-        stats.adapted_width = stats.adapted_height = -1;
-      }
-
+      stats.cropped_width = cropped_width_;
+      stats.cropped_height = cropped_height_;
+      stats.out_width = out_width_;
+      stats.out_height = out_height_;
       return stats;
     }
 
    private:
     rtc::CriticalSection crit_;
     VideoAdapter* video_adapter_;
-    VideoFormat adapted_format_;
+    int cropped_width_;
+    int cropped_height_;
+    int out_width_;
+    int out_height_;
     int captured_frames_;
     int dropped_frames_;
     bool last_adapt_was_no_op_;
@@ -104,20 +116,28 @@ class VideoAdapterTest : public testing::Test {
 
 
   void VerifyAdaptedResolution(const VideoCapturerListener::Stats& stats,
-                               int width,
-                               int height) {
-    EXPECT_EQ(width, stats.adapted_width);
-    EXPECT_EQ(height, stats.adapted_height);
+                               int cropped_width,
+                               int cropped_height,
+                               int out_width,
+                               int out_height) {
+    EXPECT_EQ(cropped_width, stats.cropped_width);
+    EXPECT_EQ(cropped_height, stats.cropped_height);
+    EXPECT_EQ(out_width, stats.out_width);
+    EXPECT_EQ(out_height, stats.out_height);
   }
 
   std::unique_ptr<FakeVideoCapturer> capturer_;
   VideoAdapter adapter_;
+  int cropped_width_;
+  int cropped_height_;
+  int out_width_;
+  int out_height_;
   std::unique_ptr<VideoCapturerListener> listener_;
   VideoFormat capture_format_;
 };
 
-// Do not adapt the frame rate or the resolution. Expect no frame drop and no
-// resolution change.
+// Do not adapt the frame rate or the resolution. Expect no frame drop, no
+// cropping, and no resolution change.
 TEST_F(VideoAdapterTest, AdaptNothing) {
   EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
   for (int i = 0; i < 10; ++i)
@@ -127,7 +147,8 @@ TEST_F(VideoAdapterTest, AdaptNothing) {
   VideoCapturerListener::Stats stats = listener_->GetStats();
   EXPECT_GE(stats.captured_frames, 10);
   EXPECT_EQ(0, stats.dropped_frames);
-  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
+  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
+                          capture_format_.width, capture_format_.height);
   EXPECT_TRUE(stats.last_adapt_was_no_op);
 }
 
@@ -143,7 +164,8 @@ TEST_F(VideoAdapterTest, AdaptZeroInterval) {
   VideoCapturerListener::Stats stats = listener_->GetStats();
   EXPECT_GE(stats.captured_frames, 10);
   EXPECT_EQ(0, stats.dropped_frames);
-  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
+  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
+                          capture_format_.width, capture_format_.height);
 }
 
 // Adapt the frame rate to be half of the capture rate at the beginning. Expect
@@ -160,7 +182,8 @@ TEST_F(VideoAdapterTest, AdaptFramerate) {
   VideoCapturerListener::Stats stats = listener_->GetStats();
   EXPECT_GE(stats.captured_frames, 10);
   EXPECT_EQ(stats.captured_frames / 2, stats.dropped_frames);
-  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
+  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
+                          capture_format_.width, capture_format_.height);
 }
 
 // Adapt the frame rate to be half of the capture rate at the beginning. Expect
@@ -178,7 +201,8 @@ TEST_F(VideoAdapterTest, AdaptFramerateVariable) {
   EXPECT_GE(stats.captured_frames, 30);
   // Verify 2 / 3 kept (20) and 1 / 3 dropped (10).
   EXPECT_EQ(stats.captured_frames * 1 / 3, stats.dropped_frames);
-  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height);
+  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
+                          capture_format_.width, capture_format_.height);
 }
 
 // Adapt the frame rate to be half of the capture rate after capturing no less
@@ -205,39 +229,49 @@ TEST_F(VideoAdapterTest, AdaptFramerateOntheFly) {
   EXPECT_GT(listener_->GetStats().dropped_frames, 0);
 }
 
-// Set a very high output pixel resolution. Expect no resolution change.
+// Set a very high output pixel resolution. Expect no cropping or resolution
+// change.
 TEST_F(VideoAdapterTest, AdaptFrameResolutionHighLimit) {
   VideoFormat output_format = capture_format_;
-  output_format.width = 2560;
-  output_format.height = 2560;
+  output_format.width *= 10;
+  output_format.height *= 10;
   adapter_.OnOutputFormatRequest(output_format);
-  VideoFormat adapted_format = adapter_.AdaptFrameResolution(
-      capture_format_.width, capture_format_.height);
-  EXPECT_EQ(capture_format_.width, adapted_format.width);
-  EXPECT_EQ(capture_format_.height, adapted_format.height);
+  adapter_.AdaptFrameResolution(capture_format_.width, capture_format_.height,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(capture_format_.width, cropped_width_);
+  EXPECT_EQ(capture_format_.height, cropped_height_);
+  EXPECT_EQ(capture_format_.width, out_width_);
+  EXPECT_EQ(capture_format_.height, out_height_);
 }
 
 // Adapt the frame resolution to be the same as capture resolution. Expect no
-// resolution change.
+// cropping or resolution change.
 TEST_F(VideoAdapterTest, AdaptFrameResolutionIdentical) {
   adapter_.OnOutputFormatRequest(capture_format_);
-  const VideoFormat adapted_format = adapter_.AdaptFrameResolution(
-      capture_format_.width, capture_format_.height);
-  EXPECT_EQ(capture_format_.width, adapted_format.width);
-  EXPECT_EQ(capture_format_.height, adapted_format.height);
+  adapter_.AdaptFrameResolution(capture_format_.width, capture_format_.height,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(capture_format_.width, cropped_width_);
+  EXPECT_EQ(capture_format_.height, cropped_height_);
+  EXPECT_EQ(capture_format_.width, out_width_);
+  EXPECT_EQ(capture_format_.height, out_height_);
 }
 
 // Adapt the frame resolution to be a quarter of the capture resolution. Expect
-// resolution change.
+// no cropping, but a resolution change.
 TEST_F(VideoAdapterTest, AdaptFrameResolutionQuarter) {
   VideoFormat request_format = capture_format_;
   request_format.width /= 2;
   request_format.height /= 2;
   adapter_.OnOutputFormatRequest(request_format);
-  const VideoFormat adapted_format = adapter_.AdaptFrameResolution(
-      request_format.width, request_format.height);
-  EXPECT_EQ(request_format.width, adapted_format.width);
-  EXPECT_EQ(request_format.height, adapted_format.height);
+  adapter_.AdaptFrameResolution(capture_format_.width, capture_format_.height,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(capture_format_.width, cropped_width_);
+  EXPECT_EQ(capture_format_.height, cropped_height_);
+  EXPECT_EQ(request_format.width, out_width_);
+  EXPECT_EQ(request_format.height, out_height_);
 }
 
 // Adapt the pixel resolution to 0. Expect frame drop.
@@ -246,14 +280,15 @@ TEST_F(VideoAdapterTest, AdaptFrameResolutionDrop) {
   output_format.width = 0;
   output_format.height = 0;
   adapter_.OnOutputFormatRequest(output_format);
-  EXPECT_TRUE(
-      adapter_
-          .AdaptFrameResolution(capture_format_.width, capture_format_.height)
-          .IsSize0x0());
+  adapter_.AdaptFrameResolution(capture_format_.width, capture_format_.height,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(0, out_width_);
+  EXPECT_EQ(0, out_height_);
 }
 
 // Adapt the frame resolution to be a quarter of the capture resolution at the
-// beginning. Expect resolution change.
+// beginning. Expect no cropping but a resolution change.
 TEST_F(VideoAdapterTest, AdaptResolution) {
   VideoFormat request_format = capture_format_;
   request_format.width /= 2;
@@ -263,10 +298,11 @@ TEST_F(VideoAdapterTest, AdaptResolution) {
   for (int i = 0; i < 10; ++i)
     capturer_->CaptureFrame();
 
-  // Verify no frame drop and resolution change.
+  // Verify no frame drop, no cropping, and resolution change.
   VideoCapturerListener::Stats stats = listener_->GetStats();
   EXPECT_EQ(0, stats.dropped_frames);
-  VerifyAdaptedResolution(stats, request_format.width, request_format.height);
+  VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
+                          request_format.width, request_format.height);
 }
 
 // Adapt the frame resolution to be a quarter of the capture resolution after
@@ -280,8 +316,9 @@ TEST_F(VideoAdapterTest, AdaptResolutionOnTheFly) {
     capturer_->CaptureFrame();
 
   // Verify no resolution change before adaptation.
-  VerifyAdaptedResolution(
-      listener_->GetStats(), request_format.width, request_format.height);
+  VerifyAdaptedResolution(listener_->GetStats(),
+                          capture_format_.width, capture_format_.height,
+                          request_format.width, request_format.height);
 
   // Adapt the frame resolution.
   request_format.width /= 2;
@@ -291,8 +328,9 @@ TEST_F(VideoAdapterTest, AdaptResolutionOnTheFly) {
     capturer_->CaptureFrame();
 
   // Verify resolution change after adaptation.
-  VerifyAdaptedResolution(
-      listener_->GetStats(), request_format.width, request_format.height);
+  VerifyAdaptedResolution(listener_->GetStats(),
+                          capture_format_.width, capture_format_.height,
+                          request_format.width, request_format.height);
 }
 
 // Drop all frames.
@@ -312,104 +350,155 @@ TEST_F(VideoAdapterTest, DropAllFrames) {
 TEST_F(VideoAdapterTest, TestOnOutputFormatRequest) {
   VideoFormat format(640, 400, VideoFormat::FpsToInterval(30), 0);
   adapter_.SetExpectedInputFrameInterval(VideoFormat::FpsToInterval(30));
-  VideoFormat out_format =
-      adapter_.AdaptFrameResolution(format.width, format.height);
-  EXPECT_EQ(format, adapter_.input_format());
-  EXPECT_EQ(format, out_format);
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(400, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(400, out_height_);
 
   // Format request 640x400.
   format.height = 400;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(400, out_format.height);
-
-  // Request 1280x720, higher than input. Adapt nothing.
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(400, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(400, out_height_);
+
+  // Request 1280x720, higher than input, but aspect 16:9. Expect cropping but
+  // no scaling.
   format.width = 1280;
   format.height = 720;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(400, out_format.height);
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 
   // Request 0x0.
   format.width = 0;
   format.height = 0;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_TRUE(out_format.IsSize0x0());
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(0, out_width_);
+  EXPECT_EQ(0, out_height_);
 
-  // Request 320x200.
+  // Request 320x200. Expect scaling, but no cropping.
   format.width = 320;
   format.height = 200;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_EQ(320, out_format.width);
-  EXPECT_EQ(200, out_format.height);
-
-  // Request resolution of 2 / 3. Expect adapt down. Scaling to 1/3 is not
-  // optimized and not allowed.
-  format.width = (640 * 2 + 1) / 3;
-  format.height = (400 * 2 + 1) / 3;
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(400, cropped_height_);
+  EXPECT_EQ(320, out_width_);
+  EXPECT_EQ(200, out_height_);
+
+  // Request resolution close to 2/3 scale. Expect adapt down. Scaling to 2/3
+  // is not optimized and not allowed, therefore 1/2 scaling will be used
+  // instead.
+  format.width = 424;
+  format.height = 265;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_EQ(320, out_format.width);
-  EXPECT_EQ(200, out_format.height);
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(400, cropped_height_);
+  EXPECT_EQ(320, out_width_);
+  EXPECT_EQ(200, out_height_);
 
   // Request resolution of 3 / 8. Expect adapt down.
   format.width = 640 * 3 / 8;
   format.height = 400 * 3 / 8;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_EQ(640 * 3 / 8, out_format.width);
-  EXPECT_EQ(400 * 3 / 8, out_format.height);
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(400, cropped_height_);
+  EXPECT_EQ(640 * 3 / 8, out_width_);
+  EXPECT_EQ(400 * 3 / 8, out_height_);
 
   // Switch back up. Expect adapt.
   format.width = 320;
   format.height = 200;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_EQ(320, out_format.width);
-  EXPECT_EQ(200, out_format.height);
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(400, cropped_height_);
+  EXPECT_EQ(320, out_width_);
+  EXPECT_EQ(200, out_height_);
 
   // Format request 480x300.
   format.width = 480;
   format.height = 300;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 400);
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(300, out_format.height);
+  adapter_.AdaptFrameResolution(640, 400,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(400, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(300, out_height_);
 }
 
 TEST_F(VideoAdapterTest, TestViewRequestPlusCameraSwitch) {
   // Start at HD.
   VideoFormat format(1280, 720, VideoFormat::FpsToInterval(30), 0);
   adapter_.SetExpectedInputFrameInterval(VideoFormat::FpsToInterval(30));
-  VideoFormat out_format =
-      adapter_.AdaptFrameResolution(format.width, format.height);
-  EXPECT_EQ(format, adapter_.input_format());
-  EXPECT_EQ(out_format, adapter_.input_format());
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(1280, out_width_);
+  EXPECT_EQ(720, out_height_);
 
   // Format request for VGA.
   format.width = 640;
   format.height = 360;
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 
   // Now, the camera reopens at VGA.
   // Both the frame and the output format should be 640x360.
-  out_format = adapter_.AdaptFrameResolution(640, 360);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(640, 360,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 
   // And another view request comes in for 640x360, which should have no
   // real impact.
   adapter_.OnOutputFormatRequest(format);
-  out_format = adapter_.AdaptFrameResolution(640, 360);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(640, 360,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 }
 
 TEST_F(VideoAdapterTest, TestVGAWidth) {
@@ -418,130 +507,305 @@ TEST_F(VideoAdapterTest, TestVGAWidth) {
   adapter_.SetExpectedInputFrameInterval(VideoFormat::FpsToInterval(30));
   adapter_.OnOutputFormatRequest(format);
 
-  VideoFormat out_format = adapter_.AdaptFrameResolution(640, 480);
-  // At this point, we have to adapt down to something lower.
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  // Expect cropping.
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 
   // But if frames come in at 640x360, we shouldn't adapt them down.
-  out_format = adapter_.AdaptFrameResolution(640, 360);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(360, out_format.height);
-
-  out_format = adapter_.AdaptFrameResolution(640, 480);
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(640, 360,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
+
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 }
 
 TEST_F(VideoAdapterTest, TestOnResolutionRequestInSmallSteps) {
-  VideoFormat out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(1280, out_format.width);
-  EXPECT_EQ(720, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(1280, out_width_);
+  EXPECT_EQ(720, out_height_);
 
   // Adapt down one step.
   adapter_.OnResolutionRequest(rtc::Optional<int>(1280 * 720 - 1),
                                rtc::Optional<int>());
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(960, out_format.width);
-  EXPECT_EQ(540, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(960, out_width_);
+  EXPECT_EQ(540, out_height_);
 
   // Adapt down one step more.
   adapter_.OnResolutionRequest(rtc::Optional<int>(960 * 540 - 1),
                                rtc::Optional<int>());
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 
   // Adapt down one step more.
   adapter_.OnResolutionRequest(rtc::Optional<int>(640 * 360 - 1),
                                rtc::Optional<int>());
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(270, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(270, out_height_);
 
   // Adapt up one step.
   adapter_.OnResolutionRequest(rtc::Optional<int>(),
                                rtc::Optional<int>(480 * 270));
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 
   // Adapt up one step more.
   adapter_.OnResolutionRequest(rtc::Optional<int>(),
                                rtc::Optional<int>(640 * 360));
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(960, out_format.width);
-  EXPECT_EQ(540, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(960, out_width_);
+  EXPECT_EQ(540, out_height_);
 
   // Adapt up one step more.
   adapter_.OnResolutionRequest(rtc::Optional<int>(),
                                rtc::Optional<int>(960 * 720));
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(1280, out_format.width);
-  EXPECT_EQ(720, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(1280, out_width_);
+  EXPECT_EQ(720, out_height_);
 }
 
 TEST_F(VideoAdapterTest, TestOnResolutionRequestMaxZero) {
-  VideoFormat out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(1280, out_format.width);
-  EXPECT_EQ(720, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(1280, out_width_);
+  EXPECT_EQ(720, out_height_);
 
   adapter_.OnResolutionRequest(rtc::Optional<int>(0), rtc::Optional<int>());
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(0, out_format.width);
-  EXPECT_EQ(0, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(0, out_width_);
+  EXPECT_EQ(0, out_height_);
 }
 
 TEST_F(VideoAdapterTest, TestOnResolutionRequestInLargeSteps) {
   adapter_.OnResolutionRequest(rtc::Optional<int>(640 * 360 - 1),
                                rtc::Optional<int>());
-  VideoFormat out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(270, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(270, out_height_);
 
   adapter_.OnResolutionRequest(rtc::Optional<int>(),
                                rtc::Optional<int>(960 * 720));
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(1280, out_format.width);
-  EXPECT_EQ(720, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(1280, out_width_);
+  EXPECT_EQ(720, out_height_);
 }
 
 TEST_F(VideoAdapterTest, TestOnOutputFormatRequestCapsMaxResolution) {
   adapter_.OnResolutionRequest(rtc::Optional<int>(640 * 360 - 1),
                                rtc::Optional<int>());
-  VideoFormat out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(270, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(270, out_height_);
 
   VideoFormat new_format(640, 360, VideoFormat::FpsToInterval(30), FOURCC_I420);
   adapter_.SetExpectedInputFrameInterval(VideoFormat::FpsToInterval(30));
   adapter_.OnOutputFormatRequest(new_format);
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(270, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(270, out_height_);
 
   adapter_.OnResolutionRequest(rtc::Optional<int>(),
                                rtc::Optional<int>(960 * 720));
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(640, out_format.width);
-  EXPECT_EQ(360, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
 }
 
 TEST_F(VideoAdapterTest, TestOnResolutionRequestReset) {
-  VideoFormat out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(1280, out_format.width);
-  EXPECT_EQ(720, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(1280, out_width_);
+  EXPECT_EQ(720, out_height_);
 
   adapter_.OnResolutionRequest(rtc::Optional<int>(640 * 360 - 1),
                                rtc::Optional<int>());
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(480, out_format.width);
-  EXPECT_EQ(270, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(270, out_height_);
 
   adapter_.OnResolutionRequest(rtc::Optional<int>(), rtc::Optional<int>());
-  out_format = adapter_.AdaptFrameResolution(1280, 720);
-  EXPECT_EQ(1280, out_format.width);
-  EXPECT_EQ(720, out_format.height);
+  adapter_.AdaptFrameResolution(1280, 720,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(1280, cropped_width_);
+  EXPECT_EQ(720, cropped_height_);
+  EXPECT_EQ(1280, out_width_);
+  EXPECT_EQ(720, out_height_);
+}
+
+TEST_F(VideoAdapterTest, TestCroppingWithResolutionRequest) {
+  // Ask for 640x360 (16:9 aspect).
+  adapter_.SetExpectedInputFrameInterval(VideoFormat::FpsToInterval(30));
+  adapter_.OnOutputFormatRequest(
+      VideoFormat(640, 360, VideoFormat::FpsToInterval(30), FOURCC_I420));
+  // Send 640x480 (4:3 aspect).
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  // Expect cropping to 16:9 format and no scaling.
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
+
+  // Adapt down one step.
+  adapter_.OnResolutionRequest(rtc::Optional<int>(640 * 360 - 1),
+                               rtc::Optional<int>());
+  // Expect cropping to 16:9 format and 3/4 scaling.
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(270, out_height_);
+
+  // Adapt down one step more.
+  adapter_.OnResolutionRequest(rtc::Optional<int>(480 * 270 - 1),
+                               rtc::Optional<int>());
+  // Expect cropping to 16:9 format and 1/2 scaling.
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(320, out_width_);
+  EXPECT_EQ(180, out_height_);
+
+  // Adapt up one step.
+  adapter_.OnResolutionRequest(rtc::Optional<int>(),
+                               rtc::Optional<int>(320 * 180));
+  // Expect cropping to 16:9 format and 3/4 scaling.
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(480, out_width_);
+  EXPECT_EQ(270, out_height_);
+
+  // Adapt up one step more.
+  adapter_.OnResolutionRequest(rtc::Optional<int>(),
+                               rtc::Optional<int>(480 * 270));
+  // Expect cropping to 16:9 format and no scaling.
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
+
+  // Try to adapt up one step more.
+  adapter_.OnResolutionRequest(rtc::Optional<int>(),
+                               rtc::Optional<int>(640 * 360));
+  // Expect cropping to 16:9 format and no scaling.
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(360, cropped_height_);
+  EXPECT_EQ(640, out_width_);
+  EXPECT_EQ(360, out_height_);
+}
+
+TEST_F(VideoAdapterTest, TestCroppingOddResolution) {
+  // Ask for 640x360 (16:9 aspect), with 3/16 scaling.
+  adapter_.SetExpectedInputFrameInterval(VideoFormat::FpsToInterval(30));
+  adapter_.OnOutputFormatRequest(
+      VideoFormat(640, 360, VideoFormat::FpsToInterval(30), FOURCC_I420));
+  adapter_.OnResolutionRequest(rtc::Optional<int>(640 * 360 * 3 / 16 * 3 / 16),
+                               rtc::Optional<int>());
+
+  // Send 640x480 (4:3 aspect).
+  adapter_.AdaptFrameResolution(640, 480,
+                                &cropped_width_, &cropped_height_,
+                                &out_width_, &out_height_);
+
+  // Instead of getting the exact aspect ratio with cropped resolution 640x360,
+  // the resolution should be adjusted to get a perfect scale factor instead.
+  EXPECT_EQ(640, cropped_width_);
+  EXPECT_EQ(368, cropped_height_);
+  EXPECT_EQ(120, out_width_);
+  EXPECT_EQ(69, out_height_);
 }
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/media/base/videobroadcaster.cc b/chromium/third_party/webrtc/media/base/videobroadcaster.cc
index 704f4081187..18c38b731d1 100644
--- a/chromium/third_party/webrtc/media/base/videobroadcaster.cc
+++ b/chromium/third_party/webrtc/media/base/videobroadcaster.cc
@@ -98,14 +98,14 @@ const cricket::VideoFrame& VideoBroadcaster::GetBlackFrame(
     const cricket::VideoFrame& frame) {
   if (black_frame_ && black_frame_->width() == frame.width() &&
       black_frame_->height() == frame.height() &&
-      black_frame_->GetVideoRotation() == frame.GetVideoRotation()) {
-    black_frame_->SetTimeStamp(frame.GetTimeStamp());
+      black_frame_->rotation() == frame.rotation()) {
+    black_frame_->set_timestamp_us(frame.timestamp_us());
     return *black_frame_;
   }
   black_frame_.reset(new cricket::WebRtcVideoFrame(
-      new rtc::RefCountedObject<webrtc::I420Buffer>(
-          frame.width(), frame.height()),
-      frame.GetTimeStamp(), frame.GetVideoRotation()));
+      new rtc::RefCountedObject<webrtc::I420Buffer>(frame.width(),
+                                                    frame.height()),
+      frame.rotation(), frame.timestamp_us()));
   black_frame_->SetToBlack();
   return *black_frame_;
 }
diff --git a/chromium/third_party/webrtc/media/base/videobroadcaster.h b/chromium/third_party/webrtc/media/base/videobroadcaster.h
index c89c7eea975..764c749b0c3 100644
--- a/chromium/third_party/webrtc/media/base/videobroadcaster.h
+++ b/chromium/third_party/webrtc/media/base/videobroadcaster.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MEDIA_BASE_VIDEOBROADCASTER_H_
 #define WEBRTC_MEDIA_BASE_VIDEOBROADCASTER_H_
 
+#include <memory>
 #include <utility>
 #include <vector>
 
@@ -55,7 +56,7 @@ class VideoBroadcaster : public VideoSourceBase,
   rtc::CriticalSection sinks_and_wants_lock_;
 
   VideoSinkWants current_wants_ GUARDED_BY(sinks_and_wants_lock_);
-  rtc::scoped_ptr<cricket::WebRtcVideoFrame> black_frame_;
+  std::unique_ptr<cricket::WebRtcVideoFrame> black_frame_;
 };
 
 }  // namespace rtc
diff --git a/chromium/third_party/webrtc/media/base/videobroadcaster_unittest.cc b/chromium/third_party/webrtc/media/base/videobroadcaster_unittest.cc
index c6a4df0e664..e4e6d325646 100644
--- a/chromium/third_party/webrtc/media/base/videobroadcaster_unittest.cc
+++ b/chromium/third_party/webrtc/media/base/videobroadcaster_unittest.cc
@@ -135,14 +135,14 @@ TEST(VideoBroadcasterTest, SinkWantsBlackFrames) {
   broadcaster.AddOrUpdateSink(&sink2, wants2);
 
   cricket::WebRtcVideoFrame frame1;
-  frame1.InitToBlack(100, 200, 10 /*ts*/);
+  frame1.InitToBlack(100, 200, 10000 /*ts*/);
   // Make it not all-black
-  frame1.GetUPlane()[0] = 0;
+  frame1.video_frame_buffer()->MutableDataU()[0] = 0;
   broadcaster.OnFrame(frame1);
   EXPECT_TRUE(sink1.black_frame());
-  EXPECT_EQ(10, sink1.timestamp());
+  EXPECT_EQ(10000, sink1.timestamp());
   EXPECT_FALSE(sink2.black_frame());
-  EXPECT_EQ(10, sink2.timestamp());
+  EXPECT_EQ(10000, sink2.timestamp());
 
   // Switch the sink wants.
   wants1.black_frames = false;
@@ -151,12 +151,12 @@ TEST(VideoBroadcasterTest, SinkWantsBlackFrames) {
   broadcaster.AddOrUpdateSink(&sink2, wants2);
 
   cricket::WebRtcVideoFrame frame2;
-  frame2.InitToBlack(100, 200, 30 /*ts*/);
+  frame2.InitToBlack(100, 200, 30000 /*ts*/);
   // Make it not all-black
-  frame2.GetUPlane()[0] = 0;
+  frame2.video_frame_buffer()->MutableDataU()[0] = 0;
   broadcaster.OnFrame(frame2);
   EXPECT_FALSE(sink1.black_frame());
-  EXPECT_EQ(30, sink1.timestamp());
+  EXPECT_EQ(30000, sink1.timestamp());
   EXPECT_TRUE(sink2.black_frame());
-  EXPECT_EQ(30, sink2.timestamp());
+  EXPECT_EQ(30000, sink2.timestamp());
 }
diff --git a/chromium/third_party/webrtc/media/base/videocapturer.cc b/chromium/third_party/webrtc/media/base/videocapturer.cc
index 94f65143531..084a9b913c7 100644
--- a/chromium/third_party/webrtc/media/base/videocapturer.cc
+++ b/chromium/third_party/webrtc/media/base/videocapturer.cc
@@ -30,7 +30,6 @@ static const int64_t kMaxDistance = ~(static_cast<int64_t>(1) << 63);
 #ifdef WEBRTC_LINUX
 static const int kYU12Penalty = 16;  // Needs to be higher than MJPG index.
 #endif
-static const int kDefaultScreencastFps = 5;
 
 }  // namespace
 
@@ -66,10 +65,7 @@ VideoCapturer::VideoCapturer() : apply_rotation_(false) {
 }
 
 void VideoCapturer::Construct() {
-  ratio_w_ = 0;
-  ratio_h_ = 0;
   enable_camera_list_ = false;
-  square_pixel_aspect_ratio_ = false;
   capture_state_ = CS_STOPPED;
   SignalFrameCaptured.connect(this, &VideoCapturer::OnFrameCaptured);
   scaled_width_ = 0;
@@ -224,158 +220,19 @@ void VideoCapturer::OnFrameCaptured(VideoCapturer*,
     return;
   }
 
-  // Use a temporary buffer to scale
-  std::unique_ptr<uint8_t[]> scale_buffer;
-  if (IsScreencast()) {
-    int scaled_width, scaled_height;
-    int desired_screencast_fps =
-        capture_format_.get()
-            ? VideoFormat::IntervalToFps(capture_format_->interval)
-            : kDefaultScreencastFps;
-    ComputeScale(captured_frame->width, captured_frame->height,
-                 desired_screencast_fps, &scaled_width, &scaled_height);
-
-    if (FOURCC_ARGB == captured_frame->fourcc &&
-        (scaled_width != captured_frame->width ||
-         scaled_height != captured_frame->height)) {
-      if (scaled_width != scaled_width_ || scaled_height != scaled_height_) {
-        LOG(LS_INFO) << "Scaling Screencast from " << captured_frame->width
-                     << "x" << captured_frame->height << " to " << scaled_width
-                     << "x" << scaled_height;
-        scaled_width_ = scaled_width;
-        scaled_height_ = scaled_height;
-      }
-      CapturedFrame* modified_frame =
-          const_cast<CapturedFrame*>(captured_frame);
-      const int modified_frame_size = scaled_width * scaled_height * 4;
-      scale_buffer.reset(new uint8_t[modified_frame_size]);
-      // Compute new width such that width * height is less than maximum but
-      // maintains original captured frame aspect ratio.
-      // Round down width to multiple of 4 so odd width won't round up beyond
-      // maximum, and so chroma channel is even width to simplify spatial
-      // resampling.
-      libyuv::ARGBScale(reinterpret_cast<const uint8_t*>(captured_frame->data),
-                        captured_frame->width * 4, captured_frame->width,
-                        captured_frame->height, scale_buffer.get(),
-                        scaled_width * 4, scaled_width, scaled_height,
-                        libyuv::kFilterBilinear);
-      modified_frame->width = scaled_width;
-      modified_frame->height = scaled_height;
-      modified_frame->data_size = scaled_width * 4 * scaled_height;
-      modified_frame->data = scale_buffer.get();
-    }
-  }
-
-  const int kYuy2Bpp = 2;
-  const int kArgbBpp = 4;
-  // TODO(fbarchard): Make a helper function to adjust pixels to square.
-  // TODO(fbarchard): Hook up experiment to scaling.
-  // Temporary buffer is scoped here so it will persist until i420_frame.Init()
-  // makes a copy of the frame, converting to I420.
-  std::unique_ptr<uint8_t[]> temp_buffer;
-  // YUY2 can be scaled vertically using an ARGB scaler.  Aspect ratio is only
-  // a problem on OSX.  OSX always converts webcams to YUY2 or UYVY.
-  bool can_scale =
-      FOURCC_YUY2 == CanonicalFourCC(captured_frame->fourcc) ||
-      FOURCC_UYVY == CanonicalFourCC(captured_frame->fourcc);
-
-  // If pixels are not square, optionally use vertical scaling to make them
-  // square.  Square pixels simplify the rest of the pipeline, including
-  // effects and rendering.
-  if (can_scale && square_pixel_aspect_ratio_ &&
-      captured_frame->pixel_width != captured_frame->pixel_height) {
-    int scaled_width, scaled_height;
-    // modified_frame points to the captured_frame but with const casted away
-    // so it can be modified.
-    CapturedFrame* modified_frame = const_cast<CapturedFrame*>(captured_frame);
-    // Compute the frame size that makes pixels square pixel aspect ratio.
-    ComputeScaleToSquarePixels(captured_frame->width, captured_frame->height,
-                               captured_frame->pixel_width,
-                               captured_frame->pixel_height,
-                               &scaled_width, &scaled_height);
-
-    if (scaled_width != scaled_width_ || scaled_height != scaled_height_) {
-      LOG(LS_INFO) << "Scaling WebCam from "
-                   << captured_frame->width << "x"
-                   << captured_frame->height << " to "
-                   << scaled_width << "x" << scaled_height
-                   << " for PAR "
-                   << captured_frame->pixel_width << "x"
-                   << captured_frame->pixel_height;
-      scaled_width_ = scaled_width;
-      scaled_height_ = scaled_height;
-    }
-    const int modified_frame_size = scaled_width * scaled_height * kYuy2Bpp;
-    uint8_t* temp_buffer_data;
-    // Pixels are wide and short; Increasing height. Requires temporary buffer.
-    if (scaled_height > captured_frame->height) {
-      temp_buffer.reset(new uint8_t[modified_frame_size]);
-      temp_buffer_data = temp_buffer.get();
-    } else {
-      // Pixels are narrow and tall; Decreasing height. Scale will be done
-      // in place.
-      temp_buffer_data = reinterpret_cast<uint8_t*>(captured_frame->data);
-    }
-
-    // Use ARGBScaler to vertically scale the YUY2 image, adjusting for 16 bpp.
-    libyuv::ARGBScale(reinterpret_cast<const uint8_t*>(captured_frame->data),
-                      captured_frame->width * kYuy2Bpp,  // Stride for YUY2.
-                      captured_frame->width * kYuy2Bpp / kArgbBpp,  // Width.
-                      abs(captured_frame->height),                  // Height.
-                      temp_buffer_data,
-                      scaled_width * kYuy2Bpp,             // Stride for YUY2.
-                      scaled_width * kYuy2Bpp / kArgbBpp,  // Width.
-                      abs(scaled_height),                  // New height.
-                      libyuv::kFilterBilinear);
-    modified_frame->width = scaled_width;
-    modified_frame->height = scaled_height;
-    modified_frame->pixel_width = 1;
-    modified_frame->pixel_height = 1;
-    modified_frame->data_size = modified_frame_size;
-    modified_frame->data = temp_buffer_data;
-  }
-
-  // Size to crop captured frame to.  This adjusts the captured frames
-  // aspect ratio to match the final view aspect ratio, considering pixel
-  // aspect ratio and rotation.  The final size may be scaled down by video
-  // adapter to better match ratio_w_ x ratio_h_.
-  // Note that abs() of frame height is passed in, because source may be
-  // inverted, but output will be positive.
   int cropped_width = captured_frame->width;
   int cropped_height = captured_frame->height;
-
-  // TODO(fbarchard): Improve logic to pad or crop.
-  // MJPG can crop vertically, but not horizontally.  This logic disables crop.
-  // Alternatively we could pad the image with black, or implement a 2 step
-  // crop.
-  bool can_crop = true;
-  if (captured_frame->fourcc == FOURCC_MJPG) {
-    float cam_aspect = static_cast<float>(captured_frame->width) /
-        static_cast<float>(captured_frame->height);
-    float view_aspect = static_cast<float>(ratio_w_) /
-        static_cast<float>(ratio_h_);
-    can_crop = cam_aspect <= view_aspect;
-  }
-  if (can_crop && !IsScreencast()) {
-    // TODO(ronghuawu): The capturer should always produce the native
-    // resolution and the cropping should be done in downstream code.
-    ComputeCrop(ratio_w_, ratio_h_, captured_frame->width,
-                abs(captured_frame->height), captured_frame->pixel_width,
-                captured_frame->pixel_height, captured_frame->rotation,
-                &cropped_width, &cropped_height);
-  }
-
-  int adapted_width = cropped_width;
-  int adapted_height = cropped_height;
+  int out_width = captured_frame->width;
+  int out_height = captured_frame->height;
   if (enable_video_adapter_ && !IsScreencast()) {
-    const VideoFormat adapted_format =
-        video_adapter_.AdaptFrameResolution(cropped_width, cropped_height);
-    if (adapted_format.IsSize0x0()) {
+    video_adapter_.AdaptFrameResolution(
+        captured_frame->width, captured_frame->height,
+        &cropped_width, &cropped_height,
+        &out_width, &out_height);
+    if (out_width == 0 || out_height == 0) {
       // VideoAdapter dropped the frame.
       return;
     }
-    adapted_width = adapted_format.width;
-    adapted_height = adapted_format.height;
   }
 
   if (!frame_factory_) {
@@ -383,16 +240,15 @@ void VideoCapturer::OnFrameCaptured(VideoCapturer*,
     return;
   }
 
-  std::unique_ptr<VideoFrame> adapted_frame(
-      frame_factory_->CreateAliasedFrame(captured_frame,
-                                         cropped_width, cropped_height,
-                                         adapted_width, adapted_height));
+  // TODO(nisse): Reorganize frame factory methods.
+  std::unique_ptr<VideoFrame> adapted_frame(frame_factory_->CreateAliasedFrame(
+      captured_frame, cropped_width, cropped_height, out_width, out_height));
 
   if (!adapted_frame) {
     // TODO(fbarchard): LOG more information about captured frame attributes.
     LOG(LS_ERROR) << "Couldn't convert to I420! "
                   << "From " << ToString(captured_frame) << " To "
-                  << cropped_width << " x " << cropped_height;
+                  << out_width << " x " << out_height;
     return;
   }
 
diff --git a/chromium/third_party/webrtc/media/base/videocapturer.h b/chromium/third_party/webrtc/media/base/videocapturer.h
index 270315582a1..edc6cd31a1b 100644
--- a/chromium/third_party/webrtc/media/base/videocapturer.h
+++ b/chromium/third_party/webrtc/media/base/videocapturer.h
@@ -19,6 +19,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/media/base/videosourceinterface.h"
 #include "webrtc/base/sigslot.h"
@@ -283,10 +284,7 @@ class VideoCapturer : public sigslot::has_slots<>,
   std::unique_ptr<VideoFormat> max_format_;
   std::vector<VideoFormat> filtered_supported_formats_;
 
-  int ratio_w_;  // View resolution. e.g. 1280 x 720.
-  int ratio_h_;
   bool enable_camera_list_;
-  bool square_pixel_aspect_ratio_;  // Enable scaling to square pixels.
   int scaled_width_;  // Current output size from ComputeScale.
   int scaled_height_;
 
diff --git a/chromium/third_party/webrtc/media/base/videocapturer_unittest.cc b/chromium/third_party/webrtc/media/base/videocapturer_unittest.cc
index bd145e33ea1..25230b5118a 100644
--- a/chromium/third_party/webrtc/media/base/videocapturer_unittest.cc
+++ b/chromium/third_party/webrtc/media/base/videocapturer_unittest.cc
@@ -9,6 +9,8 @@
  */
 
 #include <stdio.h>
+
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/gunit.h"
@@ -40,7 +42,7 @@ class VideoCapturerTest
 
  protected:
   void InitCapturer(bool is_screencast) {
-    capturer_ = rtc::scoped_ptr<FakeVideoCapturer>(
+    capturer_ = std::unique_ptr<FakeVideoCapturer>(
         new FakeVideoCapturer(is_screencast));
     capturer_->SignalStateChange.connect(this,
                                          &VideoCapturerTest::OnStateChange);
@@ -56,7 +58,7 @@ class VideoCapturerTest
   cricket::CaptureState capture_state() { return capture_state_; }
   int num_state_changes() { return num_state_changes_; }
 
-  rtc::scoped_ptr<cricket::FakeVideoCapturer> capturer_;
+  std::unique_ptr<cricket::FakeVideoCapturer> capturer_;
   cricket::CaptureState capture_state_;
   int num_state_changes_;
   cricket::FakeVideoRenderer renderer_;
@@ -321,32 +323,6 @@ TEST_F(VideoCapturerTest, SinkWantsMaxPixelAndMaxPixelCountStepUp) {
   EXPECT_EQ(720, renderer2.height());
 }
 
-TEST_F(VideoCapturerTest, ScreencastScaledSuperLarge) {
-  InitScreencast();
-
-  const int kMaxWidth = 4096;
-  const int kMaxHeight = 3072;
-  int kWidth = kMaxWidth + 4;
-  int kHeight = kMaxHeight + 4;
-
-  std::vector<cricket::VideoFormat> formats;
-  formats.push_back(cricket::VideoFormat(kWidth, kHeight,
-      cricket::VideoFormat::FpsToInterval(5), cricket::FOURCC_ARGB));
-  capturer_->ResetSupportedFormats(formats);
-
-  EXPECT_EQ(cricket::CS_RUNNING, capturer_->Start(cricket::VideoFormat(
-      kWidth,
-      kHeight,
-      cricket::VideoFormat::FpsToInterval(30),
-      cricket::FOURCC_ARGB)));
-  EXPECT_TRUE(capturer_->IsRunning());
-  EXPECT_EQ(0, renderer_.num_rendered_frames());
-  EXPECT_TRUE(capturer_->CaptureFrame());
-  EXPECT_EQ(1, renderer_.num_rendered_frames());
-  EXPECT_EQ(kWidth / 2, renderer_.width());
-  EXPECT_EQ(kHeight / 2, renderer_.height());
-}
-
 TEST_F(VideoCapturerTest, TestFourccMatch) {
   cricket::VideoFormat desired(640, 480,
                                cricket::VideoFormat::FpsToInterval(30),
diff --git a/chromium/third_party/webrtc/media/base/videocommon.cc b/chromium/third_party/webrtc/media/base/videocommon.cc
index f6f06c24e32..12a17973d64 100644
--- a/chromium/third_party/webrtc/media/base/videocommon.cc
+++ b/chromium/third_party/webrtc/media/base/videocommon.cc
@@ -51,153 +51,6 @@ uint32_t CanonicalFourCC(uint32_t fourcc) {
   return fourcc;
 }
 
-static float kScaleFactors[] = {
-  1.f / 1.f,  // Full size.
-  1.f / 2.f,  // 1/2 scale.
-  1.f / 4.f,  // 1/4 scale.
-  1.f / 8.f,  // 1/8 scale.
-  1.f / 16.f  // 1/16 scale.
-};
-
-static const int kNumScaleFactors = arraysize(kScaleFactors);
-
-// Finds the scale factor that, when applied to width and height, produces
-// fewer than num_pixels.
-static float FindLowerScale(int width, int height, int target_num_pixels) {
-  if (!target_num_pixels) {
-    return 0.f;
-  }
-  int best_distance = INT_MAX;
-  int best_index = kNumScaleFactors - 1;  // Default to max scale.
-  for (int i = 0; i < kNumScaleFactors; ++i) {
-    int test_num_pixels = static_cast<int>(width * kScaleFactors[i] *
-                                           height * kScaleFactors[i]);
-    int diff = target_num_pixels - test_num_pixels;
-    if (diff >= 0 && diff < best_distance) {
-      best_distance = diff;
-      best_index = i;
-      if (best_distance == 0) {  // Found exact match.
-        break;
-      }
-    }
-  }
-  return kScaleFactors[best_index];
-}
-
-// Computes a scale less to fit in max_pixels while maintaining aspect ratio.
-void ComputeScaleMaxPixels(int frame_width, int frame_height, int max_pixels,
-    int* scaled_width, int* scaled_height) {
-  ASSERT(scaled_width != NULL);
-  ASSERT(scaled_height != NULL);
-  ASSERT(max_pixels > 0);
-  const int kMaxWidth = 4096;
-  const int kMaxHeight = 3072;
-  int new_frame_width = frame_width;
-  int new_frame_height = frame_height;
-
-  // Limit width.
-  if (new_frame_width > kMaxWidth) {
-    new_frame_height = new_frame_height * kMaxWidth / new_frame_width;
-    new_frame_width = kMaxWidth;
-  }
-  // Limit height.
-  if (new_frame_height > kMaxHeight) {
-    new_frame_width = new_frame_width * kMaxHeight / new_frame_height;
-    new_frame_height = kMaxHeight;
-  }
-  // Limit number of pixels.
-  if (new_frame_width * new_frame_height > max_pixels) {
-    // Compute new width such that width * height is less than maximum but
-    // maintains original captured frame aspect ratio.
-    new_frame_width = static_cast<int>(sqrtf(static_cast<float>(
-        max_pixels) * new_frame_width / new_frame_height));
-    new_frame_height = max_pixels / new_frame_width;
-  }
-  // Snap to a scale factor that is less than or equal to target pixels.
-  float scale = FindLowerScale(frame_width, frame_height,
-                               new_frame_width * new_frame_height);
-  *scaled_width = static_cast<int>(frame_width * scale + .5f);
-  *scaled_height = static_cast<int>(frame_height * scale + .5f);
-}
-
-// Compute a size to scale frames to that is below maximum compression
-// and rendering size with the same aspect ratio.
-void ComputeScale(int frame_width, int frame_height, int fps,
-                  int* scaled_width, int* scaled_height) {
-  // Maximum pixels limit is set to Retina MacBookPro 15" resolution of
-  // 2880 x 1800 as of 4/18/2013.
-  // For high fps, maximum pixels limit is set based on common 24" monitor
-  // resolution of 2048 x 1280 as of 6/13/2013. The Retina resolution is
-  // therefore reduced to 1440 x 900.
-  int max_pixels = (fps > 5) ? 2048 * 1280 : 2880 * 1800;
-  ComputeScaleMaxPixels(
-      frame_width, frame_height, max_pixels, scaled_width, scaled_height);
-}
-
-// Compute size to crop video frame to.
-// If cropped_format_* is 0, return the frame_* size as is.
-void ComputeCrop(int cropped_format_width, int cropped_format_height,
-                 int frame_width, int frame_height,
-                 int pixel_width, int pixel_height,
-                 int rotation,
-                 int* cropped_width, int* cropped_height) {
-  // Transform screen crop to camera space if rotated.
-  if (rotation == 90 || rotation == 270) {
-    std::swap(cropped_format_width, cropped_format_height);
-  }
-  ASSERT(cropped_format_width >= 0);
-  ASSERT(cropped_format_height >= 0);
-  ASSERT(frame_width > 0);
-  ASSERT(frame_height > 0);
-  ASSERT(pixel_width >= 0);
-  ASSERT(pixel_height >= 0);
-  ASSERT(rotation == 0 || rotation == 90 || rotation == 180 || rotation == 270);
-  ASSERT(cropped_width != NULL);
-  ASSERT(cropped_height != NULL);
-  if (!pixel_width) {
-    pixel_width = 1;
-  }
-  if (!pixel_height) {
-    pixel_height = 1;
-  }
-  // if cropped_format is 0x0 disable cropping.
-  if (!cropped_format_height) {
-    cropped_format_height = 1;
-  }
-  float frame_aspect = static_cast<float>(frame_width * pixel_width) /
-      static_cast<float>(frame_height * pixel_height);
-  float crop_aspect = static_cast<float>(cropped_format_width) /
-      static_cast<float>(cropped_format_height);
-  // kAspectThresh is the maximum aspect ratio difference that we'll accept
-  // for cropping.  The value 1.34 allows cropping from 4:3 to 16:9.
-  // Set to zero to disable cropping entirely.
-  // TODO(fbarchard): crop to multiple of 16 width for better performance.
-  const float kAspectThresh = 1.34f;
-  // Wide aspect - crop horizontally
-  if (frame_aspect > crop_aspect &&
-      frame_aspect < crop_aspect * kAspectThresh) {
-    // Round width down to multiple of 4 to avoid odd chroma width.
-    // Width a multiple of 4 allows a half size image to have chroma channel
-    // that avoids rounding errors.
-    frame_width = static_cast<int>((crop_aspect * frame_height *
-        pixel_height) / pixel_width + 0.5f) & ~3;
-  } else if (frame_aspect < crop_aspect &&
-             frame_aspect > crop_aspect / kAspectThresh) {
-    frame_height = static_cast<int>((frame_width * pixel_width) /
-        (crop_aspect * pixel_height) + 0.5f) & ~1;
-  }
-  *cropped_width = frame_width;
-  *cropped_height = frame_height;
-}
-
-// Compute the frame size that makes pixels square pixel aspect ratio.
-void ComputeScaleToSquarePixels(int in_width, int in_height,
-                                int pixel_width, int pixel_height,
-                                int* scaled_width, int* scaled_height) {
-  *scaled_width = in_width;  // Keep width the same.
-  *scaled_height = in_height * pixel_height / pixel_width;
-}
-
 // The C++ standard requires a namespace-scope definition of static const
 // integral types even when they are initialized in the declaration (see
 // [class.static.data]/4), but MSVC with /Ze is non-conforming and treats that
diff --git a/chromium/third_party/webrtc/media/base/videocommon.h b/chromium/third_party/webrtc/media/base/videocommon.h
index d8d7f9cdd38..757100b07d4 100644
--- a/chromium/third_party/webrtc/media/base/videocommon.h
+++ b/chromium/third_party/webrtc/media/base/videocommon.h
@@ -135,31 +135,6 @@ inline std::string GetFourccName(uint32_t fourcc) {
   return name;
 }
 
-// Computes a scale less to fit in max_pixels while maintaining aspect ratio.
-void ComputeScaleMaxPixels(int frame_width, int frame_height, int max_pixels,
-                           int* scaled_width, int* scaled_height);
-
-// For low fps, max pixels limit is set to Retina MacBookPro 15" resolution of
-// 2880 x 1800 as of 4/18/2013.
-// For high fps, maximum pixels limit is set based on common 24" monitor
-// resolution of 2048 x 1280 as of 6/13/2013. The Retina resolution is
-// therefore reduced to 1440 x 900.
-void ComputeScale(int frame_width, int frame_height, int fps,
-                  int* scaled_width, int* scaled_height);
-
-// Compute the frame size that conversion should crop to based on aspect ratio.
-// Ensures size is multiple of 2 due to I420 and conversion limitations.
-void ComputeCrop(int cropped_format_width, int cropped_format_height,
-                 int frame_width, int frame_height,
-                 int pixel_width, int pixel_height,
-                 int rotation,
-                 int* cropped_width, int* cropped_height);
-
-// Compute the frame size that makes pixels square pixel aspect ratio.
-void ComputeScaleToSquarePixels(int in_width, int in_height,
-                                int pixel_width, int pixel_height,
-                                int* scaled_width, int* scaled_height);
-
 //////////////////////////////////////////////////////////////////////////////
 // Definition of VideoFormat.
 //////////////////////////////////////////////////////////////////////////////
diff --git a/chromium/third_party/webrtc/media/base/videocommon_unittest.cc b/chromium/third_party/webrtc/media/base/videocommon_unittest.cc
index 84a728b19b1..c28a5d25bae 100644
--- a/chromium/third_party/webrtc/media/base/videocommon_unittest.cc
+++ b/chromium/third_party/webrtc/media/base/videocommon_unittest.cc
@@ -91,221 +91,4 @@ TEST(VideoCommonTest, TestVideoFormatCompare) {
   EXPECT_TRUE(format.IsPixelRateLess(format2));
 }
 
-TEST(VideoCommonTest, TestComputeScaleWithLowFps) {
-  int scaled_width, scaled_height;
-
-  // Request small enough. Expect no change.
-  ComputeScale(2560, 1600, 5, &scaled_width, &scaled_height);
-  EXPECT_EQ(2560, scaled_width);
-  EXPECT_EQ(1600, scaled_height);
-
-  // Request too many pixels. Expect 1/2 size.
-  ComputeScale(4096, 2560, 5, &scaled_width, &scaled_height);
-  EXPECT_EQ(2048, scaled_width);
-  EXPECT_EQ(1280, scaled_height);
-
-  // Request too many pixels and too wide and tall. Expect 1/4 size.
-  ComputeScale(16000, 10000, 5, &scaled_width, &scaled_height);
-  EXPECT_EQ(2000, scaled_width);
-  EXPECT_EQ(1250, scaled_height);
-
-  // Request too wide. (two 30 inch monitors). Expect 1/2 size.
-  ComputeScale(5120, 1600, 5, &scaled_width, &scaled_height);
-  EXPECT_EQ(2560, scaled_width);
-  EXPECT_EQ(800, scaled_height);
-
-  // Request too wide but not too many pixels. Expect 1/2 size.
-  ComputeScale(8192, 1024, 5, &scaled_width, &scaled_height);
-  EXPECT_EQ(4096, scaled_width);
-  EXPECT_EQ(512, scaled_height);
-
-  // Request too tall. Expect 1/4 size.
-  ComputeScale(1024, 8192, 5, &scaled_width, &scaled_height);
-  EXPECT_EQ(256, scaled_width);
-  EXPECT_EQ(2048, scaled_height);
-}
-
-// Same as TestComputeScale but with 15 fps instead of 5 fps.
-// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5487
-#ifdef UNDEFINED_SANITIZER
-#define MAYBE_TestComputeScaleWithHighFps DISABLED_TestComputeScaleWithHighFps
-#else
-#define MAYBE_TestComputeScaleWithHighFps TestComputeScaleWithHighFps
-#endif
-TEST(VideoCommonTest, MAYBE_TestComputeScaleWithHighFps) {
-  int scaled_width, scaled_height;
-
-  // Request small enough but high fps. Expect 1/2 size.
-  ComputeScale(2560, 1600, 15, &scaled_width, &scaled_height);
-  EXPECT_EQ(1280, scaled_width);
-  EXPECT_EQ(800, scaled_height);
-
-  // Request too many pixels. Expect 1/2 size.
-  ComputeScale(4096, 2560, 15, &scaled_width, &scaled_height);
-  EXPECT_EQ(2048, scaled_width);
-  EXPECT_EQ(1280, scaled_height);
-
-  // Request too many pixels and too wide and tall. Expect 1/16 size.
-  ComputeScale(64000, 40000, 15, &scaled_width, &scaled_height);
-  EXPECT_EQ(4000, scaled_width);
-  EXPECT_EQ(2500, scaled_height);
-
-  // Request too wide. (two 30 inch monitors). Expect 1/2 size.
-  ComputeScale(5120, 1600, 15, &scaled_width, &scaled_height);
-  EXPECT_EQ(2560, scaled_width);
-  EXPECT_EQ(800, scaled_height);
-
-  // Request too wide but not too many pixels. Expect 1/2 size.
-  ComputeScale(8192, 1024, 15, &scaled_width, &scaled_height);
-  EXPECT_EQ(4096, scaled_width);
-  EXPECT_EQ(512, scaled_height);
-
-  // Request too tall. Expect 1/4 size.
-  ComputeScale(1024, 8192, 15, &scaled_width, &scaled_height);
-  EXPECT_EQ(256, scaled_width);
-  EXPECT_EQ(2048, scaled_height);
-}
-
-TEST(VideoCommonTest, TestComputeCrop) {
-  int cropped_width, cropped_height;
-
-  // Request 16:9 to 16:9.  Expect no cropping.
-  ComputeCrop(1280, 720,  // Crop size 16:9
-              640, 360,  // Frame is 4:3
-              1, 1,  // Normal 1:1 pixels
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(640, cropped_width);
-  EXPECT_EQ(360, cropped_height);
-
-  // Request 4:3 to 16:9.  Expect vertical.
-  ComputeCrop(640, 360,  // Crop size 16:9
-              640, 480,  // Frame is 4:3
-              1, 1,  // Normal 1:1 pixels
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(640, cropped_width);
-  EXPECT_EQ(360, cropped_height);
-
-  // Request 16:9 to 4:3.  Expect horizontal crop.
-  ComputeCrop(640, 480,  // Crop size 4:3
-              640, 360,  // Frame is 16:9
-              1, 1,  // Normal 1:1 pixels
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(480, cropped_width);
-  EXPECT_EQ(360, cropped_height);
-
-  // Request 16:9 but VGA has 3:8 pixel aspect ratio. Expect no crop.
-  // This occurs on HP4110 on OSX 10.5/10.6/10.7
-  ComputeCrop(640, 360,  // Crop size 16:9
-              640, 480,  // Frame is 4:3
-              3, 8,  // Pixel aspect ratio is tall
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(640, cropped_width);
-  EXPECT_EQ(480, cropped_height);
-
-  // Request 16:9 but QVGA has 15:11 pixel aspect ratio. Expect horizontal crop.
-  // This occurs on Logitech B910 on OSX 10.5/10.6/10.7 in Hangouts.
-  ComputeCrop(640, 360,  // Crop size 16:9
-              320, 240,  // Frame is 4:3
-              15, 11,  // Pixel aspect ratio is wide
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(312, cropped_width);
-  EXPECT_EQ(240, cropped_height);
-
-  // Request 16:10 but QVGA has 15:11 pixel aspect ratio.
-  // Expect horizontal crop.
-  // This occurs on Logitech B910 on OSX 10.5/10.6/10.7 in gmail.
-  ComputeCrop(640, 400,  // Crop size 16:10
-              320, 240,  // Frame is 4:3
-              15, 11,  // Pixel aspect ratio is wide
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(280, cropped_width);
-  EXPECT_EQ(240, cropped_height);
-
-  // Request 16:9 but VGA has 6:5 pixel aspect ratio. Expect vertical crop.
-  // This occurs on Logitech QuickCam Pro C9000 on OSX
-  ComputeCrop(640, 360,  // Crop size 16:9
-              640, 480,  // Frame is 4:3
-              6, 5,  // Pixel aspect ratio is wide
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(640, cropped_width);
-  EXPECT_EQ(432, cropped_height);
-
-  // Request 16:10 but HD is 16:9. Expect horizontal crop.
-  // This occurs in settings and local preview with HD experiment.
-  ComputeCrop(1280, 800,  // Crop size 16:10
-              1280, 720,  // Frame is 4:3
-              1, 1,  // Pixel aspect ratio is wide
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(1152, cropped_width);
-  EXPECT_EQ(720, cropped_height);
-
-  // Request 16:9 but HD has 3:4 pixel aspect ratio. Expect vertical crop.
-  // This occurs on Logitech B910 on OSX 10.5/10.6.7 but not OSX 10.6.8 or 10.7
-  ComputeCrop(1280, 720,  // Crop size 16:9
-              1280, 720,  // Frame is 4:3
-              3, 4,  // Pixel aspect ratio is wide
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(1280, cropped_width);
-  EXPECT_EQ(540, cropped_height);
-
-  // Request 16:9 to 3:4 (portrait).  Expect no cropping.
-  ComputeCrop(640, 360,  // Crop size 16:9
-              640, 480,  // Frame is 3:4 portrait
-              1, 1,  // Normal 1:1 pixels
-              90,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(640, cropped_width);
-  EXPECT_EQ(480, cropped_height);
-
-  // Request 9:16 from VGA rotated (portrait).  Expect crop.
-  ComputeCrop(360, 640,  // Crop size 9:16
-              640, 480,  // Frame is 3:4 portrait
-              1, 1,  // Normal 1:1 pixels
-              90,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(640, cropped_width);
-  EXPECT_EQ(360, cropped_height);
-
-  // Cropped size 0x0.  Expect no cropping.
-  // This is used when adding multiple capturers
-  ComputeCrop(0, 0,  // Crop size 0x0
-              1024, 768,  // Frame is 3:4 portrait
-              1, 1,  // Normal 1:1 pixels
-              0,
-              &cropped_width, &cropped_height);
-  EXPECT_EQ(1024, cropped_width);
-  EXPECT_EQ(768, cropped_height);
-}
-
-TEST(VideoCommonTest, TestComputeScaleToSquarePixels) {
-  int scaled_width, scaled_height;
-
-  // Pixel aspect ratio is 4:3.  Logical aspect ratio is 16:9.  Expect scale
-  // to square pixels with physical aspect ratio of 16:9.
-  ComputeScaleToSquarePixels(640, 480,
-                             4, 3,  // 4 x 3 pixel aspect ratio
-                             &scaled_width, &scaled_height);
-  EXPECT_EQ(640, scaled_width);
-  EXPECT_EQ(360, scaled_height);
-
-  // Pixel aspect ratio is 3:8.  Physical aspect ratio is 4:3.  Expect scale
-  // to square pixels with logical aspect ratio of 1:2.
-  // Note that 640x1280 will be scaled down by video adapter to view request
-  // of 640*360 and will end up using 320x640.
-  ComputeScaleToSquarePixels(640, 480,
-                             3, 8,  // 4 x 3 pixel aspect ratio
-                             &scaled_width, &scaled_height);
-  EXPECT_EQ(640, scaled_width);
-  EXPECT_EQ(1280, scaled_height);
-}
-
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/media/base/videoengine_unittest.h b/chromium/third_party/webrtc/media/base/videoengine_unittest.h
index 01a4f76ca76..394b9c9a8e9 100644
--- a/chromium/third_party/webrtc/media/base/videoengine_unittest.h
+++ b/chromium/third_party/webrtc/media/base/videoengine_unittest.h
@@ -74,27 +74,6 @@ inline int TimeBetweenSend(const cricket::VideoCodec& codec) {
       rtc::kNumNanosecsPerMillisec);
 }
 
-// Fake video engine that makes it possible to test enabling and disabling
-// capturer (checking that the engine state is updated and that the capturer
-// is indeed capturing) without having to create a channel. It also makes it
-// possible to test that the media processors are indeed being called when
-// registered.
-template<class T>
-class VideoEngineOverride : public T {
- public:
-  VideoEngineOverride() : T() {
-  }
-  virtual ~VideoEngineOverride() {
-  }
-  bool is_camera_on() const { return T::GetVideoCapturer()->IsRunning(); }
-
-  void TriggerMediaFrame(uint32_t ssrc,
-                         cricket::VideoFrame* frame,
-                         bool* drop_frame) {
-    T::SignalMediaFrame(ssrc, frame, drop_frame);
-  }
-};
-
 template<class E, class C>
 class VideoMediaChannelTest : public testing::Test,
                               public sigslot::has_slots<> {
@@ -125,7 +104,7 @@ class VideoMediaChannelTest : public testing::Test,
                                 cricket::VideoFormat::FpsToInterval(30),
                                 cricket::FOURCC_I420);
     EXPECT_EQ(cricket::CS_RUNNING, video_capturer_->Start(format));
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, video_capturer_.get()));
+    channel_->SetSource(kSsrc, video_capturer_.get());
   }
 
   virtual cricket::FakeVideoCapturer* CreateFakeVideoCapturer() {
@@ -162,7 +141,7 @@ class VideoMediaChannelTest : public testing::Test,
                                 cricket::FOURCC_I420);
     EXPECT_EQ(cricket::CS_RUNNING, video_capturer_2_->Start(format));
 
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc + 2, video_capturer_2_.get()));
+    channel_->SetSource(kSsrc + 2, video_capturer_2_.get());
   }
   virtual void TearDown() {
     channel_.reset();
@@ -172,7 +151,7 @@ class VideoMediaChannelTest : public testing::Test,
   }
 
   bool SetOneCodec(int pt, const char* name, int w, int h, int fr) {
-    return SetOneCodec(cricket::VideoCodec(pt, name, w, h, fr, 0));
+    return SetOneCodec(cricket::VideoCodec(pt, name, w, h, fr));
   }
   bool SetOneCodec(const cricket::VideoCodec& codec) {
     cricket::VideoFormat capture_format(codec.width, codec.height,
@@ -373,7 +352,7 @@ class VideoMediaChannelTest : public testing::Test,
   // Test that SetSend works.
   void SetSend() {
     EXPECT_FALSE(channel_->sending());
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, video_capturer_.get()));
+    channel_->SetSource(kSsrc, video_capturer_.get());
     EXPECT_TRUE(SetOneCodec(DefaultCodec()));
     EXPECT_FALSE(channel_->sending());
     EXPECT_TRUE(SetSend(true));
@@ -567,7 +546,7 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_EQ(cricket::CS_RUNNING, capturer->Start(format));
     EXPECT_TRUE(channel_->AddSendStream(
         cricket::StreamParams::CreateLegacy(5678)));
-    EXPECT_TRUE(channel_->SetCapturer(5678, capturer.get()));
+    channel_->SetSource(5678, capturer.get());
     EXPECT_TRUE(channel_->AddRecvStream(
         cricket::StreamParams::CreateLegacy(5678)));
     EXPECT_TRUE(channel_->SetSink(5678, &renderer2));
@@ -603,7 +582,7 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_EQ(kTestWidth, info.senders[1].send_frame_width);
     EXPECT_EQ(kTestHeight, info.senders[1].send_frame_height);
     // The capturer must be unregistered here as it runs out of it's scope next.
-    EXPECT_TRUE(channel_->SetCapturer(5678, NULL));
+    channel_->SetSource(5678, NULL);
   }
 
   // Test that we can set the bandwidth.
@@ -640,7 +619,7 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_TRUE(SetDefaultCodec());
     EXPECT_TRUE(channel_->AddSendStream(
         cricket::StreamParams::CreateLegacy(999)));
-    EXPECT_TRUE(channel_->SetCapturer(999u, video_capturer_.get()));
+    channel_->SetSource(999u, video_capturer_.get());
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(WaitAndSendFrame(0));
     EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
@@ -706,7 +685,7 @@ class VideoMediaChannelTest : public testing::Test,
 
     EXPECT_TRUE(channel_->AddSendStream(
         cricket::StreamParams::CreateLegacy(789u)));
-    EXPECT_TRUE(channel_->SetCapturer(789u, video_capturer_.get()));
+    channel_->SetSource(789u, video_capturer_.get());
     EXPECT_EQ(rtp_packets, NumRtpPackets());
     // Wait 30ms to guarantee the engine does not drop the frame.
     EXPECT_TRUE(WaitAndSendFrame(30));
@@ -789,7 +768,7 @@ class VideoMediaChannelTest : public testing::Test,
 
     int captured_frames = 1;
     for (int iterations = 0; iterations < 2; ++iterations) {
-      EXPECT_TRUE(channel_->SetCapturer(kSsrc, capturer.get()));
+      channel_->SetSource(kSsrc, capturer.get());
       rtc::Thread::Current()->ProcessMessages(time_between_send);
       EXPECT_TRUE(capturer->CaptureCustomFrame(format.width, format.height,
                                                cricket::FOURCC_I420));
@@ -804,7 +783,7 @@ class VideoMediaChannelTest : public testing::Test,
       EXPECT_EQ(format.height, renderer_.height());
       captured_frames = renderer_.num_rendered_frames() + 1;
       EXPECT_FALSE(renderer_.black_frame());
-      EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
+      channel_->SetSource(kSsrc, NULL);
       // Make sure a black frame is generated within the specified timeout.
       // The black frame should be the resolution of the previous frame to
       // prevent expensive encoder reconfigurations.
@@ -839,13 +818,12 @@ class VideoMediaChannelTest : public testing::Test,
     // tightly.
     rtc::Thread::Current()->ProcessMessages(30);
     // Remove the capturer.
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
+    channel_->SetSource(kSsrc, NULL);
     // Wait for one black frame for removing the capturer.
     EXPECT_FRAME_WAIT(2, 640, 400, kTimeout);
 
-    // No capturer was added, so this RemoveCapturer should
-    // fail.
-    EXPECT_FALSE(channel_->SetCapturer(kSsrc, NULL));
+    // No capturer was added, so this SetSource should be a NOP.
+    channel_->SetSource(kSsrc, NULL);
     rtc::Thread::Current()->ProcessMessages(300);
     // Verify no more frames were sent.
     EXPECT_EQ(2, renderer_.num_rendered_frames());
@@ -887,11 +865,11 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_EQ(cricket::CS_RUNNING, capturer2->Start(capture_format));
     // State for all the streams.
     EXPECT_TRUE(SetOneCodec(DefaultCodec()));
-    // A limitation in the lmi implementation requires that SetCapturer() is
+    // A limitation in the lmi implementation requires that SetSource() is
     // called after SetOneCodec().
     // TODO(hellner): this seems like an unnecessary constraint, fix it.
-    EXPECT_TRUE(channel_->SetCapturer(1, capturer1.get()));
-    EXPECT_TRUE(channel_->SetCapturer(2, capturer2.get()));
+    channel_->SetSource(1, capturer1.get());
+    channel_->SetSource(2, capturer2.get());
     EXPECT_TRUE(SetSend(true));
     // Test capturer associated with engine.
     const int kTestWidth = 160;
@@ -906,13 +884,13 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_FRAME_ON_RENDERER_WAIT(
         renderer2, 1, kTestWidth, kTestHeight, kTimeout);
     // Successfully remove the capturer.
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
+    channel_->SetSource(kSsrc, NULL);
     // Fail to re-remove the capturer.
-    EXPECT_FALSE(channel_->SetCapturer(kSsrc, NULL));
+    channel_->SetSource(kSsrc, NULL);
     // The capturers must be unregistered here as it runs out of it's scope
     // next.
-    EXPECT_TRUE(channel_->SetCapturer(1, NULL));
-    EXPECT_TRUE(channel_->SetCapturer(2, NULL));
+    channel_->SetSource(1, NULL);
+    channel_->SetSource(2, NULL);
   }
 
   void HighAspectHighHeightCapturer() {
@@ -945,13 +923,13 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_EQ(cricket::CS_RUNNING, capturer->Start(capture_format));
     // Capture frame to not get same frame timestamps as previous capturer.
     capturer->CaptureFrame();
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, capturer.get()));
+    channel_->SetSource(kSsrc, capturer.get());
     EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(30));
     EXPECT_TRUE(capturer->CaptureCustomFrame(kWidth, kHeight,
                                              cricket::FOURCC_ARGB));
     EXPECT_GT_FRAME_ON_RENDERER_WAIT(
         renderer, 2, kScaledWidth, kScaledHeight, kTimeout);
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
+    channel_->SetSource(kSsrc, NULL);
   }
 
   // Tests that we can adapt video resolution with 16:10 aspect ratio properly.
@@ -1062,57 +1040,6 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_FRAME_WAIT(2, codec.width / 2, codec.height / 2, kTimeout);
   }
 
-  // Tests that we can mute and unmute the channel properly.
-  void MuteStream() {
-    EXPECT_TRUE(SetDefaultCodec());
-    cricket::FakeVideoCapturer video_capturer;
-    video_capturer.Start(
-        cricket::VideoFormat(
-            640, 480,
-            cricket::VideoFormat::FpsToInterval(30),
-            cricket::FOURCC_I420));
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, &video_capturer));
-    EXPECT_TRUE(SetSend(true));
-    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
-    EXPECT_EQ(0, renderer_.num_rendered_frames());
-    // Mute the channel and expect black output frame.
-    int frame_count = 0;
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
-    EXPECT_TRUE(video_capturer.CaptureFrame());
-    ++frame_count;
-    EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
-    EXPECT_TRUE(renderer_.black_frame());
-    // Unmute the channel and expect non-black output frame.
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
-    EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(30));
-    EXPECT_TRUE(video_capturer.CaptureFrame());
-    ++frame_count;
-    EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
-    EXPECT_FALSE(renderer_.black_frame());
-    // Test that we can also Mute using the correct send stream SSRC.
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
-    EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(30));
-    EXPECT_TRUE(video_capturer.CaptureFrame());
-    ++frame_count;
-    EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
-    EXPECT_TRUE(renderer_.black_frame());
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
-    EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(30));
-    EXPECT_TRUE(video_capturer.CaptureFrame());
-    ++frame_count;
-    EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
-    EXPECT_FALSE(renderer_.black_frame());
-    // Test that muting an existing stream succeeds even if it's muted.
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
-    // Test that unmuting an existing stream succeeds even if it's not muted.
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
-    // Test that muting an invalid stream fails.
-    EXPECT_FALSE(channel_->SetVideoSend(kSsrc+1, false, nullptr));
-    EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
-  }
-
   // Test that multiple send streams can be created and deleted properly.
   void MultipleSendStreams() {
     // Remove stream added in Setup. I.e. remove stream corresponding to default
@@ -1144,7 +1071,7 @@ class VideoMediaChannelTest : public testing::Test,
   }
 
   const std::unique_ptr<webrtc::Call> call_;
-  VideoEngineOverride<E> engine_;
+  E engine_;
   std::unique_ptr<cricket::FakeVideoCapturer> video_capturer_;
   std::unique_ptr<cricket::FakeVideoCapturer> video_capturer_2_;
   std::unique_ptr<C> channel_;
diff --git a/chromium/third_party/webrtc/media/base/videoframe.cc b/chromium/third_party/webrtc/media/base/videoframe.cc
index 6fd8d8168b0..d5c24adf3cb 100644
--- a/chromium/third_party/webrtc/media/base/videoframe.cc
+++ b/chromium/third_party/webrtc/media/base/videoframe.cc
@@ -31,15 +31,17 @@ bool VideoFrame::CopyToPlanes(uint8_t* dst_y,
                               int32_t dst_pitch_y,
                               int32_t dst_pitch_u,
                               int32_t dst_pitch_v) const {
-  if (!GetYPlane() || !GetUPlane() || !GetVPlane()) {
-    LOG(LS_ERROR) << "NULL plane pointer.";
+  const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer =
+      video_frame_buffer();
+  if (!buffer) {
+    LOG(LS_ERROR) << "NULL video buffer.";
     return false;
   }
   int32_t src_width = width();
   int32_t src_height = height();
-  return libyuv::I420Copy(GetYPlane(), GetYPitch(),
-                          GetUPlane(), GetUPitch(),
-                          GetVPlane(), GetVPitch(),
+  return libyuv::I420Copy(buffer->DataY(), buffer->StrideY(),
+                          buffer->DataU(), buffer->StrideU(),
+                          buffer->DataV(), buffer->StrideV(),
                           dst_y, dst_pitch_y,
                           dst_u, dst_pitch_u,
                           dst_v, dst_pitch_v,
@@ -56,9 +58,11 @@ size_t VideoFrame::ConvertToRgbBuffer(uint32_t to_fourcc,
     return needed;
   }
 
-  if (libyuv::ConvertFromI420(GetYPlane(), GetYPitch(), GetUPlane(),
-                              GetUPitch(), GetVPlane(), GetVPitch(), buffer,
-                              stride_rgb, width(), height(), to_fourcc)) {
+  if (libyuv::ConvertFromI420(
+          video_frame_buffer()->DataY(), video_frame_buffer()->StrideY(),
+          video_frame_buffer()->DataU(), video_frame_buffer()->StrideU(),
+          video_frame_buffer()->DataV(), video_frame_buffer()->StrideV(),
+          buffer, stride_rgb, width(), height(), to_fourcc)) {
     LOG(LS_ERROR) << "RGB type not supported: " << to_fourcc;
     return 0;  // 0 indicates error
   }
@@ -78,8 +82,8 @@ void VideoFrame::StretchToPlanes(uint8_t* dst_y,
                                  size_t dst_height,
                                  bool interpolate,
                                  bool vert_crop) const {
-  if (!GetYPlane() || !GetUPlane() || !GetVPlane()) {
-    LOG(LS_ERROR) << "NULL plane pointer.";
+  if (!video_frame_buffer()) {
+    LOG(LS_ERROR) << "NULL frame buffer.";
     return;
   }
 
@@ -89,9 +93,9 @@ void VideoFrame::StretchToPlanes(uint8_t* dst_y,
     CopyToPlanes(dst_y, dst_u, dst_v, dst_pitch_y, dst_pitch_u, dst_pitch_v);
     return;
   }
-  const uint8_t* src_y = GetYPlane();
-  const uint8_t* src_u = GetUPlane();
-  const uint8_t* src_v = GetVPlane();
+  const uint8_t* src_y = video_frame_buffer()->DataY();
+  const uint8_t* src_u = video_frame_buffer()->DataU();
+  const uint8_t* src_v = video_frame_buffer()->DataV();
 
   if (vert_crop) {
     // Adjust the input width:height ratio to be the same as the output ratio.
@@ -108,15 +112,16 @@ void VideoFrame::StretchToPlanes(uint8_t* dst_y,
       int32_t iheight_offset =
           static_cast<int32_t>((height() - src_height) >> 2);
       iheight_offset <<= 1;  // Ensure that iheight_offset is even.
-      src_y += iheight_offset * GetYPitch();
-      src_u += iheight_offset / 2 * GetUPitch();
-      src_v += iheight_offset / 2 * GetVPitch();
+      src_y += iheight_offset * video_frame_buffer()->StrideY();
+      src_u += iheight_offset / 2 * video_frame_buffer()->StrideU();
+      src_v += iheight_offset / 2 * video_frame_buffer()->StrideV();
     }
   }
 
   // Scale to the output I420 frame.
-  libyuv::Scale(src_y, src_u, src_v,
-                GetYPitch(), GetUPitch(), GetVPitch(),
+  libyuv::Scale(src_y, src_u, src_v, video_frame_buffer()->StrideY(),
+                video_frame_buffer()->StrideU(),
+                video_frame_buffer()->StrideV(),
                 static_cast<int>(src_width), static_cast<int>(src_height),
                 dst_y, dst_u, dst_v, dst_pitch_y, dst_pitch_u, dst_pitch_v,
                 static_cast<int>(dst_width), static_cast<int>(dst_height),
@@ -130,13 +135,17 @@ void VideoFrame::StretchToFrame(VideoFrame* dst,
     return;
   }
 
-  StretchToPlanes(dst->GetYPlane(), dst->GetUPlane(), dst->GetVPlane(),
-                  dst->GetYPitch(), dst->GetUPitch(), dst->GetVPitch(),
+  StretchToPlanes(dst->video_frame_buffer()->MutableDataY(),
+                  dst->video_frame_buffer()->MutableDataU(),
+                  dst->video_frame_buffer()->MutableDataV(),
+                  dst->video_frame_buffer()->StrideY(),
+                  dst->video_frame_buffer()->StrideU(),
+                  dst->video_frame_buffer()->StrideV(),
                   dst->width(), dst->height(),
                   interpolate, vert_crop);
   dst->SetTimeStamp(GetTimeStamp());
   // Stretched frame should have the same rotation as the source.
-  dst->SetRotation(GetVideoRotation());
+  dst->set_rotation(rotation());
 }
 
 VideoFrame* VideoFrame::Stretch(size_t dst_width, size_t dst_height,
@@ -151,9 +160,12 @@ VideoFrame* VideoFrame::Stretch(size_t dst_width, size_t dst_height,
 }
 
 bool VideoFrame::SetToBlack() {
-  return libyuv::I420Rect(GetYPlane(), GetYPitch(),
-                          GetUPlane(), GetUPitch(),
-                          GetVPlane(), GetVPitch(),
+  return libyuv::I420Rect(video_frame_buffer()->MutableDataY(),
+                          video_frame_buffer()->StrideY(),
+                          video_frame_buffer()->MutableDataU(),
+                          video_frame_buffer()->StrideU(),
+                          video_frame_buffer()->MutableDataV(),
+                          video_frame_buffer()->StrideV(),
                           0, 0,
                           width(), height(),
                           16, 128, 128) == 0;
diff --git a/chromium/third_party/webrtc/media/base/videoframe.h b/chromium/third_party/webrtc/media/base/videoframe.h
index 6045bc08b47..4026c26b3b1 100644
--- a/chromium/third_party/webrtc/media/base/videoframe.h
+++ b/chromium/third_party/webrtc/media/base/videoframe.h
@@ -35,18 +35,6 @@ class VideoFrame {
   virtual size_t GetWidth() const final { return width(); }
   virtual size_t GetHeight() const final { return height(); }
 
-  // These can return NULL if the object is not backed by a buffer.
-  virtual const uint8_t* GetYPlane() const = 0;
-  virtual const uint8_t* GetUPlane() const = 0;
-  virtual const uint8_t* GetVPlane() const = 0;
-  virtual uint8_t* GetYPlane() = 0;
-  virtual uint8_t* GetUPlane() = 0;
-  virtual uint8_t* GetVPlane() = 0;
-
-  virtual int32_t GetYPitch() const = 0;
-  virtual int32_t GetUPitch() const = 0;
-  virtual int32_t GetVPitch() const = 0;
-
   // Returns the handle of the underlying video frame. This is used when the
   // frame is backed by a texture. The object should be destroyed when it is no
   // longer in use, so the underlying resource can be freed.
@@ -54,14 +42,25 @@ class VideoFrame {
 
   // Returns the underlying video frame buffer. This function is ok to call
   // multiple times, but the returned object will refer to the same memory.
-  virtual rtc::scoped_refptr<webrtc::VideoFrameBuffer> GetVideoFrameBuffer()
-      const = 0;
+  virtual const rtc::scoped_refptr<webrtc::VideoFrameBuffer>&
+  video_frame_buffer() const = 0;
+
+  // System monotonic clock, same timebase as rtc::TimeMicros().
+  virtual int64_t timestamp_us() const = 0;
+  virtual void set_timestamp_us(int64_t time_us) = 0;
 
-  virtual int64_t GetTimeStamp() const = 0;
-  virtual void SetTimeStamp(int64_t time_stamp) = 0;
+  // Deprecated methods, for backwards compatibility.
+  // TODO(nisse): Delete when usage in Chrome and other applications
+  // have been replaced.
+  virtual int64_t GetTimeStamp() const {
+    return rtc::kNumNanosecsPerMicrosec * timestamp_us();
+  }
+  virtual void SetTimeStamp(int64_t time_ns) {
+    set_timestamp_us(time_ns / rtc::kNumNanosecsPerMicrosec);
+  }
 
   // Indicates the rotation angle in degrees.
-  virtual webrtc::VideoRotation GetVideoRotation() const  = 0;
+  virtual webrtc::VideoRotation rotation() const = 0;
 
   // Make a shallow copy of the frame. The frame buffer itself is not copied.
   // Both the current and new VideoFrame will share a single reference-counted
@@ -137,9 +136,10 @@ class VideoFrame {
                             int32_t dst_pitch_v) const;
 
   // Creates an empty frame.
-  virtual VideoFrame *CreateEmptyFrame(int w, int h,
-                                       int64_t time_stamp) const = 0;
-  virtual void SetRotation(webrtc::VideoRotation rotation) = 0;
+  virtual VideoFrame* CreateEmptyFrame(int w,
+                                       int h,
+                                       int64_t timestamp_us) const = 0;
+  virtual void set_rotation(webrtc::VideoRotation rotation) = 0;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/media/base/videoframe_unittest.h b/chromium/third_party/webrtc/media/base/videoframe_unittest.h
index 9e9b7dddb1f..6ceba833b5b 100644
--- a/chromium/third_party/webrtc/media/base/videoframe_unittest.h
+++ b/chromium/third_party/webrtc/media/base/videoframe_unittest.h
@@ -267,24 +267,27 @@ class VideoFrameTest : public testing::Test {
     const uint8_t* start = reinterpret_cast<const uint8_t*>(ms->GetBuffer());
     int awidth = (width + 1) & ~1;
     frame->InitToBlack(width, height, 0);
-    int stride_y = frame->GetYPitch();
-    int stride_u = frame->GetUPitch();
-    int stride_v = frame->GetVPitch();
+    int stride_y = frame->video_frame_buffer()->StrideY();
+    int stride_u = frame->video_frame_buffer()->StrideU();
+    int stride_v = frame->video_frame_buffer()->StrideV();
+    uint8_t* plane_y = frame->video_frame_buffer()->MutableDataY();
+    uint8_t* plane_u = frame->video_frame_buffer()->MutableDataU();
+    uint8_t* plane_v = frame->video_frame_buffer()->MutableDataV();
     for (uint32_t y = 0; y < height; ++y) {
       for (uint32_t x = 0; x < width; x += 2) {
         const uint8_t* quad1 = start + (y * awidth + x) * 2;
-        frame->GetYPlane()[stride_y * y + x] = quad1[y1_pos];
+        plane_y[stride_y * y + x] = quad1[y1_pos];
         if ((x + 1) < width) {
-          frame->GetYPlane()[stride_y * y + x + 1] = quad1[y2_pos];
+          plane_y[stride_y * y + x + 1] = quad1[y2_pos];
         }
         if ((y & 1) == 0) {
           const uint8_t* quad2 = quad1 + awidth * 2;
           if ((y + 1) >= height) {
             quad2 = quad1;
           }
-          frame->GetUPlane()[stride_u * (y / 2) + x / 2] =
+          plane_u[stride_u * (y / 2) + x / 2] =
               (quad1[u_pos] + quad2[u_pos] + 1) / 2;
-          frame->GetVPlane()[stride_v * (y / 2) + x / 2] =
+          plane_v[stride_v * (y / 2) + x / 2] =
               (quad1[v_pos] + quad2[v_pos] + 1) / 2;
         }
       }
@@ -311,9 +314,12 @@ class VideoFrameTest : public testing::Test {
       pitch = -pitch;
     }
     frame->InitToBlack(width, height, 0);
-    int stride_y = frame->GetYPitch();
-    int stride_u = frame->GetUPitch();
-    int stride_v = frame->GetVPitch();
+    int stride_y = frame->video_frame_buffer()->StrideY();
+    int stride_u = frame->video_frame_buffer()->StrideU();
+    int stride_v = frame->video_frame_buffer()->StrideV();
+    uint8_t* plane_y = frame->video_frame_buffer()->MutableDataY();
+    uint8_t* plane_u = frame->video_frame_buffer()->MutableDataU();
+    uint8_t* plane_v = frame->video_frame_buffer()->MutableDataV();
     for (int32_t y = 0; y < height; y += 2) {
       for (int32_t x = 0; x < width; x += 2) {
         const uint8_t* rgb[4];
@@ -326,19 +332,19 @@ class VideoFrameTest : public testing::Test {
           ConvertRgbPixel(rgb[i][r_pos], rgb[i][g_pos], rgb[i][b_pos],
                           &yuv[i][0], &yuv[i][1], &yuv[i][2]);
         }
-        frame->GetYPlane()[stride_y * y + x] = yuv[0][0];
+        plane_y[stride_y * y + x] = yuv[0][0];
         if ((x + 1) < width) {
-          frame->GetYPlane()[stride_y * y + x + 1] = yuv[1][0];
+          plane_y[stride_y * y + x + 1] = yuv[1][0];
         }
         if ((y + 1) < height) {
-          frame->GetYPlane()[stride_y * (y + 1) + x] = yuv[2][0];
+          plane_y[stride_y * (y + 1) + x] = yuv[2][0];
           if ((x + 1) < width) {
-            frame->GetYPlane()[stride_y * (y + 1) + x + 1] = yuv[3][0];
+            plane_y[stride_y * (y + 1) + x + 1] = yuv[3][0];
           }
         }
-        frame->GetUPlane()[stride_u * (y / 2) + x / 2] =
+        plane_u[stride_u * (y / 2) + x / 2] =
             (yuv[0][1] + yuv[1][1] + yuv[2][1] + yuv[3][1] + 2) / 4;
-        frame->GetVPlane()[stride_v * (y / 2) + x / 2] =
+        plane_v[stride_v * (y / 2) + x / 2] =
             (yuv[0][2] + yuv[1][2] + yuv[2][2] + yuv[3][2] + 2) / 4;
       }
     }
@@ -395,15 +401,15 @@ class VideoFrameTest : public testing::Test {
 
   // Comparison functions for testing.
   static bool IsNull(const cricket::VideoFrame& frame) {
-    return !frame.GetYPlane();
+    return !frame.video_frame_buffer();
   }
 
   static bool IsSize(const cricket::VideoFrame& frame,
                      int width,
                      int height) {
-    return !IsNull(frame) && frame.GetYPitch() >= width &&
-           frame.GetUPitch() >= width / 2 &&
-           frame.GetVPitch() >= width / 2 &&
+    return !IsNull(frame) && frame.video_frame_buffer()->StrideY() >= width &&
+           frame.video_frame_buffer()->StrideU() >= width / 2 &&
+           frame.video_frame_buffer()->StrideV() >= width / 2 &&
            frame.width() == width && frame.height() == height;
   }
 
@@ -444,15 +450,17 @@ class VideoFrameTest : public testing::Test {
                       const uint8_t* v,
                       uint32_t vpitch,
                       int max_error) {
-    return IsSize(frame, width, height) &&
-           frame.GetTimeStamp() == time_stamp &&
-           IsPlaneEqual("y", frame.GetYPlane(), frame.GetYPitch(), y, ypitch,
+    return IsSize(frame, width, height) && frame.GetTimeStamp() == time_stamp &&
+           IsPlaneEqual("y", frame.video_frame_buffer()->DataY(),
+                        frame.video_frame_buffer()->StrideY(), y, ypitch,
                         static_cast<uint32_t>(width),
                         static_cast<uint32_t>(height), max_error) &&
-           IsPlaneEqual("u", frame.GetUPlane(), frame.GetUPitch(), u, upitch,
+           IsPlaneEqual("u", frame.video_frame_buffer()->DataU(),
+                        frame.video_frame_buffer()->StrideU(), u, upitch,
                         static_cast<uint32_t>((width + 1) / 2),
                         static_cast<uint32_t>((height + 1) / 2), max_error) &&
-           IsPlaneEqual("v", frame.GetVPlane(), frame.GetVPitch(), v, vpitch,
+           IsPlaneEqual("v", frame.video_frame_buffer()->DataV(),
+                        frame.video_frame_buffer()->StrideV(), v, vpitch,
                         static_cast<uint32_t>((width + 1) / 2),
                         static_cast<uint32_t>((height + 1) / 2), max_error);
   }
@@ -463,9 +471,12 @@ class VideoFrameTest : public testing::Test {
     return IsEqual(frame1,
                    frame2.width(), frame2.height(),
                    frame2.GetTimeStamp(),
-                   frame2.GetYPlane(), frame2.GetYPitch(),
-                   frame2.GetUPlane(), frame2.GetUPitch(),
-                   frame2.GetVPlane(), frame2.GetVPitch(),
+                   frame2.video_frame_buffer()->DataY(),
+                   frame2.video_frame_buffer()->StrideY(),
+                   frame2.video_frame_buffer()->DataU(),
+                   frame2.video_frame_buffer()->StrideU(),
+                   frame2.video_frame_buffer()->DataV(),
+                   frame2.video_frame_buffer()->StrideV(),
                    max_error);
   }
 
@@ -478,23 +489,26 @@ class VideoFrameTest : public testing::Test {
                    frame2.width() - hcrop * 2,
                    frame2.height() - vcrop * 2,
                    frame2.GetTimeStamp(),
-                   frame2.GetYPlane() + vcrop * frame2.GetYPitch()
+                   frame2.video_frame_buffer()->DataY()
+                       + vcrop * frame2.video_frame_buffer()->StrideY()
                        + hcrop,
-                   frame2.GetYPitch(),
-                   frame2.GetUPlane() + vcrop * frame2.GetUPitch() / 2
+                   frame2.video_frame_buffer()->StrideY(),
+                   frame2.video_frame_buffer()->DataU()
+                       + vcrop * frame2.video_frame_buffer()->StrideU() / 2
                        + hcrop / 2,
-                   frame2.GetUPitch(),
-                   frame2.GetVPlane() + vcrop * frame2.GetVPitch() / 2
+                   frame2.video_frame_buffer()->StrideU(),
+                   frame2.video_frame_buffer()->DataV()
+                       + vcrop * frame2.video_frame_buffer()->StrideV() / 2
                        + hcrop / 2,
-                   frame2.GetVPitch(),
+                   frame2.video_frame_buffer()->StrideV(),
                    max_error);
   }
 
   static bool IsBlack(const cricket::VideoFrame& frame) {
     return !IsNull(frame) &&
-        *frame.GetYPlane() == 16 &&
-        *frame.GetUPlane() == 128 &&
-        *frame.GetVPlane() == 128;
+           *frame.video_frame_buffer()->DataY() == 16 &&
+           *frame.video_frame_buffer()->DataU() == 128 &&
+           *frame.video_frame_buffer()->DataV() == 128;
   }
 
   ////////////////////////
@@ -541,9 +555,12 @@ class VideoFrameTest : public testing::Test {
     uint8_t* y = ALIGNP(buf.get(), kAlignment);
     uint8_t* u = y + kWidth * kHeight;
     uint8_t* v = u + (kWidth / 2) * kHeight;
-    EXPECT_EQ(0, libyuv::I420ToI422(frame1.GetYPlane(), frame1.GetYPitch(),
-                                    frame1.GetUPlane(), frame1.GetUPitch(),
-                                    frame1.GetVPlane(), frame1.GetVPitch(),
+    EXPECT_EQ(0, libyuv::I420ToI422(frame1.video_frame_buffer()->DataY(),
+                                    frame1.video_frame_buffer()->StrideY(),
+                                    frame1.video_frame_buffer()->DataU(),
+                                    frame1.video_frame_buffer()->StrideU(),
+                                    frame1.video_frame_buffer()->DataV(),
+                                    frame1.video_frame_buffer()->StrideV(),
                                     y, kWidth,
                                     u, kWidth / 2,
                                     v, kWidth / 2,
@@ -560,9 +577,12 @@ class VideoFrameTest : public testing::Test {
     size_t buf_size = kWidth * kHeight * 2;
     std::unique_ptr<uint8_t[]> buf(new uint8_t[buf_size + kAlignment]);
     uint8_t* yuy2 = ALIGNP(buf.get(), kAlignment);
-    EXPECT_EQ(0, libyuv::I420ToYUY2(frame1.GetYPlane(), frame1.GetYPitch(),
-                                    frame1.GetUPlane(), frame1.GetUPitch(),
-                                    frame1.GetVPlane(), frame1.GetVPitch(),
+    EXPECT_EQ(0, libyuv::I420ToYUY2(frame1.video_frame_buffer()->DataY(),
+                                    frame1.video_frame_buffer()->StrideY(),
+                                    frame1.video_frame_buffer()->DataU(),
+                                    frame1.video_frame_buffer()->StrideU(),
+                                    frame1.video_frame_buffer()->DataV(),
+                                    frame1.video_frame_buffer()->StrideV(),
                                     yuy2, kWidth * 2,
                                     kWidth, kHeight));
     EXPECT_TRUE(LoadFrame(yuy2, buf_size, cricket::FOURCC_YUY2,
@@ -577,9 +597,12 @@ class VideoFrameTest : public testing::Test {
     size_t buf_size = kWidth * kHeight * 2;
     std::unique_ptr<uint8_t[]> buf(new uint8_t[buf_size + kAlignment + 1]);
     uint8_t* yuy2 = ALIGNP(buf.get(), kAlignment) + 1;
-    EXPECT_EQ(0, libyuv::I420ToYUY2(frame1.GetYPlane(), frame1.GetYPitch(),
-                                    frame1.GetUPlane(), frame1.GetUPitch(),
-                                    frame1.GetVPlane(), frame1.GetVPitch(),
+    EXPECT_EQ(0, libyuv::I420ToYUY2(frame1.video_frame_buffer()->DataY(),
+                                    frame1.video_frame_buffer()->StrideY(),
+                                    frame1.video_frame_buffer()->DataU(),
+                                    frame1.video_frame_buffer()->StrideU(),
+                                    frame1.video_frame_buffer()->DataV(),
+                                    frame1.video_frame_buffer()->StrideV(),
                                     yuy2, kWidth * 2,
                                     kWidth, kHeight));
     EXPECT_TRUE(LoadFrame(yuy2, buf_size, cricket::FOURCC_YUY2,
@@ -792,16 +815,23 @@ class VideoFrameTest : public testing::Test {
     EXPECT_TRUE(frame2.Init(cricket::FOURCC_##FOURCC, kWidth, kHeight, kWidth, \
                             kHeight,                                           \
                             reinterpret_cast<uint8_t*>(ms->GetBuffer()),       \
-                            data_size, 0, webrtc::kVideoRotation_0));    \
-    int width_rotate = frame1.width();                                  \
-    int height_rotate = frame1.height();                                \
-    EXPECT_TRUE(frame3.InitToBlack(width_rotate, height_rotate, 0));     \
-    libyuv::I420Mirror(                                                        \
-        frame2.GetYPlane(), frame2.GetYPitch(), frame2.GetUPlane(),            \
-        frame2.GetUPitch(), frame2.GetVPlane(), frame2.GetVPitch(),            \
-        frame3.GetYPlane(), frame3.GetYPitch(), frame3.GetUPlane(),            \
-        frame3.GetUPitch(), frame3.GetVPlane(), frame3.GetVPitch(), kWidth,    \
-        kHeight);                                                              \
+                            data_size, 0, webrtc::kVideoRotation_0));          \
+    int width_rotate = frame1.width();                                         \
+    int height_rotate = frame1.height();                                       \
+    EXPECT_TRUE(frame3.InitToBlack(width_rotate, height_rotate, 0));           \
+    libyuv::I420Mirror(frame2.video_frame_buffer()->DataY(),                   \
+                       frame2.video_frame_buffer()->StrideY(),                 \
+                       frame2.video_frame_buffer()->DataU(),                   \
+                       frame2.video_frame_buffer()->StrideU(),                 \
+                       frame2.video_frame_buffer()->DataV(),                   \
+                       frame2.video_frame_buffer()->StrideV(),                 \
+                       frame3.video_frame_buffer()->MutableDataY(),            \
+                       frame3.video_frame_buffer()->StrideY(),                 \
+                       frame3.video_frame_buffer()->MutableDataU(),            \
+                       frame3.video_frame_buffer()->StrideU(),                 \
+                       frame3.video_frame_buffer()->MutableDataV(),            \
+                       frame3.video_frame_buffer()->StrideV(),                 \
+                       kWidth, kHeight);                                       \
     EXPECT_TRUE(IsEqual(frame1, frame3, 0));                                   \
   }
 
@@ -823,16 +853,23 @@ class VideoFrameTest : public testing::Test {
     EXPECT_TRUE(frame2.Init(cricket::FOURCC_##FOURCC, kWidth, kHeight, kWidth, \
                             kHeight,                                           \
                             reinterpret_cast<uint8_t*>(ms->GetBuffer()),       \
-                            data_size, 0, webrtc::kVideoRotation_0));    \
-    int width_rotate = frame1.width();                                  \
-    int height_rotate = frame1.height();                                \
-    EXPECT_TRUE(frame3.InitToBlack(width_rotate, height_rotate, 0));     \
-    libyuv::I420Rotate(                                                        \
-        frame2.GetYPlane(), frame2.GetYPitch(), frame2.GetUPlane(),            \
-        frame2.GetUPitch(), frame2.GetVPlane(), frame2.GetVPitch(),            \
-        frame3.GetYPlane(), frame3.GetYPitch(), frame3.GetUPlane(),            \
-        frame3.GetUPitch(), frame3.GetVPlane(), frame3.GetVPitch(), kWidth,    \
-        kHeight, libyuv::kRotate##ROTATE);                                     \
+                            data_size, 0, webrtc::kVideoRotation_0));          \
+    int width_rotate = frame1.width();                                         \
+    int height_rotate = frame1.height();                                       \
+    EXPECT_TRUE(frame3.InitToBlack(width_rotate, height_rotate, 0));           \
+    libyuv::I420Rotate(frame2.video_frame_buffer()->DataY(),                   \
+                       frame2.video_frame_buffer()->StrideY(),                 \
+                       frame2.video_frame_buffer()->DataU(),                   \
+                       frame2.video_frame_buffer()->StrideU(),                 \
+                       frame2.video_frame_buffer()->DataV(),                   \
+                       frame2.video_frame_buffer()->StrideV(),                 \
+                       frame3.video_frame_buffer()->MutableDataY(),            \
+                       frame3.video_frame_buffer()->StrideY(),                 \
+                       frame3.video_frame_buffer()->MutableDataU(),            \
+                       frame3.video_frame_buffer()->StrideU(),                 \
+                       frame3.video_frame_buffer()->MutableDataV(),            \
+                       frame3.video_frame_buffer()->StrideV(),                 \
+                       kWidth, kHeight, libyuv::kRotate##ROTATE);              \
     EXPECT_TRUE(IsEqual(frame1, frame3, 0));                                   \
   }
 
@@ -952,9 +989,9 @@ class VideoFrameTest : public testing::Test {
     }
     EXPECT_EQ(5, frame.width());
     EXPECT_EQ(5, frame.height());
-    EXPECT_EQ(5, frame.GetYPitch());
-    EXPECT_EQ(3, frame.GetUPitch());
-    EXPECT_EQ(3, frame.GetVPitch());
+    EXPECT_EQ(5, frame.video_frame_buffer()->StrideY());
+    EXPECT_EQ(3, frame.video_frame_buffer()->StrideU());
+    EXPECT_EQ(3, frame.video_frame_buffer()->StrideV());
   }
 
   // Test 1 pixel edge case image ARGB buffer.
@@ -1121,8 +1158,10 @@ class VideoFrameTest : public testing::Test {
     ASSERT_TRUE(LoadFrameNoRepeat(&frame1));
     ASSERT_TRUE(LoadFrame(kJpeg400Filename,
                           cricket::FOURCC_MJPG, kWidth, kHeight, &frame2));
-    EXPECT_TRUE(IsPlaneEqual("y", frame1.GetYPlane(), frame1.GetYPitch(),
-                             frame2.GetYPlane(), frame2.GetYPitch(),
+    EXPECT_TRUE(IsPlaneEqual("y", frame1.video_frame_buffer()->DataY(),
+                             frame1.video_frame_buffer()->StrideY(),
+                             frame2.video_frame_buffer()->DataY(),
+                             frame2.video_frame_buffer()->StrideY(),
                              kWidth, kHeight, 32));
     EXPECT_TRUE(IsEqual(frame1, frame2, 128));
   }
@@ -1304,9 +1343,7 @@ class VideoFrameTest : public testing::Test {
       EXPECT_TRUE(frame2.Init(frame1));
     }
     EXPECT_TRUE(IsEqual(frame1, frame2, 0));
-    EXPECT_EQ(frame1.GetYPlane(), frame2.GetYPlane());
-    EXPECT_EQ(frame1.GetUPlane(), frame2.GetUPlane());
-    EXPECT_EQ(frame1.GetVPlane(), frame2.GetVPlane());
+    EXPECT_EQ(frame1.video_frame_buffer(), frame2.video_frame_buffer());
   }
 
   // Test creating an empty image and initing it to black.
@@ -1419,9 +1456,12 @@ class VideoFrameTest : public testing::Test {
     EXPECT_TRUE(frame2.InitToBlack(kWidth, kHeight, 0));
     for (int i = 0; i < repeat_from; ++i) {
       EXPECT_EQ(0, RGBToI420(out, stride,
-                             frame2.GetYPlane(), frame2.GetYPitch(),
-                             frame2.GetUPlane(), frame2.GetUPitch(),
-                             frame2.GetVPlane(), frame2.GetVPitch(),
+                             frame2.video_frame_buffer()->MutableDataY(),
+                             frame2.video_frame_buffer()->StrideY(),
+                             frame2.video_frame_buffer()->MutableDataU(),
+                             frame2.video_frame_buffer()->StrideU(),
+                             frame2.video_frame_buffer()->MutableDataV(),
+                             frame2.video_frame_buffer()->StrideV(),
                              kWidth, kHeight));
     }
     if (rowpad) {
@@ -1724,9 +1764,12 @@ class VideoFrameTest : public testing::Test {
     uint8_t* v = u + (kWidth / 2) * kHeight;
     ASSERT_TRUE(LoadFrameNoRepeat(&frame1));
     for (int i = 0; i < repeat_; ++i) {
-      EXPECT_EQ(0, libyuv::I420ToI422(frame1.GetYPlane(), frame1.GetYPitch(),
-                                      frame1.GetUPlane(), frame1.GetUPitch(),
-                                      frame1.GetVPlane(), frame1.GetVPitch(),
+      EXPECT_EQ(0, libyuv::I420ToI422(frame1.video_frame_buffer()->DataY(),
+                                      frame1.video_frame_buffer()->StrideY(),
+                                      frame1.video_frame_buffer()->DataU(),
+                                      frame1.video_frame_buffer()->StrideU(),
+                                      frame1.video_frame_buffer()->DataV(),
+                                      frame1.video_frame_buffer()->StrideV(),
                                       y, kWidth,
                                       u, kWidth / 2,
                                       v, kWidth / 2,
@@ -1749,7 +1792,8 @@ class VideoFrameTest : public testing::Test {
     target.reset(source->Copy());
     EXPECT_TRUE(IsEqual(*source, *target, 0));
     source.reset();
-    EXPECT_TRUE(target->GetYPlane() != NULL);
+    ASSERT_TRUE(target->video_frame_buffer() != NULL);
+    EXPECT_TRUE(target->video_frame_buffer()->DataY() != NULL);
   }
 
   void CopyIsRef() {
@@ -1759,9 +1803,7 @@ class VideoFrameTest : public testing::Test {
     target.reset(source->Copy());
     EXPECT_TRUE(IsEqual(*source, *target, 0));
     const T* const_source = source.get();
-    EXPECT_EQ(const_source->GetYPlane(), target->GetYPlane());
-    EXPECT_EQ(const_source->GetUPlane(), target->GetUPlane());
-    EXPECT_EQ(const_source->GetVPlane(), target->GetVPlane());
+    EXPECT_EQ(const_source->video_frame_buffer(), target->video_frame_buffer());
   }
 
   void StretchToFrame() {
diff --git a/chromium/third_party/webrtc/media/engine/fakewebrtccall.cc b/chromium/third_party/webrtc/media/engine/fakewebrtccall.cc
index 8eff0ebcf8e..e9edf27bcb6 100644
--- a/chromium/third_party/webrtc/media/engine/fakewebrtccall.cc
+++ b/chromium/third_party/webrtc/media/engine/fakewebrtccall.cc
@@ -67,8 +67,17 @@ void FakeAudioReceiveStream::SetStats(
   stats_ = stats;
 }
 
-void FakeAudioReceiveStream::IncrementReceivedPackets() {
-  received_packets_++;
+bool FakeAudioReceiveStream::VerifyLastPacket(const uint8_t* data,
+                                              size_t length) const {
+  return last_packet_ == rtc::Buffer(data, length);
+}
+
+bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                        size_t length,
+                                        const webrtc::PacketTime& packet_time) {
+  ++received_packets_;
+  last_packet_.SetData(packet, length);
+  return true;
 }
 
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
@@ -409,7 +418,7 @@ FakeCall::DeliveryStatus FakeCall::DeliverPacket(
       media_type == webrtc::MediaType::AUDIO) {
     for (auto receiver : audio_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
-        receiver->IncrementReceivedPackets();
+        receiver->DeliverRtp(packet, length, packet_time);
         return DELIVERY_OK;
       }
     }
diff --git a/chromium/third_party/webrtc/media/engine/fakewebrtccall.h b/chromium/third_party/webrtc/media/engine/fakewebrtccall.h
index 9caa02a9b71..c9caf8e158e 100644
--- a/chromium/third_party/webrtc/media/engine/fakewebrtccall.h
+++ b/chromium/third_party/webrtc/media/engine/fakewebrtccall.h
@@ -25,6 +25,7 @@
 
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
+#include "webrtc/base/buffer.h"
 #include "webrtc/call.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/video_receive_stream.h"
@@ -47,15 +48,10 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
   bool IsSending() const { return sending_; }
 
  private:
-  // webrtc::SendStream implementation.
+  // webrtc::AudioSendStream implementation.
   void Start() override { sending_ = true; }
   void Stop() override { sending_ = false; }
-  void SignalNetworkState(webrtc::NetworkState state) override {}
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
-    return true;
-  }
 
-  // webrtc::AudioSendStream implementation.
   bool SendTelephoneEvent(int payload_type, int event,
                           int duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
@@ -74,24 +70,17 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
   int received_packets() const { return received_packets_; }
-  void IncrementReceivedPackets();
+  bool VerifyLastPacket(const uint8_t* data, size_t length) const;
   const webrtc::AudioSinkInterface* sink() const { return sink_.get(); }
+  bool DeliverRtp(const uint8_t* packet,
+                  size_t length,
+                  const webrtc::PacketTime& packet_time);
 
  private:
-  // webrtc::ReceiveStream implementation.
+  // webrtc::AudioReceiveStream implementation.
   void Start() override {}
   void Stop() override {}
-  void SignalNetworkState(webrtc::NetworkState state) override {}
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
-    return true;
-  }
-  bool DeliverRtp(const uint8_t* packet,
-                  size_t length,
-                  const webrtc::PacketTime& packet_time) override {
-    return true;
-  }
 
-  // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
@@ -99,6 +88,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
+  rtc::Buffer last_packet_;
 };
 
 class FakeVideoSendStream final : public webrtc::VideoSendStream,
@@ -126,15 +116,9 @@ class FakeVideoSendStream final : public webrtc::VideoSendStream,
  private:
   void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override;
 
-  // webrtc::SendStream implementation.
+  // webrtc::VideoSendStream implementation.
   void Start() override;
   void Stop() override;
-  void SignalNetworkState(webrtc::NetworkState state) override {}
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
-    return true;
-  }
-
-  // webrtc::VideoSendStream implementation.
   webrtc::VideoSendStream::Stats GetStats() override;
   void ReconfigureVideoEncoder(
       const webrtc::VideoEncoderConfig& config) override;
@@ -168,20 +152,10 @@ class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
   void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
 
  private:
-  // webrtc::ReceiveStream implementation.
+  // webrtc::VideoReceiveStream implementation.
   void Start() override;
   void Stop() override;
-  void SignalNetworkState(webrtc::NetworkState state) override {}
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override {
-    return true;
-  }
-  bool DeliverRtp(const uint8_t* packet,
-                  size_t length,
-                  const webrtc::PacketTime& packet_time) override {
-    return true;
-  }
 
-  // webrtc::VideoReceiveStream implementation.
   webrtc::VideoReceiveStream::Stats GetStats() const override;
 
   webrtc::VideoReceiveStream::Config config_;
@@ -239,6 +213,8 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
+  void OnNetworkRouteChanged(const std::string& transport_name,
+                             const rtc::NetworkRoute& network_route) override {}
   void SignalChannelNetworkState(webrtc::MediaType media,
                                  webrtc::NetworkState state) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
diff --git a/chromium/third_party/webrtc/media/engine/fakewebrtcvideocapturemodule.h b/chromium/third_party/webrtc/media/engine/fakewebrtcvideocapturemodule.h
index 3202502e6c5..2bc715b1e1f 100644
--- a/chromium/third_party/webrtc/media/engine/fakewebrtcvideocapturemodule.h
+++ b/chromium/third_party/webrtc/media/engine/fakewebrtcvideocapturemodule.h
@@ -87,11 +87,16 @@ class FakeWebRtcVideoCaptureModule : public webrtc::VideoCaptureModule {
 
   void SendFrame(int w, int h) {
     if (!running_) return;
-    webrtc::VideoFrame sample;
-    // Setting stride based on width.
-    sample.CreateEmptyFrame(w, h, w, (w + 1) / 2, (w + 1) / 2);
+
+    rtc::scoped_refptr<webrtc::I420Buffer> buffer =
+        new rtc::RefCountedObject<webrtc::I420Buffer>(w, h);
+    // Initialize memory to satisfy DrMemory tests. See
+    // https://bugs.chromium.org/p/libyuv/issues/detail?id=377
+    buffer->InitializeData();
     if (callback_) {
-      callback_->OnIncomingCapturedFrame(id_, sample);
+      callback_->OnIncomingCapturedFrame(
+          id_,
+          webrtc::VideoFrame(buffer, 0, 0, webrtc::kVideoRotation_0));
     }
   }
 
diff --git a/chromium/third_party/webrtc/media/engine/fakewebrtcvideoengine.h b/chromium/third_party/webrtc/media/engine/fakewebrtcvideoengine.h
index 475797b16d5..f8b8cbb4925 100644
--- a/chromium/third_party/webrtc/media/engine/fakewebrtcvideoengine.h
+++ b/chromium/third_party/webrtc/media/engine/fakewebrtcvideoengine.h
@@ -38,13 +38,12 @@ static const int kMaxVideoBitrate = 1000;
 // renderer for a channel or it is adding a renderer for a capturer.
 static const int kViEChannelIdBase = 0;
 static const int kViEChannelIdMax = 1000;
+static const int kEventTimeoutMs = 10000;
 
 // Fake class for mocking out webrtc::VideoDecoder
 class FakeWebRtcVideoDecoder : public webrtc::VideoDecoder {
  public:
-  FakeWebRtcVideoDecoder()
-      : num_frames_received_(0) {
-  }
+  FakeWebRtcVideoDecoder() : num_frames_received_(0) {}
 
   virtual int32_t InitDecode(const webrtc::VideoCodec*, int32_t) {
     return WEBRTC_VIDEO_CODEC_OK;
@@ -120,16 +119,20 @@ class FakeWebRtcVideoDecoderFactory : public WebRtcVideoDecoderFactory {
 // Fake class for mocking out webrtc::VideoEnoder
 class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
  public:
-  FakeWebRtcVideoEncoder() : num_frames_encoded_(0) {}
+  FakeWebRtcVideoEncoder()
+      : init_encode_event_(false, false), num_frames_encoded_(0) {}
 
   virtual int32_t InitEncode(const webrtc::VideoCodec* codecSettings,
                              int32_t numberOfCores,
                              size_t maxPayloadSize) {
     rtc::CritScope lock(&crit_);
     codec_settings_ = *codecSettings;
+    init_encode_event_.Set();
     return WEBRTC_VIDEO_CODEC_OK;
   }
 
+  bool WaitForInitEncode() { return init_encode_event_.Wait(kEventTimeoutMs); }
+
   webrtc::VideoCodec GetCodecSettings() {
     rtc::CritScope lock(&crit_);
     return codec_settings_;
@@ -140,6 +143,7 @@ class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
                          const std::vector<webrtc::FrameType>* frame_types) {
     rtc::CritScope lock(&crit_);
     ++num_frames_encoded_;
+    init_encode_event_.Set();
     return WEBRTC_VIDEO_CODEC_OK;
   }
 
@@ -165,6 +169,7 @@ class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
 
  private:
   rtc::CriticalSection crit_;
+  rtc::Event init_encode_event_;
   int num_frames_encoded_ GUARDED_BY(crit_);
   webrtc::VideoCodec codec_settings_ GUARDED_BY(crit_);
 };
@@ -173,32 +178,45 @@ class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
 class FakeWebRtcVideoEncoderFactory : public WebRtcVideoEncoderFactory {
  public:
   FakeWebRtcVideoEncoderFactory()
-      : num_created_encoders_(0), encoders_have_internal_sources_(false) {}
+      : created_video_encoder_event_(false, false),
+        num_created_encoders_(0),
+        encoders_have_internal_sources_(false) {}
 
-  virtual webrtc::VideoEncoder* CreateVideoEncoder(
-      webrtc::VideoCodecType type) {
+  webrtc::VideoEncoder* CreateVideoEncoder(
+      webrtc::VideoCodecType type) override {
+    rtc::CritScope lock(&crit_);
     if (supported_codec_types_.count(type) == 0) {
       return NULL;
     }
     FakeWebRtcVideoEncoder* encoder = new FakeWebRtcVideoEncoder();
     encoders_.push_back(encoder);
     num_created_encoders_++;
+    created_video_encoder_event_.Set();
     return encoder;
   }
 
-  virtual void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) {
+  bool WaitForCreatedVideoEncoders(int num_encoders) {
+    while (created_video_encoder_event_.Wait(kEventTimeoutMs)) {
+      if (GetNumCreatedEncoders() >= num_encoders)
+        return true;
+    }
+    return false;
+  }
+
+  void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override {
+    rtc::CritScope lock(&crit_);
     encoders_.erase(
         std::remove(encoders_.begin(), encoders_.end(), encoder),
         encoders_.end());
     delete encoder;
   }
 
-  virtual const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs()
-      const {
+  const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs()
+      const override {
     return codecs_;
   }
 
-  virtual bool EncoderTypeHasInternalSource(
+  bool EncoderTypeHasInternalSource(
       webrtc::VideoCodecType type) const override {
     return encoders_have_internal_sources_;
   }
@@ -215,18 +233,22 @@ class FakeWebRtcVideoEncoderFactory : public WebRtcVideoEncoderFactory {
   }
 
   int GetNumCreatedEncoders() {
+    rtc::CritScope lock(&crit_);
     return num_created_encoders_;
   }
 
-  const std::vector<FakeWebRtcVideoEncoder*>& encoders() {
+  const std::vector<FakeWebRtcVideoEncoder*> encoders() {
+    rtc::CritScope lock(&crit_);
     return encoders_;
   }
 
  private:
+  rtc::CriticalSection crit_;
+  rtc::Event created_video_encoder_event_;
   std::set<webrtc::VideoCodecType> supported_codec_types_;
   std::vector<WebRtcVideoEncoderFactory::VideoCodec> codecs_;
-  std::vector<FakeWebRtcVideoEncoder*> encoders_;
-  int num_created_encoders_;
+  std::vector<FakeWebRtcVideoEncoder*> encoders_ GUARDED_BY(crit_);
+  int num_created_encoders_ GUARDED_BY(crit_);
   bool encoders_have_internal_sources_;
 };
 
diff --git a/chromium/third_party/webrtc/media/engine/fakewebrtcvoiceengine.h b/chromium/third_party/webrtc/media/engine/fakewebrtcvoiceengine.h
index b5ad81c6a98..13a36968531 100644
--- a/chromium/third_party/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/chromium/third_party/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -121,15 +121,13 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
-      public webrtc::VoEHardware,
-      public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
+      public webrtc::VoEHardware, public webrtc::VoERTP_RTCP,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
     Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
-    bool external_transport = false;
     bool playout = false;
     float volume_scale = 1.0f;
     bool vad = false;
@@ -146,8 +144,6 @@ class FakeWebRtcVoiceEngine
     int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
-    webrtc::PacketTime last_rtp_packet_time;
-    std::list<std::string> packets;
     int neteq_capacity = -1;
     bool neteq_fast_accelerate = false;
   };
@@ -191,10 +187,6 @@ class FakeWebRtcVoiceEngine
   int GetNACKMaxPackets(int channel) {
     return channels_[channel]->nack_max_packets;
   }
-  const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
-    RTC_DCHECK(channels_.find(channel) != channels_.end());
-    return channels_[channel]->last_rtp_packet_time;
-  }
   int GetSendCNPayloadType(int channel, bool wideband) {
     return (wideband) ?
         channels_[channel]->cn16_type :
@@ -203,18 +195,6 @@ class FakeWebRtcVoiceEngine
   int GetSendREDPayloadType(int channel) {
     return channels_[channel]->red_type;
   }
-  bool CheckPacket(int channel, const void* data, size_t len) {
-    bool result = !CheckNoPacket(channel);
-    if (result) {
-      std::string packet = channels_[channel]->packets.front();
-      result = (packet == std::string(static_cast<const char*>(data), len));
-      channels_[channel]->packets.pop_front();
-    }
-    return result;
-  }
-  bool CheckNoPacket(int channel) {
-    return channels_[channel]->packets.empty();
-  }
   void set_playout_fail_channel(int channel) {
     playout_fail_channel_ = channel;
   }
@@ -310,7 +290,7 @@ class FakeWebRtcVoiceEngine
     channels_[channel]->associate_send_channel = accociate_send_channel;
     return 0;
   }
-  webrtc::RtcEventLog* GetEventLog() { return nullptr; }
+  webrtc::RtcEventLog* GetEventLog() override { return nullptr; }
 
   // webrtc::VoECodec
   WEBRTC_STUB(NumOfCodecs, ());
@@ -456,62 +436,16 @@ class FakeWebRtcVoiceEngine
   WEBRTC_STUB(SetPlayoutDevice, (int));
   WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
   WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
-  WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
-    recording_sample_rate_ = samples_per_sec;
-    return 0;
-  }
-  WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) {
-    *samples_per_sec = recording_sample_rate_;
-    return 0;
-  }
-  WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) {
-    playout_sample_rate_ = samples_per_sec;
-    return 0;
-  }
-  WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) {
-    *samples_per_sec = playout_sample_rate_;
-    return 0;
-  }
+  WEBRTC_STUB(SetRecordingSampleRate, (unsigned int samples_per_sec));
+  WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
+  WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
+  WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
-  virtual bool BuiltInAECIsAvailable() const { return false; }
+  bool BuiltInAECIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
-  virtual bool BuiltInAGCIsAvailable() const { return false; }
+  bool BuiltInAGCIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
-  virtual bool BuiltInNSIsAvailable() const { return false; }
-
-  // webrtc::VoENetwork
-  WEBRTC_FUNC(RegisterExternalTransport, (int channel,
-                                          webrtc::Transport& transport)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->external_transport = true;
-    return 0;
-  }
-  WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->external_transport = false;
-    return 0;
-  }
-  WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
-                                  size_t length)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (!channels_[channel]->external_transport) return -1;
-    channels_[channel]->packets.push_back(
-        std::string(static_cast<const char*>(data), length));
-    return 0;
-  }
-  WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
-                                  size_t length,
-                                  const webrtc::PacketTime& packet_time)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    if (ReceivedRTPPacket(channel, data, length) == -1) {
-      return -1;
-    }
-    channels_[channel]->last_rtp_packet_time = packet_time;
-    return 0;
-  }
-
-  WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
-                                   size_t length));
+  bool BuiltInNSIsAvailable() const override { return false; }
 
   // webrtc::VoERTP_RTCP
   WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
@@ -685,17 +619,17 @@ class FakeWebRtcVoiceEngine
                                              int reportingThreshold,
                                              int penaltyDecay,
                                              int typeEventDelay));
-  int EnableHighPassFilter(bool enable) {
+  int EnableHighPassFilter(bool enable) override {
     highpass_filter_enabled_ = enable;
     return 0;
   }
-  bool IsHighPassFilterEnabled() {
+  bool IsHighPassFilterEnabled() override {
     return highpass_filter_enabled_;
   }
-  bool IsStereoChannelSwappingEnabled() {
+  bool IsStereoChannelSwappingEnabled() override {
     return stereo_swapping_enabled_;
   }
-  void EnableStereoChannelSwapping(bool enable) {
+  void EnableStereoChannelSwapping(bool enable) override {
     stereo_swapping_enabled_ = enable;
   }
   int GetNetEqCapacity() const {
@@ -729,8 +663,6 @@ class FakeWebRtcVoiceEngine
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   int playout_fail_channel_ = -1;
-  int recording_sample_rate_ = -1;
-  int playout_sample_rate_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.cc b/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.cc
index a5405510597..71c88413f89 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.cc
@@ -278,7 +278,7 @@ CaptureState WebRtcVideoCapturer::Start(const VideoFormat& capture_format) {
     return CS_FAILED;
   }
 
-  uint32_t start = rtc::Time();
+  int64_t start = rtc::TimeMillis();
   module_->RegisterCaptureDataCallback(*this);
   if (module_->StartCapture(cap) != 0) {
     LOG(LS_ERROR) << "Camera '" << GetId() << "' failed to start";
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.h b/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.h
index b6b39386c1b..1efa4ad66fe 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.h
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideocapturer.h
@@ -61,14 +61,14 @@ class WebRtcVideoCapturer : public VideoCapturer,
  protected:
   void OnSinkWantsChanged(const rtc::VideoSinkWants& wants) override;
   // Override virtual methods of the parent class VideoCapturer.
-  virtual bool GetPreferredFourccs(std::vector<uint32_t>* fourccs);
+  bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) override;
 
  private:
   // Callback when a frame is captured by camera.
-  virtual void OnIncomingCapturedFrame(const int32_t id,
-                                       const webrtc::VideoFrame& frame);
-  virtual void OnCaptureDelayChanged(const int32_t id,
-                                     const int32_t delay);
+  void OnIncomingCapturedFrame(const int32_t id,
+                               const webrtc::VideoFrame& frame) override;
+  void OnCaptureDelayChanged(const int32_t id,
+                             const int32_t delay) override;
 
   // Used to signal captured frames on the same thread as invoked Start().
   // With WebRTC's current VideoCapturer implementations, this will mean a
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.cc b/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.cc
index b90950f0fbe..d81c849234e 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.cc
@@ -21,7 +21,6 @@
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
-#include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/engine/constants.h"
 #include "webrtc/media/engine/simulcast.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
@@ -30,6 +29,7 @@
 #include "webrtc/media/engine/webrtcvoiceengine.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/video_decoder.h"
 #include "webrtc/video_encoder.h"
@@ -160,7 +160,8 @@ bool CodecIsInternallySupported(const std::string& codec_name) {
     return true;
   }
   if (CodecNamesEq(codec_name, kVp9CodecName)) {
-    return true;
+    return webrtc::VP9Encoder::IsSupported() &&
+           webrtc::VP9Decoder::IsSupported();
   }
   if (CodecNamesEq(codec_name, kH264CodecName)) {
     return webrtc::H264Encoder::IsSupported() &&
@@ -181,7 +182,7 @@ void AddDefaultFeedbackParams(VideoCodec* codec) {
 static VideoCodec MakeVideoCodecWithDefaultFeedbackParams(int payload_type,
                                                           const char* name) {
   VideoCodec codec(payload_type, name, kDefaultVideoMaxWidth,
-                   kDefaultVideoMaxHeight, kDefaultVideoMaxFramerate, 0);
+                   kDefaultVideoMaxHeight, kDefaultVideoMaxFramerate);
   AddDefaultFeedbackParams(&codec);
   return codec;
 }
@@ -262,39 +263,6 @@ inline bool ContainsHeaderExtension(
   return false;
 }
 
-// Merges two fec configs and logs an error if a conflict arises
-// such that merging in different order would trigger a different output.
-static void MergeFecConfig(const webrtc::FecConfig& other,
-                           webrtc::FecConfig* output) {
-  if (other.ulpfec_payload_type != -1) {
-    if (output->ulpfec_payload_type != -1 &&
-        output->ulpfec_payload_type != other.ulpfec_payload_type) {
-      LOG(LS_WARNING) << "Conflict merging ulpfec_payload_type configs: "
-                      << output->ulpfec_payload_type << " and "
-                      << other.ulpfec_payload_type;
-    }
-    output->ulpfec_payload_type = other.ulpfec_payload_type;
-  }
-  if (other.red_payload_type != -1) {
-    if (output->red_payload_type != -1 &&
-        output->red_payload_type != other.red_payload_type) {
-      LOG(LS_WARNING) << "Conflict merging red_payload_type configs: "
-                      << output->red_payload_type << " and "
-                      << other.red_payload_type;
-    }
-    output->red_payload_type = other.red_payload_type;
-  }
-  if (other.red_rtx_payload_type != -1) {
-    if (output->red_rtx_payload_type != -1 &&
-        output->red_rtx_payload_type != other.red_rtx_payload_type) {
-      LOG(LS_WARNING) << "Conflict merging red_rtx_payload_type configs: "
-                      << output->red_rtx_payload_type << " and "
-                      << other.red_rtx_payload_type;
-    }
-    output->red_rtx_payload_type = other.red_rtx_payload_type;
-  }
-}
-
 // Returns true if the given codec is disallowed from doing simulcast.
 bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) {
   return CodecNamesEq(codec_name, kH264CodecName) ||
@@ -631,12 +599,19 @@ std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
   std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList();
 
   if (external_encoder_factory_ == NULL) {
+    LOG(LS_INFO) << "Supported codecs: "
+                 << CodecVectorToString(supported_codecs);
     return supported_codecs;
   }
 
+  std::stringstream out;
   const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
       external_encoder_factory_->codecs();
   for (size_t i = 0; i < codecs.size(); ++i) {
+    out << codecs[i].name;
+    if (i != codecs.size() - 1) {
+      out << ", ";
+    }
     // Don't add internally-supported codecs twice.
     if (CodecIsInternallySupported(codecs[i].name)) {
       continue;
@@ -647,16 +622,17 @@ std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
     const int kExternalVideoPayloadTypeBase = 120;
     size_t payload_type = kExternalVideoPayloadTypeBase + i;
     RTC_DCHECK(payload_type < 128);
-    VideoCodec codec(static_cast<int>(payload_type),
-                     codecs[i].name,
-                     codecs[i].max_width,
-                     codecs[i].max_height,
-                     codecs[i].max_fps,
-                     0);
+    VideoCodec codec(static_cast<int>(payload_type), codecs[i].name,
+                     codecs[i].max_width, codecs[i].max_height,
+                     codecs[i].max_fps);
 
     AddDefaultFeedbackParams(&codec);
     supported_codecs.push_back(codec);
   }
+  LOG(LS_INFO) << "Supported codecs (incl. external codecs): "
+               << CodecVectorToString(supported_codecs);
+  LOG(LS_INFO) << "Codecs supported by the external encoder factory: "
+               << out.str();
   return supported_codecs;
 }
 
@@ -673,7 +649,8 @@ WebRtcVideoChannel2::WebRtcVideoChannel2(
       video_config_(config.video),
       external_encoder_factory_(external_encoder_factory),
       external_decoder_factory_(external_decoder_factory),
-      default_send_options_(options) {
+      default_send_options_(options),
+      red_disabled_by_remote_side_(false) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
   rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
@@ -740,17 +717,7 @@ bool WebRtcVideoChannel2::ReceiveCodecsHaveChanged(
       };
   std::sort(before.begin(), before.end(), comparison);
   std::sort(after.begin(), after.end(), comparison);
-  for (size_t i = 0; i < before.size(); ++i) {
-    // For the same reason that we sort the codecs, we also ignore the
-    // preference.  We don't want a preference change on the receive
-    // side to cause recreation of the stream.
-    before[i].codec.preference = 0;
-    after[i].codec.preference = 0;
-    if (before[i] != after[i]) {
-      return true;
-    }
-  }
-  return false;
+  return before != after;
 }
 
 bool WebRtcVideoChannel2::GetChangedSendParameters(
@@ -784,7 +751,7 @@ bool WebRtcVideoChannel2::GetChangedSendParameters(
   }
 
   // Handle max bitrate.
-  if (params.max_bandwidth_bps != bitrate_config_.max_bitrate_bps &&
+  if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps &&
       params.max_bandwidth_bps >= 0) {
     // 0 uncaps max bitrate (-1).
     changed_params->max_bandwidth_bps = rtc::Optional<int>(
@@ -819,39 +786,38 @@ bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
     return false;
   }
 
-  bool bitrate_config_changed = false;
-
   if (changed_params.codec) {
     const VideoCodecSettings& codec_settings = *changed_params.codec;
     send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings);
-
     LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString();
-    // TODO(holmer): Changing the codec parameters shouldn't necessarily mean
-    // that we change the min/max of bandwidth estimation. Reevaluate this.
-    bitrate_config_ = GetBitrateConfigForCodec(codec_settings.codec);
-    bitrate_config_changed = true;
   }
 
   if (changed_params.rtp_header_extensions) {
     send_rtp_extensions_ = *changed_params.rtp_header_extensions;
   }
 
-  if (changed_params.max_bandwidth_bps) {
-    // TODO(pbos): Figure out whether b=AS means max bitrate for this
-    // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC), in
-    // which case this should not set a Call::BitrateConfig but rather
-    // reconfigure all senders.
-    int max_bitrate_bps = *changed_params.max_bandwidth_bps;
-    bitrate_config_.start_bitrate_bps = -1;
-    bitrate_config_.max_bitrate_bps = max_bitrate_bps;
-    if (max_bitrate_bps > 0 &&
-        bitrate_config_.min_bitrate_bps > max_bitrate_bps) {
-      bitrate_config_.min_bitrate_bps = max_bitrate_bps;
+  if (changed_params.codec || changed_params.max_bandwidth_bps) {
+    if (send_codec_) {
+      // TODO(holmer): Changing the codec parameters shouldn't necessarily mean
+      // that we change the min/max of bandwidth estimation. Reevaluate this.
+      bitrate_config_ = GetBitrateConfigForCodec(send_codec_->codec);
+      if (!changed_params.codec) {
+        // If the codec isn't changing, set the start bitrate to -1 which means
+        // "unchanged" so that BWE isn't affected.
+        bitrate_config_.start_bitrate_bps = -1;
+      }
+    }
+    if (params.max_bandwidth_bps >= 0) {
+      // Note that max_bandwidth_bps intentionally takes priority over the
+      // bitrate config for the codec. This allows FEC to be applied above the
+      // codec target bitrate.
+      // TODO(pbos): Figure out whether b=AS means max bitrate for this
+      // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC),
+      // in which case this should not set a Call::BitrateConfig but rather
+      // reconfigure all senders.
+      bitrate_config_.max_bitrate_bps =
+          params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
     }
-    bitrate_config_changed = true;
-  }
-
-  if (bitrate_config_changed) {
     call_->SetBitrateConfig(bitrate_config_);
   }
 
@@ -874,37 +840,106 @@ bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
                                      : webrtc::RtcpMode::kCompound);
       }
     }
+    if (changed_params.codec) {
+      bool red_was_disabled = red_disabled_by_remote_side_;
+      red_disabled_by_remote_side_ =
+          changed_params.codec->fec.red_payload_type == -1;
+      if (red_was_disabled != red_disabled_by_remote_side_) {
+        for (auto& kv : receive_streams_) {
+          // In practice VideoChannel::SetRemoteContent appears to most of the
+          // time also call UpdateRemoteStreams, which recreates the receive
+          // streams. If that's always true this call isn't needed.
+          kv.second->SetFecDisabledRemotely(red_disabled_by_remote_side_);
+        }
+      }
+    }
   }
   send_params_ = params;
   return true;
 }
-webrtc::RtpParameters WebRtcVideoChannel2::GetRtpParameters(
+
+webrtc::RtpParameters WebRtcVideoChannel2::GetRtpSendParameters(
     uint32_t ssrc) const {
   rtc::CritScope stream_lock(&stream_crit_);
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
-    LOG(LS_WARNING) << "Attempting to get RTP parameters for stream with ssrc "
-                    << ssrc << " which doesn't exist.";
+    LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
+                    << "with ssrc " << ssrc << " which doesn't exist.";
     return webrtc::RtpParameters();
   }
 
-  return it->second->GetRtpParameters();
+  webrtc::RtpParameters rtp_params = it->second->GetRtpParameters();
+  // Need to add the common list of codecs to the send stream-specific
+  // RTP parameters.
+  for (const VideoCodec& codec : send_params_.codecs) {
+    rtp_params.codecs.push_back(codec.ToCodecParameters());
+  }
+  return rtp_params;
 }
 
-bool WebRtcVideoChannel2::SetRtpParameters(
+bool WebRtcVideoChannel2::SetRtpSendParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpSendParameters");
   rtc::CritScope stream_lock(&stream_crit_);
   auto it = send_streams_.find(ssrc);
   if (it == send_streams_.end()) {
-    LOG(LS_ERROR) << "Attempting to set RTP parameters for stream with ssrc "
-                  << ssrc << " which doesn't exist.";
+    LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream "
+                  << "with ssrc " << ssrc << " which doesn't exist.";
+    return false;
+  }
+
+  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
+  // different order (which should change the send codec).
+  webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
+  if (current_parameters.codecs != parameters.codecs) {
+    LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
+                  << "is not currently supported.";
     return false;
   }
 
   return it->second->SetRtpParameters(parameters);
 }
 
+webrtc::RtpParameters WebRtcVideoChannel2::GetRtpReceiveParameters(
+    uint32_t ssrc) const {
+  rtc::CritScope stream_lock(&stream_crit_);
+  auto it = receive_streams_.find(ssrc);
+  if (it == receive_streams_.end()) {
+    LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
+                    << "with ssrc " << ssrc << " which doesn't exist.";
+    return webrtc::RtpParameters();
+  }
+
+  // TODO(deadbeef): Return stream-specific parameters.
+  webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
+  for (const VideoCodec& codec : recv_params_.codecs) {
+    rtp_params.codecs.push_back(codec.ToCodecParameters());
+  }
+  return rtp_params;
+}
+
+bool WebRtcVideoChannel2::SetRtpReceiveParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpReceiveParameters");
+  rtc::CritScope stream_lock(&stream_crit_);
+  auto it = receive_streams_.find(ssrc);
+  if (it == receive_streams_.end()) {
+    LOG(LS_ERROR) << "Attempting to set RTP receive parameters for stream "
+                  << "with ssrc " << ssrc << " which doesn't exist.";
+    return false;
+  }
+
+  webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
+  if (current_parameters != parameters) {
+    LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
+                  << "unsupported.";
+    return false;
+  }
+  return true;
+}
+
 bool WebRtcVideoChannel2::GetChangedRecvParameters(
     const VideoRecvParameters& params,
     ChangedRecvParameters* changed_params) const {
@@ -996,6 +1031,7 @@ bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
 }
 
 bool WebRtcVideoChannel2::SetSend(bool send) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSend");
   LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
   if (send && !send_codec_) {
     LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
@@ -1011,6 +1047,9 @@ bool WebRtcVideoChannel2::SetSend(bool send) {
   return true;
 }
 
+// TODO(nisse): The enable argument was used for mute logic which has
+// been moved to VideoBroadcaster. So delete this method, and use
+// SetOptions instead.
 bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable,
                                        const VideoOptions* options) {
   TRACE_EVENT0("webrtc", "SetVideoSend");
@@ -1018,11 +1057,6 @@ bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable,
                << "options: " << (options ? options->ToString() : "nullptr")
                << ").";
 
-  // TODO(solenberg): The state change should be fully rolled back if any one of
-  //                  these calls fail.
-  if (!MuteStream(ssrc, !enable)) {
-    return false;
-  }
   if (enable && options) {
     SetOptions(ssrc, *options);
   }
@@ -1184,7 +1218,7 @@ bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp,
 
   receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
       call_, sp, config, external_decoder_factory_, default_stream,
-      recv_codecs_);
+      recv_codecs_, red_disabled_by_remote_side_);
 
   return true;
 }
@@ -1220,10 +1254,6 @@ void WebRtcVideoChannel2::ConfigureReceiverRtp(
   }
 
   for (size_t i = 0; i < recv_codecs_.size(); ++i) {
-    MergeFecConfig(recv_codecs_[i].fec, &config->rtp.fec);
-  }
-
-  for (size_t i = 0; i < recv_codecs_.size(); ++i) {
     uint32_t rtx_ssrc;
     if (recv_codecs_[i].rtx_payload_type != -1 &&
         sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
@@ -1325,22 +1355,21 @@ void WebRtcVideoChannel2::FillBandwidthEstimationStats(
   video_media_info->bw_estimations.push_back(bwe_info);
 }
 
-bool WebRtcVideoChannel2::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
-  LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
-               << (capturer != NULL ? "(capturer)" : "NULL");
+void WebRtcVideoChannel2::SetSource(
+    uint32_t ssrc,
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
+  LOG(LS_INFO) << "SetSource: " << ssrc << " -> "
+               << (source ? "(source)" : "NULL");
   RTC_DCHECK(ssrc != 0);
-  {
-    rtc::CritScope stream_lock(&stream_crit_);
-    const auto& kv = send_streams_.find(ssrc);
-    if (kv == send_streams_.end()) {
-      LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
-      return false;
-    }
-    if (!kv->second->SetCapturer(capturer)) {
-      return false;
-    }
+
+  rtc::CritScope stream_lock(&stream_crit_);
+  const auto& kv = send_streams_.find(ssrc);
+  if (kv == send_streams_.end()) {
+    // Allow unknown ssrc only if source is null.
+    RTC_CHECK(source == nullptr);
+  } else {
+    kv->second->SetSource(source);
   }
-  return true;
 }
 
 void WebRtcVideoChannel2::OnPacketReceived(
@@ -1425,24 +1454,8 @@ void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
 
 void WebRtcVideoChannel2::OnNetworkRouteChanged(
     const std::string& transport_name,
-    const NetworkRoute& network_route) {
-  // TODO(honghaiz): uncomment this once the function in call is implemented.
-  // call_->OnNetworkRouteChanged(transport_name, network_route);
-}
-
-bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
-  LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
-                  << (mute ? "mute" : "unmute");
-  RTC_DCHECK(ssrc != 0);
-  rtc::CritScope stream_lock(&stream_crit_);
-  const auto& kv = send_streams_.find(ssrc);
-  if (kv == send_streams_.end()) {
-    LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
-    return false;
-  }
-
-  kv->second->MuteStream(mute);
-  return true;
+    const rtc::NetworkRoute& network_route) {
+  call_->OnNetworkRouteChanged(transport_name, network_route);
 }
 
 // TODO(pbos): Remove SetOptions in favor of SetSendParameters.
@@ -1533,7 +1546,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
       call_(call),
       cpu_restricted_counter_(0),
       number_of_cpu_adapt_changes_(0),
-      capturer_(nullptr),
+      source_(nullptr),
       external_encoder_factory_(external_encoder_factory),
       stream_(nullptr),
       parameters_(config, options, max_bitrate_bps, codec_settings),
@@ -1541,8 +1554,6 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
       pending_encoder_reconfiguration_(false),
       allocated_encoder_(nullptr, webrtc::kVideoCodecUnknown, false),
       sending_(false),
-      muted_(false),
-      first_frame_timestamp_ms_(0),
       last_frame_timestamp_ms_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
   parameters_.conference_mode = send_params.conference_mode;
@@ -1567,54 +1578,52 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
 }
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
-  DisconnectCapturer();
+  DisconnectSource();
   if (stream_ != NULL) {
     call_->DestroyVideoSendStream(stream_);
   }
   DestroyVideoEncoder(&allocated_encoder_);
 }
 
-static void CreateBlackFrame(webrtc::VideoFrame* video_frame,
-                             int width,
-                             int height,
-                             webrtc::VideoRotation rotation) {
-  video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2,
-                                (width + 1) / 2);
-  memset(video_frame->buffer(webrtc::kYPlane), 16,
-         video_frame->allocated_size(webrtc::kYPlane));
-  memset(video_frame->buffer(webrtc::kUPlane), 128,
-         video_frame->allocated_size(webrtc::kUPlane));
-  memset(video_frame->buffer(webrtc::kVPlane), 128,
-         video_frame->allocated_size(webrtc::kVPlane));
-  video_frame->set_rotation(rotation);
+static webrtc::VideoFrame CreateBlackFrame(int width,
+                                           int height,
+                                           int64_t render_time_ms_,
+                                           webrtc::VideoRotation rotation) {
+  webrtc::VideoFrame frame;
+  frame.CreateEmptyFrame(width, height, width, (width + 1) / 2,
+                         (width + 1) / 2);
+  memset(frame.video_frame_buffer()->MutableDataY(), 16,
+         frame.allocated_size(webrtc::kYPlane));
+  memset(frame.video_frame_buffer()->MutableDataU(), 128,
+         frame.allocated_size(webrtc::kUPlane));
+  memset(frame.video_frame_buffer()->MutableDataV(), 128,
+         frame.allocated_size(webrtc::kVPlane));
+  frame.set_rotation(rotation);
+  frame.set_render_time_ms(render_time_ms_);
+  return frame;
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
     const VideoFrame& frame) {
   TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::OnFrame");
-  webrtc::VideoFrame video_frame(frame.GetVideoFrameBuffer(), 0, 0,
-                                 frame.GetVideoRotation());
+  webrtc::VideoFrame video_frame(frame.video_frame_buffer(), 0, 0,
+                                 frame.rotation());
   rtc::CritScope cs(&lock_);
   if (stream_ == NULL) {
     // Frame input before send codecs are configured, dropping frame.
     return;
   }
 
-  if (muted_) {
-    // Create a black frame to transmit instead.
-    CreateBlackFrame(&video_frame,
-                     frame.width(),
-                     frame.height(),
-                     video_frame.rotation());
-  }
-
   int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
+
   // frame->GetTimeStamp() is essentially a delta, align to webrtc time
-  if (first_frame_timestamp_ms_ == 0) {
-    first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
+  if (!first_frame_timestamp_ms_) {
+    first_frame_timestamp_ms_ =
+        rtc::Optional<int64_t>(rtc::TimeMillis() - frame_delta_ms);
   }
 
-  last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms;
+  last_frame_timestamp_ms_ = *first_frame_timestamp_ms_ + frame_delta_ms;
+
   video_frame.set_render_time_ms(last_frame_timestamp_ms_);
   // Reconfigure codec if necessary.
   SetDimensions(video_frame.width(), video_frame.height());
@@ -1630,71 +1639,62 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
   stream_->Input()->IncomingCapturedFrame(video_frame);
 }
 
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
-    VideoCapturer* capturer) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
+void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSource(
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetSource");
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!DisconnectCapturer() && capturer == NULL) {
-    return false;
-  }
+
+  if (!source && !source_)
+    return;
+  DisconnectSource();
 
   {
     rtc::CritScope cs(&lock_);
 
     // Reset timestamps to realign new incoming frames to a webrtc timestamp. A
     // new capturer may have a different timestamp delta than the previous one.
-    first_frame_timestamp_ms_ = 0;
+    first_frame_timestamp_ms_ = rtc::Optional<int64_t>();
 
-    if (capturer == NULL) {
+    if (source == NULL) {
       if (stream_ != NULL) {
         LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
-        webrtc::VideoFrame black_frame;
-
-        CreateBlackFrame(&black_frame, last_dimensions_.width,
-                         last_dimensions_.height, last_rotation_);
-
         // Force this black frame not to be dropped due to timestamp order
         // check. As IncomingCapturedFrame will drop the frame if this frame's
         // timestamp is less than or equal to last frame's timestamp, it is
         // necessary to give this black frame a larger timestamp than the
         // previous one.
         last_frame_timestamp_ms_ += 1;
-        black_frame.set_render_time_ms(last_frame_timestamp_ms_);
-        stream_->Input()->IncomingCapturedFrame(black_frame);
-      }
+        stream_->Input()->IncomingCapturedFrame(
+            CreateBlackFrame(last_dimensions_.width, last_dimensions_.height,
+                             last_frame_timestamp_ms_, last_rotation_));
 
-      capturer_ = NULL;
-      return true;
+
+      }
     }
   }
-  capturer_ = capturer;
-  // |capturer_->AddOrUpdateSink| may not be called while holding |lock_| since
+  source_ = source;
+  // |source_->AddOrUpdateSink| may not be called while holding |lock_| since
   // that might cause a lock order inversion.
-  capturer_->AddOrUpdateSink(this, sink_wants_);
-  return true;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) {
-  rtc::CritScope cs(&lock_);
-  muted_ = mute;
+  if (source_) {
+    source_->AddOrUpdateSink(this, sink_wants_);
+  }
 }
 
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
+void WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectSource() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (capturer_ == NULL) {
-    return false;
+  if (source_ == NULL) {
+    return;
   }
 
-  // |capturer_->RemoveSink| may not be called while holding |lock_| since
+  // |source_->RemoveSink| may not be called while holding |lock_| since
   // that might cause a lock order inversion.
-  capturer_->RemoveSink(this);
-  capturer_ = NULL;
+  source_->RemoveSink(this);
+  source_ = nullptr;
   // Reset |cpu_restricted_counter_| if the capturer is changed. It is not
   // possible to know if the video resolution is restricted by CPU usage after
   // the capturer is changed since the next capturer might be screen capture
   // with another resolution and frame rate.
   cpu_restricted_counter_ = 0;
-  return true;
 }
 
 const std::vector<uint32_t>&
@@ -1855,8 +1855,8 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
   if (params.rtp_header_extensions) {
     sink_wants_.rotation_applied = !ContainsHeaderExtension(
         *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension);
-    if (capturer_) {
-      capturer_->AddOrUpdateSink(this, sink_wants_);
+    if (source_) {
+      source_->AddOrUpdateSink(this, sink_wants_);
     }
   }
 }
@@ -1873,6 +1873,8 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpParameters(
     pending_encoder_reconfiguration_ = true;
   }
   rtp_parameters_ = new_parameters;
+  // Codecs are currently handled at the WebRtcVideoChannel2 level.
+  rtp_parameters_.codecs.clear();
   // Encoding may have been activated/deactivated.
   UpdateSendState();
   return true;
@@ -2021,7 +2023,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) {
     return;
   }
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!capturer_) {
+  if (!source_) {
     return;
   }
   {
@@ -2077,9 +2079,9 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) {
     sink_wants_.max_pixel_count = max_pixel_count;
     sink_wants_.max_pixel_count_step_up = max_pixel_count_step_up;
   }
-  // |capturer_->AddOrUpdateSink| may not be called while holding |lock_| since
+  // |source_->AddOrUpdateSink| may not be called while holding |lock_| since
   // that might cause a lock order inversion.
-  capturer_->AddOrUpdateSink(this, sink_wants_);
+  source_->AddOrUpdateSink(this, sink_wants_);
 }
 
 VideoSenderInfo
@@ -2214,13 +2216,15 @@ WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
     const webrtc::VideoReceiveStream::Config& config,
     WebRtcVideoDecoderFactory* external_decoder_factory,
     bool default_stream,
-    const std::vector<VideoCodecSettings>& recv_codecs)
+    const std::vector<VideoCodecSettings>& recv_codecs,
+    bool red_disabled_by_remote_side)
     : call_(call),
       ssrcs_(sp.ssrcs),
       ssrc_groups_(sp.ssrc_groups),
       stream_(NULL),
       default_stream_(default_stream),
       config_(config),
+      red_disabled_by_remote_side_(red_disabled_by_remote_side),
       external_decoder_factory_(external_decoder_factory),
       sink_(NULL),
       last_width_(-1),
@@ -2396,7 +2400,13 @@ void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
   if (stream_ != NULL) {
     call_->DestroyVideoReceiveStream(stream_);
   }
-  stream_ = call_->CreateVideoReceiveStream(config_);
+  webrtc::VideoReceiveStream::Config config = config_;
+  if (red_disabled_by_remote_side_) {
+    config.rtp.fec.red_payload_type = -1;
+    config.rtp.fec.ulpfec_payload_type = -1;
+    config.rtp.fec.red_rtx_payload_type = -1;
+  }
+  stream_ = call_->CreateVideoReceiveStream(config);
   stream_->Start();
 }
 
@@ -2435,8 +2445,8 @@ void WebRtcVideoChannel2::WebRtcVideoReceiveStream::OnFrame(
   last_height_ = frame.height();
 
   const WebRtcVideoFrame render_frame(
-      frame.video_frame_buffer(),
-      frame.render_time_ms() * rtc::kNumNanosecsPerMillisec, frame.rotation());
+      frame.video_frame_buffer(), frame.rotation(),
+      frame.render_time_ms() * rtc::kNumNanosecsPerMicrosec);
   sink_->OnFrame(render_frame);
 }
 
@@ -2504,6 +2514,12 @@ WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo() {
   return info;
 }
 
+void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFecDisabledRemotely(
+    bool disable) {
+  red_disabled_by_remote_side_ = disable;
+  RecreateWebRtcStream();
+}
+
 WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
     : rtx_payload_type(-1) {}
 
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.h b/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.h
index ba2dbd8a02b..2d1b1923ed7 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.h
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideoengine2.h
@@ -74,6 +74,7 @@ class UnsignalledSsrcHandler {
   };
   virtual Action OnUnsignalledSsrc(WebRtcVideoChannel2* channel,
                                    uint32_t ssrc) = 0;
+  virtual ~UnsignalledSsrcHandler() = default;
 };
 
 // TODO(pbos): Remove, use external handlers only.
@@ -86,6 +87,7 @@ class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
   rtc::VideoSinkInterface<VideoFrame>* GetDefaultSink() const;
   void SetDefaultSink(VideoMediaChannel* channel,
                       rtc::VideoSinkInterface<VideoFrame>* sink);
+  virtual ~DefaultUnsignalledSsrcHandler() = default;
 
  private:
   uint32_t default_recv_ssrc_;
@@ -96,7 +98,7 @@ class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
 class WebRtcVideoEngine2 {
  public:
   WebRtcVideoEngine2();
-  ~WebRtcVideoEngine2();
+  virtual ~WebRtcVideoEngine2();
 
   // Basic video engine implementation.
   void Init();
@@ -145,9 +147,13 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
 
   bool SetSendParameters(const VideoSendParameters& params) override;
   bool SetRecvParameters(const VideoRecvParameters& params) override;
-  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const override;
-  bool SetRtpParameters(uint32_t ssrc,
-                        const webrtc::RtpParameters& parameters) override;
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
+  bool SetRtpSendParameters(uint32_t ssrc,
+                            const webrtc::RtpParameters& parameters) override;
+  webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
+  bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) override;
   bool GetSendCodec(VideoCodec* send_codec) override;
   bool SetSend(bool send) override;
   bool SetVideoSend(uint32_t ssrc,
@@ -161,7 +167,9 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
   bool SetSink(uint32_t ssrc,
                rtc::VideoSinkInterface<VideoFrame>* sink) override;
   bool GetStats(VideoMediaInfo* info) override;
-  bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) override;
+  void SetSource(
+      uint32_t ssrc,
+      rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
 
   void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                         const rtc::PacketTime& packet_time) override;
@@ -169,7 +177,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override;
   void OnNetworkRouteChanged(const std::string& transport_name,
-                             const NetworkRoute& network_route) override;
+                             const rtc::NetworkRoute& network_route) override;
   void SetInterface(NetworkInterface* iface) override;
 
   // Implemented for VideoMediaChannelTest.
@@ -217,8 +225,6 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
   bool GetChangedRecvParameters(const VideoRecvParameters& params,
                                 ChangedRecvParameters* changed_params) const;
 
-  bool MuteStream(uint32_t ssrc, bool mute);
-
   void SetMaxSendBandwidth(int bps);
   void SetOptions(uint32_t ssrc, const VideoOptions& options);
 
@@ -235,7 +241,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
   static std::string CodecSettingsVectorToString(
       const std::vector<VideoCodecSettings>& codecs);
 
-  // Wrapper for the sender part, this is where the capturer is connected and
+  // Wrapper for the sender part, this is where the source is connected and
   // frames are then converted from cricket frames to webrtc frames.
   class WebRtcVideoSendStream
       : public rtc::VideoSinkInterface<cricket::VideoFrame>,
@@ -261,9 +267,8 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
     webrtc::RtpParameters GetRtpParameters() const;
 
     void OnFrame(const cricket::VideoFrame& frame) override;
-    bool SetCapturer(VideoCapturer* capturer);
-    void MuteStream(bool mute);
-    bool DisconnectCapturer();
+    void SetSource(rtc::VideoSourceInterface<cricket::VideoFrame>* source);
+    void DisconnectSource();
 
     void SetSend(bool send);
 
@@ -365,12 +370,12 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
     webrtc::Call* const call_;
     rtc::VideoSinkWants sink_wants_;
     // Counter used for deciding if the video resolution is currently
-    // restricted by CPU usage. It is reset if |capturer_| is changed.
+    // restricted by CPU usage. It is reset if |source_| is changed.
     int cpu_restricted_counter_;
     // Total number of times resolution as been requested to be changed due to
     // CPU adaptation.
     int number_of_cpu_adapt_changes_;
-    VideoCapturer* capturer_;
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
     WebRtcVideoEncoderFactory* const external_encoder_factory_
         GUARDED_BY(lock_);
 
@@ -381,6 +386,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
     // entire channel.
     VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
     // Contains settings that are unique for each stream, such as max_bitrate.
+    // Does *not* contain codecs, however.
     // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
     // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
     // one stream per MediaChannel.
@@ -393,14 +399,13 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
         webrtc::kVideoRotation_0;
 
     bool sending_ GUARDED_BY(lock_);
-    bool muted_ GUARDED_BY(lock_);
 
     // The timestamp of the first frame received
     // Used to generate the timestamps of subsequent frames
-    int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_);
+    rtc::Optional<int64_t> first_frame_timestamp_ms_ GUARDED_BY(lock_);
 
     // The timestamp of the last frame received
-    // Used to generate timestamp for the black frame when capturer is removed
+    // Used to generate timestamp for the black frame when source is removed
     int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
   };
 
@@ -417,7 +422,8 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
         const webrtc::VideoReceiveStream::Config& config,
         WebRtcVideoDecoderFactory* external_decoder_factory,
         bool default_stream,
-        const std::vector<VideoCodecSettings>& recv_codecs);
+        const std::vector<VideoCodecSettings>& recv_codecs,
+        bool red_disabled_by_remote_side);
     ~WebRtcVideoReceiveStream();
 
     const std::vector<uint32_t>& GetSsrcs() const;
@@ -437,6 +443,14 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
 
     VideoReceiverInfo GetVideoReceiverInfo();
 
+    // Used to disable RED/FEC when the remote description doesn't contain those
+    // codecs. This is needed to be able to work around an RTX bug which is only
+    // happening if the remote side doesn't send RED, but the local side is
+    // configured to receive RED.
+    // TODO(holmer): Remove this after a couple of Chrome versions, M53-54
+    // time frame.
+    void SetFecDisabledRemotely(bool disable);
+
    private:
     struct AllocatedDecoder {
       AllocatedDecoder(webrtc::VideoDecoder* decoder,
@@ -467,6 +481,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
     webrtc::VideoReceiveStream* stream_;
     const bool default_stream_;
     webrtc::VideoReceiveStream::Config config_;
+    bool red_disabled_by_remote_side_;
 
     WebRtcVideoDecoderFactory* const external_decoder_factory_;
     std::vector<AllocatedDecoder> allocated_decoders_;
@@ -537,6 +552,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
   VideoSendParameters send_params_;
   VideoOptions default_send_options_;
   VideoRecvParameters recv_params_;
+  bool red_disabled_by_remote_side_;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideoengine2_unittest.cc b/chromium/third_party/webrtc/media/engine/webrtcvideoengine2_unittest.cc
index e1f540660f2..fd7a6463b8b 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideoengine2_unittest.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideoengine2_unittest.cc
@@ -31,16 +31,16 @@ namespace {
 static const int kDefaultQpMax = 56;
 static const int kDefaultFramerate = 30;
 
-static const cricket::VideoCodec kVp8Codec720p(100, "VP8", 1280, 720, 30, 0);
-static const cricket::VideoCodec kVp8Codec360p(100, "VP8", 640, 360, 30, 0);
-static const cricket::VideoCodec kVp8Codec270p(100, "VP8", 480, 270, 30, 0);
+static const cricket::VideoCodec kVp8Codec720p(100, "VP8", 1280, 720, 30);
+static const cricket::VideoCodec kVp8Codec360p(100, "VP8", 640, 360, 30);
+static const cricket::VideoCodec kVp8Codec270p(100, "VP8", 480, 270, 30);
 
-static const cricket::VideoCodec kVp8Codec(100, "VP8", 640, 400, 30, 0);
-static const cricket::VideoCodec kVp9Codec(101, "VP9", 640, 400, 30, 0);
-static const cricket::VideoCodec kH264Codec(102, "H264", 640, 400, 30, 0);
+static const cricket::VideoCodec kVp8Codec(100, "VP8", 640, 400, 30);
+static const cricket::VideoCodec kVp9Codec(101, "VP9", 640, 400, 30);
+static const cricket::VideoCodec kH264Codec(102, "H264", 640, 400, 30);
 
-static const cricket::VideoCodec kRedCodec(116, "red", 0, 0, 0, 0);
-static const cricket::VideoCodec kUlpfecCodec(117, "ulpfec", 0, 0, 0, 0);
+static const cricket::VideoCodec kRedCodec(116, "red", 0, 0, 0);
+static const cricket::VideoCodec kUlpfecCodec(117, "ulpfec", 0, 0, 0);
 
 static const uint8_t kRedRtxPayloadType = 125;
 
@@ -69,11 +69,11 @@ static void CreateBlackFrame(webrtc::VideoFrame* video_frame,
                              int height) {
   video_frame->CreateEmptyFrame(
       width, height, width, (width + 1) / 2, (width + 1) / 2);
-  memset(video_frame->buffer(webrtc::kYPlane), 16,
+  memset(video_frame->video_frame_buffer()->MutableDataY(), 16,
          video_frame->allocated_size(webrtc::kYPlane));
-  memset(video_frame->buffer(webrtc::kUPlane), 128,
+  memset(video_frame->video_frame_buffer()->MutableDataU(), 128,
          video_frame->allocated_size(webrtc::kUPlane));
-  memset(video_frame->buffer(webrtc::kVPlane), 128,
+  memset(video_frame->video_frame_buffer()->MutableDataV(), 128,
          video_frame->allocated_size(webrtc::kVPlane));
 }
 
@@ -150,6 +150,21 @@ class WebRtcVideoEngine2Test : public ::testing::Test {
   std::map<int, int> default_apt_rtx_types_;
 };
 
+TEST_F(WebRtcVideoEngine2Test, AnnouncesVp9AccordingToBuildFlags) {
+  bool claims_vp9_support = false;
+  for (const cricket::VideoCodec& codec : engine_.codecs()) {
+    if (codec.name == "VP9") {
+      claims_vp9_support = true;
+      break;
+    }
+  }
+#if defined(RTC_DISABLE_VP9)
+  EXPECT_FALSE(claims_vp9_support);
+#else
+  EXPECT_TRUE(claims_vp9_support);
+#endif  // defined(RTC_DISABLE_VP9)
+}
+
 TEST_F(WebRtcVideoEngine2Test, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
   std::vector<VideoCodec> engine_codecs = engine_.codecs();
   for (size_t i = 0; i < engine_codecs.size(); ++i) {
@@ -241,7 +256,7 @@ TEST_F(WebRtcVideoEngine2Test, CVOSetHeaderExtensionBeforeCapturer) {
   EXPECT_TRUE(channel->SetSendParameters(parameters));
 
   // Set capturer.
-  EXPECT_TRUE(channel->SetCapturer(kSsrc, &capturer));
+  channel->SetSource(kSsrc, &capturer);
 
   // Verify capturer has turned off applying rotation.
   EXPECT_FALSE(capturer.GetApplyRotation());
@@ -272,7 +287,7 @@ TEST_F(WebRtcVideoEngine2Test, CVOSetHeaderExtensionBeforeAddSendStream) {
   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
 
   // Set capturer.
-  EXPECT_TRUE(channel->SetCapturer(kSsrc, &capturer));
+  channel->SetSource(kSsrc, &capturer);
 
   // Verify capturer has turned off applying rotation.
   EXPECT_FALSE(capturer.GetApplyRotation());
@@ -293,7 +308,7 @@ TEST_F(WebRtcVideoEngine2Test, CVOSetHeaderExtensionAfterCapturer) {
   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
 
   // Set capturer.
-  EXPECT_TRUE(channel->SetCapturer(kSsrc, &capturer));
+  channel->SetSource(kSsrc, &capturer);
 
   // Verify capturer has turned on applying rotation.
   EXPECT_TRUE(capturer.GetApplyRotation());
@@ -348,19 +363,22 @@ TEST_F(WebRtcVideoEngine2Test, UseExternalFactoryForVp8WhenSupported) {
 
   EXPECT_TRUE(
       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
+  ASSERT_TRUE(encoder_factory.WaitForCreatedVideoEncoders(1));
   ASSERT_EQ(1u, encoder_factory.encoders().size());
   EXPECT_TRUE(channel->SetSend(true));
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel->SetCapturer(kSsrc, &capturer));
+  channel->SetSource(kSsrc, &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(capturer.CaptureFrame());
+  // Sending one frame will have reallocated the encoder since input size
+  // changes from a small default to the actual frame width/height. Wait for
+  // that to happen then for the frame to be sent.
+  ASSERT_TRUE(encoder_factory.WaitForCreatedVideoEncoders(2));
   EXPECT_TRUE_WAIT(encoder_factory.encoders()[0]->GetNumEncodedFrames() > 0,
                    kTimeout);
 
-  // Sending one frame will have reallocated the encoder since input size
-  // changes from a small default to the actual frame width/height.
   int num_created_encoders = encoder_factory.GetNumCreatedEncoders();
   EXPECT_EQ(num_created_encoders, 2);
 
@@ -411,6 +429,7 @@ TEST_F(WebRtcVideoEngine2Test, DisablesFullEncoderTimeForNonExternalEncoders) {
   TestExtendedEncoderOveruse(false);
 }
 
+#if !defined(RTC_DISABLE_VP9)
 TEST_F(WebRtcVideoEngine2Test, CanConstructDecoderForVp9EncoderFactory) {
   cricket::FakeWebRtcVideoEncoderFactory encoder_factory;
   encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecVP9, "VP9");
@@ -423,6 +442,7 @@ TEST_F(WebRtcVideoEngine2Test, CanConstructDecoderForVp9EncoderFactory) {
   EXPECT_TRUE(
       channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
 }
+#endif  // !defined(RTC_DISABLE_VP9)
 
 TEST_F(WebRtcVideoEngine2Test, PropagatesInputFrameTimestamp) {
   cricket::FakeWebRtcVideoEncoderFactory encoder_factory;
@@ -439,7 +459,7 @@ TEST_F(WebRtcVideoEngine2Test, PropagatesInputFrameTimestamp) {
       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
 
   FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel->SetCapturer(kSsrc, &capturer));
+  channel->SetSource(kSsrc, &capturer);
   capturer.Start(cricket::VideoFormat(1280, 720,
                                       cricket::VideoFormat::FpsToInterval(60),
                                       cricket::FOURCC_I420));
@@ -501,7 +521,7 @@ TEST_F(WebRtcVideoEngine2Test,
   FakeVideoSendStream* stream = fake_call->GetVideoSendStreams()[0];
 
   FakeVideoCapturer capturer1;
-  EXPECT_TRUE(channel->SetCapturer(kSsrc, &capturer1));
+  channel->SetSource(kSsrc, &capturer1);
 
   cricket::CapturedFrame frame;
   frame.width = 1280;
@@ -512,8 +532,8 @@ TEST_F(WebRtcVideoEngine2Test,
   std::unique_ptr<char[]> data(new char[frame.data_size]);
   frame.data = data.get();
   memset(frame.data, 1, frame.data_size);
-  const int kInitialTimestamp = 123456;
-  frame.time_stamp = kInitialTimestamp;
+  int64_t initial_timestamp = rtc::TimeNanos();
+  frame.time_stamp = initial_timestamp;
 
   // Deliver initial frame.
   capturer1.SignalCapturedFrame(&frame);
@@ -526,12 +546,12 @@ TEST_F(WebRtcVideoEngine2Test,
   // Reset input source, should still be continuous even though input-frame
   // timestamp is less than before.
   FakeVideoCapturer capturer2;
-  EXPECT_TRUE(channel->SetCapturer(kSsrc, &capturer2));
+  channel->SetSource(kSsrc, &capturer2);
 
   rtc::Thread::Current()->SleepMs(1);
   // Deliver with a timestamp (10 seconds) before the previous initial one,
   // these should not be related at all anymore and it should still work fine.
-  frame.time_stamp = kInitialTimestamp - 10000;
+  frame.time_stamp = initial_timestamp - 10 * rtc::kNumNanosecsPerSec;
   capturer2.SignalCapturedFrame(&frame);
 
   // New timestamp should be at least 1ms in the future and not old.
@@ -586,17 +606,18 @@ TEST_F(WebRtcVideoEngine2Test, UsesSimulcastAdapterForVp8Factories) {
   EXPECT_TRUE(channel->SetSend(true));
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel->SetCapturer(ssrcs.front(), &capturer));
+  channel->SetSource(ssrcs.front(), &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(capturer.CaptureFrame());
 
-  EXPECT_GT(encoder_factory.encoders().size(), 1u);
+  ASSERT_TRUE(encoder_factory.WaitForCreatedVideoEncoders(2));
 
   // Verify that encoders are configured for simulcast through adapter
   // (increasing resolution and only configured to send one stream each).
   int prev_width = -1;
   for (size_t i = 0; i < encoder_factory.encoders().size(); ++i) {
+    ASSERT_TRUE(encoder_factory.encoders()[i]->WaitForInitEncode());
     webrtc::VideoCodec codec_settings =
         encoder_factory.encoders()[i]->GetCodecSettings();
     EXPECT_EQ(0, codec_settings.numberOfSimulcastStreams);
@@ -604,7 +625,7 @@ TEST_F(WebRtcVideoEngine2Test, UsesSimulcastAdapterForVp8Factories) {
     prev_width = codec_settings.width;
   }
 
-  EXPECT_TRUE(channel->SetCapturer(ssrcs.front(), NULL));
+  channel->SetSource(ssrcs.front(), NULL);
 
   channel.reset();
   ASSERT_EQ(0u, encoder_factory.encoders().size());
@@ -667,12 +688,13 @@ TEST_F(WebRtcVideoEngine2Test,
   // encoder adapter at a low-enough size that it'll only create a single
   // encoder layer.
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel->SetCapturer(ssrcs.front(), &capturer));
+  channel->SetSource(ssrcs.front(), &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(capturer.CaptureFrame());
 
-  ASSERT_GT(encoder_factory.encoders().size(), 1u);
+  ASSERT_TRUE(encoder_factory.WaitForCreatedVideoEncoders(2));
+  ASSERT_TRUE(encoder_factory.encoders()[0]->WaitForInitEncode());
   EXPECT_EQ(webrtc::kVideoCodecVP8,
             encoder_factory.encoders()[0]->GetCodecSettings().codecType);
 
@@ -696,6 +718,7 @@ TEST_F(WebRtcVideoEngine2Test,
   EXPECT_TRUE(
       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
   ASSERT_EQ(1u, encoder_factory.encoders().size());
+  ASSERT_TRUE(encoder_factory.encoders()[0]->WaitForInitEncode());
   EXPECT_EQ(webrtc::kVideoCodecH264,
             encoder_factory.encoders()[0]->GetCodecSettings().codecType);
 
@@ -721,15 +744,16 @@ TEST_F(WebRtcVideoEngine2Test, SimulcastDisabledForH264) {
   cricket::VideoFormat format(
       1280, 720, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420);
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel->SetCapturer(ssrcs[0], &capturer));
+  channel->SetSource(ssrcs[0], &capturer);
   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(format));
   EXPECT_TRUE(capturer.CaptureFrame());
 
   ASSERT_EQ(1u, encoder_factory.encoders().size());
   FakeWebRtcVideoEncoder* encoder = encoder_factory.encoders()[0];
+  ASSERT_TRUE(encoder_factory.encoders()[0]->WaitForInitEncode());
   EXPECT_EQ(webrtc::kVideoCodecH264, encoder->GetCodecSettings().codecType);
   EXPECT_EQ(1u, encoder->GetCodecSettings().numberOfSimulcastStreams);
-  EXPECT_TRUE(channel->SetCapturer(ssrcs[0], nullptr));
+  channel->SetSource(ssrcs[0], nullptr);
 }
 
 // Test that external codecs are added to the end of the supported codec list.
@@ -849,24 +873,29 @@ WEBRTC_DISABLED_BASE_TEST(AdaptFramerate);
 
 WEBRTC_BASE_TEST(SendsLowerResolutionOnSmallerFrames);
 
-WEBRTC_BASE_TEST(MuteStream);
-
 WEBRTC_BASE_TEST(MultipleSendStreams);
 
 TEST_F(WebRtcVideoChannel2BaseTest, SendAndReceiveVp8Vga) {
-  SendAndReceive(cricket::VideoCodec(100, "VP8", 640, 400, 30, 0));
+  SendAndReceive(cricket::VideoCodec(100, "VP8", 640, 400, 30));
 }
 
 TEST_F(WebRtcVideoChannel2BaseTest, SendAndReceiveVp8Qvga) {
-  SendAndReceive(cricket::VideoCodec(100, "VP8", 320, 200, 30, 0));
+  SendAndReceive(cricket::VideoCodec(100, "VP8", 320, 200, 30));
 }
 
 TEST_F(WebRtcVideoChannel2BaseTest, SendAndReceiveVp8SvcQqvga) {
-  SendAndReceive(cricket::VideoCodec(100, "VP8", 160, 100, 30, 0));
+  SendAndReceive(cricket::VideoCodec(100, "VP8", 160, 100, 30));
 }
 
 TEST_F(WebRtcVideoChannel2BaseTest, TwoStreamsSendAndReceive) {
-  Base::TwoStreamsSendAndReceive(kVp8Codec);
+  // Set a high bitrate to not be downscaled by VP8 due to low initial start
+  // bitrates. This currently happens at <250k, and two streams sharing 300k
+  // initially will use QVGA instead of VGA.
+  // TODO(pbos): Set up the quality scaler so that both senders reliably start
+  // at QVGA, then verify that instead.
+  cricket::VideoCodec codec = kVp8Codec;
+  codec.params[kCodecParamStartBitrate] = "1000000";
+  Base::TwoStreamsSendAndReceive(codec);
 }
 
 class WebRtcVideoChannel2Test : public WebRtcVideoEngine2Test {
@@ -1097,12 +1126,13 @@ class WebRtcVideoChannel2Test : public WebRtcVideoEngine2Test {
     VideoSendParameters limited_send_params = send_parameters_;
     limited_send_params.max_bandwidth_bps = global_max;
     EXPECT_TRUE(channel_->SetSendParameters(limited_send_params));
-    webrtc::RtpParameters parameters = channel_->GetRtpParameters(last_ssrc_);
+    webrtc::RtpParameters parameters =
+        channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, parameters.encodings.size());
     parameters.encodings[0].max_bitrate_bps = stream_max;
-    EXPECT_TRUE(channel_->SetRtpParameters(last_ssrc_, parameters));
+    EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
     // Read back the parameteres and verify they have the correct value
-    parameters = channel_->GetRtpParameters(last_ssrc_);
+    parameters = channel_->GetRtpSendParameters(last_ssrc_);
     EXPECT_EQ(1UL, parameters.encodings.size());
     EXPECT_EQ(stream_max, parameters.encodings[0].max_bitrate_bps);
     // Verify that the new value propagated down to the encoder
@@ -1151,11 +1181,13 @@ TEST_F(WebRtcVideoChannel2Test, RecvStreamWithSimAndRtx) {
   // Receiver side.
   FakeVideoReceiveStream* recv_stream = AddRecvStream(
       cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
-  ASSERT_GT(recv_stream->GetConfig().rtp.rtx.size(), 0u)
-      << "No SSRCs for RTX configured by AddRecvStream.";
-  EXPECT_EQ(rtx_ssrcs[0],
-            recv_stream->GetConfig().rtp.rtx.begin()->second.ssrc);
-  // TODO(pbos): Make sure we set the RTX for correct payloads etc.
+  EXPECT_FALSE(recv_stream->GetConfig().rtp.rtx.empty());
+  EXPECT_EQ(recv_stream->GetConfig().decoders.size(),
+            recv_stream->GetConfig().rtp.rtx.size())
+      << "RTX should be mapped for all decoders/payload types.";
+  for (const auto& kv : recv_stream->GetConfig().rtp.rtx) {
+    EXPECT_EQ(rtx_ssrcs[0], kv.second.ssrc);
+  }
 }
 
 TEST_F(WebRtcVideoChannel2Test, RecvStreamWithRtx) {
@@ -1538,7 +1570,7 @@ TEST_F(WebRtcVideoChannel2Test, ReconfiguresEncodersWhenNotSending) {
   EXPECT_EQ(144u, streams[0].height);
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(capturer.CaptureFrame());
@@ -1550,7 +1582,7 @@ TEST_F(WebRtcVideoChannel2Test, ReconfiguresEncodersWhenNotSending) {
   // No frames should have been actually put in there though.
   EXPECT_EQ(0, stream->GetNumberOfSwappedFrames());
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, UsesCorrectSettingsForScreencast) {
@@ -1567,7 +1599,7 @@ TEST_F(WebRtcVideoChannel2Test, UsesCorrectSettingsForScreencast) {
   channel_->SetVideoSend(last_ssrc_, true, &min_bitrate_options);
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   cricket::VideoFormat capture_format_hd =
       capturer.GetSupportedFormats()->front();
   EXPECT_EQ(1280, capture_format_hd.width);
@@ -1591,10 +1623,10 @@ TEST_F(WebRtcVideoChannel2Test, UsesCorrectSettingsForScreencast) {
   EXPECT_EQ(0, encoder_config.min_transmit_bitrate_bps)
       << "Non-screenshare shouldn't use min-transmit bitrate.";
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, nullptr));
+  channel_->SetSource(last_ssrc_, nullptr);
   // Removing a capturer triggers a black frame to be sent.
   EXPECT_EQ(2, send_stream->GetNumberOfSwappedFrames());
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   VideoOptions screencast_options;
   screencast_options.is_screencast = rtc::Optional<bool>(true);
   EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &screencast_options));
@@ -1614,7 +1646,7 @@ TEST_F(WebRtcVideoChannel2Test, UsesCorrectSettingsForScreencast) {
   EXPECT_EQ(capture_format_hd.height, encoder_config.streams.front().height);
   EXPECT_TRUE(encoder_config.streams[0].temporal_layer_thresholds_bps.empty());
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, NoRecreateStreamForScreencast) {
@@ -1624,7 +1656,7 @@ TEST_F(WebRtcVideoChannel2Test, NoRecreateStreamForScreencast) {
   EXPECT_TRUE(channel_->SetSend(true));
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(kSsrc, &capturer));
+  channel_->SetSource(kSsrc, &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(capturer.CaptureFrame());
@@ -1665,7 +1697,7 @@ TEST_F(WebRtcVideoChannel2Test, NoRecreateStreamForScreencast) {
   EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo,
             encoder_config.content_type);
 
-  EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
+  channel_->SetSource(kSsrc, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test,
@@ -1680,7 +1712,7 @@ TEST_F(WebRtcVideoChannel2Test,
   options.is_screencast = rtc::Optional<bool>(true);
   channel_->SetVideoSend(last_ssrc_, true, &options);
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   cricket::VideoFormat capture_format_hd =
       capturer.GetSupportedFormats()->front();
   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format_hd));
@@ -1702,7 +1734,7 @@ TEST_F(WebRtcVideoChannel2Test,
   EXPECT_EQ(kConferenceScreencastTemporalBitrateBps,
             encoder_config.streams[0].temporal_layer_thresholds_bps[0]);
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, SuspendBelowMinBitrateDisabledByDefault) {
@@ -1752,7 +1784,7 @@ TEST_F(WebRtcVideoChannel2Test, VerifyVp8SpecificSettings) {
   cricket::FakeVideoCapturer capturer;
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   channel_->SetSend(true);
 
   EXPECT_TRUE(capturer.CaptureFrame());
@@ -1776,9 +1808,9 @@ TEST_F(WebRtcVideoChannel2Test, VerifyVp8SpecificSettings) {
   EXPECT_TRUE(vp8_settings.automaticResizeOn);
   EXPECT_TRUE(vp8_settings.frameDroppingOn);
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
   stream = SetUpSimulcast(true, false);
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   channel_->SetSend(true);
   EXPECT_TRUE(capturer.CaptureFrame());
 
@@ -1809,7 +1841,7 @@ TEST_F(WebRtcVideoChannel2Test, VerifyVp8SpecificSettings) {
   EXPECT_FALSE(vp8_settings.automaticResizeOn);
   EXPECT_FALSE(vp8_settings.frameDroppingOn);
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 // Test that setting the same options doesn't result in the encoder being
@@ -1819,7 +1851,7 @@ TEST_F(WebRtcVideoChannel2Test, SetIdenticalOptionsDoesntReconfigureEncoder) {
   cricket::FakeVideoCapturer capturer;
 
   FakeVideoSendStream* send_stream = AddSendStream();
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options));
@@ -1833,7 +1865,7 @@ TEST_F(WebRtcVideoChannel2Test, SetIdenticalOptionsDoesntReconfigureEncoder) {
   EXPECT_TRUE(capturer.CaptureFrame());
   EXPECT_EQ(2, send_stream->num_encoder_reconfigurations());
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, nullptr));
+  channel_->SetSource(last_ssrc_, nullptr);
 }
 
 class Vp9SettingsTest : public WebRtcVideoChannel2Test {
@@ -1871,7 +1903,7 @@ TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) {
   cricket::FakeVideoCapturer capturer;
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   channel_->SetSend(true);
 
   EXPECT_TRUE(capturer.CaptureFrame());
@@ -1912,7 +1944,7 @@ TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) {
   EXPECT_FALSE(vp9_settings.denoisingOn);
   EXPECT_FALSE(vp9_settings.frameDroppingOn);
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 class Vp9SettingsTestWithFieldTrial : public Vp9SettingsTest {
@@ -1931,7 +1963,7 @@ class Vp9SettingsTestWithFieldTrial : public Vp9SettingsTest {
     cricket::FakeVideoCapturer capturer;
     EXPECT_EQ(cricket::CS_RUNNING,
               capturer.Start(capturer.GetSupportedFormats()->front()));
-    EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+    channel_->SetSource(last_ssrc_, &capturer);
     channel_->SetSend(true);
 
     EXPECT_TRUE(capturer.CaptureFrame());
@@ -1941,7 +1973,7 @@ class Vp9SettingsTestWithFieldTrial : public Vp9SettingsTest {
     EXPECT_EQ(num_spatial_layers, vp9_settings.numberOfSpatialLayers);
     EXPECT_EQ(num_temporal_layers, vp9_settings.numberOfTemporalLayers);
 
-    EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+    channel_->SetSource(last_ssrc_, NULL);
   }
 };
 
@@ -2006,7 +2038,7 @@ TEST_F(WebRtcVideoChannel2Test, AdaptsOnOveruseAndChangeResolution) {
   AddSendStream();
 
   cricket::FakeVideoCapturer capturer;
-  ASSERT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   ASSERT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   ASSERT_TRUE(channel_->SetSend(true));
@@ -2064,7 +2096,7 @@ TEST_F(WebRtcVideoChannel2Test, AdaptsOnOveruseAndChangeResolution) {
   EXPECT_EQ(1284, send_stream->GetLastWidth());
   EXPECT_EQ(724, send_stream->GetLastHeight());
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, PreviousAdaptationDoesNotApplyToScreenshare) {
@@ -2080,7 +2112,7 @@ TEST_F(WebRtcVideoChannel2Test, PreviousAdaptationDoesNotApplyToScreenshare) {
   AddSendStream();
 
   cricket::FakeVideoCapturer capturer;
-  ASSERT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   ASSERT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   ASSERT_TRUE(channel_->SetSend(true));
@@ -2109,7 +2141,7 @@ TEST_F(WebRtcVideoChannel2Test, PreviousAdaptationDoesNotApplyToScreenshare) {
   cricket::FakeVideoCapturer screen_share(true);
   ASSERT_EQ(cricket::CS_RUNNING,
             screen_share.Start(screen_share.GetSupportedFormats()->front()));
-  ASSERT_TRUE(channel_->SetCapturer(last_ssrc_, &screen_share));
+  channel_->SetSource(last_ssrc_, &screen_share);
   cricket::VideoOptions screenshare_options;
   screenshare_options.is_screencast = rtc::Optional<bool>(true);
   channel_->SetVideoSend(last_ssrc_, true /* enable */, &screenshare_options);
@@ -2119,14 +2151,14 @@ TEST_F(WebRtcVideoChannel2Test, PreviousAdaptationDoesNotApplyToScreenshare) {
   EXPECT_EQ(724, send_stream->GetLastHeight());
 
   // Switch back to the normal capturer. Expect the frame to be CPU adapted.
-  ASSERT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   channel_->SetVideoSend(last_ssrc_, true /* enable */, &camera_options);
   EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720, cricket::FOURCC_I420));
   EXPECT_EQ(4, send_stream->GetNumberOfSwappedFrames());
   EXPECT_EQ(1280 * 3 / 4, send_stream->GetLastWidth());
   EXPECT_EQ(720 * 3 / 4, send_stream->GetLastHeight());
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 void WebRtcVideoChannel2Test::TestCpuAdaptation(bool enable_overuse,
@@ -2151,7 +2183,7 @@ void WebRtcVideoChannel2Test::TestCpuAdaptation(bool enable_overuse,
   EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options));
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
 
@@ -2172,7 +2204,7 @@ void WebRtcVideoChannel2Test::TestCpuAdaptation(bool enable_overuse,
     EXPECT_EQ(codec.width, send_stream->GetLastWidth());
     EXPECT_EQ(codec.height, send_stream->GetLastHeight());
 
-    EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+    channel_->SetSource(last_ssrc_, NULL);
     return;
   }
 
@@ -2201,7 +2233,7 @@ void WebRtcVideoChannel2Test::TestCpuAdaptation(bool enable_overuse,
   EXPECT_EQ(codec.width, send_stream->GetLastWidth());
   EXPECT_EQ(codec.height, send_stream->GetLastHeight());
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, EstimatesNtpStartTimeCorrectly) {
@@ -2280,7 +2312,7 @@ TEST_F(WebRtcVideoChannel2Test, SetSendCodecsWithoutFec) {
 TEST_F(WebRtcVideoChannel2Test,
        SetSendCodecRejectsRtxWithoutAssociatedPayloadType) {
   cricket::VideoSendParameters parameters;
-  cricket::VideoCodec rtx_codec(96, "rtx", 0, 0, 0, 0);
+  cricket::VideoCodec rtx_codec(96, "rtx", 0, 0, 0);
   parameters.codecs.push_back(rtx_codec);
   EXPECT_FALSE(channel_->SetSendParameters(parameters))
       << "RTX codec without associated payload type should be rejected.";
@@ -2332,7 +2364,7 @@ TEST_F(WebRtcVideoChannel2Test, SetSendCodecsChangesExistingStreams) {
   FakeVideoSendStream* stream = AddSendStream();
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(capturer.CaptureFrame());
@@ -2347,7 +2379,7 @@ TEST_F(WebRtcVideoChannel2Test, SetSendCodecsChangesExistingStreams) {
   streams = fake_call_->GetVideoSendStreams()[0]->GetVideoStreams();
   EXPECT_EQ(kVp8Codec360p.width, streams[0].width);
   EXPECT_EQ(kVp8Codec360p.height, streams[0].height);
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, SetSendCodecsWithBitrates) {
@@ -2378,6 +2410,43 @@ TEST_F(WebRtcVideoChannel2Test, SetSendCodecsRejectsMaxLessThanMinBitrate) {
   EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
 }
 
+// Test that when both the codec-specific bitrate params and max_bandwidth_bps
+// are present in the same send parameters, the settings are combined correctly.
+TEST_F(WebRtcVideoChannel2Test, SetSendCodecsWithBitratesAndMaxSendBandwidth) {
+  send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
+  send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
+  send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
+  send_parameters_.max_bandwidth_bps = 400000;
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
+  EXPECT_EQ(200000, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
+  // We expect max_bandwidth_bps to take priority, if set.
+  EXPECT_EQ(400000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
+
+  // Decrease max_bandwidth_bps.
+  send_parameters_.max_bandwidth_bps = 350000;
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
+  // Since the codec isn't changing, start_bitrate_bps should be -1.
+  EXPECT_EQ(-1, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
+  EXPECT_EQ(350000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
+
+  // Now try again with the values flipped around.
+  send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "400";
+  send_parameters_.max_bandwidth_bps = 300000;
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
+  EXPECT_EQ(200000, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
+  EXPECT_EQ(300000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
+
+  // If we change the codec max, max_bandwidth_bps should still apply.
+  send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "350";
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
+  EXPECT_EQ(200000, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
+  EXPECT_EQ(300000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
+}
+
 TEST_F(WebRtcVideoChannel2Test,
        SetMaxSendBandwidthShouldPreserveOtherBitrates) {
   SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
@@ -2411,7 +2480,7 @@ TEST_F(WebRtcVideoChannel2Test, SetMaxSendBitrateCanIncreaseSenderBitrate) {
   FakeVideoSendStream* stream = AddSendStream();
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
 
@@ -2425,7 +2494,7 @@ TEST_F(WebRtcVideoChannel2Test, SetMaxSendBitrateCanIncreaseSenderBitrate) {
   EXPECT_TRUE(capturer.CaptureFrame());
   streams = stream->GetVideoStreams();
   EXPECT_EQ(initial_max_bitrate_bps * 2, streams[0].max_bitrate_bps);
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, nullptr));
+  channel_->SetSource(last_ssrc_, nullptr);
 }
 
 TEST_F(WebRtcVideoChannel2Test,
@@ -2440,7 +2509,7 @@ TEST_F(WebRtcVideoChannel2Test,
 
   // Send a frame to make sure this scales up to >1 stream (simulcast).
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], &capturer));
+  channel_->SetSource(kSsrcs3[0], &capturer);
   EXPECT_EQ(cricket::CS_RUNNING,
             capturer.Start(capturer.GetSupportedFormats()->front()));
   EXPECT_TRUE(capturer.CaptureFrame());
@@ -2459,7 +2528,7 @@ TEST_F(WebRtcVideoChannel2Test,
   int increased_max_bitrate_bps = GetTotalMaxBitrateBps(streams);
   EXPECT_EQ(initial_max_bitrate_bps * 2, increased_max_bitrate_bps);
 
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], nullptr));
+  channel_->SetSource(kSsrcs3[0], nullptr);
 }
 
 TEST_F(WebRtcVideoChannel2Test, SetSendCodecsWithMaxQuantization) {
@@ -2512,6 +2581,26 @@ TEST_F(WebRtcVideoChannel2Test, SetSendCodecsAcceptAllValidPayloadTypes) {
   }
 }
 
+// Test that setting the a different set of codecs but with an identical front
+// codec doesn't result in the stream being recreated.
+// This may happen when a subsequent negotiation includes fewer codecs, as a
+// result of one of the codecs being rejected.
+TEST_F(WebRtcVideoChannel2Test,
+       SetSendCodecsIdenticalFirstCodecDoesntRecreateStream) {
+  cricket::VideoSendParameters parameters1;
+  parameters1.codecs.push_back(kVp8Codec);
+  parameters1.codecs.push_back(kVp9Codec);
+  EXPECT_TRUE(channel_->SetSendParameters(parameters1));
+
+  AddSendStream();
+  EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
+
+  cricket::VideoSendParameters parameters2;
+  parameters2.codecs.push_back(kVp8Codec);
+  EXPECT_TRUE(channel_->SetSendParameters(parameters2));
+  EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
+}
+
 TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsWithOnlyVp8) {
   cricket::VideoRecvParameters parameters;
   parameters.codecs.push_back(kVp8Codec);
@@ -2522,7 +2611,7 @@ TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsWithOnlyVp8) {
 TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsWithRtx) {
   cricket::VideoRecvParameters parameters;
   parameters.codecs.push_back(kVp8Codec);
-  cricket::VideoCodec rtx_codec(96, "rtx", 0, 0, 0, 0);
+  cricket::VideoCodec rtx_codec(96, "rtx", 0, 0, 0);
   parameters.codecs.push_back(rtx_codec);
   EXPECT_FALSE(channel_->SetRecvParameters(parameters))
       << "RTX codec without associated payload should be rejected.";
@@ -2534,7 +2623,7 @@ TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsWithRtx) {
   parameters.codecs[1].SetParam("apt", kVp8Codec.id);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
 
-  cricket::VideoCodec rtx_codec2(97, "rtx", 0, 0, 0, 0);
+  cricket::VideoCodec rtx_codec2(97, "rtx", 0, 0, 0);
   rtx_codec2.SetParam("apt", rtx_codec.id);
   parameters.codecs.push_back(rtx_codec2);
 
@@ -2564,7 +2653,7 @@ TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsAcceptDefaultCodecs) {
 TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsRejectUnsupportedCodec) {
   cricket::VideoRecvParameters parameters;
   parameters.codecs.push_back(kVp8Codec);
-  parameters.codecs.push_back(VideoCodec(101, "WTF3", 640, 400, 30, 0));
+  parameters.codecs.push_back(VideoCodec(101, "WTF3", 640, 400, 30));
   EXPECT_FALSE(channel_->SetRecvParameters(parameters));
 }
 
@@ -2589,6 +2678,7 @@ TEST_F(WebRtcVideoChannel2Test,
 TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsWithoutFecDisablesFec) {
   cricket::VideoSendParameters send_parameters;
   send_parameters.codecs.push_back(kVp8Codec);
+  send_parameters.codecs.push_back(kRedCodec);
   send_parameters.codecs.push_back(kUlpfecCodec);
   ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
 
@@ -2607,6 +2697,41 @@ TEST_F(WebRtcVideoChannel2Test, SetRecvCodecsWithoutFecDisablesFec) {
       << "SetSendCodec without FEC should disable current FEC.";
 }
 
+TEST_F(WebRtcVideoChannel2Test, SetSendParamsWithoutFecDisablesReceivingFec) {
+  FakeVideoReceiveStream* stream = AddRecvStream();
+  webrtc::VideoReceiveStream::Config config = stream->GetConfig();
+
+  EXPECT_EQ(kUlpfecCodec.id, config.rtp.fec.ulpfec_payload_type);
+
+  cricket::VideoRecvParameters recv_parameters;
+  recv_parameters.codecs.push_back(kVp8Codec);
+  recv_parameters.codecs.push_back(kRedCodec);
+  recv_parameters.codecs.push_back(kUlpfecCodec);
+  ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters));
+  stream = fake_call_->GetVideoReceiveStreams()[0];
+  ASSERT_TRUE(stream != NULL);
+  config = stream->GetConfig();
+  EXPECT_EQ(kUlpfecCodec.id, config.rtp.fec.ulpfec_payload_type)
+      << "FEC should be enabled on the recieve stream.";
+
+  cricket::VideoSendParameters send_parameters;
+  send_parameters.codecs.push_back(kVp8Codec);
+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
+  stream = fake_call_->GetVideoReceiveStreams()[0];
+  config = stream->GetConfig();
+  EXPECT_EQ(-1, config.rtp.fec.ulpfec_payload_type)
+      << "FEC should have been disabled when we know the other side won't do "
+         "FEC.";
+
+  send_parameters.codecs.push_back(kRedCodec);
+  send_parameters.codecs.push_back(kUlpfecCodec);
+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters));
+  stream = fake_call_->GetVideoReceiveStreams()[0];
+  config = stream->GetConfig();
+  EXPECT_EQ(kUlpfecCodec.id, config.rtp.fec.ulpfec_payload_type)
+      << "FEC should be enabled on the recieve stream.";
+}
+
 TEST_F(WebRtcVideoChannel2Test, SetSendCodecsRejectDuplicateFecPayloads) {
   cricket::VideoRecvParameters parameters;
   parameters.codecs.push_back(kVp8Codec);
@@ -2632,10 +2757,10 @@ TEST_F(WebRtcVideoChannel2Test,
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
 }
 
-// Test that setting the same codecs but with a different order and preference
+// Test that setting the same codecs but with a different order
 // doesn't result in the stream being recreated.
 TEST_F(WebRtcVideoChannel2Test,
-       SetRecvCodecsDifferentOrderAndPreferenceDoesntRecreateStream) {
+       SetRecvCodecsDifferentOrderDoesntRecreateStream) {
   cricket::VideoRecvParameters parameters1;
   parameters1.codecs.push_back(kVp8Codec);
   parameters1.codecs.push_back(kRedCodec);
@@ -2647,7 +2772,6 @@ TEST_F(WebRtcVideoChannel2Test,
   cricket::VideoRecvParameters parameters2;
   parameters2.codecs.push_back(kRedCodec);
   parameters2.codecs.push_back(kVp8Codec);
-  parameters2.codecs[1].preference += 1;
   EXPECT_TRUE(channel_->SetRecvParameters(parameters2));
   EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
 }
@@ -2829,10 +2953,10 @@ TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationStats) {
       video_capturer_vga.GetSupportedFormats();
   cricket::VideoFormat capture_format_vga = (*formats)[1];
   EXPECT_EQ(cricket::CS_RUNNING, video_capturer_vga.Start(capture_format_vga));
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], &video_capturer_vga));
+  channel_->SetSource(kSsrcs3[0], &video_capturer_vga);
   EXPECT_TRUE(video_capturer_vga.CaptureFrame());
 
-  cricket::VideoCodec send_codec(100, "VP8", 640, 480, 30, 0);
+  cricket::VideoCodec send_codec(100, "VP8", 640, 480, 30);
   cricket::VideoSendParameters parameters;
   parameters.codecs.push_back(send_codec);
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
@@ -2866,7 +2990,7 @@ TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationStats) {
             info.senders[0].adapt_reason);
 
   // No capturer (no adapter). Adapt changes from old adapter should be kept.
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], NULL));
+  channel_->SetSource(kSsrcs3[0], NULL);
   info.Clear();
   EXPECT_TRUE(channel_->GetStats(&info));
   ASSERT_EQ(1U, info.senders.size());
@@ -2878,7 +3002,7 @@ TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationStats) {
   cricket::FakeVideoCapturer video_capturer_hd;
   cricket::VideoFormat capture_format_hd = (*formats)[0];
   EXPECT_EQ(cricket::CS_RUNNING, video_capturer_hd.Start(capture_format_hd));
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], &video_capturer_hd));
+  channel_->SetSource(kSsrcs3[0], &video_capturer_hd);
   EXPECT_TRUE(video_capturer_hd.CaptureFrame());
 
   // Trigger overuse, HD -> adapt (OnCpuResolutionRequest downgrade) -> HD/2.
@@ -2890,7 +3014,7 @@ TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationStats) {
   EXPECT_EQ(3, info.senders[0].adapt_changes);
   EXPECT_EQ(WebRtcVideoChannel2::ADAPTREASON_CPU, info.senders[0].adapt_reason);
 
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], NULL));
+  channel_->SetSource(kSsrcs3[0], NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationAndBandwidthStats) {
@@ -2902,10 +3026,10 @@ TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationAndBandwidthStats) {
       video_capturer_vga.GetSupportedFormats();
   cricket::VideoFormat capture_format_vga = (*formats)[1];
   EXPECT_EQ(cricket::CS_RUNNING, video_capturer_vga.Start(capture_format_vga));
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], &video_capturer_vga));
+  channel_->SetSource(kSsrcs3[0], &video_capturer_vga);
   EXPECT_TRUE(video_capturer_vga.CaptureFrame());
 
-  cricket::VideoCodec send_codec(100, "VP8", 640, 480, 30, 0);
+  cricket::VideoCodec send_codec(100, "VP8", 640, 480, 30);
   cricket::VideoSendParameters parameters;
   parameters.codecs.push_back(send_codec);
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
@@ -2954,7 +3078,7 @@ TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationAndBandwidthStats) {
   EXPECT_EQ(WebRtcVideoChannel2::ADAPTREASON_NONE,
             info.senders[0].adapt_reason);
 
-  EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], NULL));
+  channel_->SetSource(kSsrcs3[0], NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test,
@@ -3131,7 +3255,7 @@ TEST_F(WebRtcVideoChannel2Test, DefaultReceiveStreamReconfiguresToUseRtx) {
   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
       << "No default receive stream created.";
   FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
-  EXPECT_EQ(0u, recv_stream->GetConfig().rtp.rtx.size())
+  EXPECT_TRUE(recv_stream->GetConfig().rtp.rtx.empty())
       << "Default receive stream should not have configured RTX";
 
   EXPECT_TRUE(channel_->AddRecvStream(
@@ -3139,9 +3263,13 @@ TEST_F(WebRtcVideoChannel2Test, DefaultReceiveStreamReconfiguresToUseRtx) {
   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
       << "AddRecvStream should've reconfigured, not added a new receiver.";
   recv_stream = fake_call_->GetVideoReceiveStreams()[0];
-  ASSERT_GE(2u, recv_stream->GetConfig().rtp.rtx.size());
-  EXPECT_EQ(rtx_ssrcs[0],
-            recv_stream->GetConfig().rtp.rtx.begin()->second.ssrc);
+  EXPECT_FALSE(recv_stream->GetConfig().rtp.rtx.empty());
+  EXPECT_EQ(recv_stream->GetConfig().decoders.size(),
+            recv_stream->GetConfig().rtp.rtx.size())
+      << "RTX should be mapped for all decoders/payload types.";
+  for (const auto& kv : recv_stream->GetConfig().rtp.rtx) {
+    EXPECT_EQ(rtx_ssrcs[0], kv.second.ssrc);
+  }
 }
 
 TEST_F(WebRtcVideoChannel2Test, RejectsAddingStreamsWithMissingSsrcsForRtx) {
@@ -3316,7 +3444,7 @@ TEST_F(WebRtcVideoChannel2Test, CanSentMaxBitrateForExistingStream) {
   AddSendStream();
 
   cricket::FakeVideoCapturer capturer;
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, &capturer));
+  channel_->SetSource(last_ssrc_, &capturer);
   cricket::VideoFormat capture_format_hd =
       capturer.GetSupportedFormats()->front();
   EXPECT_EQ(1280, capture_format_hd.width);
@@ -3341,20 +3469,21 @@ TEST_F(WebRtcVideoChannel2Test, CanSentMaxBitrateForExistingStream) {
   SetAndExpectMaxBitrate(capturer, 0, 800, 800);
   SetAndExpectMaxBitrate(capturer, 0, 0, default_encoder_bitrate);
 
-  EXPECT_TRUE(channel_->SetCapturer(last_ssrc_, NULL));
+  channel_->SetSource(last_ssrc_, NULL);
 }
 
 TEST_F(WebRtcVideoChannel2Test, CannotSetMaxBitrateForNonexistentStream) {
   webrtc::RtpParameters nonexistent_parameters =
-      channel_->GetRtpParameters(last_ssrc_);
+      channel_->GetRtpSendParameters(last_ssrc_);
   EXPECT_EQ(0, nonexistent_parameters.encodings.size());
 
   nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
-  EXPECT_FALSE(channel_->SetRtpParameters(last_ssrc_, nonexistent_parameters));
+  EXPECT_FALSE(
+      channel_->SetRtpSendParameters(last_ssrc_, nonexistent_parameters));
 }
 
 TEST_F(WebRtcVideoChannel2Test,
-       CannotSetRtpParametersWithIncorrectNumberOfEncodings) {
+       CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
   // This test verifies that setting RtpParameters succeeds only if
   // the structure contains exactly one encoding.
   // TODO(skvlad): Update this test when we start supporting setting parameters
@@ -3362,39 +3491,107 @@ TEST_F(WebRtcVideoChannel2Test,
 
   AddSendStream();
   // Setting RtpParameters with no encoding is expected to fail.
-  webrtc::RtpParameters parameters;
-  EXPECT_FALSE(channel_->SetRtpParameters(last_ssrc_, parameters));
+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
+  parameters.encodings.clear();
+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
   // Setting RtpParameters with exactly one encoding should succeed.
   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
-  EXPECT_TRUE(channel_->SetRtpParameters(last_ssrc_, parameters));
+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
   // Two or more encodings should result in failure.
   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
-  EXPECT_FALSE(channel_->SetRtpParameters(last_ssrc_, parameters));
+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
 }
 
 // Test that a stream will not be sending if its encoding is made
-// inactive through SetRtpParameters.
+// inactive through SetRtpSendParameters.
 // TODO(deadbeef): Update this test when we start supporting setting parameters
 // for each encoding individually.
-TEST_F(WebRtcVideoChannel2Test, SetRtpParametersEncodingsActive) {
+TEST_F(WebRtcVideoChannel2Test, SetRtpSendParametersEncodingsActive) {
   FakeVideoSendStream* stream = AddSendStream();
   EXPECT_TRUE(channel_->SetSend(true));
   EXPECT_TRUE(stream->IsSending());
 
   // Get current parameters and change "active" to false.
-  webrtc::RtpParameters parameters = channel_->GetRtpParameters(last_ssrc_);
+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
   ASSERT_EQ(1u, parameters.encodings.size());
   ASSERT_TRUE(parameters.encodings[0].active);
   parameters.encodings[0].active = false;
-  EXPECT_TRUE(channel_->SetRtpParameters(last_ssrc_, parameters));
+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
   EXPECT_FALSE(stream->IsSending());
 
   // Now change it back to active and verify we resume sending.
   parameters.encodings[0].active = true;
-  EXPECT_TRUE(channel_->SetRtpParameters(last_ssrc_, parameters));
+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
   EXPECT_TRUE(stream->IsSending());
 }
 
+// Test that GetRtpSendParameters returns the currently configured codecs.
+TEST_F(WebRtcVideoChannel2Test, GetRtpSendParametersCodecs) {
+  AddSendStream();
+  cricket::VideoSendParameters parameters;
+  parameters.codecs.push_back(kVp8Codec);
+  parameters.codecs.push_back(kVp9Codec);
+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
+
+  webrtc::RtpParameters rtp_parameters =
+      channel_->GetRtpSendParameters(last_ssrc_);
+  ASSERT_EQ(2u, rtp_parameters.codecs.size());
+  EXPECT_EQ(kVp8Codec.ToCodecParameters(), rtp_parameters.codecs[0]);
+  EXPECT_EQ(kVp9Codec.ToCodecParameters(), rtp_parameters.codecs[1]);
+}
+
+// Test that if we set/get parameters multiple times, we get the same results.
+TEST_F(WebRtcVideoChannel2Test, SetAndGetRtpSendParameters) {
+  AddSendStream();
+  cricket::VideoSendParameters parameters;
+  parameters.codecs.push_back(kVp8Codec);
+  parameters.codecs.push_back(kVp9Codec);
+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
+
+  webrtc::RtpParameters initial_params =
+      channel_->GetRtpSendParameters(last_ssrc_);
+
+  // We should be able to set the params we just got.
+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, initial_params));
+
+  // ... And this shouldn't change the params returned by GetRtpSendParameters.
+  EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(last_ssrc_));
+}
+
+// Test that GetRtpReceiveParameters returns the currently configured codecs.
+TEST_F(WebRtcVideoChannel2Test, GetRtpReceiveParametersCodecs) {
+  AddRecvStream();
+  cricket::VideoRecvParameters parameters;
+  parameters.codecs.push_back(kVp8Codec);
+  parameters.codecs.push_back(kVp9Codec);
+  EXPECT_TRUE(channel_->SetRecvParameters(parameters));
+
+  webrtc::RtpParameters rtp_parameters =
+      channel_->GetRtpReceiveParameters(last_ssrc_);
+  ASSERT_EQ(2u, rtp_parameters.codecs.size());
+  EXPECT_EQ(kVp8Codec.ToCodecParameters(), rtp_parameters.codecs[0]);
+  EXPECT_EQ(kVp9Codec.ToCodecParameters(), rtp_parameters.codecs[1]);
+}
+
+// Test that if we set/get parameters multiple times, we get the same results.
+TEST_F(WebRtcVideoChannel2Test, SetAndGetRtpReceiveParameters) {
+  AddRecvStream();
+  cricket::VideoRecvParameters parameters;
+  parameters.codecs.push_back(kVp8Codec);
+  parameters.codecs.push_back(kVp9Codec);
+  EXPECT_TRUE(channel_->SetRecvParameters(parameters));
+
+  webrtc::RtpParameters initial_params =
+      channel_->GetRtpReceiveParameters(last_ssrc_);
+
+  // We should be able to set the params we just got.
+  EXPECT_TRUE(channel_->SetRtpReceiveParameters(last_ssrc_, initial_params));
+
+  // ... And this shouldn't change the params returned by
+  // GetRtpReceiveParameters.
+  EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(last_ssrc_));
+}
+
 void WebRtcVideoChannel2Test::TestReceiverLocalSsrcConfiguration(
     bool receiver_first) {
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
@@ -3475,7 +3672,7 @@ class WebRtcVideoChannel2SimulcastTest : public testing::Test {
     // Send a full-size frame to trigger a stream reconfiguration to use all
     // expected simulcast layers.
     cricket::FakeVideoCapturer capturer;
-    EXPECT_TRUE(channel_->SetCapturer(ssrcs.front(), &capturer));
+    channel_->SetSource(ssrcs.front(), &capturer);
     EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(cricket::VideoFormat(
                                        codec.width, codec.height,
                                        cricket::VideoFormat::FpsToInterval(30),
@@ -3532,7 +3729,7 @@ class WebRtcVideoChannel2SimulcastTest : public testing::Test {
     ASSERT_EQ(1u, info.senders.size());
     EXPECT_EQ(total_max_bitrate_bps, info.senders[0].preferred_bitrate);
 
-    EXPECT_TRUE(channel_->SetCapturer(ssrcs.front(), NULL));
+    channel_->SetSource(ssrcs.front(), NULL);
   }
 
   FakeVideoSendStream* AddSendStream() {
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideoframe.cc b/chromium/third_party/webrtc/media/engine/webrtcvideoframe.cc
index 3a2d2def0e3..cda00275f70 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideoframe.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideoframe.cc
@@ -22,18 +22,24 @@ using webrtc::kVPlane;
 
 namespace cricket {
 
-WebRtcVideoFrame::WebRtcVideoFrame():
-    time_stamp_ns_(0),
-    rotation_(webrtc::kVideoRotation_0) {}
+WebRtcVideoFrame::WebRtcVideoFrame()
+    : timestamp_us_(0), rotation_(webrtc::kVideoRotation_0) {}
+
+WebRtcVideoFrame::WebRtcVideoFrame(
+    const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
+    webrtc::VideoRotation rotation,
+    int64_t timestamp_us)
+    : video_frame_buffer_(buffer),
+      timestamp_us_(timestamp_us),
+      rotation_(rotation) {}
 
 WebRtcVideoFrame::WebRtcVideoFrame(
     const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
     int64_t time_stamp_ns,
     webrtc::VideoRotation rotation)
-    : video_frame_buffer_(buffer),
-      time_stamp_ns_(time_stamp_ns),
-      rotation_(rotation) {
-}
+    : WebRtcVideoFrame(buffer,
+                       rotation,
+                       time_stamp_ns / rtc::kNumNanosecsPerMicrosec) {}
 
 WebRtcVideoFrame::~WebRtcVideoFrame() {}
 
@@ -47,7 +53,7 @@ bool WebRtcVideoFrame::Init(uint32_t format,
                             int64_t time_stamp_ns,
                             webrtc::VideoRotation rotation) {
   return Reset(format, w, h, dw, dh, sample, sample_size,
-               time_stamp_ns, rotation,
+               time_stamp_ns / rtc::kNumNanosecsPerMicrosec, rotation,
                true /*apply_rotation*/);
 }
 
@@ -55,7 +61,7 @@ bool WebRtcVideoFrame::Init(const CapturedFrame* frame, int dw, int dh,
                             bool apply_rotation) {
   return Reset(frame->fourcc, frame->width, frame->height, dw, dh,
                static_cast<uint8_t*>(frame->data), frame->data_size,
-               frame->time_stamp,
+               frame->time_stamp / rtc::kNumNanosecsPerMicrosec,
                frame->rotation, apply_rotation);
 }
 
@@ -73,62 +79,21 @@ int WebRtcVideoFrame::height() const {
   return video_frame_buffer_ ? video_frame_buffer_->height() : 0;
 }
 
-const uint8_t* WebRtcVideoFrame::GetYPlane() const {
-  return video_frame_buffer_ ? video_frame_buffer_->data(kYPlane) : nullptr;
-}
-
-const uint8_t* WebRtcVideoFrame::GetUPlane() const {
-  return video_frame_buffer_ ? video_frame_buffer_->data(kUPlane) : nullptr;
-}
-
-const uint8_t* WebRtcVideoFrame::GetVPlane() const {
-  return video_frame_buffer_ ? video_frame_buffer_->data(kVPlane) : nullptr;
-}
-
-uint8_t* WebRtcVideoFrame::GetYPlane() {
-  return video_frame_buffer_ ? video_frame_buffer_->MutableData(kYPlane)
-                             : nullptr;
-}
-
-uint8_t* WebRtcVideoFrame::GetUPlane() {
-  return video_frame_buffer_ ? video_frame_buffer_->MutableData(kUPlane)
-                             : nullptr;
-}
-
-uint8_t* WebRtcVideoFrame::GetVPlane() {
-  return video_frame_buffer_ ? video_frame_buffer_->MutableData(kVPlane)
-                             : nullptr;
-}
-
-int32_t WebRtcVideoFrame::GetYPitch() const {
-  return video_frame_buffer_ ? video_frame_buffer_->stride(kYPlane) : 0;
-}
-
-int32_t WebRtcVideoFrame::GetUPitch() const {
-  return video_frame_buffer_ ? video_frame_buffer_->stride(kUPlane) : 0;
-}
-
-int32_t WebRtcVideoFrame::GetVPitch() const {
-  return video_frame_buffer_ ? video_frame_buffer_->stride(kVPlane) : 0;
-}
-
 bool WebRtcVideoFrame::IsExclusive() const {
-  return video_frame_buffer_->HasOneRef();
+  return video_frame_buffer_->IsMutable();
 }
 
 void* WebRtcVideoFrame::GetNativeHandle() const {
   return video_frame_buffer_ ? video_frame_buffer_->native_handle() : nullptr;
 }
 
-rtc::scoped_refptr<webrtc::VideoFrameBuffer>
-WebRtcVideoFrame::GetVideoFrameBuffer() const {
+const rtc::scoped_refptr<webrtc::VideoFrameBuffer>&
+WebRtcVideoFrame::video_frame_buffer() const {
   return video_frame_buffer_;
 }
 
 VideoFrame* WebRtcVideoFrame::Copy() const {
-  WebRtcVideoFrame* new_frame = new WebRtcVideoFrame(
-      video_frame_buffer_, time_stamp_ns_, rotation_);
-  return new_frame;
+  return new WebRtcVideoFrame(video_frame_buffer_, rotation_, timestamp_us_);
 }
 
 size_t WebRtcVideoFrame::ConvertToRgbBuffer(uint32_t to_fourcc,
@@ -147,7 +112,7 @@ bool WebRtcVideoFrame::Reset(uint32_t format,
                              int dh,
                              uint8_t* sample,
                              size_t sample_size,
-                             int64_t time_stamp_ns,
+                             int64_t timestamp_us,
                              webrtc::VideoRotation rotation,
                              bool apply_rotation) {
   if (!Validate(format, w, h, sample, sample_size)) {
@@ -166,8 +131,7 @@ bool WebRtcVideoFrame::Reset(uint32_t format,
     new_height = dw;
   }
 
-  InitToEmptyBuffer(new_width, new_height,
-                    time_stamp_ns);
+  InitToEmptyBuffer(new_width, new_height);
   rotation_ = apply_rotation ? webrtc::kVideoRotation_0 : rotation;
 
   int horiz_crop = ((w - dw) / 2) & ~1;
@@ -178,9 +142,12 @@ bool WebRtcVideoFrame::Reset(uint32_t format,
   int idh = (h < 0) ? -dh : dh;
   int r = libyuv::ConvertToI420(
       sample, sample_size,
-      GetYPlane(), GetYPitch(),
-      GetUPlane(), GetUPitch(),
-      GetVPlane(), GetVPitch(),
+      video_frame_buffer_->MutableDataY(),
+      video_frame_buffer_->StrideY(),
+      video_frame_buffer_->MutableDataU(),
+      video_frame_buffer_->StrideU(),
+      video_frame_buffer_->MutableDataV(),
+      video_frame_buffer_->StrideV(),
       horiz_crop, vert_crop,
       w, h,
       dw, idh,
@@ -192,28 +159,34 @@ bool WebRtcVideoFrame::Reset(uint32_t format,
                   << " return code : " << r;
     return false;
   }
+  timestamp_us_ = timestamp_us;
   return true;
 }
 
-VideoFrame* WebRtcVideoFrame::CreateEmptyFrame(
-    int w, int h,
-    int64_t time_stamp_ns) const {
+VideoFrame* WebRtcVideoFrame::CreateEmptyFrame(int w,
+                                               int h,
+                                               int64_t timestamp_us) const {
   WebRtcVideoFrame* frame = new WebRtcVideoFrame();
-  frame->InitToEmptyBuffer(w, h, time_stamp_ns);
+  frame->InitToEmptyBuffer(w, h, rtc::kNumNanosecsPerMicrosec * timestamp_us);
   return frame;
 }
 
+void WebRtcVideoFrame::InitToEmptyBuffer(int w, int h) {
+  video_frame_buffer_ = new rtc::RefCountedObject<webrtc::I420Buffer>(w, h);
+  rotation_ = webrtc::kVideoRotation_0;
+}
+
 void WebRtcVideoFrame::InitToEmptyBuffer(int w, int h,
                                          int64_t time_stamp_ns) {
   video_frame_buffer_ = new rtc::RefCountedObject<webrtc::I420Buffer>(w, h);
-  time_stamp_ns_ = time_stamp_ns;
+  SetTimeStamp(time_stamp_ns);
   rotation_ = webrtc::kVideoRotation_0;
 }
 
 const VideoFrame* WebRtcVideoFrame::GetCopyWithRotationApplied() const {
   // If the frame is not rotated, the caller should reuse this frame instead of
   // making a redundant copy.
-  if (GetVideoRotation() == webrtc::kVideoRotation_0) {
+  if (rotation() == webrtc::kVideoRotation_0) {
     return this;
   }
 
@@ -231,24 +204,29 @@ const VideoFrame* WebRtcVideoFrame::GetCopyWithRotationApplied() const {
 
   int rotated_width = orig_width;
   int rotated_height = orig_height;
-  if (GetVideoRotation() == webrtc::kVideoRotation_90 ||
-      GetVideoRotation() == webrtc::kVideoRotation_270) {
+  if (rotation() == webrtc::kVideoRotation_90 ||
+      rotation() == webrtc::kVideoRotation_270) {
     rotated_width = orig_height;
     rotated_height = orig_width;
   }
 
-  rotated_frame_.reset(CreateEmptyFrame(rotated_width, rotated_height,
-                                        GetTimeStamp()));
+  rotated_frame_.reset(
+      CreateEmptyFrame(rotated_width, rotated_height, timestamp_us_));
 
   // TODO(guoweis): Add a function in webrtc_libyuv.cc to convert from
   // VideoRotation to libyuv::RotationMode.
   int ret = libyuv::I420Rotate(
-      GetYPlane(), GetYPitch(), GetUPlane(), GetUPitch(), GetVPlane(),
-      GetVPitch(), rotated_frame_->GetYPlane(), rotated_frame_->GetYPitch(),
-      rotated_frame_->GetUPlane(), rotated_frame_->GetUPitch(),
-      rotated_frame_->GetVPlane(), rotated_frame_->GetVPitch(),
+      video_frame_buffer_->DataY(), video_frame_buffer_->StrideY(),
+      video_frame_buffer_->DataU(), video_frame_buffer_->StrideU(),
+      video_frame_buffer_->DataV(), video_frame_buffer_->StrideV(),
+      rotated_frame_->video_frame_buffer()->MutableDataY(),
+      rotated_frame_->video_frame_buffer()->StrideY(),
+      rotated_frame_->video_frame_buffer()->MutableDataU(),
+      rotated_frame_->video_frame_buffer()->StrideU(),
+      rotated_frame_->video_frame_buffer()->MutableDataV(),
+      rotated_frame_->video_frame_buffer()->StrideV(),
       orig_width, orig_height,
-      static_cast<libyuv::RotationMode>(GetVideoRotation()));
+      static_cast<libyuv::RotationMode>(rotation()));
   if (ret == 0) {
     return rotated_frame_.get();
   }
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideoframe.h b/chromium/third_party/webrtc/media/engine/webrtcvideoframe.h
index cb280b5e153..ee34c414a47 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideoframe.h
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideoframe.h
@@ -27,6 +27,13 @@ struct CapturedFrame;
 class WebRtcVideoFrame : public VideoFrame {
  public:
   WebRtcVideoFrame();
+
+  // Preferred construction, with microsecond timestamp.
+  WebRtcVideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
+                   webrtc::VideoRotation rotation,
+                   int64_t timestamp_us);
+
+  // TODO(nisse): Deprecate/delete.
   WebRtcVideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
                    int64_t time_stamp_ns,
                    webrtc::VideoRotation rotation);
@@ -47,8 +54,13 @@ class WebRtcVideoFrame : public VideoFrame {
             int64_t time_stamp_ns,
             webrtc::VideoRotation rotation);
 
+  // The timestamp of the captured frame is expected to use the same
+  // timescale and epoch as rtc::Time.
+  // TODO(nisse): Consider adding a warning message, or even an RTC_DCHECK, if
+  // the time is too far off.
   bool Init(const CapturedFrame* frame, int dw, int dh, bool apply_rotation);
 
+  void InitToEmptyBuffer(int w, int h);
   void InitToEmptyBuffer(int w, int h, int64_t time_stamp_ns);
 
   bool InitToBlack(int w, int h, int64_t time_stamp_ns);
@@ -56,27 +68,15 @@ class WebRtcVideoFrame : public VideoFrame {
   int width() const override;
   int height() const override;
 
-  const uint8_t* GetYPlane() const override;
-  const uint8_t* GetUPlane() const override;
-  const uint8_t* GetVPlane() const override;
-  uint8_t* GetYPlane() override;
-  uint8_t* GetUPlane() override;
-  uint8_t* GetVPlane() override;
-  int32_t GetYPitch() const override;
-  int32_t GetUPitch() const override;
-  int32_t GetVPitch() const override;
   void* GetNativeHandle() const override;
-  rtc::scoped_refptr<webrtc::VideoFrameBuffer> GetVideoFrameBuffer()
+  const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& video_frame_buffer()
       const override;
 
-  int64_t GetTimeStamp() const override { return time_stamp_ns_; }
-  void SetTimeStamp(int64_t time_stamp_ns) override {
-    time_stamp_ns_ = time_stamp_ns;
-  }
+  /* System monotonic clock */
+  int64_t timestamp_us() const override { return timestamp_us_; }
+  void set_timestamp_us(int64_t time_us) override { timestamp_us_ = time_us; };
 
-  webrtc::VideoRotation GetVideoRotation() const override {
-    return rotation_;
-  }
+  webrtc::VideoRotation rotation() const override { return rotation_; }
 
   VideoFrame* Copy() const override;
   bool IsExclusive() const override;
@@ -88,7 +88,7 @@ class WebRtcVideoFrame : public VideoFrame {
   const VideoFrame* GetCopyWithRotationApplied() const override;
 
  protected:
-  void SetRotation(webrtc::VideoRotation rotation) override {
+  void set_rotation(webrtc::VideoRotation rotation) override {
     rotation_ = rotation;
   }
   // Creates a frame from a raw sample with FourCC |format| and size |w| x |h|.
@@ -97,15 +97,15 @@ class WebRtcVideoFrame : public VideoFrame {
   // |dh| is destination height, like |dw|, but must be a positive number.
   // Returns whether the function succeeded or failed.
   bool Reset(uint32_t format,
-                     int w,
-                     int h,
-                     int dw,
-                     int dh,
-                     uint8_t* sample,
-                     size_t sample_size,
-                     int64_t time_stamp_ns,
-                     webrtc::VideoRotation rotation,
-                     bool apply_rotation);
+             int w,
+             int h,
+             int dw,
+             int dh,
+             uint8_t* sample,
+             size_t sample_size,
+             int64_t timestamp_us,
+             webrtc::VideoRotation rotation,
+             bool apply_rotation);
 
  private:
   VideoFrame* CreateEmptyFrame(int w, int h,
@@ -113,7 +113,7 @@ class WebRtcVideoFrame : public VideoFrame {
 
   // An opaque reference counted handle that stores the pixel data.
   rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
-  int64_t time_stamp_ns_;
+  int64_t timestamp_us_;
   webrtc::VideoRotation rotation_;
 
   // This is mutable as the calculation is expensive but once calculated, it
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideoframe_unittest.cc b/chromium/third_party/webrtc/media/engine/webrtcvideoframe_unittest.cc
index 4b0e988d288..7849de61b0a 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideoframe_unittest.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideoframe_unittest.cc
@@ -20,7 +20,8 @@ namespace {
 
 class WebRtcVideoTestFrame : public cricket::WebRtcVideoFrame {
  public:
-  using cricket::WebRtcVideoFrame::SetRotation;
+  // The ApplyRotationToFrame test needs this as a public method.
+  using cricket::WebRtcVideoFrame::set_rotation;
 
   virtual VideoFrame* CreateEmptyFrame(int w,
                                        int h,
@@ -47,7 +48,7 @@ class WebRtcVideoFrameTest : public VideoFrameTest<cricket::WebRtcVideoFrame> {
     // Build the CapturedFrame.
     cricket::CapturedFrame captured_frame;
     captured_frame.fourcc = cricket::FOURCC_I420;
-    captured_frame.time_stamp = 5678;
+    captured_frame.time_stamp = rtc::TimeNanos();
     captured_frame.rotation = frame_rotation;
     captured_frame.width = frame_width;
     captured_frame.height = frame_height;
@@ -66,11 +67,12 @@ class WebRtcVideoFrameTest : public VideoFrameTest<cricket::WebRtcVideoFrame> {
                    apply_rotation));
 
     // Verify the new frame.
-    EXPECT_EQ(5678, frame.GetTimeStamp());
+    EXPECT_EQ(captured_frame.time_stamp / rtc::kNumNanosecsPerMicrosec,
+              frame.timestamp_us());
     if (apply_rotation)
-      EXPECT_EQ(webrtc::kVideoRotation_0, frame.GetVideoRotation());
+      EXPECT_EQ(webrtc::kVideoRotation_0, frame.rotation());
     else
-      EXPECT_EQ(frame_rotation, frame.GetVideoRotation());
+      EXPECT_EQ(frame_rotation, frame.rotation());
     // If |apply_rotation| and the frame rotation is 90 or 270, width and
     // height are flipped.
     if (apply_rotation && (frame_rotation == webrtc::kVideoRotation_90
@@ -271,13 +273,16 @@ TEST_F(WebRtcVideoFrameTest, TextureInitialValues) {
   webrtc::NativeHandleBuffer* buffer =
       new rtc::RefCountedObject<webrtc::test::FakeNativeHandleBuffer>(
           dummy_handle, 640, 480);
-  cricket::WebRtcVideoFrame frame(buffer, 200, webrtc::kVideoRotation_0);
+  // Timestamp is converted from ns to us, so last three digits are lost.
+  cricket::WebRtcVideoFrame frame(buffer, 20000, webrtc::kVideoRotation_0);
   EXPECT_EQ(dummy_handle, frame.GetNativeHandle());
   EXPECT_EQ(640, frame.width());
   EXPECT_EQ(480, frame.height());
-  EXPECT_EQ(200, frame.GetTimeStamp());
-  frame.SetTimeStamp(400);
-  EXPECT_EQ(400, frame.GetTimeStamp());
+  EXPECT_EQ(20000, frame.GetTimeStamp());
+  EXPECT_EQ(20, frame.timestamp_us());
+  frame.set_timestamp_us(40);
+  EXPECT_EQ(40000, frame.GetTimeStamp());
+  EXPECT_EQ(40, frame.timestamp_us());
 }
 
 TEST_F(WebRtcVideoFrameTest, CopyTextureFrame) {
@@ -286,12 +291,14 @@ TEST_F(WebRtcVideoFrameTest, CopyTextureFrame) {
   webrtc::NativeHandleBuffer* buffer =
       new rtc::RefCountedObject<webrtc::test::FakeNativeHandleBuffer>(
           dummy_handle, 640, 480);
-  cricket::WebRtcVideoFrame frame1(buffer, 200, webrtc::kVideoRotation_0);
+  // Timestamp is converted from ns to us, so last three digits are lost.
+  cricket::WebRtcVideoFrame frame1(buffer, 20000, webrtc::kVideoRotation_0);
   cricket::VideoFrame* frame2 = frame1.Copy();
   EXPECT_EQ(frame1.GetNativeHandle(), frame2->GetNativeHandle());
   EXPECT_EQ(frame1.width(), frame2->width());
   EXPECT_EQ(frame1.height(), frame2->height());
   EXPECT_EQ(frame1.GetTimeStamp(), frame2->GetTimeStamp());
+  EXPECT_EQ(frame1.timestamp_us(), frame2->timestamp_us());
   delete frame2;
 }
 
@@ -303,22 +310,22 @@ TEST_F(WebRtcVideoFrameTest, ApplyRotationToFrame) {
       LoadFrame(ms.get(), cricket::FOURCC_I420, kWidth, kHeight, &applied0));
 
   // Claim that this frame needs to be rotated for 90 degree.
-  applied0.SetRotation(webrtc::kVideoRotation_90);
+  applied0.set_rotation(webrtc::kVideoRotation_90);
 
   // Apply rotation on frame 1. Output should be different from frame 1.
   WebRtcVideoTestFrame* applied90 = const_cast<WebRtcVideoTestFrame*>(
       static_cast<const WebRtcVideoTestFrame*>(
           applied0.GetCopyWithRotationApplied()));
   EXPECT_TRUE(applied90);
-  EXPECT_EQ(applied90->GetVideoRotation(), webrtc::kVideoRotation_0);
+  EXPECT_EQ(applied90->rotation(), webrtc::kVideoRotation_0);
   EXPECT_FALSE(IsEqual(applied0, *applied90, 0));
 
   // Claim the frame 2 needs to be rotated for another 270 degree. The output
   // from frame 2 rotation should be the same as frame 1.
-  applied90->SetRotation(webrtc::kVideoRotation_270);
+  applied90->set_rotation(webrtc::kVideoRotation_270);
   const cricket::VideoFrame* applied360 =
       applied90->GetCopyWithRotationApplied();
   EXPECT_TRUE(applied360);
-  EXPECT_EQ(applied360->GetVideoRotation(), webrtc::kVideoRotation_0);
+  EXPECT_EQ(applied360->rotation(), webrtc::kVideoRotation_0);
   EXPECT_TRUE(IsEqual(applied0, *applied360, 0));
 }
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvideoframefactory_unittest.cc b/chromium/third_party/webrtc/media/engine/webrtcvideoframefactory_unittest.cc
index e2630734cae..197784b54a3 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvideoframefactory_unittest.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvideoframefactory_unittest.cc
@@ -12,12 +12,12 @@
 
 #include <memory>
 
-#include "webrtc/media/base/videoframe_unittest.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/engine/webrtcvideoframe.h"
 #include "webrtc/media/engine/webrtcvideoframefactory.h"
 
-class WebRtcVideoFrameFactoryTest
-    : public VideoFrameTest<cricket::WebRtcVideoFrameFactory> {
+class WebRtcVideoFrameFactoryTest : public testing::Test {
  public:
   WebRtcVideoFrameFactoryTest() {}
 
@@ -29,7 +29,7 @@ class WebRtcVideoFrameFactoryTest
     captured_frame_.fourcc = cricket::FOURCC_I420;
     captured_frame_.pixel_width = 1;
     captured_frame_.pixel_height = 1;
-    captured_frame_.time_stamp = 5678;
+    captured_frame_.time_stamp = rtc::TimeNanos();
     captured_frame_.rotation = frame_rotation;
     captured_frame_.width = frame_width;
     captured_frame_.height = frame_height;
@@ -48,11 +48,11 @@ class WebRtcVideoFrameFactoryTest
                    int src_height,
                    bool apply_rotation) {
     if (!apply_rotation) {
-      EXPECT_EQ(dest_frame->GetVideoRotation(), src_rotation);
+      EXPECT_EQ(dest_frame->rotation(), src_rotation);
       EXPECT_EQ(dest_frame->width(), src_width);
       EXPECT_EQ(dest_frame->height(), src_height);
     } else {
-      EXPECT_EQ(dest_frame->GetVideoRotation(), webrtc::kVideoRotation_0);
+      EXPECT_EQ(dest_frame->rotation(), webrtc::kVideoRotation_0);
       if (src_rotation == webrtc::kVideoRotation_90 ||
           src_rotation == webrtc::kVideoRotation_270) {
         EXPECT_EQ(dest_frame->width(), src_height);
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvoe.h b/chromium/third_party/webrtc/media/engine/webrtcvoe.h
index 2f7b997dae5..238b4ce7817 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvoe.h
+++ b/chromium/third_party/webrtc/media/engine/webrtcvoe.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVOE_H_
 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOE_H_
 
+#include <memory>
+
 #include "webrtc/base/common.h"
 #include "webrtc/media/engine/webrtccommon.h"
 
@@ -74,15 +76,13 @@ class VoEWrapper {
  public:
   VoEWrapper()
       : engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
-        base_(engine_), codec_(engine_),
-        hw_(engine_), network_(engine_),
-        rtp_(engine_), volume_(engine_) {
+        base_(engine_), codec_(engine_), hw_(engine_), rtp_(engine_),
+        volume_(engine_) {
   }
   VoEWrapper(webrtc::VoEAudioProcessing* processing,
              webrtc::VoEBase* base,
              webrtc::VoECodec* codec,
              webrtc::VoEHardware* hw,
-             webrtc::VoENetwork* network,
              webrtc::VoERTP_RTCP* rtp,
              webrtc::VoEVolumeControl* volume)
       : engine_(NULL),
@@ -90,7 +90,6 @@ class VoEWrapper {
         base_(base),
         codec_(codec),
         hw_(hw),
-        network_(network),
         rtp_(rtp),
         volume_(volume) {
   }
@@ -100,7 +99,6 @@ class VoEWrapper {
   webrtc::VoEBase* base() const { return base_.get(); }
   webrtc::VoECodec* codec() const { return codec_.get(); }
   webrtc::VoEHardware* hw() const { return hw_.get(); }
-  webrtc::VoENetwork* network() const { return network_.get(); }
   webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
   webrtc::VoEVolumeControl* volume() const { return volume_.get(); }
   int error() { return base_->LastError(); }
@@ -111,7 +109,6 @@ class VoEWrapper {
   scoped_voe_ptr<webrtc::VoEBase> base_;
   scoped_voe_ptr<webrtc::VoECodec> codec_;
   scoped_voe_ptr<webrtc::VoEHardware> hw_;
-  scoped_voe_ptr<webrtc::VoENetwork> network_;
   scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
   scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
 };
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.cc b/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.cc
index b899470b970..09a072a5f3d 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.cc
@@ -22,6 +22,7 @@
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
@@ -29,7 +30,6 @@
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
-#include "webrtc/media/base/audioframe.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
@@ -84,6 +84,9 @@ const int kOpusBitrateFb = 32000;
 const int kOpusMinBitrate = 6000;
 const int kOpusMaxBitrate = 510000;
 
+// iSAC bitrate should be <= 56000.
+const int kIsacMaxBitrate = 56000;
+
 // Default audio dscp value.
 // See http://tools.ietf.org/html/rfc2474 for details.
 // See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
@@ -250,31 +253,28 @@ class WebRtcVoiceCodecs final {
   // list and add a test which verifies VoE supports the listed codecs.
   static std::vector<AudioCodec> SupportedCodecs() {
     std::vector<AudioCodec> result;
-    for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
-      // Change the sample rate of G722 to 8000 to match SDP.
-      MaybeFixupG722(&voe_codec, 8000);
-      // Skip uncompressed formats.
-      if (IsCodec(voe_codec, kL16CodecName)) {
-        continue;
-      }
+    // Iterate first over our preferred codecs list, so that the results are
+    // added in order of preference.
+    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
+      const CodecPref* pref = &kCodecPrefs[i];
+      for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
+        // Change the sample rate of G722 to 8000 to match SDP.
+        MaybeFixupG722(&voe_codec, 8000);
+        // Skip uncompressed formats.
+        if (IsCodec(voe_codec, kL16CodecName)) {
+          continue;
+        }
 
-      const CodecPref* pref = NULL;
-      for (size_t j = 0; j < arraysize(kCodecPrefs); ++j) {
-        if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
-            kCodecPrefs[j].clockrate == voe_codec.plfreq &&
-            kCodecPrefs[j].channels == voe_codec.channels) {
-          pref = &kCodecPrefs[j];
-          break;
+        if (!IsCodec(voe_codec, pref->name) ||
+            pref->clockrate != voe_codec.plfreq ||
+            pref->channels != voe_codec.channels) {
+          // Not a match.
+          continue;
         }
-      }
 
-      if (pref) {
-        // Use the payload type that we've configured in our pref table;
-        // use the offset in our pref table to determine the sort order.
-        AudioCodec codec(
-            pref->payload_type, voe_codec.plname, voe_codec.plfreq,
-            voe_codec.rate, voe_codec.channels,
-            static_cast<int>(arraysize(kCodecPrefs)) - (pref - kCodecPrefs));
+        AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
+                         voe_codec.rate, voe_codec.channels);
+        LOG(LS_INFO) << "Adding supported codec: " << ToString(codec);
         if (IsCodec(codec, kIsacCodecName)) {
           // Indicate auto-bitrate in signaling.
           codec.bitrate = 0;
@@ -297,12 +297,8 @@ class WebRtcVoiceCodecs final {
           // when they can be set to values other than the default.
         }
         result.push_back(codec);
-      } else {
-        LOG(LS_INFO) << "[Unused] " << ToString(voe_codec);
       }
     }
-    // Make sure they are in local preference order.
-    std::sort(result.begin(), result.end(), &AudioCodec::Preferable);
     return result;
   }
 
@@ -312,7 +308,7 @@ class WebRtcVoiceCodecs final {
       // Change the sample rate of G722 to 8000 to match SDP.
       MaybeFixupG722(&voe_codec, 8000);
       AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
-                       voe_codec.rate, voe_codec.channels, 0);
+                       voe_codec.rate, voe_codec.channels);
       bool multi_rate = IsCodecMultiRate(voe_codec);
       // Allow arbitrary rates for ISAC to be specified.
       if (multi_rate) {
@@ -356,6 +352,16 @@ class WebRtcVoiceCodecs final {
     return false;
   }
 
+  static int MaxBitrateBps(const webrtc::CodecInst& codec) {
+    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
+      if (IsCodec(codec, kCodecPrefs[i].name) &&
+          kCodecPrefs[i].clockrate == codec.plfreq) {
+        return kCodecPrefs[i].max_bitrate_bps;
+      }
+    }
+    return 0;
+  }
+
   // If the AudioCodec param kCodecParamPTime is set, then we will set it to
   // codec pacsize if it's valid, or we will pick the next smallest value we
   // support.
@@ -426,6 +432,7 @@ class WebRtcVoiceCodecs final {
     int payload_type;
     bool is_multi_rate;
     int packet_sizes_ms[kMaxNumPacketSize];
+    int max_bitrate_bps;
   };
   // Note: keep the supported packet sizes in ascending order.
   static const CodecPref kCodecPrefs[12];
@@ -492,19 +499,19 @@ class WebRtcVoiceCodecs final {
 };
 
 const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[12] = {
-  { kOpusCodecName,   48000, 2, 111, true,  { 10, 20, 40, 60 } },
-  { kIsacCodecName,   16000, 1, 103, true,  { 30, 60 } },
-  { kIsacCodecName,   32000, 1, 104, true,  { 30 } },
-  // G722 should be advertised as 8000 Hz because of the RFC "bug".
-  { kG722CodecName,   8000,  1, 9,   false, { 10, 20, 30, 40, 50, 60 } },
-  { kIlbcCodecName,   8000,  1, 102, false, { 20, 30, 40, 60 } },
-  { kPcmuCodecName,   8000,  1, 0,   false, { 10, 20, 30, 40, 50, 60 } },
-  { kPcmaCodecName,   8000,  1, 8,   false, { 10, 20, 30, 40, 50, 60 } },
-  { kCnCodecName,     32000, 1, 106, false, { } },
-  { kCnCodecName,     16000, 1, 105, false, { } },
-  { kCnCodecName,     8000,  1, 13,  false, { } },
-  { kRedCodecName,    8000,  1, 127, false, { } },
-  { kDtmfCodecName,   8000,  1, 126, false, { } },
+    {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrate},
+    {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrate},
+    {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrate},
+    // G722 should be advertised as 8000 Hz because of the RFC "bug".
+    {kG722CodecName, 8000, 1, 9, false, {10, 20, 30, 40, 50, 60}},
+    {kIlbcCodecName, 8000, 1, 102, false, {20, 30, 40, 60}},
+    {kPcmuCodecName, 8000, 1, 0, false, {10, 20, 30, 40, 50, 60}},
+    {kPcmaCodecName, 8000, 1, 8, false, {10, 20, 30, 40, 50, 60}},
+    {kCnCodecName, 32000, 1, 106, false, {}},
+    {kCnCodecName, 16000, 1, 105, false, {}},
+    {kCnCodecName, 8000, 1, 13, false, {}},
+    {kRedCodecName, 8000, 1, 127, false, {}},
+    {kDtmfCodecName, 8000, 1, 126, false, {}},
 };
 } // namespace {
 
@@ -570,6 +577,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(webrtc::AudioDeviceModule* adm,
     options.extended_filter_aec = rtc::Optional<bool>(false);
     options.delay_agnostic_aec = rtc::Optional<bool>(false);
     options.experimental_ns = rtc::Optional<bool>(false);
+    options.intelligibility_enhancer = rtc::Optional<bool>(false);
     bool error = ApplyOptions(options);
     RTC_DCHECK(error);
   }
@@ -655,14 +663,14 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     // Android and in combination with Java based audio layer.
     // TODO(henrika): investigate possibility to support built-in EC also
     // in combination with Open SL ES audio.
-    const bool built_in_aec = voe_wrapper_->hw()->BuiltInAECIsAvailable();
+    const bool built_in_aec = adm()->BuiltInAECIsAvailable();
     if (built_in_aec) {
       // Built-in EC exists on this device and use_delay_agnostic_aec is not
       // overriding it. Enable/Disable it according to the echo_cancellation
       // audio option.
       const bool enable_built_in_aec =
           *options.echo_cancellation && !use_delay_agnostic_aec;
-      if (voe_wrapper_->hw()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
+      if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
           enable_built_in_aec) {
         // Disable internal software EC if built-in EC is enabled,
         // i.e., replace the software EC with the built-in EC.
@@ -694,10 +702,9 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
   }
 
   if (options.auto_gain_control) {
-    const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable();
+    const bool built_in_agc = adm()->BuiltInAGCIsAvailable();
     if (built_in_agc) {
-      if (voe_wrapper_->hw()->EnableBuiltInAGC(*options.auto_gain_control) ==
-              0 &&
+      if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
           *options.auto_gain_control) {
         // Disable internal software AGC if built-in AGC is enabled,
         // i.e., replace the software AGC with the built-in AGC.
@@ -740,12 +747,20 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     }
   }
 
+  if (options.intelligibility_enhancer) {
+    intelligibility_enhancer_ = options.intelligibility_enhancer;
+  }
+  if (intelligibility_enhancer_ && *intelligibility_enhancer_) {
+    LOG(LS_INFO) << "Enabling NS when Intelligibility Enhancer is active.";
+    options.noise_suppression = intelligibility_enhancer_;
+  }
+
   if (options.noise_suppression) {
-    const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable();
-    if (built_in_ns) {
-      if (voe_wrapper_->hw()->EnableBuiltInNS(*options.noise_suppression) ==
-              0 &&
-          *options.noise_suppression) {
+    if (adm()->BuiltInNSIsAvailable()) {
+      bool builtin_ns =
+          *options.noise_suppression &&
+          !(intelligibility_enhancer_ && *intelligibility_enhancer_);
+      if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) {
         // Disable internal software NS if built-in NS is enabled,
         // i.e., replace the software NS with the built-in NS.
         options.noise_suppression = rtc::Optional<bool>(false);
@@ -838,6 +853,13 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
         new webrtc::ExperimentalNs(*experimental_ns_));
   }
 
+  if (intelligibility_enhancer_) {
+    LOG(LS_INFO) << "Intelligibility Enhancer is enabled? "
+                 << *intelligibility_enhancer_;
+    config.Set<webrtc::Intelligibility>(
+        new webrtc::Intelligibility(*intelligibility_enhancer_));
+  }
+
   // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
   // returns NULL on audio_processing().
   webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
@@ -848,16 +870,14 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
   if (options.recording_sample_rate) {
     LOG(LS_INFO) << "Recording sample rate is "
                  << *options.recording_sample_rate;
-    if (voe_wrapper_->hw()->SetRecordingSampleRate(
-            *options.recording_sample_rate)) {
+    if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
       LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
     }
   }
 
   if (options.playout_sample_rate) {
     LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
-    if (voe_wrapper_->hw()->SetPlayoutSampleRate(
-            *options.playout_sample_rate)) {
+    if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) {
       LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
     }
   }
@@ -1053,11 +1073,12 @@ void WebRtcVoiceEngine::StopAecDump() {
   }
 }
 
-bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
+bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file,
+                                         int64_t max_size_bytes) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
   if (event_log) {
-    return event_log->StartLogging(file);
+    return event_log->StartLogging(file, max_size_bytes);
   }
   LOG_RTCERR0(StartRtcEventLog);
   return false;
@@ -1078,16 +1099,26 @@ int WebRtcVoiceEngine::CreateVoEChannel() {
   return voe_wrapper_->base()->CreateChannel(voe_config_);
 }
 
+webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(adm_);
+  return adm_;
+}
+
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioSource::Sink {
  public:
-  WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
-                        uint32_t ssrc, const std::string& c_name,
+  WebRtcAudioSendStream(int ch,
+                        webrtc::AudioTransport* voe_audio_transport,
+                        uint32_t ssrc,
+                        const std::string& c_name,
                         const std::vector<webrtc::RtpExtension>& extensions,
-                        webrtc::Call* call)
+                        webrtc::Call* call,
+                        webrtc::Transport* send_transport)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
-        config_(nullptr) {
+        config_(send_transport),
+        rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
     RTC_DCHECK_GE(ch, 0);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
@@ -1199,11 +1230,23 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     return config_.voe_channel_id;
   }
 
+  const webrtc::RtpParameters& rtp_parameters() const {
+    return rtp_parameters_;
+  }
+
+  void SetRtpParameters(const webrtc::RtpParameters& parameters) {
+    RTC_CHECK_EQ(1UL, parameters.encodings.size());
+    rtp_parameters_ = parameters;
+    // parameters.encodings[0].active could have changed.
+    UpdateSendState();
+  }
+
  private:
   void UpdateSendState() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
-    if (send_ && source_ != nullptr) {
+    RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size());
+    if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
       stream_->Start();
     } else {  // !send || source_ = nullptr
       stream_->Stop();
@@ -1224,6 +1267,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
   // goes away.
   AudioSource* source_ = nullptr;
   bool send_ = false;
+  webrtc::RtpParameters rtp_parameters_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
 };
@@ -1236,12 +1280,14 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
                            bool use_transport_cc,
                            const std::string& sync_group,
                            const std::vector<webrtc::RtpExtension>& extensions,
-                           webrtc::Call* call)
+                           webrtc::Call* call,
+                           webrtc::Transport* rtcp_send_transport)
       : call_(call), config_() {
     RTC_DCHECK_GE(ch, 0);
     RTC_DCHECK(call);
     config_.rtp.remote_ssrc = remote_ssrc;
     config_.rtp.local_ssrc = local_ssrc;
+    config_.rtcp_send_transport = rtcp_send_transport;
     config_.voe_channel_id = ch;
     config_.sync_group = sync_group;
     RecreateAudioReceiveStream(use_transport_cc, extensions);
@@ -1360,7 +1406,7 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
     }
   }
 
-  if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
+  if (!SetMaxSendBitrate(params.max_bandwidth_bps)) {
     return false;
   }
   return SetOptions(params.options);
@@ -1394,6 +1440,110 @@ bool WebRtcVoiceMediaChannel::SetRecvParameters(
   return true;
 }
 
+webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters(
+    uint32_t ssrc) const {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  auto it = send_streams_.find(ssrc);
+  if (it == send_streams_.end()) {
+    LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
+                    << "with ssrc " << ssrc << " which doesn't exist.";
+    return webrtc::RtpParameters();
+  }
+
+  webrtc::RtpParameters rtp_params = it->second->rtp_parameters();
+  // Need to add the common list of codecs to the send stream-specific
+  // RTP parameters.
+  for (const AudioCodec& codec : send_codecs_) {
+    rtp_params.codecs.push_back(codec.ToCodecParameters());
+  }
+  return rtp_params;
+}
+
+bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  if (!ValidateRtpParameters(parameters)) {
+    return false;
+  }
+  auto it = send_streams_.find(ssrc);
+  if (it == send_streams_.end()) {
+    LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
+                    << "with ssrc " << ssrc << " which doesn't exist.";
+    return false;
+  }
+
+  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
+  // different order (which should change the send codec).
+  webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
+  if (current_parameters.codecs != parameters.codecs) {
+    LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
+                  << "is not currently supported.";
+    return false;
+  }
+
+  if (!SetChannelSendParameters(it->second->channel(), parameters)) {
+    LOG(LS_WARNING) << "Failed to set send RtpParameters.";
+    return false;
+  }
+  // Codecs are handled at the WebRtcVoiceMediaChannel level.
+  webrtc::RtpParameters reduced_params = parameters;
+  reduced_params.codecs.clear();
+  it->second->SetRtpParameters(reduced_params);
+  return true;
+}
+
+webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
+    uint32_t ssrc) const {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  auto it = recv_streams_.find(ssrc);
+  if (it == recv_streams_.end()) {
+    LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
+                    << "with ssrc " << ssrc << " which doesn't exist.";
+    return webrtc::RtpParameters();
+  }
+
+  // TODO(deadbeef): Return stream-specific parameters.
+  webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
+  for (const AudioCodec& codec : recv_codecs_) {
+    rtp_params.codecs.push_back(codec.ToCodecParameters());
+  }
+  return rtp_params;
+}
+
+bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  if (!ValidateRtpParameters(parameters)) {
+    return false;
+  }
+  auto it = recv_streams_.find(ssrc);
+  if (it == recv_streams_.end()) {
+    LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
+                    << "with ssrc " << ssrc << " which doesn't exist.";
+    return false;
+  }
+
+  webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
+  if (current_parameters != parameters) {
+    LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
+                  << "unsupported.";
+    return false;
+  }
+  return true;
+}
+
+bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
+    const webrtc::RtpParameters& rtp_parameters) {
+  if (rtp_parameters.encodings.size() != 1) {
+    LOG(LS_ERROR)
+        << "Attempted to set RtpParameters without exactly one encoding";
+    return false;
+  }
+  return true;
+}
+
 bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "Setting voice channel options: "
@@ -1588,16 +1738,14 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
 
   // Cache the codecs in order to configure the channel created later.
   for (const auto& ch : send_streams_) {
-    if (!SetSendCodecs(ch.second->channel())) {
+    if (!SetSendCodecs(ch.second->channel(), ch.second->rtp_parameters())) {
       return false;
     }
   }
 
   // Set nack status on receive channels.
-  if (!send_streams_.empty()) {
-    for (const auto& kv : recv_streams_) {
-      SetNack(kv.second->channel(), send_codec_spec_.nack_enabled);
-    }
+  for (const auto& kv : recv_streams_) {
+    SetNack(kv.second->channel(), send_codec_spec_.nack_enabled);
   }
 
   // Check if the transport cc feedback has changed on the preferred send codec,
@@ -1611,11 +1759,14 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
     }
   }
 
+  send_codecs_ = codecs;
   return true;
 }
 
 // Apply current codec settings to a single voe::Channel used for sending.
-bool WebRtcVoiceMediaChannel::SetSendCodecs(int channel) {
+bool WebRtcVoiceMediaChannel::SetSendCodecs(
+    int channel,
+    const webrtc::RtpParameters& rtp_parameters) {
   // Disable VAD, FEC, and RED unless we know the other side wants them.
   engine()->voe()->codec()->SetVADStatus(channel, false);
   engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
@@ -1683,10 +1834,9 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(int channel) {
       }
     }
   }
-
-  if (send_bitrate_setting_) {
-    SetSendBitrateInternal(send_bitrate_bps_);
-  }
+  // TODO(solenberg): SetMaxSendBitrate() yields another call to SetSendCodec().
+  // Check if it is possible to fuse with the previous call in this function.
+  SetChannelSendParameters(channel, rtp_parameters);
 
   // Set the CN payloadtype and the VAD status.
   if (send_codec_spec_.cng_payload_type != -1) {
@@ -1801,9 +1951,18 @@ void WebRtcVoiceMediaChannel::SetSend(bool send) {
     return;
   }
 
-  // Apply channel specific options when channel is enabled for sending.
+  // Apply channel specific options, and initialize the ADM for recording (this
+  // may take time on some platforms, e.g. Android).
   if (send) {
     engine()->ApplyOptions(options_);
+
+    // InitRecording() may return an error if the ADM is already recording.
+    if (!engine()->adm()->RecordingIsInitialized() &&
+        !engine()->adm()->Recording()) {
+      if (engine()->adm()->InitRecording() != 0) {
+        LOG(LS_WARNING) << "Failed to initialize recording";
+      }
+    }
   }
 
   // Change the settings on each send channel.
@@ -1839,18 +1998,11 @@ int WebRtcVoiceMediaChannel::CreateVoEChannel() {
     LOG_RTCERR0(CreateVoEChannel);
     return -1;
   }
-  if (engine()->voe()->network()->RegisterExternalTransport(id, *this) == -1) {
-    LOG_RTCERR2(RegisterExternalTransport, id, this);
-    engine()->voe()->base()->DeleteChannel(id);
-    return -1;
-  }
+
   return id;
 }
 
 bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) {
-  if (engine()->voe()->network()->DeRegisterExternalTransport(channel) == -1) {
-    LOG_RTCERR1(DeRegisterExternalTransport, channel);
-  }
   if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
     LOG_RTCERR1(DeleteChannel, channel);
     return false;
@@ -1881,13 +2033,16 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
   // delete the channel in case failure happens below.
   webrtc::AudioTransport* audio_transport =
       engine()->voe()->base()->audio_transport();
-  send_streams_.insert(std::make_pair(ssrc, new WebRtcAudioSendStream(
-      channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_)));
+
+  WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
+      channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_,
+      this);
+  send_streams_.insert(std::make_pair(ssrc, stream));
 
   // Set the current codecs to be used for the new channel. We need to do this
   // after adding the channel to send_channels_, because of how max bitrate is
   // currently being configured by SetSendCodec().
-  if (HasSendCodec() && !SetSendCodecs(channel)) {
+  if (HasSendCodec() && !SetSendCodecs(channel, stream->rtp_parameters())) {
     RemoveSendStream(ssrc);
     return false;
   }
@@ -2012,7 +2167,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
       ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
                                          recv_transport_cc_enabled_,
                                          sp.sync_label, recv_rtp_extensions_,
-                                         call_)));
+                                         call_, this)));
 
   SetNack(channel, send_codec_spec_.nack_enabled);
   SetPlayout(channel, playout_);
@@ -2178,54 +2333,51 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
     rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 
+  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+                                              packet_time.not_before);
+  webrtc::PacketReceiver::DeliveryStatus delivery_result =
+      call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
+                                       packet->cdata(), packet->size(),
+                                       webrtc_packet_time);
+  if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
+    return;
+  }
+
+  // Create a default receive stream for this unsignalled and previously not
+  // received ssrc. If there already is a default receive stream, delete it.
+  // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
   uint32_t ssrc = 0;
   if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
     return;
   }
 
-  // If we don't have a default channel, and the SSRC is unknown, create a
-  // default channel.
-  if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) {
-    StreamParams sp;
-    sp.ssrcs.push_back(ssrc);
-    LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
-    if (!AddRecvStream(sp)) {
-      LOG(LS_WARNING) << "Could not create default receive stream.";
-      return;
-    }
-    default_recv_ssrc_ = ssrc;
-    SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
-    if (default_sink_) {
-      std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
-          new ProxySink(default_sink_.get()));
-      SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
-    }
+  if (default_recv_ssrc_ != -1) {
+    LOG(LS_INFO) << "Removing default receive stream with ssrc "
+                 << default_recv_ssrc_;
+    RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
+    RemoveRecvStream(default_recv_ssrc_);
+    default_recv_ssrc_ = -1;
   }
 
-  // Forward packet to Call. If the SSRC is unknown we'll return after this.
-  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
-                                              packet_time.not_before);
-  webrtc::PacketReceiver::DeliveryStatus delivery_result =
-      call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
-          packet->cdata(), packet->size(), webrtc_packet_time);
-  if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) {
-    // If the SSRC is unknown here, route it to the default channel, if we have
-    // one. See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
-    if (default_recv_ssrc_ == -1) {
-      return;
-    } else {
-      ssrc = default_recv_ssrc_;
-    }
+  StreamParams sp;
+  sp.ssrcs.push_back(ssrc);
+  LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
+  if (!AddRecvStream(sp)) {
+    LOG(LS_WARNING) << "Could not create default receive stream.";
+    return;
   }
-
-  // Find the channel to send this packet to. It must exist since webrtc::Call
-  // was able to demux the packet.
-  int channel = GetReceiveChannelId(ssrc);
-  RTC_DCHECK(channel != -1);
-
-  // Pass it off to the decoder.
-  engine()->voe()->network()->ReceivedRTPPacket(
-      channel, packet->cdata(), packet->size(), webrtc_packet_time);
+  default_recv_ssrc_ = ssrc;
+  SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
+  if (default_sink_) {
+    std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
+        new ProxySink(default_sink_.get()));
+    SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
+  }
+  delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
+                                                     packet->cdata(),
+                                                     packet->size(),
+                                                     webrtc_packet_time);
+  RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
 }
 
 void WebRtcVoiceMediaChannel::OnRtcpReceived(
@@ -2237,44 +2389,12 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived(
                                               packet_time.not_before);
   call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
       packet->cdata(), packet->size(), webrtc_packet_time);
-
-  // Sending channels need all RTCP packets with feedback information.
-  // Even sender reports can contain attached report blocks.
-  // Receiving channels need sender reports in order to create
-  // correct receiver reports.
-  int type = 0;
-  if (!GetRtcpType(packet->cdata(), packet->size(), &type)) {
-    LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
-    return;
-  }
-
-  // If it is a sender report, find the receive channel that is listening.
-  if (type == kRtcpTypeSR) {
-    uint32_t ssrc = 0;
-    if (!GetRtcpSsrc(packet->cdata(), packet->size(), &ssrc)) {
-      return;
-    }
-    int recv_channel_id = GetReceiveChannelId(ssrc);
-    if (recv_channel_id != -1) {
-      engine()->voe()->network()->ReceivedRTCPPacket(
-          recv_channel_id, packet->cdata(), packet->size());
-    }
-  }
-
-  // SR may continue RR and any RR entry may correspond to any one of the send
-  // channels. So all RTCP packets must be forwarded all send channels. VoE
-  // will filter out RR internally.
-  for (const auto& ch : send_streams_) {
-    engine()->voe()->network()->ReceivedRTCPPacket(
-        ch.second->channel(), packet->cdata(), packet->size());
-  }
 }
 
 void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
     const std::string& transport_name,
-    const NetworkRoute& network_route) {
-  // TODO(honghaiz): uncomment this once the function in call is implemented.
-  // call_->OnNetworkRouteChanged(transport_name, network_route);
+    const rtc::NetworkRoute& network_route) {
+  call_->OnNetworkRouteChanged(transport_name, network_route);
 }
 
 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
@@ -2311,18 +2431,37 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
   return true;
 }
 
-// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
-// SetMaxSendBitrate() in future.
-bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
-  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
-  return SetSendBitrateInternal(bps);
+bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) {
+  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBitrate.";
+  max_send_bitrate_bps_ = bps;
+
+  for (const auto& kv : send_streams_) {
+    if (!SetChannelSendParameters(kv.second->channel(),
+                                  kv.second->rtp_parameters())) {
+      return false;
+    }
+  }
+  return true;
 }
 
-bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
-  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
+bool WebRtcVoiceMediaChannel::SetChannelSendParameters(
+    int channel,
+    const webrtc::RtpParameters& parameters) {
+  RTC_CHECK_EQ(1UL, parameters.encodings.size());
+  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
+  // different order (which should change the send codec).
+  return SetMaxSendBitrate(
+      channel, MinPositive(max_send_bitrate_bps_,
+                           parameters.encodings[0].max_bitrate_bps));
+}
 
-  send_bitrate_setting_ = true;
-  send_bitrate_bps_ = bps;
+bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int channel, int bps) {
+  // Bitrate is auto by default.
+  // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
+  // SetMaxSendBandwith(0), the second call removes the previous limit.
+  if (bps <= 0) {
+    return true;
+  }
 
   if (!HasSendCodec()) {
     LOG(LS_INFO) << "The send codec has not been set up yet. "
@@ -2330,24 +2469,19 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
     return true;
   }
 
-  // Bitrate is auto by default.
-  // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
-  // SetMaxSendBandwith(0), the second call removes the previous limit.
-  if (bps <= 0)
-    return true;
-
   webrtc::CodecInst codec = send_codec_spec_.codec_inst;
   bool is_multi_rate = WebRtcVoiceCodecs::IsCodecMultiRate(codec);
 
   if (is_multi_rate) {
     // If codec is multi-rate then just set the bitrate.
-    codec.rate = bps;
-    for (const auto& ch : send_streams_) {
-      if (!SetSendCodec(ch.second->channel(), codec)) {
-        LOG(LS_INFO) << "Failed to set codec " << codec.plname
-                     << " to bitrate " << bps << " bps.";
-        return false;
-      }
+    int max_bitrate_bps = WebRtcVoiceCodecs::MaxBitrateBps(codec);
+    codec.rate = std::min(bps, max_bitrate_bps);
+    LOG(LS_INFO) << "Setting codec " << codec.plname << " to bitrate " << bps
+                 << " bps.";
+    if (!SetSendCodec(channel, codec)) {
+      LOG(LS_ERROR) << "Failed to set codec " << codec.plname << " to bitrate "
+                    << bps << " bps.";
+      return false;
     }
     return true;
   } else {
@@ -2355,9 +2489,9 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
     // then fail. If codec is not multi-rate and |bps| exceeds or equal the
     // fixed bitrate then ignore.
     if (bps < codec.rate) {
-      LOG(LS_INFO) << "Failed to set codec " << codec.plname
-                   << " to bitrate " << bps << " bps"
-                   << ", requires at least " << codec.rate << " bps.";
+      LOG(LS_ERROR) << "Failed to set codec " << codec.plname << " to bitrate "
+                    << bps << " bps"
+                    << ", requires at least " << codec.rate << " bps.";
       return false;
     }
     return true;
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.h b/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.h
index 396e3bbb8c0..5b435348e65 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.h
+++ b/chromium/third_party/webrtc/media/engine/webrtcvoiceengine.h
@@ -18,6 +18,7 @@
 
 #include "webrtc/audio_state.h"
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/stream.h"
@@ -84,9 +85,10 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   // Stops AEC dump.
   void StopAecDump();
 
-  // Starts recording an RtcEventLog using an existing file until 10 minutes
-  // pass or the StopRtcEventLog function is called.
-  bool StartRtcEventLog(rtc::PlatformFile file);
+  // Starts recording an RtcEventLog using an existing file until the log file
+  // reaches the maximum filesize or the StopRtcEventLog function is called.
+  // If the value of max_size_bytes is <= 0, no limit is used.
+  bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops recording the RtcEventLog.
   void StopRtcEventLog();
@@ -103,6 +105,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
+  webrtc::AudioDeviceModule* adm();
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
@@ -118,13 +121,14 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   bool is_dumping_aec_ = false;
 
   webrtc::AgcConfig default_agc_config_;
-  // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
-  // values, and apply them in case they are missing in the audio options. We
-  // need to do this because SetExtraOptions() will revert to defaults for
-  // options which are not provided.
+  // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns and
+  // intelligibility_enhancer values, and apply them in case they are missing
+  // in the audio options. We need to do this because SetExtraOptions() will
+  // revert to defaults for options which are not provided.
   rtc::Optional<bool> extended_filter_aec_;
   rtc::Optional<bool> delay_agnostic_aec_;
   rtc::Optional<bool> experimental_ns_;
+  rtc::Optional<bool> intelligibility_enhancer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
 };
@@ -146,6 +150,14 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
+  bool SetRtpSendParameters(uint32_t ssrc,
+                            const webrtc::RtpParameters& parameters) override;
+  webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
+  bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) override;
+
   bool SetPlayout(bool playout) override;
   bool PausePlayout();
   bool ResumePlayout();
@@ -176,7 +188,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnNetworkRouteChanged(const std::string& transport_name,
-                             const NetworkRoute& network_route) override;
+                             const rtc::NetworkRoute& network_route) override;
   void OnReadyToSend(bool ready) override;
   bool GetStats(VoiceMediaInfo* info) override;
 
@@ -206,10 +218,9 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool SetOptions(const AudioOptions& options);
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
-  bool SetSendCodecs(int channel);
+  bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters);
   void SetNack(int channel, bool nack_enabled);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
-  bool SetMaxSendBandwidth(int bps);
   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
   bool MuteStream(uint32_t ssrc, bool mute);
 
@@ -223,17 +234,22 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
-  bool SetSendBitrateInternal(int bps);
+  bool SetMaxSendBitrate(int bps);
+  bool SetChannelSendParameters(int channel,
+                                const webrtc::RtpParameters& parameters);
+  bool SetMaxSendBitrate(int channel, int bps);
   bool HasSendCodec() const {
     return send_codec_spec_.codec_inst.pltype != -1;
   }
+  bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
+  void SetupRecording();
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
+  std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
-  bool send_bitrate_setting_ = false;
-  int send_bitrate_bps_ = 0;
+  int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;
diff --git a/chromium/third_party/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/chromium/third_party/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index f2f7056a7d0..4446e27c162 100644
--- a/chromium/third_party/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/chromium/third_party/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -33,16 +33,19 @@ using testing::StrictMock;
 
 namespace {
 
-const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0);
-const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0);
-const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2, 0);
-const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 0);
-const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1, 0);
-const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1, 0);
-const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1, 0);
-const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0);
-const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000, 0,
-                                               1, 0);
+const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1);
+const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1);
+const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2);
+const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
+const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
+const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1);
+const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
+const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1);
+const cricket::AudioCodec kTelephoneEventCodec(106,
+                                               "telephone-event",
+                                               8000,
+                                               0,
+                                               1);
 const uint32_t kSsrc1 = 0x99;
 const uint32_t kSsrc2 = 2;
 const uint32_t kSsrc3 = 3;
@@ -55,7 +58,6 @@ class FakeVoEWrapper : public cricket::VoEWrapper {
                             engine,  // base
                             engine,  // codec
                             engine,  // hw
-                            engine,  // network
                             engine,  // rtp
                             engine) {  // volume
   }
@@ -67,6 +69,9 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
   StrictMock<webrtc::test::MockAudioDeviceModule> adm;
   EXPECT_CALL(adm, AddRef()).WillOnce(Return(0));
   EXPECT_CALL(adm, Release()).WillOnce(Return(0));
+  EXPECT_CALL(adm, BuiltInAECIsAvailable()).WillOnce(Return(false));
+  EXPECT_CALL(adm, BuiltInAGCIsAvailable()).WillOnce(Return(false));
+  EXPECT_CALL(adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
   cricket::FakeWebRtcVoiceEngine voe;
   EXPECT_FALSE(voe.IsInited());
   {
@@ -93,6 +98,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
       : call_(webrtc::Call::Config()), override_field_trials_(field_trials) {
     EXPECT_CALL(adm_, AddRef()).WillOnce(Return(0));
     EXPECT_CALL(adm_, Release()).WillOnce(Return(0));
+    EXPECT_CALL(adm_, BuiltInAECIsAvailable()).WillOnce(Return(false));
+    EXPECT_CALL(adm_, BuiltInAGCIsAvailable()).WillOnce(Return(false));
+    EXPECT_CALL(adm_, BuiltInNSIsAvailable()).WillOnce(Return(false));
     engine_.reset(new cricket::WebRtcVoiceEngine(&adm_,
                                                  new FakeVoEWrapper(&voe_)));
     send_parameters_.codecs.push_back(kPcmuCodec);
@@ -155,6 +163,16 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     return GetRecvStream(ssrc).GetConfig();
   }
 
+  void SetSend(cricket::VoiceMediaChannel* channel, bool enable) {
+    ASSERT_TRUE(channel);
+    if (enable) {
+      EXPECT_CALL(adm_, RecordingIsInitialized()).WillOnce(Return(false));
+      EXPECT_CALL(adm_, Recording()).WillOnce(Return(false));
+      EXPECT_CALL(adm_, InitRecording()).WillOnce(Return(0));
+    }
+    channel->SetSend(enable);
+  }
+
   void TestInsertDtmf(uint32_t ssrc, bool caller) {
     EXPECT_TRUE(SetupChannel());
     if (caller) {
@@ -166,7 +184,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 
     // Test we can only InsertDtmf when the other side supports telephone-event.
     EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-    channel_->SetSend(true);
+    SetSend(channel_, true);
     EXPECT_FALSE(channel_->CanInsertDtmf());
     EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
     send_parameters_.codecs.push_back(kTelephoneEventCodec);
@@ -199,10 +217,10 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
   // |max_bitrate| is a parameter to set to SetMaxSendBandwidth().
   // |expected_result| is the expected result from SetMaxSendBandwidth().
   // |expected_bitrate| is the expected audio bitrate afterward.
-  void TestSendBandwidth(const cricket::AudioCodec& codec,
-                         int max_bitrate,
-                         bool expected_result,
-                         int expected_bitrate) {
+  void TestMaxSendBandwidth(const cricket::AudioCodec& codec,
+                            int max_bitrate,
+                            bool expected_result,
+                            int expected_bitrate) {
     cricket::AudioSendParameters parameters;
     parameters.codecs.push_back(codec);
     parameters.max_bandwidth_bps = max_bitrate;
@@ -214,6 +232,56 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     EXPECT_EQ(expected_bitrate, temp_codec.rate);
   }
 
+  // Sets the per-stream maximum bitrate limit for the specified SSRC.
+  bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) {
+    webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrc);
+    EXPECT_EQ(1UL, parameters.encodings.size());
+
+    parameters.encodings[0].max_bitrate_bps = bitrate;
+    return channel_->SetRtpSendParameters(ssrc, parameters);
+  }
+
+  bool SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) {
+    cricket::AudioSendParameters send_parameters;
+    send_parameters.codecs.push_back(codec);
+    send_parameters.max_bandwidth_bps = bitrate;
+    return channel_->SetSendParameters(send_parameters);
+  }
+
+  int GetCodecBitrate(int32_t ssrc) {
+    cricket::WebRtcVoiceMediaChannel* media_channel =
+        static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
+    int channel = media_channel->GetSendChannelId(ssrc);
+    EXPECT_NE(-1, channel);
+    webrtc::CodecInst codec;
+    EXPECT_FALSE(voe_.GetSendCodec(channel, codec));
+    return codec.rate;
+  }
+
+  void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec,
+                              int global_max,
+                              int stream_max,
+                              bool expected_result,
+                              int expected_codec_bitrate) {
+    // Clear the bitrate limit from the previous test case.
+    EXPECT_TRUE(SetMaxBitrateForStream(kSsrc1, -1));
+
+    // Attempt to set the requested bitrate limits.
+    EXPECT_TRUE(SetGlobalMaxBitrate(codec, global_max));
+    EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrc1, stream_max));
+
+    // Verify that reading back the parameters gives results
+    // consistent with the Set() result.
+    webrtc::RtpParameters resulting_parameters =
+        channel_->GetRtpSendParameters(kSsrc1);
+    EXPECT_EQ(1UL, resulting_parameters.encodings.size());
+    EXPECT_EQ(expected_result ? stream_max : -1,
+              resulting_parameters.encodings[0].max_bitrate_bps);
+
+    // Verify that the codec settings have the expected bitrate.
+    EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrc1));
+  }
+
   void TestSetSendRtpHeaderExtensions(const std::string& ext) {
     EXPECT_TRUE(SetupSendStream());
 
@@ -426,20 +494,44 @@ TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) {
   EXPECT_TRUE(SetupChannel());
 }
 
+// Test that we can add a send stream and that it has the correct defaults.
+TEST_F(WebRtcVoiceEngineTestFake, CreateSendStream) {
+  EXPECT_TRUE(SetupChannel());
+  EXPECT_TRUE(
+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
+  const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrc1);
+  EXPECT_EQ(kSsrc1, config.rtp.ssrc);
+  EXPECT_EQ("", config.rtp.c_name);
+  EXPECT_EQ(0u, config.rtp.extensions.size());
+  EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
+            config.send_transport);
+}
+
+// Test that we can add a receive stream and that it has the correct defaults.
+TEST_F(WebRtcVoiceEngineTestFake, CreateRecvStream) {
+  EXPECT_TRUE(SetupChannel());
+  EXPECT_TRUE(
+      channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc1)));
+  const webrtc::AudioReceiveStream::Config& config =
+      GetRecvStreamConfig(kSsrc1);
+  EXPECT_EQ(kSsrc1, config.rtp.remote_ssrc);
+  EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc);
+  EXPECT_FALSE(config.rtp.transport_cc);
+  EXPECT_EQ(0u, config.rtp.extensions.size());
+  EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
+            config.rtcp_send_transport);
+  EXPECT_EQ("", config.sync_group);
+}
+
 // Tests that the list of supported codecs is created properly and ordered
-// correctly
-TEST_F(WebRtcVoiceEngineTestFake, CodecPreference) {
+// correctly (such that opus appears first).
+TEST_F(WebRtcVoiceEngineTestFake, CodecOrder) {
   const std::vector<cricket::AudioCodec>& codecs = engine_->codecs();
   ASSERT_FALSE(codecs.empty());
   EXPECT_STRCASEEQ("opus", codecs[0].name.c_str());
   EXPECT_EQ(48000, codecs[0].clockrate);
   EXPECT_EQ(2, codecs[0].channels);
   EXPECT_EQ(64000, codecs[0].bitrate);
-  int pref = codecs[0].preference;
-  for (size_t i = 1; i < codecs.size(); ++i) {
-    EXPECT_GT(pref, codecs[i].preference);
-    pref = codecs[i].preference;
-  }
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) {
@@ -523,7 +615,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) {
   EXPECT_TRUE(SetupChannel());
   cricket::AudioRecvParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
-  parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1, 0));
+  parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1));
   EXPECT_FALSE(channel_->SetRecvParameters(parameters));
 }
 
@@ -692,13 +784,13 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
   // bitrate is <= 0.
 
   // ISAC, default bitrate == 32000.
-  TestSendBandwidth(kIsacCodec, 0, true, 32000);
+  TestMaxSendBandwidth(kIsacCodec, 0, true, 32000);
 
   // PCMU, default bitrate == 64000.
-  TestSendBandwidth(kPcmuCodec, -1, true, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000);
 
   // opus, default bitrate == 64000.
-  TestSendBandwidth(kOpusCodec, -1, true, 64000);
+  TestMaxSendBandwidth(kOpusCodec, -1, true, 64000);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
@@ -707,12 +799,16 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
   // Test that the bitrate of a multi-rate codec is always the maximum.
 
   // ISAC, default bitrate == 32000.
-  TestSendBandwidth(kIsacCodec, 128000, true, 128000);
-  TestSendBandwidth(kIsacCodec, 16000, true, 16000);
+  TestMaxSendBandwidth(kIsacCodec, 40000, true, 40000);
+  TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000);
+  // Rates above the max (56000) should be capped.
+  TestMaxSendBandwidth(kIsacCodec, 100000, true, 56000);
 
   // opus, default bitrate == 64000.
-  TestSendBandwidth(kOpusCodec, 96000, true, 96000);
-  TestSendBandwidth(kOpusCodec, 48000, true, 48000);
+  TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000);
+  TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000);
+  // Rates above the max (510000) should be capped.
+  TestMaxSendBandwidth(kOpusCodec, 600000, true, 510000);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
@@ -722,13 +818,13 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
   // if it's bigger than the fixed rate.
 
   // PCMU, fixed bitrate == 64000.
-  TestSendBandwidth(kPcmuCodec, 0, true, 64000);
-  TestSendBandwidth(kPcmuCodec, 1, false, 64000);
-  TestSendBandwidth(kPcmuCodec, 128000, true, 64000);
-  TestSendBandwidth(kPcmuCodec, 32000, false, 64000);
-  TestSendBandwidth(kPcmuCodec, 64000, true, 64000);
-  TestSendBandwidth(kPcmuCodec, 63999, false, 64000);
-  TestSendBandwidth(kPcmuCodec, 64001, true, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, 0, true, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, 1, false, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, 128000, true, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, 32000, false, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, 64000, true, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, 63999, false, 64000);
+  TestMaxSendBandwidth(kPcmuCodec, 64001, true, 64000);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
@@ -772,6 +868,175 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) {
   EXPECT_EQ(64000, codec.rate);
 }
 
+// Test that the per-stream bitrate limit and the global
+// bitrate limit both apply.
+TEST_F(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) {
+  EXPECT_TRUE(SetupSendStream());
+
+  // opus, default bitrate == 64000.
+  SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 64000);
+  SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000);
+  SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000);
+  SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000);
+
+  // CBR codecs allow both maximums to exceed the bitrate.
+  SetAndExpectMaxBitrate(kPcmuCodec, 0, 0, true, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 64001, 0, true, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 0, 64001, true, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 64001, 64001, true, 64000);
+
+  // CBR codecs don't allow per stream maximums to be too low.
+  SetAndExpectMaxBitrate(kPcmuCodec, 0, 63999, false, 64000);
+  SetAndExpectMaxBitrate(kPcmuCodec, 64001, 63999, false, 64000);
+}
+
+// Test that an attempt to set RtpParameters for a stream that does not exist
+// fails.
+TEST_F(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) {
+  EXPECT_TRUE(SetupChannel());
+  webrtc::RtpParameters nonexistent_parameters =
+      channel_->GetRtpSendParameters(kSsrc1);
+  EXPECT_EQ(0, nonexistent_parameters.encodings.size());
+
+  nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, nonexistent_parameters));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake,
+       CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
+  // This test verifies that setting RtpParameters succeeds only if
+  // the structure contains exactly one encoding.
+  // TODO(skvlad): Update this test when we start supporting setting parameters
+  // for each encoding individually.
+
+  EXPECT_TRUE(SetupSendStream());
+  // Setting RtpParameters with no encoding is expected to fail.
+  webrtc::RtpParameters parameters;
+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+  // Setting RtpParameters with exactly one encoding should succeed.
+  parameters.encodings.push_back(webrtc::RtpEncodingParameters());
+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+  // Two or more encodings should result in failure.
+  parameters.encodings.push_back(webrtc::RtpEncodingParameters());
+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+}
+
+// Test that a stream will not be sending if its encoding is made
+// inactive through SetRtpSendParameters.
+TEST_F(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) {
+  EXPECT_TRUE(SetupSendStream());
+  SetSend(channel_, true);
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+  // Get current parameters and change "active" to false.
+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrc1);
+  ASSERT_EQ(1u, parameters.encodings.size());
+  ASSERT_TRUE(parameters.encodings[0].active);
+  parameters.encodings[0].active = false;
+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+  EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
+
+  // Now change it back to active and verify we resume sending.
+  parameters.encodings[0].active = true;
+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, parameters));
+  EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
+}
+
+// Test that SetRtpSendParameters configures the correct encoding channel for
+// each SSRC.
+TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
+  SetupForMultiSendStream();
+  // Create send streams.
+  for (uint32_t ssrc : kSsrcs4) {
+    EXPECT_TRUE(
+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
+  }
+  // Configure one stream to be limited by the stream config, another to be
+  // limited by the global max, and the third one with no per-stream limit
+  // (still subject to the global limit).
+  EXPECT_TRUE(SetGlobalMaxBitrate(kOpusCodec, 64000));
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 48000));
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 96000));
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1));
+
+  EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0]));
+  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[1]));
+  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[2]));
+
+  // Remove the global cap; the streams should switch to their respective
+  // maximums (or remain unchanged if there was no other limit on them.)
+  EXPECT_TRUE(SetGlobalMaxBitrate(kOpusCodec, -1));
+  EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0]));
+  EXPECT_EQ(96000, GetCodecBitrate(kSsrcs4[1]));
+  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[2]));
+}
+
+// Test that GetRtpSendParameters returns the currently configured codecs.
+TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
+  EXPECT_TRUE(SetupSendStream());
+  cricket::AudioSendParameters parameters;
+  parameters.codecs.push_back(kIsacCodec);
+  parameters.codecs.push_back(kPcmuCodec);
+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
+
+  webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc1);
+  ASSERT_EQ(2u, rtp_parameters.codecs.size());
+  EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]);
+  EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
+}
+
+// Test that if we set/get parameters multiple times, we get the same results.
+TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
+  EXPECT_TRUE(SetupSendStream());
+  cricket::AudioSendParameters parameters;
+  parameters.codecs.push_back(kIsacCodec);
+  parameters.codecs.push_back(kPcmuCodec);
+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
+
+  webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrc1);
+
+  // We should be able to set the params we just got.
+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrc1, initial_params));
+
+  // ... And this shouldn't change the params returned by GetRtpSendParameters.
+  webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrc1);
+  EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrc1));
+}
+
+// Test that GetRtpReceiveParameters returns the currently configured codecs.
+TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) {
+  EXPECT_TRUE(SetupRecvStream());
+  cricket::AudioRecvParameters parameters;
+  parameters.codecs.push_back(kIsacCodec);
+  parameters.codecs.push_back(kPcmuCodec);
+  EXPECT_TRUE(channel_->SetRecvParameters(parameters));
+
+  webrtc::RtpParameters rtp_parameters =
+      channel_->GetRtpReceiveParameters(kSsrc1);
+  ASSERT_EQ(2u, rtp_parameters.codecs.size());
+  EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]);
+  EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
+}
+
+// Test that if we set/get parameters multiple times, we get the same results.
+TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) {
+  EXPECT_TRUE(SetupRecvStream());
+  cricket::AudioRecvParameters parameters;
+  parameters.codecs.push_back(kIsacCodec);
+  parameters.codecs.push_back(kPcmuCodec);
+  EXPECT_TRUE(channel_->SetRecvParameters(parameters));
+
+  webrtc::RtpParameters initial_params =
+      channel_->GetRtpReceiveParameters(kSsrc1);
+
+  // We should be able to set the params we just got.
+  EXPECT_TRUE(channel_->SetRtpReceiveParameters(kSsrc1, initial_params));
+
+  // ... And this shouldn't change the params returned by
+  // GetRtpReceiveParameters.
+  webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrc1);
+  EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(kSsrc1));
+}
+
 // Test that we apply codecs properly.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
   EXPECT_TRUE(SetupSendStream());
@@ -1115,7 +1380,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
                              cricket::kParamValueEmpty));
   EXPECT_FALSE(voe_.GetNACK(channel_num));
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
-  EXPECT_FALSE(voe_.GetNACK(channel_num));
+  // NACK should be enabled even with no send stream.
+  EXPECT_TRUE(voe_.GetNACK(channel_num));
 
   EXPECT_TRUE(channel_->AddSendStream(
       cricket::StreamParams::CreateLegacy(kSsrc1)));
@@ -2018,9 +2284,9 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
 TEST_F(WebRtcVoiceEngineTestFake, Send) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-  channel_->SetSend(true);
+  SetSend(channel_, true);
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
-  channel_->SetSend(false);
+  SetSend(channel_, false);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
 }
 
@@ -2030,7 +2296,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SendStateWithAndWithoutSource) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
   EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
-  channel_->SetSend(true);
+  SetSend(channel_, true);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
   EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, nullptr, &fake_source_));
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
@@ -2044,7 +2310,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
 
   // Turn on sending.
-  channel_->SetSend(true);
+  SetSend(channel_, true);
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
 
   // Changing RTP header extensions will recreate the AudioSendStream.
@@ -2054,7 +2320,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
 
   // Turn off sending.
-  channel_->SetSend(false);
+  SetSend(channel_, false);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
 
   // Changing RTP header extensions will recreate the AudioSendStream.
@@ -2079,7 +2345,7 @@ TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
   SetupForMultiSendStream();
 
   // Set the global state for sending.
-  channel_->SetSend(true);
+  SetSend(channel_, true);
 
   for (uint32_t ssrc : kSsrcs4) {
     EXPECT_TRUE(channel_->AddSendStream(
@@ -2150,14 +2416,14 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
   }
 
   // Set the global state for starting sending.
-  channel_->SetSend(true);
+  SetSend(channel_, true);
   for (uint32_t ssrc : kSsrcs4) {
     // Verify that we are in a sending state for all the send streams.
     EXPECT_TRUE(GetSendStream(ssrc).IsSending());
   }
 
   // Set the global state for stopping sending.
-  channel_->SetSend(false);
+  SetSend(channel_, false);
   for (uint32_t ssrc : kSsrcs4) {
     // Verify that we are in a stop state for all the send streams.
     EXPECT_FALSE(GetSendStream(ssrc).IsSending());
@@ -2236,7 +2502,7 @@ TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
   EXPECT_TRUE(
       channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc2)));
   int channel_num2 = voe_.GetLastChannel();
-  channel_->SetSend(true);
+  SetSend(channel_, true);
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
 
   // Make sure only the new stream is played out.
@@ -2252,7 +2518,7 @@ TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
   EXPECT_TRUE(voe_.GetPlayout(channel_num3));
 
   // Stop sending.
-  channel_->SetSend(false);
+  SetSend(channel_, false);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
 
   // Stop playout.
@@ -2284,17 +2550,19 @@ TEST_F(WebRtcVoiceEngineTestFake, CodianSend) {
   EXPECT_EQ(0, agc_config.targetLeveldBOv);
   send_parameters_.options = options_adjust_agc;
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-  channel_->SetSend(true);
+  SetSend(channel_, true);
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
   EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
   EXPECT_EQ(agc_config.targetLeveldBOv, 10);  // level was attenuated
-  channel_->SetSend(false);
+  SetSend(channel_, false);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
   EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
   EXPECT_TRUE(SetupSendStream());
+  EXPECT_CALL(adm_,
+              BuiltInAGCIsAvailable()).Times(2).WillRepeatedly(Return(false));
   webrtc::AgcConfig agc_config;
   EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
   EXPECT_EQ(0, agc_config.targetLeveldBOv);
@@ -2319,16 +2587,12 @@ TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
 
 TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) {
   EXPECT_TRUE(SetupSendStream());
+  EXPECT_CALL(adm_, SetRecordingSampleRate(48000)).WillOnce(Return(0));
+  EXPECT_CALL(adm_, SetPlayoutSampleRate(44100)).WillOnce(Return(0));
   send_parameters_.options.recording_sample_rate =
       rtc::Optional<uint32_t>(48000);
   send_parameters_.options.playout_sample_rate = rtc::Optional<uint32_t>(44100);
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-
-  unsigned int recording_sample_rate, playout_sample_rate;
-  EXPECT_EQ(0, voe_.RecordingSampleRate(&recording_sample_rate));
-  EXPECT_EQ(0, voe_.PlayoutSampleRate(&playout_sample_rate));
-  EXPECT_EQ(48000u, recording_sample_rate);
-  EXPECT_EQ(44100u, playout_sample_rate);
 }
 
 // Test that we can set the outgoing SSRC properly.
@@ -2365,7 +2629,7 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStats) {
   // Start sending - this affects some reported stats.
   {
     cricket::VoiceMediaInfo info;
-    channel_->SetSend(true);
+    SetSend(channel_, true);
     EXPECT_EQ(true, channel_->GetStats(&info));
     VerifyVoiceSenderInfo(info.senders[0], true);
   }
@@ -2420,61 +2684,77 @@ TEST_F(WebRtcVoiceEngineTestFake, Recv) {
   EXPECT_TRUE(SetupChannel());
   EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
   DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
-  int channel_num = voe_.GetLastChannel();
-  EXPECT_TRUE(voe_.CheckPacket(channel_num, kPcmuFrame, sizeof(kPcmuFrame)));
+
+  EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame,
+                                                sizeof(kPcmuFrame)));
 }
 
 // Test that we can properly receive packets on multiple streams.
 TEST_F(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
   EXPECT_TRUE(SetupChannel());
-  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
-  int channel_num1 = voe_.GetLastChannel();
-  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
-  int channel_num2 = voe_.GetLastChannel();
-  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(3)));
-  int channel_num3 = voe_.GetLastChannel();
+  const uint32_t ssrc1 = 1;
+  const uint32_t ssrc2 = 2;
+  const uint32_t ssrc3 = 3;
+  EXPECT_TRUE(channel_->AddRecvStream(
+      cricket::StreamParams::CreateLegacy(ssrc1)));
+  EXPECT_TRUE(channel_->AddRecvStream(
+      cricket::StreamParams::CreateLegacy(ssrc2)));
+  EXPECT_TRUE(channel_->AddRecvStream(
+      cricket::StreamParams::CreateLegacy(ssrc3)));
   // Create packets with the right SSRCs.
-  char packets[4][sizeof(kPcmuFrame)];
+  unsigned char packets[4][sizeof(kPcmuFrame)];
   for (size_t i = 0; i < arraysize(packets); ++i) {
     memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame));
     rtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i));
   }
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+
+  const cricket::FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1);
+  const cricket::FakeAudioReceiveStream& s2 = GetRecvStream(ssrc2);
+  const cricket::FakeAudioReceiveStream& s3 = GetRecvStream(ssrc3);
+
+  EXPECT_EQ(s1.received_packets(), 0);
+  EXPECT_EQ(s2.received_packets(), 0);
+  EXPECT_EQ(s3.received_packets(), 0);
 
   DeliverPacket(packets[0], sizeof(packets[0]));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+  EXPECT_EQ(s1.received_packets(), 0);
+  EXPECT_EQ(s2.received_packets(), 0);
+  EXPECT_EQ(s3.received_packets(), 0);
 
   DeliverPacket(packets[1], sizeof(packets[1]));
-  EXPECT_TRUE(voe_.CheckPacket(channel_num1, packets[1], sizeof(packets[1])));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+  EXPECT_EQ(s1.received_packets(), 1);
+  EXPECT_TRUE(s1.VerifyLastPacket(packets[1], sizeof(packets[1])));
+  EXPECT_EQ(s2.received_packets(), 0);
+  EXPECT_EQ(s3.received_packets(), 0);
 
   DeliverPacket(packets[2], sizeof(packets[2]));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
-  EXPECT_TRUE(voe_.CheckPacket(channel_num2, packets[2], sizeof(packets[2])));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+  EXPECT_EQ(s1.received_packets(), 1);
+  EXPECT_EQ(s2.received_packets(), 1);
+  EXPECT_TRUE(s2.VerifyLastPacket(packets[2], sizeof(packets[2])));
+  EXPECT_EQ(s3.received_packets(), 0);
 
   DeliverPacket(packets[3], sizeof(packets[3]));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
-  EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
-  EXPECT_TRUE(voe_.CheckPacket(channel_num3, packets[3], sizeof(packets[3])));
+  EXPECT_EQ(s1.received_packets(), 1);
+  EXPECT_EQ(s2.received_packets(), 1);
+  EXPECT_EQ(s3.received_packets(), 1);
+  EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3])));
 
-  EXPECT_TRUE(channel_->RemoveRecvStream(3));
-  EXPECT_TRUE(channel_->RemoveRecvStream(2));
-  EXPECT_TRUE(channel_->RemoveRecvStream(1));
+  EXPECT_TRUE(channel_->RemoveRecvStream(ssrc3));
+  EXPECT_TRUE(channel_->RemoveRecvStream(ssrc2));
+  EXPECT_TRUE(channel_->RemoveRecvStream(ssrc1));
 }
 
 // Test that receiving on an unsignalled stream works (default channel will be
 // created).
 TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalled) {
   EXPECT_TRUE(SetupChannel());
+  EXPECT_EQ(0, call_.GetAudioReceiveStreams().size());
+
   DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
-  int channel_num = voe_.GetLastChannel();
-  EXPECT_TRUE(voe_.CheckPacket(channel_num, kPcmuFrame, sizeof(kPcmuFrame)));
+
+  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
+  EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame,
+                                                sizeof(kPcmuFrame)));
 }
 
 // Test that receiving on an unsignalled stream works (default channel will be
@@ -2482,48 +2762,61 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalled) {
 // regardless of their SSRCs.
 TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalledWithSsrcSwitch) {
   EXPECT_TRUE(SetupChannel());
-  char packet[sizeof(kPcmuFrame)];
+  unsigned char packet[sizeof(kPcmuFrame)];
   memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
 
-  // Note that the first unknown SSRC cannot be 0, because we only support
-  // creating receive streams for SSRC!=0.
-  DeliverPacket(packet, sizeof(packet));
-  int channel_num = voe_.GetLastChannel();
-  EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
-  // Once we have the default channel, SSRC==0 will be ok.
-  for (uint32_t ssrc = 0; ssrc < 10; ++ssrc) {
+  // Note that ssrc = 0 is not supported.
+  uint32_t ssrc = 1;
+  for (; ssrc < 10; ++ssrc) {
     rtc::SetBE32(&packet[8], ssrc);
     DeliverPacket(packet, sizeof(packet));
-    EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
+
+    // Verify we only have one default stream.
+    EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
+    EXPECT_EQ(1, GetRecvStream(ssrc).received_packets());
+    EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
   }
+
+  // Sending the same ssrc again should not create a new stream.
+  --ssrc;
+  DeliverPacket(packet, sizeof(packet));
+  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
+  EXPECT_EQ(2, GetRecvStream(ssrc).received_packets());
+  EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
 }
 
 // Test that a default channel is created even after a signalled stream has been
 // added, and that this stream will get any packets for unknown SSRCs.
 TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalledAfterSignalled) {
   EXPECT_TRUE(SetupChannel());
-  char packet[sizeof(kPcmuFrame)];
+  unsigned char packet[sizeof(kPcmuFrame)];
   memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
 
   // Add a known stream, send packet and verify we got it.
-  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
-  int signalled_channel_num = voe_.GetLastChannel();
+  const uint32_t signaled_ssrc = 1;
+  rtc::SetBE32(&packet[8], signaled_ssrc);
+  EXPECT_TRUE(channel_->AddRecvStream(
+      cricket::StreamParams::CreateLegacy(signaled_ssrc)));
   DeliverPacket(packet, sizeof(packet));
-  EXPECT_TRUE(voe_.CheckPacket(signalled_channel_num, packet, sizeof(packet)));
+  EXPECT_TRUE(GetRecvStream(signaled_ssrc).VerifyLastPacket(
+      packet, sizeof(packet)));
 
   // Note that the first unknown SSRC cannot be 0, because we only support
   // creating receive streams for SSRC!=0.
-  rtc::SetBE32(&packet[8], 7011);
+  const uint32_t unsignaled_ssrc = 7011;
+  rtc::SetBE32(&packet[8], unsignaled_ssrc);
   DeliverPacket(packet, sizeof(packet));
-  int channel_num = voe_.GetLastChannel();
-  EXPECT_NE(channel_num, signalled_channel_num);
-  EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
-  // Once we have the default channel, SSRC==0 will be ok.
-  for (uint32_t ssrc = 0; ssrc < 20; ssrc += 2) {
-    rtc::SetBE32(&packet[8], ssrc);
-    DeliverPacket(packet, sizeof(packet));
-    EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
-  }
+  EXPECT_TRUE(GetRecvStream(unsignaled_ssrc).VerifyLastPacket(
+      packet, sizeof(packet)));
+  EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
+
+  DeliverPacket(packet, sizeof(packet));
+  EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets());
+
+  rtc::SetBE32(&packet[8], signaled_ssrc);
+  DeliverPacket(packet, sizeof(packet));
+  EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets());
+  EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
 }
 
 // Test that we properly handle failures to add a receive stream.
@@ -2621,7 +2914,7 @@ TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
 TEST_F(WebRtcVoiceEngineTestFake, TestSetPlayoutError) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-  channel_->SetSend(true);
+  SetSend(channel_, true);
   EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
   EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(3)));
   EXPECT_TRUE(channel_->SetPlayout(true));
@@ -2632,7 +2925,12 @@ TEST_F(WebRtcVoiceEngineTestFake, TestSetPlayoutError) {
 
 TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
   EXPECT_TRUE(SetupSendStream());
-
+  EXPECT_CALL(adm_,
+              BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false));
+  EXPECT_CALL(adm_,
+              BuiltInAGCIsAvailable()).Times(4).WillRepeatedly(Return(false));
+  EXPECT_CALL(adm_,
+              BuiltInNSIsAvailable()).Times(2).WillRepeatedly(Return(false));
   bool ec_enabled;
   webrtc::EcModes ec_mode;
   webrtc::AecmModes aecm_mode;
@@ -2830,6 +3128,13 @@ TEST_F(WebRtcVoiceEngineTestFake, InitDoesNotOverwriteDefaultAgcConfig) {
 
 TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
   EXPECT_TRUE(SetupSendStream());
+  EXPECT_CALL(adm_,
+              BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false));
+  EXPECT_CALL(adm_,
+              BuiltInAGCIsAvailable()).Times(9).WillRepeatedly(Return(false));
+  EXPECT_CALL(adm_,
+              BuiltInNSIsAvailable()).Times(9).WillRepeatedly(Return(false));
+
   std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1(
       static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
           &call_, cricket::MediaConfig(), cricket::AudioOptions())));
@@ -2890,7 +3195,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
   EXPECT_TRUE(agc_enabled);
   EXPECT_TRUE(ns_enabled);
 
-  channel1->SetSend(true);
+  SetSend(channel1.get(), true);
   voe_.GetEcStatus(ec_enabled, ec_mode);
   voe_.GetAgcStatus(agc_enabled, agc_mode);
   voe_.GetNsStatus(ns_enabled, ns_mode);
@@ -2898,7 +3203,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
   EXPECT_TRUE(agc_enabled);
   EXPECT_FALSE(ns_enabled);
 
-  channel2->SetSend(true);
+  SetSend(channel2.get(), true);
   voe_.GetEcStatus(ec_enabled, ec_mode);
   voe_.GetAgcStatus(agc_enabled, agc_mode);
   voe_.GetNsStatus(ns_enabled, ns_mode);
@@ -3262,55 +3567,55 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
 TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
   // Check codecs by name.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "OPUS", 48000, 0, 2, 0), nullptr));
+      cricket::AudioCodec(96, "OPUS", 48000, 0, 2), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "ISAC", 16000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "ISAC", 16000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "ISAC", 32000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "ISAC", 32000, 0, 1), nullptr));
   // Check that name matching is case-insensitive.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "ILBC", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "ILBC", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "iLBC", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "iLBC", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "PCMU", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "PCMU", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "PCMA", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "G722", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "G722", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "red", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "red", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "CN", 32000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "CN", 32000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "CN", 16000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "CN", 16000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "CN", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "CN", 8000, 0, 1), nullptr));
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "telephone-event", 8000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(96, "telephone-event", 8000, 0, 1), nullptr));
   // Check codecs with an id by id.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 8000, 0, 1, 0), nullptr));   // PCMU
+      cricket::AudioCodec(0, "", 8000, 0, 1), nullptr));  // PCMU
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(8, "", 8000, 0, 1, 0), nullptr));   // PCMA
+      cricket::AudioCodec(8, "", 8000, 0, 1), nullptr));  // PCMA
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(9, "", 8000, 0, 1, 0), nullptr));  // G722
+      cricket::AudioCodec(9, "", 8000, 0, 1), nullptr));  // G722
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(13, "", 8000, 0, 1, 0), nullptr));  // CN
+      cricket::AudioCodec(13, "", 8000, 0, 1), nullptr));  // CN
   // Check sample/bitrate matching.
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "PCMU", 8000, 64000, 1, 0), nullptr));
+      cricket::AudioCodec(0, "PCMU", 8000, 64000, 1), nullptr));
   // Check that bad codecs fail.
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(99, "ABCD", 0, 0, 1, 0), nullptr));
+      cricket::AudioCodec(99, "ABCD", 0, 0, 1), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(88, "", 0, 0, 1, 0), nullptr));
+      cricket::AudioCodec(88, "", 0, 0, 1), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 0, 0, 2, 0), nullptr));
+      cricket::AudioCodec(0, "", 0, 0, 2), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 5000, 0, 1, 0), nullptr));
+      cricket::AudioCodec(0, "", 5000, 0, 1), nullptr));
   EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 0, 5000, 1, 0), nullptr));
+      cricket::AudioCodec(0, "", 0, 5000, 1), nullptr));
 
   // Verify the payload id of common audio codecs, including CN, ISAC, and G722.
   cricket::WebRtcVoiceEngine engine(nullptr);
@@ -3334,8 +3639,6 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
       EXPECT_EQ(111, it->id);
       ASSERT_TRUE(it->params.find("minptime") != it->params.end());
       EXPECT_EQ("10", it->params.find("minptime")->second);
-      ASSERT_TRUE(it->params.find("maxptime") != it->params.end());
-      EXPECT_EQ("60", it->params.find("maxptime")->second);
       ASSERT_TRUE(it->params.find("useinbandfec") != it->params.end());
       EXPECT_EQ("1", it->params.find("useinbandfec")->second);
     }
diff --git a/chromium/third_party/webrtc/media/media.gyp b/chromium/third_party/webrtc/media/media.gyp
index 621af832e87..deb730a85a8 100644
--- a/chromium/third_party/webrtc/media/media.gyp
+++ b/chromium/third_party/webrtc/media/media.gyp
@@ -15,7 +15,6 @@
       'dependencies': [
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
-        '<(webrtc_root)/modules/modules.gyp:video_render_module',
         '<(webrtc_root)/webrtc.gyp:webrtc',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
@@ -30,7 +29,6 @@
         ],
       },
       'sources': [
-        'base/audioframe.h',
         'base/audiosource.h',
         'base/codec.cc',
         'base/codec.h',
@@ -109,7 +107,6 @@
         '-Wextra',
       ],
       'cflags_cc!': [
-        '-Wnon-virtual-dtor',
         '-Woverloaded-virtual',
       ],
       'msvs_disabled_warnings': [
@@ -134,7 +131,6 @@
         ['build_with_chromium==1', {
           'dependencies': [
             '<(webrtc_root)/modules/modules.gyp:video_capture',
-            '<(webrtc_root)/modules/modules.gyp:video_render',
           ],
         }, {
           'defines': [
@@ -149,7 +145,6 @@
           },
           'dependencies': [
             '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
-            '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
           ],
         }],
         ['OS=="linux" and use_gtk==1', {
@@ -252,11 +247,6 @@
             'engine/fakewebrtcvideoengine.h',
             'engine/fakewebrtcvoiceengine.h',
           ],
-          # TODO(kjellander): Make the code compile without disabling these flags.
-          # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
-          'cflags_cc!': [
-            '-Wnon-virtual-dtor',
-          ],
         },  # target rtc_unittest_main
         {
           'target_name': 'rtc_media_unittests',
@@ -295,7 +285,6 @@
             '-Wno-sign-compare',
           ],
           'cflags_cc!': [
-            '-Wnon-virtual-dtor',
             '-Woverloaded-virtual',
           ],
           'msvs_disabled_warnings': [
@@ -304,15 +293,6 @@
           ],
           'conditions': [
             ['OS=="win"', {
-              'conditions': [
-                ['use_openssl==0', {
-                  'dependencies': [
-                    '<(DEPTH)/net/third_party/nss/ssl.gyp:libssl',
-                    '<(DEPTH)/third_party/nss/nss.gyp:nspr',
-                    '<(DEPTH)/third_party/nss/nss.gyp:nss',
-                  ],
-                }],
-              ],
               'msvs_settings': {
                 'VCLinkerTool': {
                   'AdditionalDependencies': [
diff --git a/chromium/third_party/webrtc/media/sctp/sctpdataengine.cc b/chromium/third_party/webrtc/media/sctp/sctpdataengine.cc
index ba3b8f24084..958cd9a0f63 100644
--- a/chromium/third_party/webrtc/media/sctp/sctpdataengine.cc
+++ b/chromium/third_party/webrtc/media/sctp/sctpdataengine.cc
@@ -20,6 +20,7 @@
 #include "usrsctplib/usrsctp.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/copyonwritebuffer.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/safe_conversions.h"
@@ -27,8 +28,29 @@
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 
+namespace cricket {
+// The biggest SCTP packet.  Starting from a 'safe' wire MTU value of 1280,
+// take off 80 bytes for DTLS/TURN/TCP/IP overhead.
+static const size_t kSctpMtu = 1200;
+
+// The size of the SCTP association send buffer.  256kB, the usrsctp default.
+static const int kSendBufferSize = 262144;
+
+struct SctpInboundPacket {
+  rtc::CopyOnWriteBuffer buffer;
+  ReceiveDataParams params;
+  // The |flags| parameter is used by SCTP to distinguish notification packets
+  // from other types of packets.
+  int flags;
+};
+
 namespace {
-typedef cricket::SctpDataMediaChannel::StreamSet StreamSet;
+// Set the initial value of the static SCTP Data Engines reference count.
+int g_usrsctp_usage_count = 0;
+rtc::GlobalLockPod g_usrsctp_lock_;
+
+typedef SctpDataMediaChannel::StreamSet StreamSet;
+
 // Returns a comma-separated, human-readable list of the stream IDs in 's'
 std::string ListStreams(const StreamSet& s) {
   std::stringstream result;
@@ -85,78 +107,62 @@ std::string ListArray(const uint16_t* array, int num_elems) {
   }
   return result.str();
 }
-}  // namespace
 
-namespace cricket {
 typedef rtc::ScopedMessageData<SctpInboundPacket> InboundPacketMessage;
 typedef rtc::ScopedMessageData<rtc::CopyOnWriteBuffer> OutboundPacketMessage;
 
-// The biggest SCTP packet.  Starting from a 'safe' wire MTU value of 1280,
-// take off 80 bytes for DTLS/TURN/TCP/IP overhead.
-static const size_t kSctpMtu = 1200;
-
-// The size of the SCTP association send buffer.  256kB, the usrsctp default.
-static const int kSendBufferSize = 262144;
 enum {
   MSG_SCTPINBOUNDPACKET = 1,   // MessageData is SctpInboundPacket
   MSG_SCTPOUTBOUNDPACKET = 2,  // MessageData is rtc:Buffer
 };
 
-struct SctpInboundPacket {
-  rtc::CopyOnWriteBuffer buffer;
-  ReceiveDataParams params;
-  // The |flags| parameter is used by SCTP to distinguish notification packets
-  // from other types of packets.
-  int flags;
-};
-
 // Helper for logging SCTP messages.
-static void debug_sctp_printf(const char *format, ...) {
+void DebugSctpPrintf(const char* format, ...) {
+#if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
   char s[255];
   va_list ap;
   va_start(ap, format);
   vsnprintf(s, sizeof(s), format, ap);
   LOG(LS_INFO) << "SCTP: " << s;
   va_end(ap);
+#endif
 }
 
 // Get the PPID to use for the terminating fragment of this type.
-static SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid(
-    cricket::DataMessageType type) {
+SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid(DataMessageType type) {
   switch (type) {
   default:
-  case cricket::DMT_NONE:
+  case DMT_NONE:
     return SctpDataMediaChannel::PPID_NONE;
-  case cricket::DMT_CONTROL:
+  case DMT_CONTROL:
     return SctpDataMediaChannel::PPID_CONTROL;
-  case cricket::DMT_BINARY:
+  case DMT_BINARY:
     return SctpDataMediaChannel::PPID_BINARY_LAST;
-  case cricket::DMT_TEXT:
+  case DMT_TEXT:
     return SctpDataMediaChannel::PPID_TEXT_LAST;
   };
 }
 
-static bool GetDataMediaType(
-    SctpDataMediaChannel::PayloadProtocolIdentifier ppid,
-    cricket::DataMessageType *dest) {
+bool GetDataMediaType(SctpDataMediaChannel::PayloadProtocolIdentifier ppid,
+                      DataMessageType* dest) {
   ASSERT(dest != NULL);
   switch (ppid) {
     case SctpDataMediaChannel::PPID_BINARY_PARTIAL:
     case SctpDataMediaChannel::PPID_BINARY_LAST:
-      *dest = cricket::DMT_BINARY;
+      *dest = DMT_BINARY;
       return true;
 
     case SctpDataMediaChannel::PPID_TEXT_PARTIAL:
     case SctpDataMediaChannel::PPID_TEXT_LAST:
-      *dest = cricket::DMT_TEXT;
+      *dest = DMT_TEXT;
       return true;
 
     case SctpDataMediaChannel::PPID_CONTROL:
-      *dest = cricket::DMT_CONTROL;
+      *dest = DMT_CONTROL;
       return true;
 
     case SctpDataMediaChannel::PPID_NONE:
-      *dest = cricket::DMT_NONE;
+      *dest = DMT_NONE;
       return true;
 
     default:
@@ -165,7 +171,7 @@ static bool GetDataMediaType(
 }
 
 // Log the packet in text2pcap format, if log level is at LS_VERBOSE.
-static void VerboseLogPacket(const void *data, size_t length, int direction) {
+void VerboseLogPacket(const void* data, size_t length, int direction) {
   if (LOG_CHECK_LEVEL(LS_VERBOSE) && length > 0) {
     char *dump_buf;
     // Some downstream project uses an older version of usrsctp that expects
@@ -181,8 +187,11 @@ static void VerboseLogPacket(const void *data, size_t length, int direction) {
 
 // This is the callback usrsctp uses when there's data to send on the network
 // that has been wrapped appropriatly for the SCTP protocol.
-static int OnSctpOutboundPacket(void* addr, void* data, size_t length,
-                                uint8_t tos, uint8_t set_df) {
+int OnSctpOutboundPacket(void* addr,
+                         void* data,
+                         size_t length,
+                         uint8_t tos,
+                         uint8_t set_df) {
   SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(addr);
   LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
                   << "addr: " << addr << "; length: " << length
@@ -201,10 +210,13 @@ static int OnSctpOutboundPacket(void* addr, void* data, size_t length,
 // a packet has been interpreted and parsed by usrsctp and found to contain
 // payload data. It is called by a usrsctp thread. It is assumed this function
 // will free the memory used by 'data'.
-static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr,
-                               void* data, size_t length,
-                               struct sctp_rcvinfo rcv, int flags,
-                               void* ulp_info) {
+int OnSctpInboundPacket(struct socket* sock,
+                        union sctp_sockstore addr,
+                        void* data,
+                        size_t length,
+                        struct sctp_rcvinfo rcv,
+                        int flags,
+                        void* ulp_info) {
   SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(ulp_info);
   // Post data to the channel's receiver thread (copying it).
   // TODO(ldixon): Unclear if copy is needed as this method is responsible for
@@ -212,7 +224,7 @@ static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr,
   const SctpDataMediaChannel::PayloadProtocolIdentifier ppid =
       static_cast<SctpDataMediaChannel::PayloadProtocolIdentifier>(
           rtc::HostToNetwork32(rcv.rcv_ppid));
-  cricket::DataMessageType type = cricket::DMT_NONE;
+  DataMessageType type = DMT_NONE;
   if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) {
     // It's neither a notification nor a recognized data packet.  Drop it.
     LOG(LS_ERROR) << "Received an unknown PPID " << ppid
@@ -233,78 +245,94 @@ static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr,
   return 1;
 }
 
-// Set the initial value of the static SCTP Data Engines reference count.
-int SctpDataEngine::usrsctp_engines_count = 0;
-
-SctpDataEngine::SctpDataEngine() {
-  if (usrsctp_engines_count == 0) {
-    // First argument is udp_encapsulation_port, which is not releveant for our
-    // AF_CONN use of sctp.
-    usrsctp_init(0, cricket::OnSctpOutboundPacket, debug_sctp_printf);
-
-    // To turn on/off detailed SCTP debugging. You will also need to have the
-    // SCTP_DEBUG cpp defines flag.
-    // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
-
-    // TODO(ldixon): Consider turning this on/off.
-    usrsctp_sysctl_set_sctp_ecn_enable(0);
-
-    // This is harmless, but we should find out when the library default
-    // changes.
-    int send_size = usrsctp_sysctl_get_sctp_sendspace();
-    if (send_size != kSendBufferSize) {
-      LOG(LS_ERROR) << "Got different send size than expected: " << send_size;
-    }
+void InitializeUsrSctp() {
+  LOG(LS_INFO) << __FUNCTION__;
+  // First argument is udp_encapsulation_port, which is not releveant for our
+  // AF_CONN use of sctp.
+  usrsctp_init(0, &OnSctpOutboundPacket, &DebugSctpPrintf);
 
-    // TODO(ldixon): Consider turning this on/off.
-    // This is not needed right now (we don't do dynamic address changes):
-    // If SCTP Auto-ASCONF is enabled, the peer is informed automatically
-    // when a new address is added or removed. This feature is enabled by
-    // default.
-    // usrsctp_sysctl_set_sctp_auto_asconf(0);
-
-    // TODO(ldixon): Consider turning this on/off.
-    // Add a blackhole sysctl. Setting it to 1 results in no ABORTs
-    // being sent in response to INITs, setting it to 2 results
-    // in no ABORTs being sent for received OOTB packets.
-    // This is similar to the TCP sysctl.
-    //
-    // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
-    // See: http://svnweb.freebsd.org/base?view=revision&revision=229805
-    // usrsctp_sysctl_set_sctp_blackhole(2);
-
-    // Set the number of default outgoing streams.  This is the number we'll
-    // send in the SCTP INIT message.  The 'appropriate default' in the
-    // second paragraph of
-    // http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2
-    // is cricket::kMaxSctpSid.
-    usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(
-        cricket::kMaxSctpSid);
+  // To turn on/off detailed SCTP debugging. You will also need to have the
+  // SCTP_DEBUG cpp defines flag.
+  // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
+
+  // TODO(ldixon): Consider turning this on/off.
+  usrsctp_sysctl_set_sctp_ecn_enable(0);
+
+  // This is harmless, but we should find out when the library default
+  // changes.
+  int send_size = usrsctp_sysctl_get_sctp_sendspace();
+  if (send_size != kSendBufferSize) {
+    LOG(LS_ERROR) << "Got different send size than expected: " << send_size;
   }
-  usrsctp_engines_count++;
 
-  cricket::DataCodec codec(kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, 0);
-  codec.SetParam(kCodecParamPort, kSctpDefaultPort);
-  codecs_.push_back(codec);
+  // TODO(ldixon): Consider turning this on/off.
+  // This is not needed right now (we don't do dynamic address changes):
+  // If SCTP Auto-ASCONF is enabled, the peer is informed automatically
+  // when a new address is added or removed. This feature is enabled by
+  // default.
+  // usrsctp_sysctl_set_sctp_auto_asconf(0);
+
+  // TODO(ldixon): Consider turning this on/off.
+  // Add a blackhole sysctl. Setting it to 1 results in no ABORTs
+  // being sent in response to INITs, setting it to 2 results
+  // in no ABORTs being sent for received OOTB packets.
+  // This is similar to the TCP sysctl.
+  //
+  // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
+  // See: http://svnweb.freebsd.org/base?view=revision&revision=229805
+  // usrsctp_sysctl_set_sctp_blackhole(2);
+
+  // Set the number of default outgoing streams.  This is the number we'll
+  // send in the SCTP INIT message.  The 'appropriate default' in the
+  // second paragraph of
+  // http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2
+  // is kMaxSctpSid.
+  usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpSid);
 }
 
-SctpDataEngine::~SctpDataEngine() {
-  usrsctp_engines_count--;
-  LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
+void UninitializeUsrSctp() {
+  LOG(LS_INFO) << __FUNCTION__;
+  // usrsctp_finish() may fail if it's called too soon after the channels are
+  // closed. Wait and try again until it succeeds for up to 3 seconds.
+  for (size_t i = 0; i < 300; ++i) {
+    if (usrsctp_finish() == 0) {
+      return;
+    }
 
-  if (usrsctp_engines_count == 0) {
-    // usrsctp_finish() may fail if it's called too soon after the channels are
-    // closed. Wait and try again until it succeeds for up to 3 seconds.
-    for (size_t i = 0; i < 300; ++i) {
-      if (usrsctp_finish() == 0)
-        return;
+    rtc::Thread::SleepMs(10);
+  }
+  LOG(LS_ERROR) << "Failed to shutdown usrsctp.";
+}
 
-      rtc::Thread::SleepMs(10);
-    }
-    LOG(LS_ERROR) << "Failed to shutdown usrsctp.";
+void IncrementUsrSctpUsageCount() {
+  rtc::GlobalLockScope lock(&g_usrsctp_lock_);
+  if (!g_usrsctp_usage_count) {
+    InitializeUsrSctp();
+  }
+  ++g_usrsctp_usage_count;
+}
+
+void DecrementUsrSctpUsageCount() {
+  rtc::GlobalLockScope lock(&g_usrsctp_lock_);
+  --g_usrsctp_usage_count;
+  if (!g_usrsctp_usage_count) {
+    UninitializeUsrSctp();
   }
 }
 
+DataCodec GetSctpDataCodec() {
+  DataCodec codec(kGoogleSctpDataCodecId, kGoogleSctpDataCodecName);
+  codec.SetParam(kCodecParamPort, kSctpDefaultPort);
+  return codec;
+}
+
+}  // namespace
+
+SctpDataEngine::SctpDataEngine() : codecs_(1, GetSctpDataCodec()) {}
+
+SctpDataEngine::~SctpDataEngine() {}
+
+// Called on the worker thread.
 DataMediaChannel* SctpDataEngine::CreateChannel(
     DataChannelType data_channel_type) {
   if (data_channel_type != DCT_SCTP) {
@@ -314,7 +342,7 @@ DataMediaChannel* SctpDataEngine::CreateChannel(
 }
 
 // static
-SctpDataMediaChannel* SctpDataEngine::GetChannelFromSocket(
+SctpDataMediaChannel* SctpDataMediaChannel::GetChannelFromSocket(
     struct socket* sock) {
   struct sockaddr* addrs = nullptr;
   int naddrs = usrsctp_getladdrs(sock, 0, &addrs);
@@ -336,8 +364,8 @@ SctpDataMediaChannel* SctpDataEngine::GetChannelFromSocket(
 }
 
 // static
-int SctpDataEngine::SendThresholdCallback(struct socket* sock,
-                                          uint32_t sb_free) {
+int SctpDataMediaChannel::SendThresholdCallback(struct socket* sock,
+                                                uint32_t sb_free) {
   // Fired on our I/O thread.  SctpDataMediaChannel::OnPacketReceived() gets
   // a packet containing acknowledgments, which goes into usrsctp_conninput,
   // and then back here.
@@ -389,17 +417,19 @@ bool SctpDataMediaChannel::OpenSctpSocket() {
     return false;
   }
 
+  IncrementUsrSctpUsageCount();
+
   // If kSendBufferSize isn't reflective of reality, we log an error, but we
   // still have to do something reasonable here.  Look up what the buffer's
   // real size is and set our threshold to something reasonable.
   const static int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 2;
 
-  sock_ = usrsctp_socket(AF_CONN, SOCK_STREAM, IPPROTO_SCTP,
-                         cricket::OnSctpInboundPacket,
-                         &SctpDataEngine::SendThresholdCallback,
-                         kSendThreshold, this);
+  sock_ = usrsctp_socket(
+      AF_CONN, SOCK_STREAM, IPPROTO_SCTP, OnSctpInboundPacket,
+      &SctpDataMediaChannel::SendThresholdCallback, kSendThreshold, this);
   if (!sock_) {
     LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to create SCTP socket.";
+    DecrementUsrSctpUsageCount();
     return false;
   }
 
@@ -488,6 +518,8 @@ void SctpDataMediaChannel::CloseSctpSocket() {
     usrsctp_close(sock_);
     sock_ = NULL;
     usrsctp_deregister_address(this);
+
+    DecrementUsrSctpUsageCount();
   }
 }
 
@@ -599,7 +631,7 @@ bool SctpDataMediaChannel::SendData(
     return false;
   }
 
-  if (params.type != cricket::DMT_CONTROL &&
+  if (params.type != DMT_CONTROL &&
       open_streams_.find(params.ssrc) == open_streams_.end()) {
     LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
                     << "Not sending data because ssrc is unknown: "
@@ -717,7 +749,7 @@ bool SctpDataMediaChannel::AddStream(const StreamParams& stream) {
   }
 
   const uint32_t ssrc = stream.first_ssrc();
-  if (ssrc >= cricket::kMaxSctpSid) {
+  if (ssrc >= kMaxSctpSid) {
     LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
                     << "Not adding data stream '" << stream.id
                     << "' with ssrc=" << ssrc
@@ -984,8 +1016,9 @@ void SctpDataMediaChannel::OnPacketFromSctpToNetwork(
 }
 
 bool SctpDataMediaChannel::SendQueuedStreamResets() {
-  if (!sent_reset_streams_.empty() || queued_reset_streams_.empty())
+  if (!sent_reset_streams_.empty() || queued_reset_streams_.empty()) {
     return true;
+  }
 
   LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ << "]: Sending ["
                   << ListStreams(queued_reset_streams_) << "], Open: ["
diff --git a/chromium/third_party/webrtc/media/sctp/sctpdataengine.h b/chromium/third_party/webrtc/media/sctp/sctpdataengine.h
index 8eb99564ca2..f9a3b5a6547 100644
--- a/chromium/third_party/webrtc/media/sctp/sctpdataengine.h
+++ b/chromium/third_party/webrtc/media/sctp/sctpdataengine.h
@@ -25,6 +25,7 @@ enum PreservedErrno {
 }  // namespace cricket
 
 #include "webrtc/base/copyonwritebuffer.h"
+#include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/mediaengine.h"
@@ -75,19 +76,13 @@ class SctpDataMediaChannel;
 class SctpDataEngine : public DataEngineInterface, public sigslot::has_slots<> {
  public:
   SctpDataEngine();
-  virtual ~SctpDataEngine();
+  ~SctpDataEngine() override;
 
-  virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);
-
-  virtual const std::vector<DataCodec>& data_codecs() { return codecs_; }
-
-  static int SendThresholdCallback(struct socket* sock, uint32_t sb_free);
+  DataMediaChannel* CreateChannel(DataChannelType data_channel_type) override;
+  const std::vector<DataCodec>& data_codecs() override { return codecs_; }
 
  private:
-  static int usrsctp_engines_count;
-  std::vector<DataCodec> codecs_;
-
-  static SctpDataMediaChannel* GetChannelFromSocket(struct socket* sock);
+  const std::vector<DataCodec> codecs_;
 };
 
 // TODO(ldixon): Make into a special type of TypedMessageData.
@@ -161,11 +156,16 @@ class SctpDataMediaChannel : public DataMediaChannel,
 
   void OnSendThresholdCallback();
   // Helper for debugging.
-  void set_debug_name(const std::string& debug_name) {
+  void set_debug_name_for_testing(const char* debug_name) {
     debug_name_ = debug_name;
   }
-  const std::string& debug_name() const { return debug_name_; }
   const struct socket* socket() const { return sock_; }
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(SctpDataMediaChannelTest, EngineSignalsRightChannel);
+  static int SendThresholdCallback(struct socket* sock, uint32_t sb_free);
+  static SctpDataMediaChannel* GetChannelFromSocket(struct socket* sock);
+
  private:
   sockaddr_conn GetSctpSockAddr(int port);
 
@@ -229,8 +229,8 @@ class SctpDataMediaChannel : public DataMediaChannel,
   StreamSet queued_reset_streams_;
   StreamSet sent_reset_streams_;
 
-  // A human-readable name for debugging messages.
-  std::string debug_name_;
+  // A static human-readable name for debugging messages.
+  const char* debug_name_;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/media/sctp/sctpdataengine_unittest.cc b/chromium/third_party/webrtc/media/sctp/sctpdataengine_unittest.cc
index f18437dc8ed..f41691b6016 100644
--- a/chromium/third_party/webrtc/media/sctp/sctpdataengine_unittest.cc
+++ b/chromium/third_party/webrtc/media/sctp/sctpdataengine_unittest.cc
@@ -29,13 +29,14 @@
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/sctp/sctpdataengine.h"
 
+namespace cricket {
 enum {
   MSG_PACKET = 1,
 };
 
 // Fake NetworkInterface that sends/receives sctp packets.  The one in
 // webrtc/media/base/fakenetworkinterface.h only works with rtp/rtcp.
-class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
+class SctpFakeNetworkInterface : public MediaChannel::NetworkInterface,
                                  public rtc::MessageHandler {
  public:
   explicit SctpFakeNetworkInterface(rtc::Thread* thread)
@@ -43,7 +44,7 @@ class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
       dest_(NULL) {
   }
 
-  void SetDestination(cricket::DataMediaChannel* dest) { dest_ = dest; }
+  void SetDestination(DataMediaChannel* dest) { dest_ = dest; }
 
  protected:
   // Called to send raw packet down the wire (e.g. SCTP an packet).
@@ -91,7 +92,7 @@ class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
  private:
   // Not owned by this class.
   rtc::Thread* thread_;
-  cricket::DataMediaChannel* dest_;
+  DataMediaChannel* dest_;
 };
 
 // This is essentially a buffer to hold recieved data. It stores only the last
@@ -106,11 +107,12 @@ class SctpFakeDataReceiver : public sigslot::has_slots<> {
   void Clear() {
     received_ = false;
     last_data_ = "";
-    last_params_ = cricket::ReceiveDataParams();
+    last_params_ = ReceiveDataParams();
   }
 
-  virtual void OnDataReceived(const cricket::ReceiveDataParams& params,
-                              const char* data, size_t length) {
+  virtual void OnDataReceived(const ReceiveDataParams& params,
+                              const char* data,
+                              size_t length) {
     received_ = true;
     last_data_ = std::string(data, length);
     last_params_ = params;
@@ -118,12 +120,12 @@ class SctpFakeDataReceiver : public sigslot::has_slots<> {
 
   bool received() const { return received_; }
   std::string last_data() const { return last_data_; }
-  cricket::ReceiveDataParams last_params() const { return last_params_; }
+  ReceiveDataParams last_params() const { return last_params_; }
 
  private:
   bool received_;
   std::string last_data_;
-  cricket::ReceiveDataParams last_params_;
+  ReceiveDataParams last_params_;
 };
 
 class SignalReadyToSendObserver : public sigslot::has_slots<> {
@@ -147,7 +149,7 @@ class SignalReadyToSendObserver : public sigslot::has_slots<> {
 class SignalChannelClosedObserver : public sigslot::has_slots<> {
  public:
   SignalChannelClosedObserver() {}
-  void BindSelf(cricket::SctpDataMediaChannel* channel) {
+  void BindSelf(SctpDataMediaChannel* channel) {
     channel->SignalStreamClosedRemotely.connect(
         this, &SignalChannelClosedObserver::OnStreamClosed);
   }
@@ -168,12 +170,12 @@ class SignalChannelClosedObserver : public sigslot::has_slots<> {
 
 class SignalChannelClosedReopener : public sigslot::has_slots<> {
  public:
-  SignalChannelClosedReopener(cricket::SctpDataMediaChannel* channel,
-                              cricket::SctpDataMediaChannel* peer)
+  SignalChannelClosedReopener(SctpDataMediaChannel* channel,
+                              SctpDataMediaChannel* peer)
       : channel_(channel), peer_(peer) {}
 
   void OnStreamClosed(int stream) {
-    cricket::StreamParams p(cricket::StreamParams::CreateLegacy(stream));
+    StreamParams p(StreamParams::CreateLegacy(stream));
     channel_->AddSendStream(p);
     channel_->AddRecvStream(p);
     peer_->AddSendStream(p);
@@ -186,8 +188,8 @@ class SignalChannelClosedReopener : public sigslot::has_slots<> {
   }
 
  private:
-  cricket::SctpDataMediaChannel* channel_;
-  cricket::SctpDataMediaChannel* peer_;
+  SctpDataMediaChannel* channel_;
+  SctpDataMediaChannel* peer_;
   std::vector<int> streams_;
 };
 
@@ -200,9 +202,7 @@ class SctpDataMediaChannelTest : public testing::Test,
   static void SetUpTestCase() {
   }
 
-  virtual void SetUp() {
-    engine_.reset(new cricket::SctpDataEngine());
-  }
+  virtual void SetUp() { engine_.reset(new SctpDataEngine()); }
 
   void SetupConnectedChannels() {
     net1_.reset(new SctpFakeNetworkInterface(rtc::Thread::Current()));
@@ -212,11 +212,11 @@ class SctpDataMediaChannelTest : public testing::Test,
     chan1_ready_to_send_count_ = 0;
     chan2_ready_to_send_count_ = 0;
     chan1_.reset(CreateChannel(net1_.get(), recv1_.get()));
-    chan1_->set_debug_name("chan1/connector");
+    chan1_->set_debug_name_for_testing("chan1/connector");
     chan1_->SignalReadyToSend.connect(
         this, &SctpDataMediaChannelTest::OnChan1ReadyToSend);
     chan2_.reset(CreateChannel(net2_.get(), recv2_.get()));
-    chan2_->set_debug_name("chan2/listener");
+    chan2_->set_debug_name_for_testing("chan2/listener");
     chan2_->SignalReadyToSend.connect(
         this, &SctpDataMediaChannelTest::OnChan2ReadyToSend);
     // Setup two connected channels ready to send and receive.
@@ -254,7 +254,7 @@ class SctpDataMediaChannelTest : public testing::Test,
 
   bool AddStream(int ssrc) {
     bool ret = true;
-    cricket::StreamParams p(cricket::StreamParams::CreateLegacy(ssrc));
+    StreamParams p(StreamParams::CreateLegacy(ssrc));
     ret = ret && chan1_->AddSendStream(p);
     ret = ret && chan1_->AddRecvStream(p);
     ret = ret && chan2_->AddSendStream(p);
@@ -262,11 +262,10 @@ class SctpDataMediaChannelTest : public testing::Test,
     return ret;
   }
 
-  cricket::SctpDataMediaChannel* CreateChannel(
-      SctpFakeNetworkInterface* net, SctpFakeDataReceiver* recv) {
-    cricket::SctpDataMediaChannel* channel =
-        static_cast<cricket::SctpDataMediaChannel*>(engine_->CreateChannel(
-            cricket::DCT_SCTP));
+  SctpDataMediaChannel* CreateChannel(SctpFakeNetworkInterface* net,
+                                      SctpFakeDataReceiver* recv) {
+    SctpDataMediaChannel* channel =
+        static_cast<SctpDataMediaChannel*>(engine_->CreateChannel(DCT_SCTP));
     channel->SetInterface(net);
     // When data is received, pass it to the SctpFakeDataReceiver.
     channel->SignalDataReceived.connect(
@@ -274,11 +273,11 @@ class SctpDataMediaChannelTest : public testing::Test,
     return channel;
   }
 
-  bool SendData(cricket::SctpDataMediaChannel* chan,
+  bool SendData(SctpDataMediaChannel* chan,
                 uint32_t ssrc,
                 const std::string& msg,
-                cricket::SendDataResult* result) {
-    cricket::SendDataParams params;
+                SendDataResult* result) {
+    SendDataParams params;
     params.ssrc = ssrc;
 
     return chan->SendData(params, rtc::CopyOnWriteBuffer(
@@ -304,21 +303,21 @@ class SctpDataMediaChannelTest : public testing::Test,
     return !thread->IsQuitting();
   }
 
-  cricket::SctpDataMediaChannel* channel1() { return chan1_.get(); }
-  cricket::SctpDataMediaChannel* channel2() { return chan2_.get(); }
+  SctpDataMediaChannel* channel1() { return chan1_.get(); }
+  SctpDataMediaChannel* channel2() { return chan2_.get(); }
   SctpFakeDataReceiver* receiver1() { return recv1_.get(); }
   SctpFakeDataReceiver* receiver2() { return recv2_.get(); }
 
   int channel1_ready_to_send_count() { return chan1_ready_to_send_count_; }
   int channel2_ready_to_send_count() { return chan2_ready_to_send_count_; }
  private:
-  std::unique_ptr<cricket::SctpDataEngine> engine_;
+  std::unique_ptr<SctpDataEngine> engine_;
   std::unique_ptr<SctpFakeNetworkInterface> net1_;
   std::unique_ptr<SctpFakeNetworkInterface> net2_;
   std::unique_ptr<SctpFakeDataReceiver> recv1_;
   std::unique_ptr<SctpFakeDataReceiver> recv2_;
-  std::unique_ptr<cricket::SctpDataMediaChannel> chan1_;
-  std::unique_ptr<cricket::SctpDataMediaChannel> chan2_;
+  std::unique_ptr<SctpDataMediaChannel> chan1_;
+  std::unique_ptr<SctpDataMediaChannel> chan2_;
 
   int chan1_ready_to_send_count_;
   int chan2_ready_to_send_count_;
@@ -345,12 +344,12 @@ TEST_F(SctpDataMediaChannelTest, SignalReadyToSend) {
   channel2()->SignalReadyToSend.connect(&signal_observer_2,
                                         &SignalReadyToSendObserver::OnSignaled);
 
-  cricket::SendDataResult result;
+  SendDataResult result;
   ASSERT_TRUE(SendData(channel1(), 1, "hello?", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), 1000);
   ASSERT_TRUE(SendData(channel2(), 2, "hi chan1", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver1(), 2, "hi chan1"), 1000);
 
   EXPECT_TRUE_WAIT(signal_observer_1.IsSignaled(true), 1000);
@@ -360,10 +359,10 @@ TEST_F(SctpDataMediaChannelTest, SignalReadyToSend) {
 TEST_F(SctpDataMediaChannelTest, SendData) {
   SetupConnectedChannels();
 
-  cricket::SendDataResult result;
+  SendDataResult result;
   LOG(LS_VERBOSE) << "chan1 sending: 'hello?' -----------------------------";
   ASSERT_TRUE(SendData(channel1(), 1, "hello?", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), 1000);
   LOG(LS_VERBOSE) << "recv2.received=" << receiver2()->received()
                   << ", recv2.last_params.ssrc="
@@ -376,7 +375,7 @@ TEST_F(SctpDataMediaChannelTest, SendData) {
 
   LOG(LS_VERBOSE) << "chan2 sending: 'hi chan1' -----------------------------";
   ASSERT_TRUE(SendData(channel2(), 2, "hi chan1", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver1(), 2, "hi chan1"), 1000);
   LOG(LS_VERBOSE) << "recv1.received=" << receiver1()->received()
                   << ", recv1.last_params.ssrc="
@@ -392,8 +391,8 @@ TEST_F(SctpDataMediaChannelTest, SendData) {
 TEST_F(SctpDataMediaChannelTest, SendDataBlocked) {
   SetupConnectedChannels();
 
-  cricket::SendDataResult result;
-  cricket::SendDataParams params;
+  SendDataResult result;
+  SendDataParams params;
   params.ssrc = 1;
 
   std::vector<char> buffer(1024 * 64, 0);
@@ -401,11 +400,11 @@ TEST_F(SctpDataMediaChannelTest, SendDataBlocked) {
   for (size_t i = 0; i < 100; ++i) {
     channel1()->SendData(
         params, rtc::CopyOnWriteBuffer(&buffer[0], buffer.size()), &result);
-    if (result == cricket::SDR_BLOCK)
+    if (result == SDR_BLOCK)
       break;
   }
 
-  EXPECT_EQ(cricket::SDR_BLOCK, result);
+  EXPECT_EQ(SDR_BLOCK, result);
 }
 
 TEST_F(SctpDataMediaChannelTest, ClosesRemoteStream) {
@@ -414,12 +413,12 @@ TEST_F(SctpDataMediaChannelTest, ClosesRemoteStream) {
   chan_1_sig_receiver.BindSelf(channel1());
   chan_2_sig_receiver.BindSelf(channel2());
 
-  cricket::SendDataResult result;
+  SendDataResult result;
   ASSERT_TRUE(SendData(channel1(), 1, "hello?", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), 1000);
   ASSERT_TRUE(SendData(channel2(), 2, "hi chan1", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver1(), 2, "hi chan1"), 1000);
 
   // Close channel 1.  Channel 2 should notify us.
@@ -434,12 +433,12 @@ TEST_F(SctpDataMediaChannelTest, ClosesTwoRemoteStreams) {
   chan_1_sig_receiver.BindSelf(channel1());
   chan_2_sig_receiver.BindSelf(channel2());
 
-  cricket::SendDataResult result;
+  SendDataResult result;
   ASSERT_TRUE(SendData(channel1(), 1, "hello?", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), 1000);
   ASSERT_TRUE(SendData(channel2(), 2, "hi chan1", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver1(), 2, "hi chan1"), 1000);
 
   // Close two streams on one side.
@@ -457,12 +456,12 @@ TEST_F(SctpDataMediaChannelTest, ClosesStreamsOnBothSides) {
   chan_1_sig_receiver.BindSelf(channel1());
   chan_2_sig_receiver.BindSelf(channel2());
 
-  cricket::SendDataResult result;
+  SendDataResult result;
   ASSERT_TRUE(SendData(channel1(), 1, "hello?", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), 1000);
   ASSERT_TRUE(SendData(channel2(), 2, "hi chan1", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver1(), 2, "hi chan1"), 1000);
 
   // Close one stream on channel1(), while closing three streams on
@@ -484,7 +483,7 @@ TEST_F(SctpDataMediaChannelTest, EngineSignalsRightChannel) {
   EXPECT_TRUE_WAIT(channel1()->socket() != NULL, 1000);
   struct socket *sock = const_cast<struct socket*>(channel1()->socket());
   int prior_count = channel1_ready_to_send_count();
-  cricket::SctpDataEngine::SendThresholdCallback(sock, 0);
+  SctpDataMediaChannel::SendThresholdCallback(sock, 0);
   EXPECT_GT(channel1_ready_to_send_count(), prior_count);
 }
 
@@ -503,12 +502,12 @@ TEST_F(SctpDataMediaChannelTest, RefusesHighNumberedChannels) {
 TEST_F(SctpDataMediaChannelTest, MAYBE_ReusesAStream) {
   // Shut down channel 1, then open it up again for reuse.
   SetupConnectedChannels();
-  cricket::SendDataResult result;
+  SendDataResult result;
   SignalChannelClosedObserver chan_2_sig_receiver;
   chan_2_sig_receiver.BindSelf(channel2());
 
   ASSERT_TRUE(SendData(channel1(), 1, "hello?", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), 1000);
 
   channel1()->RemoveSendStream(1);
@@ -518,8 +517,10 @@ TEST_F(SctpDataMediaChannelTest, MAYBE_ReusesAStream) {
   // Create a new channel 1.
   AddStream(1);
   ASSERT_TRUE(SendData(channel1(), 1, "hi?", &result));
-  EXPECT_EQ(cricket::SDR_SUCCESS, result);
+  EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hi?"), 1000);
   channel1()->RemoveSendStream(1);
   EXPECT_TRUE_WAIT(chan_2_sig_receiver.StreamCloseCount(1) == 2, 1000);
 }
+
+}  // namespace cricket
diff --git a/chromium/third_party/webrtc/modules/audio_codec_speed_tests_apk.isolate b/chromium/third_party/webrtc/modules/audio_codec_speed_tests_apk.isolate
new file mode 100644
index 00000000000..2122df84730
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_codec_speed_tests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'audio_codec_speed_tests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_audio_codec_speed_tests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/audio_codec_speed_tests_apk/',
+      '<(PRODUCT_DIR)/bin/run_audio_codec_speed_tests',
+      'audio_codec_speed_tests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/modules/audio_coding/BUILD.gn b/chromium/third_party/webrtc/modules/audio_coding/BUILD.gn
index 23b5fa46da5..d1f70cf105b 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/BUILD.gn
+++ b/chromium/third_party/webrtc/modules/audio_coding/BUILD.gn
@@ -39,6 +39,33 @@ if (!build_with_mozilla && !build_with_chromium) {
   audio_codec_deps += [ ":red" ]
 }
 
+source_set("audio_decoder_factory_interface") {
+  sources = [
+    "codecs/audio_decoder_factory.h",
+    "codecs/audio_format.cc",
+    "codecs/audio_format.h",
+  ]
+  configs += [ "../..:common_config" ]
+  public_configs = [ "../..:common_inherited_config" ]
+  deps = [
+    "../..:webrtc_common",
+  ]
+}
+
+source_set("builtin_audio_decoder_factory") {
+  sources = [
+    "codecs/builtin_audio_decoder_factory.cc",
+    "codecs/builtin_audio_decoder_factory.h",
+  ]
+  configs += [ "../..:common_config" ]
+  public_configs = [ "../..:common_inherited_config" ]
+  deps = [
+           "../..:webrtc_common",
+           ":audio_decoder_factory_interface",
+         ] + audio_codec_deps
+  defines = audio_codec_defines
+}
+
 source_set("rent_a_codec") {
   sources = [
     "acm2/acm_codec_database.cc",
@@ -121,6 +148,7 @@ source_set("audio_decoder_interface") {
   public_configs = [ "../..:common_inherited_config" ]
   deps = [
     "../..:webrtc_common",
+    "../../base:rtc_base_approved",
   ]
 }
 
@@ -133,6 +161,7 @@ source_set("audio_encoder_interface") {
   public_configs = [ "../..:common_inherited_config" ]
   deps = [
     "../..:webrtc_common",
+    "../../base:rtc_base_approved",
   ]
 }
 
@@ -147,9 +176,7 @@ source_set("cng") {
   sources = [
     "codecs/cng/audio_encoder_cng.cc",
     "codecs/cng/audio_encoder_cng.h",
-    "codecs/cng/cng_helpfuns.c",
-    "codecs/cng/cng_helpfuns.h",
-    "codecs/cng/webrtc_cng.c",
+    "codecs/cng/webrtc_cng.cc",
     "codecs/cng/webrtc_cng.h",
   ]
 
@@ -760,6 +787,8 @@ source_set("neteq") {
     "neteq/buffer_level_filter.h",
     "neteq/comfort_noise.cc",
     "neteq/comfort_noise.h",
+    "neteq/cross_correlation.cc",
+    "neteq/cross_correlation.h",
     "neteq/decision_logic.cc",
     "neteq/decision_logic.h",
     "neteq/decision_logic_fax.cc",
@@ -791,6 +820,8 @@ source_set("neteq") {
     "neteq/neteq_impl.h",
     "neteq/normal.cc",
     "neteq/normal.h",
+    "neteq/packet.cc",
+    "neteq/packet.h",
     "neteq/packet_buffer.cc",
     "neteq/packet_buffer.h",
     "neteq/payload_splitter.cc",
@@ -807,6 +838,8 @@ source_set("neteq") {
     "neteq/statistics_calculator.h",
     "neteq/sync_buffer.cc",
     "neteq/sync_buffer.h",
+    "neteq/tick_timer.cc",
+    "neteq/tick_timer.h",
     "neteq/time_stretch.cc",
     "neteq/time_stretch.h",
     "neteq/timestamp_scaler.cc",
@@ -822,9 +855,11 @@ source_set("neteq") {
 
   deps = [
     ":audio_decoder_interface",
+    ":builtin_audio_decoder_factory",
     ":cng",
     ":g711",
     ":pcm16b",
+    ":rent_a_codec",
     "../..:webrtc_common",
     "../../common_audio",
     "../../system_wrappers",
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
index e4a34a72946..dc8111daf3b 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -158,8 +158,11 @@ void AcmReceiveTestOldApi::Run() {
     // Pull audio until time to insert packet.
     while (clock_.TimeInMilliseconds() < packet->time_ms()) {
       AudioFrame output_frame;
-      EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
+      bool muted;
+      EXPECT_EQ(0,
+                acm_->PlayoutData10Ms(output_freq_hz_, &output_frame, &muted));
       ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
+      ASSERT_FALSE(muted);
       const size_t samples_per_block =
           static_cast<size_t>(output_freq_hz_ * 10 / 1000);
       EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index f8bacf8dc7d..8c07631e3a6 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -26,7 +26,6 @@
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -133,11 +132,13 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
   return 0;
 }
 
-int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
+int AcmReceiver::GetAudio(int desired_freq_hz,
+                          AudioFrame* audio_frame,
+                          bool* muted) {
   // Accessing members, take the lock.
   rtc::CritScope lock(&crit_sect_);
 
-  if (neteq_->GetAudio(audio_frame) != NetEq::kOK) {
+  if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     return -1;
   }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.h b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 6fec1ffdda1..f37212c067b 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -82,11 +82,13 @@ class AcmReceiver {
   // Output:
   //   -audio_frame           : an audio frame were output data and
   //                            associated parameters are written to.
+  //   -muted                 : if true, the sample data in audio_frame is not
+  //                            populated, and must be interpreted as all zero.
   //
   // Return value             : 0 if OK.
   //                           -1 if NetEq returned an error.
   //
-  int GetAudio(int desired_freq_hz, AudioFrame* audio_frame);
+  int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
 
   //
   // Adds a new codec to the NetEq codec database.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index c39a7cc797f..b57b7ef446e 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -285,7 +285,8 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
     const int num_10ms_frames = codec.inst.pacsize / (codec.inst.plfreq / 100);
     InsertOnePacketOfSilence(codec.id);
     for (int k = 0; k < num_10ms_frames; ++k) {
-      EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame));
+      bool muted;
+      EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
     }
     EXPECT_EQ(codec.inst.plfreq, receiver_->last_output_sample_rate_hz());
   }
@@ -326,13 +327,15 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
         rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
 
     AudioFrame frame;
-    EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
+    bool muted;
+    EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
     // Expect timestamp = 0 before first packet is inserted.
     EXPECT_EQ(0u, frame.timestamp_);
     for (int i = 0; i < 5; ++i) {
       InsertOnePacketOfSilence(codec.id);
       for (int k = 0; k < num_10ms_frames; ++k) {
-        EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
+        EXPECT_EQ(0,
+                  receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
         EXPECT_EQ(expected_output_ts, frame.timestamp_);
         expected_output_ts += 10 * samples_per_ms;
         EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
@@ -340,6 +343,7 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
         EXPECT_EQ(output_channels, frame.num_channels_);
         EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
         EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
+        EXPECT_FALSE(muted);
       }
     }
   }
@@ -388,8 +392,10 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
   AudioFrame frame;
   for (int n = 0; n < kNumPackets; ++n) {
     InsertOnePacketOfSilence(codec.id);
-    for (int k = 0; k < num_10ms_frames; ++k)
-      ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      bool muted;
+      ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame, &muted));
+    }
   }
   EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
 }
@@ -417,8 +423,10 @@ TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
   AudioFrame frame;
   for (int n = 0; n < kNumPackets; ++n) {
     InsertOnePacketOfSilence(codec.id);
-    for (int k = 0; k < num_10ms_frames; ++k)
-      ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame));
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      bool muted;
+      ASSERT_EQ(0, receiver_->GetAudio(codec.inst.plfreq, &frame, &muted));
+    }
   }
   EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
index cfee3530206..938e39e2e69 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
@@ -47,7 +47,7 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
   // Returns the next encoded packet. Returns NULL if the test duration was
   // exceeded. Ownership of the packet is handed over to the caller.
   // Inherited from PacketSource.
-  Packet* NextPacket();
+  Packet* NextPacket() override;
 
   // Inherited from AudioPacketizationCallback.
   int32_t SendData(FrameType frame_type,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
index 254c2f420bf..bc7197d8e79 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -29,6 +29,18 @@
 
 namespace webrtc {
 
+namespace {
+
+// Adds a codec usage sample to the histogram.
+void UpdateCodecTypeHistogram(size_t codec_type) {
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.Audio.Encoder.CodecType", static_cast<int>(codec_type),
+      static_cast<int>(
+          webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
+}
+
+}  // namespace
+
 namespace acm2 {
 
 struct EncoderFactory {
@@ -104,7 +116,6 @@ void ConvertEncodedInfoToFragmentationHeader(
 class RawAudioEncoderWrapper final : public AudioEncoder {
  public:
   RawAudioEncoderWrapper(AudioEncoder* enc) : enc_(enc) {}
-  size_t MaxEncodedBytes() const override { return enc_->MaxEncodedBytes(); }
   int SampleRateHz() const override { return enc_->SampleRateHz(); }
   size_t NumChannels() const override { return enc_->NumChannels(); }
   int RtpTimestampRateHz() const override { return enc_->RtpTimestampRateHz(); }
@@ -120,13 +131,6 @@ class RawAudioEncoderWrapper final : public AudioEncoder {
                          rtc::Buffer* encoded) override {
     return enc_->Encode(rtp_timestamp, audio, encoded);
   }
-  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             rtc::ArrayView<const int16_t> audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override {
-    return enc_->EncodeInternal(rtp_timestamp, audio, max_encoded_bytes,
-                                encoded);
-  }
   void Reset() override { return enc_->Reset(); }
   bool SetFec(bool enable) override { return enc_->SetFec(enable); }
   bool SetDtx(bool enable) override { return enc_->SetDtx(enable); }
@@ -193,7 +197,9 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
       first_10ms_data_(false),
       first_frame_(true),
       packetization_callback_(NULL),
-      vad_callback_(NULL) {
+      vad_callback_(NULL),
+      codec_histogram_bins_log_(),
+      number_of_consecutive_empty_packets_(0) {
   if (InitializeReceiverSafe() < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot initialize receiver");
@@ -239,6 +245,20 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   }
   previous_pltype = previous_pltype_;  // Read it while we have the critsect.
 
+  // Log codec type to histogram once every 500 packets.
+  if (encoded_info.encoded_bytes == 0) {
+    ++number_of_consecutive_empty_packets_;
+  } else {
+    size_t codec_type = static_cast<size_t>(encoded_info.encoder_type);
+    codec_histogram_bins_log_[codec_type] +=
+        number_of_consecutive_empty_packets_ + 1;
+    number_of_consecutive_empty_packets_ = 0;
+    if (codec_histogram_bins_log_[codec_type] >= 500) {
+      codec_histogram_bins_log_[codec_type] -= 500;
+      UpdateCodecTypeHistogram(codec_type);
+    }
+  }
+
   RTPFragmentationHeader my_fragmentation;
   ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
   FrameType frame_type;
@@ -727,10 +747,12 @@ int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked(
 
   AudioDecoder* isac_decoder = nullptr;
   if (STR_CASE_CMP(codec.plname, "isac") == 0) {
-    if (!isac_decoder_) {
-      isac_decoder_ = isac_factory();
+    std::unique_ptr<AudioDecoder>& saved_isac_decoder =
+        codec.plfreq == 16000 ? isac_decoder_16k_ : isac_decoder_32k_;
+    if (!saved_isac_decoder) {
+      saved_isac_decoder = isac_factory();
     }
-    isac_decoder = isac_decoder_.get();
+    isac_decoder = saved_isac_decoder.get();
   }
   return receiver_.AddCodec(*codec_index, codec.pltype, codec.channels,
                             codec.plfreq, isac_decoder, codec.plname);
@@ -797,9 +819,10 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
 // Get 10 milliseconds of raw audio data to play out.
 // Automatic resample to the requested frequency.
 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
-                                           AudioFrame* audio_frame) {
+                                           AudioFrame* audio_frame,
+                                           bool* muted) {
   // GetAudio always returns 10 ms, at the requested sample rate.
-  if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
+  if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "PlayoutData failed, RecOut Failed");
     return -1;
@@ -808,6 +831,14 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
   return 0;
 }
 
+int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
+                                           AudioFrame* audio_frame) {
+  bool muted;
+  int ret = PlayoutData10Ms(desired_freq_hz, audio_frame, &muted);
+  RTC_DCHECK(!muted);
+  return ret;
+}
+
 /////////////////////////////////////////
 //   Statistics
 //
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index 63dfb810567..c098e62b991 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -23,6 +23,7 @@
 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
 
@@ -163,6 +164,9 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
 
   // Get 10 milliseconds of raw audio data to play out, and
   // automatic resample to the requested frequency if > 0.
+  int PlayoutData10Ms(int desired_freq_hz,
+                      AudioFrame* audio_frame,
+                      bool* muted) override;
   int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
 
   /////////////////////////////////////////
@@ -268,7 +272,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // RegisterEncoder.
   std::unique_ptr<AudioEncoder> encoder_stack_ GUARDED_BY(acm_crit_sect_);
 
-  std::unique_ptr<AudioDecoder> isac_decoder_ GUARDED_BY(acm_crit_sect_);
+  std::unique_ptr<AudioDecoder> isac_decoder_16k_ GUARDED_BY(acm_crit_sect_);
+  std::unique_ptr<AudioDecoder> isac_decoder_32k_ GUARDED_BY(acm_crit_sect_);
 
   // This is to keep track of CN instances where we can send DTMFs.
   uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
@@ -294,6 +299,10 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
+
+  int codec_histogram_bins_log_[static_cast<size_t>(
+      AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
+  int number_of_consecutive_empty_packets_;
 };
 
 }  // namespace acm2
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 6e004f9e28d..470f690ed9c 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -205,7 +205,9 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
 
   virtual void PullAudio() {
     AudioFrame audio_frame;
-    ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame));
+    bool muted;
+    ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
   }
 
   virtual void InsertAudio() {
@@ -296,7 +298,9 @@ TEST_F(AudioCodingModuleTestOldApi, MAYBE_NetEqCalls) {
 TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   AudioFrame audio_frame;
   const int kSampleRateHz = 32000;
-  EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
+  bool muted;
+  EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
+  ASSERT_FALSE(muted);
   EXPECT_EQ(id_, audio_frame.id_);
   EXPECT_EQ(0u, audio_frame.timestamp_);
   EXPECT_GT(audio_frame.num_channels_, 0u);
@@ -307,7 +311,8 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
 
 TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
   AudioFrame audio_frame;
-  EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame));
+  bool muted;
+  EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame, &muted));
 }
 
 // Checks that the transport callback is invoked once for each speech packet.
@@ -608,7 +613,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
 
   ~AcmIsacMtTestOldApi() {}
 
-  void SetUp() {
+  void SetUp() override {
     AudioCodingModuleTestOldApi::SetUp();
     RegisterCodec();  // Must be called before the threads start below.
 
@@ -642,7 +647,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
     ASSERT_EQ(0, acm_->RegisterSendCodec(codec_));
   }
 
-  void InsertPacket() {
+  void InsertPacket() override {
     int num_calls = packet_cb_.num_calls();  // Store locally for thread safety.
     if (num_calls > last_packet_number_) {
       // Get the new payload out from the callback handler.
@@ -661,7 +666,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
             &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
   }
 
-  void InsertAudio() {
+  void InsertAudio() override {
     // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
     // this call confuses the number of samples with the number of bytes, and
     // ends up copying only half of what it should.
@@ -677,7 +682,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but
   // here it is using the constants defined in this class (i.e., shorter test
   // run).
-  virtual bool TestDone() {
+  bool TestDone() override {
     if (packet_cb_.num_calls() > kNumPackets) {
       rtc::CritScope lock(&crit_sect_);
       if (pull_audio_count_ > kNumPullCalls) {
@@ -728,7 +733,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
     clock_ = fake_clock_.get();
   }
 
-  void SetUp() {
+  void SetUp() override {
     AudioCodingModuleTestOldApi::SetUp();
     // Set up input audio source to read from specified file, loop after 5
     // seconds, and deliver blocks of 10 ms.
@@ -757,7 +762,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
     codec_registration_thread_.SetPriority(rtc::kRealtimePriority);
   }
 
-  void TearDown() {
+  void TearDown() override {
     AudioCodingModuleTestOldApi::TearDown();
     receive_thread_.Stop();
     codec_registration_thread_.Stop();
@@ -806,8 +811,13 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
     // Pull audio.
     for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) {
       AudioFrame audio_frame;
+      bool muted;
       EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */,
-                                         &audio_frame));
+                                         &audio_frame, &muted));
+      if (muted) {
+        ADD_FAILURE();
+        return false;
+      }
       fake_clock_->AdvanceTimeMilliseconds(10);
     }
     rtp_utility_->Forward(&rtp_header_);
@@ -939,34 +949,34 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
 TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
-  Run(8000, PlatformChecksum("d9334a99c7f185860028e6f08e5b7390",
-                             "946803da293ef3fa39242d3059eac491",
-                             "efb5a07480bad8afb184c4150f4b3f3a",
-                             "51717ab374871cbfa2c6977ea2aa40f3"),
+  Run(8000, PlatformChecksum("90be25dd9505005aaadf91b77ee31624",
+                             "ac6dc4b5bf6d277f693889c4c916882e",
+                             "a607f7d0ba98683c9c236217f86aaa6b",
+                             "4a54f6ec712bda58484a388e1a332b42"),
       std::vector<ExternalDecoder>());
 }
 
 TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
-  Run(16000, PlatformChecksum("9ad7d5a5f3c9fac4e880a6fbfd9d3ac8",
-                              "4fc1b82404ae33511c1cdb385774b2a4",
-                              "f580bfd4e5e29f0399b61b7512d4e3b4",
-                              "5b2ae32c590b41d0c601179e14eaae96"),
+  Run(16000, PlatformChecksum("2c713197d41becd52c1ceecbd2b9f687",
+                              "130cc2a43063c74197122e3760690e7d",
+                              "cdc3d88f6d8e497d4e00c62c0e6dbb3c",
+                              "83edb67c157d0e3a0fb9f7d7b1ce5dc7"),
       std::vector<ExternalDecoder>());
 }
 
 TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
-  Run(32000, PlatformChecksum("08e6085ccb96494b242f0ecc4c8a2dc8",
-                              "d1f853b1e046c67c9ee186786eaf2124",
-                              "fdf5166b98c43235978685e40e28fea6",
-                              "7f620312f2fa74a10048bbb7739d4bf3"),
+  Run(32000, PlatformChecksum("fe5851d43c13df66a7ad30fdb124e62f",
+                              "309d24be4b287dc92c340f10a807a11e",
+                              "c4a0e0b2e031d62c693af2a9ff4337ac",
+                              "4cbfc6ab4d704f5d9b4f10406437fda9"),
       std::vector<ExternalDecoder>());
 }
 
 TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
-  Run(48000, PlatformChecksum("31343887b7ef70772df733d072b0dd00",
-                              "f6893278d75dad42ac44bff77f674b33",
-                              "71f89e87ee1bad594f529d6c036289ad",
-                              "b64c891e99eccc9ff45541ef67c9e9bf"),
+  Run(48000, PlatformChecksum("a9241f426b4bf2ac650b6d287469a550",
+                              "30374fd4a932df942c1b1120e7b724ad",
+                              "22242dd832824046d48db9ea8a01f84c",
+                              "c7f46bf165400b266d9b57aee02d2747"),
       std::vector<ExternalDecoder>());
 }
 
@@ -1021,10 +1031,10 @@ TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
   std::vector<ExternalDecoder> external_decoders;
   external_decoders.push_back(ed);
 
-  Run(48000, PlatformChecksum("31343887b7ef70772df733d072b0dd00",
-                              "f6893278d75dad42ac44bff77f674b33",
-                              "71f89e87ee1bad594f529d6c036289ad",
-                              "b64c891e99eccc9ff45541ef67c9e9bf"),
+  Run(48000, PlatformChecksum("a9241f426b4bf2ac650b6d287469a550",
+                              "30374fd4a932df942c1b1120e7b724ad",
+                              "22242dd832824046d48db9ea8a01f84c",
+                              "c7f46bf165400b266d9b57aee02d2747"),
       external_decoders);
 
   EXPECT_CALL(mock_decoder, Die());
@@ -1737,7 +1747,7 @@ class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test,
   }
 
   // Inherited from test::AudioSink.
-  bool WriteArray(const int16_t* audio, size_t num_samples) {
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
     // Skip checking the first output frame, since it has a number of zeros
     // due to how NetEq is initialized.
     if (first_output_) {
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.cc b/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.cc
index 81adf81a83c..f028c45f991 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.cc
@@ -113,7 +113,7 @@ bool CodecManager::RegisterEncoder(const CodecInst& send_codec) {
   }
 
   send_codec_inst_ = rtc::Optional<CodecInst>(send_codec);
-  codec_stack_params_.speech_encoder.reset();  // Caller must recreate it.
+  recreate_encoder_ = true;  // Caller must recreate it.
   return true;
 }
 
@@ -190,5 +190,67 @@ bool CodecManager::SetCodecFEC(bool enable_codec_fec) {
   return true;
 }
 
+bool CodecManager::MakeEncoder(RentACodec* rac, AudioCodingModule* acm) {
+  RTC_DCHECK(rac);
+  RTC_DCHECK(acm);
+
+  if (!recreate_encoder_) {
+    bool error = false;
+    // Try to re-use the speech encoder we've given to the ACM.
+    acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+      if (!*encoder) {
+        // There is no existing encoder.
+        recreate_encoder_ = true;
+        return;
+      }
+
+      // Extract the speech encoder from the ACM.
+      std::unique_ptr<AudioEncoder> enc = std::move(*encoder);
+      while (true) {
+        auto sub_enc = enc->ReclaimContainedEncoders();
+        if (sub_enc.empty()) {
+          break;
+        }
+        RTC_CHECK_EQ(1u, sub_enc.size());
+
+        // Replace enc with its sub encoder. We need to put the sub encoder in
+        // a temporary first, since otherwise the old value of enc would be
+        // destroyed before the new value got assigned, which would be bad
+        // since the new value is a part of the old value.
+        auto tmp_enc = std::move(sub_enc[0]);
+        enc = std::move(tmp_enc);
+      }
+
+      // Wrap it in a new encoder stack and put it back.
+      codec_stack_params_.speech_encoder = std::move(enc);
+      *encoder = rac->RentEncoderStack(&codec_stack_params_);
+      if (!*encoder) {
+        error = true;
+      }
+    });
+    if (error) {
+      return false;
+    }
+    if (!recreate_encoder_) {
+      return true;
+    }
+  }
+
+  if (!send_codec_inst_) {
+    // We don't have the information we need to create a new speech encoder.
+    // (This is not an error.)
+    return true;
+  }
+
+  codec_stack_params_.speech_encoder = rac->RentEncoder(*send_codec_inst_);
+  auto stack = rac->RentEncoderStack(&codec_stack_params_);
+  if (!stack) {
+    return false;
+  }
+  acm->SetEncoder(std::move(stack));
+  recreate_encoder_ = false;
+  return true;
+}
+
 }  // namespace acm2
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.h b/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.h
index f6c6cd46d2c..b60b7e7bcbd 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/codec_manager.h
@@ -59,31 +59,13 @@ class CodecManager final {
   // Uses the provided Rent-A-Codec to create a new encoder stack, if we have a
   // complete specification; if so, it is then passed to set_encoder. On error,
   // returns false.
-  bool MakeEncoder(RentACodec* rac, AudioCodingModule* acm) {
-    RTC_DCHECK(rac);
-    RTC_DCHECK(acm);
-    if (!codec_stack_params_.speech_encoder && send_codec_inst_) {
-      // We have no speech encoder, but we have a specification for making one.
-      auto enc = rac->RentEncoder(*send_codec_inst_);
-      if (!enc)
-        return false;
-      codec_stack_params_.speech_encoder = std::move(enc);
-    }
-    auto stack = rac->RentEncoderStack(&codec_stack_params_);
-    if (stack) {
-      // Give new encoder stack to the ACM.
-      acm->SetEncoder(std::move(stack));
-    } else {
-      // The specification was good but incomplete, so we have no encoder stack
-      // to give to the ACM.
-    }
-    return true;
-  }
+  bool MakeEncoder(RentACodec* rac, AudioCodingModule* acm);
 
  private:
   rtc::ThreadChecker thread_checker_;
   rtc::Optional<CodecInst> send_codec_inst_;
   RentACodec::StackParameters codec_stack_params_;
+  bool recreate_encoder_ = true;  // Need to recreate encoder?
 
   RTC_DISALLOW_COPY_AND_ASSIGN(CodecManager);
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
index 7f1e52030dc..a61f15949d7 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
@@ -47,6 +47,59 @@
 namespace webrtc {
 namespace acm2 {
 
+rtc::Optional<SdpAudioFormat> RentACodec::NetEqDecoderToSdpAudioFormat(
+    NetEqDecoder nd) {
+  switch (nd) {
+    case NetEqDecoder::kDecoderPCMu:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcmu", 8000, 1));
+    case NetEqDecoder::kDecoderPCMa:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcma", 8000, 1));
+    case NetEqDecoder::kDecoderPCMu_2ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcmu", 8000, 2));
+    case NetEqDecoder::kDecoderPCMa_2ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcma", 8000, 2));
+    case NetEqDecoder::kDecoderILBC:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("ilbc", 8000, 1));
+    case NetEqDecoder::kDecoderISAC:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("isac", 16000, 1));
+    case NetEqDecoder::kDecoderISACswb:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("isac", 32000, 1));
+    case NetEqDecoder::kDecoderPCM16B:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 1));
+    case NetEqDecoder::kDecoderPCM16Bwb:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 16000, 1));
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 32000, 1));
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 48000, 1));
+    case NetEqDecoder::kDecoderPCM16B_2ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 2));
+    case NetEqDecoder::kDecoderPCM16Bwb_2ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 16000, 2));
+    case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 32000, 2));
+    case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 48000, 2));
+    case NetEqDecoder::kDecoderPCM16B_5ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 5));
+    case NetEqDecoder::kDecoderG722:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("g722", 8000, 1));
+    case NetEqDecoder::kDecoderG722_2ch:
+      return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("g722", 8000, 2));
+    case NetEqDecoder::kDecoderOpus:
+      return rtc::Optional<SdpAudioFormat>(
+          SdpAudioFormat("opus", 48000, 2,
+                         std::map<std::string, std::string>{{"stereo", "0"}}));
+    case NetEqDecoder::kDecoderOpus_2ch:
+      return rtc::Optional<SdpAudioFormat>(
+          SdpAudioFormat("opus", 48000, 2,
+                         std::map<std::string, std::string>{{"stereo", "1"}}));
+
+    default:
+      return rtc::Optional<SdpAudioFormat>();
+  }
+}
+
 rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
     const char* payload_name,
     int sampling_freq_hz,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.h
index a4026acd283..bac37afa963 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/acm2/rent_a_codec.h
@@ -20,6 +20,7 @@
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/typedefs.h"
@@ -135,6 +136,9 @@ class RentACodec {
     kDecoderOpus_2ch,
   };
 
+  static rtc::Optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(
+      NetEqDecoder nd);
+
   static inline size_t NumberOfCodecs() {
     return static_cast<size_t>(CodecId::kNumCodecs);
   }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/audio_coding.gypi b/chromium/third_party/webrtc/modules/audio_coding/audio_coding.gypi
index e460853ac10..af2cb3c49dc 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/audio_coding.gypi
+++ b/chromium/third_party/webrtc/modules/audio_coding/audio_coding.gypi
@@ -69,6 +69,50 @@
   },
   'targets': [
     {
+      'target_name': 'audio_decoder_factory_interface',
+      'type': 'static_library',
+      'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
+      ],
+      'include_dirs': [
+        '<(webrtc_root)',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '<(webrtc_root)',
+        ],
+      },
+      'sources': [
+        'codecs/audio_decoder_factory.h',
+        'codecs/audio_format.cc',
+        'codecs/audio_format.h',
+      ],
+    },
+    {
+      'target_name': 'builtin_audio_decoder_factory',
+      'type': 'static_library',
+      'defines': [
+        '<@(audio_codec_defines)',
+      ],
+      'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<@(audio_codec_dependencies)',
+        'audio_decoder_factory_interface',
+      ],
+      'include_dirs': [
+        '<(webrtc_root)',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          '<(webrtc_root)',
+        ],
+      },
+      'sources': [
+        'codecs/builtin_audio_decoder_factory.cc',
+        'codecs/builtin_audio_decoder_factory.h',
+      ],
+    },
+    {
       'target_name': 'rent_a_codec',
       'type': 'static_library',
       'defines': [
diff --git a/chromium/third_party/webrtc/modules/audio_coding/audio_coding_tests.gypi b/chromium/third_party/webrtc/modules/audio_coding/audio_coding_tests.gypi
index e60309a6dfa..0b1a22d8d04 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/audio_coding_tests.gypi
+++ b/chromium/third_party/webrtc/modules/audio_coding/audio_coding_tests.gypi
@@ -39,17 +39,4 @@
       ],
     },
   ],
-  'conditions': [
-    ['OS=="android"', {
-      'targets': [
-        {
-          'target_name': 'audio_codec_speed_tests_apk_target',
-          'type': 'none',
-          'dependencies': [
-            '<(apk_tests_path):audio_codec_speed_tests_apk',
-          ],
-        },
-      ],
-    }],
-  ],
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.cc
index d2984b97b09..442ddc1e4b8 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.cc
@@ -82,11 +82,6 @@ bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
   return false;
 }
 
-CNG_dec_inst* AudioDecoder::CngDecoderInstance() {
-  FATAL() << "Not a CNG decoder";
-  return NULL;
-}
-
 AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
   switch (type) {
     case 0:  // TODO(hlundin): Both iSAC and Opus return 0 for speech.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.h
index 81ac8731830..580ddbf74ff 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -14,7 +14,6 @@
 #include <stdlib.h>  // NULL
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -94,10 +93,6 @@ class AudioDecoder {
   // Returns true if the packet has FEC and false otherwise.
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 
-  // If this is a CNG decoder, return the underlying CNG_dec_inst*. If this
-  // isn't a CNG decoder, don't call this method.
-  virtual CNG_dec_inst* CngDecoderInstance();
-
   virtual size_t Channels() const = 0;
 
  protected:
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder_factory.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder_factory.h
new file mode 100644
index 00000000000..12b97780918
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder_factory.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format.h"
+
+namespace webrtc {
+
+// A factory that creates AudioDecoders.
+// NOTE: This class is still under development and may change without notice.
+class AudioDecoderFactory {
+ public:
+  virtual ~AudioDecoderFactory() = default;
+
+  virtual std::vector<SdpAudioFormat> GetSupportedFormats() = 0;
+
+  virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder_factory_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder_factory_unittest.cc
new file mode 100644
index 00000000000..12a0a4047e8
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_decoder_factory_unittest.cc
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+
+namespace webrtc {
+
+TEST(AudioDecoderFactoryTest, CreateUnknownDecoder) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1)));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcmu) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMu supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1)));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcma) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMa supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1)));
+}
+
+TEST(AudioDecoderFactoryTest, CreateIlbc) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // iLBC supports 8 kHz, 1 channel.
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1)));
+
+  // iLBC actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
+  // TODO(kwiberg): Uncomment this once AudioDecoder has a SampleRateHz method.
+  // std::unique_ptr<AudioDecoder> dec =
+  //    adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1));
+  // EXPECT_EQ(16000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTest, CreateIsac) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // iSAC supports 16 kHz, 1 channel. The float implementation additionally
+  // supports 32 kHz, 1 channel.
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1)));
+#ifdef WEBRTC_ARCH_ARM
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1)));
+#else
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1)));
+#endif
+}
+
+TEST(AudioDecoderFactoryTest, CreateL16) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // L16 supports any clock rate, any number of channels.
+  const int clockrates[] = {8000, 16000, 32000, 48000};
+  const int num_channels[] = {1, 2, 3, 4711};
+  for (int clockrate : clockrates) {
+    EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0)));
+    for (int channels : num_channels) {
+      EXPECT_TRUE(
+          adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, channels)));
+    }
+  }
+}
+
+TEST(AudioDecoderFactoryTest, CreateG722) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // g722 supports 8 kHz, 1-2 channels.
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1)));
+  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1)));
+  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1)));
+}
+
+TEST(AudioDecoderFactoryTest, CreateOpus) {
+  std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // Opus supports 48 kHz, 2 channels, and wants a "stereo" parameter whose
+  // value is either "0" or "1".
+  for (int hz : {8000, 16000, 32000, 48000}) {
+    for (int channels : {0, 1, 2, 3}) {
+      for (std::string stereo : {"XX", "0", "1", "2"}) {
+        std::map<std::string, std::string> params;
+        if (stereo != "XX") {
+          params["stereo"] = stereo;
+        }
+        bool good =
+            (hz == 48000 && channels == 2 && (stereo == "0" || stereo == "1"));
+        EXPECT_EQ(good, static_cast<bool>(adf->MakeAudioDecoder(SdpAudioFormat(
+                            "opus", hz, channels, std::move(params)))));
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 6f793e25314..6b7f5f893fd 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -16,8 +16,13 @@
 namespace webrtc {
 
 AudioEncoder::EncodedInfo::EncodedInfo() = default;
-
+AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
+    const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
+    default;
 
 int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
@@ -37,55 +42,6 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode(
   return info;
 }
 
-AudioEncoder::EncodedInfo AudioEncoder::Encode(
-    uint32_t rtp_timestamp,
-    rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded) {
-  return DEPRECATED_Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
-}
-
-AudioEncoder::EncodedInfo AudioEncoder::DEPRECATED_Encode(
-    uint32_t rtp_timestamp,
-    rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded) {
-  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
-  RTC_CHECK_EQ(audio.size(),
-               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
-  EncodedInfo info =
-      EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
-  RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
-  return info;
-}
-
-AudioEncoder::EncodedInfo AudioEncoder::EncodeImpl(
-    uint32_t rtp_timestamp,
-    rtc::ArrayView<const int16_t> audio,
-    rtc::Buffer* encoded)
-{
-  EncodedInfo info;
-  encoded->AppendData(MaxEncodedBytes(), [&] (rtc::ArrayView<uint8_t> encoded) {
-      info = EncodeInternal(rtp_timestamp, audio,
-                            encoded.size(), encoded.data());
-      return info.encoded_bytes;
-    });
-  return info;
-}
-
-AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal(
-    uint32_t rtp_timestamp,
-    rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded)
-{
-  rtc::Buffer temp_buffer;
-  EncodedInfo info = EncodeImpl(rtp_timestamp, audio, &temp_buffer);
-  RTC_DCHECK_LE(temp_buffer.size(), max_encoded_bytes);
-  std::memcpy(encoded, temp_buffer.data(), info.encoded_bytes);
-  return info;
-}
-
 bool AudioEncoder::SetFec(bool enable) {
   return !enable;
 }
@@ -104,4 +60,7 @@ void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoder::ReclaimContainedEncoders() { return nullptr; }
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 3fdee259ce7..ecc28d96a16 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -25,12 +25,32 @@ namespace webrtc {
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
+  // Used for UMA logging of codec usage. The same codecs, with the
+  // same values, must be listed in
+  // src/tools/metrics/histograms/histograms.xml in chromium to log
+  // correct values.
+  enum class CodecType {
+    kOther = 0,  // Codec not specified, and/or not listed in this enum
+    kOpus = 1,
+    kIsac = 2,
+    kPcmA = 3,
+    kPcmU = 4,
+    kG722 = 5,
+    kIlbc = 6,
+
+    // Number of histogram bins in the UMA logging of codec types. The
+    // total number of different codecs that are logged cannot exceed this
+    // number.
+    kMaxLoggedAudioCodecTypes
+  };
+
   struct EncodedInfoLeaf {
     size_t encoded_bytes = 0;
     uint32_t encoded_timestamp = 0;
     int payload_type = 0;
     bool send_even_if_empty = false;
     bool speech = true;
+    CodecType encoder_type = CodecType::kOther;
   };
 
   // This is the main struct for auxiliary encoding information. Each encoded
@@ -45,21 +65,17 @@ class AudioEncoder {
   // vector.
   struct EncodedInfo : public EncodedInfoLeaf {
     EncodedInfo();
+    EncodedInfo(const EncodedInfo&);
+    EncodedInfo(EncodedInfo&&);
     ~EncodedInfo();
+    EncodedInfo& operator=(const EncodedInfo&);
+    EncodedInfo& operator=(EncodedInfo&&);
 
     std::vector<EncodedInfoLeaf> redundant;
   };
 
   virtual ~AudioEncoder() = default;
 
-  // Returns the maximum number of bytes that can be produced by the encoder
-  // at each Encode() call. The caller can use the return value to determine
-  // the size of the buffer that needs to be allocated. This value is allowed
-  // to depend on encoder parameters like bitrate, frame size etc., so if
-  // any of these change, the caller of Encode() is responsible for checking
-  // that the buffer is large enough by calling MaxEncodedBytes() again.
-  virtual size_t MaxEncodedBytes() const = 0;
-
   // Returns the input sample rate in Hz and the number of input channels.
   // These are constants set at instantiation time.
   virtual int SampleRateHz() const = 0;
@@ -95,33 +111,6 @@ class AudioEncoder {
                      rtc::ArrayView<const int16_t> audio,
                      rtc::Buffer* encoded);
 
-  // Deprecated interface to Encode (remove eventually, bug 5591). May incur a
-  // copy. The encoder produces zero or more bytes of output in |encoded| and
-  // returns additional encoding information.  The caller is responsible for
-  // making sure that |max_encoded_bytes| is not smaller than the number of
-  // bytes actually produced by the encoder.
-  RTC_DEPRECATED EncodedInfo Encode(uint32_t rtp_timestamp,
-                                    rtc::ArrayView<const int16_t> audio,
-                                    size_t max_encoded_bytes,
-                                    uint8_t* encoded);
-
-  EncodedInfo DEPRECATED_Encode(uint32_t rtp_timestamp,
-                                rtc::ArrayView<const int16_t> audio,
-                                size_t max_encoded_bytes,
-                                uint8_t* encoded);
-
-  // Deprecated interface EncodeInternal (see bug 5591). May incur a copy.
-  // Subclasses implement this to perform the actual encoding. Called by
-  // Encode(). By default, this is implemented as a call to the newer
-  // EncodeImpl() that accepts an rtc::Buffer instead of a raw pointer.
-  // That version is protected, so see below. At least one of EncodeInternal
-  // or EncodeImpl _must_ be implemented by a subclass.
-  virtual EncodedInfo EncodeInternal(
-      uint32_t rtp_timestamp,
-      rtc::ArrayView<const int16_t> audio,
-      size_t max_encoded_bytes,
-      uint8_t* encoded);
-
   // Resets the encoder to its starting state, discarding any input that has
   // been fed to the encoder but not yet emitted in a packet.
   virtual void Reset() = 0;
@@ -160,15 +149,21 @@ class AudioEncoder {
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
 
+  // Causes this encoder to let go of any other encoders it contains, and
+  // returns a pointer to an array where they are stored (which is required to
+  // live as long as this encoder). Unless the returned array is empty, you may
+  // not call any methods on this encoder afterwards, except for the
+  // destructor. The default implementation just returns an empty array.
+  // NOTE: This method is subject to change. Do not call or override it.
+  virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+  ReclaimContainedEncoders();
+
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
-  // Encode(). For compatibility reasons, this is implemented by default as a
-  // call to the older interface EncodeInternal(). At least one of
-  // EncodeInternal or EncodeImpl _must_ be implemented by a
-  // subclass. Preferably this one.
+  // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
-                                 rtc::Buffer* encoded);
+                                 rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc
deleted file mode 100644
index 71ffcde323b..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_encoder_unittest.cc
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-
-using ::testing::_;
-using ::testing::Invoke;
-using ::testing::Return;
-
-namespace webrtc {
-
-TEST(AudioEncoderTest, EncodeInternalRedirectsOk) {
-  const size_t kPayloadSize = 16;
-  const uint8_t payload[kPayloadSize] =
-      {0xf, 0xe, 0xd, 0xc, 0xb, 0xa, 0x9, 0x8,
-       0x7, 0x6, 0x5, 0x4, 0x3, 0x2, 0x1, 0x0};
-
-  MockAudioEncoderDeprecated old_impl;
-  MockAudioEncoder new_impl;
-  MockAudioEncoderBase* impls[] = { &old_impl, &new_impl };
-  for (auto* impl : impls) {
-    EXPECT_CALL(*impl, Die());
-    EXPECT_CALL(*impl, MaxEncodedBytes())
-        .WillRepeatedly(Return(kPayloadSize * 2));
-    EXPECT_CALL(*impl, NumChannels()).WillRepeatedly(Return(1));
-    EXPECT_CALL(*impl, SampleRateHz()).WillRepeatedly(Return(8000));
-  }
-
-  EXPECT_CALL(old_impl, EncodeInternal(_, _, _, _)).WillOnce(
-      Invoke(MockAudioEncoderDeprecated::CopyEncoding(payload)));
-
-  EXPECT_CALL(new_impl, EncodeImpl(_, _, _)).WillOnce(
-      Invoke(MockAudioEncoder::CopyEncoding(payload)));
-
-  int16_t audio[80];
-  uint8_t output_array[kPayloadSize * 2];
-  rtc::Buffer output_buffer;
-
-  AudioEncoder* old_encoder = &old_impl;
-  AudioEncoder* new_encoder = &new_impl;
-  auto old_info = old_encoder->Encode(0, audio, &output_buffer);
-  auto new_info = new_encoder->DEPRECATED_Encode(0, audio,
-                                                 kPayloadSize * 2,
-                                                 output_array);
-
-  EXPECT_EQ(old_info.encoded_bytes, kPayloadSize);
-  EXPECT_EQ(new_info.encoded_bytes, kPayloadSize);
-  EXPECT_EQ(output_buffer.size(), kPayloadSize);
-
-  for (size_t i = 0; i != kPayloadSize; ++i) {
-    EXPECT_EQ(output_buffer.data()[i], payload[i]);
-    EXPECT_EQ(output_array[i], payload[i]);
-  }
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_format.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_format.cc
new file mode 100644
index 00000000000..bb69cbdb2f7
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_format.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/audio_format.h"
+
+namespace webrtc {
+
+SdpAudioFormat::SdpAudioFormat() = default;
+SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
+SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
+
+SdpAudioFormat::SdpAudioFormat(const char* name,
+                               int clockrate_hz,
+                               int num_channels)
+    : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
+
+SdpAudioFormat::SdpAudioFormat(const char* name,
+                               int clockrate_hz,
+                               int num_channels,
+                               Parameters&& param)
+    : name(name),
+      clockrate_hz(clockrate_hz),
+      num_channels(num_channels),
+      parameters(std::move(param)) {}
+
+SdpAudioFormat::~SdpAudioFormat() = default;
+SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
+SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
+
+void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
+  using std::swap;
+  swap(a.name, b.name);
+  swap(a.clockrate_hz, b.clockrate_hz);
+  swap(a.num_channels, b.num_channels);
+  swap(a.parameters, b.parameters);
+}
+
+std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
+  os << "{name: " << saf.name;
+  os << ", clockrate_hz: " << saf.clockrate_hz;
+  os << ", num_channels: " << saf.num_channels;
+  os << ", parameters: {";
+  const char* sep = "";
+  for (const auto& kv : saf.parameters) {
+    os << sep << kv.first << ": " << kv.second;
+    sep = ", ";
+  }
+  os << "}}";
+  return os;
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_format.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_format.h
new file mode 100644
index 00000000000..61c0dd9f6fa
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/audio_format.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
+
+#include <map>
+#include <ostream>
+#include <string>
+#include <utility>
+
+namespace webrtc {
+
+// SDP specification for a single audio codec.
+// NOTE: This class is still under development and may change without notice.
+struct SdpAudioFormat {
+  using Parameters = std::map<std::string, std::string>;
+
+  // TODO(kwiberg): Get rid of the default constructor when rtc::Optional no
+  // longer requires it.
+  SdpAudioFormat();
+  SdpAudioFormat(const SdpAudioFormat&);
+  SdpAudioFormat(SdpAudioFormat&&);
+  SdpAudioFormat(const char* name, int clockrate_hz, int num_channels);
+  SdpAudioFormat(const char* name,
+                 int clockrate_hz,
+                 int num_channels,
+                 Parameters&& param);
+  ~SdpAudioFormat();
+
+  SdpAudioFormat& operator=(const SdpAudioFormat&);
+  SdpAudioFormat& operator=(SdpAudioFormat&&);
+
+  std::string name;
+  int clockrate_hz;
+  int num_channels;
+  Parameters parameters;
+  // Parameters feedback_parameters; ??
+};
+
+void swap(SdpAudioFormat& a, SdpAudioFormat& b);
+std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
new file mode 100644
index 00000000000..4c7445672ac
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#ifdef WEBRTC_CODEC_G722
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#endif
+#ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#endif
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+
+namespace webrtc {
+
+namespace {
+
+struct NamedDecoderConstructor {
+  const char* name;
+  std::unique_ptr<AudioDecoder> (*constructor)(const SdpAudioFormat&);
+};
+
+std::unique_ptr<AudioDecoder> Unique(AudioDecoder* d) {
+  return std::unique_ptr<AudioDecoder>(d);
+}
+
+// TODO(kwiberg): These factory functions should probably be moved to each
+// decoder.
+NamedDecoderConstructor decoder_constructors[] = {
+    {"pcmu",
+     [](const SdpAudioFormat& format) {
+       return format.clockrate_hz == 8000 && format.num_channels >= 1
+                  ? Unique(new AudioDecoderPcmU(format.num_channels))
+                  : nullptr;
+     }},
+    {"pcma",
+     [](const SdpAudioFormat& format) {
+       return format.clockrate_hz == 8000 && format.num_channels >= 1
+                  ? Unique(new AudioDecoderPcmA(format.num_channels))
+                  : nullptr;
+     }},
+#ifdef WEBRTC_CODEC_ILBC
+    {"ilbc",
+     [](const SdpAudioFormat& format) {
+       return format.clockrate_hz == 8000 && format.num_channels == 1
+                  ? Unique(new AudioDecoderIlbc)
+                  : nullptr;
+     }},
+#endif
+#if defined(WEBRTC_CODEC_ISACFX)
+    {"isac",
+     [](const SdpAudioFormat& format) {
+       return format.clockrate_hz == 16000 && format.num_channels == 1
+                  ? Unique(new AudioDecoderIsacFix)
+                  : nullptr;
+     }},
+#elif defined(WEBRTC_CODEC_ISAC)
+    {"isac",
+     [](const SdpAudioFormat& format) {
+       return (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
+                      format.num_channels == 1
+                  ? Unique(new AudioDecoderIsac)
+                  : nullptr;
+     }},
+#endif
+    {"l16",
+     [](const SdpAudioFormat& format) {
+       return format.num_channels >= 1
+                  ? Unique(new AudioDecoderPcm16B(format.num_channels))
+                  : nullptr;
+     }},
+#ifdef WEBRTC_CODEC_G722
+    {"g722",
+     [](const SdpAudioFormat& format) {
+       if (format.clockrate_hz == 8000) {
+         if (format.num_channels == 1)
+           return Unique(new AudioDecoderG722);
+         if (format.num_channels == 2)
+           return Unique(new AudioDecoderG722Stereo);
+       }
+       return Unique(nullptr);
+     }},
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    {"opus",
+     [](const SdpAudioFormat& format) {
+       rtc::Optional<int> num_channels = [&] {
+         auto stereo = format.parameters.find("stereo");
+         if (stereo != format.parameters.end()) {
+           if (stereo->second == "0") {
+             return rtc::Optional<int>(1);
+           } else if (stereo->second == "1") {
+             return rtc::Optional<int>(2);
+           }
+         }
+         return rtc::Optional<int>();
+       }();
+       return format.clockrate_hz == 48000 && format.num_channels == 2 &&
+                      num_channels
+                  ? Unique(new AudioDecoderOpus(*num_channels))
+                  : nullptr;
+     }},
+#endif
+};
+
+class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
+ public:
+  std::vector<SdpAudioFormat> GetSupportedFormats() override {
+    FATAL() << "Not implemented yet!";
+  }
+
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format) override {
+    for (const auto& dc : decoder_constructors) {
+      if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
+        return std::unique_ptr<AudioDecoder>(dc.constructor(format));
+      }
+    }
+    return nullptr;
+  }
+};
+
+}  // namespace
+
+std::unique_ptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
+  return std::unique_ptr<AudioDecoderFactory>(new BuiltinAudioDecoderFactory);
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
new file mode 100644
index 00000000000..7234c160b5c
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+
+#include <memory>
+
+#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create the built-in types of audio decoders.
+// NOTE: This function is still under development and may change without notice.
+std::unique_ptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 3b48131a754..d2edcb5c265 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -21,33 +21,10 @@ namespace {
 
 const int kMaxFrameSizeMs = 60;
 
-std::unique_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
-    int sample_rate_hz,
-    int sid_frame_interval_ms,
-    int num_cng_coefficients) {
-  CNG_enc_inst* ci;
-  RTC_CHECK_EQ(0, WebRtcCng_CreateEnc(&ci));
-  std::unique_ptr<CNG_enc_inst, CngInstDeleter> cng_inst(ci);
-  RTC_CHECK_EQ(0,
-               WebRtcCng_InitEnc(cng_inst.get(), sample_rate_hz,
-                                 sid_frame_interval_ms, num_cng_coefficients));
-  return cng_inst;
-}
-
 }  // namespace
 
 AudioEncoderCng::Config::Config() = default;
-
-// TODO(kwiberg): =default this when Visual Studio learns to handle it.
-AudioEncoderCng::Config::Config(Config&& c)
-    : num_channels(c.num_channels),
-      payload_type(c.payload_type),
-      speech_encoder(std::move(c.speech_encoder)),
-      vad_mode(c.vad_mode),
-      sid_frame_interval_ms(c.sid_frame_interval_ms),
-      num_cng_coefficients(c.num_cng_coefficients),
-      vad(c.vad) {}
-
+AudioEncoderCng::Config::Config(Config&&) = default;
 AudioEncoderCng::Config::~Config() = default;
 
 bool AudioEncoderCng::Config::IsOk() const {
@@ -75,20 +52,14 @@ AudioEncoderCng::AudioEncoderCng(Config&& config)
       sid_frame_interval_ms_(config.sid_frame_interval_ms),
       last_frame_active_(true),
       vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
-                      : CreateVad(config.vad_mode)) {
-  cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,
-                            num_cng_coefficients_);
+           : CreateVad(config.vad_mode)),
+      cng_encoder_(new ComfortNoiseEncoder(SampleRateHz(),
+                                           sid_frame_interval_ms_,
+                                           num_cng_coefficients_)) {
 }
 
 AudioEncoderCng::~AudioEncoderCng() = default;
 
-size_t AudioEncoderCng::MaxEncodedBytes() const {
-  const size_t max_encoded_bytes_active = speech_encoder_->MaxEncodedBytes();
-  const size_t max_encoded_bytes_passive =
-      rtc::CheckedDivExact(kMaxFrameSizeMs, 10) * SamplesPer10msFrame();
-  return std::max(max_encoded_bytes_active, max_encoded_bytes_passive);
-}
-
 int AudioEncoderCng::SampleRateHz() const {
   return speech_encoder_->SampleRateHz();
 }
@@ -187,8 +158,9 @@ void AudioEncoderCng::Reset() {
   rtp_timestamps_.clear();
   last_frame_active_ = true;
   vad_->Reset();
-  cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,
-                            num_cng_coefficients_);
+  cng_encoder_.reset(
+      new ComfortNoiseEncoder(SampleRateHz(), sid_frame_interval_ms_,
+                              num_cng_coefficients_));
 }
 
 bool AudioEncoderCng::SetFec(bool enable) {
@@ -215,38 +187,38 @@ void AudioEncoderCng::SetTargetBitrate(int bits_per_second) {
   speech_encoder_->SetTargetBitrate(bits_per_second);
 }
 
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCng::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
     size_t frames_to_encode,
     rtc::Buffer* encoded) {
   bool force_sid = last_frame_active_;
   bool output_produced = false;
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
-  const size_t bytes_to_encode = frames_to_encode * samples_per_10ms_frame;
   AudioEncoder::EncodedInfo info;
 
-  encoded->AppendData(bytes_to_encode, [&] (rtc::ArrayView<uint8_t> encoded) {
-      for (size_t i = 0; i < frames_to_encode; ++i) {
-        // It's important not to pass &info.encoded_bytes directly to
-        // WebRtcCng_Encode(), since later loop iterations may return zero in
-        // that value, in which case we don't want to overwrite any value from
-        // an earlier iteration.
-        size_t encoded_bytes_tmp = 0;
-        RTC_CHECK_GE(
-            WebRtcCng_Encode(cng_inst_.get(),
-                             &speech_buffer_[i * samples_per_10ms_frame],
-                             samples_per_10ms_frame, encoded.data(),
-                             &encoded_bytes_tmp, force_sid),
-            0);
-        if (encoded_bytes_tmp > 0) {
-          RTC_CHECK(!output_produced);
-          info.encoded_bytes = encoded_bytes_tmp;
-          output_produced = true;
-          force_sid = false;
-        }
-      }
-
-      return info.encoded_bytes;
-    });
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    // It's important not to pass &info.encoded_bytes directly to
+    // WebRtcCng_Encode(), since later loop iterations may return zero in
+    // that value, in which case we don't want to overwrite any value from
+    // an earlier iteration.
+    size_t encoded_bytes_tmp =
+        cng_encoder_->Encode(
+            rtc::ArrayView<const int16_t>(
+                &speech_buffer_[i * samples_per_10ms_frame],
+                samples_per_10ms_frame),
+            force_sid, encoded);
+
+    if (encoded_bytes_tmp > 0) {
+      RTC_CHECK(!output_produced);
+      info.encoded_bytes = encoded_bytes_tmp;
+      output_produced = true;
+      force_sid = false;
+    }
+  }
 
   info.encoded_timestamp = rtp_timestamps_.front();
   info.payload_type = cng_payload_type_;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index 1384cd511ee..a895e69de44 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -21,11 +21,6 @@
 
 namespace webrtc {
 
-// Deleter for use with unique_ptr.
-struct CngInstDeleter {
-  void operator()(CNG_enc_inst* ptr) const { WebRtcCng_FreeEnc(ptr); }
-};
-
 class Vad;
 
 class AudioEncoderCng final : public AudioEncoder {
@@ -52,7 +47,6 @@ class AudioEncoderCng final : public AudioEncoder {
   explicit AudioEncoderCng(Config&& config);
   ~AudioEncoderCng() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
@@ -69,6 +63,8 @@ class AudioEncoderCng final : public AudioEncoder {
   void SetMaxPlaybackRate(int frequency_hz) override;
   void SetProjectedPacketLossRate(double fraction) override;
   void SetTargetBitrate(int target_bps) override;
+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
+      override;
 
  private:
   EncodedInfo EncodePassive(size_t frames_to_encode,
@@ -85,7 +81,7 @@ class AudioEncoderCng final : public AudioEncoder {
   std::vector<uint32_t> rtp_timestamps_;
   bool last_frame_active_;
   std::unique_ptr<Vad> vad_;
-  std::unique_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
+  std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 8f30d783ae4..eb6c6d3607e 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -12,6 +12,7 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/common_audio/vad/mock/mock_vad.h"
 #include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
@@ -25,7 +26,6 @@ using ::testing::Invoke;
 namespace webrtc {
 
 namespace {
-static const size_t kMockMaxEncodedBytes = 1000;
 static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
 static const size_t kMockReturnEncodedBytes = 17;
 static const int kCngPayloadType = 18;
@@ -74,8 +74,6 @@ class AudioEncoderCngTest : public ::testing::Test {
       // as long as it is smaller than 10.
       EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket())
           .WillOnce(Return(1u));
-      EXPECT_CALL(*mock_encoder_, MaxEncodedBytes())
-          .WillRepeatedly(Return(kMockMaxEncodedBytes));
     }
     cng_.reset(new AudioEncoderCng(std::move(config)));
   }
@@ -90,8 +88,8 @@ class AudioEncoderCngTest : public ::testing::Test {
   }
 
   // Expect |num_calls| calls to the encoder, all successful. The last call
-  // claims to have encoded |kMockMaxEncodedBytes| bytes, and all the preceding
-  // ones 0 bytes.
+  // claims to have encoded |kMockReturnEncodedBytes| bytes, and all the
+  // preceding ones 0 bytes.
   void ExpectEncodeCalls(size_t num_calls) {
     InSequence s;
     AudioEncoder::EncodedInfo info;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng.gypi b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng.gypi
index c020f4740d4..bbff9f8edfe 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng.gypi
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng.gypi
@@ -18,9 +18,7 @@
       'sources': [
         'audio_encoder_cng.cc',
         'audio_encoder_cng.h',
-        'cng_helpfuns.c',
-        'cng_helpfuns.h',
-        'webrtc_cng.c',
+        'webrtc_cng.cc',
         'webrtc_cng.h',
       ],
     },
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.c
deleted file mode 100644
index bc08d431a69..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cng_helpfuns.h"
-
-#include "signal_processing_library.h"
-#include "webrtc/typedefs.h"
-#include "webrtc_cng.h"
-
-/* Values in |k| are Q15, and |a| Q12. */
-void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a) {
-  int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
-  int16_t *aptr, *aptr2, *anyptr;
-  const int16_t *kptr;
-  int m, i;
-
-  kptr = k;
-  *a = 4096;  /* i.e., (Word16_MAX >> 3) + 1 */
-  *any = *a;
-  a[1] = (*k + 4) >> 3;
-  for (m = 1; m < useOrder; m++) {
-    kptr++;
-    aptr = a;
-    aptr++;
-    aptr2 = &a[m];
-    anyptr = any;
-    anyptr++;
-
-    any[m + 1] = (*kptr + 4) >> 3;
-    for (i = 0; i < m; i++) {
-      *anyptr++ = (*aptr++) +
-          (int16_t)((((int32_t)(*aptr2--) * (int32_t) * kptr) + 16384) >> 15);
-    }
-
-    aptr = a;
-    anyptr = any;
-    for (i = 0; i < (m + 2); i++) {
-      *aptr++ = *anyptr++;
-    }
-  }
-}
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.h
deleted file mode 100644
index a553a7615e6..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_CNG_HELPFUNS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_CNG_HELPFUNS_H_
-
-#include "webrtc/typedefs.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_CNG_HELPFUNS_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
index 1061dca69ac..95132a96178 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
@@ -7,11 +7,12 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#include <memory>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 
 namespace webrtc {
 
@@ -21,7 +22,7 @@ enum {
   kSidLongIntervalUpdate = 10000
 };
 
-enum {
+enum : size_t {
   kCNGNumParamsLow = 0,
   kCNGNumParamsNormal = 8,
   kCNGNumParamsHigh = WEBRTC_CNG_MAX_LPC_ORDER,
@@ -35,19 +36,13 @@ enum {
 
 class CngTest : public ::testing::Test {
  protected:
-  CngTest();
   virtual void SetUp();
 
-  CNG_enc_inst* cng_enc_inst_;
-  CNG_dec_inst* cng_dec_inst_;
+  void TestCngEncode(int sample_rate_hz, int quality);
+
   int16_t speech_data_[640];  // Max size of CNG internal buffers.
 };
 
-CngTest::CngTest()
-    : cng_enc_inst_(NULL),
-      cng_dec_inst_(NULL) {
-}
-
 void CngTest::SetUp() {
   FILE* input_file;
   const std::string file_name =
@@ -60,289 +55,187 @@ void CngTest::SetUp() {
   input_file = NULL;
 }
 
-// Test failing Create.
-TEST_F(CngTest, CngCreateFail) {
-  // Test to see that an invalid pointer is caught.
-  EXPECT_EQ(-1, WebRtcCng_CreateEnc(NULL));
-  EXPECT_EQ(-1, WebRtcCng_CreateDec(NULL));
-}
-
-// Test normal Create.
-TEST_F(CngTest, CngCreate) {
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
-  EXPECT_TRUE(cng_enc_inst_ != NULL);
-  EXPECT_TRUE(cng_dec_inst_ != NULL);
-  // Free encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_FreeDec(cng_dec_inst_));
+void CngTest::TestCngEncode(int sample_rate_hz, int quality) {
+  const size_t num_samples_10ms = rtc::CheckedDivExact(sample_rate_hz, 100);
+  rtc::Buffer sid_data;
+
+  ComfortNoiseEncoder cng_encoder(sample_rate_hz, kSidNormalIntervalUpdate,
+                                  quality);
+  EXPECT_EQ(0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(
+                                       speech_data_, num_samples_10ms),
+                                   kNoSid, &sid_data));
+  EXPECT_EQ(static_cast<size_t>(quality + 1),
+            cng_encoder.Encode(
+                rtc::ArrayView<const int16_t>(speech_data_, num_samples_10ms),
+                kForceSid, &sid_data));
 }
 
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 // Create CNG encoder, init with faulty values, free CNG encoder.
 TEST_F(CngTest, CngInitFail) {
-  // Create encoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-
   // Call with too few parameters.
-  EXPECT_EQ(-1, WebRtcCng_InitEnc(cng_enc_inst_, 8000, kSidNormalIntervalUpdate,
-                                  kCNGNumParamsLow));
-  EXPECT_EQ(6130, WebRtcCng_GetErrorCodeEnc(cng_enc_inst_));
-
+  EXPECT_DEATH({ ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
+                                     kCNGNumParamsLow); }, "");
   // Call with too many parameters.
-  EXPECT_EQ(-1, WebRtcCng_InitEnc(cng_enc_inst_, 8000, kSidNormalIntervalUpdate,
-                                  kCNGNumParamsTooHigh));
-  EXPECT_EQ(6130, WebRtcCng_GetErrorCodeEnc(cng_enc_inst_));
-
-  // Free encoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
-}
-
-TEST_F(CngTest, CngEncode) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  size_t number_bytes;
-
-  // Create encoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-
-  // 8 kHz, Normal number of parameters
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 8000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 80, sid_data,
-                                &number_bytes, kNoSid));
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 80, sid_data, &number_bytes, kForceSid));
-
-  // 16 kHz, Normal number of parameters
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
-                                &number_bytes, kNoSid));
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 160, sid_data, &number_bytes, kForceSid));
-
-  // 32 kHz, Max number of parameters
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 32000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsHigh));
-  EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 320, sid_data,
-                                &number_bytes, kNoSid));
-  EXPECT_EQ(kCNGNumParamsHigh + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 320, sid_data, &number_bytes, kForceSid));
-
-  // 48 kHz, Normal number of parameters
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 48000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 480, sid_data,
-                                &number_bytes, kNoSid));
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 480, sid_data, &number_bytes, kForceSid));
-
-  // 64 kHz, Normal number of parameters
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 64000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 640, sid_data,
-                                &number_bytes, kNoSid));
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 640, sid_data, &number_bytes, kForceSid));
-
-  // Free encoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
+  EXPECT_DEATH({ ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
+                                     kCNGNumParamsTooHigh); }, "");
 }
 
 // Encode Cng with too long input vector.
 TEST_F(CngTest, CngEncodeTooLong) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  size_t number_bytes;
-
-  // Create and init encoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 8000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
+  rtc::Buffer sid_data;
 
+  // Create encoder.
+  ComfortNoiseEncoder cng_encoder(8000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
   // Run encoder with too much data.
-  EXPECT_EQ(-1, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 641, sid_data,
-                                 &number_bytes, kNoSid));
-  EXPECT_EQ(6140, WebRtcCng_GetErrorCodeEnc(cng_enc_inst_));
+  EXPECT_DEATH(
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 641),
+                         kNoSid, &sid_data),
+      "");
+}
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
-  // Free encoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
+TEST_F(CngTest, CngEncode8000) {
+  TestCngEncode(8000, kCNGNumParamsNormal);
 }
 
-// Call encode without calling init.
-TEST_F(CngTest, CngEncodeNoInit) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  size_t number_bytes;
+TEST_F(CngTest, CngEncode16000) {
+  TestCngEncode(16000, kCNGNumParamsNormal);
+}
 
-  // Create encoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
+TEST_F(CngTest, CngEncode32000) {
+  TestCngEncode(32000, kCNGNumParamsHigh);
+}
 
-  // Run encoder without calling init.
-  EXPECT_EQ(-1, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 640, sid_data,
-                                 &number_bytes, kNoSid));
-  EXPECT_EQ(6120, WebRtcCng_GetErrorCodeEnc(cng_enc_inst_));
+TEST_F(CngTest, CngEncode48000) {
+  TestCngEncode(48000, kCNGNumParamsNormal);
+}
 
-  // Free encoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
+TEST_F(CngTest, CngEncode64000) {
+  TestCngEncode(64000, kCNGNumParamsNormal);
 }
 
 // Update SID parameters, for both 9 and 16 parameters.
 TEST_F(CngTest, CngUpdateSid) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  size_t number_bytes;
+  rtc::Buffer sid_data;
 
-  // Create and initialize encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  WebRtcCng_InitDec(cng_dec_inst_);
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
 
   // Run normal Encode and UpdateSid.
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 160, sid_data, &number_bytes, kForceSid));
-  EXPECT_EQ(0, WebRtcCng_UpdateSid(cng_dec_inst_, sid_data,
-                                   kCNGNumParamsNormal + 1));
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  cng_decoder.UpdateSid(sid_data);
 
   // Reinit with new length.
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsHigh));
-  WebRtcCng_InitDec(cng_dec_inst_);
+  cng_encoder.Reset(16000, kSidNormalIntervalUpdate, kCNGNumParamsHigh);
+  cng_decoder.Reset();
 
   // Expect 0 because of unstable parameters after switching length.
-  EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
-                                &number_bytes, kForceSid));
-  EXPECT_EQ(kCNGNumParamsHigh + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_ + 160, 160, sid_data, &number_bytes,
-      kForceSid));
-  EXPECT_EQ(0, WebRtcCng_UpdateSid(cng_dec_inst_, sid_data,
-                                   kCNGNumParamsNormal + 1));
-
-  // Free encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_FreeDec(cng_dec_inst_));
+  EXPECT_EQ(0U,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  EXPECT_EQ(
+      kCNGNumParamsHigh + 1,
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_ + 160, 160),
+                         kForceSid, &sid_data));
+  cng_decoder.UpdateSid(
+      rtc::ArrayView<const uint8_t>(sid_data.data(), kCNGNumParamsNormal + 1));
 }
 
 // Update SID parameters, with wrong parameters or without calling decode.
 TEST_F(CngTest, CngUpdateSidErroneous) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  size_t number_bytes;
-
-  // Create encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
+  rtc::Buffer sid_data;
 
   // Encode.
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 160, sid_data, &number_bytes, kForceSid));
-
-  // Update Sid before initializing decoder.
-  EXPECT_EQ(-1, WebRtcCng_UpdateSid(cng_dec_inst_, sid_data,
-                                    kCNGNumParamsNormal + 1));
-  EXPECT_EQ(6220, WebRtcCng_GetErrorCodeDec(cng_dec_inst_));
-
-  // Initialize decoder.
-  WebRtcCng_InitDec(cng_dec_inst_);
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
 
   // First run with valid parameters, then with too many CNG parameters.
   // The function will operate correctly by only reading the maximum number of
   // parameters, skipping the extra.
-  EXPECT_EQ(0, WebRtcCng_UpdateSid(cng_dec_inst_, sid_data,
-                                   kCNGNumParamsNormal + 1));
-  EXPECT_EQ(0, WebRtcCng_UpdateSid(cng_dec_inst_, sid_data,
-                                   kCNGNumParamsTooHigh + 1));
-
-  // Free encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_FreeDec(cng_dec_inst_));
+  EXPECT_EQ(kCNGNumParamsNormal + 1, sid_data.size());
+  cng_decoder.UpdateSid(sid_data);
+
+  // Make sure the input buffer is large enough. Since Encode() appends data, we
+  // need to set the size manually only afterwards, or the buffer will be bigger
+  // than anticipated.
+  sid_data.SetSize(kCNGNumParamsTooHigh + 1);
+  cng_decoder.UpdateSid(sid_data);
 }
 
 // Test to generate cng data, by forcing SID. Both normal and faulty condition.
 TEST_F(CngTest, CngGenerate) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  rtc::Buffer sid_data;
   int16_t out_data[640];
-  size_t number_bytes;
 
-  // Create and initialize encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  WebRtcCng_InitDec(cng_dec_inst_);
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
 
   // Normal Encode.
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 160, sid_data, &number_bytes, kForceSid));
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
 
   // Normal UpdateSid.
-  EXPECT_EQ(0, WebRtcCng_UpdateSid(cng_dec_inst_, sid_data,
-                                   kCNGNumParamsNormal + 1));
+  cng_decoder.UpdateSid(sid_data);
 
   // Two normal Generate, one with new_period.
-  EXPECT_EQ(0, WebRtcCng_Generate(cng_dec_inst_, out_data, 640, 1));
-  EXPECT_EQ(0, WebRtcCng_Generate(cng_dec_inst_, out_data, 640, 0));
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 1));
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 0));
 
   // Call Genereate with too much data.
-  EXPECT_EQ(-1, WebRtcCng_Generate(cng_dec_inst_, out_data, 641, 0));
-  EXPECT_EQ(6140, WebRtcCng_GetErrorCodeDec(cng_dec_inst_));
-
-  // Free encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_FreeDec(cng_dec_inst_));
+  EXPECT_FALSE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 641), 0));
 }
 
 // Test automatic SID.
 TEST_F(CngTest, CngAutoSid) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  size_t number_bytes;
+  rtc::Buffer sid_data;
 
-  // Create and initialize encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidNormalIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  WebRtcCng_InitDec(cng_dec_inst_);
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
 
   // Normal Encode, 100 msec, where no SID data should be generated.
   for (int i = 0; i < 10; i++) {
-    EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
-                                  &number_bytes, kNoSid));
+    EXPECT_EQ(0U, cng_encoder.Encode(
+        rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
   }
 
   // We have reached 100 msec, and SID data should be generated.
-  EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-      cng_enc_inst_, speech_data_, 160, sid_data, &number_bytes, kNoSid));
-
-  // Free encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_FreeDec(cng_dec_inst_));
+  EXPECT_EQ(kCNGNumParamsNormal + 1, cng_encoder.Encode(
+      rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
 }
 
 // Test automatic SID, with very short interval.
 TEST_F(CngTest, CngAutoSidShort) {
-  uint8_t sid_data[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  size_t number_bytes;
+  rtc::Buffer sid_data;
 
-  // Create and initialize encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_CreateEnc(&cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_CreateDec(&cng_dec_inst_));
-  EXPECT_EQ(0, WebRtcCng_InitEnc(cng_enc_inst_, 16000, kSidShortIntervalUpdate,
-                                 kCNGNumParamsNormal));
-  WebRtcCng_InitDec(cng_dec_inst_);
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidShortIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
 
   // First call will never generate SID, unless forced to.
-  EXPECT_EQ(0, WebRtcCng_Encode(cng_enc_inst_, speech_data_, 160, sid_data,
-                                &number_bytes, kNoSid));
+  EXPECT_EQ(0U, cng_encoder.Encode(
+      rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
 
   // Normal Encode, 100 msec, SID data should be generated all the time.
   for (int i = 0; i < 10; i++) {
-    EXPECT_EQ(kCNGNumParamsNormal + 1, WebRtcCng_Encode(
-        cng_enc_inst_, speech_data_, 160, sid_data, &number_bytes, kNoSid));
+    EXPECT_EQ(kCNGNumParamsNormal + 1, cng_encoder.Encode(
+        rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
   }
-
-  // Free encoder and decoder memory.
-  EXPECT_EQ(0, WebRtcCng_FreeEnc(cng_enc_inst_));
-  EXPECT_EQ(0, WebRtcCng_FreeDec(cng_dec_inst_));
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
deleted file mode 100644
index 8dddc5c717d..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ /dev/null
@@ -1,603 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc_cng.h"
-
-#include <string.h>
-#include <stdlib.h>
-
-#include "cng_helpfuns.h"
-#include "signal_processing_library.h"
-
-typedef struct WebRtcCngDecoder_ {
-  uint32_t dec_seed;
-  int32_t dec_target_energy;
-  int32_t dec_used_energy;
-  int16_t dec_target_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t dec_used_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t dec_filtstate[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t dec_filtstateLow[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t dec_Efiltstate[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t dec_EfiltstateLow[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t dec_order;
-  int16_t dec_target_scale_factor;  /* Q29 */
-  int16_t dec_used_scale_factor;  /* Q29 */
-  int16_t target_scale_factor;  /* Q13 */
-  int16_t errorcode;
-  int16_t initflag;
-} WebRtcCngDecoder;
-
-typedef struct WebRtcCngEncoder_ {
-  size_t enc_nrOfCoefs;
-  int enc_sampfreq;
-  int16_t enc_interval;
-  int16_t enc_msSinceSID;
-  int32_t enc_Energy;
-  int16_t enc_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int32_t enc_corrVector[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  uint32_t enc_seed;
-  int16_t errorcode;
-  int16_t initflag;
-} WebRtcCngEncoder;
-
-const int32_t WebRtcCng_kDbov[94] = {
-  1081109975,  858756178,  682134279,  541838517,  430397633,  341876992,
-  271562548,  215709799,  171344384,  136103682,  108110997,   85875618,
-  68213428,   54183852,   43039763,   34187699,   27156255,   21570980,
-  17134438,   13610368,   10811100,    8587562,    6821343,    5418385,
-  4303976,    3418770,    2715625,    2157098,    1713444,    1361037,
-  1081110,     858756,     682134,     541839,     430398,     341877,
-  271563,     215710,     171344,     136104,     108111,      85876,
-  68213,      54184,      43040,      34188,      27156,      21571,
-  17134,      13610,      10811,       8588,       6821,       5418,
-  4304,       3419,       2716,       2157,       1713,       1361,
-  1081,        859,        682,        542,        430,        342,
-  272,        216,        171,        136,        108,         86,
-  68,         54,         43,         34,         27,         22,
-  17,         14,         11,          9,          7,          5,
-  4,          3,          3,          2,          2,           1,
-  1,          1,          1,          1
-};
-
-const int16_t WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
-  32702, 32636, 32570, 32505, 32439, 32374,
-  32309, 32244, 32179, 32114, 32049, 31985
-};
-
-/****************************************************************************
- * WebRtcCng_CreateEnc/Dec(...)
- *
- * These functions create an instance to the specified structure
- *
- * Input:
- *      - XXX_inst      : Pointer to created instance that should be created
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int16_t WebRtcCng_CreateEnc(CNG_enc_inst** cng_inst) {
-  if (cng_inst != NULL) {
-    *cng_inst = (CNG_enc_inst*) malloc(sizeof(WebRtcCngEncoder));
-    if (*cng_inst != NULL) {
-      (*(WebRtcCngEncoder**) cng_inst)->errorcode = 0;
-      (*(WebRtcCngEncoder**) cng_inst)->initflag = 0;
-
-      /* Needed to get the right function pointers in SPLIB. */
-      WebRtcSpl_Init();
-
-      return 0;
-    } else {
-      /* The memory could not be allocated. */
-      return -1;
-    }
-  } else {
-    /* The input pointer is invalid (NULL). */
-    return -1;
-  }
-}
-
-int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst) {
-  if (cng_inst != NULL ) {
-    *cng_inst = (CNG_dec_inst*) malloc(sizeof(WebRtcCngDecoder));
-    if (*cng_inst != NULL ) {
-      (*(WebRtcCngDecoder**) cng_inst)->errorcode = 0;
-      (*(WebRtcCngDecoder**) cng_inst)->initflag = 0;
-
-      /* Needed to get the right function pointers in SPLIB. */
-      WebRtcSpl_Init();
-
-      return 0;
-    } else {
-      /* The memory could not be allocated */
-      return -1;
-    }
-  } else {
-    /* The input pointer is invalid (NULL). */
-    return -1;
-  }
-}
-
-/****************************************************************************
- * WebRtcCng_InitEnc/Dec(...)
- *
- * This function initializes a instance
- *
- * Input:
- *    - cng_inst      : Instance that should be initialized
- *
- *    - fs            : 8000 for narrowband and 16000 for wideband
- *    - interval      : generate SID data every interval ms
- *    - quality       : TBD
- *
- * Output:
- *    - cng_inst      : Initialized instance
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
-                      int16_t quality) {
-  int i;
-  WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
-  memset(inst, 0, sizeof(WebRtcCngEncoder));
-
-  /* Check LPC order */
-  if (quality > WEBRTC_CNG_MAX_LPC_ORDER || quality <= 0) {
-    inst->errorcode = CNG_DISALLOWED_LPC_ORDER;
-    return -1;
-  }
-
-  inst->enc_sampfreq = fs;
-  inst->enc_interval = interval;
-  inst->enc_nrOfCoefs = quality;
-  inst->enc_msSinceSID = 0;
-  inst->enc_seed = 7777;  /* For debugging only. */
-  inst->enc_Energy = 0;
-  for (i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER + 1); i++) {
-    inst->enc_reflCoefs[i] = 0;
-    inst->enc_corrVector[i] = 0;
-  }
-  inst->initflag = 1;
-
-  return 0;
-}
-
-void WebRtcCng_InitDec(CNG_dec_inst* cng_inst) {
-  int i;
-
-  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
-
-  memset(inst, 0, sizeof(WebRtcCngDecoder));
-  inst->dec_seed = 7777;  /* For debugging only. */
-  inst->dec_order = 5;
-  inst->dec_target_scale_factor = 0;
-  inst->dec_used_scale_factor = 0;
-  for (i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER + 1); i++) {
-    inst->dec_filtstate[i] = 0;
-    inst->dec_target_reflCoefs[i] = 0;
-    inst->dec_used_reflCoefs[i] = 0;
-  }
-  inst->dec_target_reflCoefs[0] = 0;
-  inst->dec_used_reflCoefs[0] = 0;
-  inst->dec_used_energy = 0;
-  inst->initflag = 1;
-}
-
-/****************************************************************************
- * WebRtcCng_FreeEnc/Dec(...)
- *
- * These functions frees the dynamic memory of a specified instance
- *
- * Input:
- *    - cng_inst      : Pointer to created instance that should be freed
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int16_t WebRtcCng_FreeEnc(CNG_enc_inst* cng_inst) {
-  free(cng_inst);
-  return 0;
-}
-
-int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst) {
-  free(cng_inst);
-  return 0;
-}
-
-/****************************************************************************
- * WebRtcCng_Encode(...)
- *
- * These functions analyzes background noise
- *
- * Input:
- *    - cng_inst      : Pointer to created instance
- *    - speech        : Signal (noise) to be analyzed
- *    - nrOfSamples   : Size of speech vector
- *    - bytesOut      : Nr of bytes to transmit, might be 0
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
-                     size_t nrOfSamples, uint8_t* SIDdata,
-                     size_t* bytesOut, int16_t forceSID) {
-  WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
-
-  int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int32_t corrVector[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t hanningW[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
-  int16_t ReflBeta = 19661;  /* 0.6 in q15. */
-  int16_t ReflBetaComp = 13107;  /* 0.4 in q15. */
-  int32_t outEnergy;
-  int outShifts;
-  size_t i;
-  int stab;
-  int acorrScale;
-  size_t index;
-  size_t ind, factor;
-  int32_t* bptr;
-  int32_t blo, bhi;
-  int16_t negate;
-  const int16_t* aptr;
-  int16_t speechBuf[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
-
-  /* Check if encoder initiated. */
-  if (inst->initflag != 1) {
-    inst->errorcode = CNG_ENCODER_NOT_INITIATED;
-    return -1;
-  }
-
-  /* Check framesize. */
-  if (nrOfSamples > WEBRTC_CNG_MAX_OUTSIZE_ORDER) {
-    inst->errorcode = CNG_DISALLOWED_FRAME_SIZE;
-    return -1;
-  }
-
-  for (i = 0; i < nrOfSamples; i++) {
-    speechBuf[i] = speech[i];
-  }
-
-  factor = nrOfSamples;
-
-  /* Calculate energy and a coefficients. */
-  outEnergy = WebRtcSpl_Energy(speechBuf, nrOfSamples, &outShifts);
-  while (outShifts > 0) {
-    /* We can only do 5 shifts without destroying accuracy in
-     * division factor. */
-    if (outShifts > 5) {
-      outEnergy <<= (outShifts - 5);
-      outShifts = 5;
-    } else {
-      factor /= 2;
-      outShifts--;
-    }
-  }
-  outEnergy = WebRtcSpl_DivW32W16(outEnergy, (int16_t)factor);
-
-  if (outEnergy > 1) {
-    /* Create Hanning Window. */
-    WebRtcSpl_GetHanningWindow(hanningW, nrOfSamples / 2);
-    for (i = 0; i < (nrOfSamples / 2); i++)
-      hanningW[nrOfSamples - i - 1] = hanningW[i];
-
-    WebRtcSpl_ElementwiseVectorMult(speechBuf, hanningW, speechBuf, nrOfSamples,
-                                    14);
-
-    WebRtcSpl_AutoCorrelation(speechBuf, nrOfSamples, inst->enc_nrOfCoefs,
-                              corrVector, &acorrScale);
-
-    if (*corrVector == 0)
-      *corrVector = WEBRTC_SPL_WORD16_MAX;
-
-    /* Adds the bandwidth expansion. */
-    aptr = WebRtcCng_kCorrWindow;
-    bptr = corrVector;
-
-    /* (zzz) lpc16_1 = 17+1+820+2+2 = 842 (ordo2=700). */
-    for (ind = 0; ind < inst->enc_nrOfCoefs; ind++) {
-      /* The below code multiplies the 16 b corrWindow values (Q15) with
-       * the 32 b corrvector (Q0) and shifts the result down 15 steps. */
-      negate = *bptr < 0;
-      if (negate)
-        *bptr = -*bptr;
-
-      blo = (int32_t) * aptr * (*bptr & 0xffff);
-      bhi = ((blo >> 16) & 0xffff)
-          + ((int32_t)(*aptr++) * ((*bptr >> 16) & 0xffff));
-      blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
-
-      *bptr = (((bhi >> 16) & 0x7fff) << 17) | ((uint32_t) blo >> 15);
-      if (negate)
-        *bptr = -*bptr;
-      bptr++;
-    }
-    /* End of bandwidth expansion. */
-
-    stab = WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs,
-                                    inst->enc_nrOfCoefs);
-
-    if (!stab) {
-      /* Disregard from this frame */
-      *bytesOut = 0;
-      return 0;
-    }
-
-  } else {
-    for (i = 0; i < inst->enc_nrOfCoefs; i++)
-      refCs[i] = 0;
-  }
-
-  if (forceSID) {
-    /* Read instantaneous values instead of averaged. */
-    for (i = 0; i < inst->enc_nrOfCoefs; i++)
-      inst->enc_reflCoefs[i] = refCs[i];
-    inst->enc_Energy = outEnergy;
-  } else {
-    /* Average history with new values. */
-    for (i = 0; i < (inst->enc_nrOfCoefs); i++) {
-      inst->enc_reflCoefs[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          inst->enc_reflCoefs[i], ReflBeta, 15);
-      inst->enc_reflCoefs[i] += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-          refCs[i], ReflBetaComp, 15);
-    }
-    inst->enc_Energy = (outEnergy >> 2) + (inst->enc_Energy >> 1)
-        + (inst->enc_Energy >> 2);
-  }
-
-  if (inst->enc_Energy < 1) {
-    inst->enc_Energy = 1;
-  }
-
-  if ((inst->enc_msSinceSID > (inst->enc_interval - 1)) || forceSID) {
-
-    /* Search for best dbov value. */
-    index = 0;
-    for (i = 1; i < 93; i++) {
-      /* Always round downwards. */
-      if ((inst->enc_Energy - WebRtcCng_kDbov[i]) > 0) {
-        index = i;
-        break;
-      }
-    }
-    if ((i == 93) && (index == 0))
-      index = 94;
-    SIDdata[0] = (uint8_t)index;
-
-    /* Quantize coefficients with tweak for WebRtc implementation of RFC3389. */
-    if (inst->enc_nrOfCoefs == WEBRTC_CNG_MAX_LPC_ORDER) {
-      for (i = 0; i < inst->enc_nrOfCoefs; i++) {
-        /* Q15 to Q7 with rounding. */
-        SIDdata[i + 1] = ((inst->enc_reflCoefs[i] + 128) >> 8);
-      }
-    } else {
-      for (i = 0; i < inst->enc_nrOfCoefs; i++) {
-        /* Q15 to Q7 with rounding. */
-        SIDdata[i + 1] = (127 + ((inst->enc_reflCoefs[i] + 128) >> 8));
-      }
-    }
-
-    inst->enc_msSinceSID = 0;
-    *bytesOut = inst->enc_nrOfCoefs + 1;
-
-    inst->enc_msSinceSID +=
-        (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
-    return (int)(inst->enc_nrOfCoefs + 1);
-  } else {
-    inst->enc_msSinceSID +=
-        (int16_t)((1000 * nrOfSamples) / inst->enc_sampfreq);
-    *bytesOut = 0;
-    return 0;
-  }
-}
-
-/****************************************************************************
- * WebRtcCng_UpdateSid(...)
- *
- * These functions updates the CN state, when a new SID packet arrives
- *
- * Input:
- *    - cng_inst      : Pointer to created instance that should be freed
- *    - SID           : SID packet, all headers removed
- *    - length        : Length in bytes of SID packet
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
-                            size_t length) {
-
-  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
-  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
-  int32_t targetEnergy;
-  int i;
-
-  if (inst->initflag != 1) {
-    inst->errorcode = CNG_DECODER_NOT_INITIATED;
-    return -1;
-  }
-
-  /* Throw away reflection coefficients of higher order than we can handle. */
-  if (length > (WEBRTC_CNG_MAX_LPC_ORDER + 1))
-    length = WEBRTC_CNG_MAX_LPC_ORDER + 1;
-
-  inst->dec_order = (int16_t)length - 1;
-
-  if (SID[0] > 93)
-    SID[0] = 93;
-  targetEnergy = WebRtcCng_kDbov[SID[0]];
-  /* Take down target energy to 75%. */
-  targetEnergy = targetEnergy >> 1;
-  targetEnergy += targetEnergy >> 2;
-
-  inst->dec_target_energy = targetEnergy;
-
-  /* Reconstruct coeffs with tweak for WebRtc implementation of RFC3389. */
-  if (inst->dec_order == WEBRTC_CNG_MAX_LPC_ORDER) {
-    for (i = 0; i < (inst->dec_order); i++) {
-      refCs[i] = SID[i + 1] << 8; /* Q7 to Q15*/
-      inst->dec_target_reflCoefs[i] = refCs[i];
-    }
-  } else {
-    for (i = 0; i < (inst->dec_order); i++) {
-      refCs[i] = (SID[i + 1] - 127) << 8; /* Q7 to Q15. */
-      inst->dec_target_reflCoefs[i] = refCs[i];
-    }
-  }
-
-  for (i = (inst->dec_order); i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
-    refCs[i] = 0;
-    inst->dec_target_reflCoefs[i] = refCs[i];
-  }
-
-  return 0;
-}
-
-/****************************************************************************
- * WebRtcCng_Generate(...)
- *
- * These functions generates CN data when needed
- *
- * Input:
- *    - cng_inst      : Pointer to created instance that should be freed
- *    - outData       : pointer to area to write CN data
- *    - nrOfSamples   : How much data to generate
- *
- * Return value        :  0 - Ok
- *                       -1 - Error
- */
-int16_t WebRtcCng_Generate(CNG_dec_inst* cng_inst, int16_t* outData,
-                           size_t nrOfSamples, int16_t new_period) {
-  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
-
-  size_t i;
-  int16_t excitation[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
-  int16_t low[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
-  int16_t lpPoly[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-  int16_t ReflBetaStd = 26214;  /* 0.8 in q15. */
-  int16_t ReflBetaCompStd = 6553;  /* 0.2 in q15. */
-  int16_t ReflBetaNewP = 19661;  /* 0.6 in q15. */
-  int16_t ReflBetaCompNewP = 13107;  /* 0.4 in q15. */
-  int16_t Beta, BetaC, tmp1, tmp2, tmp3;
-  int32_t targetEnergy;
-  int16_t En;
-  int16_t temp16;
-
-  if (nrOfSamples > WEBRTC_CNG_MAX_OUTSIZE_ORDER) {
-    inst->errorcode = CNG_DISALLOWED_FRAME_SIZE;
-    return -1;
-  }
-
-  if (new_period) {
-    inst->dec_used_scale_factor = inst->dec_target_scale_factor;
-    Beta = ReflBetaNewP;
-    BetaC = ReflBetaCompNewP;
-  } else {
-    Beta = ReflBetaStd;
-    BetaC = ReflBetaCompStd;
-  }
-
-  /* Here we use a 0.5 weighting, should possibly be modified to 0.6. */
-  tmp1 = inst->dec_used_scale_factor << 2; /* Q13->Q15 */
-  tmp2 = inst->dec_target_scale_factor << 2; /* Q13->Q15 */
-  tmp3 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp1, Beta, 15);
-  tmp3 += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp2, BetaC, 15);
-  inst->dec_used_scale_factor = tmp3 >> 2; /* Q15->Q13 */
-
-  inst->dec_used_energy = inst->dec_used_energy >> 1;
-  inst->dec_used_energy += inst->dec_target_energy >> 1;
-
-  /* Do the same for the reflection coeffs. */
-  for (i = 0; i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
-    inst->dec_used_reflCoefs[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-        inst->dec_used_reflCoefs[i], Beta, 15);
-    inst->dec_used_reflCoefs[i] += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-        inst->dec_target_reflCoefs[i], BetaC, 15);
-  }
-
-  /* Compute the polynomial coefficients. */
-  WebRtcCng_K2a16(inst->dec_used_reflCoefs, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
-
-
-  targetEnergy = inst->dec_used_energy;
-
-  /* Calculate scaling factor based on filter energy. */
-  En = 8192;  /* 1.0 in Q13. */
-  for (i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER); i++) {
-
-    /* Floating point value for reference.
-       E *= 1.0 - (inst->dec_used_reflCoefs[i] / 32768.0) *
-       (inst->dec_used_reflCoefs[i] / 32768.0);
-     */
-
-    /* Same in fixed point. */
-    /* K(i).^2 in Q15. */
-    temp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
-        inst->dec_used_reflCoefs[i], inst->dec_used_reflCoefs[i], 15);
-    /* 1 - K(i).^2 in Q15. */
-    temp16 = 0x7fff - temp16;
-    En = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
-  }
-
-  /* float scaling= sqrt(E * inst->dec_target_energy / (1 << 24)); */
-
-  /* Calculate sqrt(En * target_energy / excitation energy) */
-  targetEnergy = WebRtcSpl_Sqrt(inst->dec_used_energy);
-
-  En = (int16_t) WebRtcSpl_Sqrt(En) << 6;
-  En = (En * 3) >> 1;  /* 1.5 estimates sqrt(2). */
-  inst->dec_used_scale_factor = (int16_t)((En * targetEnergy) >> 12);
-
-  /* Generate excitation. */
-  /* Excitation energy per sample is 2.^24 - Q13 N(0,1). */
-  for (i = 0; i < nrOfSamples; i++) {
-    excitation[i] = WebRtcSpl_RandN(&inst->dec_seed) >> 1;
-  }
-
-  /* Scale to correct energy. */
-  WebRtcSpl_ScaleVector(excitation, excitation, inst->dec_used_scale_factor,
-                        nrOfSamples, 13);
-
-  /* |lpPoly| - Coefficients in Q12.
-   * |excitation| - Speech samples.
-   * |nst->dec_filtstate| - State preservation.
-   * |outData| - Filtered speech samples. */
-  WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
-                     nrOfSamples, inst->dec_filtstate, WEBRTC_CNG_MAX_LPC_ORDER,
-                     inst->dec_filtstateLow, WEBRTC_CNG_MAX_LPC_ORDER, outData,
-                     low, nrOfSamples);
-
-  return 0;
-}
-
-/****************************************************************************
- * WebRtcCng_GetErrorCodeEnc/Dec(...)
- *
- * This functions can be used to check the error code of a CNG instance. When
- * a function returns -1 a error code will be set for that instance. The 
- * function below extract the code of the last error that occured in the 
- * specified instance.
- *
- * Input:
- *    - CNG_inst    : CNG enc/dec instance
- *
- * Return value     : Error code
- */
-int16_t WebRtcCng_GetErrorCodeEnc(CNG_enc_inst* cng_inst) {
-  /* Typecast pointer to real structure. */
-  WebRtcCngEncoder* inst = (WebRtcCngEncoder*) cng_inst;
-  return inst->errorcode;
-}
-
-int16_t WebRtcCng_GetErrorCodeDec(CNG_dec_inst* cng_inst) {
-  /* Typecast pointer to real structure. */
-  WebRtcCngDecoder* inst = (WebRtcCngDecoder*) cng_inst;
-  return inst->errorcode;
-}
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
new file mode 100644
index 00000000000..b4da260dba2
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
@@ -0,0 +1,442 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
+
+#include <algorithm>
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kCngMaxOutsizeOrder = 640;
+
+// TODO(ossu): Rename the left-over WebRtcCng according to style guide.
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a);
+
+const int32_t WebRtcCng_kDbov[94] = {
+  1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
+  271562548,  215709799, 171344384, 136103682, 108110997, 85875618,
+  68213428,   54183852,  43039763,  34187699,  27156255,  21570980,
+  17134438,   13610368,  10811100,  8587562,   6821343,   5418385,
+  4303976,    3418770,   2715625,   2157098,   1713444,   1361037,
+  1081110,    858756,    682134,    541839,    430398,    341877,
+  271563,     215710,    171344,    136104,    108111,    85876,
+  68213,      54184,     43040,     34188,     27156,     21571,
+  17134,      13610,     10811,     8588,      6821,      5418,
+  4304,       3419,      2716,      2157,      1713,      1361,
+  1081,       859,       682,       542,       430,       342,
+  272,        216,       171,       136,       108,       86,
+  68,         54,        43,        34,        27,        22,
+  17,         14,        11,        9,         7,         5,
+  4,          3,         3,         2,         2,         1,
+  1,          1,         1,         1
+};
+
+const int16_t WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
+  32702, 32636, 32570, 32505, 32439, 32374,
+  32309, 32244, 32179, 32114, 32049, 31985
+};
+
+}  // namespace
+
+ComfortNoiseDecoder::ComfortNoiseDecoder() {
+  /* Needed to get the right function pointers in SPLIB. */
+  WebRtcSpl_Init();
+  Reset();
+}
+
+void ComfortNoiseDecoder::Reset() {
+  dec_seed_ = 7777;  /* For debugging only. */
+  dec_target_energy_ = 0;
+  dec_used_energy_ = 0;
+  for (auto& c : dec_target_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_used_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_filtstate_)
+    c = 0;
+  for (auto& c : dec_filtstateLow_)
+    c = 0;
+  dec_order_ = 5;
+  dec_target_scale_factor_ = 0;
+  dec_used_scale_factor_ = 0;
+}
+
+void ComfortNoiseDecoder::UpdateSid(rtc::ArrayView<const uint8_t> sid) {
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
+  int32_t targetEnergy;
+  size_t length = sid.size();
+  /* Throw away reflection coefficients of higher order than we can handle. */
+  if (length > (WEBRTC_CNG_MAX_LPC_ORDER + 1))
+    length = WEBRTC_CNG_MAX_LPC_ORDER + 1;
+
+  dec_order_ = static_cast<uint16_t>(length - 1);
+
+  uint8_t sid0 = std::min<uint8_t>(sid[0], 93);
+  targetEnergy = WebRtcCng_kDbov[sid0];
+  /* Take down target energy to 75%. */
+  targetEnergy = targetEnergy >> 1;
+  targetEnergy += targetEnergy >> 2;
+
+  dec_target_energy_ = targetEnergy;
+
+  /* Reconstruct coeffs with tweak for WebRtc implementation of RFC3389. */
+  if (dec_order_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = sid[i + 1] << 8; /* Q7 to Q15*/
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  } else {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = (sid[i + 1] - 127) << 8; /* Q7 to Q15. */
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  }
+
+  for (size_t i = (dec_order_); i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    refCs[i] = 0;
+    dec_target_reflCoefs_[i] = refCs[i];
+  }
+}
+
+bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
+                                   bool new_period) {
+  int16_t excitation[kCngMaxOutsizeOrder];
+  int16_t low[kCngMaxOutsizeOrder];
+  int16_t lpPoly[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t ReflBetaStd = 26214;  /* 0.8 in q15. */
+  int16_t ReflBetaCompStd = 6553;  /* 0.2 in q15. */
+  int16_t ReflBetaNewP = 19661;  /* 0.6 in q15. */
+  int16_t ReflBetaCompNewP = 13107;  /* 0.4 in q15. */
+  int16_t Beta, BetaC, tmp1, tmp2, tmp3;
+  int32_t targetEnergy;
+  int16_t En;
+  int16_t temp16;
+  const size_t num_samples = out_data.size();
+
+  if (num_samples > kCngMaxOutsizeOrder) {
+    return false;
+  }
+
+  if (new_period) {
+    dec_used_scale_factor_ = dec_target_scale_factor_;
+    Beta = ReflBetaNewP;
+    BetaC = ReflBetaCompNewP;
+  } else {
+    Beta = ReflBetaStd;
+    BetaC = ReflBetaCompStd;
+  }
+
+  /* Here we use a 0.5 weighting, should possibly be modified to 0.6. */
+  tmp1 = dec_used_scale_factor_ << 2; /* Q13->Q15 */
+  tmp2 = dec_target_scale_factor_ << 2; /* Q13->Q15 */
+  tmp3 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp1, Beta, 15);
+  tmp3 += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp2, BetaC, 15);
+  dec_used_scale_factor_ = tmp3 >> 2; /* Q15->Q13 */
+
+  dec_used_energy_  = dec_used_energy_ >> 1;
+  dec_used_energy_ += dec_target_energy_ >> 1;
+
+  /* Do the same for the reflection coeffs. */
+  for (size_t i = 0; i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    dec_used_reflCoefs_[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
+        dec_used_reflCoefs_[i], Beta, 15);
+    dec_used_reflCoefs_[i] += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
+        dec_target_reflCoefs_[i], BetaC, 15);
+  }
+
+  /* Compute the polynomial coefficients. */
+  WebRtcCng_K2a16(dec_used_reflCoefs_, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
+
+
+  targetEnergy = dec_used_energy_;
+
+  /* Calculate scaling factor based on filter energy. */
+  En = 8192;  /* 1.0 in Q13. */
+  for (size_t i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER); i++) {
+    /* Floating point value for reference.
+       E *= 1.0 - (dec_used_reflCoefs_[i] / 32768.0) *
+       (dec_used_reflCoefs_[i] / 32768.0);
+     */
+
+    /* Same in fixed point. */
+    /* K(i).^2 in Q15. */
+    temp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
+        dec_used_reflCoefs_[i], dec_used_reflCoefs_[i], 15);
+    /* 1 - K(i).^2 in Q15. */
+    temp16 = 0x7fff - temp16;
+    En = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
+  }
+
+  /* float scaling= sqrt(E * dec_target_energy_ / (1 << 24)); */
+
+  /* Calculate sqrt(En * target_energy / excitation energy) */
+  targetEnergy = WebRtcSpl_Sqrt(dec_used_energy_);
+
+  En = (int16_t) WebRtcSpl_Sqrt(En) << 6;
+  En = (En * 3) >> 1;  /* 1.5 estimates sqrt(2). */
+  dec_used_scale_factor_ = (int16_t)((En * targetEnergy) >> 12);
+
+  /* Generate excitation. */
+  /* Excitation energy per sample is 2.^24 - Q13 N(0,1). */
+  for (size_t i = 0; i < num_samples; i++) {
+    excitation[i] = WebRtcSpl_RandN(&dec_seed_) >> 1;
+  }
+
+  /* Scale to correct energy. */
+  WebRtcSpl_ScaleVector(excitation, excitation, dec_used_scale_factor_,
+                        num_samples, 13);
+
+  /* |lpPoly| - Coefficients in Q12.
+   * |excitation| - Speech samples.
+   * |nst->dec_filtstate| - State preservation.
+   * |out_data| - Filtered speech samples. */
+  WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
+                     num_samples, dec_filtstate_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     dec_filtstateLow_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     out_data.data(), low, num_samples);
+
+  return true;
+}
+
+ComfortNoiseEncoder::ComfortNoiseEncoder(int fs, int interval, int quality)
+    : enc_nrOfCoefs_(quality),
+      enc_sampfreq_(fs),
+      enc_interval_(interval),
+      enc_msSinceSid_(0),
+      enc_Energy_(0),
+      enc_reflCoefs_{0},
+      enc_corrVector_{0},
+      enc_seed_(7777)  /* For debugging only. */ {
+  RTC_CHECK(quality <= WEBRTC_CNG_MAX_LPC_ORDER && quality > 0);
+  /* Needed to get the right function pointers in SPLIB. */
+  WebRtcSpl_Init();
+}
+
+void ComfortNoiseEncoder::Reset(int fs, int interval, int quality) {
+  RTC_CHECK(quality <= WEBRTC_CNG_MAX_LPC_ORDER && quality > 0);
+  enc_nrOfCoefs_ = quality;
+  enc_sampfreq_ = fs;
+  enc_interval_ = interval;
+  enc_msSinceSid_ = 0;
+  enc_Energy_ = 0;
+  for (auto& c : enc_reflCoefs_)
+    c = 0;
+  for (auto& c : enc_corrVector_)
+    c = 0;
+  enc_seed_ = 7777;  /* For debugging only. */
+}
+
+size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
+                                   bool force_sid,
+                                   rtc::Buffer* output) {
+  int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t corrVector[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t hanningW[kCngMaxOutsizeOrder];
+  int16_t ReflBeta = 19661;     /* 0.6 in q15. */
+  int16_t ReflBetaComp = 13107; /* 0.4 in q15. */
+  int32_t outEnergy;
+  int outShifts;
+  size_t i;
+  int stab;
+  int acorrScale;
+  size_t index;
+  size_t ind, factor;
+  int32_t* bptr;
+  int32_t blo, bhi;
+  int16_t negate;
+  const int16_t* aptr;
+  int16_t speechBuf[kCngMaxOutsizeOrder];
+
+  const size_t num_samples = speech.size();
+  RTC_CHECK_LE(num_samples, static_cast<size_t>(kCngMaxOutsizeOrder));
+
+  for (i = 0; i < num_samples; i++) {
+    speechBuf[i] = speech[i];
+  }
+
+  factor = num_samples;
+
+  /* Calculate energy and a coefficients. */
+  outEnergy = WebRtcSpl_Energy(speechBuf, num_samples, &outShifts);
+  while (outShifts > 0) {
+    /* We can only do 5 shifts without destroying accuracy in
+     * division factor. */
+    if (outShifts > 5) {
+      outEnergy <<= (outShifts - 5);
+      outShifts = 5;
+    } else {
+      factor /= 2;
+      outShifts--;
+    }
+  }
+  outEnergy = WebRtcSpl_DivW32W16(outEnergy, (int16_t)factor);
+
+  if (outEnergy > 1) {
+    /* Create Hanning Window. */
+    WebRtcSpl_GetHanningWindow(hanningW, num_samples / 2);
+    for (i = 0; i < (num_samples / 2); i++)
+      hanningW[num_samples - i - 1] = hanningW[i];
+
+    WebRtcSpl_ElementwiseVectorMult(speechBuf, hanningW, speechBuf, num_samples,
+                                    14);
+
+    WebRtcSpl_AutoCorrelation(speechBuf, num_samples, enc_nrOfCoefs_,
+                              corrVector, &acorrScale);
+
+    if (*corrVector == 0)
+      *corrVector = WEBRTC_SPL_WORD16_MAX;
+
+    /* Adds the bandwidth expansion. */
+    aptr = WebRtcCng_kCorrWindow;
+    bptr = corrVector;
+
+    /* (zzz) lpc16_1 = 17+1+820+2+2 = 842 (ordo2=700). */
+    for (ind = 0; ind < enc_nrOfCoefs_; ind++) {
+      /* The below code multiplies the 16 b corrWindow values (Q15) with
+       * the 32 b corrvector (Q0) and shifts the result down 15 steps. */
+      negate = *bptr < 0;
+      if (negate)
+        *bptr = -*bptr;
+
+      blo = (int32_t) * aptr * (*bptr & 0xffff);
+      bhi = ((blo >> 16) & 0xffff)
+          + ((int32_t)(*aptr++) * ((*bptr >> 16) & 0xffff));
+      blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
+
+      *bptr = (((bhi >> 16) & 0x7fff) << 17) | ((uint32_t) blo >> 15);
+      if (negate)
+        *bptr = -*bptr;
+      bptr++;
+    }
+    /* End of bandwidth expansion. */
+
+    stab = WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs,
+                                    enc_nrOfCoefs_);
+
+    if (!stab) {
+      /* Disregard from this frame */
+      return 0;
+    }
+
+  } else {
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      refCs[i] = 0;
+  }
+
+  if (force_sid) {
+    /* Read instantaneous values instead of averaged. */
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      enc_reflCoefs_[i] = refCs[i];
+    enc_Energy_ = outEnergy;
+  } else {
+    /* Average history with new values. */
+    for (i = 0; i < enc_nrOfCoefs_; i++) {
+      enc_reflCoefs_[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
+          enc_reflCoefs_[i], ReflBeta, 15);
+      enc_reflCoefs_[i] +=
+          (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(refCs[i], ReflBetaComp, 15);
+    }
+    enc_Energy_ =
+        (outEnergy >> 2) + (enc_Energy_ >> 1) + (enc_Energy_ >> 2);
+  }
+
+  if (enc_Energy_ < 1) {
+    enc_Energy_ = 1;
+  }
+
+  if ((enc_msSinceSid_ > (enc_interval_ - 1)) || force_sid) {
+    /* Search for best dbov value. */
+    index = 0;
+    for (i = 1; i < 93; i++) {
+      /* Always round downwards. */
+      if ((enc_Energy_ - WebRtcCng_kDbov[i]) > 0) {
+        index = i;
+        break;
+      }
+    }
+    if ((i == 93) && (index == 0))
+      index = 94;
+
+    const size_t output_coefs = enc_nrOfCoefs_ + 1;
+    output->AppendData(output_coefs, [&] (rtc::ArrayView<uint8_t> output) {
+        output[0] = (uint8_t)index;
+
+        /* Quantize coefficients with tweak for WebRtc implementation of
+         * RFC3389. */
+        if (enc_nrOfCoefs_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+          for (i = 0; i < enc_nrOfCoefs_; i++) {
+            /* Q15 to Q7 with rounding. */
+            output[i + 1] = ((enc_reflCoefs_[i] + 128) >> 8);
+          }
+        } else {
+          for (i = 0; i < enc_nrOfCoefs_; i++) {
+            /* Q15 to Q7 with rounding. */
+            output[i + 1] = (127 + ((enc_reflCoefs_[i] + 128) >> 8));
+          }
+        }
+
+        return output_coefs;
+      });
+
+    enc_msSinceSid_ =
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return output_coefs;
+  } else {
+    enc_msSinceSid_ +=
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return 0;
+  }
+}
+
+namespace {
+/* Values in |k| are Q15, and |a| Q12. */
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a) {
+  int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
+  int16_t* aptr;
+  int16_t* aptr2;
+  int16_t* anyptr;
+  const int16_t* kptr;
+  int m, i;
+
+  kptr = k;
+  *a = 4096; /* i.e., (Word16_MAX >> 3) + 1 */
+  *any = *a;
+  a[1] = (*k + 4) >> 3;
+  for (m = 1; m < useOrder; m++) {
+    kptr++;
+    aptr = a;
+    aptr++;
+    aptr2 = &a[m];
+    anyptr = any;
+    anyptr++;
+
+    any[m + 1] = (*kptr + 4) >> 3;
+    for (i = 0; i < m; i++) {
+      *anyptr++ =
+          (*aptr++) +
+          (int16_t)((((int32_t)(*aptr2--) * (int32_t)*kptr) + 16384) >> 15);
+    }
+
+    aptr = a;
+    anyptr = any;
+    for (i = 0; i < (m + 2); i++) {
+      *aptr++ = *anyptr++;
+    }
+  }
+}
+
+}  // namespace
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
index 64bea1e26f6..fb0a53df270 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -12,152 +12,88 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
 
-#include <stddef.h>
-#include "webrtc/typedefs.h"
+#include <cstddef>
 
-#ifdef __cplusplus
-extern "C" {
-#endif
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/typedefs.h"
 
 #define WEBRTC_CNG_MAX_LPC_ORDER 12
-#define WEBRTC_CNG_MAX_OUTSIZE_ORDER 640
-
-/* Define Error codes. */
-
-/* 6100 Encoder */
-#define CNG_ENCODER_NOT_INITIATED               6120
-#define CNG_DISALLOWED_LPC_ORDER                6130
-#define CNG_DISALLOWED_FRAME_SIZE               6140
-#define CNG_DISALLOWED_SAMPLING_FREQUENCY       6150
-/* 6200 Decoder */
-#define CNG_DECODER_NOT_INITIATED               6220
-
-typedef struct WebRtcCngEncInst CNG_enc_inst;
-typedef struct WebRtcCngDecInst CNG_dec_inst;
-
-/****************************************************************************
- * WebRtcCng_CreateEnc/Dec(...)
- *
- * These functions create an instance to the specified structure
- *
- * Input:
- *    - XXX_inst      : Pointer to created instance that should be created
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int16_t WebRtcCng_CreateEnc(CNG_enc_inst** cng_inst);
-int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
-
-/****************************************************************************
- * WebRtcCng_InitEnc/Dec(...)
- *
- * This function initializes a instance
- *
- * Input:
- *    - cng_inst      : Instance that should be initialized
- *
- *    - fs            : 8000 for narrowband and 16000 for wideband
- *    - interval      : generate SID data every interval ms
- *    - quality       : Number of refl. coefs, maximum allowed is 12
- *
- * Output:
- *    - cng_inst      : Initialized instance
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-
-int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
-                      int16_t quality);
-void WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
-
-/****************************************************************************
- * WebRtcCng_FreeEnc/Dec(...)
- *
- * These functions frees the dynamic memory of a specified instance
- *
- * Input:
- *    - cng_inst      : Pointer to created instance that should be freed
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int16_t WebRtcCng_FreeEnc(CNG_enc_inst* cng_inst);
-int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst);
-
-/****************************************************************************
- * WebRtcCng_Encode(...)
- *
- * These functions analyzes background noise
- *
- * Input:
- *    - cng_inst      : Pointer to created instance
- *    - speech        : Signal to be analyzed
- *    - nrOfSamples   : Size of speech vector
- *    - forceSID      : not zero to force SID frame and reset
- *
- * Output:
- *    - bytesOut      : Nr of bytes to transmit, might be 0
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
-                     size_t nrOfSamples, uint8_t* SIDdata,
-                     size_t* bytesOut, int16_t forceSID);
-
-/****************************************************************************
- * WebRtcCng_UpdateSid(...)
- *
- * These functions updates the CN state, when a new SID packet arrives
- *
- * Input:
- *    - cng_inst      : Pointer to created instance that should be freed
- *    - SID           : SID packet, all headers removed
- *    - length        : Length in bytes of SID packet
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
-                            size_t length);
-
-/****************************************************************************
- * WebRtcCng_Generate(...)
- *
- * These functions generates CN data when needed
- *
- * Input:
- *    - cng_inst      : Pointer to created instance that should be freed
- *    - outData       : pointer to area to write CN data
- *    - nrOfSamples   : How much data to generate
- *    - new_period    : >0 if a new period of CNG, will reset history
- *
- * Return value       :  0 - Ok
- *                      -1 - Error
- */
-int16_t WebRtcCng_Generate(CNG_dec_inst* cng_inst, int16_t* outData,
-                           size_t nrOfSamples, int16_t new_period);
-
-/*****************************************************************************
- * WebRtcCng_GetErrorCodeEnc/Dec(...)
- *
- * This functions can be used to check the error code of a CNG instance. When
- * a function returns -1 a error code will be set for that instance. The
- * function below extract the code of the last error that occurred in the
- * specified instance.
- *
- * Input:
- *    - CNG_inst    : CNG enc/dec instance
- *
- * Return value     : Error code
- */
-int16_t WebRtcCng_GetErrorCodeEnc(CNG_enc_inst* cng_inst);
-int16_t WebRtcCng_GetErrorCodeDec(CNG_dec_inst* cng_inst);
 
-#ifdef __cplusplus
-}
-#endif
+namespace webrtc {
+
+class ComfortNoiseDecoder {
+ public:
+  ComfortNoiseDecoder();
+  ~ComfortNoiseDecoder() = default;
+
+  ComfortNoiseDecoder(const ComfortNoiseDecoder&) = delete;
+  ComfortNoiseDecoder& operator=(const ComfortNoiseDecoder&) = delete;
+
+  void Reset();
+
+  // Updates the CN state when a new SID packet arrives.
+  // |sid| is a view of the SID packet without the headers.
+  void UpdateSid(rtc::ArrayView<const uint8_t> sid);
+
+  // Generates comfort noise.
+  // |out_data| will be filled with samples - its size determines the number of
+  // samples generated. When |new_period| is true, CNG history will be reset
+  // before any audio is generated.  Returns |false| if outData is too large -
+  // currently 640 bytes (equalling 10ms at 64kHz).
+  // TODO(ossu): Specify better limits for the size of out_data. Either let it
+  //             be unbounded or limit to 10ms in the current sample rate.
+  bool Generate(rtc::ArrayView<int16_t> out_data, bool new_period);
+
+ private:
+  uint32_t dec_seed_;
+  int32_t dec_target_energy_;
+  int32_t dec_used_energy_;
+  int16_t dec_target_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_used_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstate_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstateLow_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint16_t dec_order_;
+  int16_t dec_target_scale_factor_;  /* Q29 */
+  int16_t dec_used_scale_factor_;  /* Q29 */
+};
+
+class ComfortNoiseEncoder {
+ public:
+  // Creates a comfort noise encoder.
+  // |fs| selects sample rate: 8000 for narrowband or 16000 for wideband.
+  // |interval| sets the interval at which to generate SID data (in ms).
+  // |quality| selects the number of refl. coeffs. Maximum allowed is 12.
+  ComfortNoiseEncoder(int fs, int interval, int quality);
+  ~ComfortNoiseEncoder() = default;
+
+  ComfortNoiseEncoder(const ComfortNoiseEncoder&) = delete;
+  ComfortNoiseEncoder& operator=(const ComfortNoiseEncoder&) = delete;
+
+  // Resets the comfort noise encoder to its initial state.
+  // Parameters are set as during construction.
+  void Reset(int fs, int interval, int quality);
+
+  // Analyzes background noise from |speech| and appends coefficients to
+  // |output|.  Returns the number of coefficients generated.  If |force_sid| is
+  // true, a SID frame is forced and the internal sid interval counter is reset.
+  // Will fail if the input size is too large (> 640 samples, see
+  // ComfortNoiseDecoder::Generate).
+  size_t Encode(rtc::ArrayView<const int16_t> speech,
+                bool force_sid,
+                rtc::Buffer* output);
+
+ private:
+  size_t enc_nrOfCoefs_;
+  int enc_sampfreq_;
+  int16_t enc_interval_;
+  int16_t enc_msSinceSid_;
+  int32_t enc_Energy_;
+  int16_t enc_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t enc_corrVector_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint32_t enc_seed_;
+};
+
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 9dc3a6fd7ad..7a627e757c9 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index a24b1526fd2..baa5d382d32 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -52,10 +52,6 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
 
 AudioEncoderPcm::~AudioEncoderPcm() = default;
 
-size_t AudioEncoderPcm::MaxEncodedBytes() const {
-  return full_frame_samples_ * BytesPerSample();
-}
-
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
@@ -93,13 +89,14 @@ AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.encoded_bytes =
-      encoded->AppendData(MaxEncodedBytes(),
+      encoded->AppendData(full_frame_samples_ * BytesPerSample(),
                           [&] (rtc::ArrayView<uint8_t> encoded) {
                             return EncodeCall(&speech_buffer_[0],
                                               full_frame_samples_,
                                               encoded.data());
                           });
   speech_buffer_.clear();
+  info.encoder_type = GetCodecType();
   return info;
 }
 
@@ -120,6 +117,10 @@ size_t AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
+AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmA;
+}
+
 AudioEncoderPcmU::AudioEncoderPcmU(const CodecInst& codec_inst)
     : AudioEncoderPcmU(CreateConfig<AudioEncoderPcmU>(codec_inst)) {}
 
@@ -133,4 +134,8 @@ size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
+AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmU;
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index 6b3cebfb336..721344528f8 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -35,7 +35,6 @@ class AudioEncoderPcm : public AudioEncoder {
 
   ~AudioEncoderPcm() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
@@ -56,6 +55,10 @@ class AudioEncoderPcm : public AudioEncoder {
 
   virtual size_t BytesPerSample() const = 0;
 
+  // Used to set EncodedInfoLeaf::encoder_type in
+  // AudioEncoderPcm::EncodeImpl
+  virtual AudioEncoder::CodecType GetCodecType() const = 0;
+
  private:
   const int sample_rate_hz_;
   const size_t num_channels_;
@@ -85,6 +88,8 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
 
   size_t BytesPerSample() const override;
 
+  AudioEncoder::CodecType GetCodecType() const override;
+
  private:
   static const int kSampleRateHz = 8000;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA);
@@ -107,6 +112,8 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
 
   size_t BytesPerSample() const override;
 
+  AudioEncoder::CodecType GetCodecType() const override;
+
  private:
   static const int kSampleRateHz = 8000;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
index 7cc2ea98773..1837ffabe29 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 typedef struct WebRtcG722DecInst G722DecInst;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index 9256518445d..1f3936c8eee 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -60,10 +60,6 @@ AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
 
 AudioEncoderG722::~AudioEncoderG722() = default;
 
-size_t AudioEncoderG722::MaxEncodedBytes() const {
-  return SamplesPerChannel() / 2 * num_channels_;
-}
-
 int AudioEncoderG722::SampleRateHz() const {
   return kSampleRateHz;
 }
@@ -149,6 +145,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeImpl(
       });
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kG722;
   return info;
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index dec87b2b7a4..ad49a865e25 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
@@ -35,7 +36,6 @@ class AudioEncoderG722 final : public AudioEncoder {
   explicit AudioEncoderG722(const CodecInst& codec_inst);
   ~AudioEncoderG722() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
@@ -44,7 +44,7 @@ class AudioEncoderG722 final : public AudioEncoder {
   int GetTargetBitrate() const override;
   void Reset() override;
 
-protected:
+ protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
index e890635da09..036c11fac47 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 
 typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index c7d7411c45d..ca11587dfab 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -56,10 +56,6 @@ AudioEncoderIlbc::~AudioEncoderIlbc() {
   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }
 
-size_t AudioEncoderIlbc::MaxEncodedBytes() const {
-  return RequiredOutputSizeBytes();
-}
-
 int AudioEncoderIlbc::SampleRateHz() const {
   return kSampleRateHz;
 }
@@ -131,6 +127,7 @@ AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeImpl(
   info.encoded_bytes = encoded_bytes;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = config_.payload_type;
+  info.encoder_type = CodecType::kIlbc;
   return info;
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index 27329bbc4ee..63639860f45 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -34,7 +34,6 @@ class AudioEncoderIlbc final : public AudioEncoder {
   explicit AudioEncoderIlbc(const CodecInst& codec_inst);
   ~AudioEncoderIlbc() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
index 62a686495b1..a8375afb609 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
@@ -65,15 +65,15 @@ void WebRtcIlbcfix_GetLspPoly(
     {
       /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */
       high = (int16_t)(fPtr[-1] >> 16);
-      low = (int16_t)((fPtr[-1] - ((int32_t)high << 16)) >> 1);
+      low = (int16_t)((fPtr[-1] & 0xffff) >> 1);
 
-      tmpW32 = ((high * *lspPtr) << 2) + (((low * *lspPtr) >> 15) << 2);
+      tmpW32 = 4 * high * *lspPtr + 4 * ((low * *lspPtr) >> 15);
 
       (*fPtr) += fPtr[-2];
       (*fPtr) -= tmpW32;
       fPtr--;
     }
-    *fPtr -= *lspPtr << 10;
+    *fPtr -= *lspPtr * (1 << 10);
 
     fPtr+=i;
     lspPtr+=2;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
index bd101bf30ca..8b18c047b93 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
@@ -48,7 +48,7 @@ void WebRtcIlbcfix_HpOutput(
     tmpW32 = (tmpW32>>15);
     tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
     tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
-    tmpW32 = (tmpW32<<1);
+    tmpW32 *= 2;
 
     tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
     tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
@@ -77,11 +77,11 @@ void WebRtcIlbcfix_HpOutput(
     } else if (tmpW32<-268435456) {
       tmpW32 = WEBRTC_SPL_WORD32_MIN;
     } else {
-      tmpW32 <<= 3;
+      tmpW32 *= 8;
     }
 
     y[0] = (int16_t)(tmpW32 >> 16);
-    y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1);
+    y[1] = (int16_t)((tmpW32 & 0xffff) >> 1);
 
   }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/interfaces.gypi b/chromium/third_party/webrtc/modules/audio_coding/codecs/interfaces.gypi
index d4f6a4a41e6..1aba106f909 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/interfaces.gypi
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/interfaces.gypi
@@ -15,6 +15,10 @@
         'audio_decoder.cc',
         'audio_decoder.h',
       ],
+      'dependencies': [
+        '<(webrtc_root)/base/base.gyp:rtc_base_approved',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+      ],
     },
 
     {
@@ -24,6 +28,10 @@
         'audio_encoder.cc',
         'audio_encoder.h',
       ],
+      'dependencies': [
+        '<(webrtc_root)/base/base.gyp:rtc_base_approved',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+      ],
     },
   ],
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
index d9d20ec0396..b1907bbb394 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 0da8ed71d66..f1f2714ff9c 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
@@ -56,7 +57,6 @@ class AudioEncoderIsacT final : public AudioEncoder {
       const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
   ~AudioEncoderIsacT() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 1debbeb9038..b6a1747c391 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -80,11 +80,6 @@ AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
 }
 
 template <typename T>
-size_t AudioEncoderIsacT<T>::MaxEncodedBytes() const {
-  return kSufficientEncodeBufferSizeBytes;
-}
-
-template <typename T>
 int AudioEncoderIsacT<T>::SampleRateHz() const {
   return T::EncSampRate(isac_state_);
 }
@@ -150,6 +145,7 @@ AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeImpl(
   info.encoded_bytes = encoded_bytes;
   info.encoded_timestamp = packet_timestamp_;
   info.payload_type = config_.payload_type;
+  info.encoder_type = CodecType::kIsac;
   return info;
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
index fdbb2fcb0d7..001a04f39bf 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
@@ -90,7 +90,7 @@ void WebRtcIsacfix_Spec2TimeC(int16_t* inreQ7,
                               int32_t* outre1Q16,
                               int32_t* outre2Q16);
 
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 void WebRtcIsacfix_Time2SpecNeon(int16_t* inre1Q9,
                                  int16_t* inre2Q9,
                                  int16_t* outre,
@@ -174,7 +174,7 @@ void WebRtcIsacfix_FilterMaLoopC(int16_t input0,
                                  int32_t* ptr1,
                                  int32_t* ptr2);
 
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 int WebRtcIsacfix_AutocorrNeon(int32_t* __restrict r,
                                const int16_t* __restrict x,
                                int16_t N,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
index 2c8c923cd33..1b87d0ea557 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
@@ -147,7 +147,7 @@ void WebRtcIsacfix_MatrixProduct2C(const int16_t matrix0[],
                                    const int matrix0_index_factor,
                                    const int matrix0_index_step);
 
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[],
                                       const int32_t matrix1[],
                                       int32_t matrix_product[],
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
index 0e67e300ac1..d488339b31f 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
@@ -60,7 +60,7 @@ void WebRtcIsacfix_AllpassFilter2FixDec16C(
    int32_t *filter_state_ch1,
    int32_t *filter_state_ch2);
 
-#if (defined WEBRTC_DETECT_NEON) || (defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
    int16_t *data_ch1,
    int16_t *data_ch2,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
index 0ec115414b8..4b03181e456 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
@@ -64,11 +64,7 @@ class FilterBanksTest : public testing::Test {
 
 TEST_F(FilterBanksTest, AllpassFilter2FixDec16Test) {
   CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16C);
-#ifdef WEBRTC_DETECT_NEON
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16Neon);
-  }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
   CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16Neon);
 #endif
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
index 5cce1e9f0b2..3ed57788a1f 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
@@ -59,11 +59,7 @@ class FiltersTest : public testing::Test {
 
 TEST_F(FiltersTest, AutocorrFixTest) {
   FiltersTester(WebRtcIsacfix_AutocorrC);
-#ifdef WEBRTC_DETECT_NEON
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    FiltersTester(WebRtcIsacfix_AutocorrNeon);
-  }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
   FiltersTester(WebRtcIsacfix_AutocorrNeon);
 #endif
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index aba3aa0c0bf..e7905ae81fa 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -201,7 +201,7 @@ int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst)
  * This function initializes function pointers for ARM Neon platform.
  */
 
-#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 static void WebRtcIsacfix_InitNeon(void) {
   WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrNeon;
   WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopNeon;
@@ -253,11 +253,7 @@ static void InitFunctionPointers(void) {
   WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1C;
   WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2C;
 
-#ifdef WEBRTC_DETECT_NEON
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    WebRtcIsacfix_InitNeon();
-  }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
   WebRtcIsacfix_InitNeon();
 #endif
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
index 18377dd370f..0d881e80442 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
@@ -57,8 +57,6 @@ void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) {
     ysum32 += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1] >>
         scaling;
 
-    // TODO(zhongwei.yao): Move this function into a separate NEON code file so
-    // that WEBRTC_DETECT_NEON could take advantage of it.
 #ifdef WEBRTC_HAS_NEON
     {
       int32_t vbuff[4];
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
index 58d890011fe..c5cc87ffce2 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
@@ -179,22 +179,14 @@ class TransformTest : public testing::Test {
 
 TEST_F(TransformTest, Time2SpecTest) {
   Time2SpecTester(WebRtcIsacfix_Time2SpecC);
-#ifdef WEBRTC_DETECT_NEON
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    Time2SpecTester(WebRtcIsacfix_Time2SpecNeon);
-  }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
   Time2SpecTester(WebRtcIsacfix_Time2SpecNeon);
 #endif
 }
 
 TEST_F(TransformTest, Spec2TimeTest) {
   Spec2TimeTester(WebRtcIsacfix_Spec2TimeC);
-#ifdef WEBRTC_DETECT_NEON
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    Spec2TimeTester(WebRtcIsacfix_Spec2TimeNeon);
-  }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
   Spec2TimeTester(WebRtcIsacfix_Spec2TimeNeon);
 #endif
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index 32f36c52617..276eb60e280 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -25,10 +25,10 @@ class IsacSpeedTest : public AudioCodecSpeedTest {
   IsacSpeedTest();
   void SetUp() override;
   void TearDown() override;
-  virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                             size_t max_bytes, size_t* encoded_bytes);
-  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
-                             int16_t* out_data);
+  float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
+                     size_t max_bytes, size_t* encoded_bytes) override;
+  float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
+                     int16_t* out_data) override;
   ISACFIX_MainStruct *ISACFIX_main_inst_;
 };
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
index 63e4928bd88..47bbe31b8ae 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
@@ -214,10 +214,10 @@ int WebRtcIsac_DecHistOneStepMulti(int *data,        /* output: data vector */
   if (streamdata->stream_index == 0)   /* first time decoder is called for this stream */
   {
     /* read first word from bytestream */
-    streamval = *stream_ptr << 24;
-    streamval |= *++stream_ptr << 16;
-    streamval |= *++stream_ptr << 8;
-    streamval |= *++stream_ptr;
+    streamval = (uint32_t)(*stream_ptr) << 24;
+    streamval |= (uint32_t)(*++stream_ptr) << 16;
+    streamval |= (uint32_t)(*++stream_ptr) << 8;
+    streamval |= (uint32_t)(*++stream_ptr);
   } else {
     streamval = streamdata->streamval;
   }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
index c1204ad03ad..f920dc2ef8b 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
@@ -162,9 +162,9 @@ static void FindInvArSpec(const int16_t* ARCoefQ12,
   }
 
   for (k = 0; k < FRAMESAMPLES / 8; k++) {
-    CurveQ16[FRAMESAMPLES_QUARTER - 1 - k] = CurveQ16[k] -
-        (diffQ16[k] << shftVal);
-    CurveQ16[k] += diffQ16[k] << shftVal;
+    int32_t diff_q16_shifted = (int32_t)((uint32_t)(diffQ16[k]) << shftVal);
+    CurveQ16[FRAMESAMPLES_QUARTER - 1 - k] = CurveQ16[k] - diff_q16_shifted;
+    CurveQ16[k] += diff_q16_shifted;
   }
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h
new file mode 100644
index 00000000000..6e5737c89b8
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_DECODER_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_DECODER_FACTORY_H_
+
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
+
+namespace webrtc {
+
+class MockAudioDecoderFactory : public AudioDecoderFactory {
+ public:
+  MOCK_METHOD0(GetSupportedFormats, std::vector<SdpAudioFormat>());
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format) {
+    std::unique_ptr<AudioDecoder> return_value;
+    MakeAudioDecoderMock(format, &return_value);
+    return return_value;
+  }
+  MOCK_METHOD2(MakeAudioDecoderMock,
+               void(const SdpAudioFormat& format,
+                    std::unique_ptr<AudioDecoder>* return_value));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_DECODER_FACTORY_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc
index 52849691ac6..a674eba6607 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.cc
@@ -49,26 +49,4 @@ AudioEncoder::EncodedInfo MockAudioEncoder::CopyEncoding::operator()(
   return info_;
 }
 
-MockAudioEncoderDeprecated::CopyEncoding::CopyEncoding(
-    AudioEncoder::EncodedInfo info,
-    rtc::ArrayView<const uint8_t> payload)
-    : info_(info), payload_(payload) { }
-
-MockAudioEncoderDeprecated::CopyEncoding::CopyEncoding(
-    rtc::ArrayView<const uint8_t> payload)
-    : payload_(payload) {
-  info_.encoded_bytes = payload_.size();
-}
-
-AudioEncoder::EncodedInfo MockAudioEncoderDeprecated::CopyEncoding::operator()(
-    uint32_t timestamp,
-    rtc::ArrayView<const int16_t> audio,
-    size_t max_bytes_encoded,
-    uint8_t* encoded) {
-  RTC_CHECK(encoded);
-  RTC_CHECK_LE(info_.encoded_bytes, payload_.size());
-  std::memcpy(encoded, payload_.data(), info_.encoded_bytes);
-  return info_;
-}
-
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
index 58a1e756f97..2ffb30b708a 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_
 
+#include <string>
+
 #include "webrtc/base/array_view.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
@@ -18,12 +20,15 @@
 
 namespace webrtc {
 
-class MockAudioEncoderBase : public AudioEncoder {
+class MockAudioEncoder : public AudioEncoder {
  public:
-  ~MockAudioEncoderBase() override { Die(); }
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
+  ~MockAudioEncoder() /* override */ { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_METHOD1(Mark, void(std::string desc));
-  MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
   MOCK_CONST_METHOD0(SampleRateHz, int());
   MOCK_CONST_METHOD0(NumChannels, size_t());
   MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
@@ -39,10 +44,7 @@ class MockAudioEncoderBase : public AudioEncoder {
   MOCK_METHOD1(SetTargetBitrate, void(int target_bps));
   MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
   MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
-};
 
-class MockAudioEncoder final : public MockAudioEncoderBase {
- public:
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD3(EncodeImpl,
                EncodedInfo(uint32_t timestamp,
@@ -53,11 +55,11 @@ class MockAudioEncoder final : public MockAudioEncoderBase {
    public:
     // Creates a functor that will return |info| and adjust the rtc::Buffer
     // given as input to it, so it is info.encoded_bytes larger.
-    FakeEncoding(const AudioEncoder::EncodedInfo& info);
+    explicit FakeEncoding(const AudioEncoder::EncodedInfo& info);
 
     // Shorthand version of the constructor above, for when only setting
     // encoded_bytes in the EncodedInfo object matters.
-    FakeEncoding(size_t encoded_bytes);
+    explicit FakeEncoding(size_t encoded_bytes);
 
     AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
                                          rtc::ArrayView<const int16_t> audio,
@@ -80,41 +82,12 @@ class MockAudioEncoder final : public MockAudioEncoderBase {
     // Shorthand version of the constructor above, for when you wish to append
     // the whole payload and do not care about any EncodedInfo attribute other
     // than encoded_bytes.
-    CopyEncoding(rtc::ArrayView<const uint8_t> payload);
+    explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload);
 
     AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
                                          rtc::ArrayView<const int16_t> audio,
                                          rtc::Buffer* encoded);
-   private:
-    AudioEncoder::EncodedInfo info_;
-    rtc::ArrayView<const uint8_t> payload_;
-  };
-
-};
-
-class MockAudioEncoderDeprecated final : public MockAudioEncoderBase {
- public:
-  // Note, we explicitly chose not to create a mock for the Encode method.
-  MOCK_METHOD4(EncodeInternal,
-               EncodedInfo(uint32_t timestamp,
-                           rtc::ArrayView<const int16_t> audio,
-                           size_t max_encoded_bytes,
-                           uint8_t* encoded));
 
-  // A functor like MockAudioEncoder::CopyEncoding above, but which has the
-  // deprecated Encode signature. Currently only used in one test and should be
-  // removed once that backwards compatibility is.
-  class CopyEncoding {
-   public:
-    CopyEncoding(AudioEncoder::EncodedInfo info,
-                 rtc::ArrayView<const uint8_t> payload);
-
-    CopyEncoding(rtc::ArrayView<const uint8_t> payload);
-
-    AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
-                                         rtc::ArrayView<const int16_t> audio,
-                                         size_t max_bytes_encoded,
-                                         uint8_t* encoded);
    private:
     AudioEncoder::EncodedInfo info_;
     rtc::ArrayView<const uint8_t> payload_;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index af32a84512e..be48ca988ef 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index a599e291d47..a2497c7862a 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
+#include <algorithm>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
@@ -100,16 +102,6 @@ AudioEncoderOpus::~AudioEncoderOpus() {
   RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
 }
 
-size_t AudioEncoderOpus::MaxEncodedBytes() const {
-  // Calculate the number of bytes we expect the encoder to produce,
-  // then multiply by two to give a wide margin for error.
-  const size_t bytes_per_millisecond =
-      static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
-  const size_t approx_encoded_bytes =
-      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
-  return 2 * approx_encoded_bytes;
-}
-
 int AudioEncoderOpus::SampleRateHz() const {
   return kSampleRateHz;
 }
@@ -198,7 +190,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
   RTC_CHECK_EQ(input_buffer_.size(),
                Num10msFramesPerPacket() * SamplesPer10msFrame());
 
-  const size_t max_encoded_bytes = MaxEncodedBytes();
+  const size_t max_encoded_bytes = SufficientOutputBufferSize();
   EncodedInfo info;
   info.encoded_bytes =
       encoded->AppendData(
@@ -220,6 +212,7 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl(
   info.payload_type = config_.payload_type;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
   info.speech = (info.encoded_bytes > 0);
+  info.encoder_type = CodecType::kOpus;
   return info;
 }
 
@@ -231,6 +224,16 @@ size_t AudioEncoderOpus::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
 }
 
+size_t AudioEncoderOpus::SufficientOutputBufferSize() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
 // If the given config is OK, recreate the Opus encoder instance with those
 // settings, save the config, and return true. Otherwise, do nothing and return
 // false.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 3f11af1f9e0..8900659f48e 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -54,7 +54,6 @@ class AudioEncoderOpus final : public AudioEncoder {
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
   ~AudioEncoderOpus() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
@@ -79,7 +78,7 @@ class AudioEncoderOpus final : public AudioEncoder {
   ApplicationMode application() const { return config_.application; }
   bool dtx_enabled() const { return config_.dtx_enabled; }
 
-protected:
+ protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
@@ -87,6 +86,7 @@ protected:
  private:
   size_t Num10msFramesPerPacket() const;
   size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
   bool RecreateEncoderInstance(const Config& config);
 
   Config config_;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
index 4d1aa42c89f..7165d29c8b4 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -23,10 +23,10 @@ class OpusSpeedTest : public AudioCodecSpeedTest {
   OpusSpeedTest();
   void SetUp() override;
   void TearDown() override;
-  virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                             size_t max_bytes, size_t* encoded_bytes);
-  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
-                             int16_t* out_data);
+  float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
+                     size_t max_bytes, size_t* encoded_bytes) override;
+  float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
+                     int16_t* out_data) override;
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
index f4d40223024..cafd3e851bd 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -26,6 +26,10 @@ size_t AudioEncoderPcm16B::BytesPerSample() const {
   return 2;
 }
 
+AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const {
+  return CodecType::kOther;
+}
+
 namespace {
 AudioEncoderPcm16B::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderPcm16B::Config config;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index 34a780b49de..bdc27a67e30 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 namespace webrtc {
@@ -38,6 +39,8 @@ class AudioEncoderPcm16B final : public AudioEncoderPcm {
 
   size_t BytesPerSample() const override;
 
+  AudioEncoder::CodecType GetCodecType() const override;
+
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcm16B);
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 4275f54103a..37fa55a4da1 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -19,12 +19,7 @@
 namespace webrtc {
 
 AudioEncoderCopyRed::Config::Config() = default;
-
-// TODO(kwiberg): =default this when Visual Studio learns to handle it.
-AudioEncoderCopyRed::Config::Config(Config&& c)
-    : payload_type(c.payload_type),
-      speech_encoder(std::move(c.speech_encoder)) {}
-
+AudioEncoderCopyRed::Config::Config(Config&&) = default;
 AudioEncoderCopyRed::Config::~Config() = default;
 
 AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config)
@@ -35,10 +30,6 @@ AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config)
 
 AudioEncoderCopyRed::~AudioEncoderCopyRed() = default;
 
-size_t AudioEncoderCopyRed::MaxEncodedBytes() const {
-  return 2 * speech_encoder_->MaxEncodedBytes();
-}
-
 int AudioEncoderCopyRed::SampleRateHz() const {
   return speech_encoder_->SampleRateHz();
 }
@@ -132,4 +123,9 @@ void AudioEncoderCopyRed::SetTargetBitrate(int bits_per_second) {
   speech_encoder_->SetTargetBitrate(bits_per_second);
 }
 
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCopyRed::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index a67ae486bb2..a08118364cc 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
@@ -37,7 +38,6 @@ class AudioEncoderCopyRed final : public AudioEncoder {
 
   ~AudioEncoderCopyRed() override;
 
-  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
@@ -51,8 +51,10 @@ class AudioEncoderCopyRed final : public AudioEncoder {
   void SetMaxPlaybackRate(int frequency_hz) override;
   void SetProjectedPacketLossRate(double fraction) override;
   void SetTargetBitrate(int target_bps) override;
+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
+      override;
 
-protected:
+ protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index c73cb9f2096..22b2ceb5f79 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -26,7 +26,6 @@ using ::testing::MockFunction;
 namespace webrtc {
 
 namespace {
-static const size_t kMockMaxEncodedBytes = 1000;
 static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
 }
 
@@ -46,8 +45,6 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
     EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
     EXPECT_CALL(*mock_encoder_, SampleRateHz())
         .WillRepeatedly(Return(sample_rate_hz_));
-    EXPECT_CALL(*mock_encoder_, MaxEncodedBytes())
-        .WillRepeatedly(Return(kMockMaxEncodedBytes));
   }
 
   void TearDown() override {
diff --git a/chromium/third_party/webrtc/modules/audio_coding/include/audio_coding_module.h b/chromium/third_party/webrtc/modules/audio_coding/include/audio_coding_module.h
index 381e35e639b..daf9ac8ae9a 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -686,13 +687,24 @@ class AudioCodingModule {
   //                         and other relevant parameters, c.f.
   //                         module_common_types.h for the definition of
   //                         AudioFrame.
+  //   -muted              : if true, the sample data in audio_frame is not
+  //                         populated, and must be interpreted as all zero.
   //
   // Return value:
   //   -1 if the function fails,
   //    0 if the function succeeds.
   //
   virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
-                                        AudioFrame* audio_frame) = 0;
+                                  AudioFrame* audio_frame,
+                                  bool* muted) = 0;
+
+  /////////////////////////////////////////////////////////////////////////////
+  // Same as above, but without the muted parameter. This methods should not be
+  // used if enable_fast_accelerate was set to true in NetEq::Config.
+  // TODO(henrik.lundin) Remove this method when downstream dependencies are
+  // ready.
+  virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
+                                  AudioFrame* audio_frame) = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   //   Codec specific
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index d800cc7dbe9..762c3859837 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -13,7 +13,6 @@
 #include <assert.h>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #ifdef WEBRTC_CODEC_G722
 #include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
@@ -36,43 +35,6 @@
 
 namespace webrtc {
 
-AudioDecoderCng::AudioDecoderCng() {
-  RTC_CHECK_EQ(0, WebRtcCng_CreateDec(&dec_state_));
-  WebRtcCng_InitDec(dec_state_);
-}
-
-AudioDecoderCng::~AudioDecoderCng() {
-  WebRtcCng_FreeDec(dec_state_);
-}
-
-void AudioDecoderCng::Reset() {
-  WebRtcCng_InitDec(dec_state_);
-}
-
-int AudioDecoderCng::IncomingPacket(const uint8_t* payload,
-                                    size_t payload_len,
-                                    uint16_t rtp_sequence_number,
-                                    uint32_t rtp_timestamp,
-                                    uint32_t arrival_timestamp) {
-  return -1;
-}
-
-CNG_dec_inst* AudioDecoderCng::CngDecoderInstance() {
-  return dec_state_;
-}
-
-size_t AudioDecoderCng::Channels() const {
-  return 1;
-}
-
-int AudioDecoderCng::DecodeInternal(const uint8_t* encoded,
-                                    size_t encoded_len,
-                                    int sample_rate_hz,
-                                    int16_t* decoded,
-                                    SpeechType* speech_type) {
-  return -1;
-}
-
 bool CodecSupported(NetEqDecoder codec_type) {
   switch (codec_type) {
     case NetEqDecoder::kDecoderPCMu:
@@ -175,67 +137,4 @@ int CodecSampleRateHz(NetEqDecoder codec_type) {
   }
 }
 
-AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
-  if (!CodecSupported(codec_type)) {
-    return NULL;
-  }
-  switch (codec_type) {
-    case NetEqDecoder::kDecoderPCMu:
-      return new AudioDecoderPcmU(1);
-    case NetEqDecoder::kDecoderPCMa:
-      return new AudioDecoderPcmA(1);
-    case NetEqDecoder::kDecoderPCMu_2ch:
-      return new AudioDecoderPcmU(2);
-    case NetEqDecoder::kDecoderPCMa_2ch:
-      return new AudioDecoderPcmA(2);
-#ifdef WEBRTC_CODEC_ILBC
-    case NetEqDecoder::kDecoderILBC:
-      return new AudioDecoderIlbc;
-#endif
-#if defined(WEBRTC_CODEC_ISACFX)
-    case NetEqDecoder::kDecoderISAC:
-      return new AudioDecoderIsacFix();
-#elif defined(WEBRTC_CODEC_ISAC)
-    case NetEqDecoder::kDecoderISAC:
-    case NetEqDecoder::kDecoderISACswb:
-      return new AudioDecoderIsac();
-#endif
-    case NetEqDecoder::kDecoderPCM16B:
-    case NetEqDecoder::kDecoderPCM16Bwb:
-    case NetEqDecoder::kDecoderPCM16Bswb32kHz:
-    case NetEqDecoder::kDecoderPCM16Bswb48kHz:
-      return new AudioDecoderPcm16B(1);
-    case NetEqDecoder::kDecoderPCM16B_2ch:
-    case NetEqDecoder::kDecoderPCM16Bwb_2ch:
-    case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
-    case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
-      return new AudioDecoderPcm16B(2);
-    case NetEqDecoder::kDecoderPCM16B_5ch:
-      return new AudioDecoderPcm16B(5);
-#ifdef WEBRTC_CODEC_G722
-    case NetEqDecoder::kDecoderG722:
-      return new AudioDecoderG722;
-    case NetEqDecoder::kDecoderG722_2ch:
-      return new AudioDecoderG722Stereo;
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-    case NetEqDecoder::kDecoderOpus:
-      return new AudioDecoderOpus(1);
-    case NetEqDecoder::kDecoderOpus_2ch:
-      return new AudioDecoderOpus(2);
-#endif
-    case NetEqDecoder::kDecoderCNGnb:
-    case NetEqDecoder::kDecoderCNGwb:
-    case NetEqDecoder::kDecoderCNGswb32kHz:
-    case NetEqDecoder::kDecoderCNGswb48kHz:
-      return new AudioDecoderCng;
-    case NetEqDecoder::kDecoderRED:
-    case NetEqDecoder::kDecoderAVT:
-    case NetEqDecoder::kDecoderArbitrary:
-    default: {
-      return NULL;
-    }
-  }
-}
-
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
index bc8bdd9626d..579ccb36f7a 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -16,7 +16,6 @@
 #include "webrtc/engine_configurations.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #ifdef WEBRTC_CODEC_G722
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 #endif
@@ -25,38 +24,6 @@
 
 namespace webrtc {
 
-// AudioDecoderCng is a special type of AudioDecoder. It inherits from
-// AudioDecoder just to fit in the DecoderDatabase. None of the class methods
-// should be used, except constructor, destructor, and accessors.
-// TODO(hlundin): Consider the possibility to create a super-class to
-// AudioDecoder that is stored in DecoderDatabase. Then AudioDecoder and a
-// specific CngDecoder class could both inherit from that class.
-class AudioDecoderCng : public AudioDecoder {
- public:
-  explicit AudioDecoderCng();
-  ~AudioDecoderCng() override;
-  void Reset() override;
-  int IncomingPacket(const uint8_t* payload,
-                     size_t payload_len,
-                     uint16_t rtp_sequence_number,
-                     uint32_t rtp_timestamp,
-                     uint32_t arrival_timestamp) override;
-
-  CNG_dec_inst* CngDecoderInstance() override;
-  size_t Channels() const override;
-
- protected:
-  int DecodeInternal(const uint8_t* encoded,
-                     size_t encoded_len,
-                     int sample_rate_hz,
-                     int16_t* decoded,
-                     SpeechType* speech_type) override;
-
- private:
-  CNG_dec_inst* dec_state_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderCng);
-};
-
 using NetEqDecoder = acm2::RentACodec::NetEqDecoder;
 
 // Returns true if |codec_type| is supported.
@@ -65,10 +32,5 @@ bool CodecSupported(NetEqDecoder codec_type);
 // Returns the sample rate for |codec_type|.
 int CodecSampleRateHz(NetEqDecoder codec_type);
 
-// Creates an AudioDecoder object of type |codec_type|. Returns NULL for for
-// unsupported codecs, and when creating an AudioDecoder is not applicable
-// (e.g., for RED and DTMF/AVT types).
-AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type);
-
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
index bd38c43903a..c80909d7b58 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
@@ -106,7 +106,7 @@ void AudioMultiVector::PushBackFromIndex(const AudioMultiVector& append_this,
   assert(num_channels_ == append_this.num_channels_);
   if (num_channels_ == append_this.num_channels_) {
     for (size_t i = 0; i < num_channels_; ++i) {
-      channels_[i]->PushBack(&append_this[i][index], length);
+      channels_[i]->PushBack(append_this[i], length, index);
     }
   }
 }
@@ -133,14 +133,14 @@ size_t AudioMultiVector::ReadInterleavedFromIndex(size_t start_index,
                                                   int16_t* destination) const {
   RTC_DCHECK(destination);
   size_t index = 0;  // Number of elements written to |destination| so far.
-  assert(start_index <= Size());
+  RTC_DCHECK_LE(start_index, Size());
   start_index = std::min(start_index, Size());
   if (length + start_index > Size()) {
     length = Size() - start_index;
   }
   if (num_channels_ == 1) {
     // Special case to avoid the nested for loop below.
-    memcpy(destination, &(*this)[0][start_index], length * sizeof(int16_t));
+    (*this)[0].CopyTo(length, start_index, destination);
     return length;
   }
   for (size_t i = 0; i < length; ++i) {
@@ -167,7 +167,7 @@ void AudioMultiVector::OverwriteAt(const AudioMultiVector& insert_this,
   length = std::min(length, insert_this.Size());
   if (num_channels_ == insert_this.num_channels_) {
     for (size_t i = 0; i < num_channels_; ++i) {
-      channels_[i]->OverwriteAt(&insert_this[i][0], length, position);
+      channels_[i]->OverwriteAt(insert_this[i], length, position);
     }
   }
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.cc
index 013e1d89ad9..ea737a55424 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.cc
@@ -15,124 +15,236 @@
 #include <algorithm>
 #include <memory>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 AudioVector::AudioVector()
-    : array_(new int16_t[kDefaultInitialSize]),
-      first_free_ix_(0),
-      capacity_(kDefaultInitialSize) {
+    : AudioVector(kDefaultInitialSize) {
+  Clear();
 }
 
 AudioVector::AudioVector(size_t initial_size)
-    : array_(new int16_t[initial_size]),
-      first_free_ix_(initial_size),
-      capacity_(initial_size) {
-  memset(array_.get(), 0, initial_size * sizeof(int16_t));
+    : array_(new int16_t[initial_size + 1]),
+      capacity_(initial_size + 1),
+      begin_index_(0),
+      end_index_(capacity_ - 1) {
+  memset(array_.get(), 0, capacity_ * sizeof(int16_t));
 }
 
 AudioVector::~AudioVector() = default;
 
 void AudioVector::Clear() {
-  first_free_ix_ = 0;
+  end_index_ = begin_index_ = 0;
 }
 
 void AudioVector::CopyTo(AudioVector* copy_to) const {
-  if (copy_to) {
-    copy_to->Reserve(Size());
-    assert(copy_to->capacity_ >= Size());
-    memcpy(copy_to->array_.get(), array_.get(), Size() * sizeof(int16_t));
-    copy_to->first_free_ix_ = first_free_ix_;
+  RTC_DCHECK(copy_to);
+  copy_to->Reserve(Size());
+  CopyTo(Size(), 0, copy_to->array_.get());
+  copy_to->begin_index_ = 0;
+  copy_to->end_index_ = Size();
+}
+
+void AudioVector::CopyTo(
+    size_t length, size_t position, int16_t* copy_to) const {
+  if (length == 0)
+    return;
+  length = std::min(length, Size() - position);
+  const size_t copy_index = (begin_index_ + position) % capacity_;
+  const size_t first_chunk_length =
+      std::min(length, capacity_ - copy_index);
+  memcpy(copy_to, &array_[copy_index],
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(&copy_to[first_chunk_length], array_.get(),
+           remaining_length * sizeof(int16_t));
   }
 }
 
 void AudioVector::PushFront(const AudioVector& prepend_this) {
-  size_t insert_length = prepend_this.Size();
-  Reserve(Size() + insert_length);
-  memmove(&array_[insert_length], &array_[0], Size() * sizeof(int16_t));
-  memcpy(&array_[0], &prepend_this.array_[0], insert_length * sizeof(int16_t));
-  first_free_ix_ += insert_length;
+  const size_t length = prepend_this.Size();
+  if (length == 0)
+    return;
+
+  // Although the subsequent calling to PushFront does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  Reserve(Size() + length);
+
+  const size_t first_chunk_length =
+      std::min(length, prepend_this.capacity_ - prepend_this.begin_index_);
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0)
+    PushFront(prepend_this.array_.get(), remaining_length);
+  PushFront(&prepend_this.array_[prepend_this.begin_index_],
+            first_chunk_length);
 }
 
 void AudioVector::PushFront(const int16_t* prepend_this, size_t length) {
-  // Same operation as InsertAt beginning.
-  InsertAt(prepend_this, length, 0);
+  if (length == 0)
+    return;
+  Reserve(Size() + length);
+  const size_t first_chunk_length = std::min(length, begin_index_);
+  memcpy(&array_[begin_index_ - first_chunk_length],
+         &prepend_this[length - first_chunk_length],
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(&array_[capacity_ - remaining_length], prepend_this,
+           remaining_length * sizeof(int16_t));
+  }
+  begin_index_ = (begin_index_ + capacity_ - length) % capacity_;
 }
 
 void AudioVector::PushBack(const AudioVector& append_this) {
-  PushBack(append_this.array_.get(), append_this.Size());
+  PushBack(append_this, append_this.Size(), 0);
+}
+
+void AudioVector::PushBack(
+    const AudioVector& append_this, size_t length, size_t position) {
+  RTC_DCHECK_LE(position, append_this.Size());
+  RTC_DCHECK_LE(length, append_this.Size() - position);
+
+  if (length == 0)
+    return;
+
+  // Although the subsequent calling to PushBack does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  Reserve(Size() + length);
+
+  const size_t start_index =
+      (append_this.begin_index_ + position) % append_this.capacity_;
+  const size_t first_chunk_length = std::min(
+      length, append_this.capacity_ - start_index);
+  PushBack(&append_this.array_[start_index], first_chunk_length);
+
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0)
+    PushBack(append_this.array_.get(), remaining_length);
 }
 
 void AudioVector::PushBack(const int16_t* append_this, size_t length) {
+  if (length == 0)
+    return;
   Reserve(Size() + length);
-  memcpy(&array_[first_free_ix_], append_this, length * sizeof(int16_t));
-  first_free_ix_ += length;
+  const size_t first_chunk_length = std::min(length, capacity_ - end_index_);
+  memcpy(&array_[end_index_], append_this,
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(array_.get(), &append_this[first_chunk_length],
+           remaining_length * sizeof(int16_t));
+  }
+  end_index_ = (end_index_ + length) % capacity_;
 }
 
 void AudioVector::PopFront(size_t length) {
-  if (length >= Size()) {
-    // Remove all elements.
-    Clear();
-  } else {
-    size_t remaining_samples = Size() - length;
-    memmove(&array_[0], &array_[length], remaining_samples * sizeof(int16_t));
-    first_free_ix_ -= length;
-  }
+  if (length == 0)
+    return;
+  length = std::min(length, Size());
+  begin_index_ = (begin_index_ + length) % capacity_;
 }
 
 void AudioVector::PopBack(size_t length) {
+  if (length == 0)
+    return;
   // Never remove more than what is in the array.
   length = std::min(length, Size());
-  first_free_ix_ -= length;
+  end_index_ = (end_index_ + capacity_ - length) % capacity_;
 }
 
 void AudioVector::Extend(size_t extra_length) {
-  Reserve(Size() + extra_length);
-  memset(&array_[first_free_ix_], 0, extra_length * sizeof(int16_t));
-  first_free_ix_ += extra_length;
+  if (extra_length == 0)
+    return;
+  InsertZerosByPushBack(extra_length, Size());
 }
 
 void AudioVector::InsertAt(const int16_t* insert_this,
                            size_t length,
                            size_t position) {
-  Reserve(Size() + length);
-  // Cap the position at the current vector length, to be sure the iterator
-  // does not extend beyond the end of the vector.
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
   position = std::min(Size(), position);
-  int16_t* insert_position_ptr = &array_[position];
-  size_t samples_to_move = Size() - position;
-  memmove(insert_position_ptr + length, insert_position_ptr,
-          samples_to_move * sizeof(int16_t));
-  memcpy(insert_position_ptr, insert_this, length * sizeof(int16_t));
-  first_free_ix_ += length;
+
+  // When inserting to a position closer to the beginning, it is more efficient
+  // to insert by pushing front than to insert by pushing back, since less data
+  // will be moved, vice versa.
+  if (position <= Size() - position) {
+    InsertByPushFront(insert_this, length, position);
+  } else {
+    InsertByPushBack(insert_this, length, position);
+  }
 }
 
 void AudioVector::InsertZerosAt(size_t length,
                                 size_t position) {
-  Reserve(Size() + length);
-  // Cap the position at the current vector length, to be sure the iterator
-  // does not extend beyond the end of the vector.
-  position = std::min(capacity_, position);
-  int16_t* insert_position_ptr = &array_[position];
-  size_t samples_to_move = Size() - position;
-  memmove(insert_position_ptr + length, insert_position_ptr,
-          samples_to_move * sizeof(int16_t));
-  memset(insert_position_ptr, 0, length * sizeof(int16_t));
-  first_free_ix_ += length;
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // When inserting to a position closer to the beginning, it is more efficient
+  // to insert by pushing front than to insert by pushing back, since less data
+  // will be moved, vice versa.
+  if (position <= Size() - position) {
+    InsertZerosByPushFront(length, position);
+  } else {
+    InsertZerosByPushBack(length, position);
+  }
+}
+
+void AudioVector::OverwriteAt(const AudioVector& insert_this,
+                              size_t length,
+                              size_t position) {
+  RTC_DCHECK_LE(length, insert_this.Size());
+  if (length == 0)
+    return;
+
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // Although the subsequent calling to OverwriteAt does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  size_t new_size = std::max(Size(), position + length);
+  Reserve(new_size);
+
+  const size_t first_chunk_length =
+      std::min(length, insert_this.capacity_ - insert_this.begin_index_);
+  OverwriteAt(&insert_this.array_[insert_this.begin_index_], first_chunk_length,
+              position);
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    OverwriteAt(insert_this.array_.get(), remaining_length,
+                position + first_chunk_length);
+  }
 }
 
 void AudioVector::OverwriteAt(const int16_t* insert_this,
                               size_t length,
                               size_t position) {
+  if (length == 0)
+    return;
   // Cap the insert position at the current array length.
   position = std::min(Size(), position);
-  Reserve(position + length);
-  memcpy(&array_[position], insert_this, length * sizeof(int16_t));
-  if (position + length > Size()) {
-    // Array was expanded.
-    first_free_ix_ += position + length - Size();
+
+  size_t new_size = std::max(Size(), position + length);
+  Reserve(new_size);
+
+  const size_t overwrite_index = (begin_index_ + position) % capacity_;
+  const size_t first_chunk_length =
+      std::min(length, capacity_ - overwrite_index);
+  memcpy(&array_[overwrite_index], insert_this,
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(array_.get(), &insert_this[first_chunk_length],
+           remaining_length * sizeof(int16_t));
   }
+
+  end_index_ = (begin_index_ + new_size) % capacity_;
 }
 
 void AudioVector::CrossFade(const AudioVector& append_this,
@@ -142,7 +254,7 @@ void AudioVector::CrossFade(const AudioVector& append_this,
   assert(fade_length <= append_this.Size());
   fade_length = std::min(fade_length, Size());
   fade_length = std::min(fade_length, append_this.Size());
-  size_t position = Size() - fade_length;
+  size_t position = Size() - fade_length + begin_index_;
   // Cross fade the overlapping regions.
   // |alpha| is the mixing factor in Q14.
   // TODO(hlundin): Consider skipping +1 in the denominator to produce a
@@ -151,41 +263,132 @@ void AudioVector::CrossFade(const AudioVector& append_this,
   int alpha = 16384;
   for (size_t i = 0; i < fade_length; ++i) {
     alpha -= alpha_step;
-    array_[position + i] = (alpha * array_[position + i] +
-        (16384 - alpha) * append_this[i] + 8192) >> 14;
+    array_[(position + i) % capacity_] =
+        (alpha * array_[(position + i) % capacity_] +
+            (16384 - alpha) * append_this[i] + 8192) >> 14;
   }
   assert(alpha >= 0);  // Verify that the slope was correct.
   // Append what is left of |append_this|.
   size_t samples_to_push_back = append_this.Size() - fade_length;
   if (samples_to_push_back > 0)
-    PushBack(&append_this[fade_length], samples_to_push_back);
+    PushBack(append_this, samples_to_push_back, fade_length);
 }
 
 // Returns the number of elements in this AudioVector.
 size_t AudioVector::Size() const {
-  return first_free_ix_;
+  return (end_index_ + capacity_ - begin_index_) % capacity_;
 }
 
 // Returns true if this AudioVector is empty.
 bool AudioVector::Empty() const {
-  return first_free_ix_ == 0;
+  return begin_index_ == end_index_;
 }
 
 const int16_t& AudioVector::operator[](size_t index) const {
-  return array_[index];
+  return array_[(begin_index_ + index) % capacity_];
 }
 
 int16_t& AudioVector::operator[](size_t index) {
-  return array_[index];
+  return array_[(begin_index_ + index) % capacity_];
 }
 
 void AudioVector::Reserve(size_t n) {
-  if (capacity_ < n) {
-    std::unique_ptr<int16_t[]> temp_array(new int16_t[n]);
-    memcpy(temp_array.get(), array_.get(), Size() * sizeof(int16_t));
-    array_.swap(temp_array);
-    capacity_ = n;
+  if (capacity_ > n)
+    return;
+  const size_t length = Size();
+  // Reserve one more sample to remove the ambiguity between empty vector and
+  // full vector. Therefore |begin_index_| == |end_index_| indicates empty
+  // vector, and |begin_index_| == (|end_index_| + 1) % capacity indicates
+  // full vector.
+  std::unique_ptr<int16_t[]> temp_array(new int16_t[n + 1]);
+  CopyTo(length, 0, temp_array.get());
+  array_.swap(temp_array);
+  begin_index_ = 0;
+  end_index_ = length;
+  capacity_ = n + 1;
+}
+
+void AudioVector::InsertByPushBack(const int16_t* insert_this,
+                                   size_t length,
+                                   size_t position) {
+  const size_t move_chunk_length = Size() - position;
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (move_chunk_length > 0) {
+    // TODO(minyue): see if it is possible to avoid copying to a buffer.
+    temp_array.reset(new int16_t[move_chunk_length]);
+    CopyTo(move_chunk_length, position, temp_array.get());
+    PopBack(move_chunk_length);
+  }
+
+  Reserve(Size() + length + move_chunk_length);
+  PushBack(insert_this, length);
+  if (move_chunk_length > 0)
+    PushBack(temp_array.get(), move_chunk_length);
+}
+
+void AudioVector::InsertByPushFront(const int16_t* insert_this,
+                                   size_t length,
+                                   size_t position) {
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (position > 0) {
+    // TODO(minyue): see if it is possible to avoid copying to a buffer.
+    temp_array.reset(new int16_t[position]);
+    CopyTo(position, 0, temp_array.get());
+    PopFront(position);
+  }
+
+  Reserve(Size() + length + position);
+  PushFront(insert_this, length);
+  if (position > 0)
+    PushFront(temp_array.get(), position);
+}
+
+void AudioVector::InsertZerosByPushBack(size_t length,
+                                        size_t position) {
+  const size_t move_chunk_length = Size() - position;
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (move_chunk_length > 0) {
+    temp_array.reset(new int16_t[move_chunk_length]);
+    CopyTo(move_chunk_length, position, temp_array.get());
+    PopBack(move_chunk_length);
   }
+
+  Reserve(Size() + length + move_chunk_length);
+
+  const size_t first_zero_chunk_length =
+      std::min(length, capacity_ - end_index_);
+  memset(&array_[end_index_], 0, first_zero_chunk_length * sizeof(int16_t));
+  const size_t remaining_zero_length = length - first_zero_chunk_length;
+  if (remaining_zero_length > 0)
+    memset(array_.get(), 0, remaining_zero_length * sizeof(int16_t));
+  end_index_ = (end_index_ + length) % capacity_;
+
+  if (move_chunk_length > 0)
+    PushBack(temp_array.get(), move_chunk_length);
+}
+
+void AudioVector::InsertZerosByPushFront(size_t length,
+                                         size_t position) {
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (position > 0) {
+    temp_array.reset(new int16_t[position]);
+    CopyTo(position, 0, temp_array.get());
+    PopFront(position);
+  }
+
+  Reserve(Size() + length + position);
+
+  const size_t first_zero_chunk_length = std::min(length, begin_index_);
+  memset(&array_[begin_index_ - first_zero_chunk_length], 0,
+         first_zero_chunk_length * sizeof(int16_t));
+  const size_t remaining_zero_length = length - first_zero_chunk_length;
+  if (remaining_zero_length > 0)
+    memset(&array_[capacity_ - remaining_zero_length], 0,
+           remaining_zero_length * sizeof(int16_t));
+  begin_index_ = (begin_index_ + capacity_ - length) % capacity_;
+
+  if (position > 0)
+    PushFront(temp_array.get(), position);
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.h
index 15297f9bc8c..756292aa783 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector.h
@@ -37,6 +37,9 @@ class AudioVector {
   // |copy_to| will be an exact replica of this object.
   virtual void CopyTo(AudioVector* copy_to) const;
 
+  // Copies |length| values from |position| in this vector to |copy_to|.
+  virtual void CopyTo(size_t length, size_t position, int16_t* copy_to) const;
+
   // Prepends the contents of AudioVector |prepend_this| to this object. The
   // length of this object is increased with the length of |prepend_this|.
   virtual void PushFront(const AudioVector& prepend_this);
@@ -48,6 +51,12 @@ class AudioVector {
   // Same as PushFront but will append to the end of this object.
   virtual void PushBack(const AudioVector& append_this);
 
+  // Appends a segment of |append_this| to the end of this object. The segment
+  // starts from |position| and has |length| samples.
+  virtual void PushBack(const AudioVector& append_this,
+                        size_t length,
+                        size_t position);
+
   // Same as PushFront but will append to the end of this object.
   virtual void PushBack(const int16_t* append_this, size_t length);
 
@@ -71,6 +80,15 @@ class AudioVector {
   // Like InsertAt, but inserts |length| zero elements at |position|.
   virtual void InsertZerosAt(size_t length, size_t position);
 
+  // Overwrites |length| elements of this AudioVector starting from |position|
+  // with first values in |AudioVector|. The definition of |position|
+  // is the same as for InsertAt(). If |length| and |position| are selected
+  // such that the new data extends beyond the end of the current AudioVector,
+  // the vector is extended to accommodate the new data.
+  virtual void OverwriteAt(const AudioVector& insert_this,
+                           size_t length,
+                           size_t position);
+
   // Overwrites |length| elements of this AudioVector with values taken from the
   // array |insert_this|, starting at |position|. The definition of |position|
   // is the same as for InsertAt(). If |length| and |position| are selected
@@ -100,11 +118,27 @@ class AudioVector {
 
   void Reserve(size_t n);
 
+  void InsertByPushBack(const int16_t* insert_this, size_t length,
+                        size_t position);
+
+  void InsertByPushFront(const int16_t* insert_this, size_t length,
+                         size_t position);
+
+  void InsertZerosByPushBack(size_t length, size_t position);
+
+  void InsertZerosByPushFront(size_t length, size_t position);
+
   std::unique_ptr<int16_t[]> array_;
-  size_t first_free_ix_;  // The first index after the last sample in array_.
-                          // Note that this index may point outside of array_.
+
   size_t capacity_;  // Allocated number of samples in the array.
 
+  // The index of the first sample in |array_|, except when
+  // |begin_index_ == end_index_|, which indicates an empty buffer.
+  size_t begin_index_;
+
+  // The index of the sample after the last sample in |array_|.
+  size_t end_index_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioVector);
 };
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
index 08009863455..cee7e586695 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
@@ -82,14 +82,6 @@ TEST_F(AudioVectorTest, PushBackAndCopy) {
   EXPECT_TRUE(vec_copy.Empty());
 }
 
-// Try to copy to a NULL pointer. Nothing should happen.
-TEST_F(AudioVectorTest, CopyToNull) {
-  AudioVector vec;
-  AudioVector* vec_copy = NULL;
-  vec.PushBack(array_, array_length());
-  vec.CopyTo(vec_copy);
-}
-
 // Test the PushBack method with another AudioVector as input argument.
 TEST_F(AudioVectorTest, PushBackVector) {
   static const size_t kLength = 10;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.cc
index 7e7a6325e97..9cfd6cb40ed 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/background_noise.cc
@@ -17,6 +17,7 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
 #include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
 
 namespace webrtc {
@@ -58,10 +59,7 @@ void BackgroundNoise::Update(const AudioMultiVector& input,
     ChannelParameters& parameters = channel_parameters_[channel_ix];
     int16_t temp_signal_array[kVecLen + kMaxLpcOrder] = {0};
     int16_t* temp_signal = &temp_signal_array[kMaxLpcOrder];
-    memcpy(temp_signal,
-           &input[channel_ix][input.Size() - kVecLen],
-           sizeof(int16_t) * kVecLen);
-
+    input[channel_ix].CopyTo(kVecLen, input.Size() - kVecLen, temp_signal);
     int32_t sample_energy = CalculateAutoCorrelation(temp_signal, kVecLen,
                                                      auto_correlation);
 
@@ -169,15 +167,10 @@ int16_t BackgroundNoise::ScaleShift(size_t channel) const {
 
 int32_t BackgroundNoise::CalculateAutoCorrelation(
     const int16_t* signal, size_t length, int32_t* auto_correlation) const {
-  int16_t signal_max = WebRtcSpl_MaxAbsValueW16(signal, length);
-  int correlation_scale = kLogVecLen -
-      WebRtcSpl_NormW32(signal_max * signal_max);
-  correlation_scale = std::max(0, correlation_scale);
-
   static const int kCorrelationStep = -1;
-  WebRtcSpl_CrossCorrelation(auto_correlation, signal, signal, length,
-                             kMaxLpcOrder + 1, correlation_scale,
-                             kCorrelationStep);
+  const int correlation_scale =
+      CrossCorrelationWithAutoShift(signal, signal, length, kMaxLpcOrder + 1,
+                                    kCorrelationStep, auto_correlation);
 
   // Number of shifts to normalize energy to energy/sample.
   int energy_sample_shift = kLogVecLen - correlation_scale;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.cc
index a5b08469bea..90b02daf712 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.cc
@@ -14,7 +14,6 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
@@ -23,31 +22,23 @@ namespace webrtc {
 
 void ComfortNoise::Reset() {
   first_call_ = true;
-  internal_error_code_ = 0;
 }
 
 int ComfortNoise::UpdateParameters(Packet* packet) {
   assert(packet);  // Existence is verified by caller.
   // Get comfort noise decoder.
-  AudioDecoder* cng_decoder = decoder_database_->GetDecoder(
-      packet->header.payloadType);
-  if (!cng_decoder) {
+  if (decoder_database_->SetActiveCngDecoder(packet->header.payloadType)
+      != kOK) {
     delete [] packet->payload;
     delete packet;
     return kUnknownPayloadType;
   }
-  decoder_database_->SetActiveCngDecoder(packet->header.payloadType);
-  CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
-  int16_t ret = WebRtcCng_UpdateSid(cng_inst,
-                                    packet->payload,
-                                    packet->payload_length);
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  RTC_DCHECK(cng_decoder);
+  cng_decoder->UpdateSid(rtc::ArrayView<const uint8_t>(
+      packet->payload, packet->payload_length));
   delete [] packet->payload;
   delete packet;
-  if (ret < 0) {
-    internal_error_code_ = WebRtcCng_GetErrorCodeDec(cng_inst);
-    LOG(LS_ERROR) << "WebRtcCng_UpdateSid produced " << internal_error_code_;
-    return kInternalError;
-  }
   return kOK;
 }
 
@@ -63,30 +54,31 @@ int ComfortNoise::Generate(size_t requested_length,
   }
 
   size_t number_of_samples = requested_length;
-  int16_t new_period = 0;
+  bool new_period = false;
   if (first_call_) {
     // Generate noise and overlap slightly with old data.
     number_of_samples = requested_length + overlap_length_;
-    new_period = 1;
+    new_period = true;
   }
   output->AssertSize(number_of_samples);
   // Get the decoder from the database.
-  AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
   if (!cng_decoder) {
     LOG(LS_ERROR) << "Unknwown payload type";
     return kUnknownPayloadType;
   }
-  CNG_dec_inst* cng_inst = cng_decoder->CngDecoderInstance();
-  // The expression &(*output)[0][0] is a pointer to the first element in
-  // the first channel.
-  if (WebRtcCng_Generate(cng_inst, &(*output)[0][0], number_of_samples,
-                         new_period) < 0) {
+
+  std::unique_ptr<int16_t[]> temp(new int16_t[number_of_samples]);
+  if (!cng_decoder->Generate(
+          rtc::ArrayView<int16_t>(temp.get(), number_of_samples),
+          new_period)) {
     // Error returned.
     output->Zeros(requested_length);
-    internal_error_code_ = WebRtcCng_GetErrorCodeDec(cng_inst);
-    LOG(LS_ERROR) << "WebRtcCng_Generate produced " << internal_error_code_;
+    LOG(LS_ERROR) <<
+        "ComfortNoiseDecoder::Genererate failed to generate comfort noise";
     return kInternalError;
   }
+  (*output)[0].OverwriteAt(temp.get(), number_of_samples, 0);
 
   if (first_call_) {
     // Set tapering window parameters. Values are in Q15.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.h
index 1fc22586637..f877bf63efb 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/comfort_noise.h
@@ -38,8 +38,7 @@ class ComfortNoise {
         first_call_(true),
         overlap_length_(5 * fs_hz_ / 8000),
         decoder_database_(decoder_database),
-        sync_buffer_(sync_buffer),
-        internal_error_code_(0) {
+        sync_buffer_(sync_buffer) {
   }
 
   // Resets the state. Should be called before each new comfort noise period.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/cross_correlation.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/cross_correlation.cc
new file mode 100644
index 00000000000..ad89ab8a139
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/cross_correlation.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
+
+#include <cstdlib>
+#include <limits>
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+// This function decides the overflow-protecting scaling and calls
+// WebRtcSpl_CrossCorrelation.
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+                                  const int16_t* sequence_2,
+                                  size_t sequence_1_length,
+                                  size_t cross_correlation_length,
+                                  int cross_correlation_step,
+                                  int32_t* cross_correlation) {
+  // Find the maximum absolute value of sequence_1 and 2.
+  const int16_t max_1 = WebRtcSpl_MaxAbsValueW16(sequence_1, sequence_1_length);
+  const int sequence_2_shift =
+      cross_correlation_step * (static_cast<int>(cross_correlation_length) - 1);
+  const int16_t* sequence_2_start =
+      sequence_2_shift >= 0 ? sequence_2 : sequence_2 + sequence_2_shift;
+  const size_t sequence_2_length =
+      sequence_1_length + std::abs(sequence_2_shift);
+  const int16_t max_2 =
+      WebRtcSpl_MaxAbsValueW16(sequence_2_start, sequence_2_length);
+
+  // In order to avoid overflow when computing the sum we should scale the
+  // samples so that (in_vector_length * max_1 * max_2) will not overflow.
+  // Expected scaling fulfills
+  // 1) sufficient:
+  //    sequence_1_length * (max_1 * max_2 >> scaling) <= 0x7fffffff;
+  // 2) necessary:
+  //    if (scaling > 0)
+  //      sequence_1_length * (max_1 * max_2 >> (scaling - 1)) > 0x7fffffff;
+  // The following calculation fulfills 1) and almost fulfills 2).
+  // There are some corner cases that 2) is not satisfied, e.g.,
+  // max_1 = 17, max_2 = 30848, sequence_1_length = 4095, in such case,
+  // optimal scaling is 0, while the following calculation results in 1.
+  const int32_t factor = (max_1 * max_2) / (std::numeric_limits<int32_t>::max()
+      / static_cast<int32_t>(sequence_1_length));
+  const int scaling = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+
+  WebRtcSpl_CrossCorrelation(cross_correlation, sequence_1, sequence_2,
+                             sequence_1_length, cross_correlation_length,
+                             scaling, cross_correlation_step);
+
+  return scaling;
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/cross_correlation.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/cross_correlation.h
new file mode 100644
index 00000000000..db14141027c
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/cross_correlation.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+// The function calculates the cross-correlation between two sequences
+// |sequence_1| and |sequence_2|. |sequence_1| is taken as reference, with
+// |sequence_1_length| as its length. |sequence_2| slides for the calculation of
+// cross-correlation. The result will be saved in |cross_correlation|.
+// |cross_correlation_length| correlation points are calculated.
+// The corresponding lag starts from 0, and increases with a step of
+// |cross_correlation_step|. The result is without normalization. To avoid
+// overflow, the result will be right shifted. The amount of shifts will be
+// returned.
+//
+// Input:
+//     - sequence_1     : First sequence (reference).
+//     - sequence_2     : Second sequence (sliding during calculation).
+//     - sequence_1_length : Length of |sequence_1|.
+//     - cross_correlation_length : Number of cross-correlations to calculate.
+//     - cross_correlation_step : Step in the lag for the cross-correlation.
+//
+// Output:
+//      - cross_correlation : The cross-correlation in Q(-right_shifts)
+//
+// Return:
+//      Number of right shifts in cross_correlation.
+
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+                                  const int16_t* sequence_2,
+                                  size_t sequence_1_length,
+                                  size_t cross_correlation_length,
+                                  int cross_correlation_step,
+                                  int32_t* cross_correlation);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.cc
index 39bb4662c71..545d1d62455 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -29,26 +29,19 @@ DecisionLogic* DecisionLogic::Create(int fs_hz,
                                      DecoderDatabase* decoder_database,
                                      const PacketBuffer& packet_buffer,
                                      DelayManager* delay_manager,
-                                     BufferLevelFilter* buffer_level_filter) {
+                                     BufferLevelFilter* buffer_level_filter,
+                                     const TickTimer* tick_timer) {
   switch (playout_mode) {
     case kPlayoutOn:
     case kPlayoutStreaming:
-      return new DecisionLogicNormal(fs_hz,
-                                     output_size_samples,
-                                     playout_mode,
-                                     decoder_database,
-                                     packet_buffer,
-                                     delay_manager,
-                                     buffer_level_filter);
+      return new DecisionLogicNormal(
+          fs_hz, output_size_samples, playout_mode, decoder_database,
+          packet_buffer, delay_manager, buffer_level_filter, tick_timer);
     case kPlayoutFax:
     case kPlayoutOff:
-      return new DecisionLogicFax(fs_hz,
-                                  output_size_samples,
-                                  playout_mode,
-                                  decoder_database,
-                                  packet_buffer,
-                                  delay_manager,
-                                  buffer_level_filter);
+      return new DecisionLogicFax(
+          fs_hz, output_size_samples, playout_mode, decoder_database,
+          packet_buffer, delay_manager, buffer_level_filter, tick_timer);
   }
   // This line cannot be reached, but must be here to avoid compiler errors.
   assert(false);
@@ -61,30 +54,34 @@ DecisionLogic::DecisionLogic(int fs_hz,
                              DecoderDatabase* decoder_database,
                              const PacketBuffer& packet_buffer,
                              DelayManager* delay_manager,
-                             BufferLevelFilter* buffer_level_filter)
+                             BufferLevelFilter* buffer_level_filter,
+                             const TickTimer* tick_timer)
     : decoder_database_(decoder_database),
       packet_buffer_(packet_buffer),
       delay_manager_(delay_manager),
       buffer_level_filter_(buffer_level_filter),
+      tick_timer_(tick_timer),
       cng_state_(kCngOff),
-      generated_noise_samples_(0),
       packet_length_samples_(0),
       sample_memory_(0),
       prev_time_scale_(false),
-      timescale_hold_off_(kMinTimescaleInterval),
+      timescale_countdown_(
+          tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)),
       num_consecutive_expands_(0),
       playout_mode_(playout_mode) {
   delay_manager_->set_streaming_mode(playout_mode_ == kPlayoutStreaming);
   SetSampleRate(fs_hz, output_size_samples);
 }
 
+DecisionLogic::~DecisionLogic() = default;
+
 void DecisionLogic::Reset() {
   cng_state_ = kCngOff;
-  generated_noise_samples_ = 0;
+  noise_fast_forward_ = 0;
   packet_length_samples_ = 0;
   sample_memory_ = 0;
   prev_time_scale_ = false;
-  timescale_hold_off_ = 0;
+  timescale_countdown_.reset();
   num_consecutive_expands_ = 0;
 }
 
@@ -92,7 +89,8 @@ void DecisionLogic::SoftReset() {
   packet_length_samples_ = 0;
   sample_memory_ = 0;
   prev_time_scale_ = false;
-  timescale_hold_off_ = kMinTimescaleInterval;
+  timescale_countdown_ =
+      tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1);
 }
 
 void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
@@ -107,15 +105,15 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
                                       size_t decoder_frame_length,
                                       const RTPHeader* packet_header,
                                       Modes prev_mode,
-                                      bool play_dtmf, bool* reset_decoder) {
+                                      bool play_dtmf,
+                                      size_t generated_noise_samples,
+                                      bool* reset_decoder) {
   if (prev_mode == kModeRfc3389Cng ||
       prev_mode == kModeCodecInternalCng ||
       prev_mode == kModeExpand) {
     // If last mode was CNG (or Expand, since this could be covering up for
-    // a lost CNG packet), increase the |generated_noise_samples_| counter.
-    generated_noise_samples_ += output_size_samples_;
-    // Remember that CNG is on. This is needed if comfort noise is interrupted
-    // by DTMF.
+    // a lost CNG packet), remember that CNG is on. This is needed if comfort
+    // noise is interrupted by DTMF.
     if (prev_mode == kModeRfc3389Cng) {
       cng_state_ = kCngRfc3389On;
     } else if (prev_mode == kModeCodecInternalCng) {
@@ -139,7 +137,7 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
 
   return GetDecisionSpecialized(sync_buffer, expand, decoder_frame_length,
                                 packet_header, prev_mode, play_dtmf,
-                                reset_decoder);
+                                reset_decoder, generated_noise_samples);
 }
 
 void DecisionLogic::ExpandDecision(Operations operation) {
@@ -152,10 +150,6 @@ void DecisionLogic::ExpandDecision(Operations operation) {
 
 void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
                                       Modes prev_mode) {
-  const int elapsed_time_ms =
-      static_cast<int>(output_size_samples_ / (8 * fs_mult_));
-  delay_manager_->UpdateCounters(elapsed_time_ms);
-
   // Do not update buffer history if currently playing CNG since it will bias
   // the filtered buffer level.
   if ((prev_mode != kModeRfc3389Cng) && (prev_mode != kModeCodecInternalCng)) {
@@ -170,14 +164,13 @@ void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
     int sample_memory_local = 0;
     if (prev_time_scale_) {
       sample_memory_local = sample_memory_;
-      timescale_hold_off_ = kMinTimescaleInterval;
+      timescale_countdown_ =
+          tick_timer_->GetNewCountdown(kMinTimescaleInterval);
     }
     buffer_level_filter_->Update(buffer_size_packets, sample_memory_local,
                                  packet_length_samples_);
     prev_time_scale_ = false;
   }
-
-  timescale_hold_off_ = std::max(timescale_hold_off_ - 1, 0);
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.h
index 72121b7aac5..008655d1a2b 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic.h
@@ -14,6 +14,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -39,7 +40,8 @@ class DecisionLogic {
                                DecoderDatabase* decoder_database,
                                const PacketBuffer& packet_buffer,
                                DelayManager* delay_manager,
-                               BufferLevelFilter* buffer_level_filter);
+                               BufferLevelFilter* buffer_level_filter,
+                               const TickTimer* tick_timer);
 
   // Constructor.
   DecisionLogic(int fs_hz,
@@ -48,10 +50,10 @@ class DecisionLogic {
                 DecoderDatabase* decoder_database,
                 const PacketBuffer& packet_buffer,
                 DelayManager* delay_manager,
-                BufferLevelFilter* buffer_level_filter);
+                BufferLevelFilter* buffer_level_filter,
+                const TickTimer* tick_timer);
 
-  // Destructor.
-  virtual ~DecisionLogic() {}
+  virtual ~DecisionLogic();
 
   // Resets object to a clean state.
   void Reset();
@@ -79,6 +81,7 @@ class DecisionLogic {
                          const RTPHeader* packet_header,
                          Modes prev_mode,
                          bool play_dtmf,
+                         size_t generated_noise_samples,
                          bool* reset_decoder);
 
   // These methods test the |cng_state_| for different conditions.
@@ -101,10 +104,7 @@ class DecisionLogic {
 
   // Accessors and mutators.
   void set_sample_memory(int32_t value) { sample_memory_ = value; }
-  size_t generated_noise_samples() const { return generated_noise_samples_; }
-  void set_generated_noise_samples(size_t value) {
-    generated_noise_samples_ = value;
-  }
+  size_t noise_fast_forward() const { return noise_fast_forward_; }
   size_t packet_length_samples() const { return packet_length_samples_; }
   void set_packet_length_samples(size_t value) {
     packet_length_samples_ = value;
@@ -113,8 +113,8 @@ class DecisionLogic {
   NetEqPlayoutMode playout_mode() const { return playout_mode_; }
 
  protected:
-  // The value 6 sets maximum time-stretch rate to about 100 ms/s.
-  static const int kMinTimescaleInterval = 6;
+  // The value 5 sets maximum time-stretch rate to about 100 ms/s.
+  static const int kMinTimescaleInterval = 5;
 
   enum CngState {
     kCngOff,
@@ -138,7 +138,8 @@ class DecisionLogic {
                                             const RTPHeader* packet_header,
                                             Modes prev_mode,
                                             bool play_dtmf,
-                                            bool* reset_decoder) = 0;
+                                            bool* reset_decoder,
+                                            size_t generated_noise_samples) = 0;
 
   // Updates the |buffer_level_filter_| with the current buffer level
   // |buffer_size_packets|.
@@ -148,15 +149,16 @@ class DecisionLogic {
   const PacketBuffer& packet_buffer_;
   DelayManager* delay_manager_;
   BufferLevelFilter* buffer_level_filter_;
+  const TickTimer* tick_timer_;
   int fs_mult_;
   size_t output_size_samples_;
   CngState cng_state_;  // Remember if comfort noise is interrupted by other
                         // event (e.g., DTMF).
-  size_t generated_noise_samples_;
+  size_t noise_fast_forward_ = 0;
   size_t packet_length_samples_;
   int sample_memory_;
   bool prev_time_scale_;
-  int timescale_hold_off_;
+  std::unique_ptr<TickTimer::Countdown> timescale_countdown_;
   int num_consecutive_expands_;
   const NetEqPlayoutMode playout_mode_;
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
index ddea64425f2..aace402a7de 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
@@ -26,7 +26,8 @@ Operations DecisionLogicFax::GetDecisionSpecialized(
     const RTPHeader* packet_header,
     Modes prev_mode,
     bool play_dtmf,
-    bool* reset_decoder) {
+    bool* reset_decoder,
+    size_t generated_noise_samples) {
   assert(playout_mode_ == kPlayoutFax || playout_mode_ == kPlayoutOff);
   uint32_t target_timestamp = sync_buffer.end_timestamp();
   uint32_t available_timestamp = 0;
@@ -37,7 +38,7 @@ Operations DecisionLogicFax::GetDecisionSpecialized(
         decoder_database_->IsComfortNoise(packet_header->payloadType);
   }
   if (is_cng_packet) {
-    if (static_cast<int32_t>((generated_noise_samples_ + target_timestamp)
+    if (static_cast<int32_t>((generated_noise_samples + target_timestamp)
         - available_timestamp) >= 0) {
       // Time to play this packet now.
       return kRfc3389Cng;
@@ -70,13 +71,13 @@ Operations DecisionLogicFax::GetDecisionSpecialized(
   } else if (target_timestamp == available_timestamp) {
     return kNormal;
   } else {
-    if (static_cast<int32_t>((generated_noise_samples_ + target_timestamp)
+    if (static_cast<int32_t>((generated_noise_samples + target_timestamp)
         - available_timestamp) >= 0) {
       return kNormal;
     } else {
       // If currently playing comfort noise, continue with that. Do not
-      // increase the timestamp counter since generated_noise_samples_ will
-      // be increased.
+      // increase the timestamp counter since generated_noise_stopwatch_ in
+      // NetEqImpl will take care of the time-keeping.
       if (cng_state_ == kCngRfc3389On) {
         return kRfc3389CngNoPacket;
       } else if (cng_state_ == kCngInternalOn) {
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
index 204dcc168a3..6958f908b1c 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
@@ -28,11 +28,16 @@ class DecisionLogicFax : public DecisionLogic {
                    DecoderDatabase* decoder_database,
                    const PacketBuffer& packet_buffer,
                    DelayManager* delay_manager,
-                   BufferLevelFilter* buffer_level_filter)
-      : DecisionLogic(fs_hz, output_size_samples, playout_mode,
-                      decoder_database, packet_buffer, delay_manager,
-                      buffer_level_filter) {
-  }
+                   BufferLevelFilter* buffer_level_filter,
+                   const TickTimer* tick_timer)
+      : DecisionLogic(fs_hz,
+                      output_size_samples,
+                      playout_mode,
+                      decoder_database,
+                      packet_buffer,
+                      delay_manager,
+                      buffer_level_filter,
+                      tick_timer) {}
 
  protected:
   // Returns the operation that should be done next. |sync_buffer| and |expand|
@@ -50,7 +55,8 @@ class DecisionLogicFax : public DecisionLogic {
                                     const RTPHeader* packet_header,
                                     Modes prev_mode,
                                     bool play_dtmf,
-                                    bool* reset_decoder) override;
+                                    bool* reset_decoder,
+                                    size_t generated_noise_samples) override;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
index 0252d1cdfaf..37a75d7f5ad 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -31,7 +31,8 @@ Operations DecisionLogicNormal::GetDecisionSpecialized(
     const RTPHeader* packet_header,
     Modes prev_mode,
     bool play_dtmf,
-    bool* reset_decoder) {
+    bool* reset_decoder,
+    size_t generated_noise_samples) {
   assert(playout_mode_ == kPlayoutOn || playout_mode_ == kPlayoutStreaming);
   // Guard for errors, to avoid getting stuck in error mode.
   if (prev_mode == kModeError) {
@@ -52,7 +53,8 @@ Operations DecisionLogicNormal::GetDecisionSpecialized(
   }
 
   if (is_cng_packet) {
-    return CngOperation(prev_mode, target_timestamp, available_timestamp);
+    return CngOperation(prev_mode, target_timestamp, available_timestamp,
+                        generated_noise_samples);
   }
 
   // Handle the case with no packet at all available (except maybe DTMF).
@@ -76,7 +78,8 @@ Operations DecisionLogicNormal::GetDecisionSpecialized(
                  available_timestamp, target_timestamp, five_seconds_samples)) {
     return FuturePacketAvailable(sync_buffer, expand, decoder_frame_length,
                                  prev_mode, target_timestamp,
-                                 available_timestamp, play_dtmf);
+                                 available_timestamp, play_dtmf,
+                                 generated_noise_samples);
   } else {
     // This implies that available_timestamp < target_timestamp, which can
     // happen when a new stream or codec is received. Signal for a reset.
@@ -86,10 +89,11 @@ Operations DecisionLogicNormal::GetDecisionSpecialized(
 
 Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
                                              uint32_t target_timestamp,
-                                             uint32_t available_timestamp) {
+                                             uint32_t available_timestamp,
+                                             size_t generated_noise_samples) {
   // Signed difference between target and available timestamp.
   int32_t timestamp_diff = static_cast<int32_t>(
-      static_cast<uint32_t>(generated_noise_samples_ + target_timestamp) -
+      static_cast<uint32_t>(generated_noise_samples + target_timestamp) -
       available_timestamp);
   int32_t optimal_level_samp = static_cast<int32_t>(
       (delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
@@ -97,9 +101,9 @@ Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
 
   if (excess_waiting_time_samp > optimal_level_samp / 2) {
     // The waiting time for this packet will be longer than 1.5
-    // times the wanted buffer delay. Advance the clock to cut
+    // times the wanted buffer delay. Apply fast-forward to cut the
     // waiting time down to the optimal.
-    generated_noise_samples_ += excess_waiting_time_samp;
+    noise_fast_forward_ += excess_waiting_time_samp;
     timestamp_diff += excess_waiting_time_samp;
   }
 
@@ -109,6 +113,7 @@ Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
     return kRfc3389CngNoPacket;
   } else {
     // Otherwise, go for the CNG packet now.
+    noise_fast_forward_ = 0;
     return kRfc3389Cng;
   }
 }
@@ -153,7 +158,8 @@ Operations DecisionLogicNormal::FuturePacketAvailable(
     Modes prev_mode,
     uint32_t target_timestamp,
     uint32_t available_timestamp,
-    bool play_dtmf) {
+    bool play_dtmf,
+    size_t generated_noise_samples) {
   // Required packet is not available, but a future packet is.
   // Check if we should continue with an ongoing expand because the new packet
   // is too far into the future.
@@ -184,7 +190,7 @@ Operations DecisionLogicNormal::FuturePacketAvailable(
     // safety precaution), but make sure that the number of samples in buffer
     // is no higher than 4 times the optimal level. (Note that TargetLevel()
     // is in Q8.)
-    if (static_cast<uint32_t>(generated_noise_samples_ + target_timestamp) >=
+    if (static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
             available_timestamp ||
         cur_size_samples >
             ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
index 7465906a381..aa0edf3152a 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
@@ -28,11 +28,16 @@ class DecisionLogicNormal : public DecisionLogic {
                       DecoderDatabase* decoder_database,
                       const PacketBuffer& packet_buffer,
                       DelayManager* delay_manager,
-                      BufferLevelFilter* buffer_level_filter)
-      : DecisionLogic(fs_hz, output_size_samples, playout_mode,
-                      decoder_database, packet_buffer, delay_manager,
-                      buffer_level_filter) {
-  }
+                      BufferLevelFilter* buffer_level_filter,
+                      const TickTimer* tick_timer)
+      : DecisionLogic(fs_hz,
+                      output_size_samples,
+                      playout_mode,
+                      decoder_database,
+                      packet_buffer,
+                      delay_manager,
+                      buffer_level_filter,
+                      tick_timer) {}
 
  protected:
   static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
@@ -54,7 +59,8 @@ class DecisionLogicNormal : public DecisionLogic {
                                     const RTPHeader* packet_header,
                                     Modes prev_mode,
                                     bool play_dtmf,
-                                    bool* reset_decoder) override;
+                                    bool* reset_decoder,
+                                    size_t generated_noise_samples) override;
 
   // Returns the operation to do given that the expected packet is not
   // available, but a packet further into the future is at hand.
@@ -65,7 +71,8 @@ class DecisionLogicNormal : public DecisionLogic {
       Modes prev_mode,
       uint32_t target_timestamp,
       uint32_t available_timestamp,
-      bool play_dtmf);
+      bool play_dtmf,
+      size_t generated_noise_samples);
 
   // Returns the operation to do given that the expected packet is available.
   virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
@@ -77,12 +84,16 @@ class DecisionLogicNormal : public DecisionLogic {
  private:
   // Returns the operation given that the next available packet is a comfort
   // noise payload (RFC 3389 only, not codec-internal).
-  Operations CngOperation(Modes prev_mode, uint32_t target_timestamp,
-                          uint32_t available_timestamp);
+  Operations CngOperation(Modes prev_mode,
+                          uint32_t target_timestamp,
+                          uint32_t available_timestamp,
+                          size_t generated_noise_samples);
 
   // Checks if enough time has elapsed since the last successful timescale
   // operation was done (i.e., accelerate or preemptive expand).
-  bool TimescaleAllowed() const { return timescale_hold_off_ == 0; }
+  bool TimescaleAllowed() const {
+    return !timescale_countdown_ || timescale_countdown_->Finished();
+  }
 
   // Checks if the current (filtered) buffer level is under the target level.
   bool UnderTargetLevel() const;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
index 499f9464347..ebb366890b3 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -11,45 +11,42 @@
 // Unit tests for DecisionLogic class and derived classes.
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 namespace webrtc {
 
 TEST(DecisionLogic, CreateAndDestroy) {
   int fs_hz = 8000;
   int output_size_samples = fs_hz / 100;  // Samples per 10 ms.
-  DecoderDatabase decoder_database;
-  PacketBuffer packet_buffer(10);
-  DelayPeakDetector delay_peak_detector;
-  DelayManager delay_manager(240, &delay_peak_detector);
+  DecoderDatabase decoder_database(
+      std::unique_ptr<MockAudioDecoderFactory>(new MockAudioDecoderFactory));
+  TickTimer tick_timer;
+  PacketBuffer packet_buffer(10, &tick_timer);
+  DelayPeakDetector delay_peak_detector(&tick_timer);
+  DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
   BufferLevelFilter buffer_level_filter;
-  DecisionLogic* logic = DecisionLogic::Create(fs_hz, output_size_samples,
-                                               kPlayoutOn, &decoder_database,
-                                               packet_buffer, &delay_manager,
-                                               &buffer_level_filter);
+  DecisionLogic* logic = DecisionLogic::Create(
+      fs_hz, output_size_samples, kPlayoutOn, &decoder_database, packet_buffer,
+      &delay_manager, &buffer_level_filter, &tick_timer);
   delete logic;
-  logic = DecisionLogic::Create(fs_hz, output_size_samples,
-                                kPlayoutStreaming,
-                                &decoder_database,
-                                packet_buffer, &delay_manager,
-                                &buffer_level_filter);
+  logic = DecisionLogic::Create(
+      fs_hz, output_size_samples, kPlayoutStreaming, &decoder_database,
+      packet_buffer, &delay_manager, &buffer_level_filter, &tick_timer);
   delete logic;
-  logic = DecisionLogic::Create(fs_hz, output_size_samples,
-                                kPlayoutFax,
-                                &decoder_database,
-                                packet_buffer, &delay_manager,
-                                &buffer_level_filter);
+  logic = DecisionLogic::Create(
+      fs_hz, output_size_samples, kPlayoutFax, &decoder_database, packet_buffer,
+      &delay_manager, &buffer_level_filter, &tick_timer);
   delete logic;
-  logic = DecisionLogic::Create(fs_hz, output_size_samples,
-                                kPlayoutOff,
-                                &decoder_database,
-                                packet_buffer, &delay_manager,
-                                &buffer_level_filter);
+  logic = DecisionLogic::Create(
+      fs_hz, output_size_samples, kPlayoutOff, &decoder_database, packet_buffer,
+      &delay_manager, &buffer_level_filter, &tick_timer);
   delete logic;
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.cc
index 92d4bab1e4a..4fddf75ce26 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -19,13 +19,39 @@
 
 namespace webrtc {
 
-DecoderDatabase::DecoderDatabase()
-    : active_decoder_(-1), active_cng_decoder_(-1) {}
+DecoderDatabase::DecoderDatabase(
+    std::unique_ptr<AudioDecoderFactory> decoder_factory)
+    : active_decoder_type_(-1),
+      active_cng_decoder_type_(-1),
+      decoder_factory_(std::move(decoder_factory)) {}
 
-DecoderDatabase::~DecoderDatabase() {}
+DecoderDatabase::~DecoderDatabase() = default;
 
-DecoderDatabase::DecoderInfo::~DecoderInfo() {
-  if (!external) delete decoder;
+DecoderDatabase::DecoderInfo::DecoderInfo(NetEqDecoder ct,
+                                          const std::string& nm,
+                                          int fs,
+                                          AudioDecoder* ext_dec)
+    : codec_type(ct),
+      name(nm),
+      fs_hz(fs),
+      external_decoder(ext_dec),
+      audio_format_(acm2::RentACodec::NetEqDecoderToSdpAudioFormat(ct)) {}
+
+DecoderDatabase::DecoderInfo::DecoderInfo(DecoderInfo&&) = default;
+DecoderDatabase::DecoderInfo::~DecoderInfo() = default;
+
+AudioDecoder* DecoderDatabase::DecoderInfo::GetDecoder(
+    AudioDecoderFactory* factory) {
+  if (external_decoder) {
+    RTC_DCHECK(!decoder_);
+    return external_decoder;
+  }
+  RTC_DCHECK(audio_format_);
+  if (!decoder_) {
+    decoder_ = factory->MakeAudioDecoder(*audio_format_);
+  }
+  RTC_DCHECK(decoder_) << "Failed to create: " << *audio_format_;
+  return decoder_.get();
 }
 
 bool DecoderDatabase::Empty() const { return decoders_.empty(); }
@@ -34,8 +60,8 @@ int DecoderDatabase::Size() const { return static_cast<int>(decoders_.size()); }
 
 void DecoderDatabase::Reset() {
   decoders_.clear();
-  active_decoder_ = -1;
-  active_cng_decoder_ = -1;
+  active_decoder_type_ = -1;
+  active_cng_decoder_type_ = -1;
 }
 
 int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
@@ -48,8 +74,9 @@ int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
     return kCodecNotSupported;
   }
   const int fs_hz = CodecSampleRateHz(codec_type);
-  DecoderInfo info(codec_type, name, fs_hz, NULL, false);
-  auto ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
+  DecoderInfo info(codec_type, name, fs_hz, nullptr);
+  auto ret =
+      decoders_.insert(std::make_pair(rtp_payload_type, std::move(info)));
   if (ret.second == false) {
     // Database already contains a decoder with type |rtp_payload_type|.
     return kDecoderExists;
@@ -75,8 +102,8 @@ int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
     return kInvalidPointer;
   }
   std::pair<DecoderMap::iterator, bool> ret;
-  DecoderInfo info(codec_type, codec_name, fs_hz, decoder, true);
-  ret = decoders_.insert(std::make_pair(rtp_payload_type, info));
+  DecoderInfo info(codec_type, codec_name, fs_hz, decoder);
+  ret = decoders_.insert(std::make_pair(rtp_payload_type, std::move(info)));
   if (ret.second == false) {
     // Database already contains a decoder with type |rtp_payload_type|.
     return kDecoderExists;
@@ -89,11 +116,11 @@ int DecoderDatabase::Remove(uint8_t rtp_payload_type) {
     // No decoder with that |rtp_payload_type|.
     return kDecoderNotFound;
   }
-  if (active_decoder_ == rtp_payload_type) {
-    active_decoder_ = -1;  // No active decoder.
+  if (active_decoder_type_ == rtp_payload_type) {
+    active_decoder_type_ = -1;  // No active decoder.
   }
-  if (active_cng_decoder_ == rtp_payload_type) {
-    active_cng_decoder_ = -1;  // No active CNG decoder.
+  if (active_cng_decoder_type_ == rtp_payload_type) {
+    active_cng_decoder_type_ = -1;  // No active CNG decoder.
   }
   return kOK;
 }
@@ -122,7 +149,8 @@ uint8_t DecoderDatabase::GetRtpPayloadType(
 }
 
 AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) {
-  if (IsDtmf(rtp_payload_type) || IsRed(rtp_payload_type)) {
+  if (IsDtmf(rtp_payload_type) || IsRed(rtp_payload_type) ||
+      IsComfortNoise(rtp_payload_type)) {
     // These are not real decoders.
     return NULL;
   }
@@ -132,13 +160,7 @@ AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) {
     return NULL;
   }
   DecoderInfo* info = &(*it).second;
-  if (!info->decoder) {
-    // Create the decoder object.
-    AudioDecoder* decoder = CreateAudioDecoder(info->codec_type);
-    assert(decoder);  // Should not be able to have an unsupported codec here.
-    info->decoder = decoder;
-  }
-  return info->decoder;
+  return info->GetDecoder(decoder_factory_.get());
 }
 
 bool DecoderDatabase::IsType(uint8_t rtp_payload_type,
@@ -152,14 +174,16 @@ bool DecoderDatabase::IsType(uint8_t rtp_payload_type,
 }
 
 bool DecoderDatabase::IsComfortNoise(uint8_t rtp_payload_type) const {
-  if (IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGnb) ||
-      IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGwb) ||
-      IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGswb32kHz) ||
-      IsType(rtp_payload_type, NetEqDecoder::kDecoderCNGswb48kHz)) {
-    return true;
-  } else {
+  DecoderMap::const_iterator it = decoders_.find(rtp_payload_type);
+  if (it == decoders_.end()) {
+    // Decoder not found.
     return false;
   }
+  const auto& type = it->second.codec_type;
+  return type == NetEqDecoder::kDecoderCNGnb
+      || type == NetEqDecoder::kDecoderCNGwb
+      || type == NetEqDecoder::kDecoderCNGswb32kHz
+      || type == NetEqDecoder::kDecoderCNGswb48kHz;
 }
 
 bool DecoderDatabase::IsDtmf(uint8_t rtp_payload_type) const {
@@ -178,37 +202,33 @@ int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type,
     // Decoder not found.
     return kDecoderNotFound;
   }
+  RTC_CHECK(!IsComfortNoise(rtp_payload_type));
   assert(new_decoder);
   *new_decoder = false;
-  if (active_decoder_ < 0) {
+  if (active_decoder_type_ < 0) {
     // This is the first active decoder.
     *new_decoder = true;
-  } else if (active_decoder_ != rtp_payload_type) {
+  } else if (active_decoder_type_ != rtp_payload_type) {
     // Moving from one active decoder to another. Delete the first one.
-    DecoderMap::iterator it = decoders_.find(active_decoder_);
+    DecoderMap::iterator it = decoders_.find(active_decoder_type_);
     if (it == decoders_.end()) {
       // Decoder not found. This should not be possible.
       assert(false);
       return kDecoderNotFound;
     }
-    if (!(*it).second.external) {
-      // Delete the AudioDecoder object, unless it is an externally created
-      // decoder.
-      delete (*it).second.decoder;
-      (*it).second.decoder = NULL;
-    }
+    it->second.DropDecoder();
     *new_decoder = true;
   }
-  active_decoder_ = rtp_payload_type;
+  active_decoder_type_ = rtp_payload_type;
   return kOK;
 }
 
 AudioDecoder* DecoderDatabase::GetActiveDecoder() {
-  if (active_decoder_ < 0) {
+  if (active_decoder_type_ < 0) {
     // No active decoder.
     return NULL;
   }
-  return GetDecoder(active_decoder_);
+  return GetDecoder(active_decoder_type_);
 }
 
 int DecoderDatabase::SetActiveCngDecoder(uint8_t rtp_payload_type) {
@@ -218,31 +238,32 @@ int DecoderDatabase::SetActiveCngDecoder(uint8_t rtp_payload_type) {
     // Decoder not found.
     return kDecoderNotFound;
   }
-  if (active_cng_decoder_ >= 0 && active_cng_decoder_ != rtp_payload_type) {
+  if (active_cng_decoder_type_ >= 0 &&
+      active_cng_decoder_type_ != rtp_payload_type) {
     // Moving from one active CNG decoder to another. Delete the first one.
-    DecoderMap::iterator it = decoders_.find(active_cng_decoder_);
+    DecoderMap::iterator it = decoders_.find(active_cng_decoder_type_);
     if (it == decoders_.end()) {
       // Decoder not found. This should not be possible.
       assert(false);
       return kDecoderNotFound;
     }
-    if (!(*it).second.external) {
-      // Delete the AudioDecoder object, unless it is an externally created
-      // decoder.
-      delete (*it).second.decoder;
-      (*it).second.decoder = NULL;
-    }
+    // The CNG decoder should never be provided externally.
+    RTC_CHECK(!it->second.external_decoder);
+    active_cng_decoder_.reset();
   }
-  active_cng_decoder_ = rtp_payload_type;
+  active_cng_decoder_type_ = rtp_payload_type;
   return kOK;
 }
 
-AudioDecoder* DecoderDatabase::GetActiveCngDecoder() {
-  if (active_cng_decoder_ < 0) {
+ComfortNoiseDecoder* DecoderDatabase::GetActiveCngDecoder() {
+  if (active_cng_decoder_type_ < 0) {
     // No active CNG decoder.
     return NULL;
   }
-  return GetDecoder(active_cng_decoder_);
+  if (!active_cng_decoder_) {
+    active_cng_decoder_.reset(new ComfortNoiseDecoder);
+  }
+  return active_cng_decoder_.get();
 }
 
 int DecoderDatabase::CheckPayloadTypes(const PacketList& packet_list) const {
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.h
index 01ff0c9fdb3..3a40e08c8a6 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -12,10 +12,14 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
 
 #include <map>
+#include <memory>
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/common_types.h"  // NULL
+#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 #include "webrtc/typedefs.h"
@@ -34,37 +38,38 @@ class DecoderDatabase {
     kInvalidPointer = -6
   };
 
-  // Struct used to store decoder info in the database.
-  struct DecoderInfo {
-    DecoderInfo() = default;
-    DecoderInfo(NetEqDecoder ct, int fs, AudioDecoder* dec, bool ext)
-        : DecoderInfo(ct, "", fs, dec, ext) {}
+  // Class that stores decoder info in the database.
+  class DecoderInfo {
+   public:
     DecoderInfo(NetEqDecoder ct,
                 const std::string& nm,
                 int fs,
-                AudioDecoder* dec,
-                bool ext)
-        : codec_type(ct),
-          name(nm),
-          fs_hz(fs),
-          rtp_sample_rate_hz(fs),
-          decoder(dec),
-          external(ext) {}
+                AudioDecoder* ext_dec);
+    DecoderInfo(DecoderInfo&&);
     ~DecoderInfo();
 
-    NetEqDecoder codec_type = NetEqDecoder::kDecoderArbitrary;
-    std::string name;
-    int fs_hz = 8000;
-    int rtp_sample_rate_hz = 8000;
-    AudioDecoder* decoder = nullptr;
-    bool external = false;
+    // Get the AudioDecoder object, creating it first if necessary.
+    AudioDecoder* GetDecoder(AudioDecoderFactory* factory);
+
+    // Delete the AudioDecoder object, unless it's external. (This means we can
+    // always recreate it later if we need it.)
+    void DropDecoder() { decoder_.reset(); }
+
+    const NetEqDecoder codec_type;
+    const std::string name;
+    const int fs_hz;
+    AudioDecoder* const external_decoder;
+
+   private:
+    const rtc::Optional<SdpAudioFormat> audio_format_;
+    std::unique_ptr<AudioDecoder> decoder_;
   };
 
   // Maximum value for 8 bits, and an invalid RTP payload type (since it is
   // only 7 bits).
   static const uint8_t kRtpPayloadTypeError = 0xFF;
 
-  DecoderDatabase();
+  DecoderDatabase(std::unique_ptr<AudioDecoderFactory> decoder_factory);
 
   virtual ~DecoderDatabase();
 
@@ -142,7 +147,7 @@ class DecoderDatabase {
 
   // Returns the current active comfort noise decoder, or NULL if no active
   // comfort noise decoder exists.
-  virtual AudioDecoder* GetActiveCngDecoder();
+  virtual ComfortNoiseDecoder* GetActiveCngDecoder();
 
   // Returns kOK if all packets in |packet_list| carry payload types that are
   // registered in the database. Otherwise, returns kDecoderNotFound.
@@ -152,8 +157,10 @@ class DecoderDatabase {
   typedef std::map<uint8_t, DecoderInfo> DecoderMap;
 
   DecoderMap decoders_;
-  int active_decoder_;
-  int active_cng_decoder_;
+  int active_decoder_type_;
+  int active_cng_decoder_type_;
+  std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
+  const std::unique_ptr<AudioDecoderFactory> decoder_factory_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DecoderDatabase);
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
index 85aaef11431..91ca606d65b 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -19,17 +19,21 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 
 namespace webrtc {
 
 TEST(DecoderDatabase, CreateAndDestroy) {
-  DecoderDatabase db;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase db(std::move(factory));
   EXPECT_EQ(0, db.Size());
   EXPECT_TRUE(db.Empty());
 }
 
 TEST(DecoderDatabase, InsertAndRemove) {
-  DecoderDatabase db;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase db(std::move(factory));
   const uint8_t kPayloadType = 0;
   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_EQ(
@@ -43,7 +47,8 @@ TEST(DecoderDatabase, InsertAndRemove) {
 }
 
 TEST(DecoderDatabase, GetDecoderInfo) {
-  DecoderDatabase db;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase db(std::move(factory));
   const uint8_t kPayloadType = 0;
   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_EQ(
@@ -53,16 +58,16 @@ TEST(DecoderDatabase, GetDecoderInfo) {
   info = db.GetDecoderInfo(kPayloadType);
   ASSERT_TRUE(info != NULL);
   EXPECT_EQ(NetEqDecoder::kDecoderPCMu, info->codec_type);
-  EXPECT_EQ(NULL, info->decoder);
+  EXPECT_EQ(nullptr, info->external_decoder);
   EXPECT_EQ(8000, info->fs_hz);
   EXPECT_EQ(kCodecName, info->name);
-  EXPECT_FALSE(info->external);
   info = db.GetDecoderInfo(kPayloadType + 1);  // Other payload type.
   EXPECT_TRUE(info == NULL);  // Should not be found.
 }
 
 TEST(DecoderDatabase, GetRtpPayloadType) {
-  DecoderDatabase db;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase db(std::move(factory));
   const uint8_t kPayloadType = 0;
   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_EQ(
@@ -76,7 +81,7 @@ TEST(DecoderDatabase, GetRtpPayloadType) {
 }
 
 TEST(DecoderDatabase, GetDecoder) {
-  DecoderDatabase db;
+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory());
   const uint8_t kPayloadType = 0;
   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   EXPECT_EQ(DecoderDatabase::kOK,
@@ -87,7 +92,8 @@ TEST(DecoderDatabase, GetDecoder) {
 }
 
 TEST(DecoderDatabase, TypeTests) {
-  DecoderDatabase db;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase db(std::move(factory));
   const uint8_t kPayloadTypePcmU = 0;
   const uint8_t kPayloadTypeCng = 13;
   const uint8_t kPayloadTypeDtmf = 100;
@@ -122,7 +128,8 @@ TEST(DecoderDatabase, TypeTests) {
 }
 
 TEST(DecoderDatabase, ExternalDecoder) {
-  DecoderDatabase db;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase db(std::move(factory));
   const uint8_t kPayloadType = 0;
   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
   MockAudioDecoder decoder;
@@ -139,9 +146,8 @@ TEST(DecoderDatabase, ExternalDecoder) {
   ASSERT_TRUE(info != NULL);
   EXPECT_EQ(NetEqDecoder::kDecoderPCMu, info->codec_type);
   EXPECT_EQ(kCodecName, info->name);
-  EXPECT_EQ(&decoder, info->decoder);
+  EXPECT_EQ(&decoder, info->external_decoder);
   EXPECT_EQ(8000, info->fs_hz);
-  EXPECT_TRUE(info->external);
   // Expect not to delete the decoder when removing it from the database, since
   // it was declared externally.
   EXPECT_CALL(decoder, Die()).Times(0);
@@ -152,7 +158,8 @@ TEST(DecoderDatabase, ExternalDecoder) {
 }
 
 TEST(DecoderDatabase, CheckPayloadTypes) {
-  DecoderDatabase db;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase db(std::move(factory));
   // Load a number of payloads into the database. Payload types are 0, 1, ...,
   // while the decoder type is the same for all payload types (this does not
   // matter for the test).
@@ -196,7 +203,7 @@ TEST(DecoderDatabase, CheckPayloadTypes) {
 
 // Test the methods for setting and getting active speech and CNG decoders.
 TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
-  DecoderDatabase db;
+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory());
   // Load payload types.
   ASSERT_EQ(DecoderDatabase::kOK,
             db.RegisterPayload(0, NetEqDecoder::kDecoderPCMu, "pcmu"));
@@ -233,8 +240,8 @@ TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
 
   // Set active CNG codec.
   EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveCngDecoder(13));
-  decoder = db.GetActiveCngDecoder();
-  ASSERT_FALSE(decoder == NULL);  // Should get a decoder here.
+  ComfortNoiseDecoder* cng = db.GetActiveCngDecoder();
+  ASSERT_FALSE(cng == NULL);  // Should get a decoder here.
 
   // Remove the active CNG decoder, and verify that the active becomes NULL.
   EXPECT_EQ(DecoderDatabase::kOK, db.Remove(13));
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.cc
index af49f00f8af..84bda7cf699 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -24,12 +24,13 @@
 namespace webrtc {
 
 DelayManager::DelayManager(size_t max_packets_in_buffer,
-                           DelayPeakDetector* peak_detector)
+                           DelayPeakDetector* peak_detector,
+                           const TickTimer* tick_timer)
     : first_packet_received_(false),
       max_packets_in_buffer_(max_packets_in_buffer),
       iat_vector_(kMaxIat + 1, 0),
       iat_factor_(0),
-      packet_iat_count_ms_(0),
+      tick_timer_(tick_timer),
       base_target_level_(4),  // In Q0 domain.
       target_level_(base_target_level_ << 8),  // In Q8 domain.
       packet_len_ms_(0),
@@ -41,7 +42,6 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
       maximum_delay_ms_(target_level_),
       iat_cumulative_sum_(0),
       max_iat_cumulative_sum_(0),
-      max_timer_ms_(0),
       peak_detector_(*peak_detector),
       last_pack_cng_or_dtmf_(1) {
   assert(peak_detector);  // Should never be NULL.
@@ -79,7 +79,7 @@ int DelayManager::Update(uint16_t sequence_number,
 
   if (!first_packet_received_) {
     // Prepare for next packet arrival.
-    packet_iat_count_ms_ = 0;
+    packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
     last_seq_no_ = sequence_number;
     last_timestamp_ = timestamp;
     first_packet_received_ = true;
@@ -106,7 +106,7 @@ int DelayManager::Update(uint16_t sequence_number,
     // Calculate inter-arrival time (IAT) in integer "packet times"
     // (rounding down). This is the value used as index to the histogram
     // vector |iat_vector_|.
-    int iat_packets = packet_iat_count_ms_ / packet_len_ms;
+    int iat_packets = packet_iat_stopwatch_->ElapsedMs() / packet_len_ms;
 
     if (streaming_mode_) {
       UpdateCumulativeSums(packet_len_ms, sequence_number);
@@ -137,7 +137,7 @@ int DelayManager::Update(uint16_t sequence_number,
   }  // End if (packet_len_ms > 0).
 
   // Prepare for next packet arrival.
-  packet_iat_count_ms_ = 0;
+  packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
   last_seq_no_ = sequence_number;
   last_timestamp_ = timestamp;
   return 0;
@@ -147,7 +147,8 @@ void DelayManager::UpdateCumulativeSums(int packet_len_ms,
                                         uint16_t sequence_number) {
   // Calculate IAT in Q8, including fractions of a packet (i.e., more
   // accurate than |iat_packets|.
-  int iat_packets_q8 = (packet_iat_count_ms_ << 8) / packet_len_ms;
+  int iat_packets_q8 =
+      (packet_iat_stopwatch_->ElapsedMs() << 8) / packet_len_ms;
   // Calculate cumulative sum IAT with sequence number compensation. The sum
   // is zero if there is no clock-drift.
   iat_cumulative_sum_ += (iat_packets_q8 -
@@ -159,9 +160,9 @@ void DelayManager::UpdateCumulativeSums(int packet_len_ms,
   if (iat_cumulative_sum_ > max_iat_cumulative_sum_) {
     // Found a new maximum.
     max_iat_cumulative_sum_ = iat_cumulative_sum_;
-    max_timer_ms_ = 0;
+    max_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
   }
-  if (max_timer_ms_ > kMaxStreamingPeakPeriodMs) {
+  if (max_iat_stopwatch_->ElapsedMs() > kMaxStreamingPeakPeriodMs) {
     // Too long since the last maximum was observed; decrease max value.
     max_iat_cumulative_sum_ -= kCumulativeSumDrift;
   }
@@ -299,7 +300,7 @@ int DelayManager::SetPacketAudioLength(int length_ms) {
   }
   packet_len_ms_ = length_ms;
   peak_detector_.SetPacketAudioLength(packet_len_ms_);
-  packet_iat_count_ms_ = 0;
+  packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
   last_pack_cng_or_dtmf_ = 1;  // TODO(hlundin): Legacy. Remove?
   return 0;
 }
@@ -311,8 +312,8 @@ void DelayManager::Reset() {
   peak_detector_.Reset();
   ResetHistogram();  // Resets target levels too.
   iat_factor_ = 0;  // Adapt the histogram faster for the first few packets.
-  packet_iat_count_ms_ = 0;
-  max_timer_ms_ = 0;
+  packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
+  max_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
   iat_cumulative_sum_ = 0;
   max_iat_cumulative_sum_ = 0;
   last_pack_cng_or_dtmf_ = 1;
@@ -340,14 +341,10 @@ bool DelayManager::PeakFound() const {
   return peak_detector_.peak_found();
 }
 
-void DelayManager::UpdateCounters(int elapsed_time_ms) {
-  packet_iat_count_ms_ += elapsed_time_ms;
-  peak_detector_.IncrementCounter(elapsed_time_ms);
-  max_timer_ms_ += elapsed_time_ms;
+void DelayManager::ResetPacketIatCount() {
+  packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
 }
 
-void DelayManager::ResetPacketIatCount() { packet_iat_count_ms_ = 0; }
-
 // Note that |low_limit| and |higher_limit| are not assigned to
 // |minimum_delay_ms_| and |maximum_delay_ms_| defined by the client of this
 // class. They are computed from |target_level_| and used for decision making.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.h
index 785fced15df..6f3c14aea9d 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager.h
@@ -13,10 +13,12 @@
 
 #include <string.h>  // Provide access to size_t.
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -32,7 +34,9 @@ class DelayManager {
   // buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
   // is the number of packet slots in the buffer). Supply a PeakDetector
   // object to the DelayManager.
-  DelayManager(size_t max_packets_in_buffer, DelayPeakDetector* peak_detector);
+  DelayManager(size_t max_packets_in_buffer,
+               DelayPeakDetector* peak_detector,
+               const TickTimer* tick_timer);
 
   virtual ~DelayManager();
 
@@ -75,10 +79,6 @@ class DelayManager {
   // DelayPeakDetector object.
   virtual bool PeakFound() const;
 
-  // Notifies the counters in DelayManager and DelayPeakDetector that
-  // |elapsed_time_ms| have elapsed.
-  virtual void UpdateCounters(int elapsed_time_ms);
-
   // Reset the inter-arrival time counter to 0.
   virtual void ResetPacketIatCount();
 
@@ -135,7 +135,9 @@ class DelayManager {
   const size_t max_packets_in_buffer_;  // Capacity of the packet buffer.
   IATVector iat_vector_;  // Histogram of inter-arrival times.
   int iat_factor_;  // Forgetting factor for updating the IAT histogram (Q15).
-  int packet_iat_count_ms_;  // Milliseconds elapsed since last packet.
+  const TickTimer* tick_timer_;
+  // Time elapsed since last packet.
+  std::unique_ptr<TickTimer::Stopwatch> packet_iat_stopwatch_;
   int base_target_level_;   // Currently preferred buffer level before peak
                             // detection and streaming mode (Q0).
   // TODO(turajs) change the comment according to the implementation of
@@ -153,7 +155,8 @@ class DelayManager {
   int maximum_delay_ms_;  // Externally set maximum allowed delay.
   int iat_cumulative_sum_;  // Cumulative sum of delta inter-arrival times.
   int max_iat_cumulative_sum_;  // Max of |iat_cumulative_sum_|.
-  int max_timer_ms_;  // Time elapsed since maximum was observed.
+  // Time elapsed since maximum was observed.
+  std::unique_ptr<TickTimer::Stopwatch> max_iat_stopwatch_;
   DelayPeakDetector& peak_detector_;
   int last_pack_cng_or_dtmf_;
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
index f231c3da301..3290e9cca68 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -39,21 +39,19 @@ class DelayManagerTest : public ::testing::Test {
   void IncreaseTime(int inc_ms);
 
   DelayManager* dm_;
+  TickTimer tick_timer_;
   MockDelayPeakDetector detector_;
   uint16_t seq_no_;
   uint32_t ts_;
 };
 
 DelayManagerTest::DelayManagerTest()
-    : dm_(NULL),
-      seq_no_(0x1234),
-      ts_(0x12345678) {
-}
+    : dm_(NULL), detector_(&tick_timer_), seq_no_(0x1234), ts_(0x12345678) {}
 
 void DelayManagerTest::SetUp() {
   EXPECT_CALL(detector_, Reset())
             .Times(1);
-  dm_ = new DelayManager(kMaxNumberOfPackets, &detector_);
+  dm_ = new DelayManager(kMaxNumberOfPackets, &detector_, &tick_timer_);
 }
 
 void DelayManagerTest::SetPacketAudioLength(int lengt_ms) {
@@ -69,9 +67,7 @@ void DelayManagerTest::InsertNextPacket() {
 
 void DelayManagerTest::IncreaseTime(int inc_ms) {
   for (int t = 0; t < inc_ms; t += kTimeStepMs) {
-    EXPECT_CALL(detector_, IncrementCounter(kTimeStepMs))
-        .Times(1);
-    dm_->UpdateCounters(kTimeStepMs);
+    tick_timer_.Increment();
   }
 }
 void DelayManagerTest::TearDown() {
@@ -115,13 +111,6 @@ TEST_F(DelayManagerTest, PeakFound) {
   EXPECT_FALSE(dm_->PeakFound());
 }
 
-TEST_F(DelayManagerTest, UpdateCounters) {
-  // Expect DelayManager to pass on the counter update to the detector.
-  EXPECT_CALL(detector_, IncrementCounter(kTimeStepMs))
-      .Times(1);
-  dm_->UpdateCounters(kTimeStepMs);
-}
-
 TEST_F(DelayManagerTest, UpdateNormal) {
   SetPacketAudioLength(kFrameSizeMs);
   // First packet arrival.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
index 712c7788aca..ce9133bdaed 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
@@ -12,6 +12,9 @@
 
 #include <algorithm>  // max
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+
 namespace webrtc {
 
 // The DelayPeakDetector keeps track of severe inter-arrival times, called
@@ -23,14 +26,15 @@ namespace webrtc {
 
 DelayPeakDetector::~DelayPeakDetector() = default;
 
-DelayPeakDetector::DelayPeakDetector()
-  : peak_found_(false),
-    peak_detection_threshold_(0),
-    peak_period_counter_ms_(-1) {
+DelayPeakDetector::DelayPeakDetector(const TickTimer* tick_timer)
+    : peak_found_(false),
+      peak_detection_threshold_(0),
+      tick_timer_(tick_timer) {
+  RTC_DCHECK(!peak_period_stopwatch_);
 }
 
 void DelayPeakDetector::Reset() {
-  peak_period_counter_ms_ = -1;  // Indicate that next peak is the first.
+  peak_period_stopwatch_.reset();
   peak_found_ = false;
   peak_history_.clear();
 }
@@ -55,38 +59,40 @@ int DelayPeakDetector::MaxPeakHeight() const {
   return max_height;
 }
 
-int DelayPeakDetector::MaxPeakPeriod() const {
-  int max_period = -1;  // Returns -1 for an empty history.
-  std::list<Peak>::const_iterator it;
-  for (it = peak_history_.begin(); it != peak_history_.end(); ++it) {
-    max_period = std::max(max_period, it->period_ms);
+uint64_t DelayPeakDetector::MaxPeakPeriod() const {
+  auto max_period_element = std::max_element(
+      peak_history_.begin(), peak_history_.end(),
+      [](Peak a, Peak b) { return a.period_ms < b.period_ms; });
+  if (max_period_element == peak_history_.end()) {
+    return 0;  // |peak_history_| is empty.
   }
-  return max_period;
+  RTC_DCHECK_GT(max_period_element->period_ms, 0u);
+  return max_period_element->period_ms;
 }
 
 bool DelayPeakDetector::Update(int inter_arrival_time, int target_level) {
   if (inter_arrival_time > target_level + peak_detection_threshold_ ||
       inter_arrival_time > 2 * target_level) {
     // A delay peak is observed.
-    if (peak_period_counter_ms_ == -1) {
+    if (!peak_period_stopwatch_) {
       // This is the first peak. Reset the period counter.
-      peak_period_counter_ms_ = 0;
-    } else if (peak_period_counter_ms_ <= kMaxPeakPeriodMs) {
+      peak_period_stopwatch_ = tick_timer_->GetNewStopwatch();
+    } else if (peak_period_stopwatch_->ElapsedMs() <= kMaxPeakPeriodMs) {
       // This is not the first peak, and the period is valid.
       // Store peak data in the vector.
       Peak peak_data;
-      peak_data.period_ms = peak_period_counter_ms_;
+      peak_data.period_ms = peak_period_stopwatch_->ElapsedMs();
       peak_data.peak_height_packets = inter_arrival_time;
       peak_history_.push_back(peak_data);
       while (peak_history_.size() > kMaxNumPeaks) {
         // Delete the oldest data point.
         peak_history_.pop_front();
       }
-      peak_period_counter_ms_ = 0;
-    } else if (peak_period_counter_ms_ <= 2 * kMaxPeakPeriodMs) {
+      peak_period_stopwatch_ = tick_timer_->GetNewStopwatch();
+    } else if (peak_period_stopwatch_->ElapsedMs() <= 2 * kMaxPeakPeriodMs) {
       // Invalid peak due to too long period. Reset period counter and start
       // looking for next peak.
-      peak_period_counter_ms_ = 0;
+      peak_period_stopwatch_ = tick_timer_->GetNewStopwatch();
     } else {
       // More than 2 times the maximum period has elapsed since the last peak
       // was registered. It seams that the network conditions have changed.
@@ -97,16 +103,10 @@ bool DelayPeakDetector::Update(int inter_arrival_time, int target_level) {
   return CheckPeakConditions();
 }
 
-void DelayPeakDetector::IncrementCounter(int inc_ms) {
-  if (peak_period_counter_ms_ >= 0) {
-    peak_period_counter_ms_ += inc_ms;
-  }
-}
-
 bool DelayPeakDetector::CheckPeakConditions() {
   size_t s = peak_history_.size();
   if (s >= kMinPeaksToTrigger &&
-      peak_period_counter_ms_ <= 2 * MaxPeakPeriod()) {
+      peak_period_stopwatch_->ElapsedMs() <= 2 * MaxPeakPeriod()) {
     peak_found_ = true;
   } else {
     peak_found_ = false;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
index 69433b45248..f57d3bd71e5 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
@@ -14,14 +14,16 @@
 #include <string.h>  // size_t
 
 #include <list>
+#include <memory>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 namespace webrtc {
 
 class DelayPeakDetector {
  public:
-  DelayPeakDetector();
+  DelayPeakDetector(const TickTimer* tick_timer);
   virtual ~DelayPeakDetector();
   virtual void Reset();
 
@@ -37,20 +39,15 @@ class DelayPeakDetector {
   // delay peaks have been observed recently. The unit is number of packets.
   virtual int MaxPeakHeight() const;
 
-  // Calculates and returns the maximum delay peak distance in ms.
-  // Returns -1 if no delay peaks have been observed recently.
-  virtual int MaxPeakPeriod() const;
+  // Calculates and returns the maximum delay peak distance in ms (strictly
+  // larger than 0), or 0 if no delay peaks have been observed recently.
+  virtual uint64_t MaxPeakPeriod() const;
 
   // Updates the DelayPeakDetector with a new inter-arrival time (in packets)
   // and the current target buffer level (needed to decide if a peak is observed
   // or not). Returns true if peak-mode is active, false if not.
   virtual bool Update(int inter_arrival_time, int target_level);
 
-  // Increments the |peak_period_counter_ms_| with |inc_ms|. Only increments
-  // the counter if it is non-negative. A negative denotes that no peak has
-  // been observed.
-  virtual void IncrementCounter(int inc_ms);
-
  private:
   static const size_t kMaxNumPeaks = 8;
   static const size_t kMinPeaksToTrigger = 2;
@@ -58,7 +55,7 @@ class DelayPeakDetector {
   static const int kMaxPeakPeriodMs = 10000;
 
   typedef struct {
-    int period_ms;
+    uint64_t period_ms;
     int peak_height_packets;
   } Peak;
 
@@ -67,7 +64,8 @@ class DelayPeakDetector {
   std::list<Peak> peak_history_;
   bool peak_found_;
   int peak_detection_threshold_;
-  int peak_period_counter_ms_;
+  const TickTimer* tick_timer_;
+  std::unique_ptr<TickTimer::Stopwatch> peak_period_stopwatch_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DelayPeakDetector);
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
index c40f3991b04..32b36b25ef4 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
@@ -17,22 +17,25 @@
 namespace webrtc {
 
 TEST(DelayPeakDetector, CreateAndDestroy) {
-  DelayPeakDetector* detector = new DelayPeakDetector();
+  TickTimer tick_timer;
+  DelayPeakDetector* detector = new DelayPeakDetector(&tick_timer);
   EXPECT_FALSE(detector->peak_found());
   delete detector;
 }
 
 TEST(DelayPeakDetector, EmptyHistory) {
-  DelayPeakDetector detector;
+  TickTimer tick_timer;
+  DelayPeakDetector detector(&tick_timer);
   EXPECT_EQ(-1, detector.MaxPeakHeight());
-  EXPECT_EQ(-1, detector.MaxPeakPeriod());
+  EXPECT_EQ(0u, detector.MaxPeakPeriod());
 }
 
 // Inject a series of packet arrivals into the detector. Three of the packets
 // have suffered delays. After the third delay peak, peak-mode is expected to
 // start. This should then continue until it is disengaged due to lack of peaks.
 TEST(DelayPeakDetector, TriggerPeakMode) {
-  DelayPeakDetector detector;
+  TickTimer tick_timer;
+  DelayPeakDetector detector(&tick_timer);
   const int kPacketSizeMs = 30;
   detector.SetPacketAudioLength(kPacketSizeMs);
 
@@ -52,7 +55,7 @@ TEST(DelayPeakDetector, TriggerPeakMode) {
   // Third delay peak. Trigger peak-mode after this packet.
   arrival_times_ms[400] += kPeakDelayMs;
   // The second peak period is the longest, 200 packets.
-  const int kWorstPeakPeriod = 200 * kPacketSizeMs;
+  const uint64_t kWorstPeakPeriod = 200 * kPacketSizeMs;
   int peak_mode_start_ms = arrival_times_ms[400];
   // Expect to disengage after no peaks are observed for two period times.
   int peak_mode_end_ms = peak_mode_start_ms + 2 * kWorstPeakPeriod;
@@ -74,7 +77,7 @@ TEST(DelayPeakDetector, TriggerPeakMode) {
       }
       ++next;
     }
-    detector.IncrementCounter(10);
+    tick_timer.Increment();
     time += 10;  // Increase time 10 ms.
   }
 }
@@ -83,7 +86,8 @@ TEST(DelayPeakDetector, TriggerPeakMode) {
 // 2, in order to raise the bar for delay peaks to inter-arrival times > 4.
 // The delay pattern has peaks with delay = 3, thus should not trigger.
 TEST(DelayPeakDetector, DoNotTriggerPeakMode) {
-  DelayPeakDetector detector;
+  TickTimer tick_timer;
+  DelayPeakDetector detector(&tick_timer);
   const int kPacketSizeMs = 30;
   detector.SetPacketAudioLength(kPacketSizeMs);
 
@@ -114,7 +118,7 @@ TEST(DelayPeakDetector, DoNotTriggerPeakMode) {
       EXPECT_FALSE(detector.Update(iat_packets, kTargetBufferLevel));
       ++next;
     }
-    detector.IncrementCounter(10);
+    tick_timer.Increment();
     time += 10;  // Increase time 10 ms.
   }
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.cc
index 4188914c86c..32756650942 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -80,6 +80,22 @@ int DspHelper::RampSignal(int16_t* signal,
   return RampSignal(signal, length, factor, increment, signal);
 }
 
+int DspHelper::RampSignal(AudioVector* signal,
+                          size_t start_index,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  int factor_q20 = (factor << 6) + 32;
+  // TODO(hlundin): Add 32 to factor_q20 when converting back to Q14?
+  for (size_t i = start_index; i < start_index + length; ++i) {
+    (*signal)[i] = (factor * (*signal)[i] + 8192) >> 14;
+    factor_q20 += increment;
+    factor_q20 = std::max(factor_q20, 0);  // Never go negative.
+    factor = std::min(factor_q20 >> 6, 16384);
+  }
+  return factor;
+}
+
 int DspHelper::RampSignal(AudioMultiVector* signal,
                           size_t start_index,
                           size_t length,
@@ -94,7 +110,7 @@ int DspHelper::RampSignal(AudioMultiVector* signal,
   // Loop over the channels, starting at the same |factor| each time.
   for (size_t channel = 0; channel < signal->Channels(); ++channel) {
     end_factor =
-        RampSignal(&(*signal)[channel][start_index], length, factor, increment);
+        RampSignal(&(*signal)[channel], start_index, length, factor, increment);
   }
   return end_factor;
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h
index 269c2eb0f25..23543fe383e 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/dsp_helper.h
@@ -67,6 +67,13 @@ class DspHelper {
 
   // Same as above, but processes |length| samples from |signal|, starting at
   // |start_index|.
+  static int RampSignal(AudioVector* signal,
+                        size_t start_index,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Same as above, but for an AudioMultiVector.
   static int RampSignal(AudioMultiVector* signal,
                         size_t start_index,
                         size_t length,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.cc
index ef7af46597e..963f4bdb6c0 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.cc
@@ -19,6 +19,7 @@
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
@@ -111,25 +112,33 @@ int Expand::Process(AudioMultiVector* output) {
       // Use only expand_vector0.
       assert(expansion_vector_position + temp_length <=
              parameters.expand_vector0.Size());
-      memcpy(voiced_vector_storage,
-             &parameters.expand_vector0[expansion_vector_position],
-             sizeof(int16_t) * temp_length);
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       voiced_vector_storage);
     } else if (current_lag_index_ == 1) {
+      std::unique_ptr<int16_t[]> temp_0(new int16_t[temp_length]);
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       temp_0.get());
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[temp_length]);
+      parameters.expand_vector1.CopyTo(temp_length, expansion_vector_position,
+                                       temp_1.get());
       // Mix 3/4 of expand_vector0 with 1/4 of expand_vector1.
-      WebRtcSpl_ScaleAndAddVectorsWithRound(
-          &parameters.expand_vector0[expansion_vector_position], 3,
-          &parameters.expand_vector1[expansion_vector_position], 1, 2,
-          voiced_vector_storage, temp_length);
+      WebRtcSpl_ScaleAndAddVectorsWithRound(temp_0.get(), 3, temp_1.get(), 1, 2,
+                                            voiced_vector_storage, temp_length);
     } else if (current_lag_index_ == 2) {
       // Mix 1/2 of expand_vector0 with 1/2 of expand_vector1.
       assert(expansion_vector_position + temp_length <=
              parameters.expand_vector0.Size());
       assert(expansion_vector_position + temp_length <=
              parameters.expand_vector1.Size());
-      WebRtcSpl_ScaleAndAddVectorsWithRound(
-          &parameters.expand_vector0[expansion_vector_position], 1,
-          &parameters.expand_vector1[expansion_vector_position], 1, 1,
-          voiced_vector_storage, temp_length);
+
+      std::unique_ptr<int16_t[]> temp_0(new int16_t[temp_length]);
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       temp_0.get());
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[temp_length]);
+      parameters.expand_vector1.CopyTo(temp_length, expansion_vector_position,
+                                       temp_1.get());
+      WebRtcSpl_ScaleAndAddVectorsWithRound(temp_0.get(), 1, temp_1.get(), 1, 1,
+                                            voiced_vector_storage, temp_length);
     }
 
     // Get tapering window parameters. Values are in Q15.
@@ -298,8 +307,7 @@ int Expand::Process(AudioMultiVector* output) {
     } else {
       assert(output->Size() == current_lag);
     }
-    memcpy(&(*output)[channel_ix][0], temp_data,
-           sizeof(temp_data[0]) * current_lag);
+    (*output)[channel_ix].OverwriteAt(temp_data, current_lag, 0);
   }
 
   // Increase call number and cap it.
@@ -326,6 +334,17 @@ void Expand::SetParametersForMergeAfterExpand() {
   stop_muting_ = true;
 }
 
+bool Expand::Muted() const {
+  if (first_expand_ || stop_muting_)
+    return false;
+  RTC_DCHECK(channel_parameters_);
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    if (channel_parameters_[ch].mute_factor != 0)
+      return false;
+  }
+  return true;
+}
+
 size_t Expand::overlap_length() const {
   return overlap_length_;
 }
@@ -372,19 +391,20 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   size_t fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength;
 
   const size_t signal_length = static_cast<size_t>(256 * fs_mult);
-  const int16_t* audio_history =
-      &(*sync_buffer_)[0][sync_buffer_->Size() - signal_length];
+
+  const size_t audio_history_position = sync_buffer_->Size() - signal_length;
+  std::unique_ptr<int16_t[]> audio_history(new int16_t[signal_length]);
+  (*sync_buffer_)[0].CopyTo(signal_length, audio_history_position,
+                            audio_history.get());
 
   // Initialize.
   InitializeForAnExpandPeriod();
 
   // Calculate correlation in downsampled domain (4 kHz sample rate).
-  int correlation_scale;
   size_t correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
   // If it is decided to break bit-exactness |correlation_length| should be
   // initialized to the return value of Correlation().
-  Correlation(audio_history, signal_length, correlation_vector,
-              &correlation_scale);
+  Correlation(audio_history.get(), signal_length, correlation_vector);
 
   // Find peaks in correlation vector.
   DspHelper::PeakDetection(correlation_vector, correlation_length,
@@ -455,7 +475,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
         &audio_history[signal_length - correlation_length - start_index
                        - correlation_lags],
                        correlation_length + start_index + correlation_lags - 1);
-    correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
+    int correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
         (31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
     correlation_scale = std::max(0, correlation_scale);
 
@@ -541,12 +561,14 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
         parameters.expand_vector1.Extend(
             expansion_length - parameters.expand_vector1.Size());
       }
-      WebRtcSpl_AffineTransformVector(&parameters.expand_vector1[0],
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[expansion_length]);
+      WebRtcSpl_AffineTransformVector(temp_1.get(),
                                       const_cast<int16_t*>(vector2),
                                       amplitude_ratio,
                                       4096,
                                       13,
                                       expansion_length);
+      parameters.expand_vector1.OverwriteAt(temp_1.get(), expansion_length, 0);
     } else {
       // Energy change constraint not fulfilled. Only use last vector.
       parameters.expand_vector0.Clear();
@@ -582,13 +604,6 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     }
 
     // Calculate the LPC and the gain of the filters.
-    // Calculate scale value needed for auto-correlation.
-    correlation_scale = WebRtcSpl_MaxAbsValueW16(
-        &(audio_history[signal_length - fs_mult_lpc_analysis_len]),
-        fs_mult_lpc_analysis_len);
-
-    correlation_scale = std::min(16 - WebRtcSpl_NormW32(correlation_scale), 0);
-    correlation_scale = std::max(correlation_scale * 2 + 7, 0);
 
     // Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function.
     size_t temp_index = signal_length - fs_mult_lpc_analysis_len -
@@ -601,11 +616,9 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     memcpy(&temp_signal[kUnvoicedLpcOrder],
            &audio_history[temp_index + kUnvoicedLpcOrder],
            sizeof(int16_t) * fs_mult_lpc_analysis_len);
-    WebRtcSpl_CrossCorrelation(auto_correlation,
-                               &temp_signal[kUnvoicedLpcOrder],
-                               &temp_signal[kUnvoicedLpcOrder],
-                               fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1,
-                               correlation_scale, -1);
+    CrossCorrelationWithAutoShift(
+        &temp_signal[kUnvoicedLpcOrder], &temp_signal[kUnvoicedLpcOrder],
+        fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, -1, auto_correlation);
     delete [] temp_signal;
 
     // Verify that variance is positive.
@@ -766,8 +779,7 @@ Expand::ChannelParameters::ChannelParameters()
 
 void Expand::Correlation(const int16_t* input,
                          size_t input_length,
-                         int16_t* output,
-                         int* output_scale) const {
+                         int16_t* output) const {
   // Set parameters depending on sample rate.
   const int16_t* filter_coefficients;
   size_t num_coefficients;
@@ -814,13 +826,11 @@ void Expand::Correlation(const int16_t* input,
                               downsampled_input, norm_shift);
 
   int32_t correlation[kNumCorrelationLags];
-  static const int kCorrelationShift = 6;
-  WebRtcSpl_CrossCorrelation(
-      correlation,
+  CrossCorrelationWithAutoShift(
       &downsampled_input[kDownsampledLength - kCorrelationLength],
       &downsampled_input[kDownsampledLength - kCorrelationLength
           - kCorrelationStartLag],
-      kCorrelationLength, kNumCorrelationLags, kCorrelationShift, -1);
+      kCorrelationLength, kNumCorrelationLags, -1, correlation);
 
   // Normalize and move data from 32-bit to 16-bit vector.
   int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
@@ -829,8 +839,6 @@ void Expand::Correlation(const int16_t* input,
       std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
   WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
                                    norm_shift2);
-  // Total scale factor (right shifts) of correlation value.
-  *output_scale = 2 * norm_shift + kCorrelationShift + norm_shift2;
 }
 
 void Expand::UpdateLagIndex() {
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.h
index 7f61bf3b18c..0feba3693a1 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand.h
@@ -62,6 +62,10 @@ class Expand {
     return channel_parameters_[channel].mute_factor;
   }
 
+  // Returns true if expansion has been faded down to zero amplitude (for all
+  // channels); false otherwise.
+  bool Muted() const;
+
   // Accessors and mutators.
   virtual size_t overlap_length() const;
   size_t max_lag() const { return max_lag_; }
@@ -120,12 +124,10 @@ class Expand {
 
   // Calculate the auto-correlation of |input|, with length |input_length|
   // samples. The correlation is calculated from a downsampled version of
-  // |input|, and is written to |output|. The scale factor is written to
-  // |output_scale|.
+  // |input|, and is written to |output|.
   void Correlation(const int16_t* input,
                    size_t input_length,
-                   int16_t* output,
-                   int* output_scale) const;
+                   int16_t* output) const;
 
   void UpdateLagIndex();
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand_unittest.cc
index 1441704102d..f19487ab17d 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/expand_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/expand_unittest.cc
@@ -93,8 +93,9 @@ class ExpandTest : public ::testing::Test {
     ASSERT_TRUE(input_file_.Seek(speech_start_samples));
 
     // Pre-load the sync buffer with speech data.
-    ASSERT_TRUE(
-        input_file_.Read(sync_buffer_.Size(), &sync_buffer_.Channel(0)[0]));
+    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
+    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
+    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
     ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
   }
 
@@ -169,6 +170,37 @@ TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
             statistics_.last_outage_duration_ms());
 }
 
+namespace {
+// Runs expand until Muted() returns true. Times out after 1000 calls.
+void ExpandUntilMuted(size_t num_channels, Expand* expand) {
+  EXPECT_FALSE(expand->Muted()) << "Instance is muted from the start";
+  AudioMultiVector output(num_channels);
+  int num_calls = 0;
+  while (!expand->Muted()) {
+    ASSERT_LT(num_calls++, 1000) << "Test timed out";
+    EXPECT_EQ(0, expand->Process(&output));
+  }
+}
+}  // namespace
+
+// Verifies that Muted() returns true after a long expand period. Also verifies
+// that Muted() is reset to false after calling Reset(),
+// SetParametersForMergeAfterExpand() and SetParametersForNormalAfterExpand().
+TEST_F(ExpandTest, Muted) {
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.Reset();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  expand_.Reset();  // Must reset in order to start a new expand period.
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForNormalAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+}
+
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/include/neteq.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/include/neteq.h
index 89b0c543244..3a9de1d2606 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -93,6 +93,7 @@ class NetEq {
     BackgroundNoiseMode background_noise_mode;
     NetEqPlayoutMode playout_mode;
     bool enable_fast_accelerate;
+    bool enable_muted_state = false;
   };
 
   enum ReturnCodes {
@@ -161,8 +162,12 @@ class NetEq {
   // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
   // |vad_activity_| are updated upon success. If an error is returned, some
   // fields may not have been updated.
+  // If muted state is enabled (through Config::enable_muted_state), |muted|
+  // may be set to true after a prolonged expand period. When this happens, the
+  // |data_| in |audio_frame| is not written, but should be interpreted as being
+  // all zeros.
   // Returns kOK on success, or kFail in case of an error.
-  virtual int GetAudio(AudioFrame* audio_frame) = 0;
+  virtual int GetAudio(AudioFrame* audio_frame, bool* muted) = 0;
 
   // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
   // information in the codec database. Returns 0 on success, -1 on failure.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.cc
index 9aed91f7887..299682f60d4 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.cc
@@ -18,6 +18,7 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
@@ -38,6 +39,8 @@ Merge::Merge(int fs_hz,
   assert(num_channels_ > 0);
 }
 
+Merge::~Merge() = default;
+
 size_t Merge::Process(int16_t* input, size_t input_length,
                       int16_t* external_mute_factor_array,
                       AudioMultiVector* output) {
@@ -60,13 +63,16 @@ size_t Merge::Process(int16_t* input, size_t input_length,
   size_t best_correlation_index = 0;
   size_t output_length = 0;
 
+  std::unique_ptr<int16_t[]> input_channel(
+      new int16_t[input_length_per_channel]);
+  std::unique_ptr<int16_t[]> expanded_channel(new int16_t[expanded_length]);
   for (size_t channel = 0; channel < num_channels_; ++channel) {
-    int16_t* input_channel = &input_vector[channel][0];
-    int16_t* expanded_channel = &expanded_[channel][0];
-    int16_t expanded_max, input_max;
+    input_vector[channel].CopyTo(
+        input_length_per_channel, 0, input_channel.get());
+    expanded_[channel].CopyTo(expanded_length, 0, expanded_channel.get());
+
     int16_t new_mute_factor = SignalScaling(
-        input_channel, input_length_per_channel, expanded_channel,
-        &expanded_max, &input_max);
+        input_channel.get(), input_length_per_channel, expanded_channel.get());
 
     // Adjust muting factor (product of "main" muting factor and expand muting
     // factor).
@@ -84,18 +90,16 @@ size_t Merge::Process(int16_t* input, size_t input_length,
       // Downsample, correlate, and find strongest correlation period for the
       // master (i.e., first) channel only.
       // Downsample to 4kHz sample rate.
-      Downsample(input_channel, input_length_per_channel, expanded_channel,
-                 expanded_length);
+      Downsample(input_channel.get(), input_length_per_channel,
+                 expanded_channel.get(), expanded_length);
 
       // Calculate the lag of the strongest correlation period.
       best_correlation_index = CorrelateAndPeakSearch(
-          expanded_max, input_max, old_length,
-          input_length_per_channel, expand_period);
+          old_length, input_length_per_channel, expand_period);
     }
 
-    static const int kTempDataSize = 3600;
-    int16_t temp_data[kTempDataSize];  // TODO(hlundin) Remove this.
-    int16_t* decoded_output = temp_data + best_correlation_index;
+    temp_data_.resize(input_length_per_channel + best_correlation_index);
+    int16_t* decoded_output = temp_data_.data() + best_correlation_index;
 
     // Mute the new decoded data if needed (and unmute it linearly).
     // This is the overlapping part of expanded_signal.
@@ -109,7 +113,7 @@ size_t Merge::Process(int16_t* input, size_t input_length,
       // and so on.
       int increment = 4194 / fs_mult_;
       *external_mute_factor =
-          static_cast<int16_t>(DspHelper::RampSignal(input_channel,
+          static_cast<int16_t>(DspHelper::RampSignal(input_channel.get(),
                                                      interpolation_length,
                                                      *external_mute_factor,
                                                      increment));
@@ -129,10 +133,10 @@ size_t Merge::Process(int16_t* input, size_t input_length,
     int16_t increment =
         static_cast<int16_t>(16384 / (interpolation_length + 1));  // In Q14.
     int16_t mute_factor = 16384 - increment;
-    memmove(temp_data, expanded_channel,
+    memmove(temp_data_.data(), expanded_channel.get(),
             sizeof(int16_t) * best_correlation_index);
     DspHelper::CrossFade(&expanded_channel[best_correlation_index],
-                         input_channel, interpolation_length,
+                         input_channel.get(), interpolation_length,
                          &mute_factor, increment, decoded_output);
 
     output_length = best_correlation_index + input_length_per_channel;
@@ -142,8 +146,7 @@ size_t Merge::Process(int16_t* input, size_t input_length,
     } else {
       assert(output->Size() == output_length);
     }
-    memcpy(&(*output)[channel][0], temp_data,
-           sizeof(temp_data[0]) * output_length);
+    (*output)[channel].OverwriteAt(temp_data_.data(), output_length, 0);
   }
 
   // Copy back the first part of the data to |sync_buffer_| and remove it from
@@ -204,29 +207,26 @@ size_t Merge::GetExpandedSignal(size_t* old_length, size_t* expand_period) {
 }
 
 int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
-                             const int16_t* expanded_signal,
-                             int16_t* expanded_max, int16_t* input_max) const {
+                             const int16_t* expanded_signal) const {
   // Adjust muting factor if new vector is more or less of the BGN energy.
   const size_t mod_input_length =
       std::min(static_cast<size_t>(64 * fs_mult_), input_length);
-  *expanded_max = WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
-  *input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
-
-  // Calculate energy of expanded signal.
-  // |log_fs_mult| is log2(fs_mult_), but is not exact for 48000 Hz.
-  int log_fs_mult = 30 - WebRtcSpl_NormW32(fs_mult_);
-  int expanded_shift = 6 + log_fs_mult
-      - WebRtcSpl_NormW32(*expanded_max * *expanded_max);
-  expanded_shift = std::max(expanded_shift, 0);
+  const int16_t expanded_max =
+      WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
+  int32_t factor = (expanded_max * expanded_max) /
+      (std::numeric_limits<int32_t>::max() /
+          static_cast<int32_t>(mod_input_length));
+  const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
   int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
                                                           expanded_signal,
                                                           mod_input_length,
                                                           expanded_shift);
 
   // Calculate energy of input signal.
-  int input_shift = 6 + log_fs_mult -
-      WebRtcSpl_NormW32(*input_max * *input_max);
-  input_shift = std::max(input_shift, 0);
+  const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
+  factor = (input_max * input_max) / (std::numeric_limits<int32_t>::max() /
+      static_cast<int32_t>(mod_input_length));
+  const int input_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
   int32_t energy_input = WebRtcSpl_DotProductWithScale(input, input,
                                                        mod_input_length,
                                                        input_shift);
@@ -307,22 +307,17 @@ void Merge::Downsample(const int16_t* input, size_t input_length,
   }
 }
 
-size_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
-                                     size_t start_position, size_t input_length,
+size_t Merge::CorrelateAndPeakSearch(size_t start_position, size_t input_length,
                                      size_t expand_period) const {
   // Calculate correlation without any normalization.
   const size_t max_corr_length = kMaxCorrelationLength;
   size_t stop_position_downsamp =
       std::min(max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
-  int correlation_shift = 0;
-  if (expanded_max * input_max > 26843546) {
-    correlation_shift = 3;
-  }
 
   int32_t correlation[kMaxCorrelationLength];
-  WebRtcSpl_CrossCorrelation(correlation, input_downsampled_,
-                             expanded_downsampled_, kInputDownsampLength,
-                             stop_position_downsamp, correlation_shift, 1);
+  CrossCorrelationWithAutoShift(input_downsampled_, expanded_downsampled_,
+                                kInputDownsampLength, stop_position_downsamp, 1,
+                                correlation);
 
   // Normalize correlation to 14 bits and copy to a 16-bit array.
   const size_t pad_length = expand_->overlap_length() - 1;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h
index a168502c271..48f09a16727 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h
@@ -37,7 +37,7 @@ class Merge {
         size_t num_channels,
         Expand* expand,
         SyncBuffer* sync_buffer);
-  virtual ~Merge() {}
+  virtual ~Merge();
 
   // The main method to produce the audio data. The decoded data is supplied in
   // |input|, having |input_length| samples in total for all channels
@@ -69,11 +69,10 @@ class Merge {
   // of samples that were taken from the |sync_buffer_|.
   size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
 
-  // Analyzes |input| and |expanded_signal| to find maximum values. Returns
-  // a muting factor (Q14) to be used on the new data.
+  // Analyzes |input| and |expanded_signal| and returns muting factor (Q14) to
+  // be used on the new data.
   int16_t SignalScaling(const int16_t* input, size_t input_length,
-                        const int16_t* expanded_signal,
-                        int16_t* expanded_max, int16_t* input_max) const;
+                        const int16_t* expanded_signal) const;
 
   // Downsamples |input| (|input_length| samples) and |expanded_signal| to
   // 4 kHz sample rate. The downsampled signals are written to
@@ -84,8 +83,7 @@ class Merge {
   // Calculates cross-correlation between |input_downsampled_| and
   // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
   // lag is returned.
-  size_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
-                                size_t start_position, size_t input_length,
+  size_t CorrelateAndPeakSearch(size_t start_position, size_t input_length,
                                 size_t expand_period) const;
 
   const int fs_mult_;  // fs_hz_ / 8000.
@@ -95,6 +93,7 @@ class Merge {
   int16_t expanded_downsampled_[kExpandDownsampLength];
   int16_t input_downsampled_[kInputDownsampLength];
   AudioMultiVector expanded_;
+  std::vector<int16_t> temp_data_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Merge);
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
index 1b4a3c9da5b..60ae0f6501e 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -21,6 +21,7 @@ namespace webrtc {
 
 class MockDecoderDatabase : public DecoderDatabase {
  public:
+  MockDecoderDatabase() : DecoderDatabase(nullptr) {}
   virtual ~MockDecoderDatabase() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_CONST_METHOD0(Empty,
@@ -59,7 +60,7 @@ class MockDecoderDatabase : public DecoderDatabase {
   MOCK_METHOD1(SetActiveCngDecoder,
       int(uint8_t rtp_payload_type));
   MOCK_METHOD0(GetActiveCngDecoder,
-      AudioDecoder*());
+      ComfortNoiseDecoder*());
   MOCK_CONST_METHOD1(CheckPayloadTypes,
       int(const PacketList& packet_list));
 };
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
index 6fb85854d77..7ceea70621f 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
@@ -20,8 +20,9 @@ namespace webrtc {
 class MockDelayManager : public DelayManager {
  public:
   MockDelayManager(size_t max_packets_in_buffer,
-                   DelayPeakDetector* peak_detector)
-      : DelayManager(max_packets_in_buffer, peak_detector) {}
+                   DelayPeakDetector* peak_detector,
+                   const TickTimer* tick_timer)
+      : DelayManager(max_packets_in_buffer, peak_detector, tick_timer) {}
   virtual ~MockDelayManager() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_CONST_METHOD0(iat_vector,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
index fa5cd7ed061..5564fba312c 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
@@ -19,15 +19,16 @@ namespace webrtc {
 
 class MockDelayPeakDetector : public DelayPeakDetector {
  public:
+  MockDelayPeakDetector(const TickTimer* tick_timer)
+      : DelayPeakDetector(tick_timer) {}
   virtual ~MockDelayPeakDetector() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_METHOD0(Reset, void());
   MOCK_METHOD1(SetPacketAudioLength, void(int length_ms));
   MOCK_METHOD0(peak_found, bool());
   MOCK_CONST_METHOD0(MaxPeakHeight, int());
-  MOCK_CONST_METHOD0(MaxPeakPeriod, int());
+  MOCK_CONST_METHOD0(MaxPeakPeriod, uint64_t());
   MOCK_METHOD2(Update, bool(int inter_arrival_time, int target_level));
-  MOCK_METHOD1(IncrementCounter, void(int inc_ms));
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
index 97e54d83a5e..6bb95901d8c 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -19,8 +19,8 @@ namespace webrtc {
 
 class MockPacketBuffer : public PacketBuffer {
  public:
-  MockPacketBuffer(size_t max_number_of_packets)
-      : PacketBuffer(max_number_of_packets) {}
+  MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
+      : PacketBuffer(max_number_of_packets, tick_timer) {}
   virtual ~MockPacketBuffer() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_METHOD0(Flush,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.cc
index c31dbdc1a3c..2d1ce724cab 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.cc
@@ -10,21 +10,10 @@
 
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
+#include <memory>
 #include <sstream>
 
-#include "webrtc/modules/audio_coding/neteq/accelerate.h"
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
-#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/payload_splitter.h"
-#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
-#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
 
 namespace webrtc {
 
@@ -37,41 +26,16 @@ std::string NetEq::Config::ToString() const {
      << ", max_packets_in_buffer=" << max_packets_in_buffer
      << ", background_noise_mode=" << background_noise_mode
      << ", playout_mode=" << playout_mode
-     << ", enable_fast_accelerate=" << enable_fast_accelerate;
+     << ", enable_fast_accelerate="
+     << (enable_fast_accelerate ? " true": "false")
+     << ", enable_muted_state=" << (enable_muted_state ? " true": "false");
   return ss.str();
 }
 
 // Creates all classes needed and inject them into a new NetEqImpl object.
 // Return the new object.
 NetEq* NetEq::Create(const NetEq::Config& config) {
-  BufferLevelFilter* buffer_level_filter = new BufferLevelFilter;
-  DecoderDatabase* decoder_database = new DecoderDatabase;
-  DelayPeakDetector* delay_peak_detector = new DelayPeakDetector;
-  DelayManager* delay_manager =
-      new DelayManager(config.max_packets_in_buffer, delay_peak_detector);
-  delay_manager->SetMaximumDelay(config.max_delay_ms);
-  DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
-  DtmfToneGenerator* dtmf_tone_generator = new DtmfToneGenerator;
-  PacketBuffer* packet_buffer = new PacketBuffer(config.max_packets_in_buffer);
-  PayloadSplitter* payload_splitter = new PayloadSplitter;
-  TimestampScaler* timestamp_scaler = new TimestampScaler(*decoder_database);
-  AccelerateFactory* accelerate_factory = new AccelerateFactory;
-  ExpandFactory* expand_factory = new ExpandFactory;
-  PreemptiveExpandFactory* preemptive_expand_factory =
-      new PreemptiveExpandFactory;
-  return new NetEqImpl(config,
-                       buffer_level_filter,
-                       decoder_database,
-                       delay_manager,
-                       delay_peak_detector,
-                       dtmf_buffer,
-                       dtmf_tone_generator,
-                       packet_buffer,
-                       payload_splitter,
-                       timestamp_scaler,
-                       accelerate_factory,
-                       expand_factory,
-                       preemptive_expand_factory);
+  return new NetEqImpl(config, NetEqImpl::Dependencies(config));
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi
index ead9586f5ce..e92567eef5b 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -51,6 +51,8 @@
       'dependencies': [
         '<@(neteq_dependencies)',
         '<(webrtc_root)/common.gyp:webrtc_common',
+        'builtin_audio_decoder_factory',
+        'rent_a_codec',
       ],
       'defines': [
         '<@(neteq_defines)',
@@ -73,6 +75,8 @@
         'buffer_level_filter.h',
         'comfort_noise.cc',
         'comfort_noise.h',
+        'cross_correlation.cc',
+        'cross_correlation.h',
         'decision_logic.cc',
         'decision_logic.h',
         'decision_logic_fax.cc',
@@ -105,6 +109,8 @@
         'statistics_calculator.h',
         'normal.cc',
         'normal.h',
+        'packet.cc',
+        'packet.h',
         'packet_buffer.cc',
         'packet_buffer.h',
         'payload_splitter.cc',
@@ -119,6 +125,8 @@
         'rtcp.h',
         'sync_buffer.cc',
         'sync_buffer.h',
+        'tick_timer.cc',
+        'tick_timer.h',
         'timestamp_scaler.cc',
         'timestamp_scaler.h',
         'time_stretch.cc',
@@ -206,19 +214,6 @@
           ],
         }, # neteq_unittest_tools
       ], # targets
-      'conditions': [
-        ['OS=="android"', {
-          'targets': [
-            {
-              'target_name': 'audio_decoder_unittests_apk_target',
-              'type': 'none',
-              'dependencies': [
-                '<(apk_tests_path):audio_decoder_unittests_apk',
-              ],
-            },
-          ],
-        }],
-      ],
     }], # include_tests
   ], # conditions
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 50c24a3b73a..25fa1a7365c 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -189,7 +189,9 @@ class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest,
 
   void GetAndVerifyOutput() override {
     // Get audio from internal decoder instance.
-    EXPECT_EQ(NetEq::kOK, neteq_internal_->GetAudio(&output_internal_));
+    bool muted;
+    EXPECT_EQ(NetEq::kOK, neteq_internal_->GetAudio(&output_internal_, &muted));
+    ASSERT_FALSE(muted);
     EXPECT_EQ(1u, output_internal_.num_channels_);
     EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
               output_internal_.samples_per_channel_);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index db37e716d66..7f8661bae89 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -14,6 +14,7 @@
 #include <memory.h>  // memset
 
 #include <algorithm>
+#include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
@@ -21,6 +22,7 @@
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/accelerate.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
@@ -42,6 +44,7 @@
 #include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
 #include "webrtc/modules/include/module_common_types.h"
 
@@ -52,33 +55,43 @@
 
 namespace webrtc {
 
+NetEqImpl::Dependencies::Dependencies(const NetEq::Config& config)
+    : tick_timer(new TickTimer),
+      buffer_level_filter(new BufferLevelFilter),
+      decoder_database(new DecoderDatabase(CreateBuiltinAudioDecoderFactory())),
+      delay_peak_detector(new DelayPeakDetector(tick_timer.get())),
+      delay_manager(new DelayManager(config.max_packets_in_buffer,
+                                     delay_peak_detector.get(),
+                                     tick_timer.get())),
+      dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
+      dtmf_tone_generator(new DtmfToneGenerator),
+      packet_buffer(
+          new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())),
+      payload_splitter(new PayloadSplitter),
+      timestamp_scaler(new TimestampScaler(*decoder_database)),
+      accelerate_factory(new AccelerateFactory),
+      expand_factory(new ExpandFactory),
+      preemptive_expand_factory(new PreemptiveExpandFactory) {}
+
+NetEqImpl::Dependencies::~Dependencies() = default;
+
 NetEqImpl::NetEqImpl(const NetEq::Config& config,
-                     BufferLevelFilter* buffer_level_filter,
-                     DecoderDatabase* decoder_database,
-                     DelayManager* delay_manager,
-                     DelayPeakDetector* delay_peak_detector,
-                     DtmfBuffer* dtmf_buffer,
-                     DtmfToneGenerator* dtmf_tone_generator,
-                     PacketBuffer* packet_buffer,
-                     PayloadSplitter* payload_splitter,
-                     TimestampScaler* timestamp_scaler,
-                     AccelerateFactory* accelerate_factory,
-                     ExpandFactory* expand_factory,
-                     PreemptiveExpandFactory* preemptive_expand_factory,
+                     Dependencies&& deps,
                      bool create_components)
-    : buffer_level_filter_(buffer_level_filter),
-      decoder_database_(decoder_database),
-      delay_manager_(delay_manager),
-      delay_peak_detector_(delay_peak_detector),
-      dtmf_buffer_(dtmf_buffer),
-      dtmf_tone_generator_(dtmf_tone_generator),
-      packet_buffer_(packet_buffer),
-      payload_splitter_(payload_splitter),
-      timestamp_scaler_(timestamp_scaler),
+    : tick_timer_(std::move(deps.tick_timer)),
+      buffer_level_filter_(std::move(deps.buffer_level_filter)),
+      decoder_database_(std::move(deps.decoder_database)),
+      delay_manager_(std::move(deps.delay_manager)),
+      delay_peak_detector_(std::move(deps.delay_peak_detector)),
+      dtmf_buffer_(std::move(deps.dtmf_buffer)),
+      dtmf_tone_generator_(std::move(deps.dtmf_tone_generator)),
+      packet_buffer_(std::move(deps.packet_buffer)),
+      payload_splitter_(std::move(deps.payload_splitter)),
+      timestamp_scaler_(std::move(deps.timestamp_scaler)),
       vad_(new PostDecodeVad()),
-      expand_factory_(expand_factory),
-      accelerate_factory_(accelerate_factory),
-      preemptive_expand_factory_(preemptive_expand_factory),
+      expand_factory_(std::move(deps.expand_factory)),
+      accelerate_factory_(std::move(deps.accelerate_factory)),
+      preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
       last_mode_(kModeNormal),
       decoded_buffer_length_(kMaxFrameSize),
       decoded_buffer_(new int16_t[decoded_buffer_length_]),
@@ -95,7 +108,8 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
       background_noise_mode_(config.background_noise_mode),
       playout_mode_(config.playout_mode),
       enable_fast_accelerate_(config.enable_fast_accelerate),
-      nack_enabled_(false) {
+      nack_enabled_(false),
+      enable_muted_state_(config.enable_muted_state) {
   LOG(LS_INFO) << "NetEq config: " << config.ToString();
   int fs = config.sample_rate_hz;
   if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@@ -103,6 +117,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
         "Changing to 8000 Hz.";
     fs = 8000;
   }
+  delay_manager_->SetMaximumDelay(config.max_delay_ms);
   fs_hz_ = fs;
   fs_mult_ = fs / 8000;
   last_output_sample_rate_hz_ = fs;
@@ -191,10 +206,10 @@ void SetAudioFrameActivityAndType(bool vad_enabled,
 }
 }  // namespace
 
-int NetEqImpl::GetAudio(AudioFrame* audio_frame) {
+int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted) {
   TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
   rtc::CritScope lock(&crit_sect_);
-  int error = GetAudioInternal(audio_frame);
+  int error = GetAudioInternal(audio_frame, muted);
   RTC_DCHECK_EQ(
       audio_frame->sample_rate_hz_,
       rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100));
@@ -487,6 +502,11 @@ const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
   return sync_buffer_.get();
 }
 
+Operations NetEqImpl::last_operation_for_test() const {
+  rtc::CritScope lock(&crit_sect_);
+  return last_operation_;
+}
+
 // Methods below this line are private.
 
 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
@@ -532,7 +552,8 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     packet->header.numCSRCs = 0;
     packet->payload_length = payload.size();
     packet->primary = true;
-    packet->waiting_time = 0;
+    // Waiting time will be set upon inserting the packet in the buffer.
+    RTC_DCHECK(!packet->waiting_time);
     packet->payload = new uint8_t[packet->payload_length];
     packet->sync_packet = is_sync_packet;
     if (!packet->payload) {
@@ -664,13 +685,15 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     }
   }
 
-  // Update bandwidth estimate, if the packet is not sync-packet.
-  if (!packet_list.empty() && !packet_list.front()->sync_packet) {
+  // Update bandwidth estimate, if the packet is not sync-packet nor comfort
+  // noise.
+  if (!packet_list.empty() && !packet_list.front()->sync_packet &&
+      !decoder_database_->IsComfortNoise(main_header.payloadType)) {
     // The list can be empty here if we got nothing but DTMF payloads.
     AudioDecoder* decoder =
         decoder_database_->GetDecoder(main_header.payloadType);
     assert(decoder);  // Should always get a valid object, since we have
-                      // already checked that the payload types are known.
+    // already checked that the payload types are known.
     decoder->IncomingPacket(packet_list.front()->payload,
                             packet_list.front()->payload_length,
                             packet_list.front()->header.sequenceNumber,
@@ -728,14 +751,18 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader();
     assert(rtp_header);
     int payload_type = rtp_header->payloadType;
-    AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type);
-    assert(decoder);  // Payloads are already checked to be valid.
+    size_t channels = 1;
+    if (!decoder_database_->IsComfortNoise(payload_type)) {
+      AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type);
+      assert(decoder);  // Payloads are already checked to be valid.
+      channels = decoder->Channels();
+    }
     const DecoderDatabase::DecoderInfo* decoder_info =
         decoder_database_->GetDecoderInfo(payload_type);
     assert(decoder_info);
     if (decoder_info->fs_hz != fs_hz_ ||
-        decoder->Channels() != algorithm_buffer_->Channels()) {
-      SetSampleRateAndChannels(decoder_info->fs_hz, decoder->Channels());
+        channels != algorithm_buffer_->Channels()) {
+      SetSampleRateAndChannels(decoder_info->fs_hz, channels);
     }
     if (nack_enabled_) {
       RTC_DCHECK(nack_);
@@ -783,11 +810,32 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   return 0;
 }
 
-int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
+int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
   PacketList packet_list;
   DtmfEvent dtmf_event;
   Operations operation;
   bool play_dtmf;
+  *muted = false;
+  tick_timer_->Increment();
+  stats_.IncreaseCounter(output_size_samples_, fs_hz_);
+
+  // Check for muted state.
+  if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {
+    RTC_DCHECK_EQ(last_mode_, kModeExpand);
+    playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
+    audio_frame->sample_rate_hz_ = fs_hz_;
+    audio_frame->samples_per_channel_ = output_size_samples_;
+    audio_frame->timestamp_ =
+        first_packet_
+            ? 0
+            : timestamp_scaler_->ToExternal(playout_timestamp_) -
+                  static_cast<uint32_t>(audio_frame->samples_per_channel_);
+    audio_frame->num_channels_ = sync_buffer_->Channels();
+    stats_.ExpandedNoiseSamples(output_size_samples_);
+    *muted = true;
+    return 0;
+  }
+
   int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
                                  &play_dtmf);
   if (return_value != 0) {
@@ -806,6 +854,11 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
   vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
                sid_frame_available, fs_hz_);
 
+  if (sid_frame_available || speech_type == AudioDecoder::kComfortNoise) {
+    // Start a new stopwatch since we are decoding a new CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+
   algorithm_buffer_->Clear();
   switch (operation) {
     case kNormal: {
@@ -884,6 +937,7 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
       return kInvalidOperation;
     }
   }  // End of switch.
+  last_operation_ = operation;
   if (return_value < 0) {
     return return_value;
   }
@@ -978,6 +1032,12 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
           : timestamp_scaler_->ToExternal(playout_timestamp_) -
                 static_cast<uint32_t>(audio_frame->samples_per_channel_);
 
+  if (!(last_mode_ == kModeRfc3389Cng ||
+      last_mode_ == kModeCodecInternalCng ||
+      last_mode_ == kModeExpand)) {
+    generated_noise_stopwatch_.reset();
+  }
+
   if (decode_return_value) return decode_return_value;
   return return_value;
 }
@@ -990,10 +1050,6 @@ int NetEqImpl::GetDecision(Operations* operation,
   *play_dtmf = false;
   *operation = kUndefined;
 
-  // Increment time counters.
-  packet_buffer_->IncrementWaitingTimes();
-  stats_.IncreaseCounter(output_size_samples_, fs_hz_);
-
   assert(sync_buffer_.get());
   uint32_t end_timestamp = sync_buffer_->end_timestamp();
   if (!new_codec_) {
@@ -1002,14 +1058,22 @@ int NetEqImpl::GetDecision(Operations* operation,
   }
   const RTPHeader* header = packet_buffer_->NextRtpHeader();
 
+  RTC_DCHECK(!generated_noise_stopwatch_ ||
+             generated_noise_stopwatch_->ElapsedTicks() >= 1);
+  uint64_t generated_noise_samples =
+      generated_noise_stopwatch_
+          ? (generated_noise_stopwatch_->ElapsedTicks() - 1) *
+                    output_size_samples_ +
+                decision_logic_->noise_fast_forward()
+          : 0;
+
   if (decision_logic_->CngRfc3389On() || last_mode_ == kModeRfc3389Cng) {
     // Because of timestamp peculiarities, we have to "manually" disallow using
     // a CNG packet with the same timestamp as the one that was last played.
     // This can happen when using redundancy and will cause the timing to shift.
     while (header && decoder_database_->IsComfortNoise(header->payloadType) &&
            (end_timestamp >= header->timestamp ||
-            end_timestamp + decision_logic_->generated_noise_samples() >
-                header->timestamp)) {
+            end_timestamp + generated_noise_samples > header->timestamp)) {
       // Don't use this packet, discard it.
       if (packet_buffer_->DiscardNextPacket() != PacketBuffer::kOK) {
         assert(false);  // Must be ok by design.
@@ -1037,7 +1101,7 @@ int NetEqImpl::GetDecision(Operations* operation,
   // Check if it is time to play a DTMF event.
   if (dtmf_buffer_->GetEvent(
       static_cast<uint32_t>(
-          end_timestamp + decision_logic_->generated_noise_samples()),
+          end_timestamp + generated_noise_samples),
       dtmf_event)) {
     *play_dtmf = true;
   }
@@ -1045,13 +1109,14 @@ int NetEqImpl::GetDecision(Operations* operation,
   // Get instruction.
   assert(sync_buffer_.get());
   assert(expand_.get());
-  *operation = decision_logic_->GetDecision(*sync_buffer_,
-                                            *expand_,
-                                            decoder_frame_length_,
-                                            header,
-                                            last_mode_,
-                                            *play_dtmf,
-                                            &reset_decoder_);
+  generated_noise_samples =
+      generated_noise_stopwatch_
+          ? generated_noise_stopwatch_->ElapsedTicks() * output_size_samples_ +
+                decision_logic_->noise_fast_forward()
+          : 0;
+  *operation = decision_logic_->GetDecision(
+      *sync_buffer_, *expand_, decoder_frame_length_, header, last_mode_,
+      *play_dtmf, generated_noise_samples, &reset_decoder_);
 
   // Check if we already have enough samples in the |sync_buffer_|. If so,
   // change decision to normal, unless the decision was merge, accelerate, or
@@ -1124,15 +1189,19 @@ int NetEqImpl::GetDecision(Operations* operation,
       // TODO(hlundin): Write test for this.
       // Update timestamp.
       timestamp_ = end_timestamp;
-      if (decision_logic_->generated_noise_samples() > 0 &&
-          last_mode_ != kModeDtmf) {
+      const uint64_t generated_noise_samples =
+          generated_noise_stopwatch_
+              ? generated_noise_stopwatch_->ElapsedTicks() *
+                        output_size_samples_ +
+                    decision_logic_->noise_fast_forward()
+              : 0;
+      if (generated_noise_samples > 0 && last_mode_ != kModeDtmf) {
         // Make a jump in timestamp due to the recently played comfort noise.
         uint32_t timestamp_jump =
-            static_cast<uint32_t>(decision_logic_->generated_noise_samples());
+            static_cast<uint32_t>(generated_noise_samples);
         sync_buffer_->IncreaseEndTimestamp(timestamp_jump);
         timestamp_ += timestamp_jump;
       }
-      decision_logic_->set_generated_noise_samples(0);
       return 0;
     }
     case kAccelerate:
@@ -1215,9 +1284,6 @@ int NetEqImpl::GetDecision(Operations* operation,
       // We are about to decode and use a non-CNG packet.
       decision_logic_->SetCngOff();
     }
-    // Reset CNG timestamp as a new packet will be delivered.
-    // (Also if this is a CNG packet, since playedOutTS is updated.)
-    decision_logic_->set_generated_noise_samples(0);
 
     extracted_samples = ExtractPackets(required_samples, packet_list);
     if (extracted_samples < 0) {
@@ -1297,7 +1363,7 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
       decoder->Reset();
 
     // Reset comfort noise decoder.
-    AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+    ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
     if (cng_decoder)
       cng_decoder->Reset();
 
@@ -1550,6 +1616,12 @@ int NetEqImpl::DoExpand(bool play_dtmf) {
   if (!play_dtmf) {
     dtmf_tone_generator_->Reset();
   }
+
+  if (!generated_noise_stopwatch_) {
+    // Start a new stopwatch since we may be covering for a lost CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+
   return 0;
 }
 
@@ -1920,8 +1992,7 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
       return -1;
     }
     stats_.PacketsDiscarded(discard_count);
-    // Store waiting time in ms; packets->waiting_time is in "output blocks".
-    stats_.StoreWaitingTime(packet->waiting_time * kOutputSizeMs);
+    stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs());
     assert(packet->payload_length > 0);
     packet_list->push_back(packet);  // Store packet in list.
 
@@ -1955,7 +2026,7 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
           stats_.SecondaryDecodedSamples(packet_duration);
         }
       }
-    } else {
+    } else if (!decoder_database_->IsComfortNoise(packet->header.payloadType)) {
       LOG(LS_WARNING) << "Unknown payload type "
                       << static_cast<int>(packet->header.payloadType);
       assert(false);
@@ -2023,7 +2094,7 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
     mute_factor_array_[i] = 16384;  // 1.0 in Q14.
   }
 
-  AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
   if (cng_decoder)
     cng_decoder->Reset();
 
@@ -2094,11 +2165,9 @@ NetEqImpl::OutputType NetEqImpl::LastOutputType() {
 }
 
 void NetEqImpl::CreateDecisionLogic() {
-  decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
-                                              playout_mode_,
-                                              decoder_database_.get(),
-                                              *packet_buffer_.get(),
-                                              delay_manager_.get(),
-                                              buffer_level_filter_.get()));
+  decision_logic_.reset(DecisionLogic::Create(
+      fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
+      *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
+      tick_timer_.get()));
 }
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 75055a7b47f..cc5550411f2 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -24,6 +24,7 @@
 #include "webrtc/modules/audio_coding/neteq/random_vector.h"
 #include "webrtc/modules/audio_coding/neteq/rtcp.h"
 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -65,21 +66,33 @@ class NetEqImpl : public webrtc::NetEq {
     kVadPassive
   };
 
-  // Creates a new NetEqImpl object. The object will assume ownership of all
-  // injected dependencies, and will delete them when done.
+  struct Dependencies {
+    // The constructor populates the Dependencies struct with the default
+    // implementations of the objects. They can all be replaced by the user
+    // before sending the struct to the NetEqImpl constructor. However, there
+    // are dependencies between some of the classes inside the struct, so
+    // swapping out one may make it necessary to re-create another one.
+    explicit Dependencies(const NetEq::Config& config);
+    ~Dependencies();
+
+    std::unique_ptr<TickTimer> tick_timer;
+    std::unique_ptr<BufferLevelFilter> buffer_level_filter;
+    std::unique_ptr<DecoderDatabase> decoder_database;
+    std::unique_ptr<DelayPeakDetector> delay_peak_detector;
+    std::unique_ptr<DelayManager> delay_manager;
+    std::unique_ptr<DtmfBuffer> dtmf_buffer;
+    std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
+    std::unique_ptr<PacketBuffer> packet_buffer;
+    std::unique_ptr<PayloadSplitter> payload_splitter;
+    std::unique_ptr<TimestampScaler> timestamp_scaler;
+    std::unique_ptr<AccelerateFactory> accelerate_factory;
+    std::unique_ptr<ExpandFactory> expand_factory;
+    std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
+  };
+
+  // Creates a new NetEqImpl object.
   NetEqImpl(const NetEq::Config& config,
-            BufferLevelFilter* buffer_level_filter,
-            DecoderDatabase* decoder_database,
-            DelayManager* delay_manager,
-            DelayPeakDetector* delay_peak_detector,
-            DtmfBuffer* dtmf_buffer,
-            DtmfToneGenerator* dtmf_tone_generator,
-            PacketBuffer* packet_buffer,
-            PayloadSplitter* payload_splitter,
-            TimestampScaler* timestamp_scaler,
-            AccelerateFactory* accelerate_factory,
-            ExpandFactory* expand_factory,
-            PreemptiveExpandFactory* preemptive_expand_factory,
+            Dependencies&& deps,
             bool create_components = true);
 
   ~NetEqImpl() override;
@@ -104,7 +117,7 @@ class NetEqImpl : public webrtc::NetEq {
   int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                        uint32_t receive_timestamp) override;
 
-  int GetAudio(AudioFrame* audio_frame) override;
+  int GetAudio(AudioFrame* audio_frame, bool* muted) override;
 
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
@@ -191,12 +204,15 @@ class NetEqImpl : public webrtc::NetEq {
 
   // This accessor method is only intended for testing purposes.
   const SyncBuffer* sync_buffer_for_test() const;
+  Operations last_operation_for_test() const;
 
  protected:
   static const int kOutputSizeMs = 10;
-  static const size_t kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
+  static const size_t kMaxFrameSize = 5760;  // 120 ms @ 48 kHz.
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
-  static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
+  // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
+  // calculating correlations of current frame against history.
+  static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
 
   // Inserts a new packet into NetEq. This is used by the InsertPacket method
   // above. Returns 0 on success, otherwise an error code.
@@ -209,7 +225,7 @@ class NetEqImpl : public webrtc::NetEq {
 
   // Delivers 10 ms of audio data. The data is written to |audio_frame|.
   // Returns 0 on success, otherwise an error code.
-  int GetAudioInternal(AudioFrame* audio_frame)
+  int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Provides a decision to the GetAudioInternal method. The decision what to
@@ -328,6 +344,7 @@ class NetEqImpl : public webrtc::NetEq {
   virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   rtc::CriticalSection crit_sect_;
+  const std::unique_ptr<TickTimer> tick_timer_ GUARDED_BY(crit_sect_);
   const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
       GUARDED_BY(crit_sect_);
   const std::unique_ptr<DecoderDatabase> decoder_database_
@@ -369,6 +386,7 @@ class NetEqImpl : public webrtc::NetEq {
   size_t output_size_samples_ GUARDED_BY(crit_sect_);
   size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
   Modes last_mode_ GUARDED_BY(crit_sect_);
+  Operations last_operation_ GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
   size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
@@ -387,8 +405,11 @@ class NetEqImpl : public webrtc::NetEq {
   bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
   std::unique_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
   bool nack_enabled_ GUARDED_BY(crit_sect_);
+  const bool enable_muted_state_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
       AudioFrame::kVadPassive;
+  std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
+      GUARDED_BY(crit_sect_);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 561c0459bfb..43db87f4fa7 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
 
@@ -54,111 +56,82 @@ int DeletePacketsAndReturnOk(PacketList* packet_list) {
 
 class NetEqImplTest : public ::testing::Test {
  protected:
-  NetEqImplTest()
-      : neteq_(NULL),
-        config_(),
-        mock_buffer_level_filter_(NULL),
-        buffer_level_filter_(NULL),
-        use_mock_buffer_level_filter_(true),
-        mock_decoder_database_(NULL),
-        decoder_database_(NULL),
-        use_mock_decoder_database_(true),
-        mock_delay_peak_detector_(NULL),
-        delay_peak_detector_(NULL),
-        use_mock_delay_peak_detector_(true),
-        mock_delay_manager_(NULL),
-        delay_manager_(NULL),
-        use_mock_delay_manager_(true),
-        mock_dtmf_buffer_(NULL),
-        dtmf_buffer_(NULL),
-        use_mock_dtmf_buffer_(true),
-        mock_dtmf_tone_generator_(NULL),
-        dtmf_tone_generator_(NULL),
-        use_mock_dtmf_tone_generator_(true),
-        mock_packet_buffer_(NULL),
-        packet_buffer_(NULL),
-        use_mock_packet_buffer_(true),
-        mock_payload_splitter_(NULL),
-        payload_splitter_(NULL),
-        use_mock_payload_splitter_(true),
-        timestamp_scaler_(NULL) {
-    config_.sample_rate_hz = 8000;
-  }
+  NetEqImplTest() { config_.sample_rate_hz = 8000; }
 
   void CreateInstance() {
+    NetEqImpl::Dependencies deps(config_);
+
+    // Get a local pointer to NetEq's TickTimer object.
+    tick_timer_ = deps.tick_timer.get();
+
     if (use_mock_buffer_level_filter_) {
-      mock_buffer_level_filter_ = new MockBufferLevelFilter;
-      buffer_level_filter_ = mock_buffer_level_filter_;
-    } else {
-      buffer_level_filter_ = new BufferLevelFilter;
+      std::unique_ptr<MockBufferLevelFilter> mock(new MockBufferLevelFilter);
+      mock_buffer_level_filter_ = mock.get();
+      deps.buffer_level_filter = std::move(mock);
     }
+    buffer_level_filter_ = deps.buffer_level_filter.get();
+
     if (use_mock_decoder_database_) {
-      mock_decoder_database_ = new MockDecoderDatabase;
+      std::unique_ptr<MockDecoderDatabase> mock(new MockDecoderDatabase);
+      mock_decoder_database_ = mock.get();
       EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder())
           .WillOnce(ReturnNull());
-      decoder_database_ = mock_decoder_database_;
-    } else {
-      decoder_database_ = new DecoderDatabase;
+      deps.decoder_database = std::move(mock);
     }
+    decoder_database_ = deps.decoder_database.get();
+
     if (use_mock_delay_peak_detector_) {
-      mock_delay_peak_detector_ = new MockDelayPeakDetector;
+      std::unique_ptr<MockDelayPeakDetector> mock(
+          new MockDelayPeakDetector(tick_timer_));
+      mock_delay_peak_detector_ = mock.get();
       EXPECT_CALL(*mock_delay_peak_detector_, Reset()).Times(1);
-      delay_peak_detector_ = mock_delay_peak_detector_;
-    } else {
-      delay_peak_detector_ = new DelayPeakDetector;
+      deps.delay_peak_detector = std::move(mock);
     }
+    delay_peak_detector_ = deps.delay_peak_detector.get();
+
     if (use_mock_delay_manager_) {
-      mock_delay_manager_ = new MockDelayManager(config_.max_packets_in_buffer,
-                                                 delay_peak_detector_);
+      std::unique_ptr<MockDelayManager> mock(new MockDelayManager(
+          config_.max_packets_in_buffer, delay_peak_detector_, tick_timer_));
+      mock_delay_manager_ = mock.get();
       EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
-      delay_manager_ = mock_delay_manager_;
-    } else {
-      delay_manager_ =
-          new DelayManager(config_.max_packets_in_buffer, delay_peak_detector_);
+      deps.delay_manager = std::move(mock);
     }
+    delay_manager_ = deps.delay_manager.get();
+
     if (use_mock_dtmf_buffer_) {
-      mock_dtmf_buffer_ = new MockDtmfBuffer(config_.sample_rate_hz);
-      dtmf_buffer_ = mock_dtmf_buffer_;
-    } else {
-      dtmf_buffer_ = new DtmfBuffer(config_.sample_rate_hz);
+      std::unique_ptr<MockDtmfBuffer> mock(
+          new MockDtmfBuffer(config_.sample_rate_hz));
+      mock_dtmf_buffer_ = mock.get();
+      deps.dtmf_buffer = std::move(mock);
     }
+    dtmf_buffer_ = deps.dtmf_buffer.get();
+
     if (use_mock_dtmf_tone_generator_) {
-      mock_dtmf_tone_generator_ = new MockDtmfToneGenerator;
-      dtmf_tone_generator_ = mock_dtmf_tone_generator_;
-    } else {
-      dtmf_tone_generator_ = new DtmfToneGenerator;
+      std::unique_ptr<MockDtmfToneGenerator> mock(new MockDtmfToneGenerator);
+      mock_dtmf_tone_generator_ = mock.get();
+      deps.dtmf_tone_generator = std::move(mock);
     }
+    dtmf_tone_generator_ = deps.dtmf_tone_generator.get();
+
     if (use_mock_packet_buffer_) {
-      mock_packet_buffer_ = new MockPacketBuffer(config_.max_packets_in_buffer);
-      packet_buffer_ = mock_packet_buffer_;
-    } else {
-      packet_buffer_ = new PacketBuffer(config_.max_packets_in_buffer);
+      std::unique_ptr<MockPacketBuffer> mock(
+          new MockPacketBuffer(config_.max_packets_in_buffer, tick_timer_));
+      mock_packet_buffer_ = mock.get();
+      deps.packet_buffer = std::move(mock);
     }
+    packet_buffer_ = deps.packet_buffer.get();
+
     if (use_mock_payload_splitter_) {
-      mock_payload_splitter_ = new MockPayloadSplitter;
-      payload_splitter_ = mock_payload_splitter_;
-    } else {
-      payload_splitter_ = new PayloadSplitter;
+      std::unique_ptr<MockPayloadSplitter> mock(new MockPayloadSplitter);
+      mock_payload_splitter_ = mock.get();
+      deps.payload_splitter = std::move(mock);
     }
-    timestamp_scaler_ = new TimestampScaler(*decoder_database_);
-    AccelerateFactory* accelerate_factory = new AccelerateFactory;
-    ExpandFactory* expand_factory = new ExpandFactory;
-    PreemptiveExpandFactory* preemptive_expand_factory =
-        new PreemptiveExpandFactory;
-
-    neteq_ = new NetEqImpl(config_,
-                           buffer_level_filter_,
-                           decoder_database_,
-                           delay_manager_,
-                           delay_peak_detector_,
-                           dtmf_buffer_,
-                           dtmf_tone_generator_,
-                           packet_buffer_,
-                           payload_splitter_,
-                           timestamp_scaler_,
-                           accelerate_factory,
-                           expand_factory,
-                           preemptive_expand_factory);
+    payload_splitter_ = deps.payload_splitter.get();
+
+    deps.timestamp_scaler = std::unique_ptr<TimestampScaler>(
+        new TimestampScaler(*deps.decoder_database.get()));
+
+    neteq_.reset(new NetEqImpl(config_, std::move(deps)));
     ASSERT_TRUE(neteq_ != NULL);
   }
 
@@ -196,36 +169,35 @@ class NetEqImplTest : public ::testing::Test {
     if (use_mock_packet_buffer_) {
       EXPECT_CALL(*mock_packet_buffer_, Die()).Times(1);
     }
-    delete neteq_;
   }
 
-  NetEqImpl* neteq_;
+  std::unique_ptr<NetEqImpl> neteq_;
   NetEq::Config config_;
-  MockBufferLevelFilter* mock_buffer_level_filter_;
-  BufferLevelFilter* buffer_level_filter_;
-  bool use_mock_buffer_level_filter_;
-  MockDecoderDatabase* mock_decoder_database_;
-  DecoderDatabase* decoder_database_;
-  bool use_mock_decoder_database_;
-  MockDelayPeakDetector* mock_delay_peak_detector_;
-  DelayPeakDetector* delay_peak_detector_;
-  bool use_mock_delay_peak_detector_;
-  MockDelayManager* mock_delay_manager_;
-  DelayManager* delay_manager_;
-  bool use_mock_delay_manager_;
-  MockDtmfBuffer* mock_dtmf_buffer_;
-  DtmfBuffer* dtmf_buffer_;
-  bool use_mock_dtmf_buffer_;
-  MockDtmfToneGenerator* mock_dtmf_tone_generator_;
-  DtmfToneGenerator* dtmf_tone_generator_;
-  bool use_mock_dtmf_tone_generator_;
-  MockPacketBuffer* mock_packet_buffer_;
-  PacketBuffer* packet_buffer_;
-  bool use_mock_packet_buffer_;
-  MockPayloadSplitter* mock_payload_splitter_;
-  PayloadSplitter* payload_splitter_;
-  bool use_mock_payload_splitter_;
-  TimestampScaler* timestamp_scaler_;
+  TickTimer* tick_timer_ = nullptr;
+  MockBufferLevelFilter* mock_buffer_level_filter_ = nullptr;
+  BufferLevelFilter* buffer_level_filter_ = nullptr;
+  bool use_mock_buffer_level_filter_ = true;
+  MockDecoderDatabase* mock_decoder_database_ = nullptr;
+  DecoderDatabase* decoder_database_ = nullptr;
+  bool use_mock_decoder_database_ = true;
+  MockDelayPeakDetector* mock_delay_peak_detector_ = nullptr;
+  DelayPeakDetector* delay_peak_detector_ = nullptr;
+  bool use_mock_delay_peak_detector_ = true;
+  MockDelayManager* mock_delay_manager_ = nullptr;
+  DelayManager* delay_manager_ = nullptr;
+  bool use_mock_delay_manager_ = true;
+  MockDtmfBuffer* mock_dtmf_buffer_ = nullptr;
+  DtmfBuffer* dtmf_buffer_ = nullptr;
+  bool use_mock_dtmf_buffer_ = true;
+  MockDtmfToneGenerator* mock_dtmf_tone_generator_ = nullptr;
+  DtmfToneGenerator* dtmf_tone_generator_ = nullptr;
+  bool use_mock_dtmf_tone_generator_ = true;
+  MockPacketBuffer* mock_packet_buffer_ = nullptr;
+  PacketBuffer* packet_buffer_ = nullptr;
+  bool use_mock_packet_buffer_ = true;
+  MockPayloadSplitter* mock_payload_splitter_ = nullptr;
+  PayloadSplitter* payload_splitter_ = nullptr;
+  bool use_mock_payload_splitter_ = true;
 };
 
 
@@ -301,8 +273,8 @@ TEST_F(NetEqImplTest, InsertPacket) {
       .WillRepeatedly(Return(&mock_decoder));
   EXPECT_CALL(*mock_decoder_database_, IsComfortNoise(kPayloadType))
       .WillRepeatedly(Return(false));  // This is not CNG.
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type = NetEqDecoder::kDecoderPCMu;
+  DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, "", 8000,
+                                    nullptr);
   EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(&info));
 
@@ -356,6 +328,9 @@ TEST_F(NetEqImplTest, InsertPacket) {
   }
 
   // Expectations for payload splitter.
+  EXPECT_CALL(*mock_payload_splitter_, SplitFec(_, _))
+      .Times(2)
+      .WillRepeatedly(Return(PayloadSplitter::kOK));
   EXPECT_CALL(*mock_payload_splitter_, SplitAudio(_, _))
       .Times(2)
       .WillRepeatedly(Return(PayloadSplitter::kOK));
@@ -466,7 +441,9 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   AudioFrame output;
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_FALSE(muted);
   ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
@@ -521,6 +498,8 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
   EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
       .WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
+      .WillRepeatedly(Return(kPayloadLengthSamples));
   int16_t dummy_output[kPayloadLengthSamples] = {0};
   // The below expectation will make the mock decoder write
   // |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
@@ -541,7 +520,8 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   AudioFrame output;
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
@@ -569,7 +549,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
                       Return(kPayloadLengthSamples)));
 
   // Pull audio once.
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
@@ -609,7 +589,8 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   AudioFrame output;
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
@@ -630,7 +611,7 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
 
   // Pull audio repeatedly and make sure we get normal output, that is not PLC.
   for (size_t i = 0; i < 3; ++i) {
-    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
     ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
     EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
     EXPECT_EQ(1u, output.num_channels_);
@@ -734,7 +715,8 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
       50 * kSampleRateKhz, 10 * kSampleRateKhz
   };
 
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   rtc::Optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
   ASSERT_TRUE(last_timestamp);
 
@@ -756,7 +738,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
     ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
     EXPECT_EQ(1u, output.num_channels_);
     EXPECT_EQ(expected_type[i - 1], output.speech_type_);
-    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
     SCOPED_TRACE("");
     verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
   }
@@ -772,7 +754,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
     ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
     EXPECT_EQ(1u, output.num_channels_);
     EXPECT_EQ(expected_type[i - 1], output.speech_type_);
-    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
     SCOPED_TRACE("");
     verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
   }
@@ -786,7 +768,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
 TEST_F(NetEqImplTest, UnsupportedDecoder) {
   UseNoMocks();
   CreateInstance();
-  static const size_t kNetEqMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
+  static const size_t kNetEqMaxFrameSize = 5760;  // 120 ms @ 48 kHz.
   static const size_t kChannels = 2;
 
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
@@ -796,7 +778,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
   const size_t kPayloadLengthSamples =
       static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
   const size_t kPayloadLengthBytes = 1;
-  uint8_t payload[kPayloadLengthBytes]= {0};
+  uint8_t payload[kPayloadLengthBytes] = {0};
   int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
   WebRtcRTPHeader rtp_header;
   rtp_header.header.payloadType = kPayloadType;
@@ -806,11 +788,15 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
 
   class MockAudioDecoder : public AudioDecoder {
    public:
-    void Reset() override {}
+    // TODO(nisse): Valid overrides commented out, because the gmock
+    // methods don't use any override declarations, and we want to avoid
+    // warnings from -Winconsistent-missing-override. See
+    // http://crbug.com/428099.
+    void Reset() /* override */ {}
     MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
     MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*,
                                      SpeechType*));
-    size_t Channels() const override { return kChannels; }
+    size_t Channels() const /* override */ { return kChannels; }
   } decoder_;
 
   const uint8_t kFirstPayloadValue = 1;
@@ -860,9 +846,10 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
             neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
 
   AudioFrame output;
+  bool muted;
   // First call to GetAudio will try to decode the "faulty" packet.
   // Expect kFail return value...
-  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
   // ... and kOtherDecoderError error code.
   EXPECT_EQ(NetEq::kOtherDecoderError, neteq_->LastError());
   // Output size and number of channels should be correct.
@@ -872,7 +859,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
 
   // Second call to GetAudio will decode the packet that is ok. No errors are
   // expected.
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
   EXPECT_EQ(kChannels, output.num_channels_);
 }
@@ -965,7 +952,8 @@ TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   AudioFrame output;
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
@@ -1057,13 +1045,14 @@ TEST_F(NetEqImplTest, DecodingError) {
   // Pull audio.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   AudioFrame output;
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
 
   // Pull audio again. Decoder fails.
-  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
   EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
@@ -1072,13 +1061,13 @@ TEST_F(NetEqImplTest, DecodingError) {
   // returned.
 
   // Pull audio again, should continue an expansion.
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
 
   // Pull audio again, should behave normal.
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
@@ -1166,13 +1155,14 @@ TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
   // Pull audio.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   AudioFrame output;
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
 
   // Pull audio again. Decoder fails.
-  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
   EXPECT_EQ(kDecoderErrorCode, neteq_->LastDecoderError());
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
@@ -1181,7 +1171,7 @@ TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
   // returned.
 
   // Pull audio again, should resume codec CNG.
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output));
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
   EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
   EXPECT_EQ(1u, output.num_channels_);
   EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
@@ -1198,4 +1188,235 @@ TEST_F(NetEqImplTest, InitialLastOutputSampleRate) {
   EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
 }
 
+TEST_F(NetEqImplTest, TickTimerIncrement) {
+  UseNoMocks();
+  CreateInstance();
+  ASSERT_TRUE(tick_timer_);
+  EXPECT_EQ(0u, tick_timer_->ticks());
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(1u, tick_timer_->ticks());
+}
+
+class Decoder120ms : public AudioDecoder {
+ public:
+  Decoder120ms(SpeechType speech_type)
+      : next_value_(1),
+        speech_type_(speech_type) {}
+
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    size_t decoded_len =
+        rtc::CheckedDivExact(sample_rate_hz, 1000) * 120 * Channels();
+    for (size_t i = 0; i < decoded_len; ++i) {
+      decoded[i] = next_value_++;
+    }
+    *speech_type = speech_type_;
+    return decoded_len;
+  }
+
+  void Reset() override { next_value_ = 1; }
+  size_t Channels() const override { return 2; }
+
+ private:
+  int16_t next_value_;
+  SpeechType speech_type_;
+};
+
+class NetEqImplTest120ms : public NetEqImplTest {
+ protected:
+  NetEqImplTest120ms() : NetEqImplTest() {}
+  virtual ~NetEqImplTest120ms() {}
+
+  void CreateInstanceNoMocks() {
+    UseNoMocks();
+    CreateInstance();
+  }
+
+  void CreateInstanceWithDelayManagerMock() {
+    UseNoMocks();
+    use_mock_delay_manager_ = true;
+    CreateInstance();
+  }
+
+  uint32_t timestamp_diff_between_packets() const {
+    return rtc::CheckedDivExact(kSamplingFreq_, 1000u) * 120;
+  }
+
+  uint32_t first_timestamp() const { return 10u; }
+
+  void GetFirstPacket() {
+    bool muted;
+    for (int i = 0; i < 12; i++) {
+      EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+      EXPECT_FALSE(muted);
+    }
+  }
+
+  void InsertPacket(uint32_t timestamp) {
+    WebRtcRTPHeader rtp_header;
+    rtp_header.header.payloadType = kPayloadType;
+    rtp_header.header.sequenceNumber = sequence_number_;
+    rtp_header.header.timestamp = timestamp;
+    rtp_header.header.ssrc = 15;
+    const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+    uint8_t payload[kPayloadLengthBytes] = {0};
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload, 10));
+    sequence_number_++;
+  }
+
+  void Register120msCodec(AudioDecoder::SpeechType speech_type) {
+    decoder_.reset(new Decoder120ms(speech_type));
+    ASSERT_EQ(2u, decoder_->Channels());
+    EXPECT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(
+                              decoder_.get(), NetEqDecoder::kDecoderOpus_2ch,
+                              "120ms codec", kPayloadType, kSamplingFreq_));
+  }
+
+  std::unique_ptr<Decoder120ms> decoder_;
+  AudioFrame output_;
+  const uint32_t kPayloadType = 17;
+  const uint32_t kSamplingFreq_ = 48000;
+  uint16_t sequence_number_ = 1;
+};
+
+TEST_F(NetEqImplTest120ms, AudioRepetition) {
+  config_.playout_mode = kPlayoutFax;
+  CreateInstanceNoMocks();
+  Register120msCodec(AudioDecoder::kSpeech);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kAudioRepetition, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, AlternativePlc) {
+  config_.playout_mode = kPlayoutOff;
+  CreateInstanceNoMocks();
+  Register120msCodec(AudioDecoder::kSpeech);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kAlternativePlc, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, CodecInternalCng) {
+  CreateInstanceNoMocks();
+  Register120msCodec(AudioDecoder::kComfortNoise);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kCodecInternalCng, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Normal) {
+  CreateInstanceNoMocks();
+  Register120msCodec(AudioDecoder::kSpeech);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  EXPECT_EQ(kNormal, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Merge) {
+  CreateInstanceWithDelayManagerMock();
+
+  Register120msCodec(AudioDecoder::kSpeech);
+  InsertPacket(first_timestamp());
+
+  GetFirstPacket();
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+
+  InsertPacket(first_timestamp() + 2 * timestamp_diff_between_packets());
+
+  // Delay manager reports a target level which should cause a Merge.
+  EXPECT_CALL(*mock_delay_manager_, TargetLevel()).WillOnce(Return(-10));
+
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kMerge, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Expand) {
+  CreateInstanceNoMocks();
+  Register120msCodec(AudioDecoder::kSpeech);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kExpand, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, FastAccelerate) {
+  CreateInstanceWithDelayManagerMock();
+  Register120msCodec(AudioDecoder::kSpeech);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  // Delay manager report buffer limit which should cause a FastAccelerate.
+  EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
+      .Times(1)
+      .WillOnce(DoAll(SetArgPointee<0>(0), SetArgPointee<1>(0)));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kFastAccelerate, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, PreemptiveExpand) {
+  CreateInstanceWithDelayManagerMock();
+  Register120msCodec(AudioDecoder::kSpeech);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  // Delay manager report buffer limit which should cause a PreemptiveExpand.
+  EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
+      .Times(1)
+      .WillOnce(DoAll(SetArgPointee<0>(100), SetArgPointee<1>(100)));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kPreemptiveExpand, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Accelerate) {
+  CreateInstanceWithDelayManagerMock();
+  Register120msCodec(AudioDecoder::kSpeech);
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  // Delay manager report buffer limit which should cause a Accelerate.
+  EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
+      .Times(1)
+      .WillOnce(DoAll(SetArgPointee<0>(1), SetArgPointee<1>(2)));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
+}
+
 }// namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index 770ebd57835..1a77abcd505 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -24,31 +24,36 @@ using ::testing::Return;
 
 class MockAudioDecoder final : public AudioDecoder {
  public:
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
   static const int kPacketDuration = 960;  // 48 kHz * 20 ms
 
   explicit MockAudioDecoder(size_t num_channels)
       : num_channels_(num_channels), fec_enabled_(false) {
   }
-  ~MockAudioDecoder() override { Die(); }
+  ~MockAudioDecoder() /* override */ { Die(); }
   MOCK_METHOD0(Die, void());
 
   MOCK_METHOD0(Reset, void());
 
   int PacketDuration(const uint8_t* encoded,
-                     size_t encoded_len) const override {
+                     size_t encoded_len) const /* override */ {
     return kPacketDuration;
   }
 
   int PacketDurationRedundant(const uint8_t* encoded,
-                              size_t encoded_len) const override {
+                              size_t encoded_len) const /* override */ {
     return kPacketDuration;
   }
 
-  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override {
+  bool PacketHasFec(
+      const uint8_t* encoded, size_t encoded_len) const /* override */ {
     return fec_enabled_;
   }
 
-  size_t Channels() const override { return num_channels_; }
+  size_t Channels() const /* override */ { return num_channels_; }
 
   void set_fec_enabled(bool enable_fec) { fec_enabled_ = enable_fec; }
 
@@ -60,7 +65,7 @@ class MockAudioDecoder final : public AudioDecoder {
                      size_t encoded_len,
                      int /*sample_rate_hz*/,
                      int16_t* decoded,
-                     SpeechType* speech_type) override {
+                     SpeechType* speech_type) /* override */ {
     *speech_type = kSpeech;
     memset(decoded, 0, sizeof(int16_t) * kPacketDuration * Channels());
     return kPacketDuration * Channels();
@@ -70,7 +75,7 @@ class MockAudioDecoder final : public AudioDecoder {
                               size_t encoded_len,
                               int sample_rate_hz,
                               int16_t* decoded,
-                              SpeechType* speech_type) override {
+                              SpeechType* speech_type) /* override */ {
     return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
                           speech_type);
   }
@@ -294,7 +299,3 @@ TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
 
 }  // namespace test
 }  // namespace webrtc
-
-
-
-
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 4ee17d2a446..e1a9922b0b4 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -212,11 +212,14 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
         } while (Lost());  // If lost, immediately read the next packet.
       }
       // Get audio from mono instance.
-      EXPECT_EQ(NetEq::kOK, neteq_mono_->GetAudio(&output_));
+      bool muted;
+      EXPECT_EQ(NetEq::kOK, neteq_mono_->GetAudio(&output_, &muted));
+      ASSERT_FALSE(muted);
       EXPECT_EQ(1u, output_.num_channels_);
       EXPECT_EQ(output_size_samples_, output_.samples_per_channel_);
       // Get audio from multi-channel instance.
-      ASSERT_EQ(NetEq::kOK, neteq_->GetAudio(&output_multi_channel_));
+      ASSERT_EQ(NetEq::kOK, neteq_->GetAudio(&output_multi_channel_, &muted));
+      ASSERT_FALSE(muted);
       EXPECT_EQ(num_channels_, output_multi_channel_.num_channels_);
       EXPECT_EQ(output_size_samples_,
                 output_multi_channel_.samples_per_channel_);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_tests.gypi b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
index f02d3deee9b..bb316e8a81d 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
@@ -14,9 +14,12 @@
           'target_name': 'rtc_event_log_source',
           'type': 'static_library',
           'dependencies': [
-            '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
             '<(webrtc_root)/webrtc.gyp:rtc_event_log_proto',
           ],
+          'export_dependent_settings': [
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
+          ],
           'sources': [
             'tools/rtc_event_log_source.h',
             'tools/rtc_event_log_source.cc',
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index b6efe7d7d6b..cf8e5b474c2 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -8,10 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-/*
- * This file includes unit tests for NetEQ.
- */
-
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
 #include <math.h>
@@ -26,6 +22,8 @@
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/sha1digest.h"
+#include "webrtc/base/stringencode.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
@@ -45,6 +43,23 @@ DEFINE_bool(gen_ref, false, "Generate reference files.");
 
 namespace {
 
+const std::string& PlatformChecksum(const std::string& checksum_general,
+                                    const std::string& checksum_android,
+                                    const std::string& checksum_win_32,
+                                    const std::string& checksum_win_64) {
+#ifdef WEBRTC_ANDROID
+    return checksum_android;
+#elif WEBRTC_WIN
+  #ifdef WEBRTC_ARCH_64_BITS
+    return checksum_win_64;
+  #else
+    return checksum_win_32;
+  #endif  // WEBRTC_ARCH_64_BITS
+#else
+  return checksum_general;
+#endif  // WEBRTC_WIN
+}
+
 bool IsAllZero(const int16_t* buf, size_t buf_length) {
   bool all_zero = true;
   for (size_t n = 0; n < buf_length && all_zero; ++n)
@@ -89,186 +104,141 @@ void Convert(const webrtc::RtcpStatistics& stats_raw,
   stats->set_jitter(stats_raw.jitter);
 }
 
-void WriteMessage(FILE* file, const std::string& message) {
+void AddMessage(FILE* file, rtc::MessageDigest* digest,
+                const std::string& message) {
   int32_t size = message.length();
-  ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
-  if (size <= 0)
-    return;
-  ASSERT_EQ(static_cast<size_t>(size),
-            fwrite(message.data(), sizeof(char), size, file));
+  if (file)
+    ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  digest->Update(&size, sizeof(size));
+
+  if (file)
+    ASSERT_EQ(static_cast<size_t>(size),
+              fwrite(message.data(), sizeof(char), size, file));
+  digest->Update(message.data(), sizeof(char) * size);
 }
 
-void ReadMessage(FILE* file, std::string* message) {
-  int32_t size;
-  ASSERT_EQ(1u, fread(&size, sizeof(size), 1, file));
-  if (size <= 0)
-    return;
-  std::unique_ptr<char[]> buffer(new char[size]);
-  ASSERT_EQ(static_cast<size_t>(size),
-            fread(buffer.get(), sizeof(char), size, file));
-  message->assign(buffer.get(), size);
-}
 #endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 
+void LoadDecoders(webrtc::NetEq* neteq) {
+  // Load PCMu.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderPCMu,
+                                          "pcmu", 0));
+  // Load PCMa.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderPCMa,
+                                          "pcma", 8));
+#ifdef WEBRTC_CODEC_ILBC
+  // Load iLBC.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderILBC,
+                                          "ilbc", 102));
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+  // Load iSAC.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderISAC,
+                                          "isac", 103));
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+  // Load iSAC SWB.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderISACswb,
+                                          "isac-swb", 104));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderOpus,
+                                          "opus", 111));
+#endif
+  // Load PCM16B nb.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderPCM16B,
+                                          "pcm16-nb", 93));
+  // Load PCM16B wb.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(
+                   webrtc::NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb", 94));
+  // Load PCM16B swb32.
+  ASSERT_EQ(
+      0, neteq->RegisterPayloadType(
+             webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32", 95));
+  // Load CNG 8 kHz.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderCNGnb,
+                                          "cng-nb", 13));
+  // Load CNG 16 kHz.
+  ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderCNGwb,
+                                          "cng-wb", 98));
+}
 }  // namespace
 
 namespace webrtc {
 
-class RefFiles {
+class ResultSink {
  public:
-  RefFiles(const std::string& input_file, const std::string& output_file);
-  ~RefFiles();
-  template<class T> void ProcessReference(const T& test_results);
-  template<typename T, size_t n> void ProcessReference(
-      const T (&test_results)[n],
-      size_t length);
-  template<typename T, size_t n> void WriteToFile(
-      const T (&test_results)[n],
-      size_t length);
-  template<typename T, size_t n> void ReadFromFileAndCompare(
+  explicit ResultSink(const std::string& output_file);
+  ~ResultSink();
+
+  template<typename T, size_t n> void AddResult(
       const T (&test_results)[n],
       size_t length);
-  void WriteToFile(const NetEqNetworkStatistics& stats);
-  void ReadFromFileAndCompare(const NetEqNetworkStatistics& stats);
-  void WriteToFile(const RtcpStatistics& stats);
-  void ReadFromFileAndCompare(const RtcpStatistics& stats);
 
-  FILE* input_fp_;
+  void AddResult(const NetEqNetworkStatistics& stats);
+  void AddResult(const RtcpStatistics& stats);
+
+  void VerifyChecksum(const std::string& ref_check_sum);
+
+ private:
   FILE* output_fp_;
+  std::unique_ptr<rtc::MessageDigest> digest_;
 };
 
-RefFiles::RefFiles(const std::string &input_file,
-                   const std::string &output_file)
-    : input_fp_(NULL),
-      output_fp_(NULL) {
-  if (!input_file.empty()) {
-    input_fp_ = fopen(input_file.c_str(), "rb");
-    EXPECT_TRUE(input_fp_ != NULL);
-  }
+ResultSink::ResultSink(const std::string &output_file)
+    : output_fp_(nullptr),
+      digest_(new rtc::Sha1Digest()) {
   if (!output_file.empty()) {
     output_fp_ = fopen(output_file.c_str(), "wb");
     EXPECT_TRUE(output_fp_ != NULL);
   }
 }
 
-RefFiles::~RefFiles() {
-  if (input_fp_) {
-    EXPECT_EQ(EOF, fgetc(input_fp_));  // Make sure that we reached the end.
-    fclose(input_fp_);
-  }
-  if (output_fp_) fclose(output_fp_);
-}
-
-template<class T>
-void RefFiles::ProcessReference(const T& test_results) {
-  WriteToFile(test_results);
-  ReadFromFileAndCompare(test_results);
+ResultSink::~ResultSink() {
+  if (output_fp_)
+    fclose(output_fp_);
 }
 
 template<typename T, size_t n>
-void RefFiles::ProcessReference(const T (&test_results)[n], size_t length) {
-  WriteToFile(test_results, length);
-  ReadFromFileAndCompare(test_results, length);
-}
-
-template<typename T, size_t n>
-void RefFiles::WriteToFile(const T (&test_results)[n], size_t length) {
+void ResultSink::AddResult(const T (&test_results)[n], size_t length) {
   if (output_fp_) {
     ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_));
   }
+  digest_->Update(&test_results, sizeof(T) * length);
 }
 
-template<typename T, size_t n>
-void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
-                                      size_t length) {
-  if (input_fp_) {
-    // Read from ref file.
-    T* ref = new T[length];
-    ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_));
-    // Compare
-    ASSERT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length));
-    delete [] ref;
-  }
-}
-
-void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats_raw) {
+void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
-  if (!output_fp_)
-    return;
   neteq_unittest::NetEqNetworkStatistics stats;
   Convert(stats_raw, &stats);
 
   std::string stats_string;
   ASSERT_TRUE(stats.SerializeToString(&stats_string));
-  WriteMessage(output_fp_, stats_string);
+  AddMessage(output_fp_, digest_.get(), stats_string);
 #else
   FAIL() << "Writing to reference file requires Proto Buffer.";
 #endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
-void RefFiles::ReadFromFileAndCompare(
-    const NetEqNetworkStatistics& stats) {
+void ResultSink::AddResult(const RtcpStatistics& stats_raw) {
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
-  if (!input_fp_)
-    return;
-
-  std::string stats_string;
-  ReadMessage(input_fp_, &stats_string);
-  neteq_unittest::NetEqNetworkStatistics ref_stats;
-  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
-
-  // Compare
-  ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms());
-  ASSERT_EQ(stats.preferred_buffer_size_ms,
-            ref_stats.preferred_buffer_size_ms());
-  ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found());
-  ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate());
-  ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate());
-  ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate());
-  ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate());
-  ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate());
-  ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm());
-  ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples());
-  ASSERT_EQ(stats.secondary_decoded_rate, ref_stats.secondary_decoded_rate());
-  ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate());
-#else
-  FAIL() << "Reading from reference file requires Proto Buffer.";
-#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
-}
-
-void RefFiles::WriteToFile(const RtcpStatistics& stats_raw) {
-#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
-  if (!output_fp_)
-    return;
   neteq_unittest::RtcpStatistics stats;
   Convert(stats_raw, &stats);
 
   std::string stats_string;
   ASSERT_TRUE(stats.SerializeToString(&stats_string));
-  WriteMessage(output_fp_, stats_string);
+  AddMessage(output_fp_, digest_.get(), stats_string);
 #else
   FAIL() << "Writing to reference file requires Proto Buffer.";
 #endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 }
 
-void RefFiles::ReadFromFileAndCompare(const RtcpStatistics& stats) {
-#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
-  if (!input_fp_)
-    return;
-  std::string stats_string;
-  ReadMessage(input_fp_, &stats_string);
-  neteq_unittest::RtcpStatistics ref_stats;
-  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
-
-  // Compare
-  ASSERT_EQ(stats.fraction_lost, ref_stats.fraction_lost());
-  ASSERT_EQ(stats.cumulative_lost, ref_stats.cumulative_lost());
-  ASSERT_EQ(stats.extended_max_sequence_number,
-            ref_stats.extended_max_sequence_number());
-  ASSERT_EQ(stats.jitter, ref_stats.jitter());
-#else
-  FAIL() << "Reading from reference file requires Proto Buffer.";
-#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+void ResultSink::VerifyChecksum(const std::string& checksum) {
+  std::vector<char> buffer;
+  buffer.resize(digest_->Size());
+  digest_->Finish(&buffer[0], buffer.size());
+  const std::string result = rtc::hex_encode(&buffer[0], digest_->Size());
+  EXPECT_EQ(checksum, result);
 }
 
 class NetEqDecodingTest : public ::testing::Test {
@@ -286,14 +256,14 @@ class NetEqDecodingTest : public ::testing::Test {
   virtual void SetUp();
   virtual void TearDown();
   void SelectDecoders(NetEqDecoder* used_codec);
-  void LoadDecoders();
   void OpenInputFile(const std::string &rtp_file);
   void Process();
 
   void DecodeAndCompare(const std::string& rtp_file,
-                        const std::string& ref_file,
-                        const std::string& stat_ref_file,
-                        const std::string& rtcp_ref_file);
+                        const std::string& output_checksum,
+                        const std::string& network_stats_checksum,
+                        const std::string& rtcp_stats_checksum,
+                        bool gen_ref);
 
   static void PopulateRtpInfo(int frame_index,
                               int timestamp,
@@ -350,56 +320,13 @@ void NetEqDecodingTest::SetUp() {
   ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
   algorithmic_delay_ms_ = stat.current_buffer_size_ms;
   ASSERT_TRUE(neteq_);
-  LoadDecoders();
+  LoadDecoders(neteq_);
 }
 
 void NetEqDecodingTest::TearDown() {
   delete neteq_;
 }
 
-void NetEqDecodingTest::LoadDecoders() {
-  // Load PCMu.
-  ASSERT_EQ(0,
-            neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMu, "pcmu", 0));
-  // Load PCMa.
-  ASSERT_EQ(0,
-            neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMa, "pcma", 8));
-#ifdef WEBRTC_CODEC_ILBC
-  // Load iLBC.
-  ASSERT_EQ(
-      0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderILBC, "ilbc", 102));
-#endif
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-  // Load iSAC.
-  ASSERT_EQ(
-      0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac", 103));
-#endif
-#ifdef WEBRTC_CODEC_ISAC
-  // Load iSAC SWB.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISACswb,
-                                           "isac-swb", 104));
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderOpus,
-                                           "opus", 111));
-#endif
-  // Load PCM16B nb.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16B,
-                                           "pcm16-nb", 93));
-  // Load PCM16B wb.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb,
-                                           "pcm16-wb", 94));
-  // Load PCM16B swb32.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bswb32kHz,
-                                           "pcm16-swb32", 95));
-  // Load CNG 8 kHz.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
-                                           "cng-nb", 13));
-  // Load CNG 16 kHz.
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
-                                           "cng-wb", 98));
-}
-
 void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
   rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
 }
@@ -426,7 +353,9 @@ void NetEqDecodingTest::Process() {
   }
 
   // Get audio from NetEq.
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
   ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
               (out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
               (out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
@@ -438,29 +367,25 @@ void NetEqDecodingTest::Process() {
   sim_clock_ += kTimeStepMs;
 }
 
-void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
-                                         const std::string& ref_file,
-                                         const std::string& stat_ref_file,
-                                         const std::string& rtcp_ref_file) {
+void NetEqDecodingTest::DecodeAndCompare(
+    const std::string& rtp_file,
+    const std::string& output_checksum,
+    const std::string& network_stats_checksum,
+    const std::string& rtcp_stats_checksum,
+    bool gen_ref) {
   OpenInputFile(rtp_file);
 
-  std::string ref_out_file = "";
-  if (ref_file.empty()) {
-    ref_out_file = webrtc::test::OutputPath() + "neteq_universal_ref.pcm";
-  }
-  RefFiles ref_files(ref_file, ref_out_file);
+  std::string ref_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_universal_ref.pcm" : "";
+  ResultSink output(ref_out_file);
 
-  std::string stat_out_file = "";
-  if (stat_ref_file.empty()) {
-    stat_out_file = webrtc::test::OutputPath() + "neteq_network_stats.dat";
-  }
-  RefFiles network_stat_files(stat_ref_file, stat_out_file);
+  std::string stat_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
+  ResultSink network_stats(stat_out_file);
 
-  std::string rtcp_out_file = "";
-  if (rtcp_ref_file.empty()) {
-    rtcp_out_file = webrtc::test::OutputPath() + "neteq_rtcp_stats.dat";
-  }
-  RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file);
+  std::string rtcp_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_rtcp_stats.dat" : "";
+  ResultSink rtcp_stats(rtcp_out_file);
 
   packet_.reset(rtp_source_->NextPacket());
   int i = 0;
@@ -469,25 +394,33 @@ void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
     ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     ASSERT_NO_FATAL_FAILURE(Process());
-    ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(
+    ASSERT_NO_FATAL_FAILURE(output.AddResult(
         out_frame_.data_, out_frame_.samples_per_channel_));
 
     // Query the network statistics API once per second
     if (sim_clock_ % 1000 == 0) {
       // Process NetworkStatistics.
-      NetEqNetworkStatistics network_stats;
-      ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
-      ASSERT_NO_FATAL_FAILURE(
-          network_stat_files.ProcessReference(network_stats));
+      NetEqNetworkStatistics current_network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
+      ASSERT_NO_FATAL_FAILURE(network_stats.AddResult(current_network_stats));
+
       // Compare with CurrentDelay, which should be identical.
-      EXPECT_EQ(network_stats.current_buffer_size_ms, neteq_->CurrentDelayMs());
+      EXPECT_EQ(current_network_stats.current_buffer_size_ms,
+                neteq_->CurrentDelayMs());
 
       // Process RTCPstat.
-      RtcpStatistics rtcp_stats;
-      neteq_->GetRtcpStatistics(&rtcp_stats);
-      ASSERT_NO_FATAL_FAILURE(rtcp_stat_files.ProcessReference(rtcp_stats));
+      RtcpStatistics current_rtcp_stats;
+      neteq_->GetRtcpStatistics(&current_rtcp_stats);
+      ASSERT_NO_FATAL_FAILURE(rtcp_stats.AddResult(current_rtcp_stats));
     }
   }
+
+  SCOPED_TRACE("Check output audio.");
+  output.VerifyChecksum(output_checksum);
+  SCOPED_TRACE("Check network stats.");
+  network_stats.VerifyChecksum(network_stats_checksum);
+  SCOPED_TRACE("Check rtcp stats.");
+  rtcp_stats.VerifyChecksum(rtcp_stats_checksum);
 }
 
 void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
@@ -525,31 +458,30 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
-  // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
-  // are identical. The latter could have been removed, but if clients still
-  // have a copy of the file, the test will fail.
-  const std::string input_ref_file =
-      webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
-#if defined(_MSC_VER) && (_MSC_VER >= 1700)
-  // For Visual Studio 2012 and later, we will have to use the generic reference
-  // file, rather than the windows-specific one.
-  const std::string network_stat_ref_file = webrtc::test::ProjectRootPath() +
-      "resources/audio_coding/neteq4_network_stats.dat";
-#else
-  const std::string network_stat_ref_file =
-      webrtc::test::ResourcePath("audio_coding/neteq4_network_stats", "dat");
-#endif
-  const std::string rtcp_stat_ref_file =
-      webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat");
-
-  if (FLAGS_gen_ref) {
-    DecodeAndCompare(input_rtp_file, "", "", "");
-  } else {
-    DecodeAndCompare(input_rtp_file,
-                     input_ref_file,
-                     network_stat_ref_file,
-                     rtcp_stat_ref_file);
-  }
+
+  const std::string output_checksum = PlatformChecksum(
+      "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
+      "d2a6b6ff54b340cf9f961c7f07768d86b3761073",
+      "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
+      "f9749813dbc3fb59dae761de518fec65b8407c5b");
+
+  const std::string network_stats_checksum = PlatformChecksum(
+      "2cf380a05ee07080bd72471e8ec7777a39644ec9",
+      "01be67dc4c3b8e74743a45cbd8684c0535dec9ad",
+      "2cf380a05ee07080bd72471e8ec7777a39644ec9",
+      "2cf380a05ee07080bd72471e8ec7777a39644ec9");
+
+  const std::string rtcp_stats_checksum = PlatformChecksum(
+      "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
+      "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
+      "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
+      "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
+
+  DecodeAndCompare(input_rtp_file,
+                   output_checksum,
+                   network_stats_checksum,
+                   rtcp_stats_checksum,
+                   FLAGS_gen_ref);
 }
 
 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
@@ -562,26 +494,30 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
-  const std::string input_ref_file =
-      // The pcm files were generated by using Opus v1.1.2 to decode the RTC
-      // file generated by Opus v1.1
-      webrtc::test::ResourcePath("audio_coding/neteq4_opus_ref", "pcm");
-  const std::string network_stat_ref_file =
-      // The network stats file was generated when using Opus v1.1.2 to decode
-      // the RTC file generated by Opus v1.1
-      webrtc::test::ResourcePath("audio_coding/neteq4_opus_network_stats",
-                                 "dat");
-  const std::string rtcp_stat_ref_file =
-      webrtc::test::ResourcePath("audio_coding/neteq4_opus_rtcp_stats", "dat");
-
-  if (FLAGS_gen_ref) {
-    DecodeAndCompare(input_rtp_file, "", "", "");
-  } else {
-    DecodeAndCompare(input_rtp_file,
-                     input_ref_file,
-                     network_stat_ref_file,
-                     rtcp_stat_ref_file);
-  }
+
+  const std::string output_checksum = PlatformChecksum(
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4");
+
+  const std::string network_stats_checksum = PlatformChecksum(
+      "6eab76efbde753d4dde38983445ca16b4ce59b39",
+      "6eab76efbde753d4dde38983445ca16b4ce59b39",
+      "6eab76efbde753d4dde38983445ca16b4ce59b39",
+      "6eab76efbde753d4dde38983445ca16b4ce59b39");
+
+  const std::string rtcp_stats_checksum = PlatformChecksum(
+      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
+      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
+      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
+      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
+
+  DecodeAndCompare(input_rtp_file,
+                   output_checksum,
+                   network_stats_checksum,
+                   rtcp_stats_checksum,
+                   FLAGS_gen_ref);
 }
 
 // Use fax mode to avoid time-scaling. This is to simplify the testing of
@@ -610,7 +546,8 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
   }
   // Pull out all data.
   for (size_t i = 0; i < num_frames; ++i) {
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
@@ -651,7 +588,8 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
     }
 
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
@@ -678,7 +616,8 @@ TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
     }
 
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
@@ -699,6 +638,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
   const size_t kPayloadBytes = kSamples * 2;
   double next_input_time_ms = 0.0;
   double t_ms;
+  bool muted;
 
   // Insert speech for 5 seconds.
   const int kSpeechDurationMs = 5000;
@@ -715,7 +655,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
     }
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
@@ -744,7 +684,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
     }
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
@@ -757,7 +697,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
     const double loop_end_time = t_ms + network_freeze_ms;
     for (; t_ms < loop_end_time; t_ms += 10) {
       // Pull out data once.
-      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
       ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
       EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
     }
@@ -769,7 +709,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       if (pull_once && next_input_time_ms >= pull_time_ms) {
         pull_once = false;
         // Pull out data once.
-        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
         ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
         EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
         t_ms += 10;
@@ -803,7 +743,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
       next_input_time_ms += kFrameSizeMs * drift_factor;
     }
     // Pull out data once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
     // Increase clock.
     t_ms += 10;
@@ -931,7 +871,9 @@ TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
     out_frame_.data_[i] = 1;
   }
-  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_));
+  bool muted;
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
   // Verify that there is a decoder error to check.
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
 
@@ -968,7 +910,9 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
     out_frame_.data_[i] = 1;
   }
-  EXPECT_EQ(0, neteq_->GetAudio(&out_frame_));
+  bool muted;
+  EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
   // Verify that the first block of samples is set to 0.
   static const int kExpectedOutputLength =
       kInitSampleRateHz / 100;  // 10 ms at initial sample rate.
@@ -1020,6 +964,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
     rtp_info.header.payloadType = payload_type;
 
     uint32_t receive_timestamp = 0;
+    bool muted;
     for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
       auto block = input.GetNextBlock();
       ASSERT_EQ(expected_samples_per_channel, block.size());
@@ -1031,7 +976,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
                                                       payload, enc_len_bytes),
                                         receive_timestamp));
       output.Reset();
-      ASSERT_EQ(0, neteq_->GetAudio(&output));
+      ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
       ASSERT_EQ(1u, output.num_channels_);
       ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
       ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
@@ -1047,7 +992,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
     // Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull
     // one frame without checking speech-type. This is the first frame pulled
     // without inserting any packet, and might not be labeled as PLC.
-    ASSERT_EQ(0, neteq_->GetAudio(&output));
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
     ASSERT_EQ(1u, output.num_channels_);
     ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
 
@@ -1062,7 +1007,8 @@ class NetEqBgnTest : public NetEqDecodingTest {
     for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
       output.Reset();
       memset(output.data_, 1, sizeof(output.data_));  // Set to non-zero.
-      ASSERT_EQ(0, neteq_->GetAudio(&output));
+      ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+      ASSERT_FALSE(muted);
       ASSERT_EQ(1u, output.num_channels_);
       ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
       if (output.speech_type_ == AudioFrame::kPLCCNG) {
@@ -1236,9 +1182,10 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   // Insert some packets which decode to noise. We are not interested in
   // actual decoded values.
   uint32_t receive_timestamp = 0;
+  bool muted;
   for (int n = 0; n < 100; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output));
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
     ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
     ASSERT_EQ(1u, output.num_channels_);
 
@@ -1254,7 +1201,8 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   // Insert sync-packets, the decoded sequence should be all-zero.
   for (int n = 0; n < kNumSyncPackets; ++n) {
     ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output));
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_FALSE(muted);
     ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
     ASSERT_EQ(1u, output.num_channels_);
     if (n > algorithmic_frame_delay) {
@@ -1270,7 +1218,8 @@ TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   // network statistics would show some packet loss.
   for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output));
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_FALSE(muted);
     if (n >= algorithmic_frame_delay + 1) {
       // Expect that this frame contain samples from regular RTP.
       EXPECT_TRUE(IsAllNonZero(
@@ -1306,9 +1255,10 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
   // actual decoded values.
   uint32_t receive_timestamp = 0;
   int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
+  bool muted;
   for (int n = 0; n < algorithmic_frame_delay; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
-    ASSERT_EQ(0, neteq_->GetAudio(&output));
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
     ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
     ASSERT_EQ(1u, output.num_channels_);
     rtp_info.header.sequenceNumber++;
@@ -1345,7 +1295,8 @@ TEST_F(NetEqDecodingTest, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
 
   // Decode.
   for (int n = 0; n < kNumSyncPackets; ++n) {
-    ASSERT_EQ(0, neteq_->GetAudio(&output));
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_FALSE(muted);
     ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
     ASSERT_EQ(1u, output.num_channels_);
     EXPECT_TRUE(IsAllNonZero(
@@ -1412,7 +1363,8 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
     }
     // Pull out data once.
     AudioFrame output;
-    ASSERT_EQ(0, neteq_->GetAudio(&output));
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
     ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
     ASSERT_EQ(1u, output.num_channels_);
 
@@ -1468,6 +1420,7 @@ void NetEqDecodingTest::DuplicateCng() {
   // correct.
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
+  bool muted;
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
@@ -1475,7 +1428,7 @@ void NetEqDecodingTest::DuplicateCng() {
     timestamp += kSamples;
 
     // Pull audio once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
@@ -1492,7 +1445,7 @@ void NetEqDecodingTest::DuplicateCng() {
              rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio once and make sure CNG is played.
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
   EXPECT_FALSE(PlayoutTimestamp());  // Returns empty value during CNG.
@@ -1508,7 +1461,7 @@ void NetEqDecodingTest::DuplicateCng() {
   // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
   // we have already pulled out CNG once.
   for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
     EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
     EXPECT_FALSE(PlayoutTimestamp());  // Returns empty value during CNG.
@@ -1523,7 +1476,7 @@ void NetEqDecodingTest::DuplicateCng() {
   ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
 
   // Pull audio once and verify that the output is speech again.
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
   rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
@@ -1561,7 +1514,8 @@ TEST_F(NetEqDecodingTest, CngFirst) {
   timestamp += kCngPeriodSamples;
 
   // Pull audio once and make sure CNG is played.
-  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
 
@@ -1573,10 +1527,261 @@ TEST_F(NetEqDecodingTest, CngFirst) {
     timestamp += kSamples;
 
     // Pull audio once.
-    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }
+
+class NetEqDecodingTestWithMutedState : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestWithMutedState() : NetEqDecodingTest() {
+    config_.enable_muted_state = true;
+  }
+
+ protected:
+  static constexpr size_t kSamples = 10 * 16;
+  static constexpr size_t kPayloadBytes = kSamples * 2;
+
+  void InsertPacket(uint32_t rtp_timestamp) {
+    uint8_t payload[kPayloadBytes] = {0};
+    WebRtcRTPHeader rtp_info;
+    PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
+    EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
+  }
+
+  bool GetAudioReturnMuted() {
+    bool muted;
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    return muted;
+  }
+
+  void GetAudioUntilMuted() {
+    while (!GetAudioReturnMuted()) {
+      ASSERT_LT(counter_++, 1000) << "Test timed out";
+    }
+  }
+
+  void GetAudioUntilNormal() {
+    bool muted = false;
+    while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
+      EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+      ASSERT_LT(counter_++, 1000) << "Test timed out";
+    }
+    EXPECT_FALSE(muted);
+  }
+
+  int counter_ = 0;
+};
+
+// Verifies that NetEq goes in and out of muted state as expected.
+TEST_F(NetEqDecodingTestWithMutedState, MutedState) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+
+  // Verify that output audio is not written during muted mode. Other parameters
+  // should be correct, though.
+  AudioFrame new_frame;
+  for (auto& d : new_frame.data_) {
+    d = 17;
+  }
+  bool muted;
+  EXPECT_EQ(0, neteq_->GetAudio(&new_frame, &muted));
+  EXPECT_TRUE(muted);
+  for (auto d : new_frame.data_) {
+    EXPECT_EQ(17, d);
+  }
+  EXPECT_EQ(out_frame_.timestamp_ + out_frame_.samples_per_channel_,
+            new_frame.timestamp_);
+  EXPECT_EQ(out_frame_.samples_per_channel_, new_frame.samples_per_channel_);
+  EXPECT_EQ(out_frame_.sample_rate_hz_, new_frame.sample_rate_hz_);
+  EXPECT_EQ(out_frame_.num_channels_, new_frame.num_channels_);
+  EXPECT_EQ(out_frame_.speech_type_, new_frame.speech_type_);
+  EXPECT_EQ(out_frame_.vad_activity_, new_frame.vad_activity_);
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet. Verify that normal operation resumes.
+  InsertPacket(kSamples * counter_);
+  GetAudioUntilNormal();
+
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  // NetEqNetworkStatistics::expand_rate tells the fraction of samples that were
+  // concealment samples, in Q14 (16384 = 100%) .The vast majority should be
+  // concealment samples in this test.
+  EXPECT_GT(stats.expand_rate, 14000);
+  // And, it should be greater than the speech_expand_rate.
+  EXPECT_GT(stats.expand_rate, stats.speech_expand_rate);
+}
+
+// Verifies that NetEq goes out of muted state when given a delayed packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateDelayedPacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  // Insert new data. Timestamp is only corrected for the half of the time
+  // elapsed since the last packet. That is, the new packet is delayed. Verify
+  // that normal operation resumes.
+  InsertPacket(kSamples * counter_ / 2);
+  GetAudioUntilNormal();
+}
+
+// Verifies that NetEq goes out of muted state when given a future packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateFuturePacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  // Insert new data. Timestamp is over-corrected for the time elapsed since the
+  // last packet. That is, the new packet is too early. Verify that normal
+  // operation resumes.
+  InsertPacket(kSamples * counter_ * 2);
+  GetAudioUntilNormal();
+}
+
+// Verifies that NetEq goes out of muted state when given an old packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateOldPacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+
+  EXPECT_NE(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  // Insert packet which is older than the first packet.
+  InsertPacket(kSamples * (counter_ - 1000));
+  EXPECT_FALSE(GetAudioReturnMuted());
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+}
+
+class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
+
+  void SetUp() override {
+    NetEqDecodingTest::SetUp();
+    config2_ = config_;
+  }
+
+  void CreateSecondInstance() {
+    neteq2_.reset(NetEq::Create(config2_));
+    ASSERT_TRUE(neteq2_);
+    LoadDecoders(neteq2_.get());
+  }
+
+ protected:
+  std::unique_ptr<NetEq> neteq2_;
+  NetEq::Config config2_;
+};
+
+namespace {
+::testing::AssertionResult AudioFramesEqualExceptData(const AudioFrame& a,
+                                                      const AudioFrame& b) {
+  if (a.timestamp_ != b.timestamp_)
+    return ::testing::AssertionFailure() << "timestamp_ diff (" << a.timestamp_
+                                         << " != " << b.timestamp_ << ")";
+  if (a.sample_rate_hz_ != b.sample_rate_hz_)
+    return ::testing::AssertionFailure() << "sample_rate_hz_ diff ("
+                                         << a.sample_rate_hz_
+                                         << " != " << b.sample_rate_hz_ << ")";
+  if (a.samples_per_channel_ != b.samples_per_channel_)
+    return ::testing::AssertionFailure()
+           << "samples_per_channel_ diff (" << a.samples_per_channel_
+           << " != " << b.samples_per_channel_ << ")";
+  if (a.num_channels_ != b.num_channels_)
+    return ::testing::AssertionFailure() << "num_channels_ diff ("
+                                         << a.num_channels_
+                                         << " != " << b.num_channels_ << ")";
+  if (a.speech_type_ != b.speech_type_)
+    return ::testing::AssertionFailure() << "speech_type_ diff ("
+                                         << a.speech_type_
+                                         << " != " << b.speech_type_ << ")";
+  if (a.vad_activity_ != b.vad_activity_)
+    return ::testing::AssertionFailure() << "vad_activity_ diff ("
+                                         << a.vad_activity_
+                                         << " != " << b.vad_activity_ << ")";
+  return ::testing::AssertionSuccess();
+}
+
+::testing::AssertionResult AudioFramesEqual(const AudioFrame& a,
+                                            const AudioFrame& b) {
+  ::testing::AssertionResult res = AudioFramesEqualExceptData(a, b);
+  if (!res)
+    return res;
+  if (memcmp(
+      a.data_, b.data_,
+      a.samples_per_channel_ * a.num_channels_ * sizeof(a.data_[0])) != 0) {
+    return ::testing::AssertionFailure() << "data_ diff";
+  }
+  return ::testing::AssertionSuccess();
+}
+
+}  // namespace
+
+TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
+  ASSERT_FALSE(config_.enable_muted_state);
+  config2_.enable_muted_state = true;
+  CreateSecondInstance();
+
+  // Insert one speech packet into both NetEqs.
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  uint8_t payload[kPayloadBytes] = {0};
+  WebRtcRTPHeader rtp_info;
+  PopulateRtpInfo(0, 0, &rtp_info);
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
+  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
+
+  AudioFrame out_frame1, out_frame2;
+  bool muted;
+  for (int i = 0; i < 1000; ++i) {
+    std::ostringstream ss;
+    ss << "i = " << i;
+    SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
+    EXPECT_FALSE(muted);
+    EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
+    if (muted) {
+      EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
+    } else {
+      EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
+    }
+  }
+  EXPECT_TRUE(muted);
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet.
+  PopulateRtpInfo(0, kSamples * 1000, &rtp_info);
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
+  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
+
+  int counter = 0;
+  while (out_frame1.speech_type_ != AudioFrame::kNormalSpeech) {
+    ASSERT_LT(counter++, 1000) << "Test timed out";
+    std::ostringstream ss;
+    ss << "counter = " << counter;
+    SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
+    EXPECT_FALSE(muted);
+    EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
+    if (muted) {
+      EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
+    } else {
+      EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
+    }
+  }
+  EXPECT_FALSE(muted);
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.cc
index 9bddfe77657..f99b3f200f4 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
@@ -43,7 +42,6 @@ int Normal::Process(const int16_t* input,
     return 0;
   }
   output->PushBackInterleaved(input, length);
-  int16_t* signal = &(*output)[0][0];
 
   const int fs_mult = fs_hz_ / 8000;
   assert(fs_mult > 0);
@@ -64,24 +62,26 @@ int Normal::Process(const int16_t* input,
     expand_->Process(&expanded);
     expand_->Reset();
 
+    size_t length_per_channel = length / output->Channels();
+    std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
     for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
       // Adjust muting factor (main muting factor times expand muting factor).
       external_mute_factor_array[channel_ix] = static_cast<int16_t>(
           (external_mute_factor_array[channel_ix] *
           expand_->MuteFactor(channel_ix)) >> 14);
 
-      int16_t* signal = &(*output)[channel_ix][0];
-      size_t length_per_channel = length / output->Channels();
+      (*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());
+
       // Find largest absolute value in new data.
       int16_t decoded_max =
-          WebRtcSpl_MaxAbsValueW16(signal, length_per_channel);
+          WebRtcSpl_MaxAbsValueW16(signal.get(), length_per_channel);
       // Adjust muting factor if needed (to BGN level).
       size_t energy_length =
           std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
       int scaling = 6 + fs_shift
           - WebRtcSpl_NormW32(decoded_max * decoded_max);
       scaling = std::max(scaling, 0);  // |scaling| should always be >= 0.
-      int32_t energy = WebRtcSpl_DotProductWithScale(signal, signal,
+      int32_t energy = WebRtcSpl_DotProductWithScale(signal.get(), signal.get(),
                                                      energy_length, scaling);
       int32_t scaled_energy_length =
           static_cast<int32_t>(energy_length >> scaling);
@@ -149,19 +149,18 @@ int Normal::Process(const int16_t* input,
     int16_t cng_output[kCngLength];
     // Reset mute factor and start up fresh.
     external_mute_factor_array[0] = 16384;
-    AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+    ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
 
     if (cng_decoder) {
       // Generate long enough for 32kHz.
-      if (WebRtcCng_Generate(cng_decoder->CngDecoderInstance(), cng_output,
-                             kCngLength, 0) < 0) {
+      if (!cng_decoder->Generate(cng_output, 0)) {
         // Error returned; set return vector to all zeros.
         memset(cng_output, 0, sizeof(cng_output));
       }
     } else {
       // If no CNG instance is defined, just copy from the decoded data.
       // (This will result in interpolating the decoded with itself.)
-      memcpy(cng_output, signal, fs_mult * 8 * sizeof(int16_t));
+      (*output)[0].CopyTo(fs_mult * 8, 0, cng_output);
     }
     // Interpolate the CNG into the new vector.
     // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
@@ -171,8 +170,8 @@ int Normal::Process(const int16_t* input,
     for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
       // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
       // for legacy bit-exactness.
-      signal[i] =
-          (fraction * signal[i] + (32 - fraction) * cng_output[i] + 8) >> 5;
+      (*output)[0][i] = (fraction * (*output)[0][i] +
+          (32 - fraction) * cng_output[i] + 8) >> 5;
       fraction += increment;
     }
   } else if (external_mute_factor_array[0] < 16384) {
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc
index f98e99a82d8..5e1fc131e50 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal_unittest.cc
@@ -27,9 +27,20 @@
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
 using ::testing::_;
+using ::testing::Invoke;
 
 namespace webrtc {
 
+namespace {
+
+int ExpandProcess120ms(AudioMultiVector* output) {
+  AudioMultiVector dummy_audio(1, 11520u);
+  dummy_audio.CopyTo(output);
+  return 0;
+}
+
+} // namespace
+
 TEST(Normal, CreateAndDestroy) {
   MockDecoderDatabase db;
   int fs = 8000;
@@ -121,6 +132,45 @@ TEST(Normal, InputLengthAndChannelsDoNotMatch) {
   EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
 }
 
+TEST(Normal, LastModeExpand120msPacket) {
+  WebRtcSpl_Init();
+  MockDecoderDatabase db;
+  const int kFs = 48000;
+  const size_t kPacketsizeBytes = 11520u;
+  const size_t kChannels = 1;
+  BackgroundNoise bgn(kChannels);
+  SyncBuffer sync_buffer(kChannels, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, kFs,
+                    kChannels);
+  Normal normal(kFs, &db, bgn, &expand);
+
+  int16_t input[kPacketsizeBytes] = {0};
+
+  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[kChannels]);
+  for (size_t i = 0; i < kChannels; ++i) {
+    mute_factor_array[i] = 16384;
+  }
+
+  AudioMultiVector output(kChannels);
+
+  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
+  EXPECT_CALL(expand, Process(_)).WillOnce(Invoke(ExpandProcess120ms));
+  EXPECT_CALL(expand, Reset());
+  EXPECT_EQ(static_cast<int>(kPacketsizeBytes),
+            normal.Process(input,
+                           kPacketsizeBytes,
+                           kModeExpand,
+                           mute_factor_array.get(),
+                           &output));
+
+  EXPECT_EQ(kPacketsizeBytes, output.Size());
+
+  EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
+}
+
 // TODO(hlundin): Write more tests.
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI_android.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.cc
index c62a62f39aa..8a19fe4d592 100644
--- a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI_android.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -8,8 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-int main(int argc, char* argv[]) {
-  // TODO(leozwang): Video render test app is not ready on android,
-  // make it dummy test now, will add android specific tests
-  return 0;
-}
+#include "webrtc/modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+Packet::Packet() = default;
+
+Packet::~Packet() = default;
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.h
index 64b325e027a..d6f64c7e088 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet.h
@@ -12,7 +12,9 @@
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_
 
 #include <list>
+#include <memory>
 
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
@@ -21,20 +23,15 @@ namespace webrtc {
 // Struct for holding RTP packets.
 struct Packet {
   RTPHeader header;
-  uint8_t* payload;  // Datagram excluding RTP header and header extension.
-  size_t payload_length;
-  bool primary;  // Primary, i.e., not redundant payload.
-  int waiting_time;
-  bool sync_packet;
+  // Datagram excluding RTP header and header extension.
+  uint8_t* payload = nullptr;
+  size_t payload_length = 0;
+  bool primary = true;  // Primary, i.e., not redundant payload.
+  bool sync_packet = false;
+  std::unique_ptr<TickTimer::Stopwatch> waiting_time;
 
-  // Constructor.
-  Packet()
-      : payload(NULL),
-        payload_length(0),
-        primary(true),
-        waiting_time(0),
-        sync_packet(false) {
-  }
+  Packet();
+  ~Packet();
 
   // Comparison operators. Establish a packet ordering based on (1) timestamp,
   // (2) sequence number, (3) regular packet vs sync-packet and (4) redundancy.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.cc
index c89de12318b..f1b898e34cf 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -19,6 +19,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 namespace webrtc {
 
@@ -37,8 +38,9 @@ class NewTimestampIsLarger {
   const Packet* new_packet_;
 };
 
-PacketBuffer::PacketBuffer(size_t max_number_of_packets)
-    : max_number_of_packets_(max_number_of_packets) {}
+PacketBuffer::PacketBuffer(size_t max_number_of_packets,
+                           const TickTimer* tick_timer)
+    : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {}
 
 // Destructor. All packets in the buffer will be destroyed.
 PacketBuffer::~PacketBuffer() {
@@ -65,6 +67,8 @@ int PacketBuffer::InsertPacket(Packet* packet) {
 
   int return_val = kOK;
 
+  packet->waiting_time = tick_timer_->GetNewStopwatch();
+
   if (buffer_.size() >= max_number_of_packets_) {
     // Buffer is full. Flush it.
     Flush();
@@ -268,13 +272,6 @@ size_t PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
   return num_samples;
 }
 
-void PacketBuffer::IncrementWaitingTimes(int inc) {
-  PacketList::iterator it;
-  for (it = buffer_.begin(); it != buffer_.end(); ++it) {
-    (*it)->waiting_time += inc;
-  }
-}
-
 bool PacketBuffer::DeleteFirstPacket(PacketList* packet_list) {
   if (packet_list->empty()) {
     return false;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h
index 03c11e61b6e..6867b4cb37e 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer.h
@@ -17,8 +17,8 @@
 
 namespace webrtc {
 
-// Forward declaration.
 class DecoderDatabase;
+class TickTimer;
 
 // This is the actual buffer holding the packets before decoding.
 class PacketBuffer {
@@ -34,7 +34,7 @@ class PacketBuffer {
 
   // Constructor creates a buffer which can hold a maximum of
   // |max_number_of_packets| packets.
-  PacketBuffer(size_t max_number_of_packets);
+  PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
 
   // Deletes all packets in the buffer before destroying the buffer.
   virtual ~PacketBuffer();
@@ -116,10 +116,6 @@ class PacketBuffer {
   virtual size_t NumSamplesInBuffer(DecoderDatabase* decoder_database,
                                     size_t last_decoded_length) const;
 
-  // Increase the waiting time counter for every packet in the buffer by |inc|.
-  // The default value for |inc| is 1.
-  virtual void IncrementWaitingTimes(int inc = 1);
-
   virtual void BufferStat(int* num_packets, int* max_num_packets) const;
 
   // Static method that properly deletes the first packet, and its payload
@@ -148,6 +144,7 @@ class PacketBuffer {
  private:
   size_t max_number_of_packets_;
   PacketList buffer_;
+  const TickTimer* tick_timer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 435b6c848dc..da353010857 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -16,6 +16,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 
 using ::testing::Return;
 using ::testing::_;
@@ -80,13 +81,15 @@ struct PacketsToInsert {
 // Start of test definitions.
 
 TEST(PacketBuffer, CreateAndDestroy) {
-  PacketBuffer* buffer = new PacketBuffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer* buffer = new PacketBuffer(10, &tick_timer);  // 10 packets.
   EXPECT_TRUE(buffer->Empty());
   delete buffer;
 }
 
 TEST(PacketBuffer, InsertPacket) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(17u, 4711u, 0, 10);
 
   const int payload_len = 100;
@@ -107,7 +110,8 @@ TEST(PacketBuffer, InsertPacket) {
 
 // Test to flush buffer.
 TEST(PacketBuffer, FlushBuffer) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
   const int payload_len = 10;
 
@@ -127,7 +131,8 @@ TEST(PacketBuffer, FlushBuffer) {
 
 // Test to fill the buffer over the limits, and verify that it flushes.
 TEST(PacketBuffer, OverfillBuffer) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
 
   // Insert 10 small packets; should be ok.
@@ -156,7 +161,8 @@ TEST(PacketBuffer, OverfillBuffer) {
 
 // Test inserting a list of packets.
 TEST(PacketBuffer, InsertPacketList) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
   PacketList list;
   const int payload_len = 10;
@@ -192,7 +198,8 @@ TEST(PacketBuffer, InsertPacketList) {
 // Expecting the buffer to flush.
 // TODO(hlundin): Remove this test when legacy operation is no longer needed.
 TEST(PacketBuffer, InsertPacketListChangePayloadType) {
-  PacketBuffer buffer(10);  // 10 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
   PacketGenerator gen(0, 0, 0, 10);
   PacketList list;
   const int payload_len = 10;
@@ -230,7 +237,8 @@ TEST(PacketBuffer, InsertPacketListChangePayloadType) {
 }
 
 TEST(PacketBuffer, ExtractOrderRedundancy) {
-  PacketBuffer buffer(100);  // 100 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
   const int kPackets = 18;
   const int kFrameSize = 10;
   const int kPayloadLength = 10;
@@ -289,7 +297,8 @@ TEST(PacketBuffer, ExtractOrderRedundancy) {
 }
 
 TEST(PacketBuffer, DiscardPackets) {
-  PacketBuffer buffer(100);  // 100 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
   const uint16_t start_seq_no = 17;
   const uint32_t start_ts = 4711;
   const uint32_t ts_increment = 10;
@@ -318,7 +327,8 @@ TEST(PacketBuffer, DiscardPackets) {
 }
 
 TEST(PacketBuffer, Reordering) {
-  PacketBuffer buffer(100);  // 100 packets.
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
   const uint16_t start_seq_no = 17;
   const uint32_t start_ts = 4711;
   const uint32_t ts_increment = 10;
@@ -373,8 +383,9 @@ TEST(PacketBuffer, Failures) {
   const uint32_t ts_increment = 10;
   int payload_len = 100;
   PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  TickTimer tick_timer;
 
-  PacketBuffer* buffer = new PacketBuffer(100);  // 100 packets.
+  PacketBuffer* buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
   Packet* packet = NULL;
   EXPECT_EQ(PacketBuffer::kInvalidPacket, buffer->InsertPacket(packet));
   packet = gen.NextPacket(payload_len);
@@ -404,7 +415,7 @@ TEST(PacketBuffer, Failures) {
   // Insert packet list of three packets, where the second packet has an invalid
   // payload.  Expect first packet to be inserted, and the remaining two to be
   // discarded.
-  buffer = new PacketBuffer(100);  // 100 packets.
+  buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
   PacketList list;
   list.push_back(gen.NextPacket(payload_len));  // Valid packet.
   packet = gen.NextPacket(payload_len);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.cc
index 85307181341..530e9d064dc 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter.cc
@@ -12,6 +12,7 @@
 
 #include <assert.h>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 
@@ -143,8 +144,9 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
 
     // Not an FEC packet.
     AudioDecoder* decoder = decoder_database->GetDecoder(payload_type);
-    // decoder should not return NULL.
-    assert(decoder != NULL);
+    // decoder should not return NULL, except for comfort noise payloads which
+    // are handled separately.
+    assert(decoder != NULL || decoder_database->IsComfortNoise(payload_type));
     if (!decoder ||
         !decoder->PacketHasFec(packet->payload, packet->payload_length)) {
       ++it;
@@ -167,8 +169,9 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
         memcpy(new_packet->payload, packet->payload, packet->payload_length);
         new_packet->payload_length = packet->payload_length;
         new_packet->primary = false;
-        new_packet->waiting_time = packet->waiting_time;
         new_packet->sync_packet = packet->sync_packet;
+        // Waiting time should not be set here.
+        RTC_DCHECK(!packet->waiting_time);
 
         packet_list->insert(it, new_packet);
         break;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
index a68e8d68a98..63772452da6 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
@@ -18,6 +18,8 @@
 #include <utility>  // pair
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 
@@ -309,7 +311,8 @@ TEST(RedPayloadSplitter, CheckRedPayloads) {
   // Use a real DecoderDatabase object here instead of a mock, since it is
   // easier to just register the payload types and let the actual implementation
   // do its job.
-  DecoderDatabase decoder_database;
+  std::unique_ptr<MockAudioDecoderFactory> factory(new MockAudioDecoderFactory);
+  DecoderDatabase decoder_database(std::move(factory));
   decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderCNGnb, "cng-nb");
   decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
   decoder_database.RegisterPayload(2, NetEqDecoder::kDecoderAVT, "avt");
@@ -372,33 +375,33 @@ TEST(AudioPayloadSplitter, NonSplittable) {
   // codec types.
   // Use scoped pointers to avoid having to delete them later.
   std::unique_ptr<DecoderDatabase::DecoderInfo> info0(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderISAC, 16000, NULL,
-                                       false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderISAC, "", 16000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
       .WillRepeatedly(Return(info0.get()));
   std::unique_ptr<DecoderDatabase::DecoderInfo> info1(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderISACswb, 32000,
-                                       NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderISACswb, "", 32000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(1))
       .WillRepeatedly(Return(info1.get()));
   std::unique_ptr<DecoderDatabase::DecoderInfo> info2(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderRED, 8000, NULL,
-                                       false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderRED, "", 8000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(2))
       .WillRepeatedly(Return(info2.get()));
   std::unique_ptr<DecoderDatabase::DecoderInfo> info3(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderAVT, 8000, NULL,
-                                       false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderAVT, "", 8000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(3))
       .WillRepeatedly(Return(info3.get()));
   std::unique_ptr<DecoderDatabase::DecoderInfo> info4(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderCNGnb, 8000, NULL,
-                                       false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderCNGnb, "", 8000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(4))
       .WillRepeatedly(Return(info4.get()));
   std::unique_ptr<DecoderDatabase::DecoderInfo> info5(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderArbitrary, 8000,
-                                       NULL, false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderArbitrary, "",
+                                       8000, nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(5))
       .WillRepeatedly(Return(info5.get()));
 
@@ -536,7 +539,7 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
   // Use scoped pointers to avoid having to delete them later.
   // (Sample rate is set to 8000 Hz, but does not matter.)
   std::unique_ptr<DecoderDatabase::DecoderInfo> info(
-      new DecoderDatabase::DecoderInfo(decoder_type_, 8000, NULL, false));
+      new DecoderDatabase::DecoderInfo(decoder_type_, "", 8000, nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(info.get()));
 
@@ -623,8 +626,8 @@ TEST_P(SplitIlbcTest, NumFrames) {
   // codec types.
   // Use scoped pointers to avoid having to delete them later.
   std::unique_ptr<DecoderDatabase::DecoderInfo> info(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, 8000, NULL,
-                                       false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, "", 8000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(info.get()));
 
@@ -687,8 +690,8 @@ TEST(IlbcPayloadSplitter, TooLargePayload) {
 
   MockDecoderDatabase decoder_database;
   std::unique_ptr<DecoderDatabase::DecoderInfo> info(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, 8000, NULL,
-                                       false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, "", 8000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(info.get()));
 
@@ -719,8 +722,8 @@ TEST(IlbcPayloadSplitter, UnevenPayload) {
 
   MockDecoderDatabase decoder_database;
   std::unique_ptr<DecoderDatabase::DecoderInfo> info(
-      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, 8000, NULL,
-                                       false));
+      new DecoderDatabase::DecoderInfo(NetEqDecoder::kDecoderILBC, "", 8000,
+                                       nullptr));
   EXPECT_CALL(decoder_database, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(info.get()));
 
@@ -743,7 +746,7 @@ TEST(IlbcPayloadSplitter, UnevenPayload) {
 
 TEST(FecPayloadSplitter, MixedPayload) {
   PacketList packet_list;
-  DecoderDatabase decoder_database;
+  DecoderDatabase decoder_database(CreateBuiltinAudioDecoderFactory());
 
   decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderOpus, "opus");
   decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
@@ -798,7 +801,7 @@ TEST(FecPayloadSplitter, MixedPayload) {
 
 TEST(FecPayloadSplitter, EmbedFecInRed) {
   PacketList packet_list;
-  DecoderDatabase decoder_database;
+  DecoderDatabase decoder_database(CreateBuiltinAudioDecoderFactory());
 
   const int kTimestampOffset = 20 * 48;  // 20 ms * 48 kHz.
   uint8_t payload_types[] = {0, 0};
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 45586ee111c..149f2826582 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -265,7 +265,7 @@ GSMFR_encinst_t* GSMFRenc_inst[2];
 #endif
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-CNG_enc_inst* CNGenc_inst[2];
+webrtc::ComfortNoiseEncoder *CNG_encoder[2];
 #endif
 #ifdef CODEC_SPEEX_8
 SPEEX_encinst_t* SPEEX8enc_inst[2];
@@ -928,18 +928,8 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
 
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-    ok = WebRtcCng_CreateEnc(&CNGenc_inst[k]);
-    if (ok != 0) {
-      printf("Error: Couldn't allocate memory for CNG encoding instance\n");
-      exit(0);
-    }
     if (sampfreq <= 16000) {
-      ok = WebRtcCng_InitEnc(CNGenc_inst[k], sampfreq, 200, 5);
-      if (ok == -1) {
-        printf("Error: Initialization of CNG struct failed. Error code %d\n",
-               WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
-        exit(0);
-      }
+      CNG_encoder[k] = new webrtc::ComfortNoiseEncoder(sampfreq, 200, 5);
     }
 #endif
 
@@ -1461,7 +1451,8 @@ int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
     WebRtcVad_Free(VAD_inst[k]);
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-    WebRtcCng_FreeEnc(CNGenc_inst[k]);
+    delete CNG_encoder[k];
+    CNG_encoder[k] = nullptr;
 #endif
 
     switch (coder) {
@@ -1600,7 +1591,7 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
                         size_t numChannels) {
   size_t cdlen = 0;
   int16_t* tempdata;
-  static int first_cng = 1;
+  static bool first_cng = true;
   size_t tempLen;
   *vad = 1;
 
@@ -1608,9 +1599,9 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
   if (useVAD) {
     *vad = 0;
 
-    size_t sampleRate_10 = static_cast<size_t>(10 * sampleRate / 1000);
-    size_t sampleRate_20 = static_cast<size_t>(20 * sampleRate / 1000);
-    size_t sampleRate_30 = static_cast<size_t>(30 * sampleRate / 1000);
+    const size_t sampleRate_10 = static_cast<size_t>(10 * sampleRate / 1000);
+    const size_t sampleRate_20 = static_cast<size_t>(20 * sampleRate / 1000);
+    const size_t sampleRate_30 = static_cast<size_t>(30 * sampleRate / 1000);
     for (size_t k = 0; k < numChannels; k++) {
       tempLen = frameLen;
       tempdata = &indata[k * frameLen];
@@ -1642,16 +1633,22 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
 
     if (!*vad) {
       // all channels are silent
+      rtc::Buffer workaround;
       cdlen = 0;
       for (size_t k = 0; k < numChannels; k++) {
-        WebRtcCng_Encode(CNGenc_inst[k], &indata[k * frameLen],
-                         (frameLen <= 640 ? frameLen : 640) /* max 640 */,
-                         encoded, &tempLen, first_cng);
+        workaround.Clear();
+        tempLen = CNG_encoder[k]->Encode(
+            rtc::ArrayView<const int16_t>(
+                &indata[k * frameLen],
+                (frameLen <= 640 ? frameLen : 640) /* max 640 */),
+            first_cng,
+            &workaround);
+        memcpy(encoded, workaround.data(), tempLen);
         encoded += tempLen;
         cdlen += tempLen;
       }
       *vad = 0;
-      first_cng = 0;
+      first_cng = false;
       return (cdlen);
     }
   }
@@ -1734,7 +1731,7 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
 
   }  // end for
 
-  first_cng = 1;
+  first_cng = true;
   return (totalLen);
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index 2ebd1927bc4..62bfc1b3cb5 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -43,8 +43,8 @@ class NetEqIsacQualityTest : public NetEqQualityTest {
   NetEqIsacQualityTest();
   void SetUp() override;
   void TearDown() override;
-  virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
-                          rtc::Buffer* payload, size_t max_bytes);
+  int EncodeBlock(int16_t* in_data, size_t block_size_samples,
+                  rtc::Buffer* payload, size_t max_bytes) override;
  private:
   ISACFIX_MainStruct* isac_encoder_;
   int bit_rate_kbps_;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
index baa0d67aded..a6117a4c5b6 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -103,8 +103,8 @@ class NetEqOpusQualityTest : public NetEqQualityTest {
   NetEqOpusQualityTest();
   void SetUp() override;
   void TearDown() override;
-  virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
-                          rtc::Buffer* payload, size_t max_bytes);
+  int EncodeBlock(int16_t* in_data, size_t block_size_samples,
+                  rtc::Buffer* payload, size_t max_bytes) override;
  private:
   WebRtcOpusEncInst* opus_encoder_;
   OpusRepacketizer* repacketizer_;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer.cc
new file mode 100644
index 00000000000..4a1b9b7b1fe
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer.cc
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
+
+namespace webrtc {
+
+TickTimer::Stopwatch::Stopwatch(const TickTimer& ticktimer)
+    : ticktimer_(ticktimer), starttick_(ticktimer.ticks()) {}
+
+TickTimer::Countdown::Countdown(const TickTimer& ticktimer,
+                                uint64_t ticks_to_count)
+    : stopwatch_(ticktimer.GetNewStopwatch()),
+      ticks_to_count_(ticks_to_count) {}
+
+TickTimer::Countdown::~Countdown() = default;
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer.h
new file mode 100644
index 00000000000..8f17f435967
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
+
+#include <memory>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Implements a time counter. The counter is advanced with the Increment()
+// methods, and is queried with the ticks() accessor. It is assumed that one
+// "tick" och the counter corresponds to 10 ms.
+// A TickTimer object can provide two types of associated time-measuring
+// objects: Stopwatch and Countdown.
+class TickTimer {
+ public:
+  // Stopwatch measures time elapsed since it was started, by querying the
+  // associated TickTimer for the current time. The intended use is to request a
+  // new Stopwatch object from a TickTimer object with the GetNewStopwatch()
+  // method. Note: since the Stopwatch object contains a reference to the
+  // TickTimer it is associated with, it cannot outlive the TickTimer.
+  class Stopwatch {
+   public:
+    explicit Stopwatch(const TickTimer& ticktimer);
+
+    uint64_t ElapsedTicks() const { return ticktimer_.ticks() - starttick_; }
+
+    uint64_t ElapsedMs() const {
+      const uint64_t elapsed_ticks = ticktimer_.ticks() - starttick_;
+      const int ms_per_tick = ticktimer_.ms_per_tick();
+      return elapsed_ticks < UINT64_MAX / ms_per_tick
+                 ? elapsed_ticks * ms_per_tick
+                 : UINT64_MAX;
+    }
+
+   private:
+    const TickTimer& ticktimer_;
+    const uint64_t starttick_;
+  };
+
+  // Countdown counts down from a given start value with each tick of the
+  // associated TickTimer, until zero is reached. The Finished() method will
+  // return true if zero has been reached, false otherwise. The intended use is
+  // to request a new Countdown object from a TickTimer object with the
+  // GetNewCountdown() method. Note: since the Countdown object contains a
+  // reference to the TickTimer it is associated with, it cannot outlive the
+  // TickTimer.
+  class Countdown {
+   public:
+    Countdown(const TickTimer& ticktimer, uint64_t ticks_to_count);
+
+    ~Countdown();
+
+    bool Finished() const {
+      return stopwatch_->ElapsedTicks() >= ticks_to_count_;
+    }
+
+   private:
+    const std::unique_ptr<Stopwatch> stopwatch_;
+    const uint64_t ticks_to_count_;
+  };
+
+  TickTimer() : TickTimer(10) {}
+  explicit TickTimer(int ms_per_tick) : ms_per_tick_(ms_per_tick) {
+    RTC_DCHECK_GT(ms_per_tick_, 0);
+  }
+
+  void Increment() { ++ticks_; }
+
+  // Mainly intended for testing.
+  void Increment(uint64_t x) { ticks_ += x; }
+
+  uint64_t ticks() const { return ticks_; }
+
+  int ms_per_tick() const { return ms_per_tick_; }
+
+  // Returns a new Stopwatch object, based on the current TickTimer. Note that
+  // the new Stopwatch object contains a reference to the current TickTimer,
+  // and must therefore not outlive the TickTimer.
+  std::unique_ptr<Stopwatch> GetNewStopwatch() const {
+    return std::unique_ptr<Stopwatch>(new Stopwatch(*this));
+  }
+
+  // Returns a new Countdown object, based on the current TickTimer. Note that
+  // the new Countdown object contains a reference to the current TickTimer,
+  // and must therefore not outlive the TickTimer.
+  std::unique_ptr<Countdown> GetNewCountdown(uint64_t ticks_to_count) const {
+    return std::unique_ptr<Countdown>(new Countdown(*this, ticks_to_count));
+  }
+
+ private:
+  uint64_t ticks_ = 0;
+  const int ms_per_tick_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(TickTimer);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer_unittest.cc
new file mode 100644
index 00000000000..55edcf5b292
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tick_timer_unittest.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+// Verify that the default value for ms_per_tick is 10.
+TEST(TickTimer, DefaultMsPerTick) {
+  TickTimer tt;
+  EXPECT_EQ(10, tt.ms_per_tick());
+}
+
+TEST(TickTimer, CustomMsPerTick) {
+  TickTimer tt(17);
+  EXPECT_EQ(17, tt.ms_per_tick());
+}
+
+TEST(TickTimer, Increment) {
+  TickTimer tt;
+  EXPECT_EQ(0u, tt.ticks());
+  tt.Increment();
+  EXPECT_EQ(1u, tt.ticks());
+
+  for (int i = 0; i < 17; ++i) {
+    tt.Increment();
+  }
+  EXPECT_EQ(18u, tt.ticks());
+
+  tt.Increment(17);
+  EXPECT_EQ(35u, tt.ticks());
+}
+
+TEST(TickTimer, WrapAround) {
+  TickTimer tt;
+  tt.Increment(UINT64_MAX);
+  EXPECT_EQ(UINT64_MAX, tt.ticks());
+  tt.Increment();
+  EXPECT_EQ(0u, tt.ticks());
+}
+
+TEST(TickTimer, Stopwatch) {
+  TickTimer tt;
+  // Increment it a "random" number of steps.
+  tt.Increment(17);
+
+  std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
+  ASSERT_TRUE(sw);
+
+  EXPECT_EQ(0u, sw->ElapsedTicks());  // Starts at zero.
+  EXPECT_EQ(0u, sw->ElapsedMs());
+  tt.Increment();
+  EXPECT_EQ(1u, sw->ElapsedTicks());  // Increases with the TickTimer.
+  EXPECT_EQ(10u, sw->ElapsedMs());
+}
+
+TEST(TickTimer, StopwatchWrapAround) {
+  TickTimer tt;
+  tt.Increment(UINT64_MAX);
+
+  std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
+  ASSERT_TRUE(sw);
+
+  tt.Increment();
+  EXPECT_EQ(0u, tt.ticks());
+  EXPECT_EQ(1u, sw->ElapsedTicks());
+  EXPECT_EQ(10u, sw->ElapsedMs());
+
+  tt.Increment();
+  EXPECT_EQ(1u, tt.ticks());
+  EXPECT_EQ(2u, sw->ElapsedTicks());
+  EXPECT_EQ(20u, sw->ElapsedMs());
+}
+
+TEST(TickTimer, StopwatchMsOverflow) {
+  TickTimer tt;
+  std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
+  ASSERT_TRUE(sw);
+
+  tt.Increment(UINT64_MAX / 10);
+  EXPECT_EQ(UINT64_MAX, sw->ElapsedMs());
+
+  tt.Increment();
+  EXPECT_EQ(UINT64_MAX, sw->ElapsedMs());
+
+  tt.Increment(UINT64_MAX - tt.ticks());
+  EXPECT_EQ(UINT64_MAX, tt.ticks());
+  EXPECT_EQ(UINT64_MAX, sw->ElapsedMs());
+}
+
+TEST(TickTimer, StopwatchWithCustomTicktime) {
+  const int kMsPerTick = 17;
+  TickTimer tt(kMsPerTick);
+  std::unique_ptr<TickTimer::Stopwatch> sw = tt.GetNewStopwatch();
+  ASSERT_TRUE(sw);
+
+  EXPECT_EQ(0u, sw->ElapsedMs());
+  tt.Increment();
+  EXPECT_EQ(static_cast<uint64_t>(kMsPerTick), sw->ElapsedMs());
+}
+
+TEST(TickTimer, Countdown) {
+  TickTimer tt;
+  // Increment it a "random" number of steps.
+  tt.Increment(4711);
+
+  std::unique_ptr<TickTimer::Countdown> cd = tt.GetNewCountdown(17);
+  ASSERT_TRUE(cd);
+
+  EXPECT_FALSE(cd->Finished());
+  tt.Increment();
+  EXPECT_FALSE(cd->Finished());
+
+  tt.Increment(16);  // Total increment is now 17.
+  EXPECT_TRUE(cd->Finished());
+
+  // Further increments do not change the state.
+  tt.Increment();
+  EXPECT_TRUE(cd->Finished());
+  tt.Increment(1234);
+  EXPECT_TRUE(cd->Finished());
+}
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.cc
index 6a91ea487b5..880b1f82ea5 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/time_stretch.cc
@@ -16,6 +16,7 @@
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
+#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 
 namespace webrtc {
@@ -158,20 +159,15 @@ TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
 }
 
 void TimeStretch::AutoCorrelation() {
-  // Set scaling factor for cross correlation to protect against overflow.
-  int scaling = kLogCorrelationLen - WebRtcSpl_NormW32(
-      max_input_value_ * max_input_value_);
-  scaling = std::max(0, scaling);
-
   // Calculate correlation from lag kMinLag to lag kMaxLag in 4 kHz domain.
   int32_t auto_corr[kCorrelationLen];
-  WebRtcSpl_CrossCorrelation(auto_corr, &downsampled_input_[kMaxLag],
-                             &downsampled_input_[kMaxLag - kMinLag],
-                             kCorrelationLen, kMaxLag - kMinLag, scaling, -1);
+  CrossCorrelationWithAutoShift(
+      &downsampled_input_[kMaxLag], &downsampled_input_[kMaxLag - kMinLag],
+      kCorrelationLen, kMaxLag - kMinLag, -1, auto_corr);
 
   // Normalize correlation to 14 bits and write to |auto_correlation_|.
   int32_t max_corr = WebRtcSpl_MaxAbsValueW32(auto_corr, kCorrelationLen);
-  scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
+  int scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
   WebRtcSpl_VectorBitShiftW32ToW16(auto_correlation_, kCorrelationLen,
                                    auto_corr, scaling);
 }
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
index b1cb45d2014..adaf16223b6 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -23,9 +23,9 @@ namespace webrtc {
 
 TEST(TimestampScaler, TestNoScaling) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type =
-      NetEqDecoder::kDecoderPCMu;  // Does not use scaled timestamps.
+  // Use PCMu, because it doesn't use scaled timestamps.
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, "", 8000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 0;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -44,9 +44,9 @@ TEST(TimestampScaler, TestNoScaling) {
 
 TEST(TimestampScaler, TestNoScalingLargeStep) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type =
-      NetEqDecoder::kDecoderPCMu;  // Does not use scaled timestamps.
+  // Use PCMu, because it doesn't use scaled timestamps.
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, "", 8000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 0;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -70,8 +70,9 @@ TEST(TimestampScaler, TestNoScalingLargeStep) {
 
 TEST(TimestampScaler, TestG722) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, "", 16000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -94,8 +95,9 @@ TEST(TimestampScaler, TestG722) {
 
 TEST(TimestampScaler, TestG722LargeStep) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, "", 16000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -122,10 +124,11 @@ TEST(TimestampScaler, TestG722LargeStep) {
 
 TEST(TimestampScaler, TestG722WithCng) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info_g722, info_cng;
-  info_g722.codec_type =
-      NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
-  info_cng.codec_type = NetEqDecoder::kDecoderCNGwb;
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info_g722(NetEqDecoder::kDecoderG722, "",
+                                               16000, nullptr);
+  const DecoderDatabase::DecoderInfo info_cng(NetEqDecoder::kDecoderCNGwb, "",
+                                              16000, nullptr);
   static const uint8_t kRtpPayloadTypeG722 = 17;
   static const uint8_t kRtpPayloadTypeCng = 13;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
@@ -164,9 +167,9 @@ TEST(TimestampScaler, TestG722WithCng) {
 // as many tests here.
 TEST(TimestampScaler, TestG722Packet) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type =
-      NetEqDecoder::kDecoderG722;  // Does uses a factor 2 scaling.
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, "", 16000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -193,8 +196,9 @@ TEST(TimestampScaler, TestG722Packet) {
 // we are not doing as many tests here.
 TEST(TimestampScaler, TestG722PacketList) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, "", 16000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -222,8 +226,9 @@ TEST(TimestampScaler, TestG722PacketList) {
 
 TEST(TimestampScaler, TestG722Reset) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type = NetEqDecoder::kDecoderG722;  // Uses a factor 2 scaling.
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, "", 16000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -262,8 +267,8 @@ TEST(TimestampScaler, TestG722Reset) {
 // timestamp scaler.
 TEST(TimestampScaler, TestOpusLargeStep) {
   MockDecoderDatabase db;
-  DecoderDatabase::DecoderInfo info;
-  info.codec_type = NetEqDecoder::kDecoderOpus;
+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderOpus, "", 48000,
+                                          nullptr);
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 2608d9a03b7..7a51256af2d 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -45,7 +45,9 @@ void NetEqExternalDecoderTest::InsertPacket(
 
 void NetEqExternalDecoderTest::GetOutputAudio(AudioFrame* output) {
   // Get audio from regular instance.
-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(output));
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(output, &muted));
+  ASSERT_FALSE(muted);
   EXPECT_EQ(channels_, output->num_channels_);
   EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
             output->samples_per_channel_);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 59402a2029b..d0052c28a8d 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
@@ -105,7 +106,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
     }
 
     // Get output audio, but don't do anything with it.
-    int error = neteq->GetAudio(&out_frame);
+    bool muted;
+    int error = neteq->GetAudio(&out_frame, &muted);
+    RTC_CHECK(!muted);
     if (error != NetEq::kOK)
       return -1;
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 5f874ad8dbe..2983cebe9d4 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -391,7 +391,9 @@ int NetEqQualityTest::Transmit() {
 }
 
 int NetEqQualityTest::DecodeBlock() {
-  int ret = neteq_->GetAudio(&out_frame_);
+  bool muted;
+  int ret = neteq_->GetAudio(&out_frame_, &muted);
+  RTC_CHECK(!muted);
 
   if (ret != NetEq::kOK) {
     return -1;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index fdb66714cfb..1d462b3c9f2 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -605,7 +605,9 @@ int main(int argc, char* argv[]) {
     // Check if it is time to get output audio.
     while (time_now_ms >= next_output_time_ms && output_event_available) {
       webrtc::AudioFrame out_frame;
-      int error = neteq->GetAudio(&out_frame);
+      bool muted;
+      int error = neteq->GetAudio(&out_frame, &muted);
+      RTC_CHECK(!muted);
       if (error != NetEq::kOK) {
         std::cerr << "GetAudio returned error code " <<
             neteq->LastError() << std::endl;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index dad72eaecd1..9192839be30 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -16,51 +16,15 @@
 #include <limits>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
-// Files generated at build-time by the protobuf compiler.
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
-#else
-#include "webrtc/call/rtc_event_log.pb.h"
-#endif
 
 namespace webrtc {
 namespace test {
 
-namespace {
-
-const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) {
-  if (!event.has_type() || event.type() != rtclog::Event::RTP_EVENT)
-    return nullptr;
-  if (!event.has_timestamp_us() || !event.has_rtp_packet())
-    return nullptr;
-  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
-  if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
-      !rtp_packet.has_incoming() || !rtp_packet.incoming() ||
-      !rtp_packet.has_packet_length() || rtp_packet.packet_length() == 0 ||
-      !rtp_packet.has_header() || rtp_packet.header().size() == 0 ||
-      rtp_packet.packet_length() < rtp_packet.header().size())
-    return nullptr;
-  return &rtp_packet;
-}
-
-const rtclog::AudioPlayoutEvent* GetAudioPlayoutEvent(
-    const rtclog::Event& event) {
-  if (!event.has_type() || event.type() != rtclog::Event::AUDIO_PLAYOUT_EVENT)
-    return nullptr;
-  if (!event.has_timestamp_us() || !event.has_audio_playout_event())
-    return nullptr;
-  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
-  if (!playout_event.has_local_ssrc())
-    return nullptr;
-  return &playout_event;
-}
-
-}  // namespace
-
 RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
   RtcEventLogSource* source = new RtcEventLogSource();
   RTC_CHECK(source->OpenFile(file_name));
@@ -76,42 +40,57 @@ bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
 }
 
 Packet* RtcEventLogSource::NextPacket() {
-  while (rtp_packet_index_ < event_log_->stream_size()) {
-    const rtclog::Event& event = event_log_->stream(rtp_packet_index_);
-    const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event);
-    rtp_packet_index_++;
-    if (rtp_packet) {
-      uint8_t* packet_header = new uint8_t[rtp_packet->header().size()];
-      memcpy(packet_header, rtp_packet->header().data(),
-             rtp_packet->header().size());
-      Packet* packet = new Packet(packet_header, rtp_packet->header().size(),
-                                  rtp_packet->packet_length(),
-                                  event.timestamp_us() / 1000, *parser_.get());
-      if (packet->valid_header()) {
-        // Check if the packet should not be filtered out.
-        if (!filter_.test(packet->header().payloadType) &&
-            !(use_ssrc_filter_ && packet->header().ssrc != ssrc_))
-          return packet;
-      } else {
-        std::cout << "Warning: Packet with index " << (rtp_packet_index_ - 1)
-                  << " has an invalid header and will be ignored." << std::endl;
+  while (rtp_packet_index_ < parsed_stream_.GetNumberOfEvents()) {
+    if (parsed_stream_.GetEventType(rtp_packet_index_) ==
+        ParsedRtcEventLog::RTP_EVENT) {
+      PacketDirection direction;
+      MediaType media_type;
+      size_t header_length;
+      size_t packet_length;
+      uint64_t timestamp_us = parsed_stream_.GetTimestamp(rtp_packet_index_);
+      parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, &media_type,
+                                  nullptr, &header_length, &packet_length);
+      if (direction == kIncomingPacket && media_type == MediaType::AUDIO) {
+        uint8_t* packet_header = new uint8_t[header_length];
+        parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, nullptr,
+                                    packet_header, nullptr, nullptr);
+        Packet* packet = new Packet(packet_header, header_length, packet_length,
+                                    static_cast<double>(timestamp_us) / 1000,
+                                    *parser_.get());
+        if (packet->valid_header()) {
+          // Check if the packet should not be filtered out.
+          if (!filter_.test(packet->header().payloadType) &&
+              !(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
+            rtp_packet_index_++;
+            return packet;
+          }
+        } else {
+          std::cout << "Warning: Packet with index " << rtp_packet_index_
+                    << " has an invalid header and will be ignored."
+                    << std::endl;
+        }
+        // The packet has either an invalid header or needs to be filtered out,
+        // so it can be deleted.
+        delete packet;
       }
-      // The packet has either an invalid header or needs to be filtered out, so
-      // it can be deleted.
-      delete packet;
     }
+    rtp_packet_index_++;
   }
   return nullptr;
 }
 
 int64_t RtcEventLogSource::NextAudioOutputEventMs() {
-  while (audio_output_index_ < event_log_->stream_size()) {
-    const rtclog::Event& event = event_log_->stream(audio_output_index_);
-    const rtclog::AudioPlayoutEvent* playout_event =
-        GetAudioPlayoutEvent(event);
+  while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
+    if (parsed_stream_.GetEventType(audio_output_index_) ==
+        ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
+      uint64_t timestamp_us = parsed_stream_.GetTimestamp(audio_output_index_);
+      // We call GetAudioPlayout only to check that the protobuf event is
+      // well-formed.
+      parsed_stream_.GetAudioPlayout(audio_output_index_, nullptr);
+      audio_output_index_++;
+      return timestamp_us / 1000;
+    }
     audio_output_index_++;
-    if (playout_event)
-      return event.timestamp_us() / 1000;
   }
   return std::numeric_limits<int64_t>::max();
 }
@@ -120,8 +99,7 @@ RtcEventLogSource::RtcEventLogSource()
     : PacketSource(), parser_(RtpHeaderParser::Create()) {}
 
 bool RtcEventLogSource::OpenFile(const std::string& file_name) {
-  event_log_.reset(new rtclog::EventStream());
-  return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get());
+  return parsed_stream_.ParseFile(file_name);
 }
 
 }  // namespace test
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index 312338ee087..ad7add154c5 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/call/rtc_event_log_parser.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
@@ -22,10 +23,6 @@ namespace webrtc {
 
 class RtpHeaderParser;
 
-namespace rtclog {
-class EventStream;
-}  // namespace rtclog
-
 namespace test {
 
 class Packet;
@@ -55,10 +52,10 @@ class RtcEventLogSource : public PacketSource {
 
   bool OpenFile(const std::string& file_name);
 
-  int rtp_packet_index_ = 0;
-  int audio_output_index_ = 0;
+  size_t rtp_packet_index_ = 0;
+  size_t audio_output_index_ = 0;
 
-  std::unique_ptr<rtclog::EventStream> event_log_;
+  ParsedRtcEventLog parsed_stream_;
   std::unique_ptr<RtpHeaderParser> parser_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/APITest.cc b/chromium/third_party/webrtc/modules/audio_coding/test/APITest.cc
index a2506ba0113..833398acddf 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/APITest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/APITest.cc
@@ -21,13 +21,13 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -323,7 +323,8 @@ bool APITest::APIThreadB(void* obj) {
 bool APITest::PullAudioRunA() {
   _pullEventA->Wait(100);
   AudioFrame audioFrame;
-  if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame) < 0) {
+  bool muted;
+  if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame, &muted) < 0) {
     bool thereIsDecoder;
     {
       ReadLockScoped rl(_apiTestRWLock);
@@ -343,7 +344,8 @@ bool APITest::PullAudioRunA() {
 bool APITest::PullAudioRunB() {
   _pullEventB->Wait(100);
   AudioFrame audioFrame;
-  if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame) < 0) {
+  bool muted;
+  if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame, &muted) < 0) {
     bool thereIsDecoder;
     {
       ReadLockScoped rl(_apiTestRWLock);
@@ -560,7 +562,7 @@ void APITest::Perform() {
   // Keep main thread waiting for sender/receiver
   // threads to complete
   EventWrapper* completeEvent = EventWrapper::Create();
-  uint64_t startTime = TickTime::MillisecondTimestamp();
+  uint64_t startTime = rtc::TimeMillis();
   uint64_t currentTime;
   // Run test in 2 minutes (120000 ms).
   do {
@@ -570,7 +572,7 @@ void APITest::Perform() {
     }
     //fflush(stderr);
     completeEvent->Wait(50);
-    currentTime = TickTime::MillisecondTimestamp();
+    currentTime = rtc::TimeMillis();
   } while ((currentTime - startTime) < 120000);
 
   //completeEvent->Wait(0xFFFFFFFF);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/Channel.cc b/chromium/third_party/webrtc/modules/audio_coding/test/Channel.cc
index 0507691fb4d..46c398b1b75 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/Channel.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/Channel.cc
@@ -14,7 +14,7 @@
 #include <iostream>
 
 #include "webrtc/base/format_macros.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/base/timeutils.h"
 
 namespace webrtc {
 
@@ -234,7 +234,7 @@ Channel::Channel(int16_t chID)
       _lastFrameSizeSample(0),
       _packetLoss(0),
       _useFECTestWithPacketLoss(false),
-      _beginTime(TickTime::MillisecondTimestamp()),
+      _beginTime(rtc::TimeMillis()),
       _totalBytes(0),
       external_send_timestamp_(-1),
       external_sequence_number_(-1),
@@ -286,7 +286,7 @@ void Channel::ResetStats() {
       _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
     }
   }
-  _beginTime = TickTime::MillisecondTimestamp();
+  _beginTime = rtc::TimeMillis();
   _totalBytes = 0;
   _channelCritSect.Leave();
 }
@@ -411,7 +411,7 @@ uint32_t Channel::LastInTimestamp() {
 
 double Channel::BitRate() {
   double rate;
-  uint64_t currTime = TickTime::MillisecondTimestamp();
+  uint64_t currTime = rtc::TimeMillis();
   _channelCritSect.Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
   _channelCritSect.Leave();
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/chromium/third_party/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index e0632243bf4..724502354e9 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -208,8 +208,12 @@ bool Receiver::IncomingPacket() {
 
 bool Receiver::PlayoutData() {
   AudioFrame audioFrame;
-
-  int32_t ok =_acm->PlayoutData10Ms(_frequency, &audioFrame);
+  bool muted;
+  int32_t ok = _acm->PlayoutData10Ms(_frequency, &audioFrame, &muted);
+  if (muted) {
+    ADD_FAILURE();
+    return false;
+  }
   EXPECT_EQ(0, ok);
   if (ok < 0){
     return false;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/SpatialAudio.cc b/chromium/third_party/webrtc/modules/audio_coding/test/SpatialAudio.cc
deleted file mode 100644
index c9f80808260..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/test/SpatialAudio.cc
+++ /dev/null
@@ -1,196 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-#include <math.h>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/test/SpatialAudio.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-#define NUM_PANN_COEFFS 10
-
-SpatialAudio::SpatialAudio(int testMode)
-    : _acmLeft(AudioCodingModule::Create(1)),
-      _acmRight(AudioCodingModule::Create(2)),
-      _acmReceiver(AudioCodingModule::Create(3)),
-      _testMode(testMode) {
-}
-
-SpatialAudio::~SpatialAudio() {
-  delete _channel;
-  _inFile.Close();
-  _outFile.Close();
-}
-
-int16_t SpatialAudio::Setup() {
-  _channel = new Channel;
-
-  // Register callback for the sender side.
-  CHECK_ERROR(_acmLeft->RegisterTransportCallback(_channel));
-  CHECK_ERROR(_acmRight->RegisterTransportCallback(_channel));
-  // Register the receiver ACM in channel
-  _channel->RegisterReceiverACM(_acmReceiver.get());
-
-  uint16_t sampFreqHz = 32000;
-
-  const std::string file_name = webrtc::test::ResourcePath(
-      "audio_coding/testfile32kHz", "pcm");
-  _inFile.Open(file_name, sampFreqHz, "rb", false);
-
-  std::string output_file = webrtc::test::OutputPath()
-      + "out_spatial_autotest.pcm";
-  if (_testMode == 1) {
-    output_file = webrtc::test::OutputPath() + "testspatial_out.pcm";
-    printf("\n");
-    printf("Enter the output file [%s]: ", output_file.c_str());
-    PCMFile::ChooseFile(&output_file, MAX_FILE_NAME_LENGTH_BYTE, &sampFreqHz);
-  } else {
-    output_file = webrtc::test::OutputPath() + "testspatial_out.pcm";
-  }
-  _outFile.Open(output_file, sampFreqHz, "wb", false);
-  _outFile.SaveStereo(true);
-
-  // Register all available codes as receiving codecs.
-  CodecInst codecInst;
-  int status;
-  uint8_t num_encoders = _acmReceiver->NumberOfCodecs();
-  // Register all available codes as receiving codecs once more.
-  for (uint8_t n = 0; n < num_encoders; n++) {
-    status = _acmReceiver->Codec(n, &codecInst);
-    if (status < 0) {
-      printf("Error in Codec(), no matching codec found");
-    }
-    status = _acmReceiver->RegisterReceiveCodec(codecInst);
-    if (status < 0) {
-      printf("Error in RegisterReceiveCodec() for payload type %d",
-             codecInst.pltype);
-    }
-  }
-
-  return 0;
-}
-
-void SpatialAudio::Perform() {
-  if (_testMode == 0) {
-    printf("Running SpatialAudio Test");
-    WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
-                 "---------- SpatialAudio ----------");
-  }
-
-  Setup();
-
-  CodecInst codecInst;
-  _acmLeft->Codec((uint8_t) 1, &codecInst);
-  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
-  EncodeDecode();
-
-  int16_t pannCntr = 0;
-
-  double leftPanning[NUM_PANN_COEFFS] = { 1.00, 0.95, 0.90, 0.85, 0.80, 0.75,
-      0.70, 0.60, 0.55, 0.50 };
-  double rightPanning[NUM_PANN_COEFFS] = { 0.50, 0.55, 0.60, 0.70, 0.75, 0.80,
-      0.85, 0.90, 0.95, 1.00 };
-
-  while ((pannCntr + 1) < NUM_PANN_COEFFS) {
-    _acmLeft->Codec((uint8_t) 0, &codecInst);
-    codecInst.pacsize = 480;
-    CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
-    CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
-
-    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
-    pannCntr++;
-
-    // Change codec
-    _acmLeft->Codec((uint8_t) 3, &codecInst);
-    codecInst.pacsize = 320;
-    CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
-    CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
-
-    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
-    pannCntr++;
-    if (_testMode == 0) {
-      printf(".");
-    }
-  }
-
-  _acmLeft->Codec((uint8_t) 4, &codecInst);
-  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
-  EncodeDecode();
-
-  _acmLeft->Codec((uint8_t) 0, &codecInst);
-  codecInst.pacsize = 480;
-  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
-  CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
-  pannCntr = NUM_PANN_COEFFS - 1;
-  while (pannCntr >= 0) {
-    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
-    pannCntr--;
-    if (_testMode == 0) {
-      printf(".");
-    }
-  }
-  if (_testMode == 0) {
-    printf("Done!\n");
-  }
-}
-
-void SpatialAudio::EncodeDecode(const double leftPanning,
-                                const double rightPanning) {
-  AudioFrame audioFrame;
-  int32_t outFileSampFreq = _outFile.SamplingFrequency();
-
-  const double rightToLeftRatio = rightPanning / leftPanning;
-
-  _channel->SetIsStereo(true);
-
-  while (!_inFile.EndOfFile()) {
-    _inFile.Read10MsData(audioFrame);
-    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
-      audioFrame.data_[n] = (int16_t) floor(
-          audioFrame.data_[n] * leftPanning + 0.5);
-    }
-    CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
-
-    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
-      audioFrame.data_[n] = (int16_t) floor(
-          audioFrame.data_[n] * rightToLeftRatio + 0.5);
-    }
-    CHECK_ERROR(_acmRight->Add10MsData(audioFrame));
-
-    CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
-    _outFile.Write10MsData(audioFrame);
-  }
-  _inFile.Rewind();
-}
-
-void SpatialAudio::EncodeDecode() {
-  AudioFrame audioFrame;
-  int32_t outFileSampFreq = _outFile.SamplingFrequency();
-
-  _channel->SetIsStereo(false);
-
-  while (!_inFile.EndOfFile()) {
-    _inFile.Read10MsData(audioFrame);
-    CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
-
-    CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
-    _outFile.Write10MsData(audioFrame);
-  }
-  _inFile.Rewind();
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/SpatialAudio.h b/chromium/third_party/webrtc/modules/audio_coding/test/SpatialAudio.h
deleted file mode 100644
index 270c370cf46..00000000000
--- a/chromium/third_party/webrtc/modules/audio_coding/test/SpatialAudio.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
-
-#include <memory>
-
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-
-namespace webrtc {
-
-class SpatialAudio : public ACMTest {
- public:
-  SpatialAudio(int testMode);
-  ~SpatialAudio();
-
-  void Perform();
- private:
-  int16_t Setup();
-  void EncodeDecode(double leftPanning, double rightPanning);
-  void EncodeDecode();
-
-  std::unique_ptr<AudioCodingModule> _acmLeft;
-  std::unique_ptr<AudioCodingModule> _acmRight;
-  std::unique_ptr<AudioCodingModule> _acmReceiver;
-  Channel* _channel;
-  PCMFile _inFile;
-  PCMFile _outFile;
-  int _testMode;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/TestAllCodecs.cc b/chromium/third_party/webrtc/modules/audio_coding/test/TestAllCodecs.cc
index bacfd371880..80a0464d385 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/TestAllCodecs.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -452,7 +452,9 @@ void TestAllCodecs::Run(TestPack* channel) {
     }
 
     // Run received side of ACM.
-    CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame));
+    bool muted;
+    CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
 
     // Write output speech to file.
     outfile_b_.Write10MsData(audio_frame.data_,
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/TestRedFec.cc b/chromium/third_party/webrtc/modules/audio_coding/test/TestRedFec.cc
index a1bdc04e531..24cda1122b9 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/TestRedFec.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/TestRedFec.cc
@@ -461,7 +461,9 @@ void TestRedFec::Run() {
   while (!_inFileA.EndOfFile()) {
     EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
     EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
-    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+    bool muted;
+    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
     _outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
   }
   _inFileA.Rewind();
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/TestStereo.cc b/chromium/third_party/webrtc/modules/audio_coding/test/TestStereo.cc
index 9bf560d3237..85a2ae283b2 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/TestStereo.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/TestStereo.cc
@@ -792,7 +792,9 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
     }
 
     // Run received side of ACM
-    EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+    bool muted;
+    EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
 
     // Write output speech to file
     out_file_.Write10MsData(
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/TestVADDTX.cc b/chromium/third_party/webrtc/modules/audio_coding/test/TestVADDTX.cc
index 229dc2d4745..4f53e47cacc 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/TestVADDTX.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/TestVADDTX.cc
@@ -108,7 +108,9 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
     audio_frame.timestamp_ = time_stamp;
     time_stamp += frame_size_samples;
     EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
-    acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame);
+    bool muted;
+    acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted);
+    ASSERT_FALSE(muted);
     out_file.Write10MsData(audio_frame);
   }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/TwoWayCommunication.cc b/chromium/third_party/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
index 161491b0610..b59ff1fdccc 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
@@ -261,13 +261,18 @@ void TwoWayCommunication::Perform() {
 
     EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
     EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
-    EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame));
+    bool muted;
+    EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
     _outFileA.Write10MsData(audioFrame);
-    EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame));
+    EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
     _outFileRefA.Write10MsData(audioFrame);
-    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
     _outFileB.Write10MsData(audioFrame);
-    EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+    EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
     _outFileRefB.Write10MsData(audioFrame);
 
     // Update time counters each time a second of data has passed.
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/delay_test.cc b/chromium/third_party/webrtc/modules/audio_coding/test/delay_test.cc
index 8fa1fb1a3d0..50702f96b4e 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/delay_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/delay_test.cc
@@ -204,7 +204,10 @@ class DelayTest {
 
       in_file_a_.Read10MsData(audio_frame);
       ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
-      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+      bool muted;
+      ASSERT_EQ(0,
+                acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+      RTC_DCHECK(!muted);
       out_file_b_.Write10MsData(
           audio_frame.data_,
           audio_frame.samples_per_channel_ * audio_frame.num_channels_);
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/iSACTest.cc b/chromium/third_party/webrtc/modules/audio_coding/test/iSACTest.cc
index 9f223fb81fb..f06833c2d90 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/iSACTest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/iSACTest.cc
@@ -26,7 +26,6 @@
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -199,9 +198,12 @@ void ISACTest::Run10ms() {
   EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
   EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
   EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
-  EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
+  bool muted;
+  EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame, &muted));
+  ASSERT_FALSE(muted);
   _outFileA.Write10MsData(audioFrame);
-  EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
+  EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame, &muted));
+  ASSERT_FALSE(muted);
   _outFileB.Write10MsData(audioFrame);
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc b/chromium/third_party/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
index 966f4c636c6..27a8833ac67 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -141,8 +141,10 @@ class InsertPacketWithTiming {
     // Is it time to pull audio?
     if (time_to_playout_audio_ms_ == 0) {
       time_to_playout_audio_ms_ = kPlayoutPeriodMs;
+      bool muted;
       receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
-                                    &frame_);
+                                    &frame_, &muted);
+      ASSERT_FALSE(muted);
       fwrite(frame_.data_, sizeof(frame_.data_[0]),
              frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
       *action |= kAudioPlayedOut;
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/opus_test.cc b/chromium/third_party/webrtc/modules/audio_coding/test/opus_test.cc
index 104b5e587b2..5d250280f45 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/opus_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/opus_test.cc
@@ -336,7 +336,10 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
     }
 
     // Run received side of ACM.
-    ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+    bool muted;
+    ASSERT_EQ(
+        0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
 
     // Write output speech to file.
     out_file_.Write10MsData(
diff --git a/chromium/third_party/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/chromium/third_party/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index 99c1c2da1ee..5de5bf262b2 100644
--- a/chromium/third_party/webrtc/modules/audio_coding/test/target_delay_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -150,8 +150,10 @@ class TargetDelayTest : public ::testing::Test {
   // Pull audio equivalent to the amount of audio in one RTP packet.
   void Pull() {
     AudioFrame frame;
+    bool muted;
     for (int k = 0; k < kNum10msPerFrame; ++k) {  // Pull one frame.
-      ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame));
+      ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame, &muted));
+      ASSERT_FALSE(muted);
       // Had to use ASSERT_TRUE, ASSERT_EQ generated error.
       ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
       ASSERT_EQ(1u, frame.num_channels_);
diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h b/chromium/third_party/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
index 5d58f42435e..e1c5aedc887 100644
--- a/chromium/third_party/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
+++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
 #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
@@ -25,8 +26,34 @@ public:
     // audio every time it's called.
     //
     // If it returns -1, the frame will not be added to the mix.
+    //
+    // NOTE: This function should not be called. It will remain for a short
+    // time so that subclasses can override it without getting warnings.
+    // TODO(henrik.lundin) Remove this function.
     virtual int32_t GetAudioFrame(int32_t id,
-                                  AudioFrame* audioFrame) = 0;
+                                  AudioFrame* audioFrame) {
+      RTC_CHECK(false);
+      return -1;
+    }
+
+
+    // The implementation of GetAudioFrameWithMuted should update audio_frame
+    // with new audio every time it's called. The return value will be
+    // interpreted as follows.
+    enum class AudioFrameInfo {
+      kNormal,  // The samples in audio_frame are valid and should be used.
+      kMuted,   // The samples in audio_frame should not be used, but should be
+                // implicitly interpreted as zero. Other fields in audio_frame
+                // may be read and should contain meaningful values.
+      kError    // audio_frame will not be used.
+    };
+
+    virtual AudioFrameInfo GetAudioFrameWithMuted(int32_t id,
+                                                  AudioFrame* audio_frame) {
+      return GetAudioFrame(id, audio_frame) == -1 ?
+          AudioFrameInfo::kError :
+          AudioFrameInfo::kNormal;
+    }
 
     // Returns true if the participant was mixed this mix iteration.
     bool IsMixed() const;
diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index 1d7602533e4..dce3d0b5451 100644
--- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -19,12 +19,15 @@
 namespace webrtc {
 namespace {
 
-struct ParticipantFramePair {
+struct ParticipantFrameStruct {
+  ParticipantFrameStruct(MixerParticipant* p, AudioFrame* a, bool m)
+      : participant(p), audioFrame(a), muted(m) {}
   MixerParticipant* participant;
   AudioFrame* audioFrame;
+  bool muted;
 };
 
-typedef std::list<ParticipantFramePair*> ParticipantFramePairList;
+typedef std::list<ParticipantFrameStruct*> ParticipantFrameStructList;
 
 // Mix |frame| into |mixed_frame|, with saturation protection and upmixing.
 // These effects are applied to |frame| itself prior to mixing. Assumes that
@@ -55,7 +58,7 @@ size_t MaxNumChannels(const AudioFrameList* list) {
   for (AudioFrameList::const_iterator iter = list->begin();
        iter != list->end();
        ++iter) {
-    max_num_channels = std::max(max_num_channels, (*iter)->num_channels_);
+    max_num_channels = std::max(max_num_channels, (*iter).frame->num_channels_);
   }
   return max_num_channels;
 }
@@ -529,8 +532,8 @@ void AudioConferenceMixerImpl::UpdateToMix(
     AudioFrameList activeList;
     // Struct needed by the passive lists to keep track of which AudioFrame
     // belongs to which MixerParticipant.
-    ParticipantFramePairList passiveWasNotMixedList;
-    ParticipantFramePairList passiveWasMixedList;
+    ParticipantFrameStructList passiveWasNotMixedList;
+    ParticipantFrameStructList passiveWasMixedList;
     for (MixerParticipantList::const_iterator participant =
         _participantList.begin(); participant != _participantList.end();
          ++participant) {
@@ -552,12 +555,14 @@ void AudioConferenceMixerImpl::UpdateToMix(
         }
         audioFrame->sample_rate_hz_ = _outputFrequency;
 
-        if((*participant)->GetAudioFrame(_id, audioFrame) != 0) {
+        auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
+        if (ret == MixerParticipant::AudioFrameInfo::kError) {
             WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
-                         "failed to GetAudioFrame() from participant");
+                         "failed to GetAudioFrameWithMuted() from participant");
             _audioFramePool->PushMemory(audioFrame);
             continue;
         }
+        const bool muted = (ret == MixerParticipant::AudioFrameInfo::kMuted);
         if (_participantList.size() != 1) {
           // TODO(wu): Issue 3390, add support for multiple participants case.
           audioFrame->ntp_time_ms_ = -1;
@@ -573,7 +578,7 @@ void AudioConferenceMixerImpl::UpdateToMix(
         }
 
         if(audioFrame->vad_activity_ == AudioFrame::kVadActive) {
-            if(!wasMixed) {
+            if(!wasMixed && !muted) {
                 RampIn(*audioFrame);
             }
 
@@ -581,13 +586,15 @@ void AudioConferenceMixerImpl::UpdateToMix(
                 // There are already more active participants than should be
                 // mixed. Only keep the ones with the highest energy.
                 AudioFrameList::iterator replaceItem;
-                uint32_t lowestEnergy = CalculateEnergy(*audioFrame);
+                uint32_t lowestEnergy =
+                    muted ? 0 : CalculateEnergy(*audioFrame);
 
                 bool found_replace_item = false;
                 for (AudioFrameList::iterator iter = activeList.begin();
                      iter != activeList.end();
                      ++iter) {
-                    const uint32_t energy = CalculateEnergy(**iter);
+                    const uint32_t energy =
+                        muted ? 0 : CalculateEnergy(*iter->frame);
                     if(energy < lowestEnergy) {
                         replaceItem = iter;
                         lowestEnergy = energy;
@@ -595,11 +602,12 @@ void AudioConferenceMixerImpl::UpdateToMix(
                     }
                 }
                 if(found_replace_item) {
-                    AudioFrame* replaceFrame = *replaceItem;
+                    RTC_DCHECK(!muted);  // Cannot replace with a muted frame.
+                    FrameAndMuteInfo replaceFrame = *replaceItem;
 
                     bool replaceWasMixed = false;
                     std::map<int, MixerParticipant*>::const_iterator it =
-                        mixParticipantList->find(replaceFrame->id_);
+                        mixParticipantList->find(replaceFrame.frame->id_);
 
                     // When a frame is pushed to |activeList| it is also pushed
                     // to mixParticipantList with the frame's id. This means
@@ -607,26 +615,31 @@ void AudioConferenceMixerImpl::UpdateToMix(
                     assert(it != mixParticipantList->end());
                     replaceWasMixed = it->second->_mixHistory->WasMixed();
 
-                    mixParticipantList->erase(replaceFrame->id_);
+                    mixParticipantList->erase(replaceFrame.frame->id_);
                     activeList.erase(replaceItem);
 
-                    activeList.push_front(audioFrame);
+                    activeList.push_front(FrameAndMuteInfo(audioFrame, muted));
                     (*mixParticipantList)[audioFrame->id_] = *participant;
                     assert(mixParticipantList->size() <=
                            kMaximumAmountOfMixedParticipants);
 
                     if (replaceWasMixed) {
-                      RampOut(*replaceFrame);
+                      if (!replaceFrame.muted) {
+                        RampOut(*replaceFrame.frame);
+                      }
                       rampOutList->push_back(replaceFrame);
                       assert(rampOutList->size() <=
                              kMaximumAmountOfMixedParticipants);
                     } else {
-                      _audioFramePool->PushMemory(replaceFrame);
+                      _audioFramePool->PushMemory(replaceFrame.frame);
                     }
                 } else {
                     if(wasMixed) {
-                        RampOut(*audioFrame);
-                        rampOutList->push_back(audioFrame);
+                        if (!muted) {
+                            RampOut(*audioFrame);
+                        }
+                        rampOutList->push_back(FrameAndMuteInfo(audioFrame,
+                                                                muted));
                         assert(rampOutList->size() <=
                                kMaximumAmountOfMixedParticipants);
                     } else {
@@ -634,23 +647,23 @@ void AudioConferenceMixerImpl::UpdateToMix(
                     }
                 }
             } else {
-                activeList.push_front(audioFrame);
+                activeList.push_front(FrameAndMuteInfo(audioFrame, muted));
                 (*mixParticipantList)[audioFrame->id_] = *participant;
                 assert(mixParticipantList->size() <=
                        kMaximumAmountOfMixedParticipants);
             }
         } else {
             if(wasMixed) {
-                ParticipantFramePair* pair = new ParticipantFramePair;
-                pair->audioFrame  = audioFrame;
-                pair->participant = *participant;
-                passiveWasMixedList.push_back(pair);
+                ParticipantFrameStruct* part_struct =
+                    new ParticipantFrameStruct(*participant, audioFrame, muted);
+                passiveWasMixedList.push_back(part_struct);
             } else if(mustAddToPassiveList) {
-                RampIn(*audioFrame);
-                ParticipantFramePair* pair = new ParticipantFramePair;
-                pair->audioFrame  = audioFrame;
-                pair->participant = *participant;
-                passiveWasNotMixedList.push_back(pair);
+                if (!muted) {
+                    RampIn(*audioFrame);
+                }
+                ParticipantFrameStruct* part_struct =
+                    new ParticipantFrameStruct(*participant, audioFrame, muted);
+                passiveWasNotMixedList.push_back(part_struct);
             } else {
                 _audioFramePool->PushMemory(audioFrame);
             }
@@ -668,11 +681,12 @@ void AudioConferenceMixerImpl::UpdateToMix(
     // Always mix a constant number of AudioFrames. If there aren't enough
     // active participants mix passive ones. Starting with those that was mixed
     // last iteration.
-    for (ParticipantFramePairList::const_iterator
+    for (ParticipantFrameStructList::const_iterator
         iter = passiveWasMixedList.begin(); iter != passiveWasMixedList.end();
          ++iter) {
         if(mixList->size() < *maxAudioFrameCounter + mixListStartSize) {
-            mixList->push_back((*iter)->audioFrame);
+            mixList->push_back(FrameAndMuteInfo((*iter)->audioFrame,
+                                                (*iter)->muted));
             (*mixParticipantList)[(*iter)->audioFrame->id_] =
                 (*iter)->participant;
             assert(mixParticipantList->size() <=
@@ -683,12 +697,13 @@ void AudioConferenceMixerImpl::UpdateToMix(
         delete *iter;
     }
     // And finally the ones that have not been mixed for a while.
-    for (ParticipantFramePairList::const_iterator iter =
+    for (ParticipantFrameStructList::const_iterator iter =
              passiveWasNotMixedList.begin();
          iter != passiveWasNotMixedList.end();
          ++iter) {
         if(mixList->size() <  *maxAudioFrameCounter + mixListStartSize) {
-          mixList->push_back((*iter)->audioFrame);
+          mixList->push_back(FrameAndMuteInfo((*iter)->audioFrame,
+                                              (*iter)->muted));
             (*mixParticipantList)[(*iter)->audioFrame->id_] =
                 (*iter)->participant;
             assert(mixParticipantList->size() <=
@@ -706,10 +721,10 @@ void AudioConferenceMixerImpl::GetAdditionalAudio(
     AudioFrameList* additionalFramesList) const {
     WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
                  "GetAdditionalAudio(additionalFramesList)");
-    // The GetAudioFrame() callback may result in the participant being removed
-    // from additionalParticipantList_. If that happens it will invalidate any
-    // iterators. Create a copy of the participants list such that the list of
-    // participants can be traversed safely.
+    // The GetAudioFrameWithMuted() callback may result in the participant being
+    // removed from additionalParticipantList_. If that happens it will
+    // invalidate any iterators. Create a copy of the participants list such
+    // that the list of participants can be traversed safely.
     MixerParticipantList additionalParticipantList;
     additionalParticipantList.insert(additionalParticipantList.begin(),
                                      _additionalParticipantList.begin(),
@@ -727,9 +742,10 @@ void AudioConferenceMixerImpl::GetAdditionalAudio(
             return;
         }
         audioFrame->sample_rate_hz_ = _outputFrequency;
-        if((*participant)->GetAudioFrame(_id, audioFrame) != 0) {
+        auto ret = (*participant)->GetAudioFrameWithMuted(_id, audioFrame);
+        if (ret == MixerParticipant::AudioFrameInfo::kError) {
             WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
-                         "failed to GetAudioFrame() from participant");
+                         "failed to GetAudioFrameWithMuted() from participant");
             _audioFramePool->PushMemory(audioFrame);
             continue;
         }
@@ -738,7 +754,8 @@ void AudioConferenceMixerImpl::GetAdditionalAudio(
             _audioFramePool->PushMemory(audioFrame);
             continue;
         }
-        additionalFramesList->push_back(audioFrame);
+        additionalFramesList->push_back(FrameAndMuteInfo(
+            audioFrame, ret == MixerParticipant::AudioFrameInfo::kMuted));
     }
 }
 
@@ -775,7 +792,7 @@ void AudioConferenceMixerImpl::ClearAudioFrameList(
     for (AudioFrameList::iterator iter = audioFrameList->begin();
          iter != audioFrameList->end();
          ++iter) {
-        _audioFramePool->PushMemory(*iter);
+        _audioFramePool->PushMemory(iter->frame);
     }
     audioFrameList->clear();
 }
@@ -834,8 +851,9 @@ int32_t AudioConferenceMixerImpl::MixFromList(
     uint32_t position = 0;
 
     if (_numMixedParticipants == 1) {
-      mixedAudio->timestamp_ = audioFrameList.front()->timestamp_;
-      mixedAudio->elapsed_time_ms_ = audioFrameList.front()->elapsed_time_ms_;
+      mixedAudio->timestamp_ = audioFrameList.front().frame->timestamp_;
+      mixedAudio->elapsed_time_ms_ =
+          audioFrameList.front().frame->elapsed_time_ms_;
     } else {
       // TODO(wu): Issue 3390.
       // Audio frame timestamp is only supported in one channel case.
@@ -857,7 +875,9 @@ int32_t AudioConferenceMixerImpl::MixFromList(
             assert(false);
             position = 0;
         }
-        MixFrames(mixedAudio, (*iter), use_limiter_);
+        if (!iter->muted) {
+          MixFrames(mixedAudio, iter->frame, use_limiter_);
+        }
 
         position++;
     }
@@ -877,7 +897,9 @@ int32_t AudioConferenceMixerImpl::MixAnonomouslyFromList(
     for (AudioFrameList::const_iterator iter = audioFrameList.begin();
          iter != audioFrameList.end();
          ++iter) {
-        MixFrames(mixedAudio, *iter, use_limiter_);
+        if (!iter->muted) {
+            MixFrames(mixedAudio, iter->frame, use_limiter_);
+        }
     }
     return 0;
 }
diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
index 22c6a235351..e726c08f75a 100644
--- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
@@ -25,7 +25,13 @@ namespace webrtc {
 class AudioProcessing;
 class CriticalSectionWrapper;
 
-typedef std::list<AudioFrame*> AudioFrameList;
+struct FrameAndMuteInfo {
+  FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
+  AudioFrame* frame;
+  bool muted;
+};
+
+typedef std::list<FrameAndMuteInfo> AudioFrameList;
 typedef std::list<MixerParticipant*> MixerParticipantList;
 
 // Cheshire cat implementation of MixerParticipant's non virtual functions.
diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
index 19f5bd8848a..30b2933b61c 100644
--- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
+++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
@@ -17,7 +18,7 @@ TimeScheduler::TimeScheduler(const int64_t periodicityInMs)
       _isStarted(false),
       _lastPeriodMark(),
       _periodicityInMs(periodicityInMs),
-      _periodicityInTicks(TickTime::MillisecondsToTicks(periodicityInMs)),
+      _periodicityInTicks(periodicityInMs * rtc::kNumNanosecsPerMillisec),
       _missedPeriods(0)
  {
  }
@@ -33,7 +34,7 @@ int32_t TimeScheduler::UpdateScheduler()
     if(!_isStarted)
     {
         _isStarted = true;
-        _lastPeriodMark = TickTime::Now();
+        _lastPeriodMark = rtc::TimeNanos();
         return 0;
     }
     // Don't perform any calculations until the debt of pending periods have
@@ -45,9 +46,9 @@ int32_t TimeScheduler::UpdateScheduler()
     }
 
     // Calculate the time that has past since previous call to this function.
-    TickTime tickNow = TickTime::Now();
-    TickInterval amassedTicks = tickNow - _lastPeriodMark;
-    int64_t amassedMs = amassedTicks.Milliseconds();
+    int64_t tickNow = rtc::TimeNanos();
+    int64_t amassedTicks = tickNow - _lastPeriodMark;
+    int64_t amassedMs = amassedTicks / rtc::kNumNanosecsPerMillisec;
 
     // Calculate the number of periods the time that has passed correspond to.
     int64_t periodsToClaim = amassedMs / _periodicityInMs;
@@ -89,10 +90,10 @@ int32_t TimeScheduler::TimeToNextUpdate(
 
     // Calculate the time (in ms) that has past since last call to
     // UpdateScheduler()
-    TickTime tickNow = TickTime::Now();
-    TickInterval ticksSinceLastUpdate = tickNow - _lastPeriodMark;
+    int64_t tickNow = rtc::TimeNanos();
+    int64_t ticksSinceLastUpdate = tickNow - _lastPeriodMark;
     const int64_t millisecondsSinceLastUpdate =
-        ticksSinceLastUpdate.Milliseconds();
+      ticksSinceLastUpdate / rtc::kNumNanosecsPerMillisec;
 
     updateTimeInMS = _periodicityInMs - millisecondsSinceLastUpdate;
     updateTimeInMS =  (updateTimeInMS < 0) ? 0 : updateTimeInMS;
diff --git a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.h b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
index 09d0caa66ae..d1897fa1001 100644
--- a/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
+++ b/chromium/third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
@@ -15,8 +15,6 @@
 #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
 #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
 
-#include "webrtc/system_wrappers/include/tick_util.h"
-
 namespace webrtc {
 class CriticalSectionWrapper;
 class TimeScheduler
@@ -36,7 +34,7 @@ private:
     CriticalSectionWrapper* _crit;
 
     bool _isStarted;
-    TickTime _lastPeriodMark;
+    int64_t _lastPeriodMark; // In ns
 
     int64_t _periodicityInMs;
     int64_t _periodicityInTicks;
diff --git a/chromium/third_party/webrtc/modules/audio_decoder_unittests_apk.isolate b/chromium/third_party/webrtc/modules/audio_decoder_unittests_apk.isolate
new file mode 100644
index 00000000000..427559dcb05
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_decoder_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'audio_decoder_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_audio_decoder_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/audio_decoder_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_audio_decoder_unittests',
+      'audio_decoder_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/modules/audio_device/BUILD.gn b/chromium/third_party/webrtc/modules/audio_device/BUILD.gn
index 198b67d5200..d743a6a58f7 100644
--- a/chromium/third_party/webrtc/modules/audio_device/BUILD.gn
+++ b/chromium/third_party/webrtc/modules/audio_device/BUILD.gn
@@ -138,7 +138,7 @@ source_set("audio_device") {
       ]
     }
     if (is_ios) {
-      deps += [ "../../base:rtc_base_objc" ]
+      deps += [ "../../sdk:rtc_sdk_common_objc" ]
       sources += [
         "ios/audio_device_ios.h",
         "ios/audio_device_ios.mm",
diff --git a/chromium/third_party/webrtc/modules/audio_device/android/audio_device_unittest.cc b/chromium/third_party/webrtc/modules/audio_device/android/audio_device_unittest.cc
index 7655c820ada..48944f57726 100644
--- a/chromium/third_party/webrtc/modules/audio_device/android/audio_device_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -562,7 +562,7 @@ class AudioDeviceTest : public ::testing::Test {
   rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
       AudioDeviceModule::AudioLayer audio_layer) {
     rtc::scoped_refptr<AudioDeviceModule> module(
-        AudioDeviceModuleImpl::Create(0, audio_layer));
+        AudioDeviceModule::Create(0, audio_layer));
     return module;
   }
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/android/audio_manager.cc b/chromium/third_party/webrtc/modules/audio_device/android/audio_manager.cc
index 9174a5b7ab1..01e5d5fe4f2 100644
--- a/chromium/third_party/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/android/audio_manager.cc
@@ -66,7 +66,7 @@ bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() {
 
 // AudioManager implementation
 AudioManager::AudioManager()
-    : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())),
+    : j_environment_(JVM::GetInstance()->environment()),
       audio_layer_(AudioDeviceModule::kPlatformDefaultAudio),
       initialized_(false),
       hardware_aec_(false),
@@ -80,14 +80,14 @@ AudioManager::AudioManager()
       {"nativeCacheAudioParameters",
        "(IIZZZZIIJ)V",
        reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
-  j_native_registration_ = rtc::ScopedToUnique(j_environment_->RegisterNatives(
-      "org/webrtc/voiceengine/WebRtcAudioManager",
-      native_methods, arraysize(native_methods)));
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioManager", native_methods,
+      arraysize(native_methods));
   j_audio_manager_.reset(new JavaAudioManager(
       j_native_registration_.get(),
-      rtc::ScopedToUnique(j_native_registration_->NewObject(
+      j_native_registration_->NewObject(
           "<init>", "(Landroid/content/Context;J)V",
-          JVM::GetInstance()->context(), PointerTojlong(this)))));
+          JVM::GetInstance()->context(), PointerTojlong(this))));
 }
 
 AudioManager::~AudioManager() {
diff --git a/chromium/third_party/webrtc/modules/audio_device/android/build_info.cc b/chromium/third_party/webrtc/modules/audio_device/android/build_info.cc
index c6cecc96c5b..455c12f7fd0 100644
--- a/chromium/third_party/webrtc/modules/audio_device/android/build_info.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/android/build_info.cc
@@ -15,10 +15,9 @@
 namespace webrtc {
 
 BuildInfo::BuildInfo()
-    : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())),
-      j_build_info_(JVM::GetInstance()->GetClass(
-          "org/webrtc/voiceengine/BuildInfo")) {
-}
+    : j_environment_(JVM::GetInstance()->environment()),
+      j_build_info_(
+          JVM::GetInstance()->GetClass("org/webrtc/voiceengine/BuildInfo")) {}
 
 std::string BuildInfo::GetStringFromJava(const char* name) {
   jmethodID id = j_build_info_.GetStaticMethodId(name, "()Ljava/lang/String;");
diff --git a/chromium/third_party/webrtc/modules/audio_device/android/opensles_player.h b/chromium/third_party/webrtc/modules/audio_device/android/opensles_player.h
index 084546dbf7b..4058ff99948 100644
--- a/chromium/third_party/webrtc/modules/audio_device/android/opensles_player.h
+++ b/chromium/third_party/webrtc/modules/audio_device/android/opensles_player.h
@@ -132,7 +132,7 @@ class OpenSLESPlayer {
   const AudioParameters audio_parameters_;
 
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
-  // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
   AudioDeviceBuffer* audio_device_buffer_;
 
   bool initialized_;
diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi b/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi
index abefcdb4507..a7c57896585 100644
--- a/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi
+++ b/chromium/third_party/webrtc/modules/audio_device/audio_device.gypi
@@ -167,7 +167,10 @@
             }],
             ['OS=="ios"', {
               'dependencies': [
-                '<(webrtc_root)/base/base.gyp:rtc_base_objc',
+                '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
+              ],
+              'export_dependent_settings': [
+                '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
               ],
               'sources': [
                 'ios/audio_device_ios.h',
diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc
index d881bcd40c9..7abc94d5f77 100644
--- a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.cc
@@ -9,10 +9,11 @@
  */
 
 #include "webrtc/base/refcount.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 #include <assert.h>
 #include <string.h>
@@ -75,12 +76,12 @@ namespace webrtc {
 //  AudioDeviceModule::Create()
 // ----------------------------------------------------------------------------
 
-rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModuleImpl::Create(
+rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
     const int32_t id,
-    const AudioLayer audioLayer) {
+    const AudioLayer audio_layer) {
     // Create the generic ref counted (platform independent) implementation.
     rtc::scoped_refptr<AudioDeviceModuleImpl> audioDevice(
-        new rtc::RefCountedObject<AudioDeviceModuleImpl>(id, audioLayer));
+        new rtc::RefCountedObject<AudioDeviceModuleImpl>(id, audio_layer));
 
     // Ensure that the current platform is supported.
     if (audioDevice->CheckPlatform() == -1)
@@ -122,7 +123,7 @@ AudioDeviceModuleImpl::AudioDeviceModuleImpl(const int32_t id, const AudioLayer
     _ptrAudioDevice(NULL),
     _id(id),
     _platformAudioLayer(audioLayer),
-    _lastProcessTime(TickTime::MillisecondTimestamp()),
+    _lastProcessTime(rtc::TimeMillis()),
     _platformType(kPlatformNotSupported),
     _initialized(false),
     _lastError(kAdmErrNone)
@@ -406,7 +407,7 @@ AudioDeviceModuleImpl::~AudioDeviceModuleImpl()
 
 int64_t AudioDeviceModuleImpl::TimeUntilNextProcess()
 {
-    int64_t now = TickTime::MillisecondTimestamp();
+    int64_t now = rtc::TimeMillis();
     int64_t deltaProcess = kAdmMaxIdleTimeProcess - (now - _lastProcessTime);
     return deltaProcess;
 }
@@ -421,7 +422,7 @@ int64_t AudioDeviceModuleImpl::TimeUntilNextProcess()
 void AudioDeviceModuleImpl::Process()
 {
 
-    _lastProcessTime = TickTime::MillisecondTimestamp();
+    _lastProcessTime = rtc::TimeMillis();
 
     // kPlayoutWarning
     if (_ptrAudioDevice->PlayoutWarning())
@@ -1454,6 +1455,7 @@ int32_t AudioDeviceModuleImpl::InitPlayout()
 
 int32_t AudioDeviceModuleImpl::InitRecording()
 {
+    TRACE_EVENT0("webrtc", "AudioDeviceModuleImpl::InitRecording");
     CHECK_INITIALIZED();
     _audioDeviceBuffer.InitRecording();
     return (_ptrAudioDevice->InitRecording());
@@ -1515,6 +1517,7 @@ bool AudioDeviceModuleImpl::Playing() const
 
 int32_t AudioDeviceModuleImpl::StartRecording()
 {
+    TRACE_EVENT0("webrtc", "AudioDeviceModuleImpl::StartRecording");
     CHECK_INITIALIZED();
     return (_ptrAudioDevice->StartRecording());
 }
diff --git a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h
index a112e3e3bf6..044ec4ed6b5 100644
--- a/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_device/audio_device_impl.h
@@ -16,7 +16,6 @@
 #include <memory>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
@@ -48,11 +47,6 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
   int64_t TimeUntilNextProcess() override;
   void Process() override;
 
-  // Factory methods (resource allocation/deallocation)
-  static rtc::scoped_refptr<AudioDeviceModule> Create(
-      const int32_t id,
-      const AudioLayer audioLayer = kPlatformDefaultAudio);
-
   // Retrieve the currently utilized audio layer
   int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
index 2a6ac1ffe9c..7c6d16f129d 100644
--- a/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_device/dummy/file_audio_device_factory.h"
 
+#include <cstdlib>
 #include <cstring>
 
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
@@ -26,7 +27,7 @@ FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice(
   if (!_isConfigured) {
     printf("Was compiled with WEBRTC_DUMMY_AUDIO_PLAY_STATIC_FILE "
            "but did not set input/output files to use. Bailing out.\n");
-    exit(1);
+    std::exit(1);
   }
   return new FileAudioDevice(id, _inputAudioFilename, _outputAudioFilename);
 }
@@ -45,7 +46,7 @@ void FileAudioDeviceFactory::SetFilenamesToUse(
   // Sanity: must be compiled with the right define to run this.
   printf("Trying to use dummy file devices, but is not compiled "
          "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
-  exit(1);
+  std::exit(1);
 #endif
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h b/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h
index d8df05cec89..8457a6b7718 100644
--- a/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h
+++ b/chromium/third_party/webrtc/modules/audio_device/include/audio_device.h
@@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
 #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
 
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/include/module.h"
 
@@ -51,6 +52,11 @@ class AudioDeviceModule : public RefCountedModule {
   };
 
  public:
+  // Create an ADM.
+  static rtc::scoped_refptr<AudioDeviceModule> Create(
+      const int32_t id,
+      const AudioLayer audio_layer);
+
   // Retrieve the currently utilized audio layer
   virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h
index bfa6372203e..9616a285c0a 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.h
@@ -13,8 +13,7 @@
 
 #include <memory>
 
-#include "webrtc/base/asyncinvoker.h"
-#include "webrtc/base/objc/RTCMacros.h"
+#include "WebRTC/RTCMacros.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
@@ -42,7 +41,8 @@ class FineAudioBuffer;
 // same thread.
 class AudioDeviceIOS : public AudioDeviceGeneric,
                        public AudioSessionObserver,
-                       public VoiceProcessingAudioUnitObserver {
+                       public VoiceProcessingAudioUnitObserver,
+                       public rtc::MessageHandler {
  public:
   AudioDeviceIOS();
   ~AudioDeviceIOS();
@@ -162,7 +162,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   void OnInterruptionBegin() override;
   void OnInterruptionEnd() override;
   void OnValidRouteChange() override;
-  void OnConfiguredForWebRTC() override;
+  void OnCanPlayOrRecordChange(bool can_play_or_record) override;
 
   // VoiceProcessingAudioUnitObserver methods.
   OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
@@ -176,12 +176,16 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
                             UInt32 num_frames,
                             AudioBufferList* io_data) override;
 
+  // Handles messages from posts.
+  void OnMessage(rtc::Message *msg) override;
+
  private:
   // Called by the relevant AudioSessionObserver methods on |thread_|.
   void HandleInterruptionBegin();
   void HandleInterruptionEnd();
   void HandleValidRouteChange();
-  void HandleConfiguredForWebRTC();
+  void HandleCanPlayOrRecordChange(bool can_play_or_record);
+  void HandleSampleRateChange(float sample_rate);
 
   // Uses current |playout_parameters_| and |record_parameters_| to inform the
   // audio device buffer (ADB) about our internal audio parameters.
@@ -197,9 +201,13 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   // Creates the audio unit.
   bool CreateAudioUnit();
 
-  // Restarts active audio streams using a new sample rate. Required when e.g.
-  // a BT headset is enabled or disabled.
-  bool RestartAudioUnit(float sample_rate);
+  // Updates the audio unit state based on current state.
+  void UpdateAudioUnit(bool can_play_or_record);
+
+  // Configures the audio session for WebRTC.
+  void ConfigureAudioSession();
+  // Unconfigures the audio session.
+  void UnconfigureAudioSession();
 
   // Activates our audio session, creates and initializes the voice-processing
   // audio unit and verifies that we got the preferred native audio parameters.
@@ -213,11 +221,9 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
   rtc::ThreadChecker thread_checker_;
   // Thread that this object is created on.
   rtc::Thread* thread_;
-  // Invoker used to execute methods on thread_.
-  std::unique_ptr<rtc::AsyncInvoker> async_invoker_;
 
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
-  // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
   // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
   // and therefore outlives this object.
   AudioDeviceBuffer* audio_device_buffer_;
@@ -284,6 +290,9 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
 
   // Audio interruption observer instance.
   RTCAudioSessionDelegateAdapter* audio_session_observer_;
+
+  // Set to true if we've activated the audio session.
+  bool has_configured_session_;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.mm b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.mm
index f6c339fed00..8f6fb4d9b63 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -27,7 +27,7 @@
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
 #include "webrtc/modules/utility/include/helpers_ios.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
@@ -61,6 +61,13 @@ namespace webrtc {
 const UInt16 kFixedPlayoutDelayEstimate = 30;
 const UInt16 kFixedRecordDelayEstimate = 30;
 
+enum AudioDeviceMessageType : uint32_t {
+  kMessageTypeInterruptionBegin,
+  kMessageTypeInterruptionEnd,
+  kMessageTypeValidRouteChange,
+  kMessageTypeCanPlayOrRecordChange,
+};
+
 using ios::CheckAndLogError;
 
 #if !defined(NDEBUG)
@@ -85,15 +92,15 @@ static void LogDeviceInfo() {
 #endif  // !defined(NDEBUG)
 
 AudioDeviceIOS::AudioDeviceIOS()
-    : async_invoker_(new rtc::AsyncInvoker()),
-      audio_device_buffer_(nullptr),
+    : audio_device_buffer_(nullptr),
       audio_unit_(nullptr),
       recording_(0),
       playing_(0),
       initialized_(false),
       rec_is_initialized_(false),
       play_is_initialized_(false),
-      is_interrupted_(false) {
+      is_interrupted_(false),
+      has_configured_session_(false) {
   LOGI() << "ctor" << ios::GetCurrentThreadDescription();
   thread_ = rtc::Thread::Current();
   audio_session_observer_ =
@@ -191,6 +198,7 @@ int32_t AudioDeviceIOS::StartPlayout() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(play_is_initialized_);
   RTC_DCHECK(!playing_);
+  RTC_DCHECK(audio_unit_);
   if (fine_audio_buffer_) {
     fine_audio_buffer_->ResetPlayout();
   }
@@ -209,7 +217,11 @@ int32_t AudioDeviceIOS::StartPlayout() {
 int32_t AudioDeviceIOS::StopPlayout() {
   LOGI() << "StopPlayout";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!play_is_initialized_ || !playing_) {
+  if (!play_is_initialized_) {
+    return 0;
+  }
+  if (!playing_) {
+    play_is_initialized_ = false;
     return 0;
   }
   if (!recording_) {
@@ -225,6 +237,7 @@ int32_t AudioDeviceIOS::StartRecording() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   RTC_DCHECK(rec_is_initialized_);
   RTC_DCHECK(!recording_);
+  RTC_DCHECK(audio_unit_);
   if (fine_audio_buffer_) {
     fine_audio_buffer_->ResetRecord();
   }
@@ -243,7 +256,11 @@ int32_t AudioDeviceIOS::StartRecording() {
 int32_t AudioDeviceIOS::StopRecording() {
   LOGI() << "StopRecording";
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (!rec_is_initialized_ || !recording_) {
+  if (!rec_is_initialized_) {
+    return 0;
+  }
+  if (!recording_) {
+    rec_is_initialized_ = false;
     return 0;
   }
   if (!playing_) {
@@ -318,51 +335,24 @@ int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
 }
 
 void AudioDeviceIOS::OnInterruptionBegin() {
-  RTC_DCHECK(async_invoker_);
   RTC_DCHECK(thread_);
-  if (thread_->IsCurrent()) {
-    HandleInterruptionBegin();
-    return;
-  }
-  async_invoker_->AsyncInvoke<void>(
-      thread_,
-      rtc::Bind(&webrtc::AudioDeviceIOS::HandleInterruptionBegin, this));
+  thread_->Post(this, kMessageTypeInterruptionBegin);
 }
 
 void AudioDeviceIOS::OnInterruptionEnd() {
-  RTC_DCHECK(async_invoker_);
   RTC_DCHECK(thread_);
-  if (thread_->IsCurrent()) {
-    HandleInterruptionEnd();
-    return;
-  }
-  async_invoker_->AsyncInvoke<void>(
-      thread_,
-      rtc::Bind(&webrtc::AudioDeviceIOS::HandleInterruptionEnd, this));
+  thread_->Post(this, kMessageTypeInterruptionEnd);
 }
 
 void AudioDeviceIOS::OnValidRouteChange() {
-  RTC_DCHECK(async_invoker_);
   RTC_DCHECK(thread_);
-  if (thread_->IsCurrent()) {
-    HandleValidRouteChange();
-    return;
-  }
-  async_invoker_->AsyncInvoke<void>(
-      thread_,
-      rtc::Bind(&webrtc::AudioDeviceIOS::HandleValidRouteChange, this));
+  thread_->Post(this, kMessageTypeValidRouteChange);
 }
 
-void AudioDeviceIOS::OnConfiguredForWebRTC() {
-  RTC_DCHECK(async_invoker_);
+void AudioDeviceIOS::OnCanPlayOrRecordChange(bool can_play_or_record) {
   RTC_DCHECK(thread_);
-  if (thread_->IsCurrent()) {
-    HandleValidRouteChange();
-    return;
-  }
-  async_invoker_->AsyncInvoke<void>(
-      thread_,
-      rtc::Bind(&webrtc::AudioDeviceIOS::HandleConfiguredForWebRTC, this));
+  thread_->Post(this, kMessageTypeCanPlayOrRecordChange,
+                new rtc::TypedMessageData<bool>(can_play_or_record));
 }
 
 OSStatus AudioDeviceIOS::OnDeliverRecordedData(
@@ -385,6 +375,9 @@ OSStatus AudioDeviceIOS::OnDeliverRecordedData(
     RTCLogWarning(@"Expected %u frames but got %u",
                   static_cast<unsigned int>(frames_per_buffer),
                   static_cast<unsigned int>(num_frames));
+
+    RTCAudioSession *session = [RTCAudioSession sharedInstance];
+    RTCLogWarning(@"Session:\n %@", session);
     return result;
   }
 
@@ -447,12 +440,36 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
   return noErr;
 }
 
+void AudioDeviceIOS::OnMessage(rtc::Message *msg) {
+  switch (msg->message_id) {
+    case kMessageTypeInterruptionBegin:
+      HandleInterruptionBegin();
+      break;
+    case kMessageTypeInterruptionEnd:
+      HandleInterruptionEnd();
+      break;
+    case kMessageTypeValidRouteChange:
+      HandleValidRouteChange();
+      break;
+    case kMessageTypeCanPlayOrRecordChange: {
+      rtc::TypedMessageData<bool>* data =
+          static_cast<rtc::TypedMessageData<bool>*>(msg->pdata);
+      HandleCanPlayOrRecordChange(data->data());
+      delete data;
+      break;
+    }
+  }
+}
+
 void AudioDeviceIOS::HandleInterruptionBegin() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
-  RTCLog(@"Stopping the audio unit due to interruption begin.");
-  if (!audio_unit_->Stop()) {
-    RTCLogError(@"Failed to stop the audio unit.");
+  if (audio_unit_ &&
+      audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
+    RTCLog(@"Stopping the audio unit due to interruption begin.");
+    if (!audio_unit_->Stop()) {
+      RTCLogError(@"Failed to stop the audio unit for interruption begin.");
+    }
   }
   is_interrupted_ = true;
 }
@@ -460,66 +477,95 @@ void AudioDeviceIOS::HandleInterruptionBegin() {
 void AudioDeviceIOS::HandleInterruptionEnd() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
-  RTCLog(@"Starting the audio unit due to interruption end.");
-  if (!audio_unit_->Start()) {
-    RTCLogError(@"Failed to start the audio unit.");
-  }
   is_interrupted_ = false;
+  RTCLog(@"Interruption ended. Updating audio unit state.");
+  UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord);
 }
 
 void AudioDeviceIOS::HandleValidRouteChange() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 
+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
+  HandleSampleRateChange(session.sampleRate);
+}
+
+void AudioDeviceIOS::HandleCanPlayOrRecordChange(bool can_play_or_record) {
+  RTCLog(@"Handling CanPlayOrRecord change to: %d", can_play_or_record);
+  UpdateAudioUnit(can_play_or_record);
+}
+
+void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTCLog(@"Handling sample rate change to %f.", sample_rate);
+
   // Don't do anything if we're interrupted.
   if (is_interrupted_) {
+    RTCLog(@"Ignoring sample rate change to %f due to interruption.",
+           sample_rate);
     return;
   }
 
-  // Only restart audio for a valid route change if the session sample rate
-  // has changed.
-  RTCAudioSession* session = [RTCAudioSession sharedInstance];
-  const double current_sample_rate = playout_parameters_.sample_rate();
-  const double session_sample_rate = session.sampleRate;
-  if (current_sample_rate != session_sample_rate) {
-    RTCLog(@"Route changed caused sample rate to change from %f to %f. "
-           "Restarting audio unit.", current_sample_rate, session_sample_rate);
-    if (!RestartAudioUnit(session_sample_rate)) {
-      RTCLogError(@"Audio restart failed.");
-    }
+  // If we don't have an audio unit yet, or the audio unit is uninitialized,
+  // there is no work to do.
+  if (!audio_unit_ ||
+      audio_unit_->GetState() < VoiceProcessingAudioUnit::kInitialized) {
+    return;
   }
-}
 
-void AudioDeviceIOS::HandleConfiguredForWebRTC() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-
-  // If we're not initialized we don't need to do anything. Audio unit will
-  // be initialized on initialization.
-  if (!rec_is_initialized_ && !play_is_initialized_)
+  // The audio unit is already initialized or started.
+  // Check to see if the sample rate or buffer size has changed.
+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
+  const double session_sample_rate = session.sampleRate;
+  const NSTimeInterval session_buffer_duration = session.IOBufferDuration;
+  const size_t session_frames_per_buffer =
+      static_cast<size_t>(session_sample_rate * session_buffer_duration + .5);
+  const double current_sample_rate = playout_parameters_.sample_rate();
+  const size_t current_frames_per_buffer =
+      playout_parameters_.frames_per_buffer();
+  RTCLog(@"Handling playout sample rate change to: %f\n"
+          "  Session sample rate: %f frames_per_buffer: %lu\n"
+          "  ADM sample rate: %f frames_per_buffer: %lu",
+         sample_rate,
+         session_sample_rate, (unsigned long)session_frames_per_buffer,
+         current_sample_rate, (unsigned long)current_frames_per_buffer);;
+
+  // Sample rate and buffer size are the same, no work to do.
+  if (abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON &&
+      current_frames_per_buffer == session_frames_per_buffer) {
     return;
+  }
 
-  // If we're initialized, we must have an audio unit.
-  RTC_DCHECK(audio_unit_);
+  // We need to adjust our format and buffer sizes.
+  // The stream format is about to be changed and it requires that we first
+  // stop and uninitialize the audio unit to deallocate its resources.
+  RTCLog(@"Stopping and uninitializing audio unit to adjust buffers.");
+  bool restart_audio_unit = false;
+  if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
+    audio_unit_->Stop();
+    restart_audio_unit = true;
+  }
+  if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+    audio_unit_->Uninitialize();
+  }
 
-  // Use configured audio session's settings to set up audio device buffer.
-  // TODO(tkchin): Use RTCAudioSessionConfiguration to pick up settings and
-  // pass it along.
+  // Allocate new buffers given the new stream format.
   SetupAudioBuffersForActiveAudioSession();
 
-  // Initialize the audio unit. This will affect any existing audio playback.
-  if (!audio_unit_->Initialize(playout_parameters_.sample_rate())) {
-    RTCLogError(@"Failed to initialize audio unit after configuration.");
+  // Initialize the audio unit again with the new sample rate.
+  RTC_DCHECK_EQ(playout_parameters_.sample_rate(), session_sample_rate);
+  if (!audio_unit_->Initialize(session_sample_rate)) {
+    RTCLogError(@"Failed to initialize the audio unit with sample rate: %f",
+                session_sample_rate);
     return;
   }
 
-  // If we haven't started playing or recording there's nothing more to do.
-  if (!playing_ && !recording_)
-    return;
-
-  // We are in a play or record state, start the audio unit.
-  if (!audio_unit_->Start()) {
-    RTCLogError(@"Failed to start audio unit after configuration.");
+  // Restart the audio unit if it was already running.
+  if (restart_audio_unit && !audio_unit_->Start()) {
+    RTCLogError(@"Failed to start audio unit with sample rate: %f",
+                session_sample_rate);
     return;
   }
+  RTCLog(@"Successfully handled sample rate change.");
 }
 
 void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
@@ -597,6 +643,7 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
   // at each input callback when calling AudioUnitRender().
   const int data_byte_size = record_parameters_.GetBytesPerBuffer();
   record_audio_buffer_.reset(new SInt8[data_byte_size]);
+  memset(record_audio_buffer_.get(), 0, data_byte_size);
   audio_record_buffer_list_.mNumberBuffers = 1;
   AudioBuffer* audio_buffer = &audio_record_buffer_list_.mBuffers[0];
   audio_buffer->mNumberChannels = record_parameters_.channels();
@@ -616,46 +663,117 @@ bool AudioDeviceIOS::CreateAudioUnit() {
   return true;
 }
 
-bool AudioDeviceIOS::RestartAudioUnit(float sample_rate) {
-  RTCLog(@"Restarting audio unit with new sample rate: %f", sample_rate);
+void AudioDeviceIOS::UpdateAudioUnit(bool can_play_or_record) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTCLog(@"Updating audio unit state. CanPlayOrRecord=%d IsInterrupted=%d",
+         can_play_or_record, is_interrupted_);
 
-  // Stop the active audio unit.
-  if (!audio_unit_->Stop()) {
-    RTCLogError(@"Failed to stop the audio unit.");
-    return false;
+  if (is_interrupted_) {
+    RTCLog(@"Ignoring audio unit update due to interruption.");
+    return;
   }
 
-  // The stream format is about to be changed and it requires that we first
-  // uninitialize it to deallocate its resources.
-  if (!audio_unit_->Uninitialize()) {
-    RTCLogError(@"Failed to uninitialize the audio unit.");
-    return false;
+  // If we're not initialized we don't need to do anything. Audio unit will
+  // be initialized on initialization.
+  if (!rec_is_initialized_ && !play_is_initialized_)
+    return;
+
+  // If we're initialized, we must have an audio unit.
+  RTC_DCHECK(audio_unit_);
+
+  bool should_initialize_audio_unit = false;
+  bool should_uninitialize_audio_unit = false;
+  bool should_start_audio_unit = false;
+  bool should_stop_audio_unit = false;
+
+  switch (audio_unit_->GetState()) {
+    case VoiceProcessingAudioUnit::kInitRequired:
+      RTC_NOTREACHED();
+      break;
+    case VoiceProcessingAudioUnit::kUninitialized:
+      should_initialize_audio_unit = can_play_or_record;
+      should_start_audio_unit = should_initialize_audio_unit &&
+          (playing_ || recording_);
+      break;
+    case VoiceProcessingAudioUnit::kInitialized:
+      should_start_audio_unit =
+          can_play_or_record && (playing_ || recording_);
+      should_uninitialize_audio_unit = !can_play_or_record;
+      break;
+    case VoiceProcessingAudioUnit::kStarted:
+      RTC_DCHECK(playing_ || recording_);
+      should_stop_audio_unit = !can_play_or_record;
+      should_uninitialize_audio_unit = should_stop_audio_unit;
+      break;
+  }
+
+  if (should_initialize_audio_unit) {
+    RTCLog(@"Initializing audio unit for UpdateAudioUnit");
+    ConfigureAudioSession();
+    SetupAudioBuffersForActiveAudioSession();
+    if (!audio_unit_->Initialize(playout_parameters_.sample_rate())) {
+      RTCLogError(@"Failed to initialize audio unit.");
+      return;
+    }
   }
 
-  // Allocate new buffers given the new stream format.
-  SetupAudioBuffersForActiveAudioSession();
+  if (should_start_audio_unit) {
+    RTCLog(@"Starting audio unit for UpdateAudioUnit");
+    if (!audio_unit_->Start()) {
+      RTCLogError(@"Failed to start audio unit.");
+      return;
+    }
+  }
 
-  // Initialize the audio unit again with the new sample rate.
-  RTC_DCHECK_EQ(playout_parameters_.sample_rate(), sample_rate);
-  if (!audio_unit_->Initialize(sample_rate)) {
-    RTCLogError(@"Failed to initialize the audio unit with sample rate: %f",
-                sample_rate);
-    return false;
+  if (should_stop_audio_unit) {
+    RTCLog(@"Stopping audio unit for UpdateAudioUnit");
+    if (!audio_unit_->Stop()) {
+      RTCLogError(@"Failed to stop audio unit.");
+      return;
+    }
   }
 
-  // Restart the audio unit.
-  if (!audio_unit_->Start()) {
-    RTCLogError(@"Failed to start audio unit.");
-    return false;
+  if (should_uninitialize_audio_unit) {
+    RTCLog(@"Uninitializing audio unit for UpdateAudioUnit");
+    audio_unit_->Uninitialize();
+    UnconfigureAudioSession();
   }
-  RTCLog(@"Successfully restarted audio unit.");
+}
 
-  return true;
+void AudioDeviceIOS::ConfigureAudioSession() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTCLog(@"Configuring audio session.");
+  if (has_configured_session_) {
+    RTCLogWarning(@"Audio session already configured.");
+    return;
+  }
+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
+  [session lockForConfiguration];
+  [session configureWebRTCSession:nil];
+  [session unlockForConfiguration];
+  has_configured_session_ = true;
+  RTCLog(@"Configured audio session.");
+}
+
+void AudioDeviceIOS::UnconfigureAudioSession() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTCLog(@"Unconfiguring audio session.");
+  if (!has_configured_session_) {
+    RTCLogWarning(@"Audio session already unconfigured.");
+    return;
+  }
+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
+  [session lockForConfiguration];
+  [session unconfigureWebRTCSession:nil];
+  [session unlockForConfiguration];
+  has_configured_session_ = false;
+  RTCLog(@"Unconfigured audio session.");
 }
 
 bool AudioDeviceIOS::InitPlayOrRecord() {
   LOGI() << "InitPlayOrRecord";
 
+  // There should be no audio unit at this point.
   if (!CreateAudioUnit()) {
     return false;
   }
@@ -674,14 +792,11 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
     return false;
   }
 
-  // If we are already configured properly, we can initialize the audio unit.
-  if (session.isConfiguredForWebRTC) {
-    [session unlockForConfiguration];
+  // If we are ready to play or record, initialize the audio unit.
+  if (session.canPlayOrRecord) {
+    ConfigureAudioSession();
     SetupAudioBuffersForActiveAudioSession();
-    // Audio session has been marked ready for WebRTC so we can initialize the
-    // audio unit now.
     audio_unit_->Initialize(playout_parameters_.sample_rate());
-    return true;
   }
 
   // Release the lock.
@@ -694,9 +809,7 @@ void AudioDeviceIOS::ShutdownPlayOrRecord() {
   LOGI() << "ShutdownPlayOrRecord";
 
   // Close and delete the voice-processing I/O unit.
-  if (audio_unit_) {
-    audio_unit_.reset();
-  }
+  audio_unit_.reset();
 
   // Remove audio session notification observers.
   RTCAudioSession* session = [RTCAudioSession sharedInstance];
@@ -705,6 +818,7 @@ void AudioDeviceIOS::ShutdownPlayOrRecord() {
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
   [session lockForConfiguration];
+  UnconfigureAudioSession();
   [session endWebRTCSession:nil];
   [session unlockForConfiguration];
 }
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
index 4dfb073fa9f..ec10119a113 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
@@ -537,7 +537,7 @@ class AudioDeviceTest : public ::testing::Test {
   rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
       AudioDeviceModule::AudioLayer audio_layer) {
     rtc::scoped_refptr<AudioDeviceModule> module(
-        AudioDeviceModuleImpl::Create(0, audio_layer));
+        AudioDeviceModule::Create(0, audio_layer));
     return module;
   }
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/audio_session_observer.h b/chromium/third_party/webrtc/modules/audio_device/ios/audio_session_observer.h
index 6c4a9cd9e27..def8c2322b0 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/audio_session_observer.h
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/audio_session_observer.h
@@ -28,8 +28,8 @@ class AudioSessionObserver {
   // Called when audio route changes.
   virtual void OnValidRouteChange() = 0;
 
-  // Called when audio session has been configured for WebRTC.
-  virtual void OnConfiguredForWebRTC() = 0;
+  // Called when the ability to play or record changes.
+  virtual void OnCanPlayOrRecordChange(bool can_play_or_record) = 0;
 
  protected:
   virtual ~AudioSessionObserver() {}
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm
index 06ddddd9bce..5a7600a5d39 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm
@@ -10,18 +10,34 @@
 
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
 @implementation RTCAudioSession (Configuration)
 
-- (BOOL)isConfiguredForWebRTC {
-  return self.savedConfiguration != nil;
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+                   error:(NSError **)outError {
+  return [self setConfiguration:configuration
+                         active:NO
+                shouldSetActive:NO
+                          error:outError];
+}
+
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError {
+  return [self setConfiguration:configuration
+                         active:active
+                shouldSetActive:YES
+                          error:outError];
 }
 
+#pragma mark - Private
+
 - (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
                   active:(BOOL)active
+         shouldSetActive:(BOOL)shouldSetActive
                    error:(NSError **)outError {
   NSParameterAssert(configuration);
   if (outError) {
@@ -61,8 +77,22 @@
     }
   }
 
-  // self.sampleRate is accurate only if the audio session is active.
-  if (!self.isActive || self.sampleRate != configuration.sampleRate) {
+  // Sometimes category options don't stick after setting mode.
+  if (self.categoryOptions != configuration.categoryOptions) {
+    NSError *categoryError = nil;
+    if (![self setCategory:configuration.category
+               withOptions:configuration.categoryOptions
+                     error:&categoryError]) {
+      RTCLogError(@"Failed to set category options: %@",
+                  categoryError.localizedDescription);
+      error = categoryError;
+    } else {
+      RTCLog(@"Set category options to: %ld",
+             (long)configuration.categoryOptions);
+    }
+  }
+
+  if (self.preferredSampleRate != configuration.sampleRate) {
     NSError *sampleRateError = nil;
     if (![self setPreferredSampleRate:configuration.sampleRate
                                 error:&sampleRateError]) {
@@ -75,9 +105,7 @@
     }
   }
 
-  // self.IOBufferDuration is accurate only if the audio session is active.
-  if (!self.isActive ||
-      self.IOBufferDuration != configuration.ioBufferDuration) {
+  if (self.preferredIOBufferDuration != configuration.ioBufferDuration) {
     NSError *bufferDurationError = nil;
     if (![self setPreferredIOBufferDuration:configuration.ioBufferDuration
                                       error:&bufferDurationError]) {
@@ -90,11 +118,13 @@
     }
   }
 
-  NSError *activeError = nil;
-  if (![self setActive:active error:&activeError]) {
-    RTCLogError(@"Failed to setActive to %d: %@",
-                active, activeError.localizedDescription);
-    error = activeError;
+  if (shouldSetActive) {
+    NSError *activeError = nil;
+    if (![self setActive:active error:&activeError]) {
+      RTCLogError(@"Failed to setActive to %d: %@",
+                  active, activeError.localizedDescription);
+      error = activeError;
+    }
   }
 
   if (self.isActive &&
@@ -138,84 +168,4 @@
   return error == nil;
 }
 
-- (BOOL)configureWebRTCSession:(NSError **)outError {
-  if (outError) {
-    *outError = nil;
-  }
-  if (![self checkLock:outError]) {
-    return NO;
-  }
-  RTCLog(@"Configuring audio session for WebRTC.");
-
-  if (self.isConfiguredForWebRTC) {
-    RTCLogError(@"Already configured.");
-    if (outError) {
-      *outError =
-          [self configurationErrorWithDescription:@"Already configured."];
-    }
-    return NO;
-  }
-
-  // Configure the AVAudioSession and activate it.
-  // Provide an error even if there isn't one so we can log it.
-  NSError *error = nil;
-  RTCAudioSessionConfiguration *currentConfig =
-      [RTCAudioSessionConfiguration currentConfiguration];
-  RTCAudioSessionConfiguration *webRTCConfig =
-      [RTCAudioSessionConfiguration webRTCConfiguration];
-  self.savedConfiguration = currentConfig;
-  if (![self setConfiguration:webRTCConfig active:YES error:&error]) {
-    RTCLogError(@"Failed to set WebRTC audio configuration: %@",
-                error.localizedDescription);
-    [self unconfigureWebRTCSession:nil];
-    if (outError) {
-      *outError = error;
-    }
-    return NO;
-  }
-
-  // Ensure that the device currently supports audio input.
-  // TODO(tkchin): Figure out if this is really necessary.
-  if (!self.inputAvailable) {
-    RTCLogError(@"No audio input path is available!");
-    [self unconfigureWebRTCSession:nil];
-    if (outError) {
-      *outError = [self configurationErrorWithDescription:@"No input path."];
-    }
-    return NO;
-  }
-
-  // Give delegates a chance to process the event. In particular, the audio
-  // devices listening to this event will initialize their audio units.
-  [self notifyDidConfigure];
-
-  return YES;
-}
-
-- (BOOL)unconfigureWebRTCSession:(NSError **)outError {
-  if (outError) {
-    *outError = nil;
-  }
-  if (![self checkLock:outError]) {
-    return NO;
-  }
-  RTCLog(@"Unconfiguring audio session for WebRTC.");
-
-  if (!self.isConfiguredForWebRTC) {
-    RTCLogError(@"Already unconfigured.");
-    if (outError) {
-      *outError =
-          [self configurationErrorWithDescription:@"Already unconfigured."];
-    }
-    return NO;
-  }
-
-  [self setConfiguration:self.savedConfiguration active:NO error:outError];
-  self.savedConfiguration = nil;
-
-  [self notifyDidUnconfigure];
-
-  return YES;
-}
-
 @end
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h
index c6738e7a033..cb506c345a9 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h
@@ -28,11 +28,8 @@ NS_ASSUME_NONNULL_BEGIN
  */
 @property(nonatomic, readonly) int webRTCSessionCount;
 
-/** The configuration of the audio session before configureWebRTCSession
- *  was first called.
- */
-@property(nonatomic, strong, nullable)
-    RTCAudioSessionConfiguration *savedConfiguration;
+/** Convenience BOOL that checks useManualAudio and isAudioEnebled. */
+@property(readonly) BOOL canPlayOrRecord;
 
 - (BOOL)checkLock:(NSError **)outError;
 
@@ -55,6 +52,22 @@ NS_ASSUME_NONNULL_BEGIN
  */
 - (BOOL)endWebRTCSession:(NSError **)outError;
 
+/** Configure the audio session for WebRTC. This call will fail if the session
+ *  is already configured. On other failures, we will attempt to restore the
+ *  previously used audio session configuration.
+ *  |lockForConfiguration| must be called first.
+ *  Successful calls to configureWebRTCSession must be matched by calls to
+ *  |unconfigureWebRTCSession|.
+ */
+- (BOOL)configureWebRTCSession:(NSError **)outError;
+
+/** Unconfigures the session for WebRTC. This will attempt to restore the
+ *  audio session to the settings used before |configureWebRTCSession| was
+ *  called.
+ *  |lockForConfiguration| must be called first.
+ */
+- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
+
 /** Returns a configuration error with the given description. */
 - (NSError *)configurationErrorWithDescription:(NSString *)description;
 
@@ -69,10 +82,9 @@ NS_ASSUME_NONNULL_BEGIN
     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
 - (void)notifyMediaServicesWereLost;
 - (void)notifyMediaServicesWereReset;
-- (void)notifyShouldConfigure;
-- (void)notifyShouldUnconfigure;
-- (void)notifyDidConfigure;
-- (void)notifyDidUnconfigure;
+- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+- (void)notifyDidStartPlayOrRecord;
+- (void)notifyDidStopPlayOrRecord;
 
 @end
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h
index ab8dbc8859e..274cc2bc978 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h
@@ -58,29 +58,18 @@ extern NSInteger const kRTCAudioSessionErrorConfiguration;
 
 // TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
 
-/** Called on a WebRTC thread when WebRTC needs to take over audio. Applications
- *  should call -[RTCAudioSession configureWebRTCSession] to allow WebRTC to
- *  play and record audio. Will only occur if shouldDelayAudioConfiguration is
- *  set to YES.
- */
-- (void)audioSessionShouldConfigure:(RTCAudioSession *)session;
-
-/** Called on a WebRTC thread when WebRTC no longer requires audio. Applications
- *  should call -[RTCAudioSession unconfigureWebRTCSession] to restore their
- *  audio session settings. Will only occur if shouldDelayAudioConfiguration is
- *  set to YES.
- */
-- (void)audioSessionShouldUnconfigure:(RTCAudioSession *)session;
+- (void)audioSession:(RTCAudioSession *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
 
-/** Called on a WebRTC thread when WebRTC has configured the audio session for
- *  WebRTC audio.
+/** Called on a WebRTC thread when the audio device is notified to begin
+ *  playback or recording.
  */
-- (void)audioSessionDidConfigure:(RTCAudioSession *)session;
+- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session;
 
-/** Called on a WebRTC thread when WebRTC has unconfigured the audio session for
- *  WebRTC audio.
+/** Called on a WebRTC thread when the audio device is notified to stop
+ *  playback or recording.
  */
-- (void)audioSessionDidUnconfigure:(RTCAudioSession *)session;
+- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
 
 @end
 
@@ -108,11 +97,24 @@ extern NSInteger const kRTCAudioSessionErrorConfiguration;
 
 /** If YES, WebRTC will not initialize the audio unit automatically when an
  *  audio track is ready for playout or recording. Instead, applications should
- *  listen to the delegate method |audioSessionShouldConfigure| and configure
- *  the session manually. This should be set before making WebRTC media calls
- *  and should not be changed while a call is active.
+ *  call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
+ *  as soon as an audio track is ready for playout or recording.
  */
-@property(nonatomic, assign) BOOL shouldDelayAudioConfiguration;
+@property(nonatomic, assign) BOOL useManualAudio;
+
+/** This property is only effective if useManualAudio is YES.
+ *  Represents permission for WebRTC to initialize the VoIP audio unit.
+ *  When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
+ *  stopped and uninitialized. This will stop incoming and outgoing audio.
+ *  When set to YES, WebRTC will initialize and start the audio unit when it is
+ *  needed (e.g. due to establishing an audio connection).
+ *  This property was introduced to work around an issue where if an AVPlayer is
+ *  playing audio while the VoIP audio unit is initialized, its audio would be
+ *  either cut off completely or played at a reduced volume. By preventing
+ *  the audio unit from being initialized until after the audio has completed,
+ *  we are able to prevent the abrupt cutoff.
+ */
+@property(nonatomic, assign) BOOL isAudioEnabled;
 
 // Proxy properties.
 @property(readonly) NSString *category;
@@ -134,12 +136,14 @@ extern NSInteger const kRTCAudioSessionErrorConfiguration;
 @property(readonly, nullable)
     AVAudioSessionDataSourceDescription *outputDataSource;
 @property(readonly) double sampleRate;
+@property(readonly) double preferredSampleRate;
 @property(readonly) NSInteger inputNumberOfChannels;
 @property(readonly) NSInteger outputNumberOfChannels;
 @property(readonly) float outputVolume;
 @property(readonly) NSTimeInterval inputLatency;
 @property(readonly) NSTimeInterval outputLatency;
 @property(readonly) NSTimeInterval IOBufferDuration;
+@property(readonly) NSTimeInterval preferredIOBufferDuration;
 
 /** Default constructor. */
 + (instancetype)sharedInstance;
@@ -196,36 +200,20 @@ extern NSInteger const kRTCAudioSessionErrorConfiguration;
 
 @interface RTCAudioSession (Configuration)
 
-/** Whether or not |configureWebRTCSession| has been called without a balanced
- *  call to |unconfigureWebRTCSession|. This is not an indication of whether the
- *  audio session has the right settings.
- */
-@property(readonly) BOOL isConfiguredForWebRTC;
-
 /** Applies the configuration to the current session. Attempts to set all
  *  properties even if previous ones fail. Only the last error will be
- *  returned. Also calls setActive with |active|.
+ *  returned.
  *  |lockForConfiguration| must be called first.
  */
 - (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
-                  active:(BOOL)active
                    error:(NSError **)outError;
 
-/** Configure the audio session for WebRTC. This call will fail if the session
- *  is already configured. On other failures, we will attempt to restore the
- *  previously used audio session configuration.
- *  |lockForConfiguration| must be called first.
- *  Successful calls to configureWebRTCSession must be matched by calls to
- *  |unconfigureWebRTCSession|.
- */
-- (BOOL)configureWebRTCSession:(NSError **)outError;
-
-/** Unconfigures the session for WebRTC. This will attempt to restore the
- *  audio session to the settings used before |configureWebRTCSession| was
- *  called.
+/** Convenience method that calls both setConfiguration and setActive.
  *  |lockForConfiguration| must be called first.
  */
-- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError;
 
 @end
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
index c6e3677b846..7ef5110f476 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm
@@ -15,8 +15,9 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
+#import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
 NSString * const kRTCAudioSessionErrorDomain = @"org.webrtc.RTCAudioSession";
 NSInteger const kRTCAudioSessionErrorLockRequired = -1;
@@ -32,12 +33,13 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   volatile int _lockRecursionCount;
   volatile int _webRTCSessionCount;
   BOOL _isActive;
-  BOOL _shouldDelayAudioConfiguration;
+  BOOL _useManualAudio;
+  BOOL _isAudioEnabled;
+  BOOL _canPlayOrRecord;
 }
 
 @synthesize session = _session;
 @synthesize delegates = _delegates;
-@synthesize savedConfiguration = _savedConfiguration;
 
 + (instancetype)sharedInstance {
   static dispatch_once_t onceToken;
@@ -81,6 +83,9 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
 - (NSString *)description {
   NSString *format =
       @"RTCAudioSession: {\n"
+       "  category: %@\n"
+       "  categoryOptions: %ld\n"
+       "  mode: %@\n"
        "  isActive: %d\n"
        "  sampleRate: %.2f\n"
        "  IOBufferDuration: %f\n"
@@ -90,6 +95,7 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
        "  inputLatency: %f\n"
        "}";
   NSString *description = [NSString stringWithFormat:format,
+      self.category, (long)self.categoryOptions, self.mode,
       self.isActive, self.sampleRate, self.IOBufferDuration,
       self.outputNumberOfChannels, self.inputNumberOfChannels,
       self.outputLatency, self.inputLatency];
@@ -112,20 +118,35 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   return _lockRecursionCount > 0;
 }
 
-- (void)setShouldDelayAudioConfiguration:(BOOL)shouldDelayAudioConfiguration {
+- (void)setUseManualAudio:(BOOL)useManualAudio {
   @synchronized(self) {
-    // No one should be changing this while an audio device is active.
-    RTC_DCHECK(!self.isConfiguredForWebRTC);
-    if (_shouldDelayAudioConfiguration == shouldDelayAudioConfiguration) {
+    if (_useManualAudio == useManualAudio) {
       return;
     }
-    _shouldDelayAudioConfiguration = shouldDelayAudioConfiguration;
+    _useManualAudio = useManualAudio;
   }
+  [self updateCanPlayOrRecord];
 }
 
-- (BOOL)shouldDelayAudioConfiguration {
+- (BOOL)useManualAudio {
   @synchronized(self) {
-    return _shouldDelayAudioConfiguration;
+    return _useManualAudio;
+  }
+}
+
+- (void)setIsAudioEnabled:(BOOL)isAudioEnabled {
+  @synchronized(self) {
+    if (_isAudioEnabled == isAudioEnabled) {
+      return;
+    }
+    _isAudioEnabled = isAudioEnabled;
+  }
+  [self updateCanPlayOrRecord];
+}
+
+- (BOOL)isAudioEnabled {
+  @synchronized(self) {
+    return _isAudioEnabled;
   }
 }
 
@@ -147,7 +168,8 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   @synchronized(self) {
     _delegates.erase(std::remove(_delegates.begin(),
                                  _delegates.end(),
-                                 delegate));
+                                 delegate),
+                     _delegates.end());
     [self removeZeroedDelegates];
   }
 }
@@ -231,6 +253,10 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   return self.session.sampleRate;
 }
 
+- (double)preferredSampleRate {
+  return self.session.preferredSampleRate;
+}
+
 - (NSInteger)inputNumberOfChannels {
   return self.session.inputNumberOfChannels;
 }
@@ -255,6 +281,10 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   return self.session.IOBufferDuration;
 }
 
+- (NSTimeInterval)preferredIOBufferDuration {
+  return self.session.preferredIOBufferDuration;
+}
+
 // TODO(tkchin): Simplify the amount of locking happening here. Likely that we
 // can just do atomic increments / decrements.
 - (BOOL)setActive:(BOOL)active
@@ -496,21 +526,6 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   }
 }
 
-- (void)setSavedConfiguration:(RTCAudioSessionConfiguration *)configuration {
-  @synchronized(self) {
-    if (_savedConfiguration == configuration) {
-      return;
-    }
-    _savedConfiguration = configuration;
-  }
-}
-
-- (RTCAudioSessionConfiguration *)savedConfiguration {
-  @synchronized(self) {
-    return _savedConfiguration;
-  }
-}
-
 // TODO(tkchin): check for duplicates.
 - (void)pushDelegate:(id<RTCAudioSessionDelegate>)delegate {
   @synchronized(self) {
@@ -520,11 +535,11 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
 
 - (void)removeZeroedDelegates {
   @synchronized(self) {
-    for (auto it = _delegates.begin(); it != _delegates.end(); ++it) {
-      if (!*it) {
-        _delegates.erase(it);
-      }
-    }
+    _delegates.erase(
+        std::remove_if(_delegates.begin(),
+                       _delegates.end(),
+                       [](id delegate) -> bool { return delegate == nil; }),
+        _delegates.end());
   }
 }
 
@@ -546,6 +561,10 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   return _webRTCSessionCount;
 }
 
+- (BOOL)canPlayOrRecord {
+  return !self.useManualAudio || self.isAudioEnabled;
+}
+
 - (BOOL)checkLock:(NSError **)outError {
   // Check ivar instead of trying to acquire lock so that we won't accidentally
   // acquire lock if it hasn't already been called.
@@ -565,79 +584,70 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   if (![self checkLock:outError]) {
     return NO;
   }
-  NSInteger sessionCount = rtc::AtomicOps::Increment(&_webRTCSessionCount);
-  if (sessionCount > 1) {
-    // Should already be configured.
-    RTC_DCHECK(self.isConfiguredForWebRTC);
-    return YES;
-  }
+  rtc::AtomicOps::Increment(&_webRTCSessionCount);
+  [self notifyDidStartPlayOrRecord];
+  return YES;
+}
 
-  // Only perform configuration steps once. Application might have already
-  // configured the session.
-  if (self.isConfiguredForWebRTC) {
-    // Nothing more to do, already configured.
-    return YES;
+- (BOOL)endWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
   }
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  rtc::AtomicOps::Decrement(&_webRTCSessionCount);
+  [self notifyDidStopPlayOrRecord];
+  return YES;
+}
 
-  // If application has prevented automatic configuration, return here and wait
-  // for application to call configureWebRTCSession.
-  if (self.shouldDelayAudioConfiguration) {
-    [self notifyShouldConfigure];
-    return YES;
+- (BOOL)configureWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  if (![self checkLock:outError]) {
+    return NO;
   }
+  RTCLog(@"Configuring audio session for WebRTC.");
 
-  // Configure audio session.
+  // Configure the AVAudioSession and activate it.
+  // Provide an error even if there isn't one so we can log it.
   NSError *error = nil;
-  if (![self configureWebRTCSession:&error]) {
-    RTCLogError(@"Error configuring audio session: %@",
+  RTCAudioSessionConfiguration *webRTCConfig =
+      [RTCAudioSessionConfiguration webRTCConfiguration];
+  if (![self setConfiguration:webRTCConfig active:YES error:&error]) {
+    RTCLogError(@"Failed to set WebRTC audio configuration: %@",
                 error.localizedDescription);
+    [self unconfigureWebRTCSession:nil];
     if (outError) {
       *outError = error;
     }
     return NO;
   }
 
+  // Ensure that the device currently supports audio input.
+  // TODO(tkchin): Figure out if this is really necessary.
+  if (!self.inputAvailable) {
+    RTCLogError(@"No audio input path is available!");
+    [self unconfigureWebRTCSession:nil];
+    if (outError) {
+      *outError = [self configurationErrorWithDescription:@"No input path."];
+    }
+    return NO;
+  }
+
   return YES;
 }
 
-- (BOOL)endWebRTCSession:(NSError **)outError {
+- (BOOL)unconfigureWebRTCSession:(NSError **)outError {
   if (outError) {
     *outError = nil;
   }
   if (![self checkLock:outError]) {
     return NO;
   }
-  int sessionCount = rtc::AtomicOps::Decrement(&_webRTCSessionCount);
-  RTC_DCHECK_GE(sessionCount, 0);
-  if (sessionCount != 0) {
-    // Should still be configured.
-    RTC_DCHECK(self.isConfiguredForWebRTC);
-    return YES;
-  }
-
-  // Only unconfigure if application has not done it.
-  if (!self.isConfiguredForWebRTC) {
-    // Nothing more to do, already unconfigured.
-    return YES;
-  }
-
-  // If application has prevented automatic configuration, return here and wait
-  // for application to call unconfigureWebRTCSession.
-  if (self.shouldDelayAudioConfiguration) {
-    [self notifyShouldUnconfigure];
-    return YES;
-  }
-
-  // Unconfigure audio session.
-  NSError *error = nil;
-  if (![self unconfigureWebRTCSession:&error]) {
-    RTCLogError(@"Error unconfiguring audio session: %@",
-                error.localizedDescription);
-    if (outError) {
-      *outError = error;
-    }
-    return NO;
-  }
+  RTCLog(@"Unconfiguring audio session for WebRTC.");
+  [self setActive:NO error:outError];
 
   return YES;
 }
@@ -666,6 +676,22 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   }
 }
 
+- (void)updateCanPlayOrRecord {
+  BOOL canPlayOrRecord = NO;
+  BOOL shouldNotify = NO;
+  @synchronized(self) {
+    canPlayOrRecord = !self.useManualAudio || self.isAudioEnabled;
+    if (_canPlayOrRecord == canPlayOrRecord) {
+      return;
+    }
+    _canPlayOrRecord = canPlayOrRecord;
+    shouldNotify = YES;
+  }
+  if (shouldNotify) {
+    [self notifyDidChangeCanPlayOrRecord:canPlayOrRecord];
+  }
+}
+
 - (void)notifyDidBeginInterruption {
   for (auto delegate : self.delegates) {
     SEL sel = @selector(audioSessionDidBeginInterruption:);
@@ -716,38 +742,29 @@ NSInteger const kRTCAudioSessionErrorConfiguration = -2;
   }
 }
 
-- (void)notifyShouldConfigure {
-  for (auto delegate : self.delegates) {
-    SEL sel = @selector(audioSessionShouldConfigure:);
-    if ([delegate respondsToSelector:sel]) {
-      [delegate audioSessionShouldConfigure:self];
-    }
-  }
-}
-
-- (void)notifyShouldUnconfigure {
+- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
   for (auto delegate : self.delegates) {
-    SEL sel = @selector(audioSessionShouldUnconfigure:);
+    SEL sel = @selector(audioSession:didChangeCanPlayOrRecord:);
     if ([delegate respondsToSelector:sel]) {
-      [delegate audioSessionShouldUnconfigure:self];
+      [delegate audioSession:self didChangeCanPlayOrRecord:canPlayOrRecord];
     }
   }
 }
 
-- (void)notifyDidConfigure {
+- (void)notifyDidStartPlayOrRecord {
   for (auto delegate : self.delegates) {
-    SEL sel = @selector(audioSessionDidConfigure:);
+    SEL sel = @selector(audioSessionDidStartPlayOrRecord:);
     if ([delegate respondsToSelector:sel]) {
-      [delegate audioSessionDidConfigure:self];
+      [delegate audioSessionDidStartPlayOrRecord:self];
     }
   }
 }
 
-- (void)notifyDidUnconfigure {
+- (void)notifyDidStopPlayOrRecord {
   for (auto delegate : self.delegates) {
-    SEL sel = @selector(audioSessionDidUnconfigure:);
+    SEL sel = @selector(audioSessionDidStopPlayOrRecord:);
     if ([delegate respondsToSelector:sel]) {
-      [delegate audioSessionDidUnconfigure:self];
+      [delegate audioSessionDidStopPlayOrRecord:self];
     }
   }
 }
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
index 4273d4d8634..7832a82d4f3 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h
@@ -37,6 +37,8 @@ extern const double kRTCAudioSessionLowComplexityIOBufferDuration;
 + (instancetype)currentConfiguration;
 /** Returns the configuration that WebRTC needs. */
 + (instancetype)webRTCConfiguration;
+/** Provide a way to override the default configuration. */
++ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration;
 
 @end
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m
index 086725172bd..d2d04835499 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m
@@ -10,6 +10,8 @@
 
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
+#import "WebRTC/RTCDispatcher.h"
+
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
 
 // Try to use mono to save resources. Also avoids channel format conversion
@@ -49,6 +51,8 @@ const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.01;
 // TODO(henrika): monitor this size and determine if it should be modified.
 const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
 
+static RTCAudioSessionConfiguration *gWebRTCConfiguration = nil;
+
 @implementation RTCAudioSessionConfiguration
 
 @synthesize category = _category;
@@ -96,6 +100,10 @@ const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
   return self;
 }
 
++ (void)initialize {
+  gWebRTCConfiguration = [[self alloc] init];
+}
+
 + (instancetype)currentConfiguration {
   RTCAudioSession *session = [RTCAudioSession sharedInstance];
   RTCAudioSessionConfiguration *config =
@@ -111,7 +119,15 @@ const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
 }
 
 + (instancetype)webRTCConfiguration {
-  return [[self alloc] init];
+  @synchronized(self) {
+    return (RTCAudioSessionConfiguration *)gWebRTCConfiguration;
+  }
+}
+
++ (void)setWebRTCConfiguration:(RTCAudioSessionConfiguration *)configuration {
+  @synchronized(self) {
+    gWebRTCConfiguration = configuration;
+  }
 }
 
 @end
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm
index 21e8c3e84b1..b554e51ece0 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm
@@ -12,7 +12,7 @@
 
 #include "webrtc/modules/audio_device/ios/audio_session_observer.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 
 @implementation RTCAudioSessionDelegateAdapter {
   webrtc::AudioSessionObserver *_observer;
@@ -70,14 +70,15 @@
 - (void)audioSessionMediaServicesWereReset:(RTCAudioSession *)session {
 }
 
-- (void)audioSessionShouldConfigure:(RTCAudioSession *)session {
+- (void)audioSession:(RTCAudioSession *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
+  _observer->OnCanPlayOrRecordChange(canPlayOrRecord);
 }
 
-- (void)audioSessionShouldUnconfigure:(RTCAudioSession *)session {
+- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session {
 }
 
-- (void)audioSessionDidConfigure:(RTCAudioSession *)session {
-  _observer->OnConfiguredForWebRTC();
+- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session {
 }
 
 @end
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm
index 603e450c758..7cbd2a982a9 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/objc/RTCAudioSessionTest.mm
@@ -46,6 +46,28 @@
 
 @end
 
+// A delegate that adds itself to the audio session on init and removes itself
+// in its dealloc.
+@interface RTCTestRemoveOnDeallocDelegate : RTCAudioSessionTestDelegate
+@end
+
+@implementation RTCTestRemoveOnDeallocDelegate
+
+- (instancetype)init {
+  if (self = [super init]) {
+    RTCAudioSession *session = [RTCAudioSession sharedInstance];
+    [session addDelegate:self];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  [session removeDelegate:self];
+}
+
+@end
+
 
 @interface RTCAudioSessionTest : NSObject
 
@@ -142,6 +164,18 @@
   EXPECT_TRUE(session.delegates[0]);
 }
 
+// Tests that we don't crash when removing delegates in dealloc.
+// Added as a regression test.
+- (void)testRemoveDelegateOnDealloc {
+  @autoreleasepool {
+    RTCTestRemoveOnDeallocDelegate *delegate =
+        [[RTCTestRemoveOnDeallocDelegate alloc] init];
+    EXPECT_TRUE(delegate);
+  }
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  EXPECT_EQ(0u, session.delegates.size());
+}
+
 @end
 
 namespace webrtc {
@@ -176,4 +210,9 @@ TEST_F(AudioSessionTest, ZeroingWeakDelegate) {
   [test testZeroingWeakDelegate];
 }
 
+TEST_F(AudioSessionTest, RemoveDelegateOnDealloc) {
+  RTCAudioSessionTest *test = [[RTCAudioSessionTest alloc] init];
+  [test testRemoveDelegateOnDealloc];
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm b/chromium/third_party/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
index db756a4972c..db7f42edcb9 100644
--- a/chromium/third_party/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
+++ b/chromium/third_party/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
@@ -12,7 +12,7 @@
 
 #include "webrtc/base/checks.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 #import "webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h"
 
 #if !defined(NDEBUG)
@@ -175,7 +175,7 @@ VoiceProcessingAudioUnit::State VoiceProcessingAudioUnit::GetState() const {
 
 bool VoiceProcessingAudioUnit::Initialize(Float64 sample_rate) {
   RTC_DCHECK_GE(state_, kUninitialized);
-  RTCLog(@"Initializing audio unit.");
+  RTCLog(@"Initializing audio unit with sample rate: %f", sample_rate);
 
   OSStatus result = noErr;
   AudioStreamBasicDescription format = GetFormat(sample_rate);
@@ -228,7 +228,9 @@ bool VoiceProcessingAudioUnit::Initialize(Float64 sample_rate) {
     [NSThread sleepForTimeInterval:0.1f];
     result = AudioUnitInitialize(vpio_unit_);
   }
-  RTCLog(@"Voice Processing I/O unit is now initialized.");
+  if (result == noErr) {
+    RTCLog(@"Voice Processing I/O unit is now initialized.");
+  }
   state_ = kInitialized;
   return true;
 }
@@ -241,6 +243,8 @@ bool VoiceProcessingAudioUnit::Start() {
   if (result != noErr) {
     RTCLogError(@"Failed to start audio unit. Error=%ld", (long)result);
     return false;
+  } else {
+    RTCLog(@"Started audio unit");
   }
   state_ = kStarted;
   return true;
@@ -254,7 +258,10 @@ bool VoiceProcessingAudioUnit::Stop() {
   if (result != noErr) {
     RTCLogError(@"Failed to stop audio unit. Error=%ld", (long)result);
     return false;
+  } else {
+    RTCLog(@"Stopped audio unit");
   }
+
   state_ = kInitialized;
   return true;
 }
@@ -267,7 +274,11 @@ bool VoiceProcessingAudioUnit::Uninitialize() {
   if (result != noErr) {
     RTCLogError(@"Failed to uninitialize audio unit. Error=%ld", (long)result);
     return false;
+  } else {
+    RTCLog(@"Uninitialized audio unit.");
   }
+
+  state_ = kUninitialized;
   return true;
 }
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/test/audio_device_test_api.cc b/chromium/third_party/webrtc/modules/audio_device/test/audio_device_test_api.cc
index f37d89cd9ca..dad42a0c0b3 100644
--- a/chromium/third_party/webrtc/modules/audio_device/test/audio_device_test_api.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/test/audio_device_test_api.cc
@@ -142,8 +142,7 @@ class AudioDeviceAPITest: public testing::Test {
   virtual ~AudioDeviceAPITest() {}
 
   static void SetUpTestCase() {
-    process_thread_ =
-        rtc::ScopedToUnique(ProcessThread::Create("ProcessThread"));
+    process_thread_ = ProcessThread::Create("ProcessThread");
     process_thread_->Start();
 
     // Windows:
@@ -154,75 +153,75 @@ class AudioDeviceAPITest: public testing::Test {
     const int32_t kId = 444;
 
 #if defined(_WIN32)
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
 #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
     TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is defined!\n\n");
     // create default implementation (=Core Audio) instance
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
     EXPECT_EQ(0, audio_device_.release()->Release());
     // create non-default (=Wave Audio) instance
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
     EXPECT_EQ(0, audio_device_.release()->Release());
     // explicitly specify usage of Core Audio (same as default)
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsCoreAudio)) != NULL);
 #else
     TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is *not* defined!\n");
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
     // create default implementation (=Wave Audio) instance
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
     EXPECT_EQ(0, audio_device_.release()->Release());
     // explicitly specify usage of Wave Audio (same as default)
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
 #endif
 #endif
 
 #if defined(ANDROID)
     // Fails tests
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kLinuxPulseAudio)) == NULL);
     // Create default implementation instance
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
 #elif defined(WEBRTC_LINUX)
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
     // create default implementation instance
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
     EXPECT_EQ(0, audio_device_->Terminate());
     EXPECT_EQ(0, audio_device_.release()->Release());
     // explicitly specify usage of Pulse Audio (same as default)
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kLinuxPulseAudio)) != NULL);
 #endif
 
 #if defined(WEBRTC_MAC)
     // Fails tests
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kLinuxPulseAudio)) == NULL);
     // Create default implementation instance
-    EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((audio_device_ = AudioDeviceModule::Create(
                 kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
 #endif
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/test/func_test_manager.cc b/chromium/third_party/webrtc/modules/audio_device/test/func_test_manager.cc
index bb7686c6c18..f16f296011b 100644
--- a/chromium/third_party/webrtc/modules/audio_device/test/func_test_manager.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/test/func_test_manager.cc
@@ -594,16 +594,15 @@ FuncTestManager::~FuncTestManager()
 
 int32_t FuncTestManager::Init()
 {
-    EXPECT_TRUE((_processThread = rtc::ScopedToUnique(
-                     ProcessThread::Create("ProcessThread"))) != NULL);
-    if (_processThread == NULL)
-    {
-        return -1;
+    EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) !=
+                NULL);
+    if (_processThread == NULL) {
+      return -1;
     }
     _processThread->Start();
 
     // create the Audio Device module
-    EXPECT_TRUE((_audioDevice = AudioDeviceModuleImpl::Create(
+    EXPECT_TRUE((_audioDevice = AudioDeviceModule::Create(
         555, ADM_AUDIO_LAYER)) != NULL);
     if (_audioDevice == NULL)
     {
@@ -832,8 +831,8 @@ int32_t FuncTestManager::TestAudioLayerSelection()
         // ==================================================
         // Next, try to make fresh start with new audio layer
 
-        EXPECT_TRUE((_processThread = rtc::ScopedToUnique(
-                         ProcessThread::Create("ProcessThread"))) != NULL);
+        EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) !=
+                    NULL);
         if (_processThread == NULL)
         {
             return -1;
@@ -843,12 +842,12 @@ int32_t FuncTestManager::TestAudioLayerSelection()
         // create the Audio Device module based on selected audio layer
         if (tryWinWave)
         {
-            _audioDevice = AudioDeviceModuleImpl::Create(
+            _audioDevice = AudioDeviceModule::Create(
                 555,
                 AudioDeviceModule::kWindowsWaveAudio);
         } else if (tryWinCore)
         {
-            _audioDevice = AudioDeviceModuleImpl::Create(
+            _audioDevice = AudioDeviceModule::Create(
                 555,
                 AudioDeviceModule::kWindowsCoreAudio);
         }
@@ -857,7 +856,7 @@ int32_t FuncTestManager::TestAudioLayerSelection()
         {
             TEST_LOG("\nERROR: Switch of audio layer failed!\n");
             // restore default audio layer instead
-            EXPECT_TRUE((_audioDevice = AudioDeviceModuleImpl::Create(
+            EXPECT_TRUE((_audioDevice = AudioDeviceModule::Create(
                 555, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
         }
 
diff --git a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc
index 8079051184f..c1497089926 100644
--- a/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc
+++ b/chromium/third_party/webrtc/modules/audio_device/win/audio_device_wave_win.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/win/audio_device_wave_win.h"
 
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 #include <windows.h>
@@ -206,7 +206,7 @@ int32_t AudioDeviceWindowsWave::Init()
         return 0;
     }
 
-    const uint32_t nowTime(TickTime::MillisecondTimestamp());
+    const uint32_t nowTime(rtc::TimeMillis());
 
     _recordedBytes = 0;
     _prevRecByteCheckTime = nowTime;
@@ -3038,7 +3038,7 @@ bool AudioDeviceWindowsWave::ThreadProcess()
         return true;
     }
 
-    time = TickTime::MillisecondTimestamp();
+    time = rtc::TimeMillis();
 
     if (_startPlay)
     {
diff --git a/chromium/third_party/webrtc/modules/audio_processing/BUILD.gn b/chromium/third_party/webrtc/modules/audio_processing/BUILD.gn
index 22c904d52f2..a9650f74296 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/BUILD.gn
+++ b/chromium/third_party/webrtc/modules/audio_processing/BUILD.gn
@@ -24,17 +24,16 @@ source_set("audio_processing") {
   sources = [
     "aec/aec_core.cc",
     "aec/aec_core.h",
-    "aec/aec_core_internal.h",
-    "aec/aec_rdft.c",
+    "aec/aec_core_optimized_methods.h",
+    "aec/aec_rdft.cc",
     "aec/aec_rdft.h",
     "aec/aec_resampler.cc",
     "aec/aec_resampler.h",
     "aec/echo_cancellation.cc",
     "aec/echo_cancellation.h",
-    "aec/echo_cancellation_internal.h",
-    "aecm/aecm_core.c",
+    "aecm/aecm_core.cc",
     "aecm/aecm_core.h",
-    "aecm/echo_control_mobile.c",
+    "aecm/echo_control_mobile.cc",
     "aecm/echo_control_mobile.h",
     "agc/agc.cc",
     "agc/agc.h",
@@ -81,9 +80,8 @@ source_set("audio_processing") {
     "intelligibility/intelligibility_utils.h",
     "level_estimator_impl.cc",
     "level_estimator_impl.h",
-    "logging/aec_logging.h",
-    "logging/aec_logging_file_handling.cc",
-    "logging/aec_logging_file_handling.h",
+    "logging/apm_data_dumper.cc",
+    "logging/apm_data_dumper.h",
     "noise_suppression_impl.cc",
     "noise_suppression_impl.h",
     "render_queue_item_verifier.h",
@@ -110,10 +108,10 @@ source_set("audio_processing") {
     "typing_detection.h",
     "utility/block_mean_calculator.cc",
     "utility/block_mean_calculator.h",
-    "utility/delay_estimator.c",
+    "utility/delay_estimator.cc",
     "utility/delay_estimator.h",
     "utility/delay_estimator_internal.h",
-    "utility/delay_estimator_wrapper.c",
+    "utility/delay_estimator_wrapper.cc",
     "utility/delay_estimator_wrapper.h",
     "vad/common.h",
     "vad/gmm.cc",
@@ -149,7 +147,9 @@ source_set("audio_processing") {
   ]
 
   if (aec_debug_dump) {
-    defines += [ "WEBRTC_AEC_DEBUG_DUMP" ]
+    defines += [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+  } else {
+    defines += [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
   }
 
   if (aec_untrusted_delay_for_testing) {
@@ -196,15 +196,15 @@ source_set("audio_processing") {
   }
 
   if (current_cpu == "mipsel") {
-    sources += [ "aecm/aecm_core_mips.c" ]
+    sources += [ "aecm/aecm_core_mips.cc" ]
     if (mips_float_abi == "hard") {
       sources += [
         "aec/aec_core_mips.cc",
-        "aec/aec_rdft_mips.c",
+        "aec/aec_rdft_mips.cc",
       ]
     }
   } else {
-    sources += [ "aecm/aecm_core_c.c" ]
+    sources += [ "aecm/aecm_core_c.cc" ]
   }
 
   if (is_win) {
@@ -241,7 +241,7 @@ if (current_cpu == "x86" || current_cpu == "x64") {
   source_set("audio_processing_sse2") {
     sources = [
       "aec/aec_core_sse2.cc",
-      "aec/aec_rdft_sse2.c",
+      "aec/aec_rdft_sse2.cc",
     ]
 
     if (is_posix) {
@@ -250,6 +250,12 @@ if (current_cpu == "x86" || current_cpu == "x64") {
 
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
+
+    if (aec_debug_dump) {
+      defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+    } else {
+      defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
+    }
   }
 }
 
@@ -257,8 +263,8 @@ if (rtc_build_with_neon) {
   source_set("audio_processing_neon") {
     sources = [
       "aec/aec_core_neon.cc",
-      "aec/aec_rdft_neon.c",
-      "aecm/aecm_core_neon.c",
+      "aec/aec_rdft_neon.cc",
+      "aecm/aecm_core_neon.cc",
       "ns/nsx_core_neon.c",
     ]
 
@@ -285,5 +291,11 @@ if (rtc_build_with_neon) {
     deps = [
       "../../common_audio",
     ]
+
+    if (aec_debug_dump) {
+      defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+    } else {
+      defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
+    }
   }
 }
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.cc b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.cc
index 1d1e67636bf..4c109d7d9ce 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.cc
@@ -14,10 +14,6 @@
 
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-#include <stdio.h>
-#endif
-
 #include <algorithm>
 #include <assert.h>
 #include <math.h>
@@ -29,20 +25,45 @@
 extern "C" {
 #include "webrtc/common_audio/ring_buffer.h"
 }
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
-extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
-}
-#include "webrtc/modules/audio_processing/logging/aec_logging.h"
-extern "C" {
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-}
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+namespace {
+enum class DelaySource {
+  kSystemDelay,    // The delay values come from the OS.
+  kDelayAgnostic,  // The delay values come from the DA-AEC.
+};
+
+constexpr int kMinDelayLogValue = -200;
+constexpr int kMaxDelayLogValue = 200;
+constexpr int kNumDelayLogBuckets = 100;
+
+void MaybeLogDelayAdjustment(int moved_ms, DelaySource source) {
+  if (moved_ms == 0)
+    return;
+  switch (source) {
+    case DelaySource::kSystemDelay:
+      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecDelayAdjustmentMsSystemValue",
+                           moved_ms, kMinDelayLogValue, kMaxDelayLogValue,
+                           kNumDelayLogBuckets);
+      return;
+    case DelaySource::kDelayAgnostic:
+      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecDelayAdjustmentMsAgnosticValue",
+                           moved_ms, kMinDelayLogValue, kMaxDelayLogValue,
+                           kNumDelayLogBuckets);
+      return;
+  }
+}
+}  // namespace
 
 // Buffer size (samples)
 static const size_t kBufSizePartitions = 250;  // 1 second of audio in 16 kHz.
@@ -55,8 +76,8 @@ static const int kDelayMetricsAggregationWindow = 1250;  // 5 seconds at 16 kHz.
 // Divergence metric is based on audio level, which gets updated every
 // |kCountLen + 1| * 10 milliseconds. Divergence metric takes the statistics of
 // |kDivergentFilterFractionAggregationWindowSize| samples. Current value
-// corresponds to 0.5 seconds at 16 kHz.
-static const int kDivergentFilterFractionAggregationWindowSize = 25;
+// corresponds to 1 second at 16 kHz.
+static const int kDivergentFilterFractionAggregationWindowSize = 50;
 
 // Quantities to control H band scaling for SWB input
 static const float cnScaleHband = 0.4f;  // scale for comfort noise in H band.
@@ -136,16 +157,13 @@ const float WebRtcAec_kNormalSmoothingCoefficients[2][2] = {{0.9f, 0.1f},
 // Number of partitions forming the NLP's "preferred" bands.
 enum { kPrefBandSize = 24 };
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-extern int webrtc_aec_instance_count;
-#endif
-
 WebRtcAecFilterFar WebRtcAec_FilterFar;
 WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
 WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
-WebRtcAecOverdriveAndSuppress WebRtcAec_OverdriveAndSuppress;
-WebRtcAecComfortNoise WebRtcAec_ComfortNoise;
-WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
+WebRtcAecOverdrive WebRtcAec_Overdrive;
+WebRtcAecSuppress WebRtcAec_Suppress;
+WebRtcAecComputeCoherence WebRtcAec_ComputeCoherence;
+WebRtcAecUpdateCoherenceSpectra WebRtcAec_UpdateCoherenceSpectra;
 WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
 WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
 WebRtcAecWindowData WebRtcAec_WindowData;
@@ -210,7 +228,10 @@ void DivergentFilterFraction::Clear() {
 }
 
 // TODO(minyue): Moving some initialization from WebRtcAec_CreateAec() to ctor.
-AecCore::AecCore() = default;
+AecCore::AecCore(int instance_index)
+    : data_dumper(new ApmDataDumper(instance_index)) {}
+
+AecCore::~AecCore() {}
 
 static int CmpFloat(const void* a, const void* b) {
   const float* da = (const float*)a;
@@ -316,19 +337,21 @@ static void FilterAdaptation(
   }
 }
 
-static void OverdriveAndSuppress(AecCore* aec,
-                                 float hNl[PART_LEN1],
-                                 const float hNlFb,
-                                 float efw[2][PART_LEN1]) {
-  int i;
-  for (i = 0; i < PART_LEN1; i++) {
+static void Overdrive(float overdrive_scaling,
+                      const float hNlFb,
+                      float hNl[PART_LEN1]) {
+  for (int i = 0; i < PART_LEN1; ++i) {
     // Weight subbands
     if (hNl[i] > hNlFb) {
       hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
                (1 - WebRtcAec_weightCurve[i]) * hNl[i];
     }
-    hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
+    hNl[i] = powf(hNl[i], overdrive_scaling * WebRtcAec_overDriveCurve[i]);
+  }
+}
 
+static void Suppress(const float hNl[PART_LEN1], float efw[2][PART_LEN1]) {
+  for (int i = 0; i < PART_LEN1; ++i) {
     // Suppress error signal
     efw[0][i] *= hNl[i];
     efw[1][i] *= hNl[i];
@@ -339,7 +362,9 @@ static void OverdriveAndSuppress(AecCore* aec,
   }
 }
 
-static int PartitionDelay(const AecCore* aec) {
+static int PartitionDelay(int num_partitions,
+                          float h_fft_buf[2]
+                                         [kExtendedNumPartitions * PART_LEN1]) {
   // Measures the energy in each filter partition and returns the partition with
   // highest energy.
   // TODO(bjornv): Spread computational cost by computing one partition per
@@ -348,13 +373,13 @@ static int PartitionDelay(const AecCore* aec) {
   int i;
   int delay = 0;
 
-  for (i = 0; i < aec->num_partitions; i++) {
+  for (i = 0; i < num_partitions; i++) {
     int j;
     int pos = i * PART_LEN1;
     float wfEn = 0;
     for (j = 0; j < PART_LEN1; j++) {
-      wfEn += aec->wfBuf[0][pos + j] * aec->wfBuf[0][pos + j] +
-              aec->wfBuf[1][pos + j] * aec->wfBuf[1][pos + j];
+      wfEn += h_fft_buf[0][pos + j] * h_fft_buf[0][pos + j] +
+              h_fft_buf[1][pos + j] * h_fft_buf[1][pos + j];
     }
 
     if (wfEn > wfEnMax) {
@@ -365,10 +390,46 @@ static int PartitionDelay(const AecCore* aec) {
   return delay;
 }
 
+// Update metric with 10 * log10(numerator / denominator).
+static void UpdateLogRatioMetric(Stats* metric, float numerator,
+                                 float denominator) {
+  RTC_DCHECK(metric);
+  RTC_CHECK(numerator >= 0);
+  RTC_CHECK(denominator >= 0);
+
+  const float log_numerator = log10(numerator + 1e-10f);
+  const float log_denominator = log10(denominator + 1e-10f);
+  metric->instant = 10.0f * (log_numerator - log_denominator);
+
+  // Max.
+  if (metric->instant > metric->max)
+    metric->max = metric->instant;
+
+  // Min.
+  if (metric->instant < metric->min)
+    metric->min = metric->instant;
+
+  // Average.
+  metric->counter++;
+  // This is to protect overflow, which should almost never happen.
+  RTC_CHECK_NE(0u, metric->counter);
+  metric->sum += metric->instant;
+  metric->average = metric->sum / metric->counter;
+
+  // Upper mean.
+  if (metric->instant > metric->average) {
+    metric->hicounter++;
+    // This is to protect overflow, which should almost never happen.
+    RTC_CHECK_NE(0u, metric->hicounter);
+    metric->hisum += metric->instant;
+    metric->himean = metric->hisum / metric->hicounter;
+  }
+}
+
 // Threshold to protect against the ill-effects of a zero far-end.
 const float WebRtcAec_kMinFarendPSD = 15;
 
-// Updates the following smoothed  Power Spectral Densities (PSD):
+// Updates the following smoothed Power Spectral Densities (PSD):
 //  - sd  : near-end
 //  - se  : residual echo
 //  - sx  : far-end
@@ -377,53 +438,60 @@ const float WebRtcAec_kMinFarendPSD = 15;
 //
 // In addition to updating the PSDs, also the filter diverge state is
 // determined.
-static void SmoothedPSD(AecCore* aec,
-                        float efw[2][PART_LEN1],
-                        float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1],
-                        int* extreme_filter_divergence) {
+static void UpdateCoherenceSpectra(int mult,
+                                   bool extended_filter_enabled,
+                                   float efw[2][PART_LEN1],
+                                   float dfw[2][PART_LEN1],
+                                   float xfw[2][PART_LEN1],
+                                   CoherenceState* coherence_state,
+                                   short* filter_divergence_state,
+                                   int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh =
-      aec->extended_filter_enabled
-          ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
-          : WebRtcAec_kNormalSmoothingCoefficients[aec->mult - 1];
+      extended_filter_enabled
+          ? WebRtcAec_kExtendedSmoothingCoefficients[mult - 1]
+          : WebRtcAec_kNormalSmoothingCoefficients[mult - 1];
   int i;
   float sdSum = 0, seSum = 0;
 
   for (i = 0; i < PART_LEN1; i++) {
-    aec->sd[i] = ptrGCoh[0] * aec->sd[i] +
-                 ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
-    aec->se[i] = ptrGCoh[0] * aec->se[i] +
-                 ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
+    coherence_state->sd[i] =
+        ptrGCoh[0] * coherence_state->sd[i] +
+        ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
+    coherence_state->se[i] =
+        ptrGCoh[0] * coherence_state->se[i] +
+        ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
     // We threshold here to protect against the ill-effects of a zero farend.
     // The threshold is not arbitrarily chosen, but balances protection and
     // adverse interaction with the algorithm's tuning.
     // TODO(bjornv): investigate further why this is so sensitive.
-    aec->sx[i] = ptrGCoh[0] * aec->sx[i] +
-                 ptrGCoh[1] * WEBRTC_SPL_MAX(
-                                  xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
-                                  WebRtcAec_kMinFarendPSD);
-
-    aec->sde[i][0] =
-        ptrGCoh[0] * aec->sde[i][0] +
+    coherence_state->sx[i] =
+        ptrGCoh[0] * coherence_state->sx[i] +
+        ptrGCoh[1] *
+            WEBRTC_SPL_MAX(xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
+                           WebRtcAec_kMinFarendPSD);
+
+    coherence_state->sde[i][0] =
+        ptrGCoh[0] * coherence_state->sde[i][0] +
         ptrGCoh[1] * (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
-    aec->sde[i][1] =
-        ptrGCoh[0] * aec->sde[i][1] +
+    coherence_state->sde[i][1] =
+        ptrGCoh[0] * coherence_state->sde[i][1] +
         ptrGCoh[1] * (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
 
-    aec->sxd[i][0] =
-        ptrGCoh[0] * aec->sxd[i][0] +
+    coherence_state->sxd[i][0] =
+        ptrGCoh[0] * coherence_state->sxd[i][0] +
         ptrGCoh[1] * (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
-    aec->sxd[i][1] =
-        ptrGCoh[0] * aec->sxd[i][1] +
+    coherence_state->sxd[i][1] =
+        ptrGCoh[0] * coherence_state->sxd[i][1] +
         ptrGCoh[1] * (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
 
-    sdSum += aec->sd[i];
-    seSum += aec->se[i];
+    sdSum += coherence_state->sd[i];
+    seSum += coherence_state->se[i];
   }
 
   // Divergent filter safeguard update.
-  aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
+  *filter_divergence_state =
+      (*filter_divergence_state ? 1.05f : 1.0f) * seSum > sdSum;
 
   // Signal extreme filter divergence if the error is significantly larger
   // than the nearend (13 dB).
@@ -454,26 +522,17 @@ __inline static void StoreAsComplex(const float* data,
   data_complex[1][PART_LEN] = 0;
 }
 
-static void SubbandCoherence(AecCore* aec,
-                             float efw[2][PART_LEN1],
-                             float dfw[2][PART_LEN1],
-                             float xfw[2][PART_LEN1],
-                             float* fft,
+static void ComputeCoherence(const CoherenceState* coherence_state,
                              float* cohde,
-                             float* cohxd,
-                             int* extreme_filter_divergence) {
-  int i;
-
-  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
-
+                             float* cohxd) {
   // Subband coherence
-  for (i = 0; i < PART_LEN1; i++) {
-    cohde[i] =
-        (aec->sde[i][0] * aec->sde[i][0] + aec->sde[i][1] * aec->sde[i][1]) /
-        (aec->sd[i] * aec->se[i] + 1e-10f);
-    cohxd[i] =
-        (aec->sxd[i][0] * aec->sxd[i][0] + aec->sxd[i][1] * aec->sxd[i][1]) /
-        (aec->sx[i] * aec->sd[i] + 1e-10f);
+  for (int i = 0; i < PART_LEN1; i++) {
+    cohde[i] = (coherence_state->sde[i][0] * coherence_state->sde[i][0] +
+                coherence_state->sde[i][1] * coherence_state->sde[i][1]) /
+               (coherence_state->sd[i] * coherence_state->se[i] + 1e-10f);
+    cohxd[i] = (coherence_state->sxd[i][0] * coherence_state->sxd[i][0] +
+                coherence_state->sxd[i][1] * coherence_state->sxd[i][1]) /
+               (coherence_state->sx[i] * coherence_state->sd[i] + 1e-10f);
   }
 }
 
@@ -487,94 +546,67 @@ static void GetHighbandGain(const float* lambda, float* nlpGainHband) {
   *nlpGainHband /= static_cast<float>(PART_LEN1 - 1 - freqAvgIc);
 }
 
-static void ComfortNoise(AecCore* aec,
-                         float efw[2][PART_LEN1],
-                         float comfortNoiseHband[2][PART_LEN1],
-                         const float* noisePow,
-                         const float* lambda) {
-  int i, num;
-  float rand[PART_LEN];
-  float noise, noiseAvg, tmp, tmpAvg;
+static void GenerateComplexNoise(uint32_t* seed, float noise[2][PART_LEN1]) {
+  const float kPi2 = 6.28318530717959f;
   int16_t randW16[PART_LEN];
-  float u[2][PART_LEN1];
+  WebRtcSpl_RandUArray(randW16, PART_LEN, seed);
 
-  const float pi2 = 6.28318530717959f;
-
-  // Generate a uniform random array on [0 1]
-  WebRtcSpl_RandUArray(randW16, PART_LEN, &aec->seed);
-  for (i = 0; i < PART_LEN; i++) {
-    rand[i] = static_cast<float>(randW16[i]) / 32768;
+  noise[0][0] = 0;
+  noise[1][0] = 0;
+  for (size_t i = 1; i < PART_LEN1; i++) {
+    float tmp = kPi2 * randW16[i - 1] / 32768.f;
+    noise[0][i] = cosf(tmp);
+    noise[1][i] = -sinf(tmp);
   }
+  noise[1][PART_LEN] = 0;
+}
 
-  // Reject LF noise
-  u[0][0] = 0;
-  u[1][0] = 0;
-  for (i = 1; i < PART_LEN1; i++) {
-    tmp = pi2 * rand[i - 1];
-
-    noise = sqrtf(noisePow[i]);
-    u[0][i] = noise * cosf(tmp);
-    u[1][i] = -noise * sinf(tmp);
-  }
-  u[1][PART_LEN] = 0;
-
-  for (i = 0; i < PART_LEN1; i++) {
-    // This is the proper weighting to match the background noise power
-    tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
-    // tmp = 1 - lambda[i];
-    efw[0][i] += tmp * u[0][i];
-    efw[1][i] += tmp * u[1][i];
-  }
-
-  // For H band comfort noise
-  // TODO(peah): don't compute noise and "tmp" twice. Use the previous results.
-  noiseAvg = 0.0;
-  tmpAvg = 0.0;
-  num = 0;
-  if (aec->num_bands > 1) {
-    // average noise scale
-    // average over second half of freq spectrum (i.e., 4->8khz)
-    // TODO(peah): we shouldn't need num. We know how many elements we're
-    // summing.
-    for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
-      num++;
-      noiseAvg += sqrtf(noisePow[i]);
-    }
-    noiseAvg /= static_cast<float>(num);
-
-    // average nlp scale
-    // average over second half of freq spectrum (i.e., 4->8khz)
-    // TODO(peah): we shouldn't need num. We know how many elements
-    // we're summing.
-    num = 0;
-    for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
-      num++;
-      tmpAvg += sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
-    }
-    tmpAvg /= static_cast<float>(num);
-
-    // Use average noise for H band
-    // TODO(peah): we should probably have a new random vector here.
-    // Reject LF noise
-    u[0][0] = 0;
-    u[1][0] = 0;
-    for (i = 1; i < PART_LEN1; i++) {
-      tmp = pi2 * rand[i - 1];
-
-      // Use average noise for H band
-      u[0][i] = noiseAvg * static_cast<float>(cos(tmp));
-      u[1][i] = -noiseAvg * static_cast<float>(sin(tmp));
+static void ComfortNoise(bool generate_high_frequency_noise,
+                         uint32_t* seed,
+                         float e_fft[2][PART_LEN1],
+                         float high_frequency_comfort_noise[2][PART_LEN1],
+                         const float* noise_spectrum,
+                         const float* suppressor_gain) {
+  float complex_noise[2][PART_LEN1];
+
+  GenerateComplexNoise(seed, complex_noise);
+
+  // Shape, scale and add comfort noise.
+  for (int i = 1; i < PART_LEN1; ++i) {
+    float noise_scaling =
+        sqrtf(WEBRTC_SPL_MAX(1 - suppressor_gain[i] * suppressor_gain[i], 0)) *
+        sqrtf(noise_spectrum[i]);
+    e_fft[0][i] += noise_scaling * complex_noise[0][i];
+    e_fft[1][i] += noise_scaling * complex_noise[1][i];
+  }
+
+  // Form comfort noise for higher frequencies.
+  if (generate_high_frequency_noise) {
+    // Compute average noise power and nlp gain over the second half of freq
+    // spectrum (i.e., 4->8khz).
+    int start_avg_band = PART_LEN1 / 2;
+    float upper_bands_noise_power = 0.f;
+    float upper_bands_suppressor_gain = 0.f;
+    for (int i = start_avg_band; i < PART_LEN1; ++i) {
+      upper_bands_noise_power += sqrtf(noise_spectrum[i]);
+      upper_bands_suppressor_gain +=
+          sqrtf(WEBRTC_SPL_MAX(1 - suppressor_gain[i] * suppressor_gain[i], 0));
     }
-    u[1][PART_LEN] = 0;
-
-    for (i = 0; i < PART_LEN1; i++) {
-      // Use average NLP weight for H band
-      comfortNoiseHband[0][i] = tmpAvg * u[0][i];
-      comfortNoiseHband[1][i] = tmpAvg * u[1][i];
+    upper_bands_noise_power /= (PART_LEN1 - start_avg_band);
+    upper_bands_suppressor_gain /= (PART_LEN1 - start_avg_band);
+
+    // Shape, scale and add comfort noise.
+    float noise_scaling = upper_bands_suppressor_gain * upper_bands_noise_power;
+    high_frequency_comfort_noise[0][0] = 0;
+    high_frequency_comfort_noise[1][0] = 0;
+    for (int i = 1; i < PART_LEN1; ++i) {
+      high_frequency_comfort_noise[0][i] = noise_scaling * complex_noise[0][i];
+      high_frequency_comfort_noise[1][i] = noise_scaling * complex_noise[1][i];
     }
+    high_frequency_comfort_noise[1][PART_LEN] = 0;
   } else {
-    memset(comfortNoiseHband, 0,
-           2 * PART_LEN1 * sizeof(comfortNoiseHband[0][0]));
+    memset(high_frequency_comfort_noise, 0,
+           2 * PART_LEN1 * sizeof(high_frequency_comfort_noise[0][0]));
   }
 }
 
@@ -638,16 +670,12 @@ static void UpdateLevel(PowerLevel* level, float power) {
 }
 
 static void UpdateMetrics(AecCore* aec) {
-  float dtmp;
-
   const float actThresholdNoisy = 8.0f;
   const float actThresholdClean = 40.0f;
-  const float safety = 0.99995f;
 
   const float noisyPower = 300000.0f;
 
   float actThreshold;
-  float echo, suppressedEcho;
 
   if (aec->echoState) {  // Check if echo is likely present
     aec->stateCounter++;
@@ -674,95 +702,22 @@ static void UpdateMetrics(AecCore* aec) {
         (aec->farlevel.framelevel.EndOfBlock()) &&
         (far_average_level > (actThreshold * aec->farlevel.minlevel))) {
 
+      // ERL: error return loss.
       const float near_average_level =
           aec->nearlevel.averagelevel.GetLatestMean();
+      UpdateLogRatioMetric(&aec->erl, far_average_level, near_average_level);
 
-      // Subtract noise power
-      echo = near_average_level - safety * aec->nearlevel.minlevel;
-
-      // ERL
-      dtmp = 10 * static_cast<float>(log10(far_average_level /
-                                           near_average_level + 1e-10f));
-
-      aec->erl.instant = dtmp;
-      if (dtmp > aec->erl.max) {
-        aec->erl.max = dtmp;
-      }
-
-      if (dtmp < aec->erl.min) {
-        aec->erl.min = dtmp;
-      }
-
-      aec->erl.counter++;
-      aec->erl.sum += dtmp;
-      aec->erl.average = aec->erl.sum / aec->erl.counter;
-
-      // Upper mean
-      if (dtmp > aec->erl.average) {
-        aec->erl.hicounter++;
-        aec->erl.hisum += dtmp;
-        aec->erl.himean = aec->erl.hisum / aec->erl.hicounter;
-      }
-
-      // A_NLP
+      // A_NLP: error return loss enhanced before the nonlinear suppression.
       const float linout_average_level =
           aec->linoutlevel.averagelevel.GetLatestMean();
-      dtmp = 10 * static_cast<float>(log10(near_average_level /
-                                           linout_average_level + 1e-10f));
-
-      // subtract noise power
-      suppressedEcho =
-          linout_average_level - safety * aec->linoutlevel.minlevel;
-
-      aec->aNlp.instant =
-          10 * static_cast<float>(log10(echo / suppressedEcho + 1e-10f));
-
-      if (dtmp > aec->aNlp.max) {
-        aec->aNlp.max = dtmp;
-      }
+      UpdateLogRatioMetric(&aec->aNlp, near_average_level,
+                           linout_average_level);
 
-      if (dtmp < aec->aNlp.min) {
-        aec->aNlp.min = dtmp;
-      }
-
-      aec->aNlp.counter++;
-      aec->aNlp.sum += dtmp;
-      aec->aNlp.average = aec->aNlp.sum / aec->aNlp.counter;
-
-      // Upper mean
-      if (dtmp > aec->aNlp.average) {
-        aec->aNlp.hicounter++;
-        aec->aNlp.hisum += dtmp;
-        aec->aNlp.himean = aec->aNlp.hisum / aec->aNlp.hicounter;
-      }
-
-      // ERLE
+      // ERLE: error return loss enhanced.
       const float nlpout_average_level =
           aec->nlpoutlevel.averagelevel.GetLatestMean();
-      // subtract noise power
-      suppressedEcho =
-          nlpout_average_level - safety * aec->nlpoutlevel.minlevel;
-      dtmp = 10 * static_cast<float>(log10(echo / suppressedEcho + 1e-10f));
-
-      aec->erle.instant = dtmp;
-      if (dtmp > aec->erle.max) {
-        aec->erle.max = dtmp;
-      }
-
-      if (dtmp < aec->erle.min) {
-        aec->erle.min = dtmp;
-      }
-
-      aec->erle.counter++;
-      aec->erle.sum += dtmp;
-      aec->erle.average = aec->erle.sum / aec->erle.counter;
-
-      // Upper mean
-      if (dtmp > aec->erle.average) {
-        aec->erle.hicounter++;
-        aec->erle.hisum += dtmp;
-        aec->erle.himean = aec->erle.hisum / aec->erle.hicounter;
-      }
+      UpdateLogRatioMetric(&aec->erle, near_average_level,
+                           nlpout_average_level);
     }
 
     aec->stateCounter = 0;
@@ -963,9 +918,9 @@ static void RegressorPower(int num_partitions,
   }
 }
 
-static void EchoSubtraction(AecCore* aec,
-                            int num_partitions,
+static void EchoSubtraction(int num_partitions,
                             int extended_filter_enabled,
+                            int* extreme_filter_divergence,
                             float filter_step_size,
                             float error_threshold,
                             float* x_fft,
@@ -1001,9 +956,10 @@ static void EchoSubtraction(AecCore* aec,
 
   // Conditionally reset the echo subtraction filter if the filter has diverged
   // significantly.
-  if (!aec->extended_filter_enabled && aec->extreme_filter_divergence) {
-    memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
-    aec->extreme_filter_divergence = 0;
+  if (!extended_filter_enabled && *extreme_filter_divergence) {
+    memset(h_fft_buf, 0,
+           2 * kExtendedNumPartitions * PART_LEN1 * sizeof(h_fft_buf[0][0]));
+    *extreme_filter_divergence = 0;
   }
 
   // Produce echo estimate s_fft.
@@ -1024,9 +980,6 @@ static void EchoSubtraction(AecCore* aec,
   memcpy(e_extended + PART_LEN, e, sizeof(float) * PART_LEN);
   Fft(e_extended, e_fft);
 
-  RTC_AEC_DEBUG_RAW_WRITE(aec->e_fft_file, &e_fft[0][0],
-                          sizeof(e_fft[0][0]) * PART_LEN1 * 2);
-
   // Scale error signal inversely with far power.
   WebRtcAec_ScaleErrorSignal(filter_step_size, error_threshold, x_pow, e_fft);
   WebRtcAec_FilterAdaptation(num_partitions, *x_fft_buf_block_pos, x_fft_buf,
@@ -1034,92 +987,30 @@ static void EchoSubtraction(AecCore* aec,
   memcpy(echo_subtractor_output, e, sizeof(float) * PART_LEN);
 }
 
-static void EchoSuppression(AecCore* aec,
-                            float farend[PART_LEN2],
-                            float* echo_subtractor_output,
-                            float* output,
-                            float* const* outputH) {
-  float efw[2][PART_LEN1];
-  float xfw[2][PART_LEN1];
-  float dfw[2][PART_LEN1];
-  float comfortNoiseHband[2][PART_LEN1];
-  float fft[PART_LEN2];
-  float nlpGainHband;
-  int i;
-  size_t j;
-
-  // Coherence and non-linear filter
-  float cohde[PART_LEN1], cohxd[PART_LEN1];
+static void FormSuppressionGain(AecCore* aec,
+                                float cohde[PART_LEN1],
+                                float cohxd[PART_LEN1],
+                                float hNl[PART_LEN1]) {
   float hNlDeAvg, hNlXdAvg;
-  float hNl[PART_LEN1];
   float hNlPref[kPrefBandSize];
   float hNlFb = 0, hNlFbLow = 0;
-  const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
   const int prefBandSize = kPrefBandSize / aec->mult;
+  const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
   const int minPrefBand = 4 / aec->mult;
   // Power estimate smoothing coefficients.
   const float* min_overdrive = aec->extended_filter_enabled
                                    ? kExtendedMinOverDrive
                                    : kNormalMinOverDrive;
 
-  // Filter energy
-  const int delayEstInterval = 10 * aec->mult;
-
-  float* xfw_ptr = NULL;
-
-  // Update eBuf with echo subtractor output.
-  memcpy(aec->eBuf + PART_LEN, echo_subtractor_output,
-         sizeof(float) * PART_LEN);
-
-  // Analysis filter banks for the echo suppressor.
-  // Windowed near-end ffts.
-  WindowData(fft, aec->dBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, dfw);
-
-  // Windowed echo suppressor output ffts.
-  WindowData(fft, aec->eBuf);
-  aec_rdft_forward_128(fft);
-  StoreAsComplex(fft, efw);
-
-  // NLP
-
-  // Convert far-end partition to the frequency domain with windowing.
-  WindowData(fft, farend);
-  Fft(fft, xfw);
-  xfw_ptr = &xfw[0][0];
-
-  // Buffer far.
-  memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
-
-  aec->delayEstCtr++;
-  if (aec->delayEstCtr == delayEstInterval) {
-    aec->delayEstCtr = 0;
-    aec->delayIdx = WebRtcAec_PartitionDelay(aec);
-  }
-
-  // Use delayed far.
-  memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1,
-         sizeof(xfw[0][0]) * 2 * PART_LEN1);
-
-  WebRtcAec_SubbandCoherence(aec, efw, dfw, xfw, fft, cohde, cohxd,
-                             &aec->extreme_filter_divergence);
-
-  // Select the microphone signal as output if the filter is deemed to have
-  // diverged.
-  if (aec->divergeState) {
-    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
-  }
-
   hNlXdAvg = 0;
-  for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
+  for (int i = minPrefBand; i < prefBandSize + minPrefBand; ++i) {
     hNlXdAvg += cohxd[i];
   }
   hNlXdAvg /= prefBandSize;
   hNlXdAvg = 1 - hNlXdAvg;
 
   hNlDeAvg = 0;
-  for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
+  for (int i = minPrefBand; i < prefBandSize + minPrefBand; ++i) {
     hNlDeAvg += cohde[i];
   }
   hNlDeAvg /= prefBandSize;
@@ -1139,11 +1030,11 @@ static void EchoSuppression(AecCore* aec,
     aec->overDrive = min_overdrive[aec->nlp_mode];
 
     if (aec->stNearState == 1) {
-      memcpy(hNl, cohde, sizeof(hNl));
+      memcpy(hNl, cohde, sizeof(hNl[0]) * PART_LEN1);
       hNlFb = hNlDeAvg;
       hNlFbLow = hNlDeAvg;
     } else {
-      for (i = 0; i < PART_LEN1; i++) {
+      for (int i = 0; i < PART_LEN1; ++i) {
         hNl[i] = 1 - cohxd[i];
       }
       hNlFb = hNlXdAvg;
@@ -1152,12 +1043,12 @@ static void EchoSuppression(AecCore* aec,
   } else {
     if (aec->stNearState == 1) {
       aec->echoState = 0;
-      memcpy(hNl, cohde, sizeof(hNl));
+      memcpy(hNl, cohde, sizeof(hNl[0]) * PART_LEN1);
       hNlFb = hNlDeAvg;
       hNlFbLow = hNlDeAvg;
     } else {
       aec->echoState = 1;
-      for (i = 0; i < PART_LEN1; i++) {
+      for (int i = 0; i < PART_LEN1; ++i) {
         hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]);
       }
 
@@ -1197,16 +1088,96 @@ static void EchoSuppression(AecCore* aec,
   }
 
   // Smooth the overdrive.
-  if (aec->overDrive < aec->overDriveSm) {
-    aec->overDriveSm = 0.99f * aec->overDriveSm + 0.01f * aec->overDrive;
+  if (aec->overDrive < aec->overdrive_scaling) {
+    aec->overdrive_scaling =
+        0.99f * aec->overdrive_scaling + 0.01f * aec->overDrive;
   } else {
-    aec->overDriveSm = 0.9f * aec->overDriveSm + 0.1f * aec->overDrive;
+    aec->overdrive_scaling =
+        0.9f * aec->overdrive_scaling + 0.1f * aec->overDrive;
+  }
+
+  // Apply the overdrive.
+  WebRtcAec_Overdrive(aec->overdrive_scaling, hNlFb, hNl);
+}
+
+static void EchoSuppression(AecCore* aec,
+                            float farend[PART_LEN2],
+                            float* echo_subtractor_output,
+                            float* output,
+                            float* const* outputH) {
+  float efw[2][PART_LEN1];
+  float xfw[2][PART_LEN1];
+  float dfw[2][PART_LEN1];
+  float comfortNoiseHband[2][PART_LEN1];
+  float fft[PART_LEN2];
+  float nlpGainHband;
+  int i;
+  size_t j;
+
+  // Coherence and non-linear filter
+  float cohde[PART_LEN1], cohxd[PART_LEN1];
+  float hNl[PART_LEN1];
+
+  // Filter energy
+  const int delayEstInterval = 10 * aec->mult;
+
+  float* xfw_ptr = NULL;
+
+  // Update eBuf with echo subtractor output.
+  memcpy(aec->eBuf + PART_LEN, echo_subtractor_output,
+         sizeof(float) * PART_LEN);
+
+  // Analysis filter banks for the echo suppressor.
+  // Windowed near-end ffts.
+  WindowData(fft, aec->dBuf);
+  aec_rdft_forward_128(fft);
+  StoreAsComplex(fft, dfw);
+
+  // Windowed echo suppressor output ffts.
+  WindowData(fft, aec->eBuf);
+  aec_rdft_forward_128(fft);
+  StoreAsComplex(fft, efw);
+
+  // NLP
+
+  // Convert far-end partition to the frequency domain with windowing.
+  WindowData(fft, farend);
+  Fft(fft, xfw);
+  xfw_ptr = &xfw[0][0];
+
+  // Buffer far.
+  memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
+
+  aec->delayEstCtr++;
+  if (aec->delayEstCtr == delayEstInterval) {
+    aec->delayEstCtr = 0;
+    aec->delayIdx = WebRtcAec_PartitionDelay(aec->num_partitions, aec->wfBuf);
+  }
+
+  // Use delayed far.
+  memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1,
+         sizeof(xfw[0][0]) * 2 * PART_LEN1);
+
+  WebRtcAec_UpdateCoherenceSpectra(aec->mult, aec->extended_filter_enabled == 1,
+                                   efw, dfw, xfw, &aec->coherence_state,
+                                   &aec->divergeState,
+                                   &aec->extreme_filter_divergence);
+
+  WebRtcAec_ComputeCoherence(&aec->coherence_state, cohde, cohxd);
+
+  // Select the microphone signal as output if the filter is deemed to have
+  // diverged.
+  if (aec->divergeState) {
+    memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
   }
 
-  WebRtcAec_OverdriveAndSuppress(aec, hNl, hNlFb, efw);
+  FormSuppressionGain(aec, cohde, cohxd, hNl);
+
+  WebRtcAec_Suppress(hNl, efw);
 
   // Add comfort noise.
-  WebRtcAec_ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
+  ComfortNoise(aec->num_bands > 1, &aec->seed, efw, comfortNoiseHband,
+               aec->noisePow, hNl);
 
   // Inverse error fft.
   ScaledInverseFft(efw, fft, 2.0f, 1);
@@ -1315,15 +1286,10 @@ static void ProcessBlock(AecCore* aec) {
   WebRtc_ReadBuffer(aec->far_time_buf, reinterpret_cast<void**>(&farend_ptr),
                     farend, 1);
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  {
-    // TODO(minyue): |farend_ptr| starts from buffered samples. This will be
-    // modified when |aec->far_time_buf| is revised.
-    RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, &farend_ptr[PART_LEN], PART_LEN);
-
-    RTC_AEC_DEBUG_WAV_WRITE(aec->nearFile, nearend_ptr, PART_LEN);
-  }
-#endif
+  aec->data_dumper->DumpWav("aec_far", PART_LEN, &farend_ptr[PART_LEN],
+                            std::min(aec->sampFreq, 16000), 1);
+  aec->data_dumper->DumpWav("aec_near", PART_LEN, nearend_ptr,
+                            std::min(aec->sampFreq, 16000), 1);
 
   if (aec->metricsMode == 1) {
     // Update power levels
@@ -1417,12 +1383,14 @@ static void ProcessBlock(AecCore* aec) {
   }
 
   // Perform echo subtraction.
-  EchoSubtraction(aec, aec->num_partitions, aec->extended_filter_enabled,
-                  aec->filter_step_size, aec->error_threshold, &x_fft[0][0],
-                  &aec->xfBufBlockPos, aec->xfBuf, nearend_ptr, aec->xPow,
-                  aec->wfBuf, echo_subtractor_output);
+  EchoSubtraction(aec->num_partitions, aec->extended_filter_enabled,
+                  &aec->extreme_filter_divergence, aec->filter_step_size,
+                  aec->error_threshold, &x_fft[0][0], &aec->xfBufBlockPos,
+                  aec->xfBuf, nearend_ptr, aec->xPow, aec->wfBuf,
+                  echo_subtractor_output);
 
-  RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, echo_subtractor_output, PART_LEN);
+  aec->data_dumper->DumpWav("aec_out_linear", PART_LEN, echo_subtractor_output,
+                            std::min(aec->sampFreq, 16000), 1);
 
   if (aec->metricsMode == 1) {
     UpdateLevel(&aec->linoutlevel,
@@ -1444,12 +1412,14 @@ static void ProcessBlock(AecCore* aec) {
     WebRtc_WriteBuffer(aec->outFrBufH[i], outputH[i], PART_LEN);
   }
 
-  RTC_AEC_DEBUG_WAV_WRITE(aec->outFile, output, PART_LEN);
+  aec->data_dumper->DumpWav("aec_out", PART_LEN, output,
+                            std::min(aec->sampFreq, 16000), 1);
 }
 
-AecCore* WebRtcAec_CreateAec() {
+AecCore* WebRtcAec_CreateAec(int instance_count) {
   int i;
-  AecCore* aec = new AecCore;
+  AecCore* aec = new AecCore(instance_count);
+
   if (!aec) {
     return NULL;
   }
@@ -1493,12 +1463,6 @@ AecCore* WebRtcAec_CreateAec() {
     return NULL;
   }
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  aec->instance_index = webrtc_aec_instance_count;
-
-  aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = NULL;
-  aec->debug_dump_count = 0;
-#endif
   aec->delay_estimator_farend =
       WebRtc_CreateDelayEstimatorFarend(PART_LEN1, kHistorySizeBlocks);
   if (aec->delay_estimator_farend == NULL) {
@@ -1530,9 +1494,10 @@ AecCore* WebRtcAec_CreateAec() {
   WebRtcAec_FilterFar = FilterFar;
   WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
   WebRtcAec_FilterAdaptation = FilterAdaptation;
-  WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
-  WebRtcAec_ComfortNoise = ComfortNoise;
-  WebRtcAec_SubbandCoherence = SubbandCoherence;
+  WebRtcAec_Overdrive = Overdrive;
+  WebRtcAec_Suppress = Suppress;
+  WebRtcAec_ComputeCoherence = ComputeCoherence;
+  WebRtcAec_UpdateCoherenceSpectra = UpdateCoherenceSpectra;
   WebRtcAec_StoreAsComplex = StoreAsComplex;
   WebRtcAec_PartitionDelay = PartitionDelay;
   WebRtcAec_WindowData = WindowData;
@@ -1549,10 +1514,6 @@ AecCore* WebRtcAec_CreateAec() {
 
 #if defined(WEBRTC_HAS_NEON)
   WebRtcAec_InitAec_neon();
-#elif defined(WEBRTC_DETECT_NEON)
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    WebRtcAec_InitAec_neon();
-  }
 #endif
 
   aec_rdft_init();
@@ -1576,12 +1537,6 @@ void WebRtcAec_FreeAec(AecCore* aec) {
 
   WebRtc_FreeBuffer(aec->far_time_buf);
 
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->farFile);
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->nearFile);
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->outFile);
-  RTC_AEC_DEBUG_WAV_CLOSE(aec->outLinearFile);
-  RTC_AEC_DEBUG_RAW_CLOSE(aec->e_fft_file);
-
   WebRtc_FreeDelayEstimator(aec->delay_estimator);
   WebRtc_FreeDelayEstimatorFarend(aec->delay_estimator_farend);
 
@@ -1626,6 +1581,7 @@ static void SetErrorThreshold(AecCore* aec) {
 
 int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   int i;
+  aec->data_dumper->InitiateNewSetOfRecordings();
 
   aec->sampFreq = sampFreq;
 
@@ -1648,27 +1604,6 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   // Initialize far-end buffers.
   WebRtc_InitBuffer(aec->far_time_buf);
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  {
-    int process_rate = sampFreq > 16000 ? 16000 : sampFreq;
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_far", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->farFile);
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_near", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->nearFile);
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_out", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->outFile);
-    RTC_AEC_DEBUG_WAV_REOPEN("aec_out_linear", aec->instance_index,
-                             aec->debug_dump_count, process_rate,
-                             &aec->outLinearFile);
-  }
-
-  RTC_AEC_DEBUG_RAW_OPEN("aec_e_fft", aec->debug_dump_count, &aec->e_fft_file);
-
-  ++aec->debug_dump_count;
-#endif
   aec->system_delay = 0;
 
   if (WebRtc_InitDelayEstimatorFarend(aec->delay_estimator_farend) != 0) {
@@ -1749,18 +1684,18 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   // doesn't change the output at all and yields 0.4% overall speedup.
   memset(aec->xfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
   memset(aec->wfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
-  memset(aec->sde, 0, sizeof(complex_t) * PART_LEN1);
-  memset(aec->sxd, 0, sizeof(complex_t) * PART_LEN1);
+  memset(aec->coherence_state.sde, 0, sizeof(complex_t) * PART_LEN1);
+  memset(aec->coherence_state.sxd, 0, sizeof(complex_t) * PART_LEN1);
   memset(aec->xfwBuf, 0,
          sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
-  memset(aec->se, 0, sizeof(float) * PART_LEN1);
+  memset(aec->coherence_state.se, 0, sizeof(float) * PART_LEN1);
 
   // To prevent numerical instability in the first block.
   for (i = 0; i < PART_LEN1; i++) {
-    aec->sd[i] = 1;
+    aec->coherence_state.sd[i] = 1;
   }
   for (i = 0; i < PART_LEN1; i++) {
-    aec->sx[i] = 1;
+    aec->coherence_state.sx[i] = 1;
   }
 
   memset(aec->hNs, 0, sizeof(aec->hNs));
@@ -1772,7 +1707,7 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
   aec->hNlNewMin = 0;
   aec->hNlMinCtr = 0;
   aec->overDrive = 2;
-  aec->overDriveSm = 2;
+  aec->overdrive_scaling = 2;
   aec->delayIdx = 0;
   aec->stNearState = 0;
   aec->echoState = 0;
@@ -1878,11 +1813,15 @@ void WebRtcAec_ProcessFrames(AecCore* aec,
       // rounding, like -16.
       int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
       int moved_elements = WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+      MaybeLogDelayAdjustment(moved_elements * (aec->sampFreq == 8000 ? 8 : 4),
+                              DelaySource::kSystemDelay);
       aec->knownDelay -= moved_elements * PART_LEN;
     } else {
       // 2 b) Apply signal based delay correction.
       int move_elements = SignalBasedDelayCorrection(aec);
       int moved_elements = WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+      MaybeLogDelayAdjustment(moved_elements * (aec->sampFreq == 8000 ? 8 : 4),
+                              DelaySource::kDelayAgnostic);
       int far_near_buffer_diff =
           WebRtc_available_read(aec->far_time_buf) -
           WebRtc_available_read(aec->nearFrBuf) / PART_LEN;
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h
index bd5b283eca0..1ab20201860 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core.h
@@ -17,6 +17,15 @@
 
 #include <stddef.h>
 
+#include <memory>
+
+extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
+}
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_processing/aec/aec_common.h"
+#include "webrtc/modules/audio_processing/utility/block_mean_calculator.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -27,6 +36,8 @@ namespace webrtc {
 #define PART_LEN2 (PART_LEN * 2)  // Length of partition * 2
 #define NUM_HIGH_BANDS_MAX 2      // Max number of high bands
 
+class ApmDataDumper;
+
 typedef float complex_t[2];
 // For performance reasons, some arrays of complex numbers are replaced by twice
 // as long arrays of float, all the real parts followed by all the imaginary
@@ -47,20 +58,188 @@ typedef struct Stats {
   float sum;
   float hisum;
   float himean;
-  int counter;
-  int hicounter;
+  size_t counter;
+  size_t hicounter;
 } Stats;
 
-typedef struct AecCore AecCore;
+// Number of partitions for the extended filter mode. The first one is an enum
+// to be used in array declarations, as it represents the maximum filter length.
+enum { kExtendedNumPartitions = 32 };
+static const int kNormalNumPartitions = 12;
+
+// Delay estimator constants, used for logging and delay compensation if
+// if reported delays are disabled.
+enum { kLookaheadBlocks = 15 };
+enum {
+  // 500 ms for 16 kHz which is equivalent with the limit of reported delays.
+  kHistorySizeBlocks = 125
+};
+
+typedef struct PowerLevel {
+  PowerLevel();
+
+  BlockMeanCalculator framelevel;
+  BlockMeanCalculator averagelevel;
+  float minlevel;
+} PowerLevel;
+
+class DivergentFilterFraction {
+ public:
+  DivergentFilterFraction();
+
+  // Reset.
+  void Reset();
+
+  void AddObservation(const PowerLevel& nearlevel,
+                      const PowerLevel& linoutlevel,
+                      const PowerLevel& nlpoutlevel);
+
+  // Return the latest fraction.
+  float GetLatestFraction() const;
+
+ private:
+  // Clear all values added.
+  void Clear();
+
+  size_t count_;
+  size_t occurrence_;
+  float fraction_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(DivergentFilterFraction);
+};
+
+typedef struct CoherenceState {
+  complex_t sde[PART_LEN1];  // cross-psd of nearend and error
+  complex_t sxd[PART_LEN1];  // cross-psd of farend and nearend
+  float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1];  // far, near, error psd
+} CoherenceState;
+
+struct AecCore {
+  explicit AecCore(int instance_index);
+  ~AecCore();
+
+  std::unique_ptr<ApmDataDumper> data_dumper;
+
+  CoherenceState coherence_state;
+
+  int farBufWritePos, farBufReadPos;
+
+  int knownDelay;
+  int inSamples, outSamples;
+  int delayEstCtr;
+
+  RingBuffer* nearFrBuf;
+  RingBuffer* outFrBuf;
+
+  RingBuffer* nearFrBufH[NUM_HIGH_BANDS_MAX];
+  RingBuffer* outFrBufH[NUM_HIGH_BANDS_MAX];
 
-AecCore* WebRtcAec_CreateAec();  // Returns NULL on error.
+  float dBuf[PART_LEN2];  // nearend
+  float eBuf[PART_LEN2];  // error
+
+  float dBufH[NUM_HIGH_BANDS_MAX][PART_LEN2];  // nearend
+
+  float xPow[PART_LEN1];
+  float dPow[PART_LEN1];
+  float dMinPow[PART_LEN1];
+  float dInitMinPow[PART_LEN1];
+  float* noisePow;
+
+  float xfBuf[2][kExtendedNumPartitions * PART_LEN1];  // farend fft buffer
+  float wfBuf[2][kExtendedNumPartitions * PART_LEN1];  // filter fft
+  // Farend windowed fft buffer.
+  complex_t xfwBuf[kExtendedNumPartitions * PART_LEN1];
+
+  float hNs[PART_LEN1];
+  float hNlFbMin, hNlFbLocalMin;
+  float hNlXdAvgMin;
+  int hNlNewMin, hNlMinCtr;
+  float overDrive;
+  float overdrive_scaling;
+  int nlp_mode;
+  float outBuf[PART_LEN];
+  int delayIdx;
+
+  short stNearState, echoState;
+  short divergeState;
+
+  int xfBufBlockPos;
+
+  RingBuffer* far_time_buf;
+
+  int system_delay;  // Current system delay buffered in AEC.
+
+  int mult;  // sampling frequency multiple
+  int sampFreq = 16000;
+  size_t num_bands;
+  uint32_t seed;
+
+  float filter_step_size;  // stepsize
+  float error_threshold;   // error threshold
+
+  int noiseEstCtr;
+
+  PowerLevel farlevel;
+  PowerLevel nearlevel;
+  PowerLevel linoutlevel;
+  PowerLevel nlpoutlevel;
+
+  int metricsMode;
+  int stateCounter;
+  Stats erl;
+  Stats erle;
+  Stats aNlp;
+  Stats rerl;
+  DivergentFilterFraction divergent_filter_fraction;
+
+  // Quantities to control H band scaling for SWB input
+  int freq_avg_ic;       // initial bin for averaging nlp gain
+  int flag_Hband_cn;     // for comfort noise
+  float cn_scale_Hband;  // scale for comfort noise in H band
+
+  int delay_metrics_delivered;
+  int delay_histogram[kHistorySizeBlocks];
+  int num_delay_values;
+  int delay_median;
+  int delay_std;
+  float fraction_poor_delays;
+  int delay_logging_enabled;
+  void* delay_estimator_farend;
+  void* delay_estimator;
+  // Variables associated with delay correction through signal based delay
+  // estimation feedback.
+  int signal_delay_correction;
+  int previous_delay;
+  int delay_correction_count;
+  int shift_offset;
+  float delay_quality_threshold;
+  int frame_count;
+
+  // 0 = delay agnostic mode (signal based delay correction) disabled.
+  // Otherwise enabled.
+  int delay_agnostic_enabled;
+  // 1 = extended filter mode enabled, 0 = disabled.
+  int extended_filter_enabled;
+  // 1 = next generation aec mode enabled, 0 = disabled.
+  int aec3_enabled;
+  bool refined_adaptive_filter_enabled;
+
+  // Runtime selection of number of filter partitions.
+  int num_partitions;
+
+  // Flag that extreme filter divergence has been detected by the Echo
+  // Suppressor.
+  int extreme_filter_divergence;
+};
+
+AecCore* WebRtcAec_CreateAec(int instance_count);  // Returns NULL on error.
 void WebRtcAec_FreeAec(AecCore* aec);
 int WebRtcAec_InitAec(AecCore* aec, int sampFreq);
 void WebRtcAec_InitAec_SSE2(void);
 #if defined(MIPS_FPU_LE)
 void WebRtcAec_InitAec_mips(void);
 #endif
-#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 void WebRtcAec_InitAec_neon(void);
 #endif
 
@@ -97,9 +276,6 @@ void WebRtcAec_GetEchoStats(AecCore* self,
                             Stats* erle,
                             Stats* a_nlp,
                             float* divergent_filter_fraction);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-void* WebRtcAec_far_time_buf(AecCore* self);
-#endif
 
 // Sets local configuration modes.
 void WebRtcAec_SetConfigCore(AecCore* self,
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h
index 1f7b6b541fa..d4fad9e5e63 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_internal.h
@@ -11,12 +11,16 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
 
+#include <memory>
+
 extern "C" {
 #include "webrtc/common_audio/ring_buffer.h"
 }
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/modules/audio_processing/utility/block_mean_calculator.h"
 #include "webrtc/typedefs.h"
 
@@ -68,8 +72,19 @@ class DivergentFilterFraction {
   RTC_DISALLOW_COPY_AND_ASSIGN(DivergentFilterFraction);
 };
 
+typedef struct CoherenceState {
+  complex_t sde[PART_LEN1];  // cross-psd of nearend and error
+  complex_t sxd[PART_LEN1];  // cross-psd of farend and nearend
+  float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1];  // far, near, error psd
+} CoherenceState;
+
 struct AecCore {
-  AecCore();
+  explicit AecCore(int instance_index);
+  ~AecCore();
+
+  std::unique_ptr<ApmDataDumper> data_dumper;
+
+  CoherenceState coherence_state;
 
   int farBufWritePos, farBufReadPos;
 
@@ -96,17 +111,15 @@ struct AecCore {
 
   float xfBuf[2][kExtendedNumPartitions * PART_LEN1];  // farend fft buffer
   float wfBuf[2][kExtendedNumPartitions * PART_LEN1];  // filter fft
-  complex_t sde[PART_LEN1];  // cross-psd of nearend and error
-  complex_t sxd[PART_LEN1];  // cross-psd of farend and nearend
   // Farend windowed fft buffer.
   complex_t xfwBuf[kExtendedNumPartitions * PART_LEN1];
 
-  float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1];  // far, near, error psd
   float hNs[PART_LEN1];
   float hNlFbMin, hNlFbLocalMin;
   float hNlXdAvgMin;
   int hNlNewMin, hNlMinCtr;
-  float overDrive, overDriveSm;
+  float overDrive;
+  float overdrive_scaling;
   int nlp_mode;
   float outBuf[PART_LEN];
   int delayIdx;
@@ -181,22 +194,6 @@ struct AecCore {
   // Flag that extreme filter divergence has been detected by the Echo
   // Suppressor.
   int extreme_filter_divergence;
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  // Sequence number of this AEC instance, so that different instances can
-  // choose different dump file names.
-  int instance_index;
-
-  // Number of times we've restarted dumping; used to pick new dump file names
-  // each time.
-  int debug_dump_count;
-
-  rtc_WavWriter* farFile;
-  rtc_WavWriter* nearFile;
-  rtc_WavWriter* outFile;
-  rtc_WavWriter* outLinearFile;
-  FILE* e_fft_file;
-#endif
 };
 
 typedef void (*WebRtcAecFilterFar)(
@@ -218,30 +215,34 @@ typedef void (*WebRtcAecFilterAdaptation)(
     float e_fft[2][PART_LEN1],
     float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
 extern WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
-typedef void (*WebRtcAecOverdriveAndSuppress)(AecCore* aec,
-                                              float hNl[PART_LEN1],
-                                              const float hNlFb,
-                                              float efw[2][PART_LEN1]);
-extern WebRtcAecOverdriveAndSuppress WebRtcAec_OverdriveAndSuppress;
-
-typedef void (*WebRtcAecComfortNoise)(AecCore* aec,
-                                      float efw[2][PART_LEN1],
-                                      float comfortNoiseHband[2][PART_LEN1],
-                                      const float* noisePow,
-                                      const float* lambda);
-extern WebRtcAecComfortNoise WebRtcAec_ComfortNoise;
-
-typedef void (*WebRtcAecSubBandCoherence)(AecCore* aec,
-                                          float efw[2][PART_LEN1],
-                                          float dfw[2][PART_LEN1],
-                                          float xfw[2][PART_LEN1],
-                                          float* fft,
-                                          float* cohde,
-                                          float* cohxd,
-                                          int* extreme_filter_divergence);
-extern WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
 
-typedef int (*WebRtcAecPartitionDelay)(const AecCore* aec);
+typedef void (*WebRtcAecOverdrive)(float overdrive_scaling,
+                                   const float hNlFb,
+                                   float hNl[PART_LEN1]);
+extern WebRtcAecOverdrive WebRtcAec_Overdrive;
+
+typedef void (*WebRtcAecSuppress)(const float hNl[PART_LEN1],
+                                  float efw[2][PART_LEN1]);
+extern WebRtcAecSuppress WebRtcAec_Suppress;
+
+typedef void (*WebRtcAecComputeCoherence)(const CoherenceState* coherence_state,
+                                          float* cohde,
+                                          float* cohxd);
+extern WebRtcAecComputeCoherence WebRtcAec_ComputeCoherence;
+
+typedef void (*WebRtcAecUpdateCoherenceSpectra)(int mult,
+                                                bool extended_filter_enabled,
+                                                float efw[2][PART_LEN1],
+                                                float dfw[2][PART_LEN1],
+                                                float xfw[2][PART_LEN1],
+                                                CoherenceState* coherence_state,
+                                                short* filter_divergence_state,
+                                                int* extreme_filter_divergence);
+extern WebRtcAecUpdateCoherenceSpectra WebRtcAec_UpdateCoherenceSpectra;
+
+typedef int (*WebRtcAecPartitionDelay)(
+    int num_partitions,
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
 extern WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
 
 typedef void (*WebRtcAecStoreAsComplex)(const float* data,
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_mips.cc b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_mips.cc
index 5c6d8ebb73a..a9b5cd4e60f 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_mips.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_mips.cc
@@ -19,314 +19,14 @@
 extern "C" {
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 }
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
-extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
-}
 
 namespace webrtc {
 
 extern const float WebRtcAec_weightCurve[65];
 extern const float WebRtcAec_overDriveCurve[65];
 
-void WebRtcAec_ComfortNoise_mips(AecCore* aec,
-                                 float efw[2][PART_LEN1],
-                                 float comfortNoiseHband[2][PART_LEN1],
-                                 const float* noisePow,
-                                 const float* lambda) {
-  int i, num;
-  float rand[PART_LEN];
-  float noise, noiseAvg, tmp, tmpAvg;
-  int16_t randW16[PART_LEN];
-  complex_t u[PART_LEN1];
-
-  const float pi2 = 6.28318530717959f;
-  const float pi2t = pi2 / 32768;
-
-  // Generate a uniform random array on [0 1]
-  WebRtcSpl_RandUArray(randW16, PART_LEN, &aec->seed);
-
-  int16_t* randWptr = randW16;
-  float randTemp, randTemp2, randTemp3, randTemp4;
-  int32_t tmp1s, tmp2s, tmp3s, tmp4s;
-
-  for (i = 0; i < PART_LEN; i += 4) {
-    __asm __volatile(
-      ".set     push                                           \n\t"
-      ".set     noreorder                                      \n\t"
-      "lh       %[tmp1s],       0(%[randWptr])                 \n\t"
-      "lh       %[tmp2s],       2(%[randWptr])                 \n\t"
-      "lh       %[tmp3s],       4(%[randWptr])                 \n\t"
-      "lh       %[tmp4s],       6(%[randWptr])                 \n\t"
-      "mtc1     %[tmp1s],       %[randTemp]                    \n\t"
-      "mtc1     %[tmp2s],       %[randTemp2]                   \n\t"
-      "mtc1     %[tmp3s],       %[randTemp3]                   \n\t"
-      "mtc1     %[tmp4s],       %[randTemp4]                   \n\t"
-      "cvt.s.w  %[randTemp],    %[randTemp]                    \n\t"
-      "cvt.s.w  %[randTemp2],   %[randTemp2]                   \n\t"
-      "cvt.s.w  %[randTemp3],   %[randTemp3]                   \n\t"
-      "cvt.s.w  %[randTemp4],   %[randTemp4]                   \n\t"
-      "addiu    %[randWptr],    %[randWptr],      8            \n\t"
-      "mul.s    %[randTemp],    %[randTemp],      %[pi2t]      \n\t"
-      "mul.s    %[randTemp2],   %[randTemp2],     %[pi2t]      \n\t"
-      "mul.s    %[randTemp3],   %[randTemp3],     %[pi2t]      \n\t"
-      "mul.s    %[randTemp4],   %[randTemp4],     %[pi2t]      \n\t"
-      ".set     pop                                            \n\t"
-      : [randWptr] "+r" (randWptr), [randTemp] "=&f" (randTemp),
-        [randTemp2] "=&f" (randTemp2), [randTemp3] "=&f" (randTemp3),
-        [randTemp4] "=&f" (randTemp4), [tmp1s] "=&r" (tmp1s),
-        [tmp2s] "=&r" (tmp2s), [tmp3s] "=&r" (tmp3s),
-        [tmp4s] "=&r" (tmp4s)
-      : [pi2t] "f" (pi2t)
-      : "memory");
-
-    u[i + 1][0] = cosf(randTemp);
-    u[i + 1][1] = sinf(randTemp);
-    u[i + 2][0] = cosf(randTemp2);
-    u[i + 2][1] = sinf(randTemp2);
-    u[i + 3][0] = cosf(randTemp3);
-    u[i + 3][1] = sinf(randTemp3);
-    u[i + 4][0] = cosf(randTemp4);
-    u[i + 4][1] = sinf(randTemp4);
-  }
-
-  // Reject LF noise
-  float* u_ptr = &u[1][0];
-  float noise2, noise3, noise4;
-  float tmp1f, tmp2f, tmp3f, tmp4f, tmp5f, tmp6f, tmp7f, tmp8f;
-
-  u[0][0] = 0;
-  u[0][1] = 0;
-  for (i = 1; i < PART_LEN1; i += 4) {
-    __asm __volatile(
-      ".set     push                                            \n\t"
-      ".set     noreorder                                       \n\t"
-      "lwc1     %[noise],       4(%[noisePow])                  \n\t"
-      "lwc1     %[noise2],      8(%[noisePow])                  \n\t"
-      "lwc1     %[noise3],      12(%[noisePow])                 \n\t"
-      "lwc1     %[noise4],      16(%[noisePow])                 \n\t"
-      "sqrt.s   %[noise],       %[noise]                        \n\t"
-      "sqrt.s   %[noise2],      %[noise2]                       \n\t"
-      "sqrt.s   %[noise3],      %[noise3]                       \n\t"
-      "sqrt.s   %[noise4],      %[noise4]                       \n\t"
-      "lwc1     %[tmp1f],       0(%[u_ptr])                     \n\t"
-      "lwc1     %[tmp2f],       4(%[u_ptr])                     \n\t"
-      "lwc1     %[tmp3f],       8(%[u_ptr])                     \n\t"
-      "lwc1     %[tmp4f],       12(%[u_ptr])                    \n\t"
-      "lwc1     %[tmp5f],       16(%[u_ptr])                    \n\t"
-      "lwc1     %[tmp6f],       20(%[u_ptr])                    \n\t"
-      "lwc1     %[tmp7f],       24(%[u_ptr])                    \n\t"
-      "lwc1     %[tmp8f],       28(%[u_ptr])                    \n\t"
-      "addiu    %[noisePow],    %[noisePow],      16            \n\t"
-      "mul.s    %[tmp1f],       %[tmp1f],         %[noise]      \n\t"
-      "mul.s    %[tmp2f],       %[tmp2f],         %[noise]      \n\t"
-      "mul.s    %[tmp3f],       %[tmp3f],         %[noise2]     \n\t"
-      "mul.s    %[tmp4f],       %[tmp4f],         %[noise2]     \n\t"
-      "mul.s    %[tmp5f],       %[tmp5f],         %[noise3]     \n\t"
-      "mul.s    %[tmp6f],       %[tmp6f],         %[noise3]     \n\t"
-      "swc1     %[tmp1f],       0(%[u_ptr])                     \n\t"
-      "swc1     %[tmp3f],       8(%[u_ptr])                     \n\t"
-      "mul.s    %[tmp8f],       %[tmp8f],         %[noise4]     \n\t"
-      "mul.s    %[tmp7f],       %[tmp7f],         %[noise4]     \n\t"
-      "neg.s    %[tmp2f]                                        \n\t"
-      "neg.s    %[tmp4f]                                        \n\t"
-      "neg.s    %[tmp6f]                                        \n\t"
-      "neg.s    %[tmp8f]                                        \n\t"
-      "swc1     %[tmp5f],       16(%[u_ptr])                    \n\t"
-      "swc1     %[tmp7f],       24(%[u_ptr])                    \n\t"
-      "swc1     %[tmp2f],       4(%[u_ptr])                     \n\t"
-      "swc1     %[tmp4f],       12(%[u_ptr])                    \n\t"
-      "swc1     %[tmp6f],       20(%[u_ptr])                    \n\t"
-      "swc1     %[tmp8f],       28(%[u_ptr])                    \n\t"
-      "addiu    %[u_ptr],       %[u_ptr],         32            \n\t"
-      ".set     pop                                             \n\t"
-      : [u_ptr] "+r" (u_ptr),  [noisePow] "+r" (noisePow),
-        [noise] "=&f" (noise), [noise2] "=&f" (noise2),
-        [noise3] "=&f" (noise3), [noise4] "=&f" (noise4),
-        [tmp1f] "=&f" (tmp1f), [tmp2f] "=&f" (tmp2f),
-        [tmp3f] "=&f" (tmp3f), [tmp4f] "=&f" (tmp4f),
-        [tmp5f] "=&f" (tmp5f), [tmp6f] "=&f" (tmp6f),
-        [tmp7f] "=&f" (tmp7f), [tmp8f] "=&f" (tmp8f)
-      :
-      : "memory");
-  }
-  u[PART_LEN][1] = 0;
-  noisePow -= PART_LEN;
-
-  u_ptr = &u[0][0];
-  float* u_ptr_end = &u[PART_LEN][0];
-  float* efw_ptr_0 = &efw[0][0];
-  float* efw_ptr_1 = &efw[1][0];
-  float tmp9f, tmp10f;
-  const float tmp1c = 1.0;
-
-  __asm __volatile(
-    ".set     push                                                        \n\t"
-    ".set     noreorder                                                   \n\t"
-    "1:                                                                   \n\t"
-    "lwc1     %[tmp1f],       0(%[lambda])                                \n\t"
-    "lwc1     %[tmp6f],       4(%[lambda])                                \n\t"
-    "addiu    %[lambda],      %[lambda],        8                         \n\t"
-    "c.lt.s   %[tmp1f],       %[tmp1c]                                    \n\t"
-    "bc1f     4f                                                          \n\t"
-    " nop                                                                 \n\t"
-    "c.lt.s   %[tmp6f],       %[tmp1c]                                    \n\t"
-    "bc1f     3f                                                          \n\t"
-    " nop                                                                 \n\t"
-    "2:                                                                   \n\t"
-    "mul.s    %[tmp1f],       %[tmp1f],         %[tmp1f]                  \n\t"
-    "mul.s    %[tmp6f],       %[tmp6f],         %[tmp6f]                  \n\t"
-    "sub.s    %[tmp1f],       %[tmp1c],         %[tmp1f]                  \n\t"
-    "sub.s    %[tmp6f],       %[tmp1c],         %[tmp6f]                  \n\t"
-    "sqrt.s   %[tmp1f],       %[tmp1f]                                    \n\t"
-    "sqrt.s   %[tmp6f],       %[tmp6f]                                    \n\t"
-    "lwc1     %[tmp2f],       0(%[efw_ptr_0])                             \n\t"
-    "lwc1     %[tmp3f],       0(%[u_ptr])                                 \n\t"
-    "lwc1     %[tmp7f],       4(%[efw_ptr_0])                             \n\t"
-    "lwc1     %[tmp8f],       8(%[u_ptr])                                 \n\t"
-    "lwc1     %[tmp4f],       0(%[efw_ptr_1])                             \n\t"
-    "lwc1     %[tmp5f],       4(%[u_ptr])                                 \n\t"
-    "lwc1     %[tmp9f],       4(%[efw_ptr_1])                             \n\t"
-    "lwc1     %[tmp10f],      12(%[u_ptr])                                \n\t"
-#if !defined(MIPS32_R2_LE)
-    "mul.s    %[tmp3f],       %[tmp1f],         %[tmp3f]                  \n\t"
-    "add.s    %[tmp2f],       %[tmp2f],         %[tmp3f]                  \n\t"
-    "mul.s    %[tmp3f],       %[tmp1f],         %[tmp5f]                  \n\t"
-    "add.s    %[tmp4f],       %[tmp4f],         %[tmp3f]                  \n\t"
-    "mul.s    %[tmp3f],       %[tmp6f],         %[tmp8f]                  \n\t"
-    "add.s    %[tmp7f],       %[tmp7f],         %[tmp3f]                  \n\t"
-    "mul.s    %[tmp3f],       %[tmp6f],         %[tmp10f]                 \n\t"
-    "add.s    %[tmp9f],       %[tmp9f],         %[tmp3f]                  \n\t"
-#else  // #if !defined(MIPS32_R2_LE)
-    "madd.s   %[tmp2f],       %[tmp2f],         %[tmp1f],     %[tmp3f]    \n\t"
-    "madd.s   %[tmp4f],       %[tmp4f],         %[tmp1f],     %[tmp5f]    \n\t"
-    "madd.s   %[tmp7f],       %[tmp7f],         %[tmp6f],     %[tmp8f]    \n\t"
-    "madd.s   %[tmp9f],       %[tmp9f],         %[tmp6f],     %[tmp10f]   \n\t"
-#endif  // #if !defined(MIPS32_R2_LE)
-    "swc1     %[tmp2f],       0(%[efw_ptr_0])                             \n\t"
-    "swc1     %[tmp4f],       0(%[efw_ptr_1])                             \n\t"
-    "swc1     %[tmp7f],       4(%[efw_ptr_0])                             \n\t"
-    "b        5f                                                          \n\t"
-    " swc1    %[tmp9f],       4(%[efw_ptr_1])                             \n\t"
-    "3:                                                                   \n\t"
-    "mul.s    %[tmp1f],       %[tmp1f],         %[tmp1f]                  \n\t"
-    "sub.s    %[tmp1f],       %[tmp1c],         %[tmp1f]                  \n\t"
-    "sqrt.s   %[tmp1f],       %[tmp1f]                                    \n\t"
-    "lwc1     %[tmp2f],       0(%[efw_ptr_0])                             \n\t"
-    "lwc1     %[tmp3f],       0(%[u_ptr])                                 \n\t"
-    "lwc1     %[tmp4f],       0(%[efw_ptr_1])                             \n\t"
-    "lwc1     %[tmp5f],       4(%[u_ptr])                                 \n\t"
-#if !defined(MIPS32_R2_LE)
-    "mul.s    %[tmp3f],       %[tmp1f],         %[tmp3f]                  \n\t"
-    "add.s    %[tmp2f],       %[tmp2f],         %[tmp3f]                  \n\t"
-    "mul.s    %[tmp3f],       %[tmp1f],         %[tmp5f]                  \n\t"
-    "add.s    %[tmp4f],       %[tmp4f],         %[tmp3f]                  \n\t"
-#else  // #if !defined(MIPS32_R2_LE)
-    "madd.s   %[tmp2f],       %[tmp2f],         %[tmp1f],     %[tmp3f]    \n\t"
-    "madd.s   %[tmp4f],       %[tmp4f],         %[tmp1f],     %[tmp5f]    \n\t"
-#endif  // #if !defined(MIPS32_R2_LE)
-    "swc1     %[tmp2f],       0(%[efw_ptr_0])                             \n\t"
-    "b        5f                                                          \n\t"
-    " swc1    %[tmp4f],       0(%[efw_ptr_1])                             \n\t"
-    "4:                                                                   \n\t"
-    "c.lt.s   %[tmp6f],       %[tmp1c]                                    \n\t"
-    "bc1f     5f                                                          \n\t"
-    " nop                                                                 \n\t"
-    "mul.s    %[tmp6f],       %[tmp6f],         %[tmp6f]                  \n\t"
-    "sub.s    %[tmp6f],       %[tmp1c],         %[tmp6f]                  \n\t"
-    "sqrt.s   %[tmp6f],       %[tmp6f]                                    \n\t"
-    "lwc1     %[tmp7f],       4(%[efw_ptr_0])                             \n\t"
-    "lwc1     %[tmp8f],       8(%[u_ptr])                                 \n\t"
-    "lwc1     %[tmp9f],       4(%[efw_ptr_1])                             \n\t"
-    "lwc1     %[tmp10f],      12(%[u_ptr])                                \n\t"
-#if !defined(MIPS32_R2_LE)
-    "mul.s    %[tmp3f],       %[tmp6f],         %[tmp8f]                  \n\t"
-    "add.s    %[tmp7f],       %[tmp7f],         %[tmp3f]                  \n\t"
-    "mul.s    %[tmp3f],       %[tmp6f],         %[tmp10f]                 \n\t"
-    "add.s    %[tmp9f],       %[tmp9f],         %[tmp3f]                  \n\t"
-#else  // #if !defined(MIPS32_R2_LE)
-    "madd.s   %[tmp7f],       %[tmp7f],         %[tmp6f],     %[tmp8f]    \n\t"
-    "madd.s   %[tmp9f],       %[tmp9f],         %[tmp6f],     %[tmp10f]   \n\t"
-#endif  // #if !defined(MIPS32_R2_LE)
-    "swc1     %[tmp7f],       4(%[efw_ptr_0])                             \n\t"
-    "swc1     %[tmp9f],       4(%[efw_ptr_1])                             \n\t"
-    "5:                                                                   \n\t"
-    "addiu    %[u_ptr],       %[u_ptr],         16                        \n\t"
-    "addiu    %[efw_ptr_0],   %[efw_ptr_0],     8                         \n\t"
-    "bne      %[u_ptr],       %[u_ptr_end],     1b                        \n\t"
-    " addiu   %[efw_ptr_1],   %[efw_ptr_1],     8                         \n\t"
-    ".set     pop                                                         \n\t"
-    : [lambda] "+r" (lambda), [u_ptr] "+r" (u_ptr),
-      [efw_ptr_0] "+r" (efw_ptr_0), [efw_ptr_1] "+r" (efw_ptr_1),
-      [tmp1f] "=&f" (tmp1f), [tmp2f] "=&f" (tmp2f), [tmp3f] "=&f" (tmp3f),
-      [tmp4f] "=&f" (tmp4f), [tmp5f] "=&f" (tmp5f),
-      [tmp6f] "=&f" (tmp6f), [tmp7f] "=&f" (tmp7f), [tmp8f] "=&f" (tmp8f),
-      [tmp9f] "=&f" (tmp9f), [tmp10f] "=&f" (tmp10f)
-    : [tmp1c] "f" (tmp1c), [u_ptr_end] "r" (u_ptr_end)
-    : "memory");
-
-  lambda -= PART_LEN;
-  tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[PART_LEN] * lambda[PART_LEN], 0));
-  // tmp = 1 - lambda[i];
-  efw[0][PART_LEN] += tmp * u[PART_LEN][0];
-  efw[1][PART_LEN] += tmp * u[PART_LEN][1];
-
-  // For H band comfort noise
-  // TODO(peah): don't compute noise and "tmp" twice. Use the previous results.
-  noiseAvg = 0.0;
-  tmpAvg = 0.0;
-  num = 0;
-  if (aec->num_bands > 1) {
-    for (i = 0; i < PART_LEN; i++) {
-      rand[i] = (static_cast<float>(randW16[i])) / 32768;
-    }
-
-    // average noise scale
-    // average over second half of freq spectrum (i.e., 4->8khz)
-    // TODO(peah): we shouldn't need num. We know how many elements we're
-    // summing.
-    for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
-      num++;
-      noiseAvg += sqrtf(noisePow[i]);
-    }
-    noiseAvg /= static_cast<float>(num);
-
-    // average nlp scale
-    // average over second half of freq spectrum (i.e., 4->8khz)
-    // TODO(peah): we shouldn't need num. We know how many elements we're
-    // summing.
-    num = 0;
-    for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
-      num++;
-      tmpAvg += sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
-    }
-    tmpAvg /= static_cast<float>(num);
-
-    // Use average noise for H band
-    // TODO(peah): we should probably have a new random vector here.
-    // Reject LF noise
-    u[0][0] = 0;
-    u[0][1] = 0;
-    for (i = 1; i < PART_LEN1; i++) {
-      tmp = pi2 * rand[i - 1];
-
-      // Use average noise for H band
-      u[i][0] = noiseAvg * static_cast<float>(cos(tmp));
-      u[i][1] = -noiseAvg * static_cast<float>(sin(tmp));
-    }
-    u[PART_LEN][1] = 0;
-
-    for (i = 0; i < PART_LEN1; i++) {
-      // Use average NLP weight for H band
-      comfortNoiseHband[0][i] = tmpAvg * u[i][0];
-      comfortNoiseHband[1][i] = tmpAvg * u[i][1];
-    }
-  } else {
-    memset(comfortNoiseHband, 0,
-           2 * PART_LEN1 * sizeof(comfortNoiseHband[0][0]));
-  }
-}
-
 void WebRtcAec_FilterFar_mips(
     int num_partitions,
     int x_fft_buf_block_pos,
@@ -644,24 +344,18 @@ void WebRtcAec_FilterAdaptation_mips(
   }
 }
 
-void WebRtcAec_OverdriveAndSuppress_mips(AecCore* aec,
-                                         float hNl[PART_LEN1],
-                                         const float hNlFb,
-                                         float efw[2][PART_LEN1]) {
-  int i;
+void WebRtcAec_Overdrive_mips(float overdrive_scaling,
+                              float hNlFb,
+                              float hNl[PART_LEN1]) {
   const float one = 1.0;
   float* p_hNl;
-  float* p_efw0;
-  float* p_efw1;
   const float* p_WebRtcAec_wC;
   float temp1, temp2, temp3, temp4;
 
   p_hNl = &hNl[0];
-  p_efw0 = &efw[0][0];
-  p_efw1 = &efw[1][0];
   p_WebRtcAec_wC = &WebRtcAec_weightCurve[0];
 
-  for (i = 0; i < PART_LEN1; i++) {
+  for (int i = 0; i < PART_LEN1; ++i) {
     // Weight subbands
     __asm __volatile(
       ".set      push                                              \n\t"
@@ -687,8 +381,22 @@ void WebRtcAec_OverdriveAndSuppress_mips(AecCore* aec,
       : [hNlFb] "f" (hNlFb), [one] "f" (one), [p_hNl] "r" (p_hNl)
       : "memory");
 
-    hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
+    hNl[i] = powf(hNl[i], overdrive_scaling * WebRtcAec_overDriveCurve[i]);
+  }
+}
+
+void WebRtcAec_Suppress_mips(const float hNl[PART_LEN1],
+                             float efw[2][PART_LEN1]) {
+  const float* p_hNl;
+  float* p_efw0;
+  float* p_efw1;
+  float temp1, temp2, temp3, temp4;
+
+  p_hNl = &hNl[0];
+  p_efw0 = &efw[0][0];
+  p_efw1 = &efw[1][0];
 
+  for (int i = 0; i < PART_LEN1; ++i) {
     __asm __volatile(
       "lwc1      %[temp1],    0(%[p_hNl])              \n\t"
       "lwc1      %[temp3],    0(%[p_efw1])             \n\t"
@@ -775,7 +483,7 @@ void WebRtcAec_InitAec_mips(void) {
   WebRtcAec_FilterFar = WebRtcAec_FilterFar_mips;
   WebRtcAec_FilterAdaptation = WebRtcAec_FilterAdaptation_mips;
   WebRtcAec_ScaleErrorSignal = WebRtcAec_ScaleErrorSignal_mips;
-  WebRtcAec_ComfortNoise = WebRtcAec_ComfortNoise_mips;
-  WebRtcAec_OverdriveAndSuppress = WebRtcAec_OverdriveAndSuppress_mips;
+  WebRtcAec_Overdrive = WebRtcAec_Overdrive_mips;
+  WebRtcAec_Suppress = WebRtcAec_Suppress_mips;
 }
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_neon.cc b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_neon.cc
index c08ee426e61..bc503ba3db0 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_neon.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_neon.cc
@@ -22,10 +22,8 @@ extern "C" {
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 }
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
-extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
-}
 
 namespace webrtc {
 
@@ -376,15 +374,13 @@ static float32x4_t vpowq_f32(float32x4_t a, float32x4_t b) {
   return a_exp_b;
 }
 
-static void OverdriveAndSuppressNEON(AecCore* aec,
-                                     float hNl[PART_LEN1],
-                                     const float hNlFb,
-                                     float efw[2][PART_LEN1]) {
+static void OverdriveNEON(float overdrive_scaling,
+                          float hNlFb,
+                          float hNl[PART_LEN1]) {
   int i;
   const float32x4_t vec_hNlFb = vmovq_n_f32(hNlFb);
   const float32x4_t vec_one = vdupq_n_f32(1.0f);
-  const float32x4_t vec_minus_one = vdupq_n_f32(-1.0f);
-  const float32x4_t vec_overDriveSm = vmovq_n_f32(aec->overDriveSm);
+  const float32x4_t vec_overdrive_scaling = vmovq_n_f32(overdrive_scaling);
 
   // vectorized code (four at once)
   for (i = 0; i + 3 < PART_LEN1; i += 4) {
@@ -406,28 +402,12 @@ static void OverdriveAndSuppressNEON(AecCore* aec,
 
     vec_hNl = vreinterpretq_f32_u32(vorrq_u32(vec_if0, vec_if1));
 
-    {
-      const float32x4_t vec_overDriveCurve =
-          vld1q_f32(&WebRtcAec_overDriveCurve[i]);
-      const float32x4_t vec_overDriveSm_overDriveCurve =
-          vmulq_f32(vec_overDriveSm, vec_overDriveCurve);
-      vec_hNl = vpowq_f32(vec_hNl, vec_overDriveSm_overDriveCurve);
-      vst1q_f32(&hNl[i], vec_hNl);
-    }
-
-    // Suppress error signal
-    {
-      float32x4_t vec_efw_re = vld1q_f32(&efw[0][i]);
-      float32x4_t vec_efw_im = vld1q_f32(&efw[1][i]);
-      vec_efw_re = vmulq_f32(vec_efw_re, vec_hNl);
-      vec_efw_im = vmulq_f32(vec_efw_im, vec_hNl);
-
-      // Ooura fft returns incorrect sign on imaginary component. It matters
-      // here because we are making an additive change with comfort noise.
-      vec_efw_im = vmulq_f32(vec_efw_im, vec_minus_one);
-      vst1q_f32(&efw[0][i], vec_efw_re);
-      vst1q_f32(&efw[1][i], vec_efw_im);
-    }
+    const float32x4_t vec_overDriveCurve =
+        vld1q_f32(&WebRtcAec_overDriveCurve[i]);
+    const float32x4_t vec_overDriveSm_overDriveCurve =
+        vmulq_f32(vec_overdrive_scaling, vec_overDriveCurve);
+    vec_hNl = vpowq_f32(vec_hNl, vec_overDriveSm_overDriveCurve);
+    vst1q_f32(&hNl[i], vec_hNl);
   }
 
   // scalar code for the remaining items.
@@ -438,9 +418,30 @@ static void OverdriveAndSuppressNEON(AecCore* aec,
                (1 - WebRtcAec_weightCurve[i]) * hNl[i];
     }
 
-    hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
+    hNl[i] = powf(hNl[i], overdrive_scaling * WebRtcAec_overDriveCurve[i]);
+  }
+}
 
-    // Suppress error signal
+static void SuppressNEON(const float hNl[PART_LEN1], float efw[2][PART_LEN1]) {
+  int i;
+  const float32x4_t vec_minus_one = vdupq_n_f32(-1.0f);
+  // vectorized code (four at once)
+  for (i = 0; i + 3 < PART_LEN1; i += 4) {
+    float32x4_t vec_hNl = vld1q_f32(&hNl[i]);
+    float32x4_t vec_efw_re = vld1q_f32(&efw[0][i]);
+    float32x4_t vec_efw_im = vld1q_f32(&efw[1][i]);
+    vec_efw_re = vmulq_f32(vec_efw_re, vec_hNl);
+    vec_efw_im = vmulq_f32(vec_efw_im, vec_hNl);
+
+    // Ooura fft returns incorrect sign on imaginary component. It matters
+    // here because we are making an additive change with comfort noise.
+    vec_efw_im = vmulq_f32(vec_efw_im, vec_minus_one);
+    vst1q_f32(&efw[0][i], vec_efw_re);
+    vst1q_f32(&efw[1][i], vec_efw_im);
+  }
+
+  // scalar code for the remaining items.
+  for (; i < PART_LEN1; i++) {
     efw[0][i] *= hNl[i];
     efw[1][i] *= hNl[i];
 
@@ -450,7 +451,9 @@ static void OverdriveAndSuppressNEON(AecCore* aec,
   }
 }
 
-static int PartitionDelayNEON(const AecCore* aec) {
+static int PartitionDelayNEON(
+    int num_partitions,
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
   // Measures the energy in each filter partition and returns the partition with
   // highest energy.
   // TODO(bjornv): Spread computational cost by computing one partition per
@@ -459,15 +462,15 @@ static int PartitionDelayNEON(const AecCore* aec) {
   int i;
   int delay = 0;
 
-  for (i = 0; i < aec->num_partitions; i++) {
+  for (i = 0; i < num_partitions; i++) {
     int j;
     int pos = i * PART_LEN1;
     float wfEn = 0;
     float32x4_t vec_wfEn = vdupq_n_f32(0.0f);
     // vectorized code (four at once)
     for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const float32x4_t vec_wfBuf0 = vld1q_f32(&aec->wfBuf[0][pos + j]);
-      const float32x4_t vec_wfBuf1 = vld1q_f32(&aec->wfBuf[1][pos + j]);
+      const float32x4_t vec_wfBuf0 = vld1q_f32(&h_fft_buf[0][pos + j]);
+      const float32x4_t vec_wfBuf1 = vld1q_f32(&h_fft_buf[1][pos + j]);
       vec_wfEn = vmlaq_f32(vec_wfEn, vec_wfBuf0, vec_wfBuf0);
       vec_wfEn = vmlaq_f32(vec_wfEn, vec_wfBuf1, vec_wfBuf1);
     }
@@ -483,8 +486,8 @@ static int PartitionDelayNEON(const AecCore* aec) {
 
     // scalar code for the remaining items.
     for (; j < PART_LEN1; j++) {
-      wfEn += aec->wfBuf[0][pos + j] * aec->wfBuf[0][pos + j] +
-              aec->wfBuf[1][pos + j] * aec->wfBuf[1][pos + j];
+      wfEn += h_fft_buf[0][pos + j] * h_fft_buf[0][pos + j] +
+              h_fft_buf[1][pos + j] * h_fft_buf[1][pos + j];
     }
 
     if (wfEn > wfEnMax) {
@@ -504,16 +507,19 @@ static int PartitionDelayNEON(const AecCore* aec) {
 //
 // In addition to updating the PSDs, also the filter diverge state is determined
 // upon actions are taken.
-static void SmoothedPSD(AecCore* aec,
-                        float efw[2][PART_LEN1],
-                        float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1],
-                        int* extreme_filter_divergence) {
+static void UpdateCoherenceSpectraNEON(int mult,
+                                       bool extended_filter_enabled,
+                                       float efw[2][PART_LEN1],
+                                       float dfw[2][PART_LEN1],
+                                       float xfw[2][PART_LEN1],
+                                       CoherenceState* coherence_state,
+                                       short* filter_divergence_state,
+                                       int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh =
-      aec->extended_filter_enabled
-          ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
-          : WebRtcAec_kNormalSmoothingCoefficients[aec->mult - 1];
+      extended_filter_enabled
+          ? WebRtcAec_kExtendedSmoothingCoefficients[mult - 1]
+          : WebRtcAec_kNormalSmoothingCoefficients[mult - 1];
   int i;
   float sdSum = 0, seSum = 0;
   const float32x4_t vec_15 = vdupq_n_f32(WebRtcAec_kMinFarendPSD);
@@ -527,9 +533,12 @@ static void SmoothedPSD(AecCore* aec,
     const float32x4_t vec_efw1 = vld1q_f32(&efw[1][i]);
     const float32x4_t vec_xfw0 = vld1q_f32(&xfw[0][i]);
     const float32x4_t vec_xfw1 = vld1q_f32(&xfw[1][i]);
-    float32x4_t vec_sd = vmulq_n_f32(vld1q_f32(&aec->sd[i]), ptrGCoh[0]);
-    float32x4_t vec_se = vmulq_n_f32(vld1q_f32(&aec->se[i]), ptrGCoh[0]);
-    float32x4_t vec_sx = vmulq_n_f32(vld1q_f32(&aec->sx[i]), ptrGCoh[0]);
+    float32x4_t vec_sd =
+        vmulq_n_f32(vld1q_f32(&coherence_state->sd[i]), ptrGCoh[0]);
+    float32x4_t vec_se =
+        vmulq_n_f32(vld1q_f32(&coherence_state->se[i]), ptrGCoh[0]);
+    float32x4_t vec_sx =
+        vmulq_n_f32(vld1q_f32(&coherence_state->sx[i]), ptrGCoh[0]);
     float32x4_t vec_dfw_sumsq = vmulq_f32(vec_dfw0, vec_dfw0);
     float32x4_t vec_efw_sumsq = vmulq_f32(vec_efw0, vec_efw0);
     float32x4_t vec_xfw_sumsq = vmulq_f32(vec_xfw0, vec_xfw0);
@@ -542,12 +551,12 @@ static void SmoothedPSD(AecCore* aec,
     vec_se = vmlaq_n_f32(vec_se, vec_efw_sumsq, ptrGCoh[1]);
     vec_sx = vmlaq_n_f32(vec_sx, vec_xfw_sumsq, ptrGCoh[1]);
 
-    vst1q_f32(&aec->sd[i], vec_sd);
-    vst1q_f32(&aec->se[i], vec_se);
-    vst1q_f32(&aec->sx[i], vec_sx);
+    vst1q_f32(&coherence_state->sd[i], vec_sd);
+    vst1q_f32(&coherence_state->se[i], vec_se);
+    vst1q_f32(&coherence_state->sx[i], vec_sx);
 
     {
-      float32x4x2_t vec_sde = vld2q_f32(&aec->sde[i][0]);
+      float32x4x2_t vec_sde = vld2q_f32(&coherence_state->sde[i][0]);
       float32x4_t vec_dfwefw0011 = vmulq_f32(vec_dfw0, vec_efw0);
       float32x4_t vec_dfwefw0110 = vmulq_f32(vec_dfw0, vec_efw1);
       vec_sde.val[0] = vmulq_n_f32(vec_sde.val[0], ptrGCoh[0]);
@@ -556,11 +565,11 @@ static void SmoothedPSD(AecCore* aec,
       vec_dfwefw0110 = vmlsq_f32(vec_dfwefw0110, vec_dfw1, vec_efw0);
       vec_sde.val[0] = vmlaq_n_f32(vec_sde.val[0], vec_dfwefw0011, ptrGCoh[1]);
       vec_sde.val[1] = vmlaq_n_f32(vec_sde.val[1], vec_dfwefw0110, ptrGCoh[1]);
-      vst2q_f32(&aec->sde[i][0], vec_sde);
+      vst2q_f32(&coherence_state->sde[i][0], vec_sde);
     }
 
     {
-      float32x4x2_t vec_sxd = vld2q_f32(&aec->sxd[i][0]);
+      float32x4x2_t vec_sxd = vld2q_f32(&coherence_state->sxd[i][0]);
       float32x4_t vec_dfwxfw0011 = vmulq_f32(vec_dfw0, vec_xfw0);
       float32x4_t vec_dfwxfw0110 = vmulq_f32(vec_dfw0, vec_xfw1);
       vec_sxd.val[0] = vmulq_n_f32(vec_sxd.val[0], ptrGCoh[0]);
@@ -569,7 +578,7 @@ static void SmoothedPSD(AecCore* aec,
       vec_dfwxfw0110 = vmlsq_f32(vec_dfwxfw0110, vec_dfw1, vec_xfw0);
       vec_sxd.val[0] = vmlaq_n_f32(vec_sxd.val[0], vec_dfwxfw0011, ptrGCoh[1]);
       vec_sxd.val[1] = vmlaq_n_f32(vec_sxd.val[1], vec_dfwxfw0110, ptrGCoh[1]);
-      vst2q_f32(&aec->sxd[i][0], vec_sxd);
+      vst2q_f32(&coherence_state->sxd[i][0], vec_sxd);
     }
 
     vec_sdSum = vaddq_f32(vec_sdSum, vec_sd);
@@ -593,39 +602,43 @@ static void SmoothedPSD(AecCore* aec,
 
   // scalar code for the remaining items.
   for (; i < PART_LEN1; i++) {
-    aec->sd[i] = ptrGCoh[0] * aec->sd[i] +
-                 ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
-    aec->se[i] = ptrGCoh[0] * aec->se[i] +
-                 ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
+    coherence_state->sd[i] =
+        ptrGCoh[0] * coherence_state->sd[i] +
+        ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
+    coherence_state->se[i] =
+        ptrGCoh[0] * coherence_state->se[i] +
+        ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
     // We threshold here to protect against the ill-effects of a zero farend.
     // The threshold is not arbitrarily chosen, but balances protection and
     // adverse interaction with the algorithm's tuning.
     // TODO(bjornv): investigate further why this is so sensitive.
-    aec->sx[i] = ptrGCoh[0] * aec->sx[i] +
-                 ptrGCoh[1] * WEBRTC_SPL_MAX(
-                                  xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
-                                  WebRtcAec_kMinFarendPSD);
-
-    aec->sde[i][0] =
-        ptrGCoh[0] * aec->sde[i][0] +
+    coherence_state->sx[i] =
+        ptrGCoh[0] * coherence_state->sx[i] +
+        ptrGCoh[1] *
+            WEBRTC_SPL_MAX(xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
+                           WebRtcAec_kMinFarendPSD);
+
+    coherence_state->sde[i][0] =
+        ptrGCoh[0] * coherence_state->sde[i][0] +
         ptrGCoh[1] * (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
-    aec->sde[i][1] =
-        ptrGCoh[0] * aec->sde[i][1] +
+    coherence_state->sde[i][1] =
+        ptrGCoh[0] * coherence_state->sde[i][1] +
         ptrGCoh[1] * (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
 
-    aec->sxd[i][0] =
-        ptrGCoh[0] * aec->sxd[i][0] +
+    coherence_state->sxd[i][0] =
+        ptrGCoh[0] * coherence_state->sxd[i][0] +
         ptrGCoh[1] * (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
-    aec->sxd[i][1] =
-        ptrGCoh[0] * aec->sxd[i][1] +
+    coherence_state->sxd[i][1] =
+        ptrGCoh[0] * coherence_state->sxd[i][1] +
         ptrGCoh[1] * (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
 
-    sdSum += aec->sd[i];
-    seSum += aec->se[i];
+    sdSum += coherence_state->sd[i];
+    seSum += coherence_state->se[i];
   }
 
   // Divergent filter safeguard update.
-  aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
+  *filter_divergence_state =
+      (*filter_divergence_state ? 1.05f : 1.0f) * seSum > sdSum;
 
   // Signal extreme filter divergence if the error is significantly larger
   // than the nearend (13 dB).
@@ -669,30 +682,23 @@ static void StoreAsComplexNEON(const float* data,
   data_complex[0][PART_LEN] = data[1];
 }
 
-static void SubbandCoherenceNEON(AecCore* aec,
-                                 float efw[2][PART_LEN1],
-                                 float dfw[2][PART_LEN1],
-                                 float xfw[2][PART_LEN1],
-                                 float* fft,
+static void ComputeCoherenceNEON(const CoherenceState* coherence_state,
                                  float* cohde,
-                                 float* cohxd,
-                                 int* extreme_filter_divergence) {
+                                 float* cohxd) {
   int i;
 
-  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
-
   {
     const float32x4_t vec_1eminus10 = vdupq_n_f32(1e-10f);
 
     // Subband coherence
     for (i = 0; i + 3 < PART_LEN1; i += 4) {
-      const float32x4_t vec_sd = vld1q_f32(&aec->sd[i]);
-      const float32x4_t vec_se = vld1q_f32(&aec->se[i]);
-      const float32x4_t vec_sx = vld1q_f32(&aec->sx[i]);
+      const float32x4_t vec_sd = vld1q_f32(&coherence_state->sd[i]);
+      const float32x4_t vec_se = vld1q_f32(&coherence_state->se[i]);
+      const float32x4_t vec_sx = vld1q_f32(&coherence_state->sx[i]);
       const float32x4_t vec_sdse = vmlaq_f32(vec_1eminus10, vec_sd, vec_se);
       const float32x4_t vec_sdsx = vmlaq_f32(vec_1eminus10, vec_sd, vec_sx);
-      float32x4x2_t vec_sde = vld2q_f32(&aec->sde[i][0]);
-      float32x4x2_t vec_sxd = vld2q_f32(&aec->sxd[i][0]);
+      float32x4x2_t vec_sde = vld2q_f32(&coherence_state->sde[i][0]);
+      float32x4x2_t vec_sxd = vld2q_f32(&coherence_state->sxd[i][0]);
       float32x4_t vec_cohde = vmulq_f32(vec_sde.val[0], vec_sde.val[0]);
       float32x4_t vec_cohxd = vmulq_f32(vec_sxd.val[0], vec_sxd.val[0]);
       vec_cohde = vmlaq_f32(vec_cohde, vec_sde.val[1], vec_sde.val[1]);
@@ -706,12 +712,12 @@ static void SubbandCoherenceNEON(AecCore* aec,
   }
   // scalar code for the remaining items.
   for (; i < PART_LEN1; i++) {
-    cohde[i] =
-        (aec->sde[i][0] * aec->sde[i][0] + aec->sde[i][1] * aec->sde[i][1]) /
-        (aec->sd[i] * aec->se[i] + 1e-10f);
-    cohxd[i] =
-        (aec->sxd[i][0] * aec->sxd[i][0] + aec->sxd[i][1] * aec->sxd[i][1]) /
-        (aec->sx[i] * aec->sd[i] + 1e-10f);
+    cohde[i] = (coherence_state->sde[i][0] * coherence_state->sde[i][0] +
+                coherence_state->sde[i][1] * coherence_state->sde[i][1]) /
+               (coherence_state->sd[i] * coherence_state->se[i] + 1e-10f);
+    cohxd[i] = (coherence_state->sxd[i][0] * coherence_state->sxd[i][0] +
+                coherence_state->sxd[i][1] * coherence_state->sxd[i][1]) /
+               (coherence_state->sx[i] * coherence_state->sd[i] + 1e-10f);
   }
 }
 
@@ -719,8 +725,10 @@ void WebRtcAec_InitAec_neon(void) {
   WebRtcAec_FilterFar = FilterFarNEON;
   WebRtcAec_ScaleErrorSignal = ScaleErrorSignalNEON;
   WebRtcAec_FilterAdaptation = FilterAdaptationNEON;
-  WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressNEON;
-  WebRtcAec_SubbandCoherence = SubbandCoherenceNEON;
+  WebRtcAec_Overdrive = OverdriveNEON;
+  WebRtcAec_Suppress = SuppressNEON;
+  WebRtcAec_ComputeCoherence = ComputeCoherenceNEON;
+  WebRtcAec_UpdateCoherenceSpectra = UpdateCoherenceSpectraNEON;
   WebRtcAec_StoreAsComplex = StoreAsComplexNEON;
   WebRtcAec_PartitionDelay = PartitionDelayNEON;
   WebRtcAec_WindowData = WindowDataNEON;
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
new file mode 100644
index 00000000000..d1fb6e892a7
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_OPTIMIZED_METHODS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_OPTIMIZED_METHODS_H_
+
+#include <memory>
+
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+typedef void (*WebRtcAecFilterFar)(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float y_fft[2][PART_LEN1]);
+extern WebRtcAecFilterFar WebRtcAec_FilterFar;
+typedef void (*WebRtcAecScaleErrorSignal)(float mu,
+                                          float error_threshold,
+                                          float x_pow[PART_LEN1],
+                                          float ef[2][PART_LEN1]);
+extern WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
+typedef void (*WebRtcAecFilterAdaptation)(
+    int num_partitions,
+    int x_fft_buf_block_pos,
+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+    float e_fft[2][PART_LEN1],
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
+extern WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
+
+typedef void (*WebRtcAecOverdrive)(float overdrive_scaling,
+                                   const float hNlFb,
+                                   float hNl[PART_LEN1]);
+extern WebRtcAecOverdrive WebRtcAec_Overdrive;
+
+typedef void (*WebRtcAecSuppress)(const float hNl[PART_LEN1],
+                                  float efw[2][PART_LEN1]);
+extern WebRtcAecSuppress WebRtcAec_Suppress;
+
+typedef void (*WebRtcAecComputeCoherence)(const CoherenceState* coherence_state,
+                                          float* cohde,
+                                          float* cohxd);
+extern WebRtcAecComputeCoherence WebRtcAec_ComputeCoherence;
+
+typedef void (*WebRtcAecUpdateCoherenceSpectra)(int mult,
+                                                bool extended_filter_enabled,
+                                                float efw[2][PART_LEN1],
+                                                float dfw[2][PART_LEN1],
+                                                float xfw[2][PART_LEN1],
+                                                CoherenceState* coherence_state,
+                                                short* filter_divergence_state,
+                                                int* extreme_filter_divergence);
+extern WebRtcAecUpdateCoherenceSpectra WebRtcAec_UpdateCoherenceSpectra;
+
+typedef int (*WebRtcAecPartitionDelay)(
+    int num_partitions,
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
+extern WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
+
+typedef void (*WebRtcAecStoreAsComplex)(const float* data,
+                                        float data_complex[2][PART_LEN1]);
+extern WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
+
+typedef void (*WebRtcAecWindowData)(float* x_windowed, const float* x);
+extern WebRtcAecWindowData WebRtcAec_WindowData;
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_OPTIMIZED_METHODS_H_
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_sse2.cc b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
index c1a6e3de75b..47ba12f419e 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
@@ -20,10 +20,8 @@ extern "C" {
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 }
 #include "webrtc/modules/audio_processing/aec/aec_common.h"
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
-extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
-}
 
 namespace webrtc {
 
@@ -377,15 +375,13 @@ static __m128 mm_pow_ps(__m128 a, __m128 b) {
   return a_exp_b;
 }
 
-static void OverdriveAndSuppressSSE2(AecCore* aec,
-                                     float hNl[PART_LEN1],
-                                     const float hNlFb,
-                                     float efw[2][PART_LEN1]) {
+static void OverdriveSSE2(float overdrive_scaling,
+                          float hNlFb,
+                          float hNl[PART_LEN1]) {
   int i;
   const __m128 vec_hNlFb = _mm_set1_ps(hNlFb);
   const __m128 vec_one = _mm_set1_ps(1.0f);
-  const __m128 vec_minus_one = _mm_set1_ps(-1.0f);
-  const __m128 vec_overDriveSm = _mm_set1_ps(aec->overDriveSm);
+  const __m128 vec_overdrive_scaling = _mm_set1_ps(overdrive_scaling);
   // vectorized code (four at once)
   for (i = 0; i + 3 < PART_LEN1; i += 4) {
     // Weight subbands
@@ -401,28 +397,12 @@ static void OverdriveAndSuppressSSE2(AecCore* aec,
         bigger, _mm_add_ps(vec_weightCurve_hNlFb, vec_one_weightCurve_hNl));
     vec_hNl = _mm_or_ps(vec_if0, vec_if1);
 
-    {
-      const __m128 vec_overDriveCurve =
-          _mm_loadu_ps(&WebRtcAec_overDriveCurve[i]);
-      const __m128 vec_overDriveSm_overDriveCurve =
-          _mm_mul_ps(vec_overDriveSm, vec_overDriveCurve);
-      vec_hNl = mm_pow_ps(vec_hNl, vec_overDriveSm_overDriveCurve);
-      _mm_storeu_ps(&hNl[i], vec_hNl);
-    }
-
-    // Suppress error signal
-    {
-      __m128 vec_efw_re = _mm_loadu_ps(&efw[0][i]);
-      __m128 vec_efw_im = _mm_loadu_ps(&efw[1][i]);
-      vec_efw_re = _mm_mul_ps(vec_efw_re, vec_hNl);
-      vec_efw_im = _mm_mul_ps(vec_efw_im, vec_hNl);
-
-      // Ooura fft returns incorrect sign on imaginary component. It matters
-      // here because we are making an additive change with comfort noise.
-      vec_efw_im = _mm_mul_ps(vec_efw_im, vec_minus_one);
-      _mm_storeu_ps(&efw[0][i], vec_efw_re);
-      _mm_storeu_ps(&efw[1][i], vec_efw_im);
-    }
+    const __m128 vec_overDriveCurve =
+        _mm_loadu_ps(&WebRtcAec_overDriveCurve[i]);
+    const __m128 vec_overDriveSm_overDriveCurve =
+        _mm_mul_ps(vec_overdrive_scaling, vec_overDriveCurve);
+    vec_hNl = mm_pow_ps(vec_hNl, vec_overDriveSm_overDriveCurve);
+    _mm_storeu_ps(&hNl[i], vec_hNl);
   }
   // scalar code for the remaining items.
   for (; i < PART_LEN1; i++) {
@@ -431,8 +411,30 @@ static void OverdriveAndSuppressSSE2(AecCore* aec,
       hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
                (1 - WebRtcAec_weightCurve[i]) * hNl[i];
     }
-    hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
+    hNl[i] = powf(hNl[i], overdrive_scaling * WebRtcAec_overDriveCurve[i]);
+  }
+}
+
+static void SuppressSSE2(const float hNl[PART_LEN1], float efw[2][PART_LEN1]) {
+  int i;
+  const __m128 vec_minus_one = _mm_set1_ps(-1.0f);
+  // vectorized code (four at once)
+  for (i = 0; i + 3 < PART_LEN1; i += 4) {
+    // Suppress error signal
+    __m128 vec_hNl = _mm_loadu_ps(&hNl[i]);
+    __m128 vec_efw_re = _mm_loadu_ps(&efw[0][i]);
+    __m128 vec_efw_im = _mm_loadu_ps(&efw[1][i]);
+    vec_efw_re = _mm_mul_ps(vec_efw_re, vec_hNl);
+    vec_efw_im = _mm_mul_ps(vec_efw_im, vec_hNl);
 
+    // Ooura fft returns incorrect sign on imaginary component. It matters
+    // here because we are making an additive change with comfort noise.
+    vec_efw_im = _mm_mul_ps(vec_efw_im, vec_minus_one);
+    _mm_storeu_ps(&efw[0][i], vec_efw_re);
+    _mm_storeu_ps(&efw[1][i], vec_efw_im);
+  }
+  // scalar code for the remaining items.
+  for (; i < PART_LEN1; i++) {
     // Suppress error signal
     efw[0][i] *= hNl[i];
     efw[1][i] *= hNl[i];
@@ -451,7 +453,9 @@ __inline static void _mm_add_ps_4x1(__m128 sum, float* dst) {
   _mm_store_ss(dst, sum);
 }
 
-static int PartitionDelaySSE2(const AecCore* aec) {
+static int PartitionDelaySSE2(
+    int num_partitions,
+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
   // Measures the energy in each filter partition and returns the partition with
   // highest energy.
   // TODO(bjornv): Spread computational cost by computing one partition per
@@ -460,15 +464,15 @@ static int PartitionDelaySSE2(const AecCore* aec) {
   int i;
   int delay = 0;
 
-  for (i = 0; i < aec->num_partitions; i++) {
+  for (i = 0; i < num_partitions; i++) {
     int j;
     int pos = i * PART_LEN1;
     float wfEn = 0;
     __m128 vec_wfEn = _mm_set1_ps(0.0f);
     // vectorized code (four at once)
     for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const __m128 vec_wfBuf0 = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-      const __m128 vec_wfBuf1 = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
+      const __m128 vec_wfBuf0 = _mm_loadu_ps(&h_fft_buf[0][pos + j]);
+      const __m128 vec_wfBuf1 = _mm_loadu_ps(&h_fft_buf[1][pos + j]);
       vec_wfEn = _mm_add_ps(vec_wfEn, _mm_mul_ps(vec_wfBuf0, vec_wfBuf0));
       vec_wfEn = _mm_add_ps(vec_wfEn, _mm_mul_ps(vec_wfBuf1, vec_wfBuf1));
     }
@@ -476,8 +480,8 @@ static int PartitionDelaySSE2(const AecCore* aec) {
 
     // scalar code for the remaining items.
     for (; j < PART_LEN1; j++) {
-      wfEn += aec->wfBuf[0][pos + j] * aec->wfBuf[0][pos + j] +
-              aec->wfBuf[1][pos + j] * aec->wfBuf[1][pos + j];
+      wfEn += h_fft_buf[0][pos + j] * h_fft_buf[0][pos + j] +
+              h_fft_buf[1][pos + j] * h_fft_buf[1][pos + j];
     }
 
     if (wfEn > wfEnMax) {
@@ -497,16 +501,19 @@ static int PartitionDelaySSE2(const AecCore* aec) {
 //
 // In addition to updating the PSDs, also the filter diverge state is determined
 // upon actions are taken.
-static void SmoothedPSD(AecCore* aec,
-                        float efw[2][PART_LEN1],
-                        float dfw[2][PART_LEN1],
-                        float xfw[2][PART_LEN1],
-                        int* extreme_filter_divergence) {
+static void UpdateCoherenceSpectraSSE2(int mult,
+                                       bool extended_filter_enabled,
+                                       float efw[2][PART_LEN1],
+                                       float dfw[2][PART_LEN1],
+                                       float xfw[2][PART_LEN1],
+                                       CoherenceState* coherence_state,
+                                       short* filter_divergence_state,
+                                       int* extreme_filter_divergence) {
   // Power estimate smoothing coefficients.
   const float* ptrGCoh =
-      aec->extended_filter_enabled
-          ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
-          : WebRtcAec_kNormalSmoothingCoefficients[aec->mult - 1];
+      extended_filter_enabled
+          ? WebRtcAec_kExtendedSmoothingCoefficients[mult - 1]
+          : WebRtcAec_kNormalSmoothingCoefficients[mult - 1];
   int i;
   float sdSum = 0, seSum = 0;
   const __m128 vec_15 = _mm_set1_ps(WebRtcAec_kMinFarendPSD);
@@ -522,9 +529,12 @@ static void SmoothedPSD(AecCore* aec,
     const __m128 vec_efw1 = _mm_loadu_ps(&efw[1][i]);
     const __m128 vec_xfw0 = _mm_loadu_ps(&xfw[0][i]);
     const __m128 vec_xfw1 = _mm_loadu_ps(&xfw[1][i]);
-    __m128 vec_sd = _mm_mul_ps(_mm_loadu_ps(&aec->sd[i]), vec_GCoh0);
-    __m128 vec_se = _mm_mul_ps(_mm_loadu_ps(&aec->se[i]), vec_GCoh0);
-    __m128 vec_sx = _mm_mul_ps(_mm_loadu_ps(&aec->sx[i]), vec_GCoh0);
+    __m128 vec_sd =
+        _mm_mul_ps(_mm_loadu_ps(&coherence_state->sd[i]), vec_GCoh0);
+    __m128 vec_se =
+        _mm_mul_ps(_mm_loadu_ps(&coherence_state->se[i]), vec_GCoh0);
+    __m128 vec_sx =
+        _mm_mul_ps(_mm_loadu_ps(&coherence_state->sx[i]), vec_GCoh0);
     __m128 vec_dfw_sumsq = _mm_mul_ps(vec_dfw0, vec_dfw0);
     __m128 vec_efw_sumsq = _mm_mul_ps(vec_efw0, vec_efw0);
     __m128 vec_xfw_sumsq = _mm_mul_ps(vec_xfw0, vec_xfw0);
@@ -535,13 +545,13 @@ static void SmoothedPSD(AecCore* aec,
     vec_sd = _mm_add_ps(vec_sd, _mm_mul_ps(vec_dfw_sumsq, vec_GCoh1));
     vec_se = _mm_add_ps(vec_se, _mm_mul_ps(vec_efw_sumsq, vec_GCoh1));
     vec_sx = _mm_add_ps(vec_sx, _mm_mul_ps(vec_xfw_sumsq, vec_GCoh1));
-    _mm_storeu_ps(&aec->sd[i], vec_sd);
-    _mm_storeu_ps(&aec->se[i], vec_se);
-    _mm_storeu_ps(&aec->sx[i], vec_sx);
+    _mm_storeu_ps(&coherence_state->sd[i], vec_sd);
+    _mm_storeu_ps(&coherence_state->se[i], vec_se);
+    _mm_storeu_ps(&coherence_state->sx[i], vec_sx);
 
     {
-      const __m128 vec_3210 = _mm_loadu_ps(&aec->sde[i][0]);
-      const __m128 vec_7654 = _mm_loadu_ps(&aec->sde[i + 2][0]);
+      const __m128 vec_3210 = _mm_loadu_ps(&coherence_state->sde[i][0]);
+      const __m128 vec_7654 = _mm_loadu_ps(&coherence_state->sde[i + 2][0]);
       __m128 vec_a =
           _mm_shuffle_ps(vec_3210, vec_7654, _MM_SHUFFLE(2, 0, 2, 0));
       __m128 vec_b =
@@ -556,13 +566,14 @@ static void SmoothedPSD(AecCore* aec,
           _mm_sub_ps(vec_dfwefw0110, _mm_mul_ps(vec_dfw1, vec_efw0));
       vec_a = _mm_add_ps(vec_a, _mm_mul_ps(vec_dfwefw0011, vec_GCoh1));
       vec_b = _mm_add_ps(vec_b, _mm_mul_ps(vec_dfwefw0110, vec_GCoh1));
-      _mm_storeu_ps(&aec->sde[i][0], _mm_unpacklo_ps(vec_a, vec_b));
-      _mm_storeu_ps(&aec->sde[i + 2][0], _mm_unpackhi_ps(vec_a, vec_b));
+      _mm_storeu_ps(&coherence_state->sde[i][0], _mm_unpacklo_ps(vec_a, vec_b));
+      _mm_storeu_ps(&coherence_state->sde[i + 2][0],
+                    _mm_unpackhi_ps(vec_a, vec_b));
     }
 
     {
-      const __m128 vec_3210 = _mm_loadu_ps(&aec->sxd[i][0]);
-      const __m128 vec_7654 = _mm_loadu_ps(&aec->sxd[i + 2][0]);
+      const __m128 vec_3210 = _mm_loadu_ps(&coherence_state->sxd[i][0]);
+      const __m128 vec_7654 = _mm_loadu_ps(&coherence_state->sxd[i + 2][0]);
       __m128 vec_a =
           _mm_shuffle_ps(vec_3210, vec_7654, _MM_SHUFFLE(2, 0, 2, 0));
       __m128 vec_b =
@@ -577,8 +588,9 @@ static void SmoothedPSD(AecCore* aec,
           _mm_sub_ps(vec_dfwxfw0110, _mm_mul_ps(vec_dfw1, vec_xfw0));
       vec_a = _mm_add_ps(vec_a, _mm_mul_ps(vec_dfwxfw0011, vec_GCoh1));
       vec_b = _mm_add_ps(vec_b, _mm_mul_ps(vec_dfwxfw0110, vec_GCoh1));
-      _mm_storeu_ps(&aec->sxd[i][0], _mm_unpacklo_ps(vec_a, vec_b));
-      _mm_storeu_ps(&aec->sxd[i + 2][0], _mm_unpackhi_ps(vec_a, vec_b));
+      _mm_storeu_ps(&coherence_state->sxd[i][0], _mm_unpacklo_ps(vec_a, vec_b));
+      _mm_storeu_ps(&coherence_state->sxd[i + 2][0],
+                    _mm_unpackhi_ps(vec_a, vec_b));
     }
 
     vec_sdSum = _mm_add_ps(vec_sdSum, vec_sd);
@@ -589,39 +601,43 @@ static void SmoothedPSD(AecCore* aec,
   _mm_add_ps_4x1(vec_seSum, &seSum);
 
   for (; i < PART_LEN1; i++) {
-    aec->sd[i] = ptrGCoh[0] * aec->sd[i] +
-                 ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
-    aec->se[i] = ptrGCoh[0] * aec->se[i] +
-                 ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
+    coherence_state->sd[i] =
+        ptrGCoh[0] * coherence_state->sd[i] +
+        ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
+    coherence_state->se[i] =
+        ptrGCoh[0] * coherence_state->se[i] +
+        ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
     // We threshold here to protect against the ill-effects of a zero farend.
     // The threshold is not arbitrarily chosen, but balances protection and
     // adverse interaction with the algorithm's tuning.
     // TODO(bjornv): investigate further why this is so sensitive.
-    aec->sx[i] = ptrGCoh[0] * aec->sx[i] +
-                 ptrGCoh[1] * WEBRTC_SPL_MAX(
-                                  xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
-                                  WebRtcAec_kMinFarendPSD);
-
-    aec->sde[i][0] =
-        ptrGCoh[0] * aec->sde[i][0] +
+    coherence_state->sx[i] =
+        ptrGCoh[0] * coherence_state->sx[i] +
+        ptrGCoh[1] *
+            WEBRTC_SPL_MAX(xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
+                           WebRtcAec_kMinFarendPSD);
+
+    coherence_state->sde[i][0] =
+        ptrGCoh[0] * coherence_state->sde[i][0] +
         ptrGCoh[1] * (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
-    aec->sde[i][1] =
-        ptrGCoh[0] * aec->sde[i][1] +
+    coherence_state->sde[i][1] =
+        ptrGCoh[0] * coherence_state->sde[i][1] +
         ptrGCoh[1] * (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
 
-    aec->sxd[i][0] =
-        ptrGCoh[0] * aec->sxd[i][0] +
+    coherence_state->sxd[i][0] =
+        ptrGCoh[0] * coherence_state->sxd[i][0] +
         ptrGCoh[1] * (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
-    aec->sxd[i][1] =
-        ptrGCoh[0] * aec->sxd[i][1] +
+    coherence_state->sxd[i][1] =
+        ptrGCoh[0] * coherence_state->sxd[i][1] +
         ptrGCoh[1] * (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
 
-    sdSum += aec->sd[i];
-    seSum += aec->se[i];
+    sdSum += coherence_state->sd[i];
+    seSum += coherence_state->se[i];
   }
 
   // Divergent filter safeguard update.
-  aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
+  *filter_divergence_state =
+      (*filter_divergence_state ? 1.05f : 1.0f) * seSum > sdSum;
 
   // Signal extreme filter divergence if the error is significantly larger
   // than the nearend (13 dB).
@@ -668,34 +684,27 @@ static void StoreAsComplexSSE2(const float* data,
   data_complex[0][PART_LEN] = data[1];
 }
 
-static void SubbandCoherenceSSE2(AecCore* aec,
-                                 float efw[2][PART_LEN1],
-                                 float dfw[2][PART_LEN1],
-                                 float xfw[2][PART_LEN1],
-                                 float* fft,
+static void ComputeCoherenceSSE2(const CoherenceState* coherence_state,
                                  float* cohde,
-                                 float* cohxd,
-                                 int* extreme_filter_divergence) {
+                                 float* cohxd) {
   int i;
 
-  SmoothedPSD(aec, efw, dfw, xfw, extreme_filter_divergence);
-
   {
     const __m128 vec_1eminus10 = _mm_set1_ps(1e-10f);
 
     // Subband coherence
     for (i = 0; i + 3 < PART_LEN1; i += 4) {
-      const __m128 vec_sd = _mm_loadu_ps(&aec->sd[i]);
-      const __m128 vec_se = _mm_loadu_ps(&aec->se[i]);
-      const __m128 vec_sx = _mm_loadu_ps(&aec->sx[i]);
+      const __m128 vec_sd = _mm_loadu_ps(&coherence_state->sd[i]);
+      const __m128 vec_se = _mm_loadu_ps(&coherence_state->se[i]);
+      const __m128 vec_sx = _mm_loadu_ps(&coherence_state->sx[i]);
       const __m128 vec_sdse =
           _mm_add_ps(vec_1eminus10, _mm_mul_ps(vec_sd, vec_se));
       const __m128 vec_sdsx =
           _mm_add_ps(vec_1eminus10, _mm_mul_ps(vec_sd, vec_sx));
-      const __m128 vec_sde_3210 = _mm_loadu_ps(&aec->sde[i][0]);
-      const __m128 vec_sde_7654 = _mm_loadu_ps(&aec->sde[i + 2][0]);
-      const __m128 vec_sxd_3210 = _mm_loadu_ps(&aec->sxd[i][0]);
-      const __m128 vec_sxd_7654 = _mm_loadu_ps(&aec->sxd[i + 2][0]);
+      const __m128 vec_sde_3210 = _mm_loadu_ps(&coherence_state->sde[i][0]);
+      const __m128 vec_sde_7654 = _mm_loadu_ps(&coherence_state->sde[i + 2][0]);
+      const __m128 vec_sxd_3210 = _mm_loadu_ps(&coherence_state->sxd[i][0]);
+      const __m128 vec_sxd_7654 = _mm_loadu_ps(&coherence_state->sxd[i + 2][0]);
       const __m128 vec_sde_0 =
           _mm_shuffle_ps(vec_sde_3210, vec_sde_7654, _MM_SHUFFLE(2, 0, 2, 0));
       const __m128 vec_sde_1 =
@@ -716,12 +725,12 @@ static void SubbandCoherenceSSE2(AecCore* aec,
 
     // scalar code for the remaining items.
     for (; i < PART_LEN1; i++) {
-      cohde[i] =
-          (aec->sde[i][0] * aec->sde[i][0] + aec->sde[i][1] * aec->sde[i][1]) /
-          (aec->sd[i] * aec->se[i] + 1e-10f);
-      cohxd[i] =
-          (aec->sxd[i][0] * aec->sxd[i][0] + aec->sxd[i][1] * aec->sxd[i][1]) /
-          (aec->sx[i] * aec->sd[i] + 1e-10f);
+      cohde[i] = (coherence_state->sde[i][0] * coherence_state->sde[i][0] +
+                  coherence_state->sde[i][1] * coherence_state->sde[i][1]) /
+                 (coherence_state->sd[i] * coherence_state->se[i] + 1e-10f);
+      cohxd[i] = (coherence_state->sxd[i][0] * coherence_state->sxd[i][0] +
+                  coherence_state->sxd[i][1] * coherence_state->sxd[i][1]) /
+                 (coherence_state->sx[i] * coherence_state->sd[i] + 1e-10f);
     }
   }
 }
@@ -730,8 +739,10 @@ void WebRtcAec_InitAec_SSE2(void) {
   WebRtcAec_FilterFar = FilterFarSSE2;
   WebRtcAec_ScaleErrorSignal = ScaleErrorSignalSSE2;
   WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
-  WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
-  WebRtcAec_SubbandCoherence = SubbandCoherenceSSE2;
+  WebRtcAec_Overdrive = OverdriveSSE2;
+  WebRtcAec_Suppress = SuppressSSE2;
+  WebRtcAec_ComputeCoherence = ComputeCoherenceSSE2;
+  WebRtcAec_UpdateCoherenceSpectra = UpdateCoherenceSpectraSSE2;
   WebRtcAec_StoreAsComplex = StoreAsComplexSSE2;
   WebRtcAec_PartitionDelay = PartitionDelaySSE2;
   WebRtcAec_WindowData = WindowDataSSE2;
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.cc
index 03efc103ea6..690fe9f34f9 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.cc
@@ -581,9 +581,5 @@ void aec_rdft_init(void) {
 #endif
 #if defined(WEBRTC_HAS_NEON)
   aec_rdft_init_neon();
-#elif defined(WEBRTC_DETECT_NEON)
-  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
-    aec_rdft_init_neon();
-  }
 #endif
 }
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h
index 18eb7a5c3f3..d83eb27d83f 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft.h
@@ -54,7 +54,7 @@ void aec_rdft_inverse_128(float* a);
 #if defined(MIPS_FPU_LE)
 void aec_rdft_init_mips(void);
 #endif
-#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 void aec_rdft_init_neon(void);
 #endif
 
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_mips.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_mips.cc
index 7e64e657167..7e64e657167 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_mips.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_mips.cc
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_neon.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_neon.cc
index 43b6a68cd72..43b6a68cd72 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_neon.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_neon.cc
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_sse2.cc
index b4e453ff53e..b4e453ff53e 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/aec_rdft_sse2.cc
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.cc b/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.cc
index f963a4e1d30..716da38ab88 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.cc
@@ -14,9 +14,6 @@
 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 
 #include <math.h>
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-#include <stdio.h>
-#endif
 #include <stdlib.h>
 #include <string.h>
 
@@ -26,7 +23,7 @@ extern "C" {
 }
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/modules/audio_processing/aec/aec_resampler.h"
-#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -99,9 +96,7 @@ static const int kMaxBufSizeStart = 62;  // In partitions
 static const int sampMsNb = 8;           // samples per ms in nb
 static const int initCheck = 42;
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-int webrtc_aec_instance_count = 0;
-#endif
+int Aec::instance_count = 0;
 
 // Estimates delay to set the position of the far-end buffer read pointer
 // (controlled by knownDelay)
@@ -123,13 +118,14 @@ static void ProcessExtended(Aec* self,
                             int32_t skew);
 
 void* WebRtcAec_Create() {
-  Aec* aecpc = reinterpret_cast<Aec*>(malloc(sizeof(Aec)));
+  Aec* aecpc = new Aec();
 
   if (!aecpc) {
     return NULL;
   }
+  aecpc->data_dumper.reset(new ApmDataDumper(aecpc->instance_count));
 
-  aecpc->aec = WebRtcAec_CreateAec();
+  aecpc->aec = WebRtcAec_CreateAec(aecpc->instance_count);
   if (!aecpc->aec) {
     WebRtcAec_Free(aecpc);
     return NULL;
@@ -151,22 +147,7 @@ void* WebRtcAec_Create() {
 
   aecpc->initFlag = 0;
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  {
-    char filename[64];
-    snprintf(filename, sizeof(filename), "aec_buf%d.dat",
-             webrtc_aec_instance_count);
-    aecpc->bufFile = fopen(filename, "wb");
-    snprintf(filename, sizeof(filename), "aec_skew%d.dat",
-             webrtc_aec_instance_count);
-    aecpc->skewFile = fopen(filename, "wb");
-    snprintf(filename, sizeof(filename), "aec_delay%d.dat",
-             webrtc_aec_instance_count);
-    aecpc->delayFile = fopen(filename, "wb");
-    webrtc_aec_instance_count++;
-  }
-#endif
-
+  aecpc->instance_count++;
   return aecpc;
 }
 
@@ -179,19 +160,14 @@ void WebRtcAec_Free(void* aecInst) {
 
   WebRtc_FreeBuffer(aecpc->far_pre_buf);
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  fclose(aecpc->bufFile);
-  fclose(aecpc->skewFile);
-  fclose(aecpc->delayFile);
-#endif
-
   WebRtcAec_FreeAec(aecpc->aec);
   WebRtcAec_FreeResampler(aecpc->resampler);
-  free(aecpc);
+  delete aecpc;
 }
 
 int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) {
   Aec* aecpc = reinterpret_cast<Aec*>(aecInst);
+  aecpc->data_dumper->InitiateNewSetOfRecordings();
   AecConfig aecConfig;
 
   if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000 &&
@@ -376,15 +352,9 @@ int32_t WebRtcAec_Process(void* aecInst,
                            msInSndCardBuf, skew);
   }
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  {
-    int16_t far_buf_size_ms = (int16_t)(WebRtcAec_system_delay(aecpc->aec) /
-                                        (sampMsNb * aecpc->rate_factor));
-    (void)fwrite(&far_buf_size_ms, 2, 1, aecpc->bufFile);
-    (void)fwrite(&aecpc->knownDelay, sizeof(aecpc->knownDelay), 1,
-                 aecpc->delayFile);
-  }
-#endif
+  int far_buf_size_samples = WebRtcAec_system_delay(aecpc->aec);
+  aecpc->data_dumper->DumpRaw("aec_system_delay", 1, &far_buf_size_samples);
+  aecpc->data_dumper->DumpRaw("aec_known_delay", 1, &aecpc->knownDelay);
 
   return retVal;
 }
@@ -603,9 +573,7 @@ static int ProcessNormal(Aec* aecpc,
         aecpc->skew = maxSkewEst;
       }
 
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-      (void)fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
+      aecpc->data_dumper->DumpRaw("aec_skew", 1, &aecpc->skew);
     }
   }
 
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.h b/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.h
index f4b1f20ab75..8e5e52c32bf 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation.h
@@ -11,8 +11,14 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
 
+#include <memory>
+
 #include <stddef.h>
 
+extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
+}
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -56,6 +62,54 @@ typedef struct {
 
 struct AecCore;
 
+class ApmDataDumper;
+
+typedef struct Aec {
+  std::unique_ptr<ApmDataDumper> data_dumper;
+
+  int delayCtr;
+  int sampFreq;
+  int splitSampFreq;
+  int scSampFreq;
+  float sampFactor;  // scSampRate / sampFreq
+  short skewMode;
+  int bufSizeStart;
+  int knownDelay;
+  int rate_factor;
+
+  short initFlag;  // indicates if AEC has been initialized
+
+  // Variables used for averaging far end buffer size
+  short counter;
+  int sum;
+  short firstVal;
+  short checkBufSizeCtr;
+
+  // Variables used for delay shifts
+  short msInSndCardBuf;
+  short filtDelay;  // Filtered delay estimate.
+  int timeForDelayChange;
+  int startup_phase;
+  int checkBuffSize;
+  short lastDelayDiff;
+
+  // Structures
+  void* resampler;
+
+  int skewFrCtr;
+  int resample;  // if the skew is small enough we don't resample
+  int highSkewCtr;
+  float skew;
+
+  RingBuffer* far_pre_buf;  // Time domain far-end pre-buffer.
+
+  int farend_started;
+
+  // Aec instance counter.
+  static int instance_count;
+  AecCore* aec;
+} Aec;
+
 /*
  * Allocates the memory needed by the AEC. The memory needs to be initialized
  * separately using the WebRtcAec_Init() function. Returns a pointer to the
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
deleted file mode 100644
index b4a6fd8390e..00000000000
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
-
-extern "C" {
-#include "webrtc/common_audio/ring_buffer.h"
-}
-#include "webrtc/modules/audio_processing/aec/aec_core.h"
-
-namespace webrtc {
-
-typedef struct {
-  int delayCtr;
-  int sampFreq;
-  int splitSampFreq;
-  int scSampFreq;
-  float sampFactor;  // scSampRate / sampFreq
-  short skewMode;
-  int bufSizeStart;
-  int knownDelay;
-  int rate_factor;
-
-  short initFlag;  // indicates if AEC has been initialized
-
-  // Variables used for averaging far end buffer size
-  short counter;
-  int sum;
-  short firstVal;
-  short checkBufSizeCtr;
-
-  // Variables used for delay shifts
-  short msInSndCardBuf;
-  short filtDelay;  // Filtered delay estimate.
-  int timeForDelayChange;
-  int startup_phase;
-  int checkBuffSize;
-  short lastDelayDiff;
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-  FILE* bufFile;
-  FILE* delayFile;
-  FILE* skewFile;
-#endif
-
-  // Structures
-  void* resampler;
-
-  int skewFrCtr;
-  int resample;  // if the skew is small enough we don't resample
-  int highSkewCtr;
-  float skew;
-
-  RingBuffer* far_pre_buf;  // Time domain far-end pre-buffer.
-
-  int farend_started;
-
-  AecCore* aec;
-} Aec;
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index be145898b59..51a4df2b782 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -10,7 +10,6 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
-#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/typedefs.h"
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.cc
index 6bf1cf7f3ef..a17220dbd76 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.cc
@@ -14,12 +14,16 @@
 #include <stddef.h>
 #include <stdlib.h>
 
+extern "C" {
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
+}
 #include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/system_wrappers/include/compile_assert_c.h"
+extern "C" {
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+}
+
 #include "webrtc/typedefs.h"
 
 #ifdef AEC_DEBUG
@@ -208,7 +212,7 @@ StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
 ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
 
 AecmCore* WebRtcAecm_CreateCore() {
-    AecmCore* aecm = malloc(sizeof(AecmCore));
+    AecmCore* aecm = static_cast<AecmCore*>(malloc(sizeof(AecmCore)));
 
     aecm->farFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
                                             sizeof(int16_t));
@@ -361,7 +365,7 @@ static void ResetAdaptiveChannelC(AecmCore* aecm) {
 }
 
 // Initialize function pointers for ARM Neon platform.
-#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 static void WebRtcAecm_InitNeon(void)
 {
   WebRtcAecm_StoreAdaptiveChannel = WebRtcAecm_StoreAdaptiveChannelNeon;
@@ -501,20 +505,14 @@ int WebRtcAecm_InitCore(AecmCore* const aecm, int samplingFreq) {
 
     // Assert a preprocessor definition at compile-time. It's an assumption
     // used in assembly code, so check the assembly files before any change.
-    COMPILE_ASSERT(PART_LEN % 16 == 0);
+    static_assert(PART_LEN % 16 == 0, "PART_LEN is not a multiple of 16");
 
     // Initialize function pointers.
     WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesC;
     WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
     WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
 
-#ifdef WEBRTC_DETECT_NEON
-    uint64_t features = WebRtc_GetCPUFeaturesARM();
-    if ((features & kCPUFeatureNEON) != 0)
-    {
-      WebRtcAecm_InitNeon();
-    }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
     WebRtcAecm_InitNeon();
 #endif
 
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.h b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.h
index b52bb62d2de..33d80889aa4 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core.h
@@ -13,8 +13,10 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
 
+extern "C" {
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+}
 #include "webrtc/modules/audio_processing/aecm/aecm_defines.h"
 #include "webrtc/typedefs.h"
 
@@ -400,7 +402,7 @@ extern ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
 // For the above function pointers, functions for generic platforms are declared
 // and defined as static in file aecm_core.c, while those for ARM Neon platforms
 // are declared below and defined in file aecm_core_neon.c.
-#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
                                        const uint16_t* far_spectrum,
                                        int32_t* echo_est,
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
index 3a8fafa4ece..57f859f550d 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
@@ -14,19 +14,18 @@
 #include <stddef.h>
 #include <stdlib.h>
 
+extern "C" {
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/real_fft.h"
+}
 #include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/system_wrappers/include/compile_assert_c.h"
+extern "C" {
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+}
 #include "webrtc/typedefs.h"
 
 // Square root of Hanning window in Q14.
-#if defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
-// Table is defined in an ARM assembly file.
-extern const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END;
-#else
 static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
   0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
   3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224,
@@ -37,7 +36,6 @@ static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
   15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034,
   16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384
 };
-#endif
 
 #ifdef AECM_WITH_ABS_APPROX
 //Q15 alpha = 0.99439986968132  const Factor for magnitude approximation
@@ -768,4 +766,3 @@ static void ComfortNoise(AecmCore* aecm,
     out[i].imag = WebRtcSpl_AddSatW16(out[i].imag, uImag[i]);
   }
 }
-
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_mips.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc
index 3ca9982ebfa..e625a46ec5f 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_mips.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc
@@ -1563,4 +1563,3 @@ static void ComfortNoise(AecmCore* aecm,
   sgn = ((int)tt) >> 31;
   out[PART_LEN].imag = sgn == (int16_t)(tt >> 15) ? (int16_t)tt : (16384 ^ sgn);
 }
-
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_neon.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
index 1751fcf7ada..81c7667d981 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
@@ -18,19 +18,6 @@
 // TODO(kma): Re-write the corresponding assembly file, the offset
 // generating script and makefile, to replace these C functions.
 
-// Square root of Hanning window in Q14.
-const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
-  0,
-  399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
-  3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224,
-  6591, 6954, 7313, 7668, 8019, 8364, 8705, 9040,
-  9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514,
-  11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553,
-  13773, 13985, 14189, 14384, 14571, 14749, 14918, 15079,
-  15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034,
-  16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384
-};
-
 static inline void AddLanes(uint32_t* ptr, uint32x4_t v) {
 #if defined(WEBRTC_ARCH_ARM64)
   *(ptr) = vaddvq_u32(v);
diff --git a/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c b/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
index 91e6f0e80ce..a81466e678e 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
@@ -15,8 +15,10 @@
 #endif
 #include <stdlib.h>
 
+extern "C" {
 #include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+}
 #include "webrtc/modules/audio_processing/aecm/aecm_core.h"
 
 #define BUF_SIZE_FRAMES 50 // buffer size (frames)
@@ -79,7 +81,7 @@ static int WebRtcAecm_EstBufDelay(AecMobile* aecmInst, short msInSndCardBuf);
 static int WebRtcAecm_DelayComp(AecMobile* aecmInst);
 
 void* WebRtcAecm_Create() {
-    AecMobile* aecm = malloc(sizeof(AecMobile));
+    AecMobile* aecm = static_cast<AecMobile*>(malloc(sizeof(AecMobile)));
 
     WebRtcSpl_Init();
 
@@ -114,7 +116,7 @@ void* WebRtcAecm_Create() {
 }
 
 void WebRtcAecm_Free(void* aecmInst) {
-  AecMobile* aecm = aecmInst;
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
 
     if (aecm == NULL) {
       return;
@@ -138,7 +140,7 @@ void WebRtcAecm_Free(void* aecmInst) {
 
 int32_t WebRtcAecm_Init(void *aecmInst, int32_t sampFreq)
 {
-  AecMobile* aecm = aecmInst;
+    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
     AecmConfig aecConfig;
 
     if (aecm == NULL)
@@ -196,7 +198,7 @@ int32_t WebRtcAecm_Init(void *aecmInst, int32_t sampFreq)
 // farend signal.
 int32_t WebRtcAecm_GetBufferFarendError(void *aecmInst, const int16_t *farend,
                                 size_t nrOfSamples) {
-  AecMobile* aecm = aecmInst;
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
 
   if (aecm == NULL)
     return -1;
@@ -216,7 +218,7 @@ int32_t WebRtcAecm_GetBufferFarendError(void *aecmInst, const int16_t *farend,
 
 int32_t WebRtcAecm_BufferFarend(void *aecmInst, const int16_t *farend,
                                 size_t nrOfSamples) {
-  AecMobile* aecm = aecmInst;
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
 
   const int32_t err =
       WebRtcAecm_GetBufferFarendError(aecmInst, farend, nrOfSamples);
@@ -239,7 +241,7 @@ int32_t WebRtcAecm_Process(void *aecmInst, const int16_t *nearendNoisy,
                            const int16_t *nearendClean, int16_t *out,
                            size_t nrOfSamples, int16_t msInSndCardBuf)
 {
-  AecMobile* aecm = aecmInst;
+    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
     int32_t retVal = 0;
     size_t i;
     short nmbrOfFilledBuffers;
@@ -435,7 +437,7 @@ int32_t WebRtcAecm_Process(void *aecmInst, const int16_t *nearendNoisy,
 
 int32_t WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
 {
-  AecMobile* aecm = aecmInst;
+    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
 
     if (aecm == NULL)
     {
@@ -516,8 +518,8 @@ int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
                                 const void* echo_path,
                                 size_t size_bytes)
 {
-  AecMobile* aecm = aecmInst;
-    const int16_t* echo_path_ptr = echo_path;
+    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+    const int16_t* echo_path_ptr = static_cast<const int16_t*>(echo_path);
 
     if (aecmInst == NULL) {
       return -1;
@@ -544,8 +546,8 @@ int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
                                void* echo_path,
                                size_t size_bytes)
 {
-  AecMobile* aecm = aecmInst;
-    int16_t* echo_path_ptr = echo_path;
+    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+    int16_t* echo_path_ptr = static_cast<int16_t*>(echo_path);
 
     if (aecmInst == NULL) {
       return -1;
diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi b/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi
index 264f3e5befb..e6eab36a01e 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing.gypi
@@ -9,6 +9,8 @@
 {
   'variables': {
     'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets',
+    # Outputs some low-level debug files.
+    'aec_debug_dump%': 0,
   },
   'targets': [
     {
@@ -16,7 +18,6 @@
       'type': 'static_library',
       'variables': {
         # Outputs some low-level debug files.
-        'aec_debug_dump%': 0,
         'agc_debug_dump%': 0,
 
         # Disables the usual mode where we trust the reported system delay
@@ -34,17 +35,16 @@
       'sources': [
         'aec/aec_core.cc',
         'aec/aec_core.h',
-        'aec/aec_core_internal.h',
-        'aec/aec_rdft.c',
+        'aec/aec_core_optimized_methods.h',
+        'aec/aec_rdft.cc',
         'aec/aec_rdft.h',
         'aec/aec_resampler.cc',
         'aec/aec_resampler.h',
         'aec/echo_cancellation.cc',
-        'aec/echo_cancellation_internal.h',
         'aec/echo_cancellation.h',
-        'aecm/aecm_core.c',
+        'aecm/aecm_core.cc',
         'aecm/aecm_core.h',
-        'aecm/echo_control_mobile.c',
+        'aecm/echo_control_mobile.cc',
         'aecm/echo_control_mobile.h',
         'agc/agc.cc',
         'agc/agc.h',
@@ -91,9 +91,8 @@
         'intelligibility/intelligibility_utils.h',
         'level_estimator_impl.cc',
         'level_estimator_impl.h',
-        'logging/aec_logging.h',
-        'logging/aec_logging_file_handling.cc',
-        'logging/aec_logging_file_handling.h',
+        'logging/apm_data_dumper.cc',
+        'logging/apm_data_dumper.h',
         'noise_suppression_impl.cc',
         'noise_suppression_impl.h',
         'render_queue_item_verifier.h',
@@ -120,10 +119,10 @@
         'typing_detection.h',
         'utility/block_mean_calculator.cc',
         'utility/block_mean_calculator.h',
-        'utility/delay_estimator.c',
+        'utility/delay_estimator.cc',
         'utility/delay_estimator.h',
         'utility/delay_estimator_internal.h',
-        'utility/delay_estimator_wrapper.c',
+        'utility/delay_estimator_wrapper.cc',
         'utility/delay_estimator_wrapper.h',
         'vad/common.h',
         'vad/gmm.cc',
@@ -150,7 +149,9 @@
       ],
       'conditions': [
         ['aec_debug_dump==1', {
-          'defines': ['WEBRTC_AEC_DEBUG_DUMP',],
+          'defines': ['WEBRTC_AEC_DEBUG_DUMP=1',],
+        }, {
+          'defines': ['WEBRTC_AEC_DEBUG_DUMP=0',],
         }],
         ['aec_untrusted_delay_for_testing==1', {
           'defines': ['WEBRTC_UNTRUSTED_DELAY',],
@@ -201,19 +202,19 @@
         }],
         ['target_arch=="mipsel" and mips_arch_variant!="r6"', {
           'sources': [
-            'aecm/aecm_core_mips.c',
+            'aecm/aecm_core_mips.cc',
           ],
           'conditions': [
             ['mips_float_abi=="hard"', {
               'sources': [
                 'aec/aec_core_mips.cc',
-                'aec/aec_rdft_mips.c',
+                'aec/aec_rdft_mips.cc',
               ],
             }],
           ],
         }, {
           'sources': [
-            'aecm/aecm_core_c.c',
+            'aecm/aecm_core_c.cc',
           ],
         }],
       ],
@@ -246,9 +247,14 @@
           'type': 'static_library',
           'sources': [
             'aec/aec_core_sse2.cc',
-            'aec/aec_rdft_sse2.c',
+            'aec/aec_rdft_sse2.cc',
           ],
           'conditions': [
+            ['aec_debug_dump==1', {
+              'defines': ['WEBRTC_AEC_DEBUG_DUMP=1',],
+            }, {
+              'defines': ['WEBRTC_AEC_DEBUG_DUMP=0',],
+            }],
             ['os_posix==1', {
               'cflags': [ '-msse2', ],
               'xcode_settings': {
@@ -269,10 +275,18 @@
         ],
         'sources': [
           'aec/aec_core_neon.cc',
-          'aec/aec_rdft_neon.c',
-          'aecm/aecm_core_neon.c',
+          'aec/aec_rdft_neon.cc',
+          'aecm/aecm_core_neon.cc',
           'ns/nsx_core_neon.c',
         ],
+        'conditions': [
+          ['aec_debug_dump==1', {
+            'defines': ['WEBRTC_AEC_DEBUG_DUMP=1',],
+          }],
+          ['aec_debug_dump==0', {
+            'defines': ['WEBRTC_AEC_DEBUG_DUMP=0',],
+          }],
+        ],
       }],
     }],
   ],
diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc
index 2b6f1c4ac19..e75b3280346 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -94,7 +94,7 @@ bool is_multi_band(int sample_rate_hz) {
          sample_rate_hz == AudioProcessing::kSampleRate48kHz;
 }
 
-int ClosestNativeRate(int min_proc_rate) {
+int ClosestHigherNativeRate(int min_proc_rate) {
   for (int rate : AudioProcessing::kNativeSampleRatesHz) {
     if (rate >= min_proc_rate) {
       return rate;
@@ -163,12 +163,10 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
       private_submodules_(new ApmPrivateSubmodules(beamformer)),
       constants_(config.Get<ExperimentalAgc>().startup_min_volume,
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
-                 false,
+                 false),
 #else
-                 config.Get<ExperimentalAgc>().enabled,
+                 config.Get<ExperimentalAgc>().enabled),
 #endif
-                 config.Get<Intelligibility>().enabled),
-
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
       capture_(false,
 #else
@@ -176,7 +174,8 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
 #endif
                config.Get<Beamforming>().array_geometry,
                config.Get<Beamforming>().target_direction),
-      capture_nonlocked_(config.Get<Beamforming>().enabled)
+      capture_nonlocked_(config.Get<Beamforming>().enabled,
+                         config.Get<Intelligibility>().enabled)
 {
   {
     rtc::CritScope cs_render(&crit_render_);
@@ -362,22 +361,24 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
 
   formats_.api_format = config;
 
-  capture_nonlocked_.fwd_proc_format = StreamConfig(ClosestNativeRate(std::min(
-      formats_.api_format.input_stream().sample_rate_hz(),
-      formats_.api_format.output_stream().sample_rate_hz())));
+  capture_nonlocked_.fwd_proc_format = StreamConfig(ClosestHigherNativeRate(
+      std::min(formats_.api_format.input_stream().sample_rate_hz(),
+               formats_.api_format.output_stream().sample_rate_hz())));
 
-  // We normally process the reverse stream at 16 kHz. Unless...
-  int rev_proc_rate = kSampleRate16kHz;
+  int rev_proc_rate = ClosestHigherNativeRate(std::min(
+      formats_.api_format.reverse_input_stream().sample_rate_hz(),
+      formats_.api_format.reverse_output_stream().sample_rate_hz()));
+  // TODO(aluebs): Remove this restriction once we figure out why the 3-band
+  // splitting filter degrades the AEC performance.
+  if (rev_proc_rate > kSampleRate32kHz) {
+    rev_proc_rate = is_rev_processed() ? kSampleRate32kHz : kSampleRate16kHz;
+  }
+  // If the forward sample rate is 8 kHz, the reverse stream is also processed
+  // at this rate.
   if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
-    // ...the forward stream is at 8 kHz.
     rev_proc_rate = kSampleRate8kHz;
   } else {
-    if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
-        kSampleRate32kHz) {
-      // ...or the input is at 32 kHz, in which case we use the splitting
-      // filter rather than the resampler.
-      rev_proc_rate = kSampleRate32kHz;
-    }
+    rev_proc_rate = std::max(rev_proc_rate, static_cast<int>(kSampleRate16kHz));
   }
 
   // Always downmix the reverse stream to mono for analysis. This has been
@@ -409,6 +410,13 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) {
     InitializeTransient();
   }
 
+  if(capture_nonlocked_.intelligibility_enabled !=
+     config.Get<Intelligibility>().enabled) {
+    capture_nonlocked_.intelligibility_enabled =
+        config.Get<Intelligibility>().enabled;
+    InitializeIntelligibility();
+  }
+
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
   if (capture_nonlocked_.beamformer_enabled !=
           config.Get<Beamforming>().enabled) {
@@ -702,10 +710,13 @@ int AudioProcessingImpl::ProcessStreamLocked() {
     ca->CopyLowPassToReference();
   }
   public_submodules_->noise_suppression->ProcessCaptureAudio(ca);
-  if (constants_.intelligibility_enabled) {
+  if (capture_nonlocked_.intelligibility_enabled) {
     RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
+    int gain_db = public_submodules_->gain_control->is_enabled() ?
+                  public_submodules_->gain_control->compression_gain_db() :
+                  0;
     public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate(
-        public_submodules_->noise_suppression->NoiseEstimate());
+        public_submodules_->noise_suppression->NoiseEstimate(), gain_db);
   }
 
   // Ensure that the stream delay was set before the call to the
@@ -898,7 +909,7 @@ int AudioProcessingImpl::ProcessReverseStreamLocked() {
     ra->SplitIntoFrequencyBands();
   }
 
-  if (constants_.intelligibility_enabled) {
+  if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
         ra->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
         ra->num_channels());
@@ -1146,16 +1157,16 @@ bool AudioProcessingImpl::fwd_analysis_needed() const {
 }
 
 bool AudioProcessingImpl::is_rev_processed() const {
-  return constants_.intelligibility_enabled;
+  return capture_nonlocked_.intelligibility_enabled;
 }
 
 bool AudioProcessingImpl::rev_synthesis_needed() const {
   return (is_rev_processed() &&
-          formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz);
+          is_multi_band(formats_.rev_proc_format.sample_rate_hz()));
 }
 
 bool AudioProcessingImpl::rev_analysis_needed() const {
-  return formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
+  return is_multi_band(formats_.rev_proc_format.sample_rate_hz()) &&
          (is_rev_processed() ||
           public_submodules_->echo_cancellation
               ->is_enabled_render_side_query() ||
@@ -1211,7 +1222,7 @@ void AudioProcessingImpl::InitializeBeamformer() {
 }
 
 void AudioProcessingImpl::InitializeIntelligibility() {
-  if (constants_.intelligibility_enabled) {
+  if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer.reset(
         new IntelligibilityEnhancer(capture_nonlocked_.split_rate,
                                     render_.render_audio->num_channels(),
@@ -1392,8 +1403,10 @@ int AudioProcessingImpl::WriteInitMessage() {
       formats_.api_format.reverse_input_stream().sample_rate_hz());
   msg->set_output_sample_rate(
       formats_.api_format.output_stream().sample_rate_hz());
-  // TODO(ekmeyerson): Add reverse output fields to
-  // debug_dump_.capture.event_msg.
+  msg->set_reverse_output_sample_rate(
+      formats_.api_format.reverse_output_stream().sample_rate_hz());
+  msg->set_num_reverse_output_channels(
+      formats_.api_format.reverse_output_stream().num_channels());
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
@@ -1436,6 +1449,8 @@ int AudioProcessingImpl::WriteConfigMessage(bool forced) {
 
   config.set_transient_suppression_enabled(
       capture_.transient_suppressor_enabled);
+  config.set_intelligibility_enhancer_enabled(
+      capture_nonlocked_.intelligibility_enabled);
 
   std::string experiments_description =
       public_submodules_->echo_cancellation->GetExperimentsDescription();
diff --git a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h
index 7323b85c78e..04ddabd1c7f 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -276,16 +276,12 @@ class AudioProcessingImpl : public AudioProcessing {
 
   // APM constants.
   const struct ApmConstants {
-    ApmConstants(int agc_startup_min_volume,
-                 bool use_experimental_agc,
-                 bool intelligibility_enabled)
+    ApmConstants(int agc_startup_min_volume, bool use_experimental_agc)
         :  // Format of processing streams at input/output call sites.
           agc_startup_min_volume(agc_startup_min_volume),
-          use_experimental_agc(use_experimental_agc),
-          intelligibility_enabled(intelligibility_enabled) {}
+          use_experimental_agc(use_experimental_agc) {}
     int agc_startup_min_volume;
     bool use_experimental_agc;
-    bool intelligibility_enabled;
   } constants_;
 
   struct ApmCaptureState {
@@ -325,11 +321,13 @@ class AudioProcessingImpl : public AudioProcessing {
   } capture_ GUARDED_BY(crit_capture_);
 
   struct ApmCaptureNonLockedState {
-    ApmCaptureNonLockedState(bool beamformer_enabled)
+    ApmCaptureNonLockedState(bool beamformer_enabled,
+                             bool intelligibility_enabled)
         : fwd_proc_format(kSampleRate16kHz),
           split_rate(kSampleRate16kHz),
           stream_delay_ms(0),
-          beamformer_enabled(beamformer_enabled) {}
+          beamformer_enabled(beamformer_enabled),
+          intelligibility_enabled(intelligibility_enabled) {}
     // Only the rate and samples fields of fwd_proc_format_ are used because the
     // forward processing number of channels is mutable and is tracked by the
     // capture_audio_.
@@ -337,6 +335,7 @@ class AudioProcessingImpl : public AudioProcessing {
     int split_rate;
     int stream_delay_ms;
     bool beamformer_enabled;
+    bool intelligibility_enabled;
   } capture_nonlocked_;
 
   struct ApmRenderState {
diff --git a/chromium/third_party/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_unittest.cc
index 948c5efd93f..ded75c86520 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -54,7 +54,12 @@ bool write_ref_data = false;
 const google::protobuf::int32 kChannels[] = {1, 2};
 const int kSampleRates[] = {8000, 16000, 32000, 48000};
 
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+// Android doesn't support 48kHz.
+const int kProcessSampleRates[] = {8000, 16000, 32000};
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
 const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
+#endif
 
 enum StreamDirection { kForward = 0, kReverse };
 
@@ -2692,7 +2697,7 @@ INSTANTIATE_TEST_CASE_P(
                     std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
                     std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
                     std::tr1::make_tuple(16000, 16000, 48000, 16000, 40, 20),
-                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 50, 20),
+                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
                     std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 
 #elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
@@ -2748,7 +2753,7 @@ INSTANTIATE_TEST_CASE_P(
                     std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
                     std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
                     std::tr1::make_tuple(16000, 16000, 48000, 16000, 35, 20),
-                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
+                    std::tr1::make_tuple(16000, 16000, 32000, 16000, 35, 20),
                     std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 #endif
 
diff --git a/chromium/third_party/webrtc/modules/audio_processing/debug.proto b/chromium/third_party/webrtc/modules/audio_processing/debug.proto
index 1c025fbc72c..44177735e8c 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/debug.proto
+++ b/chromium/third_party/webrtc/modules/audio_processing/debug.proto
@@ -12,6 +12,8 @@ message Init {
   optional int32 num_reverse_channels = 5;
   optional int32 reverse_sample_rate = 6;
   optional int32 output_sample_rate = 7;
+  optional int32 reverse_output_sample_rate = 8;
+  optional int32 num_reverse_output_channels = 9;
 }
 
 // May contain interleaved or deinterleaved data, but don't store both formats.
@@ -44,7 +46,7 @@ message Stream {
 // Contains the configurations of various APM component. A Config message is
 // added when any of the fields are changed.
 message Config {
-  // Next field number 18.
+  // Next field number 19.
   // Acoustic echo canceler.
   optional bool aec_enabled = 1;
   optional bool aec_delay_agnostic_enabled = 2;
@@ -70,6 +72,8 @@ message Config {
   // Semicolon-separated string containing experimental feature
   // descriptions.
   optional string experiments_description = 17;
+  // Intelligibility Enhancer
+  optional bool intelligibility_enhancer_enabled = 18;
 }
 
 message Event {
diff --git a/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
index 97f69974dba..d7b27e99f3c 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -12,6 +12,7 @@
 
 #include <string.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/system_wrappers/include/logging.h"
diff --git a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc
index 3a6fcb116d5..2461f72ad35 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
@@ -274,6 +275,11 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
   return AudioProcessing::kNoError;
 }
 
+int GainControlImpl::compression_gain_db() const {
+  rtc::CritScope cs(crit_capture_);
+  return compression_gain_db_;
+}
+
 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
   rtc::CritScope cs(crit_capture_);
@@ -413,11 +419,6 @@ int GainControlImpl::set_compression_gain_db(int gain) {
   return Configure();
 }
 
-int GainControlImpl::compression_gain_db() const {
-  rtc::CritScope cs(crit_capture_);
-  return compression_gain_db_;
-}
-
 int GainControlImpl::enable_limiter(bool enable) {
   {
     rtc::CritScope cs(crit_capture_);
diff --git a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h
index 9498ac60b54..2459ce3b4b7 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/gain_control_impl.h
@@ -51,6 +51,8 @@ class GainControlImpl : public GainControl {
   // Reads render side data that has been queued on the render call.
   void ReadQueuedRenderData();
 
+  int compression_gain_db() const override;
+
  private:
   class GainController;
 
@@ -61,7 +63,6 @@ class GainControlImpl : public GainControl {
   int set_target_level_dbfs(int level) override;
   int target_level_dbfs() const override;
   int set_compression_gain_db(int gain) override;
-  int compression_gain_db() const override;
   int enable_limiter(bool enable) override;
   int set_analog_level_limits(int minimum, int maximum) override;
   int analog_level_minimum() const override;
diff --git a/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h b/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h
index 0c93a984b24..2f8e48f82da 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/include/audio_processing.h
@@ -160,9 +160,7 @@ struct Beamforming {
   const SphericalPointf target_direction;
 };
 
-// Use to enable intelligibility enhancer in audio processing. Must be provided
-// though the constructor. It will have no impact if used with
-// AudioProcessing::SetExtraOptions().
+// Use to enable intelligibility enhancer in audio processing.
 //
 // Note: If enabled and the reverse stream has more than one output channel,
 // the reverse stream will become an upmixed mono signal.
@@ -685,7 +683,7 @@ class EchoCancellation {
     // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
     AudioProcessing::Statistic a_nlp;
 
-    // Fraction of time that the AEC linear filter is divergent, in a 0.5-second
+    // Fraction of time that the AEC linear filter is divergent, in a 1-second
     // non-overlapped aggregation window.
     float divergent_filter_fraction;
   };
diff --git a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
index de36b7a8bc7..ae7f9119213 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
@@ -38,6 +38,8 @@ const float kDecayRate = 0.994f;              // Power estimation decay rate.
 const float kMaxRelativeGainChange = 0.006f;
 const float kRho = 0.0004f;  // Default production and interpretation SNR.
 const float kPowerNormalizationFactor = 1.f / (1 << 30);
+const float kMaxActiveSNR = 128.f;  // 21dB
+const float kMinInactiveSNR = 32.f;  // 15dB
 
 // Returns dot product of vectors |a| and |b| with size |length|.
 float DotProduct(const float* a, const float* b, size_t length) {
@@ -84,6 +86,8 @@ IntelligibilityEnhancer::IntelligibilityEnhancer(int sample_rate_hz,
       audio_s16_(chunk_length_),
       chunks_since_voice_(kSpeechOffsetDelay),
       is_speech_(false),
+      snr_(kMaxActiveSNR),
+      is_active_(false),
       noise_estimation_buffer_(num_noise_bins),
       noise_estimation_queue_(kMaxNumNoiseEstimatesToBuffer,
                               std::vector<float>(num_noise_bins),
@@ -105,8 +109,12 @@ IntelligibilityEnhancer::IntelligibilityEnhancer(int sample_rate_hz,
 }
 
 void IntelligibilityEnhancer::SetCaptureNoiseEstimate(
-    std::vector<float> noise) {
+    std::vector<float> noise, int gain_db) {
   RTC_DCHECK_EQ(noise.size(), num_noise_bins_);
+  const float gain = std::pow(10.f, gain_db / 20.f);
+  for (auto& bin : noise) {
+    bin *= gain;
+  }
   // Disregarding return value since buffer overflow is acceptable, because it
   // is not critical to get each noise estimate.
   if (noise_estimation_queue_.Insert(&noise)) {
@@ -135,29 +143,55 @@ void IntelligibilityEnhancer::ProcessAudioBlock(
   if (is_speech_) {
     clear_power_estimator_.Step(in_block[0]);
   }
-  const std::vector<float>& clear_power = clear_power_estimator_.power();
-  const std::vector<float>& noise_power = noise_power_estimator_.power();
-  MapToErbBands(clear_power.data(), render_filter_bank_,
-                filtered_clear_pow_.data());
-  MapToErbBands(noise_power.data(), capture_filter_bank_,
-                filtered_noise_pow_.data());
-  SolveForGainsGivenLambda(kLambdaTop, start_freq_, gains_eq_.data());
-  const float power_target = std::accumulate(
-      filtered_clear_pow_.data(), filtered_clear_pow_.data() + bank_size_, 0.f);
-  const float power_top =
-      DotProduct(gains_eq_.data(), filtered_clear_pow_.data(), bank_size_);
-  SolveForGainsGivenLambda(kLambdaBot, start_freq_, gains_eq_.data());
-  const float power_bot =
-      DotProduct(gains_eq_.data(), filtered_clear_pow_.data(), bank_size_);
-  if (power_target >= power_bot && power_target <= power_top) {
-    SolveForLambda(power_target);
-    UpdateErbGains();
-  }  // Else experiencing power underflow, so do nothing.
+  SnrBasedEffectActivation();
+  if (is_active_) {
+    MapToErbBands(clear_power_estimator_.power().data(), render_filter_bank_,
+                  filtered_clear_pow_.data());
+    MapToErbBands(noise_power_estimator_.power().data(), capture_filter_bank_,
+                  filtered_noise_pow_.data());
+    SolveForGainsGivenLambda(kLambdaTop, start_freq_, gains_eq_.data());
+    const float power_target = std::accumulate(
+        filtered_clear_pow_.data(),
+        filtered_clear_pow_.data() + bank_size_,
+        0.f);
+    const float power_top =
+        DotProduct(gains_eq_.data(), filtered_clear_pow_.data(), bank_size_);
+    SolveForGainsGivenLambda(kLambdaBot, start_freq_, gains_eq_.data());
+    const float power_bot =
+        DotProduct(gains_eq_.data(), filtered_clear_pow_.data(), bank_size_);
+    if (power_target >= power_bot && power_target <= power_top) {
+      SolveForLambda(power_target);
+      UpdateErbGains();
+    }  // Else experiencing power underflow, so do nothing.
+  }
   for (size_t i = 0; i < in_channels; ++i) {
     gain_applier_.Apply(in_block[i], out_block[i]);
   }
 }
 
+void IntelligibilityEnhancer::SnrBasedEffectActivation() {
+  const float* clear_psd = clear_power_estimator_.power().data();
+  const float* noise_psd = noise_power_estimator_.power().data();
+  const float clear_power =
+      std::accumulate(clear_psd, clear_psd + freqs_, 0.f);
+  const float noise_power =
+      std::accumulate(noise_psd, noise_psd + freqs_, 0.f);
+  snr_ = kDecayRate * snr_ + (1.f - kDecayRate) * clear_power /
+      (noise_power + std::numeric_limits<float>::epsilon());
+  if (is_active_) {
+    if (snr_ > kMaxActiveSNR) {
+      is_active_ = false;
+      // Set the target gains to unity.
+      float* gains = gain_applier_.target();
+      for (size_t i = 0; i < freqs_; ++i) {
+        gains[i] = 1.f;
+      }
+    }
+  } else {
+    is_active_ = snr_ < kMinInactiveSNR;
+  }
+}
+
 void IntelligibilityEnhancer::SolveForLambda(float power_target) {
   const float kConvergeThresh = 0.001f;  // TODO(ekmeyerson): Find best values
   const int kMaxIters = 100;             // for these, based on experiments.
diff --git a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
index 14132129349..63ae80e2c44 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -36,7 +36,7 @@ class IntelligibilityEnhancer : public LappedTransform::Callback {
                           size_t num_noise_bins);
 
   // Sets the capture noise magnitude spectrum estimate.
-  void SetCaptureNoiseEstimate(std::vector<float> noise);
+  void SetCaptureNoiseEstimate(std::vector<float> noise, int gain_db);
 
   // Reads chunk of speech in time domain and updates with modified signal.
   void ProcessRenderAudio(float* const* audio,
@@ -56,6 +56,12 @@ class IntelligibilityEnhancer : public LappedTransform::Callback {
  private:
   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation);
   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains);
+  FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest,
+                           TestNoiseGainHasExpectedResult);
+
+  // Updates the SNR estimation and enables or disables this component using a
+  // hysteresis.
+  void SnrBasedEffectActivation();
 
   // Bisection search for optimal |lambda|.
   void SolveForLambda(float power_target);
@@ -103,6 +109,8 @@ class IntelligibilityEnhancer : public LappedTransform::Callback {
   std::vector<int16_t> audio_s16_;
   size_t chunks_since_voice_;
   bool is_speech_;
+  float snr_;
+  bool is_active_;
 
   std::vector<float> noise_estimation_buffer_;
   SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>
diff --git a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc
index 080e228cb8e..30035ab16ed 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc
@@ -237,7 +237,7 @@ void ProcessOneFrame(int sample_rate_hz,
   noise_suppressor->ProcessCaptureAudio(capture_audio_buffer);
 
   intelligibility_enhancer->SetCaptureNoiseEstimate(
-      noise_suppressor->NoiseEstimate());
+      noise_suppressor->NoiseEstimate(), 0);
 
   if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
     render_audio_buffer->MergeFrequencyBands();
@@ -311,12 +311,17 @@ void RunBitexactnessTest(int sample_rate_hz,
       output_reference, render_output, kElementErrorBound));
 }
 
+float float_rand() {
+  return std::rand() * 2.f / RAND_MAX - 1;
+}
+
 }  // namespace
 
 class IntelligibilityEnhancerTest : public ::testing::Test {
  protected:
   IntelligibilityEnhancerTest()
       : clear_data_(kSamples), noise_data_(kSamples), orig_data_(kSamples) {
+    std::srand(1);
     enh_.reset(
         new IntelligibilityEnhancer(kSampleRate, kNumChannels, kNumNoiseBins));
   }
@@ -352,8 +357,6 @@ TEST_F(IntelligibilityEnhancerTest, TestRenderUpdate) {
   std::fill(orig_data_.begin(), orig_data_.end(), 0.f);
   std::fill(clear_data_.begin(), clear_data_.end(), 0.f);
   EXPECT_FALSE(CheckUpdate());
-  std::srand(1);
-  auto float_rand = []() { return std::rand() * 2.f / RAND_MAX - 1; };
   std::generate(noise_data_.begin(), noise_data_.end(), float_rand);
   EXPECT_FALSE(CheckUpdate());
   std::generate(clear_data_.begin(), clear_data_.end(), float_rand);
@@ -403,6 +406,29 @@ TEST_F(IntelligibilityEnhancerTest, TestSolveForGains) {
   }
 }
 
+TEST_F(IntelligibilityEnhancerTest, TestNoiseGainHasExpectedResult) {
+  const int kGainDB = 6;
+  const float kGainFactor = std::pow(10.f, kGainDB / 20.f);
+  const float kTolerance = 0.003f;
+  std::vector<float> noise(kNumNoiseBins);
+  std::vector<float> noise_psd(kNumNoiseBins);
+  std::generate(noise.begin(), noise.end(), float_rand);
+  for (size_t i = 0; i < kNumNoiseBins; ++i) {
+    noise_psd[i] = kGainFactor * kGainFactor * noise[i] * noise[i];
+  }
+  float* clear_cursor = clear_data_.data();
+  for (size_t i = 0; i < kNumFramesToProcess; ++i) {
+    enh_->SetCaptureNoiseEstimate(noise, kGainDB);
+    enh_->ProcessRenderAudio(&clear_cursor, kSampleRate, kNumChannels);
+  }
+  const std::vector<float>& estimated_psd =
+      enh_->noise_power_estimator_.power();
+  for (size_t i = 0; i < kNumNoiseBins; ++i) {
+    EXPECT_LT(std::abs(estimated_psd[i] - noise_psd[i]) / noise_psd[i],
+              kTolerance);
+  }
+}
+
 TEST(IntelligibilityEnhancerBitExactnessTest, DISABLED_Mono8kHz) {
   const float kOutputReference[] = {-0.001892f, -0.003296f, -0.001953f};
 
diff --git a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
index 3a9433b4768..3675f66cafe 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
@@ -22,16 +22,15 @@ namespace intelligibility {
 
 namespace {
 
+const float kMinFactor = 0.01f;
+const float kMaxFactor = 1000.f;
+
 // Return |current| changed towards |target|, with the relative change being at
 // most |limit|.
 float UpdateFactor(float target, float current, float limit) {
   float gain = target / (current + std::numeric_limits<float>::epsilon());
-  if (gain < 1.f - limit) {
-    gain = 1.f - limit;
-  } else if (gain > 1.f + limit) {
-    gain = 1.f + limit;
-  }
-  return current * gain + std::numeric_limits<float>::epsilon();
+  gain = std::min(std::max(gain, 1.f - limit), 1.f + limit);
+  return std::min(std::max(current * gain, kMinFactor), kMaxFactor);;
 }
 
 }  // namespace
diff --git a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
index 64ccfd96efc..abd10d85165 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc
@@ -64,7 +64,7 @@ void void_main(int argc, char* argv[]) {
     capture_audio.CopyFrom(noise_buf.channels(), stream_config);
     ns.AnalyzeCaptureAudio(&capture_audio);
     ns.ProcessCaptureAudio(&capture_audio);
-    enh.SetCaptureNoiseEstimate(ns.NoiseEstimate());
+    enh.SetCaptureNoiseEstimate(ns.NoiseEstimate(), 0);
     enh.ProcessRenderAudio(in_buf.channels(), in_file.sample_rate(),
                            in_file.num_channels());
     Interleave(in_buf.channels(), in_buf.num_frames(), in_buf.num_channels(),
diff --git a/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging.h b/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging.h
deleted file mode 100644
index b062913be28..00000000000
--- a/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging.h
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
-
-#include <stdio.h>
-
-#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
-
-// To enable AEC logging, invoke GYP with -Daec_debug_dump=1.
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-// Dumps a wav data to file.
-#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
-  do {                                                   \
-    rtc_WavWriteSamples(file, data, num_samples);        \
-  } while (0)
-
-// (Re)opens a wav file for writing using the specified sample rate.
-#define RTC_AEC_DEBUG_WAV_REOPEN(name, instance_index, process_rate,     \
-                                 sample_rate, wav_file)                  \
-  do {                                                                   \
-    WebRtcAec_ReopenWav(name, instance_index, process_rate, sample_rate, \
-                        wav_file);                                       \
-  } while (0)
-
-// Closes a wav file.
-#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
-  do {                                    \
-    rtc_WavClose(wav_file);               \
-  } while (0)
-
-// Dumps a raw data to file.
-#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
-  do {                                                 \
-    (void) fwrite(data, data_size, 1, file);           \
-  } while (0)
-
-// Dumps a raw scalar int32 to file.
-#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_INT32(file, data)             \
-  do {                                                               \
-    int32_t value_to_store = data;                                   \
-    (void) fwrite(&value_to_store, sizeof(value_to_store), 1, file); \
-  } while (0)
-
-// Dumps a raw scalar double to file.
-#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_DOUBLE(file, data)            \
-  do {                                                               \
-    double value_to_store = data;                                    \
-    (void) fwrite(&value_to_store, sizeof(value_to_store), 1, file); \
-  } while (0)
-
-// Opens a raw data file for writing using the specified sample rate.
-#define RTC_AEC_DEBUG_RAW_OPEN(name, instance_counter, file) \
-  do {                                                       \
-    WebRtcAec_RawFileOpen(name, instance_counter, file);     \
-  } while (0)
-
-// Closes a raw data file.
-#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
-  do {                                \
-    fclose(file);                     \
-  } while (0)
-
-#else  // RTC_AEC_DEBUG_DUMP
-#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
-  do {                                                   \
-  } while (0)
-
-#define RTC_AEC_DEBUG_WAV_REOPEN(wav_file, name, instance_index, process_rate, \
-                                 sample_rate)                                  \
-  do {                                                                         \
-  } while (0)
-
-#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
-  do {                                    \
-  } while (0)
-
-#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
-  do {                                                 \
-  } while (0)
-
-#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_INT32(file, data) \
-  do {                                                   \
-  } while (0)
-
-#define RTC_AEC_DEBUG_RAW_WRITE_SCALAR_DOUBLE(file, data) \
-  do {                                                    \
-  } while (0)
-
-#define RTC_AEC_DEBUG_RAW_OPEN(file, name, instance_counter) \
-  do {                                                       \
-  } while (0)
-
-#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
-  do {                                \
-  } while (0)
-
-#endif  // WEBRTC_AEC_DEBUG_DUMP
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
diff --git a/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc b/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
deleted file mode 100644
index 3a434714e1e..00000000000
--- a/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
-
-#include <stdint.h>
-#include <stdio.h>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/common_audio/wav_file.h"
-#include "webrtc/typedefs.h"
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-void WebRtcAec_ReopenWav(const char* name,
-                         int instance_index,
-                         int process_rate,
-                         int sample_rate,
-                         rtc_WavWriter** wav_file) {
-  if (*wav_file) {
-    if (rtc_WavSampleRate(*wav_file) == sample_rate)
-      return;
-    rtc_WavClose(*wav_file);
-  }
-  char filename[64];
-  int written = rtc::sprintfn(filename, sizeof(filename), "%s%d-%d.wav", name,
-                              instance_index, process_rate);
-
-  // Ensure there was no buffer output error.
-  RTC_DCHECK_GE(written, 0);
-  // Ensure that the buffer size was sufficient.
-  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
-
-  *wav_file = rtc_WavOpen(filename, sample_rate, 1);
-}
-
-void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file) {
-  char filename[64];
-  int written = rtc::sprintfn(filename, sizeof(filename), "%s_%d.dat", name,
-                              instance_index);
-
-  // Ensure there was no buffer output error.
-  RTC_DCHECK_GE(written, 0);
-  // Ensure that the buffer size was sufficient.
-  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
-
-  *file = fopen(filename, "wb");
-}
-
-#endif  // WEBRTC_AEC_DEBUG_DUMP
diff --git a/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h b/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h
deleted file mode 100644
index 5ec83948726..00000000000
--- a/chromium/third_party/webrtc/modules/audio_processing/logging/aec_logging_file_handling.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
-
-#include <stdio.h>
-
-#include "webrtc/common_audio/wav_file.h"
-#include "webrtc/typedefs.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-// Opens a new Wav file for writing. If it was already open with a different
-// sample frequency, it closes it first.
-void WebRtcAec_ReopenWav(const char* name,
-                         int instance_index,
-                         int process_rate,
-                         int sample_rate,
-                         rtc_WavWriter** wav_file);
-
-// Opens dumpfile with instance-specific filename.
-void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file);
-
-#endif  // WEBRTC_AEC_DEBUG_DUMP
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_FILE_HANDLING_
diff --git a/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
new file mode 100644
index 00000000000..491196e0972
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+#include <sstream>
+
+#include "webrtc/base/stringutils.h"
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_AEC_DEBUG_DUMP) || \
+    (WEBRTC_AEC_DEBUG_DUMP != 0 && WEBRTC_AEC_DEBUG_DUMP != 1)
+#error "Set WEBRTC_AEC_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+
+namespace {
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+std::string FormFileName(const char* name,
+                         int instance_index,
+                         int reinit_index,
+                         const std::string& suffix) {
+  std::stringstream ss;
+  ss << name << "_" << instance_index << "-" << reinit_index << suffix;
+  return ss.str();
+}
+#endif
+
+}  // namespace
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+FILE* ApmDataDumper::GetRawFile(const char* name) {
+  std::string filename =
+      FormFileName(name, instance_index_, recording_set_index_, ".dat");
+  auto& f = raw_files_[filename];
+  if (!f) {
+    f.reset(fopen(filename.c_str(), "wb"));
+  }
+  return f.get();
+}
+
+WavWriter* ApmDataDumper::GetWavFile(const char* name,
+                                     int sample_rate_hz,
+                                     int num_channels) {
+  std::string filename =
+      FormFileName(name, instance_index_, recording_set_index_, ".wav");
+  auto& f = wav_files_[filename];
+  if (!f) {
+    f.reset(new WavWriter(filename.c_str(), sample_rate_hz, num_channels));
+  }
+  return f.get();
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.h b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.h
new file mode 100644
index 00000000000..93232b7c9dc
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_processing/logging/apm_data_dumper.h
@@ -0,0 +1,129 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+#include <unordered_map>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_audio/wav_file.h"
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_AEC_DEBUG_DUMP) || \
+    (WEBRTC_AEC_DEBUG_DUMP != 0 && WEBRTC_AEC_DEBUG_DUMP != 1)
+#error "Set WEBRTC_AEC_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+// Functor used to use as a custom deleter in the map of file pointers to raw
+// files.
+struct RawFileCloseFunctor {
+  void operator()(FILE* f) const { fclose(f); }
+};
+#endif
+
+// Class that handles dumping of variables into files.
+class ApmDataDumper {
+ public:
+// Constructor that takes an instance index that may
+// be used to distinguish data dumped from different
+// instances of the code.
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+  explicit ApmDataDumper(int instance_index)
+      : instance_index_(instance_index) {}
+#else
+  explicit ApmDataDumper(int instance_index) {}
+#endif
+
+  // Reinitializes the data dumping such that new versions
+  // of all files being dumped to are created.
+  void InitiateNewSetOfRecordings() {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    ++recording_set_index_;
+#endif
+  }
+
+  // Methods for performing dumping of data of various types into
+  // various formats.
+  void DumpRaw(const char* name, int v_length, const float* v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    FILE* file = GetRawFile(name);
+    fwrite(v, sizeof(v[0]), v_length, file);
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const float> v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpRaw(const char* name, int v_length, const int16_t* v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    FILE* file = GetRawFile(name);
+    fwrite(v, sizeof(v[0]), v_length, file);
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpRaw(const char* name, int v_length, const int32_t* v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    FILE* file = GetRawFile(name);
+    fwrite(v, sizeof(v[0]), v_length, file);
+#endif
+  }
+
+  void DumpRaw(const char* name, rtc::ArrayView<const int32_t> v) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpWav(const char* name,
+               int v_length,
+               const float* v,
+               int sample_rate_hz,
+               int num_channels) {
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+    WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
+    file->WriteSamples(v, v_length);
+#endif
+  }
+
+ private:
+#if WEBRTC_AEC_DEBUG_DUMP == 1
+  const int instance_index_;
+  int recording_set_index_ = 0;
+  std::unordered_map<std::string, std::unique_ptr<FILE, RawFileCloseFunctor>>
+      raw_files_;
+  std::unordered_map<std::string, std::unique_ptr<WavWriter>> wav_files_;
+
+  FILE* GetRawFile(const char* name);
+  WavWriter* GetWavFile(const char* name, int sample_rate_hz, int num_channels);
+#endif
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ApmDataDumper);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
diff --git a/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc
index 4344c56fcc6..e1c9fdcd019 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #if defined(WEBRTC_NS_FLOAT)
 #include "webrtc/modules/audio_processing/ns/noise_suppression.h"
diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c
index 25f16d26abc..94b6449776f 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -19,7 +19,7 @@
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 
-#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 /* Tables are defined in ARM assembly files. */
 extern const int16_t WebRtcNsx_kLogTable[9];
 extern const int16_t WebRtcNsx_kCounterDiv[201];
@@ -65,7 +65,7 @@ static const int16_t WebRtcNsx_kLogTableFrac[256] = {
   237, 238, 238, 239, 240, 241, 241, 242, 243, 244, 244, 245, 246, 247, 247,
   248, 249, 249, 250, 251, 252, 252, 253, 254, 255, 255
 };
-#endif  // WEBRTC_DETECT_NEON || WEBRTC_HAS_NEON
+#endif  // WEBRTC_HAS_NEON
 
 // Skip first frequency bins during estimation. (0 <= value < 64)
 static const size_t kStartBand = 5;
@@ -557,7 +557,7 @@ AnalysisUpdate WebRtcNsx_AnalysisUpdate;
 Denormalize WebRtcNsx_Denormalize;
 NormalizeRealBuffer WebRtcNsx_NormalizeRealBuffer;
 
-#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 // Initialize function pointers for ARM Neon platform.
 static void WebRtcNsx_InitNeon(void) {
   WebRtcNsx_NoiseEstimation = WebRtcNsx_NoiseEstimationNeon;
@@ -762,12 +762,7 @@ int32_t WebRtcNsx_InitCore(NoiseSuppressionFixedC* inst, uint32_t fs) {
   WebRtcNsx_Denormalize = DenormalizeC;
   WebRtcNsx_NormalizeRealBuffer = NormalizeRealBufferC;
 
-#ifdef WEBRTC_DETECT_NEON
-  uint64_t features = WebRtc_GetCPUFeaturesARM();
-  if ((features & kCPUFeatureNEON) != 0) {
-      WebRtcNsx_InitNeon();
-  }
-#elif defined(WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
   WebRtcNsx_InitNeon();
 #endif
 
diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h
index f463dbbe1a2..d1754f31eae 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core.h
@@ -215,7 +215,7 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
                                uint32_t* priorLocSnr,
                                uint32_t* postLocSnr);
 
-#if (defined WEBRTC_DETECT_NEON || defined WEBRTC_HAS_NEON)
+#if defined(WEBRTC_HAS_NEON)
 // For the above function pointers, functions for generic platforms are declared
 // and defined as static in file nsx_core.c, while those for ARM Neon platforms
 // are declared below and defined in file nsx_core_neon.c.
diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_c.c b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_c.c
index da7aa3d5dbe..213320d38c8 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_c.c
@@ -96,8 +96,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
   }
   tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14
   // compute indicator function: sigmoid map
-  tableIndex = (int16_t)(tmp32no1 >> 14);
-  if ((tableIndex < 16) && (tableIndex >= 0)) {
+  if (tmp32no1 < (16 << 14) && tmp32no1 >= 0) {
+    tableIndex = (int16_t)(tmp32no1 >> 14);
     tmp16no2 = kIndicatorTable[tableIndex];
     tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
     frac = (int16_t)(tmp32no1 & 0x00003fff); // Q14
@@ -128,8 +128,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
     // FLOAT code
     // indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
     //                          (threshPrior1 - tmpFloat1)) + 1.0);
-    tableIndex = (int16_t)(tmpU32no1 >> 14);
-    if (tableIndex < 16) {
+    if (tmpU32no1 < (16 << 14)) {
+      tableIndex = (int16_t)(tmpU32no1 >> 14);
       tmp16no2 = kIndicatorTable[tableIndex];
       tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
       frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14
@@ -175,8 +175,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
     /* FLOAT code
      indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
      */
-    tableIndex = (int16_t)(tmpU32no1 >> 14);
-    if (tableIndex < 16) {
+    if (tmpU32no1 < (16 << 14)) {
+      tableIndex = (int16_t)(tmpU32no1 >> 14);
       tmp16no2 = kIndicatorTable[tableIndex];
       tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
       frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14
diff --git a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_mips.c b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_mips.c
index 7688d82d78d..3922308c7c8 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_mips.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/ns/nsx_core_mips.c
@@ -131,8 +131,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
   }
   tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14
   // compute indicator function: sigmoid map
-  tableIndex = (int16_t)(tmp32no1 >> 14);
-  if ((tableIndex < 16) && (tableIndex >= 0)) {
+  if (tmp32no1 < (16 << 14) && tmp32no1 >= 0) {
+    tableIndex = (int16_t)(tmp32no1 >> 14);
     tmp16no2 = kIndicatorTable[tableIndex];
     tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
     frac = (int16_t)(tmp32no1 & 0x00003fff); // Q14
@@ -163,8 +163,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
     // FLOAT code
     // indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
     //                          (threshPrior1 - tmpFloat1)) + 1.0);
-    tableIndex = (int16_t)(tmpU32no1 >> 14);
-    if (tableIndex < 16) {
+    if (tmpU32no1 < (16 << 14)) {
+      tableIndex = (int16_t)(tmpU32no1 >> 14);
       tmp16no2 = kIndicatorTable[tableIndex];
       tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
       frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14
@@ -210,8 +210,8 @@ void WebRtcNsx_SpeechNoiseProb(NoiseSuppressionFixedC* inst,
     /* FLOAT code
      indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
      */
-    tableIndex = (int16_t)(tmpU32no1 >> 14);
-    if (tableIndex < 16) {
+    if (tmpU32no1 < (16 << 14)) {
+      tableIndex = (int16_t)(tmpU32no1 >> 14);
       tmp16no2 = kIndicatorTable[tableIndex];
       tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
       frac = (int16_t)(tmpU32no1 & 0x00003fff); // Q14
diff --git a/chromium/third_party/webrtc/modules/audio_processing/test/audio_file_processor.h b/chromium/third_party/webrtc/modules/audio_processing/test/audio_file_processor.h
index f7cde598210..76d5e0edb8f 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/test/audio_file_processor.h
+++ b/chromium/third_party/webrtc/modules/audio_processing/test/audio_file_processor.h
@@ -16,11 +16,11 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
@@ -33,9 +33,9 @@ namespace webrtc {
 // Holds a few statistics about a series of TickIntervals.
 struct TickIntervalStats {
   TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
-  TickInterval sum;
-  TickInterval max;
-  TickInterval min;
+  int64_t sum;
+  int64_t max;
+  int64_t min;
 };
 
 // Interface for processing an input file with an AudioProcessing instance and
@@ -60,10 +60,10 @@ class AudioFileProcessor {
   class ScopedTimer {
    public:
     explicit ScopedTimer(TickIntervalStats* proc_time)
-        : proc_time_(proc_time), start_time_(TickTime::Now()) {}
+      : proc_time_(proc_time), start_time_(rtc::TimeNanos()) {}
 
     ~ScopedTimer() {
-      TickInterval interval = TickTime::Now() - start_time_;
+      int64_t interval = rtc::TimeNanos() - start_time_;
       proc_time_->sum += interval;
       proc_time_->max = std::max(proc_time_->max, interval);
       proc_time_->min = std::min(proc_time_->min, interval);
@@ -71,7 +71,7 @@ class AudioFileProcessor {
 
    private:
     TickIntervalStats* const proc_time_;
-    TickTime start_time_;
+    int64_t start_time_;
   };
 
   TickIntervalStats* mutable_proc_time() { return &proc_time_; }
diff --git a/chromium/third_party/webrtc/modules/audio_processing/test/audioproc_float.cc b/chromium/third_party/webrtc/modules/audio_processing/test/audioproc_float.cc
index 41e45bfdc69..33790d837f4 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -25,7 +25,6 @@
 #include "webrtc/modules/audio_processing/test/audio_file_processor.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 
 namespace {
@@ -167,13 +166,14 @@ int main(int argc, char* argv[]) {
 
   if (FLAGS_perf) {
     const auto& proc_time = processor->proc_time();
-    int64_t exec_time_us = proc_time.sum.Microseconds();
+    int64_t exec_time_us = proc_time.sum / rtc::kNumNanosecsPerMicrosec;
     printf(
         "\nExecution time: %.3f s, File time: %.2f s\n"
         "Time per chunk (mean, max, min):\n%.0f us, %.0f us, %.0f us\n",
         exec_time_us * 1e-6, num_chunks * 1.f / kChunksPerSecond,
-        exec_time_us * 1.f / num_chunks, 1.f * proc_time.max.Microseconds(),
-        1.f * proc_time.min.Microseconds());
+        exec_time_us * 1.f / num_chunks,
+        1.f * proc_time.max / rtc::kNumNanosecsPerMicrosec,
+        1.f * proc_time.min / rtc::kNumNanosecsPerMicrosec);
   }
 
   return 0;
diff --git a/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_replayer.cc b/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
index fc127e610ed..fa76747c2be 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
@@ -187,6 +187,10 @@ void DebugDumpReplayer::MaybeRecreateApm(const audioproc::Config& msg) {
   config.Set<ExtendedFilter>(
       new ExtendedFilter(msg.aec_extended_filter_enabled()));
 
+  RTC_CHECK(msg.has_intelligibility_enhancer_enabled());
+  config.Set<Intelligibility>(
+      new Intelligibility(msg.intelligibility_enhancer_enabled()));
+
   // We only create APM once, since changes on these fields should not
   // happen in current implementation.
   if (!apm_.get()) {
diff --git a/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_test.cc b/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_test.cc
index 3acb69444d9..64d659ea500 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -10,6 +10,7 @@
 
 #include <stddef.h>  // size_t
 
+#include <memory>
 #include <string>
 #include <vector>
 
diff --git a/chromium/third_party/webrtc/modules/audio_processing/test/process_test.cc b/chromium/third_party/webrtc/modules/audio_processing/test/process_test.cc
index 185bc142d50..527e0a1e3e5 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/test/process_test.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/test/process_test.cc
@@ -19,13 +19,13 @@
 #include <memory>
 
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
@@ -562,7 +562,7 @@ void void_main(int argc, char* argv[]) {
   int reverse_count = 0;
   int primary_count = 0;
   int near_read_bytes = 0;
-  TickInterval acc_ticks;
+  int64_t acc_nanos = 0;
 
   AudioFrame far_frame;
   AudioFrame near_frame;
@@ -573,8 +573,8 @@ void void_main(int argc, char* argv[]) {
   int8_t stream_has_voice = 0;
   float ns_speech_prob = 0.0f;
 
-  TickTime t0 = TickTime::Now();
-  TickTime t1 = t0;
+  int64_t t0 = rtc::TimeNanos();
+  int64_t t1 = t0;
   int64_t max_time_us = 0;
   int64_t max_time_reverse_us = 0;
   int64_t min_time_us = 1e6;
@@ -676,7 +676,7 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (perf_testing) {
-          t0 = TickTime::Now();
+          t0 = rtc::TimeNanos();
         }
 
         if (msg.has_data()) {
@@ -692,14 +692,15 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (perf_testing) {
-          t1 = TickTime::Now();
-          TickInterval tick_diff = t1 - t0;
-          acc_ticks += tick_diff;
-          if (tick_diff.Microseconds() > max_time_reverse_us) {
-            max_time_reverse_us = tick_diff.Microseconds();
+          t1 = rtc::TimeNanos();
+          int64_t diff_nanos = t1 - t0;
+          acc_nanos += diff_nanos;
+          int64_t diff_us = diff_nanos / rtc::kNumNanosecsPerMicrosec;
+          if (diff_us > max_time_reverse_us) {
+            max_time_reverse_us = diff_us;
           }
-          if (tick_diff.Microseconds() < min_time_reverse_us) {
-            min_time_reverse_us = tick_diff.Microseconds();
+          if (diff_us < min_time_reverse_us) {
+            min_time_reverse_us = diff_us;
           }
         }
 
@@ -737,7 +738,7 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (perf_testing) {
-          t0 = TickTime::Now();
+          t0 = rtc::TimeNanos();
         }
 
         ASSERT_EQ(apm->kNoError,
@@ -795,14 +796,15 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (perf_testing) {
-          t1 = TickTime::Now();
-          TickInterval tick_diff = t1 - t0;
-          acc_ticks += tick_diff;
-          if (tick_diff.Microseconds() > max_time_us) {
-            max_time_us = tick_diff.Microseconds();
+          t1 = rtc::TimeNanos();
+          int64_t diff_nanos = t1 - t0;
+          acc_nanos += diff_nanos;
+          int64_t diff_us = diff_nanos / rtc::kNumNanosecsPerMicrosec;
+          if (diff_us > max_time_us) {
+            max_time_us = diff_us;
           }
-          if (tick_diff.Microseconds() < min_time_us) {
-            min_time_us = tick_diff.Microseconds();
+          if (diff_us < min_time_us) {
+            min_time_us = diff_us;
           }
         }
 
@@ -925,21 +927,22 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (perf_testing) {
-          t0 = TickTime::Now();
+          t0 = rtc::TimeNanos();
         }
 
         ASSERT_EQ(apm->kNoError,
                   apm->ProcessReverseStream(&far_frame));
 
         if (perf_testing) {
-          t1 = TickTime::Now();
-          TickInterval tick_diff = t1 - t0;
-          acc_ticks += tick_diff;
-          if (tick_diff.Microseconds() > max_time_reverse_us) {
-            max_time_reverse_us = tick_diff.Microseconds();
+          t1 = rtc::TimeNanos();
+          int64_t diff_nanos = t1 - t0;
+          acc_nanos += diff_nanos;
+          int64_t diff_us = diff_nanos / rtc::kNumNanosecsPerMicrosec;
+          if (diff_us > max_time_reverse_us) {
+            max_time_reverse_us = diff_us;
           }
-          if (tick_diff.Microseconds() < min_time_reverse_us) {
-            min_time_reverse_us = tick_diff.Microseconds();
+          if (diff_us < min_time_reverse_us) {
+            min_time_reverse_us = diff_us;
           }
         }
 
@@ -982,7 +985,7 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (perf_testing) {
-          t0 = TickTime::Now();
+          t0 = rtc::TimeNanos();
         }
 
         const int capture_level_in = capture_level;
@@ -1030,14 +1033,15 @@ void void_main(int argc, char* argv[]) {
         }
 
         if (perf_testing) {
-          t1 = TickTime::Now();
-          TickInterval tick_diff = t1 - t0;
-          acc_ticks += tick_diff;
-          if (tick_diff.Microseconds() > max_time_us) {
-            max_time_us = tick_diff.Microseconds();
+          t1 = rtc::TimeNanos();
+          int64_t diff_nanos = t1 - t0;
+          acc_nanos += diff_nanos;
+          int64_t diff_us = diff_nanos / rtc::kNumNanosecsPerMicrosec;
+          if (diff_us > max_time_us) {
+            max_time_us = diff_us;
           }
-          if (tick_diff.Microseconds() < min_time_us) {
-            min_time_us = tick_diff.Microseconds();
+          if (diff_us < min_time_us) {
+            min_time_us = diff_us;
           }
         }
 
@@ -1130,7 +1134,7 @@ void void_main(int argc, char* argv[]) {
 
   if (perf_testing) {
     if (primary_count > 0) {
-      int64_t exec_time = acc_ticks.Milliseconds();
+      int64_t exec_time = acc_nanos / rtc::kNumNanosecsPerMillisec;
       printf("\nTotal time: %.3f s, file time: %.2f s\n",
         exec_time * 0.001, primary_count * 0.01);
       printf("Time per frame: %.3f ms (average), %.3f ms (max),"
diff --git a/chromium/third_party/webrtc/modules/audio_processing/test/unpack.cc b/chromium/third_party/webrtc/modules/audio_processing/test/unpack.cc
index fbb8e85fee7..f5c0700b3f8 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/test/unpack.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/test/unpack.cc
@@ -252,6 +252,7 @@ int do_main(int argc, char* argv[]) {
       PRINT_CONFIG(ns_enabled);
       PRINT_CONFIG(ns_level);
       PRINT_CONFIG(transient_suppression_enabled);
+      PRINT_CONFIG(intelligibility_enhancer_enabled);
       if (msg.has_experiments_description()) {
         fprintf(settings_file, "  experiments_description: %s\n",
                 msg.experiments_description().c_str());
diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.c b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.cc
index 02df75a1010..56bdde890c9 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator.cc
@@ -13,6 +13,7 @@
 #include <assert.h>
 #include <stdlib.h>
 #include <string.h>
+#include <algorithm>
 
 // Number of right shifts for scaling is linearly depending on number of bits in
 // the far-end binary spectrum.
@@ -276,7 +277,8 @@ BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend(
 
   if (history_size > 1) {
     // Sanity conditions fulfilled.
-    self = malloc(sizeof(BinaryDelayEstimatorFarend));
+    self = static_cast<BinaryDelayEstimatorFarend*>(
+        malloc(sizeof(BinaryDelayEstimatorFarend)));
   }
   if (self == NULL) {
     return NULL;
@@ -296,11 +298,12 @@ int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self,
                                       int history_size) {
   assert(self != NULL);
   // (Re-)Allocate memory for history buffers.
-  self->binary_far_history =
+  self->binary_far_history = static_cast<uint32_t*>(
       realloc(self->binary_far_history,
-              history_size * sizeof(*self->binary_far_history));
-  self->far_bit_counts = realloc(self->far_bit_counts,
-                                 history_size * sizeof(*self->far_bit_counts));
+              history_size * sizeof(*self->binary_far_history)));
+  self->far_bit_counts = static_cast<int*>(
+      realloc(self->far_bit_counts,
+              history_size * sizeof(*self->far_bit_counts)));
   if ((self->binary_far_history == NULL) || (self->far_bit_counts == NULL)) {
     history_size = 0;
   }
@@ -404,7 +407,8 @@ BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
 
   if ((farend != NULL) && (max_lookahead >= 0)) {
     // Sanity conditions fulfilled.
-    self = malloc(sizeof(BinaryDelayEstimator));
+    self = static_cast<BinaryDelayEstimator*>(
+        malloc(sizeof(BinaryDelayEstimator)));
   }
   if (self == NULL) {
     return NULL;
@@ -422,8 +426,8 @@ BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
   self->mean_bit_counts = NULL;
   self->bit_counts = NULL;
   self->histogram = NULL;
-  self->binary_near_history =
-      malloc((max_lookahead + 1) * sizeof(*self->binary_near_history));
+  self->binary_near_history = static_cast<uint32_t*>(
+      malloc((max_lookahead + 1) * sizeof(*self->binary_near_history)));
   if (self->binary_near_history == NULL ||
       WebRtc_AllocateHistoryBufferMemory(self, farend->history_size) == 0) {
     WebRtc_FreeBinaryDelayEstimator(self);
@@ -444,13 +448,13 @@ int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self,
   // The extra array element in |mean_bit_counts| and |histogram| is a dummy
   // element only used while |last_delay| == -2, i.e., before we have a valid
   // estimate.
-  self->mean_bit_counts =
+  self->mean_bit_counts = static_cast<int32_t*>(
       realloc(self->mean_bit_counts,
-              (history_size + 1) * sizeof(*self->mean_bit_counts));
-  self->bit_counts =
-      realloc(self->bit_counts, history_size * sizeof(*self->bit_counts));
-  self->histogram =
-      realloc(self->histogram, (history_size + 1) * sizeof(*self->histogram));
+              (history_size + 1) * sizeof(*self->mean_bit_counts)));
+  self->bit_counts = static_cast<int32_t*>(
+      realloc(self->bit_counts, history_size * sizeof(*self->bit_counts)));
+  self->histogram = static_cast<float*>(
+      realloc(self->histogram, (history_size + 1) * sizeof(*self->histogram)));
 
   if ((self->mean_bit_counts == NULL) ||
       (self->bit_counts == NULL) ||
@@ -616,13 +620,10 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
           (value_best_candidate < self->last_delay_probability)));
 
   // Check for nonstationary farend signal.
-  int non_stationary_farend = 0;
-  for (i = 0; i < self->history_size; ++i) {
-    if (self->farend->far_bit_counts[i] > 0) {
-      non_stationary_farend = 1;
-      break;
-    }
-  }
+  const bool non_stationary_farend =
+      std::any_of(self->farend->far_bit_counts,
+                  self->farend->far_bit_counts + self->history_size,
+                  [](int a) { return a > 0; });
 
   if (non_stationary_farend) {
     // Only update the validation statistics when the farend is nonstationary
diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
index 4ebe0e61289..3e46763a6a2 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
@@ -10,11 +10,9 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
-extern "C" {
 #include "webrtc/modules/audio_processing/utility/delay_estimator.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_internal.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-}
 #include "webrtc/typedefs.h"
 
 namespace {
diff --git a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
index b5448bc5bde..75c7abea776 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c
+++ b/chromium/third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/modules/audio_processing/utility/delay_estimator.h"
 #include "webrtc/modules/audio_processing/utility/delay_estimator_internal.h"
-#include "webrtc/system_wrappers/include/compile_assert_c.h"
 
 // Only bit |kBandFirst| through bit |kBandLast| are processed and
 // |kBandFirst| - |kBandLast| must be < 32.
@@ -144,10 +143,11 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) {
 
   // Check if the sub band used in the delay estimation is small enough to fit
   // the binary spectra in a uint32_t.
-  COMPILE_ASSERT(kBandLast - kBandFirst < 32);
+  static_assert(kBandLast - kBandFirst < 32, "");
 
   if (spectrum_size >= kBandLast) {
-    self = malloc(sizeof(DelayEstimatorFarend));
+    self = static_cast<DelayEstimatorFarend*>(
+        malloc(sizeof(DelayEstimatorFarend)));
   }
 
   if (self != NULL) {
@@ -158,7 +158,8 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) {
     memory_fail |= (self->binary_farend == NULL);
 
     // Allocate memory for spectrum buffers.
-    self->mean_far_spectrum = malloc(spectrum_size * sizeof(SpectrumType));
+    self->mean_far_spectrum =
+        static_cast<SpectrumType*>(malloc(spectrum_size * sizeof(SpectrumType)));
     memory_fail |= (self->mean_far_spectrum == NULL);
 
     self->spectrum_size = spectrum_size;
@@ -275,7 +276,7 @@ void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) {
   DelayEstimatorFarend* farend = (DelayEstimatorFarend*) farend_handle;
 
   if (farend_handle != NULL) {
-    self = malloc(sizeof(DelayEstimator));
+    self = static_cast<DelayEstimator*>(malloc(sizeof(DelayEstimator)));
   }
 
   if (self != NULL) {
@@ -287,8 +288,8 @@ void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) {
     memory_fail |= (self->binary_handle == NULL);
 
     // Allocate memory for spectrum buffers.
-    self->mean_near_spectrum = malloc(farend->spectrum_size *
-                                      sizeof(SpectrumType));
+    self->mean_near_spectrum = static_cast<SpectrumType*>(
+        malloc(farend->spectrum_size * sizeof(SpectrumType)));
     memory_fail |= (self->mean_near_spectrum == NULL);
 
     self->spectrum_size = farend->spectrum_size;
@@ -328,7 +329,7 @@ int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift) {
 }
 
 int WebRtc_set_history_size(void* handle, int history_size) {
-  DelayEstimator* self = handle;
+  DelayEstimator* self = static_cast<DelayEstimator*>(handle);
 
   if ((self == NULL) || (history_size <= 1)) {
     return -1;
@@ -337,7 +338,7 @@ int WebRtc_set_history_size(void* handle, int history_size) {
 }
 
 int WebRtc_history_size(const void* handle) {
-  const DelayEstimator* self = handle;
+  const DelayEstimator* self = static_cast<const DelayEstimator*>(handle);
 
   if (self == NULL) {
     return -1;
diff --git a/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc b/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc
index 674a5197a87..5a0d37c2747 100644
--- a/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc
+++ b/chromium/third_party/webrtc/modules/audio_processing/voice_detection_impl.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 
diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
index bf608789d5d..09652d84194 100644
--- a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
+++ b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
@@ -83,6 +83,10 @@ BitrateController* BitrateController::CreateBitrateController(
   return new BitrateControllerImpl(clock, observer);
 }
 
+BitrateController* BitrateController::CreateBitrateController(Clock* clock) {
+  return new BitrateControllerImpl(clock, nullptr);
+}
+
 BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
                                              BitrateObserver* observer)
     : clock_(clock),
@@ -94,8 +98,8 @@ BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
       last_fraction_loss_(0),
       last_rtt_ms_(0),
       last_reserved_bitrate_bps_(0) {
-  // This calls the observer_, which means that the observer provided by the
-  // user must be ready to accept a bitrate update when it constructs the
+  // This calls the observer_ if set, which means that the observer provided by
+  // the user must be ready to accept a bitrate update when it constructs the
   // controller. We do this to avoid having to keep synchronized initial values
   // in both the controller and the allocator.
   MaybeTriggerOnNetworkChanged();
@@ -122,6 +126,18 @@ void BitrateControllerImpl::SetMinMaxBitrate(int min_bitrate_bps,
   MaybeTriggerOnNetworkChanged();
 }
 
+void BitrateControllerImpl::SetBitrates(int start_bitrate_bps,
+                                        int min_bitrate_bps,
+                                        int max_bitrate_bps) {
+  {
+    rtc::CritScope cs(&critsect_);
+    bandwidth_estimation_.SetBitrates(start_bitrate_bps,
+                                      min_bitrate_bps,
+                                      max_bitrate_bps);
+  }
+  MaybeTriggerOnNetworkChanged();
+}
+
 void BitrateControllerImpl::SetReservedBitrate(uint32_t reserved_bitrate_bps) {
   {
     rtc::CritScope cs(&critsect_);
@@ -187,11 +203,15 @@ void BitrateControllerImpl::OnReceivedRtcpReceiverReport(
 }
 
 void BitrateControllerImpl::MaybeTriggerOnNetworkChanged() {
-  uint32_t bitrate;
+  if (!observer_)
+    return;
+
+  uint32_t bitrate_bps;
   uint8_t fraction_loss;
   int64_t rtt;
-  if (GetNetworkParameters(&bitrate, &fraction_loss, &rtt))
-    observer_->OnNetworkChanged(bitrate, fraction_loss, rtt);
+
+  if (GetNetworkParameters(&bitrate_bps, &fraction_loss, &rtt))
+    observer_->OnNetworkChanged(bitrate_bps, fraction_loss, rtt);
 }
 
 bool BitrateControllerImpl::GetNetworkParameters(uint32_t* bitrate,
diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
index 6f776d758b4..5a61379ce01 100644
--- a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
+++ b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
@@ -20,6 +20,7 @@
 #include <list>
 #include <utility>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h"
 
@@ -27,6 +28,8 @@ namespace webrtc {
 
 class BitrateControllerImpl : public BitrateController {
  public:
+  // TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
+  // |observer| is left for project that is not yet updated.
   BitrateControllerImpl(Clock* clock, BitrateObserver* observer);
   virtual ~BitrateControllerImpl() {}
 
@@ -34,15 +37,26 @@ class BitrateControllerImpl : public BitrateController {
 
   RtcpBandwidthObserver* CreateRtcpBandwidthObserver() override;
 
+  // Deprecated
   void SetStartBitrate(int start_bitrate_bps) override;
+  // Deprecated
   void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) override;
 
+  void SetBitrates(int start_bitrate_bps,
+                   int min_bitrate_bps,
+                   int max_bitrate_bps) override;
+
   void UpdateDelayBasedEstimate(uint32_t bitrate_bps) override;
 
   void SetReservedBitrate(uint32_t reserved_bitrate_bps) override;
 
   void SetEventLog(RtcEventLog* event_log) override;
 
+  // Returns true if the parameters have changed since the last call.
+  bool GetNetworkParameters(uint32_t* bitrate,
+                            uint8_t* fraction_loss,
+                            int64_t* rtt) override;
+
   int64_t TimeUntilNextProcess() override;
   void Process() override;
 
@@ -57,20 +71,16 @@ class BitrateControllerImpl : public BitrateController {
                                     int number_of_packets,
                                     int64_t now_ms);
 
+  // Deprecated
   void MaybeTriggerOnNetworkChanged();
 
-  // Returns true if the parameters have changed since the last call.
-  bool GetNetworkParameters(uint32_t* bitrate,
-                            uint8_t* fraction_loss,
-                            int64_t* rtt);
-
   void OnNetworkChanged(uint32_t bitrate,
                         uint8_t fraction_loss,  // 0 - 255.
                         int64_t rtt) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
   // Used by process thread.
-  Clock* clock_;
-  BitrateObserver* observer_;
+  Clock* const clock_;
+  BitrateObserver* const observer_;
   int64_t last_bitrate_update_ms_;
 
   rtc::CriticalSection critsect_;
diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
index 3f467ef8a46..4f92a3884b5 100644
--- a/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/chromium/third_party/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -14,11 +14,16 @@
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
-using webrtc::RtcpBandwidthObserver;
-using webrtc::BitrateObserver;
+using ::testing::Exactly;
+using ::testing::Return;
+
 using webrtc::BitrateController;
+using webrtc::BitrateObserver;
+using webrtc::PacedSender;
+using webrtc::RtcpBandwidthObserver;
 
 uint8_t WeightedLoss(int num_packets1, uint8_t fraction_loss1,
                      int num_packets2, uint8_t fraction_loss2) {
diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index a9c247acf17..a61cf6a7a74 100644
--- a/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/chromium/third_party/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -18,6 +18,7 @@
 #include <map>
 
 #include "webrtc/modules/include/module.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
@@ -26,6 +27,8 @@ class CriticalSectionWrapper;
 class RtcEventLog;
 struct PacketInfo;
 
+// Deprecated
+// TODO(perkj): Remove BitrateObserver when no implementations use it.
 class BitrateObserver {
   // Observer class for bitrate changes announced due to change in bandwidth
   // estimate or due to bitrate allocation changes. Fraction loss and rtt is
@@ -46,16 +49,26 @@ class BitrateController : public Module {
   // estimation and divide the available bitrate between all its registered
   // BitrateObservers.
  public:
-  static const int kDefaultStartBitrateKbps = 300;
+  static const int kDefaultStartBitratebps = 300000;
 
+  // Deprecated:
+  // TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
+  // Remove this method once other other projects does not use it.
   static BitrateController* CreateBitrateController(Clock* clock,
                                                     BitrateObserver* observer);
+  static BitrateController* CreateBitrateController(Clock* clock);
+
   virtual ~BitrateController() {}
 
   virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0;
 
+  // Deprecated
   virtual void SetStartBitrate(int start_bitrate_bps) = 0;
+  // Deprecated
   virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
+  virtual void SetBitrates(int start_bitrate_bps,
+                           int min_bitrate_bps,
+                           int max_bitrate_bps) = 0;
 
   virtual void UpdateDelayBasedEstimate(uint32_t bitrate_bps) = 0;
 
@@ -66,6 +79,10 @@ class BitrateController : public Module {
   virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
 
   virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0;
+
+  virtual bool GetNetworkParameters(uint32_t* bitrate,
+                                    uint8_t* fraction_loss,
+                                    int64_t* rtt) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h b/chromium/third_party/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h
index 45b596a8688..da6169e748e 100644
--- a/chromium/third_party/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h
+++ b/chromium/third_party/webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h
@@ -31,10 +31,16 @@ class MockBitrateController : public BitrateController {
   MOCK_METHOD1(SetStartBitrate, void(int start_bitrate_bps));
   MOCK_METHOD2(SetMinMaxBitrate,
                void(int min_bitrate_bps, int max_bitrate_bps));
+  MOCK_METHOD3(SetBitrates,
+               void(int start_bitrate_bps,
+                    int min_bitrate_bps,
+                    int max_bitrate_bps));
   MOCK_METHOD1(UpdateDelayBasedEstimate, void(uint32_t bitrate_bps));
   MOCK_METHOD1(SetEventLog, void(RtcEventLog* event_log));
   MOCK_CONST_METHOD1(AvailableBandwidth, bool(uint32_t* bandwidth));
   MOCK_METHOD1(SetReservedBitrate, void(uint32_t reserved_bitrate_bps));
+  MOCK_METHOD3(GetNetworkParameters,
+               bool(uint32_t* bitrate, uint8_t* fraction_loss, int64_t* rtt));
 
   MOCK_METHOD0(Process, void());
   MOCK_METHOD0(TimeUntilNextProcess, int64_t());
diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index 785267d8c97..a1b78a257c6 100644
--- a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -67,6 +67,14 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation()
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
+void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
+                                              int min_bitrate,
+                                              int max_bitrate) {
+  if (send_bitrate > 0)
+    SetSendBitrate(send_bitrate);
+  SetMinMaxBitrate(min_bitrate, max_bitrate);
+}
+
 void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
   RTC_DCHECK_GT(bitrate, 0);
   bitrate_ = bitrate;
diff --git a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 15894f93954..402d22a6bf4 100644
--- a/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/chromium/third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -46,6 +46,9 @@ class SendSideBandwidthEstimation {
                            int number_of_packets,
                            int64_t now_ms);
 
+  void SetBitrates(int send_bitrate,
+                   int min_bitrate,
+                   int max_bitrate);
   void SetSendBitrate(int bitrate);
   void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
   int GetMinBitrate() const;
diff --git a/chromium/third_party/webrtc/modules/congestion_controller/congestion_controller.cc b/chromium/third_party/webrtc/modules/congestion_controller/congestion_controller.cc
index 11ce46adf3a..9f95fc3a1f9 100644
--- a/chromium/third_party/webrtc/modules/congestion_controller/congestion_controller.cc
+++ b/chromium/third_party/webrtc/modules/congestion_controller/congestion_controller.cc
@@ -15,11 +15,11 @@
 #include <vector>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
@@ -47,11 +47,10 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
 
   void IncomingPacket(int64_t arrival_time_ms,
                       size_t payload_size,
-                      const RTPHeader& header,
-                      bool was_paced) override {
+                      const RTPHeader& header) override {
     CriticalSectionScoped cs(crit_sect_.get());
     PickEstimatorFromHeader(header);
-    rbe_->IncomingPacket(arrival_time_ms, payload_size, header, was_paced);
+    rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
   }
 
   void Process() override {
@@ -80,7 +79,7 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
     return rbe_->LatestEstimate(ssrcs, bitrate_bps);
   }
 
-  void SetMinBitrate(int min_bitrate_bps) {
+  void SetMinBitrate(int min_bitrate_bps) override {
     CriticalSectionScoped cs(crit_sect_.get());
     rbe_->SetMinBitrate(min_bitrate_bps);
     min_bitrate_bps_ = min_bitrate_bps;
@@ -115,7 +114,7 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
   // Instantiate RBE for Time Offset or Absolute Send Time extensions.
   void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
     if (using_absolute_send_time_) {
-      rbe_.reset(new RemoteBitrateEstimatorAbsSendTime(observer_, clock_));
+      rbe_.reset(new RemoteBitrateEstimatorAbsSendTime(observer_));
     } else {
       rbe_.reset(new RemoteBitrateEstimatorSingleStream(observer_, clock_));
     }
@@ -140,31 +139,78 @@ CongestionController::CongestionController(
     BitrateObserver* bitrate_observer,
     RemoteBitrateObserver* remote_bitrate_observer)
     : clock_(clock),
-      pacer_(new PacedSender(clock_,
-             &packet_router_,
-             BitrateController::kDefaultStartBitrateKbps,
-             PacedSender::kDefaultPaceMultiplier *
-                 BitrateController::kDefaultStartBitrateKbps,
-             0)),
+      observer_(nullptr),
+      packet_router_(new PacketRouter()),
+      pacer_(new PacedSender(clock_, packet_router_.get())),
       remote_bitrate_estimator_(
           new WrappingBitrateEstimator(remote_bitrate_observer, clock_)),
-      // Constructed last as this object calls the provided callback on
-      // construction.
       bitrate_controller_(
           BitrateController::CreateBitrateController(clock_, bitrate_observer)),
-      remote_estimator_proxy_(clock_, &packet_router_),
+      remote_estimator_proxy_(clock_, packet_router_.get()),
+      transport_feedback_adapter_(bitrate_controller_.get(), clock_),
+      min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps),
+      last_reported_bitrate_bps_(0),
+      last_reported_fraction_loss_(0),
+      last_reported_rtt_(0),
+      network_state_(kNetworkUp) {
+  Init();
+}
+
+CongestionController::CongestionController(
+    Clock* clock,
+    Observer* observer,
+    RemoteBitrateObserver* remote_bitrate_observer)
+    : clock_(clock),
+      observer_(observer),
+      packet_router_(new PacketRouter()),
+      pacer_(new PacedSender(clock_, packet_router_.get())),
+      remote_bitrate_estimator_(
+          new WrappingBitrateEstimator(remote_bitrate_observer, clock_)),
+      bitrate_controller_(BitrateController::CreateBitrateController(clock_)),
+      remote_estimator_proxy_(clock_, packet_router_.get()),
+      transport_feedback_adapter_(bitrate_controller_.get(), clock_),
+      min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps),
+      last_reported_bitrate_bps_(0),
+      last_reported_fraction_loss_(0),
+      last_reported_rtt_(0),
+      network_state_(kNetworkUp) {
+  Init();
+}
+
+CongestionController::CongestionController(
+    Clock* clock,
+    Observer* observer,
+    RemoteBitrateObserver* remote_bitrate_observer,
+    std::unique_ptr<PacketRouter> packet_router,
+    std::unique_ptr<PacedSender> pacer)
+    : clock_(clock),
+      observer_(observer),
+      packet_router_(std::move(packet_router)),
+      pacer_(std::move(pacer)),
+      remote_bitrate_estimator_(
+          new WrappingBitrateEstimator(remote_bitrate_observer, clock_)),
+      // Constructed last as this object calls the provided callback on
+      // construction.
+      bitrate_controller_(BitrateController::CreateBitrateController(clock_)),
+      remote_estimator_proxy_(clock_, packet_router_.get()),
       transport_feedback_adapter_(bitrate_controller_.get(), clock_),
-      min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {
+      min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps),
+      last_reported_bitrate_bps_(0),
+      last_reported_fraction_loss_(0),
+      last_reported_rtt_(0),
+      network_state_(kNetworkUp) {
+  Init();
+}
+
+CongestionController::~CongestionController() {}
+
+void CongestionController::Init() {
   transport_feedback_adapter_.SetBitrateEstimator(
-      new RemoteBitrateEstimatorAbsSendTime(&transport_feedback_adapter_,
-                                            clock_));
+      new RemoteBitrateEstimatorAbsSendTime(&transport_feedback_adapter_));
   transport_feedback_adapter_.GetBitrateEstimator()->SetMinBitrate(
       min_bitrate_bps_);
 }
 
-CongestionController::~CongestionController() {
-}
-
 
 void CongestionController::SetBweBitrates(int min_bitrate_bps,
                                           int start_bitrate_bps,
@@ -177,16 +223,19 @@ void CongestionController::SetBweBitrates(int min_bitrate_bps,
     min_bitrate_bps = kMinBitrateBps;
   if (max_bitrate_bps > 0)
     max_bitrate_bps = std::max(min_bitrate_bps, max_bitrate_bps);
-  if (start_bitrate_bps > 0) {
+  if (start_bitrate_bps > 0)
     start_bitrate_bps = std::max(min_bitrate_bps, start_bitrate_bps);
-    bitrate_controller_->SetStartBitrate(start_bitrate_bps);
-  }
-  bitrate_controller_->SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
+
+  bitrate_controller_->SetBitrates(start_bitrate_bps,
+                                   min_bitrate_bps,
+                                   max_bitrate_bps);
+
   if (remote_bitrate_estimator_)
     remote_bitrate_estimator_->SetMinBitrate(min_bitrate_bps);
   min_bitrate_bps_ = min_bitrate_bps;
   transport_feedback_adapter_.GetBitrateEstimator()->SetMinBitrate(
       min_bitrate_bps_);
+  MaybeTriggerOnNetworkChanged();
 }
 
 BitrateController* CongestionController::GetBitrateController() const {
@@ -207,10 +256,9 @@ CongestionController::GetTransportFeedbackObserver() {
   return &transport_feedback_adapter_;
 }
 
-void CongestionController::UpdatePacerBitrate(int bitrate_kbps,
-                                              int max_bitrate_kbps,
-                                              int min_bitrate_kbps) {
-  pacer_->UpdateBitrate(bitrate_kbps, max_bitrate_kbps, min_bitrate_kbps);
+void CongestionController::SetAllocatedSendBitrate(int allocated_bitrate_bps,
+                                                   int padding_bitrate_bps) {
+  pacer_->SetAllocatedSendBitrate(allocated_bitrate_bps, padding_bitrate_bps);
 }
 
 int64_t CongestionController::GetPacerQueuingDelayMs() const {
@@ -223,6 +271,11 @@ void CongestionController::SignalNetworkState(NetworkState state) {
   } else {
     pacer_->Pause();
   }
+  {
+    rtc::CritScope cs(&critsect_);
+    network_state_ = state;
+  }
+  MaybeTriggerOnNetworkChanged();
 }
 
 void CongestionController::OnSentPacket(const rtc::SentPacket& sent_packet) {
@@ -243,6 +296,52 @@ int64_t CongestionController::TimeUntilNextProcess() {
 void CongestionController::Process() {
   bitrate_controller_->Process();
   remote_bitrate_estimator_->Process();
+  MaybeTriggerOnNetworkChanged();
+}
+
+void CongestionController::MaybeTriggerOnNetworkChanged() {
+  // TODO(perkj): |observer_| can be nullptr if the ctor that accepts a
+  // BitrateObserver is used. Remove this check once the ctor is removed.
+  if (!observer_)
+    return;
+
+  uint32_t bitrate_bps;
+  uint8_t fraction_loss;
+  int64_t rtt;
+  bool estimate_changed = bitrate_controller_->GetNetworkParameters(
+      &bitrate_bps, &fraction_loss, &rtt);
+  if (estimate_changed)
+    pacer_->SetEstimatedBitrate(bitrate_bps);
+
+  bitrate_bps = IsNetworkDown() || IsSendQueueFull() ? 0 : bitrate_bps;
+
+  if (HasNetworkParametersToReportChanged(bitrate_bps, fraction_loss, rtt)) {
+    observer_->OnNetworkChanged(bitrate_bps, fraction_loss, rtt);
+  }
+}
+
+bool CongestionController::HasNetworkParametersToReportChanged(
+    uint32_t bitrate_bps,
+    uint8_t fraction_loss,
+    int64_t rtt) {
+  rtc::CritScope cs(&critsect_);
+  bool changed =
+      last_reported_bitrate_bps_ != bitrate_bps ||
+      (bitrate_bps > 0 && (last_reported_fraction_loss_ != fraction_loss ||
+                           last_reported_rtt_ != rtt));
+  last_reported_bitrate_bps_ = bitrate_bps;
+  last_reported_fraction_loss_ = fraction_loss;
+  last_reported_rtt_ = rtt;
+  return changed;
+}
+
+bool CongestionController::IsSendQueueFull() const {
+  return pacer_->ExpectedQueueTimeMs() > PacedSender::kMaxQueueLengthMs;
+}
+
+bool CongestionController::IsNetworkDown() const {
+  rtc::CritScope cs(&critsect_);
+  return network_state_ == kNetworkDown;
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/congestion_controller/congestion_controller_unittest.cc b/chromium/third_party/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
new file mode 100644
index 00000000000..c82c75daf3c
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
+#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
+#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+using testing::_;
+using testing::NiceMock;
+using testing::Return;
+using testing::SaveArg;
+using testing::StrictMock;
+
+namespace webrtc {
+namespace test {
+
+class CongestionControllerTest : public ::testing::Test {
+ protected:
+  CongestionControllerTest() : clock_(123456) {}
+  ~CongestionControllerTest() override {}
+
+  void SetUp() override {
+    pacer_ = new NiceMock<MockPacedSender>();
+    std::unique_ptr<PacedSender> pacer(pacer_);  // Passes ownership.
+    std::unique_ptr<PacketRouter> packet_router(new PacketRouter());
+    controller_.reset(
+        new CongestionController(&clock_, &observer_, &remote_bitrate_observer_,
+                                 std::move(packet_router), std::move(pacer)));
+    bandwidth_observer_.reset(
+        controller_->GetBitrateController()->CreateRtcpBandwidthObserver());
+
+    // Set the initial bitrate estimate and expect the |observer| and |pacer_|
+    // to be updated.
+    EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _));
+    EXPECT_CALL(*pacer_, SetEstimatedBitrate(kInitialBitrateBps));
+    controller_->SetBweBitrates(0, kInitialBitrateBps, 5 * kInitialBitrateBps);
+  }
+
+  SimulatedClock clock_;
+  StrictMock<MockCongestionObserver> observer_;
+  NiceMock<MockPacedSender>* pacer_;
+  NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
+  std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
+  std::unique_ptr<CongestionController> controller_;
+  const uint32_t kInitialBitrateBps = 60000;
+};
+
+TEST_F(CongestionControllerTest, OnNetworkChanged) {
+  // Test no change.
+  clock_.AdvanceTimeMilliseconds(25);
+  controller_->Process();
+
+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 2, _, _));
+  EXPECT_CALL(*pacer_, SetEstimatedBitrate(kInitialBitrateBps * 2));
+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps * 2);
+  clock_.AdvanceTimeMilliseconds(25);
+  controller_->Process();
+
+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _));
+  EXPECT_CALL(*pacer_, SetEstimatedBitrate(kInitialBitrateBps));
+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps);
+  clock_.AdvanceTimeMilliseconds(25);
+  controller_->Process();
+}
+
+TEST_F(CongestionControllerTest, OnSendQueueFull) {
+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
+      .WillOnce(Return(PacedSender::kMaxQueueLengthMs + 1));
+
+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _));
+  controller_->Process();
+
+  // Let the pacer not be full next time the controller checks.
+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
+      .WillOnce(Return(PacedSender::kMaxQueueLengthMs - 1));
+
+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _));
+  controller_->Process();
+}
+
+TEST_F(CongestionControllerTest, OnSendQueueFullAndEstimateChange) {
+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
+      .WillOnce(Return(PacedSender::kMaxQueueLengthMs + 1));
+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _));
+  controller_->Process();
+
+  // Receive new estimate but let the queue still be full.
+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps * 2);
+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
+      .WillOnce(Return(PacedSender::kMaxQueueLengthMs + 1));
+  //  The send pacer should get the new estimate though.
+  EXPECT_CALL(*pacer_, SetEstimatedBitrate(kInitialBitrateBps * 2));
+  clock_.AdvanceTimeMilliseconds(25);
+  controller_->Process();
+
+  // Let the pacer not be full next time the controller checks.
+  // |OnNetworkChanged| should be called with the new estimate.
+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
+      .WillOnce(Return(PacedSender::kMaxQueueLengthMs - 1));
+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 2, _, _));
+  clock_.AdvanceTimeMilliseconds(25);
+  controller_->Process();
+}
+
+TEST_F(CongestionControllerTest, SignalNetworkState) {
+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _));
+  controller_->SignalNetworkState(kNetworkDown);
+
+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _));
+  controller_->SignalNetworkState(kNetworkUp);
+
+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _));
+  controller_->SignalNetworkState(kNetworkDown);
+}
+
+TEST_F(CongestionControllerTest,
+       SignalNetworkStateAndQueueIsFullAndEstimateChange) {
+  // Send queue is full
+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs + 1));
+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _));
+  controller_->Process();
+
+  // Queue is full and network is down. Expect no bitrate change.
+  controller_->SignalNetworkState(kNetworkDown);
+  controller_->Process();
+
+  // Queue is full but network is up. Expect no bitrate change.
+  controller_->SignalNetworkState(kNetworkUp);
+  controller_->Process();
+
+  // Receive new estimate but let the queue still be full.
+  EXPECT_CALL(*pacer_, SetEstimatedBitrate(kInitialBitrateBps * 2));
+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps * 2);
+  clock_.AdvanceTimeMilliseconds(25);
+  controller_->Process();
+
+  // Let the pacer not be full next time the controller checks.
+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
+      .WillOnce(Return(PacedSender::kMaxQueueLengthMs - 1));
+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 2, _, _));
+  controller_->Process();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/congestion_controller/include/congestion_controller.h b/chromium/third_party/webrtc/modules/congestion_controller/include/congestion_controller.h
index 65bf5be3ec1..da8719d33a7 100644
--- a/chromium/third_party/webrtc/modules/congestion_controller/include/congestion_controller.h
+++ b/chromium/third_party/webrtc/modules/congestion_controller/include/congestion_controller.h
@@ -11,13 +11,16 @@
 #ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_CONGESTION_CONTROLLER_H_
 #define WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_CONGESTION_CONTROLLER_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_types.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
 #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
-#include "webrtc/stream.h"
 
 namespace rtc {
 struct SentPacket;
@@ -28,7 +31,6 @@ namespace webrtc {
 class BitrateController;
 class BitrateObserver;
 class Clock;
-class PacedSender;
 class ProcessThread;
 class RemoteBitrateEstimator;
 class RemoteBitrateObserver;
@@ -36,9 +38,33 @@ class TransportFeedbackObserver;
 
 class CongestionController : public CallStatsObserver, public Module {
  public:
+  // Observer class for bitrate changes announced due to change in bandwidth
+  // estimate or due to that the send pacer is full. Fraction loss and rtt is
+  // also part of this callback to allow the observer to optimize its settings
+  // for different types of network environments. The bitrate does not include
+  // packet headers and is measured in bits per second.
+  class Observer {
+   public:
+    virtual void OnNetworkChanged(uint32_t bitrate_bps,
+                                  uint8_t fraction_loss,  // 0 - 255.
+                                  int64_t rtt_ms) = 0;
+
+   protected:
+    virtual ~Observer() {}
+  };
+  // Deprecated
+  // TODO(perkj): Remove once no other clients use this ctor.
   CongestionController(Clock* clock,
                        BitrateObserver* bitrate_observer,
                        RemoteBitrateObserver* remote_bitrate_observer);
+  CongestionController(Clock* clock,
+                       Observer* observer,
+                       RemoteBitrateObserver* remote_bitrate_observer);
+  CongestionController(Clock* clock,
+                       Observer* observer,
+                       RemoteBitrateObserver* remote_bitrate_observer,
+                       std::unique_ptr<PacketRouter> packet_router,
+                       std::unique_ptr<PacedSender> pacer);
   virtual ~CongestionController();
 
   virtual void SetBweBitrates(int min_bitrate_bps,
@@ -50,12 +76,11 @@ class CongestionController : public CallStatsObserver, public Module {
       bool send_side_bwe);
   virtual int64_t GetPacerQueuingDelayMs() const;
   virtual PacedSender* pacer() { return pacer_.get(); }
-  virtual PacketRouter* packet_router() { return &packet_router_; }
+  virtual PacketRouter* packet_router() { return packet_router_.get(); }
   virtual TransportFeedbackObserver* GetTransportFeedbackObserver();
 
-  virtual void UpdatePacerBitrate(int bitrate_kbps,
-                                  int max_bitrate_kbps,
-                                  int min_bitrate_kbps);
+  void SetAllocatedSendBitrate(int allocated_bitrate_bps,
+                               int padding_bitrate_bps);
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet);
 
@@ -67,14 +92,28 @@ class CongestionController : public CallStatsObserver, public Module {
   void Process() override;
 
  private:
+  void Init();
+  void MaybeTriggerOnNetworkChanged();
+
+  bool IsSendQueueFull() const;
+  bool IsNetworkDown() const;
+  bool HasNetworkParametersToReportChanged(uint32_t bitrate_bps,
+                                           uint8_t fraction_loss,
+                                           int64_t rtt);
   Clock* const clock_;
-  const rtc::scoped_ptr<PacedSender> pacer_;
-  const rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
-  const rtc::scoped_ptr<BitrateController> bitrate_controller_;
-  PacketRouter packet_router_;
+  Observer* const observer_;
+  const std::unique_ptr<PacketRouter> packet_router_;
+  const std::unique_ptr<PacedSender> pacer_;
+  const std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+  const std::unique_ptr<BitrateController> bitrate_controller_;
   RemoteEstimatorProxy remote_estimator_proxy_;
   TransportFeedbackAdapter transport_feedback_adapter_;
   int min_bitrate_bps_;
+  rtc::CriticalSection critsect_;
+  uint32_t last_reported_bitrate_bps_ GUARDED_BY(critsect_);
+  uint8_t last_reported_fraction_loss_ GUARDED_BY(critsect_);
+  int64_t last_reported_rtt_ GUARDED_BY(critsect_);
+  NetworkState network_state_ GUARDED_BY(critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CongestionController);
 };
diff --git a/chromium/third_party/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h b/chromium/third_party/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h
index 0813c3d42cd..20955ea81a8 100644
--- a/chromium/third_party/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h
+++ b/chromium/third_party/webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h
@@ -12,20 +12,27 @@
 #define WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_CONGESTION_CONTROLLER_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 
 namespace webrtc {
 namespace test {
 
+class MockCongestionObserver : public CongestionController::Observer {
+ public:
+  MOCK_METHOD3(OnNetworkChanged,
+               void(uint32_t bitrate_bps,
+                    uint8_t fraction_loss,
+                    int64_t rtt_ms));
+};
+
 class MockCongestionController : public CongestionController {
  public:
   MockCongestionController(Clock* clock,
-                           BitrateObserver* bitrate_observer,
+                           Observer* observer,
                            RemoteBitrateObserver* remote_bitrate_observer)
-      : CongestionController(clock,
-                             bitrate_observer,
-                             remote_bitrate_observer) {}
+      : CongestionController(clock, observer, remote_bitrate_observer) {}
   MOCK_METHOD3(SetBweBitrates,
                void(int min_bitrate_bps,
                     int start_bitrate_bps,
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/BUILD.gn b/chromium/third_party/webrtc/modules/desktop_capture/BUILD.gn
index aa33993192a..894d9308e36 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/BUILD.gn
+++ b/chromium/third_party/webrtc/modules/desktop_capture/BUILD.gn
@@ -60,9 +60,7 @@ source_set("desktop_capture") {
     "mouse_cursor_monitor.h",
     "mouse_cursor_monitor_mac.mm",
     "mouse_cursor_monitor_win.cc",
-    "screen_capture_frame_queue.cc",
     "screen_capture_frame_queue.h",
-    "screen_capturer.cc",
     "screen_capturer.h",
     "screen_capturer_helper.cc",
     "screen_capturer_helper.h",
@@ -87,7 +85,6 @@ source_set("desktop_capture") {
     "win/screen_capturer_win_magnifier.h",
     "win/window_capture_utils.cc",
     "win/window_capture_utils.h",
-    "window_capturer.cc",
     "window_capturer.h",
     "window_capturer_mac.mm",
     "window_capturer_win.cc",
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/cropped_desktop_frame.cc b/chromium/third_party/webrtc/modules/desktop_capture/cropped_desktop_frame.cc
index f57fc572b65..733fe9b2adf 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/cropped_desktop_frame.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/cropped_desktop_frame.cc
@@ -12,6 +12,8 @@
 
 #include "webrtc/modules/desktop_capture/cropped_desktop_frame.h"
 
+#include "webrtc/base/constructormagic.h"
+
 namespace webrtc {
 
 // A DesktopFrame that is a sub-rect of another DesktopFrame.
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.cc b/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.cc
index 0dd564f1705..cbe7d96e5d6 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.cc
@@ -32,7 +32,7 @@ void CroppingWindowCapturer::Start(DesktopCapturer::Callback* callback) {
 }
 
 void CroppingWindowCapturer::SetSharedMemoryFactory(
-    rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) {
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
   window_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory));
 }
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.h b/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.h
index 177b5443a31..dfeb447e449 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/cropping_window_capturer.h
@@ -13,7 +13,6 @@
 
 #include <memory>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 #include "webrtc/modules/desktop_capture/window_capturer.h"
@@ -32,7 +31,7 @@ class CroppingWindowCapturer : public WindowCapturer,
   // DesktopCapturer implementation.
   void Start(DesktopCapturer::Callback* callback) override;
   void SetSharedMemoryFactory(
-      rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override;
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
   void Capture(const DesktopRegion& region) override;
   void SetExcludedWindow(WindowId window) override;
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
index 55afedeabf1..4c6e27e561e 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
@@ -12,6 +12,7 @@
 
 #include <string.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_capturer.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
@@ -137,7 +138,7 @@ void DesktopAndCursorComposer::Start(DesktopCapturer::Callback* callback) {
 }
 
 void DesktopAndCursorComposer::SetSharedMemoryFactory(
-    rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) {
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
   desktop_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory));
 }
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
index cd0b2cfbb65..dcbe6129e65 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
@@ -13,7 +13,7 @@
 
 #include <memory>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_capturer.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor_monitor.h"
 
@@ -36,7 +36,7 @@ class DesktopAndCursorComposer : public DesktopCapturer,
   // DesktopCapturer interface.
   void Start(DesktopCapturer::Callback* callback) override;
   void SetSharedMemoryFactory(
-      rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override;
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
   void Capture(const DesktopRegion& region) override;
   void SetExcludedWindow(WindowId window) override;
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi
index b92447c3497..c4fbabf6b24 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capture.gypi
@@ -56,9 +56,7 @@
         "mouse_cursor_monitor_mac.mm",
         "mouse_cursor_monitor_win.cc",
         "mouse_cursor_monitor_x11.cc",
-        "screen_capture_frame_queue.cc",
         "screen_capture_frame_queue.h",
-        "screen_capturer.cc",
         "screen_capturer.h",
         "screen_capturer_helper.cc",
         "screen_capturer_helper.h",
@@ -84,7 +82,6 @@
         "win/screen_capture_utils.h",
         "win/window_capture_utils.cc",
         "win/window_capture_utils.h",
-        "window_capturer.cc",
         "window_capturer.h",
         "window_capturer_mac.mm",
         "window_capturer_win.cc",
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h
index 47f78dc3ff4..ba70e015537 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_capturer.h
@@ -13,7 +13,8 @@
 
 #include <stddef.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/desktop_capture/desktop_capture_types.h"
 #include "webrtc/modules/desktop_capture/shared_memory.h"
 
@@ -28,11 +29,6 @@ class DesktopCapturer {
   // Interface that must be implemented by the DesktopCapturer consumers.
   class Callback {
    public:
-    // Deprecated.
-    // TODO(sergeyu): Remove this method once all references to it are removed
-    // from chromium.
-    virtual SharedMemory* CreateSharedMemory(size_t size) { return nullptr; }
-
     // Called after a frame has been captured. Handler must take ownership of
     // |frame|. If capture has failed for any reason |frame| is set to NULL
     // (e.g. the window has been closed).
@@ -53,7 +49,7 @@ class DesktopCapturer {
   // where Capture() is called. It will be destroyed on the same thread. Shared
   // memory is currently supported only by some DesktopCapturer implementations.
   virtual void SetSharedMemoryFactory(
-      rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) {}
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {}
 
   // Captures next frame. |region| specifies region of the capture target that
   // should be fresh in the resulting frame. The frame may also include fresh
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.cc b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.cc
index 6bc7b2e38f6..3278ed46dcb 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.cc
@@ -84,8 +84,7 @@ std::unique_ptr<DesktopFrame> SharedMemoryDesktopFrame::Create(
   size_t buffer_size =
       size.width() * size.height() * DesktopFrame::kBytesPerPixel;
   std::unique_ptr<SharedMemory> shared_memory;
-  shared_memory = rtc::ScopedToUnique(
-      shared_memory_factory->CreateSharedMemory(buffer_size));
+  shared_memory = shared_memory_factory->CreateSharedMemory(buffer_size);
   if (!shared_memory)
     return nullptr;
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.h
index 3cd839ca1b8..53091239e40 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame.h
@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
 #include "webrtc/modules/desktop_capture/shared_memory.h"
@@ -55,11 +56,6 @@ class DesktopFrame {
   int64_t capture_time_ms() const { return capture_time_ms_; }
   void set_capture_time_ms(int64_t time_ms) { capture_time_ms_ = time_ms; }
 
-  // Optional shape for the frame. Frames may be shaped e.g. if
-  // capturing the contents of a shaped window.
-  const DesktopRegion* shape() const { return shape_.get(); }
-  void set_shape(DesktopRegion* shape) { shape_.reset(shape); }
-
   // Copies pixels from a buffer or another frame. |dest_rect| rect must lay
   // within bounds of this frame.
   void CopyPixelsFrom(uint8_t* src_buffer, int src_stride,
@@ -89,7 +85,6 @@ class DesktopFrame {
   DesktopRegion updated_region_;
   DesktopVector dpi_;
   int64_t capture_time_ms_;
-  std::unique_ptr<DesktopRegion> shape_;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrame);
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.cc b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.cc
index f139fb5cddf..624b729203b 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.cc
@@ -49,10 +49,8 @@ DesktopFrameWin* DesktopFrameWin::Create(
   std::unique_ptr<SharedMemory> shared_memory;
   HANDLE section_handle = nullptr;
   if (shared_memory_factory) {
-    shared_memory = rtc::ScopedToUnique(
-        shared_memory_factory->CreateSharedMemory(buffer_size));
-    if (shared_memory)
-      section_handle = shared_memory->handle();
+    shared_memory = shared_memory_factory->CreateSharedMemory(buffer_size);
+    section_handle = shared_memory->handle();
   }
   void* data = nullptr;
   HBITMAP bitmap = CreateDIBSection(hdc, &bmi, DIB_RGB_COLORS, &data,
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.h
index 929d23c0e56..3513e14ffb7 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_frame_win.h
@@ -15,6 +15,7 @@
 
 #include <windows.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/typedefs.h"
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.cc b/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.cc
index bc9972660ad..e130c103811 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.cc
@@ -20,6 +20,9 @@ DesktopRegion::RowSpan::RowSpan(int32_t left, int32_t right)
     : left(left), right(right) {
 }
 
+DesktopRegion::Row::Row(const Row&) = default;
+DesktopRegion::Row::Row(Row&&) = default;
+
 DesktopRegion::Row::Row(int32_t top, int32_t bottom)
     : top(top), bottom(bottom) {
 }
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h b/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h
index c86da56e173..5278159412a 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/desktop_region.h
@@ -47,6 +47,8 @@ class DesktopRegion {
   // Row represents a single row of a region. A row is set of rectangles that
   // have the same vertical position.
   struct Row {
+    Row(const Row&);
+    Row(Row&&);
     Row(int32_t top, int32_t bottom);
     ~Row();
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/differ.h b/chromium/third_party/webrtc/modules/desktop_capture/differ.h
index c3dcd4b0e1d..9ab059bcaa2 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/differ.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/differ.h
@@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc
index df4e6a45c0a..543910de289 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/differ_unittest.cc
@@ -11,6 +11,7 @@
 #include <memory>
 
 #include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/differ.h"
 #include "webrtc/modules/desktop_capture/differ_block.h"
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
index 2f2dd72a6e5..329beef23d3 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
@@ -16,6 +16,7 @@
 #include <memory>
 #include <set>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/mac/desktop_configuration.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc b/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
index 2d3c2d90479..451acb3db5e 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "webrtc/base/macutils.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/desktop_capture/mac/window_list_utils.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
@@ -141,7 +142,7 @@ bool IsChromeWindow(CGWindowID id) {
 }  // namespace
 
 FullScreenChromeWindowDetector::FullScreenChromeWindowDetector()
-    : ref_count_(0) {}
+    : ref_count_(0), last_update_time_ns_(0) {}
 
 FullScreenChromeWindowDetector::~FullScreenChromeWindowDetector() {}
 
@@ -161,10 +162,7 @@ CGWindowID FullScreenChromeWindowDetector::FindFullScreenWindow(
     if (static_cast<CGWindowID>(it->id) != full_screen_window_id)
       continue;
 
-    int64_t time_interval =
-        (TickTime::Now() - last_udpate_time_).Milliseconds();
-    LOG(LS_WARNING) << "The full-screen window exists in the list, "
-                    << "which was updated " << time_interval << "ms ago.";
+    LOG(LS_WARNING) << "The full-screen window exists in the list.";
     return kCGNullWindowID;
   }
 
@@ -174,7 +172,7 @@ CGWindowID FullScreenChromeWindowDetector::FindFullScreenWindow(
 void FullScreenChromeWindowDetector::UpdateWindowListIfNeeded(
     CGWindowID original_window) {
   if (IsChromeWindow(original_window) &&
-      (TickTime::Now() - last_udpate_time_).Milliseconds()
+      (rtc::TimeNanos() - last_update_time_ns_) / rtc::kNumNanosecsPerMillisec
           > kUpdateIntervalMs) {
     previous_window_list_.clear();
     previous_window_list_.swap(current_window_list_);
@@ -186,7 +184,7 @@ void FullScreenChromeWindowDetector::UpdateWindowListIfNeeded(
     }
 
     GetWindowList(&current_window_list_);
-    last_udpate_time_ = TickTime::Now();
+    last_update_time_ns_ = rtc::TimeNanos();
   }
 }
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h b/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h
index 4e6008966e6..838966d46eb 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h
@@ -13,9 +13,9 @@
 
 #include <ApplicationServices/ApplicationServices.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/window_capturer.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -59,7 +59,7 @@ class FullScreenChromeWindowDetector {
   // full-screen window exists in the list) if Capture() is called too soon.
   WindowCapturer::WindowList current_window_list_;
   WindowCapturer::WindowList previous_window_list_;
-  TickTime last_udpate_time_;
+  int64_t last_update_time_ns_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(FullScreenChromeWindowDetector);
 };
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
index 204bb00b160..479a39a0217 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
@@ -11,10 +11,12 @@
 #include "webrtc/modules/desktop_capture/mouse_cursor_monitor.h"
 
 #include <assert.h>
+#include <string.h>
 
 #include <memory>
 
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
+#include "webrtc/modules/desktop_capture/desktop_geometry.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor.h"
 #include "webrtc/modules/desktop_capture/win/cursor.h"
 #include "webrtc/modules/desktop_capture/win/window_capture_utils.h"
@@ -22,6 +24,17 @@
 
 namespace webrtc {
 
+namespace {
+
+bool IsSameCursorShape(const CURSORINFO& left, const CURSORINFO& right) {
+  // If the cursors are not showing, we do not care the hCursor handle.
+  return left.flags == right.flags &&
+         (left.flags != CURSOR_SHOWING ||
+          left.hCursor == right.hCursor);
+}
+
+}  // namespace
+
 class MouseCursorMonitorWin : public MouseCursorMonitor {
  public:
   explicit MouseCursorMonitorWin(HWND window);
@@ -45,7 +58,8 @@ class MouseCursorMonitorWin : public MouseCursorMonitor {
 
   HDC desktop_dc_;
 
-  HCURSOR last_cursor_;
+  // The last CURSORINFO (converted to MouseCursor) we have sent to the client.
+  CURSORINFO last_cursor_;
 };
 
 MouseCursorMonitorWin::MouseCursorMonitorWin(HWND window)
@@ -53,8 +67,8 @@ MouseCursorMonitorWin::MouseCursorMonitorWin(HWND window)
       screen_(kInvalidScreenId),
       callback_(NULL),
       mode_(SHAPE_AND_POSITION),
-      desktop_dc_(NULL),
-      last_cursor_(NULL) {
+      desktop_dc_(NULL) {
+  memset(&last_cursor_, 0, sizeof(CURSORINFO));
 }
 
 MouseCursorMonitorWin::MouseCursorMonitorWin(ScreenId screen)
@@ -62,9 +76,9 @@ MouseCursorMonitorWin::MouseCursorMonitorWin(ScreenId screen)
       screen_(screen),
       callback_(NULL),
       mode_(SHAPE_AND_POSITION),
-      desktop_dc_(NULL),
-      last_cursor_(NULL) {
+      desktop_dc_(NULL) {
   assert(screen >= kFullDesktopScreenId);
+  memset(&last_cursor_, 0, sizeof(CURSORINFO));
 }
 
 MouseCursorMonitorWin::~MouseCursorMonitorWin() {
@@ -92,13 +106,31 @@ void MouseCursorMonitorWin::Capture() {
     return;
   }
 
-  if (last_cursor_ != cursor_info.hCursor) {
-    last_cursor_ = cursor_info.hCursor;
-    // Note that |cursor_info.hCursor| does not need to be freed.
-    std::unique_ptr<MouseCursor> cursor(
-        CreateMouseCursorFromHCursor(desktop_dc_, cursor_info.hCursor));
-    if (cursor.get())
-      callback_->OnMouseCursor(cursor.release());
+  if (!IsSameCursorShape(cursor_info, last_cursor_)) {
+    if (cursor_info.flags == CURSOR_SUPPRESSED) {
+      // The cursor is intentionally hidden now, send an empty bitmap.
+      last_cursor_ = cursor_info;
+      callback_->OnMouseCursor(new MouseCursor(
+          new BasicDesktopFrame(DesktopSize()), DesktopVector()));
+    } else {
+      // According to MSDN https://goo.gl/u6gyuC, HCURSOR instances returned by
+      // functions other than CreateCursor do not need to be actively destroyed.
+      // And CloseHandle function (https://goo.gl/ja5ycW) does not close a
+      // cursor, so assume a HCURSOR does not need to be closed.
+      if (cursor_info.flags == 0) {
+        // Host machine does not have a hardware mouse attached, we will send a
+        // default one instead.
+        // Note, Windows automatically caches cursor resource, so we do not need
+        // to cache the result of LoadCursor.
+        cursor_info.hCursor = LoadCursor(nullptr, IDC_ARROW);
+      }
+      std::unique_ptr<MouseCursor> cursor(
+          CreateMouseCursorFromHCursor(desktop_dc_, cursor_info.hCursor));
+      if (cursor) {
+        last_cursor_ = cursor_info;
+        callback_->OnMouseCursor(cursor.release());
+      }
+    }
   }
 
   if (mode_ != SHAPE_AND_POSITION)
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_shape.h b/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_shape.h
deleted file mode 100644
index 57120a0b3fa..00000000000
--- a/chromium/third_party/webrtc/modules/desktop_capture/mouse_cursor_shape.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_SHAPE_H_
-#define WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_SHAPE_H_
-
-// This file is no longer needed, but some code in chromium still includes it.
-// TODO(sergeyu): Cleanup dependencies in chromium and remove this file.
-
-#endif  // WEBRTC_MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_SHAPE_H_
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc
deleted file mode 100644
index 94d8a27b137..00000000000
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.cc
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
-
-#include <assert.h>
-#include <algorithm>
-
-#include "webrtc/modules/desktop_capture/desktop_frame.h"
-#include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
-#include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-ScreenCaptureFrameQueue::ScreenCaptureFrameQueue() : current_(0) {}
-
-ScreenCaptureFrameQueue::~ScreenCaptureFrameQueue() {}
-
-void ScreenCaptureFrameQueue::MoveToNextFrame() {
-  current_ = (current_ + 1) % kQueueLength;
-
-  // Verify that the frame is not shared, i.e. that consumer has released it
-  // before attempting to capture again.
-  assert(!frames_[current_].get() || !frames_[current_]->IsShared());
-}
-
-void ScreenCaptureFrameQueue::ReplaceCurrentFrame(DesktopFrame* frame) {
-  frames_[current_].reset(SharedDesktopFrame::Wrap(frame));
-}
-
-void ScreenCaptureFrameQueue::Reset() {
-  for (int i = 0; i < kQueueLength; ++i)
-    frames_[i].reset();
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.h b/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
index 21af0f320fc..97f3b810e9d 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
@@ -13,12 +13,12 @@
 
 #include <memory>
 
-#include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
-#include "webrtc/typedefs.h"
+#include "webrtc/base/constructormagic.h"
+// TODO(zijiehe): These headers are not used in this file, but to avoid build
+// break in remoting/host. We should add headers in each individual files.
+#include "webrtc/modules/desktop_capture/desktop_frame.h"  // Remove
+#include "webrtc/modules/desktop_capture/shared_desktop_frame.h"  // Remove
 
-namespace webrtc {
-class DesktopFrame;
-}  // namespace webrtc
 
 namespace webrtc {
 
@@ -35,28 +35,38 @@ namespace webrtc {
 // Frame consumer is expected to never hold more than kQueueLength frames
 // created by this function and it should release the earliest one before trying
 // to capture a new frame (i.e. before MoveToNextFrame() is called).
+template <typename FrameType>
 class ScreenCaptureFrameQueue {
  public:
-  ScreenCaptureFrameQueue();
-  ~ScreenCaptureFrameQueue();
+  ScreenCaptureFrameQueue() : current_(0) {}
+  ~ScreenCaptureFrameQueue() = default;
 
   // Moves to the next frame in the queue, moving the 'current' frame to become
   // the 'previous' one.
-  void MoveToNextFrame();
+  void MoveToNextFrame() {
+    current_ = (current_ + 1) % kQueueLength;
+  }
 
   // Replaces the current frame with a new one allocated by the caller. The
   // existing frame (if any) is destroyed. Takes ownership of |frame|.
-  void ReplaceCurrentFrame(DesktopFrame* frame);
+  void ReplaceCurrentFrame(FrameType* frame) {
+    frames_[current_].reset(frame);
+  }
 
   // Marks all frames obsolete and resets the previous frame pointer. No
   // frames are freed though as the caller can still access them.
-  void Reset();
+  void Reset() {
+    for (int i = 0; i < kQueueLength; i++) {
+      frames_[i].reset();
+    }
+    current_ = 0;
+  }
 
-  SharedDesktopFrame* current_frame() const {
+  FrameType* current_frame() const {
     return frames_[current_].get();
   }
 
-  SharedDesktopFrame* previous_frame() const {
+  FrameType* previous_frame() const {
     return frames_[(current_ + kQueueLength - 1) % kQueueLength].get();
   }
 
@@ -65,7 +75,7 @@ class ScreenCaptureFrameQueue {
   int current_;
 
   static const int kQueueLength = 2;
-  std::unique_ptr<SharedDesktopFrame> frames_[kQueueLength];
+  std::unique_ptr<FrameType> frames_[kQueueLength];
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCaptureFrameQueue);
 };
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.cc
deleted file mode 100644
index 97f69d3baff..00000000000
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.cc
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/desktop_capture/screen_capturer.h"
-
-#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
-
-namespace webrtc {
-
-ScreenCapturer* ScreenCapturer::Create() {
-  return Create(DesktopCaptureOptions::CreateDefault());
-}
-
-#if defined(WEBRTC_LINUX)
-ScreenCapturer* ScreenCapturer::CreateWithXDamage(
-    bool use_update_notifications) {
-  DesktopCaptureOptions options;
-  options.set_use_update_notifications(use_update_notifications);
-  return Create(options);
-}
-#elif defined(WEBRTC_WIN)
-ScreenCapturer* ScreenCapturer::CreateWithDisableAero(bool disable_effects) {
-  DesktopCaptureOptions options;
-  options.set_disable_effects(disable_effects);
-  return Create(options);
-}
-#endif
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h
index b4e34887662..48ecc31fbd8 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer.h
@@ -48,34 +48,10 @@ class ScreenCapturer : public DesktopCapturer {
   };
   typedef std::vector<Screen> ScreenList;
 
-  // TODO(sergeyu): Remove this class once all dependencies are removed from
-  // chromium.
-  class MouseShapeObserver {
-  };
-
   virtual ~ScreenCapturer() {}
 
-  // Creates platform-specific capturer.
-  //
-  // TODO(sergeyu): Remove all Create() methods except the first one.
-  // crbug.com/172183
+  // Creates a platform-specific capturer.
   static ScreenCapturer* Create(const DesktopCaptureOptions& options);
-  static ScreenCapturer* Create();
-
-#if defined(WEBRTC_LINUX)
-  // Creates platform-specific capturer and instructs it whether it should use
-  // X DAMAGE support.
-  static ScreenCapturer* CreateWithXDamage(bool use_x_damage);
-#elif defined(WEBRTC_WIN)
-  // Creates Windows-specific capturer and instructs it whether or not to
-  // disable desktop compositing.
-  static ScreenCapturer* CreateWithDisableAero(bool disable_aero);
-#endif  // defined(WEBRTC_WIN)
-
-  // TODO(sergeyu): Remove this method once all dependencies are removed from
-  // chromium.
-  virtual void SetMouseShapeObserver(
-      MouseShapeObserver* mouse_shape_observer) {};
 
   // Get the list of screens (not containing kFullDesktopScreenId). Returns
   // false in case of a failure.
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.h b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.h
index f912378333c..458bccc1ed3 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_helper.h
@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm
index c41dc4d7a3b..bf6c72950c2 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac.mm
@@ -22,7 +22,10 @@
 #include <OpenGL/CGLMacro.h>
 #include <OpenGL/OpenGL.h>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/macutils.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
@@ -32,8 +35,8 @@
 #include "webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h"
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
+#include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
 #include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -234,7 +237,7 @@ class ScreenCapturerMac : public ScreenCapturer {
   ScopedPixelBufferObject pixel_buffer_object_;
 
   // Queue of the frames buffers.
-  ScreenCaptureFrameQueue queue_;
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
 
   // Current display configuration.
   MacDesktopConfiguration desktop_config_;
@@ -381,9 +384,10 @@ void ScreenCapturerMac::Start(Callback* callback) {
 }
 
 void ScreenCapturerMac::Capture(const DesktopRegion& region_to_capture) {
-  TickTime capture_start_time = TickTime::Now();
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
 
   queue_.MoveToNextFrame();
+  RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
 
   desktop_config_monitor_->Lock();
   MacDesktopConfiguration new_config =
@@ -405,7 +409,7 @@ void ScreenCapturerMac::Capture(const DesktopRegion& region_to_capture) {
   // Note that we can't reallocate other buffers at this point, since the caller
   // may still be reading from them.
   if (!queue_.current_frame())
-    queue_.ReplaceCurrentFrame(CreateFrame());
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(CreateFrame()));
 
   DesktopFrame* current_frame = queue_.current_frame();
 
@@ -444,7 +448,8 @@ void ScreenCapturerMac::Capture(const DesktopRegion& region_to_capture) {
   desktop_config_monitor_->Unlock();
 
   new_frame->set_capture_time_ms(
-      (TickTime::Now() - capture_start_time).Milliseconds());
+      (rtc::TimeNanos() - capture_start_time_nanos) /
+      rtc::kNumNanosecsPerMillisec);
   callback_->OnCaptureCompleted(new_frame);
 }
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
index 64d649cd9d1..815c7f53afe 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
@@ -16,6 +16,7 @@
 #include <ostream>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
@@ -38,7 +39,10 @@ class ScreenCapturerMacTest : public testing::Test {
   void CaptureDoneCallback2(DesktopFrame* frame);
 
  protected:
-  void SetUp() override { capturer_.reset(ScreenCapturer::Create()); }
+  void SetUp() override {
+    capturer_.reset(
+        ScreenCapturer::Create(DesktopCaptureOptions::CreateDefault()));
+  }
 
   std::unique_ptr<ScreenCapturer> capturer_;
   MockScreenCapturerCallback callback_;
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h
index b1f64e41c97..7264249e0fb 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_mock_objects.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_MOCK_OBJECTS_H_
 
 #include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
index 72105acca47..bc87ed3eba8 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
@@ -14,6 +14,7 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_region.h"
@@ -59,8 +60,8 @@ class FakeSharedMemoryFactory : public SharedMemoryFactory {
   FakeSharedMemoryFactory() {}
   ~FakeSharedMemoryFactory() override {}
 
-  rtc::scoped_ptr<SharedMemory> CreateSharedMemory(size_t size) override {
-    return rtc::scoped_ptr<SharedMemory>(
+  std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) override {
+    return std::unique_ptr<SharedMemory>(
         new FakeSharedMemory(new char[size], size));
   }
 
@@ -117,7 +118,7 @@ TEST_F(ScreenCapturerTest, UseSharedBuffers) {
 
   capturer_->Start(&callback_);
   capturer_->SetSharedMemoryFactory(
-      rtc::scoped_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory()));
+      std::unique_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory()));
   capturer_->Capture(DesktopRegion());
 
   ASSERT_TRUE(frame);
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc
index 65e682b6f8b..5540e6820fa 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/screen_capturer_x11.cc
@@ -21,14 +21,16 @@
 #include <X11/Xutil.h>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/differ.h"
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
+#include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
 #include "webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h"
 #include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace {
@@ -106,7 +108,7 @@ class ScreenCapturerLinux : public ScreenCapturer,
   ScreenCapturerHelper helper_;
 
   // Queue of the frames buffers.
-  ScreenCaptureFrameQueue queue_;
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
 
   // Invalid region from the previous capture. This is used to synchronize the
   // current with the last buffer used.
@@ -234,9 +236,10 @@ void ScreenCapturerLinux::Start(Callback* callback) {
 }
 
 void ScreenCapturerLinux::Capture(const DesktopRegion& region) {
-  TickTime capture_start_time = TickTime::Now();
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
 
   queue_.MoveToNextFrame();
+  RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
 
   // Process XEvents for XDamage and cursor shape tracking.
   options_.x_display()->ProcessPendingXEvents();
@@ -256,7 +259,7 @@ void ScreenCapturerLinux::Capture(const DesktopRegion& region) {
   if (!queue_.current_frame()) {
     std::unique_ptr<DesktopFrame> frame(
         new BasicDesktopFrame(x_server_pixel_buffer_.window_size()));
-    queue_.ReplaceCurrentFrame(frame.release());
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(frame.release()));
   }
 
   // Refresh the Differ helper used by CaptureFrame(), if needed.
@@ -274,7 +277,8 @@ void ScreenCapturerLinux::Capture(const DesktopRegion& region) {
   DesktopFrame* result = CaptureScreen();
   last_invalid_region_ = result->updated_region();
   result->set_capture_time_ms(
-      (TickTime::Now() - capture_start_time).Milliseconds());
+      (rtc::TimeNanos() - capture_start_time_nanos) /
+      rtc::kNumNanosecsPerMillisec);
   callback_->OnCaptureCompleted(result);
 }
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.cc b/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.cc
index 309bac55add..8d10827e29c 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.cc
@@ -12,6 +12,7 @@
 
 #include <memory>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
@@ -48,8 +49,7 @@ class SharedDesktopFrame::Core {
 SharedDesktopFrame::~SharedDesktopFrame() {}
 
 // static
-SharedDesktopFrame* SharedDesktopFrame::Wrap(
-    DesktopFrame* desktop_frame) {
+SharedDesktopFrame* SharedDesktopFrame::Wrap(DesktopFrame* desktop_frame) {
   rtc::scoped_refptr<Core> core(new Core(desktop_frame));
   return new SharedDesktopFrame(core);
 }
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.h b/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.h
index 7d18db153cd..4f6a2bb7c69 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/shared_desktop_frame.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_SHARED_DESKTOP_FRAME_H_
 #define WEBRTC_MODULES_DESKTOP_CAPTURE_SHARED_DESKTOP_FRAME_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h b/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h
index 45f531e0d53..6e15f23f6b8 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/shared_memory.h
@@ -17,8 +17,9 @@
 #include <windows.h>
 #endif
 
+#include <memory>
+
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -69,7 +70,7 @@ class SharedMemoryFactory {
   SharedMemoryFactory() {}
   virtual ~SharedMemoryFactory() {}
 
-  virtual rtc::scoped_ptr<SharedMemory> CreateSharedMemory(size_t size) = 0;
+  virtual std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) = 0;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(SharedMemoryFactory);
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
index d3035a15ca5..9df2e5fc9b2 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
@@ -14,6 +14,8 @@
 
 #include <utility>
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
@@ -24,7 +26,6 @@
 #include "webrtc/modules/desktop_capture/win/desktop.h"
 #include "webrtc/modules/desktop_capture/win/screen_capture_utils.h"
 #include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -36,22 +37,6 @@ const UINT DWM_EC_ENABLECOMPOSITION = 1;
 
 const wchar_t kDwmapiLibraryName[] = L"dwmapi.dll";
 
-// SharedMemoryFactory that creates SharedMemory using the deprecated
-// DesktopCapturer::Callback::CreateSharedMemory().
-class CallbackSharedMemoryFactory : public SharedMemoryFactory {
- public:
-  CallbackSharedMemoryFactory(DesktopCapturer::Callback* callback)
-      : callback_(callback) {}
-  ~CallbackSharedMemoryFactory() override {}
-
-  rtc::scoped_ptr<SharedMemory> CreateSharedMemory(size_t size) override {
-    return rtc::scoped_ptr<SharedMemory>(callback_->CreateSharedMemory(size));
-  }
-
- private:
-  DesktopCapturer::Callback* callback_;
-};
-
 }  // namespace
 
 ScreenCapturerWinGdi::ScreenCapturerWinGdi(const DesktopCaptureOptions& options)
@@ -89,15 +74,15 @@ ScreenCapturerWinGdi::~ScreenCapturerWinGdi() {
 }
 
 void ScreenCapturerWinGdi::SetSharedMemoryFactory(
-    rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) {
-  shared_memory_factory_ =
-      rtc::ScopedToUnique(std::move(shared_memory_factory));
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  shared_memory_factory_ = std::move(shared_memory_factory);
 }
 
 void ScreenCapturerWinGdi::Capture(const DesktopRegion& region) {
-  TickTime capture_start_time = TickTime::Now();
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
 
   queue_.MoveToNextFrame();
+  RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
 
   // Request that the system not power-down the system, or the display hardware.
   if (!SetThreadExecutionState(ES_DISPLAY_REQUIRED | ES_SYSTEM_REQUIRED)) {
@@ -152,7 +137,8 @@ void ScreenCapturerWinGdi::Capture(const DesktopRegion& region) {
   frame->mutable_updated_region()->Clear();
   helper_.TakeInvalidRegion(frame->mutable_updated_region());
   frame->set_capture_time_ms(
-      (TickTime::Now() - capture_start_time).Milliseconds());
+      (rtc::TimeNanos() - capture_start_time_nanos) /
+      rtc::kNumNanosecsPerMillisec);
   callback_->OnCaptureCompleted(frame);
 }
 
@@ -172,8 +158,6 @@ void ScreenCapturerWinGdi::Start(Callback* callback) {
   assert(callback);
 
   callback_ = callback;
-  if (!shared_memory_factory_)
-    shared_memory_factory_.reset(new CallbackSharedMemoryFactory(callback));
 
   // Vote to disable Aero composited desktop effects while capturing. Windows
   // will restore Aero automatically if the process exits. This has no effect
@@ -265,9 +249,9 @@ bool ScreenCapturerWinGdi::CaptureImage() {
 
     std::unique_ptr<DesktopFrame> buffer(DesktopFrameWin::Create(
         size, shared_memory_factory_.get(), desktop_dc_));
-    if (!buffer.get())
+    if (!buffer)
       return false;
-    queue_.ReplaceCurrentFrame(buffer.release());
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(buffer.release()));
   }
 
   // Select the target bitmap into the memory dc and copy the rect from desktop
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
index 17cb0aa1940..5a50580e690 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
@@ -17,9 +17,10 @@
 
 #include <windows.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
+#include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
 #include "webrtc/modules/desktop_capture/win/scoped_thread_desktop.h"
 
 namespace webrtc {
@@ -37,7 +38,7 @@ class ScreenCapturerWinGdi : public ScreenCapturer {
   // Overridden from ScreenCapturer:
   void Start(Callback* callback) override;
   void SetSharedMemoryFactory(
-      rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override;
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
   void Capture(const DesktopRegion& region) override;
   bool GetScreenList(ScreenList* screens) override;
   bool SelectScreen(ScreenId id) override;
@@ -71,7 +72,7 @@ class ScreenCapturerWinGdi : public ScreenCapturer {
   HDC memory_dc_;
 
   // Queue of the frames buffers.
-  ScreenCaptureFrameQueue queue_;
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
 
   // Rectangle describing the bounds of the desktop device context, relative to
   // the primary display's top-left.
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
index 8af9779ce80..6b4308bb7a3 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
@@ -14,6 +14,7 @@
 
 #include <utility>
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
@@ -24,7 +25,6 @@
 #include "webrtc/modules/desktop_capture/win/desktop.h"
 #include "webrtc/modules/desktop_capture/win/screen_capture_utils.h"
 #include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -82,13 +82,12 @@ void ScreenCapturerWinMagnifier::Start(Callback* callback) {
 }
 
 void ScreenCapturerWinMagnifier::SetSharedMemoryFactory(
-    rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) {
-  shared_memory_factory_ =
-      rtc::ScopedToUnique(std::move(shared_memory_factory));
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  shared_memory_factory_ = std::move(shared_memory_factory);
 }
 
 void ScreenCapturerWinMagnifier::Capture(const DesktopRegion& region) {
-  TickTime capture_start_time = TickTime::Now();
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
 
   queue_.MoveToNextFrame();
 
@@ -169,7 +168,8 @@ void ScreenCapturerWinMagnifier::Capture(const DesktopRegion& region) {
   frame->mutable_updated_region()->Clear();
   helper_.TakeInvalidRegion(frame->mutable_updated_region());
   frame->set_capture_time_ms(
-      (TickTime::Now() - capture_start_time).Milliseconds());
+      (rtc::TimeNanos() - capture_start_time_nanos) /
+      rtc::kNumNanosecsPerMillisec);
   callback_->OnCaptureCompleted(frame);
 }
 
@@ -433,7 +433,7 @@ void ScreenCapturerWinMagnifier::CreateCurrentFrameIfNecessary(
             ? SharedMemoryDesktopFrame::Create(size,
                                                shared_memory_factory_.get())
             : std::unique_ptr<DesktopFrame>(new BasicDesktopFrame(size));
-    queue_.ReplaceCurrentFrame(frame.release());
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(frame.release()));
   }
 }
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
index d5e3946d627..623c8a30034 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
@@ -18,10 +18,10 @@
 #include <wincodec.h>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
+#include "webrtc/modules/desktop_capture/shared_desktop_frame.h"
 #include "webrtc/modules/desktop_capture/win/scoped_thread_desktop.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 
@@ -47,7 +47,7 @@ class ScreenCapturerWinMagnifier : public ScreenCapturer {
   // Overridden from ScreenCapturer:
   void Start(Callback* callback) override;
   void SetSharedMemoryFactory(
-      rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override;
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
   void Capture(const DesktopRegion& region) override;
   bool GetScreenList(ScreenList* screens) override;
   bool SelectScreen(ScreenId id) override;
@@ -118,7 +118,7 @@ class ScreenCapturerWinMagnifier : public ScreenCapturer {
   ScreenCapturerHelper helper_;
 
   // Queue of the frames buffers.
-  ScreenCaptureFrameQueue queue_;
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
 
   // Class to calculate the difference between two screen bitmaps.
   std::unique_ptr<Differ> differ_;
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.h b/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.h
index 7c80490f609..85bca4c8246 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/win/window_capture_utils.h
@@ -10,6 +10,7 @@
 
 #include <windows.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.cc
deleted file mode 100644
index c5176d5e605..00000000000
--- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.cc
+++ /dev/null
@@ -1,22 +0,0 @@
-  /*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/desktop_capture/window_capturer.h"
-
-#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
-
-namespace webrtc {
-
-// static
-WindowCapturer* WindowCapturer::Create() {
-  return Create(DesktopCaptureOptions::CreateDefault());
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h
index 9ba441a8ecb..eb9b9feaf04 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer.h
@@ -38,9 +38,6 @@ class WindowCapturer : public DesktopCapturer {
 
   static WindowCapturer* Create(const DesktopCaptureOptions& options);
 
-  // TODO(sergeyu): Remove this method. crbug.com/172183
-  static WindowCapturer* Create();
-
   virtual ~WindowCapturer() {}
 
   // Get list of windows. Returns false in case of a failure.
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.mm b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.mm
index 22061edbe7a..ac5fdb6bc1a 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.mm
+++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_mac.mm
@@ -15,6 +15,7 @@
 #include <Cocoa/Cocoa.h>
 #include <CoreFoundation/CoreFoundation.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/macutils.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
@@ -23,7 +24,6 @@
 #include "webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.h"
 #include "webrtc/modules/desktop_capture/mac/window_list_utils.h"
 #include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc
index b74f17e39bf..5f32c3d71cc 100755
--- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_null.cc
@@ -12,6 +12,7 @@
 
 #include <assert.h>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc
index 0d594a2a096..702324372bd 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_win.cc
@@ -15,6 +15,7 @@
 #include <memory>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/win32.h"
 #include "webrtc/modules/desktop_capture/desktop_frame_win.h"
 #include "webrtc/modules/desktop_capture/win/window_capture_utils.h"
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc
index 68e1725db31..8ead98109a8 100755
--- a/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc
+++ b/chromium/third_party/webrtc/modules/desktop_capture/window_capturer_x11.cc
@@ -19,6 +19,7 @@
 
 #include <algorithm>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/desktop_frame.h"
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/x11/shared_x_display.h b/chromium/third_party/webrtc/modules/desktop_capture/x11/shared_x_display.h
index d905b9e51c1..6cade2decc0 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/x11/shared_x_display.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/x11/shared_x_display.h
@@ -19,6 +19,7 @@
 
 #include <string>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 
diff --git a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
index d1e6632f082..b868b044a81 100644
--- a/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
+++ b/chromium/third_party/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
@@ -13,6 +13,7 @@
 #ifndef WEBRTC_MODULES_DESKTOP_CAPTURE_X11_X_SERVER_PIXEL_BUFFER_H_
 #define WEBRTC_MODULES_DESKTOP_CAPTURE_X11_X_SERVER_PIXEL_BUFFER_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/desktop_capture/desktop_geometry.h"
 
 #include <X11/Xutil.h>
diff --git a/chromium/third_party/webrtc/modules/include/module_common_types.h b/chromium/third_party/webrtc/modules/include/module_common_types.h
index 82d87d5c5c7..3572cd6fc5c 100644
--- a/chromium/third_party/webrtc/modules/include/module_common_types.h
+++ b/chromium/third_party/webrtc/modules/include/module_common_types.h
@@ -71,9 +71,10 @@ struct RTPVideoHeaderVP8 {
 };
 
 enum TemporalStructureMode {
-  kTemporalStructureMode1,    // 1 temporal layer structure - i.e., IPPP...
-  kTemporalStructureMode2,    // 2 temporal layers 0-1-0-1...
-  kTemporalStructureMode3     // 3 temporal layers 0-2-1-2-0-2-1-2...
+  kTemporalStructureMode1,  // 1 temporal layer structure - i.e., IPPP...
+  kTemporalStructureMode2,  // 2 temporal layers 01...
+  kTemporalStructureMode3,  // 3 temporal layers 0212...
+  kTemporalStructureMode4   // 3 temporal layers 02120212...
 };
 
 struct GofInfoVP9 {
@@ -121,6 +122,52 @@ struct GofInfoVP9 {
         pid_diff[3][0] = 1;
         pid_diff[3][1] = 2;
         break;
+      case kTemporalStructureMode4:
+        num_frames_in_gof = 8;
+        temporal_idx[0] = 0;
+        temporal_up_switch[0] = false;
+        num_ref_pics[0] = 1;
+        pid_diff[0][0] = 4;
+
+        temporal_idx[1] = 2;
+        temporal_up_switch[1] = true;
+        num_ref_pics[1] = 1;
+        pid_diff[1][0] = 1;
+
+        temporal_idx[2] = 1;
+        temporal_up_switch[2] = true;
+        num_ref_pics[2] = 1;
+        pid_diff[2][0] = 2;
+
+        temporal_idx[3] = 2;
+        temporal_up_switch[3] = false;
+        num_ref_pics[3] = 2;
+        pid_diff[3][0] = 1;
+        pid_diff[3][1] = 2;
+
+        temporal_idx[4] = 0;
+        temporal_up_switch[0] = false;
+        num_ref_pics[4] = 1;
+        pid_diff[4][0] = 4;
+
+        temporal_idx[5] = 2;
+        temporal_up_switch[1] = false;
+        num_ref_pics[5] = 2;
+        pid_diff[5][0] = 1;
+        pid_diff[5][1] = 2;
+
+        temporal_idx[6] = 1;
+        temporal_up_switch[2] = false;
+        num_ref_pics[6] = 2;
+        pid_diff[6][0] = 2;
+        pid_diff[6][1] = 4;
+
+        temporal_idx[7] = 2;
+        temporal_up_switch[3] = false;
+        num_ref_pics[7] = 2;
+        pid_diff[7][0] = 1;
+        pid_diff[7][1] = 2;
+        break;
       default:
         assert(false);
     }
@@ -143,6 +190,7 @@ struct GofInfoVP9 {
   bool temporal_up_switch[kMaxVp9FramesInGof];
   uint8_t num_ref_pics[kMaxVp9FramesInGof];
   uint8_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
+  uint16_t pid_start;
 };
 
 struct RTPVideoHeaderVP9 {
@@ -432,7 +480,6 @@ enum FecMaskType {
 // Struct containing forward error correction settings.
 struct FecProtectionParams {
   int fec_rate;
-  bool use_uep_protection;
   int max_fec_frames;
   FecMaskType fec_mask_type;
 };
@@ -447,25 +494,6 @@ class CallStatsObserver {
   virtual ~CallStatsObserver() {}
 };
 
-struct VideoContentMetrics {
-  VideoContentMetrics()
-      : motion_magnitude(0.0f),
-        spatial_pred_err(0.0f),
-        spatial_pred_err_h(0.0f),
-        spatial_pred_err_v(0.0f) {}
-
-  void Reset() {
-    motion_magnitude = 0.0f;
-    spatial_pred_err = 0.0f;
-    spatial_pred_err_h = 0.0f;
-    spatial_pred_err_v = 0.0f;
-  }
-  float motion_magnitude;
-  float spatial_pred_err;
-  float spatial_pred_err_h;
-  float spatial_pred_err_v;
-};
-
 /* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
  * allows for adding and subtracting frames while keeping track of the resulting
  * states.
diff --git a/chromium/third_party/webrtc/modules/media_file/media_file_impl.cc b/chromium/third_party/webrtc/modules/media_file/media_file_impl.cc
index 76bcca74d2c..27fe9613a3b 100644
--- a/chromium/third_party/webrtc/modules/media_file/media_file_impl.cc
+++ b/chromium/third_party/webrtc/modules/media_file/media_file_impl.cc
@@ -14,7 +14,6 @@
 #include "webrtc/modules/media_file/media_file_impl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/modules.gyp b/chromium/third_party/webrtc/modules/modules.gyp
index d5df8539051..e11bbfe88b3 100644
--- a/chromium/third_party/webrtc/modules/modules.gyp
+++ b/chromium/third_party/webrtc/modules/modules.gyp
@@ -26,7 +26,6 @@
     'video_coding/video_coding.gypi',
     'video_capture/video_capture.gypi',
     'video_processing/video_processing.gypi',
-    'video_render/video_render.gypi',
   ],
   'conditions': [
     ['include_tests==1', {
@@ -64,7 +63,6 @@
             '<(webrtc_root)/common.gyp:webrtc_common',
             '<(webrtc_root)/common_video/common_video.gyp:common_video',
             '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
-            '<(webrtc_root)/modules/video_coding/codecs/vp9/vp9.gyp:webrtc_vp9',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/test/metrics.gyp:metrics',
             '<(webrtc_root)/test/test.gyp:test_support',
@@ -86,7 +84,6 @@
             'audio_coding/test/PCMFile.cc',
             'audio_coding/test/PacketLossTest.cc',
             'audio_coding/test/RTPFile.cc',
-            'audio_coding/test/SpatialAudio.cc',
             'audio_coding/test/TestAllCodecs.cc',
             'audio_coding/test/TestRedFec.cc',
             'audio_coding/test/TestStereo.cc',
@@ -133,6 +130,7 @@
             'audio_processing',
             'audioproc_test_utils',
             'bitrate_controller',
+            'builtin_audio_decoder_factory',
             'bwe_simulator',
             'cng',
             'isac_fix',
@@ -167,7 +165,7 @@
             '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
           ],
           'sources': [
-            'audio_coding/codecs/audio_encoder_unittest.cc',
+            'audio_coding/codecs/audio_decoder_factory_unittest.cc',
             'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
             'audio_coding/acm2/acm_receiver_unittest_oldapi.cc',
             'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc',
@@ -214,6 +212,7 @@
             'audio_coding/neteq/post_decode_vad_unittest.cc',
             'audio_coding/neteq/random_vector_unittest.cc',
             'audio_coding/neteq/sync_buffer_unittest.cc',
+            'audio_coding/neteq/tick_timer_unittest.cc',
             'audio_coding/neteq/timestamp_scaler_unittest.cc',
             'audio_coding/neteq/time_stretch_unittest.cc',
             'audio_coding/neteq/mock/mock_audio_decoder.h',
@@ -270,6 +269,7 @@
             'audio_processing/vad/voice_activity_detector_unittest.cc',
             'bitrate_controller/bitrate_controller_unittest.cc',
             'bitrate_controller/send_side_bandwidth_estimation_unittest.cc',
+            'congestion_controller/congestion_controller_unittest.cc',
             'media_file/media_file_unittest.cc',
             'module_common_types_unittest.cc',
             'pacing/bitrate_prober_unittest.cc',
@@ -338,6 +338,7 @@
             'rtp_rtcp/source/rtp_format_vp8_unittest.cc',
             'rtp_rtcp/source/rtp_format_vp9_unittest.cc',
             'rtp_rtcp/source/rtp_packet_history_unittest.cc',
+            'rtp_rtcp/source/rtp_packet_unittest.cc',
             'rtp_rtcp/source/rtp_payload_registry_unittest.cc',
             'rtp_rtcp/source/rtp_rtcp_impl_unittest.cc',
             'rtp_rtcp/source/rtp_header_extension_unittest.cc',
@@ -361,9 +362,8 @@
             'video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc',
             'video_coding/codecs/vp8/simulcast_unittest.cc',
             'video_coding/codecs/vp8/simulcast_unittest.h',
-            'video_coding/codecs/vp9/screenshare_layers_unittest.cc',
+            'video_coding/frame_buffer2_unittest.cc',
             'video_coding/include/mock/mock_vcm_callbacks.h',
-            'video_coding/bitrate_adjuster_unittest.cc',
             'video_coding/decoding_state_unittest.cc',
             'video_coding/histogram_unittest.cc',
             'video_coding/jitter_buffer_unittest.cc',
@@ -379,19 +379,21 @@
             'video_coding/video_coding_robustness_unittest.cc',
             'video_coding/video_receiver_unittest.cc',
             'video_coding/video_sender_unittest.cc',
-            'video_coding/qm_select_unittest.cc',
             'video_coding/test/stream_generator.cc',
             'video_coding/test/stream_generator.h',
             'video_coding/utility/frame_dropper_unittest.cc',
+            'video_coding/utility/ivf_file_writer_unittest.cc',
             'video_coding/utility/quality_scaler_unittest.cc',
-            'video_processing/test/brightness_detection_test.cc',
-            'video_processing/test/content_metrics_test.cc',
-            'video_processing/test/deflickering_test.cc',
             'video_processing/test/denoiser_test.cc',
             'video_processing/test/video_processing_unittest.cc',
             'video_processing/test/video_processing_unittest.h',
           ],
           'conditions': [
+            ['libvpx_build_vp9==1', {
+              'sources': [
+                'video_coding/codecs/vp9/screenshare_layers_unittest.cc',
+              ],
+            }],
             ['enable_bwe_test_logging==1', {
               'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
             }, {
@@ -444,6 +446,7 @@
               'sources': [
                 'audio_processing/audio_processing_impl_locking_unittest.cc',
                 'audio_processing/audio_processing_impl_unittest.cc',
+                'audio_processing/audio_processing_unittest.cc',
                 'audio_processing/echo_control_mobile_unittest.cc',
                 'audio_processing/echo_cancellation_unittest.cc',
                 'audio_processing/gain_control_unittest.cc',
@@ -453,7 +456,6 @@
                 'audio_processing/voice_detection_unittest.cc',
                 'audio_processing/test/audio_buffer_tools.cc',
                 'audio_processing/test/audio_buffer_tools.h',
-                'audio_processing/test/audio_processing_unittest.cc',
                 'audio_processing/test/bitexactness_tools.cc',
                 'audio_processing/test/bitexactness_tools.h',
                 'audio_processing/test/debug_dump_replayer.cc',
@@ -504,16 +506,6 @@
                 '<(DEPTH)/data/voice_engine/audio_tiny48.wav',
                 '<(DEPTH)/resources/att-downlink.rx',
                 '<(DEPTH)/resources/att-uplink.rx',
-                '<(DEPTH)/resources/audio_coding/neteq4_network_stats.dat',
-                '<(DEPTH)/resources/audio_coding/neteq4_opus_network_stats.dat',
-                '<(DEPTH)/resources/audio_coding/neteq4_opus_ref.pcm',
-                '<(DEPTH)/resources/audio_coding/neteq4_opus_ref_win_32.pcm',
-                '<(DEPTH)/resources/audio_coding/neteq4_opus_ref_win_64.pcm',
-                '<(DEPTH)/resources/audio_coding/neteq4_opus_rtcp_stats.dat',
-                '<(DEPTH)/resources/audio_coding/neteq4_rtcp_stats.dat',
-                '<(DEPTH)/resources/audio_coding/neteq4_universal_ref.pcm',
-                '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_32.pcm',
-                '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_64.pcm',
                 '<(DEPTH)/resources/audio_coding/neteq_opus.rtp',
                 '<(DEPTH)/resources/audio_coding/neteq_universal_new.rtp',
                 '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
@@ -626,10 +618,17 @@
         ['OS=="android"', {
           'targets': [
             {
-              'target_name': 'modules_unittests_apk_target',
+              'target_name': 'audio_codec_speed_tests_apk_target',
               'type': 'none',
               'dependencies': [
-                '<(apk_tests_path):modules_unittests_apk',
+                '<(apk_tests_path):audio_codec_speed_tests_apk',
+              ],
+            },
+            {
+              'target_name': 'audio_decoder_unittests_apk_target',
+              'type': 'none',
+              'dependencies': [
+                '<(apk_tests_path):audio_decoder_unittests_apk',
               ],
             },
             {
@@ -639,8 +638,75 @@
                 '<(apk_tests_path):modules_tests_apk',
               ],
             },
+            {
+              'target_name': 'modules_unittests_apk_target',
+              'type': 'none',
+              'dependencies': [
+                '<(apk_tests_path):modules_unittests_apk',
+              ],
+            },
           ],
-        }],
+          'conditions': [
+            ['test_isolation_mode != "noop"',
+              {
+                'targets': [
+                  {
+                    'target_name': 'audio_codec_speed_tests_apk_run',
+                    'type': 'none',
+                    'dependencies': [
+                      '<(apk_tests_path):audio_codec_speed_tests_apk',
+                    ],
+                    'includes': [
+                      '../build/isolate.gypi',
+                    ],
+                    'sources': [
+                      'audio_codec_speed_tests_apk.isolate',
+                    ],
+                  },
+                  {
+                    'target_name': 'audio_decoder_unittests_apk_run',
+                    'type': 'none',
+                    'dependencies': [
+                      '<(apk_tests_path):audio_decoder_unittests_apk',
+                    ],
+                    'includes': [
+                      '../build/isolate.gypi',
+                    ],
+                    'sources': [
+                      'audio_decoder_unittests_apk.isolate',
+                    ],
+                  },
+                  {
+                    'target_name': 'modules_tests_apk_run',
+                    'type': 'none',
+                    'dependencies': [
+                      '<(apk_tests_path):modules_tests_apk',
+                    ],
+                    'includes': [
+                      '../build/isolate.gypi',
+                    ],
+                    'sources': [
+                      'modules_tests_apk.isolate',
+                    ],
+                  },
+                  {
+                    'target_name': 'modules_unittests_apk_run',
+                    'type': 'none',
+                    'dependencies': [
+                      '<(apk_tests_path):modules_unittests_apk',
+                    ],
+                    'includes': [
+                      '../build/isolate.gypi',
+                    ],
+                    'sources': [
+                      'modules_unittests_apk.isolate',
+                    ],
+                  },
+                ],
+              },
+            ],
+          ],
+        }],  # OS=="android"
         ['test_isolation_mode != "noop"', {
           'targets': [
             {
@@ -708,19 +774,6 @@
                 'modules_unittests.isolate',
               ],
             },
-            {
-              'target_name': 'video_render_tests_run',
-              'type': 'none',
-              'dependencies': [
-                'video_render_tests',
-              ],
-              'includes': [
-                '../build/isolate.gypi',
-              ],
-              'sources': [
-                'video_render_tests.isolate',
-              ],
-            },
           ],
         }],
       ],
diff --git a/chromium/third_party/webrtc/modules/modules_java.gyp b/chromium/third_party/webrtc/modules/modules_java.gyp
index 060de2a0678..2a72fb30a4c 100644
--- a/chromium/third_party/webrtc/modules/modules_java.gyp
+++ b/chromium/third_party/webrtc/modules/modules_java.gyp
@@ -18,14 +18,5 @@
 
       'includes': [ '../../build/java.gypi' ],
     }, # audio_device_module_java
-    {
-      'target_name': 'video_render_module_java',
-      'type': 'none',
-      'variables': {
-        'java_in_dir': 'video_render/android/java',
-        'additional_src_dirs': [ '../base/java/src', ],
-      },
-      'includes': [ '../../build/java.gypi' ],
-    }, # video_render_module_java
   ],
 }
diff --git a/chromium/third_party/webrtc/modules/modules_java_chromium.gyp b/chromium/third_party/webrtc/modules/modules_java_chromium.gyp
index 32d2d8d24e8..ebc53d60ff8 100644
--- a/chromium/third_party/webrtc/modules/modules_java_chromium.gyp
+++ b/chromium/third_party/webrtc/modules/modules_java_chromium.gyp
@@ -16,13 +16,5 @@
       },
       'includes': [ '../../../build/java.gypi' ],
     }, # audio_device_module_java
-    {
-      'target_name': 'video_render_module_java',
-      'type': 'none',
-      'variables': {
-        'java_in_dir': 'video_render/android/java',
-      },
-      'includes': [ '../../../build/java.gypi' ],
-    }, # video_render_module_java
   ],
 }
diff --git a/chromium/third_party/webrtc/modules/modules_tests_apk.isolate b/chromium/third_party/webrtc/modules/modules_tests_apk.isolate
new file mode 100644
index 00000000000..ffdd967b4fc
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/modules_tests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'modules_tests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_modules_tests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/modules_tests_apk/',
+      '<(PRODUCT_DIR)/bin/run_modules_tests',
+      'modules_tests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/modules/modules_unittests.isolate b/chromium/third_party/webrtc/modules/modules_unittests.isolate
index a7fc4e8661f..af7e6ef46e8 100644
--- a/chromium/third_party/webrtc/modules/modules_unittests.isolate
+++ b/chromium/third_party/webrtc/modules/modules_unittests.isolate
@@ -11,9 +11,6 @@
       'variables': {
         'files': [
           '<(DEPTH)/data/audio_processing/output_data_fixed.pb',
-          '<(DEPTH)/resources/audio_coding/neteq4_network_stats_android.dat',
-          '<(DEPTH)/resources/audio_coding/neteq4_rtcp_stats_android.dat',
-          '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_android.pcm',
         ],
       },
     }],
@@ -25,16 +22,6 @@
           '<(DEPTH)/data/voice_engine/audio_tiny48.wav',
           '<(DEPTH)/resources/att-downlink.rx',
           '<(DEPTH)/resources/att-uplink.rx',
-          '<(DEPTH)/resources/audio_coding/neteq4_network_stats.dat',
-          '<(DEPTH)/resources/audio_coding/neteq4_opus_network_stats.dat',
-          '<(DEPTH)/resources/audio_coding/neteq4_opus_ref.pcm',
-          '<(DEPTH)/resources/audio_coding/neteq4_opus_ref_win_32.pcm',
-          '<(DEPTH)/resources/audio_coding/neteq4_opus_ref_win_64.pcm',
-          '<(DEPTH)/resources/audio_coding/neteq4_opus_rtcp_stats.dat',
-          '<(DEPTH)/resources/audio_coding/neteq4_rtcp_stats.dat',
-          '<(DEPTH)/resources/audio_coding/neteq4_universal_ref.pcm',
-          '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_32.pcm',
-          '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_win_64.pcm',
           '<(DEPTH)/resources/audio_coding/neteq_opus.rtp',
           '<(DEPTH)/resources/audio_coding/neteq_universal_new.rtp',
           '<(DEPTH)/resources/audio_coding/speech_mono_16kHz.pcm',
diff --git a/chromium/third_party/webrtc/modules/modules_unittests_apk.isolate b/chromium/third_party/webrtc/modules/modules_unittests_apk.isolate
new file mode 100644
index 00000000000..cd60cf75c86
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/modules_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'modules_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_modules_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/modules_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_modules_unittests',
+      'modules_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/modules/pacing/bitrate_prober.cc b/chromium/third_party/webrtc/modules/pacing/bitrate_prober.cc
index fbd9b817419..8e8e36ea34b 100644
--- a/chromium/third_party/webrtc/modules/pacing/bitrate_prober.cc
+++ b/chromium/third_party/webrtc/modules/pacing/bitrate_prober.cc
@@ -15,13 +15,14 @@
 #include <limits>
 #include <sstream>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 
 namespace webrtc {
 
 namespace {
-int ComputeDeltaFromBitrate(size_t packet_size, int bitrate_bps) {
+int ComputeDeltaFromBitrate(size_t packet_size, uint32_t bitrate_bps) {
   assert(bitrate_bps > 0);
   // Compute the time delta needed to send packet_size bytes at bitrate_bps
   // bps. Result is in milliseconds.
@@ -33,8 +34,8 @@ int ComputeDeltaFromBitrate(size_t packet_size, int bitrate_bps) {
 BitrateProber::BitrateProber()
     : probing_state_(kDisabled),
       packet_size_last_send_(0),
-      time_last_send_ms_(-1) {
-}
+      time_last_send_ms_(-1),
+      next_cluster_id_(0) {}
 
 void BitrateProber::SetEnabled(bool enable) {
   if (enable) {
@@ -52,7 +53,7 @@ bool BitrateProber::IsProbing() const {
   return probing_state_ == kProbing;
 }
 
-void BitrateProber::OnIncomingPacket(int bitrate_bps,
+void BitrateProber::OnIncomingPacket(uint32_t bitrate_bps,
                                      size_t packet_size,
                                      int64_t now_ms) {
   // Don't initialize probing unless we have something large enough to start
@@ -61,24 +62,24 @@ void BitrateProber::OnIncomingPacket(int bitrate_bps,
     return;
   if (probing_state_ != kAllowedToProbe)
     return;
-  probe_bitrates_.clear();
   // Max number of packets used for probing.
   const int kMaxNumProbes = 2;
   const int kPacketsPerProbe = 5;
   const float kProbeBitrateMultipliers[kMaxNumProbes] = {3, 6};
-  int bitrates_bps[kMaxNumProbes];
   std::stringstream bitrate_log;
-  bitrate_log << "Start probing for bandwidth, bitrates:";
+  bitrate_log << "Start probing for bandwidth, (bitrate:packets): ";
   for (int i = 0; i < kMaxNumProbes; ++i) {
-    bitrates_bps[i] = kProbeBitrateMultipliers[i] * bitrate_bps;
-    bitrate_log << " " << bitrates_bps[i];
-    // We need one extra to get 5 deltas for the first probe.
-    if (i == 0)
-      probe_bitrates_.push_back(bitrates_bps[i]);
-    for (int j = 0; j < kPacketsPerProbe; ++j)
-      probe_bitrates_.push_back(bitrates_bps[i]);
+    ProbeCluster cluster;
+    // We need one extra to get 5 deltas for the first probe, therefore (i == 0)
+    cluster.max_probe_packets = kPacketsPerProbe + (i == 0 ? 1 : 0);
+    cluster.probe_bitrate_bps = kProbeBitrateMultipliers[i] * bitrate_bps;
+    cluster.id = next_cluster_id_++;
+
+    bitrate_log << "(" << cluster.probe_bitrate_bps << ":"
+                << cluster.max_probe_packets << ") ";
+
+    clusters_.push(cluster);
   }
-  bitrate_log << ", num packets: " << probe_bitrates_.size();
   LOG(LS_INFO) << bitrate_log.str().c_str();
   // Set last send time to current time so TimeUntilNextProbe doesn't short
   // circuit due to inactivity.
@@ -87,10 +88,11 @@ void BitrateProber::OnIncomingPacket(int bitrate_bps,
 }
 
 int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
-  if (probing_state_ != kDisabled && probe_bitrates_.empty()) {
+  if (probing_state_ != kDisabled && clusters_.empty()) {
     probing_state_ = kWait;
   }
-  if (probe_bitrates_.empty() || time_last_send_ms_ == -1) {
+
+  if (clusters_.empty() || time_last_send_ms_ == -1) {
     // No probe started, probe finished, or too long since last probe packet.
     return -1;
   }
@@ -107,8 +109,8 @@ int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
   // sent before.
   int time_until_probe_ms = 0;
   if (packet_size_last_send_ != 0 && probing_state_ == kProbing) {
-    int next_delta_ms = ComputeDeltaFromBitrate(packet_size_last_send_,
-                                                probe_bitrates_.front());
+    int next_delta_ms = ComputeDeltaFromBitrate(
+        packet_size_last_send_, clusters_.front().probe_bitrate_bps);
     time_until_probe_ms = next_delta_ms - elapsed_time_ms;
     // There is no point in trying to probe with less than 1 ms between packets
     // as it essentially means trying to probe at infinite bandwidth.
@@ -129,6 +131,12 @@ int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
   return std::max(time_until_probe_ms, 0);
 }
 
+int BitrateProber::CurrentClusterId() const {
+  RTC_DCHECK(!clusters_.empty());
+  RTC_DCHECK_EQ(kProbing, probing_state_);
+  return clusters_.front().id;
+}
+
 size_t BitrateProber::RecommendedPacketSize() const {
   return packet_size_last_send_;
 }
@@ -141,7 +149,11 @@ void BitrateProber::PacketSent(int64_t now_ms, size_t packet_size) {
   time_last_send_ms_ = now_ms;
   if (probing_state_ != kProbing)
     return;
-  if (!probe_bitrates_.empty())
-    probe_bitrates_.pop_front();
+  if (!clusters_.empty()) {
+    ProbeCluster* cluster = &clusters_.front();
+    ++cluster->sent_probe_packets;
+    if (cluster->sent_probe_packets == cluster->max_probe_packets)
+      clusters_.pop();
+  }
 }
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/pacing/bitrate_prober.h b/chromium/third_party/webrtc/modules/pacing/bitrate_prober.h
index 84fbc522fc9..e8967abde45 100644
--- a/chromium/third_party/webrtc/modules/pacing/bitrate_prober.h
+++ b/chromium/third_party/webrtc/modules/pacing/bitrate_prober.h
@@ -13,6 +13,7 @@
 
 #include <cstddef>
 #include <list>
+#include <queue>
 
 #include "webrtc/typedefs.h"
 
@@ -34,12 +35,17 @@ class BitrateProber {
   // Initializes a new probing session if the prober is allowed to probe. Does
   // not initialize the prober unless the packet size is large enough to probe
   // with.
-  void OnIncomingPacket(int bitrate_bps, size_t packet_size, int64_t now_ms);
+  void OnIncomingPacket(uint32_t bitrate_bps,
+                        size_t packet_size,
+                        int64_t now_ms);
 
   // Returns the number of milliseconds until the next packet should be sent to
   // get accurate probing.
   int TimeUntilNextProbe(int64_t now_ms);
 
+  // Which cluster that is currently being used for probing.
+  int CurrentClusterId() const;
+
   // Returns the number of bytes that the prober recommends for the next probe
   // packet.
   size_t RecommendedPacketSize() const;
@@ -51,13 +57,21 @@ class BitrateProber {
  private:
   enum ProbingState { kDisabled, kAllowedToProbe, kProbing, kWait };
 
+  struct ProbeCluster {
+    int max_probe_packets = 0;
+    int sent_probe_packets = 0;
+    int probe_bitrate_bps = 0;
+    int id = -1;
+  };
+
   ProbingState probing_state_;
   // Probe bitrate per packet. These are used to compute the delta relative to
   // the previous probe packet based on the size and time when that packet was
   // sent.
-  std::list<int> probe_bitrates_;
+  std::queue<ProbeCluster> clusters_;
   size_t packet_size_last_send_;
   int64_t time_last_send_ms_;
+  int next_cluster_id_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_PACING_BITRATE_PROBER_H_
diff --git a/chromium/third_party/webrtc/modules/pacing/bitrate_prober_unittest.cc b/chromium/third_party/webrtc/modules/pacing/bitrate_prober_unittest.cc
index 59ee479973d..9e38220e012 100644
--- a/chromium/third_party/webrtc/modules/pacing/bitrate_prober_unittest.cc
+++ b/chromium/third_party/webrtc/modules/pacing/bitrate_prober_unittest.cc
@@ -26,6 +26,7 @@ TEST(BitrateProberTest, VerifyStatesAndTimeBetweenProbes) {
 
   prober.OnIncomingPacket(300000, 1000, now_ms);
   EXPECT_TRUE(prober.IsProbing());
+  EXPECT_EQ(0, prober.CurrentClusterId());
 
   // First packet should probe as soon as possible.
   EXPECT_EQ(0, prober.TimeUntilNextProbe(now_ms));
@@ -37,12 +38,14 @@ TEST(BitrateProberTest, VerifyStatesAndTimeBetweenProbes) {
     EXPECT_EQ(4, prober.TimeUntilNextProbe(now_ms));
     now_ms += 4;
     EXPECT_EQ(0, prober.TimeUntilNextProbe(now_ms));
+    EXPECT_EQ(0, prober.CurrentClusterId());
     prober.PacketSent(now_ms, 1000);
   }
   for (int i = 0; i < 5; ++i) {
     EXPECT_EQ(4, prober.TimeUntilNextProbe(now_ms));
     now_ms += 4;
     EXPECT_EQ(0, prober.TimeUntilNextProbe(now_ms));
+    EXPECT_EQ(1, prober.CurrentClusterId());
     prober.PacketSent(now_ms, 1000);
   }
 
diff --git a/chromium/third_party/webrtc/modules/pacing/mock/mock_paced_sender.h b/chromium/third_party/webrtc/modules/pacing/mock/mock_paced_sender.h
index c710dbcbea5..bd7d7aaa499 100644
--- a/chromium/third_party/webrtc/modules/pacing/mock/mock_paced_sender.h
+++ b/chromium/third_party/webrtc/modules/pacing/mock/mock_paced_sender.h
@@ -22,15 +22,17 @@ namespace webrtc {
 
 class MockPacedSender : public PacedSender {
  public:
-  MockPacedSender() : PacedSender(Clock::GetRealTimeClock(), NULL, 0, 0, 0) {}
+  MockPacedSender() : PacedSender(Clock::GetRealTimeClock(), nullptr) {}
   MOCK_METHOD6(SendPacket, bool(Priority priority,
                                 uint32_t ssrc,
                                 uint16_t sequence_number,
                                 int64_t capture_time_ms,
                                 size_t bytes,
                                 bool retransmission));
+  MOCK_METHOD1(SetEstimatedBitrate, void(uint32_t));
   MOCK_CONST_METHOD0(QueueInMs, int64_t());
   MOCK_CONST_METHOD0(QueueInPackets, int());
+  MOCK_CONST_METHOD0(ExpectedQueueTimeMs, int64_t());
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/pacing/paced_sender.cc b/chromium/third_party/webrtc/modules/pacing/paced_sender.cc
index b56d28510f2..167be23ab62 100644
--- a/chromium/third_party/webrtc/modules/pacing/paced_sender.cc
+++ b/chromium/third_party/webrtc/modules/pacing/paced_sender.cc
@@ -246,20 +246,18 @@ const int64_t PacedSender::kMaxQueueLengthMs = 2000;
 const float PacedSender::kDefaultPaceMultiplier = 2.5f;
 
 PacedSender::PacedSender(Clock* clock,
-                         Callback* callback,
-                         int bitrate_kbps,
-                         int max_bitrate_kbps,
-                         int min_bitrate_kbps)
+                         PacketSender* packet_sender)
     : clock_(clock),
-      callback_(callback),
+      packet_sender_(packet_sender),
       critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       paused_(false),
       probing_enabled_(true),
-      media_budget_(new paced_sender::IntervalBudget(max_bitrate_kbps)),
-      padding_budget_(new paced_sender::IntervalBudget(min_bitrate_kbps)),
+      media_budget_(new paced_sender::IntervalBudget(0)),
+      padding_budget_(new paced_sender::IntervalBudget(0)),
       prober_(new BitrateProber()),
-      bitrate_bps_(1000 * bitrate_kbps),
-      max_bitrate_kbps_(max_bitrate_kbps),
+      estimated_bitrate_bps_(0),
+      min_send_bitrate_kbps_(0u),
+      pacing_bitrate_kbps_(0),
       time_last_update_us_(clock->TimeInMicroseconds()),
       packets_(new paced_sender::PacketQueue(clock)),
       packet_counter_(0) {
@@ -283,16 +281,22 @@ void PacedSender::SetProbingEnabled(bool enabled) {
   probing_enabled_ = enabled;
 }
 
-void PacedSender::UpdateBitrate(int bitrate_kbps,
-                                int max_bitrate_kbps,
-                                int min_bitrate_kbps) {
+void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
   CriticalSectionScoped cs(critsect_.get());
-  // Don't set media bitrate here as it may be boosted in order to meet max
-  // queue time constraint. Just update max_bitrate_kbps_ and let media_budget_
-  // be updated in Process().
-  padding_budget_->set_target_rate_kbps(min_bitrate_kbps);
-  bitrate_bps_ = 1000 * bitrate_kbps;
-  max_bitrate_kbps_ = max_bitrate_kbps;
+  estimated_bitrate_bps_ = bitrate_bps;
+  pacing_bitrate_kbps_ =
+      std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
+      kDefaultPaceMultiplier;
+}
+
+void PacedSender::SetAllocatedSendBitrate(int allocated_bitrate,
+                                          int padding_bitrate) {
+  CriticalSectionScoped cs(critsect_.get());
+  min_send_bitrate_kbps_ = allocated_bitrate / 1000;
+  pacing_bitrate_kbps_ =
+      std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
+      kDefaultPaceMultiplier;
+  padding_budget_->set_target_rate_kbps(padding_bitrate / 1000);
 }
 
 void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
@@ -302,11 +306,13 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
                                size_t bytes,
                                bool retransmission) {
   CriticalSectionScoped cs(critsect_.get());
+  RTC_DCHECK(estimated_bitrate_bps_ > 0)
+        << "SetEstimatedBitrate must be called before InsertPacket.";
 
   if (probing_enabled_ && !prober_->IsProbing())
     prober_->SetEnabled(true);
   int64_t now_ms = clock_->TimeInMilliseconds();
-  prober_->OnIncomingPacket(bitrate_bps_, bytes, now_ms);
+  prober_->OnIncomingPacket(estimated_bitrate_bps_, bytes, now_ms);
 
   if (capture_time_ms < 0)
     capture_time_ms = now_ms;
@@ -318,8 +324,9 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
 
 int64_t PacedSender::ExpectedQueueTimeMs() const {
   CriticalSectionScoped cs(critsect_.get());
-  RTC_DCHECK_GT(max_bitrate_kbps_, 0);
-  return static_cast<int64_t>(packets_->SizeInBytes() * 8 / max_bitrate_kbps_);
+  RTC_DCHECK_GT(pacing_bitrate_kbps_, 0u);
+  return static_cast<int64_t>(packets_->SizeInBytes() * 8 /
+                              pacing_bitrate_kbps_);
 }
 
 size_t PacedSender::QueueSizePackets() const {
@@ -360,7 +367,7 @@ void PacedSender::Process() {
   CriticalSectionScoped cs(critsect_.get());
   int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
   time_last_update_us_ = now_us;
-  int target_bitrate_kbps = max_bitrate_kbps_;
+  int target_bitrate_kbps = pacing_bitrate_kbps_;
   // TODO(holmer): Remove the !paused_ check when issue 5307 has been fixed.
   if (!paused_ && elapsed_time_ms > 0) {
     size_t queue_size_bytes = packets_->SizeInBytes();
@@ -390,8 +397,10 @@ void PacedSender::Process() {
     // element from the priority queue but keep it in storage, so that we can
     // reinsert it if send fails.
     const paced_sender::Packet& packet = packets_->BeginPop();
+    int probe_cluster_id =
+        prober_->IsProbing() ? prober_->CurrentClusterId() : -1;
 
-    if (SendPacket(packet)) {
+    if (SendPacket(packet, probe_cluster_id)) {
       // Send succeeded, remove it from the queue.
       packets_->FinalizePop(packet);
       if (prober_->IsProbing())
@@ -418,17 +427,17 @@ void PacedSender::Process() {
     SendPadding(static_cast<size_t>(padding_needed));
 }
 
-bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
+bool PacedSender::SendPacket(const paced_sender::Packet& packet,
+                             int probe_cluster_id) {
   // TODO(holmer): Because of this bug issue 5307 we have to send audio
   // packets even when the pacer is paused. Here we assume audio packets are
   // always high priority and that they are the only high priority packets.
   if (paused_ && packet.priority != kHighPriority)
     return false;
   critsect_->Leave();
-  const bool success = callback_->TimeToSendPacket(packet.ssrc,
-                                                   packet.sequence_number,
-                                                   packet.capture_time_ms,
-                                                   packet.retransmission);
+  const bool success = packet_sender_->TimeToSendPacket(
+      packet.ssrc, packet.sequence_number, packet.capture_time_ms,
+      packet.retransmission, probe_cluster_id);
   critsect_->Enter();
 
   if (success) {
@@ -447,7 +456,7 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet) {
 
 void PacedSender::SendPadding(size_t padding_needed) {
   critsect_->Leave();
-  size_t bytes_sent = callback_->TimeToSendPadding(padding_needed);
+  size_t bytes_sent = packet_sender_->TimeToSendPadding(padding_needed);
   critsect_->Enter();
 
   if (bytes_sent > 0) {
diff --git a/chromium/third_party/webrtc/modules/pacing/paced_sender.h b/chromium/third_party/webrtc/modules/pacing/paced_sender.h
index 16569b04045..d42b9b38489 100644
--- a/chromium/third_party/webrtc/modules/pacing/paced_sender.h
+++ b/chromium/third_party/webrtc/modules/pacing/paced_sender.h
@@ -33,7 +33,7 @@ class PacketQueue;
 
 class PacedSender : public Module, public RtpPacketSender {
  public:
-  class Callback {
+  class PacketSender {
    public:
     // Note: packets sent as a result of a callback should not pass by this
     // module again.
@@ -42,13 +42,14 @@ class PacedSender : public Module, public RtpPacketSender {
     virtual bool TimeToSendPacket(uint32_t ssrc,
                                   uint16_t sequence_number,
                                   int64_t capture_time_ms,
-                                  bool retransmission) = 0;
+                                  bool retransmission,
+                                  int probe_cluster_id) = 0;
     // Called when it's a good time to send a padding data.
     // Returns the number of bytes sent.
     virtual size_t TimeToSendPadding(size_t bytes) = 0;
 
    protected:
-    virtual ~Callback() {}
+    virtual ~PacketSender() {}
   };
 
   // Expected max pacer delay in ms. If ExpectedQueueTimeMs() is higher than
@@ -56,8 +57,6 @@ class PacedSender : public Module, public RtpPacketSender {
   // encoding them). Bitrate sent may temporarily exceed target set by
   // UpdateBitrate() so that this limit will be upheld.
   static const int64_t kMaxQueueLengthMs;
-  // Pace in kbits/s until we receive first estimate.
-  static const int kDefaultInitialPaceKbps = 2000;
   // Pacing-rate relative to our target send rate.
   // Multiplicative factor that is applied to the target bitrate to calculate
   // the number of bytes that can be transmitted per interval.
@@ -68,10 +67,7 @@ class PacedSender : public Module, public RtpPacketSender {
   static const size_t kMinProbePacketSize = 200;
 
   PacedSender(Clock* clock,
-              Callback* callback,
-              int bitrate_kbps,
-              int max_bitrate_kbps,
-              int min_bitrate_kbps);
+              PacketSender* packet_sender);
 
   virtual ~PacedSender();
 
@@ -86,14 +82,20 @@ class PacedSender : public Module, public RtpPacketSender {
   // effect.
   void SetProbingEnabled(bool enabled);
 
-  // Set target bitrates for the pacer.
-  // We will pace out bursts of packets at a bitrate of |max_bitrate_kbps|.
-  // |bitrate_kbps| is our estimate of what we are allowed to send on average.
-  // Padding packets will be utilized to reach |min_bitrate| unless enough media
-  // packets are available.
-  void UpdateBitrate(int bitrate_kbps,
-                     int max_bitrate_kbps,
-                     int min_bitrate_kbps);
+  // Sets the estimated capacity of the network. Must be called once before
+  // packets can be sent.
+  // |bitrate_bps| is our estimate of what we are allowed to send on average.
+  // We will pace out bursts of packets at a bitrate of
+  // |bitrate_bps| * kDefaultPaceMultiplier.
+  virtual void SetEstimatedBitrate(uint32_t bitrate_bps);
+
+  // Sets the bitrate that has been allocated for encoders.
+  // |allocated_bitrate| might be higher that the estimated available network
+  // bitrate and if so, the pacer will send with |allocated_bitrate|.
+  // Padding packets will be utilized to reach |padding_bitrate| unless enough
+  // media packets are available.
+  void SetAllocatedSendBitrate(int allocated_bitrate_bps,
+                               int padding_bitrate_bps);
 
   // Returns true if we send the packet now, else it will add the packet
   // information to the queue and call TimeToSendPacket when it's time to send.
@@ -129,12 +131,12 @@ class PacedSender : public Module, public RtpPacketSender {
   void UpdateBytesPerInterval(int64_t delta_time_in_ms)
       EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
-  bool SendPacket(const paced_sender::Packet& packet)
+  bool SendPacket(const paced_sender::Packet& packet, int probe_cluster_id)
       EXCLUSIVE_LOCKS_REQUIRED(critsect_);
   void SendPadding(size_t padding_needed) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
   Clock* const clock_;
-  Callback* const callback_;
+  PacketSender* const packet_sender_;
 
   std::unique_ptr<CriticalSectionWrapper> critsect_;
   bool paused_ GUARDED_BY(critsect_);
@@ -152,8 +154,9 @@ class PacedSender : public Module, public RtpPacketSender {
   std::unique_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
   // Actual configured bitrates (media_budget_ may temporarily be higher in
   // order to meet pace time constraint).
-  int bitrate_bps_ GUARDED_BY(critsect_);
-  int max_bitrate_kbps_ GUARDED_BY(critsect_);
+  uint32_t estimated_bitrate_bps_ GUARDED_BY(critsect_);
+  uint32_t min_send_bitrate_kbps_ GUARDED_BY(critsect_);
+  uint32_t pacing_bitrate_kbps_ GUARDED_BY(critsect_);
 
   int64_t time_last_update_us_ GUARDED_BY(critsect_);
 
diff --git a/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc b/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc
index 941c81335b6..6a0a006c326 100644
--- a/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc
+++ b/chromium/third_party/webrtc/modules/pacing/paced_sender_unittest.cc
@@ -22,32 +22,33 @@ using testing::Return;
 namespace webrtc {
 namespace test {
 
-static const int kTargetBitrate = 800;
-static const float kPaceMultiplier = 1.5f;
+static const int kTargetBitrateBps = 800000;
 
-class MockPacedSenderCallback : public PacedSender::Callback {
+class MockPacedSenderCallback : public PacedSender::PacketSender {
  public:
-  MOCK_METHOD4(TimeToSendPacket,
+  MOCK_METHOD5(TimeToSendPacket,
                bool(uint32_t ssrc,
                     uint16_t sequence_number,
                     int64_t capture_time_ms,
-                    bool retransmission));
+                    bool retransmission,
+                    int probe_cluster_id));
   MOCK_METHOD1(TimeToSendPadding,
       size_t(size_t bytes));
 };
 
-class PacedSenderPadding : public PacedSender::Callback {
+class PacedSenderPadding : public PacedSender::PacketSender {
  public:
   PacedSenderPadding() : padding_sent_(0) {}
 
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_time_ms,
-                        bool retransmission) {
+                        bool retransmission,
+                        int probe_cluster_id) override {
     return true;
   }
 
-  size_t TimeToSendPadding(size_t bytes) {
+  size_t TimeToSendPadding(size_t bytes) override {
     const size_t kPaddingPacketSize = 224;
     size_t num_packets = (bytes + kPaddingPacketSize - 1) / kPaddingPacketSize;
     padding_sent_ += kPaddingPacketSize * num_packets;
@@ -60,7 +61,7 @@ class PacedSenderPadding : public PacedSender::Callback {
   size_t padding_sent_;
 };
 
-class PacedSenderProbing : public PacedSender::Callback {
+class PacedSenderProbing : public PacedSender::PacketSender {
  public:
   PacedSenderProbing(const std::list<int>& expected_deltas, Clock* clock)
       : prev_packet_time_ms_(-1),
@@ -71,12 +72,13 @@ class PacedSenderProbing : public PacedSender::Callback {
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_time_ms,
-                        bool retransmission) {
+                        bool retransmission,
+                        int probe_cluster_id) override {
     ExpectAndCountPacket();
     return true;
   }
 
-  size_t TimeToSendPadding(size_t bytes) {
+  size_t TimeToSendPadding(size_t bytes) override {
     ExpectAndCountPacket();
     return bytes;
   }
@@ -108,15 +110,14 @@ class PacedSenderTest : public ::testing::Test {
   PacedSenderTest() : clock_(123456) {
     srand(0);
     // Need to initialize PacedSender after we initialize clock.
-    send_bucket_.reset(new PacedSender(&clock_,
-                                       &callback_,
-                                       kTargetBitrate,
-                                       kPaceMultiplier * kTargetBitrate,
-                                       0));
+    send_bucket_.reset(new PacedSender(&clock_, &callback_));
     // Default to bitrate probing disabled for testing purposes. Probing tests
     // have to enable probing, either by creating a new PacedSender instance or
     // by calling SetProbingEnabled(true).
     send_bucket_->SetProbingEnabled(false);
+    send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
+
+    clock_.AdvanceTimeMilliseconds(send_bucket_->TimeUntilNextProcess());
   }
 
   void SendAndExpectPacket(PacedSender::Priority priority,
@@ -127,8 +128,8 @@ class PacedSenderTest : public ::testing::Test {
                            bool retransmission) {
     send_bucket_->InsertPacket(priority, ssrc, sequence_number, capture_time_ms,
                                size, retransmission);
-    EXPECT_CALL(callback_,
-                TimeToSendPacket(ssrc, sequence_number, capture_time_ms, false))
+    EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number,
+                                            capture_time_ms, false, _))
         .Times(1)
         .WillRepeatedly(Return(true));
   }
@@ -141,29 +142,21 @@ class PacedSenderTest : public ::testing::Test {
 TEST_F(PacedSenderTest, QueuePacket) {
   uint32_t ssrc = 12345;
   uint16_t sequence_number = 1234;
-  // Due to the multiplicative factor we can send 3 packets not 2 packets.
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send =
+      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
+  }
+
   int64_t queued_packet_timestamp = clock_.TimeInMilliseconds();
   send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                              sequence_number, queued_packet_timestamp, 250,
                              false);
+  EXPECT_EQ(packets_to_send + 1, send_bucket_->QueueSizePackets());
   send_bucket_->Process();
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
@@ -171,86 +164,79 @@ TEST_F(PacedSenderTest, QueuePacket) {
   EXPECT_EQ(1, send_bucket_->TimeUntilNextProcess());
   clock_.AdvanceTimeMilliseconds(1);
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
-  EXPECT_CALL(
-      callback_,
-      TimeToSendPacket(ssrc, sequence_number++, queued_packet_timestamp, false))
+  EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number++,
+                                          queued_packet_timestamp, false, _))
       .Times(1)
       .WillRepeatedly(Return(true));
   send_bucket_->Process();
   sequence_number++;
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
+  EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
+
+  // We can send packets_to_send -1 packets of size 250 during the current
+  // interval since one packet has already been sent.
+  for (size_t i = 0; i < packets_to_send - 1; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
+  }
   send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                              sequence_number++, clock_.TimeInMilliseconds(),
                              250, false);
+  EXPECT_EQ(packets_to_send, send_bucket_->QueueSizePackets());
   send_bucket_->Process();
+  EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
 }
 
 TEST_F(PacedSenderTest, PaceQueuedPackets) {
   uint32_t ssrc = 12345;
   uint16_t sequence_number = 1234;
 
-  // Due to the multiplicative factor we can send 3 packets not 2 packets.
-  for (int i = 0; i < 3; ++i) {
-    SendAndExpectPacket(PacedSender::kNormalPriority,
-                        ssrc,
-                        sequence_number++,
-                        clock_.TimeInMilliseconds(),
-                        250,
-                        false);
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
   }
-  for (int j = 0; j < 30; ++j) {
+
+  for (size_t j = 0; j < packets_to_send_per_interval * 10; ++j) {
     send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                                sequence_number++, clock_.TimeInMilliseconds(),
                                250, false);
   }
+  EXPECT_EQ(packets_to_send_per_interval + packets_to_send_per_interval * 10,
+            send_bucket_->QueueSizePackets());
   send_bucket_->Process();
+  EXPECT_EQ(packets_to_send_per_interval * 10,
+            send_bucket_->QueueSizePackets());
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
   for (int k = 0; k < 10; ++k) {
     EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
     clock_.AdvanceTimeMilliseconds(5);
-    EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, _, false))
-        .Times(3)
+    EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, _, false, _))
+        .Times(packets_to_send_per_interval)
         .WillRepeatedly(Return(true));
     EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
     send_bucket_->Process();
   }
+  EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   clock_.AdvanceTimeMilliseconds(5);
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
+  EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
   send_bucket_->Process();
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
+
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
+  }
   send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                              sequence_number, clock_.TimeInMilliseconds(), 250,
                              false);
   send_bucket_->Process();
+  EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
 }
 
 TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
@@ -258,18 +244,18 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
   uint16_t sequence_number = 1234;
   uint16_t queued_sequence_number;
 
-  // Due to the multiplicative factor we can send 3 packets not 2 packets.
-  for (int i = 0; i < 3; ++i) {
-    SendAndExpectPacket(PacedSender::kNormalPriority,
-                        ssrc,
-                        sequence_number++,
-                        clock_.TimeInMilliseconds(),
-                        250,
-                        false);
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
   }
   queued_sequence_number = sequence_number;
 
-  for (int j = 0; j < 30; ++j) {
+  for (size_t j = 0; j < packets_to_send_per_interval * 10; ++j) {
     // Send in duplicate packets.
     send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                                sequence_number, clock_.TimeInMilliseconds(),
@@ -284,9 +270,9 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
     EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
     clock_.AdvanceTimeMilliseconds(5);
 
-    for (int i = 0; i < 3; ++i) {
+    for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
       EXPECT_CALL(callback_,
-                  TimeToSendPacket(ssrc, queued_sequence_number++, _, false))
+                  TimeToSendPacket(ssrc, queued_sequence_number++, _, false, _))
           .Times(1)
           .WillRepeatedly(Return(true));
     }
@@ -297,28 +283,16 @@ TEST_F(PacedSenderTest, PaceQueuedPacketsWithDuplicates) {
   clock_.AdvanceTimeMilliseconds(5);
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
   send_bucket_->Process();
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
+
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
+  }
   send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                              sequence_number++, clock_.TimeInMilliseconds(),
                              250, false);
   send_bucket_->Process();
+  EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
 }
 
 TEST_F(PacedSenderTest, CanQueuePacketsWithSameSequenceNumberOnDifferentSsrcs) {
@@ -348,29 +322,27 @@ TEST_F(PacedSenderTest, Padding) {
   uint32_t ssrc = 12345;
   uint16_t sequence_number = 1234;
 
-  send_bucket_->UpdateBitrate(
-      kTargetBitrate, kPaceMultiplier * kTargetBitrate, kTargetBitrate);
-  // Due to the multiplicative factor we can send 3 packets not 2 packets.
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      clock_.TimeInMilliseconds(),
-                      250,
-                      false);
+  send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
+  send_bucket_->SetAllocatedSendBitrate(kTargetBitrateBps, kTargetBitrateBps);
+
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
+  }
   // No padding is expected since we have sent too much already.
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
+  EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
+  send_bucket_->Process();
+  EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
+
+  // 5 milliseconds later should not send padding since we filled the buffers
+  // initially.
+  EXPECT_CALL(callback_, TimeToSendPadding(250)).Times(0);
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   clock_.AdvanceTimeMilliseconds(5);
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
@@ -391,8 +363,9 @@ TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) {
   int64_t capture_time_ms = 56789;
   const int kTimeStep = 5;
   const int64_t kBitrateWindow = 100;
-  send_bucket_->UpdateBitrate(
-      kTargetBitrate, kPaceMultiplier * kTargetBitrate, kTargetBitrate);
+  send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
+  send_bucket_->SetAllocatedSendBitrate(kTargetBitrateBps, kTargetBitrateBps);
+
   int64_t start_time = clock_.TimeInMilliseconds();
   while (clock_.TimeInMilliseconds() - start_time < kBitrateWindow) {
     SendAndExpectPacket(PacedSender::kNormalPriority,
@@ -401,10 +374,10 @@ TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) {
                         capture_time_ms,
                         250,
                         false);
-    clock_.AdvanceTimeMilliseconds(kTimeStep);
     EXPECT_CALL(callback_, TimeToSendPadding(250)).Times(1).
         WillOnce(Return(250));
     send_bucket_->Process();
+    clock_.AdvanceTimeMilliseconds(kTimeStep);
   }
 }
 
@@ -415,11 +388,11 @@ TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
   const int kTimeStep = 5;
   const int64_t kBitrateWindow = 10000;
   PacedSenderPadding callback;
-  send_bucket_.reset(new PacedSender(
-      &clock_, &callback, kTargetBitrate, kPaceMultiplier * kTargetBitrate, 0));
+  send_bucket_.reset(new PacedSender(&clock_, &callback));
   send_bucket_->SetProbingEnabled(false);
-  send_bucket_->UpdateBitrate(
-      kTargetBitrate, kPaceMultiplier * kTargetBitrate, kTargetBitrate);
+  send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
+  send_bucket_->SetAllocatedSendBitrate(kTargetBitrateBps, kTargetBitrateBps);
+
   int64_t start_time = clock_.TimeInMilliseconds();
   size_t media_bytes = 0;
   while (clock_.TimeInMilliseconds() - start_time < kBitrateWindow) {
@@ -432,9 +405,10 @@ TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
     clock_.AdvanceTimeMilliseconds(kTimeStep);
     send_bucket_->Process();
   }
-  EXPECT_NEAR(kTargetBitrate,
+  EXPECT_NEAR(kTargetBitrateBps / 1000,
               static_cast<int>(8 * (media_bytes + callback.padding_sent()) /
-                  kBitrateWindow), 1);
+                               kBitrateWindow),
+              1);
 }
 
 TEST_F(PacedSenderTest, Priority) {
@@ -444,54 +418,45 @@ TEST_F(PacedSenderTest, Priority) {
   int64_t capture_time_ms = 56789;
   int64_t capture_time_ms_low_priority = 1234567;
 
-  // Due to the multiplicative factor we can send 3 packets not 2 packets.
-  SendAndExpectPacket(PacedSender::kLowPriority,
-                      ssrc,
-                      sequence_number++,
-                      capture_time_ms,
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      capture_time_ms,
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      capture_time_ms,
-                      250,
-                      false);
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
+  }
   send_bucket_->Process();
+  EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
 
   // Expect normal and low priority to be queued and high to pass through.
   send_bucket_->InsertPacket(PacedSender::kLowPriority, ssrc_low_priority,
                              sequence_number++, capture_time_ms_low_priority,
                              250, false);
-  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
-                             sequence_number++, capture_time_ms, 250, false);
-  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
-                             sequence_number++, capture_time_ms, 250, false);
-  send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
-                             sequence_number++, capture_time_ms, 250, false);
+
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number++, capture_time_ms, 250, false);
+  }
   send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
                              sequence_number++, capture_time_ms, 250, false);
 
   // Expect all high and normal priority to be sent out first.
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
-  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, capture_time_ms, false))
-      .Times(4)
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, capture_time_ms, false, _))
+      .Times(packets_to_send_per_interval + 1)
       .WillRepeatedly(Return(true));
 
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   clock_.AdvanceTimeMilliseconds(5);
   EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
   send_bucket_->Process();
+  EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
 
   EXPECT_CALL(callback_,
-              TimeToSendPacket(
-                  ssrc_low_priority, _, capture_time_ms_low_priority, false))
+              TimeToSendPacket(ssrc_low_priority, _,
+                               capture_time_ms_low_priority, false, _))
       .Times(1)
       .WillRepeatedly(Return(true));
 
@@ -513,23 +478,30 @@ TEST_F(PacedSenderTest, HighPrioDoesntAffectBudget) {
                         capture_time_ms, 250, false);
   }
   send_bucket_->Process();
-  // Low prio packets does affect the budget, so we should only be able to send
-  // 3 at once, the 4th should be queued.
-  for (int i = 0; i < 3; ++i) {
+  // Low prio packets does affect the budget.
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
     SendAndExpectPacket(PacedSender::kLowPriority, ssrc, sequence_number++,
-                        capture_time_ms, 250, false);
+                        clock_.TimeInMilliseconds(), 250, false);
   }
   send_bucket_->InsertPacket(PacedSender::kLowPriority, ssrc, sequence_number,
                              capture_time_ms, 250, false);
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   clock_.AdvanceTimeMilliseconds(5);
   send_bucket_->Process();
-  EXPECT_CALL(callback_,
-              TimeToSendPacket(ssrc, sequence_number++, capture_time_ms, false))
-      .Times(1);
+  EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number++,
+                                          capture_time_ms, false, _))
+      .Times(1)
+      .WillRepeatedly(Return(true));
   EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
   clock_.AdvanceTimeMilliseconds(5);
   send_bucket_->Process();
+  EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
 }
 
 TEST_F(PacedSenderTest, Pause) {
@@ -540,25 +512,16 @@ TEST_F(PacedSenderTest, Pause) {
 
   EXPECT_EQ(0, send_bucket_->QueueInMs());
 
-  // Due to the multiplicative factor we can send 3 packets not 2 packets.
-  SendAndExpectPacket(PacedSender::kLowPriority,
-                      ssrc,
-                      sequence_number++,
-                      capture_time_ms,
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      capture_time_ms,
-                      250,
-                      false);
-  SendAndExpectPacket(PacedSender::kNormalPriority,
-                      ssrc,
-                      sequence_number++,
-                      capture_time_ms,
-                      250,
-                      false);
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), 250, false);
+  }
+
   send_bucket_->Process();
 
   send_bucket_->Pause();
@@ -583,7 +546,7 @@ TEST_F(PacedSenderTest, Pause) {
 
   // Expect no packet to come out while paused.
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
-  EXPECT_CALL(callback_, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_CALL(callback_, TimeToSendPacket(_, _, _, _, _)).Times(0);
 
   for (int i = 0; i < 10; ++i) {
     clock_.AdvanceTimeMilliseconds(5);
@@ -592,10 +555,11 @@ TEST_F(PacedSenderTest, Pause) {
   }
   // Expect high prio packets to come out first followed by all packets in the
   // way they were added.
-  EXPECT_CALL(callback_, TimeToSendPacket(_, _, capture_time_ms, false))
+  EXPECT_CALL(callback_, TimeToSendPacket(_, _, capture_time_ms, false, _))
       .Times(3)
       .WillRepeatedly(Return(true));
-  EXPECT_CALL(callback_, TimeToSendPacket(_, _, second_capture_time_ms, false))
+  EXPECT_CALL(callback_,
+              TimeToSendPacket(_, _, second_capture_time_ms, false, _))
       .Times(1)
       .WillRepeatedly(Return(true));
   send_bucket_->Resume();
@@ -624,8 +588,8 @@ TEST_F(PacedSenderTest, ResendPacket) {
   EXPECT_EQ(clock_.TimeInMilliseconds() - capture_time_ms,
             send_bucket_->QueueInMs());
   // Fails to send first packet so only one call.
-  EXPECT_CALL(callback_,
-              TimeToSendPacket(ssrc, sequence_number, capture_time_ms, false))
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number,
+                                          capture_time_ms, false, _))
       .Times(1)
       .WillOnce(Return(false));
   clock_.AdvanceTimeMilliseconds(10000);
@@ -636,13 +600,12 @@ TEST_F(PacedSenderTest, ResendPacket) {
             send_bucket_->QueueInMs());
 
   // Fails to send second packet.
-  EXPECT_CALL(callback_,
-              TimeToSendPacket(ssrc, sequence_number, capture_time_ms, false))
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number,
+                                          capture_time_ms, false, _))
       .Times(1)
       .WillOnce(Return(true));
-  EXPECT_CALL(
-      callback_,
-      TimeToSendPacket(ssrc, sequence_number + 1, capture_time_ms + 1, false))
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 1,
+                                          capture_time_ms + 1, false, _))
       .Times(1)
       .WillOnce(Return(false));
   clock_.AdvanceTimeMilliseconds(10000);
@@ -653,9 +616,8 @@ TEST_F(PacedSenderTest, ResendPacket) {
             send_bucket_->QueueInMs());
 
   // Send second packet and queue becomes empty.
-  EXPECT_CALL(
-      callback_,
-      TimeToSendPacket(ssrc, sequence_number + 1, capture_time_ms + 1, false))
+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 1,
+                                          capture_time_ms + 1, false, _))
       .Times(1)
       .WillOnce(Return(true));
   clock_.AdvanceTimeMilliseconds(10000);
@@ -668,18 +630,18 @@ TEST_F(PacedSenderTest, ExpectedQueueTimeMs) {
   uint16_t sequence_number = 1234;
   const size_t kNumPackets = 60;
   const size_t kPacketSize = 1200;
-  const int32_t kMaxBitrate = kPaceMultiplier * 30;
+  const int32_t kMaxBitrate = PacedSender::kDefaultPaceMultiplier * 30000;
   EXPECT_EQ(0, send_bucket_->ExpectedQueueTimeMs());
 
-  send_bucket_->UpdateBitrate(30, kMaxBitrate, 0);
+  send_bucket_->SetEstimatedBitrate(30000);
   for (size_t i = 0; i < kNumPackets; ++i) {
     SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
                         clock_.TimeInMilliseconds(), kPacketSize, false);
   }
 
-  // Queue in ms = 1000 * (bytes in queue) / (kbit per second * 1000 / 8)
+  // Queue in ms = 1000 * (bytes in queue) *8 / (bits per second)
   int64_t queue_in_ms =
-      static_cast<int64_t>(kNumPackets * kPacketSize * 8 / kMaxBitrate);
+      static_cast<int64_t>(1000 * kNumPackets * kPacketSize * 8 / kMaxBitrate);
   EXPECT_EQ(queue_in_ms, send_bucket_->ExpectedQueueTimeMs());
 
   int64_t time_start = clock_.TimeInMilliseconds();
@@ -697,7 +659,7 @@ TEST_F(PacedSenderTest, ExpectedQueueTimeMs) {
 
   // Allow for aliasing, duration should be within one pack of max time limit.
   EXPECT_NEAR(duration, PacedSender::kMaxQueueLengthMs,
-              static_cast<int64_t>(kPacketSize * 8 / kMaxBitrate));
+              static_cast<int64_t>(1000 * kPacketSize * 8 / kMaxBitrate));
 }
 
 TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) {
@@ -705,7 +667,7 @@ TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) {
   uint16_t sequence_number = 1234;
   EXPECT_EQ(0, send_bucket_->QueueInMs());
 
-  send_bucket_->UpdateBitrate(30, kPaceMultiplier * 30, 0);
+  send_bucket_->SetEstimatedBitrate(30000);
   SendAndExpectPacket(PacedSender::kNormalPriority,
                       ssrc,
                       sequence_number,
@@ -723,25 +685,23 @@ TEST_F(PacedSenderTest, ProbingWithInitialFrame) {
   const int kNumPackets = 11;
   const int kNumDeltas = kNumPackets - 1;
   const size_t kPacketSize = 1200;
-  const int kInitialBitrateKbps = 300;
+  const int kInitialBitrateBps = 300000;
   uint32_t ssrc = 12346;
   uint16_t sequence_number = 1234;
+
   const int expected_deltas[kNumDeltas] = {10, 10, 10, 10, 10, 5, 5, 5, 5, 5};
   std::list<int> expected_deltas_list(expected_deltas,
                                       expected_deltas + kNumDeltas);
   PacedSenderProbing callback(expected_deltas_list, &clock_);
-  send_bucket_.reset(
-      new PacedSender(&clock_,
-                      &callback,
-                      kInitialBitrateKbps,
-                      kPaceMultiplier * kInitialBitrateKbps,
-                      0));
+  send_bucket_.reset(new PacedSender(&clock_, &callback));
+  send_bucket_->SetEstimatedBitrate(kInitialBitrateBps);
 
   for (int i = 0; i < kNumPackets; ++i) {
     send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
                                sequence_number++, clock_.TimeInMilliseconds(),
                                kPacketSize, false);
   }
+
   while (callback.packets_sent() < kNumPackets) {
     int time_until_process = send_bucket_->TimeUntilNextProcess();
     if (time_until_process <= 0) {
@@ -756,15 +716,15 @@ TEST_F(PacedSenderTest, ProbingWithTooSmallInitialFrame) {
   const int kNumPackets = 11;
   const int kNumDeltas = kNumPackets - 1;
   const size_t kPacketSize = 1200;
-  const int kInitialBitrateKbps = 300;
+  const int kInitialBitrateBps = 300000;
   uint32_t ssrc = 12346;
   uint16_t sequence_number = 1234;
   const int expected_deltas[kNumDeltas] = {10, 10, 10, 10, 10, 5, 5, 5, 5, 5};
   std::list<int> expected_deltas_list(expected_deltas,
                                       expected_deltas + kNumDeltas);
   PacedSenderProbing callback(expected_deltas_list, &clock_);
-  send_bucket_.reset(new PacedSender(&clock_, &callback, kInitialBitrateKbps,
-                                     kPaceMultiplier * kInitialBitrateKbps, 0));
+  send_bucket_.reset(new PacedSender(&clock_, &callback));
+  send_bucket_->SetEstimatedBitrate(kInitialBitrateBps);
 
   for (int i = 0; i < kNumPackets - 5; ++i) {
     send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
@@ -810,18 +770,22 @@ TEST_F(PacedSenderTest, PriorityInversion) {
   // Packets from earlier frames should be sent first.
   {
     ::testing::InSequence sequence;
-    EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number,
-                                            clock_.TimeInMilliseconds(), true))
+    EXPECT_CALL(callback_,
+                TimeToSendPacket(ssrc, sequence_number,
+                                 clock_.TimeInMilliseconds(), true, _))
+        .WillOnce(Return(true));
+    EXPECT_CALL(callback_,
+                TimeToSendPacket(ssrc, sequence_number + 1,
+                                 clock_.TimeInMilliseconds(), true, _))
+        .WillOnce(Return(true));
+    EXPECT_CALL(callback_,
+                TimeToSendPacket(ssrc, sequence_number + 3,
+                                 clock_.TimeInMilliseconds() + 33, true, _))
         .WillOnce(Return(true));
-    EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 1,
-                                            clock_.TimeInMilliseconds(), true))
+    EXPECT_CALL(callback_,
+                TimeToSendPacket(ssrc, sequence_number + 2,
+                                 clock_.TimeInMilliseconds() + 33, true, _))
         .WillOnce(Return(true));
-    EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 3,
-                                            clock_.TimeInMilliseconds() + 33,
-                                            true)).WillOnce(Return(true));
-    EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 2,
-                                            clock_.TimeInMilliseconds() + 33,
-                                            true)).WillOnce(Return(true));
 
     while (send_bucket_->QueueSizePackets() > 0) {
       int time_until_process = send_bucket_->TimeUntilNextProcess();
@@ -839,21 +803,22 @@ TEST_F(PacedSenderTest, PaddingOveruse) {
   uint16_t sequence_number = 1234;
   const size_t kPacketSize = 1200;
 
-  // Min bitrate 0 => no padding, padding budget will stay at 0.
-  send_bucket_->UpdateBitrate(60, 90, 0);
+  send_bucket_->Process();
+  send_bucket_->SetEstimatedBitrate(60000);
+  send_bucket_->SetAllocatedSendBitrate(60000, 0);
+
   SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
                       clock_.TimeInMilliseconds(), kPacketSize, false);
   send_bucket_->Process();
 
   // Add 30kbit padding. When increasing budget, media budget will increase from
-  // negative (overuse) while padding budget will increase form 0.
+  // negative (overuse) while padding budget will increase from 0.
   clock_.AdvanceTimeMilliseconds(5);
-  send_bucket_->UpdateBitrate(60, 90, 30);
-
-  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
-                             sequence_number++, clock_.TimeInMilliseconds(),
-                             kPacketSize, false);
+  send_bucket_->SetAllocatedSendBitrate(60000, 30000);
 
+  SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
+                      clock_.TimeInMilliseconds(), kPacketSize, false);
+  EXPECT_LT(5u, send_bucket_->ExpectedQueueTimeMs());
   // Don't send padding if queue is non-empty, even if padding budget > 0.
   EXPECT_CALL(callback_, TimeToSendPadding(_)).Times(0);
   send_bucket_->Process();
@@ -864,9 +829,8 @@ TEST_F(PacedSenderTest, AverageQueueTime) {
   uint16_t sequence_number = 1234;
   const size_t kPacketSize = 1200;
   const int kBitrateBps = 10 * kPacketSize * 8;  // 10 packets per second.
-  const int kBitrateKbps = (kBitrateBps + 500) / 1000;
 
-  send_bucket_->UpdateBitrate(kBitrateKbps, kBitrateKbps, kBitrateKbps);
+  send_bucket_->SetEstimatedBitrate(kBitrateBps);
 
   EXPECT_EQ(0, send_bucket_->AverageQueueTimeMs());
 
@@ -885,7 +849,7 @@ TEST_F(PacedSenderTest, AverageQueueTime) {
   // Only first packet (queued for 20ms) should be removed, leave the second
   // packet (queued for 10ms) alone in the queue.
   EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number,
-                                          first_capture_time, false))
+                                          first_capture_time, false, _))
       .Times(1)
       .WillRepeatedly(Return(true));
   send_bucket_->Process();
@@ -894,7 +858,7 @@ TEST_F(PacedSenderTest, AverageQueueTime) {
 
   clock_.AdvanceTimeMilliseconds(10);
   EXPECT_CALL(callback_, TimeToSendPacket(ssrc, sequence_number + 1,
-                                          first_capture_time + 10, false))
+                                          first_capture_time + 10, false, _))
       .Times(1)
       .WillRepeatedly(Return(true));
   for (int i = 0; i < 3; ++i) {
@@ -905,5 +869,37 @@ TEST_F(PacedSenderTest, AverageQueueTime) {
   EXPECT_EQ(0, send_bucket_->AverageQueueTimeMs());
 }
 
+TEST_F(PacedSenderTest, ProbeClusterId) {
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 1200;
+
+  send_bucket_->SetProbingEnabled(true);
+  for (int i = 0; i < 11; ++i) {
+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
+                               sequence_number + i, clock_.TimeInMilliseconds(),
+                               kPacketSize, false);
+  }
+
+  // First probing cluster.
+  EXPECT_CALL(callback_, TimeToSendPacket(_, _, _, _, 0))
+      .Times(6)
+      .WillRepeatedly(Return(true));
+  for (int i = 0; i < 6; ++i)
+    send_bucket_->Process();
+
+  // Second probing cluster.
+  EXPECT_CALL(callback_, TimeToSendPacket(_, _, _, _, 1))
+      .Times(5)
+      .WillRepeatedly(Return(true));
+  for (int i = 0; i < 5; ++i)
+    send_bucket_->Process();
+
+  // No more probing packets.
+  EXPECT_CALL(callback_, TimeToSendPadding(_))
+      .Times(1);
+  send_bucket_->Process();
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/pacing/packet_router.cc b/chromium/third_party/webrtc/modules/pacing/packet_router.cc
index 5c7a7ab29a3..1884958aca4 100644
--- a/chromium/third_party/webrtc/modules/pacing/packet_router.cc
+++ b/chromium/third_party/webrtc/modules/pacing/packet_router.cc
@@ -43,7 +43,8 @@ void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
                                     uint16_t sequence_number,
                                     int64_t capture_timestamp,
-                                    bool retransmission) {
+                                    bool retransmission,
+                                    int probe_cluster_id) {
   RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
   rtc::CritScope cs(&modules_crit_);
   for (auto* rtp_module : rtp_modules_) {
diff --git a/chromium/third_party/webrtc/modules/pacing/packet_router.h b/chromium/third_party/webrtc/modules/pacing/packet_router.h
index 635b931225c..81d85404eee 100644
--- a/chromium/third_party/webrtc/modules/pacing/packet_router.h
+++ b/chromium/third_party/webrtc/modules/pacing/packet_router.h
@@ -30,7 +30,7 @@ class TransportFeedback;
 
 // PacketRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
-class PacketRouter : public PacedSender::Callback,
+class PacketRouter : public PacedSender::PacketSender,
                      public TransportSequenceNumberAllocator {
  public:
   PacketRouter();
@@ -43,7 +43,8 @@ class PacketRouter : public PacedSender::Callback,
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_timestamp,
-                        bool retransmission) override;
+                        bool retransmission,
+                        int probe_cluster_id) override;
 
   size_t TimeToSendPadding(size_t bytes) override;
 
diff --git a/chromium/third_party/webrtc/modules/pacing/packet_router_unittest.cc b/chromium/third_party/webrtc/modules/pacing/packet_router_unittest.cc
index faf270ced37..006b9f2bf48 100644
--- a/chromium/third_party/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/chromium/third_party/webrtc/modules/pacing/packet_router_unittest.cc
@@ -53,7 +53,7 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
       .WillOnce(Return(true));
   EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
   EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
-                                               timestamp, retransmission));
+                                               timestamp, retransmission, -1));
 
   // Send on the second module by letting rtp_2 be sending, but not rtp_1.
   ++sequence_number;
@@ -69,7 +69,7 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
       .Times(1)
       .WillOnce(Return(true));
   EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number,
-                                               timestamp, retransmission));
+                                               timestamp, retransmission, -1));
 
   // No module is sending, hence no packet should be sent.
   EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
@@ -77,7 +77,7 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
   EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
   EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
   EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
-                                               timestamp, retransmission));
+                                               timestamp, retransmission, -1));
 
   // Add a packet with incorrect ssrc and test it's dropped in the router.
   EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
@@ -87,7 +87,7 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
   EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
   EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
   EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1 + kSsrc2, sequence_number,
-                                               timestamp, retransmission));
+                                               timestamp, retransmission, -1));
 
   packet_router_->RemoveRtpModule(&rtp_1);
 
@@ -97,7 +97,7 @@ TEST_F(PacketRouterTest, TimeToSendPacket) {
   EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
   EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
   EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
-                                               timestamp, retransmission));
+                                               timestamp, retransmission, -1));
 
   packet_router_->RemoveRtpModule(&rtp_2);
 }
@@ -167,7 +167,7 @@ TEST_F(PacketRouterTest, SenderOnlyFunctionsRespectSendingMedia) {
 
   // Verify that TimeToSendPacket does not end up in a receiver.
   EXPECT_CALL(rtp, TimeToSendPacket(_, _, _, _)).Times(0);
-  EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc, 1, 1, false));
+  EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc, 1, 1, false, -1));
   // Verify that TimeToSendPadding does not end up in a receiver.
   EXPECT_CALL(rtp, TimeToSendPadding(_)).Times(0);
   EXPECT_EQ(0u, packet_router_->TimeToSendPadding(200));
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
index 4967913558c..d44788be274 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
@@ -11,6 +11,7 @@
 #include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h
index 2c35df872bc..61773037d74 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h
@@ -28,7 +28,7 @@ class MockRemoteBitrateEstimator : public RemoteBitrateEstimator {
  public:
   MOCK_METHOD1(IncomingPacketFeedbackVector,
                void(const std::vector<PacketInfo>&));
-  MOCK_METHOD4(IncomingPacket, void(int64_t, size_t, const RTPHeader&, bool));
+  MOCK_METHOD3(IncomingPacket, void(int64_t, size_t, const RTPHeader&));
   MOCK_METHOD1(RemoveStream, void(uint32_t));
   MOCK_CONST_METHOD2(LatestEstimate, bool(std::vector<uint32_t>*, uint32_t*));
 
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
index e56c273fc4b..d7d8d2c8d2e 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
@@ -58,8 +58,7 @@ class RemoteBitrateEstimator : public CallStatsObserver, public Module {
   // Note that |arrival_time_ms| can be of an arbitrary time base.
   virtual void IncomingPacket(int64_t arrival_time_ms,
                               size_t payload_size,
-                              const RTPHeader& header,
-                              bool was_paced) = 0;
+                              const RTPHeader& header) = 0;
 
   // Removes all data for |ssrc|.
   virtual void RemoveStream(uint32_t ssrc) = 0;
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
index a643c1f1030..83e87f95712 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
@@ -24,7 +24,7 @@ class SendTimeHistory {
   SendTimeHistory(Clock* clock, int64_t packet_age_limit);
   virtual ~SendTimeHistory();
 
-  void AddAndRemoveOld(uint16_t sequence_number, size_t length, bool was_paced);
+  void AddAndRemoveOld(uint16_t sequence_number, size_t length);
   bool OnSentPacket(uint16_t sequence_number, int64_t timestamp);
   // Look up PacketInfo for a sent packet, based on the sequence number, and
   // populate all fields except for receive_time. The packet parameter must
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
index f75bc2b03ea..1b7ce07583d 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
@@ -18,7 +18,7 @@
 
 namespace webrtc {
 
-static const int kBurstDeltaThresholdMs  = 5;
+static const int kBurstDeltaThresholdMs = 5;
 
 InterArrival::InterArrival(uint32_t timestamp_group_length_ticks,
                            double timestamp_to_ms_coeff,
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
index 32663d729b8..7b20cf7f9db 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
@@ -62,6 +62,7 @@
           'type': 'static_library',
           'dependencies': [
             '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(DEPTH)/testing/gmock.gyp:gmock',
           ],
           'sources': [
             'test/bwe.cc',
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 7c2abb2f08f..82335055b43 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -14,12 +14,12 @@
 
 #include <algorithm>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
@@ -79,23 +79,19 @@ bool RemoteBitrateEstimatorAbsSendTime::IsWithinClusterBounds(
   }
 
   RemoteBitrateEstimatorAbsSendTime::RemoteBitrateEstimatorAbsSendTime(
-      RemoteBitrateObserver* observer,
-      Clock* clock)
+      RemoteBitrateObserver* observer)
       : observer_(observer),
         inter_arrival_(),
-        estimator_(OverUseDetectorOptions()),
+        estimator_(),
         detector_(OverUseDetectorOptions()),
         incoming_bitrate_(kBitrateWindowMs, 8000),
         total_probes_received_(0),
         first_packet_time_ms_(-1),
         last_update_ms_(-1),
-        ssrcs_(),
-        clock_(clock) {
+        ssrcs_() {
     RTC_DCHECK(observer_);
-    RTC_DCHECK(clock_);
     LOG(LS_INFO) << "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
     network_thread_.DetachFromThread();
-    process_thread_.DetachFromThread();
 }
 
 void RemoteBitrateEstimatorAbsSendTime::ComputeClusters(
@@ -180,8 +176,7 @@ RemoteBitrateEstimatorAbsSendTime::ProcessClusters(int64_t now_ms) {
         std::min(best_it->GetSendBitrateBps(), best_it->GetRecvBitrateBps());
     // Make sure that a probe sent on a lower bitrate than our estimate can't
     // reduce the estimate.
-    if (IsBitrateImproving(probe_bitrate_bps) &&
-        probe_bitrate_bps > static_cast<int>(incoming_bitrate_.Rate(now_ms))) {
+    if (IsBitrateImproving(probe_bitrate_bps)) {
       LOG(LS_INFO) << "Probe successful, sent at "
                    << best_it->GetSendBitrateBps() << " bps, received at "
                    << best_it->GetRecvBitrateBps()
@@ -215,14 +210,14 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketFeedbackVector(
   for (const auto& packet_info : packet_feedback_vector) {
     IncomingPacketInfo(packet_info.arrival_time_ms,
                        ConvertMsTo24Bits(packet_info.send_time_ms),
-                       packet_info.payload_size, 0, packet_info.was_paced);
+                       packet_info.payload_size, 0);
   }
 }
 
-void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(int64_t arrival_time_ms,
-                                                       size_t payload_size,
-                                                       const RTPHeader& header,
-                                                       bool was_paced) {
+void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(
+    int64_t arrival_time_ms,
+    size_t payload_size,
+    const RTPHeader& header) {
   RTC_DCHECK(network_thread_.CalledOnValidThread());
   if (!header.extension.hasAbsoluteSendTime) {
     LOG(LS_WARNING) << "RemoteBitrateEstimatorAbsSendTimeImpl: Incoming packet "
@@ -230,36 +225,31 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(int64_t arrival_time_ms,
     return;
   }
   IncomingPacketInfo(arrival_time_ms, header.extension.absoluteSendTime,
-                     payload_size, header.ssrc, was_paced);
+                     payload_size, header.ssrc);
 }
 
 void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
     int64_t arrival_time_ms,
     uint32_t send_time_24bits,
     size_t payload_size,
-    uint32_t ssrc,
-    bool was_paced) {
+    uint32_t ssrc) {
   assert(send_time_24bits < (1ul << 24));
   // Shift up send time to use the full 32 bits that inter_arrival works with,
   // so wrapping works properly.
   uint32_t timestamp = send_time_24bits << kAbsSendTimeInterArrivalUpshift;
   int64_t send_time_ms = static_cast<int64_t>(timestamp) * kTimestampToMs;
 
-  int64_t now_ms = clock_->TimeInMilliseconds();
+  int64_t now_ms = arrival_time_ms;
   // TODO(holmer): SSRCs are only needed for REMB, should be broken out from
   // here.
   incoming_bitrate_.Update(payload_size, now_ms);
 
   if (first_packet_time_ms_ == -1)
-    first_packet_time_ms_ = clock_->TimeInMilliseconds();
+    first_packet_time_ms_ = arrival_time_ms;
 
   uint32_t ts_delta = 0;
   int64_t t_delta = 0;
   int size_delta = 0;
-  // For now only try to detect probes while we don't have a valid estimate, and
-  // make sure the packet was paced. We currently assume that only packets
-  // larger than 200 bytes are paced by the sender.
-  was_paced = was_paced && payload_size > PacedSender::kMinProbePacketSize;
   bool update_estimate = false;
   uint32_t target_bitrate_bps = 0;
   std::vector<uint32_t> ssrcs;
@@ -267,9 +257,14 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
     rtc::CritScope lock(&crit_);
 
     TimeoutStreams(now_ms);
+    RTC_DCHECK(inter_arrival_.get());
+    RTC_DCHECK(estimator_.get());
     ssrcs_[ssrc] = now_ms;
 
-    if (was_paced &&
+    // For now only try to detect probes while we don't have a valid estimate.
+    // We currently assume that only packets larger than 200 bytes are paced by
+    // the sender.
+    if (payload_size > PacedSender::kMinProbePacketSize &&
         (!remote_rate_.ValidEstimate() ||
          now_ms - first_packet_time_ms_ < kInitialProbingIntervalMs)) {
       // TODO(holmer): Use a map instead to get correct order?
@@ -295,9 +290,9 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
     if (inter_arrival_->ComputeDeltas(timestamp, arrival_time_ms, payload_size,
                                       &ts_delta, &t_delta, &size_delta)) {
       double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
-      estimator_.Update(t_delta, ts_delta_ms, size_delta, detector_.State());
-      detector_.Detect(estimator_.offset(), ts_delta_ms,
-                       estimator_.num_of_deltas(), arrival_time_ms);
+      estimator_->Update(t_delta, ts_delta_ms, size_delta, detector_.State());
+      detector_.Detect(estimator_->offset(), ts_delta_ms,
+                       estimator_->num_of_deltas(), arrival_time_ms);
     }
 
     if (!update_estimate) {
@@ -319,7 +314,7 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
       // and the target bitrate is too high compared to what we are receiving.
       const RateControlInput input(detector_.State(),
                                    incoming_bitrate_.Rate(now_ms),
-                                   estimator_.var_noise());
+                                   estimator_->var_noise());
       remote_rate_.Update(&input, now_ms);
       target_bitrate_bps = remote_rate_.UpdateBandwidthEstimate(now_ms);
       update_estimate = remote_rate_.ValidEstimate();
@@ -352,6 +347,7 @@ void RemoteBitrateEstimatorAbsSendTime::TimeoutStreams(int64_t now_ms) {
     inter_arrival_.reset(
         new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
                          kTimestampToMs, true));
+    estimator_.reset(new OveruseEstimator(OverUseDetectorOptions()));
     // We deliberately don't reset the first_packet_time_ms_ here for now since
     // we only probe for bandwidth in the beginning of a call right now.
   }
@@ -359,7 +355,6 @@ void RemoteBitrateEstimatorAbsSendTime::TimeoutStreams(int64_t now_ms) {
 
 void RemoteBitrateEstimatorAbsSendTime::OnRttUpdate(int64_t avg_rtt_ms,
                                                     int64_t max_rtt_ms) {
-  RTC_DCHECK(process_thread_.CalledOnValidThread());
   rtc::CritScope lock(&crit_);
   remote_rate_.SetRtt(avg_rtt_ms);
 }
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
index 1f47dc3b244..9403a6cc8b9 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
@@ -17,6 +17,7 @@
 #include <vector>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/base/thread_checker.h"
@@ -67,8 +68,7 @@ struct Cluster {
 
 class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
  public:
-  RemoteBitrateEstimatorAbsSendTime(RemoteBitrateObserver* observer,
-                                    Clock* clock);
+  explicit RemoteBitrateEstimatorAbsSendTime(RemoteBitrateObserver* observer);
   virtual ~RemoteBitrateEstimatorAbsSendTime() {}
 
   void IncomingPacketFeedbackVector(
@@ -76,8 +76,7 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
 
   void IncomingPacket(int64_t arrival_time_ms,
                       size_t payload_size,
-                      const RTPHeader& header,
-                      bool was_paced) override;
+                      const RTPHeader& header) override;
   // This class relies on Process() being called periodically (at least once
   // every other second) for streams to be timed out properly. Therefore it
   // shouldn't be detached from the ProcessThread except if it's about to be
@@ -102,8 +101,7 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
   void IncomingPacketInfo(int64_t arrival_time_ms,
                           uint32_t send_time_24bits,
                           size_t payload_size,
-                          uint32_t ssrc,
-                          bool was_paced);
+                          uint32_t ssrc);
 
   void ComputeClusters(std::list<Cluster>* clusters) const;
 
@@ -121,7 +119,7 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
   rtc::ThreadChecker network_thread_;
   RemoteBitrateObserver* const observer_;
   std::unique_ptr<InterArrival> inter_arrival_;
-  OveruseEstimator estimator_;
+  std::unique_ptr<OveruseEstimator> estimator_;
   OveruseDetector detector_;
   RateStatistics incoming_bitrate_;
   std::vector<int> recent_propagation_delta_ms_;
@@ -131,11 +129,9 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
   int64_t first_packet_time_ms_;
   int64_t last_update_ms_;
 
-  rtc::ThreadChecker process_thread_;
   rtc::CriticalSection crit_;
   Ssrcs ssrcs_ GUARDED_BY(&crit_);
   AimdRateControl remote_rate_ GUARDED_BY(&crit_);
-  Clock* const clock_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RemoteBitrateEstimatorAbsSendTime);
 };
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
index e8026a5764d..a4e4150e769 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
@@ -20,7 +20,7 @@ class RemoteBitrateEstimatorAbsSendTimeTest :
   RemoteBitrateEstimatorAbsSendTimeTest() {}
   virtual void SetUp() {
     bitrate_estimator_.reset(new RemoteBitrateEstimatorAbsSendTime(
-        bitrate_observer_.get(), &clock_));
+        bitrate_observer_.get()));
   }
  protected:
   RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorAbsSendTimeTest);
@@ -35,15 +35,15 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseReordering) {
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseRtpTimestamps) {
-  RateIncreaseRtpTimestampsTestHelper(1232);
+  RateIncreaseRtpTimestampsTestHelper(1229);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropOneStream) {
-  CapacityDropTestHelper(1, false, 633);
+  CapacityDropTestHelper(1, false, 667);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropOneStreamWrap) {
-  CapacityDropTestHelper(1, true, 633);
+  CapacityDropTestHelper(1, true, 667);
 }
 
 TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropTwoStreamsWrap) {
@@ -90,7 +90,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProcessAfterTimeout) {
   // RemoteBitrateEstimator.
   const int64_t kStreamTimeOutMs = 2000;
   const int64_t kProcessIntervalMs = 1000;
-  IncomingPacket(0, 1000, clock_.TimeInMilliseconds(), 0, 0, true);
+  IncomingPacket(0, 1000, clock_.TimeInMilliseconds(), 0, 0);
   clock_.AdvanceTimeMilliseconds(kStreamTimeOutMs + 1);
   // Trigger timeout.
   bitrate_estimator_->Process();
@@ -106,16 +106,14 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetection) {
   for (int i = 0; i < kProbeLength; ++i) {
     clock_.AdvanceTimeMilliseconds(10);
     now_ms = clock_.TimeInMilliseconds();
-    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
-                   true);
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
   }
 
   // Second burst sent at 8 * 1000 / 5 = 1600 kbps.
   for (int i = 0; i < kProbeLength; ++i) {
     clock_.AdvanceTimeMilliseconds(5);
     now_ms = clock_.TimeInMilliseconds();
-    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
-                   true);
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -132,12 +130,10 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
   for (int i = 0; i < kProbeLength; ++i) {
     clock_.AdvanceTimeMilliseconds(5);
     now_ms = clock_.TimeInMilliseconds();
-    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
-                   true);
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
     // Non-paced packet, arriving 5 ms after.
     clock_.AdvanceTimeMilliseconds(5);
-    IncomingPacket(0, 100, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
-                   false);
+    IncomingPacket(0, 100, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -158,7 +154,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     now_ms = clock_.TimeInMilliseconds();
     send_time_ms += 10;
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000), true);
+                   AbsSendTime(send_time_ms, 1000));
   }
 
   // Second burst sent at 8 * 1000 / 5 = 1600 kbps, arriving at 8 * 1000 / 8 =
@@ -168,7 +164,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     now_ms = clock_.TimeInMilliseconds();
     send_time_ms += 5;
     IncomingPacket(0, 1000, now_ms, send_time_ms,
-                   AbsSendTime(send_time_ms, 1000), true);
+                   AbsSendTime(send_time_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -188,7 +184,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     send_time_ms += 10;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000), true);
+                   AbsSendTime(send_time_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -207,7 +203,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetectionFasterArrival) {
     send_time_ms += 10;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000), true);
+                   AbsSendTime(send_time_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -225,7 +221,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetectionSlowerArrival) {
     send_time_ms += 5;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000), true);
+                   AbsSendTime(send_time_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -245,7 +241,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
     send_time_ms += 1;
     now_ms = clock_.TimeInMilliseconds();
     IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
-                   AbsSendTime(send_time_ms, 1000), true);
+                   AbsSendTime(send_time_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -261,8 +257,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, ProbingIgnoresSmallPackets) {
   for (int i = 0; i < kProbeLength; ++i) {
     clock_.AdvanceTimeMilliseconds(10);
     now_ms = clock_.TimeInMilliseconds();
-    IncomingPacket(0, 200, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
-                   true);
+    IncomingPacket(0, 200, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
   }
 
   bitrate_estimator_->Process();
@@ -273,8 +268,7 @@ TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, ProbingIgnoresSmallPackets) {
   for (int i = 0; i < kProbeLength; ++i) {
     clock_.AdvanceTimeMilliseconds(10);
     now_ms = clock_.TimeInMilliseconds();
-    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000),
-                   true);
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
   }
 
   // Wait long enough so that we can call Process again.
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index f38ef783067..b5adb9fa70a 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -66,10 +66,10 @@ RemoteBitrateEstimatorSingleStream::~RemoteBitrateEstimatorSingleStream() {
   }
 }
 
-void RemoteBitrateEstimatorSingleStream::IncomingPacket(int64_t arrival_time_ms,
-                                                        size_t payload_size,
-                                                        const RTPHeader& header,
-                                                        bool was_paced) {
+void RemoteBitrateEstimatorSingleStream::IncomingPacket(
+    int64_t arrival_time_ms,
+    size_t payload_size,
+    const RTPHeader& header) {
   uint32_t ssrc = header.ssrc;
   uint32_t rtp_timestamp = header.timestamp +
       header.extension.transmissionTimeOffset;
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
index 5516ea781da..2f74e1c4af0 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
@@ -15,6 +15,7 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
@@ -30,8 +31,7 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
 
   void IncomingPacket(int64_t arrival_time_ms,
                       size_t payload_size,
-                      const RTPHeader& header,
-                      bool was_paced) override;
+                      const RTPHeader& header) override;
   void Process() override;
   int64_t TimeUntilNextProcess() override;
   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
index 6fd0ad11b57..97e3abaa32b 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
@@ -47,7 +47,7 @@ TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropOneStreamWrap) {
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropTwoStreamsWrap) {
-  CapacityDropTestHelper(2, true, 767);
+  CapacityDropTestHelper(2, true, 600);
 }
 
 TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThreeStreamsWrap) {
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
index 8bfb8ed0fd3..4530053a869 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
@@ -221,8 +221,7 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
                                                 size_t payload_size,
                                                 int64_t arrival_time,
                                                 uint32_t rtp_timestamp,
-                                                uint32_t absolute_send_time,
-                                                bool was_paced) {
+                                                uint32_t absolute_send_time) {
   RTPHeader header;
   memset(&header, 0, sizeof(header));
   header.ssrc = ssrc;
@@ -230,7 +229,7 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
   header.extension.hasAbsoluteSendTime = true;
   header.extension.absoluteSendTime = absolute_send_time;
   bitrate_estimator_->IncomingPacket(arrival_time + kArrivalTimeClockOffsetMs,
-                                     payload_size, header, was_paced);
+                                     payload_size, header);
 }
 
 // Generates a frame of packets belonging to a stream at a given bitrate and
@@ -255,7 +254,7 @@ bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(uint32_t ssrc,
                                    clock_.TimeInMicroseconds());
     IncomingPacket(packet->ssrc, packet->size,
                    (packet->arrival_time + 500) / 1000, packet->rtp_timestamp,
-                   AbsSendTime(packet->send_time, 1000000), true);
+                   AbsSendTime(packet->send_time, 1000000));
     if (bitrate_observer_->updated()) {
       if (bitrate_observer_->latest_bitrate() < bitrate_bps)
         overuse = true;
@@ -319,7 +318,7 @@ void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper(
   clock_.AdvanceTimeMilliseconds(1000);
   // Inserting a packet. Still no valid estimate. We need to wait 5 seconds.
   IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
-                 absolute_send_time, true);
+                 absolute_send_time);
   bitrate_estimator_->Process();
   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
   EXPECT_EQ(0u, ssrcs.size());
@@ -328,7 +327,7 @@ void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper(
   // Inserting packets for 5 seconds to get a valid estimate.
   for (int i = 0; i < 5 * kFramerate + 1; ++i) {
     IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time, true);
+                   absolute_send_time);
     clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
     timestamp += 90 * kFrameIntervalMs;
     absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -356,13 +355,13 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
   uint32_t timestamp = 0;
   uint32_t absolute_send_time = 0;
   IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                 absolute_send_time, true);
+                 absolute_send_time);
   bitrate_estimator_->Process();
   EXPECT_FALSE(bitrate_observer_->updated());  // No valid estimate.
   // Inserting packets for one second to get a valid estimate.
   for (int i = 0; i < 5 * kFramerate + 1; ++i) {
     IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time, true);
+                   absolute_send_time);
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     timestamp += 90 * kFrameIntervalMs;
     absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -379,12 +378,12 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
     absolute_send_time = AddAbsSendTime(absolute_send_time,
                                         2 * kFrameIntervalAbsSendTime);
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time, true);
-    IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
-                   timestamp - 90 * kFrameIntervalMs,
-                   AddAbsSendTime(absolute_send_time,
-                                  -static_cast<int>(kFrameIntervalAbsSendTime)),
-                   true);
+                   absolute_send_time);
+    IncomingPacket(
+        kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
+        timestamp - 90 * kFrameIntervalMs,
+        AddAbsSendTime(absolute_send_time,
+                       -static_cast<int>(kFrameIntervalAbsSendTime)));
   }
   bitrate_estimator_->Process();
   EXPECT_TRUE(bitrate_observer_->updated());
@@ -517,7 +516,7 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
   // time for the first estimate to be generated and for Process() to be called.
   for (int i = 0; i <= 6 * kFramerate; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time, true);
+                   absolute_send_time);
     bitrate_estimator_->Process();
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     timestamp += 90 * kFrameIntervalMs;
@@ -538,7 +537,7 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
       // Insert |kTimestampGroupLength| frames with just 1 timestamp ticks in
       // between. Should be treated as part of the same group by the estimator.
       IncomingPacket(kDefaultSsrc, 100, clock_.TimeInMilliseconds(), timestamp,
-                     absolute_send_time, true);
+                     absolute_send_time);
       clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
       timestamp += 1;
       absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -568,7 +567,7 @@ void RemoteBitrateEstimatorTest::TestWrappingHelper(
 
   for (size_t i = 0; i < 3000; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time, true);
+                   absolute_send_time);
     timestamp += kFrameIntervalMs;
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     absolute_send_time = AddAbsSendTime(absolute_send_time,
@@ -583,9 +582,9 @@ void RemoteBitrateEstimatorTest::TestWrappingHelper(
   absolute_send_time = AddAbsSendTime(absolute_send_time,
                                       AbsSendTime(silence_time_s, 1));
   bitrate_estimator_->Process();
-  for (size_t i = 0; i < 10; ++i) {
+  for (size_t i = 0; i < 21; ++i) {
     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
-                   absolute_send_time, true);
+                   absolute_send_time);
     timestamp += kFrameIntervalMs;
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     absolute_send_time = AddAbsSendTime(absolute_send_time,
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
index 12ac9e8502d..b4bff670e75 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
@@ -173,8 +173,7 @@ class RemoteBitrateEstimatorTest : public ::testing::Test {
                       size_t payload_size,
                       int64_t arrival_time,
                       uint32_t rtp_timestamp,
-                      uint32_t absolute_send_time,
-                      bool was_paced);
+                      uint32_t absolute_send_time);
 
   // Generates a frame of packets belonging to a stream at a given bitrate and
   // with a given ssrc. The stream is pushed through a very simple simulated
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
index 31cd9f98028..e307242f8d3 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
@@ -16,6 +16,7 @@
 #include <algorithm>
 #include <sstream>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/random.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
index eace9fc8b0a..2172bce9380 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
@@ -43,8 +43,7 @@ void RemoteEstimatorProxy::IncomingPacketFeedbackVector(
 
 void RemoteEstimatorProxy::IncomingPacket(int64_t arrival_time_ms,
                                           size_t payload_size,
-                                          const RTPHeader& header,
-                                          bool was_paced) {
+                                          const RTPHeader& header) {
   if (!header.extension.hasTransportSequenceNumber) {
     LOG(LS_WARNING) << "RemoteEstimatorProxy: Incoming packet "
                        "is missing the transport sequence number extension!";
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
index 93d5244b67b..66373e29778 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
@@ -39,8 +39,7 @@ class RemoteEstimatorProxy : public RemoteBitrateEstimator {
       const std::vector<PacketInfo>& packet_feedback_vector) override;
   void IncomingPacket(int64_t arrival_time_ms,
                       size_t payload_size,
-                      const RTPHeader& header,
-                      bool was_paced) override;
+                      const RTPHeader& header) override;
   void RemoveStream(uint32_t ssrc) override {}
   bool LatestEstimate(std::vector<unsigned int>* ssrcs,
                       unsigned int* bitrate_bps) const override;
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
index 3c3c7297e18..a1264b2ff95 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
@@ -37,7 +37,7 @@ class RemoteEstimatorProxyTest : public ::testing::Test {
     header.extension.hasTransportSequenceNumber = true;
     header.extension.transportSequenceNumber = seq;
     header.ssrc = kMediaSsrc;
-    proxy_.IncomingPacket(time_ms, kDefaultPacketSize, header, true);
+    proxy_.IncomingPacket(time_ms, kDefaultPacketSize, header);
   }
 
   void Process() {
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
index a58d12a1600..f4fe2208a81 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
@@ -26,9 +26,7 @@ void SendTimeHistory::Clear() {
   history_.clear();
 }
 
-void SendTimeHistory::AddAndRemoveOld(uint16_t sequence_number,
-                                      size_t length,
-                                      bool was_paced) {
+void SendTimeHistory::AddAndRemoveOld(uint16_t sequence_number, size_t length) {
   EraseOld();
 
   if (history_.empty())
@@ -36,7 +34,7 @@ void SendTimeHistory::AddAndRemoveOld(uint16_t sequence_number,
 
   history_.insert(std::pair<uint16_t, PacketInfo>(
       sequence_number, PacketInfo(clock_->TimeInMilliseconds(), 0, -1,
-                                  sequence_number, length, was_paced)));
+                                  sequence_number, length)));
 }
 
 bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc
index b525813cdca..7500f575fb7 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc
@@ -33,9 +33,8 @@ class SendTimeHistoryTest : public ::testing::Test {
 
   void AddPacketWithSendTime(uint16_t sequence_number,
                              size_t length,
-                             bool was_paced,
                              int64_t send_time_ms) {
-    history_.AddAndRemoveOld(sequence_number, length, was_paced);
+    history_.AddAndRemoveOld(sequence_number, length);
     history_.OnSentPacket(sequence_number, send_time_ms);
   }
 
@@ -46,42 +45,40 @@ class SendTimeHistoryTest : public ::testing::Test {
 // Help class extended so we can do EXPECT_EQ and collections.
 class PacketInfo : public webrtc::PacketInfo {
  public:
-  PacketInfo() : webrtc::PacketInfo(-1, 0, 0, 0, 0, false) {}
+  PacketInfo() : webrtc::PacketInfo(-1, 0, 0, 0, 0) {}
   PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
-      : PacketInfo(arrival_time_ms, 0, sequence_number, 0, false) {}
+      : PacketInfo(arrival_time_ms, 0, sequence_number, 0) {}
   PacketInfo(int64_t arrival_time_ms,
              int64_t send_time_ms,
              uint16_t sequence_number,
-             size_t payload_size,
-             bool was_paced)
+             size_t payload_size)
       : webrtc::PacketInfo(-1,
                            arrival_time_ms,
                            send_time_ms,
                            sequence_number,
-                           payload_size,
-                           was_paced) {}
+                           payload_size) {}
   bool operator==(const PacketInfo& other) const {
     return arrival_time_ms == other.arrival_time_ms &&
            send_time_ms == other.send_time_ms &&
            sequence_number == other.sequence_number &&
-           payload_size == other.payload_size && was_paced == other.was_paced;
+           payload_size == other.payload_size;
   }
 };
 
 TEST_F(SendTimeHistoryTest, AddRemoveOne) {
   const uint16_t kSeqNo = 10;
-  const PacketInfo kSentPacket(0, 1, kSeqNo, 1, true);
-  AddPacketWithSendTime(kSeqNo, 1, true, 1);
+  const PacketInfo kSentPacket(0, 1, kSeqNo, 1);
+  AddPacketWithSendTime(kSeqNo, 1, 1);
 
-  PacketInfo received_packet(0, 0, kSeqNo, 0, false);
+  PacketInfo received_packet(0, 0, kSeqNo, 0);
   EXPECT_TRUE(history_.GetInfo(&received_packet, false));
   EXPECT_EQ(kSentPacket, received_packet);
 
-  PacketInfo received_packet2(0, 0, kSeqNo, 0, false);
+  PacketInfo received_packet2(0, 0, kSeqNo, 0);
   EXPECT_TRUE(history_.GetInfo(&received_packet2, true));
   EXPECT_EQ(kSentPacket, received_packet2);
 
-  PacketInfo received_packet3(0, 0, kSeqNo, 0, false);
+  PacketInfo received_packet3(0, 0, kSeqNo, 0);
   EXPECT_FALSE(history_.GetInfo(&received_packet3, true));
 }
 
@@ -90,9 +87,8 @@ TEST_F(SendTimeHistoryTest, PopulatesExpectedFields) {
   const int64_t kSendTime = 1000;
   const int64_t kReceiveTime = 2000;
   const size_t kPayloadSize = 42;
-  const bool kPaced = true;
 
-  AddPacketWithSendTime(kSeqNo, kPayloadSize, kPaced, kSendTime);
+  AddPacketWithSendTime(kSeqNo, kPayloadSize, kSendTime);
 
   PacketInfo info(kReceiveTime, kSeqNo);
   EXPECT_TRUE(history_.GetInfo(&info, true));
@@ -100,7 +96,6 @@ TEST_F(SendTimeHistoryTest, PopulatesExpectedFields) {
   EXPECT_EQ(kSendTime, info.send_time_ms);
   EXPECT_EQ(kSeqNo, info.sequence_number);
   EXPECT_EQ(kPayloadSize, info.payload_size);
-  EXPECT_EQ(kPaced, info.was_paced);
 }
 
 TEST_F(SendTimeHistoryTest, AddThenRemoveOutOfOrder) {
@@ -109,19 +104,16 @@ TEST_F(SendTimeHistoryTest, AddThenRemoveOutOfOrder) {
   const size_t num_items = 100;
   const size_t kPacketSize = 400;
   const size_t kTransmissionTime = 1234;
-  const bool kPaced = true;
   for (size_t i = 0; i < num_items; ++i) {
     sent_packets.push_back(PacketInfo(0, static_cast<int64_t>(i),
-                                      static_cast<uint16_t>(i), kPacketSize,
-                                      kPaced));
+                                      static_cast<uint16_t>(i), kPacketSize));
     received_packets.push_back(
         PacketInfo(static_cast<int64_t>(i) + kTransmissionTime, 0,
-                   static_cast<uint16_t>(i), kPacketSize, false));
+                   static_cast<uint16_t>(i), kPacketSize));
   }
   for (size_t i = 0; i < num_items; ++i) {
     history_.AddAndRemoveOld(sent_packets[i].sequence_number,
-                             sent_packets[i].payload_size,
-                             sent_packets[i].was_paced);
+                             sent_packets[i].payload_size);
   }
   for (size_t i = 0; i < num_items; ++i)
     history_.OnSentPacket(sent_packets[i].sequence_number,
@@ -143,19 +135,19 @@ TEST_F(SendTimeHistoryTest, HistorySize) {
   const int kItems = kDefaultHistoryLengthMs / 100;
   for (int i = 0; i < kItems; ++i) {
     clock_.AdvanceTimeMilliseconds(100);
-    AddPacketWithSendTime(i, 0, false, i * 100);
+    AddPacketWithSendTime(i, 0, i * 100);
   }
   for (int i = 0; i < kItems; ++i) {
-    PacketInfo info(0, 0, static_cast<uint16_t>(i), 0, false);
+    PacketInfo info(0, 0, static_cast<uint16_t>(i), 0);
     EXPECT_TRUE(history_.GetInfo(&info, false));
     EXPECT_EQ(i * 100, info.send_time_ms);
   }
   clock_.AdvanceTimeMilliseconds(101);
-  AddPacketWithSendTime(kItems, 0, false, kItems * 101);
-  PacketInfo info(0, 0, 0, 0, false);
+  AddPacketWithSendTime(kItems, 0, kItems * 101);
+  PacketInfo info(0, 0, 0, 0);
   EXPECT_FALSE(history_.GetInfo(&info, false));
   for (int i = 1; i < (kItems + 1); ++i) {
-    PacketInfo info2(0, 0, static_cast<uint16_t>(i), 0, false);
+    PacketInfo info2(0, 0, static_cast<uint16_t>(i), 0);
     EXPECT_TRUE(history_.GetInfo(&info2, false));
     int64_t expected_time_ms = (i == kItems) ? i * 101 : i * 100;
     EXPECT_EQ(expected_time_ms, info2.send_time_ms);
@@ -164,16 +156,16 @@ TEST_F(SendTimeHistoryTest, HistorySize) {
 
 TEST_F(SendTimeHistoryTest, HistorySizeWithWraparound) {
   const uint16_t kMaxSeqNo = std::numeric_limits<uint16_t>::max();
-  AddPacketWithSendTime(kMaxSeqNo - 2, 0, false, 0);
+  AddPacketWithSendTime(kMaxSeqNo - 2, 0, 0);
 
   clock_.AdvanceTimeMilliseconds(100);
-  AddPacketWithSendTime(kMaxSeqNo - 1, 1, false, 100);
+  AddPacketWithSendTime(kMaxSeqNo - 1, 1, 100);
 
   clock_.AdvanceTimeMilliseconds(100);
-  AddPacketWithSendTime(kMaxSeqNo, 0, false, 200);
+  AddPacketWithSendTime(kMaxSeqNo, 0, 200);
 
   clock_.AdvanceTimeMilliseconds(kDefaultHistoryLengthMs - 200 + 1);
-  AddPacketWithSendTime(0, 0, false, kDefaultHistoryLengthMs);
+  AddPacketWithSendTime(0, 0, kDefaultHistoryLengthMs);
 
   PacketInfo info(0, static_cast<uint16_t>(kMaxSeqNo - 2));
   EXPECT_FALSE(history_.GetInfo(&info, false));
@@ -189,7 +181,7 @@ TEST_F(SendTimeHistoryTest, HistorySizeWithWraparound) {
   EXPECT_TRUE(history_.GetInfo(&info5, true));
 
   clock_.AdvanceTimeMilliseconds(100);
-  AddPacketWithSendTime(1, 0, false, 1100);
+  AddPacketWithSendTime(1, 0, 1100);
 
   PacketInfo info6(0, static_cast<uint16_t>(kMaxSeqNo - 2));
   EXPECT_FALSE(history_.GetInfo(&info6, false));
@@ -206,26 +198,26 @@ TEST_F(SendTimeHistoryTest, HistorySizeWithWraparound) {
 TEST_F(SendTimeHistoryTest, InterlievedGetAndRemove) {
   const uint16_t kSeqNo = 1;
   const int64_t kTimestamp = 2;
-  PacketInfo packets[3] = {{0, kTimestamp, kSeqNo, 0, false},
-                           {0, kTimestamp + 1, kSeqNo + 1, 0, false},
-                           {0, kTimestamp + 2, kSeqNo + 2, 0, false}};
+  PacketInfo packets[3] = {{0, kTimestamp, kSeqNo, 0},
+                           {0, kTimestamp + 1, kSeqNo + 1, 0},
+                           {0, kTimestamp + 2, kSeqNo + 2, 0}};
 
   AddPacketWithSendTime(packets[0].sequence_number, packets[0].payload_size,
-                        packets[0].was_paced, packets[0].send_time_ms);
+                        packets[0].send_time_ms);
   AddPacketWithSendTime(packets[1].sequence_number, packets[1].payload_size,
-                        packets[1].was_paced, packets[1].send_time_ms);
-  PacketInfo info(0, 0, packets[0].sequence_number, 0, false);
+                        packets[1].send_time_ms);
+  PacketInfo info(0, 0, packets[0].sequence_number, 0);
   EXPECT_TRUE(history_.GetInfo(&info, true));
   EXPECT_EQ(packets[0], info);
 
   AddPacketWithSendTime(packets[2].sequence_number, packets[2].payload_size,
-                        packets[2].was_paced, packets[2].send_time_ms);
+                        packets[2].send_time_ms);
 
-  PacketInfo info2(0, 0, packets[1].sequence_number, 0, false);
+  PacketInfo info2(0, 0, packets[1].sequence_number, 0);
   EXPECT_TRUE(history_.GetInfo(&info2, true));
   EXPECT_EQ(packets[1], info2);
 
-  PacketInfo info3(0, 0, packets[2].sequence_number, 0, false);
+  PacketInfo info3(0, 0, packets[2].sequence_number, 0);
   EXPECT_TRUE(history_.GetInfo(&info3, true));
   EXPECT_EQ(packets[2], info3);
 }
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
index 5e282c6f087..b21a269ec6b 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
@@ -13,6 +13,7 @@
 #include <limits>
 
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h"
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.h
index bd016cf318b..fc3018f9967 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe.h
@@ -16,6 +16,7 @@
 #include <sstream>
 #include <string>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
index 41bf836c9e9..a9fd617118a 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
@@ -14,6 +14,8 @@
 
 #include <sstream>
 
+#include "webrtc/base/constructormagic.h"
+
 namespace webrtc {
 namespace testing {
 namespace bwe {
@@ -97,18 +99,14 @@ Packet::Packet()
       creation_time_us_(-1),
       send_time_us_(-1),
       sender_timestamp_us_(-1),
-      payload_size_(0),
-      paced_(false) {
-}
+      payload_size_(0) {}
 
 Packet::Packet(int flow_id, int64_t send_time_us, size_t payload_size)
     : flow_id_(flow_id),
       creation_time_us_(send_time_us),
       send_time_us_(send_time_us),
       sender_timestamp_us_(send_time_us),
-      payload_size_(payload_size),
-      paced_(false) {
-}
+      payload_size_(payload_size) {}
 
 Packet::~Packet() {
 }
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
index 223b20f21c5..1fe3a228e4b 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
@@ -25,6 +25,7 @@
 #include <vector>
 
 #include "webrtc/base/common.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/random.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/include/module_common_types.h"
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
index 37009c77018..2a33440bc3b 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
@@ -21,6 +21,7 @@
 #include <map>
 #include <memory>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/voice_engine/channel.h"
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
index d469e675e41..e2d3da9632a 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
@@ -69,7 +69,7 @@ RembReceiver::RembReceiver(int flow_id, bool plot)
       recv_stats_(ReceiveStatistics::Create(&clock_)),
       latest_estimate_bps_(-1),
       last_feedback_ms_(-1),
-      estimator_(new RemoteBitrateEstimatorAbsSendTime(this, &clock_)) {
+      estimator_(new RemoteBitrateEstimatorAbsSendTime(this)) {
   std::stringstream ss;
   ss << "Estimate_" << flow_id_ << "#1";
   estimate_log_prefix_ = ss.str();
@@ -95,7 +95,7 @@ void RembReceiver::ReceivePacket(int64_t arrival_time_ms,
     step_ms = std::max<int64_t>(estimator_->TimeUntilNextProcess(), 0);
   }
   estimator_->IncomingPacket(arrival_time_ms, media_packet.payload_size(),
-                             media_packet.header(), true);
+                             media_packet.header());
   clock_.AdvanceTimeMilliseconds(arrival_time_ms - clock_.TimeInMilliseconds());
   ASSERT_TRUE(arrival_time_ms == clock_.TimeInMilliseconds());
 
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
index 5e6b6edb256..3dc4f388c8f 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
index 36dff1fb2ac..c54a7b05949 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
@@ -23,7 +23,7 @@ const int kFeedbackIntervalMs = 50;
 FullBweSender::FullBweSender(int kbps, BitrateObserver* observer, Clock* clock)
     : bitrate_controller_(
           BitrateController::CreateBitrateController(clock, observer)),
-      rbe_(new RemoteBitrateEstimatorAbsSendTime(this, clock)),
+      rbe_(new RemoteBitrateEstimatorAbsSendTime(this)),
       feedback_observer_(bitrate_controller_->CreateRtcpBandwidthObserver()),
       clock_(clock),
       send_time_history_(clock_, 10000),
@@ -93,8 +93,7 @@ void FullBweSender::OnPacketsSent(const Packets& packets) {
     if (packet->GetPacketType() == Packet::kMedia) {
       MediaPacket* media_packet = static_cast<MediaPacket*>(packet);
       send_time_history_.AddAndRemoveOld(media_packet->header().sequenceNumber,
-                                         media_packet->payload_size(),
-                                         packet->paced());
+                                         media_packet->payload_size());
       send_time_history_.OnSentPacket(media_packet->header().sequenceNumber,
                                       media_packet->sender_timestamp_ms());
     }
@@ -126,7 +125,7 @@ void SendSideBweReceiver::ReceivePacket(int64_t arrival_time_ms,
                                         const MediaPacket& media_packet) {
   packet_feedback_vector_.push_back(PacketInfo(
       -1, arrival_time_ms, media_packet.sender_timestamp_ms(),
-      media_packet.header().sequenceNumber, media_packet.payload_size(), true));
+      media_packet.header().sequenceNumber, media_packet.payload_size()));
 
   // Log received packet information.
   BweReceiver::ReceivePacket(arrival_time_ms, media_packet);
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet.h
index 4a361c4dc20..9aa596c1bd7 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet.h
@@ -42,8 +42,6 @@ class Packet {
   virtual void set_sender_timestamp_us(int64_t sender_timestamp_us) {
     sender_timestamp_us_ = sender_timestamp_us;
   }
-  virtual void set_paced(bool paced) { paced_ = paced; }
-  virtual bool paced() const { return paced_; }
   virtual int64_t creation_time_ms() const {
     return (creation_time_us_ + 500) / 1000;
   }
@@ -58,7 +56,6 @@ class Packet {
   int64_t send_time_us_;  // Time the packet left last processor touching it.
   int64_t sender_timestamp_us_;  // Time the packet left the Sender.
   size_t payload_size_;  // Size of the (non-existent, simulated) payload.
-  bool paced_;           // True if sent through paced sender.
 };
 
 class MediaPacket : public Packet {
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
index 3bcbc0a071b..7ffeb5584b0 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
@@ -157,13 +157,9 @@ PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener,
                                    VideoSource* source,
                                    BandwidthEstimatorType estimator)
     : VideoSender(listener, source, estimator),
-      pacer_(&clock_,
-             this,
-             source->bits_per_second() / 1000,
-             PacedSender::kDefaultPaceMultiplier * source->bits_per_second() /
-                 1000,
-             0) {
+      pacer_(&clock_, this) {
   modules_.push_back(&pacer_);
+  pacer_.SetEstimatedBitrate(source->bits_per_second());
 }
 
 PacedVideoSender::~PacedVideoSender() {
@@ -271,8 +267,6 @@ void PacedVideoSender::QueuePackets(Packets* batch,
   }
   Packets to_transfer;
   to_transfer.splice(to_transfer.begin(), queue_, queue_.begin(), it);
-  for (Packet* packet : to_transfer)
-    packet->set_paced(true);
   bwe_->OnPacketsSent(to_transfer);
   batch->merge(to_transfer, DereferencingComparator<Packet>);
 }
@@ -280,7 +274,8 @@ void PacedVideoSender::QueuePackets(Packets* batch,
 bool PacedVideoSender::TimeToSendPacket(uint32_t ssrc,
                                         uint16_t sequence_number,
                                         int64_t capture_time_ms,
-                                        bool retransmission) {
+                                        bool retransmission,
+                                        int probe_cluster_id) {
   for (Packets::iterator it = pacer_queue_.begin(); it != pacer_queue_.end();
        ++it) {
     MediaPacket* media_packet = static_cast<MediaPacket*>(*it);
@@ -310,9 +305,7 @@ void PacedVideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
                                         uint8_t fraction_lost,
                                         int64_t rtt) {
   VideoSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
-  pacer_.UpdateBitrate(
-      target_bitrate_bps / 1000,
-      PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000, 0);
+  pacer_.SetEstimatedBitrate(target_bitrate_bps);
 }
 
 const int kNoLimit = std::numeric_limits<int>::max();
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
index 5ed4a3bc380..4990574bdec 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
@@ -100,7 +100,7 @@ class VideoSender : public PacketSender, public BitrateObserver {
   RTC_DISALLOW_COPY_AND_ASSIGN(VideoSender);
 };
 
-class PacedVideoSender : public VideoSender, public PacedSender::Callback {
+class PacedVideoSender : public VideoSender, public PacedSender::PacketSender {
  public:
   PacedVideoSender(PacketProcessorListener* listener,
                    VideoSource* source,
@@ -113,7 +113,8 @@ class PacedVideoSender : public VideoSender, public PacedSender::Callback {
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_time_ms,
-                        bool retransmission) override;
+                        bool retransmission,
+                        int probe_cluster_id) override;
   size_t TimeToSendPadding(size_t bytes) override;
 
   // Implements BitrateObserver.
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
index f138035de53..7ae6ede363e 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -117,7 +117,7 @@ bool ParseArgsAndSetupEstimator(int argc,
     switch (extension) {
       case webrtc::kRtpExtensionAbsoluteSendTime: {
         *estimator =
-            new webrtc::RemoteBitrateEstimatorAbsSendTime(observer, clock);
+            new webrtc::RemoteBitrateEstimatorAbsSendTime(observer);
           *estimator_used = "AbsoluteSendTimeRemoteBitrateEstimator";
           break;
         }
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
index 08dc0e63a02..f5dbaef3717 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
@@ -83,7 +83,7 @@ int main(int argc, char** argv) {
           packet_length = packet.original_length;
         }
         rbe->IncomingPacket(clock.TimeInMilliseconds(),
-                            packet_length - header.headerLength, header, true);
+                            packet_length - header.headerLength, header);
         ++packet_counter;
       }
       if (!rtp_reader->NextPacket(&packet)) {
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
index f7e07a5dc51..a02f407d072 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
 
+#include <algorithm>
 #include <limits>
 
 #include "webrtc/base/checks.h"
@@ -26,6 +27,17 @@ const int64_t kBaseTimestampScaleFactor =
     rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
 
+class PacketInfoComparator {
+ public:
+  inline bool operator()(const PacketInfo& lhs, const PacketInfo& rhs) {
+    if (lhs.arrival_time_ms != rhs.arrival_time_ms)
+      return lhs.arrival_time_ms < rhs.arrival_time_ms;
+    if (lhs.send_time_ms != rhs.send_time_ms)
+      return lhs.send_time_ms < rhs.send_time_ms;
+    return lhs.sequence_number < rhs.sequence_number;
+  }
+};
+
 TransportFeedbackAdapter::TransportFeedbackAdapter(
     BitrateController* bitrate_controller,
     Clock* clock)
@@ -46,10 +58,9 @@ void TransportFeedbackAdapter::SetBitrateEstimator(
 }
 
 void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
-                                         size_t length,
-                                         bool was_paced) {
+                                         size_t length) {
   rtc::CritScope cs(&lock_);
-  send_time_history_.AddAndRemoveOld(sequence_number, length, was_paced);
+  send_time_history_.AddAndRemoveOld(sequence_number, length);
 }
 
 void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
@@ -104,6 +115,8 @@ void TransportFeedbackAdapter::OnTransportFeedback(
       }
       ++sequence_number;
     }
+    std::sort(packet_feedback_vector.begin(), packet_feedback_vector.end(),
+              PacketInfoComparator());
     RTC_DCHECK(delta_it == delta_vec.end());
     if (failed_lookups > 0) {
       LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
index c97ef57cf05..867570f26b9 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
@@ -38,9 +38,7 @@ class TransportFeedbackAdapter : public TransportFeedbackObserver,
   }
 
   // Implements TransportFeedbackObserver.
-  void AddPacket(uint16_t sequence_number,
-                 size_t length,
-                 bool was_paced) override;
+  void AddPacket(uint16_t sequence_number, size_t length) override;
   void OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
 
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
index f3be09206ec..239f2ec561d 100644
--- a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
@@ -92,15 +92,13 @@ class TransportFeedbackAdapterTest : public ::testing::Test {
       EXPECT_EQ(truth[i].send_time_ms, input[i].send_time_ms);
       EXPECT_EQ(truth[i].sequence_number, input[i].sequence_number);
       EXPECT_EQ(truth[i].payload_size, input[i].payload_size);
-      EXPECT_EQ(truth[i].was_paced, input[i].was_paced);
     }
   }
 
   // Utility method, to reset arrival_time_ms before adding send time.
   void OnSentPacket(PacketInfo info) {
     info.arrival_time_ms = 0;
-    adapter_->AddPacket(info.sequence_number, info.payload_size,
-                        info.was_paced);
+    adapter_->AddPacket(info.sequence_number, info.payload_size);
     adapter_->OnSentPacket(info.sequence_number, info.send_time_ms);
   }
 
@@ -114,11 +112,11 @@ class TransportFeedbackAdapterTest : public ::testing::Test {
 
 TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
   std::vector<PacketInfo> packets;
-  packets.push_back(PacketInfo(100, 200, 0, 1500, true));
-  packets.push_back(PacketInfo(110, 210, 1, 1500, true));
-  packets.push_back(PacketInfo(120, 220, 2, 1500, true));
-  packets.push_back(PacketInfo(130, 230, 3, 1500, true));
-  packets.push_back(PacketInfo(140, 240, 4, 1500, true));
+  packets.push_back(PacketInfo(100, 200, 0, 1500));
+  packets.push_back(PacketInfo(110, 210, 1, 1500));
+  packets.push_back(PacketInfo(120, 220, 2, 1500));
+  packets.push_back(PacketInfo(130, 230, 3, 1500));
+  packets.push_back(PacketInfo(140, 240, 4, 1500));
 
   for (const PacketInfo& packet : packets)
     OnSentPacket(packet);
@@ -145,11 +143,11 @@ TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
 
 TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
   std::vector<PacketInfo> packets;
-  packets.push_back(PacketInfo(100, 200, 0, 1500, true));
-  packets.push_back(PacketInfo(110, 210, 1, 1500, true));
-  packets.push_back(PacketInfo(120, 220, 2, 1500, true));
-  packets.push_back(PacketInfo(130, 230, 3, 1500, true));
-  packets.push_back(PacketInfo(140, 240, 4, 1500, true));
+  packets.push_back(PacketInfo(100, 200, 0, 1500));
+  packets.push_back(PacketInfo(110, 210, 1, 1500));
+  packets.push_back(PacketInfo(120, 220, 2, 1500));
+  packets.push_back(PacketInfo(130, 230, 3, 1500));
+  packets.push_back(PacketInfo(140, 240, 4, 1500));
 
   const uint16_t kSendSideDropBefore = 1;
   const uint16_t kReceiveSideDropAfter = 3;
@@ -190,9 +188,9 @@ TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
                                static_cast<int64_t>(1 << 8) *
                                static_cast<int64_t>((1 << 23) - 1) / 1000;
   std::vector<PacketInfo> packets;
-  packets.push_back(PacketInfo(kHighArrivalTimeMs - 64, 200, 0, 1500, true));
-  packets.push_back(PacketInfo(kHighArrivalTimeMs + 64, 210, 1, 1500, true));
-  packets.push_back(PacketInfo(kHighArrivalTimeMs, 220, 2, 1500, true));
+  packets.push_back(PacketInfo(kHighArrivalTimeMs - 64, 200, 0, 1500));
+  packets.push_back(PacketInfo(kHighArrivalTimeMs + 64, 210, 1, 1500));
+  packets.push_back(PacketInfo(kHighArrivalTimeMs, 220, 2, 1500));
 
   for (const PacketInfo& packet : packets)
     OnSentPacket(packet);
@@ -207,8 +205,8 @@ TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
         packets[i].sequence_number, packets[i].arrival_time_ms * 1000));
 
     rtc::Buffer raw_packet = feedback->Build();
-    feedback = rtc::ScopedToUnique(rtcp::TransportFeedback::ParseFrom(
-        raw_packet.data(), raw_packet.size()));
+    feedback = rtcp::TransportFeedback::ParseFrom(raw_packet.data(),
+                                                  raw_packet.size());
 
     std::vector<PacketInfo> expected_packets;
     expected_packets.push_back(packets[i]);
@@ -223,6 +221,39 @@ TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
   }
 }
 
+TEST_F(TransportFeedbackAdapterTest, HandlesReordering) {
+  std::vector<PacketInfo> packets;
+  packets.push_back(PacketInfo(120, 200, 0, 1500));
+  packets.push_back(PacketInfo(110, 210, 1, 1500));
+  packets.push_back(PacketInfo(100, 220, 2, 1500));
+  std::vector<PacketInfo> expected_packets;
+  expected_packets.push_back(packets[2]);
+  expected_packets.push_back(packets[1]);
+  expected_packets.push_back(packets[0]);
+
+  for (const PacketInfo& packet : packets)
+    OnSentPacket(packet);
+
+  rtcp::TransportFeedback feedback;
+  feedback.WithBase(packets[0].sequence_number,
+                    packets[0].arrival_time_ms * 1000);
+
+  for (const PacketInfo& packet : packets) {
+    EXPECT_TRUE(feedback.WithReceivedPacket(packet.sequence_number,
+                                            packet.arrival_time_ms * 1000));
+  }
+
+  feedback.Build();
+
+  EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+      .Times(1)
+      .WillOnce(Invoke([expected_packets,
+                        this](const std::vector<PacketInfo>& feedback_vector) {
+        ComparePacketVectors(expected_packets, feedback_vector);
+      }));
+  adapter_->OnTransportFeedback(feedback);
+}
+
 TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
   std::vector<PacketInfo> sent_packets;
   const int64_t kSmallDeltaUs =
@@ -257,6 +288,14 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
   info.arrival_time_ms += (kLargePositiveDeltaUs + 1000) / 1000;
   ++info.sequence_number;
 
+  // Expected to be ordered on arrival time when the feedback message has been
+  // parsed.
+  std::vector<PacketInfo> expected_packets;
+  expected_packets.push_back(sent_packets[0]);
+  expected_packets.push_back(sent_packets[3]);
+  expected_packets.push_back(sent_packets[1]);
+  expected_packets.push_back(sent_packets[2]);
+
   // Packets will be added to send history.
   for (const PacketInfo& packet : sent_packets)
     OnSentPacket(packet);
@@ -276,17 +315,17 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
                                             info.arrival_time_ms * 1000));
 
   rtc::Buffer raw_packet = feedback->Build();
-  feedback = rtc::ScopedToUnique(
-      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size()));
+  feedback =
+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
 
   std::vector<PacketInfo> received_feedback;
 
   EXPECT_TRUE(feedback.get() != nullptr);
   EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
       .Times(1)
-      .WillOnce(Invoke([sent_packets, &received_feedback](
+      .WillOnce(Invoke([expected_packets, &received_feedback](
           const std::vector<PacketInfo>& feedback_vector) {
-        EXPECT_EQ(sent_packets.size(), feedback_vector.size());
+        EXPECT_EQ(expected_packets.size(), feedback_vector.size());
         received_feedback = feedback_vector;
       }));
   adapter_->OnTransportFeedback(*feedback.get());
@@ -297,8 +336,8 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
   EXPECT_TRUE(feedback->WithReceivedPacket(info.sequence_number,
                                            info.arrival_time_ms * 1000));
   raw_packet = feedback->Build();
-  feedback = rtc::ScopedToUnique(
-      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size()));
+  feedback =
+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
 
   EXPECT_TRUE(feedback.get() != nullptr);
   EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
@@ -310,9 +349,9 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
           }));
   adapter_->OnTransportFeedback(*feedback.get());
 
-  sent_packets.push_back(info);
+  expected_packets.push_back(info);
 
-  ComparePacketVectors(sent_packets, received_feedback);
+  ComparePacketVectors(expected_packets, received_feedback);
 }
 
 }  // namespace test
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/BUILD.gn b/chromium/third_party/webrtc/modules/rtp_rtcp/BUILD.gn
index d386951cb00..9d69811ef31 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -118,9 +118,15 @@ source_set("rtp_rtcp") {
     "source/rtp_format_vp9.h",
     "source/rtp_header_extension.cc",
     "source/rtp_header_extension.h",
+    "source/rtp_header_extensions.cc",
+    "source/rtp_header_extensions.h",
     "source/rtp_header_parser.cc",
+    "source/rtp_packet.cc",
+    "source/rtp_packet.h",
     "source/rtp_packet_history.cc",
     "source/rtp_packet_history.h",
+    "source/rtp_packet_received.h",
+    "source/rtp_packet_to_send.h",
     "source/rtp_payload_registry.cc",
     "source/rtp_receiver_audio.cc",
     "source/rtp_receiver_audio.h",
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/chromium/third_party/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
index 56c6e48691c..207e749a02a 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -11,7 +11,9 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 
 namespace webrtc {
@@ -40,7 +42,7 @@ class RemoteNtpTimeEstimator {
 
  private:
   Clock* clock_;
-  rtc::scoped_ptr<TimestampExtrapolator> ts_extrapolator_;
+  std::unique_ptr<TimestampExtrapolator> ts_extrapolator_;
   RtcpList rtcp_list_;
   int64_t last_timing_log_ms_;
   RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index fae864107f0..a199755aafe 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -12,8 +12,9 @@
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
 
 #include <map>
+#include <memory>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
@@ -85,15 +86,6 @@ class RTPPayloadRegistry {
 
   bool IsRtx(const RTPHeader& header) const;
 
-  // DEPRECATED. Use RestoreOriginalPacket below that takes a uint8_t*
-  // restored_packet, instead of a uint8_t**.
-  // TODO(noahric): Remove this when all callers have been updated.
-  bool RestoreOriginalPacket(uint8_t** restored_packet,
-                             const uint8_t* packet,
-                             size_t* packet_length,
-                             uint32_t original_ssrc,
-                             const RTPHeader& header) const;
-
   bool RestoreOriginalPacket(uint8_t* restored_packet,
                              const uint8_t* packet,
                              size_t* packet_length,
@@ -110,19 +102,10 @@ class RTPPayloadRegistry {
 
   int GetPayloadTypeFrequency(uint8_t payload_type) const;
 
-  // DEPRECATED. Use PayloadTypeToPayload below that returns const Payload*
-  // instead of taking output parameter.
-  // TODO(danilchap): Remove this when all callers have been updated.
-  bool PayloadTypeToPayload(const uint8_t payload_type,
-                            RtpUtility::Payload*& payload) const {  // NOLINT
-    payload =
-        const_cast<RtpUtility::Payload*>(PayloadTypeToPayload(payload_type));
-    return payload != nullptr;
-  }
   const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
 
   void ResetLastReceivedPayloadTypes() {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     last_received_payload_type_ = -1;
     last_received_media_payload_type_ = -1;
   }
@@ -136,34 +119,34 @@ class RTPPayloadRegistry {
   bool ReportMediaPayloadType(uint8_t media_payload_type);
 
   int8_t red_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     return red_payload_type_;
   }
   int8_t ulpfec_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     return ulpfec_payload_type_;
   }
   int8_t last_received_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     return last_received_payload_type_;
   }
   void set_last_received_payload_type(int8_t last_received_payload_type) {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     last_received_payload_type_ = last_received_payload_type;
   }
 
   int8_t last_received_media_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     return last_received_media_payload_type_;
   }
 
   bool use_rtx_payload_mapping_on_restore() const {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     return use_rtx_payload_mapping_on_restore_;
   }
 
   void set_use_rtx_payload_mapping_on_restore(bool val) {
-    CriticalSectionScoped cs(crit_sect_.get());
+    rtc::CritScope cs(&crit_sect_);
     use_rtx_payload_mapping_on_restore_ = val;
   }
 
@@ -178,9 +161,9 @@ class RTPPayloadRegistry {
 
   bool IsRtxInternal(const RTPHeader& header) const;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  rtc::CriticalSection crit_sect_;
   RtpUtility::PayloadTypeMap payload_type_map_;
-  rtc::scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
+  std::unique_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
   int8_t  red_payload_type_;
   int8_t ulpfec_payload_type_;
   int8_t incoming_payload_type_;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
index f393e41eabd..9db1c63da78 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
@@ -75,12 +75,6 @@ class RtpReceiver {
                                  PayloadUnion payload_specific,
                                  bool in_order) = 0;
 
-  // Returns the currently configured NACK method.
-  virtual NACKMethod NACK() const = 0;
-
-  // Turn negative acknowledgement (NACK) requests on/off.
-  virtual void SetNACKStatus(const NACKMethod method) = 0;
-
   // Gets the last received timestamp. Returns true if a packet has been
   // received, false otherwise.
   virtual bool Timestamp(uint32_t* timestamp) const = 0;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index d01465b9f8f..66589888bda 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -16,6 +16,7 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
@@ -77,7 +78,7 @@ class RtpRtcp : public Module {
     FrameCountObserver* send_frame_count_observer;
     SendSideDelayObserver* send_side_delay_observer;
     RtcEventLog* event_log;
-
+    SendPacketObserver* send_packet_observer;
     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
   };
 
@@ -210,10 +211,10 @@ class RtpRtcp : public Module {
     */
     virtual void SetSequenceNumber(uint16_t seq) = 0;
 
-    // Returns true if the ssrc matched this module, false otherwise.
-    virtual bool SetRtpStateForSsrc(uint32_t ssrc,
-                                    const RtpState& rtp_state) = 0;
-    virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
+    virtual void SetRtpState(const RtpState& rtp_state) = 0;
+    virtual void SetRtxState(const RtpState& rtp_state) = 0;
+    virtual RtpState GetRtpState() const = 0;
+    virtual RtpState GetRtxState() const = 0;
 
     /*
     *   Get SSRC
@@ -539,6 +540,9 @@ class RtpRtcp : public Module {
 
     /*
     *   Send NACK for the packets specified.
+    *
+    *   Note: This assumes the caller keeps track of timing and doesn't rely on
+    *   the RTP module to do this.
     */
     virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
 
@@ -594,12 +598,6 @@ class RtpRtcp : public Module {
     *
     *   return -1 on failure else 0
     */
-    // DEPRECATED. Use SendREDPayloadType below that takes output parameter
-    // by pointer instead of by reference.
-    // TODO(danilchap): Remove this when all callers have been updated.
-    int32_t SendREDPayloadType(int8_t& payloadType) const {  // NOLINT
-      return SendREDPayloadType(&payloadType);
-    }
     virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 0;
      /*
      * Store the audio level in dBov for header-extension-for-audio-level-
@@ -632,14 +630,6 @@ class RtpRtcp : public Module {
     /*
     *   Get generic FEC setting
     */
-    // DEPRECATED. Use GenericFECStatus below that takes output parameters
-    // by pointers instead of by references.
-    // TODO(danilchap): Remove this when all callers have been updated.
-    void GenericFECStatus(bool& enable,               // NOLINT
-                          uint8_t& payloadTypeRED,    // NOLINT
-                          uint8_t& payloadTypeFEC) {  // NOLINT
-      GenericFECStatus(&enable, &payloadTypeRED, &payloadTypeFEC);
-    }
     virtual void GenericFECStatus(bool* enable,
                                   uint8_t* payload_type_red,
                                   uint8_t* payload_type_fec) = 0;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 9acef79f383..8dc8b0651d2 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -99,8 +99,6 @@ enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
 
 enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 };
 
-enum NACKMethod { kNackOff = 0, kNackRtcp = 2 };
-
 enum RetransmissionMode : uint8_t {
   kRetransmitOff = 0x0,
   kRetransmitFECPackets = 0x1,
@@ -247,32 +245,28 @@ class RtcpBandwidthObserver {
 
 struct PacketInfo {
   PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
-      : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {}
+      : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0) {}
 
   PacketInfo(int64_t arrival_time_ms,
              int64_t send_time_ms,
              uint16_t sequence_number,
-             size_t payload_size,
-             bool was_paced)
+             size_t payload_size)
       : PacketInfo(-1,
                    arrival_time_ms,
                    send_time_ms,
                    sequence_number,
-                   payload_size,
-                   was_paced) {}
+                   payload_size) {}
 
   PacketInfo(int64_t creation_time_ms,
              int64_t arrival_time_ms,
              int64_t send_time_ms,
              uint16_t sequence_number,
-             size_t payload_size,
-             bool was_paced)
+             size_t payload_size)
       : creation_time_ms(creation_time_ms),
         arrival_time_ms(arrival_time_ms),
         send_time_ms(send_time_ms),
         sequence_number(sequence_number),
-        payload_size(payload_size),
-        was_paced(was_paced) {}
+        payload_size(payload_size) {}
 
   // Time corresponding to when this object was created.
   int64_t creation_time_ms;
@@ -287,8 +281,6 @@ struct PacketInfo {
   uint16_t sequence_number;
   // Size of the packet excluding RTP headers.
   size_t payload_size;
-  // True if the packet was paced out by the pacer.
-  bool was_paced;
 };
 
 class TransportFeedbackObserver {
@@ -298,9 +290,7 @@ class TransportFeedbackObserver {
 
   // Note: Transport-wide sequence number as sequence number. Arrival time
   // must be set to 0.
-  virtual void AddPacket(uint16_t sequence_number,
-                         size_t length,
-                         bool was_paced) = 0;
+  virtual void AddPacket(uint16_t sequence_number, size_t length) = 0;
 
   virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
 };
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 9991aa21108..bf5e9369215 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -85,12 +85,12 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD0(StartTimestamp,
       uint32_t());
   MOCK_METHOD1(SetStartTimestamp, void(const uint32_t timestamp));
-  MOCK_CONST_METHOD0(SequenceNumber,
-      uint16_t());
+  MOCK_CONST_METHOD0(SequenceNumber, uint16_t());
   MOCK_METHOD1(SetSequenceNumber, void(const uint16_t seq));
-  MOCK_METHOD2(SetRtpStateForSsrc,
-               bool(uint32_t ssrc, const RtpState& rtp_state));
-  MOCK_METHOD2(GetRtpStateForSsrc, bool(uint32_t ssrc, RtpState* rtp_state));
+  MOCK_METHOD1(SetRtpState, void(const RtpState& rtp_state));
+  MOCK_METHOD1(SetRtxState, void(const RtpState& rtp_state));
+  MOCK_CONST_METHOD0(GetRtpState, RtpState());
+  MOCK_CONST_METHOD0(GetRtxState, RtpState());
   MOCK_CONST_METHOD0(SSRC,
       uint32_t());
   MOCK_METHOD1(SetSSRC,
@@ -202,10 +202,6 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_METHOD1(SetTMMBRStatus, void(const bool enable));
   MOCK_METHOD1(OnBandwidthEstimateUpdate,
       void(uint16_t bandWidthKbit));
-  MOCK_CONST_METHOD0(NACK,
-      NACKMethod());
-  MOCK_METHOD2(SetNACKStatus,
-      int32_t(const NACKMethod method, int oldestSequenceNumberToNack));
   MOCK_CONST_METHOD0(SelectiveRetransmissions,
       int());
   MOCK_METHOD1(SetSelectiveRetransmissions,
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/chromium/third_party/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
index 23a64b752f7..3f1e935b2a6 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
@@ -35,10 +35,6 @@
         'source/receive_statistics_impl.cc',
         'source/receive_statistics_impl.h',
         'source/remote_ntp_time_estimator.cc',
-        'source/rtp_header_parser.cc',
-        'source/rtp_rtcp_config.h',
-        'source/rtp_rtcp_impl.cc',
-        'source/rtp_rtcp_impl.h',
         'source/rtcp_packet.cc',
         'source/rtcp_packet.h',
         'source/rtcp_packet/app.cc',
@@ -103,8 +99,18 @@
         'source/rtcp_utility.h',
         'source/rtp_header_extension.cc',
         'source/rtp_header_extension.h',
+        'source/rtp_header_extensions.cc',
+        'source/rtp_header_extensions.h',
+        'source/rtp_header_parser.cc',
+        'source/rtp_packet.cc',
+        'source/rtp_packet.h',
+        'source/rtp_packet_received.h',
+        'source/rtp_packet_to_send.h',
         'source/rtp_receiver_impl.cc',
         'source/rtp_receiver_impl.h',
+        'source/rtp_rtcp_config.h',
+        'source/rtp_rtcp_impl.cc',
+        'source/rtp_rtcp_impl.h',
         'source/rtp_sender.cc',
         'source/rtp_sender.h',
         'source/rtp_utility.cc',
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.cc
index 4e9fc72c1f8..49a23592bfe 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.cc
@@ -11,13 +11,11 @@
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 Bitrate::Bitrate(Clock* clock, Observer* observer)
     : clock_(clock),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
       packet_rate_(0),
       bitrate_(0),
       bitrate_next_idx_(0),
@@ -33,23 +31,23 @@ Bitrate::Bitrate(Clock* clock, Observer* observer)
 Bitrate::~Bitrate() {}
 
 void Bitrate::Update(const size_t bytes) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   bytes_count_ += bytes;
   packet_count_++;
 }
 
 uint32_t Bitrate::PacketRate() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   return packet_rate_;
 }
 
 uint32_t Bitrate::BitrateLast() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   return bitrate_;
 }
 
 uint32_t Bitrate::BitrateNow() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   int64_t now = clock_->TimeInMilliseconds();
   int64_t diff_ms = now - time_last_rate_update_;
 
@@ -67,7 +65,7 @@ uint32_t Bitrate::BitrateNow() const {
 }
 
 int64_t Bitrate::time_last_rate_update() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   return time_last_rate_update_;
 }
 
@@ -75,7 +73,7 @@ int64_t Bitrate::time_last_rate_update() const {
 void Bitrate::Process() {
   BitrateStatistics stats;
   {
-    CriticalSectionScoped cs(crit_.get());
+    rtc::CritScope cs(&crit_);
     int64_t now = clock_->CurrentNtpInMilliseconds();
     int64_t diff_ms = now - time_last_rate_update_;
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.h
index 393d05d3e3e..7aaaead42d2 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/bitrate.h
@@ -15,7 +15,7 @@
 
 #include <list>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
@@ -23,7 +23,6 @@
 namespace webrtc {
 
 class Clock;
-class CriticalSectionWrapper;
 
 class Bitrate {
  public:
@@ -60,7 +59,7 @@ class Bitrate {
   Clock* clock_;
 
  private:
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  rtc::CriticalSection crit_;
   uint32_t packet_rate_;
   uint32_t bitrate_;
   uint8_t bitrate_next_idx_;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
index ab21b8704a5..81e8b5926e2 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -13,20 +13,16 @@
 #include <string.h>
 
 namespace webrtc {
-DTMFqueue::DTMFqueue()
-    : dtmf_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
-      next_empty_index_(0) {
+DTMFqueue::DTMFqueue() : next_empty_index_(0) {
   memset(dtmf_key_, 0, sizeof(dtmf_key_));
   memset(dtmf_length, 0, sizeof(dtmf_length));
   memset(dtmf_level_, 0, sizeof(dtmf_level_));
 }
 
-DTMFqueue::~DTMFqueue() {
-  delete dtmf_critsect_;
-}
+DTMFqueue::~DTMFqueue() {}
 
 int32_t DTMFqueue::AddDTMF(uint8_t key, uint16_t len, uint8_t level) {
-  CriticalSectionScoped lock(dtmf_critsect_);
+  rtc::CritScope lock(&dtmf_critsect_);
 
   if (next_empty_index_ >= DTMF_OUTBAND_MAX) {
     return -1;
@@ -40,7 +36,7 @@ int32_t DTMFqueue::AddDTMF(uint8_t key, uint16_t len, uint8_t level) {
 }
 
 int8_t DTMFqueue::NextDTMF(uint8_t* dtmf_key, uint16_t* len, uint8_t* level) {
-  CriticalSectionScoped lock(dtmf_critsect_);
+  rtc::CritScope lock(&dtmf_critsect_);
   if (next_empty_index_ == 0)
     return -1;
 
@@ -60,12 +56,12 @@ int8_t DTMFqueue::NextDTMF(uint8_t* dtmf_key, uint16_t* len, uint8_t* level) {
 }
 
 bool DTMFqueue::PendingDTMF() {
-  CriticalSectionScoped lock(dtmf_critsect_);
+  rtc::CritScope lock(&dtmf_critsect_);
   return next_empty_index_ > 0;
 }
 
 void DTMFqueue::ResetDTMF() {
-  CriticalSectionScoped lock(dtmf_critsect_);
+  rtc::CritScope lock(&dtmf_critsect_);
   next_empty_index_ = 0;
 }
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
index d1b3f5667cf..c0e616f9827 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -27,7 +27,7 @@ class DTMFqueue {
   void ResetDTMF();
 
  private:
-  CriticalSectionWrapper* dtmf_critsect_;
+  rtc::CriticalSection dtmf_critsect_;
   uint8_t next_empty_index_;
   uint8_t dtmf_key_[DTMF_OUTBAND_MAX];
   uint16_t dtmf_length[DTMF_OUTBAND_MAX];
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
index 2109574e396..83bd2849df4 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
@@ -12,11 +12,11 @@
 
 #include <assert.h>
 
+#include <memory>
+
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // RFC 5109
 namespace webrtc {
@@ -26,8 +26,7 @@ FecReceiver* FecReceiver::Create(RtpData* callback) {
 }
 
 FecReceiverImpl::FecReceiverImpl(RtpData* callback)
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      recovered_packet_callback_(callback),
+    : recovered_packet_callback_(callback),
       fec_(new ForwardErrorCorrection()) {}
 
 FecReceiverImpl::~FecReceiverImpl() {
@@ -42,7 +41,7 @@ FecReceiverImpl::~FecReceiverImpl() {
 }
 
 FecPacketCounter FecReceiverImpl::GetPacketCounter() const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   return packet_counter_;
 }
 
@@ -77,7 +76,7 @@ FecPacketCounter FecReceiverImpl::GetPacketCounter() const {
 int32_t FecReceiverImpl::AddReceivedRedPacket(
     const RTPHeader& header, const uint8_t* incoming_rtp_packet,
     size_t packet_length, uint8_t ulpfec_payload_type) {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   uint8_t REDHeaderLength = 1;
   size_t payload_data_length = packet_length - header.headerLength;
 
@@ -89,7 +88,7 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
   // Add to list without RED header, aka a virtual RTP packet
   // we remove the RED header
 
-  rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
       new ForwardErrorCorrection::ReceivedPacket);
   received_packet->pkt = new ForwardErrorCorrection::Packet;
 
@@ -137,7 +136,7 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
   }
   ++packet_counter_.num_packets;
 
-  rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket>
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>
       second_received_packet;
   if (blockLength > 0) {
     // handle block length, split into 2 packets
@@ -219,21 +218,21 @@ int32_t FecReceiverImpl::AddReceivedRedPacket(
 }
 
 int32_t FecReceiverImpl::ProcessReceivedFec() {
-  crit_sect_->Enter();
+  crit_sect_.Enter();
   if (!received_packet_list_.empty()) {
     // Send received media packet to VCM.
     if (!received_packet_list_.front()->is_fec) {
       ForwardErrorCorrection::Packet* packet =
           received_packet_list_.front()->pkt;
-      crit_sect_->Leave();
+      crit_sect_.Leave();
       if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
                                                          packet->length)) {
         return -1;
       }
-      crit_sect_->Enter();
+      crit_sect_.Enter();
     }
     if (fec_->DecodeFEC(&received_packet_list_, &recovered_packet_list_) != 0) {
-      crit_sect_->Leave();
+      crit_sect_.Leave();
       return -1;
     }
     assert(received_packet_list_.empty());
@@ -246,15 +245,15 @@ int32_t FecReceiverImpl::ProcessReceivedFec() {
       continue;
     ForwardErrorCorrection::Packet* packet = (*it)->pkt;
     ++packet_counter_.num_recovered_packets;
-    crit_sect_->Leave();
+    crit_sect_.Leave();
     if (!recovered_packet_callback_->OnRecoveredPacket(packet->data,
                                                        packet->length)) {
       return -1;
     }
-    crit_sect_->Enter();
+    crit_sect_.Enter();
     (*it)->returned = true;
   }
-  crit_sect_->Leave();
+  crit_sect_.Leave();
   return 0;
 }
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index 6a63813f408..0ebca9bce2d 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -13,7 +13,7 @@
 
 // This header is included to get the nested declaration of Packet structure.
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
@@ -21,8 +21,6 @@
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-
 class FecReceiverImpl : public FecReceiver {
  public:
   explicit FecReceiverImpl(RtpData* callback);
@@ -38,7 +36,7 @@ class FecReceiverImpl : public FecReceiver {
   FecPacketCounter GetPacketCounter() const override;
 
  private:
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  rtc::CriticalSection crit_sect_;
   RtpData* recovered_packet_callback_;
   ForwardErrorCorrection* fec_;
   // TODO(holmer): In the current version received_packet_list_ is never more
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
index ee8f408720e..cd60d9b094c 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -11,10 +11,10 @@
 #include <string.h>
 
 #include <list>
+#include <memory>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
@@ -92,9 +92,9 @@ class ReceiverFecTest : public ::testing::Test {
                                       uint8_t ulpfec_payload_type);
 
   MockRtpData rtp_data_callback_;
-  rtc::scoped_ptr<ForwardErrorCorrection> fec_;
-  rtc::scoped_ptr<FecReceiver> receiver_fec_;
-  rtc::scoped_ptr<FrameGenerator> generator_;
+  std::unique_ptr<ForwardErrorCorrection> fec_;
+  std::unique_ptr<FecReceiver> receiver_fec_;
+  std::unique_ptr<FrameGenerator> generator_;
 };
 
 void DeletePackets(std::list<Packet*>* packets) {
@@ -415,12 +415,12 @@ void ReceiverFecTest::SurvivesMaliciousPacket(const uint8_t* data,
                                               size_t length,
                                               uint8_t ulpfec_payload_type) {
   webrtc::RTPHeader header;
-  rtc::scoped_ptr<webrtc::RtpHeaderParser> parser(
+  std::unique_ptr<webrtc::RtpHeaderParser> parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(parser->Parse(data, length, &header));
 
   webrtc::NullRtpData null_callback;
-  rtc::scoped_ptr<webrtc::FecReceiver> receiver_fec(
+  std::unique_ptr<webrtc::FecReceiver> receiver_fec(
       webrtc::FecReceiver::Create(&null_callback));
 
   receiver_fec->AddReceivedRedPacket(header, data, length, ulpfec_payload_type);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
index b85d813790d..623c658a174 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -15,6 +15,7 @@
 
 #include <algorithm>
 #include <iterator>
+#include <memory>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
@@ -163,7 +164,7 @@ int32_t ForwardErrorCorrection::GenerateFEC(const PacketList& media_packet_list,
 
   // -- Generate packet masks --
   // Always allocate space for a large mask.
-  rtc::scoped_ptr<uint8_t[]> packet_mask(
+  std::unique_ptr<uint8_t[]> packet_mask(
       new uint8_t[num_fec_packets * kMaskSizeLBitSet]);
   memset(packet_mask.get(), 0, num_fec_packets * num_mask_bytes);
   internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc
index 6d8b4074598..e23a3fa629b 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc
@@ -9,13 +9,13 @@
  */
 #include "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h"
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/bitbuffer.h"
 #include "webrtc/base/bytebuffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 namespace {
@@ -103,7 +103,7 @@ bool H264BitstreamParser::ParseSpsNalu(const uint8_t* sps, size_t length) {
   sps_parsed_ = false;
   // Parse out the SPS RBSP. It should be small, so it's ok that we create a
   // copy. We'll eventually write this back.
-  rtc::scoped_ptr<rtc::ByteBufferWriter> sps_rbsp(
+  std::unique_ptr<rtc::ByteBufferWriter> sps_rbsp(
       ParseRbsp(sps + kNaluHeaderAndTypeSize, length - kNaluHeaderAndTypeSize));
   rtc::BitBuffer sps_parser(reinterpret_cast<const uint8_t*>(sps_rbsp->Data()),
                             sps_rbsp->Length());
@@ -209,7 +209,7 @@ bool H264BitstreamParser::ParsePpsNalu(const uint8_t* pps, size_t length) {
   // We're starting a new stream, so reset picture type rewriting values.
   pps_ = PpsState();
   pps_parsed_ = false;
-  rtc::scoped_ptr<rtc::ByteBufferWriter> buffer(
+  std::unique_ptr<rtc::ByteBufferWriter> buffer(
       ParseRbsp(pps + kNaluHeaderAndTypeSize, length - kNaluHeaderAndTypeSize));
   rtc::BitBuffer parser(reinterpret_cast<const uint8_t*>(buffer->Data()),
                         buffer->Length());
@@ -317,7 +317,7 @@ bool H264BitstreamParser::ParseNonParameterSetNalu(const uint8_t* source,
   RTC_CHECK(sps_parsed_);
   RTC_CHECK(pps_parsed_);
   last_slice_qp_delta_parsed_ = false;
-  rtc::scoped_ptr<rtc::ByteBufferWriter> slice_rbsp(ParseRbsp(
+  std::unique_ptr<rtc::ByteBufferWriter> slice_rbsp(ParseRbsp(
       source + kNaluHeaderAndTypeSize, source_length - kNaluHeaderAndTypeSize));
   rtc::BitBuffer slice_reader(
       reinterpret_cast<const uint8_t*>(slice_rbsp->Data()),
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index e19c31bfece..9a7b9d3f7ec 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -11,10 +11,10 @@
 #include <algorithm>
 #include <iterator>
 #include <list>
+#include <memory>
 #include <set>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -105,7 +105,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
     size_t packet_length = len;
     uint8_t restored_packet[1500];
     RTPHeader header;
-    rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+    std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     if (!parser->Parse(ptr, len, &header)) {
       return false;
     }
@@ -191,7 +191,6 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
 
     rtp_rtcp_module_->SetSSRC(kTestSsrc);
     rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
-    rtp_receiver_->SetNACKStatus(kNackRtcp);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
     rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
@@ -279,11 +278,11 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
 
   void TearDown() override { delete rtp_rtcp_module_; }
 
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RTPPayloadRegistry rtp_payload_registry_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
+  std::unique_ptr<RtpReceiver> rtp_receiver_;
   RtpRtcp* rtp_rtcp_module_;
-  rtc::scoped_ptr<TestRtpFeedback> rtp_feedback_;
+  std::unique_ptr<TestRtpFeedback> rtp_feedback_;
   RtxLoopBackTransport transport_;
   VerifyingRtxReceiver receiver_;
   uint8_t payload_data[65000];
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec.cc
index 69a28ed4dbc..c7ea19db586 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec.cc
@@ -157,12 +157,11 @@ int ProducerFec::AddRtpPacketAndGenerateFec(const uint8_t* data_buffer,
           (ExcessOverheadBelowMax() && MinimumMediaPacketsReached()))) {
     assert(num_first_partition_ <=
            static_cast<int>(ForwardErrorCorrection::kMaxMediaPackets));
-    int ret = fec_->GenerateFEC(media_packets_fec_,
-                                params_.fec_rate,
-                                num_first_partition_,
-                                params_.use_uep_protection,
-                                params_.fec_mask_type,
-                                &fec_packets_);
+    // TODO(pbos): Consider whether unequal protection should be enabled or not,
+    // it is currently always disabled.
+    int ret = fec_->GenerateFEC(media_packets_fec_, params_.fec_rate,
+                                num_first_partition_, false,
+                                params_.fec_mask_type, &fec_packets_);
     if (fec_packets_.empty()) {
       num_frames_ = 0;
       DeletePackets();
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
index fad0f502f59..ec5228afd5b 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include <list>
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
@@ -78,7 +79,7 @@ TEST_F(ProducerFecTest, NoEmptyFecWithSeqNumGaps) {
   protected_packets.push_back({12, 3, 54, 0});
   protected_packets.push_back({21, 0, 55, 0});
   protected_packets.push_back({13, 3, 57, 1});
-  FecProtectionParams params = {117, 0, 3, kFecMaskBursty};
+  FecProtectionParams params = {117, 3, kFecMaskBursty};
   producer_->SetFecParameters(&params, 0);
   uint8_t packet[28] = {0};
   for (Packet p : protected_packets) {
@@ -111,7 +112,7 @@ TEST_F(ProducerFecTest, OneFrameFec) {
   // of packets is within |kMaxExcessOverhead|, and (2) the total number of
   // media packets for 1 frame is at least |minimum_media_packets_fec_|.
   const int kNumPackets = 4;
-  FecProtectionParams params = {15, false, 3};
+  FecProtectionParams params = {15, 3, kFecMaskRandom};
   std::list<test::RawRtpPacket*> rtp_packets;
   generator_->NewFrame(kNumPackets);
   producer_->SetFecParameters(&params, 0);  // Expecting one FEC packet.
@@ -152,7 +153,7 @@ TEST_F(ProducerFecTest, TwoFrameFec) {
   const int kNumPackets = 2;
   const int kNumFrames = 2;
 
-  FecProtectionParams params = {15, 0, 3};
+  FecProtectionParams params = {15, 3, kFecMaskRandom};
   std::list<test::RawRtpPacket*> rtp_packets;
   producer_->SetFecParameters(&params, 0);  // Expecting one FEC packet.
   uint32_t last_timestamp = 0;
@@ -188,7 +189,7 @@ TEST_F(ProducerFecTest, TwoFrameFec) {
 TEST_F(ProducerFecTest, BuildRedPacket) {
   generator_->NewFrame(1);
   test::RawRtpPacket* packet = generator_->NextPacket(0, 10);
-  rtc::scoped_ptr<RedPacket> red_packet(producer_->BuildRedPacket(
+  std::unique_ptr<RedPacket> red_packet(producer_->BuildRedPacket(
       packet->data, packet->length - kRtpHeaderSize, kRtpHeaderSize,
       kRedPayloadType));
   EXPECT_EQ(packet->length + 1, red_packet->length());
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 022fc9610f4..932be1bb9e1 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -12,10 +12,10 @@
 
 #include <math.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include <cstdlib>
+
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -29,7 +29,6 @@ StreamStatisticianImpl::StreamStatisticianImpl(
     RtcpStatisticsCallback* rtcp_callback,
     StreamDataCountersCallback* rtp_callback)
     : clock_(clock),
-      stream_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       incoming_bitrate_(clock, NULL),
       ssrc_(0),
       max_reordering_threshold_(kDefaultMaxReorderingThreshold),
@@ -59,7 +58,7 @@ void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
 void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
                                             size_t packet_length,
                                             bool retransmitted) {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   bool in_order = InOrderPacketInternal(header.sequenceNumber);
   ssrc_ = header.ssrc;
   incoming_bitrate_.Update(packet_length);
@@ -116,7 +115,7 @@ void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
   int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
       (header.timestamp - last_received_timestamp_);
 
-  time_diff_samples = abs(time_diff_samples);
+  time_diff_samples = std::abs(time_diff_samples);
 
   // lib_jingle sometimes deliver crazy jumps in TS for the same stream.
   // If this happens, don't update jitter value. Use 5 secs video frequency
@@ -136,7 +135,7 @@ void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
      (last_received_timestamp_ +
       last_received_transmission_time_offset_));
 
-  time_diff_samples_ext = abs(time_diff_samples_ext);
+  time_diff_samples_ext = std::abs(time_diff_samples_ext);
 
   if (time_diff_samples_ext < 450000) {
     int32_t jitter_diffQ4TransmissionTimeOffset =
@@ -150,7 +149,7 @@ void StreamStatisticianImpl::NotifyRtpCallback() {
   StreamDataCounters data;
   uint32_t ssrc;
   {
-    CriticalSectionScoped cs(stream_lock_.get());
+    rtc::CritScope cs(&stream_lock_);
     data = receive_counters_;
     ssrc = ssrc_;
   }
@@ -161,7 +160,7 @@ void StreamStatisticianImpl::NotifyRtcpCallback() {
   RtcpStatistics data;
   uint32_t ssrc;
   {
-    CriticalSectionScoped cs(stream_lock_.get());
+    rtc::CritScope cs(&stream_lock_);
     data = last_reported_statistics_;
     ssrc = ssrc_;
   }
@@ -171,7 +170,7 @@ void StreamStatisticianImpl::NotifyRtcpCallback() {
 void StreamStatisticianImpl::FecPacketReceived(const RTPHeader& header,
                                                size_t packet_length) {
   {
-    CriticalSectionScoped cs(stream_lock_.get());
+    rtc::CritScope cs(&stream_lock_);
     receive_counters_.fec.AddPacket(packet_length, header);
   }
   NotifyRtpCallback();
@@ -179,14 +178,14 @@ void StreamStatisticianImpl::FecPacketReceived(const RTPHeader& header,
 
 void StreamStatisticianImpl::SetMaxReorderingThreshold(
     int max_reordering_threshold) {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   max_reordering_threshold_ = max_reordering_threshold;
 }
 
 bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
                                            bool reset) {
   {
-    CriticalSectionScoped cs(stream_lock_.get());
+    rtc::CritScope cs(&stream_lock_);
     if (received_seq_first_ == 0 &&
         receive_counters_.transmitted.payload_bytes == 0) {
       // We have not received anything.
@@ -282,7 +281,7 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
 
 void StreamStatisticianImpl::GetDataCounters(
     size_t* bytes_received, uint32_t* packets_received) const {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   if (bytes_received) {
     *bytes_received = receive_counters_.transmitted.payload_bytes +
                       receive_counters_.transmitted.header_bytes +
@@ -295,30 +294,30 @@ void StreamStatisticianImpl::GetDataCounters(
 
 void StreamStatisticianImpl::GetReceiveStreamDataCounters(
     StreamDataCounters* data_counters) const {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   *data_counters = receive_counters_;
 }
 
 uint32_t StreamStatisticianImpl::BitrateReceived() const {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   return incoming_bitrate_.BitrateNow();
 }
 
 void StreamStatisticianImpl::ProcessBitrate() {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   incoming_bitrate_.Process();
 }
 
 void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
                                                 uint32_t* frac) const {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   *secs = last_receive_time_ntp_.seconds();
   *frac = last_receive_time_ntp_.fractions();
 }
 
 bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
     const RTPHeader& header, int64_t min_rtt) const {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   if (InOrderPacketInternal(header.sequenceNumber)) {
     return false;
   }
@@ -352,7 +351,7 @@ bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
 }
 
 bool StreamStatisticianImpl::IsPacketInOrder(uint16_t sequence_number) const {
-  CriticalSectionScoped cs(stream_lock_.get());
+  rtc::CritScope cs(&stream_lock_);
   return InOrderPacketInternal(sequence_number);
 }
 
@@ -377,7 +376,6 @@ ReceiveStatistics* ReceiveStatistics::Create(Clock* clock) {
 
 ReceiveStatisticsImpl::ReceiveStatisticsImpl(Clock* clock)
     : clock_(clock),
-      receive_statistics_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       last_rate_update_ms_(0),
       rtcp_stats_callback_(NULL),
       rtp_stats_callback_(NULL) {}
@@ -394,7 +392,7 @@ void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header,
                                            bool retransmitted) {
   StreamStatisticianImpl* impl;
   {
-    CriticalSectionScoped cs(receive_statistics_lock_.get());
+    rtc::CritScope cs(&receive_statistics_lock_);
     StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
     if (it != statisticians_.end()) {
       impl = it->second;
@@ -412,7 +410,7 @@ void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header,
 
 void ReceiveStatisticsImpl::FecPacketReceived(const RTPHeader& header,
                                               size_t packet_length) {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
   // Ignore FEC if it is the first packet.
   if (it != statisticians_.end()) {
@@ -421,7 +419,7 @@ void ReceiveStatisticsImpl::FecPacketReceived(const RTPHeader& header,
 }
 
 StatisticianMap ReceiveStatisticsImpl::GetActiveStatisticians() const {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   StatisticianMap active_statisticians;
   for (StatisticianImplMap::const_iterator it = statisticians_.begin();
        it != statisticians_.end(); ++it) {
@@ -438,7 +436,7 @@ StatisticianMap ReceiveStatisticsImpl::GetActiveStatisticians() const {
 
 StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
     uint32_t ssrc) const {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   StatisticianImplMap::const_iterator it = statisticians_.find(ssrc);
   if (it == statisticians_.end())
     return NULL;
@@ -447,7 +445,7 @@ StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
 
 void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
     int max_reordering_threshold) {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   for (StatisticianImplMap::iterator it = statisticians_.begin();
        it != statisticians_.end(); ++it) {
     it->second->SetMaxReorderingThreshold(max_reordering_threshold);
@@ -455,7 +453,7 @@ void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
 }
 
 void ReceiveStatisticsImpl::Process() {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   for (StatisticianImplMap::iterator it = statisticians_.begin();
        it != statisticians_.end(); ++it) {
     it->second->ProcessBitrate();
@@ -464,7 +462,7 @@ void ReceiveStatisticsImpl::Process() {
 }
 
 int64_t ReceiveStatisticsImpl::TimeUntilNextProcess() {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   int64_t time_since_last_update = clock_->TimeInMilliseconds() -
       last_rate_update_ms_;
   return std::max<int64_t>(
@@ -473,7 +471,7 @@ int64_t ReceiveStatisticsImpl::TimeUntilNextProcess() {
 
 void ReceiveStatisticsImpl::RegisterRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   if (callback != NULL)
     assert(rtcp_stats_callback_ == NULL);
   rtcp_stats_callback_ = callback;
@@ -481,20 +479,20 @@ void ReceiveStatisticsImpl::RegisterRtcpStatisticsCallback(
 
 void ReceiveStatisticsImpl::StatisticsUpdated(const RtcpStatistics& statistics,
                                               uint32_t ssrc) {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   if (rtcp_stats_callback_)
     rtcp_stats_callback_->StatisticsUpdated(statistics, ssrc);
 }
 
 void ReceiveStatisticsImpl::CNameChanged(const char* cname, uint32_t ssrc) {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   if (rtcp_stats_callback_)
     rtcp_stats_callback_->CNameChanged(cname, ssrc);
 }
 
 void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   if (callback != NULL)
     assert(rtp_stats_callback_ == NULL);
   rtp_stats_callback_ = callback;
@@ -502,7 +500,7 @@ void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback(
 
 void ReceiveStatisticsImpl::DataCountersUpdated(const StreamDataCounters& stats,
                                                 uint32_t ssrc) {
-  CriticalSectionScoped cs(receive_statistics_lock_.get());
+  rtc::CritScope cs(&receive_statistics_lock_);
   if (rtp_stats_callback_) {
     rtp_stats_callback_->DataCountersUpdated(stats, ssrc);
   }
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index 6da8334da6e..39679673d08 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -16,15 +16,12 @@
 #include <algorithm>
 #include <map>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/ntp_time.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-
 class StreamStatisticianImpl : public StreamStatistician {
  public:
   StreamStatisticianImpl(Clock* clock,
@@ -57,11 +54,11 @@ class StreamStatisticianImpl : public StreamStatistician {
   void UpdateCounters(const RTPHeader& rtp_header,
                       size_t packet_length,
                       bool retransmitted);
-  void NotifyRtpCallback() LOCKS_EXCLUDED(stream_lock_.get());
-  void NotifyRtcpCallback() LOCKS_EXCLUDED(stream_lock_.get());
+  void NotifyRtpCallback() LOCKS_EXCLUDED(stream_lock_);
+  void NotifyRtcpCallback() LOCKS_EXCLUDED(stream_lock_);
 
   Clock* clock_;
-  rtc::scoped_ptr<CriticalSectionWrapper> stream_lock_;
+  rtc::CriticalSection stream_lock_;
   Bitrate incoming_bitrate_;
   uint32_t ssrc_;
   int max_reordering_threshold_;  // In number of packets or sequence numbers.
@@ -131,7 +128,7 @@ class ReceiveStatisticsImpl : public ReceiveStatistics,
   typedef std::map<uint32_t, StreamStatisticianImpl*> StatisticianImplMap;
 
   Clock* clock_;
-  rtc::scoped_ptr<CriticalSectionWrapper> receive_statistics_lock_;
+  rtc::CriticalSection receive_statistics_lock_;
   int64_t last_rate_update_ms_;
   StatisticianImplMap statisticians_;
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index c265c17c04b..f6cbe74e478 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
@@ -36,7 +37,7 @@ class ReceiveStatisticsTest : public ::testing::Test {
 
  protected:
   SimulatedClock clock_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RTPHeader header1_;
   RTPHeader header2_;
 };
@@ -156,8 +157,8 @@ TEST_F(ReceiveStatisticsTest, RtcpCallbacks) {
         : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {}
     virtual ~TestCallback() {}
 
-    virtual void StatisticsUpdated(const RtcpStatistics& statistics,
-                                   uint32_t ssrc) {
+    void StatisticsUpdated(const RtcpStatistics& statistics,
+                           uint32_t ssrc) override {
       ssrc_ = ssrc;
       stats_ = statistics;
       ++num_calls_;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 183076ff591..bbfb52c6cc9 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_types.h"
@@ -74,13 +76,13 @@ class RtcpFormatRembTest : public ::testing::Test {
   OverUseDetectorOptions over_use_detector_options_;
   Clock* system_clock_;
   ModuleRtpRtcpImpl* dummy_rtp_rtcp_impl_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RTCPSender* rtcp_sender_;
   RTCPReceiver* rtcp_receiver_;
   TestTransport* test_transport_;
   test::NullTransport null_transport_;
   MockRemoteBitrateObserver remote_bitrate_observer_;
-  rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+  std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
 };
 
 void RtcpFormatRembTest::SetUp() {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
index 16bd3fc2a2d..f5a885cfdd7 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
 
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
index af3fbacc809..ad28cb39e7b 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
index 4fb92facc73..06b67c375cf 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
index 49de7be1a80..34eb57f3484 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
index a85576db2da..d7e715bc9ec 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
index fb2be113a2b..b6acae5aab1 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
index 10fafd229da..64caf1b5c41 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
@@ -11,6 +11,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
index 866eb2ce529..8568e7327c7 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
index 237d923cd7a..0630adbac24 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 
@@ -49,7 +50,7 @@ class ReceiverReport : public RtcpPacket {
   static const size_t kRrBaseLength = 4;
   static const size_t kMaxNumberOfReportBlocks = 0x1F;
 
-  size_t BlockLength() const {
+  size_t BlockLength() const override {
     return kHeaderLength + kRrBaseLength +
            report_blocks_.size() * ReportBlock::kLength;
   }
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
index 3b33982a838..2f59fbbd55f 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
@@ -13,11 +13,11 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-
-using webrtc::RTCPUtility::RtcpCommonHeader;
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 
 namespace webrtc {
 namespace rtcp {
+constexpr uint8_t Remb::kFeedbackMessageType;
 // Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
 //
 //     0                   1                   2                   3
@@ -36,32 +36,39 @@ namespace rtcp {
 // 16 |   SSRC feedback                                               |
 //    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //    :  ...                                                          :
-bool Remb::Parse(const RtcpCommonHeader& header, const uint8_t* payload) {
-  RTC_DCHECK(header.packet_type == kPacketType);
-  RTC_DCHECK(header.count_or_format == kFeedbackMessageType);
+bool Remb::Parse(const CommonHeader& packet) {
+  RTC_DCHECK(packet.type() == kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
 
-  if (header.payload_size_bytes < 16) {
-    LOG(LS_WARNING) << "Payload length " << header.payload_size_bytes
+  if (packet.payload_size_bytes() < 16) {
+    LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
                     << " is too small for Remb packet.";
     return false;
   }
+  const uint8_t* const payload = packet.payload();
   if (kUniqueIdentifier != ByteReader<uint32_t>::ReadBigEndian(&payload[8])) {
     LOG(LS_WARNING) << "REMB identifier not found, not a REMB packet.";
     return false;
   }
   uint8_t number_of_ssrcs = payload[12];
-  if (header.payload_size_bytes !=
+  if (packet.payload_size_bytes() !=
       kCommonFeedbackLength + (2 + number_of_ssrcs) * 4) {
-    LOG(LS_WARNING) << "Payload size " << header.payload_size_bytes
+    LOG(LS_WARNING) << "Payload size " << packet.payload_size_bytes()
                     << " does not match " << number_of_ssrcs << " ssrcs.";
     return false;
   }
 
   ParseCommonFeedback(payload);
   uint8_t exponenta = payload[13] >> 2;
-  uint32_t mantissa = (static_cast<uint32_t>(payload[13] & 0x03) << 16) |
+  uint64_t mantissa = (static_cast<uint32_t>(payload[13] & 0x03) << 16) |
                       ByteReader<uint16_t>::ReadBigEndian(&payload[14]);
   bitrate_bps_ = (mantissa << exponenta);
+  bool shift_overflow = (bitrate_bps_ >> exponenta) != mantissa;
+  if (shift_overflow) {
+    LOG(LS_ERROR) << "Invalid remb bitrate value : " << mantissa
+                  << "*2^" << static_cast<int>(exponenta);
+    return false;
+  }
 
   const uint8_t* next_ssrc = payload + 16;
   ssrcs_.clear();
@@ -111,7 +118,7 @@ bool Remb::Create(uint8_t* packet,
   ByteWriter<uint32_t>::WriteBigEndian(packet + *index, kUniqueIdentifier);
   *index += sizeof(uint32_t);
   const uint32_t kMaxMantissa = 0x3ffff;  // 18 bits.
-  uint32_t mantissa = bitrate_bps_;
+  uint64_t mantissa = bitrate_bps_;
   uint8_t exponenta = 0;
   while (mantissa > kMaxMantissa) {
     mantissa >>= 1;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
index d58f052b145..9f10921c994 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
@@ -14,28 +14,29 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
 namespace webrtc {
 namespace rtcp {
+class CommonHeader;
+
 // Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
 class Remb : public Psfb {
  public:
-  static const uint8_t kFeedbackMessageType = 15;
+  static constexpr uint8_t kFeedbackMessageType = 15;
 
   Remb() : bitrate_bps_(0) {}
   ~Remb() override {}
 
   // Parse assumes header is already parsed and validated.
-  bool Parse(const RTCPUtility::RtcpCommonHeader& header,
-             const uint8_t* payload);  // Size of the payload is in the header.
+  bool Parse(const CommonHeader& packet);
 
   bool AppliesTo(uint32_t ssrc);
   bool AppliesToMany(const std::vector<uint32_t>& ssrcs);
-  void WithBitrateBps(uint32_t bitrate_bps) { bitrate_bps_ = bitrate_bps; }
+  void WithBitrateBps(uint64_t bitrate_bps) { bitrate_bps_ = bitrate_bps; }
 
-  uint32_t bitrate_bps() const { return bitrate_bps_; }
+  uint64_t bitrate_bps() const { return bitrate_bps_; }
   const std::vector<uint32_t>& ssrcs() const { return ssrcs_; }
 
  protected:
@@ -49,14 +50,14 @@ class Remb : public Psfb {
   }
 
  private:
-  static const size_t kMaxNumberOfSsrcs = 0xff;
-  static const uint32_t kUniqueIdentifier = 0x52454D42;  // 'R' 'E' 'M' 'B'.
+  static constexpr size_t kMaxNumberOfSsrcs = 0xff;
+  static constexpr uint32_t kUniqueIdentifier = 0x52454D42;  // 'R' 'E' 'M' 'B'.
 
   // Media ssrc is unused, shadow base class setter and getter.
   void To(uint32_t);
   uint32_t media_ssrc() const;
 
-  uint32_t bitrate_bps_;
+  uint64_t bitrate_bps_;
   std::vector<uint32_t> ssrcs_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Remb);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb_unittest.cc
index ee06972e2b0..d504143f6f6 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb_unittest.cc
@@ -12,32 +12,25 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/test/rtcp_packet_parser.h"
 
 using testing::ElementsAreArray;
 using testing::IsEmpty;
 using testing::make_tuple;
 using webrtc::rtcp::Remb;
-using webrtc::RTCPUtility::RtcpCommonHeader;
-using webrtc::RTCPUtility::RtcpParseCommonHeader;
 
 namespace webrtc {
 namespace {
-
 const uint32_t kSenderSsrc = 0x12345678;
 const uint32_t kRemoteSsrcs[] = {0x23456789, 0x2345678a, 0x2345678b};
 const uint32_t kBitrateBps = 0x3fb93 * 2;  // 522022;
+const uint64_t kBitrateBps64bit = 0x3fb93ULL << 30;
 const uint8_t kPacket[] = {0x8f, 206,  0x00, 0x07, 0x12, 0x34, 0x56, 0x78,
                            0x00, 0x00, 0x00, 0x00, 'R',  'E',  'M',  'B',
                            0x03, 0x07, 0xfb, 0x93, 0x23, 0x45, 0x67, 0x89,
                            0x23, 0x45, 0x67, 0x8a, 0x23, 0x45, 0x67, 0x8b};
 const size_t kPacketLength = sizeof(kPacket);
-
-bool ParseRemb(const uint8_t* buffer, size_t length, Remb* remb) {
-  RtcpCommonHeader header;
-  EXPECT_TRUE(RtcpParseCommonHeader(buffer, length, &header));
-  EXPECT_EQ(length, header.BlockSize());
-  return remb->Parse(header, buffer + RtcpCommonHeader::kHeaderSizeBytes);
-}
+}  // namespace
 
 TEST(RtcpPacketRembTest, Create) {
   Remb remb;
@@ -55,7 +48,7 @@ TEST(RtcpPacketRembTest, Create) {
 
 TEST(RtcpPacketRembTest, Parse) {
   Remb remb;
-  EXPECT_TRUE(ParseRemb(kPacket, kPacketLength, &remb));
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &remb));
   const Remb& parsed = remb;
 
   EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
@@ -70,19 +63,31 @@ TEST(RtcpPacketRembTest, CreateAndParseWithoutSsrcs) {
   rtc::Buffer packet = remb.Build();
 
   Remb parsed;
-  EXPECT_TRUE(ParseRemb(packet.data(), packet.size(), &parsed));
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
   EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
   EXPECT_EQ(kBitrateBps, parsed.bitrate_bps());
   EXPECT_THAT(parsed.ssrcs(), IsEmpty());
 }
 
+TEST(RtcpPacketRembTest, CreateAndParse64bitBitrate) {
+  Remb remb;
+  remb.WithBitrateBps(kBitrateBps64bit);
+  rtc::Buffer packet = remb.Build();
+
+  Remb parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+  EXPECT_EQ(kBitrateBps64bit, parsed.bitrate_bps());
+}
+
 TEST(RtcpPacketRembTest, ParseFailsOnTooSmallPacketToBeRemb) {
-  uint8_t packet[kPacketLength];
-  memcpy(packet, kPacket, kPacketLength);
-  packet[3] = 3;  // Make it too small.
+  // Make it too small.
+  constexpr size_t kTooSmallSize = (1 + 3) * 4;
+  uint8_t packet[kTooSmallSize];
+  memcpy(packet, kPacket, kTooSmallSize);
+  packet[3] = 3;
 
   Remb remb;
-  EXPECT_FALSE(ParseRemb(packet, (1 + 3) * 4, &remb));
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
 }
 
 TEST(RtcpPacketRembTest, ParseFailsWhenUniqueIdentifierIsNotRemb) {
@@ -91,7 +96,17 @@ TEST(RtcpPacketRembTest, ParseFailsWhenUniqueIdentifierIsNotRemb) {
   packet[12] = 'N';  // Swap 'R' -> 'N' in the 'REMB' unique identifier.
 
   Remb remb;
-  EXPECT_FALSE(ParseRemb(packet, kPacketLength, &remb));
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
+}
+
+TEST(RtcpPacketRembTest, ParseFailsWhenBitrateDoNotFitIn64bits) {
+  uint8_t packet[kPacketLength];
+  memcpy(packet, kPacket, kPacketLength);
+  packet[17] |= 0xfc;  // Set exponenta component to maximum of 63.
+  packet[19] |= 0x02;  // Ensure mantissa is at least 2.
+
+  Remb remb;
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
 }
 
 TEST(RtcpPacketRembTest, ParseFailsWhenSsrcCountMismatchLength) {
@@ -100,7 +115,7 @@ TEST(RtcpPacketRembTest, ParseFailsWhenSsrcCountMismatchLength) {
   packet[16]++;  // Swap 3 -> 4 in the ssrcs count.
 
   Remb remb;
-  EXPECT_FALSE(ParseRemb(packet, kPacketLength, &remb));
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
 }
 
 TEST(RtcpPacketRembTest, TooManySsrcs) {
@@ -126,5 +141,5 @@ TEST(RtcpPacketRembTest, TooManySsrcsForBatchAssign) {
   // But not for another one.
   EXPECT_FALSE(remb.AppliesTo(kRemoteSsrc));
 }
-}  // namespace
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rpsi.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rpsi.h
index 1fa3352335e..7d4895b09cd 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rpsi.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/rpsi.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RPSI_H_
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
index 5940edbb116..19d5b42b557 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
index e11bdb9a942..a544017f149 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "webrtc/system_wrappers/include/ntp_time.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h
index 5d9e6c93e95..7b6b24f3b8d 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
index 26a44082d35..c84d0dfe0b9 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
index 4028563d071..15bfc5856fa 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
index 4ad49561b80..5cdaa3aaa47 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
@@ -651,10 +651,10 @@ bool TransportFeedback::Create(uint8_t* packet,
 //    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
 // De-serialize packet.
-rtc::scoped_ptr<TransportFeedback> TransportFeedback::ParseFrom(
+std::unique_ptr<TransportFeedback> TransportFeedback::ParseFrom(
     const uint8_t* buffer,
     size_t length) {
-  rtc::scoped_ptr<TransportFeedback> packet(new TransportFeedback());
+  std::unique_ptr<TransportFeedback> packet(new TransportFeedback());
 
   if (length < kMinSizeBytes) {
     LOG(LS_WARNING) << "Buffer too small (" << length
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
index ad6fd166f2b..7a74d7ffe83 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_
 
 #include <deque>
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
@@ -58,7 +59,7 @@ class TransportFeedback : public RtcpPacket {
   static const uint8_t kFeedbackMessageType = 15;  // TODO(sprang): IANA reg?
   static const uint8_t kPayloadType = 205;         // RTPFB, see RFC4585.
 
-  static rtc::scoped_ptr<TransportFeedback> ParseFrom(const uint8_t* buffer,
+  static std::unique_ptr<TransportFeedback> ParseFrom(const uint8_t* buffer,
                                                       size_t length);
 
  protected:
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
index 3615065351d..203d70fab15 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 #include <limits>
+#include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -43,7 +44,7 @@ class FeedbackTester {
   void WithInput(const uint16_t received_seq[],
                  const int64_t received_ts[],
                  uint16_t length) {
-    rtc::scoped_ptr<int64_t[]> temp_deltas;
+    std::unique_ptr<int64_t[]> temp_deltas;
     if (received_ts == nullptr) {
       temp_deltas.reset(new int64_t[length]);
       GenerateDeltas(received_seq, length, temp_deltas.get());
@@ -136,7 +137,7 @@ class FeedbackTester {
   std::vector<int64_t> expected_deltas_;
   size_t expected_size_;
   int64_t default_delta_;
-  rtc::scoped_ptr<TransportFeedback> feedback_;
+  std::unique_ptr<TransportFeedback> feedback_;
   rtc::Buffer serialized_;
 };
 
@@ -356,7 +357,7 @@ TEST(RtcpPacketTest, TransportFeedback_Aliasing) {
 
 TEST(RtcpPacketTest, TransportFeedback_Limits) {
   // Sequence number wrap above 0x8000.
-  rtc::scoped_ptr<TransportFeedback> packet(new TransportFeedback());
+  std::unique_ptr<TransportFeedback> packet(new TransportFeedback());
   packet->WithBase(0, 0);
   EXPECT_TRUE(packet->WithReceivedPacket(0x8000, 1000));
 
@@ -446,7 +447,7 @@ TEST(RtcpPacketTest, TransportFeedback_Padding) {
       &mod_buffer[2], ByteReader<uint16_t>::ReadBigEndian(&mod_buffer[2]) +
                           ((kPaddingBytes + 3) / 4));
 
-  rtc::scoped_ptr<TransportFeedback> parsed_packet(
+  std::unique_ptr<TransportFeedback> parsed_packet(
       TransportFeedback::ParseFrom(mod_buffer, kExpectedSizeWithPadding));
   ASSERT_TRUE(parsed_packet.get() != nullptr);
   EXPECT_EQ(kExpectedSizeWords * 4, packet.size());  // Padding not included.
@@ -468,7 +469,7 @@ TEST(RtcpPacketTest, TransportFeedback_CorrectlySplitsVectorChunks) {
     feedback.WithReceivedPacket(deltas, deltas * 1000 + kLargeTimeDelta);
 
     rtc::Buffer serialized_packet = feedback.Build();
-    rtc::scoped_ptr<TransportFeedback> deserialized_packet =
+    std::unique_ptr<TransportFeedback> deserialized_packet =
         TransportFeedback::ParseFrom(serialized_packet.data(),
                                      serialized_packet.size());
     EXPECT_TRUE(deserialized_packet.get() != nullptr);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
index 1b5d4f32a62..58218ddbfd1 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
@@ -6,8 +6,6 @@
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
- *
- * This file includes unit tests for the RtcpPacket.
  */
 
 #include "testing/gmock/include/gmock/gmock.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 0faf2a42575..3e8e47fbd37 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -49,13 +49,9 @@ RTCPReceiver::RTCPReceiver(
       receiver_only_(receiver_only),
       _lastReceived(0),
       _rtpRtcp(*owner),
-      _criticalSectionFeedbacks(
-          CriticalSectionWrapper::CreateCriticalSection()),
       _cbRtcpBandwidthObserver(rtcp_bandwidth_observer),
       _cbRtcpIntraFrameObserver(rtcp_intra_frame_observer),
       _cbTransportFeedbackObserver(transport_feedback_observer),
-      _criticalSectionRTCPReceiver(
-          CriticalSectionWrapper::CreateCriticalSection()),
       main_ssrc_(0),
       _remoteSSRC(0),
       _remoteSenderInfo(),
@@ -76,9 +72,6 @@ RTCPReceiver::RTCPReceiver(
 }
 
 RTCPReceiver::~RTCPReceiver() {
-  delete _criticalSectionRTCPReceiver;
-  delete _criticalSectionFeedbacks;
-
   ReportBlockMap::iterator it = _receivedReportBlockMap.begin();
   for (; it != _receivedReportBlockMap.end(); ++it) {
     ReportBlockInfoMap* info_map = &(it->second);
@@ -103,12 +96,12 @@ RTCPReceiver::~RTCPReceiver() {
 }
 
 int64_t RTCPReceiver::LastReceived() {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   return _lastReceived;
 }
 
 int64_t RTCPReceiver::LastReceivedReceiverReport() const {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   int64_t last_received_rr = -1;
   for (ReceivedInfoMap::const_iterator it = _receivedInfoMap.begin();
        it != _receivedInfoMap.end(); ++it) {
@@ -120,7 +113,7 @@ int64_t RTCPReceiver::LastReceivedReceiverReport() const {
 }
 
 void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   // new SSRC reset old reports
   memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
@@ -131,7 +124,7 @@ void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
 }
 
 uint32_t RTCPReceiver::RemoteSSRC() const {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   return _remoteSSRC;
 }
 
@@ -139,7 +132,7 @@ void RTCPReceiver::SetSsrcs(uint32_t main_ssrc,
                             const std::set<uint32_t>& registered_ssrcs) {
   uint32_t old_ssrc = 0;
   {
-    CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+    rtc::CritScope lock(&_criticalSectionRTCPReceiver);
     old_ssrc = main_ssrc_;
     main_ssrc_ = main_ssrc;
     registered_ssrcs_ = registered_ssrcs;
@@ -156,7 +149,7 @@ int32_t RTCPReceiver::RTT(uint32_t remoteSSRC,
                           int64_t* avgRTT,
                           int64_t* minRTT,
                           int64_t* maxRTT) const {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   RTCPReportBlockInformation* reportBlock =
       GetReportBlockInformation(remoteSSRC, main_ssrc_);
@@ -180,13 +173,13 @@ int32_t RTCPReceiver::RTT(uint32_t remoteSSRC,
 }
 
 void RTCPReceiver::SetRtcpXrRrtrStatus(bool enable) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   xr_rrtr_status_ = enable;
 }
 
 bool RTCPReceiver::GetAndResetXrRrRtt(int64_t* rtt_ms) {
   assert(rtt_ms);
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   if (xr_rr_rtt_ms_ == 0) {
     return false;
   }
@@ -202,7 +195,7 @@ bool RTCPReceiver::NTP(uint32_t* ReceivedNTPsecs,
                        uint32_t* RTCPArrivalTimeFrac,
                        uint32_t* rtcp_timestamp) const
 {
-    CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+    rtc::CritScope lock(&_criticalSectionRTCPReceiver);
     if(ReceivedNTPsecs)
     {
         *ReceivedNTPsecs = _remoteSenderInfo.NTPseconds; // NTP from incoming SendReport
@@ -228,7 +221,7 @@ bool RTCPReceiver::NTP(uint32_t* ReceivedNTPsecs,
 bool RTCPReceiver::LastReceivedXrReferenceTimeInfo(
     RtcpReceiveTimeInfo* info) const {
   assert(info);
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   if (_lastReceivedXRNTPsecs == 0 && _lastReceivedXRNTPfrac == 0) {
     return false;
   }
@@ -251,7 +244,7 @@ bool RTCPReceiver::LastReceivedXrReferenceTimeInfo(
 
 int32_t RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const {
   assert(senderInfo);
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   if (_lastReceivedSRNTPsecs == 0) {
     return -1;
   }
@@ -264,7 +257,7 @@ int32_t RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const {
 int32_t RTCPReceiver::StatisticsReceived(
     std::vector<RTCPReportBlock>* receiveBlocks) const {
   assert(receiveBlocks);
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   ReportBlockMap::const_iterator it = _receivedReportBlockMap.begin();
   for (; it != _receivedReportBlockMap.end(); ++it) {
     const ReportBlockInfoMap* info_map = &(it->second);
@@ -280,7 +273,7 @@ int32_t
 RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
                                  RTCPUtility::RTCPParserV2* rtcpParser)
 {
-    CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+    rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
     _lastReceived = _clock->TimeInMilliseconds();
 
@@ -590,7 +583,7 @@ RTCPReportBlockInformation* RTCPReceiver::GetReportBlockInformation(
 
 RTCPCnameInformation*
 RTCPReceiver::CreateCnameInformation(uint32_t remoteSSRC) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPCnameInformation*>::iterator it =
       _receivedCnameMap.find(remoteSSRC);
@@ -606,7 +599,7 @@ RTCPReceiver::CreateCnameInformation(uint32_t remoteSSRC) {
 
 RTCPCnameInformation*
 RTCPReceiver::GetCnameInformation(uint32_t remoteSSRC) const {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPCnameInformation*>::const_iterator it =
       _receivedCnameMap.find(remoteSSRC);
@@ -619,7 +612,7 @@ RTCPReceiver::GetCnameInformation(uint32_t remoteSSRC) const {
 
 RTCPReceiveInformation*
 RTCPReceiver::CreateReceiveInformation(uint32_t remoteSSRC) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPReceiveInformation*>::iterator it =
       _receivedInfoMap.find(remoteSSRC);
@@ -634,7 +627,7 @@ RTCPReceiver::CreateReceiveInformation(uint32_t remoteSSRC) {
 
 RTCPReceiveInformation*
 RTCPReceiver::GetReceiveInformation(uint32_t remoteSSRC) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPReceiveInformation*>::iterator it =
       _receivedInfoMap.find(remoteSSRC);
@@ -651,7 +644,7 @@ void RTCPReceiver::UpdateReceiveInformation(
 }
 
 bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   if (_lastReceivedRrMs == 0)
     return false;
 
@@ -665,7 +658,7 @@ bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
 }
 
 bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   if (_lastIncreasedSequenceNumberMs == 0)
     return false;
 
@@ -680,7 +673,7 @@ bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) {
 }
 
 bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   bool updateBoundingSet = false;
   int64_t timeNow = _clock->TimeInMilliseconds();
@@ -724,7 +717,7 @@ bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
 }
 
 int32_t RTCPReceiver::BoundingSet(bool* tmmbrOwner, TMMBRSet* boundingSetRec) {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPReceiveInformation*>::iterator receiveInfoIt =
       _receivedInfoMap.find(_remoteSSRC);
@@ -773,7 +766,7 @@ void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) {
   cnameInfo->name[RTCP_CNAME_SIZE - 1] = 0;
   strncpy(cnameInfo->name, rtcpPacket.CName.CName, RTCP_CNAME_SIZE - 1);
   {
-    CriticalSectionScoped lock(_criticalSectionFeedbacks);
+    rtc::CritScope lock(&_criticalSectionFeedbacks);
     if (stats_callback_ != NULL) {
       stats_callback_->CNameChanged(rtcpPacket.CName.CName,
                                     rtcpPacket.CName.SenderSSRC);
@@ -1283,12 +1276,12 @@ int32_t RTCPReceiver::UpdateTMMBR() {
 
 void RTCPReceiver::RegisterRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {
-  CriticalSectionScoped cs(_criticalSectionFeedbacks);
+  rtc::CritScope cs(&_criticalSectionFeedbacks);
   stats_callback_ = callback;
 }
 
 RtcpStatisticsCallback* RTCPReceiver::GetRtcpStatisticsCallback() {
-  CriticalSectionScoped cs(_criticalSectionFeedbacks);
+  rtc::CritScope cs(&_criticalSectionFeedbacks);
   return stats_callback_;
 }
 
@@ -1305,7 +1298,7 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
   std::set<uint32_t> registered_ssrcs;
   {
     // We don't want to hold this critsect when triggering the callbacks below.
-    CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+    rtc::CritScope lock(&_criticalSectionRTCPReceiver);
     local_ssrc = main_ssrc_;
     registered_ssrcs = registered_ssrcs_;
   }
@@ -1378,7 +1371,7 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
   }
 
   if (!receiver_only_) {
-    CriticalSectionScoped cs(_criticalSectionFeedbacks);
+    rtc::CritScope cs(&_criticalSectionFeedbacks);
     if (stats_callback_) {
       for (ReportBlockList::const_iterator it =
           rtcpPacketInformation.report_blocks.begin();
@@ -1400,7 +1393,7 @@ int32_t RTCPReceiver::CNAME(uint32_t remoteSSRC,
                             char cName[RTCP_CNAME_SIZE]) const {
   assert(cName);
 
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
   RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC);
   if (cnameInfo == NULL) {
     return -1;
@@ -1414,7 +1407,7 @@ int32_t RTCPReceiver::CNAME(uint32_t remoteSSRC,
 int32_t RTCPReceiver::TMMBRReceived(uint32_t size,
                                     uint32_t accNumCandidates,
                                     TMMBRSet* candidateSet) const {
-  CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+  rtc::CritScope lock(&_criticalSectionRTCPReceiver);
 
   std::map<uint32_t, RTCPReceiveInformation*>::const_iterator
       receiveInfoIt = _receivedInfoMap.begin();
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 475ab1e26f7..28c28cb69bb 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -15,6 +15,7 @@
 #include <set>
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
@@ -267,12 +268,12 @@ protected:
   int64_t _lastReceived;
   ModuleRtpRtcpImpl& _rtpRtcp;
 
-  CriticalSectionWrapper* _criticalSectionFeedbacks;
+  rtc::CriticalSection _criticalSectionFeedbacks;
   RtcpBandwidthObserver* const _cbRtcpBandwidthObserver;
   RtcpIntraFrameObserver* const _cbRtcpIntraFrameObserver;
   TransportFeedbackObserver* const _cbTransportFeedbackObserver;
 
-  CriticalSectionWrapper* _criticalSectionRTCPReceiver;
+  rtc::CriticalSection _criticalSectionRTCPReceiver;
   uint32_t main_ssrc_ GUARDED_BY(_criticalSectionRTCPReceiver);
   uint32_t _remoteSSRC GUARDED_BY(_criticalSectionRTCPReceiver);
   std::set<uint32_t> registered_ssrcs_ GUARDED_BY(_criticalSectionRTCPReceiver);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
index a5c0e282828..bfcc1bdfde0 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
@@ -42,14 +42,13 @@ RTCPPacketInformation::RTCPPacketInformation()
 RTCPPacketInformation::~RTCPPacketInformation()
 {
     delete [] applicationData;
-    delete VoIPMetric;
 }
 
 void
 RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
 {
-    VoIPMetric = new RTCPVoIPMetric();
-    memcpy(VoIPMetric, metric, sizeof(RTCPVoIPMetric));
+    VoIPMetric.reset(new RTCPVoIPMetric());
+    memcpy(VoIPMetric.get(), metric, sizeof(RTCPVoIPMetric));
 }
 
 void RTCPPacketInformation::AddApplicationData(const uint8_t* data,
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
index a7928419627..40d1220069c 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -12,10 +12,10 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 
 #include <list>
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPReportBlock
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
@@ -87,9 +87,9 @@ public:
 
     uint32_t xr_originator_ssrc;
     bool xr_dlrr_item;
-    RTCPVoIPMetric*  VoIPMetric;
+    std::unique_ptr<RTCPVoIPMetric>  VoIPMetric;
 
-    rtc::scoped_ptr<rtcp::TransportFeedback> transport_feedback_;
+    std::unique_ptr<rtcp::TransportFeedback> transport_feedback_;
 
 private:
     RTC_DISALLOW_COPY_AND_ASSIGN(RTCPPacketInformation);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 08f109bc297..924d009883d 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -8,10 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 
-/*
- * This file includes unit tests for the RTCPReceiver.
- */
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -136,7 +134,7 @@ class RtcpReceiverTest : public ::testing::Test {
     rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp;
     rtcp_packet_info_.xr_dlrr_item = rtcpPacketInformation.xr_dlrr_item;
     if (rtcpPacketInformation.VoIPMetric)
-      rtcp_packet_info_.AddVoIPMetric(rtcpPacketInformation.VoIPMetric);
+      rtcp_packet_info_.AddVoIPMetric(rtcpPacketInformation.VoIPMetric.get());
     rtcp_packet_info_.transport_feedback_.reset(
         rtcpPacketInformation.transport_feedback_.release());
     return 0;
@@ -149,7 +147,7 @@ class RtcpReceiverTest : public ::testing::Test {
   TestTransport* test_transport_;
   RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
   MockRemoteBitrateObserver remote_bitrate_observer_;
-  rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+  std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
 };
 
 
@@ -1244,6 +1242,20 @@ TEST_F(RtcpReceiverTest, ReceivesTransportFeedback) {
   EXPECT_TRUE(rtcp_packet_info_.transport_feedback_.get() != nullptr);
 }
 
+TEST_F(RtcpReceiverTest, ReceivesRemb) {
+  const uint32_t kSenderSsrc = 0x123456;
+  const uint32_t kBitrateBps = 500000;
+  rtcp::Remb remb;
+  remb.From(kSenderSsrc);
+  remb.WithBitrateBps(kBitrateBps);
+  rtc::Buffer built_packet = remb.Build();
+
+  EXPECT_EQ(0, InjectRtcpPacket(built_packet.data(), built_packet.size()));
+
+  EXPECT_EQ(kRtcpRemb, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpRemb);
+  EXPECT_EQ(kBitrateBps, rtcp_packet_info_.receiverEstimatedMaxBitrate);
+}
+
 TEST_F(RtcpReceiverTest, HandlesInvalidTransportFeedback) {
   const uint32_t kSenderSsrc = 0x10203;
   const uint32_t kSourceSsrc = 0x123456;
@@ -1261,7 +1273,7 @@ TEST_F(RtcpReceiverTest, HandlesInvalidTransportFeedback) {
 
   static uint32_t kBitrateBps = 50000;
   rtcp::Remb remb;
-  remb.From(kSourceSsrc);
+  remb.From(kSenderSsrc);
   remb.WithBitrateBps(kBitrateBps);
   rtcp::CompoundPacket compound;
   compound.Append(&packet);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 95bfeeea1f6..4a509b001ea 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -13,6 +13,7 @@
 #include <string.h>  // memcpy
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
@@ -638,16 +639,11 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBR(
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBN(
     const RtcpContext& ctx) {
-  TMMBRSet* boundingSet = tmmbr_help_.BoundingSetToSend();
-  if (boundingSet == nullptr)
-    return nullptr;
-
   rtcp::Tmmbn* tmmbn = new rtcp::Tmmbn();
   tmmbn->From(ssrc_);
-  for (uint32_t i = 0; i < boundingSet->lengthOfSet(); i++) {
-    if (boundingSet->Tmmbr(i) > 0) {
-      tmmbn->WithTmmbr(boundingSet->Ssrc(i), boundingSet->Tmmbr(i),
-                       boundingSet->PacketOH(i));
+  for (const rtcp::TmmbItem& tmmbr : tmmbn_to_send_) {
+    if (tmmbr.bitrate_bps() > 0) {
+      tmmbn->WithTmmbr(tmmbr);
     }
   }
 
@@ -871,11 +867,13 @@ void RTCPSender::PrepareReport(const std::set<RTCPPacketType>& packetTypes,
         random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
     next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
 
-    StatisticianMap statisticians =
-        receive_statistics_->GetActiveStatisticians();
-    RTC_DCHECK(report_blocks_.empty());
-    for (auto& it : statisticians) {
-      AddReportBlock(feedback_state, it.first, it.second);
+    if (receive_statistics_) {
+      StatisticianMap statisticians =
+          receive_statistics_->GetActiveStatisticians();
+      RTC_DCHECK(report_blocks_.empty());
+      for (auto& it : statisticians) {
+        AddReportBlock(feedback_state, it.first, it.second);
+      }
     }
   }
 }
@@ -972,14 +970,14 @@ bool RTCPSender::RtcpXrReceiverReferenceTime() const {
 }
 
 // no callbacks allowed inside this function
-int32_t RTCPSender::SetTMMBN(const TMMBRSet* boundingSet) {
+void RTCPSender::SetTMMBN(const std::vector<rtcp::TmmbItem>* bounding_set) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
-
-  if (0 == tmmbr_help_.SetTMMBRBoundingSetToSend(boundingSet)) {
-    SetFlag(kRtcpTmmbn, true);
-    return 0;
+  if (bounding_set) {
+    tmmbn_to_send_ = *bounding_set;
+  } else {
+    tmmbn_to_send_.clear();
   }
-  return -1;
+  SetFlag(kRtcpTmmbn, true);
 }
 
 void RTCPSender::SetFlag(RTCPPacketType type, bool is_volatile) {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index ba6fb700558..02719aa1c50 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -18,6 +18,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
@@ -134,7 +135,7 @@ class RTCPSender {
 
   void SetMaxPayloadLength(size_t max_payload_length);
 
-  int32_t SetTMMBN(const TMMBRSet* boundingSet);
+  void SetTMMBN(const std::vector<rtcp::TmmbItem>* boundingSet);
 
   int32_t SetApplicationSpecificData(uint8_t subType,
                                      uint32_t name,
@@ -240,6 +241,8 @@ class RTCPSender {
   std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(critical_section_rtcp_sender_);
 
   TMMBRHelp tmmbr_help_ GUARDED_BY(critical_section_rtcp_sender_);
+  std::vector<rtcp::TmmbItem> tmmbn_to_send_
+      GUARDED_BY(critical_section_rtcp_sender_);
   uint32_t tmmbr_send_ GUARDED_BY(critical_section_rtcp_sender_);
   uint32_t packet_oh_send_ GUARDED_BY(critical_section_rtcp_sender_);
   size_t max_payload_length_;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index dafc2e0be66..a4d6e59c8f8 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -8,10 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
-/*
- * This file includes unit tests for the RTCPSender.
- */
+#include <memory>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -261,9 +258,9 @@ class RtcpSenderTest : public ::testing::Test {
 
   SimulatedClock clock_;
   TestTransport test_transport_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
-  rtc::scoped_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
-  rtc::scoped_ptr<RTCPSender> rtcp_sender_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
+  std::unique_ptr<RTCPSender> rtcp_sender_;
 };
 
 TEST_F(RtcpSenderTest, SetRtcpStatus) {
@@ -691,13 +688,14 @@ TEST_F(RtcpSenderTest, TmmbrIncludedInCompoundPacketIfEnabled) {
 
 TEST_F(RtcpSenderTest, SendTmmbn) {
   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
-  TMMBRSet bounding_set;
-  bounding_set.VerifyAndAllocateSet(1);
+  std::vector<rtcp::TmmbItem> bounding_set;
   const uint32_t kBitrateKbps = 32768;
   const uint32_t kPacketOh = 40;
   const uint32_t kSourceSsrc = 12345;
-  bounding_set.AddEntry(kBitrateKbps, kPacketOh, kSourceSsrc);
-  EXPECT_EQ(0, rtcp_sender_->SetTMMBN(&bounding_set));
+  const rtcp::TmmbItem tmmbn(kSourceSsrc, kBitrateKbps * 1000, kPacketOh);
+  bounding_set.push_back(tmmbn);
+  rtcp_sender_->SetTMMBN(&bounding_set);
+
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
   EXPECT_EQ(1, parser()->sender_report()->num_packets());
   EXPECT_EQ(1, parser()->tmmbn()->num_packets());
@@ -716,8 +714,8 @@ TEST_F(RtcpSenderTest, SendTmmbn) {
 // situation where this caused confusion.
 TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndEmpty) {
   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
-  TMMBRSet bounding_set;
-  EXPECT_EQ(0, rtcp_sender_->SetTMMBN(&bounding_set));
+  std::vector<rtcp::TmmbItem> bounding_set;
+  rtcp_sender_->SetTMMBN(&bounding_set);
   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpSr));
   EXPECT_EQ(1, parser()->sender_report()->num_packets());
   EXPECT_EQ(1, parser()->tmmbn()->num_packets());
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
index 9b5a83515f4..c4f688aac44 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc
@@ -14,12 +14,17 @@
 #include <math.h>   // ceil
 #include <string.h> // memcpy
 
+#include <limits>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 namespace webrtc {
+namespace {
+constexpr uint64_t kMaxBitrateBps = std::numeric_limits<uint32_t>::max();
+}  // namespace
 
 namespace RTCPUtility {
 
@@ -1440,14 +1445,23 @@ RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
     }
 
     _packet.REMBItem.NumberOfSSRCs = *_ptrRTCPData++;
-    const uint8_t brExp = (_ptrRTCPData[0] >> 2) & 0x3F;
+    const uint8_t exp = (_ptrRTCPData[0] >> 2) & 0x3F;
 
-    uint32_t brMantissa = (_ptrRTCPData[0] & 0x03) << 16;
-    brMantissa += (_ptrRTCPData[1] << 8);
-    brMantissa += (_ptrRTCPData[2]);
+    uint64_t mantissa = (_ptrRTCPData[0] & 0x03) << 16;
+    mantissa += (_ptrRTCPData[1] << 8);
+    mantissa += (_ptrRTCPData[2]);
 
     _ptrRTCPData += 3; // Fwd read data
-    _packet.REMBItem.BitRate = (brMantissa << brExp);
+    uint64_t bitrate_bps = (mantissa << exp);
+    bool shift_overflow = exp > 0 && (mantissa >> (64 - exp)) != 0;
+    if (shift_overflow || bitrate_bps > kMaxBitrateBps) {
+      LOG(LS_ERROR) << "Unhandled remb bitrate value : " << mantissa
+                    << "*2^" << static_cast<int>(exp);
+      _state = ParseState::State_TopLevel;
+      EndCurrentBlock();
+      return false;
+    }
+    _packet.REMBItem.BitRate = bitrate_bps;
 
     const ptrdiff_t length_ssrcs = _ptrRTCPBlockEnd - _ptrRTCPData;
     if (length_ssrcs < 4 * _packet.REMBItem.NumberOfSSRCs)
@@ -1492,18 +1506,28 @@ RTCPUtility::RTCPParserV2::ParseTMMBRItem()
     _packet.TMMBRItem.SSRC += *_ptrRTCPData++ << 8;
     _packet.TMMBRItem.SSRC += *_ptrRTCPData++;
 
-    uint8_t mxtbrExp = (_ptrRTCPData[0] >> 2) & 0x3F;
+    uint8_t exp = (_ptrRTCPData[0] >> 2) & 0x3F;
 
-    uint32_t mxtbrMantissa = (_ptrRTCPData[0] & 0x03) << 15;
-    mxtbrMantissa += (_ptrRTCPData[1] << 7);
-    mxtbrMantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
+    uint64_t mantissa = (_ptrRTCPData[0] & 0x03) << 15;
+    mantissa += (_ptrRTCPData[1] << 7);
+    mantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
 
     uint32_t measuredOH = (_ptrRTCPData[2] & 0x01) << 8;
     measuredOH += _ptrRTCPData[3];
 
     _ptrRTCPData += 4; // Fwd read data
 
-    _packet.TMMBRItem.MaxTotalMediaBitRate = ((mxtbrMantissa << mxtbrExp) / 1000);
+    uint64_t bitrate_bps = (mantissa << exp);
+    bool shift_overflow = exp > 0 && (mantissa >> (64 - exp)) != 0;
+    if (shift_overflow || bitrate_bps > kMaxBitrateBps) {
+      LOG(LS_ERROR) << "Unhandled tmmbr bitrate value : " << mantissa
+                    << "*2^" << static_cast<int>(exp);
+      _state = ParseState::State_TopLevel;
+      EndCurrentBlock();
+      return false;
+    }
+
+    _packet.TMMBRItem.MaxTotalMediaBitRate = bitrate_bps / 1000;
     _packet.TMMBRItem.MeasuredOverhead     = measuredOH;
 
     return true;
@@ -1531,18 +1555,28 @@ RTCPUtility::RTCPParserV2::ParseTMMBNItem()
     _packet.TMMBNItem.SSRC += *_ptrRTCPData++ << 8;
     _packet.TMMBNItem.SSRC += *_ptrRTCPData++;
 
-    uint8_t mxtbrExp = (_ptrRTCPData[0] >> 2) & 0x3F;
+    uint8_t exp = (_ptrRTCPData[0] >> 2) & 0x3F;
 
-    uint32_t mxtbrMantissa = (_ptrRTCPData[0] & 0x03) << 15;
-    mxtbrMantissa += (_ptrRTCPData[1] << 7);
-    mxtbrMantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
+    uint64_t mantissa = (_ptrRTCPData[0] & 0x03) << 15;
+    mantissa += (_ptrRTCPData[1] << 7);
+    mantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
 
     uint32_t measuredOH = (_ptrRTCPData[2] & 0x01) << 8;
     measuredOH += _ptrRTCPData[3];
 
     _ptrRTCPData += 4; // Fwd read data
 
-    _packet.TMMBNItem.MaxTotalMediaBitRate = ((mxtbrMantissa << mxtbrExp) / 1000);
+    uint64_t bitrate_bps = (mantissa << exp);
+    bool shift_overflow = exp > 0 && (mantissa >> (64 - exp)) != 0;
+    if (shift_overflow || bitrate_bps > kMaxBitrateBps) {
+      LOG(LS_ERROR) << "Unhandled tmmbn bitrate value : " << mantissa
+                    << "*2^" << static_cast<int>(exp);
+      _state = ParseState::State_TopLevel;
+      EndCurrentBlock();
+      return false;
+    }
+
+    _packet.TMMBNItem.MaxTotalMediaBitRate = bitrate_bps / 1000;
     _packet.TMMBNItem.MeasuredOverhead     = measuredOH;
 
     return true;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
index 4067a40886f..629de4e99ec 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
@@ -13,7 +13,8 @@
 
 #include <stddef.h> // size_t, ptrdiff_t
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
@@ -468,7 +469,7 @@ class RTCPParserV2 {
 
   RTCPPacketTypes _packetType;
   RTCPPacket _packet;
-  rtc::scoped_ptr<webrtc::rtcp::RtcpPacket> rtcp_packet_;
+  std::unique_ptr<webrtc::rtcp::RtcpPacket> rtcp_packet_;
 };
 
 class RTCPPacketIterator {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
index e32433fe904..88258df8bcd 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -14,6 +14,7 @@
 #include <queue>
 #include <string>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index d29e3d4f212..12c2db564bd 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
@@ -72,7 +72,7 @@ void VerifyFua(size_t fua_index,
 void TestFua(size_t frame_size,
              size_t max_payload_size,
              const std::vector<size_t>& expected_sizes) {
-  rtc::scoped_ptr<uint8_t[]> frame;
+  std::unique_ptr<uint8_t[]> frame;
   frame.reset(new uint8_t[frame_size]);
   frame[0] = 0x05;  // F=0, NRI=0, Type=5.
   for (size_t i = 0; i < frame_size - kNalHeaderSize; ++i) {
@@ -82,11 +82,11 @@ void TestFua(size_t frame_size,
   fragmentation.VerifyAndAllocateFragmentationHeader(1);
   fragmentation.fragmentationOffset[0] = 0;
   fragmentation.fragmentationLength[0] = frame_size;
-  rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
+  std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
       kRtpVideoH264, max_payload_size, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame.get(), frame_size, &fragmentation);
 
-  rtc::scoped_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]);
+  std::unique_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]);
   size_t length = 0;
   bool last = false;
   size_t offset = kNalHeaderSize;
@@ -156,7 +156,7 @@ TEST(RtpPacketizerH264Test, TestSingleNalu) {
   fragmentation.VerifyAndAllocateFragmentationHeader(1);
   fragmentation.fragmentationOffset[0] = 0;
   fragmentation.fragmentationLength[0] = sizeof(frame);
-  rtc::scoped_ptr<RtpPacketizer> packetizer(
+  std::unique_ptr<RtpPacketizer> packetizer(
       RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, sizeof(frame), &fragmentation);
   uint8_t packet[kMaxPayloadSize] = {0};
@@ -185,7 +185,7 @@ TEST(RtpPacketizerH264Test, TestSingleNaluTwoPackets) {
   frame[fragmentation.fragmentationOffset[0]] = 0x01;
   frame[fragmentation.fragmentationOffset[1]] = 0x01;
 
-  rtc::scoped_ptr<RtpPacketizer> packetizer(
+  std::unique_ptr<RtpPacketizer> packetizer(
       RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
@@ -222,7 +222,7 @@ TEST(RtpPacketizerH264Test, TestStapA) {
   fragmentation.fragmentationOffset[2] = 4;
   fragmentation.fragmentationLength[2] =
       kNalHeaderSize + kFrameSize - kPayloadOffset;
-  rtc::scoped_ptr<RtpPacketizer> packetizer(
+  std::unique_ptr<RtpPacketizer> packetizer(
       RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
@@ -257,7 +257,7 @@ TEST(RtpPacketizerH264Test, TestTooSmallForStapAHeaders) {
   fragmentation.fragmentationOffset[2] = 4;
   fragmentation.fragmentationLength[2] =
       kNalHeaderSize + kFrameSize - kPayloadOffset;
-  rtc::scoped_ptr<RtpPacketizer> packetizer(
+  std::unique_ptr<RtpPacketizer> packetizer(
       RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
@@ -305,7 +305,7 @@ TEST(RtpPacketizerH264Test, TestMixedStapA_FUA) {
       frame[nalu_offset + j] = i + j;
     }
   }
-  rtc::scoped_ptr<RtpPacketizer> packetizer(
+  std::unique_ptr<RtpPacketizer> packetizer(
       RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame));
   packetizer->SetPayloadData(frame, kFrameSize, &fragmentation);
 
@@ -394,7 +394,7 @@ class RtpDepacketizerH264Test : public ::testing::Test {
                 ::testing::ElementsAreArray(data, length));
   }
 
-  rtc::scoped_ptr<RtpDepacketizer> depacketizer_;
+  std::unique_ptr<RtpDepacketizer> depacketizer_;
 };
 
 TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
index 3bf72e9dd35..e72fe310cfc 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
@@ -12,6 +12,7 @@
 
 #include <string>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
index 4283a778d00..079d9647545 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -8,9 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-/*
- * This file includes unit tests for the VP8 packetizer.
- */
+#include <memory>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -421,7 +419,7 @@ class RtpDepacketizerVp8Test : public ::testing::Test {
                 ::testing::ElementsAreArray(data, length));
   }
 
-  rtc::scoped_ptr<RtpDepacketizer> depacketizer_;
+  std::unique_ptr<RtpDepacketizer> depacketizer_;
 };
 
 TEST_F(RtpDepacketizerVp8Test, BasicHeader) {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
index 5bbafe459d2..f9514ad4bdb 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
@@ -76,7 +77,7 @@ void ParseAndCheckPacket(const uint8_t* packet,
                          const RTPVideoHeaderVP9& expected,
                          size_t expected_hdr_length,
                          size_t expected_length) {
-  rtc::scoped_ptr<RtpDepacketizer> depacketizer(new RtpDepacketizerVp9());
+  std::unique_ptr<RtpDepacketizer> depacketizer(new RtpDepacketizerVp9());
   RtpDepacketizer::ParsedPayload parsed;
   ASSERT_TRUE(depacketizer->Parse(&parsed, packet, expected_length));
   EXPECT_EQ(kRtpVideoVp9, parsed.type.Video.codec);
@@ -127,12 +128,12 @@ class RtpPacketizerVp9Test : public ::testing::Test {
     expected_.InitRTPVideoHeaderVP9();
   }
 
-  rtc::scoped_ptr<uint8_t[]> packet_;
-  rtc::scoped_ptr<uint8_t[]> payload_;
+  std::unique_ptr<uint8_t[]> packet_;
+  std::unique_ptr<uint8_t[]> payload_;
   size_t payload_size_;
   size_t payload_pos_;
   RTPVideoHeaderVP9 expected_;
-  rtc::scoped_ptr<RtpPacketizerVp9> packetizer_;
+  std::unique_ptr<RtpPacketizerVp9> packetizer_;
 
   void Init(size_t payload_size, size_t packet_size) {
     payload_.reset(new uint8_t[payload_size]);
@@ -469,7 +470,7 @@ class RtpDepacketizerVp9Test : public ::testing::Test {
   }
 
   RTPVideoHeaderVP9 expected_;
-  rtc::scoped_ptr<RtpDepacketizer> depacketizer_;
+  std::unique_ptr<RtpDepacketizer> depacketizer_;
 };
 
 TEST_F(RtpDepacketizerVp9Test, ParseBasicHeader) {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
index 8605925785e..2c2a0a13566 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
@@ -112,6 +112,14 @@ int32_t RtpHeaderExtensionMap::GetType(const uint8_t id,
   return 0;
 }
 
+RTPExtensionType RtpHeaderExtensionMap::GetType(uint8_t id) const {
+  auto it = extensionMap_.find(id);
+  if (it == extensionMap_.end()) {
+    return kInvalidType;
+  }
+  return it->second->type;
+}
+
 int32_t RtpHeaderExtensionMap::GetId(const RTPExtensionType type,
                                      uint8_t* id) const {
   assert(id);
@@ -129,6 +137,14 @@ int32_t RtpHeaderExtensionMap::GetId(const RTPExtensionType type,
   return -1;
 }
 
+uint8_t RtpHeaderExtensionMap::GetId(RTPExtensionType type) const {
+  for (auto kv : extensionMap_) {
+    if (kv.second->type == type)
+      return kv.first;
+  }
+  return kInvalidId;
+}
+
 size_t RtpHeaderExtensionMap::GetTotalLengthInBytes() const {
   // Get length for each extension block.
   size_t length = 0;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 342e38a1f2a..beaf989c895 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -70,6 +70,8 @@ struct HeaderExtension {
 
 class RtpHeaderExtensionMap {
  public:
+  static constexpr RTPExtensionType kInvalidType = kRtpExtensionNone;
+  static constexpr uint8_t kInvalidId = 0;
   RtpHeaderExtensionMap();
   ~RtpHeaderExtensionMap();
 
@@ -89,8 +91,12 @@ class RtpHeaderExtensionMap {
   bool IsRegistered(RTPExtensionType type) const;
 
   int32_t GetType(const uint8_t id, RTPExtensionType* type) const;
+  // Return kInvalidType if not found.
+  RTPExtensionType GetType(uint8_t id) const;
 
   int32_t GetId(const RTPExtensionType type, uint8_t* id) const;
+  // Return kInvalidId if not found.
+  uint8_t GetId(RTPExtensionType type) const;
 
   //
   // Methods below ignore any inactive rtp header extensions.
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
index ca37750621c..0b4f893e2c8 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
@@ -8,11 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
-/*
- * This file includes unit tests for the RtpHeaderExtensionMap.
- */
-
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
new file mode 100644
index 00000000000..a551b15617b
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+// Absolute send time in RTP streams.
+//
+// The absolute send time is signaled to the receiver in-band using the
+// general mechanism for RTP header extensions [RFC5285]. The payload
+// of this extension (the transmitted value) is a 24-bit unsigned integer
+// containing the sender's current time in seconds as a fixed point number
+// with 18 bits fractional part.
+//
+// The form of the absolute send time extension block:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=2 |              absolute send time               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* AbsoluteSendTime::kName =
+    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+bool AbsoluteSendTime::IsSupportedFor(MediaType type) {
+  return true;
+}
+
+bool AbsoluteSendTime::Parse(const uint8_t* data, uint32_t* value) {
+  *value = ByteReader<uint32_t, 3>::ReadBigEndian(data);
+  return true;
+}
+
+bool AbsoluteSendTime::Write(uint8_t* data, int64_t time_ms) {
+  const uint32_t kAbsSendTimeFraction = 18;
+  uint32_t time_24_bits =
+      static_cast<uint32_t>(((time_ms << kAbsSendTimeFraction) + 500) / 1000) &
+      0x00FFFFFF;
+
+  ByteWriter<uint32_t, 3>::WriteBigEndian(data, time_24_bits);
+  return true;
+}
+
+// An RTP Header Extension for Client-to-Mixer Audio Level Indication
+//
+// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
+//
+// The form of the audio level extension block:
+//
+//    0                   1
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=0 |V|   level     |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+const char* AudioLevel::kName = "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+bool AudioLevel::IsSupportedFor(MediaType type) {
+  switch (type) {
+    case MediaType::ANY:
+    case MediaType::AUDIO:
+      return true;
+    case MediaType::VIDEO:
+    case MediaType::DATA:
+      return false;
+  }
+  RTC_NOTREACHED();
+  return false;
+}
+
+bool AudioLevel::Parse(const uint8_t* data,
+                       bool* voice_activity,
+                       uint8_t* audio_level) {
+  *voice_activity = (data[0] & 0x80) != 0;
+  *audio_level = data[0] & 0x7F;
+  return true;
+}
+
+bool AudioLevel::Write(uint8_t* data,
+                       bool voice_activity,
+                       uint8_t audio_level) {
+  RTC_CHECK_LE(audio_level, 0x7f);
+  data[0] = (voice_activity ? 0x80 : 0x00) | audio_level;
+  return true;
+}
+
+// From RFC 5450: Transmission Time Offsets in RTP Streams.
+//
+// The transmission time is signaled to the receiver in-band using the
+// general mechanism for RTP header extensions [RFC5285]. The payload
+// of this extension (the transmitted value) is a 24-bit signed integer.
+// When added to the RTP timestamp of the packet, it represents the
+// "effective" RTP transmission time of the packet, on the RTP
+// timescale.
+//
+// The form of the transmission offset extension block:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=2 |              transmission offset              |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* TransmissionOffset::kName = "urn:ietf:params:rtp-hdrext:toffset";
+bool TransmissionOffset::IsSupportedFor(MediaType type) {
+  switch (type) {
+    case MediaType::ANY:
+    case MediaType::VIDEO:
+      return true;
+    case MediaType::AUDIO:
+    case MediaType::DATA:
+      return false;
+  }
+  RTC_NOTREACHED();
+  return false;
+}
+
+bool TransmissionOffset::Parse(const uint8_t* data, int32_t* value) {
+  *value = ByteReader<int32_t, 3>::ReadBigEndian(data);
+  return true;
+}
+
+bool TransmissionOffset::Write(uint8_t* data, int64_t value) {
+  RTC_CHECK_LE(value, 0x00ffffff);
+  ByteWriter<int32_t, 3>::WriteBigEndian(data, value);
+  return true;
+}
+
+//   0                   1                   2
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  ID   | L=1   |transport wide sequence number |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* TransportSequenceNumber::kName =
+    "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
+bool TransportSequenceNumber::IsSupportedFor(MediaType type) {
+  return true;
+}
+
+bool TransportSequenceNumber::Parse(const uint8_t* data, uint16_t* value) {
+  *value = ByteReader<uint16_t>::ReadBigEndian(data);
+  return true;
+}
+
+bool TransportSequenceNumber::Write(uint8_t* data, uint16_t value) {
+  ByteWriter<uint16_t>::WriteBigEndian(data, value);
+  return true;
+}
+
+// Coordination of Video Orientation in RTP streams.
+//
+// Coordination of Video Orientation consists in signaling of the current
+// orientation of the image captured on the sender side to the receiver for
+// appropriate rendering and displaying.
+//
+//    0                   1
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=0 |0 0 0 0 C F R R|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+const char* VideoOrientation::kName = "urn:3gpp:video-orientation";
+bool VideoOrientation::IsSupportedFor(MediaType type) {
+  switch (type) {
+    case MediaType::ANY:
+    case MediaType::VIDEO:
+      return true;
+    case MediaType::AUDIO:
+    case MediaType::DATA:
+      return false;
+  }
+  RTC_NOTREACHED();
+  return false;
+}
+
+bool VideoOrientation::Parse(const uint8_t* data, VideoRotation* rotation) {
+  *rotation = ConvertCVOByteToVideoRotation(data[0] & 0x03);
+  return true;
+}
+
+bool VideoOrientation::Write(uint8_t* data, VideoRotation rotation) {
+  data[0] = ConvertVideoRotationToCVOByte(rotation);
+  return true;
+}
+
+bool VideoOrientation::Parse(const uint8_t* data, uint8_t* value) {
+  *value = data[0];
+  return true;
+}
+
+bool VideoOrientation::Write(uint8_t* data, uint8_t value) {
+  data[0] = value;
+  return true;
+}
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
new file mode 100644
index 00000000000..cdbf806170d
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/call.h"
+#include "webrtc/common_video/rotation.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class AbsoluteSendTime {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
+  static constexpr uint8_t kValueSizeBytes = 3;
+  static const char* kName;
+  static bool IsSupportedFor(MediaType type);
+  static bool Parse(const uint8_t* data, uint32_t* time_ms);
+  static bool Write(uint8_t* data, int64_t time_ms);
+};
+
+class AudioLevel {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionAudioLevel;
+  static constexpr uint8_t kValueSizeBytes = 1;
+  static const char* kName;
+  static bool IsSupportedFor(MediaType type);
+  static bool Parse(const uint8_t* data,
+                    bool* voice_activity,
+                    uint8_t* audio_level);
+  static bool Write(uint8_t* data, bool voice_activity, uint8_t audio_level);
+};
+
+class TransmissionOffset {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionTransmissionTimeOffset;
+  static constexpr uint8_t kValueSizeBytes = 3;
+  static const char* kName;
+  static bool IsSupportedFor(MediaType type);
+  static bool Parse(const uint8_t* data, int32_t* time_ms);
+  static bool Write(uint8_t* data, int64_t time_ms);
+};
+
+class TransportSequenceNumber {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionTransportSequenceNumber;
+  static constexpr uint8_t kValueSizeBytes = 2;
+  static const char* kName;
+  static bool IsSupportedFor(MediaType type);
+  static bool Parse(const uint8_t* data, uint16_t* value);
+  static bool Write(uint8_t* data, uint16_t value);
+};
+
+class VideoOrientation {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
+  static constexpr uint8_t kValueSizeBytes = 1;
+  static const char* kName;
+  static bool IsSupportedFor(MediaType type);
+  static bool Parse(const uint8_t* data, VideoRotation* value);
+  static bool Write(uint8_t* data, VideoRotation value);
+  static bool Parse(const uint8_t* data, uint8_t* value);
+  static bool Write(uint8_t* data, uint8_t value);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index d4cbe544cc6..2cec8a3e0f6 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
@@ -9,10 +9,9 @@
  */
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -30,7 +29,7 @@ class RtpHeaderParserImpl : public RtpHeaderParser {
   bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
 
  private:
-  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_;
+  rtc::CriticalSection critical_section_;
   RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(critical_section_);
 };
 
@@ -38,8 +37,7 @@ RtpHeaderParser* RtpHeaderParser::Create() {
   return new RtpHeaderParserImpl;
 }
 
-RtpHeaderParserImpl::RtpHeaderParserImpl()
-    : critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {}
+RtpHeaderParserImpl::RtpHeaderParserImpl() {}
 
 bool RtpHeaderParser::IsRtcp(const uint8_t* packet, size_t length) {
   RtpUtility::RtpHeaderParser rtp_parser(packet, length);
@@ -54,7 +52,7 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
 
   RtpHeaderExtensionMap map;
   {
-    CriticalSectionScoped cs(critical_section_.get());
+    rtc::CritScope cs(&critical_section_);
     rtp_header_extension_map_.GetCopy(&map);
   }
 
@@ -67,12 +65,12 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
 
 bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
                                                      uint8_t id) {
-  CriticalSectionScoped cs(critical_section_.get());
+  rtc::CritScope cs(&critical_section_);
   return rtp_header_extension_map_.Register(type, id) == 0;
 }
 
 bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
-  CriticalSectionScoped cs(critical_section_.get());
+  rtc::CritScope cs(&critical_section_);
   return rtp_header_extension_map_.Deregister(type) == 0;
 }
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
new file mode 100644
index 00000000000..f6634867f68
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
@@ -0,0 +1,509 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet.h"
+
+#include <cstring>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/random.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtp {
+namespace {
+constexpr size_t kFixedHeaderSize = 12;
+constexpr uint8_t kRtpVersion = 2;
+constexpr uint16_t kOneByteExtensionId = 0xBEDE;
+constexpr size_t kOneByteHeaderSize = 1;
+constexpr size_t kDefaultPacketSize = 1500;
+}  // namespace
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |V=2|P|X|  CC   |M|     PT      |       sequence number         |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |                           timestamp                           |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |           synchronization source (SSRC) identifier            |
+// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// |            Contributing source (CSRC) identifiers             |
+// |                             ....                              |
+// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// |One-byte eXtensions id = 0xbede|       length in 32bits        |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |                          Extensions                           |
+// |                             ....                              |
+// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// |                           Payload                             |
+// |             ....              :  padding...                   |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |               padding         | Padding size  |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Packet::Packet(const ExtensionManager* extensions)
+    : extensions_(extensions), buffer_(kDefaultPacketSize) {
+  Clear();
+}
+
+Packet::Packet(const ExtensionManager* extensions, size_t capacity)
+    : extensions_(extensions), buffer_(capacity) {
+  RTC_DCHECK_GE(capacity, kFixedHeaderSize);
+  Clear();
+}
+
+Packet::~Packet() {}
+
+void Packet::IdentifyExtensions(const ExtensionManager* extensions) {
+  RTC_DCHECK(extensions);
+  extensions_ = extensions;
+  for (size_t i = 0; i < num_extensions_; ++i) {
+    uint8_t id = data()[extension_entries_[i].offset - 1] >> 4;
+    extension_entries_[i].type = extensions_->GetType(id);
+  }
+}
+
+bool Packet::Parse(const uint8_t* buffer, size_t buffer_size) {
+  if (!ParseBuffer(buffer, buffer_size)) {
+    Clear();
+    return false;
+  }
+  RTC_DCHECK_EQ(size(), buffer_size);
+  buffer_.SetData(buffer, buffer_size);
+  return true;
+}
+
+bool Packet::Parse(rtc::Buffer buffer) {
+  if (!ParseBuffer(buffer.data(), buffer.size())) {
+    Clear();
+    return false;
+  }
+  RTC_DCHECK_EQ(size(), buffer.size());
+  buffer_ = std::move(buffer);
+  return true;
+}
+
+bool Packet::Marker() const {
+  RTC_DCHECK_EQ(marker_, (data()[1] & 0x80) != 0);
+  return marker_;
+}
+
+uint8_t Packet::PayloadType() const {
+  RTC_DCHECK_EQ(payload_type_, data()[1] & 0x7f);
+  return payload_type_;
+}
+
+uint16_t Packet::SequenceNumber() const {
+  RTC_DCHECK_EQ(sequence_number_,
+                ByteReader<uint16_t>::ReadBigEndian(data() + 2));
+  return sequence_number_;
+}
+
+uint32_t Packet::Timestamp() const {
+  RTC_DCHECK_EQ(timestamp_, ByteReader<uint32_t>::ReadBigEndian(data() + 4));
+  return timestamp_;
+}
+
+uint32_t Packet::Ssrc() const {
+  RTC_DCHECK_EQ(ssrc_, ByteReader<uint32_t>::ReadBigEndian(data() + 8));
+  return ssrc_;
+}
+
+std::vector<uint32_t> Packet::Csrcs() const {
+  size_t num_csrc = data()[0] & 0x0F;
+  RTC_DCHECK_GE(capacity(), kFixedHeaderSize + num_csrc * 4);
+  std::vector<uint32_t> csrcs(num_csrc);
+  for (size_t i = 0; i < num_csrc; ++i) {
+    csrcs[i] =
+        ByteReader<uint32_t>::ReadBigEndian(&data()[kFixedHeaderSize + i * 4]);
+  }
+  return csrcs;
+}
+
+void Packet::GetHeader(RTPHeader* header) const {
+  header->markerBit = Marker();
+  header->payloadType = PayloadType();
+  header->sequenceNumber = SequenceNumber();
+  header->timestamp = Timestamp();
+  header->ssrc = Ssrc();
+  std::vector<uint32_t> csrcs = Csrcs();
+  header->numCSRCs = csrcs.size();
+  for (size_t i = 0; i < csrcs.size(); ++i) {
+    header->arrOfCSRCs[i] = csrcs[i];
+  }
+  header->paddingLength = padding_size();
+  header->headerLength = headers_size();
+  header->payload_type_frequency = 0;
+  header->extension.hasTransmissionTimeOffset =
+      GetExtension<TransmissionOffset>(
+          &header->extension.transmissionTimeOffset);
+  header->extension.hasAbsoluteSendTime =
+      GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
+  header->extension.hasTransportSequenceNumber =
+      GetExtension<TransportSequenceNumber>(
+          &header->extension.transportSequenceNumber);
+  header->extension.hasAudioLevel = GetExtension<AudioLevel>(
+      &header->extension.voiceActivity, &header->extension.audioLevel);
+  header->extension.hasVideoRotation =
+      GetExtension<VideoOrientation>(&header->extension.videoRotation);
+}
+
+size_t Packet::headers_size() const {
+  return payload_offset_;
+}
+
+size_t Packet::payload_size() const {
+  return payload_size_;
+}
+
+size_t Packet::padding_size() const {
+  return padding_size_;
+}
+
+const uint8_t* Packet::payload() const {
+  return data() + payload_offset_;
+}
+
+size_t Packet::capacity() const {
+  return buffer_.size();
+}
+
+size_t Packet::size() const {
+  return payload_offset_ + payload_size_ + padding_size_;
+}
+
+const uint8_t* Packet::data() const {
+  return buffer_.data();
+}
+
+size_t Packet::FreeCapacity() const {
+  return capacity() - size();
+}
+
+size_t Packet::MaxPayloadSize() const {
+  return capacity() - payload_offset_;
+}
+
+void Packet::CopyHeader(const Packet& packet) {
+  RTC_DCHECK_GE(capacity(), packet.headers_size());
+
+  marker_ = packet.marker_;
+  payload_type_ = packet.payload_type_;
+  sequence_number_ = packet.sequence_number_;
+  timestamp_ = packet.timestamp_;
+  ssrc_ = packet.ssrc_;
+  payload_offset_ = packet.payload_offset_;
+  num_extensions_ = packet.num_extensions_;
+  for (size_t i = 0; i < num_extensions_; ++i) {
+    extension_entries_[i] = packet.extension_entries_[i];
+  }
+  extensions_size_ = packet.extensions_size_;
+  buffer_.SetData(packet.data(), packet.headers_size());
+  // Reset payload and padding.
+  payload_size_ = 0;
+  padding_size_ = 0;
+}
+
+void Packet::SetMarker(bool marker_bit) {
+  marker_ = marker_bit;
+  if (marker_) {
+    WriteAt(1, data()[1] | 0x80);
+  } else {
+    WriteAt(1, data()[1] & 0x7F);
+  }
+}
+
+void Packet::SetPayloadType(uint8_t payload_type) {
+  RTC_DCHECK_LE(payload_type, 0x7Fu);
+  payload_type_ = payload_type;
+  WriteAt(1, (data()[1] & 0x80) | payload_type);
+}
+
+void Packet::SetSequenceNumber(uint16_t seq_no) {
+  sequence_number_ = seq_no;
+  ByteWriter<uint16_t>::WriteBigEndian(WriteAt(2), seq_no);
+}
+
+void Packet::SetTimestamp(uint32_t timestamp) {
+  timestamp_ = timestamp;
+  ByteWriter<uint32_t>::WriteBigEndian(WriteAt(4), timestamp);
+}
+
+void Packet::SetSsrc(uint32_t ssrc) {
+  ssrc_ = ssrc;
+  ByteWriter<uint32_t>::WriteBigEndian(WriteAt(8), ssrc);
+}
+
+void Packet::SetCsrcs(const std::vector<uint32_t>& csrcs) {
+  RTC_DCHECK_EQ(num_extensions_, 0u);
+  RTC_DCHECK_EQ(payload_size_, 0u);
+  RTC_DCHECK_EQ(padding_size_, 0u);
+  RTC_DCHECK_LE(csrcs.size(), 0x0fu);
+  RTC_DCHECK_LE(kFixedHeaderSize + 4 * csrcs.size(), capacity());
+  payload_offset_ = kFixedHeaderSize + 4 * csrcs.size();
+  WriteAt(0, (data()[0] & 0xF0) | csrcs.size());
+  size_t offset = kFixedHeaderSize;
+  for (uint32_t csrc : csrcs) {
+    ByteWriter<uint32_t>::WriteBigEndian(WriteAt(offset), csrc);
+    offset += 4;
+  }
+}
+
+uint8_t* Packet::AllocatePayload(size_t size_bytes) {
+  RTC_DCHECK_EQ(padding_size_, 0u);
+  if (payload_offset_ + size_bytes > capacity()) {
+    LOG(LS_WARNING) << "Cannot set payload, not enough space in buffer.";
+    return nullptr;
+  }
+  payload_size_ = size_bytes;
+  return WriteAt(payload_offset_);
+}
+
+void Packet::SetPayloadSize(size_t size_bytes) {
+  RTC_DCHECK_EQ(padding_size_, 0u);
+  RTC_DCHECK_LE(size_bytes, payload_size_);
+  payload_size_ = size_bytes;
+}
+
+bool Packet::SetPadding(uint8_t size_bytes, Random* random) {
+  RTC_DCHECK(random);
+  if (payload_offset_ + payload_size_ + size_bytes > capacity()) {
+    LOG(LS_WARNING) << "Cannot set padding size " << size_bytes << ", only "
+                    << (capacity() - payload_offset_ - payload_size_)
+                    << " bytes left in buffer.";
+    return false;
+  }
+  padding_size_ = size_bytes;
+  if (padding_size_ > 0) {
+    size_t padding_offset = payload_offset_ + payload_size_;
+    size_t padding_end = padding_offset + padding_size_;
+    for (size_t offset = padding_offset; offset < padding_end - 1; ++offset) {
+      WriteAt(offset, random->Rand<uint8_t>());
+    }
+    WriteAt(padding_end - 1, padding_size_);
+    WriteAt(0, data()[0] | 0x20);  // Set padding bit.
+  } else {
+    WriteAt(0, data()[0] & ~0x20);  // Clear padding bit.
+  }
+  return true;
+}
+
+void Packet::Clear() {
+  marker_ = false;
+  payload_type_ = 0;
+  sequence_number_ = 0;
+  timestamp_ = 0;
+  ssrc_ = 0;
+  payload_offset_ = kFixedHeaderSize;
+  payload_size_ = 0;
+  padding_size_ = 0;
+  num_extensions_ = 0;
+  extensions_size_ = 0;
+
+  memset(WriteAt(0), 0, kFixedHeaderSize);
+  WriteAt(0, kRtpVersion << 6);
+}
+
+bool Packet::ParseBuffer(const uint8_t* buffer, size_t size) {
+  if (size < kFixedHeaderSize) {
+    return false;
+  }
+  const uint8_t version = buffer[0] >> 6;
+  if (version != kRtpVersion) {
+    return false;
+  }
+  const bool has_padding = (buffer[0] & 0x20) != 0;
+  const bool has_extension = (buffer[0] & 0x10) != 0;
+  const uint8_t number_of_crcs = buffer[0] & 0x0f;
+  marker_ = (buffer[1] & 0x80) != 0;
+  payload_type_ = buffer[1] & 0x7f;
+
+  sequence_number_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]);
+  timestamp_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  if (size < kFixedHeaderSize + number_of_crcs * 4) {
+    return false;
+  }
+  payload_offset_ = kFixedHeaderSize + number_of_crcs * 4;
+
+  if (has_padding) {
+    padding_size_ = buffer[size - 1];
+    if (padding_size_ == 0) {
+      LOG(LS_WARNING) << "Padding was set, but padding size is zero";
+      return false;
+    }
+  } else {
+    padding_size_ = 0;
+  }
+
+  num_extensions_ = 0;
+  extensions_size_ = 0;
+  if (has_extension) {
+    /* RTP header extension, RFC 3550.
+     0                   1                   2                   3
+     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |      defined by profile       |           length              |
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |                        header extension                       |
+    |                             ....                              |
+    */
+    size_t extension_offset = payload_offset_ + 4;
+    if (extension_offset > size) {
+      return false;
+    }
+    uint16_t profile =
+        ByteReader<uint16_t>::ReadBigEndian(&buffer[payload_offset_]);
+    size_t extensions_capacity =
+        ByteReader<uint16_t>::ReadBigEndian(&buffer[payload_offset_ + 2]);
+    extensions_capacity *= 4;
+    if (extension_offset + extensions_capacity > size) {
+      return false;
+    }
+    if (profile != kOneByteExtensionId) {
+      LOG(LS_WARNING) << "Unsupported rtp extension " << profile;
+    } else {
+      constexpr uint8_t kPaddingId = 0;
+      constexpr uint8_t kReservedId = 15;
+      while (extensions_size_ + kOneByteHeaderSize < extensions_capacity) {
+        uint8_t id = buffer[extension_offset + extensions_size_] >> 4;
+        if (id == kReservedId) {
+          break;
+        } else if (id == kPaddingId) {
+          extensions_size_++;
+          continue;
+        }
+        uint8_t length =
+            1 + (buffer[extension_offset + extensions_size_] & 0xf);
+        extensions_size_ += kOneByteHeaderSize;
+        if (num_extensions_ >= kMaxExtensionHeaders) {
+          LOG(LS_WARNING) << "Too many extensions.";
+          return false;
+        }
+        extension_entries_[num_extensions_].type =
+            extensions_ ? extensions_->GetType(id)
+                        : ExtensionManager::kInvalidType;
+        extension_entries_[num_extensions_].length = length;
+        extension_entries_[num_extensions_].offset =
+            extension_offset + extensions_size_;
+        num_extensions_++;
+        extensions_size_ += length;
+      }
+    }
+    payload_offset_ = extension_offset + extensions_capacity;
+  }
+
+  if (payload_offset_ + padding_size_ > size) {
+    return false;
+  }
+  payload_size_ = size - payload_offset_ - padding_size_;
+  return true;
+}
+
+bool Packet::FindExtension(ExtensionType type,
+                           uint8_t length,
+                           uint16_t* offset) const {
+  RTC_DCHECK(offset);
+  for (size_t i = 0; i < num_extensions_; ++i) {
+    if (extension_entries_[i].type == type) {
+      RTC_CHECK_EQ(length, extension_entries_[i].length)
+          << "Length mismatch for extension '" << type << "'"
+          << "should be " << length << ", received "
+          << extension_entries_[i].length;
+      *offset = extension_entries_[i].offset;
+      return true;
+    }
+  }
+  return false;
+}
+
+bool Packet::AllocateExtension(ExtensionType type,
+                               uint8_t length,
+                               uint16_t* offset) {
+  if (!extensions_) {
+    return false;
+  }
+  if (FindExtension(type, length, offset)) {
+    return true;
+  }
+
+  // Can't add new extension after payload/padding was set.
+  if (payload_size_ > 0) {
+    return false;
+  }
+  if (padding_size_ > 0) {
+    return false;
+  }
+
+  uint8_t extension_id = extensions_->GetId(type);
+  if (extension_id == ExtensionManager::kInvalidId) {
+    return false;
+  }
+  RTC_DCHECK_GT(length, 0u);
+  RTC_DCHECK_LE(length, 16u);
+
+  size_t num_csrc = data()[0] & 0x0F;
+  size_t extensions_offset = kFixedHeaderSize + (num_csrc * 4) + 4;
+  if (extensions_offset + extensions_size_ + kOneByteHeaderSize + length >
+      capacity()) {
+    LOG(LS_WARNING) << "Extension cannot be registered: "
+                       "Not enough space left in buffer.";
+    return false;
+  }
+
+  uint16_t new_extensions_size =
+      extensions_size_ + kOneByteHeaderSize + length;
+  uint16_t extensions_words =
+      (new_extensions_size + 3) / 4;  // Wrap up to 32bit.
+
+  // All checks passed, write down the extension.
+  if (num_extensions_ == 0) {
+    RTC_DCHECK_EQ(payload_offset_, kFixedHeaderSize + (num_csrc * 4));
+    RTC_DCHECK_EQ(extensions_size_, 0);
+    WriteAt(0, data()[0] | 0x10);  // Set extension bit.
+    // Profile specific ID always set to OneByteExtensionHeader.
+    ByteWriter<uint16_t>::WriteBigEndian(WriteAt(extensions_offset - 4),
+                                         kOneByteExtensionId);
+  }
+
+  WriteAt(extensions_offset + extensions_size_,
+          (extension_id << 4) | (length - 1));
+  RTC_DCHECK(num_extensions_ < kMaxExtensionHeaders);
+  extension_entries_[num_extensions_].type = type;
+  extension_entries_[num_extensions_].length = length;
+  *offset = extensions_offset + kOneByteHeaderSize + extensions_size_;
+  extension_entries_[num_extensions_].offset = *offset;
+  ++num_extensions_;
+  extensions_size_ = new_extensions_size;
+
+  // Update header length field.
+  ByteWriter<uint16_t>::WriteBigEndian(WriteAt(extensions_offset - 2),
+                                       extensions_words);
+  // Fill extension padding place with zeroes.
+  size_t extension_padding_size = 4 * extensions_words - extensions_size_;
+  memset(WriteAt(extensions_offset + extensions_size_), 0,
+         extension_padding_size);
+  payload_offset_ = extensions_offset + 4 * extensions_words;
+  return true;
+}
+
+uint8_t* Packet::WriteAt(size_t offset) {
+  return buffer_.data() + offset;
+}
+
+void Packet::WriteAt(size_t offset, uint8_t byte) {
+  buffer_.data()[offset] = byte;
+}
+
+}  // namespace rtp
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.h
new file mode 100644
index 00000000000..b2687ca9bab
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.h
@@ -0,0 +1,187 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
+
+#include <vector>
+
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+struct RTPHeader;
+class RtpHeaderExtensionMap;
+class Random;
+
+namespace rtp {
+class Packet {
+ public:
+  using ExtensionType = RTPExtensionType;
+  using ExtensionManager = RtpHeaderExtensionMap;
+  static constexpr size_t kMaxExtensionHeaders = 14;
+
+  // Parse and copy given buffer into Packet.
+  bool Parse(const uint8_t* buffer, size_t size);
+
+  // Parse and move given buffer into Packet.
+  bool Parse(rtc::Buffer packet);
+
+  // Maps parsed extensions to their types to allow use of GetExtension.
+  // Used after parsing when |extensions| can't be provided until base rtp
+  // header is parsed.
+  void IdentifyExtensions(const ExtensionManager* extensions);
+
+  // Header.
+  bool Marker() const;
+  uint8_t PayloadType() const;
+  uint16_t SequenceNumber() const;
+  uint32_t Timestamp() const;
+  uint32_t Ssrc() const;
+  std::vector<uint32_t> Csrcs() const;
+
+  // TODO(danilchap): Remove this function when all code update to use RtpPacket
+  // directly. Function is there just for easier backward compatibilty.
+  void GetHeader(RTPHeader* header) const;
+
+  size_t headers_size() const;
+
+  // Payload.
+  size_t payload_size() const;
+  size_t padding_size() const;
+  const uint8_t* payload() const;
+
+  // Buffer.
+  size_t capacity() const;
+  size_t size() const;
+  const uint8_t* data() const;
+  size_t FreeCapacity() const;
+  size_t MaxPayloadSize() const;
+
+  // Reset fields and buffer.
+  void Clear();
+
+  // Header setters.
+  void CopyHeader(const Packet& packet);
+  void SetMarker(bool marker_bit);
+  void SetPayloadType(uint8_t payload_type);
+  void SetSequenceNumber(uint16_t seq_no);
+  void SetTimestamp(uint32_t timestamp);
+  void SetSsrc(uint32_t ssrc);
+
+  // Writes csrc list. Assumes:
+  // a) There is enough room left in buffer.
+  // b) Extension headers, payload or padding data has not already been added.
+  void SetCsrcs(const std::vector<uint32_t>& csrcs);
+
+  // Header extensions.
+  template <typename Extension, typename... Values>
+  bool GetExtension(Values...) const;
+
+  template <typename Extension, typename... Values>
+  bool SetExtension(Values...);
+
+  template <typename Extension>
+  bool ReserveExtension();
+
+  // Reserve size_bytes for payload. Returns nullptr on failure.
+  uint8_t* AllocatePayload(size_t size_bytes);
+  void SetPayloadSize(size_t size_bytes);
+  bool SetPadding(uint8_t size_bytes, Random* random);
+
+ protected:
+  // |extensions| required for SetExtension/ReserveExtension functions during
+  // packet creating and used if available in Parse function.
+  // Adding and getting extensions will fail until |extensions| is
+  // provided via constructor or IdentifyExtensions function.
+  explicit Packet(const ExtensionManager* extensions);
+  Packet(const ExtensionManager* extensions, size_t capacity);
+  virtual ~Packet();
+
+ private:
+  struct ExtensionInfo {
+    ExtensionType type;
+    uint16_t offset;
+    uint8_t length;
+  };
+
+  // Helper function for Parse. Fill header fields using data in given buffer,
+  // but does not touch packet own buffer, leaving packet in invalid state.
+  bool ParseBuffer(const uint8_t* buffer, size_t size);
+
+  // Find an extension based on the type field of the parameter.
+  // If found, length field would be validated, the offset field will be set
+  // and true returned,
+  // otherwise the parameter will be unchanged and false is returned.
+  bool FindExtension(ExtensionType type,
+                     uint8_t length,
+                     uint16_t* offset) const;
+
+  // Find or allocate an extension, based on the type field of the parameter.
+  // If found, the length field be checked against what is already registered
+  // and the offset field will be set, then true is returned. If allocated, the
+  // length field will be used for allocation and the offset update to indicate
+  // position, the true is returned.
+  // If not found and allocations fails, false is returned and parameter remains
+  // unchanged.
+  bool AllocateExtension(ExtensionType type, uint8_t length, uint16_t* offset);
+
+  uint8_t* WriteAt(size_t offset);
+  void WriteAt(size_t offset, uint8_t byte);
+
+  const ExtensionManager* extensions_;
+
+  // Header.
+  bool marker_;
+  uint8_t payload_type_;
+  uint8_t padding_size_;
+  uint16_t sequence_number_;
+  uint32_t timestamp_;
+  uint32_t ssrc_;
+  size_t payload_offset_;  // Match header size with csrcs and extensions.
+  size_t payload_size_;
+
+  uint8_t num_extensions_ = 0;
+  ExtensionInfo extension_entries_[kMaxExtensionHeaders];
+  uint16_t extensions_size_ = 0;  // Unaligned.
+  rtc::Buffer buffer_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Packet);
+};
+
+template <typename Extension, typename... Values>
+bool Packet::GetExtension(Values... values) const {
+  uint16_t offset = 0;
+  if (!FindExtension(Extension::kId, Extension::kValueSizeBytes, &offset))
+    return false;
+  return Extension::Parse(data() + offset, values...);
+}
+
+template <typename Extension, typename... Values>
+bool Packet::SetExtension(Values... values) {
+  uint16_t offset = 0;
+  if (!AllocateExtension(Extension::kId, Extension::kValueSizeBytes, &offset))
+    return false;
+  return Extension::Write(WriteAt(offset), values...);
+}
+
+template <typename Extension>
+bool Packet::ReserveExtension() {
+  uint16_t offset = 0;
+  if (!AllocateExtension(Extension::kId, Extension::kValueSizeBytes, &offset))
+    return false;
+  memset(WriteAt(offset), 0, Extension::kValueSizeBytes);
+  return true;
+}
+}  // namespace rtp
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index 49f9d8530a9..713fba87707 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -21,7 +21,6 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -29,7 +28,6 @@ static const int kMinPacketRequestBytes = 50;
 
 RTPPacketHistory::RTPPacketHistory(Clock* clock)
     : clock_(clock),
-      critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       store_(false),
       prev_index_(0) {}
 
@@ -38,7 +36,7 @@ RTPPacketHistory::~RTPPacketHistory() {
 
 void RTPPacketHistory::SetStorePacketsStatus(bool enable,
                                              uint16_t number_to_store) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   if (enable) {
     if (store_) {
       LOG(LS_WARNING) << "Purging packet history in order to re-set status.";
@@ -70,7 +68,7 @@ void RTPPacketHistory::Free() {
 }
 
 bool RTPPacketHistory::StorePackets() const {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   return store_;
 }
 
@@ -78,7 +76,7 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
                                        size_t packet_length,
                                        int64_t capture_time_ms,
                                        StorageType type) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   if (!store_) {
     return 0;
   }
@@ -131,7 +129,7 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
 }
 
 bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   if (!store_) {
     return false;
   }
@@ -150,7 +148,7 @@ bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const {
 }
 
 bool RTPPacketHistory::SetSent(uint16_t sequence_number) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   if (!store_) {
     return false;
   }
@@ -176,7 +174,7 @@ bool RTPPacketHistory::GetPacketAndSetSendTime(uint16_t sequence_number,
                                                uint8_t* packet,
                                                size_t* packet_length,
                                                int64_t* stored_time_ms) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   RTC_CHECK_GE(*packet_length, static_cast<size_t>(IP_PACKET_SIZE));
   if (!store_)
     return false;
@@ -232,7 +230,7 @@ void RTPPacketHistory::GetPacket(int index,
 bool RTPPacketHistory::GetBestFittingPacket(uint8_t* packet,
                                             size_t* packet_length,
                                             int64_t* stored_time_ms) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   if (!store_)
     return false;
   int index = FindBestFittingPacket(*packet_length);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index 8e1a732b199..b4d48aa2ced 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -15,6 +15,7 @@
 
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -23,7 +24,6 @@
 namespace webrtc {
 
 class Clock;
-class CriticalSectionWrapper;
 
 static const size_t kMaxHistoryCapacity = 9600;
 
@@ -71,19 +71,19 @@ class RTPPacketHistory {
                  uint8_t* packet,
                  size_t* packet_length,
                  int64_t* stored_time_ms) const
-      EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
-  void Allocate(size_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
-  void Free() EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+      EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+  void Allocate(size_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+  void Free() EXCLUSIVE_LOCKS_REQUIRED(critsect_);
   void VerifyAndAllocatePacketLength(size_t packet_length, uint32_t start_index)
-      EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+      EXCLUSIVE_LOCKS_REQUIRED(critsect_);
   bool FindSeqNum(uint16_t sequence_number, int32_t* index) const
-      EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+      EXCLUSIVE_LOCKS_REQUIRED(critsect_);
   int FindBestFittingPacket(size_t size) const
-      EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
+      EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
  private:
   Clock* clock_;
-  rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
+  rtc::CriticalSection critsect_;
   bool store_ GUARDED_BY(critsect_);
   uint32_t prev_index_ GUARDED_BY(critsect_);
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index a406d8bc9b4..7580a809235 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -6,8 +6,6 @@
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
- *
- * This file includes unit tests for the RTPPacketHistory.
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
new file mode 100644
index 00000000000..e2222b9200a
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet.h"
+#include "webrtc/system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+// Class to hold rtp packet with metadata for receiver side.
+class RtpPacketReceived : public rtp::Packet {
+ public:
+  RtpPacketReceived() : Packet(nullptr) {}
+  explicit RtpPacketReceived(const ExtensionManager* extensions)
+      : Packet(extensions) {}
+
+  void GetHeader(RTPHeader* header) const {
+    Packet::GetHeader(header);
+    header->payload_type_frequency = payload_type_frequency();
+  }
+
+  // Time in local time base as close as it can to packet arrived on the
+  // network.
+  int64_t arrival_time_ms() const { return arrival_time_ms_; }
+  void set_arrival_time_ms(int64_t time) { arrival_time_ms_ = time; }
+
+  // Estimated from Timestamp() using rtcp Sender Reports.
+  NtpTime capture_ntp_time() const { return capture_time_; }
+  void set_capture_ntp_time(NtpTime time) { capture_time_ = time; }
+
+  // Flag if packet arrived via rtx.
+  bool retransmit() const { return retransmit_; }
+  void set_retransmit(bool value) { retransmit_ = value; }
+
+  int payload_type_frequency() const { return payload_type_frequency_; }
+  void set_payload_type_frequency(int value) {
+    payload_type_frequency_ = value;
+  }
+
+ private:
+  NtpTime capture_time_;
+  int64_t arrival_time_ms_ = 0;
+  int payload_type_frequency_ = 0;
+  bool retransmit_ = false;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
new file mode 100644
index 00000000000..ad749ffb61e
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet.h"
+
+namespace webrtc {
+// Class to hold rtp packet with metadata for sender side.
+class RtpPacketToSend : public rtp::Packet {
+ public:
+  explicit RtpPacketToSend(const ExtensionManager* extensions)
+      : Packet(extensions) {}
+  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
+      : Packet(extensions, capacity) {}
+
+  // Time in local time base as close as it can to frame capture time.
+  int64_t capture_time_ms() const { return capture_time_ms_; }
+  void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
+
+ private:
+  int64_t capture_time_ms_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
new file mode 100644
index 00000000000..b992d2da909
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -0,0 +1,252 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+
+using testing::ElementsAreArray;
+using testing::make_tuple;
+
+namespace webrtc {
+namespace {
+constexpr int8_t kPayloadType = 100;
+constexpr uint32_t kSsrc = 0x12345678;
+constexpr uint16_t kSeqNum = 88;
+constexpr uint32_t kTimestamp = 0x65431278;
+constexpr uint8_t kTransmissionOffsetExtensionId = 1;
+constexpr uint8_t kAudioLevelExtensionId = 9;
+constexpr int32_t kTimeOffset = 0x56ce;
+constexpr bool kVoiceActive = true;
+constexpr uint8_t kAudioLevel = 0x5a;
+constexpr size_t kMaxPaddingSize = 224u;
+constexpr uint8_t kMinimumPacket[] = {
+    0x80, kPayloadType, 0x00, kSeqNum,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78};
+constexpr uint8_t kPacketWithTO[] = {
+    0x90, kPayloadType, 0x00, kSeqNum,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x01,
+    0x12, 0x00, 0x56, 0xce};
+
+constexpr uint8_t kPacketWithTOAndAL[] = {
+    0x90, kPayloadType, 0x00, kSeqNum,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x02,
+    0x12, 0x00, 0x56, 0xce,
+    0x90, 0x80|kAudioLevel, 0x00, 0x00};
+
+constexpr uint32_t kCsrcs[] = {0x34567890, 0x32435465};
+constexpr uint8_t kPayload[] = {'p', 'a', 'y', 'l', 'o', 'a', 'd'};
+constexpr uint8_t kPacketPaddingSize = 8;
+constexpr uint8_t kPacket[] = {
+    0xb2, kPayloadType, 0x00, kSeqNum,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x34, 0x56, 0x78, 0x90,
+    0x32, 0x43, 0x54, 0x65,
+    0xbe, 0xde, 0x00, 0x01,
+    0x12, 0x00, 0x56, 0xce,
+    'p', 'a', 'y', 'l', 'o', 'a', 'd',
+    'p', 'a', 'd', 'd', 'i', 'n', 'g', kPacketPaddingSize};
+
+}  // namespace
+
+TEST(RtpPacketTest, CreateMinimum) {
+  RtpPacketToSend packet(nullptr);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  EXPECT_THAT(kMinimumPacket, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWithExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWith2Extensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel);
+  EXPECT_THAT(kPacketWithTOAndAL,
+              ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
+  const size_t kPayloadSize = 4;
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+
+  EXPECT_TRUE(packet.ReserveExtension<TransmissionOffset>());
+  packet.AllocatePayload(kPayloadSize);
+  // Can't set extension after payload.
+  EXPECT_FALSE(packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel));
+  // Unless reserved.
+  EXPECT_TRUE(packet.SetExtension<TransmissionOffset>(kTimeOffset));
+}
+
+TEST(RtpPacketTest, CreatePurePadding) {
+  const size_t kPaddingSize = kMaxPaddingSize - 1;
+  RtpPacketToSend packet(nullptr, 12 + kPaddingSize);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  Random random(0x123456789);
+
+  EXPECT_LT(packet.size(), packet.capacity());
+  EXPECT_FALSE(packet.SetPadding(kPaddingSize + 1, &random));
+  EXPECT_TRUE(packet.SetPadding(kPaddingSize, &random));
+  EXPECT_EQ(packet.size(), packet.capacity());
+}
+
+TEST(RtpPacketTest, CreateUnalignedPadding) {
+  const size_t kPayloadSize = 3;  // Make padding start at unaligned address.
+  RtpPacketToSend packet(nullptr, 12 + kPayloadSize + kMaxPaddingSize);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.AllocatePayload(kPayloadSize);
+  Random r(0x123456789);
+
+  EXPECT_LT(packet.size(), packet.capacity());
+  EXPECT_TRUE(packet.SetPadding(kMaxPaddingSize, &r));
+  EXPECT_EQ(packet.size(), packet.capacity());
+}
+
+TEST(RtpPacketTest, ParseMinimum) {
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(kMinimumPacket, sizeof(kMinimumPacket)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_EQ(0u, packet.padding_size());
+  EXPECT_EQ(0u, packet.payload_size());
+}
+
+TEST(RtpPacketTest, ParseBuffer) {
+  rtc::Buffer unparsed(kMinimumPacket);
+  const uint8_t* raw = unparsed.data();
+
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(std::move(unparsed)));
+  EXPECT_EQ(raw, packet.data());  // Expect packet took over the buffer.
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_EQ(0u, packet.padding_size());
+  EXPECT_EQ(0u, packet.payload_size());
+}
+
+TEST(RtpPacketTest, ParseWithExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  EXPECT_EQ(0u, packet.payload_size());
+  EXPECT_EQ(0u, packet.padding_size());
+}
+
+TEST(RtpPacketTest, ParseWith2Extensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTOAndAL, sizeof(kPacketWithTOAndAL)));
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  bool voice_active;
+  uint8_t audio_level;
+  EXPECT_TRUE(packet.GetExtension<AudioLevel>(&voice_active, &audio_level));
+  EXPECT_EQ(kVoiceActive, voice_active);
+  EXPECT_EQ(kAudioLevel, audio_level);
+}
+
+TEST(RtpPacketTest, ParseWithAllFeatures) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacket, sizeof(kPacket)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_THAT(packet.Csrcs(), ElementsAreArray(kCsrcs));
+  EXPECT_THAT(make_tuple(packet.payload(), packet.payload_size()),
+              ElementsAreArray(kPayload));
+  EXPECT_EQ(kPacketPaddingSize, packet.padding_size());
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+}
+
+TEST(RtpPacketTest, ParseWithExtensionDelayed) {
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+
+  int32_t time_offset;
+  EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  packet.IdentifyExtensions(&extensions);
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  EXPECT_EQ(0u, packet.payload_size());
+  EXPECT_EQ(0u, packet.padding_size());
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index f3793d0901e..283c2846e1b 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -16,8 +16,7 @@
 namespace webrtc {
 
 RTPPayloadRegistry::RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      rtp_payload_strategy_(rtp_payload_strategy),
+    : rtp_payload_strategy_(rtp_payload_strategy),
       red_payload_type_(-1),
       ulpfec_payload_type_(-1),
       incoming_payload_type_(-1),
@@ -67,7 +66,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
 
   size_t payload_name_length = strlen(payload_name);
 
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
 
   RtpUtility::PayloadTypeMap::iterator it =
       payload_type_map_.find(payload_type);
@@ -122,7 +121,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
 
 int32_t RTPPayloadRegistry::DeRegisterReceivePayload(
     const int8_t payload_type) {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   RtpUtility::PayloadTypeMap::iterator it =
       payload_type_map_.find(payload_type);
   assert(it != payload_type_map_.end());
@@ -176,7 +175,7 @@ int32_t RTPPayloadRegistry::ReceivePayloadType(
   assert(payload_type);
   size_t payload_name_length = strlen(payload_name);
 
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
 
   RtpUtility::PayloadTypeMap::const_iterator it = payload_type_map_.begin();
 
@@ -218,12 +217,12 @@ int32_t RTPPayloadRegistry::ReceivePayloadType(
 }
 
 bool RTPPayloadRegistry::RtxEnabled() const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   return rtx_;
 }
 
 bool RTPPayloadRegistry::IsRtx(const RTPHeader& header) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   return IsRtxInternal(header);
 }
 
@@ -231,15 +230,6 @@ bool RTPPayloadRegistry::IsRtxInternal(const RTPHeader& header) const {
   return rtx_ && ssrc_rtx_ == header.ssrc;
 }
 
-bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet,
-                                               const uint8_t* packet,
-                                               size_t* packet_length,
-                                               uint32_t original_ssrc,
-                                               const RTPHeader& header) const {
-  return RestoreOriginalPacket(*restored_packet, packet, packet_length,
-                               original_ssrc, header);
-}
-
 bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t* restored_packet,
                                                const uint8_t* packet,
                                                size_t* packet_length,
@@ -264,7 +254,7 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t* restored_packet,
                                        original_sequence_number);
   ByteWriter<uint32_t>::WriteBigEndian(restored_packet + 8, original_ssrc);
 
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   if (!rtx_)
     return true;
 
@@ -290,20 +280,20 @@ bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t* restored_packet,
 }
 
 void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   ssrc_rtx_ = ssrc;
   rtx_ = true;
 }
 
 bool RTPPayloadRegistry::GetRtxSsrc(uint32_t* ssrc) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   *ssrc = ssrc_rtx_;
   return rtx_;
 }
 
 void RTPPayloadRegistry::SetRtxPayloadType(int payload_type,
                                            int associated_payload_type) {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   if (payload_type < 0) {
     LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
     return;
@@ -315,7 +305,7 @@ void RTPPayloadRegistry::SetRtxPayloadType(int payload_type,
 }
 
 bool RTPPayloadRegistry::IsRed(const RTPHeader& header) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   return red_payload_type_ == header.payloadType;
 }
 
@@ -325,7 +315,7 @@ bool RTPPayloadRegistry::IsEncapsulated(const RTPHeader& header) const {
 
 bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type,
                                              PayloadUnion* payload) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   RtpUtility::PayloadTypeMap::const_iterator it =
       payload_type_map_.find(payload_type);
 
@@ -343,13 +333,13 @@ int RTPPayloadRegistry::GetPayloadTypeFrequency(
   if (!payload) {
     return -1;
   }
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   return rtp_payload_strategy_->GetPayloadTypeFrequency(*payload);
 }
 
 const RtpUtility::Payload* RTPPayloadRegistry::PayloadTypeToPayload(
     uint8_t payload_type) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
 
   RtpUtility::PayloadTypeMap::const_iterator it =
       payload_type_map_.find(payload_type);
@@ -363,13 +353,13 @@ const RtpUtility::Payload* RTPPayloadRegistry::PayloadTypeToPayload(
 }
 
 void RTPPayloadRegistry::SetIncomingPayloadType(const RTPHeader& header) {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   if (!IsRtxInternal(header))
     incoming_payload_type_ = header.payloadType;
 }
 
 bool RTPPayloadRegistry::ReportMediaPayloadType(uint8_t media_payload_type) {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   if (last_received_media_payload_type_ == media_payload_type) {
     // Media type unchanged.
     return true;
@@ -416,7 +406,8 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy {
     return payload;
   }
 
-  int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const {
+  int GetPayloadTypeFrequency(
+      const RtpUtility::Payload& payload) const override {
     return payload.typeSpecific.Audio.frequency;
   }
 };
@@ -466,7 +457,8 @@ class RTPPayloadVideoStrategy : public RTPPayloadStrategy {
     return payload;
   }
 
-  int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const {
+  int GetPayloadTypeFrequency(
+      const RtpUtility::Payload& payload) const override {
     return kVideoPayloadTypeFrequency;
   }
 };
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index cbded6872d3..5bbe97a32ce 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
@@ -58,7 +59,7 @@ class RtpPayloadRegistryTest : public ::testing::Test {
     return returned_payload_on_heap;
   }
 
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   testing::NiceMock<MockRTPPayloadStrategy>* mock_payload_strategy_;
 };
 
@@ -296,9 +297,9 @@ void TestRtxPacket(RTPPayloadRegistry* rtp_payload_registry,
   uint16_t original_sequence_number = 1234;
   uint32_t original_ssrc = 500;
 
-  rtc::scoped_ptr<const uint8_t[]> packet(GenerateRtxPacket(
+  std::unique_ptr<const uint8_t[]> packet(GenerateRtxPacket(
       header_length, payload_length, original_sequence_number));
-  rtc::scoped_ptr<uint8_t[]> restored_packet(
+  std::unique_ptr<uint8_t[]> restored_packet(
       new uint8_t[header_length + payload_length]);
   size_t length = original_length;
   bool success = rtp_payload_registry->RestoreOriginalPacket(
@@ -312,7 +313,7 @@ void TestRtxPacket(RTPPayloadRegistry* rtp_payload_registry,
   EXPECT_EQ(original_length - kRtxHeaderSize, length)
       << "The restored packet should be exactly kRtxHeaderSize smaller.";
 
-  rtc::scoped_ptr<RtpHeaderParser> header_parser(RtpHeaderParser::Create());
+  std::unique_ptr<RtpHeaderParser> header_parser(RtpHeaderParser::Create());
   RTPHeader restored_header;
   ASSERT_TRUE(
       header_parser->Parse(restored_packet.get(), length, &restored_header));
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 7a3b6fd8290..38b2830b79c 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
@@ -46,26 +45,26 @@ RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
 // Outband TelephoneEvent(DTMF) detection
 void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
     bool forward_to_decoder) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   telephone_event_forward_to_decoder_ = forward_to_decoder;
 }
 
 // Is forwarding of outband telephone events turned on/off?
 bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return telephone_event_forward_to_decoder_;
 }
 
 bool RTPReceiverAudio::TelephoneEventPayloadType(
     int8_t payload_type) const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return telephone_event_payload_type_ == payload_type;
 }
 
 bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type,
                                       uint32_t* frequency,
                                       bool* cng_payload_type_has_changed) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   *cng_payload_type_has_changed = false;
 
   //  We can have four CNG on 8000Hz, 16000Hz, 32000Hz and 48000Hz.
@@ -152,7 +151,7 @@ int32_t RTPReceiverAudio::OnNewPayloadTypeCreated(
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     int8_t payload_type,
     uint32_t frequency) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
 
   if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) {
     telephone_event_payload_type_ = payload_type;
@@ -194,6 +193,10 @@ int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
            rtp_header->type.Audio.numEnergy);
   }
 
+  if (first_packet_received_()) {
+    LOG(LS_INFO) << "Received first audio RTP packet";
+  }
+
   return ParseAudioCodecSpecific(rtp_header,
                                  payload,
                                  payload_length,
@@ -202,7 +205,7 @@ int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
 }
 
 int RTPReceiverAudio::GetPayloadTypeFrequency() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (last_received_g722_) {
     return 8000;
   }
@@ -245,7 +248,7 @@ void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
 }
 
 int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
 
   assert(num_energy_ <= kRtpCsrcSize);
 
@@ -287,7 +290,7 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
   bool telephone_event_packet =
       TelephoneEventPayloadType(rtp_header->header.payloadType);
   if (telephone_event_packet) {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
 
     // RFC 4733 2.3
     // 0                   1                   2                   3
@@ -332,7 +335,7 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
   }
 
   {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
 
     if (!telephone_event_packet) {
       last_received_frequency_ = audio_specific.frequency;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index b68febbb82b..d5d89bae2d4 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -13,7 +13,7 @@
 
 #include <set>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -22,8 +22,6 @@
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-
 // Handles audio RTP packets. This class is thread-safe.
 class RTPReceiverAudio : public RTPReceiverStrategy,
                          public TelephoneEventHandler {
@@ -33,15 +31,15 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
 
   // The following three methods implement the TelephoneEventHandler interface.
   // Forward DTMFs to decoder for playout.
-  void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
+  void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override;
 
   // Is forwarding of outband telephone events turned on/off?
-  bool TelephoneEventForwardToDecoder() const;
+  bool TelephoneEventForwardToDecoder() const override;
 
   // Is TelephoneEvent configured with payload type payload_type
-  bool TelephoneEventPayloadType(const int8_t payload_type) const;
+  bool TelephoneEventPayloadType(const int8_t payload_type) const override;
 
-  TelephoneEventHandler* GetTelephoneEventHandler() { return this; }
+  TelephoneEventHandler* GetTelephoneEventHandler() override { return this; }
 
   // Returns true if CNG is configured with payload type payload_type. If so,
   // the frequency and cng_payload_type_has_changed are filled in.
@@ -118,6 +116,8 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
 
   uint8_t num_energy_;
   uint8_t current_remote_energy_[kRtpCsrcSize];
+
+  ThreadUnsafeOneTimeEvent first_packet_received_;
 };
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 6f2efe783a1..190449b3ddf 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -61,8 +61,6 @@ RtpReceiverImpl::RtpReceiverImpl(
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
       cb_rtp_feedback_(incoming_messages_callback),
-      critical_section_rtp_receiver_(
-          CriticalSectionWrapper::CreateCriticalSection()),
       last_receive_time_(0),
       last_received_payload_length_(0),
       ssrc_(0),
@@ -70,8 +68,7 @@ RtpReceiverImpl::RtpReceiverImpl(
       current_remote_csrc_(),
       last_received_timestamp_(0),
       last_received_frame_time_ms_(-1),
-      last_received_sequence_number_(0),
-      nack_method_(kNackOff) {
+      last_received_sequence_number_(0) {
   assert(incoming_messages_callback);
 
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
@@ -89,7 +86,7 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(
     const uint32_t frequency,
     const size_t channels,
     const uint32_t rate) {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
 
   // TODO(phoglund): Try to streamline handling of the RED codec and some other
   // cases which makes it necessary to keep track of whether we created a
@@ -111,29 +108,18 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(
 
 int32_t RtpReceiverImpl::DeRegisterReceivePayload(
     const int8_t payload_type) {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
   return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
 }
 
-NACKMethod RtpReceiverImpl::NACK() const {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
-  return nack_method_;
-}
-
-// Turn negative acknowledgment requests on/off.
-void RtpReceiverImpl::SetNACKStatus(const NACKMethod method) {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
-  nack_method_ = method;
-}
-
 uint32_t RtpReceiverImpl::SSRC() const {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
   return ssrc_;
 }
 
 // Get remote CSRC.
 int32_t RtpReceiverImpl::CSRCs(uint32_t array_of_csrcs[kRtpCsrcSize]) const {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
 
   assert(num_csrcs_ <= kRtpCsrcSize);
 
@@ -179,7 +165,7 @@ bool RtpReceiverImpl::IncomingRtpPacket(
 
   bool is_first_packet_in_frame = false;
   {
-    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+    rtc::CritScope lock(&critical_section_rtp_receiver_);
     if (HaveReceivedFrame()) {
       is_first_packet_in_frame =
           last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
@@ -198,7 +184,7 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   }
 
   {
-    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+    rtc::CritScope lock(&critical_section_rtp_receiver_);
 
     last_receive_time_ = clock_->TimeInMilliseconds();
     last_received_payload_length_ = payload_data_length;
@@ -219,7 +205,7 @@ TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
 }
 
 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!HaveReceivedFrame())
     return false;
   *timestamp = last_received_timestamp_;
@@ -227,7 +213,7 @@ bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
 }
 
 bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
-  CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!HaveReceivedFrame())
     return false;
   *receive_time_ms = last_received_frame_time_ms_;
@@ -247,7 +233,7 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
   uint32_t rate = 0;
 
   {
-    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+    rtc::CritScope lock(&critical_section_rtp_receiver_);
 
     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
@@ -318,7 +304,7 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
   int8_t payload_type = rtp_header.payloadType;
 
   {
-    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+    rtc::CritScope lock(&critical_section_rtp_receiver_);
 
     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
@@ -401,7 +387,7 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
   uint8_t old_num_csrcs = 0;
 
   {
-    CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
+    rtc::CritScope lock(&critical_section_rtp_receiver_);
 
     if (!rtp_media_receiver_->ShouldReportCsrcChanges(
         rtp_header.header.payloadType)) {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 63b65fefd8e..1ae1c9168a6 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -11,11 +11,12 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -46,11 +47,6 @@ class RtpReceiverImpl : public RtpReceiver {
                          PayloadUnion payload_specific,
                          bool in_order) override;
 
-  NACKMethod NACK() const override;
-
-  // Turn negative acknowledgement requests on/off.
-  void SetNACKStatus(const NACKMethod method) override;
-
   // Returns the last received timestamp.
   bool Timestamp(uint32_t* timestamp) const override;
   bool LastReceivedTimeMs(int64_t* receive_time_ms) const override;
@@ -75,11 +71,11 @@ class RtpReceiverImpl : public RtpReceiver {
 
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
-  rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
+  std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
 
   RtpFeedback* cb_rtp_feedback_;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
+  rtc::CriticalSection critical_section_rtp_receiver_;
   int64_t last_receive_time_;
   size_t last_received_payload_length_;
 
@@ -91,8 +87,6 @@ class RtpReceiverImpl : public RtpReceiver {
   uint32_t last_received_timestamp_;
   int64_t last_received_frame_time_ms_;
   uint16_t last_received_sequence_number_;
-
-  NACKMethod nack_method_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
index 3797b1bcc20..69d079f9aa3 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
@@ -12,25 +12,22 @@
 
 #include <stdlib.h>
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-
 namespace webrtc {
 
 RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback)
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      data_callback_(data_callback) {
+    : data_callback_(data_callback) {
   memset(&last_payload_, 0, sizeof(last_payload_));
 }
 
 void RTPReceiverStrategy::GetLastMediaSpecificPayload(
     PayloadUnion* payload) const {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   memcpy(payload, &last_payload_, sizeof(*payload));
 }
 
 void RTPReceiverStrategy::SetLastMediaSpecificPayload(
     const PayloadUnion& payload) {
-  CriticalSectionScoped cs(crit_sect_.get());
+  rtc::CritScope cs(&crit_sect_);
   memcpy(&last_payload_, &payload, sizeof(last_payload_));
 }
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index f2a60ff855a..663b883295d 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -11,11 +11,10 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -95,7 +94,7 @@ class RTPReceiverStrategy {
   // packet.
   explicit RTPReceiverStrategy(RtpData* data_callback);
 
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  rtc::CriticalSection crit_sect_;
   PayloadUnion last_payload_;
   RtpData* data_callback_;
 };
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 406acc23c20..9d76c1a6163 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -13,6 +13,8 @@
 #include <assert.h>
 #include <string.h>
 
+#include <memory>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
@@ -21,7 +23,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -70,8 +71,12 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                                                                            : -1;
   }
 
+  if (first_packet_received_()) {
+    LOG(LS_INFO) << "Received first video RTP packet";
+  }
+
   // We are not allowed to hold a critical section when calling below functions.
-  rtc::scoped_ptr<RtpDepacketizer> depacketizer(
+  std::unique_ptr<RtpDepacketizer> depacketizer(
       RtpDepacketizer::Create(rtp_header->type.Video.codec));
   if (depacketizer.get() == NULL) {
     LOG(LS_ERROR) << "Failed to create depacketizer.";
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 56f761a2e1c..486eced3641 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -34,7 +34,7 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
                          int64_t timestamp,
                          bool is_first_packet) override;
 
-  TelephoneEventHandler* GetTelephoneEventHandler() { return NULL; }
+  TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }
 
   int GetPayloadTypeFrequency() const override;
 
@@ -54,6 +54,9 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
       const PayloadUnion& specific_payload) const override;
 
   void SetPacketOverHead(uint16_t packet_over_head);
+
+ private:
+  OneTimeEvent first_packet_received_;
 };
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 5875529cfb2..214472f81ae 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -60,7 +60,8 @@ RtpRtcp::Configuration::Configuration()
       send_bitrate_observer(nullptr),
       send_frame_count_observer(nullptr),
       send_side_delay_observer(nullptr),
-      event_log(nullptr) {}
+      event_log(nullptr),
+      send_packet_observer(nullptr) {}
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
   if (configuration.clock) {
@@ -85,7 +86,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
                   configuration.send_bitrate_observer,
                   configuration.send_frame_count_observer,
                   configuration.send_side_delay_observer,
-                  configuration.event_log),
+                  configuration.event_log,
+                  configuration.send_packet_observer),
       rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
@@ -105,14 +107,13 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
       last_process_time_(configuration.clock->TimeInMilliseconds()),
       last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
       last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
-      packet_overhead_(28),                     // IPV4 UDP.
+      packet_overhead_(28),  // IPV4 UDP.
       nack_last_time_sent_full_(0),
       nack_last_time_sent_full_prev_(0),
       nack_last_seq_number_sent_(0),
       key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
-      critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()),
       rtt_ms_(0) {
   // Make sure that RTCP objects are aware of our SSRC.
   uint32_t SSRC = rtp_sender_.SSRC();
@@ -301,30 +302,21 @@ void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
   rtp_sender_.SetSequenceNumber(seq_num);
 }
 
-bool ModuleRtpRtcpImpl::SetRtpStateForSsrc(uint32_t ssrc,
-                                           const RtpState& rtp_state) {
-  if (rtp_sender_.SSRC() == ssrc) {
-    SetStartTimestamp(rtp_state.start_timestamp);
-    rtp_sender_.SetRtpState(rtp_state);
-    return true;
-  }
-  if (rtp_sender_.RtxSsrc() == ssrc) {
-    rtp_sender_.SetRtxRtpState(rtp_state);
-    return true;
-  }
-  return false;
+void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
+  SetStartTimestamp(rtp_state.start_timestamp);
+  rtp_sender_.SetRtpState(rtp_state);
 }
 
-bool ModuleRtpRtcpImpl::GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) {
-  if (rtp_sender_.SSRC() == ssrc) {
-    *rtp_state = rtp_sender_.GetRtpState();
-    return true;
-  }
-  if (rtp_sender_.RtxSsrc() == ssrc) {
-    *rtp_state = rtp_sender_.GetRtxRtpState();
-    return true;
-  }
-  return false;
+void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
+  rtp_sender_.SetRtxRtpState(rtp_state);
+}
+
+RtpState ModuleRtpRtcpImpl::GetRtpState() const {
+  return rtp_sender_.GetRtpState();
+}
+
+RtpState ModuleRtpRtcpImpl::GetRtxState() const {
+  return rtp_sender_.GetRtxRtpState();
 }
 
 uint32_t ModuleRtpRtcpImpl::SSRC() const {
@@ -688,8 +680,9 @@ void ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) {
   rtcp_sender_.SetTMMBRStatus(enable);
 }
 
-int32_t ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* bounding_set) {
-  return rtcp_sender_.SetTMMBN(bounding_set);
+void ModuleRtpRtcpImpl::SetTMMBN(
+    const std::vector<rtcp::TmmbItem>* bounding_set) {
+  rtcp_sender_.SetTMMBN(bounding_set);
 }
 
 // Returns the currently configured retransmission mode.
@@ -982,12 +975,12 @@ void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
 }
 
 void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
-  CriticalSectionScoped cs(critical_section_rtt_.get());
+  rtc::CritScope cs(&critical_section_rtt_);
   rtt_ms_ = rtt_ms;
 }
 
 int64_t ModuleRtpRtcpImpl::rtt_ms() const {
-  CriticalSectionScoped cs(critical_section_rtt_.get());
+  rtc::CritScope cs(&critical_section_rtt_);
   return rtt_ms_;
 }
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 76faca0f7eb..7bbb06e5289 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -16,8 +16,8 @@
 #include <utility>
 #include <vector>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gtest_prod_util.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
@@ -75,8 +75,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Set SequenceNumber, default is a random number.
   void SetSequenceNumber(uint16_t seq) override;
 
-  bool SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) override;
-  bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) override;
+  void SetRtpState(const RtpState& rtp_state) override;
+  void SetRtxState(const RtpState& rtp_state) override;
+  RtpState GetRtpState() const override;
+  RtpState GetRtxState() const override;
 
   uint32_t SSRC() const override;
 
@@ -200,7 +202,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   void SetTMMBRStatus(bool enable) override;
 
-  int32_t SetTMMBN(const TMMBRSet* bounding_set);
+  void SetTMMBN(const std::vector<rtcp::TmmbItem>* bounding_set);
 
   uint16_t MaxPayloadLength() const override;
 
@@ -362,7 +364,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
-  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
+  rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 708b9af1e09..7e0ac312c80 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include <map>
+#include <memory>
 #include <set>
 
 #include "testing/gmock/include/gmock/gmock.h"
@@ -68,7 +69,7 @@ class SendTransport : public Transport,
                size_t len,
                const PacketOptions& options) override {
     RTPHeader header;
-    rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+    std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), len, &header));
     ++rtp_packets_sent_;
     last_rtp_header_ = header;
@@ -115,10 +116,10 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
 
   RtcpPacketTypeCounter packets_sent_;
   RtcpPacketTypeCounter packets_received_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
   SendTransport transport_;
   RtcpRttStatsTestImpl rtt_stats_;
-  rtc::scoped_ptr<ModuleRtpRtcpImpl> impl_;
+  std::unique_ptr<ModuleRtpRtcpImpl> impl_;
   uint32_t remote_ssrc_;
 
   void SetRemoteSsrc(uint32_t ssrc) {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 3fbca7b67d8..b58a94d457f 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -17,6 +17,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
@@ -24,8 +25,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -114,12 +113,11 @@ RTPSender::RTPSender(
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
-    RtcEventLog* event_log)
+    RtcEventLog* event_log,
+    SendPacketObserver* send_packet_observer)
     : clock_(clock),
-      // TODO(holmer): Remove this conversion when we remove the use of
-      // TickTime.
-      clock_delta_ms_(clock_->TimeInMilliseconds() -
-                      TickTime::MillisecondTimestamp()),
+      // TODO(holmer): Remove this conversion?
+      clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
       random_(clock_->TimeInMicroseconds()),
       bitrates_(bitrate_callback),
       total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
@@ -147,11 +145,11 @@ RTPSender::RTPSender(
       nack_bitrate_(clock, bitrates_.retransmit_bitrate_observer()),
       packet_history_(clock),
       // Statistics
-      statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_stats_callback_(NULL),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
       event_log_(event_log),
+      send_packet_observer_(send_packet_observer),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
@@ -166,7 +164,6 @@ RTPSender::RTPSender(
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
-      target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       target_bitrate_(0) {
   memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
   memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
@@ -210,12 +207,12 @@ RTPSender::~RTPSender() {
 }
 
 void RTPSender::SetTargetBitrate(uint32_t bitrate) {
-  CriticalSectionScoped cs(target_bitrate_critsect_.get());
+  rtc::CritScope cs(&target_bitrate_critsect_);
   target_bitrate_ = bitrate;
 }
 
 uint32_t RTPSender::GetTargetBitrate() {
-  CriticalSectionScoped cs(target_bitrate_critsect_.get());
+  rtc::CritScope cs(&target_bitrate_critsect_);
   return target_bitrate_;
 }
 
@@ -532,7 +529,7 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
                           payload_size, fragmentation, rtp_hdr);
   }
 
-  CriticalSectionScoped cs(statistics_crit_.get());
+  rtc::CritScope cs(&statistics_crit_);
   // Note: This is currently only counting for video.
   if (frame_type == kVideoFrameKey) {
     ++frame_counts_.key_frames;
@@ -675,13 +672,12 @@ size_t RTPSender::SendPadData(size_t bytes,
     UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
 
     PacketOptions options;
-    if (using_transport_seq) {
-      options.packet_id =
-          UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
-    }
-
-    if (using_transport_seq && transport_feedback_observer_) {
-      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+    if (AllocateTransportSequenceNumber(&options.packet_id)) {
+      if (UpdateTransportSequenceNumber(options.packet_id, padding_packet,
+                                        length, rtp_header)) {
+        if (transport_feedback_observer_)
+          transport_feedback_observer_->AddPacket(options.packet_id, length);
+      }
     }
 
     if (!SendPacketToNetwork(padding_packet, length, options))
@@ -886,9 +882,7 @@ bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
     // Packet cannot be found. Allow sending to continue.
     return true;
   }
-  if (!retransmission && capture_time_ms > 0) {
-    UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds());
-  }
+
   int rtx;
   {
     rtc::CritScope lock(&send_critsect_);
@@ -932,19 +926,18 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
                                diff_ms);
   UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
 
-  // TODO(sprang): Potentially too much overhead in IsRegistered()?
-  bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
-                                 kRtpExtensionTransportSequenceNumber) &&
-                             transport_sequence_number_allocator_;
-
   PacketOptions options;
-  if (using_transport_seq) {
-    options.packet_id =
-        UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
+  if (AllocateTransportSequenceNumber(&options.packet_id)) {
+    if (UpdateTransportSequenceNumber(options.packet_id, buffer_to_send_ptr,
+                                      length, rtp_header)) {
+      if (transport_feedback_observer_)
+        transport_feedback_observer_->AddPacket(options.packet_id, length);
+    }
   }
 
-  if (using_transport_seq && transport_feedback_observer_) {
-    transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+  if (!is_retransmit && !send_over_rtx) {
+    UpdateDelayStatistics(capture_time_ms, now_ms);
+    UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
   }
 
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
@@ -966,7 +959,7 @@ void RTPSender::UpdateRtpStats(const uint8_t* buffer,
   // Get ssrc before taking statistics_crit_ to avoid possible deadlock.
   uint32_t ssrc = is_rtx ? RtxSsrc() : SSRC();
 
-  CriticalSectionScoped lock(statistics_crit_.get());
+  rtc::CritScope lock(&statistics_crit_);
   if (is_rtx) {
     counters = &rtx_rtp_stats_;
   } else {
@@ -1061,23 +1054,17 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
     }
     return 0;
   }
-  if (capture_time_ms > 0) {
-    UpdateDelayStatistics(capture_time_ms, now_ms);
-  }
-
-  // TODO(sprang): Potentially too much overhead in IsRegistered()?
-  bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
-                                 kRtpExtensionTransportSequenceNumber) &&
-                             transport_sequence_number_allocator_;
 
   PacketOptions options;
-  if (using_transport_seq) {
-    options.packet_id =
-        UpdateTransportSequenceNumber(buffer, length, rtp_header);
-    if (transport_feedback_observer_) {
-      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+  if (AllocateTransportSequenceNumber(&options.packet_id)) {
+    if (UpdateTransportSequenceNumber(options.packet_id, buffer, length,
+                                      rtp_header)) {
+      if (transport_feedback_observer_)
+        transport_feedback_observer_->AddPacket(options.packet_id, length);
     }
   }
+  UpdateDelayStatistics(capture_time_ms, now_ms);
+  UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
 
   bool sent = SendPacketToNetwork(buffer, length, options);
 
@@ -1098,7 +1085,7 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
-  if (!send_side_delay_observer_)
+  if (!send_side_delay_observer_ || capture_time_ms <= 0)
     return;
 
   uint32_t ssrc;
@@ -1109,7 +1096,7 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
     ssrc = ssrc_;
   }
   {
-    CriticalSectionScoped cs(statistics_crit_.get());
+    rtc::CritScope cs(&statistics_crit_);
     // TODO(holmer): Compute this iteratively instead.
     send_delays_[now_ms] = now_ms - capture_time_ms;
     send_delays_.erase(send_delays_.begin(),
@@ -1130,6 +1117,15 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
                                                   ssrc);
 }
 
+void RTPSender::UpdateOnSendPacket(int packet_id,
+                                   int64_t capture_time_ms,
+                                   uint32_t ssrc) {
+  if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1)
+    return;
+
+  send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
+}
+
 void RTPSender::ProcessBitrate() {
   rtc::CritScope lock(&send_critsect_);
   total_bitrate_sent_.Process();
@@ -1157,7 +1153,7 @@ uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
 
 void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
                                 StreamDataCounters* rtx_stats) const {
-  CriticalSectionScoped lock(statistics_crit_.get());
+  rtc::CritScope lock(&statistics_crit_);
   *rtp_stats = rtp_stats_;
   *rtx_stats = rtx_rtp_stats_;
 }
@@ -1613,7 +1609,8 @@ void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet,
                                           ConvertMsTo24Bits(now_ms));
 }
 
-uint16_t RTPSender::UpdateTransportSequenceNumber(
+bool RTPSender::UpdateTransportSequenceNumber(
+    uint16_t sequence_number,
     uint8_t* rtp_packet,
     size_t rtp_packet_length,
     const RTPHeader& rtp_header) const {
@@ -1624,19 +1621,26 @@ uint16_t RTPSender::UpdateTransportSequenceNumber(
                           rtp_packet_length, rtp_header,
                           kTransportSequenceNumberLength, &offset)) {
     case ExtensionStatus::kNotRegistered:
-      return 0;
+      return false;
     case ExtensionStatus::kError:
       LOG(LS_WARNING) << "Failed to update transport sequence number";
-      return 0;
+      return false;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
-  uint16_t seq = transport_sequence_number_allocator_->AllocateSequenceNumber();
-  BuildTransportSequenceNumberExtension(rtp_packet + offset, seq);
-  return seq;
+  BuildTransportSequenceNumberExtension(rtp_packet + offset, sequence_number);
+  return true;
+}
+
+bool RTPSender::AllocateTransportSequenceNumber(int* packet_id) const {
+  if (!transport_sequence_number_allocator_)
+    return false;
+
+  *packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
+  return true;
 }
 
 void RTPSender::SetSendingStatus(bool enabled) {
@@ -1858,12 +1862,12 @@ void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
 
 void RTPSender::RegisterRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
-  CriticalSectionScoped cs(statistics_crit_.get());
+  rtc::CritScope cs(&statistics_crit_);
   rtp_stats_callback_ = callback;
 }
 
 StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
-  CriticalSectionScoped cs(statistics_crit_.get());
+  rtc::CritScope cs(&statistics_crit_);
   return rtp_stats_callback_;
 }
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 4344df67451..f501d27a723 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -13,9 +13,11 @@
 
 #include <list>
 #include <map>
+#include <memory>
 #include <utility>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
@@ -95,7 +97,9 @@ class RTPSender : public RTPSenderInterface {
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer,
-            RtcEventLog* event_log);
+            RtcEventLog* event_log,
+            SendPacketObserver* send_packet_observer);
+
   virtual ~RTPSender();
 
   void ProcessBitrate();
@@ -351,6 +355,9 @@ class RTPSender : public RTPSenderInterface {
                            const PacketOptions& options);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
+  void UpdateOnSendPacket(int packet_id,
+                          int64_t capture_time_ms,
+                          uint32_t ssrc);
 
   // Find the byte position of the RTP extension as indicated by |type| in
   // |rtp_packet|. Return false if such extension doesn't exist.
@@ -368,12 +375,13 @@ class RTPSender : public RTPSenderInterface {
                               size_t rtp_packet_length,
                               const RTPHeader& rtp_header,
                               int64_t now_ms) const;
-  // Update the transport sequence number of the packet using a new sequence
-  // number allocated by SequenceNumberAllocator. Returns the assigned sequence
-  // number, or 0 if extension could not be updated.
-  uint16_t UpdateTransportSequenceNumber(uint8_t* rtp_packet,
-                                         size_t rtp_packet_length,
-                                         const RTPHeader& rtp_header) const;
+
+  bool UpdateTransportSequenceNumber(uint16_t sequence_number,
+                                     uint8_t* rtp_packet,
+                                     size_t rtp_packet_length,
+                                     const RTPHeader& rtp_header) const;
+
+  bool AllocateTransportSequenceNumber(int* packet_id) const;
 
   void UpdateRtpStats(const uint8_t* buffer,
                       size_t packet_length,
@@ -422,8 +430,8 @@ class RTPSender : public RTPSenderInterface {
   Bitrate total_bitrate_sent_;
 
   const bool audio_configured_;
-  const rtc::scoped_ptr<RTPSenderAudio> audio_;
-  const rtc::scoped_ptr<RTPSenderVideo> video_;
+  const std::unique_ptr<RTPSenderAudio> audio_;
+  const std::unique_ptr<RTPSenderVideo> video_;
 
   RtpPacketSender* const paced_sender_;
   TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
@@ -454,7 +462,7 @@ class RTPSender : public RTPSenderInterface {
   RTPPacketHistory packet_history_;
 
   // Statistics
-  rtc::scoped_ptr<CriticalSectionWrapper> statistics_crit_;
+  rtc::CriticalSection statistics_crit_;
   SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
   FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
@@ -463,6 +471,7 @@ class RTPSender : public RTPSenderInterface {
   FrameCountObserver* const frame_count_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
   RtcEventLog* const event_log_;
+  SendPacketObserver* const send_packet_observer_;
 
   // RTP variables
   bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
@@ -489,7 +498,7 @@ class RTPSender : public RTPSenderInterface {
   // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
   // that by the time the function returns there is no guarantee
   // that the target bitrate is still valid.
-  rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
+  rtc::CriticalSection target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 804294ac540..4236e1f37d4 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -12,10 +12,11 @@
 
 #include <string.h>
 
+#include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -24,7 +25,6 @@ static const int kDtmfFrequencyHz = 8000;
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
     : _clock(clock),
       _rtpSender(rtpSender),
-      _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
       _packetSizeSamples(160),
       _dtmfEventIsOn(false),
       _dtmfEventFirstPacketSent(false),
@@ -53,7 +53,7 @@ int RTPSenderAudio::AudioFrequency() const {
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
 int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packetSizeSamples) {
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
 
   _packetSizeSamples = packetSizeSamples;
   return 0;
@@ -67,7 +67,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
     const uint32_t rate,
     RtpUtility::Payload** payload) {
   if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     //  we can have multiple CNG payload types
     switch (frequency) {
       case 8000:
@@ -86,7 +86,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
         return -1;
     }
   } else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     // Don't add it to the list
     // we dont want to allow send with a DTMF payloadtype
     _dtmfPayloadType = payloadType;
@@ -104,7 +104,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
 }
 
 bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   // for audio true for first packet in a speech burst
   bool markerBit = false;
   if (_lastPayloadType != payload_type) {
@@ -162,7 +162,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
   int8_t dtmf_payload_type;
   uint16_t packet_size_samples;
   {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     red_payload_type = _REDPayloadType;
     audio_level_dbov = _audioLevel_dBov;
     dtmf_payload_type = _dtmfPayloadType;
@@ -333,8 +333,9 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
       memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
     }
   }
+
   {
-    CriticalSectionScoped cs(_sendAudioCritsect.get());
+    rtc::CritScope cs(&_sendAudioCritsect);
     _lastPayloadType = payloadType;
   }
   // Update audio level extension, if included.
@@ -348,10 +349,14 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
   TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
                          _rtpSender->Timestamp(), "seqnum",
                          _rtpSender->SequenceNumber());
-  return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
-                                   TickTime::MillisecondTimestamp(),
-                                   kAllowRetransmission,
-                                   RtpPacketSender::kHighPriority);
+  int32_t send_result = _rtpSender->SendToNetwork(
+      dataBuffer, payloadSize, rtpHeaderLength,
+      rtc::TimeMillis(), kAllowRetransmission,
+      RtpPacketSender::kHighPriority);
+  if (first_packet_sent_()) {
+    LOG(LS_INFO) << "First audio RTP packet sent to pacer";
+  }
+  return send_result;
 }
 
 // Audio level magnitude and voice activity flag are set for each RTP packet
@@ -359,7 +364,7 @@ int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) {
   if (level_dBov > 127) {
     return -1;
   }
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   _audioLevel_dBov = level_dBov;
   return 0;
 }
@@ -369,14 +374,14 @@ int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
   if (payloadType < -1) {
     return -1;
   }
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   _REDPayloadType = payloadType;
   return 0;
 }
 
 // Get payload type for Redundant Audio Data RFC 2198
 int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
-  CriticalSectionScoped cs(_sendAudioCritsect.get());
+  rtc::CritScope cs(&_sendAudioCritsect);
   if (_REDPayloadType == -1) {
     // not configured
     return -1;
@@ -390,7 +395,7 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
                                            uint16_t time_ms,
                                            uint8_t level) {
   {
-    CriticalSectionScoped lock(_sendAudioCritsect.get());
+    rtc::CritScope lock(&_sendAudioCritsect);
     if (_dtmfPayloadType < 0) {
       // TelephoneEvent payloadtype not configured
       return -1;
@@ -445,7 +450,7 @@ int32_t RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                          "Audio::SendTelephoneEvent", "timestamp",
                          dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
     retVal = _rtpSender->SendToNetwork(
-        dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
+        dtmfbuffer, 4, 12, rtc::TimeMillis(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;
   } while (sendCount > 0 && retVal == 0);
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 25c5e4dd88a..4bc0266b7d2 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -12,6 +12,8 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 
 #include "webrtc/common_types.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
@@ -72,7 +74,7 @@ class RTPSenderAudio : public DTMFqueue {
   Clock* const _clock;
   RTPSender* const _rtpSender;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+  rtc::CriticalSection _sendAudioCritsect;
 
   uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
 
@@ -100,6 +102,7 @@ class RTPSenderAudio : public DTMFqueue {
   // Audio level indication
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
   uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
+  OneTimeEvent first_packet_sent_;
 };
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index b7238d26a22..d04ff4d200a 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -8,17 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-/*
- * This file includes unit tests for the RTPSender.
- */
-
 #include <list>
+#include <memory>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -126,6 +122,11 @@ class MockTransportSequenceNumberAllocator
   MOCK_METHOD0(AllocateSequenceNumber, uint16_t());
 };
 
+class MockSendPacketObserver : public SendPacketObserver {
+ public:
+  MOCK_METHOD3(OnSendPacket, void(uint16_t, int64_t, uint32_t));
+};
+
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
@@ -141,10 +142,10 @@ class RtpSenderTest : public ::testing::Test {
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
-    rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
-                                    pacer ? &mock_paced_sender_ : nullptr,
-                                    &seq_num_allocator_, nullptr, nullptr,
-                                    nullptr, nullptr, &mock_rtc_event_log_));
+    rtp_sender_.reset(new RTPSender(
+        false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
+        &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
+        &mock_rtc_event_log_, &send_packet_observer_));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
@@ -152,7 +153,8 @@ class RtpSenderTest : public ::testing::Test {
   MockRtcEventLog mock_rtc_event_log_;
   MockRtpPacketSender mock_paced_sender_;
   MockTransportSequenceNumberAllocator seq_num_allocator_;
-  rtc::scoped_ptr<RTPSender> rtp_sender_;
+  MockSendPacketObserver send_packet_observer_;
+  std::unique_ptr<RTPSender> rtp_sender_;
   int payload_;
   LoopbackTransportTest transport_;
   const bool kMarkerBit;
@@ -189,6 +191,20 @@ class RtpSenderTest : public ::testing::Test {
                      packet_, payload_length, rtp_length, capture_time_ms,
                      kAllowRetransmission, RtpPacketSender::kNormalPriority));
   }
+
+  void SendGenericPayload() {
+    const uint8_t kPayload[] = {47, 11, 32, 93, 89};
+    const uint32_t kTimestamp = 1234;
+    const uint8_t kPayloadType = 127;
+    const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
+    char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
+    EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
+                                              0, 1500));
+
+    EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
+                     kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
+                     kPayload, sizeof(kPayload), nullptr));
+  }
 };
 
 // TODO(pbos): Move tests over from WithoutPacer to RtpSenderTest as this is our
@@ -206,7 +222,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
     rtp_sender_video_.reset(
         new RTPSenderVideo(&fake_clock_, rtp_sender_.get()));
   }
-  rtc::scoped_ptr<RTPSenderVideo> rtp_sender_video_;
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video_;
 
   void VerifyCVOPacket(uint8_t* data,
                        size_t len,
@@ -483,21 +499,13 @@ TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
 
-  char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
-  const uint8_t payload_type = 127;
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
-                                            0, 1500));
-  // Create a dummy payload of 5 bytes.
-  uint8_t payload[] = {47, 11, 32, 93, 89};
-
-  const uint16_t kTransportSequenceNumber = 17;
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
-  const uint32_t kTimestamp = 1234;
-  const int64_t kCaptureTimeMs = 4321;
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
-                   kVideoFrameKey, payload_type, kTimestamp, kCaptureTimeMs,
-                   payload, sizeof(payload), nullptr));
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber, _, _))
+      .Times(1);
+
+  SendGenericPayload();
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
@@ -513,6 +521,19 @@ TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
             rtp_header.extension.transportSequenceNumber);
 }
 
+TEST_F(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
+  EXPECT_CALL(mock_rtc_event_log_,  // Ignore rtc event calls.
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber, _, _))
+      .Times(1);
+
+  SendGenericPayload();
+}
+
 // Test CVO header extension is only set when marker bit is true.
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
   rtp_sender_->SetVideoRotation(kRotation);
@@ -853,7 +874,7 @@ TEST_F(RtpSenderTest, SendPadding) {
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
   // Create and set up parser.
-  rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
+  std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
@@ -951,11 +972,66 @@ TEST_F(RtpSenderTest, SendPadding) {
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
+TEST_F(RtpSenderTest, OnSendPacketUpdated) {
+  EXPECT_CALL(mock_rtc_event_log_,  // Ignore rtc event calls.
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber, _, _))
+      .Times(1);
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
+
+  SendGenericPayload();  // Packet passed to pacer.
+  const bool kIsRetransmit = false;
+  rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
+                                kIsRetransmit);
+  EXPECT_EQ(1, transport_.packets_sent_);
+}
+
+TEST_F(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) {
+  EXPECT_CALL(mock_rtc_event_log_,  // Ignore rtc event calls.
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+
+  EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
+
+  SendGenericPayload();  // Packet passed to pacer.
+  const bool kIsRetransmit = true;
+  rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
+                                kIsRetransmit);
+  EXPECT_EQ(1, transport_.packets_sent_);
+}
+
+TEST_F(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, &mock_paced_sender_,
+      nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
+      nullptr, nullptr, &send_packet_observer_));
+  rtp_sender_->SetSequenceNumber(kSeqNum);
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+
+  EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
+
+  SendGenericPayload();  // Packet passed to pacer.
+  const bool kIsRetransmit = false;
+  rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
+                                kIsRetransmit);
+  EXPECT_EQ(1, transport_.packets_sent_);
+}
+
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
-      nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_));
+      false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
+      nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr));
+
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
 
@@ -972,7 +1048,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
   rtp_sender_->SetRtxSsrc(1234);
 
   // Create and set up parser.
-  rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
+  std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
@@ -1096,9 +1172,9 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
     FrameCounts frame_counts_;
   } callback;
 
-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
-                                  &mock_paced_sender_, nullptr, nullptr,
-                                  nullptr, &callback, nullptr, nullptr));
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,
+      nullptr, &callback, nullptr, nullptr, nullptr));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
@@ -1152,8 +1228,8 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
     BitrateStatistics total_stats_;
     BitrateStatistics retransmit_stats_;
   } callback;
-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
-                                  nullptr, nullptr, nullptr, &callback, nullptr,
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
+                                  nullptr, nullptr, &callback, nullptr, nullptr,
                                   nullptr, nullptr));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals.
@@ -1205,7 +1281,7 @@ class RtpSenderAudioTest : public RtpSenderTest {
 
   void SetUp() override {
     payload_ = kAudioPayload;
-    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_,
+    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
                                     nullptr, nullptr, nullptr, nullptr, nullptr,
                                     nullptr, nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
@@ -1297,7 +1373,6 @@ TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   fec_params.fec_mask_type = kFecMaskRandom;
   fec_params.fec_rate = 1;
   fec_params.max_fec_frames = 1;
-  fec_params.use_uep_protection = false;
   rtp_sender_->SetFecParameters(&fec_params, &fec_params);
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
                                              1234, 4321, payload,
@@ -1407,7 +1482,7 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms + 2000, 0, nullptr,
                                              0, nullptr));
-  rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
+  std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   webrtc::RTPHeader rtp_header;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 32ba26f54b8..e10b5b2edab 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -13,6 +13,7 @@
 #include <stdlib.h>
 #include <string.h>
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/checks.h"
@@ -24,21 +25,19 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 enum { REDForFECHeaderLength = 1 };
 
 RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
     : _rtpSender(*rtpSender),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
       _videoType(kRtpVideoGeneric),
       _retransmissionSettings(kRetransmitBaseLayer),
       // Generic FEC
       fec_(),
       fec_enabled_(false),
-      red_payload_type_(-1),
-      fec_payload_type_(-1),
+      red_payload_type_(0),
+      fec_payload_type_(0),
       delta_fec_params_(),
       key_fec_params_(),
       producer_fec_(&fec_),
@@ -113,13 +112,13 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
                                           int64_t capture_time_ms,
                                           StorageType media_packet_storage,
                                           bool protect) {
-  rtc::scoped_ptr<RedPacket> red_packet;
+  std::unique_ptr<RedPacket> red_packet;
   std::vector<RedPacket*> fec_packets;
   StorageType fec_storage = kDontRetransmit;
   uint16_t next_fec_sequence_number = 0;
   {
     // Only protect while creating RED and FEC packets, not when sending.
-    CriticalSectionScoped cs(crit_.get());
+    rtc::CritScope cs(&crit_);
     red_packet.reset(producer_fec_.BuildRedPacket(
         data_buffer, payload_length, rtp_header_length, red_payload_type_));
     if (protect) {
@@ -170,7 +169,7 @@ void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
 void RTPSenderVideo::SetGenericFECStatus(const bool enable,
                                          const uint8_t payloadTypeRED,
                                          const uint8_t payloadTypeFEC) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   fec_enabled_ = enable;
   red_payload_type_ = payloadTypeRED;
   fec_payload_type_ = payloadTypeFEC;
@@ -184,33 +183,38 @@ void RTPSenderVideo::SetGenericFECStatus(const bool enable,
 void RTPSenderVideo::GenericFECStatus(bool* enable,
                                       uint8_t* payloadTypeRED,
                                       uint8_t* payloadTypeFEC) const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   *enable = fec_enabled_;
   *payloadTypeRED = red_payload_type_;
   *payloadTypeFEC = fec_payload_type_;
 }
 
 size_t RTPSenderVideo::FECPacketOverhead() const {
-  CriticalSectionScoped cs(crit_.get());
-  if (fec_enabled_) {
+  rtc::CritScope cs(&crit_);
+  size_t overhead = 0;
+  if (red_payload_type_ != 0) {
     // Overhead is FEC headers plus RED for FEC header plus anything in RTP
     // header beyond the 12 bytes base header (CSRC list, extensions...)
     // This reason for the header extensions to be included here is that
     // from an FEC viewpoint, they are part of the payload to be protected.
     // (The base RTP header is already protected by the FEC header.)
-    return ForwardErrorCorrection::PacketOverhead() + REDForFECHeaderLength +
-           (_rtpSender.RTPHeaderLength() - kRtpHeaderSize);
+    overhead = REDForFECHeaderLength + (_rtpSender.RTPHeaderLength() -
+      kRtpHeaderSize);
   }
-  return 0;
+  if (fec_enabled_)
+    overhead += ForwardErrorCorrection::PacketOverhead();
+  return overhead;
 }
 
 void RTPSenderVideo::SetFecParameters(const FecProtectionParams* delta_params,
                                       const FecProtectionParams* key_params) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   RTC_DCHECK(delta_params);
   RTC_DCHECK(key_params);
-  delta_fec_params_ = *delta_params;
-  key_fec_params_ = *key_params;
+  if (fec_enabled_) {
+    delta_fec_params_ = *delta_params;
+    key_fec_params_ = *key_params;
+  }
 }
 
 int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
@@ -226,19 +230,20 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
     return -1;
   }
 
-  rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
+  std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
       videoType, _rtpSender.MaxDataPayloadLength(),
       video_header ? &(video_header->codecHeader) : nullptr, frameType));
 
   StorageType storage;
-  bool fec_enabled;
+  int red_payload_type;
+  bool first_frame = first_frame_sent_();
   {
-    CriticalSectionScoped cs(crit_.get());
+    rtc::CritScope cs(&crit_);
     FecProtectionParams* fec_params =
         frameType == kVideoFrameKey ? &key_fec_params_ : &delta_fec_params_;
     producer_fec_.SetFecParameters(fec_params, 0);
     storage = packetizer->GetStorageType(_retransmissionSettings);
-    fec_enabled = fec_enabled_;
+    red_payload_type = red_payload_type_;
   }
 
   // Register CVO rtp header extension at the first time when we receive a frame
@@ -260,6 +265,7 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
 
   packetizer->SetPayloadData(data, payload_bytes_to_send, frag);
 
+  bool first = true;
   bool last = false;
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
@@ -268,6 +274,7 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
                                 &payload_bytes_in_packet, &last)) {
       return -1;
     }
+
     // Write RTP header.
     // Set marker bit true if this is the last packet in frame.
     _rtpSender.BuildRTPheader(
@@ -299,7 +306,7 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
       _rtpSender.UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
                                      video_header->rotation);
     }
-    if (fec_enabled) {
+    if (red_payload_type != 0) {
       SendVideoPacketAsRed(dataBuffer, payload_bytes_in_packet,
                            rtp_header_length, _rtpSender.SequenceNumber(),
                            captureTimeStamp, capture_time_ms, storage,
@@ -309,6 +316,18 @@ int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
                       _rtpSender.SequenceNumber(), captureTimeStamp,
                       capture_time_ms, storage);
     }
+
+    if (first_frame) {
+      if (first) {
+        LOG(LS_INFO)
+            << "Sent first RTP packet of the first video frame (pre-pacer)";
+      }
+      if (last) {
+        LOG(LS_INFO)
+            << "Sent last RTP packet of the first video frame (pre-pacer)";
+      }
+    }
+    first = false;
   }
 
   TRACE_EVENT_ASYNC_END1(
@@ -330,12 +349,12 @@ uint32_t RTPSenderVideo::FecOverheadRate() const {
 }
 
 int RTPSenderVideo::SelectiveRetransmissions() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   return _retransmissionSettings;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   _retransmissionSettings = settings;
 }
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index dc1088a3f7c..8307b83864d 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -13,7 +13,8 @@
 
 #include <list>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -27,7 +28,6 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-class CriticalSectionWrapper;
 
 class RTPSenderVideo {
  public:
@@ -97,7 +97,7 @@ class RTPSenderVideo {
   RTPSenderInterface& _rtpSender;
 
   // Should never be held when calling out of this class.
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  const rtc::CriticalSection crit_;
 
   RtpVideoCodecTypes _videoType;
   int32_t _retransmissionSettings GUARDED_BY(crit_);
@@ -116,6 +116,7 @@ class RTPSenderVideo {
   Bitrate _fecOverheadRate;
   // Bitrate used for video payload and RTP headers
   Bitrate _videoBitrate;
+  OneTimeEvent first_frame_sent_;
 };
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
index bdae3c4806f..439cd01a9aa 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -319,6 +319,13 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
       return;
     }
 
+    if (ptrRTPDataExtensionEnd - ptr < (len + 1)) {
+      LOG(LS_WARNING) << "Incorrect one-byte extension len: " << (len + 1)
+                      << ", bytes left in buffer: "
+                      << (ptrRTPDataExtensionEnd - ptr);
+      return;
+    }
+
     RTPExtensionType type;
     if (ptrExtensionMap->GetType(id, &type) != 0) {
       // If we encounter an unknown extension, just skip over it.
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index 1a242853952..474bc6e04e0 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -52,11 +52,6 @@ class RtpHeaderParser {
   bool ParseRtcp(RTPHeader* header) const;
   bool Parse(RTPHeader* parsedPacket,
              RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
-  RTC_DEPRECATED bool Parse(
-      RTPHeader& parsedPacket,  // NOLINT(runtime/references)
-      RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const {
-    return Parse(&parsedPacket, ptrExtensionMap);
-  }
 
  private:
   void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
index f1d1549e279..a96d05db468 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc
@@ -9,9 +9,8 @@
  */
 
 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h"
-
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -45,7 +44,7 @@ void SSRCDatabase::ReturnSSRC(uint32_t ssrc) {
   ssrcs_.erase(ssrc);
 }
 
-SSRCDatabase::SSRCDatabase() : random_(TickTime::Now().Ticks()) {}
+SSRCDatabase::SSRCDatabase() : random_(rtc::TimeMicros()) {}
 
 SSRCDatabase::~SSRCDatabase() {}
 
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
index da43204b093..43d3a82ab28 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -67,10 +67,8 @@ void TMMBRSet::ClearEntry(uint32_t idx) {
 }
 
 TMMBRHelp::TMMBRHelp()
-    : _criticalSection(CriticalSectionWrapper::CreateCriticalSection()),
-      _candidateSet(),
+    : _candidateSet(),
       _boundingSet(),
-      _boundingSetToSend(),
       _ptrIntersectionBoundingSet(NULL),
       _ptrMaxPRBoundingSet(NULL) {
 }
@@ -80,13 +78,12 @@ TMMBRHelp::~TMMBRHelp() {
   delete [] _ptrMaxPRBoundingSet;
   _ptrIntersectionBoundingSet = 0;
   _ptrMaxPRBoundingSet = 0;
-  delete _criticalSection;
 }
 
 TMMBRSet*
 TMMBRHelp::VerifyAndAllocateBoundingSet(uint32_t minimumSize)
 {
-    CriticalSectionScoped lock(_criticalSection);
+    rtc::CritScope lock(&_criticalSection);
 
     if(minimumSize > _boundingSet.capacity())
     {
@@ -107,43 +104,10 @@ TMMBRSet* TMMBRHelp::BoundingSet() {
   return &_boundingSet;
 }
 
-int32_t
-TMMBRHelp::SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend)
-{
-    CriticalSectionScoped lock(_criticalSection);
-
-    if (boundingSetToSend == NULL)
-    {
-        _boundingSetToSend.clearSet();
-        return 0;
-    }
-
-    VerifyAndAllocateBoundingSetToSend(boundingSetToSend->lengthOfSet());
-    _boundingSetToSend.clearSet();
-    for (uint32_t i = 0; i < boundingSetToSend->lengthOfSet(); i++)
-    {
-        // cap at our configured max bitrate
-        uint32_t bitrate = boundingSetToSend->Tmmbr(i);
-        _boundingSetToSend.SetEntry(i, bitrate,
-                                    boundingSetToSend->PacketOH(i),
-                                    boundingSetToSend->Ssrc(i));
-    }
-    return 0;
-}
-
-int32_t
-TMMBRHelp::VerifyAndAllocateBoundingSetToSend(uint32_t minimumSize)
-{
-    CriticalSectionScoped lock(_criticalSection);
-
-    _boundingSetToSend.VerifyAndAllocateSet(minimumSize);
-    return 0;
-}
-
 TMMBRSet*
 TMMBRHelp::VerifyAndAllocateCandidateSet(uint32_t minimumSize)
 {
-    CriticalSectionScoped lock(_criticalSection);
+    rtc::CritScope lock(&_criticalSection);
 
     _candidateSet.VerifyAndAllocateSet(minimumSize);
     return &_candidateSet;
@@ -155,16 +119,10 @@ TMMBRHelp::CandidateSet()
     return &_candidateSet;
 }
 
-TMMBRSet*
-TMMBRHelp::BoundingSetToSend()
-{
-    return &_boundingSetToSend;
-}
-
 int32_t
 TMMBRHelp::FindTMMBRBoundingSet(TMMBRSet*& boundingSet)
 {
-    CriticalSectionScoped lock(_criticalSection);
+    rtc::CritScope lock(&_criticalSection);
 
     // Work on local variable, will be modified
     TMMBRSet    candidateSet;
@@ -207,7 +165,7 @@ TMMBRHelp::FindTMMBRBoundingSet(TMMBRSet*& boundingSet)
 int32_t
 TMMBRHelp::FindTMMBRBoundingSet(int32_t numCandidates, TMMBRSet& candidateSet)
 {
-    CriticalSectionScoped lock(_criticalSection);
+    rtc::CritScope lock(&_criticalSection);
 
     uint32_t numBoundingSet = 0;
     VerifyAndAllocateBoundingSet(candidateSet.capacity());
@@ -412,7 +370,7 @@ TMMBRHelp::FindTMMBRBoundingSet(int32_t numCandidates, TMMBRSet& candidateSet)
 
 bool TMMBRHelp::IsOwner(const uint32_t ssrc,
                         const uint32_t length) const {
-  CriticalSectionScoped lock(_criticalSection);
+  rtc::CritScope lock(&_criticalSection);
 
   if (length == 0) {
     // Empty bounding set.
@@ -428,7 +386,7 @@ bool TMMBRHelp::IsOwner(const uint32_t ssrc,
 }
 
 bool TMMBRHelp::CalcMinBitRate( uint32_t* minBitrateKbit) const {
-  CriticalSectionScoped lock(_criticalSection);
+  rtc::CritScope lock(&_criticalSection);
 
   if (_candidateSet.size() == 0) {
     // Empty bounding set.
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.h b/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
index 6236d5d43bd..ffafb1409b8 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
@@ -12,8 +12,8 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
 
 #include <vector>
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -63,11 +63,9 @@ public:
 
     TMMBRSet* BoundingSet(); // used for debuging
     TMMBRSet* CandidateSet();
-    TMMBRSet* BoundingSetToSend();
 
     TMMBRSet* VerifyAndAllocateCandidateSet(const uint32_t minimumSize);
     int32_t FindTMMBRBoundingSet(TMMBRSet*& boundingSet);
-    int32_t SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend);
 
     bool IsOwner(const uint32_t ssrc, const uint32_t length) const;
 
@@ -75,15 +73,12 @@ public:
 
 protected:
     TMMBRSet*   VerifyAndAllocateBoundingSet(uint32_t minimumSize);
-    int32_t VerifyAndAllocateBoundingSetToSend(uint32_t minimumSize);
-
     int32_t FindTMMBRBoundingSet(int32_t numCandidates, TMMBRSet& candidateSet);
 
 private:
-    CriticalSectionWrapper* _criticalSection;
+    rtc::CriticalSection    _criticalSection;
     TMMBRSet                _candidateSet;
     TMMBRSet                _boundingSet;
-    TMMBRSet                _boundingSetToSend;
 
     float*                  _ptrIntersectionBoundingSet;
     float*                  _ptrMaxPRBoundingSet;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 67e8a65c4df..89c9cbebeba 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 
 #include "webrtc/test/null_transport.h"
@@ -41,7 +42,7 @@ bool LoopBackTransport::SendRtp(const uint8_t* data,
     }
   }
   RTPHeader header;
-  rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+  std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
   if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
     return false;
   }
@@ -100,9 +101,9 @@ class RtpRtcpAPITest : public ::testing::Test {
         &fake_clock_, NULL, NULL, rtp_payload_registry_.get()));
   }
 
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
-  rtc::scoped_ptr<RtpRtcp> module_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
+  std::unique_ptr<RtpReceiver> rtp_receiver_;
+  std::unique_ptr<RtpRtcp> module_;
   uint32_t test_ssrc_;
   uint32_t test_timestamp_;
   uint16_t test_sequence_number_;
@@ -151,10 +152,6 @@ TEST_F(RtpRtcpAPITest, RTCP) {
   EXPECT_TRUE(module_->TMMBR());
   module_->SetTMMBRStatus(false);
   EXPECT_FALSE(module_->TMMBR());
-
-  EXPECT_EQ(kNackOff, rtp_receiver_->NACK());
-  rtp_receiver_->SetNACKStatus(kNackRtcp);
-  EXPECT_EQ(kNackRtcp, rtp_receiver_->NACK());
 }
 
 TEST_F(RtpRtcpAPITest, RtxSender) {
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index d8040f79027..44de00a55f9 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -11,7 +11,6 @@
 #define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 9b44c4f40db..8069b0950b6 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -9,6 +9,7 @@
  */
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -135,12 +136,12 @@ class RtpRtcpAudioTest : public ::testing::Test {
 
   RtpRtcp* module1;
   RtpRtcp* module2;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics2_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver1_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver2_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics1_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics2_;
+  std::unique_ptr<RtpReceiver> rtp_receiver1_;
+  std::unique_ptr<RtpReceiver> rtp_receiver2_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
   VerifyingAudioReceiver* data_receiver1;
   VerifyingAudioReceiver* data_receiver2;
   LoopBackTransport* transport1;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index d4b36412736..c1359df8646 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -9,6 +9,7 @@
  */
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
@@ -175,14 +176,14 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     delete receiver;
   }
 
-  rtc::scoped_ptr<TestRtpFeedback> rtp_feedback1_;
-  rtc::scoped_ptr<TestRtpFeedback> rtp_feedback2_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics2_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
-  rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver1_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver2_;
+  std::unique_ptr<TestRtpFeedback> rtp_feedback1_;
+  std::unique_ptr<TestRtpFeedback> rtp_feedback2_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics1_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics2_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
+  std::unique_ptr<RtpReceiver> rtp_receiver1_;
+  std::unique_ptr<RtpReceiver> rtp_receiver2_;
   RtpRtcp* module1;
   RtpRtcp* module2;
   TestRtpReceiver* receiver;
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index 16ea540bd58..d84ff37be7a 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -11,6 +11,7 @@
 #include <stdlib.h>
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
@@ -55,7 +56,6 @@ class RtpRtcpVideoTest : public ::testing::Test {
 
     video_module_->SetRTCPStatus(RtcpMode::kCompound);
     video_module_->SetSSRC(test_ssrc_);
-    rtp_receiver_->SetNACKStatus(kNackRtcp);
     video_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, video_module_->SetSendingStatus(true));
 
@@ -127,9 +127,9 @@ class RtpRtcpVideoTest : public ::testing::Test {
   }
 
   int test_id_;
-  rtc::scoped_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RTPPayloadRegistry rtp_payload_registry_;
-  rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
+  std::unique_ptr<RtpReceiver> rtp_receiver_;
   RtpRtcp* video_module_;
   LoopBackTransport* transport_;
   TestRtpReceiver* receiver_;
@@ -170,7 +170,7 @@ TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
                                          kPadSize);
       ++seq_num;
       RTPHeader header;
-      rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
+      std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
       EXPECT_TRUE(parser->Parse(padding_packet, packet_size, &header));
       PayloadUnion payload_specific;
       EXPECT_TRUE(rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
diff --git a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
index 466214c740e..b7c4ef5506b 100644
--- a/chromium/third_party/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
+++ b/chromium/third_party/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -45,8 +45,9 @@
 
 #include <math.h>
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h"
 #include "webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -191,7 +192,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
   int RecoveredMediaPackets(int num_media_packets,
                             int num_fec_packets,
                             uint8_t* state) {
-    rtc::scoped_ptr<uint8_t[]> state_tmp(
+    std::unique_ptr<uint8_t[]> state_tmp(
         new uint8_t[num_media_packets + num_fec_packets]);
     memcpy(state_tmp.get(), state, num_media_packets + num_fec_packets);
     int num_recovered_packets = 0;
@@ -385,7 +386,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
   // (which containes the code size parameters/protection length).
   void ComputeMetricsForCode(CodeType code_type,
                              int code_index) {
-    rtc::scoped_ptr<double[]> prob_weight(new double[kNumLossModels]);
+    std::unique_ptr<double[]> prob_weight(new double[kNumLossModels]);
     memset(prob_weight.get() , 0, sizeof(double) * kNumLossModels);
     MetricsFecCode metrics_code;
     SetMetricsZero(&metrics_code);
@@ -393,7 +394,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
     int num_media_packets = code_params_[code_index].num_media_packets;
     int num_fec_packets = code_params_[code_index].num_fec_packets;
     int tot_num_packets = num_media_packets + num_fec_packets;
-    rtc::scoped_ptr<uint8_t[]> state(new uint8_t[tot_num_packets]);
+    std::unique_ptr<uint8_t[]> state(new uint8_t[tot_num_packets]);
     memset(state.get() , 0, tot_num_packets);
 
     int num_loss_configurations = static_cast<int>(pow(2.0f, tot_num_packets));
diff --git a/chromium/third_party/webrtc/modules/utility/OWNERS b/chromium/third_party/webrtc/modules/utility/OWNERS
index 65cb70c9b90..9456ae08c20 100644
--- a/chromium/third_party/webrtc/modules/utility/OWNERS
+++ b/chromium/third_party/webrtc/modules/utility/OWNERS
@@ -1,4 +1,3 @@
-asapersson@webrtc.org
 perkj@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
diff --git a/chromium/third_party/webrtc/modules/utility/include/file_recorder.h b/chromium/third_party/webrtc/modules/utility/include/file_recorder.h
index 09ed8ae3507..480a4a97996 100644
--- a/chromium/third_party/webrtc/modules/utility/include/file_recorder.h
+++ b/chromium/third_party/webrtc/modules/utility/include/file_recorder.h
@@ -15,7 +15,6 @@
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_frame.h"
 
@@ -58,8 +57,7 @@ public:
 
     // Write frame to file. Frame should contain 10ms of un-ecoded audio data.
     virtual int32_t RecordAudioToFile(
-        const AudioFrame& frame,
-        const TickTime* playoutTS = NULL) = 0;
+        const AudioFrame& frame) = 0;
 
     // Open/create the file specified by fileName for writing audio/video data
     // (relative path is allowed). audioCodecInst specifies the encoding of the
diff --git a/chromium/third_party/webrtc/modules/utility/include/jvm_android.h b/chromium/third_party/webrtc/modules/utility/include/jvm_android.h
index f527dff6322..574c977cd04 100644
--- a/chromium/third_party/webrtc/modules/utility/include/jvm_android.h
+++ b/chromium/third_party/webrtc/modules/utility/include/jvm_android.h
@@ -12,9 +12,10 @@
 #define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
 
 #include <jni.h>
+
+#include <memory>
 #include <string>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/utility/include/helpers_android.h"
 
@@ -76,7 +77,7 @@ class NativeRegistration : public JavaClass {
   NativeRegistration(JNIEnv* jni, jclass clazz);
   ~NativeRegistration();
 
-  rtc::scoped_ptr<GlobalRef> NewObject(
+  std::unique_ptr<GlobalRef> NewObject(
       const char* name, const char* signature, ...);
 
  private:
@@ -96,7 +97,7 @@ class JNIEnvironment {
   // Note that the class name must be one of the names in the static
   // |loaded_classes| array defined in jvm_android.cc.
   // This method must be called on the construction thread.
-  rtc::scoped_ptr<NativeRegistration> RegisterNatives(
+  std::unique_ptr<NativeRegistration> RegisterNatives(
       const char* name, const JNINativeMethod *methods, int num_methods);
 
   // Converts from Java string to std::string.
@@ -120,9 +121,9 @@ class JNIEnvironment {
 //   webrtc::JVM::Initialize(jvm, context);
 //
 //   // Header (.h) file of example class called User.
-//   rtc::scoped_ptr<JNIEnvironment> env;
-//   rtc::scoped_ptr<NativeRegistration> reg;
-//   rtc::scoped_ptr<GlobalRef> obj;
+//   std::unique_ptr<JNIEnvironment> env;
+//   std::unique_ptr<NativeRegistration> reg;
+//   std::unique_ptr<GlobalRef> obj;
 //
 //   // Construction (in .cc file) of User class.
 //   User::User() {
@@ -156,7 +157,7 @@ class JVM {
   // Creates a JNIEnvironment object.
   // This method returns a NULL pointer if AttachCurrentThread() has not been
   // called successfully. Use the AttachCurrentThreadIfNeeded class if needed.
-  rtc::scoped_ptr<JNIEnvironment> environment();
+  std::unique_ptr<JNIEnvironment> environment();
 
   // Returns a JavaClass object given class |name|.
   // Note that the class name must be one of the names in the static
diff --git a/chromium/third_party/webrtc/modules/utility/include/mock/mock_process_thread.h b/chromium/third_party/webrtc/modules/utility/include/mock/mock_process_thread.h
index 9560e408e87..621fcee8182 100644
--- a/chromium/third_party/webrtc/modules/utility/include/mock/mock_process_thread.h
+++ b/chromium/third_party/webrtc/modules/utility/include/mock/mock_process_thread.h
@@ -21,6 +21,10 @@ namespace webrtc {
 
 class MockProcessThread : public ProcessThread {
  public:
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
   MOCK_METHOD0(Start, void());
   MOCK_METHOD0(Stop, void());
   MOCK_METHOD1(WakeUp, void(Module* module));
@@ -31,7 +35,7 @@ class MockProcessThread : public ProcessThread {
   // MOCK_METHOD1 gets confused with mocking this method, so we work around it
   // by overriding the method from the interface and forwarding the call to a
   // mocked, simpler method.
-  void PostTask(rtc::scoped_ptr<ProcessTask> task) override {
+  void PostTask(std::unique_ptr<ProcessTask> task) /* override */ {
     PostTask(task.get());
   }
 };
diff --git a/chromium/third_party/webrtc/modules/utility/include/process_thread.h b/chromium/third_party/webrtc/modules/utility/include/process_thread.h
index 285a5ea5876..f6913ea3167 100644
--- a/chromium/third_party/webrtc/modules/utility/include/process_thread.h
+++ b/chromium/third_party/webrtc/modules/utility/include/process_thread.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
 #define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
 
+#include <memory>
+
 #include "webrtc/typedefs.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 class Module;
@@ -29,7 +30,7 @@ class ProcessThread {
  public:
   virtual ~ProcessThread();
 
-  static rtc::scoped_ptr<ProcessThread> Create(const char* thread_name);
+  static std::unique_ptr<ProcessThread> Create(const char* thread_name);
 
   // Starts the worker thread.  Must be called from the construction thread.
   virtual void Start() = 0;
@@ -50,7 +51,7 @@ class ProcessThread {
   // construction thread of the ProcessThread instance, if the task did not
   // get a chance to run (e.g. posting the task while shutting down or when
   // the thread never runs).
-  virtual void PostTask(rtc::scoped_ptr<ProcessTask> task) = 0;
+  virtual void PostTask(std::unique_ptr<ProcessTask> task) = 0;
 
   // Adds a module that will start to receive callbacks on the worker thread.
   // Can be called from any thread.
diff --git a/chromium/third_party/webrtc/modules/utility/source/coder.cc b/chromium/third_party/webrtc/modules/utility/source/coder.cc
index 1476e02d9c5..3c065e7c2be 100644
--- a/chromium/third_party/webrtc/modules/utility/source/coder.cc
+++ b/chromium/third_party/webrtc/modules/utility/source/coder.cc
@@ -13,96 +13,101 @@
 #include "webrtc/modules/utility/source/coder.h"
 
 namespace webrtc {
-AudioCoder::AudioCoder(uint32_t instanceID)
-    : _acm(AudioCodingModule::Create(instanceID)),
-      _receiveCodec(),
-      _encodeTimestamp(0),
-      _encodedData(NULL),
-      _encodedLengthInBytes(0),
-      _decodeTimestamp(0)
-{
-    _acm->InitializeReceiver();
-    _acm->RegisterTransportCallback(this);
+namespace {
+AudioCodingModule::Config GetAcmConfig(uint32_t id) {
+  AudioCodingModule::Config config;
+  // This class does not handle muted output.
+  config.neteq_config.enable_muted_state = false;
+  config.id = id;
+  return config;
 }
+}  // namespace
 
-AudioCoder::~AudioCoder()
-{
+AudioCoder::AudioCoder(uint32_t instance_id)
+    : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
+      receive_codec_(),
+      encode_timestamp_(0),
+      encoded_data_(nullptr),
+      encoded_length_in_bytes_(0),
+      decode_timestamp_(0) {
+  acm_->InitializeReceiver();
+  acm_->RegisterTransportCallback(this);
 }
 
-int32_t AudioCoder::SetEncodeCodec(const CodecInst& codecInst) {
-  const bool success = codec_manager_.RegisterEncoder(codecInst) &&
-                       codec_manager_.MakeEncoder(&rent_a_codec_, _acm.get());
+AudioCoder::~AudioCoder() {}
+
+int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
+  const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
+                       codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
   return success ? 0 : -1;
 }
 
-int32_t AudioCoder::SetDecodeCodec(const CodecInst& codecInst) {
-  if (_acm->RegisterReceiveCodec(
-          codecInst, [&] { return rent_a_codec_.RentIsacDecoder(); }) == -1) {
+int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
+  if (acm_->RegisterReceiveCodec(
+          codec_inst, [&] { return rent_a_codec_.RentIsacDecoder(); }) == -1) {
     return -1;
   }
-  memcpy(&_receiveCodec, &codecInst, sizeof(CodecInst));
+  memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
   return 0;
 }
 
-int32_t AudioCoder::Decode(AudioFrame& decodedAudio,
-                           uint32_t sampFreqHz,
-                           const int8_t*  incomingPayload,
-                           size_t  payloadLength)
-{
-    if (payloadLength > 0)
-    {
-        const uint8_t payloadType = _receiveCodec.pltype;
-        _decodeTimestamp += _receiveCodec.pacsize;
-        if(_acm->IncomingPayload((const uint8_t*) incomingPayload,
-                                 payloadLength,
-                                 payloadType,
-                                 _decodeTimestamp) == -1)
-        {
-            return -1;
-        }
+int32_t AudioCoder::Decode(AudioFrame& decoded_audio,
+                           uint32_t samp_freq_hz,
+                           const int8_t* incoming_payload,
+                           size_t payload_length) {
+  if (payload_length > 0) {
+    const uint8_t payload_type = receive_codec_.pltype;
+    decode_timestamp_ += receive_codec_.pacsize;
+    if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length,
+                              payload_type, decode_timestamp_) == -1) {
+      return -1;
     }
-    return _acm->PlayoutData10Ms((uint16_t)sampFreqHz, &decodedAudio);
+  }
+  bool muted;
+  int32_t ret =
+      acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, &decoded_audio, &muted);
+  RTC_DCHECK(!muted);
+  return ret;
 }
 
-int32_t AudioCoder::PlayoutData(AudioFrame& decodedAudio,
-                                uint16_t& sampFreqHz)
-{
-    return _acm->PlayoutData10Ms(sampFreqHz, &decodedAudio);
+int32_t AudioCoder::PlayoutData(AudioFrame& decoded_audio,
+                                uint16_t& samp_freq_hz) {
+  bool muted;
+  int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, &decoded_audio, &muted);
+  RTC_DCHECK(!muted);
+  return ret;
 }
 
 int32_t AudioCoder::Encode(const AudioFrame& audio,
-                           int8_t* encodedData,
-                           size_t& encodedLengthInBytes)
-{
-    // Fake a timestamp in case audio doesn't contain a correct timestamp.
-    // Make a local copy of the audio frame since audio is const
-    AudioFrame audioFrame;
-    audioFrame.CopyFrom(audio);
-    audioFrame.timestamp_ = _encodeTimestamp;
-    _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_);
+                           int8_t* encoded_data,
+                           size_t& encoded_length_in_bytes) {
+  // Fake a timestamp in case audio doesn't contain a correct timestamp.
+  // Make a local copy of the audio frame since audio is const
+  AudioFrame audio_frame;
+  audio_frame.CopyFrom(audio);
+  audio_frame.timestamp_ = encode_timestamp_;
+  encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
 
-    // For any codec with a frame size that is longer than 10 ms the encoded
-    // length in bytes should be zero until a a full frame has been encoded.
-    _encodedLengthInBytes = 0;
-    if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1)
-    {
-        return -1;
-    }
-    _encodedData = encodedData;
-    encodedLengthInBytes = _encodedLengthInBytes;
-    return 0;
+  // For any codec with a frame size that is longer than 10 ms the encoded
+  // length in bytes should be zero until a a full frame has been encoded.
+  encoded_length_in_bytes_ = 0;
+  if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
+    return -1;
+  }
+  encoded_data_ = encoded_data;
+  encoded_length_in_bytes = encoded_length_in_bytes_;
+  return 0;
 }
 
-int32_t AudioCoder::SendData(
-    FrameType /* frameType */,
-    uint8_t   /* payloadType */,
-    uint32_t  /* timeStamp */,
-    const uint8_t*  payloadData,
-    size_t  payloadSize,
-    const RTPFragmentationHeader* /* fragmentation*/)
-{
-    memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
-    _encodedLengthInBytes = payloadSize;
-    return 0;
+int32_t AudioCoder::SendData(FrameType /* frame_type */,
+                             uint8_t /* payload_type */,
+                             uint32_t /* time_stamp */,
+                             const uint8_t* payload_data,
+                             size_t payload_size,
+                             const RTPFragmentationHeader* /* fragmentation*/) {
+  memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
+  encoded_length_in_bytes_ = payload_size;
+  return 0;
 }
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/utility/source/coder.h b/chromium/third_party/webrtc/modules/utility/source/coder.h
index 9536a027d0e..5f441904bee 100644
--- a/chromium/third_party/webrtc/modules/utility/source/coder.h
+++ b/chromium/third_party/webrtc/modules/utility/source/coder.h
@@ -22,45 +22,47 @@
 namespace webrtc {
 class AudioFrame;
 
-class AudioCoder : public AudioPacketizationCallback
-{
-public:
-    AudioCoder(uint32_t instanceID);
-    ~AudioCoder();
+class AudioCoder : public AudioPacketizationCallback {
+ public:
+  AudioCoder(uint32_t instance_id);
+  ~AudioCoder();
 
-    int32_t SetEncodeCodec(const CodecInst& codecInst);
+  int32_t SetEncodeCodec(const CodecInst& codec_inst);
 
-    int32_t SetDecodeCodec(const CodecInst& codecInst);
+  int32_t SetDecodeCodec(const CodecInst& codec_inst);
 
-    int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
-                   const int8_t* incomingPayload, size_t payloadLength);
+  int32_t Decode(AudioFrame& decoded_audio,
+                 uint32_t samp_freq_hz,
+                 const int8_t* incoming_payload,
+                 size_t payload_length);
 
-    int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
+  int32_t PlayoutData(AudioFrame& decoded_audio, uint16_t& samp_freq_hz);
 
-    int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
-                   size_t& encodedLengthInBytes);
+  int32_t Encode(const AudioFrame& audio,
+                 int8_t* encoded_data,
+                 size_t& encoded_length_in_bytes);
 
-protected:
- int32_t SendData(FrameType frameType,
-                  uint8_t payloadType,
-                  uint32_t timeStamp,
-                  const uint8_t* payloadData,
-                  size_t payloadSize,
-                  const RTPFragmentationHeader* fragmentation) override;
+ protected:
+  int32_t SendData(FrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t time_stamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size,
+                   const RTPFragmentationHeader* fragmentation) override;
 
-private:
- std::unique_ptr<AudioCodingModule> _acm;
- acm2::CodecManager codec_manager_;
- acm2::RentACodec rent_a_codec_;
+ private:
+  std::unique_ptr<AudioCodingModule> acm_;
+  acm2::CodecManager codec_manager_;
+  acm2::RentACodec rent_a_codec_;
 
-    CodecInst _receiveCodec;
+  CodecInst receive_codec_;
 
-    uint32_t _encodeTimestamp;
-    int8_t*  _encodedData;
-    size_t _encodedLengthInBytes;
+  uint32_t encode_timestamp_;
+  int8_t* encoded_data_;
+  size_t encoded_length_in_bytes_;
 
-    uint32_t _decodeTimestamp;
+  uint32_t decode_timestamp_;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+#endif  // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
diff --git a/chromium/third_party/webrtc/modules/utility/source/file_player_impl.h b/chromium/third_party/webrtc/modules/utility/source/file_player_impl.h
index beb6379ff0c..62887da13b8 100644
--- a/chromium/third_party/webrtc/modules/utility/source/file_player_impl.h
+++ b/chromium/third_party/webrtc/modules/utility/source/file_player_impl.h
@@ -19,7 +19,6 @@
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc
index 88b20eeac2a..b0a766f22e1 100644
--- a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -130,8 +130,7 @@ bool FileRecorderImpl::IsRecording() const
 }
 
 int32_t FileRecorderImpl::RecordAudioToFile(
-    const AudioFrame& incomingAudioFrame,
-    const TickTime* playoutTS)
+    const AudioFrame& incomingAudioFrame)
 {
     if (codec_info_.plfreq == 0)
     {
diff --git a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h
index 697d7593757..96f811d49ea 100644
--- a/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h
+++ b/chromium/third_party/webrtc/modules/utility/source/file_recorder_impl.h
@@ -27,7 +27,6 @@
 #include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -45,23 +44,21 @@ public:
     virtual ~FileRecorderImpl();
 
     // FileRecorder functions.
-    virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
-    virtual FileFormats RecordingFileFormat() const;
-    virtual int32_t StartRecordingAudioFile(
+    int32_t RegisterModuleFileCallback(FileCallback* callback) override;
+    FileFormats RecordingFileFormat() const override;
+    int32_t StartRecordingAudioFile(
         const char* fileName,
         const CodecInst& codecInst,
         uint32_t notificationTimeMs) override;
-    virtual int32_t StartRecordingAudioFile(
+    int32_t StartRecordingAudioFile(
         OutStream& destStream,
         const CodecInst& codecInst,
         uint32_t notificationTimeMs) override;
-    virtual int32_t StopRecording();
-    virtual bool IsRecording() const;
-    virtual int32_t codec_info(CodecInst& codecInst) const;
-    virtual int32_t RecordAudioToFile(
-        const AudioFrame& frame,
-        const TickTime* playoutTS = NULL);
-    virtual int32_t StartRecordingVideoFile(
+    int32_t StopRecording() override;
+    bool IsRecording() const override;
+    int32_t codec_info(CodecInst& codecInst) const override;
+    int32_t RecordAudioToFile(const AudioFrame& frame) override;
+    int32_t StartRecordingVideoFile(
         const char* fileName,
         const CodecInst& audioCodecInst,
         const VideoCodec& videoCodecInst,
@@ -69,7 +66,7 @@ public:
     {
         return -1;
     }
-    virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) {
+    int32_t RecordVideoToFile(const VideoFrame& videoFrame) override {
         return -1;
     }
 
diff --git a/chromium/third_party/webrtc/modules/utility/source/jvm_android.cc b/chromium/third_party/webrtc/modules/utility/source/jvm_android.cc
index eb37fda0405..d53d1b5eadf 100644
--- a/chromium/third_party/webrtc/modules/utility/source/jvm_android.cc
+++ b/chromium/third_party/webrtc/modules/utility/source/jvm_android.cc
@@ -10,6 +10,8 @@
 
 #include <android/log.h>
 
+#include <memory>
+
 #include "webrtc/modules/utility/include/jvm_android.h"
 
 #include "webrtc/base/checks.h"
@@ -139,7 +141,7 @@ NativeRegistration::~NativeRegistration() {
   CHECK_EXCEPTION(jni_) << "Error during UnregisterNatives";
 }
 
-rtc::scoped_ptr<GlobalRef> NativeRegistration::NewObject(
+std::unique_ptr<GlobalRef> NativeRegistration::NewObject(
     const char* name, const char* signature, ...) {
   ALOGD("NativeRegistration::NewObject%s", GetThreadInfo().c_str());
   va_list args;
@@ -149,7 +151,7 @@ rtc::scoped_ptr<GlobalRef> NativeRegistration::NewObject(
                                  args);
   CHECK_EXCEPTION(jni_) << "Error during NewObjectV";
   va_end(args);
-  return rtc::scoped_ptr<GlobalRef>(new GlobalRef(jni_, obj));
+  return std::unique_ptr<GlobalRef>(new GlobalRef(jni_, obj));
 }
 
 // JavaClass implementation.
@@ -181,14 +183,14 @@ JNIEnvironment::~JNIEnvironment() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
-rtc::scoped_ptr<NativeRegistration> JNIEnvironment::RegisterNatives(
+std::unique_ptr<NativeRegistration> JNIEnvironment::RegisterNatives(
     const char* name, const JNINativeMethod *methods, int num_methods) {
   ALOGD("JNIEnvironment::RegisterNatives(%s)", name);
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   jclass clazz = LookUpClass(name);
   jni_->RegisterNatives(clazz, methods, num_methods);
   CHECK_EXCEPTION(jni_) << "Error during RegisterNatives";
-  return rtc::scoped_ptr<NativeRegistration>(
+  return std::unique_ptr<NativeRegistration>(
       new NativeRegistration(jni_, clazz));
 }
 
@@ -240,7 +242,7 @@ JVM::~JVM() {
   DeleteGlobalRef(jni(), context_);
 }
 
-rtc::scoped_ptr<JNIEnvironment> JVM::environment() {
+std::unique_ptr<JNIEnvironment> JVM::environment() {
   ALOGD("JVM::environment%s", GetThreadInfo().c_str());
   // The JNIEnv is used for thread-local storage. For this reason, we cannot
   // share a JNIEnv between threads. If a piece of code has no other way to get
@@ -250,9 +252,9 @@ rtc::scoped_ptr<JNIEnvironment> JVM::environment() {
   JNIEnv* jni = GetEnv(jvm_);
   if (!jni) {
     ALOGE("AttachCurrentThread() has not been called on this thread.");
-    return rtc::scoped_ptr<JNIEnvironment>();
+    return std::unique_ptr<JNIEnvironment>();
   }
-  return rtc::scoped_ptr<JNIEnvironment>(new JNIEnvironment(jni));
+  return std::unique_ptr<JNIEnvironment>(new JNIEnvironment(jni));
 }
 
 JavaClass JVM::GetClass(const char* name) {
diff --git a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc
index 8cdf01634cb..4e3606ca08f 100644
--- a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc
+++ b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.cc
@@ -11,9 +11,9 @@
 #include "webrtc/modules/utility/source/process_thread_impl.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace {
@@ -36,9 +36,9 @@ int64_t GetNextCallbackTime(Module* module, int64_t time_now) {
 ProcessThread::~ProcessThread() {}
 
 // static
-rtc::scoped_ptr<ProcessThread> ProcessThread::Create(
+std::unique_ptr<ProcessThread> ProcessThread::Create(
     const char* thread_name) {
-  return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl(thread_name));
+  return std::unique_ptr<ProcessThread>(new ProcessThreadImpl(thread_name));
 }
 
 ProcessThreadImpl::ProcessThreadImpl(const char* thread_name)
@@ -119,7 +119,7 @@ void ProcessThreadImpl::WakeUp(Module* module) {
   wake_up_->Set();
 }
 
-void ProcessThreadImpl::PostTask(rtc::scoped_ptr<ProcessTask> task) {
+void ProcessThreadImpl::PostTask(std::unique_ptr<ProcessTask> task) {
   // Allowed to be called on any thread.
   {
     rtc::CritScope lock(&lock_);
@@ -188,7 +188,7 @@ bool ProcessThreadImpl::Run(void* obj) {
 }
 
 bool ProcessThreadImpl::Process() {
-  int64_t now = TickTime::MillisecondTimestamp();
+  int64_t now = rtc::TimeMillis();
   int64_t next_checkpoint = now + (1000 * 60);
 
   {
@@ -209,7 +209,7 @@ bool ProcessThreadImpl::Process() {
         // Use a new 'now' reference to calculate when the next callback
         // should occur.  We'll continue to use 'now' above for the baseline
         // of calculating how long we should wait, to reduce variance.
-        int64_t new_now = TickTime::MillisecondTimestamp();
+        int64_t new_now = rtc::TimeMillis();
         m.next_callback = GetNextCallbackTime(m.module, new_now);
       }
 
@@ -227,7 +227,7 @@ bool ProcessThreadImpl::Process() {
     }
   }
 
-  int64_t time_to_wait = next_checkpoint - TickTime::MillisecondTimestamp();
+  int64_t time_to_wait = next_checkpoint - rtc::TimeMillis();
   if (time_to_wait > 0)
     wake_up_->Wait(static_cast<unsigned long>(time_to_wait));
 
diff --git a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h
index 2855ed9d850..330aec946c9 100644
--- a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h
+++ b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl.h
@@ -33,7 +33,7 @@ class ProcessThreadImpl : public ProcessThread {
   void Stop() override;
 
   void WakeUp(Module* module) override;
-  void PostTask(rtc::scoped_ptr<ProcessTask> task) override;
+  void PostTask(std::unique_ptr<ProcessTask> task) override;
 
   void RegisterModule(Module* module) override;
   void DeRegisterModule(Module* module) override;
diff --git a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl_unittest.cc
index 9fa9edfa24a..5b31870ac41 100644
--- a/chromium/third_party/webrtc/modules/utility/source/process_thread_impl_unittest.cc
+++ b/chromium/third_party/webrtc/modules/utility/source/process_thread_impl_unittest.cc
@@ -13,9 +13,9 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/utility/source/process_thread_impl.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -51,7 +51,7 @@ ACTION_P(Increment, counter) {
 }
 
 ACTION_P(SetTimestamp, ptr) {
-  *ptr = TickTime::MillisecondTimestamp();
+  *ptr = rtc::TimeMillis();
 }
 
 TEST(ProcessThreadImpl, StartStop) {
@@ -297,7 +297,7 @@ TEST(ProcessThreadImpl, PostTask) {
   std::unique_ptr<EventWrapper> task_ran(EventWrapper::Create());
   std::unique_ptr<RaiseEventTask> task(new RaiseEventTask(task_ran.get()));
   thread.Start();
-  thread.PostTask(rtc::UniqueToScoped(std::move(task)));
+  thread.PostTask(std::move(task));
   EXPECT_EQ(kEventSignaled, task_ran->Wait(100));
   thread.Stop();
 }
diff --git a/chromium/third_party/webrtc/modules/video_capture/test/video_capture_unittest.cc b/chromium/third_party/webrtc/modules/video_capture/test/video_capture_unittest.cc
index 7ab33ffeaba..839ab804ad9 100644
--- a/chromium/third_party/webrtc/modules/video_capture/test/video_capture_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -16,19 +16,19 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_factory.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/test/frame_utils.h"
 #include "webrtc/video_frame.h"
 
 using webrtc::CriticalSectionWrapper;
 using webrtc::CriticalSectionScoped;
 using webrtc::SleepMs;
-using webrtc::TickTime;
 using webrtc::VideoCaptureAlarm;
 using webrtc::VideoCaptureCapability;
 using webrtc::VideoCaptureDataCallback;
@@ -40,8 +40,8 @@ using webrtc::VideoCaptureModule;
 #define WAIT_(ex, timeout, res) \
   do { \
     res = (ex); \
-    int64_t start = TickTime::MillisecondTimestamp(); \
-    while (!res && TickTime::MillisecondTimestamp() < start + timeout) { \
+    int64_t start = rtc::TimeMillis(); \
+    while (!res && rtc::TimeMillis() < start + timeout) { \
       SleepMs(5); \
       res = (ex); \
     } \
@@ -60,32 +60,6 @@ static const int kTestHeight = 288;
 static const int kTestWidth = 352;
 static const int kTestFramerate = 30;
 
-// Compares the content of two video frames.
-static bool CompareFrames(const webrtc::VideoFrame& frame1,
-                          const webrtc::VideoFrame& frame2) {
-  bool result =
-      (frame1.stride(webrtc::kYPlane) == frame2.stride(webrtc::kYPlane)) &&
-      (frame1.stride(webrtc::kUPlane) == frame2.stride(webrtc::kUPlane)) &&
-      (frame1.stride(webrtc::kVPlane) == frame2.stride(webrtc::kVPlane)) &&
-      (frame1.width() == frame2.width()) &&
-      (frame1.height() == frame2.height());
-
-  if (!result)
-    return false;
-  for (int plane = 0; plane < webrtc::kNumOfPlanes; plane ++) {
-      webrtc::PlaneType plane_type = static_cast<webrtc::PlaneType>(plane);
-      int allocated_size1 = frame1.allocated_size(plane_type);
-      int allocated_size2 = frame2.allocated_size(plane_type);
-      if (allocated_size1 != allocated_size2)
-        return false;
-      const uint8_t* plane_buffer1 = frame1.buffer(plane_type);
-      const uint8_t* plane_buffer2 = frame2.buffer(plane_type);
-      if (memcmp(plane_buffer1, plane_buffer2, allocated_size1))
-        return false;
-    }
-    return true;
-}
-
 class TestVideoCaptureCallback : public VideoCaptureDataCallback {
  public:
   TestVideoCaptureCallback()
@@ -118,8 +92,8 @@ class TestVideoCaptureCallback : public VideoCaptureDataCallback {
 #endif
     // RenderTimstamp should be the time now.
     EXPECT_TRUE(
-        videoFrame.render_time_ms() >= TickTime::MillisecondTimestamp()-30 &&
-        videoFrame.render_time_ms() <= TickTime::MillisecondTimestamp());
+        videoFrame.render_time_ms() >= rtc::TimeMillis()-30 &&
+        videoFrame.render_time_ms() <= rtc::TimeMillis());
 
     if ((videoFrame.render_time_ms() >
             last_render_time_ms_ + (1000 * 1.1) / capability_.maxFPS &&
@@ -132,7 +106,7 @@ class TestVideoCaptureCallback : public VideoCaptureDataCallback {
 
     incoming_frames_++;
     last_render_time_ms_ = videoFrame.render_time_ms();
-    last_frame_.CopyFrame(videoFrame);
+    last_frame_ = videoFrame.video_frame_buffer();
   }
 
   virtual void OnCaptureDelayChanged(const int32_t id,
@@ -168,7 +142,8 @@ class TestVideoCaptureCallback : public VideoCaptureDataCallback {
 
   bool CompareLastFrame(const webrtc::VideoFrame& frame) {
     CriticalSectionScoped cs(capture_cs_.get());
-    return CompareFrames(last_frame_, frame);
+    return webrtc::test::FrameBufsEqual(last_frame_,
+                                        frame.video_frame_buffer());
   }
 
   void SetExpectedCaptureRotation(webrtc::VideoRotation rotation) {
@@ -183,7 +158,7 @@ class TestVideoCaptureCallback : public VideoCaptureDataCallback {
   int64_t last_render_time_ms_;
   int incoming_frames_;
   int timing_warnings_;
-  webrtc::VideoFrame last_frame_;
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> last_frame_;
   webrtc::VideoRotation rotate_frame_;
 };
 
@@ -277,7 +252,7 @@ class VideoCaptureTest : public testing::Test {
 #endif
 TEST_F(VideoCaptureTest, MAYBE_CreateDelete) {
   for (int i = 0; i < 5; ++i) {
-    int64_t start_time = TickTime::MillisecondTimestamp();
+    int64_t start_time = rtc::TimeMillis();
     TestVideoCaptureCallback capture_observer;
     rtc::scoped_refptr<VideoCaptureModule> module(
         OpenVideoCaptureDevice(0, &capture_observer));
@@ -296,19 +271,19 @@ TEST_F(VideoCaptureTest, MAYBE_CreateDelete) {
     ASSERT_NO_FATAL_FAILURE(StartCapture(module.get(), capability));
 
     // Less than 4s to start the camera.
-    EXPECT_LE(TickTime::MillisecondTimestamp() - start_time, 4000);
+    EXPECT_LE(rtc::TimeMillis() - start_time, 4000);
 
     // Make sure 5 frames are captured.
     EXPECT_TRUE_WAIT(capture_observer.incoming_frames() >= 5, kTimeOut);
 
     EXPECT_GE(capture_observer.capture_delay(), 0);
 
-    int64_t stop_time = TickTime::MillisecondTimestamp();
+    int64_t stop_time = rtc::TimeMillis();
     EXPECT_EQ(0, module->StopCapture());
     EXPECT_FALSE(module->CaptureStarted());
 
     // Less than 3s to stop the camera.
-    EXPECT_LE(TickTime::MillisecondTimestamp() - stop_time, 3000);
+    EXPECT_LE(rtc::TimeMillis() - stop_time, 3000);
   }
 }
 
@@ -434,8 +409,7 @@ class VideoCaptureExternalTest : public testing::Test {
  public:
   void SetUp() {
     capture_module_ = VideoCaptureFactory::Create(0, capture_input_interface_);
-    process_module_ =
-        rtc::ScopedToUnique(webrtc::ProcessThread::Create("ProcessThread"));
+    process_module_ = webrtc::ProcessThread::Create("ProcessThread");
     process_module_->Start();
     process_module_->RegisterModule(capture_module_);
 
@@ -449,10 +423,11 @@ class VideoCaptureExternalTest : public testing::Test {
     test_frame_.CreateEmptyFrame(kTestWidth, kTestHeight, kTestWidth,
                                  ((kTestWidth + 1) / 2), (kTestWidth + 1) / 2);
     SleepMs(1); // Wait 1ms so that two tests can't have the same timestamp.
-    memset(test_frame_.buffer(webrtc::kYPlane), 127, kTestWidth * kTestHeight);
-    memset(test_frame_.buffer(webrtc::kUPlane), 127,
+    memset(test_frame_.video_frame_buffer()->MutableDataY(), 127,
+           kTestWidth * kTestHeight);
+    memset(test_frame_.video_frame_buffer()->MutableDataU(), 127,
            ((kTestWidth + 1) / 2) * ((kTestHeight + 1) / 2));
-    memset(test_frame_.buffer(webrtc::kVPlane), 127,
+    memset(test_frame_.video_frame_buffer()->MutableDataV(), 127,
            ((kTestWidth + 1) / 2) * ((kTestHeight + 1) / 2));
 
     capture_module_->RegisterCaptureDataCallback(capture_callback_);
@@ -493,10 +468,10 @@ TEST_F(VideoCaptureExternalTest, TestExternalCapture) {
 #define MAYBE_FrameRate FrameRate
 #endif
 TEST_F(VideoCaptureExternalTest, MAYBE_FrameRate) {
-  int64_t testTime = 3;
-  TickTime startTime = TickTime::Now();
+  uint64_t testTime = 3 * rtc::kNumNanosecsPerSec;
+  uint64_t startTime = rtc::TimeNanos();
 
-  while ((TickTime::Now() - startTime).Milliseconds() < testTime * 1000) {
+  while ((rtc::TimeNanos() - startTime) < testTime) {
      size_t length = webrtc::CalcBufferSize(webrtc::kI420,
                                             test_frame_.width(),
                                             test_frame_.height());
@@ -511,8 +486,8 @@ TEST_F(VideoCaptureExternalTest, MAYBE_FrameRate) {
   SleepMs(500);
   EXPECT_EQ(webrtc::Raised, capture_feedback_.alarm());
 
-  startTime = TickTime::Now();
-  while ((TickTime::Now() - startTime).Milliseconds() < testTime * 1000) {
+  startTime = rtc::TimeNanos();
+  while ((rtc::TimeNanos() - startTime) < testTime) {
     size_t length = webrtc::CalcBufferSize(webrtc::kI420,
                                            test_frame_.width(),
                                            test_frame_.height());
diff --git a/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi b/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi
index c80f2bf5b56..a2b2f58c9f2 100644
--- a/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi
+++ b/chromium/third_party/webrtc/modules/video_capture/video_capture.gypi
@@ -172,6 +172,7 @@
             'video_capture_module_internal_impl',
             'webrtc_utility',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/test/test.gyp:video_test_common',
             '<(DEPTH)/testing/gtest.gyp:gtest',
           ],
           'sources': [
diff --git a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc
index b1e697edc22..c3d5f370911 100644
--- a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc
+++ b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.cc
@@ -13,6 +13,7 @@
 #include <stdlib.h>
 
 #include "webrtc/base/refcount.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/include/module_common_types.h"
@@ -20,7 +21,6 @@
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace videocapturemodule {
@@ -85,7 +85,8 @@ int64_t VideoCaptureImpl::TimeUntilNextProcess()
     CriticalSectionScoped cs(&_callBackCs);
     const int64_t kProcessIntervalMs = 300;
     return kProcessIntervalMs -
-        (TickTime::Now() - _lastProcessTime).Milliseconds();
+           (rtc::TimeNanos() - _lastProcessTimeNanos) /
+           rtc::kNumNanosecsPerMillisec;
 }
 
 // Process any pending tasks such as timeouts
@@ -93,12 +94,12 @@ void VideoCaptureImpl::Process()
 {
     CriticalSectionScoped cs(&_callBackCs);
 
-    const TickTime now = TickTime::Now();
-    _lastProcessTime = TickTime::Now();
+    const int64_t now_ns = rtc::TimeNanos();
+    _lastProcessTimeNanos = rtc::TimeNanos();
 
     // Handle No picture alarm
 
-    if (_lastProcessFrameCount.Ticks() == _incomingFrameTimes[0].Ticks() &&
+    if (_lastProcessFrameTimeNanos == _incomingFrameTimesNanos[0] &&
         _captureAlarm != Raised)
     {
         if (_noPictureAlarmCallBack && _captureCallBack)
@@ -107,7 +108,7 @@ void VideoCaptureImpl::Process()
             _captureCallBack->OnNoPictureAlarm(_id, _captureAlarm);
         }
     }
-    else if (_lastProcessFrameCount.Ticks() != _incomingFrameTimes[0].Ticks() &&
+    else if (_lastProcessFrameTimeNanos != _incomingFrameTimesNanos[0] &&
              _captureAlarm != Cleared)
     {
         if (_noPictureAlarmCallBack && _captureCallBack)
@@ -119,19 +120,21 @@ void VideoCaptureImpl::Process()
     }
 
     // Handle frame rate callback
-    if ((now - _lastFrameRateCallbackTime).Milliseconds()
+    if ((now_ns - _lastFrameRateCallbackTimeNanos) /
+        rtc::kNumNanosecsPerMillisec
         > kFrameRateCallbackInterval)
     {
         if (_frameRateCallBack && _captureCallBack)
         {
-            const uint32_t frameRate = CalculateFrameRate(now);
+            const uint32_t frameRate = CalculateFrameRate(now_ns);
             _captureCallBack->OnCaptureFrameRate(_id, frameRate);
         }
-        _lastFrameRateCallbackTime = now; // Can be set by EnableFrameRateCallback
+        // Can be set by EnableFrameRateCallback
+        _lastFrameRateCallbackTimeNanos = now_ns;
 
     }
 
-    _lastProcessFrameCount = _incomingFrameTimes[0];
+    _lastProcessFrameTimeNanos = _incomingFrameTimesNanos[0];
 }
 
 VideoCaptureImpl::VideoCaptureImpl(const int32_t id)
@@ -141,15 +144,15 @@ VideoCaptureImpl::VideoCaptureImpl(const int32_t id)
       _captureDelay(0),
       _requestedCapability(),
       _callBackCs(*CriticalSectionWrapper::CreateCriticalSection()),
-      _lastProcessTime(TickTime::Now()),
-      _lastFrameRateCallbackTime(TickTime::Now()),
+      _lastProcessTimeNanos(rtc::TimeNanos()),
+      _lastFrameRateCallbackTimeNanos(rtc::TimeNanos()),
       _frameRateCallBack(false),
       _noPictureAlarmCallBack(false),
       _captureAlarm(Cleared),
       _setCaptureDelay(0),
       _dataCallBack(NULL),
       _captureCallBack(NULL),
-      _lastProcessFrameCount(TickTime::Now()),
+      _lastProcessFrameTimeNanos(rtc::TimeNanos()),
       _rotateFrame(kVideoRotation_0),
       apply_rotation_(false) {
     _requestedCapability.width = kDefaultWidth;
@@ -157,7 +160,7 @@ VideoCaptureImpl::VideoCaptureImpl(const int32_t id)
     _requestedCapability.maxFPS = 30;
     _requestedCapability.rawType = kVideoI420;
     _requestedCapability.codecType = kVideoCodecUnknown;
-    memset(_incomingFrameTimes, 0, sizeof(_incomingFrameTimes));
+    memset(_incomingFrameTimesNanos, 0, sizeof(_incomingFrameTimesNanos));
 }
 
 VideoCaptureImpl::~VideoCaptureImpl()
@@ -295,7 +298,7 @@ int32_t VideoCaptureImpl::IncomingFrame(
           _captureFrame.set_rotation(kVideoRotation_0);
         }
         _captureFrame.set_ntp_time_ms(captureTime);
-        _captureFrame.set_render_time_ms(TickTime::MillisecondTimestamp());
+        _captureFrame.set_render_time_ms(rtc::TimeMillis());
 
         DeliverCapturedFrame(_captureFrame);
     }
@@ -321,7 +324,7 @@ void VideoCaptureImpl::EnableFrameRateCallback(const bool enable) {
     _frameRateCallBack = enable;
     if (enable)
     {
-        _lastFrameRateCallbackTime = TickTime::Now();
+      _lastFrameRateCallbackTimeNanos = rtc::TimeNanos();
     }
 }
 
@@ -341,7 +344,7 @@ void VideoCaptureImpl::EnableNoPictureAlarm(const bool enable) {
 
 void VideoCaptureImpl::UpdateFrameCount()
 {
-    if (_incomingFrameTimes[0].MicrosecondTimestamp() == 0)
+  if (_incomingFrameTimesNanos[0] / rtc::kNumNanosecsPerMicrosec == 0)
     {
         // first no shift
     }
@@ -350,20 +353,22 @@ void VideoCaptureImpl::UpdateFrameCount()
         // shift
         for (int i = (kFrameRateCountHistorySize - 2); i >= 0; i--)
         {
-            _incomingFrameTimes[i + 1] = _incomingFrameTimes[i];
+            _incomingFrameTimesNanos[i + 1] = _incomingFrameTimesNanos[i];
         }
     }
-    _incomingFrameTimes[0] = TickTime::Now();
+    _incomingFrameTimesNanos[0] = rtc::TimeNanos();
 }
 
-uint32_t VideoCaptureImpl::CalculateFrameRate(const TickTime& now)
+uint32_t VideoCaptureImpl::CalculateFrameRate(int64_t now_ns)
 {
     int32_t num = 0;
     int32_t nrOfFrames = 0;
     for (num = 1; num < (kFrameRateCountHistorySize - 1); num++)
     {
-        if (_incomingFrameTimes[num].Ticks() <= 0
-            || (now - _incomingFrameTimes[num]).Milliseconds() > kFrameRateHistoryWindowMs) // don't use data older than 2sec
+        if (_incomingFrameTimesNanos[num] <= 0 ||
+            (now_ns - _incomingFrameTimesNanos[num]) /
+            rtc::kNumNanosecsPerMillisec >
+                kFrameRateHistoryWindowMs) // don't use data older than 2sec
         {
             break;
         }
@@ -374,7 +379,8 @@ uint32_t VideoCaptureImpl::CalculateFrameRate(const TickTime& now)
     }
     if (num > 1)
     {
-        int64_t diff = (now - _incomingFrameTimes[num - 1]).Milliseconds();
+        int64_t diff = (now_ns - _incomingFrameTimesNanos[num - 1]) /
+                       rtc::kNumNanosecsPerMillisec;
         if (diff > 0)
         {
             return uint32_t((nrOfFrames * 1000.0f / diff) + 0.5f);
diff --git a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h
index 9c2cad7c95c..7d785c3a908 100644
--- a/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h
+++ b/chromium/third_party/webrtc/modules/video_capture/video_capture_impl.h
@@ -20,7 +20,6 @@
 #include "webrtc/common_video/rotation.h"
 #include "webrtc/modules/video_capture/video_capture.h"
 #include "webrtc/modules/video_capture/video_capture_config.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video_frame.h"
 
 namespace webrtc
@@ -116,12 +115,14 @@ protected:
     VideoCaptureCapability _requestedCapability; // Should be set by platform dependent code in StartCapture.
 private:
     void UpdateFrameCount();
-    uint32_t CalculateFrameRate(const TickTime& now);
+    uint32_t CalculateFrameRate(int64_t now_ns);
 
     CriticalSectionWrapper& _callBackCs;
 
-    TickTime _lastProcessTime; // last time the module process function was called.
-    TickTime _lastFrameRateCallbackTime; // last time the frame rate callback function was called.
+    // last time the module process function was called.
+    int64_t _lastProcessTimeNanos;
+    // last time the frame rate callback function was called.
+    int64_t _lastFrameRateCallbackTimeNanos;
     bool _frameRateCallBack; // true if EnableFrameRateCallback
     bool _noPictureAlarmCallBack; //true if EnableNoPictureAlarm
     VideoCaptureAlarm _captureAlarm; // current value of the noPictureAlarm
@@ -130,8 +131,9 @@ private:
     VideoCaptureDataCallback* _dataCallBack;
     VideoCaptureFeedBack* _captureCallBack;
 
-    TickTime _lastProcessFrameCount;
-    TickTime _incomingFrameTimes[kFrameRateCountHistorySize];// timestamp for local captured frames
+    int64_t _lastProcessFrameTimeNanos;
+    // timestamp for local captured frames
+    int64_t _incomingFrameTimesNanos[kFrameRateCountHistorySize];
     VideoRotation _rotateFrame;  // Set if the frame should be rotated by the
                                  // capture module.
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/BUILD.gn b/chromium/third_party/webrtc/modules/video_coding/BUILD.gn
index bc6f595b51d..755e6efa9da 100644
--- a/chromium/third_party/webrtc/modules/video_coding/BUILD.gn
+++ b/chromium/third_party/webrtc/modules/video_coding/BUILD.gn
@@ -10,14 +10,10 @@ import("../../build/webrtc.gni")
 
 source_set("video_coding") {
   sources = [
-    "bitrate_adjuster.cc",
-    "bitrate_adjuster.h",
     "codec_database.cc",
     "codec_database.h",
     "codec_timer.cc",
     "codec_timer.h",
-    "content_metrics_processing.cc",
-    "content_metrics_processing.h",
     "decoding_state.cc",
     "decoding_state.h",
     "encoded_frame.cc",
@@ -25,6 +21,8 @@ source_set("video_coding") {
     "fec_tables_xor.h",
     "frame_buffer.cc",
     "frame_buffer.h",
+    "frame_buffer2.cc",
+    "frame_buffer2.h",
     "frame_object.cc",
     "frame_object.h",
     "generic_decoder.cc",
@@ -56,11 +54,10 @@ source_set("video_coding") {
     "packet_buffer.h",
     "percentile_filter.cc",
     "percentile_filter.h",
-    "qm_select.cc",
-    "qm_select.h",
-    "qm_select_data.h",
     "receiver.cc",
     "receiver.h",
+    "rtp_frame_reference_finder.cc",
+    "rtp_frame_reference_finder.h",
     "rtt_filter.cc",
     "rtt_filter.h",
     "session_info.cc",
@@ -107,6 +104,8 @@ source_set("video_coding_utility") {
   sources = [
     "utility/frame_dropper.cc",
     "utility/frame_dropper.h",
+    "utility/ivf_file_writer.cc",
+    "utility/ivf_file_writer.h",
     "utility/moving_average.h",
     "utility/qp_parser.cc",
     "utility/qp_parser.h",
diff --git a/chromium/third_party/webrtc/modules/video_coding/OWNERS b/chromium/third_party/webrtc/modules/video_coding/OWNERS
index 389d632dfdf..acf09399ac7 100644
--- a/chromium/third_party/webrtc/modules/video_coding/OWNERS
+++ b/chromium/third_party/webrtc/modules/video_coding/OWNERS
@@ -1,5 +1,6 @@
-stefan@webrtc.org
 marpan@webrtc.org
+pbos@webrtc.org
+stefan@webrtc.org
 
 # These are for the common case of adding or renaming files. If you're doing
 # structural changes, please get a review from a reviewer in this file.
diff --git a/chromium/third_party/webrtc/modules/video_coding/codec_database.cc b/chromium/third_party/webrtc/modules/video_coding/codec_database.cc
index a5a7c1ea999..1baa414bce7 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codec_database.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codec_database.cc
@@ -238,7 +238,6 @@ bool VCMCodecDataBase::SetSendCodec(const VideoCodec* send_codec,
   memcpy(&send_codec_, &new_send_codec, sizeof(send_codec_));
 
   if (!reset_required) {
-    encoded_frame_callback_->SetPayloadType(send_codec_.plType);
     return true;
   }
 
@@ -249,7 +248,6 @@ bool VCMCodecDataBase::SetSendCodec(const VideoCodec* send_codec,
   ptr_encoder_.reset(
       new VCMGenericEncoder(external_encoder_, encoder_rate_observer_,
                             encoded_frame_callback_, internal_source_));
-  encoded_frame_callback_->SetPayloadType(send_codec_.plType);
   encoded_frame_callback_->SetInternalSource(internal_source_);
   if (ptr_encoder_->InitEncode(&send_codec_, number_of_cores_,
                                max_payload_size_) < 0) {
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264.gypi b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264.gypi
index 9cb58a85986..92489c39665 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264.gypi
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264.gypi
@@ -64,7 +64,7 @@
           'type': 'static_library',
           'includes': [ '../../../../build/objc_common.gypi' ],
           'dependencies': [
-            '<(webrtc_root)/base/base.gyp:rtc_base_objc',
+            '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
           ],
           'link_settings': {
             'xcode_settings': {
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index e98666d0736..f560a37d0ec 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -129,10 +129,12 @@ int H264DecoderImpl::AVGetBuffer2(
   video_frame->set_video_frame_buffer(
       decoder->pool_.CreateBuffer(width, height));
   // DCHECK that we have a continuous buffer as is required.
-  RTC_DCHECK_EQ(video_frame->buffer(kUPlane),
-      video_frame->buffer(kYPlane) + video_frame->allocated_size(kYPlane));
-  RTC_DCHECK_EQ(video_frame->buffer(kVPlane),
-      video_frame->buffer(kUPlane) + video_frame->allocated_size(kUPlane));
+  RTC_DCHECK_EQ(video_frame->video_frame_buffer()->DataU(),
+                video_frame->video_frame_buffer()->DataY() +
+                video_frame->allocated_size(kYPlane));
+  RTC_DCHECK_EQ(video_frame->video_frame_buffer()->DataV(),
+                video_frame->video_frame_buffer()->DataU() +
+                video_frame->allocated_size(kUPlane));
   int total_size = video_frame->allocated_size(kYPlane) +
                    video_frame->allocated_size(kUPlane) +
                    video_frame->allocated_size(kVPlane);
@@ -141,12 +143,18 @@ int H264DecoderImpl::AVGetBuffer2(
   av_frame->reordered_opaque = context->reordered_opaque;
 
   // Set |av_frame| members as required by FFmpeg.
-  av_frame->data[kYPlaneIndex] = video_frame->buffer(kYPlane);
-  av_frame->linesize[kYPlaneIndex] = video_frame->stride(kYPlane);
-  av_frame->data[kUPlaneIndex] = video_frame->buffer(kUPlane);
-  av_frame->linesize[kUPlaneIndex] = video_frame->stride(kUPlane);
-  av_frame->data[kVPlaneIndex] = video_frame->buffer(kVPlane);
-  av_frame->linesize[kVPlaneIndex] = video_frame->stride(kVPlane);
+  av_frame->data[kYPlaneIndex] =
+      video_frame->video_frame_buffer()->MutableDataY();
+  av_frame->linesize[kYPlaneIndex] =
+      video_frame->video_frame_buffer()->StrideY();
+  av_frame->data[kUPlaneIndex] =
+      video_frame->video_frame_buffer()->MutableDataU();
+  av_frame->linesize[kUPlaneIndex] =
+      video_frame->video_frame_buffer()->StrideU();
+  av_frame->data[kVPlaneIndex] =
+      video_frame->video_frame_buffer()->MutableDataV();
+  av_frame->linesize[kVPlaneIndex] =
+      video_frame->video_frame_buffer()->StrideV();
   RTC_DCHECK_EQ(av_frame->extended_data, av_frame->data);
 
   av_frame->buf[0] = av_buffer_create(av_frame->data[kYPlaneIndex],
@@ -339,9 +347,12 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
   VideoFrame* video_frame = static_cast<VideoFrame*>(
       av_buffer_get_opaque(av_frame_->buf[0]));
   RTC_DCHECK(video_frame);
-  RTC_CHECK_EQ(av_frame_->data[kYPlane], video_frame->buffer(kYPlane));
-  RTC_CHECK_EQ(av_frame_->data[kUPlane], video_frame->buffer(kUPlane));
-  RTC_CHECK_EQ(av_frame_->data[kVPlane], video_frame->buffer(kVPlane));
+  RTC_CHECK_EQ(av_frame_->data[kYPlane],
+               video_frame->video_frame_buffer()->DataY());
+  RTC_CHECK_EQ(av_frame_->data[kUPlane],
+               video_frame->video_frame_buffer()->DataU());
+  RTC_CHECK_EQ(av_frame_->data[kVPlane],
+               video_frame->video_frame_buffer()->DataV());
   video_frame->set_timestamp(input_image._timeStamp);
 
   // The decoded image may be larger than what is supposed to be visible, see
@@ -352,9 +363,9 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
     video_frame->set_video_frame_buffer(
         new rtc::RefCountedObject<WrappedI420Buffer>(
             av_frame_->width, av_frame_->height,
-            buf->data(kYPlane), buf->stride(kYPlane),
-            buf->data(kUPlane), buf->stride(kUPlane),
-            buf->data(kVPlane), buf->stride(kVPlane),
+            buf->DataY(), buf->StrideY(),
+            buf->DataU(), buf->StrideU(),
+            buf->DataV(), buf->StrideV(),
             rtc::KeepRefUntilDone(buf)));
   }
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
index 0e065c5e497..4d85858a162 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -50,12 +50,8 @@ int NumberOfThreads(int width, int height, int number_of_cores) {
   return 1;
 }
 
-}  // namespace
-
-static FrameType EVideoFrameType_to_FrameType(EVideoFrameType type) {
+FrameType ConvertToVideoFrameType(EVideoFrameType type) {
   switch (type) {
-    case videoFrameTypeInvalid:
-      return kEmptyFrame;
     case videoFrameTypeIDR:
       return kVideoFrameKey;
     case videoFrameTypeSkip:
@@ -63,12 +59,15 @@ static FrameType EVideoFrameType_to_FrameType(EVideoFrameType type) {
     case videoFrameTypeP:
     case videoFrameTypeIPMixed:
       return kVideoFrameDelta;
-    default:
-      LOG(LS_WARNING) << "Unknown EVideoFrameType: " << type;
-      return kVideoFrameDelta;
+    case videoFrameTypeInvalid:
+      break;
   }
+  RTC_NOTREACHED() << "Unexpected/invalid frame type: " << type;
+  return kEmptyFrame;
 }
 
+}  // namespace
+
 // Helper method used by H264EncoderImpl::Encode.
 // Copies the encoded bytes from |info| to |encoded_image| and updates the
 // fragmentation information of |frag_header|. The |encoded_image->_buffer| may
@@ -368,12 +367,12 @@ int32_t H264EncoderImpl::Encode(
   picture.iPicHeight = frame.height();
   picture.iColorFormat = EVideoFormatType::videoFormatI420;
   picture.uiTimeStamp = frame.ntp_time_ms();
-  picture.iStride[0] = frame.stride(kYPlane);
-  picture.iStride[1] = frame.stride(kUPlane);
-  picture.iStride[2] = frame.stride(kVPlane);
-  picture.pData[0] = const_cast<uint8_t*>(frame.buffer(kYPlane));
-  picture.pData[1] = const_cast<uint8_t*>(frame.buffer(kUPlane));
-  picture.pData[2] = const_cast<uint8_t*>(frame.buffer(kVPlane));
+  picture.iStride[0] = frame.video_frame_buffer()->StrideY();
+  picture.iStride[1] = frame.video_frame_buffer()->StrideU();
+  picture.iStride[2] = frame.video_frame_buffer()->StrideV();
+  picture.pData[0] = const_cast<uint8_t*>(frame.video_frame_buffer()->DataY());
+  picture.pData[1] = const_cast<uint8_t*>(frame.video_frame_buffer()->DataU());
+  picture.pData[2] = const_cast<uint8_t*>(frame.video_frame_buffer()->DataV());
 
   // EncodeFrame output.
   SFrameBSInfo info;
@@ -393,7 +392,8 @@ int32_t H264EncoderImpl::Encode(
   encoded_image_._timeStamp = frame.timestamp();
   encoded_image_.ntp_time_ms_ = frame.ntp_time_ms();
   encoded_image_.capture_time_ms_ = frame.render_time_ms();
-  encoded_image_._frameType = EVideoFrameType_to_FrameType(info.eFrameType);
+  encoded_image_.rotation_ = frame.rotation();
+  encoded_image_._frameType = ConvertToVideoFrameType(info.eFrameType);
 
   // Split encoded image up into fragments. This also updates |encoded_image_|.
   RTPFragmentationHeader frag_header;
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
index 0ea2600197e..18820d3ded7 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc
@@ -15,12 +15,12 @@
 
 #include <memory>
 
+#if defined(WEBRTC_IOS)
+#include "RTCUIApplication.h"
+#endif
 #include "libyuv/convert.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#if defined(WEBRTC_IOS)
-#include "webrtc/base/objc/RTCUIApplication.h"
-#endif
 #include "webrtc/common_video/include/corevideo_frame_buffer.h"
 #include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
 #include "webrtc/video_frame.h"
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
index 8cfe63dbb78..5f6a231288d 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.cc
@@ -17,12 +17,12 @@
 #include <string>
 #include <vector>
 
+#if defined(WEBRTC_IOS)
+#include "RTCUIApplication.h"
+#endif
 #include "libyuv/convert_from.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#if defined(WEBRTC_IOS)
-#include "webrtc/base/objc/RTCUIApplication.h"
-#endif
 #include "webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
@@ -118,8 +118,14 @@ struct FrameEncodeParams {
                     int32_t w,
                     int32_t h,
                     int64_t rtms,
-                    uint32_t ts)
-      : encoder(e), width(w), height(h), render_time_ms(rtms), timestamp(ts) {
+                    uint32_t ts,
+                    webrtc::VideoRotation r)
+      : encoder(e),
+        width(w),
+        height(h),
+        render_time_ms(rtms),
+        timestamp(ts),
+        rotation(r) {
     if (csi) {
       codec_specific_info = *csi;
     } else {
@@ -133,6 +139,7 @@ struct FrameEncodeParams {
   int32_t height;
   int64_t render_time_ms;
   uint32_t timestamp;
+  webrtc::VideoRotation rotation;
 };
 
 // We receive I420Frames as input, but we need to feed CVPixelBuffers into the
@@ -161,10 +168,14 @@ bool CopyVideoFrameToPixelBuffer(const webrtc::VideoFrame& frame,
   int dst_stride_uv = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer, 1);
   // Convert I420 to NV12.
   int ret = libyuv::I420ToNV12(
-      frame.buffer(webrtc::kYPlane), frame.stride(webrtc::kYPlane),
-      frame.buffer(webrtc::kUPlane), frame.stride(webrtc::kUPlane),
-      frame.buffer(webrtc::kVPlane), frame.stride(webrtc::kVPlane), dst_y,
-      dst_stride_y, dst_uv, dst_stride_uv, frame.width(), frame.height());
+      frame.video_frame_buffer()->DataY(),
+      frame.video_frame_buffer()->StrideY(),
+      frame.video_frame_buffer()->DataU(),
+      frame.video_frame_buffer()->StrideU(),
+      frame.video_frame_buffer()->DataV(),
+      frame.video_frame_buffer()->StrideV(),
+      dst_y, dst_stride_y, dst_uv, dst_stride_uv,
+      frame.width(), frame.height());
   CVPixelBufferUnlockBaseAddress(pixel_buffer, 0);
   if (ret) {
     LOG(LS_ERROR) << "Error converting I420 VideoFrame to NV12 :" << ret;
@@ -185,7 +196,8 @@ void VTCompressionOutputCallback(void* encoder,
   encode_params->encoder->OnEncodedFrame(
       status, info_flags, sample_buffer, encode_params->codec_specific_info,
       encode_params->width, encode_params->height,
-      encode_params->render_time_ms, encode_params->timestamp);
+      encode_params->render_time_ms, encode_params->timestamp,
+      encode_params->rotation);
 }
 
 }  // namespace internal
@@ -248,6 +260,7 @@ int H264VideoToolboxEncoder::Encode(
     return WEBRTC_VIDEO_CODEC_OK;
   }
 #endif
+  bool is_keyframe_required = false;
   // Get a pixel buffer from the pool and copy frame data over.
   CVPixelBufferPoolRef pixel_buffer_pool =
       VTCompressionSessionGetPixelBufferPool(compression_session_);
@@ -257,9 +270,11 @@ int H264VideoToolboxEncoder::Encode(
     // invalidated, which causes this pool call to fail when the application
     // is foregrounded and frames are being sent for encoding again.
     // Resetting the session when this happens fixes the issue.
+    // In addition we request a keyframe so video can recover quickly.
     ResetCompressionSession();
     pixel_buffer_pool =
         VTCompressionSessionGetPixelBufferPool(compression_session_);
+    is_keyframe_required = true;
   }
 #endif
   if (!pixel_buffer_pool) {
@@ -283,8 +298,7 @@ int H264VideoToolboxEncoder::Encode(
   }
 
   // Check if we need a keyframe.
-  bool is_keyframe_required = false;
-  if (frame_types) {
+  if (!is_keyframe_required && frame_types) {
     for (auto frame_type : *frame_types) {
       if (frame_type == kVideoFrameKey) {
         is_keyframe_required = true;
@@ -304,7 +318,7 @@ int H264VideoToolboxEncoder::Encode(
   std::unique_ptr<internal::FrameEncodeParams> encode_params;
   encode_params.reset(new internal::FrameEncodeParams(
       this, codec_specific_info, width_, height_, input_image.render_time_ms(),
-      input_image.timestamp()));
+      input_image.timestamp(), input_image.rotation()));
 
   // Update the bitrate if needed.
   SetBitrateBps(bitrate_adjuster_.GetAdjustedBitrateBps());
@@ -469,7 +483,8 @@ void H264VideoToolboxEncoder::OnEncodedFrame(
     int32_t width,
     int32_t height,
     int64_t render_time_ms,
-    uint32_t timestamp) {
+    uint32_t timestamp,
+    VideoRotation rotation) {
   if (status != noErr) {
     LOG(LS_ERROR) << "H264 encode failed.";
     return;
@@ -509,6 +524,7 @@ void H264VideoToolboxEncoder::OnEncodedFrame(
       is_keyframe ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta;
   frame.capture_time_ms_ = render_time_ms;
   frame._timeStamp = timestamp;
+  frame.rotation_ = rotation;
 
   int result = callback_->Encoded(frame, &codec_specific_info, header.get());
   if (result != 0) {
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
index 779889d43cc..d54fa612c3d 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_encoder.h
@@ -12,8 +12,9 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_ENCODER_H_
 
+#include "webrtc/common_video/include/bitrate_adjuster.h"
+#include "webrtc/common_video/rotation.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
-#include "webrtc/modules/video_coding/include/bitrate_adjuster.h"
 
 #if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
 
@@ -58,7 +59,8 @@ class H264VideoToolboxEncoder : public H264Encoder {
                       int32_t width,
                       int32_t height,
                       int64_t render_time_ms,
-                      uint32_t timestamp);
+                      uint32_t timestamp,
+                      VideoRotation rotation);
 
  private:
   int ResetCompressionSession();
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
index e64babd599f..9a9a0ddf165 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
@@ -17,6 +17,7 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 
 namespace webrtc {
@@ -198,7 +199,7 @@ bool VideoProcessorImpl::ProcessFrame(int frame_number) {
     // Ensure we have a new statistics data object we can fill:
     FrameStatistic& stat = stats_->NewFrame(frame_number);
 
-    encode_start_ = TickTime::Now();
+    encode_start_ns_ = rtc::TimeNanos();
     // Use the frame number as "timestamp" to identify frames
     source_frame_.set_timestamp(frame_number);
 
@@ -248,11 +249,11 @@ void VideoProcessorImpl::FrameEncoded(
 
   encoded_frame_type_ = encoded_image._frameType;
 
-  TickTime encode_stop = TickTime::Now();
+  int64_t encode_stop_ns = rtc::TimeNanos();
   int frame_number = encoded_image._timeStamp;
   FrameStatistic& stat = stats_->stats_[frame_number];
   stat.encode_time_in_us =
-      GetElapsedTimeMicroseconds(encode_start_, encode_stop);
+      GetElapsedTimeMicroseconds(encode_start_ns_, encode_stop_ns);
   stat.encoding_successful = true;
   stat.encoded_frame_length_in_bytes = encoded_image._length;
   stat.frame_number = encoded_image._timeStamp;
@@ -299,7 +300,7 @@ void VideoProcessorImpl::FrameEncoded(
 
   // Keep track of if frames are lost due to packet loss so we can tell
   // this to the encoder (this is handled by the RTP logic in the full stack)
-  decode_start_ = TickTime::Now();
+  decode_start_ns_ = rtc::TimeNanos();
   // TODO(kjellander): Pass fragmentation header to the decoder when
   // CL 172001 has been submitted and PacketManipulator supports this.
   int32_t decode_result =
@@ -315,12 +316,12 @@ void VideoProcessorImpl::FrameEncoded(
 }
 
 void VideoProcessorImpl::FrameDecoded(const VideoFrame& image) {
-  TickTime decode_stop = TickTime::Now();
+  int64_t decode_stop_ns = rtc::TimeNanos();
   int frame_number = image.timestamp();
   // Report stats
   FrameStatistic& stat = stats_->stats_[frame_number];
   stat.decode_time_in_us =
-      GetElapsedTimeMicroseconds(decode_start_, decode_stop);
+      GetElapsedTimeMicroseconds(decode_start_ns_, decode_stop_ns);
   stat.decoding_successful = true;
 
   // Check for resize action (either down or up):
@@ -378,10 +379,9 @@ void VideoProcessorImpl::FrameDecoded(const VideoFrame& image) {
   }
 }
 
-int VideoProcessorImpl::GetElapsedTimeMicroseconds(
-    const webrtc::TickTime& start,
-    const webrtc::TickTime& stop) {
-  uint64_t encode_time = (stop - start).Microseconds();
+int VideoProcessorImpl::GetElapsedTimeMicroseconds(int64_t start,
+                                                   int64_t stop) {
+  uint64_t encode_time = (stop - start) / rtc::kNumNanosecsPerMicrosec;
   assert(encode_time <
          static_cast<unsigned int>(std::numeric_limits<int>::max()));
   return static_cast<int>(encode_time);
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h
index cd1c7b9d62c..f0322dd67b4 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor.h
@@ -19,7 +19,6 @@
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h"
 #include "webrtc/modules/video_coding/codecs/test/stats.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/frame_reader.h"
 #include "webrtc/test/testsupport/frame_writer.h"
 #include "webrtc/video_frame.h"
@@ -179,8 +178,7 @@ class VideoProcessorImpl : public VideoProcessor {
   void FrameDecoded(const webrtc::VideoFrame& image);
   // Used for getting a 32-bit integer representing time
   // (checks the size is within signed 32-bit bounds before casting it)
-  int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
-                                 const webrtc::TickTime& stop);
+  int GetElapsedTimeMicroseconds(int64_t start, int64_t stop);
   // Updates the encoder with the target bit rate and the frame rate.
   void SetRates(int bit_rate, int frame_rate) override;
   // Return the size of the encoded frame in bytes.
@@ -225,8 +223,8 @@ class VideoProcessorImpl : public VideoProcessor {
 
   // Statistics
   double bit_rate_factor_;  // multiply frame length with this to get bit rate
-  webrtc::TickTime encode_start_;
-  webrtc::TickTime decode_start_;
+  int64_t encode_start_ns_;
+  int64_t decode_start_ns_;
 
   // Callback class required to implement according to the VideoEncoder API.
   class VideoProcessorEncodeCompleteCallback
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index 897870a2717..9f361dc6261 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -626,6 +626,7 @@ TEST_F(VideoProcessorIntegrationTest, Process0PercentPacketLossH264) {
 // Fails on iOS. See webrtc:4755.
 #if !defined(WEBRTC_IOS)
 
+#if !defined(RTC_DISABLE_VP9)
 // VP9: Run with no packet loss and fixed bitrate. Quality should be very high.
 // One key frame (first frame only) in sequence. Setting |key_frame_interval|
 // to -1 below means no periodic key frames in test.
@@ -780,6 +781,8 @@ TEST_F(VideoProcessorIntegrationTest, ProcessNoLossSpatialResizeFrameDropVP9) {
 // TODO(marpan): Add temporal layer test for VP9, once changes are in
 // vp9 wrapper for this.
 
+#endif  // !defined(RTC_DISABLE_VP9)
+
 // VP8: Run with no packet loss and fixed bitrate. Quality should be very high.
 // One key frame (first frame only) in sequence. Setting |key_frame_interval|
 // to -1 below means no periodic key frames in test.
@@ -961,7 +964,7 @@ TEST_F(VideoProcessorIntegrationTest,
   SetQualityMetrics(&quality_metrics, 25.0, 15.0, 0.70, 0.40);
   // Metrics for rate control.
   RateControlMetrics rc_metrics[1];
-  SetRateControlMetrics(rc_metrics, 0, 160, 60, 120, 20, 70, 1, 2);
+  SetRateControlMetrics(rc_metrics, 0, 160, 80, 120, 20, 70, 1, 2);
   ProcessFramesAndVerify(quality_metrics, rate_profile, process_settings,
                          rc_metrics);
 }
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
index d22601358f2..b9721cde1bc 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
@@ -101,10 +101,10 @@ class RealTimeTemporalLayers : public TemporalLayers {
 
   virtual ~RealTimeTemporalLayers() {}
 
-  virtual bool ConfigureBitrates(int bitrate_kbit,
-                                 int max_bitrate_kbit,
-                                 int framerate,
-                                 vpx_codec_enc_cfg_t* cfg) {
+  bool ConfigureBitrates(int bitrate_kbit,
+                         int max_bitrate_kbit,
+                         int framerate,
+                         vpx_codec_enc_cfg_t* cfg) override {
     temporal_layers_ =
         CalculateNumberOfTemporalLayers(temporal_layers_, framerate);
     temporal_layers_ = std::min(temporal_layers_, max_temporal_layers_);
@@ -184,7 +184,7 @@ class RealTimeTemporalLayers : public TemporalLayers {
     return true;
   }
 
-  virtual int EncodeFlags(uint32_t timestamp) {
+  int EncodeFlags(uint32_t timestamp) override {
     frame_counter_++;
     return CurrentEncodeFlags();
   }
@@ -196,16 +196,16 @@ class RealTimeTemporalLayers : public TemporalLayers {
     return encode_flags_[index];
   }
 
-  virtual int CurrentLayerId() const {
+  int CurrentLayerId() const override {
     assert(layer_ids_length_ > 0 && layer_ids_ != NULL);
     int index = frame_counter_ % layer_ids_length_;
     assert(index >= 0 && index < layer_ids_length_);
     return layer_ids_[index];
   }
 
-  virtual void PopulateCodecSpecific(bool base_layer_sync,
-                                     CodecSpecificInfoVP8* vp8_info,
-                                     uint32_t timestamp) {
+  void PopulateCodecSpecific(bool base_layer_sync,
+                             CodecSpecificInfoVP8* vp8_info,
+                             uint32_t timestamp) override {
     assert(temporal_layers_ > 0);
 
     if (temporal_layers_ == 1) {
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
index 55a4402cbe4..be55133dd6b 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.cc
@@ -301,14 +301,21 @@ int SimulcastEncoderAdapter::Encode(
       // Aligning stride values based on width.
       dst_frame.CreateEmptyFrame(dst_width, dst_height, dst_width,
                                  (dst_width + 1) / 2, (dst_width + 1) / 2);
-      libyuv::I420Scale(
-          input_image.buffer(kYPlane), input_image.stride(kYPlane),
-          input_image.buffer(kUPlane), input_image.stride(kUPlane),
-          input_image.buffer(kVPlane), input_image.stride(kVPlane), src_width,
-          src_height, dst_frame.buffer(kYPlane), dst_frame.stride(kYPlane),
-          dst_frame.buffer(kUPlane), dst_frame.stride(kUPlane),
-          dst_frame.buffer(kVPlane), dst_frame.stride(kVPlane), dst_width,
-          dst_height, libyuv::kFilterBilinear);
+      libyuv::I420Scale(input_image.video_frame_buffer()->DataY(),
+                        input_image.video_frame_buffer()->StrideY(),
+                        input_image.video_frame_buffer()->DataU(),
+                        input_image.video_frame_buffer()->StrideU(),
+                        input_image.video_frame_buffer()->DataV(),
+                        input_image.video_frame_buffer()->StrideV(),
+                        src_width, src_height,
+                        dst_frame.video_frame_buffer()->MutableDataY(),
+                        dst_frame.video_frame_buffer()->StrideY(),
+                        dst_frame.video_frame_buffer()->MutableDataU(),
+                        dst_frame.video_frame_buffer()->StrideU(),
+                        dst_frame.video_frame_buffer()->MutableDataV(),
+                        dst_frame.video_frame_buffer()->StrideV(),
+                        dst_width, dst_height,
+                        libyuv::kFilterBilinear);
       dst_frame.set_timestamp(input_image.timestamp());
       dst_frame.set_render_time_ms(input_image.render_time_ms());
       streaminfos_[stream_idx].encoder->Encode(dst_frame, codec_specific_info,
@@ -494,10 +501,6 @@ void SimulcastEncoderAdapter::OnDroppedFrame() {
   streaminfos_[0].encoder->OnDroppedFrame();
 }
 
-int SimulcastEncoderAdapter::GetTargetFramerate() {
-  return streaminfos_[0].encoder->GetTargetFramerate();
-}
-
 bool SimulcastEncoderAdapter::SupportsNativeHandle() const {
   // We should not be calling this method before streaminfos_ are configured.
   RTC_DCHECK(!streaminfos_.empty());
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
index 777ac1ba368..fca16df6fad 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h
@@ -58,7 +58,6 @@ class SimulcastEncoderAdapter : public VP8Encoder {
 
   void OnDroppedFrame() override;
 
-  int GetTargetFramerate() override;
   bool SupportsNativeHandle() const override;
   const char* ImplementationName() const override;
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
index 9a7e1b2e7ca..aafcd797ac5 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
@@ -107,34 +107,40 @@ TEST_F(TestSimulcastEncoderAdapter, DISABLED_TestRPSIEncoder) {
 
 class MockVideoEncoder : public VideoEncoder {
  public:
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
   int32_t InitEncode(const VideoCodec* codecSettings,
                      int32_t numberOfCores,
-                     size_t maxPayloadSize) override {
+                     size_t maxPayloadSize) /* override */ {
     codec_ = *codecSettings;
     return 0;
   }
 
   int32_t Encode(const VideoFrame& inputImage,
                  const CodecSpecificInfo* codecSpecificInfo,
-                 const std::vector<FrameType>* frame_types) override {
+                 const std::vector<FrameType>* frame_types) /* override */ {
     return 0;
   }
 
   int32_t RegisterEncodeCompleteCallback(
-      EncodedImageCallback* callback) override {
+      EncodedImageCallback* callback) /* override */ {
     callback_ = callback;
     return 0;
   }
 
-  int32_t Release() override { return 0; }
+  int32_t Release() /* override */ { return 0; }
 
-  int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override {
+  int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) /* override */ {
     return 0;
   }
 
   MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
 
-  bool SupportsNativeHandle() const override { return supports_native_handle_; }
+  bool SupportsNativeHandle() const /* override */ {
+    return supports_native_handle_;
+  }
 
   virtual ~MockVideoEncoder() {}
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
index 2b2aa5de69f..b277ad2ee48 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
@@ -119,13 +119,13 @@ class Vp8TestDecodedImageCallback : public DecodedImageCallback {
   Vp8TestDecodedImageCallback() : decoded_frames_(0) {}
   int32_t Decoded(VideoFrame& decoded_image) override {
     for (int i = 0; i < decoded_image.width(); ++i) {
-      EXPECT_NEAR(kColorY, decoded_image.buffer(kYPlane)[i], 1);
+      EXPECT_NEAR(kColorY, decoded_image.video_frame_buffer()->DataY()[i], 1);
     }
 
     // TODO(mikhal): Verify the difference between U,V and the original.
     for (int i = 0; i < ((decoded_image.width() + 1) / 2); ++i) {
-      EXPECT_NEAR(kColorU, decoded_image.buffer(kUPlane)[i], 4);
-      EXPECT_NEAR(kColorV, decoded_image.buffer(kVPlane)[i], 4);
+      EXPECT_NEAR(kColorU, decoded_image.video_frame_buffer()->DataU()[i], 4);
+      EXPECT_NEAR(kColorV, decoded_image.video_frame_buffer()->DataV()[i], 4);
     }
     decoded_frames_++;
     return 0;
@@ -168,7 +168,7 @@ class SkipEncodingUnusedStreamsTest {
 
     virtual ~SpyingTemporalLayers() { delete layers_; }
 
-    virtual int EncodeFlags(uint32_t timestamp) {
+    int EncodeFlags(uint32_t timestamp) override {
       return layers_->EncodeFlags(timestamp);
     }
 
@@ -222,26 +222,40 @@ class TestVp8Simulcast : public ::testing::Test {
   TestVp8Simulcast(VP8Encoder* encoder, VP8Decoder* decoder)
       : encoder_(encoder), decoder_(decoder) {}
 
-  // Creates an VideoFrame from |plane_colors|.
-  static void CreateImage(VideoFrame* frame, int plane_colors[kNumOfPlanes]) {
-    for (int plane_num = 0; plane_num < kNumOfPlanes; ++plane_num) {
-      int width =
-          (plane_num != kYPlane ? (frame->width() + 1) / 2 : frame->width());
-      int height =
-          (plane_num != kYPlane ? (frame->height() + 1) / 2 : frame->height());
-      PlaneType plane_type = static_cast<PlaneType>(plane_num);
-      uint8_t* data = frame->buffer(plane_type);
+  static void SetPlane(uint8_t* data,
+                       uint8_t value,
+                       int width,
+                       int height,
+                       int stride) {
+    for (int i = 0; i < height; i++, data += stride) {
       // Setting allocated area to zero - setting only image size to
       // requested values - will make it easier to distinguish between image
       // size and frame size (accounting for stride).
-      memset(frame->buffer(plane_type), 0, frame->allocated_size(plane_type));
-      for (int i = 0; i < height; i++) {
-        memset(data, plane_colors[plane_num], width);
-        data += frame->stride(plane_type);
-      }
+      memset(data, value, width);
+      memset(data + width, 0, stride - width);
     }
   }
 
+  // Fills in an VideoFrameBuffer from |plane_colors|.
+  static void CreateImage(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
+                          int plane_colors[kNumOfPlanes]) {
+    int width = buffer->width();
+    int height = buffer->height();
+    int chroma_width = (width + 1) / 2;
+    int chroma_height = (height + 1) / 2;
+
+    SetPlane(buffer->MutableDataY(), plane_colors[0],
+             width, height, buffer->StrideY());
+
+    SetPlane(buffer->MutableDataU(), plane_colors[1],
+             chroma_width, chroma_height,
+             buffer->StrideU());
+
+    SetPlane(buffer->MutableDataV(), plane_colors[2],
+             chroma_width, chroma_height,
+             buffer->StrideV());
+  }
+
   static void DefaultSettings(VideoCodec* settings,
                               const int* temporal_layer_profile) {
     assert(settings);
@@ -305,11 +319,11 @@ class TestVp8Simulcast : public ::testing::Test {
     int half_width = (kDefaultWidth + 1) / 2;
     input_frame_.CreateEmptyFrame(kDefaultWidth, kDefaultHeight, kDefaultWidth,
                                   half_width, half_width);
-    memset(input_frame_.buffer(kYPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataY(), 0,
            input_frame_.allocated_size(kYPlane));
-    memset(input_frame_.buffer(kUPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataU(), 0,
            input_frame_.allocated_size(kUPlane));
-    memset(input_frame_.buffer(kVPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataV(), 0,
            input_frame_.allocated_size(kVPlane));
   }
 
@@ -555,11 +569,11 @@ class TestVp8Simulcast : public ::testing::Test {
     int half_width = (settings_.width + 1) / 2;
     input_frame_.CreateEmptyFrame(settings_.width, settings_.height,
                                   settings_.width, half_width, half_width);
-    memset(input_frame_.buffer(kYPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataY(), 0,
            input_frame_.allocated_size(kYPlane));
-    memset(input_frame_.buffer(kUPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataU(), 0,
            input_frame_.allocated_size(kUPlane));
-    memset(input_frame_.buffer(kVPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataV(), 0,
            input_frame_.allocated_size(kVPlane));
 
     // The for loop above did not set the bitrate of the highest layer.
@@ -596,11 +610,11 @@ class TestVp8Simulcast : public ::testing::Test {
     half_width = (settings_.width + 1) / 2;
     input_frame_.CreateEmptyFrame(settings_.width, settings_.height,
                                   settings_.width, half_width, half_width);
-    memset(input_frame_.buffer(kYPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataY(), 0,
            input_frame_.allocated_size(kYPlane));
-    memset(input_frame_.buffer(kUPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataU(), 0,
            input_frame_.allocated_size(kUPlane));
-    memset(input_frame_.buffer(kVPlane), 0,
+    memset(input_frame_.video_frame_buffer()->MutableDataV(), 0,
            input_frame_.allocated_size(kVPlane));
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, &frame_types));
   }
@@ -691,7 +705,7 @@ class TestVp8Simulcast : public ::testing::Test {
     plane_offset[kYPlane] = kColorY;
     plane_offset[kUPlane] = kColorU;
     plane_offset[kVPlane] = kColorV;
-    CreateImage(&input_frame_, plane_offset);
+    CreateImage(input_frame_.video_frame_buffer(), plane_offset);
 
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
     int picture_id = -1;
@@ -707,7 +721,7 @@ class TestVp8Simulcast : public ::testing::Test {
     plane_offset[kYPlane] += 1;
     plane_offset[kUPlane] += 1;
     plane_offset[kVPlane] += 1;
-    CreateImage(&input_frame_, plane_offset);
+    CreateImage(input_frame_.video_frame_buffer(), plane_offset);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
 
@@ -715,7 +729,7 @@ class TestVp8Simulcast : public ::testing::Test {
     plane_offset[kYPlane] += 1;
     plane_offset[kUPlane] += 1;
     plane_offset[kVPlane] += 1;
-    CreateImage(&input_frame_, plane_offset);
+    CreateImage(input_frame_.video_frame_buffer(), plane_offset);
 
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
@@ -724,7 +738,7 @@ class TestVp8Simulcast : public ::testing::Test {
     plane_offset[kYPlane] += 1;
     plane_offset[kUPlane] += 1;
     plane_offset[kVPlane] += 1;
-    CreateImage(&input_frame_, plane_offset);
+    CreateImage(input_frame_.video_frame_buffer(), plane_offset);
 
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
@@ -739,7 +753,7 @@ class TestVp8Simulcast : public ::testing::Test {
     plane_offset[kYPlane] = kColorY;
     plane_offset[kUPlane] = kColorU;
     plane_offset[kVPlane] = kColorV;
-    CreateImage(&input_frame_, plane_offset);
+    CreateImage(input_frame_.video_frame_buffer(), plane_offset);
 
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, &codec_specific, NULL));
@@ -898,7 +912,7 @@ class TestVp8Simulcast : public ::testing::Test {
     plane_offset[kYPlane] = kColorY;
     plane_offset[kUPlane] = kColorU;
     plane_offset[kVPlane] = kColorV;
-    CreateImage(&input_frame_, plane_offset);
+    CreateImage(input_frame_.video_frame_buffer(), plane_offset);
 
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
 
@@ -906,7 +920,7 @@ class TestVp8Simulcast : public ::testing::Test {
     plane_offset[kYPlane] += 1;
     plane_offset[kUPlane] += 1;
     plane_offset[kVPlane] += 1;
-    CreateImage(&input_frame_, plane_offset);
+    CreateImage(input_frame_.video_frame_buffer(), plane_offset);
     input_frame_.set_timestamp(input_frame_.timestamp() + 3000);
     EXPECT_EQ(0, encoder_->Encode(input_frame_, NULL, NULL));
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index f3ebfa1f766..d562dd4bb97 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -14,9 +14,9 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -159,8 +159,8 @@ class TestVp8Impl : public ::testing::Test {
   }
 
   size_t WaitForEncodedFrame() const {
-    int64_t startTime = TickTime::MillisecondTimestamp();
-    while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs) {
+    int64_t startTime = rtc::TimeMillis();
+    while (rtc::TimeMillis() - startTime < kMaxWaitEncTimeMs) {
       if (encode_complete_callback_->EncodeComplete()) {
         return encoded_frame_._length;
       }
@@ -169,8 +169,8 @@ class TestVp8Impl : public ::testing::Test {
   }
 
   size_t WaitForDecodedFrame() const {
-    int64_t startTime = TickTime::MillisecondTimestamp();
-    while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitDecTimeMs) {
+    int64_t startTime = rtc::TimeMillis();
+    while (rtc::TimeMillis() - startTime < kMaxWaitDecTimeMs) {
       if (decode_complete_callback_->DecodeComplete()) {
         return CalcBufferSize(kI420, decoded_frame_.width(),
                               decoded_frame_.height());
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
index b34288632cd..f035568355d 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
@@ -20,6 +20,7 @@
 #include "libyuv/convert.h"  // NOLINT
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@@ -29,7 +30,6 @@
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
 #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace {
@@ -166,7 +166,7 @@ VP8EncoderImpl::VP8EncoderImpl()
       tl1_frame_dropper_(kTl1MaxTimeToDropFrames),
       key_frame_request_(kMaxSimulcastStreams, false),
       quality_scaler_enabled_(false) {
-  uint32_t seed = static_cast<uint32_t>(TickTime::MillisecondTimestamp());
+  uint32_t seed = rtc::Time32();
   srand(seed);
 
   picture_id_.reserve(kMaxSimulcastStreams);
@@ -598,14 +598,9 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
   }
 
   rps_.Init();
-  // Disable both high-QP limits and framedropping. Both are handled by libvpx
-  // internally.
-  const int kDisabledBadQpThreshold = 64;
-  // TODO(glaznev/sprang): consider passing codec initial bitrate to quality
-  // scaler to avoid starting with HD for low initial bitrates.
-  quality_scaler_.Init(codec_.qpMax / QualityScaler::kDefaultLowQpDenominator,
-                       kDisabledBadQpThreshold, false, 0, 0, 0,
-                       codec_.maxFramerate);
+  quality_scaler_.Init(QualityScaler::kLowVp8QpThreshold,
+                       QualityScaler::kBadVp8QpThreshold, codec_.startBitrate,
+                       codec_.width, codec_.height, codec_.maxFramerate);
 
   // Only apply scaling to improve for single-layer streams. The scaling metrics
   // use frame drops as a signal and is only applicable when we drop frames.
@@ -751,15 +746,18 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
   // Image in vpx_image_t format.
   // Input image is const. VP8's raw image is not defined as const.
   raw_images_[0].planes[VPX_PLANE_Y] =
-      const_cast<uint8_t*>(input_image.buffer(kYPlane));
+      const_cast<uint8_t*>(input_image.video_frame_buffer()->DataY());
   raw_images_[0].planes[VPX_PLANE_U] =
-      const_cast<uint8_t*>(input_image.buffer(kUPlane));
+      const_cast<uint8_t*>(input_image.video_frame_buffer()->DataU());
   raw_images_[0].planes[VPX_PLANE_V] =
-      const_cast<uint8_t*>(input_image.buffer(kVPlane));
+      const_cast<uint8_t*>(input_image.video_frame_buffer()->DataV());
 
-  raw_images_[0].stride[VPX_PLANE_Y] = input_image.stride(kYPlane);
-  raw_images_[0].stride[VPX_PLANE_U] = input_image.stride(kUPlane);
-  raw_images_[0].stride[VPX_PLANE_V] = input_image.stride(kVPlane);
+  raw_images_[0].stride[VPX_PLANE_Y] =
+      input_image.video_frame_buffer()->StrideY();
+  raw_images_[0].stride[VPX_PLANE_U] =
+      input_image.video_frame_buffer()->StrideU();
+  raw_images_[0].stride[VPX_PLANE_V] =
+      input_image.video_frame_buffer()->StrideV();
 
   for (size_t i = 1; i < encoders_.size(); ++i) {
     // Scale the image down a number of times by downsampling factor
@@ -1020,6 +1018,7 @@ int VP8EncoderImpl::GetEncodedPartitions(const VideoFrame& input_image,
     encoded_images_[encoder_idx]._timeStamp = input_image.timestamp();
     encoded_images_[encoder_idx].capture_time_ms_ =
         input_image.render_time_ms();
+    encoded_images_[encoder_idx].rotation_ = input_image.rotation();
 
     int qp = -1;
     vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
@@ -1053,9 +1052,9 @@ int VP8EncoderImpl::GetEncodedPartitions(const VideoFrame& input_image,
   }
   if (encoders_.size() == 1 && send_stream_[0]) {
     if (encoded_images_[0]._length > 0) {
-      int qp;
-      vpx_codec_control(&encoders_[0], VP8E_GET_LAST_QUANTIZER_64, &qp);
-      quality_scaler_.ReportQP(qp);
+      int qp_128;
+      vpx_codec_control(&encoders_[0], VP8E_GET_LAST_QUANTIZER, &qp_128);
+      quality_scaler_.ReportQP(qp_128);
     } else {
       quality_scaler_.ReportDroppedFrame();
     }
@@ -1355,9 +1354,12 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
   libyuv::I420Copy(img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
                    img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
                    img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
-                   decoded_image.buffer(kYPlane), decoded_image.stride(kYPlane),
-                   decoded_image.buffer(kUPlane), decoded_image.stride(kUPlane),
-                   decoded_image.buffer(kVPlane), decoded_image.stride(kVPlane),
+                   decoded_image.video_frame_buffer()->MutableDataY(),
+                   decoded_image.video_frame_buffer()->StrideY(),
+                   decoded_image.video_frame_buffer()->MutableDataU(),
+                   decoded_image.video_frame_buffer()->StrideU(),
+                   decoded_image.video_frame_buffer()->MutableDataV(),
+                   decoded_image.video_frame_buffer()->StrideV(),
                    img->d_w, img->d_h);
   decoded_image.set_ntp_time_ms(ntp_time_ms);
   int ret = decode_complete_callback_->Decoded(decoded_image);
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
index 6906a322afc..f8af6422538 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
@@ -40,21 +40,21 @@ class VP8EncoderImpl : public VP8Encoder {
 
   virtual ~VP8EncoderImpl();
 
-  virtual int Release();
+  int Release() override;
 
-  virtual int InitEncode(const VideoCodec* codec_settings,
-                         int number_of_cores,
-                         size_t max_payload_size);
+  int InitEncode(const VideoCodec* codec_settings,
+                 int number_of_cores,
+                 size_t max_payload_size) override;
 
-  virtual int Encode(const VideoFrame& input_image,
-                     const CodecSpecificInfo* codec_specific_info,
-                     const std::vector<FrameType>* frame_types);
+  int Encode(const VideoFrame& input_image,
+             const CodecSpecificInfo* codec_specific_info,
+             const std::vector<FrameType>* frame_types) override;
 
-  virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
+  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
 
-  virtual int SetChannelParameters(uint32_t packet_loss, int64_t rtt);
+  int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
 
-  virtual int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate);
+  int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate) override;
 
   void OnDroppedFrame() override {}
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
index 33dae8d8e4c..28027009537 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
@@ -12,10 +12,10 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/include/video_image.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/metrics/video_metrics.h"
 #include "webrtc/tools/simple_command_line_parser.h"
@@ -158,7 +158,7 @@ int SequenceCoder(webrtc::test::CommandLineParser* parser) {
   decoder->RegisterDecodeCompleteCallback(&decoder_callback);
   // Read->Encode->Decode sequence.
   // num_frames = -1 implies unlimited encoding (entire sequence).
-  int64_t starttime = webrtc::TickTime::MillisecondTimestamp();
+  int64_t starttime = rtc::TimeMillis();
   int frame_cnt = 1;
   int frames_processed = 0;
   input_frame.CreateEmptyFrame(width, height, width, half_width, half_width);
@@ -176,7 +176,7 @@ int SequenceCoder(webrtc::test::CommandLineParser* parser) {
     ++frame_cnt;
   }
   printf("\nProcessed %d frames\n", frames_processed);
-  int64_t endtime = webrtc::TickTime::MillisecondTimestamp();
+  int64_t endtime = rtc::TimeMillis();
   int64_t totalExecutionTime = endtime - starttime;
   printf("Total execution time: %.2lf ms\n",
          static_cast<double>(totalExecutionTime));
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/include/vp9.h b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
index 3bcbe46b3a8..3b726a0cc5d 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
@@ -18,6 +18,7 @@ namespace webrtc {
 
 class VP9Encoder : public VideoEncoder {
  public:
+  static bool IsSupported();
   static VP9Encoder* Create();
 
   virtual ~VP9Encoder() {}
@@ -25,6 +26,7 @@ class VP9Encoder : public VideoEncoder {
 
 class VP9Decoder : public VideoDecoder {
  public:
+  static bool IsSupported();
   static VP9Decoder* Create();
 
   virtual ~VP9Decoder() {}
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9.gyp b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
index cd5201f8aac..9124e5fad30 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9.gyp
@@ -20,6 +20,21 @@
             '<(libvpx_dir)/libvpx.gyp:libvpx',
           ],
         }],
+        ['libvpx_build_vp9==1', {
+          'sources': [
+            'screenshare_layers.cc',
+            'screenshare_layers.h',
+            'vp9_frame_buffer_pool.cc',
+            'vp9_frame_buffer_pool.h',
+            'vp9_impl.cc',
+            'vp9_impl.h',
+          ],
+        }, {
+          'sources': [
+            'vp9_noop.cc',
+          ],
+        }
+        ],
       ],
       'dependencies': [
         '<(webrtc_root)/common_video/common_video.gyp:common_video',
@@ -28,12 +43,6 @@
       ],
       'sources': [
         'include/vp9.h',
-        'screenshare_layers.cc',
-        'screenshare_layers.h',
-        'vp9_frame_buffer_pool.cc',
-        'vp9_frame_buffer_pool.h',
-        'vp9_impl.cc',
-        'vp9_impl.h',
       ],
     },
   ],
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
index fcefdb89fa2..750f7427cdd 100644
--- a/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -22,13 +22,13 @@
 #include "vpx/vp8dx.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/keep_ref_until_done.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -47,6 +47,10 @@ int GetCpuSpeed(int width, int height) {
 #endif
 }
 
+bool VP9Encoder::IsSupported() {
+  return true;
+}
+
 VP9Encoder* VP9Encoder::Create() {
   return new VP9EncoderImpl();
 }
@@ -77,7 +81,7 @@ VP9EncoderImpl::VP9EncoderImpl()
       // Use two spatial when screensharing with flexible mode.
       spatial_layer_(new ScreenshareLayersVP9(2)) {
   memset(&codec_, 0, sizeof(codec_));
-  uint32_t seed = static_cast<uint32_t>(TickTime::MillisecondTimestamp());
+  uint32_t seed = rtc::Time32();
   srand(seed);
 }
 
@@ -500,12 +504,15 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
 
   // Image in vpx_image_t format.
   // Input image is const. VPX's raw image is not defined as const.
-  raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(input_image.buffer(kYPlane));
-  raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(input_image.buffer(kUPlane));
-  raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(input_image.buffer(kVPlane));
-  raw_->stride[VPX_PLANE_Y] = input_image.stride(kYPlane);
-  raw_->stride[VPX_PLANE_U] = input_image.stride(kUPlane);
-  raw_->stride[VPX_PLANE_V] = input_image.stride(kVPlane);
+  raw_->planes[VPX_PLANE_Y] =
+      const_cast<uint8_t*>(input_image.video_frame_buffer()->DataY());
+  raw_->planes[VPX_PLANE_U] =
+      const_cast<uint8_t*>(input_image.video_frame_buffer()->DataU());
+  raw_->planes[VPX_PLANE_V] =
+      const_cast<uint8_t*>(input_image.video_frame_buffer()->DataV());
+  raw_->stride[VPX_PLANE_Y] = input_image.video_frame_buffer()->StrideY();
+  raw_->stride[VPX_PLANE_U] = input_image.video_frame_buffer()->StrideU();
+  raw_->stride[VPX_PLANE_V] = input_image.video_frame_buffer()->StrideV();
 
   vpx_enc_frame_flags_t flags = 0;
   bool send_keyframe = (frame_type == kVideoFrameKey);
@@ -692,8 +699,12 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
     TRACE_COUNTER1("webrtc", "EncodedFrameSize", encoded_image_._length);
     encoded_image_._timeStamp = input_image_->timestamp();
     encoded_image_.capture_time_ms_ = input_image_->render_time_ms();
+    encoded_image_.rotation_ = input_image_->rotation();
     encoded_image_._encodedHeight = raw_->d_h;
     encoded_image_._encodedWidth = raw_->d_w;
+    int qp = -1;
+    vpx_codec_control(encoder_, VP8E_GET_LAST_QUANTIZER, &qp);
+    encoded_image_.qp_ = qp;
     encoded_complete_callback_->Encoded(encoded_image_, &codec_specific,
                                         &frag_info);
   }
@@ -816,6 +827,10 @@ const char* VP9EncoderImpl::ImplementationName() const {
   return "libvpx";
 }
 
+bool VP9Decoder::IsSupported() {
+  return true;
+}
+
 VP9Decoder* VP9Decoder::Create() {
   return new VP9DecoderImpl();
 }
diff --git a/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc
new file mode 100644
index 00000000000..cc37e1adbbd
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#if !defined(RTC_DISABLE_VP9)
+#error
+#endif  // !defined(RTC_DISABLE_VP9)
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
+
+namespace webrtc {
+
+bool VP9Encoder::IsSupported() {
+  return false;
+}
+
+VP9Encoder* VP9Encoder::Create() {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+bool VP9Decoder::IsSupported() {
+  return false;
+}
+
+VP9Decoder* VP9Decoder::Create() {
+  RTC_NOTREACHED();
+  return nullptr;
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/content_metrics_processing.cc b/chromium/third_party/webrtc/modules/video_coding/content_metrics_processing.cc
deleted file mode 100644
index b2586fce3fb..00000000000
--- a/chromium/third_party/webrtc/modules/video_coding/content_metrics_processing.cc
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/content_metrics_processing.h"
-
-#include <math.h>
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/video_coding/include/video_coding_defines.h"
-
-namespace webrtc {
-//////////////////////////////////
-/// VCMContentMetricsProcessing //
-//////////////////////////////////
-
-VCMContentMetricsProcessing::VCMContentMetricsProcessing()
-    : recursive_avg_factor_(1 / 150.0f),  // matched to  30fps.
-      frame_cnt_uniform_avg_(0),
-      avg_motion_level_(0.0f),
-      avg_spatial_level_(0.0f) {
-  recursive_avg_ = new VideoContentMetrics();
-  uniform_avg_ = new VideoContentMetrics();
-}
-
-VCMContentMetricsProcessing::~VCMContentMetricsProcessing() {
-  delete recursive_avg_;
-  delete uniform_avg_;
-}
-
-int VCMContentMetricsProcessing::Reset() {
-  recursive_avg_->Reset();
-  uniform_avg_->Reset();
-  frame_cnt_uniform_avg_ = 0;
-  avg_motion_level_ = 0.0f;
-  avg_spatial_level_ = 0.0f;
-  return VCM_OK;
-}
-
-void VCMContentMetricsProcessing::UpdateFrameRate(uint32_t frameRate) {
-  if (frameRate == 0)
-    frameRate = 1;
-  // Update factor for recursive averaging.
-  recursive_avg_factor_ = static_cast<float>(1000.0f) /
-                          static_cast<float>(frameRate * kQmMinIntervalMs);
-}
-
-VideoContentMetrics* VCMContentMetricsProcessing::LongTermAvgData() {
-  return recursive_avg_;
-}
-
-VideoContentMetrics* VCMContentMetricsProcessing::ShortTermAvgData() {
-  if (frame_cnt_uniform_avg_ == 0) {
-    return NULL;
-  }
-  // Two metrics are used: motion and spatial level.
-  uniform_avg_->motion_magnitude =
-      avg_motion_level_ / static_cast<float>(frame_cnt_uniform_avg_);
-  uniform_avg_->spatial_pred_err =
-      avg_spatial_level_ / static_cast<float>(frame_cnt_uniform_avg_);
-  return uniform_avg_;
-}
-
-void VCMContentMetricsProcessing::ResetShortTermAvgData() {
-  // Reset.
-  avg_motion_level_ = 0.0f;
-  avg_spatial_level_ = 0.0f;
-  frame_cnt_uniform_avg_ = 0;
-}
-
-int VCMContentMetricsProcessing::UpdateContentData(
-    const VideoContentMetrics* contentMetrics) {
-  if (contentMetrics == NULL) {
-    return VCM_OK;
-  }
-  return ProcessContent(contentMetrics);
-}
-
-int VCMContentMetricsProcessing::ProcessContent(
-    const VideoContentMetrics* contentMetrics) {
-  // Update the recursive averaged metrics: average is over longer window
-  // of time: over QmMinIntervalMs ms.
-  UpdateRecursiveAvg(contentMetrics);
-  // Update the uniform averaged metrics: average is over shorter window
-  // of time: based on ~RTCP reports.
-  UpdateUniformAvg(contentMetrics);
-  return VCM_OK;
-}
-
-void VCMContentMetricsProcessing::UpdateUniformAvg(
-    const VideoContentMetrics* contentMetrics) {
-  // Update frame counter.
-  frame_cnt_uniform_avg_ += 1;
-  // Update averaged metrics: motion and spatial level are used.
-  avg_motion_level_ += contentMetrics->motion_magnitude;
-  avg_spatial_level_ += contentMetrics->spatial_pred_err;
-  return;
-}
-
-void VCMContentMetricsProcessing::UpdateRecursiveAvg(
-    const VideoContentMetrics* contentMetrics) {
-  // Spatial metrics: 2x2, 1x2(H), 2x1(V).
-  recursive_avg_->spatial_pred_err =
-      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err +
-      recursive_avg_factor_ * contentMetrics->spatial_pred_err;
-
-  recursive_avg_->spatial_pred_err_h =
-      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err_h +
-      recursive_avg_factor_ * contentMetrics->spatial_pred_err_h;
-
-  recursive_avg_->spatial_pred_err_v =
-      (1 - recursive_avg_factor_) * recursive_avg_->spatial_pred_err_v +
-      recursive_avg_factor_ * contentMetrics->spatial_pred_err_v;
-
-  // Motion metric: Derived from NFD (normalized frame difference).
-  recursive_avg_->motion_magnitude =
-      (1 - recursive_avg_factor_) * recursive_avg_->motion_magnitude +
-      recursive_avg_factor_ * contentMetrics->motion_magnitude;
-}
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/content_metrics_processing.h b/chromium/third_party/webrtc/modules/video_coding/content_metrics_processing.h
deleted file mode 100644
index 3f67ec19c98..00000000000
--- a/chromium/third_party/webrtc/modules/video_coding/content_metrics_processing.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-struct VideoContentMetrics;
-
-// QM interval time (in ms)
-enum { kQmMinIntervalMs = 10000 };
-
-// Flag for NFD metric vs motion metric
-enum { kNfdMetric = 1 };
-
-/**********************************/
-/* Content Metrics Processing     */
-/**********************************/
-class VCMContentMetricsProcessing {
- public:
-  VCMContentMetricsProcessing();
-  ~VCMContentMetricsProcessing();
-
-  // Update class with latest metrics.
-  int UpdateContentData(const VideoContentMetrics* contentMetrics);
-
-  // Reset the short-term averaged content data.
-  void ResetShortTermAvgData();
-
-  // Initialize.
-  int Reset();
-
-  // Inform class of current frame rate.
-  void UpdateFrameRate(uint32_t frameRate);
-
-  // Returns the long-term averaged content data: recursive average over longer
-  // time scale.
-  VideoContentMetrics* LongTermAvgData();
-
-  // Returns the short-term averaged content data: uniform average over
-  // shorter time scalE.
-  VideoContentMetrics* ShortTermAvgData();
-
- private:
-  // Compute working average.
-  int ProcessContent(const VideoContentMetrics* contentMetrics);
-
-  // Update the recursive averaged metrics: longer time average (~5/10 secs).
-  void UpdateRecursiveAvg(const VideoContentMetrics* contentMetrics);
-
-  // Update the uniform averaged metrics: shorter time average (~RTCP report).
-  void UpdateUniformAvg(const VideoContentMetrics* contentMetrics);
-
-  VideoContentMetrics* recursive_avg_;
-  VideoContentMetrics* uniform_avg_;
-  float recursive_avg_factor_;
-  uint32_t frame_cnt_uniform_avg_;
-  float avg_motion_level_;
-  float avg_spatial_level_;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/frame_buffer2.cc b/chromium/third_party/webrtc/modules/video_coding/frame_buffer2.cc
new file mode 100644
index 00000000000..c6a1a06e756
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/frame_buffer2.cc
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/frame_buffer2.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/video_coding/frame_object.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
+#include "webrtc/modules/video_coding/sequence_number_util.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace video_coding {
+
+namespace {
+// The maximum age of decoded frames tracked by frame buffer, compared to
+// |newest_picture_id_|.
+constexpr int kMaxFrameAge = 4096;
+
+// The maximum number of decoded frames being tracked by the frame buffer.
+constexpr int kMaxNumHistoryFrames = 256;
+}  // namespace
+
+bool FrameBuffer::FrameComp::operator()(const FrameKey& f1,
+                                        const FrameKey& f2) const {
+  // first = picture id
+  // second = spatial layer
+  if (f1.first == f2.first)
+    return f1.second < f2.second;
+  return AheadOf(f2.first, f1.first);
+}
+
+FrameBuffer::FrameBuffer(Clock* clock,
+                         VCMJitterEstimator* jitter_estimator,
+                         const VCMTiming* timing)
+    : clock_(clock),
+      frame_inserted_event_(false, false),
+      jitter_estimator_(jitter_estimator),
+      timing_(timing),
+      newest_picture_id_(-1) {}
+
+std::unique_ptr<FrameObject> FrameBuffer::NextFrame(int64_t max_wait_time_ms) {
+  int64_t latest_return_time = clock_->TimeInMilliseconds() + max_wait_time_ms;
+  while (true) {
+    int64_t now = clock_->TimeInMilliseconds();
+    int64_t wait_ms = max_wait_time_ms;
+
+    crit_.Enter();
+    frame_inserted_event_.Reset();
+    auto next_frame = frames_.end();
+    for (auto frame_it = frames_.begin(); frame_it != frames_.end();
+         ++frame_it) {
+      const FrameObject& frame = *frame_it->second;
+      if (IsContinuous(frame)) {
+        next_frame = frame_it;
+        int64_t render_time = timing_->RenderTimeMs(frame.timestamp, now);
+        wait_ms = timing_->MaxWaitingTime(render_time, now);
+
+        // This will cause the frame buffer to prefer high framerate rather
+        // than high resolution in the case of the decoder not decoding fast
+        // enough and the stream has multiple spatial and temporal layers.
+        if (wait_ms == 0)
+          continue;
+
+        break;
+      }
+    }
+    crit_.Leave();
+
+    // If the timout occures, return. Otherwise a new frame has been inserted
+    // and the best frame to decode next will be selected again.
+    wait_ms = std::min<int64_t>(wait_ms, latest_return_time - now);
+    wait_ms = std::max<int64_t>(wait_ms, 0);
+    if (!frame_inserted_event_.Wait(wait_ms)) {
+      crit_.Enter();
+      if (next_frame != frames_.end()) {
+        // TODO(philipel): update jitter estimator with correct values.
+        jitter_estimator_->UpdateEstimate(100, 100);
+
+        decoded_frames_.insert(next_frame->first);
+        std::unique_ptr<FrameObject> frame = std::move(next_frame->second);
+        frames_.erase(frames_.begin(), ++next_frame);
+        crit_.Leave();
+        return frame;
+      } else {
+        crit_.Leave();
+        return std::unique_ptr<FrameObject>();
+      }
+    }
+  }
+}
+
+void FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
+  rtc::CritScope lock(&crit_);
+  if (newest_picture_id_ == -1)
+    newest_picture_id_ = frame->picture_id;
+
+  if (AheadOf<uint16_t>(frame->picture_id, newest_picture_id_))
+    newest_picture_id_ = frame->picture_id;
+
+  // Remove frames as long as we have too many, |kMaxNumHistoryFrames|.
+  while (decoded_frames_.size() > kMaxNumHistoryFrames)
+    decoded_frames_.erase(decoded_frames_.begin());
+
+  // Remove frames that are too old, |kMaxNumHistoryFrames|.
+  uint16_t old_picture_id = Subtract<1 << 16>(newest_picture_id_, kMaxFrameAge);
+  auto old_decoded_it =
+      decoded_frames_.lower_bound(FrameKey(old_picture_id, 0));
+  decoded_frames_.erase(decoded_frames_.begin(), old_decoded_it);
+
+  FrameKey key(frame->picture_id, frame->spatial_layer);
+  frames_[key] = std::move(frame);
+  frame_inserted_event_.Set();
+}
+
+bool FrameBuffer::IsContinuous(const FrameObject& frame) const {
+  // If a frame with an earlier picture id was inserted compared to the last
+  // decoded frames picture id then that frame arrived too late.
+  if (!decoded_frames_.empty() &&
+      AheadOf(decoded_frames_.rbegin()->first, frame.picture_id)) {
+    return false;
+  }
+
+  // Have we decoded all frames that this frame depend on?
+  for (size_t r = 0; r < frame.num_references; ++r) {
+    FrameKey ref_key(frame.references[r], frame.spatial_layer);
+    if (decoded_frames_.find(ref_key) == decoded_frames_.end())
+      return false;
+  }
+
+  // If this is a layer frame, have we decoded the lower layer of this
+  // super frame.
+  if (frame.inter_layer_predicted) {
+    RTC_DCHECK_GT(frame.spatial_layer, 0);
+    FrameKey ref_key(frame.picture_id, frame.spatial_layer - 1);
+    if (decoded_frames_.find(ref_key) == decoded_frames_.end())
+      return false;
+  }
+
+  return true;
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/frame_buffer2.h b/chromium/third_party/webrtc/modules/video_coding/frame_buffer2.h
new file mode 100644
index 00000000000..10cae426f62
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/frame_buffer2.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER2_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER2_H_
+
+#include <array>
+#include <map>
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+class VCMJitterEstimator;
+class VCMTiming;
+
+namespace video_coding {
+
+class FrameObject;
+
+class FrameBuffer {
+ public:
+  FrameBuffer(Clock* clock,
+              VCMJitterEstimator* jitter_estimator,
+              const VCMTiming* timing);
+
+  // Insert a frame into the frame buffer.
+  void InsertFrame(std::unique_ptr<FrameObject> frame);
+
+  // Get the next frame for decoding. Will return at latest after
+  // |max_wait_time_ms|, with either a managed FrameObject or an empty
+  // unique ptr if there is no available frame for decoding.
+  std::unique_ptr<FrameObject> NextFrame(int64_t max_wait_time_ms);
+
+ private:
+  // FrameKey is a pair of (picture id, spatial layer).
+  using FrameKey = std::pair<uint16_t, uint8_t>;
+
+  // Comparator used to sort frames, first on their picture id, and second
+  // on their spatial layer.
+  struct FrameComp {
+    bool operator()(const FrameKey& f1, const FrameKey& f2) const;
+  };
+
+  // Determines whether a frame is continuous.
+  bool IsContinuous(const FrameObject& frame) const
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Keep track of decoded frames.
+  std::set<FrameKey, FrameComp> decoded_frames_ GUARDED_BY(crit_);
+
+  // The actual buffer that holds the FrameObjects.
+  std::map<FrameKey, std::unique_ptr<FrameObject>, FrameComp> frames_
+      GUARDED_BY(crit_);
+
+  rtc::CriticalSection crit_;
+  Clock* const clock_;
+  rtc::Event frame_inserted_event_;
+  VCMJitterEstimator* const jitter_estimator_;
+  const VCMTiming* const timing_;
+  int newest_picture_id_ GUARDED_BY(crit_);
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameBuffer);
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER2_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/frame_buffer2_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/frame_buffer2_unittest.cc
new file mode 100644
index 00000000000..67706ce0581
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/frame_buffer2_unittest.cc
@@ -0,0 +1,329 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/frame_buffer2.h"
+
+#include <algorithm>
+#include <cstring>
+#include <limits>
+#include <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/random.h"
+#include "webrtc/modules/video_coding/frame_object.h"
+#include "webrtc/modules/video_coding/jitter_estimator.h"
+#include "webrtc/modules/video_coding/sequence_number_util.h"
+#include "webrtc/modules/video_coding/timing.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class VCMTimingFake : public VCMTiming {
+ public:
+  explicit VCMTimingFake(Clock* clock) : VCMTiming(clock) {}
+
+  int64_t RenderTimeMs(uint32_t frame_timestamp,
+                       int64_t now_ms) const override {
+    if (last_ms_ == -1) {
+      last_ms_ = now_ms + kDelayMs;
+      last_timestamp_ = frame_timestamp;
+    }
+
+    uint32_t diff = MinDiff(frame_timestamp, last_timestamp_);
+    if (AheadOf(frame_timestamp, last_timestamp_))
+      last_ms_ += diff / 90;
+    else
+      last_ms_ -= diff / 90;
+
+    last_timestamp_ = frame_timestamp;
+    return last_ms_;
+  }
+
+  uint32_t MaxWaitingTime(int64_t render_time_ms,
+                          int64_t now_ms) const override {
+    return std::max<int>(0, render_time_ms - now_ms - kDecodeTime);
+  }
+
+ private:
+  static constexpr int kDelayMs = 50;
+  static constexpr int kDecodeTime = kDelayMs / 2;
+  mutable uint32_t last_timestamp_ = 0;
+  mutable int64_t last_ms_ = -1;
+};
+
+class VCMJitterEstimatorMock : public VCMJitterEstimator {
+ public:
+  explicit VCMJitterEstimatorMock(Clock* clock) : VCMJitterEstimator(clock) {}
+
+  MOCK_METHOD1(UpdateRtt, void(int64_t rttMs));
+  MOCK_METHOD3(UpdateEstimate,
+               void(int64_t frameDelayMs,
+                    uint32_t frameSizeBytes,
+                    bool incompleteFrame));
+};
+
+class FrameObjectMock : public FrameObject {
+ public:
+  MOCK_CONST_METHOD1(GetBitstream, bool(uint8_t* destination));
+};
+
+class TestFrameBuffer2 : public ::testing::Test {
+ protected:
+  static constexpr int kMaxReferences = 5;
+  static constexpr int kFps1 = 1000;
+  static constexpr int kFps10 = kFps1 / 10;
+  static constexpr int kFps20 = kFps1 / 20;
+
+  TestFrameBuffer2()
+      : clock_(0),
+        timing_(&clock_),
+        jitter_estimator_(&clock_),
+        buffer_(&clock_, &jitter_estimator_, &timing_),
+        rand_(0x34678213),
+        tear_down_(false),
+        extract_thread_(&ExtractLoop, this, "Extract Thread"),
+        trigger_extract_event_(false, false),
+        crit_acquired_event_(false, false) {}
+
+  void SetUp() override { extract_thread_.Start(); }
+
+  void TearDown() override {
+    tear_down_ = true;
+    trigger_extract_event_.Set();
+    extract_thread_.Stop();
+  }
+
+  template <typename... T>
+  void InsertFrame(uint16_t picture_id,
+                   uint8_t spatial_layer,
+                   int64_t ts_ms,
+                   bool inter_layer_predicted,
+                   T... refs) {
+    static_assert(sizeof...(refs) <= kMaxReferences,
+                  "To many references specified for FrameObject.");
+    std::array<uint16_t, sizeof...(refs)> references = {{refs...}};
+
+    std::unique_ptr<FrameObjectMock> frame(new FrameObjectMock());
+    frame->picture_id = picture_id;
+    frame->spatial_layer = spatial_layer;
+    frame->timestamp = ts_ms * 90;
+    frame->num_references = references.size();
+    frame->inter_layer_predicted = inter_layer_predicted;
+    for (size_t r = 0; r < references.size(); ++r)
+      frame->references[r] = references[r];
+
+    buffer_.InsertFrame(std::move(frame));
+  }
+
+  void ExtractFrame(int64_t max_wait_time = 0) {
+    crit_.Enter();
+    if (max_wait_time == 0) {
+      frames_.emplace_back(buffer_.NextFrame(0));
+      crit_.Leave();
+    } else {
+      max_wait_time_ = max_wait_time;
+      trigger_extract_event_.Set();
+      crit_.Leave();
+      // Make sure |crit_| is aquired by |extract_thread_| before returning.
+      crit_acquired_event_.Wait(rtc::Event::kForever);
+    }
+  }
+
+  void CheckFrame(size_t index, int picture_id, int spatial_layer) {
+    rtc::CritScope lock(&crit_);
+    ASSERT_LT(index, frames_.size());
+    ASSERT_TRUE(frames_[index]);
+    ASSERT_EQ(picture_id, frames_[index]->picture_id);
+    ASSERT_EQ(spatial_layer, frames_[index]->spatial_layer);
+  }
+
+  void CheckNoFrame(size_t index) {
+    rtc::CritScope lock(&crit_);
+    ASSERT_LT(index, frames_.size());
+    ASSERT_FALSE(frames_[index]);
+  }
+
+  static bool ExtractLoop(void* obj) {
+    TestFrameBuffer2* tfb = static_cast<TestFrameBuffer2*>(obj);
+    while (true) {
+      tfb->trigger_extract_event_.Wait(rtc::Event::kForever);
+      {
+        rtc::CritScope lock(&tfb->crit_);
+        tfb->crit_acquired_event_.Set();
+        if (tfb->tear_down_)
+          return false;
+
+        tfb->frames_.emplace_back(tfb->buffer_.NextFrame(tfb->max_wait_time_));
+      }
+    }
+  }
+
+  uint32_t Rand() { return rand_.Rand<uint32_t>(); }
+
+  SimulatedClock clock_;
+  VCMTimingFake timing_;
+  VCMJitterEstimatorMock jitter_estimator_;
+  FrameBuffer buffer_;
+  std::vector<std::unique_ptr<FrameObject>> frames_;
+  Random rand_;
+
+  int64_t max_wait_time_;
+  bool tear_down_;
+  rtc::PlatformThread extract_thread_;
+  rtc::Event trigger_extract_event_;
+  rtc::Event crit_acquired_event_;
+  rtc::CriticalSection crit_;
+};
+
+TEST_F(TestFrameBuffer2, ExtractFromEmptyBuffer) {
+  ExtractFrame();
+  CheckNoFrame(0);
+}
+
+TEST_F(TestFrameBuffer2, WaitForFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  ExtractFrame(20);
+  InsertFrame(pid, 0, ts, false);
+  CheckFrame(0, pid, 0);
+}
+
+TEST_F(TestFrameBuffer2, OneSuperFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  ExtractFrame(20);
+  InsertFrame(pid, 1, ts, true);
+  InsertFrame(pid, 0, ts, false);
+  ExtractFrame();
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid, 1);
+}
+
+TEST_F(TestFrameBuffer2, OneLayerStream) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false);
+  ExtractFrame();
+  CheckFrame(0, pid, 0);
+  for (int i = 1; i < 10; ++i) {
+    InsertFrame(pid + i, 0, ts + i * kFps10, false, pid + i - 1);
+    ExtractFrame();
+    clock_.AdvanceTimeMilliseconds(kFps10);
+    CheckFrame(i, pid + i, 0);
+  }
+}
+
+TEST_F(TestFrameBuffer2, OneLayerStreamReordered) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false);
+  ExtractFrame();
+  CheckFrame(0, pid, 0);
+  for (int i = 1; i < 10; i += 2) {
+    ExtractFrame(15);
+    InsertFrame(pid + i + 1, 0, ts + (i + 1) * kFps10, false, pid + i);
+    clock_.AdvanceTimeMilliseconds(kFps10);
+    InsertFrame(pid + i, 0, ts + i * kFps10, false, pid + i - 1);
+    clock_.AdvanceTimeMilliseconds(kFps10);
+    ExtractFrame();
+    CheckFrame(i, pid + i, 0);
+    CheckFrame(i + 1, pid + i + 1, 0);
+  }
+}
+
+TEST_F(TestFrameBuffer2, DropTemporalLayerSlowDecoder) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false);
+  InsertFrame(pid + 1, 0, ts + kFps20, false);
+  for (int i = 2; i < 10; i += 2) {
+    uint32_t ts_tl0 = ts + i / 2 * kFps10;
+    InsertFrame(pid + i, 0, ts_tl0, false, pid + i - 2);
+    InsertFrame(pid + i + 1, 0, ts_tl0 + kFps20, false, pid + i, pid + i - 1);
+  }
+
+  for (int i = 0; i < 10; ++i) {
+    ExtractFrame();
+    clock_.AdvanceTimeMilliseconds(60);
+  }
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid + 1, 0);
+  CheckFrame(2, pid + 2, 0);
+  CheckFrame(3, pid + 4, 0);
+  CheckFrame(4, pid + 6, 0);
+  CheckFrame(5, pid + 8, 0);
+  CheckNoFrame(6);
+  CheckNoFrame(7);
+  CheckNoFrame(8);
+  CheckNoFrame(9);
+}
+
+TEST_F(TestFrameBuffer2, DropSpatialLayerSlowDecoder) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false);
+  InsertFrame(pid, 1, ts, false);
+  for (int i = 1; i < 6; ++i) {
+    uint32_t ts_tl0 = ts + i * kFps10;
+    InsertFrame(pid + i, 0, ts_tl0, false, pid + i - 1);
+    InsertFrame(pid + i, 1, ts_tl0, false, pid + i - 1);
+  }
+
+  ExtractFrame();
+  ExtractFrame();
+  clock_.AdvanceTimeMilliseconds(55);
+  for (int i = 2; i < 12; ++i) {
+    ExtractFrame();
+    clock_.AdvanceTimeMilliseconds(55);
+  }
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid, 1);
+  CheckFrame(2, pid + 1, 0);
+  CheckFrame(3, pid + 1, 1);
+  CheckFrame(4, pid + 2, 0);
+  CheckFrame(5, pid + 2, 1);
+  CheckFrame(6, pid + 3, 0);
+  CheckFrame(7, pid + 4, 0);
+  CheckFrame(8, pid + 5, 0);
+  CheckNoFrame(9);
+  CheckNoFrame(10);
+  CheckNoFrame(11);
+}
+
+TEST_F(TestFrameBuffer2, InsertLateFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false);
+  ExtractFrame();
+  InsertFrame(pid + 2, 0, ts, false);
+  ExtractFrame();
+  InsertFrame(pid + 1, 0, ts, false, pid);
+  ExtractFrame();
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid + 2, 0);
+  CheckNoFrame(2);
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/frame_object.cc b/chromium/third_party/webrtc/modules/video_coding/frame_object.cc
index 363c8a70357..7b9ec0d2835 100644
--- a/chromium/third_party/webrtc/modules/video_coding/frame_object.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/frame_object.cc
@@ -15,33 +15,56 @@
 namespace webrtc {
 namespace video_coding {
 
+FrameObject::FrameObject()
+    : picture_id(0),
+      spatial_layer(0),
+      timestamp(0),
+      num_references(0),
+      inter_layer_predicted(false) {}
+
 RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
-                               uint16_t picture_id,
-                               uint16_t first_packet,
-                               uint16_t last_packet)
+                               uint16_t first_seq_num,
+                               uint16_t last_seq_num)
     : packet_buffer_(packet_buffer),
-      first_packet_(first_packet),
-      last_packet_(last_packet) {}
+      first_seq_num_(first_seq_num),
+      last_seq_num_(last_seq_num) {
+  VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num);
+  if (packet) {
+    frame_type_ = packet->frameType;
+    codec_type_ = packet->codec;
+  }
+}
 
 RtpFrameObject::~RtpFrameObject() {
   packet_buffer_->ReturnFrame(this);
 }
 
-uint16_t RtpFrameObject::first_packet() const {
-  return first_packet_;
+uint16_t RtpFrameObject::first_seq_num() const {
+  return first_seq_num_;
+}
+
+uint16_t RtpFrameObject::last_seq_num() const {
+  return last_seq_num_;
 }
 
-uint16_t RtpFrameObject::last_packet() const {
-  return last_packet_;
+FrameType RtpFrameObject::frame_type() const {
+  return frame_type_;
 }
 
-uint16_t RtpFrameObject::picture_id() const {
-  return picture_id_;
+VideoCodecType RtpFrameObject::codec_type() const {
+  return codec_type_;
 }
 
 bool RtpFrameObject::GetBitstream(uint8_t* destination) const {
   return packet_buffer_->GetBitstream(*this, destination);
 }
 
+RTPVideoTypeHeader* RtpFrameObject::GetCodecHeader() const {
+  VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
+  if (!packet)
+    return nullptr;
+  return &packet->codecSpecificHeader.codecHeader;
+}
+
 }  // namespace video_coding
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/frame_object.h b/chromium/third_party/webrtc/modules/video_coding/frame_object.h
index 2a68293d638..e8bb4811e6e 100644
--- a/chromium/third_party/webrtc/modules/video_coding/frame_object.h
+++ b/chromium/third_party/webrtc/modules/video_coding/frame_object.h
@@ -11,16 +11,31 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 #define WEBRTC_MODULES_VIDEO_CODING_FRAME_OBJECT_H_
 
-#include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace video_coding {
 
 class FrameObject {
  public:
-  virtual uint16_t picture_id() const = 0;
+  static const uint8_t kMaxFrameReferences = 5;
+
+  FrameObject();
+
   virtual bool GetBitstream(uint8_t* destination) const = 0;
   virtual ~FrameObject() {}
+
+  // The tuple (|picture_id|, |spatial_layer|) uniquely identifies a frame
+  // object. For codec types that don't necessarily have picture ids they
+  // have to be constructed from the header data relevant to that codec.
+  uint16_t picture_id;
+  uint8_t spatial_layer;
+  uint32_t timestamp;
+
+  size_t num_references;
+  uint16_t references[kMaxFrameReferences];
+  bool inter_layer_predicted;
 };
 
 class PacketBuffer;
@@ -28,20 +43,23 @@ class PacketBuffer;
 class RtpFrameObject : public FrameObject {
  public:
   RtpFrameObject(PacketBuffer* packet_buffer,
-                 uint16_t picture_id,
-                 uint16_t first_packet,
-                 uint16_t last_packet);
+                 uint16_t first_seq_num,
+                 uint16_t last_seq_num);
+
   ~RtpFrameObject();
-  uint16_t first_packet() const;
-  uint16_t last_packet() const;
-  uint16_t picture_id() const override;
+  uint16_t first_seq_num() const;
+  uint16_t last_seq_num() const;
+  FrameType frame_type() const;
+  VideoCodecType codec_type() const;
   bool GetBitstream(uint8_t* destination) const override;
+  RTPVideoTypeHeader* GetCodecHeader() const;
 
  private:
   PacketBuffer* packet_buffer_;
-  uint16_t picture_id_;
-  uint16_t first_packet_;
-  uint16_t last_packet_;
+  FrameType frame_type_;
+  VideoCodecType codec_type_;
+  uint16_t first_seq_num_;
+  uint16_t last_seq_num_;
 };
 
 }  // namespace video_coding
diff --git a/chromium/third_party/webrtc/modules/video_coding/generic_encoder.cc b/chromium/third_party/webrtc/modules/video_coding/generic_encoder.cc
index 2463cf5c7a5..abc6369a005 100644
--- a/chromium/third_party/webrtc/modules/video_coding/generic_encoder.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/generic_encoder.cc
@@ -1,12 +1,12 @@
 /*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
+*  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+*
+*  Use of this source code is governed by a BSD-style license
+*  that can be found in the LICENSE file in the root of the source
+*  tree. An additional intellectual property rights grant can be found
+*  in the file PATENTS.  All contributing project authors may
+*  be found in the AUTHORS file in the root of the source tree.
+*/
 
 #include "webrtc/modules/video_coding/generic_encoder.h"
 
@@ -21,89 +21,16 @@
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
-namespace {
-// Map information from info into rtp. If no relevant information is found
-// in info, rtp is set to NULL.
-void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
-  RTC_DCHECK(info);
-  switch (info->codecType) {
-    case kVideoCodecVP8: {
-      rtp->codec = kRtpVideoVp8;
-      rtp->codecHeader.VP8.InitRTPVideoHeaderVP8();
-      rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
-      rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference;
-      rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
-      rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync;
-      rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx;
-      rtp->codecHeader.VP8.keyIdx = info->codecSpecific.VP8.keyIdx;
-      rtp->simulcastIdx = info->codecSpecific.VP8.simulcastIdx;
-      return;
-    }
-    case kVideoCodecVP9: {
-      rtp->codec = kRtpVideoVp9;
-      rtp->codecHeader.VP9.InitRTPVideoHeaderVP9();
-      rtp->codecHeader.VP9.inter_pic_predicted =
-          info->codecSpecific.VP9.inter_pic_predicted;
-      rtp->codecHeader.VP9.flexible_mode =
-          info->codecSpecific.VP9.flexible_mode;
-      rtp->codecHeader.VP9.ss_data_available =
-          info->codecSpecific.VP9.ss_data_available;
-      rtp->codecHeader.VP9.picture_id = info->codecSpecific.VP9.picture_id;
-      rtp->codecHeader.VP9.tl0_pic_idx = info->codecSpecific.VP9.tl0_pic_idx;
-      rtp->codecHeader.VP9.temporal_idx = info->codecSpecific.VP9.temporal_idx;
-      rtp->codecHeader.VP9.spatial_idx = info->codecSpecific.VP9.spatial_idx;
-      rtp->codecHeader.VP9.temporal_up_switch =
-          info->codecSpecific.VP9.temporal_up_switch;
-      rtp->codecHeader.VP9.inter_layer_predicted =
-          info->codecSpecific.VP9.inter_layer_predicted;
-      rtp->codecHeader.VP9.gof_idx = info->codecSpecific.VP9.gof_idx;
-      rtp->codecHeader.VP9.num_spatial_layers =
-          info->codecSpecific.VP9.num_spatial_layers;
-
-      if (info->codecSpecific.VP9.ss_data_available) {
-        rtp->codecHeader.VP9.spatial_layer_resolution_present =
-            info->codecSpecific.VP9.spatial_layer_resolution_present;
-        if (info->codecSpecific.VP9.spatial_layer_resolution_present) {
-          for (size_t i = 0; i < info->codecSpecific.VP9.num_spatial_layers;
-               ++i) {
-            rtp->codecHeader.VP9.width[i] = info->codecSpecific.VP9.width[i];
-            rtp->codecHeader.VP9.height[i] = info->codecSpecific.VP9.height[i];
-          }
-        }
-        rtp->codecHeader.VP9.gof.CopyGofInfoVP9(info->codecSpecific.VP9.gof);
-      }
-
-      rtp->codecHeader.VP9.num_ref_pics = info->codecSpecific.VP9.num_ref_pics;
-      for (int i = 0; i < info->codecSpecific.VP9.num_ref_pics; ++i)
-        rtp->codecHeader.VP9.pid_diff[i] = info->codecSpecific.VP9.p_diff[i];
-      return;
-    }
-    case kVideoCodecH264:
-      rtp->codec = kRtpVideoH264;
-      return;
-    case kVideoCodecGeneric:
-      rtp->codec = kRtpVideoGeneric;
-      rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
-      return;
-    default:
-      return;
-  }
-}
-}  // namespace
-
-// #define DEBUG_ENCODER_BIT_STREAM
-
 VCMGenericEncoder::VCMGenericEncoder(
     VideoEncoder* encoder,
     VideoEncoderRateObserver* rate_observer,
     VCMEncodedFrameCallback* encoded_frame_callback,
-    bool internalSource)
+    bool internal_source)
     : encoder_(encoder),
       rate_observer_(rate_observer),
       vcm_encoded_frame_callback_(encoded_frame_callback),
-      internal_source_(internalSource),
+      internal_source_(internal_source),
       encoder_params_({0, 0, 0, 0}),
-      rotation_(kVideoRotation_0),
       is_screenshare_(false) {}
 
 VCMGenericEncoder::~VCMGenericEncoder() {}
@@ -114,8 +41,8 @@ int32_t VCMGenericEncoder::Release() {
 }
 
 int32_t VCMGenericEncoder::InitEncode(const VideoCodec* settings,
-                                      int32_t numberOfCores,
-                                      size_t maxPayloadSize) {
+                                      int32_t number_of_cores,
+                                      size_t max_payload_size) {
   TRACE_EVENT0("webrtc", "VCMGenericEncoder::InitEncode");
   {
     rtc::CritScope lock(&params_lock_);
@@ -124,7 +51,7 @@ int32_t VCMGenericEncoder::InitEncode(const VideoCodec* settings,
   }
 
   is_screenshare_ = settings->mode == VideoCodecMode::kScreensharing;
-  if (encoder_->InitEncode(settings, numberOfCores, maxPayloadSize) != 0) {
+  if (encoder_->InitEncode(settings, number_of_cores, max_payload_size) != 0) {
     LOG(LS_ERROR) << "Failed to initialize the encoder associated with "
                      "payload name: "
                   << settings->plName;
@@ -134,40 +61,30 @@ int32_t VCMGenericEncoder::InitEncode(const VideoCodec* settings,
   return 0;
 }
 
-int32_t VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
-                                  const CodecSpecificInfo* codecSpecificInfo,
-                                  const std::vector<FrameType>& frameTypes) {
+int32_t VCMGenericEncoder::Encode(const VideoFrame& frame,
+                                  const CodecSpecificInfo* codec_specific,
+                                  const std::vector<FrameType>& frame_types) {
   TRACE_EVENT1("webrtc", "VCMGenericEncoder::Encode", "timestamp",
-               inputFrame.timestamp());
+               frame.timestamp());
 
-  for (FrameType frame_type : frameTypes)
+  for (FrameType frame_type : frame_types)
     RTC_DCHECK(frame_type == kVideoFrameKey || frame_type == kVideoFrameDelta);
 
-  rotation_ = inputFrame.rotation();
-
-  // Keep track of the current frame rotation and apply to the output of the
-  // encoder. There might not be exact as the encoder could have one frame delay
-  // but it should be close enough.
-  // TODO(pbos): Map from timestamp, this is racy (even if rotation_ is locked
-  // properly, which it isn't). More than one frame may be in the pipeline.
-  vcm_encoded_frame_callback_->SetRotation(rotation_);
-
-  int32_t result = encoder_->Encode(inputFrame, codecSpecificInfo, &frameTypes);
-
-  if (vcm_encoded_frame_callback_) {
-    vcm_encoded_frame_callback_->SignalLastEncoderImplementationUsed(
-        encoder_->ImplementationName());
-  }
+  int32_t result = encoder_->Encode(frame, codec_specific, &frame_types);
 
   if (is_screenshare_ &&
       result == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT) {
     // Target bitrate exceeded, encoder state has been reset - try again.
-    return encoder_->Encode(inputFrame, codecSpecificInfo, &frameTypes);
+    return encoder_->Encode(frame, codec_specific, &frame_types);
   }
 
   return result;
 }
 
+const char* VCMGenericEncoder::ImplementationName() const {
+  return encoder_->ImplementationName();
+}
+
 void VCMGenericEncoder::SetEncoderParameters(const EncoderParameters& params) {
   bool channel_parameters_have_changed;
   bool rates_have_changed;
@@ -186,7 +103,7 @@ void VCMGenericEncoder::SetEncoderParameters(const EncoderParameters& params) {
   if (rates_have_changed) {
     uint32_t target_bitrate_kbps = (params.target_bitrate + 500) / 1000;
     encoder_->SetRates(target_bitrate_kbps, params.input_frame_rate);
-    if (rate_observer_ != nullptr) {
+    if (rate_observer_) {
       rate_observer_->OnSetRates(params.target_bitrate,
                                  params.input_frame_rate);
     }
@@ -220,93 +137,32 @@ bool VCMGenericEncoder::SupportsNativeHandle() const {
   return encoder_->SupportsNativeHandle();
 }
 
-int VCMGenericEncoder::GetTargetFramerate() {
-  return encoder_->GetTargetFramerate();
-}
-
-/***************************
- * Callback Implementation
- ***************************/
 VCMEncodedFrameCallback::VCMEncodedFrameCallback(
-    EncodedImageCallback* post_encode_callback)
-    : send_callback_(),
-      _mediaOpt(NULL),
-      _payloadType(0),
-      _internalSource(false),
-      _rotation(kVideoRotation_0),
-      post_encode_callback_(post_encode_callback)
-#ifdef DEBUG_ENCODER_BIT_STREAM
-      ,
-      _bitStreamAfterEncoder(NULL)
-#endif
-{
-#ifdef DEBUG_ENCODER_BIT_STREAM
-  _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
-#endif
-}
+    EncodedImageCallback* post_encode_callback,
+    media_optimization::MediaOptimization* media_opt)
+    : internal_source_(false),
+      post_encode_callback_(post_encode_callback),
+      media_opt_(media_opt) {}
 
-VCMEncodedFrameCallback::~VCMEncodedFrameCallback() {
-#ifdef DEBUG_ENCODER_BIT_STREAM
-  fclose(_bitStreamAfterEncoder);
-#endif
-}
-
-int32_t VCMEncodedFrameCallback::SetTransportCallback(
-    VCMPacketizationCallback* transport) {
-  send_callback_ = transport;
-  return VCM_OK;
-}
+VCMEncodedFrameCallback::~VCMEncodedFrameCallback() {}
 
 int32_t VCMEncodedFrameCallback::Encoded(
     const EncodedImage& encoded_image,
-    const CodecSpecificInfo* codecSpecificInfo,
-    const RTPFragmentationHeader* fragmentationHeader) {
+    const CodecSpecificInfo* codec_specific,
+    const RTPFragmentationHeader* fragmentation_header) {
   TRACE_EVENT_INSTANT1("webrtc", "VCMEncodedFrameCallback::Encoded",
                        "timestamp", encoded_image._timeStamp);
-  post_encode_callback_->Encoded(encoded_image, NULL, NULL);
-
-  if (send_callback_ == NULL) {
-    return VCM_UNINITIALIZED;
-  }
-
-#ifdef DEBUG_ENCODER_BIT_STREAM
-  if (_bitStreamAfterEncoder != NULL) {
-    fwrite(encoded_image._buffer, 1, encoded_image._length,
-           _bitStreamAfterEncoder);
-  }
-#endif
-
-  RTPVideoHeader rtpVideoHeader;
-  memset(&rtpVideoHeader, 0, sizeof(RTPVideoHeader));
-  RTPVideoHeader* rtpVideoHeaderPtr = &rtpVideoHeader;
-  if (codecSpecificInfo) {
-    CopyCodecSpecific(codecSpecificInfo, rtpVideoHeaderPtr);
-  }
-  rtpVideoHeader.rotation = _rotation;
-
-  int32_t callbackReturn = send_callback_->SendData(
-      _payloadType, encoded_image, fragmentationHeader, rtpVideoHeaderPtr);
-  if (callbackReturn < 0) {
-    return callbackReturn;
-  }
-
-  if (_mediaOpt != NULL) {
-    _mediaOpt->UpdateWithEncodedData(encoded_image);
-    if (_internalSource)
-      return _mediaOpt->DropFrame();  // Signal to encoder to drop next frame.
+  int ret_val = post_encode_callback_->Encoded(encoded_image, codec_specific,
+                                               fragmentation_header);
+  if (ret_val < 0)
+    return ret_val;
+
+  if (media_opt_) {
+    media_opt_->UpdateWithEncodedData(encoded_image);
+    if (internal_source_)
+      return media_opt_->DropFrame();  // Signal to encoder to drop next frame.
   }
   return VCM_OK;
 }
 
-void VCMEncodedFrameCallback::SetMediaOpt(
-    media_optimization::MediaOptimization* mediaOpt) {
-  _mediaOpt = mediaOpt;
-}
-
-void VCMEncodedFrameCallback::SignalLastEncoderImplementationUsed(
-    const char* implementation_name) {
-  if (send_callback_)
-    send_callback_->OnEncoderImplementationName(implementation_name);
-}
-
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/generic_encoder.h b/chromium/third_party/webrtc/modules/video_coding/generic_encoder.h
index e96d9957815..469f04ded40 100644
--- a/chromium/third_party/webrtc/modules/video_coding/generic_encoder.h
+++ b/chromium/third_party/webrtc/modules/video_coding/generic_encoder.h
@@ -33,60 +33,26 @@ struct EncoderParameters {
   uint32_t input_frame_rate;
 };
 
-/*************************************/
-/* VCMEncodeFrameCallback class     */
-/***********************************/
 class VCMEncodedFrameCallback : public EncodedImageCallback {
  public:
-    explicit VCMEncodedFrameCallback(
-      EncodedImageCallback* post_encode_callback);
-    virtual ~VCMEncodedFrameCallback();
-
-    /*
-    * Callback implementation - codec encode complete
-    */
-    int32_t Encoded(
-        const EncodedImage& encodedImage,
-        const CodecSpecificInfo* codecSpecificInfo = NULL,
-        const RTPFragmentationHeader* fragmentationHeader = NULL);
-    /*
-    * Callback implementation - generic encoder encode complete
-    */
-    int32_t SetTransportCallback(VCMPacketizationCallback* transport);
-    /**
-    * Set media Optimization
-    */
-    void SetMediaOpt(media_optimization::MediaOptimization* mediaOpt);
-
-    void SetPayloadType(uint8_t payloadType) {
-      _payloadType = payloadType;
-    }
-
-    void SetInternalSource(bool internalSource) {
-      _internalSource = internalSource;
-    }
-
-    void SetRotation(VideoRotation rotation) { _rotation = rotation; }
-    void SignalLastEncoderImplementationUsed(
-        const char* encoder_implementation_name);
+  VCMEncodedFrameCallback(EncodedImageCallback* post_encode_callback,
+                          media_optimization::MediaOptimization* media_opt);
+  virtual ~VCMEncodedFrameCallback();
+
+  // Implements EncodedImageCallback.
+  int32_t Encoded(const EncodedImage& encoded_image,
+                  const CodecSpecificInfo* codec_specific,
+                  const RTPFragmentationHeader* fragmentation_header) override;
+  void SetInternalSource(bool internal_source) {
+    internal_source_ = internal_source;
+  }
 
  private:
-    VCMPacketizationCallback* send_callback_;
-    media_optimization::MediaOptimization* _mediaOpt;
-    uint8_t _payloadType;
-    bool _internalSource;
-    VideoRotation _rotation;
-
-    EncodedImageCallback* post_encode_callback_;
-
-#ifdef DEBUG_ENCODER_BIT_STREAM
-    FILE* _bitStreamAfterEncoder;
-#endif
-};  // end of VCMEncodeFrameCallback class
-
-/******************************/
-/* VCMGenericEncoder class    */
-/******************************/
+  bool internal_source_;
+  EncodedImageCallback* const post_encode_callback_;
+  media_optimization::MediaOptimization* const media_opt_;
+};
+
 class VCMGenericEncoder {
   friend class VCMCodecDataBase;
 
@@ -94,44 +60,27 @@ class VCMGenericEncoder {
   VCMGenericEncoder(VideoEncoder* encoder,
                     VideoEncoderRateObserver* rate_observer,
                     VCMEncodedFrameCallback* encoded_frame_callback,
-                    bool internalSource);
+                    bool internal_source);
   ~VCMGenericEncoder();
-  /**
-  * Free encoder memory
-  */
   int32_t Release();
-  /**
-  * Initialize the encoder with the information from the VideoCodec
-  */
   int32_t InitEncode(const VideoCodec* settings,
-                     int32_t numberOfCores,
-                     size_t maxPayloadSize);
-  /**
-  * Encode raw image
-  * inputFrame        : Frame containing raw image
-  * codecSpecificInfo : Specific codec data
-  * cameraFrameRate   : Request or information from the remote side
-  * frameType         : The requested frame type to encode
-  */
-  int32_t Encode(const VideoFrame& inputFrame,
-                 const CodecSpecificInfo* codecSpecificInfo,
-                 const std::vector<FrameType>& frameTypes);
+                     int32_t number_of_cores,
+                     size_t max_payload_size);
+  int32_t Encode(const VideoFrame& frame,
+                 const CodecSpecificInfo* codec_specific,
+                 const std::vector<FrameType>& frame_types);
+
+  const char* ImplementationName() const;
 
   void SetEncoderParameters(const EncoderParameters& params);
   EncoderParameters GetEncoderParameters() const;
 
   int32_t SetPeriodicKeyFrames(bool enable);
-
   int32_t RequestFrame(const std::vector<FrameType>& frame_types);
-
   bool InternalSource() const;
-
   void OnDroppedFrame();
-
   bool SupportsNativeHandle() const;
 
-  int GetTargetFramerate();
-
  private:
   VideoEncoder* const encoder_;
   VideoEncoderRateObserver* const rate_observer_;
@@ -139,9 +88,8 @@ class VCMGenericEncoder {
   const bool internal_source_;
   rtc::CriticalSection params_lock_;
   EncoderParameters encoder_params_ GUARDED_BY(params_lock_);
-  VideoRotation rotation_;
   bool is_screenshare_;
-};  // end of VCMGenericEncoder class
+};
 
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/include/video_coding.h b/chromium/third_party/webrtc/modules/video_coding/include/video_coding.h
index 0c508b7739a..0f8567963a0 100644
--- a/chromium/third_party/webrtc/modules/video_coding/include/video_coding.h
+++ b/chromium/third_party/webrtc/modules/video_coding/include/video_coding.h
@@ -31,6 +31,10 @@ namespace webrtc {
 
 class Clock;
 class EncodedImageCallback;
+// TODO(pbos): Remove VCMQMSettingsCallback completely. This might be done by
+// removing the VCM and use VideoSender/VideoReceiver as a public interface
+// directly.
+class VCMQMSettingsCallback;
 class VideoEncoder;
 class VideoDecoder;
 struct CodecSpecificInfo;
@@ -78,7 +82,8 @@ class VideoCodingModule : public Module {
       VideoEncoderRateObserver* encoder_rate_observer,
       VCMQMSettingsCallback* qm_settings_callback,
       NackSender* nack_sender,
-      KeyFrameRequestSender* keyframe_request_sender);
+      KeyFrameRequestSender* keyframe_request_sender,
+      EncodedImageCallback* pre_decode_image_callback);
 
   static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory);
 
@@ -183,32 +188,6 @@ class VideoCodingModule : public Module {
   //                     < 0,    on error.
   virtual int32_t SetReceiveChannelParameters(int64_t rtt) = 0;
 
-  // Register a transport callback which will be called to deliver the encoded
-  // data and
-  // side information.
-  //
-  // Input:
-  //      - transport  : The callback object to register.
-  //
-  // Return value      : VCM_OK, on success.
-  //                     < 0,    on error.
-  virtual int32_t RegisterTransportCallback(
-      VCMPacketizationCallback* transport) = 0;
-
-  // Register video output information callback which will be called to deliver
-  // information
-  // about the video stream produced by the encoder, for instance the average
-  // frame rate and
-  // bit rate.
-  //
-  // Input:
-  //      - outputInformation  : The callback object to register.
-  //
-  // Return value      : VCM_OK, on success.
-  //                     < 0,    on error.
-  virtual int32_t RegisterSendStatisticsCallback(
-      VCMSendStatisticsCallback* sendStats) = 0;
-
   // Register a video protection callback which will be called to deliver
   // the requested FEC rate and NACK status (on/off).
   //
@@ -248,14 +227,13 @@ class VideoCodingModule : public Module {
   //                     < 0,    on error.
   virtual int32_t AddVideoFrame(
       const VideoFrame& videoFrame,
-      const VideoContentMetrics* contentMetrics = NULL,
       const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
 
   // Next frame encoded should be an intra frame (keyframe).
   //
   // Return value      : VCM_OK, on success.
   //                     < 0,    on error.
-  virtual int32_t IntraFrameRequest(int stream_index) = 0;
+  virtual int32_t IntraFrameRequest(size_t stream_index) = 0;
 
   // Frame Dropper enable. Can be used to disable the frame dropping when the
   // encoder
@@ -391,10 +369,6 @@ class VideoCodingModule : public Module {
   //                     < 0,    on error.
   virtual int32_t Decode(uint16_t maxWaitTimeMs = 200) = 0;
 
-  // Registers a callback which conveys the size of the render buffer.
-  virtual int RegisterRenderBufferSizeCallback(
-      VCMRenderBufferSizeCallback* callback) = 0;
-
   // API to get the codec which is currently used for decoding by the module.
   //
   // Input:
@@ -511,8 +485,6 @@ class VideoCodingModule : public Module {
   // suspended due to bandwidth limitations; otherwise false.
   virtual bool VideoSuspended() const = 0;
 
-  virtual void RegisterPreDecodeImageCallback(
-      EncodedImageCallback* observer) = 0;
   virtual void RegisterPostEncodeImageCallback(
       EncodedImageCallback* post_encode_callback) = 0;
   // Releases pending decode calls, permitting faster thread shutdown.
diff --git a/chromium/third_party/webrtc/modules/video_coding/include/video_coding_defines.h b/chromium/third_party/webrtc/modules/video_coding/include/video_coding_defines.h
index 4fe8c797933..ba71803c7c9 100644
--- a/chromium/third_party/webrtc/modules/video_coding/include/video_coding_defines.h
+++ b/chromium/third_party/webrtc/modules/video_coding/include/video_coding_defines.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
 #define WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
 
+#include <string>
 #include <vector>
 
 #include "webrtc/modules/include/module_common_types.h"
@@ -56,20 +57,6 @@ struct VCMFrameCount {
   uint32_t numDeltaFrames;
 };
 
-// Callback class used for sending data ready to be packetized
-class VCMPacketizationCallback {
- public:
-  virtual int32_t SendData(uint8_t payloadType,
-                           const EncodedImage& encoded_image,
-                           const RTPFragmentationHeader* fragmentationHeader,
-                           const RTPVideoHeader* rtpVideoHdr) = 0;
-
-  virtual void OnEncoderImplementationName(const char* implementation_name) {}
-
- protected:
-  virtual ~VCMPacketizationCallback() {}
-};
-
 // Callback class used for passing decoded frames which are ready to be
 // rendered.
 class VCMReceiveCallback {
@@ -86,13 +73,13 @@ class VCMReceiveCallback {
   virtual ~VCMReceiveCallback() {}
 };
 
-// Callback class used for informing the user of the bit rate and frame rate
-// produced by the
-// encoder.
+// Callback class used for informing the user of the bit rate and frame rate,
+// and the name of the encoder.
 class VCMSendStatisticsCallback {
  public:
-  virtual int32_t SendStatistics(const uint32_t bitRate,
-                                 const uint32_t frameRate) = 0;
+  virtual void SendStatistics(uint32_t bitRate,
+                              uint32_t frameRate,
+                              const std::string& encoder_name) = 0;
 
  protected:
   virtual ~VCMSendStatisticsCallback() {}
@@ -189,30 +176,6 @@ class KeyFrameRequestSender {
   virtual ~KeyFrameRequestSender() {}
 };
 
-// Callback used to inform the user of the the desired resolution
-// as subscribed by Media Optimization (Quality Modes)
-class VCMQMSettingsCallback {
- public:
-  virtual int32_t SetVideoQMSettings(const uint32_t frameRate,
-                                     const uint32_t width,
-                                     const uint32_t height) = 0;
-
-  virtual void SetTargetFramerate(int frame_rate) = 0;
-
- protected:
-  virtual ~VCMQMSettingsCallback() {}
-};
-
-// Callback class used for telling the user about the size (in time) of the
-// render buffer, that is the size in time of the complete continuous frames.
-class VCMRenderBufferSizeCallback {
- public:
-  virtual void RenderBufferSizeMs(int buffer_size_ms) = 0;
-
- protected:
-  virtual ~VCMRenderBufferSizeCallback() {}
-};
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.cc b/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.cc
index f048b0a883c..9c50a945191 100644
--- a/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.cc
@@ -601,7 +601,8 @@ VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) {
   // Frame pulled out from jitter buffer, update the jitter estimate.
   const bool retransmitted = (frame->GetNackCount() > 0);
   if (retransmitted) {
-    jitter_estimate_.FrameNacked();
+    if (WaitForRetransmissions())
+      jitter_estimate_.FrameNacked();
   } else if (frame->Length() > 0) {
     // Ignore retransmitted and empty frames.
     if (waiting_for_completion_.latest_packet_time >= 0) {
@@ -958,6 +959,8 @@ void VCMJitterBuffer::UpdateRtt(int64_t rtt_ms) {
   jitter_estimate_.UpdateRtt(rtt_ms);
   if (nack_module_)
     nack_module_->UpdateRtt(rtt_ms);
+  if (!WaitForRetransmissions())
+    jitter_estimate_.ResetNackCount();
 }
 
 void VCMJitterBuffer::SetNackMode(VCMNackMode mode,
@@ -1194,19 +1197,6 @@ int64_t VCMJitterBuffer::LastDecodedTimestamp() const {
   return last_decoded_state_.time_stamp();
 }
 
-void VCMJitterBuffer::RenderBufferSize(uint32_t* timestamp_start,
-                                       uint32_t* timestamp_end) {
-  CriticalSectionScoped cs(crit_sect_);
-  CleanUpOldOrEmptyFrames();
-  *timestamp_start = 0;
-  *timestamp_end = 0;
-  if (decodable_frames_.empty()) {
-    return;
-  }
-  *timestamp_start = decodable_frames_.Front()->TimeStamp();
-  *timestamp_end = decodable_frames_.Back()->TimeStamp();
-}
-
 void VCMJitterBuffer::RegisterStatsCallback(
     VCMReceiveStatisticsCallback* callback) {
   CriticalSectionScoped cs(crit_sect_);
@@ -1282,9 +1272,13 @@ void VCMJitterBuffer::CountFrame(const VCMFrameBuffer& frame) {
   if (frame.IsSessionComplete()) {
     if (frame.FrameType() == kVideoFrameKey) {
       ++receive_statistics_.key_frames;
+      if (receive_statistics_.key_frames == 1) {
+        LOG(LS_INFO) << "Received first complete key frame";
+      }
     } else {
       ++receive_statistics_.delta_frames;
     }
+
     if (stats_callback_ != NULL)
       stats_callback_->OnFrameCountsUpdated(receive_statistics_);
   }
diff --git a/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.h b/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.h
index 0cc03dd8107..e36f2cd6d52 100644
--- a/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.h
+++ b/chromium/third_party/webrtc/modules/video_coding/jitter_buffer.h
@@ -212,10 +212,6 @@ class VCMJitterBuffer {
   int64_t LastDecodedTimestamp() const;
   VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
 
-  // Used to compute time of complete continuous frames. Returns the timestamps
-  // corresponding to the start and end of the continuous complete buffer.
-  void RenderBufferSize(uint32_t* timestamp_start, uint32_t* timestamp_end);
-
   void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
 
   int64_t TimeUntilNextProcess();
diff --git a/chromium/third_party/webrtc/modules/video_coding/jitter_buffer_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/jitter_buffer_unittest.cc
index df70ea98261..af9c20aaefe 100644
--- a/chromium/third_party/webrtc/modules/video_coding/jitter_buffer_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/jitter_buffer_unittest.cc
@@ -195,7 +195,7 @@ class ProcessThreadMock : public ProcessThread {
   MOCK_METHOD1(WakeUp, void(Module* module));
   MOCK_METHOD1(RegisterModule, void(Module* module));
   MOCK_METHOD1(DeRegisterModule, void(Module* module));
-  void PostTask(rtc::scoped_ptr<ProcessTask> task) {}
+  void PostTask(std::unique_ptr<ProcessTask> task) {}
 };
 
 class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
@@ -215,7 +215,7 @@ class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
 
  protected:
   TestBasicJitterBuffer() : scoped_field_trial_(GetParam()) {}
-  virtual void SetUp() {
+  void SetUp() override {
     clock_.reset(new SimulatedClock(0));
     jitter_buffer_.reset(new VCMJitterBuffer(
         clock_.get(),
diff --git a/chromium/third_party/webrtc/modules/video_coding/media_opt_util.cc b/chromium/third_party/webrtc/modules/video_coding/media_opt_util.cc
index 69cf757f2b4..42db2facf1c 100644
--- a/chromium/third_party/webrtc/modules/video_coding/media_opt_util.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/media_opt_util.cc
@@ -34,19 +34,10 @@ VCMProtectionMethod::VCMProtectionMethod()
       _protectionFactorD(0),
       _scaleProtKey(2.0f),
       _maxPayloadSize(1460),
-      _qmRobustness(new VCMQmRobustness()),
-      _useUepProtectionK(false),
-      _useUepProtectionD(true),
       _corrFecCost(1.0),
       _type(kNone) {}
 
-VCMProtectionMethod::~VCMProtectionMethod() {
-  delete _qmRobustness;
-}
-void VCMProtectionMethod::UpdateContentMetrics(
-    const VideoContentMetrics* contentMetrics) {
-  _qmRobustness->UpdateContent(contentMetrics);
-}
+VCMProtectionMethod::~VCMProtectionMethod() {}
 
 VCMNackFecMethod::VCMNackFecMethod(int64_t lowRttNackThresholdMs,
                                    int64_t highRttNackThresholdMs)
@@ -333,17 +324,6 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
     codeRateDelta = kPacketLossMax - 1;
   }
 
-  float adjustFec = 1.0f;
-  // Avoid additional adjustments when layers are active.
-  // TODO(mikhal/marco): Update adjusmtent based on layer info.
-  if (parameters->numLayers == 1) {
-    adjustFec = _qmRobustness->AdjustFecFactor(
-        codeRateDelta, parameters->bitRate, parameters->frameRate,
-        parameters->rtt, packetLoss);
-  }
-
-  codeRateDelta = static_cast<uint8_t>(codeRateDelta * adjustFec);
-
   // For Key frame:
   // Effectively at a higher rate, so we scale/boost the rate
   // The boost factor may depend on several factors: ratio of packet
@@ -411,13 +391,6 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
     _corrFecCost = 0.0f;
   }
 
-  // TODO(marpan): Set the UEP protection on/off for Key and Delta frames
-  _useUepProtectionK = _qmRobustness->SetUepProtection(
-      codeRateKey, parameters->bitRate, packetLoss, 0);
-
-  _useUepProtectionD = _qmRobustness->SetUepProtection(
-      codeRateDelta, parameters->bitRate, packetLoss, 1);
-
   // DONE WITH FEC PROTECTION SETTINGS
   return true;
 }
diff --git a/chromium/third_party/webrtc/modules/video_coding/media_opt_util.h b/chromium/third_party/webrtc/modules/video_coding/media_opt_util.h
index 6b47e3b2d99..ad314aca8c2 100644
--- a/chromium/third_party/webrtc/modules/video_coding/media_opt_util.h
+++ b/chromium/third_party/webrtc/modules/video_coding/media_opt_util.h
@@ -18,7 +18,6 @@
 
 #include "webrtc/base/exp_filter.h"
 #include "webrtc/modules/video_coding/internal_defines.h"
-#include "webrtc/modules/video_coding/qm_select.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/typedefs.h"
 
@@ -45,6 +44,10 @@ enum FilterPacketLossMode {
 // common to media optimization and the jitter buffer.
 const int64_t kLowRttNackMs = 20;
 
+// If the RTT is higher than this an extra RTT wont be added to to the jitter
+// buffer delay.
+const int kMaxRttDelayThreshold = 500;
+
 struct VCMProtectionParameters {
   VCMProtectionParameters()
       : rtt(0),
@@ -138,9 +141,6 @@ class VCMProtectionMethod {
 
   virtual int MaxFramesFec() const { return 1; }
 
-  // Updates content metrics
-  void UpdateContentMetrics(const VideoContentMetrics* contentMetrics);
-
  protected:
   uint8_t _effectivePacketLoss;
   uint8_t _protectionFactorK;
@@ -149,7 +149,6 @@ class VCMProtectionMethod {
   float _scaleProtKey;
   int32_t _maxPayloadSize;
 
-  VCMQmRobustness* _qmRobustness;
   bool _useUepProtectionK;
   bool _useUepProtectionD;
   float _corrFecCost;
diff --git a/chromium/third_party/webrtc/modules/video_coding/media_optimization.cc b/chromium/third_party/webrtc/modules/video_coding/media_optimization.cc
index a234a06f9b4..d5fbadc122f 100644
--- a/chromium/third_party/webrtc/modules/video_coding/media_optimization.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/media_optimization.cc
@@ -11,8 +11,6 @@
 #include "webrtc/modules/video_coding/media_optimization.h"
 
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/video_coding/content_metrics_processing.h"
-#include "webrtc/modules/video_coding/qm_select.h"
 #include "webrtc/modules/video_coding/utility/frame_dropper.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
@@ -33,13 +31,6 @@ void UpdateProtectionCallback(
   // Get the FEC code rate for Delta frames (set to 0 when NA).
   delta_fec_params.fec_rate = selected_method->RequiredProtectionFactorD();
 
-  // Get the FEC-UEP protection status for Key frames: UEP on/off.
-  key_fec_params.use_uep_protection = selected_method->RequiredUepProtectionK();
-
-  // Get the FEC-UEP protection status for Delta frames: UEP on/off.
-  delta_fec_params.use_uep_protection =
-      selected_method->RequiredUepProtectionD();
-
   // The RTP module currently requires the same |max_fec_frames| for both
   // key and delta frames.
   delta_fec_params.max_fec_frames = selected_method->MaxFramesFec();
@@ -88,16 +79,11 @@ MediaOptimization::MediaOptimization(Clock* clock)
       max_payload_size_(1460),
       video_target_bitrate_(0),
       incoming_frame_rate_(0),
-      enable_qm_(false),
       encoded_frame_samples_(),
       avg_sent_bit_rate_bps_(0),
       avg_sent_framerate_(0),
       key_frame_cnt_(0),
       delta_frame_cnt_(0),
-      content_(new VCMContentMetricsProcessing()),
-      qm_resolution_(new VCMQmResolution()),
-      last_qm_update_time_(0),
-      last_change_time_(0),
       num_layers_(0),
       suspension_enabled_(false),
       video_suspended_(false),
@@ -120,8 +106,6 @@ void MediaOptimization::Reset() {
   frame_dropper_->Reset();
   loss_prot_logic_->Reset(clock_->TimeInMilliseconds());
   frame_dropper_->SetRates(0, 0);
-  content_->Reset();
-  qm_resolution_->Reset();
   loss_prot_logic_->UpdateFrameRate(incoming_frame_rate_);
   loss_prot_logic_->Reset(clock_->TimeInMilliseconds());
   send_statistics_zero_encode_ = 0;
@@ -131,8 +115,6 @@ void MediaOptimization::Reset() {
   user_frame_rate_ = 0;
   key_frame_cnt_ = 0;
   delta_frame_cnt_ = 0;
-  last_qm_update_time_ = 0;
-  last_change_time_ = 0;
   encoded_frame_samples_.clear();
   avg_sent_bit_rate_bps_ = 0;
   num_layers_ = 1;
@@ -160,12 +142,7 @@ void MediaOptimization::SetEncodingDataInternal(VideoCodecType send_codec_type,
                                                 int num_layers,
                                                 int32_t mtu) {
   // Everything codec specific should be reset here since this means the codec
-  // has changed. If native dimension values have changed, then either user
-  // initiated change, or QM initiated change. Will be able to determine only
-  // after the processing of the first frame.
-  last_change_time_ = clock_->TimeInMilliseconds();
-  content_->Reset();
-  content_->UpdateFrameRate(frame_rate);
+  // has changed.
 
   max_bit_rate_ = max_bit_rate;
   send_codec_type_ = send_codec_type;
@@ -182,16 +159,13 @@ void MediaOptimization::SetEncodingDataInternal(VideoCodecType send_codec_type,
   codec_height_ = height;
   num_layers_ = (num_layers <= 1) ? 1 : num_layers;  // Can also be zero.
   max_payload_size_ = mtu;
-  qm_resolution_->Initialize(target_bitrate_kbps, user_frame_rate_,
-                             codec_width_, codec_height_, num_layers_);
 }
 
 uint32_t MediaOptimization::SetTargetRates(
     uint32_t target_bitrate,
     uint8_t fraction_lost,
     int64_t round_trip_time_ms,
-    VCMProtectionCallback* protection_callback,
-    VCMQMSettingsCallback* qmsettings_callback) {
+    VCMProtectionCallback* protection_callback) {
   CriticalSectionScoped lock(crit_sect_.get());
   VCMProtectionMethod* selected_method = loss_prot_logic_->SelectedMethod();
   float target_bitrate_kbps = static_cast<float>(target_bitrate) / 1000.0f;
@@ -227,11 +201,7 @@ uint32_t MediaOptimization::SetTargetRates(
   float protection_overhead_rate = 0.0f;
 
   // Update protection settings, when applicable.
-  float sent_video_rate_kbps = 0.0f;
   if (loss_prot_logic_->SelectedType() != kNone) {
-    // Update protection method with content metrics.
-    selected_method->UpdateContentMetrics(content_->ShortTermAvgData());
-
     // Update method will compute the robustness settings for the given
     // protection method and the overhead cost
     // the protection method is set by the user via SetVideoProtection.
@@ -265,7 +235,6 @@ uint32_t MediaOptimization::SetTargetRates(
     // Get the effective packet loss for encoder ER when applicable. Should be
     // passed to encoder via fraction_lost.
     packet_loss_enc = selected_method->RequiredPacketLossER();
-    sent_video_rate_kbps = static_cast<float>(sent_video_rate_bps) / 1000.0f;
   }
 
   // Source coding rate: total rate - protection overhead.
@@ -281,19 +250,6 @@ uint32_t MediaOptimization::SetTargetRates(
       static_cast<float>(video_target_bitrate_) / 1000.0f;
   frame_dropper_->SetRates(target_video_bitrate_kbps, incoming_frame_rate_);
 
-  if (enable_qm_ && qmsettings_callback) {
-    // Update QM with rates.
-    qm_resolution_->UpdateRates(target_video_bitrate_kbps, sent_video_rate_kbps,
-                                incoming_frame_rate_, fraction_lost_);
-    // Check for QM selection.
-    bool select_qm = CheckStatusForQMchange();
-    if (select_qm) {
-      SelectQuality(qmsettings_callback);
-    }
-    // Reset the short-term averaged content data.
-    content_->ResetShortTermAvgData();
-  }
-
   CheckSuspendConditions();
 
   return video_target_bitrate_;
@@ -367,11 +323,6 @@ int32_t MediaOptimization::UpdateWithEncodedData(
         loss_prot_logic_->UpdatePacketsPerFrameKey(
             min_packets_per_frame, clock_->TimeInMilliseconds());
       }
-
-      if (enable_qm_) {
-        // Update quality select with encoded length.
-        qm_resolution_->UpdateEncodedSize(encoded_length);
-      }
     }
     if (!delta_frame && encoded_length > 0) {
       loss_prot_logic_->UpdateKeyFrameSize(static_cast<float>(encoded_length));
@@ -388,11 +339,6 @@ int32_t MediaOptimization::UpdateWithEncodedData(
   return VCM_OK;
 }
 
-void MediaOptimization::EnableQM(bool enable) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  enable_qm_ = enable;
-}
-
 void MediaOptimization::EnableFrameDropper(bool enable) {
   CriticalSectionScoped lock(crit_sect_.get());
   frame_dropper_->Enable(enable);
@@ -424,19 +370,6 @@ bool MediaOptimization::DropFrame() {
   return frame_dropper_->DropFrame();
 }
 
-void MediaOptimization::UpdateContentData(
-    const VideoContentMetrics* content_metrics) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  // Updating content metrics.
-  if (content_metrics == NULL) {
-    // Disable QM if metrics are NULL.
-    enable_qm_ = false;
-    qm_resolution_->Reset();
-  } else {
-    content_->UpdateContentData(content_metrics);
-  }
-}
-
 void MediaOptimization::UpdateIncomingFrameRate() {
   int64_t now = clock_->TimeInMilliseconds();
   if (incoming_frame_times_[0] == 0) {
@@ -451,36 +384,6 @@ void MediaOptimization::UpdateIncomingFrameRate() {
   ProcessIncomingFrameRate(now);
 }
 
-int32_t MediaOptimization::SelectQuality(
-    VCMQMSettingsCallback* video_qmsettings_callback) {
-  // Reset quantities for QM select.
-  qm_resolution_->ResetQM();
-
-  // Update QM will long-term averaged content metrics.
-  qm_resolution_->UpdateContent(content_->LongTermAvgData());
-
-  // Select quality mode.
-  VCMResolutionScale* qm = NULL;
-  int32_t ret = qm_resolution_->SelectResolution(&qm);
-  if (ret < 0) {
-    return ret;
-  }
-
-  // Check for updates to spatial/temporal modes.
-  QMUpdate(qm, video_qmsettings_callback);
-
-  // Reset all the rate and related frame counters quantities.
-  qm_resolution_->ResetRates();
-
-  // Reset counters.
-  last_qm_update_time_ = clock_->TimeInMilliseconds();
-
-  // Reset content metrics.
-  content_->Reset();
-
-  return VCM_OK;
-}
-
 void MediaOptimization::PurgeOldFrameSamples(int64_t now_ms) {
   while (!encoded_frame_samples_.empty()) {
     if (now_ms - encoded_frame_samples_.front().time_complete_ms >
@@ -527,65 +430,6 @@ void MediaOptimization::UpdateSentFramerate() {
   }
 }
 
-bool MediaOptimization::QMUpdate(
-    VCMResolutionScale* qm,
-    VCMQMSettingsCallback* video_qmsettings_callback) {
-  // Check for no change.
-  if (!qm->change_resolution_spatial && !qm->change_resolution_temporal) {
-    return false;
-  }
-
-  // Check for change in frame rate.
-  if (qm->change_resolution_temporal) {
-    incoming_frame_rate_ = qm->frame_rate;
-    // Reset frame rate estimate.
-    memset(incoming_frame_times_, -1, sizeof(incoming_frame_times_));
-  }
-
-  // Check for change in frame size.
-  if (qm->change_resolution_spatial) {
-    codec_width_ = qm->codec_width;
-    codec_height_ = qm->codec_height;
-  }
-
-  LOG(LS_INFO) << "Media optimizer requests the video resolution to be changed "
-                  "to "
-               << qm->codec_width << "x" << qm->codec_height << "@"
-               << qm->frame_rate;
-
-  // Update VPM with new target frame rate and frame size.
-  // Note: use |qm->frame_rate| instead of |_incoming_frame_rate| for updating
-  // target frame rate in VPM frame dropper. The quantity |_incoming_frame_rate|
-  // will vary/fluctuate, and since we don't want to change the state of the
-  // VPM frame dropper, unless a temporal action was selected, we use the
-  // quantity |qm->frame_rate| for updating.
-  video_qmsettings_callback->SetVideoQMSettings(qm->frame_rate, codec_width_,
-                                                codec_height_);
-  content_->UpdateFrameRate(qm->frame_rate);
-  qm_resolution_->UpdateCodecParameters(qm->frame_rate, codec_width_,
-                                        codec_height_);
-  return true;
-}
-
-// Check timing constraints and look for significant change in:
-// (1) scene content,
-// (2) target bit rate.
-bool MediaOptimization::CheckStatusForQMchange() {
-  bool status = true;
-
-  // Check that we do not call QMSelect too often, and that we waited some time
-  // (to sample the metrics) from the event last_change_time
-  // last_change_time is the time where user changed the size/rate/frame rate
-  // (via SetEncodingData).
-  int64_t now = clock_->TimeInMilliseconds();
-  if ((now - last_qm_update_time_) < kQmMinIntervalMs ||
-      (now - last_change_time_) < kQmMinIntervalMs) {
-    status = false;
-  }
-
-  return status;
-}
-
 // Allowing VCM to keep track of incoming frame rate.
 void MediaOptimization::ProcessIncomingFrameRate(int64_t now) {
   int32_t num = 0;
diff --git a/chromium/third_party/webrtc/modules/video_coding/media_optimization.h b/chromium/third_party/webrtc/modules/video_coding/media_optimization.h
index 060cd893ffe..081b2a900a2 100644
--- a/chromium/third_party/webrtc/modules/video_coding/media_optimization.h
+++ b/chromium/third_party/webrtc/modules/video_coding/media_optimization.h
@@ -17,7 +17,6 @@
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/modules/video_coding/media_opt_util.h"
-#include "webrtc/modules/video_coding/qm_select.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
@@ -59,11 +58,9 @@ class MediaOptimization {
   uint32_t SetTargetRates(uint32_t target_bitrate,
                           uint8_t fraction_lost,
                           int64_t round_trip_time_ms,
-                          VCMProtectionCallback* protection_callback,
-                          VCMQMSettingsCallback* qmsettings_callback);
+                          VCMProtectionCallback* protection_callback);
 
   void SetProtectionMethod(VCMProtectionMethodEnum method);
-  void EnableQM(bool enable);
   void EnableFrameDropper(bool enable);
 
   // Lets the sender suspend video when the rate drops below
@@ -74,8 +71,6 @@ class MediaOptimization {
 
   bool DropFrame();
 
-  void UpdateContentData(const VideoContentMetrics* content_metrics);
-
   // Informs Media Optimization of encoded output.
   int32_t UpdateWithEncodedData(const EncodedImage& encoded_image);
 
@@ -98,19 +93,6 @@ class MediaOptimization {
   void UpdateSentBitrate(int64_t now_ms) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
   void UpdateSentFramerate() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
-  // Computes new Quality Mode.
-  int32_t SelectQuality(VCMQMSettingsCallback* qmsettings_callback)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
-  // Verifies if QM settings differ from default, i.e. if an update is required.
-  // Computes actual values, as will be sent to the encoder.
-  bool QMUpdate(VCMResolutionScale* qm,
-                VCMQMSettingsCallback* qmsettings_callback)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
-  // Checks if we should make a QM change. Return true if yes, false otherwise.
-  bool CheckStatusForQMchange() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
   void ProcessIncomingFrameRate(int64_t now)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
@@ -152,16 +134,11 @@ class MediaOptimization {
   int video_target_bitrate_ GUARDED_BY(crit_sect_);
   float incoming_frame_rate_ GUARDED_BY(crit_sect_);
   int64_t incoming_frame_times_[kFrameCountHistorySize] GUARDED_BY(crit_sect_);
-  bool enable_qm_ GUARDED_BY(crit_sect_);
   std::list<EncodedFrameSample> encoded_frame_samples_ GUARDED_BY(crit_sect_);
   uint32_t avg_sent_bit_rate_bps_ GUARDED_BY(crit_sect_);
   uint32_t avg_sent_framerate_ GUARDED_BY(crit_sect_);
   uint32_t key_frame_cnt_ GUARDED_BY(crit_sect_);
   uint32_t delta_frame_cnt_ GUARDED_BY(crit_sect_);
-  std::unique_ptr<VCMContentMetricsProcessing> content_ GUARDED_BY(crit_sect_);
-  std::unique_ptr<VCMQmResolution> qm_resolution_ GUARDED_BY(crit_sect_);
-  int64_t last_qm_update_time_ GUARDED_BY(crit_sect_);
-  int64_t last_change_time_ GUARDED_BY(crit_sect_);  // Content/user triggered.
   int num_layers_ GUARDED_BY(crit_sect_);
   bool suspension_enabled_ GUARDED_BY(crit_sect_);
   bool video_suspended_ GUARDED_BY(crit_sect_);
diff --git a/chromium/third_party/webrtc/modules/video_coding/media_optimization_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/media_optimization_unittest.cc
index 3f8ac5d0752..e6a1bcccd96 100644
--- a/chromium/third_party/webrtc/modules/video_coding/media_optimization_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/media_optimization_unittest.cc
@@ -66,7 +66,7 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
   media_opt_.SetTargetRates(target_bitrate_kbps * 1000,
                             0,    // Lossrate.
                             100,  // RTT in ms.
-                            nullptr, nullptr);
+                            nullptr);
   media_opt_.EnableFrameDropper(true);
   for (int time = 0; time < 2000; time += frame_time_ms_) {
     ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, false));
@@ -76,7 +76,7 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
   media_opt_.SetTargetRates(kThresholdBps - 1000,
                             0,    // Lossrate.
                             100,  // RTT in ms.
-                            nullptr, nullptr);
+                            nullptr);
   // Expect the muter to engage immediately and stay muted.
   // Test during 2 seconds.
   for (int time = 0; time < 2000; time += frame_time_ms_) {
@@ -89,7 +89,7 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
   media_opt_.SetTargetRates(kThresholdBps + 1000,
                             0,    // Lossrate.
                             100,  // RTT in ms.
-                            nullptr, nullptr);
+                            nullptr);
   // Expect the muter to stay muted.
   // Test during 2 seconds.
   for (int time = 0; time < 2000; time += frame_time_ms_) {
@@ -101,7 +101,7 @@ TEST_F(TestMediaOptimization, VerifyMuting) {
   media_opt_.SetTargetRates(kThresholdBps + kWindowBps + 1000,
                             0,    // Lossrate.
                             100,  // RTT in ms.
-                            nullptr, nullptr);
+                            nullptr);
   // Expect the muter to disengage immediately.
   // Test during 2 seconds.
   for (int time = 0; time < 2000; time += frame_time_ms_) {
@@ -138,7 +138,7 @@ TEST_F(TestMediaOptimization, ProtectsUsingFecBitrateAboveCodecMax) {
   // Using 10% of codec bitrate for FEC, should still be able to use all of it.
   protection_callback.fec_rate_bps_ = kCodecBitrateBps / 10;
   uint32_t target_bitrate = media_opt_.SetTargetRates(
-      kMaxBitrateBps, 0, 0, &protection_callback, nullptr);
+      kMaxBitrateBps, 0, 0, &protection_callback);
 
   EXPECT_EQ(kCodecBitrateBps, static_cast<int>(target_bitrate));
 
@@ -146,7 +146,7 @@ TEST_F(TestMediaOptimization, ProtectsUsingFecBitrateAboveCodecMax) {
   // both equally, but only be half of max (since that ceiling should be hit).
   protection_callback.fec_rate_bps_ = kCodecBitrateBps;
   target_bitrate = media_opt_.SetTargetRates(kMaxBitrateBps, 128, 100,
-                                             &protection_callback, nullptr);
+                                             &protection_callback);
   EXPECT_EQ(kMaxBitrateBps / 2, static_cast<int>(target_bitrate));
 }
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/nack_module.cc b/chromium/third_party/webrtc/modules/video_coding/nack_module.cc
index 1b12afe0f0d..43244321ea0 100644
--- a/chromium/third_party/webrtc/modules/video_coding/nack_module.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/nack_module.cc
@@ -49,17 +49,17 @@ NackModule::NackModule(Clock* clock,
       running_(true),
       initialized_(false),
       rtt_ms_(kDefaultRttMs),
-      last_seq_num_(0),
+      newest_seq_num_(0),
       next_process_time_ms_(-1) {
   RTC_DCHECK(clock_);
   RTC_DCHECK(nack_sender_);
   RTC_DCHECK(keyframe_request_sender_);
 }
 
-void NackModule::OnReceivedPacket(const VCMPacket& packet) {
+int NackModule::OnReceivedPacket(const VCMPacket& packet) {
   rtc::CritScope lock(&crit_);
   if (!running_)
-    return;
+    return -1;
   uint16_t seq_num = packet.seqNum;
   // TODO(philipel): When the packet includes information whether it is
   //                 retransmitted or not, use that value instead. For
@@ -69,40 +69,48 @@ void NackModule::OnReceivedPacket(const VCMPacket& packet) {
   bool is_keyframe = packet.isFirstPacket && packet.frameType == kVideoFrameKey;
 
   if (!initialized_) {
-    last_seq_num_ = seq_num;
+    newest_seq_num_ = seq_num;
     if (is_keyframe)
       keyframe_list_.insert(seq_num);
     initialized_ = true;
-    return;
+    return 0;
   }
 
-  if (seq_num == last_seq_num_)
-    return;
+  // Since the |newest_seq_num_| is a packet we have actually received we know
+  // that packet has never been Nacked.
+  if (seq_num == newest_seq_num_)
+    return 0;
 
-  if (AheadOf(last_seq_num_, seq_num)) {
+  if (AheadOf(newest_seq_num_, seq_num)) {
     // An out of order packet has been received.
-    nack_list_.erase(seq_num);
+    auto nack_list_it = nack_list_.find(seq_num);
+    int nacks_sent_for_packet = 0;
+    if (nack_list_it != nack_list_.end()) {
+      nacks_sent_for_packet = nack_list_it->second.retries;
+      nack_list_.erase(nack_list_it);
+    }
     if (!is_retransmitted)
       UpdateReorderingStatistics(seq_num);
-    return;
-  } else {
-    AddPacketsToNack(last_seq_num_ + 1, seq_num);
-    last_seq_num_ = seq_num;
+    return nacks_sent_for_packet;
+  }
+  AddPacketsToNack(newest_seq_num_ + 1, seq_num);
+  newest_seq_num_ = seq_num;
 
-    // Keep track of new keyframes.
-    if (is_keyframe)
-      keyframe_list_.insert(seq_num);
+  // Keep track of new keyframes.
+  if (is_keyframe)
+    keyframe_list_.insert(seq_num);
 
-    // And remove old ones so we don't accumulate keyframes.
-    auto it = keyframe_list_.lower_bound(seq_num - kMaxPacketAge);
-    if (it != keyframe_list_.begin())
-      keyframe_list_.erase(keyframe_list_.begin(), it);
+  // And remove old ones so we don't accumulate keyframes.
+  auto it = keyframe_list_.lower_bound(seq_num - kMaxPacketAge);
+  if (it != keyframe_list_.begin())
+    keyframe_list_.erase(keyframe_list_.begin(), it);
 
-    // Are there any nacks that are waiting for this seq_num.
-    std::vector<uint16_t> nack_batch = GetNackBatch(kSeqNumOnly);
-    if (!nack_batch.empty())
-      nack_sender_->SendNack(nack_batch);
-  }
+  // Are there any nacks that are waiting for this seq_num.
+  std::vector<uint16_t> nack_batch = GetNackBatch(kSeqNumOnly);
+  if (!nack_batch.empty())
+    nack_sender_->SendNack(nack_batch);
+
+  return 0;
 }
 
 void NackModule::ClearUpTo(uint16_t seq_num) {
@@ -215,7 +223,7 @@ std::vector<uint16_t> NackModule::GetNackBatch(NackFilterOptions options) {
   auto it = nack_list_.begin();
   while (it != nack_list_.end()) {
     if (consider_seq_num && it->second.sent_at_time == -1 &&
-        AheadOrAt(last_seq_num_, it->second.send_at_seq_num)) {
+        AheadOrAt(newest_seq_num_, it->second.send_at_seq_num)) {
       nack_batch.emplace_back(it->second.seq_num);
       ++it->second.retries;
       it->second.sent_at_time = now_ms;
@@ -248,8 +256,8 @@ std::vector<uint16_t> NackModule::GetNackBatch(NackFilterOptions options) {
 }
 
 void NackModule::UpdateReorderingStatistics(uint16_t seq_num) {
-  RTC_DCHECK(AheadOf(last_seq_num_, seq_num));
-  uint16_t diff = ReverseDiff(last_seq_num_, seq_num);
+  RTC_DCHECK(AheadOf(newest_seq_num_, seq_num));
+  uint16_t diff = ReverseDiff(newest_seq_num_, seq_num);
   reordering_histogram_.Add(diff);
 }
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/nack_module.h b/chromium/third_party/webrtc/modules/video_coding/nack_module.h
index 7163a8e9054..58d6cfa985e 100644
--- a/chromium/third_party/webrtc/modules/video_coding/nack_module.h
+++ b/chromium/third_party/webrtc/modules/video_coding/nack_module.h
@@ -32,7 +32,7 @@ class NackModule : public Module {
              NackSender* nack_sender,
              KeyFrameRequestSender* keyframe_request_sender);
 
-  void OnReceivedPacket(const VCMPacket& packet);
+  int OnReceivedPacket(const VCMPacket& packet);
   void ClearUpTo(uint16_t seq_num);
   void UpdateRtt(int64_t rtt_ms);
   void Clear();
@@ -59,11 +59,6 @@ class NackModule : public Module {
     int64_t sent_at_time;
     int retries;
   };
-
-  struct SeqNumComparator {
-    bool operator()(uint16_t s1, uint16_t s2) const { return AheadOf(s2, s1); }
-  };
-
   void AddPacketsToNack(uint16_t seq_num_start, uint16_t seq_num_end)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
@@ -87,13 +82,15 @@ class NackModule : public Module {
   NackSender* const nack_sender_;
   KeyFrameRequestSender* const keyframe_request_sender_;
 
-  std::map<uint16_t, NackInfo, SeqNumComparator> nack_list_ GUARDED_BY(crit_);
-  std::set<uint16_t, SeqNumComparator> keyframe_list_ GUARDED_BY(crit_);
+  std::map<uint16_t, NackInfo, DescendingSeqNumComp<uint16_t>> nack_list_
+      GUARDED_BY(crit_);
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> keyframe_list_
+      GUARDED_BY(crit_);
   video_coding::Histogram reordering_histogram_ GUARDED_BY(crit_);
   bool running_ GUARDED_BY(crit_);
   bool initialized_ GUARDED_BY(crit_);
   int64_t rtt_ms_ GUARDED_BY(crit_);
-  uint16_t last_seq_num_ GUARDED_BY(crit_);
+  uint16_t newest_seq_num_ GUARDED_BY(crit_);
   int64_t next_process_time_ms_ GUARDED_BY(crit_);
 };
 
diff --git a/chromium/third_party/webrtc/modules/video_coding/nack_module_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/nack_module_unittest.cc
index 3870742016a..9c2eb4ac0c3 100644
--- a/chromium/third_party/webrtc/modules/video_coding/nack_module_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/nack_module_unittest.cc
@@ -290,4 +290,28 @@ TEST_F(TestNackModule, ClearUpToWrap) {
   EXPECT_EQ(0, sent_nacks_[0]);
 }
 
+TEST_F(TestNackModule, PacketNackCount) {
+  VCMPacket packet;
+  packet.seqNum = 0;
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(packet));
+  packet.seqNum = 2;
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(packet));
+  packet.seqNum = 1;
+  EXPECT_EQ(1, nack_module_.OnReceivedPacket(packet));
+
+  sent_nacks_.clear();
+  nack_module_.UpdateRtt(100);
+  packet.seqNum = 5;
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(packet));
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.Process();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.Process();
+  packet.seqNum = 3;
+  EXPECT_EQ(3, nack_module_.OnReceivedPacket(packet));
+  packet.seqNum = 4;
+  EXPECT_EQ(3, nack_module_.OnReceivedPacket(packet));
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(packet));
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/packet_buffer.cc b/chromium/third_party/webrtc/modules/video_coding/packet_buffer.cc
index 0a05baa16ab..09fb2499074 100644
--- a/chromium/third_party/webrtc/modules/video_coding/packet_buffer.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/packet_buffer.cc
@@ -14,8 +14,8 @@
 #include <limits>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/video_coding/frame_object.h"
-#include "webrtc/modules/video_coding/sequence_number_util.h"
 
 namespace webrtc {
 namespace video_coding {
@@ -25,12 +25,12 @@ PacketBuffer::PacketBuffer(size_t start_buffer_size,
                            OnCompleteFrameCallback* frame_callback)
     : size_(start_buffer_size),
       max_size_(max_buffer_size),
-      last_seq_num_(0),
       first_seq_num_(0),
-      initialized_(false),
+      last_seq_num_(0),
+      first_packet_received_(false),
       data_buffer_(start_buffer_size),
       sequence_buffer_(start_buffer_size),
-      frame_callback_(frame_callback) {
+      reference_finder_(frame_callback) {
   RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
   // Buffer size must always be a power of 2.
   RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
@@ -40,12 +40,12 @@ PacketBuffer::PacketBuffer(size_t start_buffer_size,
 bool PacketBuffer::InsertPacket(const VCMPacket& packet) {
   rtc::CritScope lock(&crit_);
   uint16_t seq_num = packet.seqNum;
-  int index = seq_num % size_;
+  size_t index = seq_num % size_;
 
-  if (!initialized_) {
+  if (!first_packet_received_) {
     first_seq_num_ = seq_num - 1;
     last_seq_num_ = seq_num;
-    initialized_ = true;
+    first_packet_received_ = true;
   }
 
   if (sequence_buffer_[index].used) {
@@ -70,16 +70,17 @@ bool PacketBuffer::InsertPacket(const VCMPacket& packet) {
   sequence_buffer_[index].frame_end = packet.markerBit;
   sequence_buffer_[index].seq_num = packet.seqNum;
   sequence_buffer_[index].continuous = false;
+  sequence_buffer_[index].frame_created = false;
   sequence_buffer_[index].used = true;
   data_buffer_[index] = packet;
 
-  FindCompleteFrames(seq_num);
+  FindFrames(seq_num);
   return true;
 }
 
 void PacketBuffer::ClearTo(uint16_t seq_num) {
   rtc::CritScope lock(&crit_);
-  int index = first_seq_num_ % size_;
+  size_t index = first_seq_num_ % size_;
   while (AheadOf<uint16_t>(seq_num, first_seq_num_ + 1)) {
     index = (index + 1) % size_;
     first_seq_num_ = Add<1 << 16>(first_seq_num_, 1);
@@ -96,7 +97,7 @@ bool PacketBuffer::ExpandBufferSize() {
   std::vector<ContinuityInfo> new_sequence_buffer(new_size);
   for (size_t i = 0; i < size_; ++i) {
     if (sequence_buffer_[i].used) {
-      int index = sequence_buffer_[i].seq_num % new_size;
+      size_t index = sequence_buffer_[i].seq_num % new_size;
       new_sequence_buffer[index] = sequence_buffer_[i];
       new_data_buffer[index] = data_buffer_[i];
     }
@@ -108,38 +109,47 @@ bool PacketBuffer::ExpandBufferSize() {
 }
 
 bool PacketBuffer::IsContinuous(uint16_t seq_num) const {
-  int index = seq_num % size_;
+  size_t index = seq_num % size_;
   int prev_index = index > 0 ? index - 1 : size_ - 1;
+
   if (!sequence_buffer_[index].used)
     return false;
+  if (sequence_buffer_[index].frame_created)
+    return false;
   if (sequence_buffer_[index].frame_begin)
     return true;
   if (!sequence_buffer_[prev_index].used)
     return false;
+  if (sequence_buffer_[prev_index].seq_num !=
+      static_cast<uint16_t>(seq_num - 1))
+    return false;
   if (sequence_buffer_[prev_index].continuous)
     return true;
 
   return false;
 }
 
-void PacketBuffer::FindCompleteFrames(uint16_t seq_num) {
-  int index = seq_num % size_;
+void PacketBuffer::FindFrames(uint16_t seq_num) {
+  size_t index = seq_num % size_;
   while (IsContinuous(seq_num)) {
     sequence_buffer_[index].continuous = true;
 
-    // If the frame is complete, find the first packet of the frame and
-    // create a FrameObject.
+    // If all packets of the frame is continuous, find the first packet of the
+    // frame and create an RtpFrameObject.
     if (sequence_buffer_[index].frame_end) {
-      int rindex = index;
+      int start_index = index;
       uint16_t start_seq_num = seq_num;
-      while (!sequence_buffer_[rindex].frame_begin) {
-        rindex = rindex > 0 ? rindex - 1 : size_ - 1;
+
+      while (!sequence_buffer_[start_index].frame_begin) {
+        sequence_buffer_[start_index].frame_created = true;
+        start_index = start_index > 0 ? start_index - 1 : size_ - 1;
         start_seq_num--;
       }
+      sequence_buffer_[start_index].frame_created = true;
 
-      std::unique_ptr<FrameObject> frame(
-          new RtpFrameObject(this, 1, start_seq_num, seq_num));
-      frame_callback_->OnCompleteFrame(std::move(frame));
+      std::unique_ptr<RtpFrameObject> frame(
+          new RtpFrameObject(this, start_seq_num, seq_num));
+      reference_finder_.ManageFrame(std::move(frame));
     }
 
     index = (index + 1) % size_;
@@ -149,14 +159,13 @@ void PacketBuffer::FindCompleteFrames(uint16_t seq_num) {
 
 void PacketBuffer::ReturnFrame(RtpFrameObject* frame) {
   rtc::CritScope lock(&crit_);
-  int index = frame->first_packet() % size_;
-  int end = (frame->last_packet() + 1) % size_;
-  uint16_t seq_num = frame->first_packet();
+  size_t index = frame->first_seq_num() % size_;
+  size_t end = (frame->last_seq_num() + 1) % size_;
+  uint16_t seq_num = frame->first_seq_num();
   while (index != end) {
-    if (sequence_buffer_[index].seq_num == seq_num) {
+    if (sequence_buffer_[index].seq_num == seq_num)
       sequence_buffer_[index].used = false;
-      sequence_buffer_[index].continuous = false;
-    }
+
     index = (index + 1) % size_;
     ++seq_num;
   }
@@ -173,9 +182,9 @@ bool PacketBuffer::GetBitstream(const RtpFrameObject& frame,
                                 uint8_t* destination) {
   rtc::CritScope lock(&crit_);
 
-  int index = frame.first_packet() % size_;
-  int end = (frame.last_packet() + 1) % size_;
-  uint16_t seq_num = frame.first_packet();
+  size_t index = frame.first_seq_num() % size_;
+  size_t end = (frame.last_seq_num() + 1) % size_;
+  uint16_t seq_num = frame.first_seq_num();
   while (index != end) {
     if (!sequence_buffer_[index].used ||
         sequence_buffer_[index].seq_num != seq_num) {
@@ -192,12 +201,22 @@ bool PacketBuffer::GetBitstream(const RtpFrameObject& frame,
   return true;
 }
 
-void PacketBuffer::Flush() {
+VCMPacket* PacketBuffer::GetPacket(uint16_t seq_num) {
   rtc::CritScope lock(&crit_);
-  for (size_t i = 0; i < size_; ++i) {
-    sequence_buffer_[i].used = false;
-    sequence_buffer_[i].continuous = false;
+  size_t index = seq_num % size_;
+  if (!sequence_buffer_[index].used ||
+      seq_num != sequence_buffer_[index].seq_num) {
+    return nullptr;
   }
+  return &data_buffer_[index];
+}
+
+void PacketBuffer::Clear() {
+  rtc::CritScope lock(&crit_);
+  for (size_t i = 0; i < size_; ++i)
+    sequence_buffer_[i].used = false;
+
+  first_packet_received_ = false;
 }
 
 }  // namespace video_coding
diff --git a/chromium/third_party/webrtc/modules/video_coding/packet_buffer.h b/chromium/third_party/webrtc/modules/video_coding/packet_buffer.h
index 6ca514536ef..ae0916a75f3 100644
--- a/chromium/third_party/webrtc/modules/video_coding/packet_buffer.h
+++ b/chromium/third_party/webrtc/modules/video_coding/packet_buffer.h
@@ -14,9 +14,11 @@
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/packet.h"
+#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
+#include "webrtc/modules/video_coding/sequence_number_util.h"
 
 namespace webrtc {
 namespace video_coding {
@@ -39,25 +41,50 @@ class PacketBuffer {
 
   bool InsertPacket(const VCMPacket& packet);
   void ClearTo(uint16_t seq_num);
-  void Flush();
+  void Clear();
 
  private:
   friend RtpFrameObject;
   // Since we want the packet buffer to be as packet type agnostic
   // as possible we extract only the information needed in order
-  // to determin whether a sequence of packets is continuous or not.
+  // to determine whether a sequence of packets is continuous or not.
   struct ContinuityInfo {
+    // The sequence number of the packet.
     uint16_t seq_num = 0;
+
+    // If this is the first packet of the frame.
     bool frame_begin = false;
+
+    // If this is the last packet of the frame.
     bool frame_end = false;
+
+    // If this slot is currently used.
     bool used = false;
+
+    // If all its previous packets have been inserted into the packet buffer.
     bool continuous = false;
+
+    // If this packet has been used to create a frame already.
+    bool frame_created = false;
   };
 
+  // Tries to expand the buffer.
   bool ExpandBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Test if all previous packets has arrived for the given sequence number.
   bool IsContinuous(uint16_t seq_num) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  void FindCompleteFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Test if all packets of a frame has arrived, and if so, creates a frame.
+  // May create multiple frames per invocation.
+  void FindFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Copy the bitstream for |frame| to |destination|.
   bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);
+
+  // Get the packet with sequence number |seq_num|.
+  VCMPacket* GetPacket(uint16_t seq_num);
+
+  // Mark all slots used by |frame| as not used.
   void ReturnFrame(RtpFrameObject* frame);
 
   rtc::CriticalSection crit_;
@@ -66,13 +93,25 @@ class PacketBuffer {
   size_t size_ GUARDED_BY(crit_);
   const size_t max_size_;
 
-  uint16_t last_seq_num_ GUARDED_BY(crit_);
+  // The fist sequence number currently in the buffer.
   uint16_t first_seq_num_ GUARDED_BY(crit_);
-  bool initialized_ GUARDED_BY(crit_);
+
+  // The last sequence number currently in the buffer.
+  uint16_t last_seq_num_ GUARDED_BY(crit_);
+
+  // If the packet buffer has received its first packet.
+  bool first_packet_received_ GUARDED_BY(crit_);
+
+  // Buffer that holds the inserted packets.
   std::vector<VCMPacket> data_buffer_ GUARDED_BY(crit_);
+
+  // Buffer that holds the information about which slot that is currently in use
+  // and information needed to determine the continuity between packets.
   std::vector<ContinuityInfo> sequence_buffer_ GUARDED_BY(crit_);
 
-  OnCompleteFrameCallback* const frame_callback_;
+  // Frames that have received all their packets are handed off to the
+  // |reference_finder_| which finds the dependencies between the frames.
+  RtpFrameReferenceFinder reference_finder_;
 };
 
 }  // namespace video_coding
diff --git a/chromium/third_party/webrtc/modules/video_coding/packet_buffer_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/packet_buffer_unittest.cc
index bc06940391c..b50074d8f60 100644
--- a/chromium/third_party/webrtc/modules/video_coding/packet_buffer_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/packet_buffer_unittest.cc
@@ -10,6 +10,9 @@
 
 #include <cstring>
 #include <limits>
+#include <map>
+#include <set>
+#include <utility>
 
 #include "webrtc/modules/video_coding/frame_object.h"
 #include "webrtc/modules/video_coding/packet_buffer.h"
@@ -24,142 +27,301 @@ class TestPacketBuffer : public ::testing::Test,
                          public OnCompleteFrameCallback {
  protected:
   TestPacketBuffer()
-      : rand_(0x8739211), packet_buffer_(kStartSize, kMaxSize, this) {}
+      : rand_(0x8739211),
+        packet_buffer_(new PacketBuffer(kStartSize, kMaxSize, this)),
+        frames_from_callback_(FrameComp()) {}
 
   uint16_t Rand() { return rand_.Rand(std::numeric_limits<uint16_t>::max()); }
 
   void OnCompleteFrame(std::unique_ptr<FrameObject> frame) override {
-    frames_from_callback_.emplace_back(std::move(frame));
+    uint16_t pid = frame->picture_id;
+    uint16_t sidx = frame->spatial_layer;
+    auto frame_it = frames_from_callback_.find(std::make_pair(pid, sidx));
+    if (frame_it != frames_from_callback_.end()) {
+      ADD_FAILURE() << "Already received frame with (pid:sidx): ("
+                    << pid << ":" << sidx << ")";
+      return;
+    }
+
+    frames_from_callback_.insert(
+        std::make_pair(std::make_pair(pid, sidx), std::move(frame)));
   }
 
   void TearDown() override {
-    // All FrameObjects must be destroyed before the PacketBuffer since
-    // a FrameObject will try to remove itself from the packet buffer
+    // All frame objects must be destroyed before the packet buffer since
+    // a frame object will try to remove itself from the packet buffer
     // upon destruction.
     frames_from_callback_.clear();
   }
 
+  // Short version of true and false.
+  enum {
+    kT = true,
+    kF = false
+  };
+
+  // Insert a generic packet into the packet buffer.
+  void InsertGeneric(uint16_t seq_num,              // packet sequence number
+                     bool keyframe,                 // is keyframe
+                     bool first,                    // is first packet of frame
+                     bool last,                     // is last packet of frame
+                     size_t data_size = 0,          // size of data
+                     uint8_t* data = nullptr) {     // data pointer
+    VCMPacket packet;
+    packet.codec = kVideoCodecGeneric;
+    packet.seqNum = seq_num;
+    packet.frameType = keyframe ? kVideoFrameKey : kVideoFrameDelta;
+    packet.isFirstPacket = first;
+    packet.markerBit = last;
+    packet.sizeBytes = data_size;
+    packet.dataPtr = data;
+
+    EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
+  }
+
+  // Insert a Vp8 packet into the packet buffer.
+  void InsertVp8(uint16_t seq_num,              // packet sequence number
+                 bool keyframe,                 // is keyframe
+                 bool first,                    // is first packet of frame
+                 bool last,                     // is last packet of frame
+                 bool sync = false,             // is sync frame
+                 int32_t pid = kNoPictureId,    // picture id
+                 uint8_t tid = kNoTemporalIdx,  // temporal id
+                 int32_t tl0 = kNoTl0PicIdx,    // tl0 pic index
+                 size_t data_size = 0,          // size of data
+                 uint8_t* data = nullptr) {     // data pointer
+    VCMPacket packet;
+    packet.codec = kVideoCodecVP8;
+    packet.seqNum = seq_num;
+    packet.frameType = keyframe ? kVideoFrameKey : kVideoFrameDelta;
+    packet.isFirstPacket = first;
+    packet.markerBit = last;
+    packet.sizeBytes = data_size;
+    packet.dataPtr = data;
+    packet.codecSpecificHeader.codecHeader.VP8.pictureId = pid % (1 << 15);
+    packet.codecSpecificHeader.codecHeader.VP8.temporalIdx = tid;
+    packet.codecSpecificHeader.codecHeader.VP8.tl0PicIdx = tl0;
+    packet.codecSpecificHeader.codecHeader.VP8.layerSync = sync;
+
+    EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
+  }
+
+  // Insert a Vp9 packet into the packet buffer.
+  void InsertVp9Gof(uint16_t seq_num,              // packet sequence number
+                    bool keyframe,                 // is keyframe
+                    bool first,                    // is first packet of frame
+                    bool last,                     // is last packet of frame
+                    bool up = false,               // frame is up-switch point
+                    int32_t pid = kNoPictureId,    // picture id
+                    uint8_t sid = kNoSpatialIdx,   // spatial id
+                    uint8_t tid = kNoTemporalIdx,  // temporal id
+                    int32_t tl0 = kNoTl0PicIdx,    // tl0 pic index
+                    GofInfoVP9* ss = nullptr,      // scalability structure
+                    size_t data_size = 0,          // size of data
+                    uint8_t* data = nullptr) {     // data pointer
+    VCMPacket packet;
+    packet.codec = kVideoCodecVP9;
+    packet.seqNum = seq_num;
+    packet.frameType = keyframe ? kVideoFrameKey : kVideoFrameDelta;
+    packet.isFirstPacket = first;
+    packet.markerBit = last;
+    packet.sizeBytes = data_size;
+    packet.dataPtr = data;
+    packet.codecSpecificHeader.codecHeader.VP9.flexible_mode = false;
+    packet.codecSpecificHeader.codecHeader.VP9.picture_id = pid % (1 << 15);
+    packet.codecSpecificHeader.codecHeader.VP9.temporal_idx = tid;
+    packet.codecSpecificHeader.codecHeader.VP9.spatial_idx = sid;
+    packet.codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = tl0;
+    packet.codecSpecificHeader.codecHeader.VP9.temporal_up_switch = up;
+    if (ss != nullptr) {
+      packet.codecSpecificHeader.codecHeader.VP9.ss_data_available = true;
+      packet.codecSpecificHeader.codecHeader.VP9.gof = *ss;
+    }
+
+    EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
+  }
+
+  // Insert a Vp9 packet into the packet buffer.
+  void InsertVp9Flex(uint16_t seq_num,              // packet sequence number
+                     bool keyframe,                 // is keyframe
+                     bool first,                    // is first packet of frame
+                     bool last,                     // is last packet of frame
+                     bool inter,                    // depends on S-1 layer
+                     int32_t pid = kNoPictureId,    // picture id
+                     uint8_t sid = kNoSpatialIdx,   // spatial id
+                     uint8_t tid = kNoTemporalIdx,  // temporal id
+                     int32_t tl0 = kNoTl0PicIdx,    // tl0 pic index
+                     std::vector<uint8_t> refs =
+                       std::vector<uint8_t>(),      // frame references
+                     size_t data_size = 0,          // size of data
+                     uint8_t* data = nullptr) {     // data pointer
+    VCMPacket packet;
+    packet.codec = kVideoCodecVP9;
+    packet.seqNum = seq_num;
+    packet.frameType = keyframe ? kVideoFrameKey : kVideoFrameDelta;
+    packet.isFirstPacket = first;
+    packet.markerBit = last;
+    packet.sizeBytes = data_size;
+    packet.dataPtr = data;
+    packet.codecSpecificHeader.codecHeader.VP9.inter_layer_predicted = inter;
+    packet.codecSpecificHeader.codecHeader.VP9.flexible_mode = true;
+    packet.codecSpecificHeader.codecHeader.VP9.picture_id = pid % (1 << 15);
+    packet.codecSpecificHeader.codecHeader.VP9.temporal_idx = tid;
+    packet.codecSpecificHeader.codecHeader.VP9.spatial_idx = sid;
+    packet.codecSpecificHeader.codecHeader.VP9.tl0_pic_idx = tl0;
+    packet.codecSpecificHeader.codecHeader.VP9.num_ref_pics = refs.size();
+    for (size_t i = 0; i < refs.size(); ++i)
+      packet.codecSpecificHeader.codecHeader.VP9.pid_diff[i] = refs[i];
+
+    EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
+  }
+
+  // Check if a frame with picture id |pid| and spatial index |sidx| has been
+  // delivered from the packet buffer, and if so, if it has the references
+  // specified by |refs|.
+  template <typename... T>
+  void CheckReferences(uint16_t pid, uint16_t sidx, T... refs) const {
+    auto frame_it = frames_from_callback_.find(std::make_pair(pid, sidx));
+    if (frame_it == frames_from_callback_.end()) {
+      ADD_FAILURE() << "Could not find frame with (pid:sidx): ("
+                    << pid << ":" << sidx << ")";
+      return;
+    }
+
+    std::set<uint16_t> actual_refs;
+    for (uint8_t r = 0; r < frame_it->second->num_references; ++r) {
+      actual_refs.insert(frame_it->second->references[r]);
+    }
+
+    std::set<uint16_t> expected_refs;
+    RefsToSet(&expected_refs, refs...);
+
+    ASSERT_EQ(expected_refs, actual_refs);
+  }
+
+  template <typename... T>
+  void CheckReferencesGeneric(uint16_t pid, T... refs) const {
+    CheckReferences(pid, 0, refs...);
+  }
+
+  template <typename... T>
+  void CheckReferencesVp8(uint16_t pid, T... refs) const {
+    CheckReferences(pid, 0, refs...);
+  }
+
+  template <typename... T>
+  void CheckReferencesVp9(uint16_t pid, uint8_t sidx, T... refs) const {
+    CheckReferences(pid, sidx, refs...);
+  }
+
+  template <typename... T>
+  void RefsToSet(std::set<uint16_t>* m, uint16_t ref, T... refs) const {
+    m->insert(ref);
+    RefsToSet(m, refs...);
+  }
+
+  void RefsToSet(std::set<uint16_t>* m) const {}
+
   const int kStartSize = 16;
   const int kMaxSize = 64;
 
   Random rand_;
-  PacketBuffer packet_buffer_;
-  std::vector<std::unique_ptr<FrameObject>> frames_from_callback_;
+  std::unique_ptr<PacketBuffer> packet_buffer_;
+  struct FrameComp {
+    bool operator()(const std::pair<uint16_t, uint8_t> f1,
+                    const std::pair<uint16_t, uint8_t> f2) const {
+                      if (f1.first == f2.first)
+                        return f1.second < f2.second;
+                      return f1.first < f2.first;
+                    }
+  };
+  std::map<std::pair<uint16_t, uint8_t>,
+           std::unique_ptr<FrameObject>,
+           FrameComp> frames_from_callback_;
 };
 
 TEST_F(TestPacketBuffer, InsertOnePacket) {
   VCMPacket packet;
   packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
 }
 
 TEST_F(TestPacketBuffer, InsertMultiplePackets) {
   VCMPacket packet;
   packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
 }
 
 TEST_F(TestPacketBuffer, InsertDuplicatePacket) {
   VCMPacket packet;
   packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
 }
 
 TEST_F(TestPacketBuffer, ExpandBuffer) {
-  VCMPacket packet;
-  packet.seqNum = Rand();
+  uint16_t seq_num = Rand();
 
   for (int i = 0; i < kStartSize + 1; ++i) {
-    EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-    ++packet.seqNum;
+    //            seq_num    , kf, frst, lst
+    InsertGeneric(seq_num + i, kT    , kT, kT);
   }
 }
 
 TEST_F(TestPacketBuffer, ExpandBufferOverflow) {
-  VCMPacket packet;
-  packet.seqNum = Rand();
+  uint16_t seq_num = Rand();
 
   for (int i = 0; i < kMaxSize; ++i) {
-    EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-    ++packet.seqNum;
+    //            seq_num    , kf, frst, lst
+    InsertGeneric(seq_num + i, kT, kT  , kT);
   }
 
-  EXPECT_FALSE(packet_buffer_.InsertPacket(packet));
+  VCMPacket packet;
+  packet.seqNum = seq_num + kMaxSize + 1;
+  packet.sizeBytes = 1;
+  EXPECT_FALSE(packet_buffer_->InsertPacket(packet));
 }
 
-TEST_F(TestPacketBuffer, OnePacketOneFrame) {
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.markerBit = true;
-  packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(1UL, frames_from_callback_.size());
+TEST_F(TestPacketBuffer, GenericOnePacketOneFrame) {
+  //            seq_num, kf, frst, lst
+  InsertGeneric(Rand() , kT, kT  , kT);
+  ASSERT_EQ(1UL, frames_from_callback_.size());
 }
 
-TEST_F(TestPacketBuffer, TwoPacketsTwoFrames) {
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.markerBit = true;
-  packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+TEST_F(TestPacketBuffer, GenericTwoPacketsTwoFrames) {
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , kf, frst, lst
+  InsertGeneric(seq_num    , kT, kT  , kT);
+  InsertGeneric(seq_num + 1, kT, kT  , kT);
+
   EXPECT_EQ(2UL, frames_from_callback_.size());
 }
 
-TEST_F(TestPacketBuffer, TwoPacketsOneFrames) {
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  packet.markerBit = true;
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(1UL, frames_from_callback_.size());
-}
+TEST_F(TestPacketBuffer, GenericTwoPacketsOneFrames) {
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , kf, frst, lst
+  InsertGeneric(seq_num    , kT, kT  , kF);
+  InsertGeneric(seq_num + 1, kT, kF  , kT);
 
-TEST_F(TestPacketBuffer, ThreePacketReorderingOneFrame) {
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  packet.isFirstPacket = false;
-  packet.markerBit = true;
-  packet.seqNum += 2;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  packet.markerBit = false;
-  packet.seqNum -= 1;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
   EXPECT_EQ(1UL, frames_from_callback_.size());
 }
 
-TEST_F(TestPacketBuffer, IndexWrapOneFrame) {
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.seqNum = kStartSize - 1;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  packet.isFirstPacket = false;
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  packet.markerBit = true;
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+TEST_F(TestPacketBuffer, GenericThreePacketReorderingOneFrame) {
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , kf, frst, lst
+  InsertGeneric(seq_num    , kT, kT  , kF);
+  InsertGeneric(seq_num + 2, kT, kF  , kT);
+  InsertGeneric(seq_num + 1, kT, kF  , kF);
+
   EXPECT_EQ(1UL, frames_from_callback_.size());
 }
 
@@ -167,45 +329,77 @@ TEST_F(TestPacketBuffer, DiscardOldPacket) {
   uint16_t seq_num = Rand();
   VCMPacket packet;
   packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   packet.seqNum += 2;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
 
   for (int i = 3; i < kMaxSize; ++i) {
     ++packet.seqNum;
-    EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+    EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   }
 
   ++packet.seqNum;
-  EXPECT_FALSE(packet_buffer_.InsertPacket(packet));
-  packet_buffer_.ClearTo(seq_num + 1);
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_FALSE(packet_buffer_->InsertPacket(packet));
+  packet_buffer_->ClearTo(seq_num + 1);
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
 }
 
 TEST_F(TestPacketBuffer, DiscardMultipleOldPackets) {
   uint16_t seq_num = Rand();
   VCMPacket packet;
   packet.seqNum = seq_num;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   packet.seqNum += 2;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
 
   for (int i = 3; i < kMaxSize; ++i) {
     ++packet.seqNum;
-    EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+    EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   }
 
-  packet_buffer_.ClearTo(seq_num + 15);
+  packet_buffer_->ClearTo(seq_num + 15);
   for (int i = 0; i < 15; ++i) {
     ++packet.seqNum;
-    EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+    EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   }
   for (int i = 15; i < kMaxSize; ++i) {
     ++packet.seqNum;
-    EXPECT_FALSE(packet_buffer_.InsertPacket(packet));
+    EXPECT_FALSE(packet_buffer_->InsertPacket(packet));
   }
 }
 
+TEST_F(TestPacketBuffer, GenericFrames) {
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , keyf , first, last
+  InsertGeneric(seq_num    , true , true , true);
+  InsertGeneric(seq_num + 1, false, true , true);
+  InsertGeneric(seq_num + 2, false, true , true);
+  InsertGeneric(seq_num + 3, false, true , true);
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesGeneric(seq_num);
+  CheckReferencesGeneric(seq_num + 1, seq_num);
+  CheckReferencesGeneric(seq_num + 2, seq_num + 1);
+  CheckReferencesGeneric(seq_num + 3, seq_num + 2);
+}
+
+TEST_F(TestPacketBuffer, GenericFramesReordered) {
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , keyf , first, last
+  InsertGeneric(seq_num + 1, false, true , true);
+  InsertGeneric(seq_num    , true , true , true);
+  InsertGeneric(seq_num + 3, false, true , true);
+  InsertGeneric(seq_num + 2, false, true , true);
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesGeneric(seq_num);
+  CheckReferencesGeneric(seq_num + 1, seq_num);
+  CheckReferencesGeneric(seq_num + 2, seq_num + 1);
+  CheckReferencesGeneric(seq_num + 3, seq_num + 2);
+}
+
 TEST_F(TestPacketBuffer, GetBitstreamFromFrame) {
   // "many bitstream, such data" with null termination.
   uint8_t many[] = {0x6d, 0x61, 0x6e, 0x79, 0x20};
@@ -216,89 +410,997 @@ TEST_F(TestPacketBuffer, GetBitstreamFromFrame) {
   uint8_t
       result[sizeof(many) + sizeof(bitstream) + sizeof(such) + sizeof(data)];
 
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.seqNum = 0xfffe;
-  packet.dataPtr = many;
-  packet.sizeBytes = sizeof(many);
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  packet.isFirstPacket = false;
-  ++packet.seqNum;
-  packet.dataPtr = bitstream;
-  packet.sizeBytes = sizeof(bitstream);
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  ++packet.seqNum;
-  packet.dataPtr = such;
-  packet.sizeBytes = sizeof(such);
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  packet.markerBit = true;
-  ++packet.seqNum;
-  packet.dataPtr = data;
-  packet.sizeBytes = sizeof(data);
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  ASSERT_EQ(1UL, frames_from_callback_.size());
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , kf, frst, lst, data_size        , data
+  InsertGeneric(seq_num    , kT, kT  , kF , sizeof(many)     , many);
+  InsertGeneric(seq_num + 1, kF, kF  , kF , sizeof(bitstream), bitstream);
+  InsertGeneric(seq_num + 2, kF, kF  , kF , sizeof(such)     , such);
+  InsertGeneric(seq_num + 3, kF, kF  , kT , sizeof(data)     , data);
 
-  EXPECT_TRUE(frames_from_callback_[0]->GetBitstream(result));
-  EXPECT_EQ(
-      std::strcmp("many bitstream, such data", reinterpret_cast<char*>(result)),
-      0);
+  ASSERT_EQ(1UL, frames_from_callback_.size());
+  CheckReferencesVp8(seq_num + 3);
+  EXPECT_TRUE(frames_from_callback_[std::make_pair(seq_num + 3, 0)]->
+                                                          GetBitstream(result));
+  EXPECT_EQ(std::strcmp("many bitstream, such data",
+            reinterpret_cast<char*>(result)),
+            0);
 }
 
 TEST_F(TestPacketBuffer, FreeSlotsOnFrameDestruction) {
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  packet.isFirstPacket = false;
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  ++packet.seqNum;
-  packet.markerBit = true;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , kf, frst, lst
+  InsertGeneric(seq_num    , kT, kT  , kF);
+  InsertGeneric(seq_num + 1, kF, kF  , kF);
+  InsertGeneric(seq_num + 2, kF, kF  , kT);
   EXPECT_EQ(1UL, frames_from_callback_.size());
 
   frames_from_callback_.clear();
 
-  packet.isFirstPacket = true;
-  packet.markerBit = false;
-  packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  packet.isFirstPacket = false;
-  ++packet.seqNum;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_EQ(0UL, frames_from_callback_.size());
-  ++packet.seqNum;
-  packet.markerBit = true;
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  //            seq_num    , kf, frst, lst
+  InsertGeneric(seq_num    , kT, kT  , kF);
+  InsertGeneric(seq_num + 1, kF, kF  , kF);
+  InsertGeneric(seq_num + 2, kF, kF  , kT);
   EXPECT_EQ(1UL, frames_from_callback_.size());
 }
 
-TEST_F(TestPacketBuffer, Flush) {
-  VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.markerBit = true;
-  packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
-  packet_buffer_.Flush();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+TEST_F(TestPacketBuffer, Clear) {
+  uint16_t seq_num = Rand();
+
+  //            seq_num    , kf, frst, lst
+  InsertGeneric(seq_num    , kT, kT  , kF);
+  InsertGeneric(seq_num + 1, kF, kF  , kF);
+  InsertGeneric(seq_num + 2, kF, kF  , kT);
+  EXPECT_EQ(1UL, frames_from_callback_.size());
+
+  packet_buffer_->Clear();
+
+  //            seq_num                 , kf, frst, lst
+  InsertGeneric(seq_num + kStartSize    , kT, kT  , kF);
+  InsertGeneric(seq_num + kStartSize + 1, kF, kF  , kF);
+  InsertGeneric(seq_num + kStartSize + 2, kF, kF  , kT);
   EXPECT_EQ(2UL, frames_from_callback_.size());
 }
 
-TEST_F(TestPacketBuffer, InvalidateFrameByFlushing) {
+TEST_F(TestPacketBuffer, InvalidateFrameByClearing) {
   VCMPacket packet;
-  packet.isFirstPacket = true;
-  packet.markerBit = true;
+  packet.codec = kVideoCodecGeneric;
+  packet.frameType = kVideoFrameKey;
+  packet.isFirstPacket = kT;
+  packet.markerBit = kT;
   packet.seqNum = Rand();
-  EXPECT_TRUE(packet_buffer_.InsertPacket(packet));
+  EXPECT_TRUE(packet_buffer_->InsertPacket(packet));
   ASSERT_EQ(1UL, frames_from_callback_.size());
 
-  packet_buffer_.Flush();
-  EXPECT_FALSE(frames_from_callback_[0]->GetBitstream(nullptr));
+  packet_buffer_->Clear();
+  EXPECT_FALSE(frames_from_callback_.begin()->second->GetBitstream(nullptr));
+}
+
+TEST_F(TestPacketBuffer, Vp8NoPictureId) {
+  uint16_t seq_num = Rand();
+
+  //        seq_num     , kf, frst, lst
+  InsertVp8(seq_num     , kT, kT  , kF);
+  InsertVp8(seq_num + 1 , kF, kF  , kF);
+  InsertVp8(seq_num + 2 , kF, kF  , kT);
+  ASSERT_EQ(1UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 3 , kF, kT  , kF);
+  InsertVp8(seq_num + 4 , kF, kF  , kT);
+  ASSERT_EQ(2UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 5 , kF, kT  , kF);
+  InsertVp8(seq_num + 6 , kF, kF  , kF);
+  InsertVp8(seq_num + 7 , kF, kF  , kF);
+  InsertVp8(seq_num + 8 , kF, kF  , kT);
+  ASSERT_EQ(3UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 9 , kF, kT  , kT);
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 10, kF, kT  , kF);
+  InsertVp8(seq_num + 11, kF, kF  , kT);
+  ASSERT_EQ(5UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 12, kT, kT  , kT);
+  ASSERT_EQ(6UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 13, kF, kT  , kF);
+  InsertVp8(seq_num + 14, kF, kF  , kF);
+  InsertVp8(seq_num + 15, kF, kF  , kF);
+  InsertVp8(seq_num + 16, kF, kF  , kF);
+  InsertVp8(seq_num + 17, kF, kF  , kT);
+  ASSERT_EQ(7UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 18, kF, kT  , kT);
+  ASSERT_EQ(8UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 19, kF, kT  , kF);
+  InsertVp8(seq_num + 20, kF, kF  , kT);
+  ASSERT_EQ(9UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 21, kF, kT  , kT);
+
+  ASSERT_EQ(10UL, frames_from_callback_.size());
+  CheckReferencesVp8(seq_num + 2);
+  CheckReferencesVp8(seq_num + 4, seq_num + 2);
+  CheckReferencesVp8(seq_num + 8, seq_num + 4);
+  CheckReferencesVp8(seq_num + 9, seq_num + 8);
+  CheckReferencesVp8(seq_num + 11, seq_num + 9);
+  CheckReferencesVp8(seq_num + 12);
+  CheckReferencesVp8(seq_num + 17, seq_num + 12);
+  CheckReferencesVp8(seq_num + 18, seq_num + 17);
+  CheckReferencesVp8(seq_num + 20, seq_num + 18);
+  CheckReferencesVp8(seq_num + 21, seq_num + 20);
+}
+
+TEST_F(TestPacketBuffer, Vp8NoPictureIdReordered) {
+  uint16_t seq_num = 0xfffa;
+
+  //        seq_num     , kf, frst, lst
+  InsertVp8(seq_num + 1 , kF, kF  , kF);
+  InsertVp8(seq_num     , kT, kT  , kF);
+  InsertVp8(seq_num + 2 , kF, kF  , kT);
+  InsertVp8(seq_num + 4 , kF, kF  , kT);
+  InsertVp8(seq_num + 6 , kF, kF  , kF);
+  InsertVp8(seq_num + 3 , kF, kT  , kF);
+  InsertVp8(seq_num + 7 , kF, kF  , kF);
+  InsertVp8(seq_num + 5 , kF, kT  , kF);
+  InsertVp8(seq_num + 9 , kF, kT  , kT);
+  InsertVp8(seq_num + 10, kF, kT  , kF);
+  InsertVp8(seq_num + 8 , kF, kF  , kT);
+  InsertVp8(seq_num + 13, kF, kT  , kF);
+  InsertVp8(seq_num + 14, kF, kF  , kF);
+  InsertVp8(seq_num + 12, kT, kT  , kT);
+  InsertVp8(seq_num + 11, kF, kF  , kT);
+  InsertVp8(seq_num + 16, kF, kF  , kF);
+  InsertVp8(seq_num + 19, kF, kT  , kF);
+  InsertVp8(seq_num + 15, kF, kF  , kF);
+  InsertVp8(seq_num + 17, kF, kF  , kT);
+  InsertVp8(seq_num + 20, kF, kF  , kT);
+  InsertVp8(seq_num + 21, kF, kT  , kT);
+  InsertVp8(seq_num + 18, kF, kT  , kT);
+
+  ASSERT_EQ(10UL, frames_from_callback_.size());
+  CheckReferencesVp8(seq_num + 2);
+  CheckReferencesVp8(seq_num + 4, seq_num + 2);
+  CheckReferencesVp8(seq_num + 8, seq_num + 4);
+  CheckReferencesVp8(seq_num + 9, seq_num + 8);
+  CheckReferencesVp8(seq_num + 11, seq_num + 9);
+  CheckReferencesVp8(seq_num + 12);
+  CheckReferencesVp8(seq_num + 17, seq_num + 12);
+  CheckReferencesVp8(seq_num + 18, seq_num + 17);
+  CheckReferencesVp8(seq_num + 20, seq_num + 18);
+  CheckReferencesVp8(seq_num + 21, seq_num + 20);
+}
+
+
+TEST_F(TestPacketBuffer, Vp8KeyFrameReferences) {
+  uint16_t pid = Rand();
+  //        seq_num, kf, frst, lst, sync, pid, tid, tl0
+  InsertVp8(Rand() , kT, kT  , kT , kF  , pid, 0  , 0);
+
+  ASSERT_EQ(1UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+}
+
+// Test with 1 temporal layer.
+TEST_F(TestPacketBuffer, Vp8TemporalLayers_0) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  //        seq_num    , kf, frst, lst, sync, pid    , tid, tl0
+  InsertVp8(seq_num    , kT, kT  , kT , kF  , pid    , 0  , 1);
+  InsertVp8(seq_num + 1, kF, kT  , kT , kF  , pid + 1, 0  , 2);
+  InsertVp8(seq_num + 2, kF, kT  , kT , kF  , pid + 2, 0  , 3);
+  InsertVp8(seq_num + 3, kF, kT  , kT , kF  , pid + 3, 0  , 4);
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1, pid);
+  CheckReferencesVp8(pid + 2, pid + 1);
+  CheckReferencesVp8(pid + 3, pid + 2);
+}
+
+// Test with 1 temporal layer.
+TEST_F(TestPacketBuffer, Vp8TemporalLayersReordering_0) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  //        seq_num    , kf, frst, lst, sync, pid    , tid, tl0
+  InsertVp8(seq_num    , kT, kT  , kT , kF  , pid    , 0  , 1);
+  InsertVp8(seq_num + 1, kF, kT  , kT , kF  , pid + 1, 0  , 2);
+  InsertVp8(seq_num + 3, kF, kT  , kT , kF  , pid + 3, 0  , 4);
+  InsertVp8(seq_num + 2, kF, kT  , kT , kF  , pid + 2, 0  , 3);
+  InsertVp8(seq_num + 5, kF, kT  , kT , kF  , pid + 5, 0  , 6);
+  InsertVp8(seq_num + 6, kF, kT  , kT , kF  , pid + 6, 0  , 7);
+  InsertVp8(seq_num + 4, kF, kT  , kT , kF  , pid + 4, 0  , 5);
+
+  ASSERT_EQ(7UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1, pid);
+  CheckReferencesVp8(pid + 2, pid + 1);
+  CheckReferencesVp8(pid + 3, pid + 2);
+  CheckReferencesVp8(pid + 4, pid + 3);
+  CheckReferencesVp8(pid + 5, pid + 4);
+  CheckReferencesVp8(pid + 6, pid + 5);
+}
+
+// Test with 2 temporal layers in a 01 pattern.
+TEST_F(TestPacketBuffer, Vp8TemporalLayers_01) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  //        seq_num    , kf, frst, lst, sync, pid    , tid, tl0
+  InsertVp8(seq_num    , kT, kT  , kT , kF  , pid    , 0, 255);
+  InsertVp8(seq_num + 1, kF, kT  , kT , kT  , pid + 1, 1, 255);
+  InsertVp8(seq_num + 2, kF, kT  , kT , kF  , pid + 2, 0, 0);
+  InsertVp8(seq_num + 3, kF, kT  , kT , kF  , pid + 3, 1, 0);
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1, pid);
+  CheckReferencesVp8(pid + 2, pid);
+  CheckReferencesVp8(pid + 3, pid + 1, pid + 2);
+}
+
+// Test with 2 temporal layers in a 01 pattern.
+TEST_F(TestPacketBuffer, Vp8TemporalLayersReordering_01) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  //        seq_num    , kf, frst, lst, sync, pid    , tid, tl0
+  InsertVp8(seq_num + 1, kF, kT  , kT , kT  , pid + 1, 1  , 255);
+  InsertVp8(seq_num    , kT, kT  , kT , kF  , pid    , 0  , 255);
+  InsertVp8(seq_num + 3, kF, kT  , kT , kF  , pid + 3, 1  , 0);
+  InsertVp8(seq_num + 5, kF, kT  , kT , kF  , pid + 5, 1  , 1);
+  InsertVp8(seq_num + 2, kF, kT  , kT , kF  , pid + 2, 0  , 0);
+  InsertVp8(seq_num + 4, kF, kT  , kT , kF  , pid + 4, 0  , 1);
+  InsertVp8(seq_num + 6, kF, kT  , kT , kF  , pid + 6, 0  , 2);
+  InsertVp8(seq_num + 7, kF, kT  , kT , kF  , pid + 7, 1  , 2);
+
+  ASSERT_EQ(8UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1, pid);
+  CheckReferencesVp8(pid + 2, pid);
+  CheckReferencesVp8(pid + 3, pid + 1, pid + 2);
+  CheckReferencesVp8(pid + 4, pid + 2);
+  CheckReferencesVp8(pid + 5, pid + 3, pid + 4);
+  CheckReferencesVp8(pid + 6, pid + 4);
+  CheckReferencesVp8(pid + 7, pid + 5, pid + 6);
+}
+
+// Test with 3 temporal layers in a 0212 pattern.
+TEST_F(TestPacketBuffer, Vp8TemporalLayers_0212) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  //        seq_num     , kf, frst, lst, sync, pid     , tid, tl0
+  InsertVp8(seq_num     , kT, kT  , kT , kF  , pid     , 0  , 55);
+  InsertVp8(seq_num + 1 , kF, kT  , kT , kT  , pid + 1 , 2  , 55);
+  InsertVp8(seq_num + 2 , kF, kT  , kT , kT  , pid + 2 , 1  , 55);
+  InsertVp8(seq_num + 3 , kF, kT  , kT , kF  , pid + 3 , 2  , 55);
+  InsertVp8(seq_num + 4 , kF, kT  , kT , kF  , pid + 4 , 0  , 56);
+  InsertVp8(seq_num + 5 , kF, kT  , kT , kF  , pid + 5 , 2  , 56);
+  InsertVp8(seq_num + 6 , kF, kT  , kT , kF  , pid + 6 , 1  , 56);
+  InsertVp8(seq_num + 7 , kF, kT  , kT , kF  , pid + 7 , 2  , 56);
+  InsertVp8(seq_num + 8 , kF, kT  , kT , kF  , pid + 8 , 0  , 57);
+  InsertVp8(seq_num + 9 , kF, kT  , kT , kT  , pid + 9 , 2  , 57);
+  InsertVp8(seq_num + 10, kF, kT  , kT , kT  , pid + 10, 1  , 57);
+  InsertVp8(seq_num + 11, kF, kT  , kT , kF  , pid + 11, 2  , 57);
+
+  ASSERT_EQ(12UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1 , pid);
+  CheckReferencesVp8(pid + 2 , pid);
+  CheckReferencesVp8(pid + 3 , pid, pid + 1, pid + 2);
+  CheckReferencesVp8(pid + 4 , pid);
+  CheckReferencesVp8(pid + 5 , pid + 2, pid + 3, pid + 4);
+  CheckReferencesVp8(pid + 6 , pid + 2, pid + 4);
+  CheckReferencesVp8(pid + 7 , pid + 4, pid + 5, pid + 6);
+  CheckReferencesVp8(pid + 8 , pid + 4);
+  CheckReferencesVp8(pid + 9 , pid + 8);
+  CheckReferencesVp8(pid + 10, pid + 8);
+  CheckReferencesVp8(pid + 11, pid + 8, pid + 9, pid + 10);
+}
+
+// Test with 3 temporal layers in a 0212 pattern.
+TEST_F(TestPacketBuffer, Vp8TemporalLayersReordering_0212) {
+  uint16_t pid = 126;
+  uint16_t seq_num = Rand();
+
+  //        seq_num     , kf, frst, lst, sync, pid     , tid, tl0
+  InsertVp8(seq_num + 1 , kF, kT  , kT , kT  , pid + 1 , 2  , 55);
+  InsertVp8(seq_num     , kT, kT  , kT , kF  , pid     , 0  , 55);
+  InsertVp8(seq_num + 2 , kF, kT  , kT , kT  , pid + 2 , 1  , 55);
+  InsertVp8(seq_num + 4 , kF, kT  , kT , kF  , pid + 4 , 0  , 56);
+  InsertVp8(seq_num + 5 , kF, kT  , kT , kF  , pid + 5 , 2  , 56);
+  InsertVp8(seq_num + 3 , kF, kT  , kT , kF  , pid + 3 , 2  , 55);
+  InsertVp8(seq_num + 7 , kF, kT  , kT , kF  , pid + 7 , 2  , 56);
+  InsertVp8(seq_num + 9 , kF, kT  , kT , kT  , pid + 9 , 2  , 57);
+  InsertVp8(seq_num + 6 , kF, kT  , kT , kF  , pid + 6 , 1  , 56);
+  InsertVp8(seq_num + 8 , kF, kT  , kT , kF  , pid + 8 , 0  , 57);
+  InsertVp8(seq_num + 11, kF, kT  , kT , kF  , pid + 11, 2  , 57);
+  InsertVp8(seq_num + 10, kF, kT  , kT , kT  , pid + 10, 1  , 57);
+
+  ASSERT_EQ(12UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1 , pid);
+  CheckReferencesVp8(pid + 2 , pid);
+  CheckReferencesVp8(pid + 3 , pid, pid + 1, pid + 2);
+  CheckReferencesVp8(pid + 4 , pid);
+  CheckReferencesVp8(pid + 5 , pid + 2, pid + 3, pid + 4);
+  CheckReferencesVp8(pid + 6 , pid + 2, pid + 4);
+  CheckReferencesVp8(pid + 7 , pid + 4, pid + 5, pid + 6);
+  CheckReferencesVp8(pid + 8 , pid + 4);
+  CheckReferencesVp8(pid + 9 , pid + 8);
+  CheckReferencesVp8(pid + 10, pid + 8);
+  CheckReferencesVp8(pid + 11, pid + 8, pid + 9, pid + 10);
+}
+
+TEST_F(TestPacketBuffer, Vp8InsertManyFrames_0212) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  const int keyframes_to_insert = 50;
+  const int frames_per_keyframe = 120;  // Should be a multiple of 4.
+  uint8_t tl0 = 128;
+
+  for (int k = 0; k < keyframes_to_insert; ++k) {
+    //        seq_num    , keyf, frst, lst, sync, pid    , tid, tl0
+    InsertVp8(seq_num    , kT  , kT  , kT , kF  , pid    , 0  , tl0);
+    InsertVp8(seq_num + 1, kF  , kT  , kT , kT  , pid + 1, 2  , tl0);
+    InsertVp8(seq_num + 2, kF  , kT  , kT , kT  , pid + 2, 1  , tl0);
+    InsertVp8(seq_num + 3, kF  , kT  , kT , kF  , pid + 3, 2  , tl0);
+    CheckReferencesVp8(pid);
+    CheckReferencesVp8(pid + 1, pid);
+    CheckReferencesVp8(pid + 2, pid);
+    CheckReferencesVp8(pid + 3, pid, pid + 1, pid + 2);
+    frames_from_callback_.clear();
+    ++tl0;
+
+    for (int f = 4; f < frames_per_keyframe; f += 4) {
+      uint16_t sf = seq_num + f;
+      uint16_t pidf = pid + f;
+
+      //        seq_num, keyf, frst, lst, sync, pid     , tid, tl0
+      InsertVp8(sf     , kF  , kT  , kT , kF  , pidf    , 0  , tl0);
+      InsertVp8(sf + 1 , kF  , kT  , kT , kF  , pidf + 1, 2  , tl0);
+      InsertVp8(sf + 2 , kF  , kT  , kT , kF  , pidf + 2, 1  , tl0);
+      InsertVp8(sf + 3 , kF  , kT  , kT , kF  , pidf + 3, 2  , tl0);
+      CheckReferencesVp8(pidf, pidf - 4);
+      CheckReferencesVp8(pidf + 1, pidf, pidf - 1, pidf - 2);
+      CheckReferencesVp8(pidf + 2, pidf, pidf - 2);
+      CheckReferencesVp8(pidf + 3, pidf, pidf + 1, pidf + 2);
+      frames_from_callback_.clear();
+      ++tl0;
+    }
+
+    pid += frames_per_keyframe;
+    seq_num += frames_per_keyframe;
+  }
+}
+
+TEST_F(TestPacketBuffer, Vp8LayerSync) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  //        seq_num     , keyf, frst, lst, sync, pid     , tid, tl0
+  InsertVp8(seq_num     , kT  , kT  , kT , kF  , pid     , 0  , 0);
+  InsertVp8(seq_num + 1 , kF  , kT  , kT , kT  , pid + 1 , 1  , 0);
+  InsertVp8(seq_num + 2 , kF  , kT  , kT , kF  , pid + 2 , 0  , 1);
+  ASSERT_EQ(3UL, frames_from_callback_.size());
+
+  InsertVp8(seq_num + 4 , kF  , kT  , kT , kF  , pid + 4 , 0  , 2);
+  InsertVp8(seq_num + 5 , kF  , kT  , kT , kT  , pid + 5 , 1  , 2);
+  InsertVp8(seq_num + 6 , kF  , kT  , kT , kF  , pid + 6 , 0  , 3);
+  InsertVp8(seq_num + 7 , kF  , kT  , kT , kF  , pid + 7 , 1  , 3);
+
+  ASSERT_EQ(7UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1, pid);
+  CheckReferencesVp8(pid + 2, pid);
+  CheckReferencesVp8(pid + 4, pid + 2);
+  CheckReferencesVp8(pid + 5, pid + 4);
+  CheckReferencesVp8(pid + 6, pid + 4);
+  CheckReferencesVp8(pid + 7, pid + 6, pid + 5);
+}
+
+TEST_F(TestPacketBuffer, Vp8InsertLargeFrames) {
+  packet_buffer_.reset(new PacketBuffer(1 << 3, 1 << 12, this));
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+
+  const uint16_t packets_per_frame = 1000;
+  uint16_t current = seq_num;
+  uint16_t end = current + packets_per_frame;
+
+  //        seq_num  , keyf, frst, lst, sync, pid, tid, tl0
+  InsertVp8(current++, kT  , kT  , kF , kF  , pid, 0  , 0);
+  while (current != end)
+    InsertVp8(current++, kF  , kF  , kF , kF  , pid, 0  , 0);
+  InsertVp8(current++, kF  , kF  , kT , kF  , pid, 0  , 0);
+  end = current + packets_per_frame;
+
+  for (int f = 1; f < 4; ++f) {
+    InsertVp8(current++, kF  , kT  , kF , kF  , pid + f, 0, f);
+    while (current != end)
+      InsertVp8(current++, kF  , kF  , kF , kF  , pid + f, 0, f);
+    InsertVp8(current++, kF  , kF  , kT , kF  , pid + f, 0, f);
+    end = current + packets_per_frame;
+  }
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesVp8(pid);
+  CheckReferencesVp8(pid + 1, pid);
+  CheckReferencesVp8(pid + 2, pid + 1);
+  CheckReferencesVp8(pid + 3, pid + 2);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofInsertOneFrame) {
+  uint16_t pid = Rand();
+  uint16_t seq_num = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);
+
+  //           seq_num, keyf, frst, lst, up, pid, sid, tid, tl0, ss
+  InsertVp9Gof(seq_num, kT  , kT  , kT , kF, pid, 0  , 0  , 0  , &ss);
+
+  CheckReferencesVp9(pid, 0);
+}
+
+TEST_F(TestPacketBuffer, Vp9NoPictureIdReordered) {
+  uint16_t sn = 0xfffa;
+
+  //           sn     , kf, frst, lst
+  InsertVp9Gof(sn + 1 , kF, kF  , kF);
+  InsertVp9Gof(sn     , kT, kT  , kF);
+  InsertVp9Gof(sn + 2 , kF, kF  , kT);
+  InsertVp9Gof(sn + 4 , kF, kF  , kT);
+  InsertVp9Gof(sn + 6 , kF, kF  , kF);
+  InsertVp9Gof(sn + 3 , kF, kT  , kF);
+  InsertVp9Gof(sn + 7 , kF, kF  , kF);
+  InsertVp9Gof(sn + 5 , kF, kT  , kF);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT);
+  InsertVp9Gof(sn + 10, kF, kT  , kF);
+  InsertVp9Gof(sn + 8 , kF, kF  , kT);
+  InsertVp9Gof(sn + 13, kF, kT  , kF);
+  InsertVp9Gof(sn + 14, kF, kF  , kF);
+  InsertVp9Gof(sn + 12, kT, kT  , kT);
+  InsertVp9Gof(sn + 11, kF, kF  , kT);
+  InsertVp9Gof(sn + 16, kF, kF  , kF);
+  InsertVp9Gof(sn + 19, kF, kT  , kF);
+  InsertVp9Gof(sn + 15, kF, kF  , kF);
+  InsertVp9Gof(sn + 17, kF, kF  , kT);
+  InsertVp9Gof(sn + 20, kF, kF  , kT);
+  InsertVp9Gof(sn + 21, kF, kT  , kT);
+  InsertVp9Gof(sn + 18, kF, kT  , kT);
+
+  ASSERT_EQ(10UL, frames_from_callback_.size());
+  CheckReferencesVp9(sn + 2 , 0);
+  CheckReferencesVp9(sn + 4 , 0, sn + 2);
+  CheckReferencesVp9(sn + 8 , 0, sn + 4);
+  CheckReferencesVp9(sn + 9 , 0, sn + 8);
+  CheckReferencesVp9(sn + 11, 0, sn + 9);
+  CheckReferencesVp9(sn + 12, 0);
+  CheckReferencesVp9(sn + 17, 0, sn + 12);
+  CheckReferencesVp9(sn + 18, 0, sn + 17);
+  CheckReferencesVp9(sn + 20, 0, sn + 18);
+  CheckReferencesVp9(sn + 21, 0, sn + 20);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayers_0) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);  // Only 1 spatial layer.
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 0  , 3);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 4);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 0  , 5);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kF, pid + 6 , 0  , 0  , 6);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 0  , 7);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kF, pid + 8 , 0  , 0  , 8);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 0  , 9);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 0  , 10);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kF, pid + 11, 0  , 0  , 11);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 12);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 0  , 13);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 0  , 14);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 0  , 15);
+  InsertVp9Gof(sn + 16, kF, kT  , kT , kF, pid + 16, 0  , 0  , 16);
+  InsertVp9Gof(sn + 17, kF, kT  , kT , kF, pid + 17, 0  , 0  , 17);
+  InsertVp9Gof(sn + 18, kF, kT  , kT , kF, pid + 18, 0  , 0  , 18);
+  InsertVp9Gof(sn + 19, kF, kT  , kT , kF, pid + 19, 0  , 0  , 19);
+
+  ASSERT_EQ(20UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid + 1);
+  CheckReferencesVp9(pid + 3 , 0, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid + 3);
+  CheckReferencesVp9(pid + 5 , 0, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 5);
+  CheckReferencesVp9(pid + 7 , 0, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 7);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 9);
+  CheckReferencesVp9(pid + 11, 0, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 11);
+  CheckReferencesVp9(pid + 13, 0, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 13);
+  CheckReferencesVp9(pid + 15, 0, pid + 14);
+  CheckReferencesVp9(pid + 16, 0, pid + 15);
+  CheckReferencesVp9(pid + 17, 0, pid + 16);
+  CheckReferencesVp9(pid + 18, 0, pid + 17);
+  CheckReferencesVp9(pid + 19, 0, pid + 18);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayersReordered_0) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);  // Only 1 spatial layer.
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 0  , 1);
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 4);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 0  , 3);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 0  , 5);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 0  , 7);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kF, pid + 6 , 0  , 0  , 6);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kF, pid + 8 , 0  , 0  , 8);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 0  , 10);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 0  , 13);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kF, pid + 11, 0  , 0  , 11);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 0  , 9);
+  InsertVp9Gof(sn + 16, kF, kT  , kT , kF, pid + 16, 0  , 0  , 16);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 0  , 14);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 0  , 15);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 12);
+  InsertVp9Gof(sn + 17, kF, kT  , kT , kF, pid + 17, 0  , 0  , 17);
+  InsertVp9Gof(sn + 19, kF, kT  , kT , kF, pid + 19, 0  , 0  , 19);
+  InsertVp9Gof(sn + 18, kF, kT  , kT , kF, pid + 18, 0  , 0  , 18);
+
+  ASSERT_EQ(20UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid + 1);
+  CheckReferencesVp9(pid + 3 , 0, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid + 3);
+  CheckReferencesVp9(pid + 5 , 0, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 5);
+  CheckReferencesVp9(pid + 7 , 0, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 7);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 9);
+  CheckReferencesVp9(pid + 11, 0, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 11);
+  CheckReferencesVp9(pid + 13, 0, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 13);
+  CheckReferencesVp9(pid + 15, 0, pid + 14);
+  CheckReferencesVp9(pid + 16, 0, pid + 15);
+  CheckReferencesVp9(pid + 17, 0, pid + 16);
+  CheckReferencesVp9(pid + 18, 0, pid + 17);
+  CheckReferencesVp9(pid + 19, 0, pid + 18);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayers_01) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);  // 0101 pattern
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 1  , 0);
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 1  , 1);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 1  , 2);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kF, pid + 6 , 0  , 0  , 3);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 1  , 3);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kF, pid + 8 , 0  , 0  , 4);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 1  , 4);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 0  , 5);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kF, pid + 11, 0  , 1  , 5);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 6);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 1  , 6);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 0  , 7);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 1  , 7);
+  InsertVp9Gof(sn + 16, kF, kT  , kT , kF, pid + 16, 0  , 0  , 8);
+  InsertVp9Gof(sn + 17, kF, kT  , kT , kF, pid + 17, 0  , 1  , 8);
+  InsertVp9Gof(sn + 18, kF, kT  , kT , kF, pid + 18, 0  , 0  , 9);
+  InsertVp9Gof(sn + 19, kF, kT  , kT , kF, pid + 19, 0  , 1  , 9);
+
+  ASSERT_EQ(20UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid);
+  CheckReferencesVp9(pid + 3 , 0, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid + 2);
+  CheckReferencesVp9(pid + 5 , 0, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 4);
+  CheckReferencesVp9(pid + 7 , 0, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 6);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 8);
+  CheckReferencesVp9(pid + 11, 0, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 10);
+  CheckReferencesVp9(pid + 13, 0, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 12);
+  CheckReferencesVp9(pid + 15, 0, pid + 14);
+  CheckReferencesVp9(pid + 16, 0, pid + 14);
+  CheckReferencesVp9(pid + 17, 0, pid + 16);
+  CheckReferencesVp9(pid + 18, 0, pid + 16);
+  CheckReferencesVp9(pid + 19, 0, pid + 18);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayersReordered_01) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);  // 01 pattern
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 1  , 0);
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 1  , 1);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 1  , 2);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 1  , 3);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kF, pid + 6 , 0  , 0  , 3);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 0  , 5);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kF, pid + 8 , 0  , 0  , 4);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 1  , 4);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kF, pid + 11, 0  , 1  , 5);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 1  , 6);
+  InsertVp9Gof(sn + 16, kF, kT  , kT , kF, pid + 16, 0  , 0  , 8);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 6);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 0  , 7);
+  InsertVp9Gof(sn + 17, kF, kT  , kT , kF, pid + 17, 0  , 1  , 8);
+  InsertVp9Gof(sn + 19, kF, kT  , kT , kF, pid + 19, 0  , 1  , 9);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 1  , 7);
+  InsertVp9Gof(sn + 18, kF, kT  , kT , kF, pid + 18, 0  , 0  , 9);
+
+  ASSERT_EQ(20UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid);
+  CheckReferencesVp9(pid + 3 , 0, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid + 2);
+  CheckReferencesVp9(pid + 5 , 0, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 4);
+  CheckReferencesVp9(pid + 7 , 0, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 6);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 8);
+  CheckReferencesVp9(pid + 11, 0, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 10);
+  CheckReferencesVp9(pid + 13, 0, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 12);
+  CheckReferencesVp9(pid + 15, 0, pid + 14);
+  CheckReferencesVp9(pid + 16, 0, pid + 14);
+  CheckReferencesVp9(pid + 17, 0, pid + 16);
+  CheckReferencesVp9(pid + 18, 0, pid + 16);
+  CheckReferencesVp9(pid + 19, 0, pid + 18);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayers_0212) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode3);  // 0212 pattern
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 2  , 0);
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 1  , 0);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 2  , 0);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kF, pid + 6 , 0  , 1  , 1);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kF, pid + 8 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 2  , 2);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 1  , 2);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kF, pid + 11, 0  , 2  , 2);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 3);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 2  , 3);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 1  , 3);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 2  , 3);
+  InsertVp9Gof(sn + 16, kF, kT  , kT , kF, pid + 16, 0  , 0  , 4);
+  InsertVp9Gof(sn + 17, kF, kT  , kT , kF, pid + 17, 0  , 2  , 4);
+  InsertVp9Gof(sn + 18, kF, kT  , kT , kF, pid + 18, 0  , 1  , 4);
+  InsertVp9Gof(sn + 19, kF, kT  , kT , kF, pid + 19, 0  , 2  , 4);
+
+  ASSERT_EQ(20UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid);
+  CheckReferencesVp9(pid + 3 , 0, pid + 1, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid);
+  CheckReferencesVp9(pid + 5 , 0, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 4);
+  CheckReferencesVp9(pid + 7 , 0, pid + 5, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 4);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 8);
+  CheckReferencesVp9(pid + 11, 0, pid + 9, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 8);
+  CheckReferencesVp9(pid + 13, 0, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 12);
+  CheckReferencesVp9(pid + 15, 0, pid + 13, pid + 14);
+  CheckReferencesVp9(pid + 16, 0, pid + 12);
+  CheckReferencesVp9(pid + 17, 0, pid + 16);
+  CheckReferencesVp9(pid + 18, 0, pid + 16);
+  CheckReferencesVp9(pid + 19, 0, pid + 17, pid + 18);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayersReordered_0212) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode3);  // 0212 pattern
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 1  , 0);
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 2  , 0);
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 2  , 0);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kF, pid + 6 , 0  , 1  , 1);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 2  , 2);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kF, pid + 8 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kF, pid + 11, 0  , 2  , 2);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 1  , 2);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 2  , 3);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 3);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 1  , 3);
+  InsertVp9Gof(sn + 16, kF, kT  , kT , kF, pid + 16, 0  , 0  , 4);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 2  , 3);
+  InsertVp9Gof(sn + 17, kF, kT  , kT , kF, pid + 17, 0  , 2  , 4);
+  InsertVp9Gof(sn + 19, kF, kT  , kT , kF, pid + 19, 0  , 2  , 4);
+  InsertVp9Gof(sn + 18, kF, kT  , kT , kF, pid + 18, 0  , 1  , 4);
+
+  ASSERT_EQ(20UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid);
+  CheckReferencesVp9(pid + 3 , 0, pid + 1, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid);
+  CheckReferencesVp9(pid + 5 , 0, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 4);
+  CheckReferencesVp9(pid + 7 , 0, pid + 5, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 4);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 8);
+  CheckReferencesVp9(pid + 11, 0, pid + 9, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 8);
+  CheckReferencesVp9(pid + 13, 0, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 12);
+  CheckReferencesVp9(pid + 15, 0, pid + 13, pid + 14);
+  CheckReferencesVp9(pid + 16, 0, pid + 12);
+  CheckReferencesVp9(pid + 17, 0, pid + 16);
+  CheckReferencesVp9(pid + 18, 0, pid + 16);
+  CheckReferencesVp9(pid + 19, 0, pid + 17, pid + 18);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayersUpSwitch_02120212) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode4);  // 02120212 pattern
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 2  , 0);
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 1  , 0);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 2  , 0);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kT, pid + 6 , 0  , 1  , 1);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kT, pid + 8 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 2  , 2);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 1  , 2);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kT, pid + 11, 0  , 2  , 2);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 3);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 2  , 3);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 1  , 3);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 2  , 3);
+
+  ASSERT_EQ(16UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid);
+  CheckReferencesVp9(pid + 3 , 0, pid + 1, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid);
+  CheckReferencesVp9(pid + 5 , 0, pid + 3, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 2, pid + 4);
+  CheckReferencesVp9(pid + 7 , 0, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 4);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 8);
+  CheckReferencesVp9(pid + 11, 0, pid + 9, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 8);
+  CheckReferencesVp9(pid + 13, 0, pid + 11, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 10, pid + 12);
+  CheckReferencesVp9(pid + 15, 0, pid + 13, pid + 14);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayersUpSwitchReordered_02120212) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode4);  // 02120212 pattern
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 2  , 0);
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 1  , 0);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 2  , 0);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 2  , 1);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 2  , 2);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kT, pid + 6 , 0  , 1  , 1);
+  InsertVp9Gof(sn + 12, kF, kT  , kT , kF, pid + 12, 0  , 0  , 3);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 1  , 2);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kT, pid + 8 , 0  , 0  , 2);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kT, pid + 11, 0  , 2  , 2);
+  InsertVp9Gof(sn + 13, kF, kT  , kT , kF, pid + 13, 0  , 2  , 3);
+  InsertVp9Gof(sn + 15, kF, kT  , kT , kF, pid + 15, 0  , 2  , 3);
+  InsertVp9Gof(sn + 14, kF, kT  , kT , kF, pid + 14, 0  , 1  , 3);
+
+  ASSERT_EQ(16UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid);
+  CheckReferencesVp9(pid + 3 , 0, pid + 1, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid);
+  CheckReferencesVp9(pid + 5 , 0, pid + 3, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 2, pid + 4);
+  CheckReferencesVp9(pid + 7 , 0, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 4);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 8);
+  CheckReferencesVp9(pid + 11, 0, pid + 9, pid + 10);
+  CheckReferencesVp9(pid + 12, 0, pid + 8);
+  CheckReferencesVp9(pid + 13, 0, pid + 11, pid + 12);
+  CheckReferencesVp9(pid + 14, 0, pid + 10, pid + 12);
+  CheckReferencesVp9(pid + 15, 0, pid + 13, pid + 14);
+}
+
+TEST_F(TestPacketBuffer, Vp9GofTemporalLayersReordered_01_0212) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);  // 01 pattern
+
+  //           sn     , kf, frst, lst, up, pid     , sid, tid, tl0, ss
+  InsertVp9Gof(sn + 1 , kF, kT  , kT , kF, pid + 1 , 0  , 1  , 0);
+  InsertVp9Gof(sn     , kT, kT  , kT , kF, pid     , 0  , 0  , 0  , &ss);
+  InsertVp9Gof(sn + 3 , kF, kT  , kT , kF, pid + 3 , 0  , 1  , 1);
+  InsertVp9Gof(sn + 6 , kF, kT  , kT , kF, pid + 6 , 0  , 1  , 2);
+  ss.SetGofInfoVP9(kTemporalStructureMode3);  // 0212 pattern
+  InsertVp9Gof(sn + 4 , kF, kT  , kT , kF, pid + 4 , 0  , 0  , 2  , &ss);
+  InsertVp9Gof(sn + 2 , kF, kT  , kT , kF, pid + 2 , 0  , 0  , 1);
+  InsertVp9Gof(sn + 5 , kF, kT  , kT , kF, pid + 5 , 0  , 2  , 2);
+  InsertVp9Gof(sn + 8 , kF, kT  , kT , kF, pid + 8 , 0  , 0  , 3);
+  InsertVp9Gof(sn + 10, kF, kT  , kT , kF, pid + 10, 0  , 1  , 3);
+  InsertVp9Gof(sn + 7 , kF, kT  , kT , kF, pid + 7 , 0  , 2  , 2);
+  InsertVp9Gof(sn + 11, kF, kT  , kT , kF, pid + 11, 0  , 2  , 3);
+  InsertVp9Gof(sn + 9 , kF, kT  , kT , kF, pid + 9 , 0  , 2  , 3);
+
+  ASSERT_EQ(12UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+  CheckReferencesVp9(pid + 1 , 0, pid);
+  CheckReferencesVp9(pid + 2 , 0, pid);
+  CheckReferencesVp9(pid + 3 , 0, pid + 2);
+  CheckReferencesVp9(pid + 4 , 0, pid);
+  CheckReferencesVp9(pid + 5 , 0, pid + 4);
+  CheckReferencesVp9(pid + 6 , 0, pid + 4);
+  CheckReferencesVp9(pid + 7 , 0, pid + 5, pid + 6);
+  CheckReferencesVp9(pid + 8 , 0, pid + 4);
+  CheckReferencesVp9(pid + 9 , 0, pid + 8);
+  CheckReferencesVp9(pid + 10, 0, pid + 8);
+  CheckReferencesVp9(pid + 11, 0, pid + 9, pid + 10);
+}
+
+TEST_F(TestPacketBuffer, Vp9FlexibleModeOneFrame) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+
+  //            sn, kf, frst, lst, intr, pid, sid, tid, tl0
+  InsertVp9Flex(sn, kT, kT  , kT , kF  , pid, 0  , 0  , 0);
+
+  ASSERT_EQ(1UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid, 0);
+}
+
+TEST_F(TestPacketBuffer, Vp9FlexibleModeTwoSpatialLayers) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+
+  //            sn     , kf, frst, lst, intr, pid    , sid, tid, tl0, refs
+  InsertVp9Flex(sn     , kT, kT  , kT , kF  , pid    , 0  , 0  , 0);
+  InsertVp9Flex(sn + 1 , kT, kT  , kT , kT  , pid    , 1  , 0  , 0);
+  InsertVp9Flex(sn + 2 , kF, kT  , kT , kF  , pid + 1, 1  , 0  , 0  , {1});
+  InsertVp9Flex(sn + 3 , kF, kT  , kT , kF  , pid + 2, 0  , 0  , 1  , {2});
+  InsertVp9Flex(sn + 4 , kF, kT  , kT , kF  , pid + 2, 1  , 0  , 1  , {1});
+  InsertVp9Flex(sn + 5 , kF, kT  , kT , kF  , pid + 3, 1  , 0  , 1  , {1});
+  InsertVp9Flex(sn + 6 , kF, kT  , kT , kF  , pid + 4, 0  , 0  , 2  , {2});
+  InsertVp9Flex(sn + 7 , kF, kT  , kT , kF  , pid + 4, 1  , 0  , 2  , {1});
+  InsertVp9Flex(sn + 8 , kF, kT  , kT , kF  , pid + 5, 1  , 0  , 2  , {1});
+  InsertVp9Flex(sn + 9 , kF, kT  , kT , kF  , pid + 6, 0  , 0  , 3  , {2});
+  InsertVp9Flex(sn + 10, kF, kT  , kT , kF  , pid + 6, 1  , 0  , 3  , {1});
+  InsertVp9Flex(sn + 11, kF, kT  , kT , kF  , pid + 7, 1  , 0  , 3  , {1});
+  InsertVp9Flex(sn + 12, kF, kT  , kT , kF  , pid + 8, 0  , 0  , 4  , {2});
+  InsertVp9Flex(sn + 13, kF, kT  , kT , kF  , pid + 8, 1  , 0  , 4  , {1});
+
+  ASSERT_EQ(14UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid    , 0);
+  CheckReferencesVp9(pid    , 1);
+  CheckReferencesVp9(pid + 1, 1, pid);
+  CheckReferencesVp9(pid + 2, 0, pid);
+  CheckReferencesVp9(pid + 2, 1, pid + 1);
+  CheckReferencesVp9(pid + 3, 1, pid + 2);
+  CheckReferencesVp9(pid + 4, 0, pid + 2);
+  CheckReferencesVp9(pid + 4, 1, pid + 3);
+  CheckReferencesVp9(pid + 5, 1, pid + 4);
+  CheckReferencesVp9(pid + 6, 0, pid + 4);
+  CheckReferencesVp9(pid + 6, 1, pid + 5);
+  CheckReferencesVp9(pid + 7, 1, pid + 6);
+  CheckReferencesVp9(pid + 8, 0, pid + 6);
+  CheckReferencesVp9(pid + 8, 1, pid + 7);
+}
+
+TEST_F(TestPacketBuffer, Vp9FlexibleModeTwoSpatialLayersReordered) {
+  uint16_t pid = Rand();
+  uint16_t sn = Rand();
+
+  //            sn     , kf, frst, lst, intr, pid    , sid, tid, tl0, refs
+  InsertVp9Flex(sn + 1 , kT, kT  , kT , kT  , pid    , 1  , 0  , 0);
+  InsertVp9Flex(sn + 2 , kF, kT  , kT , kF  , pid + 1, 1  , 0  , 0  , {1});
+  InsertVp9Flex(sn     , kT, kT  , kT , kF  , pid    , 0  , 0  , 0);
+  InsertVp9Flex(sn + 4 , kF, kT  , kT , kF  , pid + 2, 1  , 0  , 1  , {1});
+  InsertVp9Flex(sn + 5 , kF, kT  , kT , kF  , pid + 3, 1  , 0  , 1  , {1});
+  InsertVp9Flex(sn + 3 , kF, kT  , kT , kF  , pid + 2, 0  , 0  , 1  , {2});
+  InsertVp9Flex(sn + 7 , kF, kT  , kT , kF  , pid + 4, 1  , 0  , 2  , {1});
+  InsertVp9Flex(sn + 6 , kF, kT  , kT , kF  , pid + 4, 0  , 0  , 2  , {2});
+  InsertVp9Flex(sn + 8 , kF, kT  , kT , kF  , pid + 5, 1  , 0  , 2  , {1});
+  InsertVp9Flex(sn + 9 , kF, kT  , kT , kF  , pid + 6, 0  , 0  , 3  , {2});
+  InsertVp9Flex(sn + 11, kF, kT  , kT , kF  , pid + 7, 1  , 0  , 3  , {1});
+  InsertVp9Flex(sn + 10, kF, kT  , kT , kF  , pid + 6, 1  , 0  , 3  , {1});
+  InsertVp9Flex(sn + 13, kF, kT  , kT , kF  , pid + 8, 1  , 0  , 4  , {1});
+  InsertVp9Flex(sn + 12, kF, kT  , kT , kF  , pid + 8, 0  , 0  , 4  , {2});
+
+  ASSERT_EQ(14UL, frames_from_callback_.size());
+  CheckReferencesVp9(pid    , 0);
+  CheckReferencesVp9(pid    , 1);
+  CheckReferencesVp9(pid + 1, 1, pid);
+  CheckReferencesVp9(pid + 2, 0, pid);
+  CheckReferencesVp9(pid + 2, 1, pid + 1);
+  CheckReferencesVp9(pid + 3, 1, pid + 2);
+  CheckReferencesVp9(pid + 4, 0, pid + 2);
+  CheckReferencesVp9(pid + 4, 1, pid + 3);
+  CheckReferencesVp9(pid + 5, 1, pid + 4);
+  CheckReferencesVp9(pid + 6, 0, pid + 4);
+  CheckReferencesVp9(pid + 6, 1, pid + 5);
+  CheckReferencesVp9(pid + 7, 1, pid + 6);
+  CheckReferencesVp9(pid + 8, 0, pid + 6);
+  CheckReferencesVp9(pid + 8, 1, pid + 7);
 }
 
 }  // namespace video_coding
diff --git a/chromium/third_party/webrtc/modules/video_coding/qm_select.cc b/chromium/third_party/webrtc/modules/video_coding/qm_select.cc
deleted file mode 100644
index 9da42bb33c6..00000000000
--- a/chromium/third_party/webrtc/modules/video_coding/qm_select.cc
+++ /dev/null
@@ -1,953 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_coding/qm_select.h"
-
-#include <math.h>
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/video_coding/include/video_coding_defines.h"
-#include "webrtc/modules/video_coding/internal_defines.h"
-#include "webrtc/modules/video_coding/qm_select_data.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-// QM-METHOD class
-
-VCMQmMethod::VCMQmMethod()
-    : content_metrics_(NULL),
-      width_(0),
-      height_(0),
-      user_frame_rate_(0.0f),
-      native_width_(0),
-      native_height_(0),
-      native_frame_rate_(0.0f),
-      image_type_(kVGA),
-      framerate_level_(kFrameRateHigh),
-      init_(false) {
-  ResetQM();
-}
-
-VCMQmMethod::~VCMQmMethod() {}
-
-void VCMQmMethod::ResetQM() {
-  aspect_ratio_ = 1.0f;
-  motion_.Reset();
-  spatial_.Reset();
-  content_class_ = 0;
-}
-
-uint8_t VCMQmMethod::ComputeContentClass() {
-  ComputeMotionNFD();
-  ComputeSpatial();
-  return content_class_ = 3 * motion_.level + spatial_.level;
-}
-
-void VCMQmMethod::UpdateContent(const VideoContentMetrics* contentMetrics) {
-  content_metrics_ = contentMetrics;
-}
-
-void VCMQmMethod::ComputeMotionNFD() {
-  if (content_metrics_) {
-    motion_.value = content_metrics_->motion_magnitude;
-  }
-  // Determine motion level.
-  if (motion_.value < kLowMotionNfd) {
-    motion_.level = kLow;
-  } else if (motion_.value > kHighMotionNfd) {
-    motion_.level = kHigh;
-  } else {
-    motion_.level = kDefault;
-  }
-}
-
-void VCMQmMethod::ComputeSpatial() {
-  float spatial_err = 0.0;
-  float spatial_err_h = 0.0;
-  float spatial_err_v = 0.0;
-  if (content_metrics_) {
-    spatial_err = content_metrics_->spatial_pred_err;
-    spatial_err_h = content_metrics_->spatial_pred_err_h;
-    spatial_err_v = content_metrics_->spatial_pred_err_v;
-  }
-  // Spatial measure: take average of 3 prediction errors.
-  spatial_.value = (spatial_err + spatial_err_h + spatial_err_v) / 3.0f;
-
-  // Reduce thresholds for large scenes/higher pixel correlation.
-  float scale2 = image_type_ > kVGA ? kScaleTexture : 1.0;
-
-  if (spatial_.value > scale2 * kHighTexture) {
-    spatial_.level = kHigh;
-  } else if (spatial_.value < scale2 * kLowTexture) {
-    spatial_.level = kLow;
-  } else {
-    spatial_.level = kDefault;
-  }
-}
-
-ImageType VCMQmMethod::GetImageType(uint16_t width, uint16_t height) {
-  // Get the image type for the encoder frame size.
-  uint32_t image_size = width * height;
-  if (image_size == kSizeOfImageType[kQCIF]) {
-    return kQCIF;
-  } else if (image_size == kSizeOfImageType[kHCIF]) {
-    return kHCIF;
-  } else if (image_size == kSizeOfImageType[kQVGA]) {
-    return kQVGA;
-  } else if (image_size == kSizeOfImageType[kCIF]) {
-    return kCIF;
-  } else if (image_size == kSizeOfImageType[kHVGA]) {
-    return kHVGA;
-  } else if (image_size == kSizeOfImageType[kVGA]) {
-    return kVGA;
-  } else if (image_size == kSizeOfImageType[kQFULLHD]) {
-    return kQFULLHD;
-  } else if (image_size == kSizeOfImageType[kWHD]) {
-    return kWHD;
-  } else if (image_size == kSizeOfImageType[kFULLHD]) {
-    return kFULLHD;
-  } else {
-    // No exact match, find closet one.
-    return FindClosestImageType(width, height);
-  }
-}
-
-ImageType VCMQmMethod::FindClosestImageType(uint16_t width, uint16_t height) {
-  float size = static_cast<float>(width * height);
-  float min = size;
-  int isel = 0;
-  for (int i = 0; i < kNumImageTypes; ++i) {
-    float dist = fabs(size - kSizeOfImageType[i]);
-    if (dist < min) {
-      min = dist;
-      isel = i;
-    }
-  }
-  return static_cast<ImageType>(isel);
-}
-
-FrameRateLevelClass VCMQmMethod::FrameRateLevel(float avg_framerate) {
-  if (avg_framerate <= kLowFrameRate) {
-    return kFrameRateLow;
-  } else if (avg_framerate <= kMiddleFrameRate) {
-    return kFrameRateMiddle1;
-  } else if (avg_framerate <= kHighFrameRate) {
-    return kFrameRateMiddle2;
-  } else {
-    return kFrameRateHigh;
-  }
-}
-
-// RESOLUTION CLASS
-
-VCMQmResolution::VCMQmResolution() : qm_(new VCMResolutionScale()) {
-  Reset();
-}
-
-VCMQmResolution::~VCMQmResolution() {
-  delete qm_;
-}
-
-void VCMQmResolution::ResetRates() {
-  sum_target_rate_ = 0.0f;
-  sum_incoming_framerate_ = 0.0f;
-  sum_rate_MM_ = 0.0f;
-  sum_rate_MM_sgn_ = 0.0f;
-  sum_packet_loss_ = 0.0f;
-  buffer_level_ = kInitBufferLevel * target_bitrate_;
-  frame_cnt_ = 0;
-  frame_cnt_delta_ = 0;
-  low_buffer_cnt_ = 0;
-  update_rate_cnt_ = 0;
-}
-
-void VCMQmResolution::ResetDownSamplingState() {
-  state_dec_factor_spatial_ = 1.0;
-  state_dec_factor_temporal_ = 1.0;
-  for (int i = 0; i < kDownActionHistorySize; i++) {
-    down_action_history_[i].spatial = kNoChangeSpatial;
-    down_action_history_[i].temporal = kNoChangeTemporal;
-  }
-}
-
-void VCMQmResolution::Reset() {
-  target_bitrate_ = 0.0f;
-  incoming_framerate_ = 0.0f;
-  buffer_level_ = 0.0f;
-  per_frame_bandwidth_ = 0.0f;
-  avg_target_rate_ = 0.0f;
-  avg_incoming_framerate_ = 0.0f;
-  avg_ratio_buffer_low_ = 0.0f;
-  avg_rate_mismatch_ = 0.0f;
-  avg_rate_mismatch_sgn_ = 0.0f;
-  avg_packet_loss_ = 0.0f;
-  encoder_state_ = kStableEncoding;
-  num_layers_ = 1;
-  ResetRates();
-  ResetDownSamplingState();
-  ResetQM();
-}
-
-EncoderState VCMQmResolution::GetEncoderState() {
-  return encoder_state_;
-}
-
-// Initialize state after re-initializing the encoder,
-// i.e., after SetEncodingData() in mediaOpt.
-int VCMQmResolution::Initialize(float bitrate,
-                                float user_framerate,
-                                uint16_t width,
-                                uint16_t height,
-                                int num_layers) {
-  if (user_framerate == 0.0f || width == 0 || height == 0) {
-    return VCM_PARAMETER_ERROR;
-  }
-  Reset();
-  target_bitrate_ = bitrate;
-  incoming_framerate_ = user_framerate;
-  UpdateCodecParameters(user_framerate, width, height);
-  native_width_ = width;
-  native_height_ = height;
-  native_frame_rate_ = user_framerate;
-  num_layers_ = num_layers;
-  // Initial buffer level.
-  buffer_level_ = kInitBufferLevel * target_bitrate_;
-  // Per-frame bandwidth.
-  per_frame_bandwidth_ = target_bitrate_ / user_framerate;
-  init_ = true;
-  return VCM_OK;
-}
-
-void VCMQmResolution::UpdateCodecParameters(float frame_rate,
-                                            uint16_t width,
-                                            uint16_t height) {
-  width_ = width;
-  height_ = height;
-  // |user_frame_rate| is the target frame rate for VPM frame dropper.
-  user_frame_rate_ = frame_rate;
-  image_type_ = GetImageType(width, height);
-}
-
-// Update rate data after every encoded frame.
-void VCMQmResolution::UpdateEncodedSize(size_t encoded_size) {
-  frame_cnt_++;
-  // Convert to Kbps.
-  float encoded_size_kbits = 8.0f * static_cast<float>(encoded_size) / 1000.0f;
-
-  // Update the buffer level:
-  // Note this is not the actual encoder buffer level.
-  // |buffer_level_| is reset to an initial value after SelectResolution is
-  // called, and does not account for frame dropping by encoder or VCM.
-  buffer_level_ += per_frame_bandwidth_ - encoded_size_kbits;
-
-  // Counter for occurrences of low buffer level:
-  // low/negative values means encoder is likely dropping frames.
-  if (buffer_level_ <= kPercBufferThr * kInitBufferLevel * target_bitrate_) {
-    low_buffer_cnt_++;
-  }
-}
-
-// Update various quantities after SetTargetRates in MediaOpt.
-void VCMQmResolution::UpdateRates(float target_bitrate,
-                                  float encoder_sent_rate,
-                                  float incoming_framerate,
-                                  uint8_t packet_loss) {
-  // Sum the target bitrate: this is the encoder rate from previous update
-  // (~1sec), i.e, before the update for next ~1sec.
-  sum_target_rate_ += target_bitrate_;
-  update_rate_cnt_++;
-
-  // Sum the received (from RTCP reports) packet loss rates.
-  sum_packet_loss_ += static_cast<float>(packet_loss / 255.0);
-
-  // Sum the sequence rate mismatch:
-  // Mismatch here is based on the difference between the target rate
-  // used (in previous ~1sec) and the average actual encoding rate measured
-  // at previous ~1sec.
-  float diff = target_bitrate_ - encoder_sent_rate;
-  if (target_bitrate_ > 0.0)
-    sum_rate_MM_ += fabs(diff) / target_bitrate_;
-  int sgnDiff = diff > 0 ? 1 : (diff < 0 ? -1 : 0);
-  // To check for consistent under(+)/over_shooting(-) of target rate.
-  sum_rate_MM_sgn_ += sgnDiff;
-
-  // Update with the current new target and frame rate:
-  // these values are ones the encoder will use for the current/next ~1sec.
-  target_bitrate_ = target_bitrate;
-  incoming_framerate_ = incoming_framerate;
-  sum_incoming_framerate_ += incoming_framerate_;
-  // Update the per_frame_bandwidth:
-  // this is the per_frame_bw for the current/next ~1sec.
-  per_frame_bandwidth_ = 0.0f;
-  if (incoming_framerate_ > 0.0f) {
-    per_frame_bandwidth_ = target_bitrate_ / incoming_framerate_;
-  }
-}
-
-// Select the resolution factors: frame size and frame rate change (qm scales).
-// Selection is for going down in resolution, or for going back up
-// (if a previous down-sampling action was taken).
-
-// In the current version the following constraints are imposed:
-// 1) We only allow for one action, either down or up, at a given time.
-// 2) The possible down-sampling actions are: spatial by 1/2x1/2, 3/4x3/4;
-//    temporal/frame rate reduction by 1/2 and 2/3.
-// 3) The action for going back up is the reverse of last (spatial or temporal)
-//    down-sampling action. The list of down-sampling actions from the
-//    Initialize() state are kept in |down_action_history_|.
-// 4) The total amount of down-sampling (spatial and/or temporal) from the
-//    Initialize() state (native resolution) is limited by various factors.
-int VCMQmResolution::SelectResolution(VCMResolutionScale** qm) {
-  if (!init_) {
-    return VCM_UNINITIALIZED;
-  }
-  if (content_metrics_ == NULL) {
-    Reset();
-    *qm = qm_;
-    return VCM_OK;
-  }
-
-  // Check conditions on down-sampling state.
-  assert(state_dec_factor_spatial_ >= 1.0f);
-  assert(state_dec_factor_temporal_ >= 1.0f);
-  assert(state_dec_factor_spatial_ <= kMaxSpatialDown);
-  assert(state_dec_factor_temporal_ <= kMaxTempDown);
-  assert(state_dec_factor_temporal_ * state_dec_factor_spatial_ <=
-         kMaxTotalDown);
-
-  // Compute content class for selection.
-  content_class_ = ComputeContentClass();
-  // Compute various rate quantities for selection.
-  ComputeRatesForSelection();
-
-  // Get the encoder state.
-  ComputeEncoderState();
-
-  // Default settings: no action.
-  SetDefaultAction();
-  *qm = qm_;
-
-  // Check for going back up in resolution, if we have had some down-sampling
-  // relative to native state in Initialize().
-  if (down_action_history_[0].spatial != kNoChangeSpatial ||
-      down_action_history_[0].temporal != kNoChangeTemporal) {
-    if (GoingUpResolution()) {
-      *qm = qm_;
-      return VCM_OK;
-    }
-  }
-
-  // Check for going down in resolution.
-  if (GoingDownResolution()) {
-    *qm = qm_;
-    return VCM_OK;
-  }
-  return VCM_OK;
-}
-
-void VCMQmResolution::SetDefaultAction() {
-  qm_->codec_width = width_;
-  qm_->codec_height = height_;
-  qm_->frame_rate = user_frame_rate_;
-  qm_->change_resolution_spatial = false;
-  qm_->change_resolution_temporal = false;
-  qm_->spatial_width_fact = 1.0f;
-  qm_->spatial_height_fact = 1.0f;
-  qm_->temporal_fact = 1.0f;
-  action_.spatial = kNoChangeSpatial;
-  action_.temporal = kNoChangeTemporal;
-}
-
-void VCMQmResolution::ComputeRatesForSelection() {
-  avg_target_rate_ = 0.0f;
-  avg_incoming_framerate_ = 0.0f;
-  avg_ratio_buffer_low_ = 0.0f;
-  avg_rate_mismatch_ = 0.0f;
-  avg_rate_mismatch_sgn_ = 0.0f;
-  avg_packet_loss_ = 0.0f;
-  if (frame_cnt_ > 0) {
-    avg_ratio_buffer_low_ =
-        static_cast<float>(low_buffer_cnt_) / static_cast<float>(frame_cnt_);
-  }
-  if (update_rate_cnt_ > 0) {
-    avg_rate_mismatch_ =
-        static_cast<float>(sum_rate_MM_) / static_cast<float>(update_rate_cnt_);
-    avg_rate_mismatch_sgn_ = static_cast<float>(sum_rate_MM_sgn_) /
-                             static_cast<float>(update_rate_cnt_);
-    avg_target_rate_ = static_cast<float>(sum_target_rate_) /
-                       static_cast<float>(update_rate_cnt_);
-    avg_incoming_framerate_ = static_cast<float>(sum_incoming_framerate_) /
-                              static_cast<float>(update_rate_cnt_);
-    avg_packet_loss_ = static_cast<float>(sum_packet_loss_) /
-                       static_cast<float>(update_rate_cnt_);
-  }
-  // For selection we may want to weight some quantities more heavily
-  // with the current (i.e., next ~1sec) rate values.
-  avg_target_rate_ =
-      kWeightRate * avg_target_rate_ + (1.0 - kWeightRate) * target_bitrate_;
-  avg_incoming_framerate_ = kWeightRate * avg_incoming_framerate_ +
-                            (1.0 - kWeightRate) * incoming_framerate_;
-  // Use base layer frame rate for temporal layers: this will favor spatial.
-  assert(num_layers_ > 0);
-  framerate_level_ = FrameRateLevel(avg_incoming_framerate_ /
-                                    static_cast<float>(1 << (num_layers_ - 1)));
-}
-
-void VCMQmResolution::ComputeEncoderState() {
-  // Default.
-  encoder_state_ = kStableEncoding;
-
-  // Assign stressed state if:
-  // 1) occurrences of low buffer levels is high, or
-  // 2) rate mis-match is high, and consistent over-shooting by encoder.
-  if ((avg_ratio_buffer_low_ > kMaxBufferLow) ||
-      ((avg_rate_mismatch_ > kMaxRateMisMatch) &&
-       (avg_rate_mismatch_sgn_ < -kRateOverShoot))) {
-    encoder_state_ = kStressedEncoding;
-  }
-  // Assign easy state if:
-  // 1) rate mis-match is high, and
-  // 2) consistent under-shooting by encoder.
-  if ((avg_rate_mismatch_ > kMaxRateMisMatch) &&
-      (avg_rate_mismatch_sgn_ > kRateUnderShoot)) {
-    encoder_state_ = kEasyEncoding;
-  }
-}
-
-bool VCMQmResolution::GoingUpResolution() {
-  // For going up, we check for undoing the previous down-sampling action.
-
-  float fac_width = kFactorWidthSpatial[down_action_history_[0].spatial];
-  float fac_height = kFactorHeightSpatial[down_action_history_[0].spatial];
-  float fac_temp = kFactorTemporal[down_action_history_[0].temporal];
-  // For going up spatially, we allow for going up by 3/4x3/4 at each stage.
-  // So if the last spatial action was 1/2x1/2 it would be undone in 2 stages.
-  // Modify the fac_width/height for this case.
-  if (down_action_history_[0].spatial == kOneQuarterSpatialUniform) {
-    fac_width = kFactorWidthSpatial[kOneQuarterSpatialUniform] /
-                kFactorWidthSpatial[kOneHalfSpatialUniform];
-    fac_height = kFactorHeightSpatial[kOneQuarterSpatialUniform] /
-                 kFactorHeightSpatial[kOneHalfSpatialUniform];
-  }
-
-  // Check if we should go up both spatially and temporally.
-  if (down_action_history_[0].spatial != kNoChangeSpatial &&
-      down_action_history_[0].temporal != kNoChangeTemporal) {
-    if (ConditionForGoingUp(fac_width, fac_height, fac_temp,
-                            kTransRateScaleUpSpatialTemp)) {
-      action_.spatial = down_action_history_[0].spatial;
-      action_.temporal = down_action_history_[0].temporal;
-      UpdateDownsamplingState(kUpResolution);
-      return true;
-    }
-  }
-  // Check if we should go up either spatially or temporally.
-  bool selected_up_spatial = false;
-  bool selected_up_temporal = false;
-  if (down_action_history_[0].spatial != kNoChangeSpatial) {
-    selected_up_spatial = ConditionForGoingUp(fac_width, fac_height, 1.0f,
-                                              kTransRateScaleUpSpatial);
-  }
-  if (down_action_history_[0].temporal != kNoChangeTemporal) {
-    selected_up_temporal =
-        ConditionForGoingUp(1.0f, 1.0f, fac_temp, kTransRateScaleUpTemp);
-  }
-  if (selected_up_spatial && !selected_up_temporal) {
-    action_.spatial = down_action_history_[0].spatial;
-    action_.temporal = kNoChangeTemporal;
-    UpdateDownsamplingState(kUpResolution);
-    return true;
-  } else if (!selected_up_spatial && selected_up_temporal) {
-    action_.spatial = kNoChangeSpatial;
-    action_.temporal = down_action_history_[0].temporal;
-    UpdateDownsamplingState(kUpResolution);
-    return true;
-  } else if (selected_up_spatial && selected_up_temporal) {
-    PickSpatialOrTemporal();
-    UpdateDownsamplingState(kUpResolution);
-    return true;
-  }
-  return false;
-}
-
-bool VCMQmResolution::ConditionForGoingUp(float fac_width,
-                                          float fac_height,
-                                          float fac_temp,
-                                          float scale_fac) {
-  float estimated_transition_rate_up =
-      GetTransitionRate(fac_width, fac_height, fac_temp, scale_fac);
-  // Go back up if:
-  // 1) target rate is above threshold and current encoder state is stable, or
-  // 2) encoder state is easy (encoder is significantly under-shooting target).
-  if (((avg_target_rate_ > estimated_transition_rate_up) &&
-       (encoder_state_ == kStableEncoding)) ||
-      (encoder_state_ == kEasyEncoding)) {
-    return true;
-  } else {
-    return false;
-  }
-}
-
-bool VCMQmResolution::GoingDownResolution() {
-  float estimated_transition_rate_down =
-      GetTransitionRate(1.0f, 1.0f, 1.0f, 1.0f);
-  float max_rate = kFrameRateFac[framerate_level_] * kMaxRateQm[image_type_];
-  // Resolution reduction if:
-  // (1) target rate is below transition rate, or
-  // (2) encoder is in stressed state and target rate below a max threshold.
-  if ((avg_target_rate_ < estimated_transition_rate_down) ||
-      (encoder_state_ == kStressedEncoding && avg_target_rate_ < max_rate)) {
-    // Get the down-sampling action: based on content class, and how low
-    // average target rate is relative to transition rate.
-    uint8_t spatial_fact =
-        kSpatialAction[content_class_ +
-                       9 * RateClass(estimated_transition_rate_down)];
-    uint8_t temp_fact =
-        kTemporalAction[content_class_ +
-                        9 * RateClass(estimated_transition_rate_down)];
-
-    switch (spatial_fact) {
-      case 4: {
-        action_.spatial = kOneQuarterSpatialUniform;
-        break;
-      }
-      case 2: {
-        action_.spatial = kOneHalfSpatialUniform;
-        break;
-      }
-      case 1: {
-        action_.spatial = kNoChangeSpatial;
-        break;
-      }
-      default: { assert(false); }
-    }
-    switch (temp_fact) {
-      case 3: {
-        action_.temporal = kTwoThirdsTemporal;
-        break;
-      }
-      case 2: {
-        action_.temporal = kOneHalfTemporal;
-        break;
-      }
-      case 1: {
-        action_.temporal = kNoChangeTemporal;
-        break;
-      }
-      default: { assert(false); }
-    }
-    // Only allow for one action (spatial or temporal) at a given time.
-    assert(action_.temporal == kNoChangeTemporal ||
-           action_.spatial == kNoChangeSpatial);
-
-    // Adjust cases not captured in tables, mainly based on frame rate, and
-    // also check for odd frame sizes.
-    AdjustAction();
-
-    // Update down-sampling state.
-    if (action_.spatial != kNoChangeSpatial ||
-        action_.temporal != kNoChangeTemporal) {
-      UpdateDownsamplingState(kDownResolution);
-      return true;
-    }
-  }
-  return false;
-}
-
-float VCMQmResolution::GetTransitionRate(float fac_width,
-                                         float fac_height,
-                                         float fac_temp,
-                                         float scale_fac) {
-  ImageType image_type =
-      GetImageType(static_cast<uint16_t>(fac_width * width_),
-                   static_cast<uint16_t>(fac_height * height_));
-
-  FrameRateLevelClass framerate_level =
-      FrameRateLevel(fac_temp * avg_incoming_framerate_);
-  // If we are checking for going up temporally, and this is the last
-  // temporal action, then use native frame rate.
-  if (down_action_history_[1].temporal == kNoChangeTemporal &&
-      fac_temp > 1.0f) {
-    framerate_level = FrameRateLevel(native_frame_rate_);
-  }
-
-  // The maximum allowed rate below which down-sampling is allowed:
-  // Nominal values based on image format (frame size and frame rate).
-  float max_rate = kFrameRateFac[framerate_level] * kMaxRateQm[image_type];
-
-  uint8_t image_class = image_type > kVGA ? 1 : 0;
-  uint8_t table_index = image_class * 9 + content_class_;
-  // Scale factor for down-sampling transition threshold:
-  // factor based on the content class and the image size.
-  float scaleTransRate = kScaleTransRateQm[table_index];
-  // Threshold bitrate for resolution action.
-  return static_cast<float>(scale_fac * scaleTransRate * max_rate);
-}
-
-void VCMQmResolution::UpdateDownsamplingState(UpDownAction up_down) {
-  if (up_down == kUpResolution) {
-    qm_->spatial_width_fact = 1.0f / kFactorWidthSpatial[action_.spatial];
-    qm_->spatial_height_fact = 1.0f / kFactorHeightSpatial[action_.spatial];
-    // If last spatial action was 1/2x1/2, we undo it in two steps, so the
-    // spatial scale factor in this first step is modified as (4.0/3.0 / 2.0).
-    if (action_.spatial == kOneQuarterSpatialUniform) {
-      qm_->spatial_width_fact = 1.0f *
-                                kFactorWidthSpatial[kOneHalfSpatialUniform] /
-                                kFactorWidthSpatial[kOneQuarterSpatialUniform];
-      qm_->spatial_height_fact =
-          1.0f * kFactorHeightSpatial[kOneHalfSpatialUniform] /
-          kFactorHeightSpatial[kOneQuarterSpatialUniform];
-    }
-    qm_->temporal_fact = 1.0f / kFactorTemporal[action_.temporal];
-    RemoveLastDownAction();
-  } else if (up_down == kDownResolution) {
-    ConstrainAmountOfDownSampling();
-    ConvertSpatialFractionalToWhole();
-    qm_->spatial_width_fact = kFactorWidthSpatial[action_.spatial];
-    qm_->spatial_height_fact = kFactorHeightSpatial[action_.spatial];
-    qm_->temporal_fact = kFactorTemporal[action_.temporal];
-    InsertLatestDownAction();
-  } else {
-    // This function should only be called if either the Up or Down action
-    // has been selected.
-    assert(false);
-  }
-  UpdateCodecResolution();
-  state_dec_factor_spatial_ = state_dec_factor_spatial_ *
-                              qm_->spatial_width_fact *
-                              qm_->spatial_height_fact;
-  state_dec_factor_temporal_ = state_dec_factor_temporal_ * qm_->temporal_fact;
-}
-
-void VCMQmResolution::UpdateCodecResolution() {
-  if (action_.spatial != kNoChangeSpatial) {
-    qm_->change_resolution_spatial = true;
-    qm_->codec_width =
-        static_cast<uint16_t>(width_ / qm_->spatial_width_fact + 0.5f);
-    qm_->codec_height =
-        static_cast<uint16_t>(height_ / qm_->spatial_height_fact + 0.5f);
-    // Size should not exceed native sizes.
-    assert(qm_->codec_width <= native_width_);
-    assert(qm_->codec_height <= native_height_);
-    // New sizes should be multiple of 2, otherwise spatial should not have
-    // been selected.
-    assert(qm_->codec_width % 2 == 0);
-    assert(qm_->codec_height % 2 == 0);
-  }
-  if (action_.temporal != kNoChangeTemporal) {
-    qm_->change_resolution_temporal = true;
-    // Update the frame rate based on the average incoming frame rate.
-    qm_->frame_rate = avg_incoming_framerate_ / qm_->temporal_fact + 0.5f;
-    if (down_action_history_[0].temporal == 0) {
-      // When we undo the last temporal-down action, make sure we go back up
-      // to the native frame rate. Since the incoming frame rate may
-      // fluctuate over time, |avg_incoming_framerate_| scaled back up may
-      // be smaller than |native_frame rate_|.
-      qm_->frame_rate = native_frame_rate_;
-    }
-  }
-}
-
-uint8_t VCMQmResolution::RateClass(float transition_rate) {
-  return avg_target_rate_ < (kFacLowRate * transition_rate)
-             ? 0
-             : (avg_target_rate_ >= transition_rate ? 2 : 1);
-}
-
-// TODO(marpan): Would be better to capture these frame rate adjustments by
-// extending the table data (qm_select_data.h).
-void VCMQmResolution::AdjustAction() {
-  // If the spatial level is default state (neither low or high), motion level
-  // is not high, and spatial action was selected, switch to 2/3 frame rate
-  // reduction if the average incoming frame rate is high.
-  if (spatial_.level == kDefault && motion_.level != kHigh &&
-      action_.spatial != kNoChangeSpatial &&
-      framerate_level_ == kFrameRateHigh) {
-    action_.spatial = kNoChangeSpatial;
-    action_.temporal = kTwoThirdsTemporal;
-  }
-  // If both motion and spatial level are low, and temporal down action was
-  // selected, switch to spatial 3/4x3/4 if the frame rate is not above the
-  // lower middle level (|kFrameRateMiddle1|).
-  if (motion_.level == kLow && spatial_.level == kLow &&
-      framerate_level_ <= kFrameRateMiddle1 &&
-      action_.temporal != kNoChangeTemporal) {
-    action_.spatial = kOneHalfSpatialUniform;
-    action_.temporal = kNoChangeTemporal;
-  }
-  // If spatial action is selected, and there has been too much spatial
-  // reduction already (i.e., 1/4), then switch to temporal action if the
-  // average frame rate is not low.
-  if (action_.spatial != kNoChangeSpatial &&
-      down_action_history_[0].spatial == kOneQuarterSpatialUniform &&
-      framerate_level_ != kFrameRateLow) {
-    action_.spatial = kNoChangeSpatial;
-    action_.temporal = kTwoThirdsTemporal;
-  }
-  // Never use temporal action if number of temporal layers is above 2.
-  if (num_layers_ > 2) {
-    if (action_.temporal != kNoChangeTemporal) {
-      action_.spatial = kOneHalfSpatialUniform;
-    }
-    action_.temporal = kNoChangeTemporal;
-  }
-  // If spatial action was selected, we need to make sure the frame sizes
-  // are multiples of two. Otherwise switch to 2/3 temporal.
-  if (action_.spatial != kNoChangeSpatial && !EvenFrameSize()) {
-    action_.spatial = kNoChangeSpatial;
-    // Only one action (spatial or temporal) is allowed at a given time, so need
-    // to check whether temporal action is currently selected.
-    action_.temporal = kTwoThirdsTemporal;
-  }
-}
-
-void VCMQmResolution::ConvertSpatialFractionalToWhole() {
-  // If 3/4 spatial is selected, check if there has been another 3/4,
-  // and if so, combine them into 1/2. 1/2 scaling is more efficient than 9/16.
-  // Note we define 3/4x3/4 spatial as kOneHalfSpatialUniform.
-  if (action_.spatial == kOneHalfSpatialUniform) {
-    bool found = false;
-    int isel = kDownActionHistorySize;
-    for (int i = 0; i < kDownActionHistorySize; ++i) {
-      if (down_action_history_[i].spatial == kOneHalfSpatialUniform) {
-        isel = i;
-        found = true;
-        break;
-      }
-    }
-    if (found) {
-      action_.spatial = kOneQuarterSpatialUniform;
-      state_dec_factor_spatial_ =
-          state_dec_factor_spatial_ /
-          (kFactorWidthSpatial[kOneHalfSpatialUniform] *
-           kFactorHeightSpatial[kOneHalfSpatialUniform]);
-      // Check if switching to 1/2x1/2 (=1/4) spatial is allowed.
-      ConstrainAmountOfDownSampling();
-      if (action_.spatial == kNoChangeSpatial) {
-        // Not allowed. Go back to 3/4x3/4 spatial.
-        action_.spatial = kOneHalfSpatialUniform;
-        state_dec_factor_spatial_ =
-            state_dec_factor_spatial_ *
-            kFactorWidthSpatial[kOneHalfSpatialUniform] *
-            kFactorHeightSpatial[kOneHalfSpatialUniform];
-      } else {
-        // Switching is allowed. Remove 3/4x3/4 from the history, and update
-        // the frame size.
-        for (int i = isel; i < kDownActionHistorySize - 1; ++i) {
-          down_action_history_[i].spatial = down_action_history_[i + 1].spatial;
-        }
-        width_ = width_ * kFactorWidthSpatial[kOneHalfSpatialUniform];
-        height_ = height_ * kFactorHeightSpatial[kOneHalfSpatialUniform];
-      }
-    }
-  }
-}
-
-// Returns false if the new frame sizes, under the current spatial action,
-// are not multiples of two.
-bool VCMQmResolution::EvenFrameSize() {
-  if (action_.spatial == kOneHalfSpatialUniform) {
-    if ((width_ * 3 / 4) % 2 != 0 || (height_ * 3 / 4) % 2 != 0) {
-      return false;
-    }
-  } else if (action_.spatial == kOneQuarterSpatialUniform) {
-    if ((width_ * 1 / 2) % 2 != 0 || (height_ * 1 / 2) % 2 != 0) {
-      return false;
-    }
-  }
-  return true;
-}
-
-void VCMQmResolution::InsertLatestDownAction() {
-  if (action_.spatial != kNoChangeSpatial) {
-    for (int i = kDownActionHistorySize - 1; i > 0; --i) {
-      down_action_history_[i].spatial = down_action_history_[i - 1].spatial;
-    }
-    down_action_history_[0].spatial = action_.spatial;
-  }
-  if (action_.temporal != kNoChangeTemporal) {
-    for (int i = kDownActionHistorySize - 1; i > 0; --i) {
-      down_action_history_[i].temporal = down_action_history_[i - 1].temporal;
-    }
-    down_action_history_[0].temporal = action_.temporal;
-  }
-}
-
-void VCMQmResolution::RemoveLastDownAction() {
-  if (action_.spatial != kNoChangeSpatial) {
-    // If the last spatial action was 1/2x1/2 we replace it with 3/4x3/4.
-    if (action_.spatial == kOneQuarterSpatialUniform) {
-      down_action_history_[0].spatial = kOneHalfSpatialUniform;
-    } else {
-      for (int i = 0; i < kDownActionHistorySize - 1; ++i) {
-        down_action_history_[i].spatial = down_action_history_[i + 1].spatial;
-      }
-      down_action_history_[kDownActionHistorySize - 1].spatial =
-          kNoChangeSpatial;
-    }
-  }
-  if (action_.temporal != kNoChangeTemporal) {
-    for (int i = 0; i < kDownActionHistorySize - 1; ++i) {
-      down_action_history_[i].temporal = down_action_history_[i + 1].temporal;
-    }
-    down_action_history_[kDownActionHistorySize - 1].temporal =
-        kNoChangeTemporal;
-  }
-}
-
-void VCMQmResolution::ConstrainAmountOfDownSampling() {
-  // Sanity checks on down-sampling selection:
-  // override the settings for too small image size and/or frame rate.
-  // Also check the limit on current down-sampling states.
-
-  float spatial_width_fact = kFactorWidthSpatial[action_.spatial];
-  float spatial_height_fact = kFactorHeightSpatial[action_.spatial];
-  float temporal_fact = kFactorTemporal[action_.temporal];
-  float new_dec_factor_spatial =
-      state_dec_factor_spatial_ * spatial_width_fact * spatial_height_fact;
-  float new_dec_factor_temp = state_dec_factor_temporal_ * temporal_fact;
-
-  // No spatial sampling if current frame size is too small, or if the
-  // amount of spatial down-sampling is above maximum spatial down-action.
-  if ((width_ * height_) <= kMinImageSize ||
-      new_dec_factor_spatial > kMaxSpatialDown) {
-    action_.spatial = kNoChangeSpatial;
-    new_dec_factor_spatial = state_dec_factor_spatial_;
-  }
-  // No frame rate reduction if average frame rate is below some point, or if
-  // the amount of temporal down-sampling is above maximum temporal down-action.
-  if (avg_incoming_framerate_ <= kMinFrameRate ||
-      new_dec_factor_temp > kMaxTempDown) {
-    action_.temporal = kNoChangeTemporal;
-    new_dec_factor_temp = state_dec_factor_temporal_;
-  }
-  // Check if the total (spatial-temporal) down-action is above maximum allowed,
-  // if so, disallow the current selected down-action.
-  if (new_dec_factor_spatial * new_dec_factor_temp > kMaxTotalDown) {
-    if (action_.spatial != kNoChangeSpatial) {
-      action_.spatial = kNoChangeSpatial;
-    } else if (action_.temporal != kNoChangeTemporal) {
-      action_.temporal = kNoChangeTemporal;
-    } else {
-      // We only allow for one action (spatial or temporal) at a given time, so
-      // either spatial or temporal action is selected when this function is
-      // called. If the selected action is disallowed from one of the above
-      // 2 prior conditions (on spatial & temporal max down-action), then this
-      // condition "total down-action > |kMaxTotalDown|" would not be entered.
-      assert(false);
-    }
-  }
-}
-
-void VCMQmResolution::PickSpatialOrTemporal() {
-  // Pick the one that has had the most down-sampling thus far.
-  if (state_dec_factor_spatial_ > state_dec_factor_temporal_) {
-    action_.spatial = down_action_history_[0].spatial;
-    action_.temporal = kNoChangeTemporal;
-  } else {
-    action_.spatial = kNoChangeSpatial;
-    action_.temporal = down_action_history_[0].temporal;
-  }
-}
-
-// TODO(marpan): Update when we allow for directional spatial down-sampling.
-void VCMQmResolution::SelectSpatialDirectionMode(float transition_rate) {
-  // Default is 4/3x4/3
-  // For bit rates well below transitional rate, we select 2x2.
-  if (avg_target_rate_ < transition_rate * kRateRedSpatial2X2) {
-    qm_->spatial_width_fact = 2.0f;
-    qm_->spatial_height_fact = 2.0f;
-  }
-  // Otherwise check prediction errors and aspect ratio.
-  float spatial_err = 0.0f;
-  float spatial_err_h = 0.0f;
-  float spatial_err_v = 0.0f;
-  if (content_metrics_) {
-    spatial_err = content_metrics_->spatial_pred_err;
-    spatial_err_h = content_metrics_->spatial_pred_err_h;
-    spatial_err_v = content_metrics_->spatial_pred_err_v;
-  }
-
-  // Favor 1x2 if aspect_ratio is 16:9.
-  if (aspect_ratio_ >= 16.0f / 9.0f) {
-    // Check if 1x2 has lowest prediction error.
-    if (spatial_err_h < spatial_err && spatial_err_h < spatial_err_v) {
-      qm_->spatial_width_fact = 2.0f;
-      qm_->spatial_height_fact = 1.0f;
-    }
-  }
-  // Check for 4/3x4/3 selection: favor 2x2 over 1x2 and 2x1.
-  if (spatial_err < spatial_err_h * (1.0f + kSpatialErr2x2VsHoriz) &&
-      spatial_err < spatial_err_v * (1.0f + kSpatialErr2X2VsVert)) {
-    qm_->spatial_width_fact = 4.0f / 3.0f;
-    qm_->spatial_height_fact = 4.0f / 3.0f;
-  }
-  // Check for 2x1 selection.
-  if (spatial_err_v < spatial_err_h * (1.0f - kSpatialErrVertVsHoriz) &&
-      spatial_err_v < spatial_err * (1.0f - kSpatialErr2X2VsVert)) {
-    qm_->spatial_width_fact = 1.0f;
-    qm_->spatial_height_fact = 2.0f;
-  }
-}
-
-// ROBUSTNESS CLASS
-
-VCMQmRobustness::VCMQmRobustness() {
-  Reset();
-}
-
-VCMQmRobustness::~VCMQmRobustness() {}
-
-void VCMQmRobustness::Reset() {
-  prev_total_rate_ = 0.0f;
-  prev_rtt_time_ = 0;
-  prev_packet_loss_ = 0;
-  prev_code_rate_delta_ = 0;
-  ResetQM();
-}
-
-// Adjust the FEC rate based on the content and the network state
-// (packet loss rate, total rate/bandwidth, round trip time).
-// Note that packetLoss here is the filtered loss value.
-float VCMQmRobustness::AdjustFecFactor(uint8_t code_rate_delta,
-                                       float total_rate,
-                                       float framerate,
-                                       int64_t rtt_time,
-                                       uint8_t packet_loss) {
-  // Default: no adjustment
-  float adjust_fec = 1.0f;
-  if (content_metrics_ == NULL) {
-    return adjust_fec;
-  }
-  // Compute class state of the content.
-  ComputeMotionNFD();
-  ComputeSpatial();
-
-  // TODO(marpan): Set FEC adjustment factor.
-
-  // Keep track of previous values of network state:
-  // adjustment may be also based on pattern of changes in network state.
-  prev_total_rate_ = total_rate;
-  prev_rtt_time_ = rtt_time;
-  prev_packet_loss_ = packet_loss;
-  prev_code_rate_delta_ = code_rate_delta;
-  return adjust_fec;
-}
-
-// Set the UEP (unequal-protection across packets) on/off for the FEC.
-bool VCMQmRobustness::SetUepProtection(uint8_t code_rate_delta,
-                                       float total_rate,
-                                       uint8_t packet_loss,
-                                       bool frame_type) {
-  // Default.
-  return false;
-}
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/qm_select.h b/chromium/third_party/webrtc/modules/video_coding/qm_select.h
deleted file mode 100644
index 764b5ed8e37..00000000000
--- a/chromium/third_party/webrtc/modules/video_coding/qm_select.h
+++ /dev/null
@@ -1,356 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
-#define WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
-
-#include "webrtc/common_types.h"
-#include "webrtc/typedefs.h"
-
-/******************************************************/
-/* Quality Modes: Resolution and Robustness settings  */
-/******************************************************/
-
-namespace webrtc {
-struct VideoContentMetrics;
-
-struct VCMResolutionScale {
-  VCMResolutionScale()
-      : codec_width(640),
-        codec_height(480),
-        frame_rate(30.0f),
-        spatial_width_fact(1.0f),
-        spatial_height_fact(1.0f),
-        temporal_fact(1.0f),
-        change_resolution_spatial(false),
-        change_resolution_temporal(false) {}
-  uint16_t codec_width;
-  uint16_t codec_height;
-  float frame_rate;
-  float spatial_width_fact;
-  float spatial_height_fact;
-  float temporal_fact;
-  bool change_resolution_spatial;
-  bool change_resolution_temporal;
-};
-
-enum ImageType {
-  kQCIF = 0,  // 176x144
-  kHCIF,      // 264x216 = half(~3/4x3/4) CIF.
-  kQVGA,      // 320x240 = quarter VGA.
-  kCIF,       // 352x288
-  kHVGA,      // 480x360 = half(~3/4x3/4) VGA.
-  kVGA,       // 640x480
-  kQFULLHD,   // 960x540 = quarter FULLHD, and half(~3/4x3/4) WHD.
-  kWHD,       // 1280x720
-  kFULLHD,    // 1920x1080
-  kNumImageTypes
-};
-
-const uint32_t kSizeOfImageType[kNumImageTypes] = {
-    25344, 57024, 76800, 101376, 172800, 307200, 518400, 921600, 2073600};
-
-enum FrameRateLevelClass {
-  kFrameRateLow,
-  kFrameRateMiddle1,
-  kFrameRateMiddle2,
-  kFrameRateHigh
-};
-
-enum ContentLevelClass { kLow, kHigh, kDefault };
-
-struct VCMContFeature {
-  VCMContFeature() : value(0.0f), level(kDefault) {}
-  void Reset() {
-    value = 0.0f;
-    level = kDefault;
-  }
-  float value;
-  ContentLevelClass level;
-};
-
-enum UpDownAction { kUpResolution, kDownResolution };
-
-enum SpatialAction {
-  kNoChangeSpatial,
-  kOneHalfSpatialUniform,     // 3/4 x 3/4: 9/6 ~1/2 pixel reduction.
-  kOneQuarterSpatialUniform,  // 1/2 x 1/2: 1/4 pixel reduction.
-  kNumModesSpatial
-};
-
-enum TemporalAction {
-  kNoChangeTemporal,
-  kTwoThirdsTemporal,  // 2/3 frame rate reduction
-  kOneHalfTemporal,    // 1/2 frame rate reduction
-  kNumModesTemporal
-};
-
-struct ResolutionAction {
-  ResolutionAction() : spatial(kNoChangeSpatial), temporal(kNoChangeTemporal) {}
-  SpatialAction spatial;
-  TemporalAction temporal;
-};
-
-// Down-sampling factors for spatial (width and height), and temporal.
-const float kFactorWidthSpatial[kNumModesSpatial] = {1.0f, 4.0f / 3.0f, 2.0f};
-
-const float kFactorHeightSpatial[kNumModesSpatial] = {1.0f, 4.0f / 3.0f, 2.0f};
-
-const float kFactorTemporal[kNumModesTemporal] = {1.0f, 1.5f, 2.0f};
-
-enum EncoderState {
-  kStableEncoding,    // Low rate mis-match, stable buffer levels.
-  kStressedEncoding,  // Significant over-shooting of target rate,
-                      // Buffer under-flow, etc.
-  kEasyEncoding       // Significant under-shooting of target rate.
-};
-
-// QmMethod class: main class for resolution and robustness settings
-
-class VCMQmMethod {
- public:
-  VCMQmMethod();
-  virtual ~VCMQmMethod();
-
-  // Reset values
-  void ResetQM();
-  virtual void Reset() = 0;
-
-  // Compute content class.
-  uint8_t ComputeContentClass();
-
-  // Update with the content metrics.
-  void UpdateContent(const VideoContentMetrics* content_metrics);
-
-  // Compute spatial texture magnitude and level.
-  // Spatial texture is a spatial prediction error measure.
-  void ComputeSpatial();
-
-  // Compute motion magnitude and level for NFD metric.
-  // NFD is normalized frame difference (normalized by spatial variance).
-  void ComputeMotionNFD();
-
-  // Get the imageType (CIF, VGA, HD, etc) for the system width/height.
-  ImageType GetImageType(uint16_t width, uint16_t height);
-
-  // Return the closest image type.
-  ImageType FindClosestImageType(uint16_t width, uint16_t height);
-
-  // Get the frame rate level.
-  FrameRateLevelClass FrameRateLevel(float frame_rate);
-
- protected:
-  // Content Data.
-  const VideoContentMetrics* content_metrics_;
-
-  // Encoder frame sizes and native frame sizes.
-  uint16_t width_;
-  uint16_t height_;
-  float user_frame_rate_;
-  uint16_t native_width_;
-  uint16_t native_height_;
-  float native_frame_rate_;
-  float aspect_ratio_;
-  // Image type and frame rate leve, for the current encoder resolution.
-  ImageType image_type_;
-  FrameRateLevelClass framerate_level_;
-  // Content class data.
-  VCMContFeature motion_;
-  VCMContFeature spatial_;
-  uint8_t content_class_;
-  bool init_;
-};
-
-// Resolution settings class
-
-class VCMQmResolution : public VCMQmMethod {
- public:
-  VCMQmResolution();
-  virtual ~VCMQmResolution();
-
-  // Reset all quantities.
-  virtual void Reset();
-
-  // Reset rate quantities and counters after every SelectResolution() call.
-  void ResetRates();
-
-  // Reset down-sampling state.
-  void ResetDownSamplingState();
-
-  // Get the encoder state.
-  EncoderState GetEncoderState();
-
-  // Initialize after SetEncodingData in media_opt.
-  int Initialize(float bitrate,
-                 float user_framerate,
-                 uint16_t width,
-                 uint16_t height,
-                 int num_layers);
-
-  // Update the encoder frame size.
-  void UpdateCodecParameters(float frame_rate, uint16_t width, uint16_t height);
-
-  // Update with actual bit rate (size of the latest encoded frame)
-  // and frame type, after every encoded frame.
-  void UpdateEncodedSize(size_t encoded_size);
-
-  // Update with new target bitrate, actual encoder sent rate, frame_rate,
-  // loss rate: every ~1 sec from SetTargetRates in media_opt.
-  void UpdateRates(float target_bitrate,
-                   float encoder_sent_rate,
-                   float incoming_framerate,
-                   uint8_t packet_loss);
-
-  // Extract ST (spatio-temporal) resolution action.
-  // Inputs: qm: Reference to the quality modes pointer.
-  // Output: the spatial and/or temporal scale change.
-  int SelectResolution(VCMResolutionScale** qm);
-
- private:
-  // Set the default resolution action.
-  void SetDefaultAction();
-
-  // Compute rates for the selection of down-sampling action.
-  void ComputeRatesForSelection();
-
-  // Compute the encoder state.
-  void ComputeEncoderState();
-
-  // Return true if the action is to go back up in resolution.
-  bool GoingUpResolution();
-
-  // Return true if the action is to go down in resolution.
-  bool GoingDownResolution();
-
-  // Check the condition for going up in resolution by the scale factors:
-  // |facWidth|, |facHeight|, |facTemp|.
-  // |scaleFac| is a scale factor for the transition rate.
-  bool ConditionForGoingUp(float fac_width,
-                           float fac_height,
-                           float fac_temp,
-                           float scale_fac);
-
-  // Get the bitrate threshold for the resolution action.
-  // The case |facWidth|=|facHeight|=|facTemp|==1 is for down-sampling action.
-  // |scaleFac| is a scale factor for the transition rate.
-  float GetTransitionRate(float fac_width,
-                          float fac_height,
-                          float fac_temp,
-                          float scale_fac);
-
-  // Update the down-sampling state.
-  void UpdateDownsamplingState(UpDownAction up_down);
-
-  // Update the codec frame size and frame rate.
-  void UpdateCodecResolution();
-
-  // Return a state based on average target rate relative transition rate.
-  uint8_t RateClass(float transition_rate);
-
-  // Adjust the action selected from the table.
-  void AdjustAction();
-
-  // Covert 2 stages of 3/4 (=9/16) spatial decimation to 1/2.
-  void ConvertSpatialFractionalToWhole();
-
-  // Returns true if the new frame sizes, under the selected spatial action,
-  // are of even size.
-  bool EvenFrameSize();
-
-  // Insert latest down-sampling action into the history list.
-  void InsertLatestDownAction();
-
-  // Remove the last (first element) down-sampling action from the list.
-  void RemoveLastDownAction();
-
-  // Check constraints on the amount of down-sampling allowed.
-  void ConstrainAmountOfDownSampling();
-
-  // For going up in resolution: pick spatial or temporal action,
-  // if both actions were separately selected.
-  void PickSpatialOrTemporal();
-
-  // Select the directional (1x2 or 2x1) spatial down-sampling action.
-  void SelectSpatialDirectionMode(float transition_rate);
-
-  enum { kDownActionHistorySize = 10 };
-
-  VCMResolutionScale* qm_;
-  // Encoder rate control parameters.
-  float target_bitrate_;
-  float incoming_framerate_;
-  float per_frame_bandwidth_;
-  float buffer_level_;
-
-  // Data accumulated every ~1sec from MediaOpt.
-  float sum_target_rate_;
-  float sum_incoming_framerate_;
-  float sum_rate_MM_;
-  float sum_rate_MM_sgn_;
-  float sum_packet_loss_;
-  // Counters.
-  uint32_t frame_cnt_;
-  uint32_t frame_cnt_delta_;
-  uint32_t update_rate_cnt_;
-  uint32_t low_buffer_cnt_;
-
-  // Resolution state parameters.
-  float state_dec_factor_spatial_;
-  float state_dec_factor_temporal_;
-
-  // Quantities used for selection.
-  float avg_target_rate_;
-  float avg_incoming_framerate_;
-  float avg_ratio_buffer_low_;
-  float avg_rate_mismatch_;
-  float avg_rate_mismatch_sgn_;
-  float avg_packet_loss_;
-  EncoderState encoder_state_;
-  ResolutionAction action_;
-  // Short history of the down-sampling actions from the Initialize() state.
-  // This is needed for going up in resolution. Since the total amount of
-  // down-sampling actions are constrained, the length of the list need not be
-  // large: i.e., (4/3) ^{kDownActionHistorySize} <= kMaxDownSample.
-  ResolutionAction down_action_history_[kDownActionHistorySize];
-  int num_layers_;
-};
-
-// Robustness settings class.
-
-class VCMQmRobustness : public VCMQmMethod {
- public:
-  VCMQmRobustness();
-  ~VCMQmRobustness();
-
-  virtual void Reset();
-
-  // Adjust FEC rate based on content: every ~1 sec from SetTargetRates.
-  // Returns an adjustment factor.
-  float AdjustFecFactor(uint8_t code_rate_delta,
-                        float total_rate,
-                        float framerate,
-                        int64_t rtt_time,
-                        uint8_t packet_loss);
-
-  // Set the UEP protection on/off.
-  bool SetUepProtection(uint8_t code_rate_delta,
-                        float total_rate,
-                        uint8_t packet_loss,
-                        bool frame_type);
-
- private:
-  // Previous state of network parameters.
-  float prev_total_rate_;
-  int64_t prev_rtt_time_;
-  uint8_t prev_packet_loss_;
-  uint8_t prev_code_rate_delta_;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/qm_select_data.h b/chromium/third_party/webrtc/modules/video_coding/qm_select_data.h
deleted file mode 100644
index 49190ef53b9..00000000000
--- a/chromium/third_party/webrtc/modules/video_coding/qm_select_data.h
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
-#define WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
-
-/***************************************************************
-*QMSelectData.h
-* This file includes parameters for content-aware media optimization
-****************************************************************/
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-//
-// PARAMETERS FOR RESOLUTION ADAPTATION
-//
-
-// Initial level of buffer in secs.
-const float kInitBufferLevel = 0.5f;
-
-// Threshold of (max) buffer size below which we consider too low (underflow).
-const float kPercBufferThr = 0.10f;
-
-// Threshold on the occurrences of low buffer levels.
-const float kMaxBufferLow = 0.30f;
-
-// Threshold on rate mismatch.
-const float kMaxRateMisMatch = 0.5f;
-
-// Threshold on amount of under/over encoder shooting.
-const float kRateOverShoot = 0.75f;
-const float kRateUnderShoot = 0.75f;
-
-// Factor to favor weighting the average rates with the current/last data.
-const float kWeightRate = 0.70f;
-
-// Factor for transitional rate for going back up in resolution.
-const float kTransRateScaleUpSpatial = 1.25f;
-const float kTransRateScaleUpTemp = 1.25f;
-const float kTransRateScaleUpSpatialTemp = 1.25f;
-
-// Threshold on packet loss rate, above which favor resolution reduction.
-const float kPacketLossThr = 0.1f;
-
-// Factor for reducing transitional bitrate under packet loss.
-const float kPacketLossRateFac = 1.0f;
-
-// Maximum possible transitional rate for down-sampling:
-// (units in kbps), for 30fps.
-const uint16_t kMaxRateQm[9] = {
-    0,     // QCIF
-    50,    // kHCIF
-    125,   // kQVGA
-    200,   // CIF
-    280,   // HVGA
-    400,   // VGA
-    700,   // QFULLHD
-    1000,  // WHD
-    1500   // FULLHD
-};
-
-// Frame rate scale for maximum transition rate.
-const float kFrameRateFac[4] = {
-    0.5f,   // Low
-    0.7f,   // Middle level 1
-    0.85f,  // Middle level 2
-    1.0f,   // High
-};
-
-// Scale for transitional rate: based on content class
-// motion=L/H/D,spatial==L/H/D: for low, high, middle levels
-const float kScaleTransRateQm[18] = {
-    // VGA and lower
-    0.40f,  // L, L
-    0.50f,  // L, H
-    0.40f,  // L, D
-    0.60f,  // H ,L
-    0.60f,  // H, H
-    0.60f,  // H, D
-    0.50f,  // D, L
-    0.50f,  // D, D
-    0.50f,  // D, H
-
-    // over VGA
-    0.40f,  // L, L
-    0.50f,  // L, H
-    0.40f,  // L, D
-    0.60f,  // H ,L
-    0.60f,  // H, H
-    0.60f,  // H, D
-    0.50f,  // D, L
-    0.50f,  // D, D
-    0.50f,  // D, H
-};
-
-// Threshold on the target rate relative to transitional rate.
-const float kFacLowRate = 0.5f;
-
-// Action for down-sampling:
-// motion=L/H/D,spatial==L/H/D, for low, high, middle levels;
-// rate = 0/1/2, for target rate state relative to transition rate.
-const uint8_t kSpatialAction[27] = {
-    // rateClass = 0:
-    1,  // L, L
-    1,  // L, H
-    1,  // L, D
-    4,  // H ,L
-    1,  // H, H
-    4,  // H, D
-    4,  // D, L
-    1,  // D, H
-    2,  // D, D
-
-    // rateClass = 1:
-    1,  // L, L
-    1,  // L, H
-    1,  // L, D
-    2,  // H ,L
-    1,  // H, H
-    2,  // H, D
-    2,  // D, L
-    1,  // D, H
-    2,  // D, D
-
-    // rateClass = 2:
-    1,  // L, L
-    1,  // L, H
-    1,  // L, D
-    2,  // H ,L
-    1,  // H, H
-    2,  // H, D
-    2,  // D, L
-    1,  // D, H
-    2,  // D, D
-};
-
-const uint8_t kTemporalAction[27] = {
-    // rateClass = 0:
-    3,  // L, L
-    2,  // L, H
-    2,  // L, D
-    1,  // H ,L
-    3,  // H, H
-    1,  // H, D
-    1,  // D, L
-    2,  // D, H
-    1,  // D, D
-
-    // rateClass = 1:
-    3,  // L, L
-    3,  // L, H
-    3,  // L, D
-    1,  // H ,L
-    3,  // H, H
-    1,  // H, D
-    1,  // D, L
-    3,  // D, H
-    1,  // D, D
-
-    // rateClass = 2:
-    1,  // L, L
-    3,  // L, H
-    3,  // L, D
-    1,  // H ,L
-    3,  // H, H
-    1,  // H, D
-    1,  // D, L
-    3,  // D, H
-    1,  // D, D
-};
-
-// Control the total amount of down-sampling allowed.
-const float kMaxSpatialDown = 8.0f;
-const float kMaxTempDown = 3.0f;
-const float kMaxTotalDown = 9.0f;
-
-// Minimum image size for a spatial down-sampling.
-const int kMinImageSize = 176 * 144;
-
-// Minimum frame rate for temporal down-sampling:
-// no frame rate reduction if incomingFrameRate <= MIN_FRAME_RATE.
-const int kMinFrameRate = 8;
-
-//
-// PARAMETERS FOR FEC ADJUSTMENT: TODO (marpan)
-//
-
-//
-// PARAMETETS FOR SETTING LOW/HIGH STATES OF CONTENT METRICS:
-//
-
-// Thresholds for frame rate:
-const int kLowFrameRate = 10;
-const int kMiddleFrameRate = 15;
-const int kHighFrameRate = 25;
-
-// Thresholds for motion: motion level is from NFD.
-const float kHighMotionNfd = 0.075f;
-const float kLowMotionNfd = 0.03f;
-
-// Thresholds for spatial prediction error:
-// this is applied on the average of (2x2,1x2,2x1).
-const float kHighTexture = 0.035f;
-const float kLowTexture = 0.020f;
-
-// Used to reduce thresholds for larger/HD scenes: correction factor since
-// higher correlation in HD scenes means lower spatial prediction error.
-const float kScaleTexture = 0.9f;
-
-// Percentage reduction in transitional bitrate for 2x2 selected over 1x2/2x1.
-const float kRateRedSpatial2X2 = 0.6f;
-
-const float kSpatialErr2x2VsHoriz = 0.1f;   // percentage to favor 2x2 over H
-const float kSpatialErr2X2VsVert = 0.1f;    // percentage to favor 2x2 over V
-const float kSpatialErrVertVsHoriz = 0.1f;  // percentage to favor H over V
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_DATA_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/qm_select_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/qm_select_unittest.cc
deleted file mode 100644
index f8542ec6763..00000000000
--- a/chromium/third_party/webrtc/modules/video_coding/qm_select_unittest.cc
+++ /dev/null
@@ -1,1307 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes unit tests the QmResolution class
- * In particular, for the selection of spatial and/or temporal down-sampling.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/video_coding/qm_select.h"
-
-namespace webrtc {
-
-// Representative values of content metrics for: low/high/medium(default) state,
-// based on parameters settings in qm_select_data.h.
-const float kSpatialLow = 0.01f;
-const float kSpatialMedium = 0.03f;
-const float kSpatialHigh = 0.1f;
-const float kTemporalLow = 0.01f;
-const float kTemporalMedium = 0.06f;
-const float kTemporalHigh = 0.1f;
-
-class QmSelectTest : public ::testing::Test {
- protected:
-  QmSelectTest()
-      : qm_resolution_(new VCMQmResolution()),
-        content_metrics_(new VideoContentMetrics()),
-        qm_scale_(NULL) {}
-  VCMQmResolution* qm_resolution_;
-  VideoContentMetrics* content_metrics_;
-  VCMResolutionScale* qm_scale_;
-
-  void InitQmNativeData(float initial_bit_rate,
-                        int user_frame_rate,
-                        int native_width,
-                        int native_height,
-                        int num_layers);
-
-  void UpdateQmEncodedFrame(size_t* encoded_size, size_t num_updates);
-
-  void UpdateQmRateData(int* target_rate,
-                        int* encoder_sent_rate,
-                        int* incoming_frame_rate,
-                        uint8_t* fraction_lost,
-                        int num_updates);
-
-  void UpdateQmContentData(float motion_metric,
-                           float spatial_metric,
-                           float spatial_metric_horiz,
-                           float spatial_metric_vert);
-
-  bool IsSelectedActionCorrect(VCMResolutionScale* qm_scale,
-                               float fac_width,
-                               float fac_height,
-                               float fac_temp,
-                               uint16_t new_width,
-                               uint16_t new_height,
-                               float new_frame_rate);
-
-  void TearDown() {
-    delete qm_resolution_;
-    delete content_metrics_;
-  }
-};
-
-TEST_F(QmSelectTest, HandleInputs) {
-  // Expect parameter error. Initialize with invalid inputs.
-  EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 0, 640, 480, 1));
-  EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 30, 640, 0, 1));
-  EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 30, 0, 480, 1));
-
-  // Expect uninitialized error.: No valid initialization before selection.
-  EXPECT_EQ(-7, qm_resolution_->SelectResolution(&qm_scale_));
-
-  VideoContentMetrics* content_metrics = NULL;
-  EXPECT_EQ(0, qm_resolution_->Initialize(1000, 30, 640, 480, 1));
-  qm_resolution_->UpdateContent(content_metrics);
-  // Content metrics are NULL: Expect success and no down-sampling action.
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0, 1.0, 1.0, 640, 480, 30.0f));
-}
-
-// TODO(marpan): Add a test for number of temporal layers > 1.
-
-// No down-sampling action at high rates.
-TEST_F(QmSelectTest, NoActionHighRate) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(800, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {800, 800, 800};
-  int encoder_sent_rate[] = {800, 800, 800};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  UpdateQmContentData(kTemporalLow, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
-}
-
-// Rate is well below transition, down-sampling action is taken,
-// depending on the content state.
-TEST_F(QmSelectTest, DownActionLowRate) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(50, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {50, 50, 50};
-  int encoder_sent_rate[] = {50, 50, 50};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial: 2x2 spatial expected.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Low motion, low spatial: 2/3 temporal is expected.
-  UpdateQmContentData(kTemporalLow, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Medium motion, low spatial: 2x2 spatial expected.
-  UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // High motion, high spatial: 2/3 temporal expected.
-  UpdateQmContentData(kTemporalHigh, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(4, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Low motion, high spatial: 1/2 temporal expected.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Medium motion, high spatial: 1/2 temporal expected.
-  UpdateQmContentData(kTemporalMedium, kSpatialHigh, kSpatialHigh,
-                      kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // High motion, medium spatial: 2x2 spatial expected.
-  UpdateQmContentData(kTemporalHigh, kSpatialMedium, kSpatialMedium,
-                      kSpatialMedium);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
-  // Target frame rate for frame dropper should be the same as previous == 15.
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Low motion, medium spatial: high frame rate, so 1/2 temporal expected.
-  UpdateQmContentData(kTemporalLow, kSpatialMedium, kSpatialMedium,
-                      kSpatialMedium);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Medium motion, medium spatial: high frame rate, so 2/3 temporal expected.
-  UpdateQmContentData(kTemporalMedium, kSpatialMedium, kSpatialMedium,
-                      kSpatialMedium);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(8, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
-}
-
-// Rate mis-match is high, and we have over-shooting.
-// since target rate is below max for down-sampling, down-sampling is selected.
-TEST_F(QmSelectTest, DownActionHighRateMMOvershoot) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(300, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {300, 300, 300};
-  int encoder_sent_rate[] = {900, 900, 900};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
-                                      480, 360, 30.0f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Low motion, high spatial
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
-}
-
-// Rate mis-match is high, target rate is below max for down-sampling,
-// but since we have consistent under-shooting, no down-sampling action.
-TEST_F(QmSelectTest, NoActionHighRateMMUndershoot) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(300, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {300, 300, 300};
-  int encoder_sent_rate[] = {100, 100, 100};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Low motion, high spatial
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
-}
-
-// Buffer is underflowing, and target rate is below max for down-sampling,
-// so action is taken.
-TEST_F(QmSelectTest, DownActionBufferUnderflow) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(300, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update with encoded size over a number of frames.
-  // per-frame bandwidth = 15 = 450/30: simulate (decoder) buffer underflow:
-  size_t encoded_size[] = {200, 100, 50, 30, 60, 40, 20, 30, 20, 40};
-  UpdateQmEncodedFrame(encoded_size, GTEST_ARRAY_SIZE_(encoded_size));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {300, 300, 300};
-  int encoder_sent_rate[] = {450, 450, 450};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
-                                      480, 360, 30.0f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Low motion, high spatial
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 640, 480, 20.5f));
-}
-
-// Target rate is below max for down-sampling, but buffer level is stable,
-// so no action is taken.
-TEST_F(QmSelectTest, NoActionBufferStable) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(350, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update with encoded size over a number of frames.
-  // per-frame bandwidth = 15 = 450/30: simulate stable (decoder) buffer levels.
-  size_t encoded_size[] = {40, 10, 10, 16, 18, 20, 17, 20, 16, 15};
-  UpdateQmEncodedFrame(encoded_size, GTEST_ARRAY_SIZE_(encoded_size));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {350, 350, 350};
-  int encoder_sent_rate[] = {350, 450, 450};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
-
-  qm_resolution_->ResetDownSamplingState();
-  // Low motion, high spatial
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 30.0f));
-}
-
-// Very low rate, but no spatial down-sampling below some size (QCIF).
-TEST_F(QmSelectTest, LimitDownSpatialAction) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(10, 30, 176, 144, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 176;
-  uint16_t codec_height = 144;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(0, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {10, 10, 10};
-  int encoder_sent_rate[] = {10, 10, 10};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 176, 144, 30.0f));
-}
-
-// Very low rate, but no frame reduction below some frame_rate (8fps).
-TEST_F(QmSelectTest, LimitDownTemporalAction) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(10, 8, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(8.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {10, 10, 10};
-  int encoder_sent_rate[] = {10, 10, 10};
-  int incoming_frame_rate[] = {8, 8, 8};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, medium spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialMedium, kSpatialMedium,
-                      kSpatialMedium);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 8.0f));
-}
-
-// Two stages: spatial down-sample and then back up spatially,
-// as rate as increased.
-TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(50, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {50, 50, 50};
-  int encoder_sent_rate[] = {50, 50, 50};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  // Reset and go up in rate: expected to go back up, in 2 stages of 3/4.
-  qm_resolution_->ResetRates();
-  qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
-  EXPECT_EQ(2, qm_resolution_->GetImageType(320, 240));
-  // Update rates for a sequence of intervals.
-  int target_rate2[] = {400, 400, 400, 400, 400};
-  int encoder_sent_rate2[] = {400, 400, 400, 400, 400};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360, 30.0f));
-
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
-                                      640, 480, 30.0f));
-}
-
-// Two stages: spatial down-sample and then back up spatially, since encoder
-// is under-shooting target even though rate has not increased much.
-TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(50, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {50, 50, 50};
-  int encoder_sent_rate[] = {50, 50, 50};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  // Reset rates and simulate under-shooting scenario.: expect to go back up.
-  // Goes up spatially in two stages for 1/2x1/2 down-sampling.
-  qm_resolution_->ResetRates();
-  qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
-  EXPECT_EQ(2, qm_resolution_->GetImageType(320, 240));
-  // Update rates for a sequence of intervals.
-  int target_rate2[] = {200, 200, 200, 200, 200};
-  int encoder_sent_rate2[] = {50, 50, 50, 50, 50};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
-  float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, scale, scale, 1.0f, 480, 360, 30.0f));
-
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
-                                      640, 480, 30.0f));
-}
-
-// Two stages: spatial down-sample and then no action to go up,
-// as encoding rate mis-match is too high.
-TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(50, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {50, 50, 50};
-  int encoder_sent_rate[] = {50, 50, 50};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  // Reset and simulate large rate mis-match: expect no action to go back up.
-  qm_resolution_->ResetRates();
-  qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
-  EXPECT_EQ(2, qm_resolution_->GetImageType(320, 240));
-  // Update rates for a sequence of intervals.
-  int target_rate2[] = {400, 400, 400, 400, 400};
-  int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 320, 240, 30.0f));
-}
-
-// Two stages: temporally down-sample and then back up temporally,
-// as rate as increased.
-TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporal) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(50, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {50, 50, 50};
-  int encoder_sent_rate[] = {50, 50, 50};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, high spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
-
-  // Reset rates and go up in rate: expect to go back up.
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate2[] = {400, 400, 400, 400, 400};
-  int encoder_sent_rate2[] = {400, 400, 400, 400, 400};
-  int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480, 30.0f));
-}
-
-// Two stages: temporal down-sample and then back up temporally, since encoder
-// is under-shooting target even though rate has not increased much.
-TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(50, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {50, 50, 50};
-  int encoder_sent_rate[] = {50, 50, 50};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, high spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f, 640, 480, 15.5f));
-
-  // Reset rates and simulate under-shooting scenario.: expect to go back up.
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate2[] = {150, 150, 150, 150, 150};
-  int encoder_sent_rate2[] = {50, 50, 50, 50, 50};
-  int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 0.5f, 640, 480, 30.0f));
-}
-
-// Two stages: temporal down-sample and then no action to go up,
-// as encoding rate mis-match is too high.
-TEST_F(QmSelectTest, 2StageDownTemporalNoActionUp) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(50, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {50, 50, 50};
-  int encoder_sent_rate[] = {50, 50, 50};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, high spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2, 640, 480, 15.5f));
-
-  // Reset and simulate large rate mis-match: expect no action to go back up.
-  qm_resolution_->UpdateCodecParameters(15.0f, codec_width, codec_height);
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate2[] = {600, 600, 600, 600, 600};
-  int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
-  int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 640, 480, 15.0f));
-}
-// 3 stages: spatial down-sample, followed by temporal down-sample,
-// and then go up to full state, as encoding rate has increased.
-TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(80, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {80, 80, 80};
-  int encoder_sent_rate[] = {80, 80, 80};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  // Change content data: expect temporal down-sample.
-  qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
-  EXPECT_EQ(2, qm_resolution_->GetImageType(320, 240));
-
-  // Reset rates and go lower in rate.
-  qm_resolution_->ResetRates();
-  int target_rate2[] = {40, 40, 40, 40, 40};
-  int encoder_sent_rate2[] = {40, 40, 40, 40, 40};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, high spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
-
-  // Reset rates and go high up in rate: expect to go back up both spatial
-  // and temporally. The 1/2x1/2 spatial is undone in two stages.
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate3[] = {1000, 1000, 1000, 1000, 1000};
-  int encoder_sent_rate3[] = {1000, 1000, 1000, 1000, 1000};
-  int incoming_frame_rate3[] = {20, 20, 20, 20, 20};
-  uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
-                   fraction_lost3, 5);
-
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f, 480,
-                                      360, 30.0f));
-
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
-                                      640, 480, 30.0f));
-}
-
-// No down-sampling below some total amount.
-TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(150, 30, 1280, 720, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 1280;
-  uint16_t codec_height = 720;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(7, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {150, 150, 150};
-  int encoder_sent_rate[] = {150, 150, 150};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 640, 360, 30.0f));
-
-  // Reset and lower rates to get another spatial action (3/4x3/4).
-  // Lower the frame rate for spatial to be selected again.
-  qm_resolution_->ResetRates();
-  qm_resolution_->UpdateCodecParameters(10.0f, 640, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(640, 360));
-  // Update rates for a sequence of intervals.
-  int target_rate2[] = {70, 70, 70, 70, 70};
-  int encoder_sent_rate2[] = {70, 70, 70, 70, 70};
-  int incoming_frame_rate2[] = {10, 10, 10, 10, 10};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, medium spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialMedium, kSpatialMedium,
-                      kSpatialMedium);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
-                                      480, 270, 10.0f));
-
-  // Reset and go to very low rate: no action should be taken,
-  // we went down too much already.
-  qm_resolution_->ResetRates();
-  qm_resolution_->UpdateCodecParameters(10.0f, 480, 270);
-  EXPECT_EQ(3, qm_resolution_->GetImageType(480, 270));
-  // Update rates for a sequence of intervals.
-  int target_rate3[] = {10, 10, 10, 10, 10};
-  int encoder_sent_rate3[] = {10, 10, 10, 10, 10};
-  int incoming_frame_rate3[] = {10, 10, 10, 10, 10};
-  uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
-                   fraction_lost3, 5);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.0f, 480, 270, 10.0f));
-}
-
-// Multiple down-sampling stages and then undo all of them.
-// Spatial down-sample 3/4x3/4, followed by temporal down-sample 2/3,
-// followed by spatial 3/4x3/4. Then go up to full state,
-// as encoding rate has increased.
-TEST_F(QmSelectTest, MultipleStagesCheckActionHistory1) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(150, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Go down spatial 3/4x3/4.
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {150, 150, 150};
-  int encoder_sent_rate[] = {150, 150, 150};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Medium motion, low spatial.
-  UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
-                                      480, 360, 30.0f));
-  // Go down 2/3 temporal.
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  qm_resolution_->ResetRates();
-  int target_rate2[] = {100, 100, 100, 100, 100};
-  int encoder_sent_rate2[] = {100, 100, 100, 100, 100};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, high spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360, 20.5f));
-
-  // Go down 3/4x3/4 spatial:
-  qm_resolution_->UpdateCodecParameters(20.0f, 480, 360);
-  qm_resolution_->ResetRates();
-  int target_rate3[] = {80, 80, 80, 80, 80};
-  int encoder_sent_rate3[] = {80, 80, 80, 80, 80};
-  int incoming_frame_rate3[] = {20, 20, 20, 20, 20};
-  uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
-                   fraction_lost3, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // High motion, low spatial.
-  UpdateQmContentData(kTemporalHigh, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  // The two spatial actions of 3/4x3/4 are converted to 1/2x1/2,
-  // so scale factor is 2.0.
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 20.0f));
-
-  // Reset rates and go high up in rate: expect to go up:
-  // 1/2x1x2 spatial and 1/2 temporally.
-
-  // Go up 1/2x1/2 spatially and 1/2 temporally. Spatial is done in 2 stages.
-  qm_resolution_->UpdateCodecParameters(15.0f, 320, 240);
-  EXPECT_EQ(2, qm_resolution_->GetImageType(320, 240));
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate4[] = {1000, 1000, 1000, 1000, 1000};
-  int encoder_sent_rate4[] = {1000, 1000, 1000, 1000, 1000};
-  int incoming_frame_rate4[] = {15, 15, 15, 15, 15};
-  uint8_t fraction_lost4[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate4, encoder_sent_rate4, incoming_frame_rate4,
-                   fraction_lost4, 5);
-
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f, 480,
-                                      360, 30.0f));
-
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
-                                      640, 480, 30.0f));
-}
-
-// Multiple down-sampling and up-sample stages, with partial undoing.
-// Spatial down-sample 1/2x1/2, followed by temporal down-sample 2/3, undo the
-// temporal, then another temporal, and then undo both spatial and temporal.
-TEST_F(QmSelectTest, MultipleStagesCheckActionHistory2) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(80, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Go down 1/2x1/2 spatial.
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {80, 80, 80};
-  int encoder_sent_rate[] = {80, 80, 80};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Medium motion, low spatial.
-  UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-
-  // Go down 2/3 temporal.
-  qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
-  EXPECT_EQ(2, qm_resolution_->GetImageType(320, 240));
-  qm_resolution_->ResetRates();
-  int target_rate2[] = {40, 40, 40, 40, 40};
-  int encoder_sent_rate2[] = {40, 40, 40, 40, 40};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Medium motion, high spatial.
-  UpdateQmContentData(kTemporalMedium, kSpatialHigh, kSpatialHigh,
-                      kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
-
-  // Go up 2/3 temporally.
-  qm_resolution_->UpdateCodecParameters(20.0f, 320, 240);
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate3[] = {150, 150, 150, 150, 150};
-  int encoder_sent_rate3[] = {150, 150, 150, 150, 150};
-  int incoming_frame_rate3[] = {20, 20, 20, 20, 20};
-  uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
-                   fraction_lost3, 5);
-
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f / 3.0f, 320,
-                                      240, 30.0f));
-
-  // Go down 2/3 temporal.
-  qm_resolution_->UpdateCodecParameters(30.0f, 320, 240);
-  EXPECT_EQ(2, qm_resolution_->GetImageType(320, 240));
-  qm_resolution_->ResetRates();
-  int target_rate4[] = {40, 40, 40, 40, 40};
-  int encoder_sent_rate4[] = {40, 40, 40, 40, 40};
-  int incoming_frame_rate4[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost4[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate4, encoder_sent_rate4, incoming_frame_rate4,
-                   fraction_lost4, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, high spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 320, 240, 20.5f));
-
-  // Go up spatial and temporal. Spatial undoing is done in 2 stages.
-  qm_resolution_->UpdateCodecParameters(20.5f, 320, 240);
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate5[] = {1000, 1000, 1000, 1000, 1000};
-  int encoder_sent_rate5[] = {1000, 1000, 1000, 1000, 1000};
-  int incoming_frame_rate5[] = {20, 20, 20, 20, 20};
-  uint8_t fraction_lost5[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate5, encoder_sent_rate5, incoming_frame_rate5,
-                   fraction_lost5, 5);
-
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  float scale = (4.0f / 3.0f) / 2.0f;
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, scale, scale, 2.0f / 3.0f, 480,
-                                      360, 30.0f));
-
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
-                                      640, 480, 30.0f));
-}
-
-// Multiple down-sampling and up-sample stages, with partial undoing.
-// Spatial down-sample 3/4x3/4, followed by temporal down-sample 2/3,
-// undo the temporal 2/3, and then undo the spatial.
-TEST_F(QmSelectTest, MultipleStagesCheckActionHistory3) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(100, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Go down 3/4x3/4 spatial.
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {100, 100, 100};
-  int encoder_sent_rate[] = {100, 100, 100};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Medium motion, low spatial.
-  UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
-                                      480, 360, 30.0f));
-
-  // Go down 2/3 temporal.
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  qm_resolution_->ResetRates();
-  int target_rate2[] = {100, 100, 100, 100, 100};
-  int encoder_sent_rate2[] = {100, 100, 100, 100, 100};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Low motion, high spatial.
-  UpdateQmContentData(kTemporalLow, kSpatialHigh, kSpatialHigh, kSpatialHigh);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 1.5f, 480, 360, 20.5f));
-
-  // Go up 2/3 temporal.
-  qm_resolution_->UpdateCodecParameters(20.5f, 480, 360);
-  qm_resolution_->ResetRates();
-  // Update rates for a sequence of intervals.
-  int target_rate3[] = {250, 250, 250, 250, 250};
-  int encoder_sent_rate3[] = {250, 250, 250, 250, 250};
-  int incoming_frame_rate3[] = {20, 20, 20, 20, 120};
-  uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
-                   fraction_lost3, 5);
-
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1.0f, 1.0f, 2.0f / 3.0f, 480,
-                                      360, 30.0f));
-
-  // Go up spatial.
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  qm_resolution_->ResetRates();
-  int target_rate4[] = {500, 500, 500, 500, 500};
-  int encoder_sent_rate4[] = {500, 500, 500, 500, 500};
-  int incoming_frame_rate4[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost4[] = {30, 30, 30, 30, 30};
-  UpdateQmRateData(target_rate4, encoder_sent_rate4, incoming_frame_rate4,
-                   fraction_lost4, 5);
-
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 3.0f / 4.0f, 3.0f / 4.0f, 1.0f,
-                                      640, 480, 30.0f));
-}
-
-// Two stages of 3/4x3/4 converted to one stage of 1/2x1/2.
-TEST_F(QmSelectTest, ConvertThreeQuartersToOneHalf) {
-  // Initialize with bitrate, frame rate, native system width/height, and
-  // number of temporal layers.
-  InitQmNativeData(150, 30, 640, 480, 1);
-
-  // Update with encoder frame size.
-  uint16_t codec_width = 640;
-  uint16_t codec_height = 480;
-  qm_resolution_->UpdateCodecParameters(30.0f, codec_width, codec_height);
-  EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
-  // Go down 3/4x3/4 spatial.
-  // Update rates for a sequence of intervals.
-  int target_rate[] = {150, 150, 150};
-  int encoder_sent_rate[] = {150, 150, 150};
-  int incoming_frame_rate[] = {30, 30, 30};
-  uint8_t fraction_lost[] = {10, 10, 10};
-  UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
-                   fraction_lost, 3);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Medium motion, low spatial.
-  UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 4.0f / 3.0f, 4.0f / 3.0f, 1.0f,
-                                      480, 360, 30.0f));
-
-  // Set rates to go down another 3/4 spatial. Should be converted ton 1/2.
-  qm_resolution_->UpdateCodecParameters(30.0f, 480, 360);
-  EXPECT_EQ(4, qm_resolution_->GetImageType(480, 360));
-  qm_resolution_->ResetRates();
-  int target_rate2[] = {100, 100, 100, 100, 100};
-  int encoder_sent_rate2[] = {100, 100, 100, 100, 100};
-  int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
-  uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
-  UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
-                   fraction_lost2, 5);
-
-  // Update content: motion level, and 3 spatial prediction errors.
-  // Medium motion, low spatial.
-  UpdateQmContentData(kTemporalMedium, kSpatialLow, kSpatialLow, kSpatialLow);
-  EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
-  EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
-  EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
-  EXPECT_TRUE(
-      IsSelectedActionCorrect(qm_scale_, 2.0f, 2.0f, 1.0f, 320, 240, 30.0f));
-}
-
-void QmSelectTest::InitQmNativeData(float initial_bit_rate,
-                                    int user_frame_rate,
-                                    int native_width,
-                                    int native_height,
-                                    int num_layers) {
-  EXPECT_EQ(
-      0, qm_resolution_->Initialize(initial_bit_rate, user_frame_rate,
-                                    native_width, native_height, num_layers));
-}
-
-void QmSelectTest::UpdateQmContentData(float motion_metric,
-                                       float spatial_metric,
-                                       float spatial_metric_horiz,
-                                       float spatial_metric_vert) {
-  content_metrics_->motion_magnitude = motion_metric;
-  content_metrics_->spatial_pred_err = spatial_metric;
-  content_metrics_->spatial_pred_err_h = spatial_metric_horiz;
-  content_metrics_->spatial_pred_err_v = spatial_metric_vert;
-  qm_resolution_->UpdateContent(content_metrics_);
-}
-
-void QmSelectTest::UpdateQmEncodedFrame(size_t* encoded_size,
-                                        size_t num_updates) {
-  for (size_t i = 0; i < num_updates; ++i) {
-    // Convert to bytes.
-    size_t encoded_size_update = 1000 * encoded_size[i] / 8;
-    qm_resolution_->UpdateEncodedSize(encoded_size_update);
-  }
-}
-
-void QmSelectTest::UpdateQmRateData(int* target_rate,
-                                    int* encoder_sent_rate,
-                                    int* incoming_frame_rate,
-                                    uint8_t* fraction_lost,
-                                    int num_updates) {
-  for (int i = 0; i < num_updates; ++i) {
-    float target_rate_update = target_rate[i];
-    float encoder_sent_rate_update = encoder_sent_rate[i];
-    float incoming_frame_rate_update = incoming_frame_rate[i];
-    uint8_t fraction_lost_update = fraction_lost[i];
-    qm_resolution_->UpdateRates(target_rate_update, encoder_sent_rate_update,
-                                incoming_frame_rate_update,
-                                fraction_lost_update);
-  }
-}
-
-// Check is the selected action from the QmResolution class is the same
-// as the expected scales from |fac_width|, |fac_height|, |fac_temp|.
-bool QmSelectTest::IsSelectedActionCorrect(VCMResolutionScale* qm_scale,
-                                           float fac_width,
-                                           float fac_height,
-                                           float fac_temp,
-                                           uint16_t new_width,
-                                           uint16_t new_height,
-                                           float new_frame_rate) {
-  if (qm_scale->spatial_width_fact == fac_width &&
-      qm_scale->spatial_height_fact == fac_height &&
-      qm_scale->temporal_fact == fac_temp &&
-      qm_scale->codec_width == new_width &&
-      qm_scale->codec_height == new_height &&
-      qm_scale->frame_rate == new_frame_rate) {
-    return true;
-  } else {
-    return false;
-  }
-}
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/receiver.cc b/chromium/third_party/webrtc/modules/video_coding/receiver.cc
index a02fd01de6a..1954df94e73 100644
--- a/chromium/third_party/webrtc/modules/video_coding/receiver.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/receiver.cc
@@ -42,12 +42,17 @@ VCMReceiver::VCMReceiver(VCMTiming* timing,
                          EventFactory* event_factory,
                          NackSender* nack_sender,
                          KeyFrameRequestSender* keyframe_request_sender)
-    : VCMReceiver(timing,
-                  clock,
-                  std::unique_ptr<EventWrapper>(event_factory->CreateEvent()),
-                  std::unique_ptr<EventWrapper>(event_factory->CreateEvent()),
-                  nack_sender,
-                  keyframe_request_sender) {}
+    : VCMReceiver(
+          timing,
+          clock,
+          std::unique_ptr<EventWrapper>(event_factory
+                                            ? event_factory->CreateEvent()
+                                            : EventWrapper::Create()),
+          std::unique_ptr<EventWrapper>(event_factory
+                                            ? event_factory->CreateEvent()
+                                            : EventWrapper::Create()),
+          nack_sender,
+          keyframe_request_sender) {}
 
 VCMReceiver::VCMReceiver(VCMTiming* timing,
                          Clock* clock,
@@ -281,24 +286,6 @@ int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
   return 0;
 }
 
-int VCMReceiver::RenderBufferSizeMs() {
-  uint32_t timestamp_start = 0u;
-  uint32_t timestamp_end = 0u;
-  // Render timestamps are computed just prior to decoding. Therefore this is
-  // only an estimate based on frames' timestamps and current timing state.
-  jitter_buffer_.RenderBufferSize(&timestamp_start, &timestamp_end);
-  if (timestamp_start == timestamp_end) {
-    return 0;
-  }
-  // Update timing.
-  const int64_t now_ms = clock_->TimeInMilliseconds();
-  timing_->SetJitterDelay(jitter_buffer_.EstimatedJitterMs());
-  // Get render timestamps.
-  uint32_t render_start = timing_->RenderTimeMs(timestamp_start, now_ms);
-  uint32_t render_end = timing_->RenderTimeMs(timestamp_end, now_ms);
-  return render_end - render_start;
-}
-
 void VCMReceiver::RegisterStatsCallback(
     VCMReceiveStatisticsCallback* callback) {
   jitter_buffer_.RegisterStatsCallback(callback);
diff --git a/chromium/third_party/webrtc/modules/video_coding/receiver.h b/chromium/third_party/webrtc/modules/video_coding/receiver.h
index a4c55e967cb..dbef62a716c 100644
--- a/chromium/third_party/webrtc/modules/video_coding/receiver.h
+++ b/chromium/third_party/webrtc/modules/video_coding/receiver.h
@@ -90,11 +90,6 @@ class VCMReceiver {
   void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode);
   VCMDecodeErrorMode DecodeErrorMode() const;
 
-  // Returns size in time (milliseconds) of complete continuous frames in the
-  // jitter buffer. The render time is estimated based on the render delay at
-  // the time this function is called.
-  int RenderBufferSizeMs();
-
   void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
 
   void TriggerDecoderShutdown();
diff --git a/chromium/third_party/webrtc/modules/video_coding/receiver_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/receiver_unittest.cc
index 42cc9ac0a88..d05957f6f0d 100644
--- a/chromium/third_party/webrtc/modules/video_coding/receiver_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/receiver_unittest.cc
@@ -92,66 +92,6 @@ class TestVCMReceiver : public ::testing::Test {
   std::unique_ptr<StreamGenerator> stream_generator_;
 };
 
-TEST_F(TestVCMReceiver, RenderBufferSize_AllComplete) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  int num_of_frames = 10;
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ(num_of_frames * kDefaultFramePeriodMs,
-            receiver_.RenderBufferSizeMs());
-}
-
-TEST_F(TestVCMReceiver, RenderBufferSize_SkipToKeyFrame) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  const int kNumOfNonDecodableFrames = 2;
-  for (int i = 0; i < kNumOfNonDecodableFrames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  const int kNumOfFrames = 10;
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  for (int i = 0; i < kNumOfFrames - 1; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ((kNumOfFrames - 1) * kDefaultFramePeriodMs,
-            receiver_.RenderBufferSizeMs());
-}
-
-TEST_F(TestVCMReceiver, RenderBufferSize_NotAllComplete) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
-  int num_of_frames = 10;
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  num_of_frames++;
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ((num_of_frames - 1) * kDefaultFramePeriodMs,
-            receiver_.RenderBufferSizeMs());
-}
-
-TEST_F(TestVCMReceiver, RenderBufferSize_NoKeyFrame) {
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-  int num_of_frames = 10;
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  int64_t next_render_time_ms = 0;
-  VCMEncodedFrame* frame =
-      receiver_.FrameForDecoding(10, &next_render_time_ms, false);
-  EXPECT_TRUE(frame == NULL);
-  receiver_.ReleaseFrame(frame);
-  EXPECT_GE(InsertFrame(kVideoFrameDelta, false), kNoError);
-  for (int i = 0; i < num_of_frames; ++i) {
-    EXPECT_GE(InsertFrame(kVideoFrameDelta, true), kNoError);
-  }
-  EXPECT_EQ(0, receiver_.RenderBufferSizeMs());
-}
-
 TEST_F(TestVCMReceiver, NonDecodableDuration_Empty) {
   // Enable NACK and with no RTT thresholds for disabling retransmission delay.
   receiver_.SetNackMode(kNack, -1, -1);
diff --git a/chromium/third_party/webrtc/modules/video_coding/rtp_frame_reference_finder.cc b/chromium/third_party/webrtc/modules/video_coding/rtp_frame_reference_finder.cc
new file mode 100644
index 00000000000..2ddfada74e5
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/rtp_frame_reference_finder.cc
@@ -0,0 +1,486 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/frame_object.h"
+#include "webrtc/modules/video_coding/packet_buffer.h"
+
+namespace webrtc {
+namespace video_coding {
+
+RtpFrameReferenceFinder::RtpFrameReferenceFinder(
+    OnCompleteFrameCallback* frame_callback)
+    : last_picture_id_(-1),
+      last_unwrap_(-1),
+      current_ss_idx_(0),
+      frame_callback_(frame_callback) {}
+
+void RtpFrameReferenceFinder::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame) {
+  rtc::CritScope lock(&crit_);
+  switch (frame->codec_type()) {
+    case kVideoCodecULPFEC:
+    case kVideoCodecRED:
+    case kVideoCodecUnknown:
+      RTC_NOTREACHED();
+      break;
+    case kVideoCodecVP8:
+      ManageFrameVp8(std::move(frame));
+      break;
+    case kVideoCodecVP9:
+      ManageFrameVp9(std::move(frame));
+      break;
+    case kVideoCodecH264:
+    case kVideoCodecI420:
+    case kVideoCodecGeneric:
+      ManageFrameGeneric(std::move(frame));
+      break;
+  }
+}
+
+void RtpFrameReferenceFinder::RetryStashedFrames() {
+  size_t num_stashed_frames = stashed_frames_.size();
+
+  // Clean up stashed frames if there are too many.
+  while (stashed_frames_.size() > kMaxStashedFrames)
+    stashed_frames_.pop();
+
+  // Since frames are stashed if there is not enough data to determine their
+  // frame references we should at most check |stashed_frames_.size()| in
+  // order to not pop and push frames in and endless loop.
+  for (size_t i = 0; i < num_stashed_frames && !stashed_frames_.empty(); ++i) {
+    std::unique_ptr<RtpFrameObject> frame = std::move(stashed_frames_.front());
+    stashed_frames_.pop();
+    ManageFrame(std::move(frame));
+  }
+}
+
+void RtpFrameReferenceFinder::ManageFrameGeneric(
+    std::unique_ptr<RtpFrameObject> frame) {
+  if (frame->frame_type() == kVideoFrameKey)
+    last_seq_num_gop_[frame->last_seq_num()] = frame->last_seq_num();
+
+  // We have received a frame but not yet a keyframe, stash this frame.
+  if (last_seq_num_gop_.empty()) {
+    stashed_frames_.emplace(std::move(frame));
+    return;
+  }
+
+  // Clean up info for old keyframes but make sure to keep info
+  // for the last keyframe.
+  auto clean_to = last_seq_num_gop_.lower_bound(frame->last_seq_num() - 100);
+  if (clean_to != last_seq_num_gop_.end())
+    last_seq_num_gop_.erase(last_seq_num_gop_.begin(), clean_to);
+
+  // Find the last sequence number of the last frame for the keyframe
+  // that this frame indirectly references.
+  auto seq_num_it = last_seq_num_gop_.upper_bound(frame->last_seq_num());
+  seq_num_it--;
+
+  // Make sure the packet sequence numbers are continuous, otherwise stash
+  // this frame.
+  if (frame->frame_type() == kVideoFrameDelta) {
+    if (seq_num_it->second !=
+        static_cast<uint16_t>(frame->first_seq_num() - 1)) {
+      stashed_frames_.emplace(std::move(frame));
+      return;
+    }
+  }
+
+  RTC_DCHECK(AheadOrAt(frame->last_seq_num(), seq_num_it->first));
+
+  // Since keyframes can cause reordering we can't simply assign the
+  // picture id according to some incrementing counter.
+  frame->picture_id = frame->last_seq_num();
+  frame->num_references = frame->frame_type() == kVideoFrameDelta;
+  frame->references[0] = seq_num_it->second;
+  seq_num_it->second = frame->picture_id;
+
+  last_picture_id_ = frame->picture_id;
+  frame_callback_->OnCompleteFrame(std::move(frame));
+  RetryStashedFrames();
+}
+
+void RtpFrameReferenceFinder::ManageFrameVp8(
+    std::unique_ptr<RtpFrameObject> frame) {
+  RTPVideoTypeHeader* rtp_codec_header = frame->GetCodecHeader();
+  if (!rtp_codec_header)
+    return;
+
+  const RTPVideoHeaderVP8& codec_header = rtp_codec_header->VP8;
+
+  if (codec_header.pictureId == kNoPictureId ||
+      codec_header.temporalIdx == kNoTemporalIdx ||
+      codec_header.tl0PicIdx == kNoTl0PicIdx) {
+    ManageFrameGeneric(std::move(frame));
+    return;
+  }
+
+  frame->picture_id = codec_header.pictureId % kPicIdLength;
+
+  if (last_unwrap_ == -1)
+    last_unwrap_ = codec_header.pictureId;
+
+  if (last_picture_id_ == -1)
+    last_picture_id_ = frame->picture_id;
+
+  // Find if there has been a gap in fully received frames and save the picture
+  // id of those frames in |not_yet_received_frames_|.
+  if (AheadOf<uint16_t, kPicIdLength>(frame->picture_id, last_picture_id_)) {
+    last_picture_id_ = Add<kPicIdLength>(last_picture_id_, 1);
+    while (last_picture_id_ != frame->picture_id) {
+      not_yet_received_frames_.insert(last_picture_id_);
+      last_picture_id_ = Add<kPicIdLength>(last_picture_id_, 1);
+    }
+  }
+
+  // Clean up info for base layers that are too old.
+  uint8_t old_tl0_pic_idx = codec_header.tl0PicIdx - kMaxLayerInfo;
+  auto clean_layer_info_to = layer_info_.lower_bound(old_tl0_pic_idx);
+  layer_info_.erase(layer_info_.begin(), clean_layer_info_to);
+
+  // Clean up info about not yet received frames that are too old.
+  uint16_t old_picture_id =
+      Subtract<kPicIdLength>(frame->picture_id, kMaxNotYetReceivedFrames);
+  auto clean_frames_to = not_yet_received_frames_.lower_bound(old_picture_id);
+  not_yet_received_frames_.erase(not_yet_received_frames_.begin(),
+                                 clean_frames_to);
+
+  if (frame->frame_type() == kVideoFrameKey) {
+    frame->num_references = 0;
+    layer_info_[codec_header.tl0PicIdx].fill(-1);
+    CompletedFrameVp8(std::move(frame));
+    return;
+  }
+
+  auto layer_info_it = layer_info_.find(codec_header.temporalIdx == 0
+                                            ? codec_header.tl0PicIdx - 1
+                                            : codec_header.tl0PicIdx);
+
+  // If we don't have the base layer frame yet, stash this frame.
+  if (layer_info_it == layer_info_.end()) {
+    stashed_frames_.emplace(std::move(frame));
+    return;
+  }
+
+  // A non keyframe base layer frame has been received, copy the layer info
+  // from the previous base layer frame and set a reference to the previous
+  // base layer frame.
+  if (codec_header.temporalIdx == 0) {
+    layer_info_it =
+        layer_info_
+            .insert(make_pair(codec_header.tl0PicIdx, layer_info_it->second))
+            .first;
+    frame->num_references = 1;
+    frame->references[0] = layer_info_it->second[0];
+    CompletedFrameVp8(std::move(frame));
+    return;
+  }
+
+  // Layer sync frame, this frame only references its base layer frame.
+  if (codec_header.layerSync) {
+    frame->num_references = 1;
+    frame->references[0] = layer_info_it->second[0];
+
+    CompletedFrameVp8(std::move(frame));
+    return;
+  }
+
+  // Find all references for this frame.
+  frame->num_references = 0;
+  for (uint8_t layer = 0; layer <= codec_header.temporalIdx; ++layer) {
+    RTC_DCHECK_NE(-1, layer_info_it->second[layer]);
+
+    // If we have not yet received a frame between this frame and the referenced
+    // frame then we have to wait for that frame to be completed first.
+    auto not_received_frame_it =
+        not_yet_received_frames_.upper_bound(layer_info_it->second[layer]);
+    if (not_received_frame_it != not_yet_received_frames_.end() &&
+        AheadOf<uint16_t, kPicIdLength>(frame->picture_id,
+                                        *not_received_frame_it)) {
+      stashed_frames_.emplace(std::move(frame));
+      return;
+    }
+
+    ++frame->num_references;
+    frame->references[layer] = layer_info_it->second[layer];
+  }
+
+  CompletedFrameVp8(std::move(frame));
+}
+
+void RtpFrameReferenceFinder::CompletedFrameVp8(
+    std::unique_ptr<RtpFrameObject> frame) {
+  RTPVideoTypeHeader* rtp_codec_header = frame->GetCodecHeader();
+  if (!rtp_codec_header)
+    return;
+
+  const RTPVideoHeaderVP8& codec_header = rtp_codec_header->VP8;
+
+  uint8_t tl0_pic_idx = codec_header.tl0PicIdx;
+  uint8_t temporal_index = codec_header.temporalIdx;
+  auto layer_info_it = layer_info_.find(tl0_pic_idx);
+
+  // Update this layer info and newer.
+  while (layer_info_it != layer_info_.end()) {
+    if (layer_info_it->second[temporal_index] != -1 &&
+        AheadOf<uint16_t, kPicIdLength>(layer_info_it->second[temporal_index],
+                                        frame->picture_id)) {
+      // The frame was not newer, then no subsequent layer info have to be
+      // update.
+      break;
+    }
+
+    layer_info_it->second[codec_header.temporalIdx] = frame->picture_id;
+    ++tl0_pic_idx;
+    layer_info_it = layer_info_.find(tl0_pic_idx);
+  }
+  not_yet_received_frames_.erase(frame->picture_id);
+
+  for (size_t i = 0; i < frame->num_references; ++i)
+    frame->references[i] = UnwrapPictureId(frame->references[i]);
+  frame->picture_id = UnwrapPictureId(frame->picture_id);
+
+  frame_callback_->OnCompleteFrame(std::move(frame));
+  RetryStashedFrames();
+}
+
+void RtpFrameReferenceFinder::ManageFrameVp9(
+    std::unique_ptr<RtpFrameObject> frame) {
+  RTPVideoTypeHeader* rtp_codec_header = frame->GetCodecHeader();
+  if (!rtp_codec_header)
+    return;
+
+  const RTPVideoHeaderVP9& codec_header = rtp_codec_header->VP9;
+
+  if (codec_header.picture_id == kNoPictureId) {
+    ManageFrameGeneric(std::move(frame));
+    return;
+  }
+
+  frame->spatial_layer = codec_header.spatial_idx;
+  frame->inter_layer_predicted = codec_header.inter_layer_predicted;
+  frame->picture_id = codec_header.picture_id % kPicIdLength;
+
+  if (last_unwrap_ == -1)
+    last_unwrap_ = codec_header.picture_id;
+
+  if (last_picture_id_ == -1)
+    last_picture_id_ = frame->picture_id;
+
+  if (codec_header.flexible_mode) {
+    frame->num_references = codec_header.num_ref_pics;
+    for (size_t i = 0; i < frame->num_references; ++i) {
+      frame->references[i] =
+          Subtract<1 << 16>(frame->picture_id, codec_header.pid_diff[i]);
+    }
+
+    CompletedFrameVp9(std::move(frame));
+    return;
+  }
+
+  if (codec_header.ss_data_available) {
+    // Scalability structures can only be sent with tl0 frames.
+    if (codec_header.temporal_idx != 0) {
+      LOG(LS_WARNING) << "Received scalability structure on a non base layer"
+                         " frame. Scalability structure ignored.";
+    } else {
+      current_ss_idx_ = Add<kMaxGofSaved>(current_ss_idx_, 1);
+      scalability_structures_[current_ss_idx_] = codec_header.gof;
+      scalability_structures_[current_ss_idx_].pid_start = frame->picture_id;
+
+      auto pid_and_gof = std::make_pair(
+          frame->picture_id, &scalability_structures_[current_ss_idx_]);
+      gof_info_.insert(std::make_pair(codec_header.tl0_pic_idx, pid_and_gof));
+    }
+  }
+
+  // Clean up info for base layers that are too old.
+  uint8_t old_tl0_pic_idx = codec_header.tl0_pic_idx - kMaxGofSaved;
+  auto clean_gof_info_to = gof_info_.lower_bound(old_tl0_pic_idx);
+  gof_info_.erase(gof_info_.begin(), clean_gof_info_to);
+
+  if (frame->frame_type() == kVideoFrameKey) {
+    // When using GOF all keyframes must include the scalability structure.
+    if (!codec_header.ss_data_available)
+      LOG(LS_WARNING) << "Received keyframe without scalability structure";
+
+    frame->num_references = 0;
+    GofInfoVP9* gof = gof_info_.find(codec_header.tl0_pic_idx)->second.second;
+    FrameReceivedVp9(frame->picture_id, *gof);
+    CompletedFrameVp9(std::move(frame));
+    return;
+  }
+
+  auto gof_info_it = gof_info_.find(
+      (codec_header.temporal_idx == 0 && !codec_header.ss_data_available)
+          ? codec_header.tl0_pic_idx - 1
+          : codec_header.tl0_pic_idx);
+
+  // Gof info for this frame is not available yet, stash this frame.
+  if (gof_info_it == gof_info_.end()) {
+    stashed_frames_.emplace(std::move(frame));
+    return;
+  }
+
+  GofInfoVP9* gof = gof_info_it->second.second;
+  uint16_t picture_id_tl0 = gof_info_it->second.first;
+
+  FrameReceivedVp9(frame->picture_id, *gof);
+
+  // Make sure we don't miss any frame that could potentially have the
+  // up switch flag set.
+  if (MissingRequiredFrameVp9(frame->picture_id, *gof)) {
+    stashed_frames_.emplace(std::move(frame));
+    return;
+  }
+
+  if (codec_header.temporal_up_switch) {
+    auto pid_tidx =
+        std::make_pair(frame->picture_id, codec_header.temporal_idx);
+    up_switch_.insert(pid_tidx);
+  }
+
+  // If this is a base layer frame that contains a scalability structure
+  // then gof info has already been inserted earlier, so we only want to
+  // insert if we haven't done so already.
+  if (codec_header.temporal_idx == 0 && !codec_header.ss_data_available) {
+    auto pid_and_gof = std::make_pair(frame->picture_id, gof);
+    gof_info_.insert(std::make_pair(codec_header.tl0_pic_idx, pid_and_gof));
+  }
+
+  // Clean out old info about up switch frames.
+  uint16_t old_picture_id = Subtract<kPicIdLength>(last_picture_id_, 50);
+  auto up_switch_erase_to = up_switch_.lower_bound(old_picture_id);
+  up_switch_.erase(up_switch_.begin(), up_switch_erase_to);
+
+  RTC_DCHECK(
+      (AheadOrAt<uint16_t, kPicIdLength>(frame->picture_id, picture_id_tl0)));
+
+  size_t diff =
+      ForwardDiff<uint16_t, kPicIdLength>(gof->pid_start, frame->picture_id);
+  size_t gof_idx = diff % gof->num_frames_in_gof;
+
+  // Populate references according to the scalability structure.
+  frame->num_references = gof->num_ref_pics[gof_idx];
+  for (size_t i = 0; i < frame->num_references; ++i) {
+    frame->references[i] =
+        Subtract<kPicIdLength>(frame->picture_id, gof->pid_diff[gof_idx][i]);
+
+    // If this is a reference to a frame earlier than the last up switch point,
+    // then ignore this reference.
+    if (UpSwitchInIntervalVp9(frame->picture_id, codec_header.temporal_idx,
+                              frame->references[i])) {
+      --frame->num_references;
+    }
+  }
+
+  CompletedFrameVp9(std::move(frame));
+}
+
+bool RtpFrameReferenceFinder::MissingRequiredFrameVp9(uint16_t picture_id,
+                                                      const GofInfoVP9& gof) {
+  size_t diff = ForwardDiff<uint16_t, kPicIdLength>(gof.pid_start, picture_id);
+  size_t gof_idx = diff % gof.num_frames_in_gof;
+  size_t temporal_idx = gof.temporal_idx[gof_idx];
+
+  // For every reference this frame has, check if there is a frame missing in
+  // the interval (|ref_pid|, |picture_id|) in any of the lower temporal
+  // layers. If so, we are missing a required frame.
+  uint8_t num_references = gof.num_ref_pics[gof_idx];
+  for (size_t i = 0; i < num_references; ++i) {
+    uint16_t ref_pid =
+        Subtract<kPicIdLength>(picture_id, gof.pid_diff[gof_idx][i]);
+    for (size_t l = 0; l < temporal_idx; ++l) {
+      auto missing_frame_it = missing_frames_for_layer_[l].lower_bound(ref_pid);
+      if (missing_frame_it != missing_frames_for_layer_[l].end() &&
+          AheadOf<uint16_t, kPicIdLength>(picture_id, *missing_frame_it)) {
+        return true;
+      }
+    }
+  }
+  return false;
+}
+
+void RtpFrameReferenceFinder::FrameReceivedVp9(uint16_t picture_id,
+                                               const GofInfoVP9& gof) {
+  RTC_DCHECK_NE(-1, last_picture_id_);
+
+  // If there is a gap, find which temporal layer the missing frames
+  // belong to and add the frame as missing for that temporal layer.
+  // Otherwise, remove this frame from the set of missing frames.
+  if (AheadOf<uint16_t, kPicIdLength>(picture_id, last_picture_id_)) {
+    size_t diff =
+        ForwardDiff<uint16_t, kPicIdLength>(gof.pid_start, last_picture_id_);
+    size_t gof_idx = diff % gof.num_frames_in_gof;
+
+    last_picture_id_ = Add<kPicIdLength>(last_picture_id_, 1);
+    while (last_picture_id_ != picture_id) {
+      ++gof_idx;
+      RTC_DCHECK_NE(0ul, gof_idx % gof.num_frames_in_gof);
+      size_t temporal_idx = gof.temporal_idx[gof_idx];
+      missing_frames_for_layer_[temporal_idx].insert(last_picture_id_);
+      last_picture_id_ = Add<kPicIdLength>(last_picture_id_, 1);
+    }
+  } else {
+    size_t diff =
+        ForwardDiff<uint16_t, kPicIdLength>(gof.pid_start, picture_id);
+    size_t gof_idx = diff % gof.num_frames_in_gof;
+    size_t temporal_idx = gof.temporal_idx[gof_idx];
+    missing_frames_for_layer_[temporal_idx].erase(picture_id);
+  }
+}
+
+bool RtpFrameReferenceFinder::UpSwitchInIntervalVp9(uint16_t picture_id,
+                                                    uint8_t temporal_idx,
+                                                    uint16_t pid_ref) {
+  for (auto up_switch_it = up_switch_.upper_bound(pid_ref);
+       up_switch_it != up_switch_.end() &&
+       AheadOf<uint16_t, kPicIdLength>(picture_id, up_switch_it->first);
+       ++up_switch_it) {
+    if (up_switch_it->second < temporal_idx)
+      return true;
+  }
+
+  return false;
+}
+
+void RtpFrameReferenceFinder::CompletedFrameVp9(
+    std::unique_ptr<RtpFrameObject> frame) {
+  for (size_t i = 0; i < frame->num_references; ++i)
+    frame->references[i] = UnwrapPictureId(frame->references[i]);
+  frame->picture_id = UnwrapPictureId(frame->picture_id);
+
+  frame_callback_->OnCompleteFrame(std::move(frame));
+  RetryStashedFrames();
+}
+
+uint16_t RtpFrameReferenceFinder::UnwrapPictureId(uint16_t picture_id) {
+  RTC_DCHECK_NE(-1, last_unwrap_);
+
+  uint16_t unwrap_truncated = last_unwrap_ % kPicIdLength;
+  uint16_t diff = MinDiff<uint16_t, kPicIdLength>(unwrap_truncated, picture_id);
+
+  if (AheadOf<uint16_t, kPicIdLength>(picture_id, unwrap_truncated))
+    last_unwrap_ = Add<1 << 16>(last_unwrap_, diff);
+  else
+    last_unwrap_ = Subtract<1 << 16>(last_unwrap_, diff);
+
+  return last_unwrap_;
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/rtp_frame_reference_finder.h b/chromium/third_party/webrtc/modules/video_coding/rtp_frame_reference_finder.h
new file mode 100644
index 00000000000..7289b803bd0
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/rtp_frame_reference_finder.h
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_
+
+#include <array>
+#include <map>
+#include <memory>
+#include <queue>
+#include <set>
+#include <utility>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/sequence_number_util.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class RtpFrameObject;
+class OnCompleteFrameCallback;
+
+class RtpFrameReferenceFinder {
+ public:
+  explicit RtpFrameReferenceFinder(OnCompleteFrameCallback* frame_callback);
+  void ManageFrame(std::unique_ptr<RtpFrameObject> frame);
+
+ private:
+  static const uint16_t kPicIdLength = 1 << 7;
+  static const uint8_t kMaxTemporalLayers = 5;
+  static const int kMaxLayerInfo = 10;
+  static const int kMaxStashedFrames = 10;
+  static const int kMaxNotYetReceivedFrames = 20;
+  static const int kMaxGofSaved = 15;
+
+  rtc::CriticalSection crit_;
+
+  // Retry finding references for all frames that previously didn't have
+  // all information needed.
+  void RetryStashedFrames() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Find references for generic frames.
+  void ManageFrameGeneric(std::unique_ptr<RtpFrameObject> frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Find references for Vp8 frames
+  void ManageFrameVp8(std::unique_ptr<RtpFrameObject> frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Updates all necessary state used to determine frame references
+  // for Vp8 and then calls the |frame_callback| callback with the
+  // completed frame.
+  void CompletedFrameVp8(std::unique_ptr<RtpFrameObject> frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Find references for Vp9 frames
+  void ManageFrameVp9(std::unique_ptr<RtpFrameObject> frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Unwrap the picture id and the frame references  and then call the
+  // |frame_callback| callback with the completed frame.
+  void CompletedFrameVp9(std::unique_ptr<RtpFrameObject> frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Check if we are missing a frame necessary to determine the references
+  // for this frame.
+  bool MissingRequiredFrameVp9(uint16_t picture_id, const GofInfoVP9& gof)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Updates which frames that have been received. If there is a gap,
+  // missing frames will be added to |missing_frames_for_layer_| or
+  // if this is an already missing frame then it will be removed.
+  void FrameReceivedVp9(uint16_t picture_id, const GofInfoVP9& gof)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Check if there is a frame with the up-switch flag set in the interval
+  // (|pid_ref|, |picture_id|) with temporal layer smaller than |temporal_idx|.
+  bool UpSwitchInIntervalVp9(uint16_t picture_id,
+                             uint8_t temporal_idx,
+                             uint16_t pid_ref) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // All picture ids are unwrapped to 16 bits.
+  uint16_t UnwrapPictureId(uint16_t picture_id) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Holds the last sequence number of the last frame that has been created
+  // given the last sequence number of a given keyframe.
+  std::map<uint16_t, uint16_t, DescendingSeqNumComp<uint16_t>> last_seq_num_gop_
+      GUARDED_BY(crit_);
+
+  // Save the last picture id in order to detect when there is a gap in frames
+  // that have not yet been fully received.
+  int last_picture_id_ GUARDED_BY(crit_);
+
+  // The last unwrapped picture id. Used to unwrap the picture id from a length
+  // of |kPicIdLength| to 16 bits.
+  int last_unwrap_ GUARDED_BY(crit_);
+
+  // Frames earlier than the last received frame that have not yet been
+  // fully received.
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t, kPicIdLength>>
+      not_yet_received_frames_ GUARDED_BY(crit_);
+
+  // Frames that have been fully received but didn't have all the information
+  // needed to determine their references.
+  std::queue<std::unique_ptr<RtpFrameObject>> stashed_frames_ GUARDED_BY(crit_);
+
+  // Holds the information about the last completed frame for a given temporal
+  // layer given a Tl0 picture index.
+  std::map<uint8_t,
+           std::array<int16_t, kMaxTemporalLayers>,
+           DescendingSeqNumComp<uint8_t>>
+      layer_info_ GUARDED_BY(crit_);
+
+  // Where the current scalability structure is in the
+  // |scalability_structures_| array.
+  uint8_t current_ss_idx_;
+
+  // Holds received scalability structures.
+  std::array<GofInfoVP9, kMaxGofSaved> scalability_structures_
+      GUARDED_BY(crit_);
+
+  // Holds the picture id and the Gof information for a given TL0 picture index.
+  std::map<uint8_t,
+           std::pair<uint16_t, GofInfoVP9*>,
+           DescendingSeqNumComp<uint8_t>>
+      gof_info_ GUARDED_BY(crit_);
+
+  // Keep track of which picture id and which temporal layer that had the
+  // up switch flag set.
+  std::map<uint16_t, uint8_t> up_switch_ GUARDED_BY(crit_);
+
+  // For every temporal layer, keep a set of which frames that are missing.
+  std::array<std::set<uint16_t, DescendingSeqNumComp<uint16_t, kPicIdLength>>,
+             kMaxTemporalLayers>
+      missing_frames_for_layer_ GUARDED_BY(crit_);
+
+  OnCompleteFrameCallback* frame_callback_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/test/rtp_player.cc b/chromium/third_party/webrtc/modules/video_coding/test/rtp_player.cc
index 97d63e0fb54..d5fa9ae936c 100644
--- a/chromium/third_party/webrtc/modules/video_coding/test/rtp_player.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/test/rtp_player.cc
@@ -12,9 +12,11 @@
 
 #include <stdio.h>
 
+#include <cstdlib>
 #include <map>
 #include <memory>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
@@ -227,7 +229,6 @@ class SsrcHandlers {
       return -1;
     }
 
-    handler->rtp_module_->SetNACKStatus(kNackOff);
     handler->rtp_header_parser_->RegisterRtpHeaderExtension(
         kRtpExtensionTransmissionTimeOffset,
         kDefaultTransmissionTimeOffsetExtensionId);
@@ -341,7 +342,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
     assert(packet_source);
     assert(packet_source->get());
     packet_source_.swap(*packet_source);
-    srand(321);
+    std::srand(321);
   }
 
   virtual ~RtpPlayerImpl() {}
@@ -434,7 +435,8 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 
       if (no_loss_startup_ > 0) {
         no_loss_startup_--;
-      } else if ((rand() + 1.0) / (RAND_MAX + 1.0) < loss_rate_) {  // NOLINT
+      } else if ((std::rand() + 1.0) / (RAND_MAX + 1.0) <
+                 loss_rate_) {  // NOLINT
         uint16_t seq_num = header.sequenceNumber;
         lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
         DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
diff --git a/chromium/third_party/webrtc/modules/video_coding/test/stream_generator.h b/chromium/third_party/webrtc/modules/video_coding/test/stream_generator.h
index 36b26db92e4..9eb957194f1 100644
--- a/chromium/third_party/webrtc/modules/video_coding/test/stream_generator.h
+++ b/chromium/third_party/webrtc/modules/video_coding/test/stream_generator.h
@@ -12,6 +12,7 @@
 
 #include <list>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/video_coding/packet.h"
 #include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/typedefs.h"
diff --git a/chromium/third_party/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc b/chromium/third_party/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
index e774db16520..8c674ef5042 100644
--- a/chromium/third_party/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
@@ -14,6 +14,7 @@
 
 #include <algorithm>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/test/test_util.h"
 #include "webrtc/system_wrappers/include/clock.h"
diff --git a/chromium/third_party/webrtc/modules/video_coding/timing.h b/chromium/third_party/webrtc/modules/video_coding/timing.h
index a45eee38c6c..e593c9acbc3 100644
--- a/chromium/third_party/webrtc/modules/video_coding/timing.h
+++ b/chromium/third_party/webrtc/modules/video_coding/timing.h
@@ -28,7 +28,7 @@ class VCMTiming {
   // The primary timing component should be passed
   // if this is the dual timing component.
   explicit VCMTiming(Clock* clock, VCMTiming* master_timing = NULL);
-  ~VCMTiming();
+  virtual ~VCMTiming();
 
   // Resets the timing to the initial state.
   void Reset();
@@ -69,11 +69,11 @@ class VCMTiming {
   // Returns the receiver system time when the frame with timestamp
   // frame_timestamp should be rendered, assuming that the system time currently
   // is now_ms.
-  int64_t RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms) const;
+  virtual int64_t RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms) const;
 
   // Returns the maximum time in ms that we can wait for a frame to become
   // complete before we must pass it to the decoder.
-  uint32_t MaxWaitingTime(int64_t render_time_ms, int64_t now_ms) const;
+  virtual uint32_t MaxWaitingTime(int64_t render_time_ms, int64_t now_ms) const;
 
   // Returns the current target delay which is required delay + decode time +
   // render delay.
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/frame_dropper.cc b/chromium/third_party/webrtc/modules/video_coding/utility/frame_dropper.cc
index c95048c0743..5de7526ac24 100644
--- a/chromium/third_party/webrtc/modules/video_coding/utility/frame_dropper.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/frame_dropper.cc
@@ -73,6 +73,7 @@ void FrameDropper::Reset() {
   incoming_frame_rate_ = kDefaultIncomingFrameRate;
 
   large_frame_accumulation_count_ = 0;
+  large_frame_accumulation_chunk_size_ = 0;
   large_frame_accumulation_spread_ = 0.5 * kDefaultIncomingFrameRate;
 
   drop_next_ = false;
@@ -129,13 +130,6 @@ void FrameDropper::Fill(size_t framesize_bytes, bool delta_frame) {
   // Change the level of the accumulator (bucket)
   accumulator_ += framesize_kbits;
   CapAccumulator();
-  LOG(LS_VERBOSE) << "FILL acc " << accumulator_ << " max " << accumulator_max_
-                  << " count " << large_frame_accumulation_count_ << " chunk "
-                  << large_frame_accumulation_chunk_size_ << " spread "
-                  << large_frame_accumulation_spread_ << " delta avg "
-                  << delta_frame_size_avg_kbits_.filtered() << " SIZE "
-                  << framesize_kbits << "key frame ratio "
-                  << key_frame_ratio_.filtered();
 }
 
 void FrameDropper::Leak(uint32_t input_framerate) {
@@ -160,10 +154,6 @@ void FrameDropper::Leak(uint32_t input_framerate) {
   if (accumulator_ < 0.0f) {
     accumulator_ = 0.0f;
   }
-  LOG(LS_VERBOSE) << "LEAK acc " << accumulator_ << " max " << accumulator_max_
-                  << " count " << large_frame_accumulation_count_ << " spread "
-                  << large_frame_accumulation_spread_ << " delta avg "
-                  << delta_frame_size_avg_kbits_.filtered();
   UpdateRatio();
 }
 
@@ -201,8 +191,6 @@ bool FrameDropper::DropFrame() {
     drop_next_ = false;
     drop_count_ = 0;
   }
-  LOG(LS_VERBOSE) << " drop_ratio_ " << drop_ratio_.filtered()
-                  << " drop_count_ " << drop_count_;
 
   if (drop_ratio_.filtered() >= 0.5f) {  // Drops per keep
     // limit is the number of frames we should drop between each kept frame
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer.cc b/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer.cc
new file mode 100644
index 00000000000..97f1da30e40
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer.cc
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+
+IvfFileWriter::IvfFileWriter(const std::string& file_name,
+                             std::unique_ptr<FileWrapper> file,
+                             VideoCodecType codec_type)
+    : codec_type_(codec_type),
+      num_frames_(0),
+      width_(0),
+      height_(0),
+      last_timestamp_(-1),
+      using_capture_timestamps_(false),
+      file_name_(file_name),
+      file_(std::move(file)) {}
+
+IvfFileWriter::~IvfFileWriter() {
+  Close();
+}
+
+const size_t kIvfHeaderSize = 32;
+
+std::unique_ptr<IvfFileWriter> IvfFileWriter::Open(const std::string& file_name,
+                                                   VideoCodecType codec_type) {
+  std::unique_ptr<IvfFileWriter> file_writer;
+  std::unique_ptr<FileWrapper> file(FileWrapper::Create());
+  if (file->OpenFile(file_name.c_str(), false) != 0)
+    return file_writer;
+
+  file_writer.reset(new IvfFileWriter(
+      file_name, std::unique_ptr<FileWrapper>(std::move(file)), codec_type));
+  if (!file_writer->WriteHeader())
+    file_writer.reset();
+
+  return file_writer;
+}
+
+bool IvfFileWriter::WriteHeader() {
+  if (file_->Rewind() != 0) {
+    LOG(LS_WARNING) << "Unable to rewind output file " << file_name_;
+    return false;
+  }
+
+  uint8_t ivf_header[kIvfHeaderSize] = {0};
+  ivf_header[0] = 'D';
+  ivf_header[1] = 'K';
+  ivf_header[2] = 'I';
+  ivf_header[3] = 'F';
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[4], 0);   // Version.
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[6], 32);  // Header size.
+
+  switch (codec_type_) {
+    case kVideoCodecVP8:
+      ivf_header[8] = 'V';
+      ivf_header[9] = 'P';
+      ivf_header[10] = '8';
+      ivf_header[11] = '0';
+      break;
+    case kVideoCodecVP9:
+      ivf_header[8] = 'V';
+      ivf_header[9] = 'P';
+      ivf_header[10] = '9';
+      ivf_header[11] = '0';
+      break;
+    case kVideoCodecH264:
+      ivf_header[8] = 'H';
+      ivf_header[9] = '2';
+      ivf_header[10] = '6';
+      ivf_header[11] = '4';
+      break;
+    default:
+      LOG(LS_ERROR) << "Unknown CODEC type: " << codec_type_;
+      return false;
+  }
+
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[12], width_);
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[14], height_);
+  // Render timestamps are in ms (1/1000 scale), while RTP timestamps use a
+  // 90kHz clock.
+  ByteWriter<uint32_t>::WriteLittleEndian(
+      &ivf_header[16], using_capture_timestamps_ ? 1000 : 90000);
+  ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[20], 1);
+  ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[24],
+                                          static_cast<uint32_t>(num_frames_));
+  ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[28], 0);  // Reserved.
+
+  if (!file_->Write(ivf_header, kIvfHeaderSize)) {
+    LOG(LS_ERROR) << "Unable to write IVF header for file " << file_name_;
+    return false;
+  }
+
+  return true;
+}
+
+bool IvfFileWriter::InitFromFirstFrame(const EncodedImage& encoded_image) {
+  width_ = encoded_image._encodedWidth;
+  height_ = encoded_image._encodedHeight;
+  RTC_CHECK_GT(width_, 0);
+  RTC_CHECK_GT(height_, 0);
+  using_capture_timestamps_ = encoded_image._timeStamp == 0;
+
+  if (!WriteHeader())
+    return false;
+
+  std::string codec_name;
+  switch (codec_type_) {
+    case kVideoCodecVP8:
+      codec_name = "VP8";
+      break;
+    case kVideoCodecVP9:
+      codec_name = "VP9";
+      break;
+    case kVideoCodecH264:
+      codec_name = "H264";
+      break;
+    default:
+      codec_name = "Unknown";
+  }
+  LOG(LS_WARNING) << "Created IVF file " << file_name_
+                  << " for codec data of type " << codec_name
+                  << " at resolution " << width_ << " x " << height_
+                  << ", using " << (using_capture_timestamps_ ? "1" : "90")
+                  << "kHz clock resolution.";
+  return true;
+}
+
+bool IvfFileWriter::WriteFrame(const EncodedImage& encoded_image) {
+  RTC_DCHECK(file_->Open());
+
+  if (num_frames_ == 0 && !InitFromFirstFrame(encoded_image))
+    return false;
+
+  if ((encoded_image._encodedWidth > 0 || encoded_image._encodedHeight > 0) &&
+      (encoded_image._encodedHeight != height_ ||
+       encoded_image._encodedWidth != width_)) {
+    LOG(LS_WARNING)
+        << "Incomig frame has diffferent resolution then previous: (" << width_
+        << "x" << height_ << ") -> (" << encoded_image._encodedWidth << "x"
+        << encoded_image._encodedHeight << ")";
+  }
+
+  int64_t timestamp = using_capture_timestamps_
+                          ? encoded_image.capture_time_ms_
+                          : wrap_handler_.Unwrap(encoded_image._timeStamp);
+  if (last_timestamp_ != -1 && timestamp <= last_timestamp_) {
+    LOG(LS_WARNING) << "Timestamp no increasing: " << last_timestamp_ << " -> "
+                    << timestamp;
+  }
+  last_timestamp_ = timestamp;
+
+  const size_t kFrameHeaderSize = 12;
+  uint8_t frame_header[kFrameHeaderSize] = {};
+  ByteWriter<uint32_t>::WriteLittleEndian(
+      &frame_header[0], static_cast<uint32_t>(encoded_image._length));
+  ByteWriter<uint64_t>::WriteLittleEndian(&frame_header[4], timestamp);
+  if (!file_->Write(frame_header, kFrameHeaderSize) ||
+      !file_->Write(encoded_image._buffer, encoded_image._length)) {
+    LOG(LS_ERROR) << "Unable to write frame to file " << file_name_;
+    return false;
+  }
+
+  ++num_frames_;
+  return true;
+}
+
+bool IvfFileWriter::Close() {
+  if (!file_->Open())
+    return false;
+
+  if (num_frames_ == 0) {
+    // No frame written to file, close and remove it entirely if possible.
+    file_->CloseFile();
+    if (remove(file_name_.c_str()) != 0)
+      LOG(LS_WARNING) << "Failed to remove empty IVF file " << file_name_;
+
+    return true;
+  }
+
+  return WriteHeader() && (file_->CloseFile() == 0);
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer.h b/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer.h
new file mode 100644
index 00000000000..25d68a28034
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_IVF_FILE_WRITER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_IVF_FILE_WRITER_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/video_frame.h"
+#include "webrtc/system_wrappers/include/file_wrapper.h"
+
+namespace webrtc {
+
+class IvfFileWriter {
+ public:
+  ~IvfFileWriter();
+
+  static std::unique_ptr<IvfFileWriter> Open(const std::string& file_name,
+                                             VideoCodecType codec_type);
+  bool WriteFrame(const EncodedImage& encoded_image);
+  bool Close();
+
+ private:
+  IvfFileWriter(const std::string& path_name,
+                std::unique_ptr<FileWrapper> file,
+                VideoCodecType codec_type);
+  bool WriteHeader();
+  bool InitFromFirstFrame(const EncodedImage& encoded_image);
+
+  const VideoCodecType codec_type_;
+  size_t num_frames_;
+  uint16_t width_;
+  uint16_t height_;
+  int64_t last_timestamp_;
+  bool using_capture_timestamps_;
+  rtc::TimestampWrapAroundHandler wrap_handler_;
+  const std::string file_name_;
+  std::unique_ptr<FileWrapper> file_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(IvfFileWriter);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_IVF_FILE_WRITER_H_
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer_unittest.cc
new file mode 100644
index 00000000000..bdeef2abd5b
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/ivf_file_writer_unittest.cc
@@ -0,0 +1,176 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
+
+#include <memory>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+namespace {
+static const int kHeaderSize = 32;
+static const int kFrameHeaderSize = 12;
+static uint8_t dummy_payload[4] = {0, 1, 2, 3};
+}  // namespace
+
+class IvfFileWriterTest : public ::testing::Test {
+ protected:
+  void SetUp() override {
+    const int64_t start_id =
+        reinterpret_cast<int64_t>(this) ^ rtc::TimeMicros();
+    int64_t id = start_id;
+    do {
+      std::ostringstream oss;
+      oss << test::OutputPath() << "ivf_test_file_" << id++ << ".ivf";
+      file_name_ = oss.str();
+    } while (id < start_id + 100 && FileExists());
+    ASSERT_LT(id, start_id + 100);
+  }
+
+  bool WriteDummyTestFrames(int width,
+                            int height,
+                            int num_frames,
+                            bool use_capture_tims_ms) {
+    EncodedImage frame;
+    frame._buffer = dummy_payload;
+    frame._encodedWidth = width;
+    frame._encodedHeight = height;
+    for (int i = 1; i <= num_frames; ++i) {
+      frame._length = i % sizeof(dummy_payload);
+      if (use_capture_tims_ms) {
+        frame.capture_time_ms_ = i;
+      } else {
+        frame._timeStamp = i;
+      }
+      if (!file_writer_->WriteFrame(frame))
+        return false;
+    }
+    return true;
+  }
+
+  void VerifyIvfHeader(FileWrapper* file,
+                       const uint8_t fourcc[4],
+                       int width,
+                       int height,
+                       uint32_t num_frames,
+                       bool use_capture_tims_ms) {
+    uint8_t data[kHeaderSize];
+    ASSERT_EQ(kHeaderSize, file->Read(data, kHeaderSize));
+
+    uint8_t dkif[4] = {'D', 'K', 'I', 'F'};
+    EXPECT_EQ(0, memcmp(dkif, data, 4));
+    EXPECT_EQ(0u, ByteReader<uint16_t>::ReadLittleEndian(&data[4]));
+    EXPECT_EQ(32u, ByteReader<uint16_t>::ReadLittleEndian(&data[6]));
+    EXPECT_EQ(0, memcmp(fourcc, &data[8], 4));
+    EXPECT_EQ(width, ByteReader<uint16_t>::ReadLittleEndian(&data[12]));
+    EXPECT_EQ(height, ByteReader<uint16_t>::ReadLittleEndian(&data[14]));
+    EXPECT_EQ(use_capture_tims_ms ? 1000u : 90000u,
+              ByteReader<uint32_t>::ReadLittleEndian(&data[16]));
+    EXPECT_EQ(1u, ByteReader<uint32_t>::ReadLittleEndian(&data[20]));
+    EXPECT_EQ(num_frames, ByteReader<uint32_t>::ReadLittleEndian(&data[24]));
+    EXPECT_EQ(0u, ByteReader<uint32_t>::ReadLittleEndian(&data[28]));
+  }
+
+  void VerifyDummyTestFrames(FileWrapper* file, uint32_t num_frames) {
+    const int kMaxFrameSize = 4;
+    for (uint32_t i = 1; i <= num_frames; ++i) {
+      uint8_t frame_header[kFrameHeaderSize];
+      ASSERT_EQ(kFrameHeaderSize, file->Read(frame_header, kFrameHeaderSize));
+      uint32_t frame_length =
+          ByteReader<uint32_t>::ReadLittleEndian(&frame_header[0]);
+      EXPECT_EQ(i % 4, frame_length);
+      uint64_t timestamp =
+          ByteReader<uint64_t>::ReadLittleEndian(&frame_header[4]);
+      EXPECT_EQ(i, timestamp);
+
+      uint8_t data[kMaxFrameSize] = {};
+      ASSERT_EQ(frame_length,
+                static_cast<uint32_t>(file->Read(data, frame_length)));
+      EXPECT_EQ(0, memcmp(data, dummy_payload, frame_length));
+    }
+  }
+
+  void RunBasicFileStructureTest(VideoCodecType codec_type,
+                                 const uint8_t fourcc[4],
+                                 bool use_capture_tims_ms) {
+    file_writer_ = IvfFileWriter::Open(file_name_, codec_type);
+    ASSERT_TRUE(file_writer_.get());
+    const int kWidth = 320;
+    const int kHeight = 240;
+    const int kNumFrames = 257;
+    EXPECT_TRUE(
+        WriteDummyTestFrames(kWidth, kHeight, kNumFrames, use_capture_tims_ms));
+    EXPECT_TRUE(file_writer_->Close());
+
+    std::unique_ptr<FileWrapper> out_file(FileWrapper::Create());
+    ASSERT_EQ(0, out_file->OpenFile(file_name_.c_str(), true));
+    VerifyIvfHeader(out_file.get(), fourcc, kWidth, kHeight, kNumFrames,
+                    use_capture_tims_ms);
+    VerifyDummyTestFrames(out_file.get(), kNumFrames);
+
+    EXPECT_EQ(0, out_file->CloseFile());
+    EXPECT_EQ(0, remove(file_name_.c_str()));
+  }
+
+  bool FileExists() {
+    std::unique_ptr<FileWrapper> file_wrapper(FileWrapper::Create());
+    return file_wrapper->OpenFile(file_name_.c_str(), true) == 0;
+  }
+
+  std::string file_name_;
+  std::unique_ptr<IvfFileWriter> file_writer_;
+};
+
+TEST_F(IvfFileWriterTest, RemovesUnusedFile) {
+  file_writer_ = IvfFileWriter::Open(file_name_, kVideoCodecVP8);
+  ASSERT_TRUE(file_writer_.get() != nullptr);
+  EXPECT_TRUE(FileExists());
+  EXPECT_TRUE(file_writer_->Close());
+  EXPECT_FALSE(FileExists());
+  EXPECT_FALSE(file_writer_->Close());  // Can't close twice.
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicVP8FileNtpTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP8, fourcc, false);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicVP8FileMsTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP8, fourcc, true);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicVP9FileNtpTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '9', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP9, fourcc, false);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicVP9FileMsTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '9', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP9, fourcc, true);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicH264FileNtpTimestamp) {
+  const uint8_t fourcc[4] = {'H', '2', '6', '4'};
+  RunBasicFileStructureTest(kVideoCodecH264, fourcc, false);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicH264FileMsTimestamp) {
+  const uint8_t fourcc[4] = {'H', '2', '6', '4'};
+  RunBasicFileStructureTest(kVideoCodecH264, fourcc, true);
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.cc b/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.cc
index c6e56697310..bb60ee036e3 100644
--- a/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.cc
@@ -11,29 +11,39 @@
 
 namespace webrtc {
 
+namespace {
 static const int kMinFps = 5;
-static const int kMeasureSecondsDownscale = 3;
 // Threshold constant used until first downscale (to permit fast rampup).
 static const int kMeasureSecondsFastUpscale = 2;
 static const int kMeasureSecondsUpscale = 5;
+static const int kMeasureSecondsDownscale = 5;
 static const int kFramedropPercentThreshold = 60;
-static const int kHdResolutionThreshold = 700 * 500;
-static const int kHdBitrateThresholdKbps = 500;
-
-const int QualityScaler::kDefaultLowQpDenominator = 3;
-// Note that this is the same for width and height to permit 120x90 in both
-// portrait and landscape mode.
-const int QualityScaler::kDefaultMinDownscaleDimension = 90;
-
-QualityScaler::QualityScaler()
-    : low_qp_threshold_(-1),
-      framerate_down_(false),
-      min_width_(kDefaultMinDownscaleDimension),
-      min_height_(kDefaultMinDownscaleDimension) {}
+// Min width/height to downscale to, set to not go below QVGA, but with some
+// margin to permit "almost-QVGA" resolutions, such as QCIF.
+static const int kMinDownscaleDimension = 140;
+// Initial resolutions corresponding to a bitrate. Aa bit above their actual
+// values to permit near-VGA and near-QVGA resolutions to use the same
+// mechanism.
+static const int kVgaBitrateThresholdKbps = 500;
+static const int kVgaNumPixels = 700 * 500;  // 640x480
+static const int kQvgaBitrateThresholdKbps = 250;
+static const int kQvgaNumPixels = 400 * 300;  // 320x240
+}  // namespace
+
+// QP thresholds are chosen to be high enough to be hit in practice when quality
+// is good, but also low enough to not cause a flip-flop behavior (e.g. going up
+// in resolution shouldn't give so bad quality that we should go back down).
+
+const int QualityScaler::kLowVp8QpThreshold = 29;
+const int QualityScaler::kBadVp8QpThreshold = 95;
+
+const int QualityScaler::kLowH264QpThreshold = 22;
+const int QualityScaler::kBadH264QpThreshold = 35;
+
+QualityScaler::QualityScaler() : low_qp_threshold_(-1) {}
 
 void QualityScaler::Init(int low_qp_threshold,
                          int high_qp_threshold,
-                         bool use_framerate_reduction,
                          int initial_bitrate_kbps,
                          int width,
                          int height,
@@ -41,7 +51,6 @@ void QualityScaler::Init(int low_qp_threshold,
   ClearSamples();
   low_qp_threshold_ = low_qp_threshold;
   high_qp_threshold_ = high_qp_threshold;
-  use_framerate_reduction_ = use_framerate_reduction;
   downscale_shift_ = 0;
   // Use a faster window for upscaling initially (but be more graceful later).
   // This enables faster initial rampups without risking strong up-down
@@ -49,25 +58,24 @@ void QualityScaler::Init(int low_qp_threshold,
   measure_seconds_upscale_ = kMeasureSecondsFastUpscale;
   const int init_width = width;
   const int init_height = height;
-  // TODO(glaznev): Investigate using thresholds for other resolutions
-  // or threshold tables.
-  if (initial_bitrate_kbps > 0 &&
-      initial_bitrate_kbps < kHdBitrateThresholdKbps) {
-    // Start scaling to roughly VGA.
-    while (width * height > kHdResolutionThreshold) {
+  if (initial_bitrate_kbps > 0) {
+    int init_num_pixels = width * height;
+    if (initial_bitrate_kbps < kVgaBitrateThresholdKbps)
+      init_num_pixels = kVgaNumPixels;
+    if (initial_bitrate_kbps < kQvgaBitrateThresholdKbps)
+      init_num_pixels = kQvgaNumPixels;
+    while (width * height > init_num_pixels) {
       ++downscale_shift_;
       width /= 2;
       height /= 2;
     }
   }
+
+  // Zero out width/height so they can be checked against inside
+  // UpdateTargetResolution.
+  res_.width = res_.height = 0;
   UpdateTargetResolution(init_width, init_height);
   ReportFramerate(fps);
-  target_framerate_ = -1;
-}
-
-void QualityScaler::SetMinResolution(int min_width, int min_height) {
-  min_width_ = min_width;
-  min_height_ = min_height;
 }
 
 // Report framerate(fps) to estimate # of samples.
@@ -96,34 +104,14 @@ void QualityScaler::OnEncodeFrame(const VideoFrame& frame) {
   int avg_drop = 0;
   int avg_qp = 0;
 
-  // When encoder consistently overshoots, framerate reduction and spatial
-  // resizing will be triggered to get a smoother video.
   if ((framedrop_percent_.GetAverage(num_samples_downscale_, &avg_drop) &&
        avg_drop >= kFramedropPercentThreshold) ||
       (average_qp_downscale_.GetAverage(num_samples_downscale_, &avg_qp) &&
        avg_qp > high_qp_threshold_)) {
-    // Reducing frame rate before spatial resolution change.
-    // Reduce frame rate only when it is above a certain number.
-    // Only one reduction is allowed for now.
-    // TODO(jackychen): Allow more than one framerate reduction.
-    if (use_framerate_reduction_ && !framerate_down_ && framerate_ >= 20) {
-      target_framerate_ = framerate_ / 2;
-      framerate_down_ = true;
-      // If frame rate has been updated, clear the buffer. We don't want
-      // spatial resolution to change right after frame rate change.
-      ClearSamples();
-    } else {
-      AdjustScale(false);
-    }
+    AdjustScale(false);
   } else if (average_qp_upscale_.GetAverage(num_samples_upscale_, &avg_qp) &&
              avg_qp <= low_qp_threshold_) {
-    if (use_framerate_reduction_ && framerate_down_) {
-      target_framerate_ = -1;
-      framerate_down_ = false;
-      ClearSamples();
-    } else {
-      AdjustScale(true);
-    }
+    AdjustScale(true);
   }
   UpdateTargetResolution(frame.width(), frame.height());
 }
@@ -132,10 +120,6 @@ QualityScaler::Resolution QualityScaler::GetScaledResolution() const {
   return res_;
 }
 
-int QualityScaler::GetTargetFramerate() const {
-  return target_framerate_;
-}
-
 const VideoFrame& QualityScaler::GetScaledFrame(const VideoFrame& frame) {
   Resolution res = GetScaledResolution();
   if (res.width == frame.width())
@@ -146,24 +130,39 @@ const VideoFrame& QualityScaler::GetScaledFrame(const VideoFrame& frame) {
   if (scaler_.Scale(frame, &scaled_frame_) != 0)
     return frame;
 
+  // TODO(perkj): Refactor the scaler to not own |scaled_frame|. VideoFrame are
+  // just thin wrappers so instead the scaler should return a
+  // rtc::scoped_refptr<VideoFrameBuffer> and a new VideoFrame be created with
+  // the meta data from |frame|. That way we would not have to set all these
+  // meta data.
   scaled_frame_.set_ntp_time_ms(frame.ntp_time_ms());
   scaled_frame_.set_timestamp(frame.timestamp());
   scaled_frame_.set_render_time_ms(frame.render_time_ms());
+  scaled_frame_.set_rotation(frame.rotation());
 
   return scaled_frame_;
 }
 
 void QualityScaler::UpdateTargetResolution(int frame_width, int frame_height) {
   assert(downscale_shift_ >= 0);
-  res_.width = frame_width;
-  res_.height = frame_height;
+  int shifts_performed = 0;
   for (int shift = downscale_shift_;
-       shift > 0 && (res_.width / 2 >= min_width_) &&
-       (res_.height / 2 >= min_height_);
-       --shift) {
-    res_.width /= 2;
-    res_.height /= 2;
+       shift > 0 && (frame_width / 2 >= kMinDownscaleDimension) &&
+       (frame_height / 2 >= kMinDownscaleDimension);
+       --shift, ++shifts_performed) {
+    frame_width /= 2;
+    frame_height /= 2;
+  }
+  // Clamp to number of shifts actually performed to not be stuck trying to
+  // scale way beyond QVGA.
+  downscale_shift_ = shifts_performed;
+  if (res_.width == frame_width && res_.height == frame_height) {
+    // No reset done/needed, using same resolution.
+    return;
   }
+  res_.width = frame_width;
+  res_.height = frame_height;
+  ClearSamples();
 }
 
 void QualityScaler::ClearSamples() {
@@ -184,11 +183,10 @@ void QualityScaler::AdjustScale(bool up) {
   if (downscale_shift_ < 0)
     downscale_shift_ = 0;
   if (!up) {
-    // Hit first downscale, start using a slower threshold for going up.
+    // First downscale hit, start using a slower threshold for going up.
     measure_seconds_upscale_ = kMeasureSecondsUpscale;
     UpdateSampleCounts();
   }
-  ClearSamples();
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.h b/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.h
index 34dda0e9f37..fe70393c21c 100644
--- a/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.h
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler.h
@@ -17,8 +17,6 @@
 namespace webrtc {
 class QualityScaler {
  public:
-  static const int kDefaultLowQpDenominator;
-  static const int kDefaultMinDownscaleDimension;
   struct Resolution {
     int width;
     int height;
@@ -27,22 +25,27 @@ class QualityScaler {
   QualityScaler();
   void Init(int low_qp_threshold,
             int high_qp_threshold,
-            bool use_framerate_reduction,
             int initial_bitrate_kbps,
             int width,
             int height,
             int fps);
-  void SetMinResolution(int min_width, int min_height);
   void ReportFramerate(int framerate);
   void ReportQP(int qp);
   void ReportDroppedFrame();
-  void Reset(int framerate, int bitrate, int width, int height);
   void OnEncodeFrame(const VideoFrame& frame);
   Resolution GetScaledResolution() const;
   const VideoFrame& GetScaledFrame(const VideoFrame& frame);
-  int GetTargetFramerate() const;
   int downscale_shift() const { return downscale_shift_; }
 
+  // QP is obtained from VP8-bitstream for HW, so the QP corresponds to the
+  // bitstream range of [0, 127] and not the user-level range of [0,63].
+  static const int kLowVp8QpThreshold;
+  static const int kBadVp8QpThreshold;
+
+  // H264 QP is in the range [0, 51].
+  static const int kLowH264QpThreshold;
+  static const int kBadH264QpThreshold;
+
  private:
   void AdjustScale(bool up);
   void UpdateTargetResolution(int frame_width, int frame_height);
@@ -59,17 +62,12 @@ class QualityScaler {
   MovingAverage<int> average_qp_downscale_;
 
   int framerate_;
-  int target_framerate_;
   int low_qp_threshold_;
   int high_qp_threshold_;
   MovingAverage<int> framedrop_percent_;
   Resolution res_;
 
   int downscale_shift_;
-  int framerate_down_;
-  bool use_framerate_reduction_;
-  int min_width_;
-  int min_height_;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
index 72e9db405ed..fdec081c900 100644
--- a/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
@@ -16,33 +16,24 @@ namespace webrtc {
 namespace {
 static const int kNumSeconds = 10;
 static const int kWidth = 1920;
-static const int kWidthVga = 640;
 static const int kHalfWidth = kWidth / 2;
 static const int kHeight = 1080;
-static const int kHeightVga = 480;
 static const int kFramerate = 30;
 static const int kLowQp = 15;
 static const int kNormalQp = 30;
+static const int kLowQpThreshold = 18;
 static const int kHighQp = 40;
-static const int kMaxQp = 56;
-static const int kDisabledBadQpThreshold = kMaxQp + 1;
+static const int kDisabledBadQpThreshold = 64;
 static const int kLowInitialBitrateKbps = 300;
 // These values need to be in sync with corresponding constants
 // in quality_scaler.cc
-static const int kMeasureSecondsDownscale = 3;
 static const int kMeasureSecondsFastUpscale = 2;
 static const int kMeasureSecondsUpscale = 5;
+static const int kMeasureSecondsDownscale = 5;
+static const int kMinDownscaleDimension = 140;
 }  // namespace
 
 class QualityScalerTest : public ::testing::Test {
- public:
-  // Temporal and spatial resolution.
-  struct Resolution {
-    int framerate;
-    int width;
-    int height;
-  };
-
  protected:
   enum ScaleDirection {
     kKeepScaleAtHighQp,
@@ -50,13 +41,11 @@ class QualityScalerTest : public ::testing::Test {
     kScaleDownAboveHighQp,
     kScaleUp
   };
-  enum BadQualityMetric { kDropFrame, kReportLowQP };
 
   QualityScalerTest() {
     input_frame_.CreateEmptyFrame(kWidth, kHeight, kWidth, kHalfWidth,
                                   kHalfWidth);
-    qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator, kHighQp, false,
-             0, 0, 0, kFramerate);
+    qs_.Init(kLowQpThreshold, kHighQp, 0, 0, 0, kFramerate);
     qs_.OnEncodeFrame(input_frame_);
   }
 
@@ -103,16 +92,6 @@ class QualityScalerTest : public ::testing::Test {
 
   void DoesNotDownscaleFrameDimensions(int width, int height);
 
-  Resolution TriggerResolutionChange(BadQualityMetric dropframe_lowqp,
-                                     int num_second,
-                                     int initial_framerate);
-
-  void VerifyQualityAdaptation(int initial_framerate,
-                               int seconds_downscale,
-                               int seconds_upscale,
-                               bool expect_spatial_resize,
-                               bool expect_framerate_reduction);
-
   void DownscaleEndsAt(int input_width,
                        int input_height,
                        int end_width,
@@ -200,7 +179,7 @@ void QualityScalerTest::ContinuouslyDownscalesByHalfDimensionsAndBackUp() {
   int min_dimension = initial_min_dimension;
   int current_shift = 0;
   // Drop all frames to force-trigger downscaling.
-  while (min_dimension >= 2 * QualityScaler::kDefaultMinDownscaleDimension) {
+  while (min_dimension >= 2 * kMinDownscaleDimension) {
     EXPECT_TRUE(TriggerScale(kScaleDown)) << "No downscale within "
                                           << kNumSeconds << " seconds.";
     qs_.OnEncodeFrame(input_frame_);
@@ -270,133 +249,50 @@ TEST_F(QualityScalerTest, DoesNotDownscaleFrom1Px) {
   DoesNotDownscaleFrameDimensions(1, 1);
 }
 
-QualityScalerTest::Resolution QualityScalerTest::TriggerResolutionChange(
-    BadQualityMetric dropframe_lowqp,
-    int num_second,
-    int initial_framerate) {
-  QualityScalerTest::Resolution res;
-  res.framerate = initial_framerate;
-  qs_.OnEncodeFrame(input_frame_);
-  res.width = qs_.GetScaledResolution().width;
-  res.height = qs_.GetScaledResolution().height;
-  for (int i = 0; i < kFramerate * num_second; ++i) {
-    switch (dropframe_lowqp) {
-      case kReportLowQP:
-        qs_.ReportQP(kLowQp);
-        break;
-      case kDropFrame:
-        qs_.ReportDroppedFrame();
-        break;
-    }
-    qs_.OnEncodeFrame(input_frame_);
-    // Simulate the case when SetRates is called right after reducing
-    // framerate.
-    qs_.ReportFramerate(initial_framerate);
-    res.framerate = qs_.GetTargetFramerate();
-    if (res.framerate != -1)
-      qs_.ReportFramerate(res.framerate);
-    res.width = qs_.GetScaledResolution().width;
-    res.height = qs_.GetScaledResolution().height;
-  }
-  return res;
-}
-
-void QualityScalerTest::VerifyQualityAdaptation(
-    int initial_framerate,
-    int seconds_downscale,
-    int seconds_upscale,
-    bool expect_spatial_resize,
-    bool expect_framerate_reduction) {
-  qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator,
-           kDisabledBadQpThreshold, true, 0, 0, 0, initial_framerate);
-  qs_.OnEncodeFrame(input_frame_);
-  int init_width = qs_.GetScaledResolution().width;
-  int init_height = qs_.GetScaledResolution().height;
-
-  // Test reducing framerate by dropping frame continuously.
-  QualityScalerTest::Resolution res =
-      TriggerResolutionChange(kDropFrame, seconds_downscale, initial_framerate);
-
-  if (expect_framerate_reduction) {
-    EXPECT_LT(res.framerate, initial_framerate);
-  } else {
-    // No framerate reduction, video decimator should be disabled.
-    EXPECT_EQ(-1, res.framerate);
-  }
-
-  if (expect_spatial_resize) {
-    EXPECT_LT(res.width, init_width);
-    EXPECT_LT(res.height, init_height);
-  } else {
-    EXPECT_EQ(init_width, res.width);
-    EXPECT_EQ(init_height, res.height);
-  }
-
-  // The "seconds * 1.5" is to ensure spatial resolution to recover.
-  // For example, in 6 seconds test, framerate reduction happens in the first
-  // 3 seconds from 30fps to 15fps and causes the buffer size to be half of the
-  // original one. Then it will take only 45 samples to downscale (twice in 90
-  // samples). So to recover the resolution changes, we need more than 10
-  // seconds (i.e, seconds_upscale * 1.5). This is because the framerate
-  // increases before spatial size recovers, so it will take 150 samples to
-  // recover spatial size (300 for twice).
-  res = TriggerResolutionChange(kReportLowQP, seconds_upscale * 1.5,
-                                initial_framerate);
-  EXPECT_EQ(-1, res.framerate);
-  EXPECT_EQ(init_width, res.width);
-  EXPECT_EQ(init_height, res.height);
-}
-
-// In 3 seconds test, only framerate adjusting should happen and 5 second
-// upscaling duration, only a framerate adjusting should happen.
-TEST_F(QualityScalerTest, ChangeFramerateOnly) {
-  VerifyQualityAdaptation(kFramerate, kMeasureSecondsDownscale,
-                          kMeasureSecondsUpscale, false, true);
-}
-
-// In 6 seconds test, framerate adjusting and scaling are both
-// triggered, it shows that scaling would happen after framerate
-// adjusting.
-TEST_F(QualityScalerTest, ChangeFramerateAndSpatialSize) {
-  VerifyQualityAdaptation(kFramerate, kMeasureSecondsDownscale * 2,
-                          kMeasureSecondsUpscale * 2, true, true);
-}
-
-// When starting from a low framerate, only spatial size will be changed.
-TEST_F(QualityScalerTest, ChangeSpatialSizeOnly) {
-  qs_.ReportFramerate(kFramerate >> 1);
-  VerifyQualityAdaptation(kFramerate >> 1, kMeasureSecondsDownscale * 2,
-                          kMeasureSecondsUpscale * 2, true, false);
-}
-
 TEST_F(QualityScalerTest, DoesNotDownscaleBelow2xDefaultMinDimensionsWidth) {
   DoesNotDownscaleFrameDimensions(
-      2 * QualityScaler::kDefaultMinDownscaleDimension - 1, 1000);
+      2 * kMinDownscaleDimension - 1, 1000);
 }
 
 TEST_F(QualityScalerTest, DoesNotDownscaleBelow2xDefaultMinDimensionsHeight) {
   DoesNotDownscaleFrameDimensions(
-      1000, 2 * QualityScaler::kDefaultMinDownscaleDimension - 1);
+      1000, 2 * kMinDownscaleDimension - 1);
 }
 
 TEST_F(QualityScalerTest, DownscaleToVgaOnLowInitialBitrate) {
-  qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator,
-           kDisabledBadQpThreshold, true,
-           kLowInitialBitrateKbps, kWidth, kHeight, kFramerate);
+  static const int kWidth720p = 1280;
+  static const int kHeight720p = 720;
+  static const int kInitialBitrateKbps = 300;
+  input_frame_.CreateEmptyFrame(kWidth720p, kHeight720p, kWidth720p,
+                                kWidth720p / 2, kWidth720p / 2);
+  qs_.Init(kLowQpThreshold, kDisabledBadQpThreshold, kInitialBitrateKbps,
+           kWidth720p, kHeight720p, kFramerate);
   qs_.OnEncodeFrame(input_frame_);
   int init_width = qs_.GetScaledResolution().width;
   int init_height = qs_.GetScaledResolution().height;
-  EXPECT_LE(init_width, kWidthVga);
-  EXPECT_LE(init_height, kHeightVga);
+  EXPECT_EQ(640, init_width);
+  EXPECT_EQ(360, init_height);
+}
+
+TEST_F(QualityScalerTest, DownscaleToQvgaOnLowerInitialBitrate) {
+  static const int kWidth720p = 1280;
+  static const int kHeight720p = 720;
+  static const int kInitialBitrateKbps = 200;
+  input_frame_.CreateEmptyFrame(kWidth720p, kHeight720p, kWidth720p,
+                                kWidth720p / 2, kWidth720p / 2);
+  qs_.Init(kLowQpThreshold, kDisabledBadQpThreshold, kInitialBitrateKbps,
+           kWidth720p, kHeight720p, kFramerate);
+  qs_.OnEncodeFrame(input_frame_);
+  int init_width = qs_.GetScaledResolution().width;
+  int init_height = qs_.GetScaledResolution().height;
+  EXPECT_EQ(320, init_width);
+  EXPECT_EQ(180, init_height);
 }
 
 TEST_F(QualityScalerTest, DownscaleAfterMeasuredSecondsThenSlowerBackUp) {
-  QualityScalerTest::Resolution initial_res;
-  qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator, kHighQp, false, 0,
-           kWidth, kHeight, kFramerate);
+  qs_.Init(kLowQpThreshold, kHighQp, 0, kWidth, kHeight, kFramerate);
   qs_.OnEncodeFrame(input_frame_);
-  initial_res.width = qs_.GetScaledResolution().width;
-  initial_res.height = qs_.GetScaledResolution().height;
+  QualityScaler::Resolution initial_res = qs_.GetScaledResolution();
 
   // Should not downscale if less than kMeasureSecondsDownscale seconds passed.
   for (int i = 0; i < kFramerate * kMeasureSecondsDownscale - 1; ++i) {
@@ -431,12 +327,10 @@ TEST_F(QualityScalerTest, DownscaleAfterMeasuredSecondsThenSlowerBackUp) {
 }
 
 TEST_F(QualityScalerTest, UpscaleQuicklyInitiallyAfterMeasuredSeconds) {
-  QualityScalerTest::Resolution initial_res;
-  qs_.Init(kMaxQp / QualityScaler::kDefaultLowQpDenominator, kHighQp, false,
-           kLowInitialBitrateKbps, kWidth, kHeight, kFramerate);
+  qs_.Init(kLowQpThreshold, kHighQp, kLowInitialBitrateKbps, kWidth, kHeight,
+           kFramerate);
   qs_.OnEncodeFrame(input_frame_);
-  initial_res.width = qs_.GetScaledResolution().width;
-  initial_res.height = qs_.GetScaledResolution().height;
+  QualityScaler::Resolution initial_res = qs_.GetScaledResolution();
 
   // Should not upscale if less than kMeasureSecondsFastUpscale seconds passed.
   for (int i = 0; i < kFramerate * kMeasureSecondsFastUpscale - 1; ++i) {
@@ -480,36 +374,20 @@ void QualityScalerTest::DownscaleEndsAt(int input_width,
   }
 }
 
-TEST_F(QualityScalerTest, DefaultDownscalesTo160x90) {
-  DownscaleEndsAt(320, 180, 160, 90);
-}
-
-TEST_F(QualityScalerTest, DefaultDownscalesTo90x160) {
-  DownscaleEndsAt(180, 320, 90, 160);
-}
-
-TEST_F(QualityScalerTest, DefaultDownscalesFrom1280x720To160x90) {
-  DownscaleEndsAt(1280, 720, 160, 90);
-}
-
-TEST_F(QualityScalerTest, DefaultDoesntDownscaleBelow160x90) {
-  DownscaleEndsAt(320 - 1, 180 - 1, 320 - 1, 180 - 1);
+TEST_F(QualityScalerTest, DownscalesTo320x180) {
+  DownscaleEndsAt(640, 360, 320, 180);
 }
 
-TEST_F(QualityScalerTest, DefaultDoesntDownscaleBelow90x160) {
-  DownscaleEndsAt(180 - 1, 320 - 1, 180 - 1, 320 - 1);
+TEST_F(QualityScalerTest, DownscalesTo180x320) {
+  DownscaleEndsAt(360, 640, 180, 320);
 }
 
-TEST_F(QualityScalerTest, RespectsMinResolutionWidth) {
-  // Should end at 200x100, as width can't go lower.
-  qs_.SetMinResolution(200, 10);
-  DownscaleEndsAt(1600, 800, 200, 100);
+TEST_F(QualityScalerTest, DownscalesFrom1280x720To320x180) {
+  DownscaleEndsAt(1280, 720, 320, 180);
 }
 
-TEST_F(QualityScalerTest, RespectsMinResolutionHeight) {
-  // Should end at 100x200, as height can't go lower.
-  qs_.SetMinResolution(10, 200);
-  DownscaleEndsAt(800, 1600, 100, 200);
+TEST_F(QualityScalerTest, DoesntDownscaleInitialQvga) {
+  DownscaleEndsAt(320, 180, 320, 180);
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/video_coding_utility.gyp b/chromium/third_party/webrtc/modules/video_coding/utility/video_coding_utility.gyp
index 42cbb3d4e03..8edfd619863 100644
--- a/chromium/third_party/webrtc/modules/video_coding/utility/video_coding_utility.gyp
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/video_coding_utility.gyp
@@ -20,6 +20,8 @@
       'sources': [
         'frame_dropper.cc',
         'frame_dropper.h',
+        'ivf_file_writer.cc',
+        'ivf_file_writer.h',
         'moving_average.h',
         'qp_parser.cc',
         'qp_parser.h',
diff --git a/chromium/third_party/webrtc/modules/video_coding/utility/vp8_header_parser.cc b/chromium/third_party/webrtc/modules/video_coding/utility/vp8_header_parser.cc
index 631385d0f25..d88fb6cc21e 100644
--- a/chromium/third_party/webrtc/modules/video_coding/utility/vp8_header_parser.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/utility/vp8_header_parser.cc
@@ -74,8 +74,9 @@ static int VP8GetBit(VP8BitReader* const br, int prob) {
   uint8_t range = br->range_;
   if (br->bits_ < 0) {
     VP8LoadNewBytes(br);
+    if (br->eof_)
+      return 0;
   }
-
   const int pos = br->bits_;
   const uint8_t split = (range * prob) >> 8;
   const uint8_t value = static_cast<uint8_t>(br->value_ >> pos);
diff --git a/chromium/third_party/webrtc/modules/video_coding/video_coding.gypi b/chromium/third_party/webrtc/modules/video_coding/video_coding.gypi
index 7cfefed3ee4..27454a47115 100644
--- a/chromium/third_party/webrtc/modules/video_coding/video_coding.gypi
+++ b/chromium/third_party/webrtc/modules/video_coding/video_coding.gypi
@@ -22,19 +22,19 @@
       ],
       'sources': [
         # interfaces
-        'include/bitrate_adjuster.h',
         'include/video_coding.h',
         'include/video_coding_defines.h',
 
         # headers
         'codec_database.h',
         'codec_timer.h',
-        'content_metrics_processing.h',
         'decoding_state.h',
         'encoded_frame.h',
         'fec_tables_xor.h',
         'frame_buffer.h',
+        'frame_buffer2.h',
         'frame_object.h',
+        'rtp_frame_reference_finder.h',
         'generic_decoder.h',
         'generic_encoder.h',
         'histogram.h',
@@ -50,8 +50,6 @@
         'packet.h',
         'packet_buffer.h',
         'percentile_filter.h',
-        'qm_select_data.h',
-        'qm_select.h',
         'receiver.h',
         'rtt_filter.h',
         'session_info.h',
@@ -60,14 +58,14 @@
         'video_coding_impl.h',
 
         # sources
-        'bitrate_adjuster.cc',
         'codec_database.cc',
         'codec_timer.cc',
-        'content_metrics_processing.cc',
         'decoding_state.cc',
         'encoded_frame.cc',
         'frame_buffer.cc',
+        'frame_buffer2.cc',
         'frame_object.cc',
+        'rtp_frame_reference_finder.cc',
         'generic_decoder.cc',
         'generic_encoder.cc',
         'inter_frame_delay.cc',
@@ -80,7 +78,6 @@
         'packet.cc',
         'packet_buffer.cc',
         'percentile_filter.cc',
-        'qm_select.cc',
         'receiver.cc',
         'rtt_filter.cc',
         'session_info.cc',
diff --git a/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.cc b/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.cc
index e5f0ee12222..72bcc9a0594 100644
--- a/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.cc
@@ -14,6 +14,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/encoded_frame.h"
 #include "webrtc/modules/video_coding/jitter_buffer.h"
@@ -53,7 +54,6 @@ class EncodedImageCallbackWrapper : public EncodedImageCallback {
     callback_ = callback;
   }
 
-  // TODO(andresp): Change to void as return value is ignored.
   virtual int32_t Encoded(const EncodedImage& encoded_image,
                           const CodecSpecificInfo* codec_specific_info,
                           const RTPFragmentationHeader* fragmentation) {
@@ -73,23 +73,19 @@ class VideoCodingModuleImpl : public VideoCodingModule {
  public:
   VideoCodingModuleImpl(Clock* clock,
                         EventFactory* event_factory,
-                        bool owns_event_factory,
                         VideoEncoderRateObserver* encoder_rate_observer,
-                        VCMQMSettingsCallback* qm_settings_callback,
                         NackSender* nack_sender,
-                        KeyFrameRequestSender* keyframe_request_sender)
+                        KeyFrameRequestSender* keyframe_request_sender,
+                        EncodedImageCallback* pre_decode_image_callback)
       : VideoCodingModule(),
-        sender_(clock,
-                &post_encode_callback_,
-                encoder_rate_observer,
-                qm_settings_callback),
+        sender_(clock, &post_encode_callback_, encoder_rate_observer, nullptr),
         receiver_(clock,
                   event_factory,
+                  pre_decode_image_callback,
                   nack_sender,
-                  keyframe_request_sender),
-        own_event_factory_(owns_event_factory ? event_factory : NULL) {}
+                  keyframe_request_sender) {}
 
-  virtual ~VideoCodingModuleImpl() { own_event_factory_.reset(); }
+  virtual ~VideoCodingModuleImpl() {}
 
   int64_t TimeUntilNextProcess() override {
     int64_t sender_time = sender_.TimeUntilNextProcess();
@@ -132,16 +128,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
     return sender_.SetChannelParameters(target_bitrate, lossRate, rtt);
   }
 
-  int32_t RegisterTransportCallback(
-      VCMPacketizationCallback* transport) override {
-    return sender_.RegisterTransportCallback(transport);
-  }
-
-  int32_t RegisterSendStatisticsCallback(
-      VCMSendStatisticsCallback* sendStats) override {
-    return sender_.RegisterSendStatisticsCallback(sendStats);
-  }
-
   int32_t RegisterProtectionCallback(
       VCMProtectionCallback* protection) override {
     return sender_.RegisterProtectionCallback(protection);
@@ -156,12 +142,11 @@ class VideoCodingModuleImpl : public VideoCodingModule {
   }
 
   int32_t AddVideoFrame(const VideoFrame& videoFrame,
-                        const VideoContentMetrics* contentMetrics,
                         const CodecSpecificInfo* codecSpecificInfo) override {
-    return sender_.AddVideoFrame(videoFrame, contentMetrics, codecSpecificInfo);
+    return sender_.AddVideoFrame(videoFrame, codecSpecificInfo);
   }
 
-  int32_t IntraFrameRequest(int stream_index) override {
+  int32_t IntraFrameRequest(size_t stream_index) override {
     return sender_.IntraFrameRequest(stream_index);
   }
 
@@ -212,11 +197,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
     return receiver_.RegisterPacketRequestCallback(callback);
   }
 
-  int RegisterRenderBufferSizeCallback(
-      VCMRenderBufferSizeCallback* callback) override {
-    return receiver_.RegisterRenderBufferSizeCallback(callback);
-  }
-
   int32_t Decode(uint16_t maxWaitTimeMs) override {
     return receiver_.Decode(maxWaitTimeMs);
   }
@@ -273,10 +253,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
     return receiver_.SetReceiveChannelParameters(rtt);
   }
 
-  void RegisterPreDecodeImageCallback(EncodedImageCallback* observer) override {
-    receiver_.RegisterPreDecodeImageCallback(observer);
-  }
-
   void RegisterPostEncodeImageCallback(
       EncodedImageCallback* observer) override {
     post_encode_callback_.Register(observer);
@@ -288,7 +264,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
   EncodedImageCallbackWrapper post_encode_callback_;
   vcm::VideoSender sender_;
   vcm::VideoReceiver receiver_;
-  std::unique_ptr<EventFactory> own_event_factory_;
 };
 }  // namespace
 
@@ -305,7 +280,8 @@ VideoCodingModule* VideoCodingModule::Create(
   return VideoCodingModule::Create(clock, encoder_rate_observer,
                                    qm_settings_callback,
                                    nullptr,   // NackSender
-                                   nullptr);  // KeyframeRequestSender
+                                   nullptr,   // KeyframeRequestSender
+                                   nullptr);  // Pre-decode image callback
 }
 
 // Create method for the new jitter buffer.
@@ -314,11 +290,11 @@ VideoCodingModule* VideoCodingModule::Create(
     VideoEncoderRateObserver* encoder_rate_observer,
     VCMQMSettingsCallback* qm_settings_callback,
     NackSender* nack_sender,
-    KeyFrameRequestSender* keyframe_request_sender) {
-  return new VideoCodingModuleImpl(clock, new EventFactoryImpl, true,
-                                   encoder_rate_observer, qm_settings_callback,
-                                   nack_sender,
-                                   keyframe_request_sender);
+    KeyFrameRequestSender* keyframe_request_sender,
+    EncodedImageCallback* pre_decode_image_callback) {
+  return new VideoCodingModuleImpl(clock, nullptr, encoder_rate_observer,
+                                   nack_sender, keyframe_request_sender,
+                                   pre_decode_image_callback);
 }
 
 // Create method for current interface, will be removed when the
@@ -338,9 +314,8 @@ VideoCodingModule* VideoCodingModule::Create(
     KeyFrameRequestSender* keyframe_request_sender) {
   assert(clock);
   assert(event_factory);
-  return new VideoCodingModuleImpl(clock, event_factory, false, nullptr,
-                                   nullptr, nack_sender,
-                                   keyframe_request_sender);
+  return new VideoCodingModuleImpl(clock, event_factory, nullptr, nack_sender,
+                                   keyframe_request_sender, nullptr);
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.h b/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.h
index f5f9b00206c..c9992b7f9ce 100644
--- a/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.h
+++ b/chromium/third_party/webrtc/modules/video_coding/video_coding_impl.h
@@ -14,10 +14,13 @@
 #include "webrtc/modules/video_coding/include/video_coding.h"
 
 #include <memory>
+#include <string>
 #include <vector>
 
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/video_coding/codec_database.h"
 #include "webrtc/modules/video_coding/frame_buffer.h"
 #include "webrtc/modules/video_coding/generic_decoder.h"
@@ -28,7 +31,6 @@
 #include "webrtc/modules/video_coding/timing.h"
 #include "webrtc/modules/video_coding/utility/qp_parser.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -50,14 +52,14 @@ class VCMProcessTimer {
   int64_t _latestMs;
 };
 
-class VideoSender {
+class VideoSender : public Module {
  public:
   typedef VideoCodingModule::SenderNackMode SenderNackMode;
 
   VideoSender(Clock* clock,
               EncodedImageCallback* post_encode_callback,
               VideoEncoderRateObserver* encoder_rate_observer,
-              VCMQMSettingsCallback* qm_settings_callback);
+              VCMSendStatisticsCallback* send_stats_callback);
 
   ~VideoSender();
 
@@ -78,23 +80,20 @@ class VideoSender {
                                uint8_t lossRate,
                                int64_t rtt);
 
-  int32_t RegisterTransportCallback(VCMPacketizationCallback* transport);
-  int32_t RegisterSendStatisticsCallback(VCMSendStatisticsCallback* sendStats);
   int32_t RegisterProtectionCallback(VCMProtectionCallback* protection);
   void SetVideoProtection(VCMVideoProtection videoProtection);
 
   int32_t AddVideoFrame(const VideoFrame& videoFrame,
-                        const VideoContentMetrics* _contentMetrics,
                         const CodecSpecificInfo* codecSpecificInfo);
 
-  int32_t IntraFrameRequest(int stream_index);
+  int32_t IntraFrameRequest(size_t stream_index);
   int32_t EnableFrameDropper(bool enable);
 
   void SuspendBelowMinBitrate();
   bool VideoSuspended() const;
 
-  int64_t TimeUntilNextProcess();
-  void Process();
+  int64_t TimeUntilNextProcess() override;
+  void Process() override;
 
  private:
   void SetEncoderParameters(EncoderParameters params)
@@ -102,12 +101,11 @@ class VideoSender {
 
   Clock* const clock_;
 
-  std::unique_ptr<CriticalSectionWrapper> process_crit_sect_;
   rtc::CriticalSection encoder_crit_;
   VCMGenericEncoder* _encoder;
-  VCMEncodedFrameCallback _encodedFrameCallback GUARDED_BY(encoder_crit_);
   media_optimization::MediaOptimization _mediaOpt;
-  VCMSendStatisticsCallback* _sendStatsCallback GUARDED_BY(process_crit_sect_);
+  VCMEncodedFrameCallback _encodedFrameCallback GUARDED_BY(encoder_crit_);
+  VCMSendStatisticsCallback* const send_stats_callback_;
   VCMCodecDataBase _codecDataBase GUARDED_BY(encoder_crit_);
   bool frame_dropper_enabled_ GUARDED_BY(encoder_crit_);
   VCMProcessTimer _sendStatsTimer;
@@ -116,21 +114,22 @@ class VideoSender {
   VideoCodec current_codec_;
   rtc::ThreadChecker main_thread_;
 
-  VCMQMSettingsCallback* const qm_settings_callback_;
   VCMProtectionCallback* protection_callback_;
 
   rtc::CriticalSection params_crit_;
   EncoderParameters encoder_params_ GUARDED_BY(params_crit_);
   bool encoder_has_internal_source_ GUARDED_BY(params_crit_);
+  std::string encoder_name_ GUARDED_BY(params_crit_);
   std::vector<FrameType> next_frame_types_ GUARDED_BY(params_crit_);
 };
 
-class VideoReceiver {
+class VideoReceiver : public Module {
  public:
   typedef VideoCodingModule::ReceiverRobustness ReceiverRobustness;
 
   VideoReceiver(Clock* clock,
                 EventFactory* event_factory,
+                EncodedImageCallback* pre_decode_image_callback,
                 NackSender* nack_sender = nullptr,
                 KeyFrameRequestSender* keyframe_request_sender = nullptr);
   ~VideoReceiver();
@@ -148,7 +147,6 @@ class VideoReceiver {
       VCMDecoderTimingCallback* decoderTiming);
   int32_t RegisterFrameTypeCallback(VCMFrameTypeCallback* frameTypeCallback);
   int32_t RegisterPacketRequestCallback(VCMPacketRequestCallback* callback);
-  int RegisterRenderBufferSizeCallback(VCMRenderBufferSizeCallback* callback);
 
   int32_t Decode(uint16_t maxWaitTimeMs);
 
@@ -175,50 +173,43 @@ class VideoReceiver {
   int32_t SetReceiveChannelParameters(int64_t rtt);
   int32_t SetVideoProtection(VCMVideoProtection videoProtection, bool enable);
 
-  int64_t TimeUntilNextProcess();
-  void Process();
+  int64_t TimeUntilNextProcess() override;
+  void Process() override;
 
-  void RegisterPreDecodeImageCallback(EncodedImageCallback* observer);
   void TriggerDecoderShutdown();
 
  protected:
   int32_t Decode(const webrtc::VCMEncodedFrame& frame)
-      EXCLUSIVE_LOCKS_REQUIRED(_receiveCritSect);
+      EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
   int32_t RequestKeyFrame();
   int32_t RequestSliceLossIndication(const uint64_t pictureID) const;
 
  private:
   Clock* const clock_;
-  std::unique_ptr<CriticalSectionWrapper> process_crit_sect_;
-  CriticalSectionWrapper* _receiveCritSect;
+  rtc::CriticalSection process_crit_;
+  rtc::CriticalSection receive_crit_;
   VCMTiming _timing;
   VCMReceiver _receiver;
   VCMDecodedFrameCallback _decodedFrameCallback;
-  VCMFrameTypeCallback* _frameTypeCallback GUARDED_BY(process_crit_sect_);
-  VCMReceiveStatisticsCallback* _receiveStatsCallback
-      GUARDED_BY(process_crit_sect_);
-  VCMDecoderTimingCallback* _decoderTimingCallback
-      GUARDED_BY(process_crit_sect_);
-  VCMPacketRequestCallback* _packetRequestCallback
-      GUARDED_BY(process_crit_sect_);
-  VCMRenderBufferSizeCallback* render_buffer_callback_
-      GUARDED_BY(process_crit_sect_);
+  VCMFrameTypeCallback* _frameTypeCallback GUARDED_BY(process_crit_);
+  VCMReceiveStatisticsCallback* _receiveStatsCallback GUARDED_BY(process_crit_);
+  VCMDecoderTimingCallback* _decoderTimingCallback GUARDED_BY(process_crit_);
+  VCMPacketRequestCallback* _packetRequestCallback GUARDED_BY(process_crit_);
   VCMGenericDecoder* _decoder;
-#ifdef DEBUG_DECODER_BIT_STREAM
-  FILE* _bitStreamBeforeDecoder;
-#endif
+
   VCMFrameBuffer _frameFromFile;
-  bool _scheduleKeyRequest GUARDED_BY(process_crit_sect_);
-  bool drop_frames_until_keyframe_ GUARDED_BY(process_crit_sect_);
-  size_t max_nack_list_size_ GUARDED_BY(process_crit_sect_);
+  bool _scheduleKeyRequest GUARDED_BY(process_crit_);
+  bool drop_frames_until_keyframe_ GUARDED_BY(process_crit_);
+  size_t max_nack_list_size_ GUARDED_BY(process_crit_);
 
-  VCMCodecDataBase _codecDataBase GUARDED_BY(_receiveCritSect);
-  EncodedImageCallback* pre_decode_image_callback_ GUARDED_BY(_receiveCritSect);
+  VCMCodecDataBase _codecDataBase GUARDED_BY(receive_crit_);
+  EncodedImageCallback* pre_decode_image_callback_;
 
   VCMProcessTimer _receiveStatsTimer;
   VCMProcessTimer _retransmissionTimer;
   VCMProcessTimer _keyRequestTimer;
   QpParser qp_parser_;
+  ThreadUnsafeOneTimeEvent first_frame_received_;
 };
 
 }  // namespace vcm
diff --git a/chromium/third_party/webrtc/modules/video_coding/video_receiver.cc b/chromium/third_party/webrtc/modules/video_coding/video_receiver.cc
index 5aadcf91e0e..a832e2180b9 100644
--- a/chromium/third_party/webrtc/modules/video_coding/video_receiver.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/video_receiver.cc
@@ -20,18 +20,15 @@
 #include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
-// #define DEBUG_DECODER_BIT_STREAM
-
 namespace webrtc {
 namespace vcm {
 
 VideoReceiver::VideoReceiver(Clock* clock,
                              EventFactory* event_factory,
+                             EncodedImageCallback* pre_decode_image_callback,
                              NackSender* nack_sender,
                              KeyFrameRequestSender* keyframe_request_sender)
     : clock_(clock),
-      process_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      _receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()),
       _timing(clock_),
       _receiver(&_timing,
                 clock_,
@@ -39,50 +36,36 @@ VideoReceiver::VideoReceiver(Clock* clock,
                 nack_sender,
                 keyframe_request_sender),
       _decodedFrameCallback(&_timing, clock_),
-      _frameTypeCallback(NULL),
-      _receiveStatsCallback(NULL),
-      _decoderTimingCallback(NULL),
-      _packetRequestCallback(NULL),
-      render_buffer_callback_(NULL),
-      _decoder(NULL),
-#ifdef DEBUG_DECODER_BIT_STREAM
-      _bitStreamBeforeDecoder(NULL),
-#endif
+      _frameTypeCallback(nullptr),
+      _receiveStatsCallback(nullptr),
+      _decoderTimingCallback(nullptr),
+      _packetRequestCallback(nullptr),
+      _decoder(nullptr),
       _frameFromFile(),
       _scheduleKeyRequest(false),
       drop_frames_until_keyframe_(false),
       max_nack_list_size_(0),
       _codecDataBase(nullptr, nullptr),
-      pre_decode_image_callback_(NULL),
+      pre_decode_image_callback_(pre_decode_image_callback),
       _receiveStatsTimer(1000, clock_),
       _retransmissionTimer(10, clock_),
-      _keyRequestTimer(500, clock_) {
-  assert(clock_);
-#ifdef DEBUG_DECODER_BIT_STREAM
-  _bitStreamBeforeDecoder = fopen("decoderBitStream.bit", "wb");
-#endif
-}
+      _keyRequestTimer(500, clock_) {}
 
-VideoReceiver::~VideoReceiver() {
-  delete _receiveCritSect;
-#ifdef DEBUG_DECODER_BIT_STREAM
-  fclose(_bitStreamBeforeDecoder);
-#endif
-}
+VideoReceiver::~VideoReceiver() {}
 
 void VideoReceiver::Process() {
   // Receive-side statistics
   if (_receiveStatsTimer.TimeUntilProcess() == 0) {
     _receiveStatsTimer.Processed();
-    CriticalSectionScoped cs(process_crit_sect_.get());
-    if (_receiveStatsCallback != NULL) {
+    rtc::CritScope cs(&process_crit_);
+    if (_receiveStatsCallback != nullptr) {
       uint32_t bitRate;
       uint32_t frameRate;
       _receiver.ReceiveStatistics(&bitRate, &frameRate);
       _receiveStatsCallback->OnReceiveRatesUpdated(bitRate, frameRate);
     }
 
-    if (_decoderTimingCallback != NULL) {
+    if (_decoderTimingCallback != nullptr) {
       int decode_ms;
       int max_decode_ms;
       int current_delay_ms;
@@ -97,12 +80,6 @@ void VideoReceiver::Process() {
           decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
           jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
     }
-
-    // Size of render buffer.
-    if (render_buffer_callback_) {
-      int buffer_size_ms = _receiver.RenderBufferSizeMs();
-      render_buffer_callback_->RenderBufferSizeMs(buffer_size_ms);
-    }
   }
 
   // Key frame requests
@@ -110,8 +87,8 @@ void VideoReceiver::Process() {
     _keyRequestTimer.Processed();
     bool request_key_frame = false;
     {
-      CriticalSectionScoped cs(process_crit_sect_.get());
-      request_key_frame = _scheduleKeyRequest && _frameTypeCallback != NULL;
+      rtc::CritScope cs(&process_crit_);
+      request_key_frame = _scheduleKeyRequest && _frameTypeCallback != nullptr;
     }
     if (request_key_frame)
       RequestKeyFrame();
@@ -129,9 +106,9 @@ void VideoReceiver::Process() {
     bool callback_registered = false;
     uint16_t length;
     {
-      CriticalSectionScoped cs(process_crit_sect_.get());
+      rtc::CritScope cs(&process_crit_);
       length = max_nack_list_size_;
-      callback_registered = _packetRequestCallback != NULL;
+      callback_registered = _packetRequestCallback != nullptr;
     }
     if (callback_registered && length > 0) {
       // Collect sequence numbers from the default receiver.
@@ -142,8 +119,8 @@ void VideoReceiver::Process() {
         ret = RequestKeyFrame();
       }
       if (ret == VCM_OK && !nackList.empty()) {
-        CriticalSectionScoped cs(process_crit_sect_.get());
-        if (_packetRequestCallback != NULL) {
+        rtc::CritScope cs(&process_crit_);
+        if (_packetRequestCallback != nullptr) {
           _packetRequestCallback->ResendPackets(&nackList[0], nackList.size());
         }
       }
@@ -168,7 +145,7 @@ int64_t VideoReceiver::TimeUntilNextProcess() {
 }
 
 int32_t VideoReceiver::SetReceiveChannelParameters(int64_t rtt) {
-  CriticalSectionScoped receiveCs(_receiveCritSect);
+  rtc::CritScope cs(&receive_crit_);
   _receiver.UpdateRtt(rtt);
   return 0;
 }
@@ -189,9 +166,11 @@ int32_t VideoReceiver::SetVideoProtection(VCMVideoProtection videoProtection,
     }
 
     case kProtectionNackFEC: {
-      CriticalSectionScoped cs(_receiveCritSect);
+      rtc::CritScope cs(&receive_crit_);
       RTC_DCHECK(enable);
-      _receiver.SetNackMode(kNack, media_optimization::kLowRttNackMs, -1);
+      _receiver.SetNackMode(kNack,
+                            media_optimization::kLowRttNackMs,
+                            media_optimization::kMaxRttDelayThreshold);
       _receiver.SetDecodeErrorMode(kNoErrors);
       break;
     }
@@ -210,14 +189,14 @@ int32_t VideoReceiver::SetVideoProtection(VCMVideoProtection videoProtection,
 // ready for rendering.
 int32_t VideoReceiver::RegisterReceiveCallback(
     VCMReceiveCallback* receiveCallback) {
-  CriticalSectionScoped cs(_receiveCritSect);
+  rtc::CritScope cs(&receive_crit_);
   _decodedFrameCallback.SetUserReceiveCallback(receiveCallback);
   return VCM_OK;
 }
 
 int32_t VideoReceiver::RegisterReceiveStatisticsCallback(
     VCMReceiveStatisticsCallback* receiveStats) {
-  CriticalSectionScoped cs(process_crit_sect_.get());
+  rtc::CritScope cs(&process_crit_);
   _receiver.RegisterStatsCallback(receiveStats);
   _receiveStatsCallback = receiveStats;
   return VCM_OK;
@@ -225,7 +204,7 @@ int32_t VideoReceiver::RegisterReceiveStatisticsCallback(
 
 int32_t VideoReceiver::RegisterDecoderTimingCallback(
     VCMDecoderTimingCallback* decoderTiming) {
-  CriticalSectionScoped cs(process_crit_sect_.get());
+  rtc::CritScope cs(&process_crit_);
   _decoderTimingCallback = decoderTiming;
   return VCM_OK;
 }
@@ -233,10 +212,10 @@ int32_t VideoReceiver::RegisterDecoderTimingCallback(
 // Register an externally defined decoder object.
 void VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
                                             uint8_t payloadType) {
-  CriticalSectionScoped cs(_receiveCritSect);
-  if (externalDecoder == NULL) {
+  rtc::CritScope cs(&receive_crit_);
+  if (externalDecoder == nullptr) {
     // Make sure the VCM updates the decoder next time it decodes.
-    _decoder = NULL;
+    _decoder = nullptr;
     RTC_CHECK(_codecDataBase.DeregisterExternalDecoder(payloadType));
     return;
   }
@@ -246,25 +225,18 @@ void VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
 // Register a frame type request callback.
 int32_t VideoReceiver::RegisterFrameTypeCallback(
     VCMFrameTypeCallback* frameTypeCallback) {
-  CriticalSectionScoped cs(process_crit_sect_.get());
+  rtc::CritScope cs(&process_crit_);
   _frameTypeCallback = frameTypeCallback;
   return VCM_OK;
 }
 
 int32_t VideoReceiver::RegisterPacketRequestCallback(
     VCMPacketRequestCallback* callback) {
-  CriticalSectionScoped cs(process_crit_sect_.get());
+  rtc::CritScope cs(&process_crit_);
   _packetRequestCallback = callback;
   return VCM_OK;
 }
 
-int VideoReceiver::RegisterRenderBufferSizeCallback(
-    VCMRenderBufferSizeCallback* callback) {
-  CriticalSectionScoped cs(process_crit_sect_.get());
-  render_buffer_callback_ = callback;
-  return VCM_OK;
-}
-
 void VideoReceiver::TriggerDecoderShutdown() {
   _receiver.TriggerDecoderShutdown();
 }
@@ -275,7 +247,7 @@ int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
   int64_t nextRenderTimeMs;
   bool prefer_late_decoding = false;
   {
-    CriticalSectionScoped cs(_receiveCritSect);
+    rtc::CritScope cs(&receive_crit_);
     prefer_late_decoding = _codecDataBase.PrefersLateDecoding();
   }
 
@@ -286,7 +258,7 @@ int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
     return VCM_FRAME_NOT_READY;
 
   {
-    CriticalSectionScoped cs(process_crit_sect_.get());
+    rtc::CritScope cs(&process_crit_);
     if (drop_frames_until_keyframe_) {
       // Still getting delta frames, schedule another keyframe request as if
       // decode failed.
@@ -298,11 +270,6 @@ int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
       drop_frames_until_keyframe_ = false;
     }
   }
-  CriticalSectionScoped cs(_receiveCritSect);
-
-  // If this frame was too late, we should adjust the delay accordingly
-  _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
-                             clock_->TimeInMilliseconds());
 
   if (pre_decode_image_callback_) {
     EncodedImage encoded_image(frame->EncodedImage());
@@ -311,18 +278,20 @@ int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
       encoded_image.qp_ = qp;
     }
     pre_decode_image_callback_->Encoded(encoded_image, frame->CodecSpecific(),
-                                        NULL);
+                                        nullptr);
   }
 
-#ifdef DEBUG_DECODER_BIT_STREAM
-  if (_bitStreamBeforeDecoder != NULL) {
-    // Write bit stream to file for debugging purposes
-    if (fwrite(frame->Buffer(), 1, frame->Length(), _bitStreamBeforeDecoder) !=
-        frame->Length()) {
-      return -1;
-    }
+  rtc::CritScope cs(&receive_crit_);
+  // If this frame was too late, we should adjust the delay accordingly
+  _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
+                             clock_->TimeInMilliseconds());
+
+  if (first_frame_received_()) {
+    LOG(LS_INFO) << "Received first "
+                 << (frame->Complete() ? "complete" : "incomplete")
+                 << " decodable video frame";
   }
-#endif
+
   const int32_t ret = Decode(*frame);
   _receiver.ReleaseFrame(frame);
   return ret;
@@ -331,8 +300,8 @@ int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
 int32_t VideoReceiver::RequestSliceLossIndication(
     const uint64_t pictureID) const {
   TRACE_EVENT1("webrtc", "RequestSLI", "picture_id", pictureID);
-  CriticalSectionScoped cs(process_crit_sect_.get());
-  if (_frameTypeCallback != NULL) {
+  rtc::CritScope cs(&process_crit_);
+  if (_frameTypeCallback != nullptr) {
     const int32_t ret =
         _frameTypeCallback->SliceLossIndicationRequest(pictureID);
     if (ret < 0) {
@@ -346,8 +315,8 @@ int32_t VideoReceiver::RequestSliceLossIndication(
 
 int32_t VideoReceiver::RequestKeyFrame() {
   TRACE_EVENT0("webrtc", "RequestKeyFrame");
-  CriticalSectionScoped process_cs(process_crit_sect_.get());
-  if (_frameTypeCallback != NULL) {
+  rtc::CritScope cs(&process_crit_);
+  if (_frameTypeCallback != nullptr) {
     const int32_t ret = _frameTypeCallback->RequestKeyFrame();
     if (ret < 0) {
       return ret;
@@ -365,7 +334,7 @@ int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
                           "type", frame.FrameType());
   // Change decoder if payload type has changed
   _decoder = _codecDataBase.GetDecoder(frame, &_decodedFrameCallback);
-  if (_decoder == NULL) {
+  if (_decoder == nullptr) {
     return VCM_NO_CODEC_REGISTERED;
   }
   // Decode a frame
@@ -389,7 +358,7 @@ int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
     ret = VCM_OK;
   }
   if (request_key_frame) {
-    CriticalSectionScoped cs(process_crit_sect_.get());
+    rtc::CritScope cs(&process_crit_);
     _scheduleKeyRequest = true;
   }
   TRACE_EVENT_ASYNC_END0("webrtc", "Video", frame.TimeStamp());
@@ -400,8 +369,8 @@ int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
 int32_t VideoReceiver::RegisterReceiveCodec(const VideoCodec* receiveCodec,
                                             int32_t numberOfCores,
                                             bool requireKeyFrame) {
-  CriticalSectionScoped cs(_receiveCritSect);
-  if (receiveCodec == NULL) {
+  rtc::CritScope cs(&receive_crit_);
+  if (receiveCodec == nullptr) {
     return VCM_PARAMETER_ERROR;
   }
   if (!_codecDataBase.RegisterReceiveCodec(receiveCodec, numberOfCores,
@@ -413,8 +382,8 @@ int32_t VideoReceiver::RegisterReceiveCodec(const VideoCodec* receiveCodec,
 
 // Get current received codec
 int32_t VideoReceiver::ReceiveCodec(VideoCodec* currentReceiveCodec) const {
-  CriticalSectionScoped cs(_receiveCritSect);
-  if (currentReceiveCodec == NULL) {
+  rtc::CritScope cs(&receive_crit_);
+  if (currentReceiveCodec == nullptr) {
     return VCM_PARAMETER_ERROR;
   }
   return _codecDataBase.ReceiveCodec(currentReceiveCodec) ? 0 : -1;
@@ -422,7 +391,7 @@ int32_t VideoReceiver::ReceiveCodec(VideoCodec* currentReceiveCodec) const {
 
 // Get current received codec
 VideoCodecType VideoReceiver::ReceiveCodec() const {
-  CriticalSectionScoped cs(_receiveCritSect);
+  rtc::CritScope cs(&receive_crit_);
   return _codecDataBase.ReceiveCodec();
 }
 
@@ -434,7 +403,7 @@ int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload,
     TRACE_EVENT1("webrtc", "VCM::PacketKeyFrame", "seqnum",
                  rtpInfo.header.sequenceNumber);
   }
-  if (incomingPayload == NULL) {
+  if (incomingPayload == nullptr) {
     // The jitter buffer doesn't handle non-zero payload lengths for packets
     // without payload.
     // TODO(holmer): We should fix this in the jitter buffer.
@@ -443,11 +412,12 @@ int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload,
   const VCMPacket packet(incomingPayload, payloadLength, rtpInfo);
   int32_t ret = _receiver.InsertPacket(packet, rtpInfo.type.Video.width,
                                        rtpInfo.type.Video.height);
+
   // TODO(holmer): Investigate if this somehow should use the key frame
   // request scheduling to throttle the requests.
   if (ret == VCM_FLUSH_INDICATOR) {
     {
-      CriticalSectionScoped process_cs(process_crit_sect_.get());
+      rtc::CritScope cs(&process_crit_);
       drop_frames_until_keyframe_ = true;
     }
     RequestKeyFrame();
@@ -484,7 +454,7 @@ uint32_t VideoReceiver::DiscardedPackets() const {
 int VideoReceiver::SetReceiverRobustnessMode(
     ReceiverRobustness robustnessMode,
     VCMDecodeErrorMode decode_error_mode) {
-  CriticalSectionScoped cs(_receiveCritSect);
+  rtc::CritScope cs(&receive_crit_);
   switch (robustnessMode) {
     case VideoCodingModule::kNone:
       _receiver.SetNackMode(kNoNack, -1, -1);
@@ -520,7 +490,7 @@ int VideoReceiver::SetReceiverRobustnessMode(
 }
 
 void VideoReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
-  CriticalSectionScoped cs(_receiveCritSect);
+  rtc::CritScope cs(&receive_crit_);
   _receiver.SetDecodeErrorMode(decode_error_mode);
 }
 
@@ -528,7 +498,7 @@ void VideoReceiver::SetNackSettings(size_t max_nack_list_size,
                                     int max_packet_age_to_nack,
                                     int max_incomplete_time_ms) {
   if (max_nack_list_size != 0) {
-    CriticalSectionScoped process_cs(process_crit_sect_.get());
+    rtc::CritScope cs(&process_crit_);
     max_nack_list_size_ = max_nack_list_size;
   }
   _receiver.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
@@ -539,11 +509,5 @@ int VideoReceiver::SetMinReceiverDelay(int desired_delay_ms) {
   return _receiver.SetMinReceiverDelay(desired_delay_ms);
 }
 
-void VideoReceiver::RegisterPreDecodeImageCallback(
-    EncodedImageCallback* observer) {
-  CriticalSectionScoped cs(_receiveCritSect);
-  pre_decode_image_callback_ = observer;
-}
-
 }  // namespace vcm
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_coding/video_receiver_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/video_receiver_unittest.cc
index 05656a5c254..3c414053bc2 100644
--- a/chromium/third_party/webrtc/modules/video_coding/video_receiver_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/video_receiver_unittest.cc
@@ -33,7 +33,7 @@ class TestVideoReceiver : public ::testing::Test {
   TestVideoReceiver() : clock_(0) {}
 
   virtual void SetUp() {
-    receiver_.reset(new VideoReceiver(&clock_, &event_factory_));
+    receiver_.reset(new VideoReceiver(&clock_, &event_factory_, nullptr));
     receiver_->RegisterExternalDecoder(&decoder_, kUnusedPayloadType);
     const size_t kMaxNackListSize = 250;
     const int kMaxPacketAgeToNack = 450;
diff --git a/chromium/third_party/webrtc/modules/video_coding/video_sender.cc b/chromium/third_party/webrtc/modules/video_coding/video_sender.cc
index 49690697889..f52b1c56e7b 100644
--- a/chromium/third_party/webrtc/modules/video_coding/video_sender.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/video_sender.cc
@@ -27,27 +27,24 @@ namespace vcm {
 VideoSender::VideoSender(Clock* clock,
                          EncodedImageCallback* post_encode_callback,
                          VideoEncoderRateObserver* encoder_rate_observer,
-                         VCMQMSettingsCallback* qm_settings_callback)
+                         VCMSendStatisticsCallback* send_stats_callback)
     : clock_(clock),
-      process_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       _encoder(nullptr),
-      _encodedFrameCallback(post_encode_callback),
       _mediaOpt(clock_),
-      _sendStatsCallback(nullptr),
+      _encodedFrameCallback(post_encode_callback, &_mediaOpt),
+      send_stats_callback_(send_stats_callback),
       _codecDataBase(encoder_rate_observer, &_encodedFrameCallback),
       frame_dropper_enabled_(true),
       _sendStatsTimer(1000, clock_),
       current_codec_(),
-      qm_settings_callback_(qm_settings_callback),
       protection_callback_(nullptr),
       encoder_params_({0, 0, 0, 0}),
       encoder_has_internal_source_(false),
       next_frame_types_(1, kVideoFrameDelta) {
+  _mediaOpt.Reset();
   // Allow VideoSender to be created on one thread but used on another, post
   // construction. This is currently how this class is being used by at least
   // one external project (diffractor).
-  _mediaOpt.EnableQM(qm_settings_callback_ != nullptr);
-  _mediaOpt.Reset();
   main_thread_.DetachFromThread();
 }
 
@@ -55,12 +52,19 @@ VideoSender::~VideoSender() {}
 
 void VideoSender::Process() {
   if (_sendStatsTimer.TimeUntilProcess() == 0) {
+    // |_sendStatsTimer.Processed()| must be called. Otherwise
+    // VideoSender::Process() will be called in an infinite loop.
     _sendStatsTimer.Processed();
-    CriticalSectionScoped cs(process_crit_sect_.get());
-    if (_sendStatsCallback != nullptr) {
+    if (send_stats_callback_) {
       uint32_t bitRate = _mediaOpt.SentBitRate();
       uint32_t frameRate = _mediaOpt.SentFrameRate();
-      _sendStatsCallback->SendStatistics(bitRate, frameRate);
+      std::string encoder_name;
+      {
+        rtc::CritScope cs(&params_crit_);
+        // Copy the string here so that we don't hold |params_crit_| in the CB.
+        encoder_name = encoder_name_;
+      }
+      send_stats_callback_->SendStatistics(bitRate, frameRate, encoder_name);
     }
   }
 
@@ -196,19 +200,38 @@ int VideoSender::FrameRate(unsigned int* framerate) const {
 int32_t VideoSender::SetChannelParameters(uint32_t target_bitrate,
                                           uint8_t lossRate,
                                           int64_t rtt) {
-  uint32_t target_rate =
-      _mediaOpt.SetTargetRates(target_bitrate, lossRate, rtt,
-                               protection_callback_, qm_settings_callback_);
+  uint32_t target_rate = _mediaOpt.SetTargetRates(target_bitrate, lossRate, rtt,
+                                                  protection_callback_);
 
   uint32_t input_frame_rate = _mediaOpt.InputFrameRate();
 
-  rtc::CritScope cs(&params_crit_);
-  encoder_params_ = {target_rate, lossRate, rtt, input_frame_rate};
+  EncoderParameters encoder_params = {target_rate, lossRate, rtt,
+                                      input_frame_rate};
+  bool encoder_has_internal_source;
+  {
+    rtc::CritScope cs(&params_crit_);
+    encoder_params_ = encoder_params;
+    encoder_has_internal_source = encoder_has_internal_source_;
+  }
+
+  // For encoders with internal sources, we need to tell the encoder directly,
+  // instead of waiting for an AddVideoFrame that will never come (internal
+  // source encoders don't get input frames).
+  if (encoder_has_internal_source) {
+    rtc::CritScope cs(&encoder_crit_);
+    if (_encoder) {
+      SetEncoderParameters(encoder_params);
+    }
+  }
 
   return VCM_OK;
 }
 
 void VideoSender::SetEncoderParameters(EncoderParameters params) {
+  // |target_bitrate == 0 | means that the network is down or the send pacer is
+  // full.
+  // TODO(perkj): Consider setting |target_bitrate| == 0 to the encoders.
+  // Especially if |encoder_has_internal_source_ | == true.
   if (params.target_bitrate == 0)
     return;
 
@@ -220,24 +243,6 @@ void VideoSender::SetEncoderParameters(EncoderParameters params) {
     _encoder->SetEncoderParameters(params);
 }
 
-int32_t VideoSender::RegisterTransportCallback(
-    VCMPacketizationCallback* transport) {
-  rtc::CritScope lock(&encoder_crit_);
-  _encodedFrameCallback.SetMediaOpt(&_mediaOpt);
-  _encodedFrameCallback.SetTransportCallback(transport);
-  return VCM_OK;
-}
-
-// Register video output information callback which will be called to deliver
-// information about the video stream produced by the encoder, for instance the
-// average frame rate and bit rate.
-int32_t VideoSender::RegisterSendStatisticsCallback(
-    VCMSendStatisticsCallback* sendStats) {
-  CriticalSectionScoped cs(process_crit_sect_.get());
-  _sendStatsCallback = sendStats;
-  return VCM_OK;
-}
-
 // Register a video protection callback which will be called to deliver the
 // requested FEC rate and NACK status (on/off).
 // Note: this callback is assumed to only be registered once and before it is
@@ -269,7 +274,6 @@ void VideoSender::SetVideoProtection(VCMVideoProtection videoProtection) {
 }
 // Add one raw video frame to the encoder, blocking.
 int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
-                                   const VideoContentMetrics* contentMetrics,
                                    const CodecSpecificInfo* codecSpecificInfo) {
   EncoderParameters encoder_params;
   std::vector<FrameType> next_frame_types;
@@ -291,7 +295,6 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
     _encoder->OnDroppedFrame();
     return VCM_OK;
   }
-  _mediaOpt.UpdateContentData(contentMetrics);
   // TODO(pbos): Make sure setting send codec is synchronized with video
   // processing so frame size always matches.
   if (!_codecDataBase.MatchesCurrentResolution(videoFrame.width(),
@@ -300,7 +303,8 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
     return VCM_PARAMETER_ERROR;
   }
   VideoFrame converted_frame = videoFrame;
-  if (converted_frame.native_handle() && !_encoder->SupportsNativeHandle()) {
+  if (converted_frame.video_frame_buffer()->native_handle() &&
+      !_encoder->SupportsNativeHandle()) {
     // This module only supports software encoding.
     // TODO(pbos): Offload conversion from the encoder thread.
     converted_frame = converted_frame.ConvertNativeToI420Frame();
@@ -313,9 +317,12 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
     LOG(LS_ERROR) << "Failed to encode frame. Error code: " << ret;
     return ret;
   }
+
   {
-    // Change all keyframe requests to encode delta frames the next time.
     rtc::CritScope lock(&params_crit_);
+    encoder_name_ = _encoder->ImplementationName();
+
+    // Change all keyframe requests to encode delta frames the next time.
     for (size_t i = 0; i < next_frame_types_.size(); ++i) {
       // Check for equality (same requested as before encoding) to not
       // accidentally drop a keyframe request while encoding.
@@ -323,16 +330,13 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
         next_frame_types_[i] = kVideoFrameDelta;
     }
   }
-  if (qm_settings_callback_)
-    qm_settings_callback_->SetTargetFramerate(_encoder->GetTargetFramerate());
   return VCM_OK;
 }
 
-int32_t VideoSender::IntraFrameRequest(int stream_index) {
+int32_t VideoSender::IntraFrameRequest(size_t stream_index) {
   {
     rtc::CritScope lock(&params_crit_);
-    if (stream_index < 0 ||
-        static_cast<size_t>(stream_index) >= next_frame_types_.size()) {
+    if (stream_index >= next_frame_types_.size()) {
       return -1;
     }
     next_frame_types_[stream_index] = kVideoFrameKey;
@@ -346,7 +350,7 @@ int32_t VideoSender::IntraFrameRequest(int stream_index) {
   // encoder_crit_.
   rtc::CritScope lock(&encoder_crit_);
   rtc::CritScope params_lock(&params_crit_);
-  if (static_cast<size_t>(stream_index) >= next_frame_types_.size())
+  if (stream_index >= next_frame_types_.size())
     return -1;
   if (_encoder != nullptr && _encoder->InternalSource()) {
     // Try to request the frame if we have an external encoder with
diff --git a/chromium/third_party/webrtc/modules/video_coding/video_sender_unittest.cc b/chromium/third_party/webrtc/modules/video_coding/video_sender_unittest.cc
index 3f9ba4eadda..5324ceeb0b8 100644
--- a/chromium/third_party/webrtc/modules/video_coding/video_sender_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_coding/video_sender_unittest.cc
@@ -86,19 +86,19 @@ class EmptyFrameGenerator : public FrameGenerator {
   std::unique_ptr<VideoFrame> frame_;
 };
 
-class PacketizationCallback : public VCMPacketizationCallback {
+class EncodedImageCallbackImpl : public EncodedImageCallback {
  public:
-  explicit PacketizationCallback(Clock* clock)
+  explicit EncodedImageCallbackImpl(Clock* clock)
       : clock_(clock), start_time_ms_(clock_->TimeInMilliseconds()) {}
 
-  virtual ~PacketizationCallback() {}
+  virtual ~EncodedImageCallbackImpl() {}
 
-  int32_t SendData(uint8_t payload_type,
-                   const EncodedImage& encoded_image,
-                   const RTPFragmentationHeader* fragmentation_header,
-                   const RTPVideoHeader* rtp_video_header) override {
-    assert(rtp_video_header);
-    frame_data_.push_back(FrameData(encoded_image._length, *rtp_video_header));
+  int32_t Encoded(const EncodedImage& encoded_image,
+                  const CodecSpecificInfo* codec_specific_info,
+                  const RTPFragmentationHeader* fragmentation) override {
+    assert(codec_specific_info);
+    frame_data_.push_back(
+        FrameData(encoded_image._length, *codec_specific_info));
     return 0;
   }
 
@@ -130,11 +130,12 @@ class PacketizationCallback : public VCMPacketizationCallback {
   struct FrameData {
     FrameData() {}
 
-    FrameData(size_t payload_size, const RTPVideoHeader& rtp_video_header)
-        : payload_size(payload_size), rtp_video_header(rtp_video_header) {}
+    FrameData(size_t payload_size, const CodecSpecificInfo& codec_specific_info)
+        : payload_size(payload_size),
+          codec_specific_info(codec_specific_info) {}
 
     size_t payload_size;
-    RTPVideoHeader rtp_video_header;
+    CodecSpecificInfo codec_specific_info;
   };
 
   int64_t interval_ms() {
@@ -146,9 +147,9 @@ class PacketizationCallback : public VCMPacketizationCallback {
   int CountFramesWithinTemporalLayer(int temporal_layer) {
     int frames = 0;
     for (size_t i = 0; i < frame_data_.size(); ++i) {
-      EXPECT_EQ(kRtpVideoVp8, frame_data_[i].rtp_video_header.codec);
+      EXPECT_EQ(kVideoCodecVP8, frame_data_[i].codec_specific_info.codecType);
       const uint8_t temporal_idx =
-          frame_data_[i].rtp_video_header.codecHeader.VP8.temporalIdx;
+          frame_data_[i].codec_specific_info.codecSpecific.VP8.temporalIdx;
       if (temporal_idx <= temporal_layer || temporal_idx == kNoTemporalIdx)
         frames++;
     }
@@ -158,9 +159,9 @@ class PacketizationCallback : public VCMPacketizationCallback {
   size_t SumPayloadBytesWithinTemporalLayer(int temporal_layer) {
     size_t payload_size = 0;
     for (size_t i = 0; i < frame_data_.size(); ++i) {
-      EXPECT_EQ(kRtpVideoVp8, frame_data_[i].rtp_video_header.codec);
+      EXPECT_EQ(kVideoCodecVP8, frame_data_[i].codec_specific_info.codecType);
       const uint8_t temporal_idx =
-          frame_data_[i].rtp_video_header.codecHeader.VP8.temporalIdx;
+          frame_data_[i].codec_specific_info.codecSpecific.VP8.temporalIdx;
       if (temporal_idx <= temporal_layer || temporal_idx == kNoTemporalIdx)
         payload_size += frame_data_[i].payload_size;
     }
@@ -176,22 +177,20 @@ class TestVideoSender : public ::testing::Test {
  protected:
   // Note: simulated clock starts at 1 seconds, since parts of webrtc use 0 as
   // a special case (e.g. frame rate in media optimization).
-  TestVideoSender() : clock_(1000), packetization_callback_(&clock_) {}
+  TestVideoSender() : clock_(1000), encoded_frame_callback_(&clock_) {}
 
   void SetUp() override {
     sender_.reset(
-        new VideoSender(&clock_, &post_encode_callback_, nullptr, nullptr));
-    EXPECT_EQ(0, sender_->RegisterTransportCallback(&packetization_callback_));
+        new VideoSender(&clock_, &encoded_frame_callback_, nullptr, nullptr));
   }
 
   void AddFrame() {
     assert(generator_.get());
-    sender_->AddVideoFrame(*generator_->NextFrame(), NULL, NULL);
+    sender_->AddVideoFrame(*generator_->NextFrame(), NULL);
   }
 
   SimulatedClock clock_;
-  PacketizationCallback packetization_callback_;
-  MockEncodedImageCallback post_encode_callback_;
+  EncodedImageCallbackImpl encoded_frame_callback_;
   // Used by subclassing tests, need to outlive sender_.
   std::unique_ptr<VideoEncoder> encoder_;
   std::unique_ptr<VideoSender> sender_;
@@ -291,9 +290,17 @@ TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequests) {
   EXPECT_EQ(-1, sender_->IntraFrameRequest(3));
   ExpectIntraRequest(-1);
   AddFrame();
+}
 
-  EXPECT_EQ(-1, sender_->IntraFrameRequest(-1));
-  ExpectIntraRequest(-1);
+TEST_F(TestVideoSenderWithMockEncoder, TestSetRate) {
+  const uint32_t new_bitrate = settings_.startBitrate + 300;
+  EXPECT_CALL(encoder_, SetRates(new_bitrate, _)).Times(1).WillOnce(Return(0));
+  sender_->SetChannelParameters(new_bitrate * 1000, 0, 200);
+  AddFrame();
+
+  // Expect no call to encoder_.SetRates if the new bitrate is zero.
+  EXPECT_CALL(encoder_, SetRates(new_bitrate, _)).Times(0);
+  sender_->SetChannelParameters(0, 0, 200);
   AddFrame();
 }
 
@@ -314,7 +321,19 @@ TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequestsInternalCapture) {
   EXPECT_EQ(0, sender_->IntraFrameRequest(2));
   // No requests expected since these indices are out of bounds.
   EXPECT_EQ(-1, sender_->IntraFrameRequest(3));
-  EXPECT_EQ(-1, sender_->IntraFrameRequest(-1));
+}
+
+TEST_F(TestVideoSenderWithMockEncoder, TestEncoderParametersForInternalSource) {
+  // De-register current external encoder.
+  sender_->RegisterExternalEncoder(nullptr, kUnusedPayloadType, false);
+  // Register encoder with internal capture.
+  sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, true);
+  EXPECT_EQ(0, sender_->RegisterSendCodec(&settings_, 1, 1200));
+  // Update encoder bitrate parameters. We expect that to immediately call
+  // SetRates on the encoder without waiting for AddFrame processing.
+  const uint32_t new_bitrate = settings_.startBitrate + 300;
+  EXPECT_CALL(encoder_, SetRates(new_bitrate, _)).Times(1).WillOnce(Return(0));
+  sender_->SetChannelParameters(new_bitrate * 1000, 0, 200);
 }
 
 TEST_F(TestVideoSenderWithMockEncoder, EncoderFramerateUpdatedViaProcess) {
@@ -402,8 +421,6 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
   void InsertFrames(float framerate, float seconds) {
     for (int i = 0; i < seconds * framerate; ++i) {
       clock_.AdvanceTimeMilliseconds(1000.0f / framerate);
-      EXPECT_CALL(post_encode_callback_, Encoded(_, NULL, NULL))
-          .WillOnce(Return(0));
       AddFrame();
       // SetChannelParameters needs to be called frequently to propagate
       // framerate from the media optimization into the encoder.
@@ -422,10 +439,10 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
     // It appears that this 5 seconds simulation is needed to allow
     // bitrate and framerate to stabilize.
     InsertFrames(framerate, short_simulation_interval);
-    packetization_callback_.Reset();
+    encoded_frame_callback_.Reset();
 
     InsertFrames(framerate, long_simulation_interval);
-    return packetization_callback_.CalculateVp8StreamInfo();
+    return encoded_frame_callback_.CalculateVp8StreamInfo();
   }
 
  protected:
diff --git a/chromium/third_party/webrtc/modules/video_processing/BUILD.gn b/chromium/third_party/webrtc/modules/video_processing/BUILD.gn
index 43a8de12557..1177d9b7f0d 100644
--- a/chromium/third_party/webrtc/modules/video_processing/BUILD.gn
+++ b/chromium/third_party/webrtc/modules/video_processing/BUILD.gn
@@ -13,12 +13,6 @@ build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64"
 
 source_set("video_processing") {
   sources = [
-    "brightness_detection.cc",
-    "brightness_detection.h",
-    "content_analysis.cc",
-    "content_analysis.h",
-    "deflickering.cc",
-    "deflickering.h",
     "frame_preprocessor.cc",
     "frame_preprocessor.h",
     "include/video_processing.h",
@@ -67,7 +61,6 @@ source_set("video_processing") {
 if (build_video_processing_sse2) {
   source_set("video_processing_sse2") {
     sources = [
-      "content_analysis_sse2.cc",
       "util/denoiser_filter_sse2.cc",
       "util/denoiser_filter_sse2.h",
     ]
diff --git a/chromium/third_party/webrtc/modules/video_processing/brightness_detection.cc b/chromium/third_party/webrtc/modules/video_processing/brightness_detection.cc
deleted file mode 100644
index 7455cf97591..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/brightness_detection.cc
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_processing/brightness_detection.h"
-
-#include <math.h>
-
-#include "webrtc/modules/video_processing/include/video_processing.h"
-
-namespace webrtc {
-
-VPMBrightnessDetection::VPMBrightnessDetection() {
-  Reset();
-}
-
-VPMBrightnessDetection::~VPMBrightnessDetection() {}
-
-void VPMBrightnessDetection::Reset() {
-  frame_cnt_bright_ = 0;
-  frame_cnt_dark_ = 0;
-}
-
-int32_t VPMBrightnessDetection::ProcessFrame(
-    const VideoFrame& frame,
-    const VideoProcessing::FrameStats& stats) {
-  if (frame.IsZeroSize()) {
-    return VPM_PARAMETER_ERROR;
-  }
-  int width = frame.width();
-  int height = frame.height();
-
-  if (!VideoProcessing::ValidFrameStats(stats)) {
-    return VPM_PARAMETER_ERROR;
-  }
-
-  const uint8_t frame_cnt_alarm = 2;
-
-  // Get proportion in lowest bins.
-  uint8_t low_th = 20;
-  float prop_low = 0;
-  for (uint32_t i = 0; i < low_th; i++) {
-    prop_low += stats.hist[i];
-  }
-  prop_low /= stats.num_pixels;
-
-  // Get proportion in highest bins.
-  unsigned char high_th = 230;
-  float prop_high = 0;
-  for (uint32_t i = high_th; i < 256; i++) {
-    prop_high += stats.hist[i];
-  }
-  prop_high /= stats.num_pixels;
-
-  if (prop_high < 0.4) {
-    if (stats.mean < 90 || stats.mean > 170) {
-      // Standard deviation of Y
-      const uint8_t* buffer = frame.buffer(kYPlane);
-      float std_y = 0;
-      for (int h = 0; h < height; h += (1 << stats.sub_sampling_factor)) {
-        int row = h * width;
-        for (int w = 0; w < width; w += (1 << stats.sub_sampling_factor)) {
-          std_y +=
-              (buffer[w + row] - stats.mean) * (buffer[w + row] - stats.mean);
-        }
-      }
-      std_y = sqrt(std_y / stats.num_pixels);
-
-      // Get percentiles.
-      uint32_t sum = 0;
-      uint32_t median_y = 140;
-      uint32_t perc05 = 0;
-      uint32_t perc95 = 255;
-      float pos_perc05 = stats.num_pixels * 0.05f;
-      float pos_median = stats.num_pixels * 0.5f;
-      float posPerc95 = stats.num_pixels * 0.95f;
-      for (uint32_t i = 0; i < 256; i++) {
-        sum += stats.hist[i];
-        if (sum < pos_perc05)
-          perc05 = i;  // 5th perc.
-        if (sum < pos_median)
-          median_y = i;  // 50th perc.
-        if (sum < posPerc95)
-          perc95 = i;  // 95th perc.
-        else
-          break;
-      }
-
-      // Check if image is too dark
-      if ((std_y < 55) && (perc05 < 50)) {
-        if (median_y < 60 || stats.mean < 80 || perc95 < 130 ||
-            prop_low > 0.20) {
-          frame_cnt_dark_++;
-        } else {
-          frame_cnt_dark_ = 0;
-        }
-      } else {
-        frame_cnt_dark_ = 0;
-      }
-
-      // Check if image is too bright
-      if ((std_y < 52) && (perc95 > 200) && (median_y > 160)) {
-        if (median_y > 185 || stats.mean > 185 || perc05 > 140 ||
-            prop_high > 0.25) {
-          frame_cnt_bright_++;
-        } else {
-          frame_cnt_bright_ = 0;
-        }
-      } else {
-        frame_cnt_bright_ = 0;
-      }
-    } else {
-      frame_cnt_dark_ = 0;
-      frame_cnt_bright_ = 0;
-    }
-  } else {
-    frame_cnt_bright_++;
-    frame_cnt_dark_ = 0;
-  }
-
-  if (frame_cnt_dark_ > frame_cnt_alarm) {
-    return VideoProcessing::kDarkWarning;
-  } else if (frame_cnt_bright_ > frame_cnt_alarm) {
-    return VideoProcessing::kBrightWarning;
-  } else {
-    return VideoProcessing::kNoWarning;
-  }
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/brightness_detection.h b/chromium/third_party/webrtc/modules/video_processing/brightness_detection.h
deleted file mode 100644
index 78a7ac5e0bf..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/brightness_detection.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
-#define WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
-
-#include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class VPMBrightnessDetection {
- public:
-  VPMBrightnessDetection();
-  ~VPMBrightnessDetection();
-
-  void Reset();
-  int32_t ProcessFrame(const VideoFrame& frame,
-                       const VideoProcessing::FrameStats& stats);
-
- private:
-  uint32_t frame_cnt_bright_;
-  uint32_t frame_cnt_dark_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_BRIGHTNESS_DETECTION_H_
diff --git a/chromium/third_party/webrtc/modules/video_processing/content_analysis.cc b/chromium/third_party/webrtc/modules/video_processing/content_analysis.cc
deleted file mode 100644
index 54c04da4668..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/content_analysis.cc
+++ /dev/null
@@ -1,281 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#include "webrtc/modules/video_processing/content_analysis.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-namespace webrtc {
-
-VPMContentAnalysis::VPMContentAnalysis(bool runtime_cpu_detection)
-    : orig_frame_(NULL),
-      prev_frame_(NULL),
-      width_(0),
-      height_(0),
-      skip_num_(1),
-      border_(8),
-      motion_magnitude_(0.0f),
-      spatial_pred_err_(0.0f),
-      spatial_pred_err_h_(0.0f),
-      spatial_pred_err_v_(0.0f),
-      first_frame_(true),
-      ca_Init_(false),
-      content_metrics_(NULL) {
-  ComputeSpatialMetrics = &VPMContentAnalysis::ComputeSpatialMetrics_C;
-  TemporalDiffMetric = &VPMContentAnalysis::TemporalDiffMetric_C;
-
-  if (runtime_cpu_detection) {
-#if defined(WEBRTC_ARCH_X86_FAMILY)
-    if (WebRtc_GetCPUInfo(kSSE2)) {
-      ComputeSpatialMetrics = &VPMContentAnalysis::ComputeSpatialMetrics_SSE2;
-      TemporalDiffMetric = &VPMContentAnalysis::TemporalDiffMetric_SSE2;
-    }
-#endif
-  }
-  Release();
-}
-
-VPMContentAnalysis::~VPMContentAnalysis() {
-  Release();
-}
-
-VideoContentMetrics* VPMContentAnalysis::ComputeContentMetrics(
-    const VideoFrame& inputFrame) {
-  if (inputFrame.IsZeroSize())
-    return NULL;
-
-  // Init if needed (native dimension change).
-  if (width_ != inputFrame.width() || height_ != inputFrame.height()) {
-    if (VPM_OK != Initialize(inputFrame.width(), inputFrame.height()))
-      return NULL;
-  }
-  // Only interested in the Y plane.
-  orig_frame_ = inputFrame.buffer(kYPlane);
-
-  // Compute spatial metrics: 3 spatial prediction errors.
-  (this->*ComputeSpatialMetrics)();
-
-  // Compute motion metrics
-  if (first_frame_ == false)
-    ComputeMotionMetrics();
-
-  // Saving current frame as previous one: Y only.
-  memcpy(prev_frame_, orig_frame_, width_ * height_);
-
-  first_frame_ = false;
-  ca_Init_ = true;
-
-  return ContentMetrics();
-}
-
-int32_t VPMContentAnalysis::Release() {
-  if (content_metrics_ != NULL) {
-    delete content_metrics_;
-    content_metrics_ = NULL;
-  }
-
-  if (prev_frame_ != NULL) {
-    delete[] prev_frame_;
-    prev_frame_ = NULL;
-  }
-
-  width_ = 0;
-  height_ = 0;
-  first_frame_ = true;
-
-  return VPM_OK;
-}
-
-int32_t VPMContentAnalysis::Initialize(int width, int height) {
-  width_ = width;
-  height_ = height;
-  first_frame_ = true;
-
-  // skip parameter: # of skipped rows: for complexity reduction
-  //  temporal also currently uses it for column reduction.
-  skip_num_ = 1;
-
-  // use skipNum = 2 for 4CIF, WHD
-  if ((height_ >= 576) && (width_ >= 704)) {
-    skip_num_ = 2;
-  }
-  // use skipNum = 4 for FULLL_HD images
-  if ((height_ >= 1080) && (width_ >= 1920)) {
-    skip_num_ = 4;
-  }
-
-  if (content_metrics_ != NULL) {
-    delete content_metrics_;
-  }
-
-  if (prev_frame_ != NULL) {
-    delete[] prev_frame_;
-  }
-
-  // Spatial Metrics don't work on a border of 8. Minimum processing
-  // block size is 16 pixels.  So make sure the width and height support this.
-  if (width_ <= 32 || height_ <= 32) {
-    ca_Init_ = false;
-    return VPM_PARAMETER_ERROR;
-  }
-
-  content_metrics_ = new VideoContentMetrics();
-  if (content_metrics_ == NULL) {
-    return VPM_MEMORY;
-  }
-
-  prev_frame_ = new uint8_t[width_ * height_];  // Y only.
-  if (prev_frame_ == NULL)
-    return VPM_MEMORY;
-
-  return VPM_OK;
-}
-
-// Compute motion metrics: magnitude over non-zero motion vectors,
-//  and size of zero cluster
-int32_t VPMContentAnalysis::ComputeMotionMetrics() {
-  // Motion metrics: only one is derived from normalized
-  //  (MAD) temporal difference
-  (this->*TemporalDiffMetric)();
-  return VPM_OK;
-}
-
-// Normalized temporal difference (MAD): used as a motion level metric
-// Normalize MAD by spatial contrast: images with more contrast
-//  (pixel variance) likely have larger temporal difference
-// To reduce complexity, we compute the metric for a reduced set of points.
-int32_t VPMContentAnalysis::TemporalDiffMetric_C() {
-  // size of original frame
-  int sizei = height_;
-  int sizej = width_;
-  uint32_t tempDiffSum = 0;
-  uint32_t pixelSum = 0;
-  uint64_t pixelSqSum = 0;
-
-  uint32_t num_pixels = 0;  // Counter for # of pixels.
-  const int width_end = ((width_ - 2 * border_) & -16) + border_;
-
-  for (int i = border_; i < sizei - border_; i += skip_num_) {
-    for (int j = border_; j < width_end; j++) {
-      num_pixels += 1;
-      int ssn = i * sizej + j;
-
-      uint8_t currPixel = orig_frame_[ssn];
-      uint8_t prevPixel = prev_frame_[ssn];
-
-      tempDiffSum +=
-          static_cast<uint32_t>(abs((int16_t)(currPixel - prevPixel)));
-      pixelSum += static_cast<uint32_t>(currPixel);
-      pixelSqSum += static_cast<uint64_t>(currPixel * currPixel);
-    }
-  }
-
-  // Default.
-  motion_magnitude_ = 0.0f;
-
-  if (tempDiffSum == 0)
-    return VPM_OK;
-
-  // Normalize over all pixels.
-  float const tempDiffAvg =
-      static_cast<float>(tempDiffSum) / static_cast<float>(num_pixels);
-  float const pixelSumAvg =
-      static_cast<float>(pixelSum) / static_cast<float>(num_pixels);
-  float const pixelSqSumAvg =
-      static_cast<float>(pixelSqSum) / static_cast<float>(num_pixels);
-  float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
-
-  if (contrast > 0.0) {
-    contrast = sqrt(contrast);
-    motion_magnitude_ = tempDiffAvg / contrast;
-  }
-  return VPM_OK;
-}
-
-// Compute spatial metrics:
-// To reduce complexity, we compute the metric for a reduced set of points.
-// The spatial metrics are rough estimates of the prediction error cost for
-//  each QM spatial mode: 2x2,1x2,2x1
-// The metrics are a simple estimate of the up-sampling prediction error,
-// estimated assuming sub-sampling for decimation (no filtering),
-// and up-sampling back up with simple bilinear interpolation.
-int32_t VPMContentAnalysis::ComputeSpatialMetrics_C() {
-  const int sizei = height_;
-  const int sizej = width_;
-
-  // Pixel mean square average: used to normalize the spatial metrics.
-  uint32_t pixelMSA = 0;
-
-  uint32_t spatialErrSum = 0;
-  uint32_t spatialErrVSum = 0;
-  uint32_t spatialErrHSum = 0;
-
-  // make sure work section is a multiple of 16
-  const int width_end = ((sizej - 2 * border_) & -16) + border_;
-
-  for (int i = border_; i < sizei - border_; i += skip_num_) {
-    for (int j = border_; j < width_end; j++) {
-      int ssn1 = i * sizej + j;
-      int ssn2 = (i + 1) * sizej + j;  // bottom
-      int ssn3 = (i - 1) * sizej + j;  // top
-      int ssn4 = i * sizej + j + 1;    // right
-      int ssn5 = i * sizej + j - 1;    // left
-
-      uint16_t refPixel1 = orig_frame_[ssn1] << 1;
-      uint16_t refPixel2 = orig_frame_[ssn1] << 2;
-
-      uint8_t bottPixel = orig_frame_[ssn2];
-      uint8_t topPixel = orig_frame_[ssn3];
-      uint8_t rightPixel = orig_frame_[ssn4];
-      uint8_t leftPixel = orig_frame_[ssn5];
-
-      spatialErrSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
-          refPixel2 - static_cast<uint16_t>(bottPixel + topPixel + leftPixel +
-                                            rightPixel))));
-      spatialErrVSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
-          refPixel1 - static_cast<uint16_t>(bottPixel + topPixel))));
-      spatialErrHSum += static_cast<uint32_t>(abs(static_cast<int16_t>(
-          refPixel1 - static_cast<uint16_t>(leftPixel + rightPixel))));
-      pixelMSA += orig_frame_[ssn1];
-    }
-  }
-
-  // Normalize over all pixels.
-  const float spatialErr = static_cast<float>(spatialErrSum >> 2);
-  const float spatialErrH = static_cast<float>(spatialErrHSum >> 1);
-  const float spatialErrV = static_cast<float>(spatialErrVSum >> 1);
-  const float norm = static_cast<float>(pixelMSA);
-
-  // 2X2:
-  spatial_pred_err_ = spatialErr / norm;
-  // 1X2:
-  spatial_pred_err_h_ = spatialErrH / norm;
-  // 2X1:
-  spatial_pred_err_v_ = spatialErrV / norm;
-  return VPM_OK;
-}
-
-VideoContentMetrics* VPMContentAnalysis::ContentMetrics() {
-  if (ca_Init_ == false)
-    return NULL;
-
-  content_metrics_->spatial_pred_err = spatial_pred_err_;
-  content_metrics_->spatial_pred_err_h = spatial_pred_err_h_;
-  content_metrics_->spatial_pred_err_v = spatial_pred_err_v_;
-  // Motion metric: normalized temporal difference (MAD).
-  content_metrics_->motion_magnitude = motion_magnitude_;
-
-  return content_metrics_;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/content_analysis.h b/chromium/third_party/webrtc/modules/video_processing/content_analysis.h
deleted file mode 100644
index d3a11bd091d..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/content_analysis.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
-#define WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/video_processing/include/video_processing_defines.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_frame.h"
-
-namespace webrtc {
-
-class VPMContentAnalysis {
- public:
-  // When |runtime_cpu_detection| is true, runtime selection of an optimized
-  // code path is allowed.
-  explicit VPMContentAnalysis(bool runtime_cpu_detection);
-  ~VPMContentAnalysis();
-
-  // Initialize ContentAnalysis - should be called prior to
-  //  extractContentFeature
-  // Inputs:         width, height
-  // Return value:   0 if OK, negative value upon error
-  int32_t Initialize(int width, int height);
-
-  // Extract content Feature - main function of ContentAnalysis
-  // Input:           new frame
-  // Return value:    pointer to structure containing content Analysis
-  //                  metrics or NULL value upon error
-  VideoContentMetrics* ComputeContentMetrics(const VideoFrame& inputFrame);
-
-  // Release all allocated memory
-  // Output: 0 if OK, negative value upon error
-  int32_t Release();
-
- private:
-  // return motion metrics
-  VideoContentMetrics* ContentMetrics();
-
-  // Normalized temporal difference metric: for motion magnitude
-  typedef int32_t (VPMContentAnalysis::*TemporalDiffMetricFunc)();
-  TemporalDiffMetricFunc TemporalDiffMetric;
-  int32_t TemporalDiffMetric_C();
-
-  // Motion metric method: call 2 metrics (magnitude and size)
-  int32_t ComputeMotionMetrics();
-
-  // Spatial metric method: computes the 3 frame-average spatial
-  //  prediction errors (1x2,2x1,2x2)
-  typedef int32_t (VPMContentAnalysis::*ComputeSpatialMetricsFunc)();
-  ComputeSpatialMetricsFunc ComputeSpatialMetrics;
-  int32_t ComputeSpatialMetrics_C();
-
-#if defined(WEBRTC_ARCH_X86_FAMILY)
-  int32_t ComputeSpatialMetrics_SSE2();
-  int32_t TemporalDiffMetric_SSE2();
-#endif
-
-  const uint8_t* orig_frame_;
-  uint8_t* prev_frame_;
-  int width_;
-  int height_;
-  int skip_num_;
-  int border_;
-
-  // Content Metrics: Stores the local average of the metrics.
-  float motion_magnitude_;    // motion class
-  float spatial_pred_err_;    // spatial class
-  float spatial_pred_err_h_;  // spatial class
-  float spatial_pred_err_v_;  // spatial class
-  bool first_frame_;
-  bool ca_Init_;
-
-  VideoContentMetrics* content_metrics_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_CONTENT_ANALYSIS_H_
diff --git a/chromium/third_party/webrtc/modules/video_processing/content_analysis_sse2.cc b/chromium/third_party/webrtc/modules/video_processing/content_analysis_sse2.cc
deleted file mode 100644
index 7a60a89b454..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/content_analysis_sse2.cc
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_processing/content_analysis.h"
-
-#include <emmintrin.h>
-#include <math.h>
-
-namespace webrtc {
-
-int32_t VPMContentAnalysis::TemporalDiffMetric_SSE2() {
-  uint32_t num_pixels = 0;  // counter for # of pixels
-  const uint8_t* imgBufO = orig_frame_ + border_ * width_ + border_;
-  const uint8_t* imgBufP = prev_frame_ + border_ * width_ + border_;
-
-  const int32_t width_end = ((width_ - 2 * border_) & -16) + border_;
-
-  __m128i sad_64 = _mm_setzero_si128();
-  __m128i sum_64 = _mm_setzero_si128();
-  __m128i sqsum_64 = _mm_setzero_si128();
-  const __m128i z = _mm_setzero_si128();
-
-  for (uint16_t i = 0; i < (height_ - 2 * border_); i += skip_num_) {
-    __m128i sqsum_32 = _mm_setzero_si128();
-
-    const uint8_t* lineO = imgBufO;
-    const uint8_t* lineP = imgBufP;
-
-    // Work on 16 pixels at a time.  For HD content with a width of 1920
-    // this loop will run ~67 times (depending on border).  Maximum for
-    // abs(o-p) and sum(o) will be 255. _mm_sad_epu8 produces 2 64 bit
-    // results which are then accumulated.  There is no chance of
-    // rollover for these two accumulators.
-    // o*o will have a maximum of 255*255 = 65025.  This will roll over
-    // a 16 bit accumulator as 67*65025 > 65535, but will fit in a
-    // 32 bit accumulator.
-    for (uint16_t j = 0; j < width_end - border_; j += 16) {
-      const __m128i o = _mm_loadu_si128((__m128i*)(lineO));
-      const __m128i p = _mm_loadu_si128((__m128i*)(lineP));
-
-      lineO += 16;
-      lineP += 16;
-
-      // Abs pixel difference between frames.
-      sad_64 = _mm_add_epi64(sad_64, _mm_sad_epu8(o, p));
-
-      // sum of all pixels in frame
-      sum_64 = _mm_add_epi64(sum_64, _mm_sad_epu8(o, z));
-
-      // Squared sum of all pixels in frame.
-      const __m128i olo = _mm_unpacklo_epi8(o, z);
-      const __m128i ohi = _mm_unpackhi_epi8(o, z);
-
-      const __m128i sqsum_32_lo = _mm_madd_epi16(olo, olo);
-      const __m128i sqsum_32_hi = _mm_madd_epi16(ohi, ohi);
-
-      sqsum_32 = _mm_add_epi32(sqsum_32, sqsum_32_lo);
-      sqsum_32 = _mm_add_epi32(sqsum_32, sqsum_32_hi);
-    }
-
-    // Add to 64 bit running sum as to not roll over.
-    sqsum_64 =
-        _mm_add_epi64(sqsum_64, _mm_add_epi64(_mm_unpackhi_epi32(sqsum_32, z),
-                                              _mm_unpacklo_epi32(sqsum_32, z)));
-
-    imgBufO += width_ * skip_num_;
-    imgBufP += width_ * skip_num_;
-    num_pixels += (width_end - border_);
-  }
-
-  __m128i sad_final_128;
-  __m128i sum_final_128;
-  __m128i sqsum_final_128;
-
-  // Bring sums out of vector registers and into integer register
-  // domain, summing them along the way.
-  _mm_store_si128(&sad_final_128, sad_64);
-  _mm_store_si128(&sum_final_128, sum_64);
-  _mm_store_si128(&sqsum_final_128, sqsum_64);
-
-  uint64_t* sad_final_64 = reinterpret_cast<uint64_t*>(&sad_final_128);
-  uint64_t* sum_final_64 = reinterpret_cast<uint64_t*>(&sum_final_128);
-  uint64_t* sqsum_final_64 = reinterpret_cast<uint64_t*>(&sqsum_final_128);
-
-  const uint32_t pixelSum = sum_final_64[0] + sum_final_64[1];
-  const uint64_t pixelSqSum = sqsum_final_64[0] + sqsum_final_64[1];
-  const uint32_t tempDiffSum = sad_final_64[0] + sad_final_64[1];
-
-  // Default.
-  motion_magnitude_ = 0.0f;
-
-  if (tempDiffSum == 0)
-    return VPM_OK;
-
-  // Normalize over all pixels.
-  const float tempDiffAvg =
-      static_cast<float>(tempDiffSum) / static_cast<float>(num_pixels);
-  const float pixelSumAvg =
-      static_cast<float>(pixelSum) / static_cast<float>(num_pixels);
-  const float pixelSqSumAvg =
-      static_cast<float>(pixelSqSum) / static_cast<float>(num_pixels);
-  float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
-
-  if (contrast > 0.0) {
-    contrast = sqrt(contrast);
-    motion_magnitude_ = tempDiffAvg / contrast;
-  }
-
-  return VPM_OK;
-}
-
-int32_t VPMContentAnalysis::ComputeSpatialMetrics_SSE2() {
-  const uint8_t* imgBuf = orig_frame_ + border_ * width_;
-  const int32_t width_end = ((width_ - 2 * border_) & -16) + border_;
-
-  __m128i se_32 = _mm_setzero_si128();
-  __m128i sev_32 = _mm_setzero_si128();
-  __m128i seh_32 = _mm_setzero_si128();
-  __m128i msa_32 = _mm_setzero_si128();
-  const __m128i z = _mm_setzero_si128();
-
-  // Error is accumulated as a 32 bit value.  Looking at HD content with a
-  // height of 1080 lines, or about 67 macro blocks.  If the 16 bit row
-  // value is maxed out at 65529 for every row, 65529*1080 = 70777800, which
-  // will not roll over a 32 bit accumulator.
-  // skip_num_ is also used to reduce the number of rows
-  for (int32_t i = 0; i < (height_ - 2 * border_); i += skip_num_) {
-    __m128i se_16 = _mm_setzero_si128();
-    __m128i sev_16 = _mm_setzero_si128();
-    __m128i seh_16 = _mm_setzero_si128();
-    __m128i msa_16 = _mm_setzero_si128();
-
-    // Row error is accumulated as a 16 bit value.  There are 8
-    // accumulators.  Max value of a 16 bit number is 65529.  Looking
-    // at HD content, 1080p, has a width of 1920, 120 macro blocks.
-    // A mb at a time is processed at a time.  Absolute max error at
-    // a point would be abs(0-255+255+255+255) which equals 1020.
-    // 120*1020 = 122400.  The probability of hitting this is quite low
-    // on well behaved content.  A specially crafted image could roll over.
-    // border_ could also be adjusted to concentrate on just the center of
-    // the images for an HD capture in order to reduce the possiblity of
-    // rollover.
-    const uint8_t* lineTop = imgBuf - width_ + border_;
-    const uint8_t* lineCen = imgBuf + border_;
-    const uint8_t* lineBot = imgBuf + width_ + border_;
-
-    for (int32_t j = 0; j < width_end - border_; j += 16) {
-      const __m128i t = _mm_loadu_si128((__m128i*)(lineTop));
-      const __m128i l = _mm_loadu_si128((__m128i*)(lineCen - 1));
-      const __m128i c = _mm_loadu_si128((__m128i*)(lineCen));
-      const __m128i r = _mm_loadu_si128((__m128i*)(lineCen + 1));
-      const __m128i b = _mm_loadu_si128((__m128i*)(lineBot));
-
-      lineTop += 16;
-      lineCen += 16;
-      lineBot += 16;
-
-      // center pixel unpacked
-      __m128i clo = _mm_unpacklo_epi8(c, z);
-      __m128i chi = _mm_unpackhi_epi8(c, z);
-
-      // left right pixels unpacked and added together
-      const __m128i lrlo =
-          _mm_add_epi16(_mm_unpacklo_epi8(l, z), _mm_unpacklo_epi8(r, z));
-      const __m128i lrhi =
-          _mm_add_epi16(_mm_unpackhi_epi8(l, z), _mm_unpackhi_epi8(r, z));
-
-      // top & bottom pixels unpacked and added together
-      const __m128i tblo =
-          _mm_add_epi16(_mm_unpacklo_epi8(t, z), _mm_unpacklo_epi8(b, z));
-      const __m128i tbhi =
-          _mm_add_epi16(_mm_unpackhi_epi8(t, z), _mm_unpackhi_epi8(b, z));
-
-      // running sum of all pixels
-      msa_16 = _mm_add_epi16(msa_16, _mm_add_epi16(chi, clo));
-
-      clo = _mm_slli_epi16(clo, 1);
-      chi = _mm_slli_epi16(chi, 1);
-      const __m128i sevtlo = _mm_subs_epi16(clo, tblo);
-      const __m128i sevthi = _mm_subs_epi16(chi, tbhi);
-      const __m128i sehtlo = _mm_subs_epi16(clo, lrlo);
-      const __m128i sehthi = _mm_subs_epi16(chi, lrhi);
-
-      clo = _mm_slli_epi16(clo, 1);
-      chi = _mm_slli_epi16(chi, 1);
-      const __m128i setlo = _mm_subs_epi16(clo, _mm_add_epi16(lrlo, tblo));
-      const __m128i sethi = _mm_subs_epi16(chi, _mm_add_epi16(lrhi, tbhi));
-
-      // Add to 16 bit running sum
-      se_16 =
-          _mm_add_epi16(se_16, _mm_max_epi16(setlo, _mm_subs_epi16(z, setlo)));
-      se_16 =
-          _mm_add_epi16(se_16, _mm_max_epi16(sethi, _mm_subs_epi16(z, sethi)));
-      sev_16 = _mm_add_epi16(sev_16,
-                             _mm_max_epi16(sevtlo, _mm_subs_epi16(z, sevtlo)));
-      sev_16 = _mm_add_epi16(sev_16,
-                             _mm_max_epi16(sevthi, _mm_subs_epi16(z, sevthi)));
-      seh_16 = _mm_add_epi16(seh_16,
-                             _mm_max_epi16(sehtlo, _mm_subs_epi16(z, sehtlo)));
-      seh_16 = _mm_add_epi16(seh_16,
-                             _mm_max_epi16(sehthi, _mm_subs_epi16(z, sehthi)));
-    }
-
-    // Add to 32 bit running sum as to not roll over.
-    se_32 = _mm_add_epi32(se_32, _mm_add_epi32(_mm_unpackhi_epi16(se_16, z),
-                                               _mm_unpacklo_epi16(se_16, z)));
-    sev_32 =
-        _mm_add_epi32(sev_32, _mm_add_epi32(_mm_unpackhi_epi16(sev_16, z),
-                                            _mm_unpacklo_epi16(sev_16, z)));
-    seh_32 =
-        _mm_add_epi32(seh_32, _mm_add_epi32(_mm_unpackhi_epi16(seh_16, z),
-                                            _mm_unpacklo_epi16(seh_16, z)));
-    msa_32 =
-        _mm_add_epi32(msa_32, _mm_add_epi32(_mm_unpackhi_epi16(msa_16, z),
-                                            _mm_unpacklo_epi16(msa_16, z)));
-
-    imgBuf += width_ * skip_num_;
-  }
-
-  __m128i se_128;
-  __m128i sev_128;
-  __m128i seh_128;
-  __m128i msa_128;
-
-  // Bring sums out of vector registers and into integer register
-  // domain, summing them along the way.
-  _mm_store_si128(&se_128, _mm_add_epi64(_mm_unpackhi_epi32(se_32, z),
-                                         _mm_unpacklo_epi32(se_32, z)));
-  _mm_store_si128(&sev_128, _mm_add_epi64(_mm_unpackhi_epi32(sev_32, z),
-                                          _mm_unpacklo_epi32(sev_32, z)));
-  _mm_store_si128(&seh_128, _mm_add_epi64(_mm_unpackhi_epi32(seh_32, z),
-                                          _mm_unpacklo_epi32(seh_32, z)));
-  _mm_store_si128(&msa_128, _mm_add_epi64(_mm_unpackhi_epi32(msa_32, z),
-                                          _mm_unpacklo_epi32(msa_32, z)));
-
-  uint64_t* se_64 = reinterpret_cast<uint64_t*>(&se_128);
-  uint64_t* sev_64 = reinterpret_cast<uint64_t*>(&sev_128);
-  uint64_t* seh_64 = reinterpret_cast<uint64_t*>(&seh_128);
-  uint64_t* msa_64 = reinterpret_cast<uint64_t*>(&msa_128);
-
-  const uint32_t spatialErrSum = se_64[0] + se_64[1];
-  const uint32_t spatialErrVSum = sev_64[0] + sev_64[1];
-  const uint32_t spatialErrHSum = seh_64[0] + seh_64[1];
-  const uint32_t pixelMSA = msa_64[0] + msa_64[1];
-
-  // Normalize over all pixels.
-  const float spatialErr = static_cast<float>(spatialErrSum >> 2);
-  const float spatialErrH = static_cast<float>(spatialErrHSum >> 1);
-  const float spatialErrV = static_cast<float>(spatialErrVSum >> 1);
-  const float norm = static_cast<float>(pixelMSA);
-
-  // 2X2:
-  spatial_pred_err_ = spatialErr / norm;
-
-  // 1X2:
-  spatial_pred_err_h_ = spatialErrH / norm;
-
-  // 2X1:
-  spatial_pred_err_v_ = spatialErrV / norm;
-
-  return VPM_OK;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/deflickering.cc b/chromium/third_party/webrtc/modules/video_processing/deflickering.cc
deleted file mode 100644
index 0e936ce9b77..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/deflickering.cc
+++ /dev/null
@@ -1,402 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_processing/deflickering.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "webrtc/base/logging.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/system_wrappers/include/sort.h"
-
-namespace webrtc {
-
-// Detection constants
-// (Q4) Maximum allowed deviation for detection.
-enum { kFrequencyDeviation = 39 };
-// (Q4) Minimum frequency that can be detected.
-enum { kMinFrequencyToDetect = 32 };
-// Number of flickers before we accept detection
-enum { kNumFlickerBeforeDetect = 2 };
-enum { kmean_valueScaling = 4 };  // (Q4) In power of 2
-// Dead-zone region in terms of pixel values
-enum { kZeroCrossingDeadzone = 10 };
-// Deflickering constants.
-// Compute the quantiles over 1 / DownsamplingFactor of the image.
-enum { kDownsamplingFactor = 8 };
-enum { kLog2OfDownsamplingFactor = 3 };
-
-// To generate in Matlab:
-// >> probUW16 = round(2^11 *
-//     [0.05,0.1,0.2,0.3,0.4,0.5,0.6,0.7,0.8,0.9,0.95,0.97]);
-// >> fprintf('%d, ', probUW16)
-// Resolution reduced to avoid overflow when multiplying with the
-// (potentially) large number of pixels.
-const uint16_t VPMDeflickering::prob_uw16_[kNumProbs] = {
-    102,  205,  410,  614,  819,  1024,
-    1229, 1434, 1638, 1843, 1946, 1987};  // <Q11>
-
-// To generate in Matlab:
-// >> numQuants = 14; maxOnlyLength = 5;
-// >> weightUW16 = round(2^15 *
-//    [linspace(0.5, 1.0, numQuants - maxOnlyLength)]);
-// >> fprintf('%d, %d,\n ', weightUW16);
-const uint16_t VPMDeflickering::weight_uw16_[kNumQuants - kMaxOnlyLength] = {
-    16384, 18432, 20480, 22528, 24576, 26624, 28672, 30720, 32768};  // <Q15>
-
-VPMDeflickering::VPMDeflickering() {
-  Reset();
-}
-
-VPMDeflickering::~VPMDeflickering() {}
-
-void VPMDeflickering::Reset() {
-  mean_buffer_length_ = 0;
-  detection_state_ = 0;
-  frame_rate_ = 0;
-
-  memset(mean_buffer_, 0, sizeof(int32_t) * kMeanBufferLength);
-  memset(timestamp_buffer_, 0, sizeof(int32_t) * kMeanBufferLength);
-
-  // Initialize the history with a uniformly distributed histogram.
-  quant_hist_uw8_[0][0] = 0;
-  quant_hist_uw8_[0][kNumQuants - 1] = 255;
-  for (int32_t i = 0; i < kNumProbs; i++) {
-    // Unsigned round. <Q0>
-    quant_hist_uw8_[0][i + 1] =
-        static_cast<uint8_t>((prob_uw16_[i] * 255 + (1 << 10)) >> 11);
-  }
-
-  for (int32_t i = 1; i < kFrameHistory_size; i++) {
-    memcpy(quant_hist_uw8_[i], quant_hist_uw8_[0],
-           sizeof(uint8_t) * kNumQuants);
-  }
-}
-
-int32_t VPMDeflickering::ProcessFrame(VideoFrame* frame,
-                                      VideoProcessing::FrameStats* stats) {
-  assert(frame);
-  uint32_t frame_memory;
-  uint8_t quant_uw8[kNumQuants];
-  uint8_t maxquant_uw8[kNumQuants];
-  uint8_t minquant_uw8[kNumQuants];
-  uint16_t target_quant_uw16[kNumQuants];
-  uint16_t increment_uw16;
-  uint8_t map_uw8[256];
-
-  uint16_t tmp_uw16;
-  uint32_t tmp_uw32;
-  int width = frame->width();
-  int height = frame->height();
-
-  if (frame->IsZeroSize()) {
-    return VPM_GENERAL_ERROR;
-  }
-
-  // Stricter height check due to subsampling size calculation below.
-  if (height < 2) {
-    LOG(LS_ERROR) << "Invalid frame size.";
-    return VPM_GENERAL_ERROR;
-  }
-
-  if (!VideoProcessing::ValidFrameStats(*stats)) {
-    return VPM_GENERAL_ERROR;
-  }
-
-  if (PreDetection(frame->timestamp(), *stats) == -1)
-    return VPM_GENERAL_ERROR;
-
-  // Flicker detection
-  int32_t det_flicker = DetectFlicker();
-  if (det_flicker < 0) {
-    return VPM_GENERAL_ERROR;
-  } else if (det_flicker != 1) {
-    return 0;
-  }
-
-  // Size of luminance component.
-  const uint32_t y_size = height * width;
-
-  const uint32_t y_sub_size =
-      width * (((height - 1) >> kLog2OfDownsamplingFactor) + 1);
-  uint8_t* y_sorted = new uint8_t[y_sub_size];
-  uint32_t sort_row_idx = 0;
-  for (int i = 0; i < height; i += kDownsamplingFactor) {
-    memcpy(y_sorted + sort_row_idx * width, frame->buffer(kYPlane) + i * width,
-           width);
-    sort_row_idx++;
-  }
-
-  webrtc::Sort(y_sorted, y_sub_size, webrtc::TYPE_UWord8);
-
-  uint32_t prob_idx_uw32 = 0;
-  quant_uw8[0] = 0;
-  quant_uw8[kNumQuants - 1] = 255;
-
-  // Ensure we won't get an overflow below.
-  // In practice, the number of subsampled pixels will not become this large.
-  if (y_sub_size > (1 << 21) - 1) {
-    LOG(LS_ERROR) << "Subsampled number of pixels too large.";
-    return -1;
-  }
-
-  for (int32_t i = 0; i < kNumProbs; i++) {
-    // <Q0>.
-    prob_idx_uw32 = WEBRTC_SPL_UMUL_32_16(y_sub_size, prob_uw16_[i]) >> 11;
-    quant_uw8[i + 1] = y_sorted[prob_idx_uw32];
-  }
-
-  delete[] y_sorted;
-  y_sorted = NULL;
-
-  // Shift history for new frame.
-  memmove(quant_hist_uw8_[1], quant_hist_uw8_[0],
-          (kFrameHistory_size - 1) * kNumQuants * sizeof(uint8_t));
-  // Store current frame in history.
-  memcpy(quant_hist_uw8_[0], quant_uw8, kNumQuants * sizeof(uint8_t));
-
-  // We use a frame memory equal to the ceiling of half the frame rate to
-  // ensure we capture an entire period of flicker.
-  frame_memory = (frame_rate_ + (1 << 5)) >> 5;  // Unsigned ceiling. <Q0>
-                                                 // frame_rate_ in Q4.
-  if (frame_memory > kFrameHistory_size) {
-    frame_memory = kFrameHistory_size;
-  }
-
-  // Get maximum and minimum.
-  for (int32_t i = 0; i < kNumQuants; i++) {
-    maxquant_uw8[i] = 0;
-    minquant_uw8[i] = 255;
-    for (uint32_t j = 0; j < frame_memory; j++) {
-      if (quant_hist_uw8_[j][i] > maxquant_uw8[i]) {
-        maxquant_uw8[i] = quant_hist_uw8_[j][i];
-      }
-
-      if (quant_hist_uw8_[j][i] < minquant_uw8[i]) {
-        minquant_uw8[i] = quant_hist_uw8_[j][i];
-      }
-    }
-  }
-
-  // Get target quantiles.
-  for (int32_t i = 0; i < kNumQuants - kMaxOnlyLength; i++) {
-    // target = w * maxquant_uw8 + (1 - w) * minquant_uw8
-    // Weights w = |weight_uw16_| are in Q15, hence the final output has to be
-    // right shifted by 8 to end up in Q7.
-    target_quant_uw16[i] = static_cast<uint16_t>(
-        (weight_uw16_[i] * maxquant_uw8[i] +
-         ((1 << 15) - weight_uw16_[i]) * minquant_uw8[i]) >>
-        8);  // <Q7>
-  }
-
-  for (int32_t i = kNumQuants - kMaxOnlyLength; i < kNumQuants; i++) {
-    target_quant_uw16[i] = ((uint16_t)maxquant_uw8[i]) << 7;
-  }
-
-  // Compute the map from input to output pixels.
-  uint16_t mapUW16;  // <Q7>
-  for (int32_t i = 1; i < kNumQuants; i++) {
-    // As quant and targetQuant are limited to UWord8, it's safe to use Q7 here.
-    tmp_uw32 =
-        static_cast<uint32_t>(target_quant_uw16[i] - target_quant_uw16[i - 1]);
-    tmp_uw16 = static_cast<uint16_t>(quant_uw8[i] - quant_uw8[i - 1]);  // <Q0>
-
-    if (tmp_uw16 > 0) {
-      increment_uw16 =
-          static_cast<uint16_t>(WebRtcSpl_DivU32U16(tmp_uw32,
-                                                    tmp_uw16));  // <Q7>
-    } else {
-      // The value is irrelevant; the loop below will only iterate once.
-      increment_uw16 = 0;
-    }
-
-    mapUW16 = target_quant_uw16[i - 1];
-    for (uint32_t j = quant_uw8[i - 1]; j < (uint32_t)(quant_uw8[i] + 1); j++) {
-      // Unsigned round. <Q0>
-      map_uw8[j] = (uint8_t)((mapUW16 + (1 << 6)) >> 7);
-      mapUW16 += increment_uw16;
-    }
-  }
-
-  // Map to the output frame.
-  uint8_t* buffer = frame->buffer(kYPlane);
-  for (uint32_t i = 0; i < y_size; i++) {
-    buffer[i] = map_uw8[buffer[i]];
-  }
-
-  // Frame was altered, so reset stats.
-  VideoProcessing::ClearFrameStats(stats);
-
-  return VPM_OK;
-}
-
-/**
-   Performs some pre-detection operations. Must be called before
-   DetectFlicker().
-
-   \param[in] timestamp Timestamp of the current frame.
-   \param[in] stats     Statistics of the current frame.
-
-   \return 0: Success\n
-           2: Detection not possible due to flickering frequency too close to
-              zero.\n
-          -1: Error
-*/
-int32_t VPMDeflickering::PreDetection(
-    const uint32_t timestamp,
-    const VideoProcessing::FrameStats& stats) {
-  int32_t mean_val;  // Mean value of frame (Q4)
-  uint32_t frame_rate = 0;
-  int32_t meanBufferLength;  // Temp variable.
-
-  mean_val = ((stats.sum << kmean_valueScaling) / stats.num_pixels);
-  // Update mean value buffer.
-  // This should be done even though we might end up in an unreliable detection.
-  memmove(mean_buffer_ + 1, mean_buffer_,
-          (kMeanBufferLength - 1) * sizeof(int32_t));
-  mean_buffer_[0] = mean_val;
-
-  // Update timestamp buffer.
-  // This should be done even though we might end up in an unreliable detection.
-  memmove(timestamp_buffer_ + 1, timestamp_buffer_,
-          (kMeanBufferLength - 1) * sizeof(uint32_t));
-  timestamp_buffer_[0] = timestamp;
-
-  /* Compute current frame rate (Q4) */
-  if (timestamp_buffer_[kMeanBufferLength - 1] != 0) {
-    frame_rate = ((90000 << 4) * (kMeanBufferLength - 1));
-    frame_rate /=
-        (timestamp_buffer_[0] - timestamp_buffer_[kMeanBufferLength - 1]);
-  } else if (timestamp_buffer_[1] != 0) {
-    frame_rate = (90000 << 4) / (timestamp_buffer_[0] - timestamp_buffer_[1]);
-  }
-
-  /* Determine required size of mean value buffer (mean_buffer_length_) */
-  if (frame_rate == 0) {
-    meanBufferLength = 1;
-  } else {
-    meanBufferLength =
-        (kNumFlickerBeforeDetect * frame_rate) / kMinFrequencyToDetect;
-  }
-  /* Sanity check of buffer length */
-  if (meanBufferLength >= kMeanBufferLength) {
-    /* Too long buffer. The flickering frequency is too close to zero, which
-     * makes the estimation unreliable.
-     */
-    mean_buffer_length_ = 0;
-    return 2;
-  }
-  mean_buffer_length_ = meanBufferLength;
-
-  if ((timestamp_buffer_[mean_buffer_length_ - 1] != 0) &&
-      (mean_buffer_length_ != 1)) {
-    frame_rate = ((90000 << 4) * (mean_buffer_length_ - 1));
-    frame_rate /=
-        (timestamp_buffer_[0] - timestamp_buffer_[mean_buffer_length_ - 1]);
-  } else if (timestamp_buffer_[1] != 0) {
-    frame_rate = (90000 << 4) / (timestamp_buffer_[0] - timestamp_buffer_[1]);
-  }
-  frame_rate_ = frame_rate;
-
-  return VPM_OK;
-}
-
-/**
-   This function detects flicker in the video stream. As a side effect the
-   mean value buffer is updated with the new mean value.
-
-   \return 0: No flickering detected\n
-           1: Flickering detected\n
-           2: Detection not possible due to unreliable frequency interval
-          -1: Error
-*/
-int32_t VPMDeflickering::DetectFlicker() {
-  uint32_t i;
-  int32_t freqEst;  // (Q4) Frequency estimate to base detection upon
-  int32_t ret_val = -1;
-
-  /* Sanity check for mean_buffer_length_ */
-  if (mean_buffer_length_ < 2) {
-    /* Not possible to estimate frequency */
-    return 2;
-  }
-  // Count zero crossings with a dead zone to be robust against noise. If the
-  // noise std is 2 pixel this corresponds to about 95% confidence interval.
-  int32_t deadzone = (kZeroCrossingDeadzone << kmean_valueScaling);  // Q4
-  int32_t meanOfBuffer = 0;  // Mean value of mean value buffer.
-  int32_t numZeros = 0;      // Number of zeros that cross the dead-zone.
-  int32_t cntState = 0;      // State variable for zero crossing regions.
-  int32_t cntStateOld = 0;   // Previous state for zero crossing regions.
-
-  for (i = 0; i < mean_buffer_length_; i++) {
-    meanOfBuffer += mean_buffer_[i];
-  }
-  meanOfBuffer += (mean_buffer_length_ >> 1);  // Rounding, not truncation.
-  meanOfBuffer /= mean_buffer_length_;
-
-  // Count zero crossings.
-  cntStateOld = (mean_buffer_[0] >= (meanOfBuffer + deadzone));
-  cntStateOld -= (mean_buffer_[0] <= (meanOfBuffer - deadzone));
-  for (i = 1; i < mean_buffer_length_; i++) {
-    cntState = (mean_buffer_[i] >= (meanOfBuffer + deadzone));
-    cntState -= (mean_buffer_[i] <= (meanOfBuffer - deadzone));
-    if (cntStateOld == 0) {
-      cntStateOld = -cntState;
-    }
-    if (((cntState + cntStateOld) == 0) && (cntState != 0)) {
-      numZeros++;
-      cntStateOld = cntState;
-    }
-  }
-  // END count zero crossings.
-
-  /* Frequency estimation according to:
-  * freqEst = numZeros * frame_rate / 2 / mean_buffer_length_;
-  *
-  * Resolution is set to Q4
-  */
-  freqEst = ((numZeros * 90000) << 3);
-  freqEst /=
-      (timestamp_buffer_[0] - timestamp_buffer_[mean_buffer_length_ - 1]);
-
-  /* Translate frequency estimate to regions close to 100 and 120 Hz */
-  uint8_t freqState = 0;  // Current translation state;
-                          // (0) Not in interval,
-                          // (1) Within valid interval,
-                          // (2) Out of range
-  int32_t freqAlias = freqEst;
-  if (freqEst > kMinFrequencyToDetect) {
-    uint8_t aliasState = 1;
-    while (freqState == 0) {
-      /* Increase frequency */
-      freqAlias += (aliasState * frame_rate_);
-      freqAlias += ((freqEst << 1) * (1 - (aliasState << 1)));
-      /* Compute state */
-      freqState = (abs(freqAlias - (100 << 4)) <= kFrequencyDeviation);
-      freqState += (abs(freqAlias - (120 << 4)) <= kFrequencyDeviation);
-      freqState += 2 * (freqAlias > ((120 << 4) + kFrequencyDeviation));
-      /* Switch alias state */
-      aliasState++;
-      aliasState &= 0x01;
-    }
-  }
-  /* Is frequency estimate within detection region? */
-  if (freqState == 1) {
-    ret_val = 1;
-  } else if (freqState == 0) {
-    ret_val = 2;
-  } else {
-    ret_val = 0;
-  }
-  return ret_val;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/deflickering.h b/chromium/third_party/webrtc/modules/video_processing/deflickering.h
deleted file mode 100644
index 3ff2723aba8..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/deflickering.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
-#define WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
-
-#include <string.h>  // NULL
-
-#include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class VPMDeflickering {
- public:
-  VPMDeflickering();
-  ~VPMDeflickering();
-
-  void Reset();
-  int32_t ProcessFrame(VideoFrame* frame, VideoProcessing::FrameStats* stats);
-
- private:
-  int32_t PreDetection(uint32_t timestamp,
-                       const VideoProcessing::FrameStats& stats);
-
-  int32_t DetectFlicker();
-
-  enum { kMeanBufferLength = 32 };
-  enum { kFrameHistory_size = 15 };
-  enum { kNumProbs = 12 };
-  enum { kNumQuants = kNumProbs + 2 };
-  enum { kMaxOnlyLength = 5 };
-
-  uint32_t mean_buffer_length_;
-  uint8_t detection_state_;  // 0: No flickering
-                             // 1: Flickering detected
-                             // 2: In flickering
-  int32_t mean_buffer_[kMeanBufferLength];
-  uint32_t timestamp_buffer_[kMeanBufferLength];
-  uint32_t frame_rate_;
-  static const uint16_t prob_uw16_[kNumProbs];
-  static const uint16_t weight_uw16_[kNumQuants - kMaxOnlyLength];
-  uint8_t quant_hist_uw8_[kFrameHistory_size][kNumQuants];
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_DEFLICKERING_H_
diff --git a/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.cc b/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.cc
index fd0d0efb97d..100cdb519ab 100644
--- a/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.cc
@@ -15,29 +15,22 @@
 namespace webrtc {
 
 VPMFramePreprocessor::VPMFramePreprocessor()
-    : content_metrics_(nullptr),
-      resampled_frame_(),
-      enable_ca_(false),
-      frame_cnt_(0) {
+    : resampled_frame_(), frame_cnt_(0) {
   spatial_resampler_ = new VPMSimpleSpatialResampler();
-  ca_ = new VPMContentAnalysis(true);
   vd_ = new VPMVideoDecimator();
-  EnableDenosing(false);
+  EnableDenoising(false);
+  denoised_frame_toggle_ = 0;
 }
 
 VPMFramePreprocessor::~VPMFramePreprocessor() {
   Reset();
-  delete ca_;
   delete vd_;
   delete spatial_resampler_;
 }
 
 void VPMFramePreprocessor::Reset() {
-  ca_->Release();
   vd_->Reset();
-  content_metrics_ = nullptr;
   spatial_resampler_->Reset();
-  enable_ca_ = false;
   frame_cnt_ = 0;
 }
 
@@ -45,10 +38,6 @@ void VPMFramePreprocessor::EnableTemporalDecimation(bool enable) {
   vd_->EnableTemporalDecimation(enable);
 }
 
-void VPMFramePreprocessor::EnableContentAnalysis(bool enable) {
-  enable_ca_ = enable;
-}
-
 void VPMFramePreprocessor::SetInputFrameResampleMode(
     VideoFrameResampling resampling_mode) {
   spatial_resampler_->SetInputFrameResampleMode(resampling_mode);
@@ -70,15 +59,6 @@ int32_t VPMFramePreprocessor::SetTargetResolution(uint32_t width,
   return VPM_OK;
 }
 
-void VPMFramePreprocessor::SetTargetFramerate(int frame_rate) {
-  if (frame_rate == -1) {
-    vd_->EnableTemporalDecimation(false);
-  } else {
-    vd_->EnableTemporalDecimation(true);
-    vd_->SetTargetFramerate(frame_rate);
-  }
-}
-
 void VPMFramePreprocessor::UpdateIncomingframe_rate() {
   vd_->UpdateIncomingframe_rate();
 }
@@ -95,7 +75,7 @@ uint32_t VPMFramePreprocessor::GetDecimatedHeight() const {
   return spatial_resampler_->TargetHeight();
 }
 
-void VPMFramePreprocessor::EnableDenosing(bool enable) {
+void VPMFramePreprocessor::EnableDenoising(bool enable) {
   if (enable) {
     denoiser_.reset(new VideoDenoiser(true));
   } else {
@@ -116,9 +96,18 @@ const VideoFrame* VPMFramePreprocessor::PreprocessFrame(
 
   const VideoFrame* current_frame = &frame;
   if (denoiser_) {
-    denoiser_->DenoiseFrame(*current_frame, &denoised_frame_,
-                            &denoised_frame_prev_, 0);
-    current_frame = &denoised_frame_;
+    VideoFrame* denoised_frame = &denoised_frame_[0];
+    VideoFrame* denoised_frame_prev = &denoised_frame_[1];
+    // Swap the buffer to save one memcpy in DenoiseFrame.
+    if (denoised_frame_toggle_) {
+      denoised_frame = &denoised_frame_[1];
+      denoised_frame_prev = &denoised_frame_[0];
+    }
+    // Invert the flag.
+    denoised_frame_toggle_ ^= 1;
+    denoiser_->DenoiseFrame(*current_frame, denoised_frame, denoised_frame_prev,
+                            true);
+    current_frame = denoised_frame;
   }
 
   if (spatial_resampler_->ApplyResample(current_frame->width(),
@@ -130,18 +119,8 @@ const VideoFrame* VPMFramePreprocessor::PreprocessFrame(
     current_frame = &resampled_frame_;
   }
 
-  // Perform content analysis on the frame to be encoded.
-  if (enable_ca_ && frame_cnt_ % kSkipFrameCA == 0) {
-    // Compute new metrics every |kSkipFramesCA| frames, starting with
-    // the first frame.
-    content_metrics_ = ca_->ComputeContentMetrics(*current_frame);
-  }
   ++frame_cnt_;
   return current_frame;
 }
 
-VideoContentMetrics* VPMFramePreprocessor::GetContentMetrics() const {
-  return content_metrics_;
-}
-
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.h b/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.h
index c35dd0d7aff..4ac6b76e830 100644
--- a/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.h
+++ b/chromium/third_party/webrtc/modules/video_processing/frame_preprocessor.h
@@ -14,7 +14,6 @@
 #include <memory>
 
 #include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/modules/video_processing/content_analysis.h"
 #include "webrtc/modules/video_processing/spatial_resampler.h"
 #include "webrtc/modules/video_processing/video_decimator.h"
 #include "webrtc/typedefs.h"
@@ -38,17 +37,11 @@ class VPMFramePreprocessor {
 
   void SetInputFrameResampleMode(VideoFrameResampling resampling_mode);
 
-  // Enable content analysis.
-  void EnableContentAnalysis(bool enable);
-
   // Set target resolution: frame rate and dimension.
   int32_t SetTargetResolution(uint32_t width,
                               uint32_t height,
                               uint32_t frame_rate);
 
-  // Set target frame rate.
-  void SetTargetFramerate(int frame_rate);
-
   // Update incoming frame rate/dimension.
   void UpdateIncomingframe_rate();
 
@@ -60,24 +53,20 @@ class VPMFramePreprocessor {
   uint32_t GetDecimatedHeight() const;
 
   // Preprocess output:
-  void EnableDenosing(bool enable);
+  void EnableDenoising(bool enable);
   const VideoFrame* PreprocessFrame(const VideoFrame& frame);
-  VideoContentMetrics* GetContentMetrics() const;
 
  private:
   // The content does not change so much every frame, so to reduce complexity
   // we can compute new content metrics every |kSkipFrameCA| frames.
   enum { kSkipFrameCA = 2 };
 
-  VideoContentMetrics* content_metrics_;
-  VideoFrame denoised_frame_;
-  VideoFrame denoised_frame_prev_;
+  VideoFrame denoised_frame_[2];
   VideoFrame resampled_frame_;
   VPMSpatialResampler* spatial_resampler_;
-  VPMContentAnalysis* ca_;
   VPMVideoDecimator* vd_;
   std::unique_ptr<VideoDenoiser> denoiser_;
-  bool enable_ca_;
+  uint8_t denoised_frame_toggle_;
   uint32_t frame_cnt_;
 };
 
diff --git a/chromium/third_party/webrtc/modules/video_processing/include/video_processing.h b/chromium/third_party/webrtc/modules/video_processing/include/video_processing.h
index a8d63588876..e2069ddbe7a 100644
--- a/chromium/third_party/webrtc/modules/video_processing/include/video_processing.h
+++ b/chromium/third_party/webrtc/modules/video_processing/include/video_processing.h
@@ -28,46 +28,9 @@ namespace webrtc {
 
 class VideoProcessing {
  public:
-  struct FrameStats {
-    uint32_t hist[256];  // Frame histogram.
-    uint32_t mean;
-    uint32_t sum;
-    uint32_t num_pixels;
-    uint32_t sub_sampling_factor;  // Sub-sampling factor, in powers of 2.
-  };
-
-  enum BrightnessWarning { kNoWarning, kDarkWarning, kBrightWarning };
-
   static VideoProcessing* Create();
   virtual ~VideoProcessing() {}
 
-  // Retrieves statistics for the input frame. This function must be used to
-  // prepare a FrameStats struct for use in certain VPM functions.
-  static void GetFrameStats(const VideoFrame& frame, FrameStats* stats);
-
-  // Checks the validity of a FrameStats struct. Currently, valid implies only
-  // that is had changed from its initialized state.
-  static bool ValidFrameStats(const FrameStats& stats);
-
-  static void ClearFrameStats(FrameStats* stats);
-
-  // Increases/decreases the luminance value. 'delta' can be in the range {}
-  static void Brighten(int delta, VideoFrame* frame);
-
-  // Detects and removes camera flicker from a video stream. Every frame from
-  // the stream must be passed in. A frame will only be altered if flicker has
-  // been detected. Has a fixed-point implementation.
-  // Frame statistics provided by GetFrameStats(). On return the stats will
-  // be reset to zero if the frame was altered. Call GetFrameStats() again
-  // if the statistics for the altered frame are required.
-  virtual int32_t Deflickering(VideoFrame* frame, FrameStats* stats) = 0;
-
-  // Detects if a video frame is excessively bright or dark. Returns a
-  // warning if this is the case. Multiple frames should be passed in before
-  // expecting a warning. Has a floating-point implementation.
-  virtual int32_t BrightnessDetection(const VideoFrame& frame,
-                                      const FrameStats& stats) = 0;
-
   // The following functions refer to the pre-processor unit within VPM. The
   // pre-processor perfoms spatial/temporal decimation and content analysis on
   // the frames prior to encoding.
@@ -79,8 +42,6 @@ class VideoProcessing {
                                       uint32_t height,
                                       uint32_t frame_rate) = 0;
 
-  virtual void SetTargetFramerate(int frame_rate) = 0;
-
   virtual uint32_t GetDecimatedFrameRate() = 0;
   virtual uint32_t GetDecimatedWidth() const = 0;
   virtual uint32_t GetDecimatedHeight() const = 0;
@@ -90,11 +51,8 @@ class VideoProcessing {
   virtual void SetInputFrameResampleMode(
       VideoFrameResampling resampling_mode) = 0;
 
-  virtual void EnableDenosing(bool enable) = 0;
+  virtual void EnableDenoising(bool enable) = 0;
   virtual const VideoFrame* PreprocessFrame(const VideoFrame& frame) = 0;
-
-  virtual VideoContentMetrics* GetContentMetrics() const = 0;
-  virtual void EnableContentAnalysis(bool enable) = 0;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/test/brightness_detection_test.cc b/chromium/third_party/webrtc/modules/video_processing/test/brightness_detection_test.cc
deleted file mode 100644
index abce518e584..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/test/brightness_detection_test.cc
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <memory>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
-
-namespace webrtc {
-
-#if defined(WEBRTC_IOS)
-#define MAYBE_BrightnessDetection DISABLED_BrightnessDetection
-#else
-#define MAYBE_BrightnessDetection BrightnessDetection
-#endif
-TEST_F(VideoProcessingTest, MAYBE_BrightnessDetection) {
-  uint32_t frameNum = 0;
-  int32_t brightnessWarning = 0;
-  uint32_t warningCount = 0;
-  std::unique_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-         frame_length_) {
-    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                               0, kVideoRotation_0, &video_frame_));
-    frameNum++;
-    VideoProcessing::FrameStats stats;
-    vp_->GetFrameStats(video_frame_, &stats);
-    EXPECT_GT(stats.num_pixels, 0u);
-    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
-              0);
-    if (brightnessWarning != VideoProcessing::kNoWarning) {
-      warningCount++;
-    }
-  }
-  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-  // Expect few warnings
-  float warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-  printf("\nWarning proportions:\n");
-  printf("Stock foreman: %.1f %%\n", warningProportion);
-  EXPECT_LT(warningProportion, 10);
-
-  rewind(source_file_);
-  frameNum = 0;
-  warningCount = 0;
-  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-             frame_length_ &&
-         frameNum < 300) {
-    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                               0, kVideoRotation_0, &video_frame_));
-    frameNum++;
-
-    uint8_t* frame = video_frame_.buffer(kYPlane);
-    uint32_t yTmp = 0;
-    for (int yIdx = 0; yIdx < width_ * height_; yIdx++) {
-      yTmp = frame[yIdx] << 1;
-      if (yTmp > 255) {
-        yTmp = 255;
-      }
-      frame[yIdx] = static_cast<uint8_t>(yTmp);
-    }
-
-    VideoProcessing::FrameStats stats;
-    vp_->GetFrameStats(video_frame_, &stats);
-    EXPECT_GT(stats.num_pixels, 0u);
-    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
-              0);
-    EXPECT_NE(VideoProcessing::kDarkWarning, brightnessWarning);
-    if (brightnessWarning == VideoProcessing::kBrightWarning) {
-      warningCount++;
-    }
-  }
-  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-  // Expect many brightness warnings
-  warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-  printf("Bright foreman: %.1f %%\n", warningProportion);
-  EXPECT_GT(warningProportion, 95);
-
-  rewind(source_file_);
-  frameNum = 0;
-  warningCount = 0;
-  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-             frame_length_ &&
-         frameNum < 300) {
-    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                               0, kVideoRotation_0, &video_frame_));
-    frameNum++;
-
-    uint8_t* y_plane = video_frame_.buffer(kYPlane);
-    int32_t yTmp = 0;
-    for (int yIdx = 0; yIdx < width_ * height_; yIdx++) {
-      yTmp = y_plane[yIdx] >> 1;
-      y_plane[yIdx] = static_cast<uint8_t>(yTmp);
-    }
-
-    VideoProcessing::FrameStats stats;
-    vp_->GetFrameStats(video_frame_, &stats);
-    EXPECT_GT(stats.num_pixels, 0u);
-    ASSERT_GE(brightnessWarning = vp_->BrightnessDetection(video_frame_, stats),
-              0);
-    EXPECT_NE(VideoProcessing::kBrightWarning, brightnessWarning);
-    if (brightnessWarning == VideoProcessing::kDarkWarning) {
-      warningCount++;
-    }
-  }
-  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-  // Expect many darkness warnings
-  warningProportion = static_cast<float>(warningCount) / frameNum * 100;
-  printf("Dark foreman: %.1f %%\n\n", warningProportion);
-  EXPECT_GT(warningProportion, 90);
-}
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/test/content_metrics_test.cc b/chromium/third_party/webrtc/modules/video_processing/test/content_metrics_test.cc
deleted file mode 100644
index 80bb56489bb..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/test/content_metrics_test.cc
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <memory>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/modules/video_processing/content_analysis.h"
-#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
-
-namespace webrtc {
-
-#if defined(WEBRTC_IOS)
-TEST_F(VideoProcessingTest, DISABLED_ContentAnalysis) {
-#else
-TEST_F(VideoProcessingTest, ContentAnalysis) {
-#endif
-  VPMContentAnalysis ca__c(false);
-  VPMContentAnalysis ca__sse(true);
-  VideoContentMetrics* _cM_c;
-  VideoContentMetrics* _cM_SSE;
-
-  ca__c.Initialize(width_, height_);
-  ca__sse.Initialize(width_, height_);
-
-  std::unique_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-         frame_length_) {
-    // Using ConvertToI420 to add stride to the image.
-    EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                               0, kVideoRotation_0, &video_frame_));
-    _cM_c = ca__c.ComputeContentMetrics(video_frame_);
-    _cM_SSE = ca__sse.ComputeContentMetrics(video_frame_);
-
-    ASSERT_EQ(_cM_c->spatial_pred_err, _cM_SSE->spatial_pred_err);
-    ASSERT_EQ(_cM_c->spatial_pred_err_v, _cM_SSE->spatial_pred_err_v);
-    ASSERT_EQ(_cM_c->spatial_pred_err_h, _cM_SSE->spatial_pred_err_h);
-    ASSERT_EQ(_cM_c->motion_magnitude, _cM_SSE->motion_magnitude);
-  }
-  ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/test/deflickering_test.cc b/chromium/third_party/webrtc/modules/video_processing/test/deflickering_test.cc
deleted file mode 100644
index 5ff5692cce9..00000000000
--- a/chromium/third_party/webrtc/modules/video_processing/test/deflickering_test.cc
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <memory>
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/modules/video_processing/test/video_processing_unittest.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-#if defined(WEBRTC_IOS)
-TEST_F(VideoProcessingTest, DISABLED_Deflickering) {
-#else
-TEST_F(VideoProcessingTest, Deflickering) {
-#endif
-  enum { NumRuns = 30 };
-  uint32_t frameNum = 0;
-  const uint32_t frame_rate = 15;
-
-  int64_t min_runtime = 0;
-  int64_t avg_runtime = 0;
-
-  // Close automatically opened Foreman.
-  fclose(source_file_);
-  const std::string input_file =
-      webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
-  source_file_ = fopen(input_file.c_str(), "rb");
-  ASSERT_TRUE(source_file_ != NULL) << "Cannot read input file: " << input_file
-                                    << "\n";
-
-  const std::string output_file =
-      webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
-  FILE* deflickerFile = fopen(output_file.c_str(), "wb");
-  ASSERT_TRUE(deflickerFile != NULL)
-      << "Could not open output file: " << output_file << "\n";
-
-  printf("\nRun time [us / frame]:\n");
-  std::unique_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++) {
-    TickTime t0;
-    TickTime t1;
-    TickInterval acc_ticks;
-    uint32_t timeStamp = 1;
-
-    frameNum = 0;
-    while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
-           frame_length_) {
-      frameNum++;
-      EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_,
-                                 height_, 0, kVideoRotation_0, &video_frame_));
-      video_frame_.set_timestamp(timeStamp);
-
-      t0 = TickTime::Now();
-      VideoProcessing::FrameStats stats;
-      vp_->GetFrameStats(video_frame_, &stats);
-      EXPECT_GT(stats.num_pixels, 0u);
-      ASSERT_EQ(0, vp_->Deflickering(&video_frame_, &stats));
-      t1 = TickTime::Now();
-      acc_ticks += (t1 - t0);
-
-      if (run_idx == 0) {
-        if (PrintVideoFrame(video_frame_, deflickerFile) < 0) {
-          return;
-        }
-      }
-      timeStamp += (90000 / frame_rate);
-    }
-    ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
-
-    printf("%u\n", static_cast<int>(acc_ticks.Microseconds() / frameNum));
-    if (acc_ticks.Microseconds() < min_runtime || run_idx == 0) {
-      min_runtime = acc_ticks.Microseconds();
-    }
-    avg_runtime += acc_ticks.Microseconds();
-
-    rewind(source_file_);
-  }
-  ASSERT_EQ(0, fclose(deflickerFile));
-  // TODO(kjellander): Add verification of deflicker output file.
-
-  printf("\nAverage run time = %d us / frame\n",
-         static_cast<int>(avg_runtime / frameNum / NumRuns));
-  printf("Min run time = %d us / frame\n\n",
-         static_cast<int>(min_runtime / frameNum));
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/test/denoiser_test.cc b/chromium/third_party/webrtc/modules/video_processing/test/denoiser_test.cc
index a45f933bb54..4c13a05d635 100644
--- a/chromium/third_party/webrtc/modules/video_processing/test/denoiser_test.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/test/denoiser_test.cc
@@ -31,18 +31,10 @@ TEST_F(VideoProcessingTest, CopyMem) {
     }
   }
 
-  memset(dst, 0, 8 * 8);
-  df_c->CopyMem8x8(src, 8, dst, 8);
-  EXPECT_EQ(0, memcmp(src, dst, 8 * 8));
-
   memset(dst, 0, 16 * 16);
   df_c->CopyMem16x16(src, 16, dst, 16);
   EXPECT_EQ(0, memcmp(src, dst, 16 * 16));
 
-  memset(dst, 0, 8 * 8);
-  df_sse_neon->CopyMem16x16(src, 8, dst, 8);
-  EXPECT_EQ(0, memcmp(src, dst, 8 * 8));
-
   memset(dst, 0, 16 * 16);
   df_sse_neon->CopyMem16x16(src, 16, dst, 16);
   EXPECT_EQ(0, memcmp(src, dst, 16 * 16));
@@ -87,10 +79,9 @@ TEST_F(VideoProcessingTest, MbDenoise) {
     }
   }
   memset(dst, 0, 16 * 16);
-  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1, false);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
   memset(dst_sse_neon, 0, 16 * 16);
-  df_sse_neon->MbDenoise(running_src, 16, dst_sse_neon, 16, src, 16, 0, 1,
-                         false);
+  df_sse_neon->MbDenoise(running_src, 16, dst_sse_neon, 16, src, 16, 0, 1);
   EXPECT_EQ(0, memcmp(dst, dst_sse_neon, 16 * 16));
 
   // Test case: |diff| >= |4 + shift_inc1|
@@ -101,10 +92,9 @@ TEST_F(VideoProcessingTest, MbDenoise) {
     }
   }
   memset(dst, 0, 16 * 16);
-  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1, false);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
   memset(dst_sse_neon, 0, 16 * 16);
-  df_sse_neon->MbDenoise(running_src, 16, dst_sse_neon, 16, src, 16, 0, 1,
-                         false);
+  df_sse_neon->MbDenoise(running_src, 16, dst_sse_neon, 16, src, 16, 0, 1);
   EXPECT_EQ(0, memcmp(dst, dst_sse_neon, 16 * 16));
 
   // Test case: |diff| >= 8
@@ -115,10 +105,9 @@ TEST_F(VideoProcessingTest, MbDenoise) {
     }
   }
   memset(dst, 0, 16 * 16);
-  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1, false);
+  df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
   memset(dst_sse_neon, 0, 16 * 16);
-  df_sse_neon->MbDenoise(running_src, 16, dst_sse_neon, 16, src, 16, 0, 1,
-                         false);
+  df_sse_neon->MbDenoise(running_src, 16, dst_sse_neon, 16, src, 16, 0, 1);
   EXPECT_EQ(0, memcmp(dst, dst_sse_neon, 16 * 16));
 
   // Test case: |diff| > 15
@@ -130,22 +119,23 @@ TEST_F(VideoProcessingTest, MbDenoise) {
   }
   memset(dst, 0, 16 * 16);
   DenoiserDecision decision =
-      df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1, false);
+      df_c->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
   EXPECT_EQ(COPY_BLOCK, decision);
-  decision =
-      df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1, false);
+  decision = df_sse_neon->MbDenoise(running_src, 16, dst, 16, src, 16, 0, 1);
   EXPECT_EQ(COPY_BLOCK, decision);
 }
 
 TEST_F(VideoProcessingTest, Denoiser) {
+  // Used in swap buffer.
+  int denoised_frame_toggle = 0;
   // Create pure C denoiser.
   VideoDenoiser denoiser_c(false);
   // Create SSE or NEON denoiser.
   VideoDenoiser denoiser_sse_neon(true);
   VideoFrame denoised_frame_c;
-  VideoFrame denoised_frame_track_c;
+  VideoFrame denoised_frame_prev_c;
   VideoFrame denoised_frame_sse_neon;
-  VideoFrame denoised_frame_track_sse_neon;
+  VideoFrame denoised_frame_prev_sse_neon;
 
   std::unique_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
   while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
@@ -154,13 +144,25 @@ TEST_F(VideoProcessingTest, Denoiser) {
     EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
                                0, kVideoRotation_0, &video_frame_));
 
-    denoiser_c.DenoiseFrame(video_frame_, &denoised_frame_c,
-                            &denoised_frame_track_c, -1);
-    denoiser_sse_neon.DenoiseFrame(video_frame_, &denoised_frame_sse_neon,
-                                   &denoised_frame_track_sse_neon, -1);
-
+    VideoFrame* p_denoised_c = &denoised_frame_c;
+    VideoFrame* p_denoised_prev_c = &denoised_frame_prev_c;
+    VideoFrame* p_denoised_sse_neon = &denoised_frame_sse_neon;
+    VideoFrame* p_denoised_prev_sse_neon = &denoised_frame_prev_sse_neon;
+    // Swap the buffer to save one memcpy in DenoiseFrame.
+    if (denoised_frame_toggle) {
+      p_denoised_c = &denoised_frame_prev_c;
+      p_denoised_prev_c = &denoised_frame_c;
+      p_denoised_sse_neon = &denoised_frame_prev_sse_neon;
+      p_denoised_prev_sse_neon = &denoised_frame_sse_neon;
+    }
+    denoiser_c.DenoiseFrame(video_frame_, p_denoised_c, p_denoised_prev_c,
+                            false);
+    denoiser_sse_neon.DenoiseFrame(video_frame_, p_denoised_sse_neon,
+                                   p_denoised_prev_sse_neon, false);
+    // Invert the flag.
+    denoised_frame_toggle ^= 1;
     // Denoising results should be the same for C and SSE/NEON denoiser.
-    ASSERT_TRUE(test::FramesEqual(denoised_frame_c, denoised_frame_sse_neon));
+    ASSERT_TRUE(test::FramesEqual(*p_denoised_c, *p_denoised_sse_neon));
   }
   ASSERT_NE(0, feof(source_file_)) << "Error reading source file";
 }
diff --git a/chromium/third_party/webrtc/modules/video_processing/test/video_processing_unittest.cc b/chromium/third_party/webrtc/modules/video_processing/test/video_processing_unittest.cc
index 0d18d0a4c89..9e61b51884f 100644
--- a/chromium/third_party/webrtc/modules/video_processing/test/video_processing_unittest.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/test/video_processing_unittest.cc
@@ -15,8 +15,8 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ -51,8 +51,6 @@ static void TestSize(const VideoFrame& source_frame,
                      int target_height,
                      double expected_psnr,
                      VideoProcessing* vpm);
-static bool CompareFrames(const webrtc::VideoFrame& frame1,
-                          const webrtc::VideoFrame& frame2);
 static void WriteProcessedFrameForVisualInspection(const VideoFrame& source,
                                                    const VideoFrame& processed);
 
@@ -73,9 +71,12 @@ void VideoProcessingTest::SetUp() {
   video_frame_.CreateEmptyFrame(width_, height_, width_,
                                 half_width_, half_width_);
   // Clear video frame so DrMemory/Valgrind will allow reads of the buffer.
-  memset(video_frame_.buffer(kYPlane), 0, video_frame_.allocated_size(kYPlane));
-  memset(video_frame_.buffer(kUPlane), 0, video_frame_.allocated_size(kUPlane));
-  memset(video_frame_.buffer(kVPlane), 0, video_frame_.allocated_size(kVPlane));
+  memset(video_frame_.video_frame_buffer()->MutableDataY(), 0,
+         video_frame_.allocated_size(kYPlane));
+  memset(video_frame_.video_frame_buffer()->MutableDataU(), 0,
+         video_frame_.allocated_size(kUPlane));
+  memset(video_frame_.video_frame_buffer()->MutableDataV(), 0,
+         video_frame_.allocated_size(kVPlane));
   const std::string video_file =
       webrtc::test::ResourcePath("foreman_cif", "yuv");
   source_file_ = fopen(video_file.c_str(), "rb");
@@ -93,108 +94,6 @@ void VideoProcessingTest::TearDown() {
 }
 
 #if defined(WEBRTC_IOS)
-TEST_F(VideoProcessingTest, DISABLED_HandleNullBuffer) {
-#else
-TEST_F(VideoProcessingTest, HandleNullBuffer) {
-#endif
-  // TODO(mikhal/stefan): Do we need this one?
-  VideoProcessing::FrameStats stats;
-  // Video frame with unallocated buffer.
-  VideoFrame videoFrame;
-
-  vp_->GetFrameStats(videoFrame, &stats);
-  EXPECT_EQ(stats.num_pixels, 0u);
-
-  EXPECT_EQ(-1, vp_->Deflickering(&videoFrame, &stats));
-
-  EXPECT_EQ(-3, vp_->BrightnessDetection(videoFrame, stats));
-}
-
-#if defined(WEBRTC_IOS)
-TEST_F(VideoProcessingTest, DISABLED_HandleBadStats) {
-#else
-TEST_F(VideoProcessingTest, HandleBadStats) {
-#endif
-  VideoProcessing::FrameStats stats;
-  vp_->ClearFrameStats(&stats);
-  std::unique_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_,
-            fread(video_buffer.get(), 1, frame_length_, source_file_));
-  EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                             0, kVideoRotation_0, &video_frame_));
-
-  EXPECT_EQ(-1, vp_->Deflickering(&video_frame_, &stats));
-
-  EXPECT_EQ(-3, vp_->BrightnessDetection(video_frame_, stats));
-}
-
-#if defined(WEBRTC_IOS)
-TEST_F(VideoProcessingTest, DISABLED_IdenticalResultsAfterReset) {
-#else
-TEST_F(VideoProcessingTest, IdenticalResultsAfterReset) {
-#endif
-  VideoFrame video_frame2;
-  VideoProcessing::FrameStats stats;
-  // Only testing non-static functions here.
-  std::unique_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_,
-            fread(video_buffer.get(), 1, frame_length_, source_file_));
-  EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                             0, kVideoRotation_0, &video_frame_));
-  vp_->GetFrameStats(video_frame_, &stats);
-  EXPECT_GT(stats.num_pixels, 0u);
-  video_frame2.CopyFrame(video_frame_);
-  ASSERT_EQ(0, vp_->Deflickering(&video_frame_, &stats));
-
-  // Retrieve frame stats again in case Deflickering() has zeroed them.
-  vp_->GetFrameStats(video_frame2, &stats);
-  EXPECT_GT(stats.num_pixels, 0u);
-  ASSERT_EQ(0, vp_->Deflickering(&video_frame2, &stats));
-  EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
-
-  ASSERT_EQ(frame_length_,
-            fread(video_buffer.get(), 1, frame_length_, source_file_));
-  EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                             0, kVideoRotation_0, &video_frame_));
-  vp_->GetFrameStats(video_frame_, &stats);
-  EXPECT_GT(stats.num_pixels, 0u);
-  video_frame2.CopyFrame(video_frame_);
-  ASSERT_EQ(0, vp_->BrightnessDetection(video_frame_, stats));
-
-  ASSERT_EQ(0, vp_->BrightnessDetection(video_frame2, stats));
-  EXPECT_TRUE(CompareFrames(video_frame_, video_frame2));
-}
-
-#if defined(WEBRTC_IOS)
-TEST_F(VideoProcessingTest, DISABLED_FrameStats) {
-#else
-TEST_F(VideoProcessingTest, FrameStats) {
-#endif
-  VideoProcessing::FrameStats stats;
-  vp_->ClearFrameStats(&stats);
-  std::unique_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
-  ASSERT_EQ(frame_length_,
-            fread(video_buffer.get(), 1, frame_length_, source_file_));
-  EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0, width_, height_,
-                             0, kVideoRotation_0, &video_frame_));
-
-  EXPECT_FALSE(vp_->ValidFrameStats(stats));
-  vp_->GetFrameStats(video_frame_, &stats);
-  EXPECT_GT(stats.num_pixels, 0u);
-  EXPECT_TRUE(vp_->ValidFrameStats(stats));
-
-  printf("\nFrameStats\n");
-  printf("mean: %u\nnum_pixels: %u\nsubSamplFactor: %u\nsum: %u\n\n",
-         static_cast<unsigned int>(stats.mean),
-         static_cast<unsigned int>(stats.num_pixels),
-         static_cast<unsigned int>(stats.sub_sampling_factor),
-         static_cast<unsigned int>(stats.sum));
-
-  vp_->ClearFrameStats(&stats);
-  EXPECT_FALSE(vp_->ValidFrameStats(stats));
-}
-
-#if defined(WEBRTC_IOS)
 TEST_F(VideoProcessingTest, DISABLED_PreprocessorLogic) {
 #else
 TEST_F(VideoProcessingTest, PreprocessorLogic) {
@@ -230,8 +129,6 @@ TEST_F(VideoProcessingTest, Resampler) {
   rewind(source_file_);
   ASSERT_TRUE(source_file_ != NULL) << "Cannot read input file \n";
 
-  // CA not needed here
-  vp_->EnableContentAnalysis(false);
   // no temporal decimation
   vp_->EnableTemporalDecimation(false);
 
@@ -248,11 +145,12 @@ TEST_F(VideoProcessingTest, Resampler) {
 
   for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++) {
     // Initiate test timer.
-    const TickTime time_start = TickTime::Now();
+    const int64_t time_start = rtc::TimeNanos();
 
     // Init the sourceFrame with a timestamp.
-    video_frame_.set_render_time_ms(time_start.MillisecondTimestamp());
-    video_frame_.set_timestamp(time_start.MillisecondTimestamp() * 90);
+    int64_t time_start_ms = time_start / rtc::kNumNanosecsPerMillisec;
+    video_frame_.set_render_time_ms(time_start_ms);
+    video_frame_.set_timestamp(time_start_ms * 90);
 
     // Test scaling to different sizes: source is of |width|/|height| = 352/288.
     // Pure scaling:
@@ -295,7 +193,8 @@ TEST_F(VideoProcessingTest, Resampler) {
     TestSize(video_frame_, cropped_source_frame, 281, 175, 29.3, vp_);
 
     // Stop timer.
-    const int64_t runtime = (TickTime::Now() - time_start).Microseconds();
+    const int64_t runtime =
+        (rtc::TimeNanos() - time_start) / rtc::kNumNanosecsPerMicrosec;
     if (runtime < min_runtime || run_idx == 0) {
       min_runtime = runtime;
     }
@@ -378,22 +277,6 @@ void TestSize(const VideoFrame& source_frame,
       target_height);
 }
 
-bool CompareFrames(const webrtc::VideoFrame& frame1,
-                   const webrtc::VideoFrame& frame2) {
-  for (int plane = 0; plane < webrtc::kNumOfPlanes; plane++) {
-    webrtc::PlaneType plane_type = static_cast<webrtc::PlaneType>(plane);
-    int allocated_size1 = frame1.allocated_size(plane_type);
-    int allocated_size2 = frame2.allocated_size(plane_type);
-    if (allocated_size1 != allocated_size2)
-      return false;
-    const uint8_t* plane_buffer1 = frame1.buffer(plane_type);
-    const uint8_t* plane_buffer2 = frame2.buffer(plane_type);
-    if (memcmp(plane_buffer1, plane_buffer2, allocated_size1))
-      return false;
-  }
-  return true;
-}
-
 void WriteProcessedFrameForVisualInspection(const VideoFrame& source,
                                             const VideoFrame& processed) {
   // Skip if writing to files is not enabled.
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.cc b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.cc
index b111a0e4123..376dec74a35 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.cc
@@ -45,14 +45,6 @@ std::unique_ptr<DenoiserFilter> DenoiserFilter::Create(
     filter.reset(new DenoiserFilterNEON());
     if (cpu_type != nullptr)
       *cpu_type = CPU_NEON;
-#elif defined(WEBRTC_DETECT_NEON)
-    if (WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) {
-      filter.reset(new DenoiserFilterNEON());
-      if (cpu_type != nullptr)
-        *cpu_type = CPU_NEON;
-    } else {
-      filter.reset(new DenoiserFilterC());
-    }
 #else
     filter.reset(new DenoiserFilterC());
 #endif
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.h b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.h
index f2c7570083d..1254a88d3c9 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.h
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter.h
@@ -25,12 +25,6 @@ extern const int kSumDiffThresholdHigh;
 
 enum DenoiserDecision { COPY_BLOCK, FILTER_BLOCK };
 enum CpuType { CPU_NEON, CPU_NOT_NEON };
-struct DenoiseMetrics {
-  uint32_t var;
-  uint32_t sad;
-  uint8_t denoise;
-  bool is_skin;
-};
 
 class DenoiserFilter {
  public:
@@ -43,10 +37,6 @@ class DenoiserFilter {
                             int src_stride,
                             uint8_t* dst,
                             int dst_stride) = 0;
-  virtual void CopyMem8x8(const uint8_t* src,
-                          int src_stride,
-                          uint8_t* dst,
-                          int dst_stride) = 0;
   virtual uint32_t Variance16x8(const uint8_t* a,
                                 int a_stride,
                                 const uint8_t* b,
@@ -59,8 +49,7 @@ class DenoiserFilter {
                                      const uint8_t* sig,
                                      int sig_stride,
                                      uint8_t motion_magnitude,
-                                     int increase_denoising,
-                                     bool denoise_always) = 0;
+                                     int increase_denoising) = 0;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.cc b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.cc
index 8c84f4989c2..1b3c0b70987 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.cc
@@ -25,17 +25,6 @@ void DenoiserFilterC::CopyMem16x16(const uint8_t* src,
   }
 }
 
-void DenoiserFilterC::CopyMem8x8(const uint8_t* src,
-                                 int src_stride,
-                                 uint8_t* dst,
-                                 int dst_stride) {
-  for (int i = 0; i < 8; i++) {
-    memcpy(dst, src, 8);
-    src += src_stride;
-    dst += dst_stride;
-  }
-}
-
 uint32_t DenoiserFilterC::Variance16x8(const uint8_t* a,
                                        int a_stride,
                                        const uint8_t* b,
@@ -66,8 +55,7 @@ DenoiserDecision DenoiserFilterC::MbDenoise(uint8_t* mc_running_avg_y,
                                             const uint8_t* sig,
                                             int sig_stride,
                                             uint8_t motion_magnitude,
-                                            int increase_denoising,
-                                            bool denoise_always) {
+                                            int increase_denoising) {
   int sum_diff_thresh = 0;
   int sum_diff = 0;
   int adj_val[3] = {3, 4, 6};
@@ -137,60 +125,10 @@ DenoiserDecision DenoiserFilterC::MbDenoise(uint8_t* mc_running_avg_y,
     sum_diff += col_sum[c];
   }
 
-  if (denoise_always)
-    sum_diff_thresh = INT_MAX;
-  else if (increase_denoising)
-    sum_diff_thresh = kSumDiffThresholdHigh;
-  else
-    sum_diff_thresh = kSumDiffThreshold;
-  if (abs(sum_diff) > sum_diff_thresh) {
-    int delta = ((abs(sum_diff) - sum_diff_thresh) >> 8) + 1;
-    // Only apply the adjustment for max delta up to 3.
-    if (delta < 4) {
-      sig -= sig_stride * 16;
-      mc_running_avg_y -= mc_avg_y_stride * 16;
-      running_avg_y -= avg_y_stride * 16;
-      for (int r = 0; r < 16; ++r) {
-        for (int c = 0; c < 16; ++c) {
-          int diff = mc_running_avg_y[c] - sig[c];
-          int adjustment = abs(diff);
-          if (adjustment > delta)
-            adjustment = delta;
-          if (diff > 0) {
-            // Bring denoised signal down.
-            if (running_avg_y[c] - adjustment < 0)
-              running_avg_y[c] = 0;
-            else
-              running_avg_y[c] = running_avg_y[c] - adjustment;
-            col_sum[c] -= adjustment;
-          } else if (diff < 0) {
-            // Bring denoised signal up.
-            if (running_avg_y[c] + adjustment > 255)
-              running_avg_y[c] = 255;
-            else
-              running_avg_y[c] = running_avg_y[c] + adjustment;
-            col_sum[c] += adjustment;
-          }
-        }
-        sig += sig_stride;
-        mc_running_avg_y += mc_avg_y_stride;
-        running_avg_y += avg_y_stride;
-      }
-
-      sum_diff = 0;
-      for (int c = 0; c < 16; ++c) {
-        if (col_sum[c] >= 128) {
-          col_sum[c] = 127;
-        }
-        sum_diff += col_sum[c];
-      }
-
-      if (abs(sum_diff) > sum_diff_thresh)
-        return COPY_BLOCK;
-    } else {
-      return COPY_BLOCK;
-    }
-  }
+  sum_diff_thresh =
+      increase_denoising ? kSumDiffThresholdHigh : kSumDiffThreshold;
+  if (abs(sum_diff) > sum_diff_thresh)
+    return COPY_BLOCK;
 
   return FILTER_BLOCK;
 }
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.h b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.h
index 3e52c3e47c9..d8b6c5eb797 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.h
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_c.h
@@ -22,10 +22,6 @@ class DenoiserFilterC : public DenoiserFilter {
                     int src_stride,
                     uint8_t* dst,
                     int dst_stride) override;
-  void CopyMem8x8(const uint8_t* src,
-                  int src_stride,
-                  uint8_t* dst,
-                  int dst_stride) override;
   uint32_t Variance16x8(const uint8_t* a,
                         int a_stride,
                         const uint8_t* b,
@@ -38,8 +34,7 @@ class DenoiserFilterC : public DenoiserFilter {
                              const uint8_t* sig,
                              int sig_stride,
                              uint8_t motion_magnitude,
-                             int increase_denoising,
-                             bool denoise_always) override;
+                             int increase_denoising) override;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.cc b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.cc
index 2920305f71b..68c94cbdb7a 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.cc
@@ -14,6 +14,8 @@
 
 namespace webrtc {
 
+const int kSumDiffThresholdHighNeon = 600;
+
 static int HorizontalAddS16x8(const int16x8_t v_16x8) {
   const int32x4_t a = vpaddlq_s16(v_16x8);
   const int64x2_t b = vpaddlq_s32(a);
@@ -75,20 +77,6 @@ void DenoiserFilterNEON::CopyMem16x16(const uint8_t* src,
   }
 }
 
-void DenoiserFilterNEON::CopyMem8x8(const uint8_t* src,
-                                    int src_stride,
-                                    uint8_t* dst,
-                                    int dst_stride) {
-  uint8x8_t vtmp;
-
-  for (int r = 0; r < 8; r++) {
-    vtmp = vld1_u8(src);
-    vst1_u8(dst, vtmp);
-    src += src_stride;
-    dst += dst_stride;
-  }
-}
-
 uint32_t DenoiserFilterNEON::Variance16x8(const uint8_t* a,
                                           int a_stride,
                                           const uint8_t* b,
@@ -106,8 +94,7 @@ DenoiserDecision DenoiserFilterNEON::MbDenoise(uint8_t* mc_running_avg_y,
                                                const uint8_t* sig,
                                                int sig_stride,
                                                uint8_t motion_magnitude,
-                                               int increase_denoising,
-                                               bool denoise_always) {
+                                               int increase_denoising) {
   // If motion_magnitude is small, making the denoiser more aggressive by
   // increasing the adjustment for each level, level1 adjustment is
   // increased, the deltas stay the same.
@@ -190,92 +177,13 @@ DenoiserDecision DenoiserFilterNEON::MbDenoise(uint8_t* mc_running_avg_y,
   }
 
   // Too much adjustments => copy block.
-  {
-    int64x1_t x = vqadd_s64(vget_high_s64(v_sum_diff_total),
-                            vget_low_s64(v_sum_diff_total));
-    int sum_diff = vget_lane_s32(vabs_s32(vreinterpret_s32_s64(x)), 0);
-    if (denoise_always)
-      sum_diff_thresh = INT_MAX;
-    else if (increase_denoising)
-      sum_diff_thresh = kSumDiffThresholdHigh;
-    else
-      sum_diff_thresh = kSumDiffThreshold;
-    if (sum_diff > sum_diff_thresh) {
-      // Before returning to copy the block (i.e., apply no denoising),
-      // checK if we can still apply some (weaker) temporal filtering to
-      // this block, that would otherwise not be denoised at all. Simplest
-      // is to apply an additional adjustment to running_avg_y to bring it
-      // closer to sig. The adjustment is capped by a maximum delta, and
-      // chosen such that in most cases the resulting sum_diff will be
-      // within the accceptable range given by sum_diff_thresh.
-
-      // The delta is set by the excess of absolute pixel diff over the
-      // threshold.
-      int delta = ((sum_diff - sum_diff_thresh) >> 8) + 1;
-      // Only apply the adjustment for max delta up to 3.
-      if (delta < 4) {
-        const uint8x16_t k_delta = vmovq_n_u8(delta);
-        sig -= sig_stride * 16;
-        mc_running_avg_y -= mc_running_avg_y_stride * 16;
-        running_avg_y -= running_avg_y_stride * 16;
-        for (int r = 0; r < 16; ++r) {
-          uint8x16_t v_running_avg_y = vld1q_u8(running_avg_y);
-          const uint8x16_t v_sig = vld1q_u8(sig);
-          const uint8x16_t v_mc_running_avg_y = vld1q_u8(mc_running_avg_y);
-
-          // Calculate absolute difference and sign masks.
-          const uint8x16_t v_abs_diff = vabdq_u8(v_sig, v_mc_running_avg_y);
-          const uint8x16_t v_diff_pos_mask =
-              vcltq_u8(v_sig, v_mc_running_avg_y);
-          const uint8x16_t v_diff_neg_mask =
-              vcgtq_u8(v_sig, v_mc_running_avg_y);
-          // Clamp absolute difference to delta to get the adjustment.
-          const uint8x16_t v_abs_adjustment = vminq_u8(v_abs_diff, (k_delta));
-
-          const uint8x16_t v_pos_adjustment =
-              vandq_u8(v_diff_pos_mask, v_abs_adjustment);
-          const uint8x16_t v_neg_adjustment =
-              vandq_u8(v_diff_neg_mask, v_abs_adjustment);
-
-          v_running_avg_y = vqsubq_u8(v_running_avg_y, v_pos_adjustment);
-          v_running_avg_y = vqaddq_u8(v_running_avg_y, v_neg_adjustment);
-
-          // Store results.
-          vst1q_u8(running_avg_y, v_running_avg_y);
-
-          {
-            const int8x16_t v_sum_diff =
-                vqsubq_s8(vreinterpretq_s8_u8(v_neg_adjustment),
-                          vreinterpretq_s8_u8(v_pos_adjustment));
-
-            const int16x8_t fe_dc_ba_98_76_54_32_10 = vpaddlq_s8(v_sum_diff);
-            const int32x4_t fedc_ba98_7654_3210 =
-                vpaddlq_s16(fe_dc_ba_98_76_54_32_10);
-            const int64x2_t fedcba98_76543210 =
-                vpaddlq_s32(fedc_ba98_7654_3210);
-
-            v_sum_diff_total = vqaddq_s64(v_sum_diff_total, fedcba98_76543210);
-          }
-          // Update pointers for next iteration.
-          sig += sig_stride;
-          mc_running_avg_y += mc_running_avg_y_stride;
-          running_avg_y += running_avg_y_stride;
-        }
-        {
-          // Update the sum of all pixel differences of this MB.
-          x = vqadd_s64(vget_high_s64(v_sum_diff_total),
-                        vget_low_s64(v_sum_diff_total));
-          sum_diff = vget_lane_s32(vabs_s32(vreinterpret_s32_s64(x)), 0);
-
-          if (sum_diff > sum_diff_thresh) {
-            return COPY_BLOCK;
-          }
-        }
-      } else {
-        return COPY_BLOCK;
-      }
-    }
-  }
+  int64x1_t x = vqadd_s64(vget_high_s64(v_sum_diff_total),
+                          vget_low_s64(v_sum_diff_total));
+  int sum_diff = vget_lane_s32(vabs_s32(vreinterpret_s32_s64(x)), 0);
+  sum_diff_thresh =
+      increase_denoising ? kSumDiffThresholdHighNeon : kSumDiffThreshold;
+  if (sum_diff > sum_diff_thresh)
+    return COPY_BLOCK;
 
   // Tell above level that block was filtered.
   running_avg_y -= running_avg_y_stride * 16;
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.h b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.h
index 2e3ea268290..55850bd1ea5 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.h
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_neon.h
@@ -22,10 +22,6 @@ class DenoiserFilterNEON : public DenoiserFilter {
                     int src_stride,
                     uint8_t* dst,
                     int dst_stride) override;
-  void CopyMem8x8(const uint8_t* src,
-                  int src_stride,
-                  uint8_t* dst,
-                  int dst_stride) override;
   uint32_t Variance16x8(const uint8_t* a,
                         int a_stride,
                         const uint8_t* b,
@@ -38,8 +34,7 @@ class DenoiserFilterNEON : public DenoiserFilter {
                              const uint8_t* sig,
                              int sig_stride,
                              uint8_t motion_magnitude,
-                             int increase_denoising,
-                             bool denoise_always) override;
+                             int increase_denoising) override;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
index 614b6c94859..0545a97398e 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
@@ -9,7 +9,6 @@
  */
 
 #include <emmintrin.h>
-
 #include "webrtc/modules/video_processing/util/denoiser_filter_sse2.h"
 
 namespace webrtc {
@@ -110,18 +109,6 @@ void DenoiserFilterSSE2::CopyMem16x16(const uint8_t* src,
   }
 }
 
-// TODO(jackychen): Optimize this function using SSE2.
-void DenoiserFilterSSE2::CopyMem8x8(const uint8_t* src,
-                                    int src_stride,
-                                    uint8_t* dst,
-                                    int dst_stride) {
-  for (int i = 0; i < 8; i++) {
-    memcpy(dst, src, 8);
-    src += src_stride;
-    dst += dst_stride;
-  }
-}
-
 uint32_t DenoiserFilterSSE2::Variance16x8(const uint8_t* src,
                                           int src_stride,
                                           const uint8_t* ref,
@@ -139,8 +126,8 @@ DenoiserDecision DenoiserFilterSSE2::MbDenoise(uint8_t* mc_running_avg_y,
                                                const uint8_t* sig,
                                                int sig_stride,
                                                uint8_t motion_magnitude,
-                                               int increase_denoising,
-                                               bool denoise_always) {
+                                               int increase_denoising) {
+  DenoiserDecision decision = FILTER_BLOCK;
   unsigned int sum_diff_thresh = 0;
   int shift_inc =
       (increase_denoising && motion_magnitude <= kMotionMagnitudeThreshold) ? 1
@@ -210,76 +197,13 @@ DenoiserDecision DenoiserFilterSSE2::MbDenoise(uint8_t* mc_running_avg_y,
     running_avg_y += avg_y_stride;
   }
 
-  {
-    // Compute the sum of all pixel differences of this MB.
-    unsigned int abs_sum_diff = AbsSumDiff16x1(acc_diff);
-    if (denoise_always)
-      sum_diff_thresh = INT_MAX;
-    else if (increase_denoising)
-      sum_diff_thresh = kSumDiffThresholdHigh;
-    else
-      sum_diff_thresh = kSumDiffThreshold;
-    if (abs_sum_diff > sum_diff_thresh) {
-      // Before returning to copy the block (i.e., apply no denoising),
-      // check if we can still apply some (weaker) temporal filtering to
-      // this block, that would otherwise not be denoised at all. Simplest
-      // is to apply an additional adjustment to running_avg_y to bring it
-      // closer to sig. The adjustment is capped by a maximum delta, and
-      // chosen such that in most cases the resulting sum_diff will be
-      // within the acceptable range given by sum_diff_thresh.
-
-      // The delta is set by the excess of absolute pixel diff over the
-      // threshold.
-      int delta = ((abs_sum_diff - sum_diff_thresh) >> 8) + 1;
-      // Only apply the adjustment for max delta up to 3.
-      if (delta < 4) {
-        const __m128i k_delta = _mm_set1_epi8(delta);
-        sig -= sig_stride * 16;
-        mc_running_avg_y -= mc_avg_y_stride * 16;
-        running_avg_y -= avg_y_stride * 16;
-        for (int r = 0; r < 16; ++r) {
-          __m128i v_running_avg_y =
-              _mm_loadu_si128(reinterpret_cast<__m128i*>(&running_avg_y[0]));
-          // Calculate differences.
-          const __m128i v_sig =
-              _mm_loadu_si128(reinterpret_cast<const __m128i*>(&sig[0]));
-          const __m128i v_mc_running_avg_y =
-              _mm_loadu_si128(reinterpret_cast<__m128i*>(&mc_running_avg_y[0]));
-          const __m128i pdiff = _mm_subs_epu8(v_mc_running_avg_y, v_sig);
-          const __m128i ndiff = _mm_subs_epu8(v_sig, v_mc_running_avg_y);
-          // Obtain the sign. FF if diff is negative.
-          const __m128i diff_sign = _mm_cmpeq_epi8(pdiff, k_0);
-          // Clamp absolute difference to delta to get the adjustment.
-          const __m128i adj = _mm_min_epu8(_mm_or_si128(pdiff, ndiff), k_delta);
-          // Restore the sign and get positive and negative adjustments.
-          __m128i padj, nadj;
-          padj = _mm_andnot_si128(diff_sign, adj);
-          nadj = _mm_and_si128(diff_sign, adj);
-          // Calculate filtered value.
-          v_running_avg_y = _mm_subs_epu8(v_running_avg_y, padj);
-          v_running_avg_y = _mm_adds_epu8(v_running_avg_y, nadj);
-          _mm_storeu_si128(reinterpret_cast<__m128i*>(running_avg_y),
-                           v_running_avg_y);
-
-          // Accumulate the adjustments.
-          acc_diff = _mm_subs_epi8(acc_diff, padj);
-          acc_diff = _mm_adds_epi8(acc_diff, nadj);
-
-          // Update pointers for next iteration.
-          sig += sig_stride;
-          mc_running_avg_y += mc_avg_y_stride;
-          running_avg_y += avg_y_stride;
-        }
-        abs_sum_diff = AbsSumDiff16x1(acc_diff);
-        if (abs_sum_diff > sum_diff_thresh) {
-          return COPY_BLOCK;
-        }
-      } else {
-        return COPY_BLOCK;
-      }
-    }
-  }
-  return FILTER_BLOCK;
+  // Compute the sum of all pixel differences of this MB.
+  unsigned int abs_sum_diff = AbsSumDiff16x1(acc_diff);
+  sum_diff_thresh =
+      increase_denoising ? kSumDiffThresholdHigh : kSumDiffThreshold;
+  if (abs_sum_diff > sum_diff_thresh)
+    decision = COPY_BLOCK;
+  return decision;
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.h b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.h
index 395fa10eca0..731344c809c 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.h
+++ b/chromium/third_party/webrtc/modules/video_processing/util/denoiser_filter_sse2.h
@@ -22,10 +22,6 @@ class DenoiserFilterSSE2 : public DenoiserFilter {
                     int src_stride,
                     uint8_t* dst,
                     int dst_stride) override;
-  void CopyMem8x8(const uint8_t* src,
-                  int src_stride,
-                  uint8_t* dst,
-                  int dst_stride) override;
   uint32_t Variance16x8(const uint8_t* a,
                         int a_stride,
                         const uint8_t* b,
@@ -38,8 +34,7 @@ class DenoiserFilterSSE2 : public DenoiserFilter {
                              const uint8_t* sig,
                              int sig_stride,
                              uint8_t motion_magnitude,
-                             int increase_denoising,
-                             bool denoise_always) override;
+                             int increase_denoising) override;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.cc b/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.cc
index 87beac38ae5..3b0d59ef717 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.cc
@@ -9,6 +9,9 @@
  */
 
 #include "webrtc/modules/video_processing/util/noise_estimation.h"
+#if DISPLAYNEON
+#include <android/log.h>
+#endif
 
 namespace webrtc {
 
@@ -27,10 +30,10 @@ void NoiseEstimation::GetNoise(int mb_index, uint32_t var, uint32_t luma) {
   consec_low_var_[mb_index]++;
   num_static_block_++;
   if (consec_low_var_[mb_index] >= kConsecLowVarFrame &&
-      (luma >> 8) < kAverageLumaMax && (luma >> 8) > kAverageLumaMin) {
+      (luma >> 6) < kAverageLumaMax && (luma >> 6) > kAverageLumaMin) {
     // Normalized var by the average luma value, this gives more weight to
     // darker blocks.
-    int nor_var = var / (luma >> 12);
+    int nor_var = var / (luma >> 10);
     noise_var_ +=
         nor_var > kBlockSelectionVarMax ? kBlockSelectionVarMax : nor_var;
     num_noisy_block_++;
@@ -46,25 +49,36 @@ void NoiseEstimation::UpdateNoiseLevel() {
   // condition more reasonable.
   // No enough samples implies the motion of the camera or too many moving
   // objects in the frame.
-  if (num_static_block_ < (0.65 * mb_cols_ * mb_rows_) || !num_noisy_block_) {
+  if (num_static_block_ <
+          (0.65 * mb_cols_ * mb_rows_ / NOISE_SUBSAMPLE_INTERVAL) ||
+      !num_noisy_block_) {
+#if DISPLAY
+    printf("Not enough samples. %d \n", num_static_block_);
+#elif DISPLAYNEON
+    __android_log_print(ANDROID_LOG_DEBUG, "DISPLAY",
+                        "Not enough samples. %d \n", num_static_block_);
+#endif
     noise_var_ = 0;
     noise_var_accum_ = 0;
-    num_static_block_ = 0;
     num_noisy_block_ = 0;
-#if DISPLAY
-    printf("Not enough samples.\n");
-#endif
+    num_static_block_ = 0;
     return;
   } else {
-    // Normalized by the number of noisy blocks.
-    noise_var_ /= num_noisy_block_;
-    // Get the percentage of static blocks.
-    percent_static_block_ =
-        static_cast<double>(num_static_block_) / (mb_cols_ * mb_rows_);
 #if DISPLAY
-    printf("%d %d fraction = %.3f\n", num_static_block_, mb_cols_ * mb_rows_,
+    printf("%d %d fraction = %.3f\n", num_static_block_,
+           mb_cols_ * mb_rows_ / NOISE_SUBSAMPLE_INTERVAL,
            percent_static_block_);
+#elif DISPLAYNEON
+    __android_log_print(ANDROID_LOG_DEBUG, "DISPLAY", "%d %d fraction = %.3f\n",
+                        num_static_block_,
+                        mb_cols_ * mb_rows_ / NOISE_SUBSAMPLE_INTERVAL,
+                        percent_static_block_);
 #endif
+    // Normalized by the number of noisy blocks.
+    noise_var_ /= num_noisy_block_;
+    // Get the percentage of static blocks.
+    percent_static_block_ = static_cast<double>(num_static_block_) /
+                            (mb_cols_ * mb_rows_ / NOISE_SUBSAMPLE_INTERVAL);
     num_noisy_block_ = 0;
     num_static_block_ = 0;
   }
@@ -75,12 +89,16 @@ void NoiseEstimation::UpdateNoiseLevel() {
   } else {
     noise_var_accum_ = (noise_var_accum_ * 15 + noise_var_) / 16;
   }
-  // Reset noise_var_ for the next frame.
-  noise_var_ = 0;
 #if DISPLAY
   printf("noise_var_accum_ = %.1f, noise_var_ = %d.\n", noise_var_accum_,
          noise_var_);
+#elif DISPLAYNEON
+  __android_log_print(ANDROID_LOG_DEBUG, "DISPLAY",
+                      "noise_var_accum_ = %.1f, noise_var_ = %d.\n",
+                      noise_var_accum_, noise_var_);
 #endif
+  // Reset noise_var_ for the next frame.
+  noise_var_ = 0;
 }
 
 uint8_t NoiseEstimation::GetNoiseLevel() {
diff --git a/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.h b/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.h
index ca5cc2324fb..294bfb3a731 100644
--- a/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.h
+++ b/chromium/third_party/webrtc/modules/video_processing/util/noise_estimation.h
@@ -11,28 +11,36 @@
 #ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_NOISE_ESTIMATION_H_
 #define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_NOISE_ESTIMATION_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_processing/include/video_processing_defines.h"
 #include "webrtc/modules/video_processing/util/denoiser_filter.h"
 
 namespace webrtc {
 
-#define EXPERIMENTAL 0
-#define DISPLAY 0
+#define DISPLAY 0      // Rectangle diagnostics
+#define DISPLAYNEON 0  // Rectangle diagnostics on NEON
 
-const int kNoiseThreshold = 200;
+const int kNoiseThreshold = 150;
 const int kNoiseThresholdNeon = 70;
 const int kConsecLowVarFrame = 6;
 const int kAverageLumaMin = 20;
 const int kAverageLumaMax = 220;
 const int kBlockSelectionVarMax = kNoiseThreshold << 1;
 
+// TODO(jackychen): To test different sampling strategy.
+// Collect noise data every NOISE_SUBSAMPLE_INTERVAL blocks.
+#define NOISE_SUBSAMPLE_INTERVAL 41
+
 class NoiseEstimation {
  public:
   void Init(int width, int height, CpuType cpu_type);
+  // Collect noise data from one qualified block.
   void GetNoise(int mb_index, uint32_t var, uint32_t luma);
+  // Reset the counter for consecutive low-var blocks.
   void ResetConsecLowVar(int mb_index);
+  // Update noise level for current frame.
   void UpdateNoiseLevel();
   // 0: low noise, 1: high noise
   uint8_t GetNoiseLevel();
@@ -42,13 +50,13 @@ class NoiseEstimation {
   int height_;
   int mb_rows_;
   int mb_cols_;
+  int num_noisy_block_;
+  int num_static_block_;
   CpuType cpu_type_;
   uint32_t noise_var_;
   double noise_var_accum_;
-  int num_noisy_block_;
-  int num_static_block_;
   double percent_static_block_;
-  rtc::scoped_ptr<uint32_t[]> consec_low_var_;
+  std::unique_ptr<uint32_t[]> consec_low_var_;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/video_decimator.cc b/chromium/third_party/webrtc/modules/video_processing/video_decimator.cc
index 63e347b026e..c6623fa836c 100644
--- a/chromium/third_party/webrtc/modules/video_processing/video_decimator.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/video_decimator.cc
@@ -9,9 +9,9 @@
  */
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/modules/video_processing/video_decimator.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 #define VD_MIN(a, b) ((a) < (b)) ? (a) : (b)
 
@@ -95,7 +95,7 @@ bool VPMVideoDecimator::DropFrame() {
 }
 
 uint32_t VPMVideoDecimator::GetDecimatedFrameRate() {
-  ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
+  ProcessIncomingframe_rate(rtc::TimeMillis());
   if (!enable_temporal_decimation_) {
     return static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
   }
@@ -104,12 +104,12 @@ uint32_t VPMVideoDecimator::GetDecimatedFrameRate() {
 }
 
 uint32_t VPMVideoDecimator::Inputframe_rate() {
-  ProcessIncomingframe_rate(TickTime::MillisecondTimestamp());
+  ProcessIncomingframe_rate(rtc::TimeMillis());
   return static_cast<uint32_t>(incoming_frame_rate_ + 0.5f);
 }
 
 void VPMVideoDecimator::UpdateIncomingframe_rate() {
-  int64_t now = TickTime::MillisecondTimestamp();
+  int64_t now = rtc::TimeMillis();
   if (incoming_frame_times_[0] == 0) {
     // First no shift.
   } else {
diff --git a/chromium/third_party/webrtc/modules/video_processing/video_denoiser.cc b/chromium/third_party/webrtc/modules/video_processing/video_denoiser.cc
index b00da5c90a1..f116f882cd6 100644
--- a/chromium/third_party/webrtc/modules/video_processing/video_denoiser.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/video_denoiser.cc
@@ -7,305 +7,347 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+
 #include "webrtc/common_video/libyuv/include/scaler.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_processing/video_denoiser.h"
 
 namespace webrtc {
 
+#if DISPLAY || DISPLAYNEON
+static void CopyMem8x8(const uint8_t* src,
+                       int src_stride,
+                       uint8_t* dst,
+                       int dst_stride) {
+  for (int i = 0; i < 8; i++) {
+    memcpy(dst, src, 8);
+    src += src_stride;
+    dst += dst_stride;
+  }
+}
+
+static void ShowRect(const std::unique_ptr<DenoiserFilter>& filter,
+                     const std::unique_ptr<uint8_t[]>& d_status,
+                     const std::unique_ptr<uint8_t[]>& moving_edge_red,
+                     const std::unique_ptr<uint8_t[]>& x_density,
+                     const std::unique_ptr<uint8_t[]>& y_density,
+                     const uint8_t* u_src,
+                     const uint8_t* v_src,
+                     uint8_t* u_dst,
+                     uint8_t* v_dst,
+                     int mb_rows_,
+                     int mb_cols_,
+                     int stride_u_,
+                     int stride_v_) {
+  for (int mb_row = 0; mb_row < mb_rows_; ++mb_row) {
+    for (int mb_col = 0; mb_col < mb_cols_; ++mb_col) {
+      int mb_index = mb_row * mb_cols_ + mb_col;
+      const uint8_t* mb_src_u =
+          u_src + (mb_row << 3) * stride_u_ + (mb_col << 3);
+      const uint8_t* mb_src_v =
+          v_src + (mb_row << 3) * stride_v_ + (mb_col << 3);
+      uint8_t* mb_dst_u = u_dst + (mb_row << 3) * stride_u_ + (mb_col << 3);
+      uint8_t* mb_dst_v = v_dst + (mb_row << 3) * stride_v_ + (mb_col << 3);
+      uint8_t uv_tmp[8 * 8];
+      memset(uv_tmp, 200, 8 * 8);
+      if (d_status[mb_index] == 1) {
+        // Paint to red.
+        CopyMem8x8(mb_src_u, stride_u_, mb_dst_u, stride_u_);
+        CopyMem8x8(uv_tmp, 8, mb_dst_v, stride_v_);
+      } else if (moving_edge_red[mb_row * mb_cols_ + mb_col] &&
+                 x_density[mb_col] * y_density[mb_row]) {
+        // Paint to blue.
+        CopyMem8x8(uv_tmp, 8, mb_dst_u, stride_u_);
+        CopyMem8x8(mb_src_v, stride_v_, mb_dst_v, stride_v_);
+      } else {
+        CopyMem8x8(mb_src_u, stride_u_, mb_dst_u, stride_u_);
+        CopyMem8x8(mb_src_v, stride_v_, mb_dst_v, stride_v_);
+      }
+    }
+  }
+}
+#endif
+
 VideoDenoiser::VideoDenoiser(bool runtime_cpu_detection)
     : width_(0),
       height_(0),
       filter_(DenoiserFilter::Create(runtime_cpu_detection, &cpu_type_)),
       ne_(new NoiseEstimation()) {}
 
-#if EXPERIMENTAL
-// Check the mb position(1: close to the center, 3: close to the border).
-static int PositionCheck(int mb_row, int mb_col, int mb_rows, int mb_cols) {
-  if ((mb_row >= (mb_rows >> 3)) && (mb_row <= (7 * mb_rows >> 3)) &&
-      (mb_col >= (mb_cols >> 3)) && (mb_col <= (7 * mb_cols >> 3)))
+void VideoDenoiser::DenoiserReset(const VideoFrame& frame,
+                                  VideoFrame* denoised_frame,
+                                  VideoFrame* denoised_frame_prev) {
+  width_ = frame.width();
+  height_ = frame.height();
+  mb_cols_ = width_ >> 4;
+  mb_rows_ = height_ >> 4;
+  stride_y_ = frame.video_frame_buffer()->StrideY();
+  stride_u_ = frame.video_frame_buffer()->StrideU();
+  stride_v_ = frame.video_frame_buffer()->StrideV();
+
+  // Allocate an empty buffer for denoised_frame_prev.
+  denoised_frame_prev->CreateEmptyFrame(width_, height_, stride_y_, stride_u_,
+                                        stride_v_);
+  // Allocate and initialize denoised_frame with key frame.
+  denoised_frame->CreateFrame(
+      frame.video_frame_buffer()->DataY(),
+      frame.video_frame_buffer()->DataU(),
+      frame.video_frame_buffer()->DataV(),
+      width_, height_, stride_y_, stride_u_, stride_v_, kVideoRotation_0);
+  // Set time parameters to the output frame.
+  denoised_frame->set_timestamp(frame.timestamp());
+  denoised_frame->set_render_time_ms(frame.render_time_ms());
+
+  // Init noise estimator and allocate buffers.
+  ne_->Init(width_, height_, cpu_type_);
+  moving_edge_.reset(new uint8_t[mb_cols_ * mb_rows_]);
+  mb_filter_decision_.reset(new DenoiserDecision[mb_cols_ * mb_rows_]);
+  x_density_.reset(new uint8_t[mb_cols_]);
+  y_density_.reset(new uint8_t[mb_rows_]);
+  moving_object_.reset(new uint8_t[mb_cols_ * mb_rows_]);
+}
+
+int VideoDenoiser::PositionCheck(int mb_row, int mb_col, int noise_level) {
+  if (noise_level == 0)
     return 1;
-  else if ((mb_row >= (mb_rows >> 4)) && (mb_row <= (15 * mb_rows >> 4)) &&
-           (mb_col >= (mb_cols >> 4)) && (mb_col <= (15 * mb_cols >> 4)))
+  if ((mb_row <= (mb_rows_ >> 4)) || (mb_col <= (mb_cols_ >> 4)) ||
+      (mb_col >= (15 * mb_cols_ >> 4)))
+    return 3;
+  else if ((mb_row <= (mb_rows_ >> 3)) || (mb_col <= (mb_cols_ >> 3)) ||
+           (mb_col >= (7 * mb_cols_ >> 3)))
     return 2;
   else
-    return 3;
+    return 1;
 }
 
-static void ReduceFalseDetection(const std::unique_ptr<uint8_t[]>& d_status,
-                                 std::unique_ptr<uint8_t[]>* d_status_tmp1,
-                                 std::unique_ptr<uint8_t[]>* d_status_tmp2,
-                                 int noise_level,
-                                 int mb_rows,
-                                 int mb_cols) {
-  // Draft. This can be optimized. This code block is to reduce false detection
-  // in moving object detection.
-  int mb_row_min = noise_level ? mb_rows >> 3 : 1;
-  int mb_col_min = noise_level ? mb_cols >> 3 : 1;
-  int mb_row_max = noise_level ? (7 * mb_rows >> 3) : mb_rows - 2;
-  int mb_col_max = noise_level ? (7 * mb_cols >> 3) : mb_cols - 2;
-  memcpy((*d_status_tmp1).get(), d_status.get(), mb_rows * mb_cols);
-  // Up left.
-  for (int mb_row = mb_row_min; mb_row <= mb_row_max; ++mb_row) {
-    for (int mb_col = mb_col_min; mb_col <= mb_col_max; ++mb_col) {
-      (*d_status_tmp1)[mb_row * mb_cols + mb_col] |=
-          ((*d_status_tmp1)[(mb_row - 1) * mb_cols + mb_col] |
-           (*d_status_tmp1)[mb_row * mb_cols + mb_col - 1]);
+void VideoDenoiser::ReduceFalseDetection(
+    const std::unique_ptr<uint8_t[]>& d_status,
+    std::unique_ptr<uint8_t[]>* moving_edge_red,
+    int noise_level) {
+  // From up left corner.
+  int mb_col_stop = mb_cols_ - 1;
+  for (int mb_row = 0; mb_row <= mb_rows_ - 1; ++mb_row) {
+    for (int mb_col = 0; mb_col <= mb_col_stop; ++mb_col) {
+      if (d_status[mb_row * mb_cols_ + mb_col]) {
+        mb_col_stop = mb_col - 1;
+        break;
+      }
+      (*moving_edge_red)[mb_row * mb_cols_ + mb_col] = 0;
     }
   }
-  memcpy((*d_status_tmp2).get(), (*d_status_tmp1).get(), mb_rows * mb_cols);
-  memcpy((*d_status_tmp1).get(), d_status.get(), mb_rows * mb_cols);
-  // Bottom left.
-  for (int mb_row = mb_row_max; mb_row >= mb_row_min; --mb_row) {
-    for (int mb_col = mb_col_min; mb_col <= mb_col_max; ++mb_col) {
-      (*d_status_tmp1)[mb_row * mb_cols + mb_col] |=
-          ((*d_status_tmp1)[(mb_row + 1) * mb_cols + mb_col] |
-           (*d_status_tmp1)[mb_row * mb_cols + mb_col - 1]);
-      (*d_status_tmp2)[mb_row * mb_cols + mb_col] &=
-          (*d_status_tmp1)[mb_row * mb_cols + mb_col];
+  // From bottom left corner.
+  mb_col_stop = mb_cols_ - 1;
+  for (int mb_row = mb_rows_ - 1; mb_row >= 0; --mb_row) {
+    for (int mb_col = 0; mb_col <= mb_col_stop; ++mb_col) {
+      if (d_status[mb_row * mb_cols_ + mb_col]) {
+        mb_col_stop = mb_col - 1;
+        break;
+      }
+      (*moving_edge_red)[mb_row * mb_cols_ + mb_col] = 0;
     }
   }
-  memcpy((*d_status_tmp1).get(), d_status.get(), mb_rows * mb_cols);
-  // Up right.
-  for (int mb_row = mb_row_min; mb_row <= mb_row_max; ++mb_row) {
-    for (int mb_col = mb_col_max; mb_col >= mb_col_min; --mb_col) {
-      (*d_status_tmp1)[mb_row * mb_cols + mb_col] |=
-          ((*d_status_tmp1)[(mb_row - 1) * mb_cols + mb_col] |
-           (*d_status_tmp1)[mb_row * mb_cols + mb_col + 1]);
-      (*d_status_tmp2)[mb_row * mb_cols + mb_col] &=
-          (*d_status_tmp1)[mb_row * mb_cols + mb_col];
+  // From up right corner.
+  mb_col_stop = 0;
+  for (int mb_row = 0; mb_row <= mb_rows_ - 1; ++mb_row) {
+    for (int mb_col = mb_cols_ - 1; mb_col >= mb_col_stop; --mb_col) {
+      if (d_status[mb_row * mb_cols_ + mb_col]) {
+        mb_col_stop = mb_col + 1;
+        break;
+      }
+      (*moving_edge_red)[mb_row * mb_cols_ + mb_col] = 0;
     }
   }
-  memcpy((*d_status_tmp1).get(), d_status.get(), mb_rows * mb_cols);
-  // Bottom right.
-  for (int mb_row = mb_row_max; mb_row >= mb_row_min; --mb_row) {
-    for (int mb_col = mb_col_max; mb_col >= mb_col_min; --mb_col) {
-      (*d_status_tmp1)[mb_row * mb_cols + mb_col] |=
-          ((*d_status_tmp1)[(mb_row + 1) * mb_cols + mb_col] |
-           (*d_status_tmp1)[mb_row * mb_cols + mb_col + 1]);
-      (*d_status_tmp2)[mb_row * mb_cols + mb_col] &=
-          (*d_status_tmp1)[mb_row * mb_cols + mb_col];
+  // From bottom right corner.
+  mb_col_stop = 0;
+  for (int mb_row = mb_rows_ - 1; mb_row >= 0; --mb_row) {
+    for (int mb_col = mb_cols_ - 1; mb_col >= mb_col_stop; --mb_col) {
+      if (d_status[mb_row * mb_cols_ + mb_col]) {
+        mb_col_stop = mb_col + 1;
+        break;
+      }
+      (*moving_edge_red)[mb_row * mb_cols_ + mb_col] = 0;
     }
   }
 }
 
-static bool TrailingBlock(const std::unique_ptr<uint8_t[]>& d_status,
-                          int mb_row,
-                          int mb_col,
-                          int mb_rows,
-                          int mb_cols) {
-  int mb_index = mb_row * mb_cols + mb_col;
-  if (!mb_row || !mb_col || mb_row == mb_rows - 1 || mb_col == mb_cols - 1)
-    return false;
-  return d_status[mb_index + 1] || d_status[mb_index - 1] ||
-         d_status[mb_index + mb_cols] || d_status[mb_index - mb_cols];
+bool VideoDenoiser::IsTrailingBlock(const std::unique_ptr<uint8_t[]>& d_status,
+                                    int mb_row,
+                                    int mb_col) {
+  bool ret = false;
+  int mb_index = mb_row * mb_cols_ + mb_col;
+  if (!mb_row || !mb_col || mb_row == mb_rows_ - 1 || mb_col == mb_cols_ - 1)
+    ret = false;
+  else
+    ret = d_status[mb_index + 1] || d_status[mb_index - 1] ||
+          d_status[mb_index + mb_cols_] || d_status[mb_index - mb_cols_];
+  return ret;
 }
-#endif
 
-#if DISPLAY
-void ShowRect(const std::unique_ptr<DenoiserFilter>& filter,
-              const std::unique_ptr<uint8_t[]>& d_status,
-              const std::unique_ptr<uint8_t[]>& d_status_tmp2,
-              const std::unique_ptr<uint8_t[]>& x_density,
-              const std::unique_ptr<uint8_t[]>& y_density,
-              const uint8_t* u_src,
-              const uint8_t* v_src,
-              uint8_t* u_dst,
-              uint8_t* v_dst,
-              int mb_rows,
-              int mb_cols,
-              int stride_u,
-              int stride_v) {
-  for (int mb_row = 0; mb_row < mb_rows; ++mb_row) {
-    for (int mb_col = 0; mb_col < mb_cols; ++mb_col) {
-      int mb_index = mb_row * mb_cols + mb_col;
-      const uint8_t* mb_src_u =
-          u_src + (mb_row << 3) * stride_u + (mb_col << 3);
-      const uint8_t* mb_src_v =
-          v_src + (mb_row << 3) * stride_v + (mb_col << 3);
-      uint8_t* mb_dst_u = u_dst + (mb_row << 3) * stride_u + (mb_col << 3);
-      uint8_t* mb_dst_v = v_dst + (mb_row << 3) * stride_v + (mb_col << 3);
-      uint8_t y_tmp_255[8 * 8];
-      memset(y_tmp_255, 200, 8 * 8);
-      // x_density_[mb_col] * y_density_[mb_row]
-      if (d_status[mb_index] == 1) {
-        // Paint to red.
-        filter->CopyMem8x8(mb_src_u, stride_u, mb_dst_u, stride_u);
-        filter->CopyMem8x8(y_tmp_255, 8, mb_dst_v, stride_v);
-#if EXPERIMENTAL
-      } else if (d_status_tmp2[mb_row * mb_cols + mb_col] &&
-                 x_density[mb_col] * y_density[mb_row]) {
-#else
-      } else if (x_density[mb_col] * y_density[mb_row]) {
-#endif
-        // Paint to blue.
-        filter->CopyMem8x8(y_tmp_255, 8, mb_dst_u, stride_u);
-        filter->CopyMem8x8(mb_src_v, stride_v, mb_dst_v, stride_v);
-      } else {
-        filter->CopyMem8x8(mb_src_u, stride_u, mb_dst_u, stride_u);
-        filter->CopyMem8x8(mb_src_v, stride_v, mb_dst_v, stride_v);
+void VideoDenoiser::CopySrcOnMOB(const uint8_t* y_src, uint8_t* y_dst) {
+  // Loop over to copy src block if the block is marked as moving object block
+  // or if the block may cause trailing artifacts.
+  for (int mb_row = 0; mb_row < mb_rows_; ++mb_row) {
+    const int mb_index_base = mb_row * mb_cols_;
+    const int offset_base = (mb_row << 4) * stride_y_;
+    const uint8_t* mb_src_base = y_src + offset_base;
+    uint8_t* mb_dst_base = y_dst + offset_base;
+    for (int mb_col = 0; mb_col < mb_cols_; ++mb_col) {
+      const int mb_index = mb_index_base + mb_col;
+      const uint32_t offset_col = mb_col << 4;
+      const uint8_t* mb_src = mb_src_base + offset_col;
+      uint8_t* mb_dst = mb_dst_base + offset_col;
+      // Check if the block is a moving object block or may cause a trailing
+      // artifacts.
+      if (mb_filter_decision_[mb_index] != FILTER_BLOCK ||
+          IsTrailingBlock(moving_edge_, mb_row, mb_col) ||
+          (x_density_[mb_col] * y_density_[mb_row] &&
+           moving_object_[mb_row * mb_cols_ + mb_col])) {
+        // Copy y source.
+        filter_->CopyMem16x16(mb_src, stride_y_, mb_dst, stride_y_);
+      }
+    }
+  }
+}
+
+void VideoDenoiser::CopyLumaOnMargin(const uint8_t* y_src, uint8_t* y_dst) {
+  if ((mb_rows_ << 4) != height_) {
+    const uint8_t* margin_y_src = y_src + (mb_rows_ << 4) * stride_y_;
+    uint8_t* margin_y_dst = y_dst + (mb_rows_ << 4) * stride_y_;
+    memcpy(margin_y_dst, margin_y_src, (height_ - (mb_rows_ << 4)) * stride_y_);
+  }
+  if ((mb_cols_ << 4) != width_) {
+    const uint8_t* margin_y_src = y_src + (mb_cols_ << 4);
+    uint8_t* margin_y_dst = y_dst + (mb_cols_ << 4);
+    for (int i = 0; i < height_; ++i) {
+      for (int j = mb_cols_ << 4; j < width_; ++j) {
+        margin_y_dst[i * stride_y_ + j] = margin_y_src[i * stride_y_ + j];
       }
     }
   }
 }
-#endif
 
 void VideoDenoiser::DenoiseFrame(const VideoFrame& frame,
                                  VideoFrame* denoised_frame,
                                  VideoFrame* denoised_frame_prev,
-                                 int noise_level_prev) {
-  int stride_y = frame.stride(kYPlane);
-  int stride_u = frame.stride(kUPlane);
-  int stride_v = frame.stride(kVPlane);
-  // If previous width and height are different from current frame's, then no
-  // denoising for the current frame.
+                                 bool noise_estimation_enabled) {
+  // If previous width and height are different from current frame's, need to
+  // reallocate the buffers and no denoising for the current frame.
   if (width_ != frame.width() || height_ != frame.height()) {
-    width_ = frame.width();
-    height_ = frame.height();
-    denoised_frame->CreateFrame(frame.buffer(kYPlane), frame.buffer(kUPlane),
-                                frame.buffer(kVPlane), width_, height_,
-                                stride_y, stride_u, stride_v, kVideoRotation_0);
-    denoised_frame_prev->CreateFrame(
-        frame.buffer(kYPlane), frame.buffer(kUPlane), frame.buffer(kVPlane),
-        width_, height_, stride_y, stride_u, stride_v, kVideoRotation_0);
-    // Setting time parameters to the output frame.
-    denoised_frame->set_timestamp(frame.timestamp());
-    denoised_frame->set_render_time_ms(frame.render_time_ms());
-    ne_->Init(width_, height_, cpu_type_);
+    DenoiserReset(frame, denoised_frame, denoised_frame_prev);
     return;
   }
-  // For 16x16 block.
-  int mb_cols = width_ >> 4;
-  int mb_rows = height_ >> 4;
-  if (metrics_.get() == nullptr)
-    metrics_.reset(new DenoiseMetrics[mb_cols * mb_rows]());
-  if (d_status_.get() == nullptr) {
-    d_status_.reset(new uint8_t[mb_cols * mb_rows]());
-#if EXPERIMENTAL
-    d_status_tmp1_.reset(new uint8_t[mb_cols * mb_rows]());
-    d_status_tmp2_.reset(new uint8_t[mb_cols * mb_rows]());
-#endif
-    x_density_.reset(new uint8_t[mb_cols]());
-    y_density_.reset(new uint8_t[mb_rows]());
-  }
 
-  // Denoise on Y plane.
-  uint8_t* y_dst = denoised_frame->buffer(kYPlane);
-  uint8_t* u_dst = denoised_frame->buffer(kUPlane);
-  uint8_t* v_dst = denoised_frame->buffer(kVPlane);
-  uint8_t* y_dst_prev = denoised_frame_prev->buffer(kYPlane);
-  const uint8_t* y_src = frame.buffer(kYPlane);
-  const uint8_t* u_src = frame.buffer(kUPlane);
-  const uint8_t* v_src = frame.buffer(kVPlane);
-  uint8_t noise_level = noise_level_prev == -1 ? 0 : ne_->GetNoiseLevel();
-  // Temporary buffer to store denoising result.
-  uint8_t y_tmp[16 * 16] = {0};
-  memset(x_density_.get(), 0, mb_cols);
-  memset(y_density_.get(), 0, mb_rows);
+  // Set buffer pointers.
+  const uint8_t* y_src = frame.video_frame_buffer()->DataY();
+  const uint8_t* u_src = frame.video_frame_buffer()->DataU();
+  const uint8_t* v_src = frame.video_frame_buffer()->DataV();
+  uint8_t* y_dst = denoised_frame->video_frame_buffer()->MutableDataY();
+  uint8_t* u_dst = denoised_frame->video_frame_buffer()->MutableDataU();
+  uint8_t* v_dst = denoised_frame->video_frame_buffer()->MutableDataV();
+  uint8_t* y_dst_prev =
+      denoised_frame_prev->video_frame_buffer()->MutableDataY();
+  memset(x_density_.get(), 0, mb_cols_);
+  memset(y_density_.get(), 0, mb_rows_);
+  memset(moving_object_.get(), 1, mb_cols_ * mb_rows_);
 
+  uint8_t noise_level = noise_estimation_enabled ? ne_->GetNoiseLevel() : 0;
+  int thr_var_base = 16 * 16 * 2;
   // Loop over blocks to accumulate/extract noise level and update x/y_density
   // factors for moving object detection.
-  for (int mb_row = 0; mb_row < mb_rows; ++mb_row) {
-    for (int mb_col = 0; mb_col < mb_cols; ++mb_col) {
-      const uint8_t* mb_src = y_src + (mb_row << 4) * stride_y + (mb_col << 4);
-      uint8_t* mb_dst_prev =
-          y_dst_prev + (mb_row << 4) * stride_y + (mb_col << 4);
-      int mb_index = mb_row * mb_cols + mb_col;
-#if EXPERIMENTAL
-      int pos_factor = PositionCheck(mb_row, mb_col, mb_rows, mb_cols);
-      uint32_t thr_var_adp = 16 * 16 * 5 * (noise_level ? pos_factor : 1);
-#else
-      uint32_t thr_var_adp = 16 * 16 * 5;
-#endif
-      int brightness = 0;
-      for (int i = 0; i < 16; ++i) {
-        for (int j = 0; j < 16; ++j) {
-          brightness += mb_src[i * stride_y + j];
+  for (int mb_row = 0; mb_row < mb_rows_; ++mb_row) {
+    const int mb_index_base = mb_row * mb_cols_;
+    const int offset_base = (mb_row << 4) * stride_y_;
+    const uint8_t* mb_src_base = y_src + offset_base;
+    uint8_t* mb_dst_base = y_dst + offset_base;
+    uint8_t* mb_dst_prev_base = y_dst_prev + offset_base;
+    for (int mb_col = 0; mb_col < mb_cols_; ++mb_col) {
+      const int mb_index = mb_index_base + mb_col;
+      const bool ne_enable = (mb_index % NOISE_SUBSAMPLE_INTERVAL == 0);
+      const int pos_factor = PositionCheck(mb_row, mb_col, noise_level);
+      const uint32_t thr_var_adp = thr_var_base * pos_factor;
+      const uint32_t offset_col = mb_col << 4;
+      const uint8_t* mb_src = mb_src_base + offset_col;
+      uint8_t* mb_dst = mb_dst_base + offset_col;
+      uint8_t* mb_dst_prev = mb_dst_prev_base + offset_col;
+
+      // TODO(jackychen): Need SSE2/NEON opt.
+      int luma = 0;
+      if (ne_enable) {
+        for (int i = 4; i < 12; ++i) {
+          for (int j = 4; j < 12; ++j) {
+            luma += mb_src[i * stride_y_ + j];
+          }
         }
       }
 
-      // Get the denoised block.
-      filter_->MbDenoise(mb_dst_prev, stride_y, y_tmp, 16, mb_src, stride_y, 0,
-                         1, true);
-      // The variance is based on the denoised blocks in time T and T-1.
-      metrics_[mb_index].var = filter_->Variance16x8(
-          mb_dst_prev, stride_y, y_tmp, 16, &metrics_[mb_index].sad);
+      // Get the filtered block and filter_decision.
+      mb_filter_decision_[mb_index] =
+          filter_->MbDenoise(mb_dst_prev, stride_y_, mb_dst, stride_y_, mb_src,
+                             stride_y_, 0, noise_level);
 
-      if (metrics_[mb_index].var > thr_var_adp) {
-        ne_->ResetConsecLowVar(mb_index);
-        d_status_[mb_index] = 1;
-#if EXPERIMENTAL
-        if (noise_level == 0 || pos_factor < 3) {
-          x_density_[mb_col] += 1;
-          y_density_[mb_row] += 1;
+      // If filter decision is FILTER_BLOCK, no need to check moving edge.
+      // It is unlikely for a moving edge block to be filtered in current
+      // setting.
+      if (mb_filter_decision_[mb_index] == FILTER_BLOCK) {
+        uint32_t sse_t = 0;
+        if (ne_enable) {
+          // The variance used in noise estimation is based on the src block in
+          // time t (mb_src) and filtered block in time t-1 (mb_dist_prev).
+          uint32_t noise_var = filter_->Variance16x8(mb_dst_prev, stride_y_,
+                                                     mb_src, stride_y_, &sse_t);
+          ne_->GetNoise(mb_index, noise_var, luma);
         }
-#else
-        x_density_[mb_col] += 1;
-        y_density_[mb_row] += 1;
-#endif
+        moving_edge_[mb_index] = 0;  // Not a moving edge block.
       } else {
         uint32_t sse_t = 0;
-        // The variance is based on the src blocks in time T and denoised block
-        // in time T-1.
-        uint32_t noise_var = filter_->Variance16x8(mb_dst_prev, stride_y,
-                                                   mb_src, stride_y, &sse_t);
-        ne_->GetNoise(mb_index, noise_var, brightness);
-        d_status_[mb_index] = 0;
-      }
-      // Track denoised frame.
-      filter_->CopyMem16x16(y_tmp, 16, mb_dst_prev, stride_y);
-    }
-  }
-
-#if EXPERIMENTAL
-  ReduceFalseDetection(d_status_, &d_status_tmp1_, &d_status_tmp2_, noise_level,
-                       mb_rows, mb_cols);
-#endif
-
-  // Denoise each MB based on the results of moving objects detection.
-  for (int mb_row = 0; mb_row < mb_rows; ++mb_row) {
-    for (int mb_col = 0; mb_col < mb_cols; ++mb_col) {
-      const uint8_t* mb_src = y_src + (mb_row << 4) * stride_y + (mb_col << 4);
-      uint8_t* mb_dst = y_dst + (mb_row << 4) * stride_y + (mb_col << 4);
-      const uint8_t* mb_src_u =
-          u_src + (mb_row << 3) * stride_u + (mb_col << 3);
-      const uint8_t* mb_src_v =
-          v_src + (mb_row << 3) * stride_v + (mb_col << 3);
-      uint8_t* mb_dst_u = u_dst + (mb_row << 3) * stride_u + (mb_col << 3);
-      uint8_t* mb_dst_v = v_dst + (mb_row << 3) * stride_v + (mb_col << 3);
-#if EXPERIMENTAL
-      if ((!d_status_tmp2_[mb_row * mb_cols + mb_col] ||
-           x_density_[mb_col] * y_density_[mb_row] == 0) &&
-          !TrailingBlock(d_status_, mb_row, mb_col, mb_rows, mb_cols)) {
-#else
-      if (x_density_[mb_col] * y_density_[mb_row] == 0) {
-#endif
-        if (filter_->MbDenoise(mb_dst, stride_y, y_tmp, 16, mb_src, stride_y, 0,
-                               noise_level, false) == FILTER_BLOCK) {
-          filter_->CopyMem16x16(y_tmp, 16, mb_dst, stride_y);
+        // The variance used in MOD is based on the filtered blocks in time
+        // T (mb_dst) and T-1 (mb_dst_prev).
+        uint32_t noise_var = filter_->Variance16x8(mb_dst_prev, stride_y_,
+                                                   mb_dst, stride_y_, &sse_t);
+        if (noise_var > thr_var_adp) {  // Moving edge checking.
+          if (ne_enable) {
+            ne_->ResetConsecLowVar(mb_index);
+          }
+          moving_edge_[mb_index] = 1;  // Mark as moving edge block.
+          x_density_[mb_col] += (pos_factor < 3);
+          y_density_[mb_row] += (pos_factor < 3);
         } else {
-          // Copy y source.
-          filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
+          moving_edge_[mb_index] = 0;
+          if (ne_enable) {
+            // The variance used in noise estimation is based on the src block
+            // in time t (mb_src) and filtered block in time t-1 (mb_dist_prev).
+            uint32_t noise_var = filter_->Variance16x8(
+                mb_dst_prev, stride_y_, mb_src, stride_y_, &sse_t);
+            ne_->GetNoise(mb_index, noise_var, luma);
+          }
         }
-      } else {
-        // Copy y source.
-        filter_->CopyMem16x16(mb_src, stride_y, mb_dst, stride_y);
       }
-      filter_->CopyMem8x8(mb_src_u, stride_u, mb_dst_u, stride_u);
-      filter_->CopyMem8x8(mb_src_v, stride_v, mb_dst_v, stride_v);
-    }
-  }
+    }  // End of for loop
+  }    // End of for loop
 
-#if DISPLAY  // Rectangle diagnostics
-  // Show rectangular region
-  ShowRect(filter_, d_status_, d_status_tmp2_, x_density_, y_density_, u_src,
-           v_src, u_dst, v_dst, mb_rows, mb_cols, stride_u, stride_v);
-#endif
+  ReduceFalseDetection(moving_edge_, &moving_object_, noise_level);
+
+  CopySrcOnMOB(y_src, y_dst);
+
+  // When frame width/height not divisible by 16, copy the margin to
+  // denoised_frame.
+  if ((mb_rows_ << 4) != height_ || (mb_cols_ << 4) != width_)
+    CopyLumaOnMargin(y_src, y_dst);
 
-  // Setting time parameters to the output frame.
+  // TODO(jackychen): Need SSE2/NEON opt.
+  // Copy u/v planes.
+  memcpy(u_dst, u_src, (height_ >> 1) * stride_u_);
+  memcpy(v_dst, v_src, (height_ >> 1) * stride_v_);
+
+  // Set time parameters to the output frame.
   denoised_frame->set_timestamp(frame.timestamp());
   denoised_frame->set_render_time_ms(frame.render_time_ms());
-  return;
+
+#if DISPLAY || DISPLAYNEON
+  // Show rectangular region
+  ShowRect(filter_, moving_edge_, moving_object_, x_density_, y_density_, u_src,
+           v_src, u_dst, v_dst, mb_rows_, mb_cols_, stride_u_, stride_v_);
+#endif
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/video_denoiser.h b/chromium/third_party/webrtc/modules/video_processing/video_denoiser.h
index 03b30d91c7f..114f663c03d 100644
--- a/chromium/third_party/webrtc/modules/video_processing/video_denoiser.h
+++ b/chromium/third_party/webrtc/modules/video_processing/video_denoiser.h
@@ -22,25 +22,58 @@ namespace webrtc {
 class VideoDenoiser {
  public:
   explicit VideoDenoiser(bool runtime_cpu_detection);
+
   void DenoiseFrame(const VideoFrame& frame,
                     VideoFrame* denoised_frame,
-                    VideoFrame* denoised_frame_track,
-                    int noise_level_prev);
+                    VideoFrame* denoised_frame_prev,
+                    bool noise_estimation_enabled);
 
  private:
+  void DenoiserReset(const VideoFrame& frame,
+                     VideoFrame* denoised_frame,
+                     VideoFrame* denoised_frame_prev);
+
+  // Check the mb position, return 1: close to the frame center (between 1/8
+  // and 7/8 of width/height), 3: close to the border (out of 1/16 and 15/16
+  // of width/height), 2: in between.
+  int PositionCheck(int mb_row, int mb_col, int noise_level);
+
+  // To reduce false detection in moving object detection (MOD).
+  void ReduceFalseDetection(const std::unique_ptr<uint8_t[]>& d_status,
+                            std::unique_ptr<uint8_t[]>* d_status_red,
+                            int noise_level);
+
+  // Return whether a block might cause trailing artifact by checking if one of
+  // its neighbor blocks is a moving edge block.
+  bool IsTrailingBlock(const std::unique_ptr<uint8_t[]>& d_status,
+                       int mb_row,
+                       int mb_col);
+
+  // Copy input blocks to dst buffer on moving object blocks (MOB).
+  void CopySrcOnMOB(const uint8_t* y_src, uint8_t* y_dst);
+
+  // Copy luma margin blocks when frame width/height not divisible by 16.
+  void CopyLumaOnMargin(const uint8_t* y_src, uint8_t* y_dst);
+
   int width_;
   int height_;
+  int mb_rows_;
+  int mb_cols_;
+  int stride_y_;
+  int stride_u_;
+  int stride_v_;
   CpuType cpu_type_;
-  std::unique_ptr<DenoiseMetrics[]> metrics_;
   std::unique_ptr<DenoiserFilter> filter_;
   std::unique_ptr<NoiseEstimation> ne_;
-  std::unique_ptr<uint8_t[]> d_status_;
-#if EXPERIMENTAL
-  std::unique_ptr<uint8_t[]> d_status_tmp1_;
-  std::unique_ptr<uint8_t[]> d_status_tmp2_;
-#endif
+  // 1 for moving edge block, 0 for static block.
+  std::unique_ptr<uint8_t[]> moving_edge_;
+  // 1 for moving object block, 0 for static block.
+  std::unique_ptr<uint8_t[]> moving_object_;
+  // x_density_ and y_density_ are used in MOD process.
   std::unique_ptr<uint8_t[]> x_density_;
   std::unique_ptr<uint8_t[]> y_density_;
+  // Save the return values by MbDenoise for each block.
+  std::unique_ptr<DenoiserDecision[]> mb_filter_decision_;
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/video_processing.gypi b/chromium/third_party/webrtc/modules/video_processing/video_processing.gypi
index 5bf0ea36c36..3e90fd21571 100644
--- a/chromium/third_party/webrtc/modules/video_processing/video_processing.gypi
+++ b/chromium/third_party/webrtc/modules/video_processing/video_processing.gypi
@@ -20,12 +20,6 @@
       'sources': [
         'include/video_processing.h',
         'include/video_processing_defines.h',
-        'brightness_detection.cc',
-        'brightness_detection.h',
-        'content_analysis.cc',
-        'content_analysis.h',
-        'deflickering.cc',
-        'deflickering.h',
         'frame_preprocessor.cc',
         'frame_preprocessor.h',
         'spatial_resampler.cc',
@@ -62,7 +56,6 @@
           'target_name': 'video_processing_sse2',
           'type': 'static_library',
           'sources': [
-            'content_analysis_sse2.cc',
             'util/denoiser_filter_sse2.cc',
             'util/denoiser_filter_sse2.h',
           ],
diff --git a/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.cc b/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.cc
index f34886f10f2..86f75bf239d 100644
--- a/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.cc
+++ b/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.cc
@@ -18,21 +18,6 @@
 
 namespace webrtc {
 
-namespace {
-
-int GetSubSamplingFactor(int width, int height) {
-  if (width * height >= 640 * 480) {
-    return 3;
-  } else if (width * height >= 352 * 288) {
-    return 2;
-  } else if (width * height >= 176 * 144) {
-    return 1;
-  } else {
-    return 0;
-  }
-}
-}  // namespace
-
 VideoProcessing* VideoProcessing::Create() {
   return new VideoProcessingImpl();
 }
@@ -40,83 +25,6 @@ VideoProcessing* VideoProcessing::Create() {
 VideoProcessingImpl::VideoProcessingImpl() {}
 VideoProcessingImpl::~VideoProcessingImpl() {}
 
-void VideoProcessing::GetFrameStats(const VideoFrame& frame,
-                                    FrameStats* stats) {
-  ClearFrameStats(stats);  // The histogram needs to be zeroed out.
-  if (frame.IsZeroSize()) {
-    return;
-  }
-
-  int width = frame.width();
-  int height = frame.height();
-  stats->sub_sampling_factor = GetSubSamplingFactor(width, height);
-
-  const uint8_t* buffer = frame.buffer(kYPlane);
-  // Compute histogram and sum of frame
-  for (int i = 0; i < height; i += (1 << stats->sub_sampling_factor)) {
-    int k = i * width;
-    for (int j = 0; j < width; j += (1 << stats->sub_sampling_factor)) {
-      stats->hist[buffer[k + j]]++;
-      stats->sum += buffer[k + j];
-    }
-  }
-
-  stats->num_pixels = (width * height) / ((1 << stats->sub_sampling_factor) *
-                                          (1 << stats->sub_sampling_factor));
-  assert(stats->num_pixels > 0);
-
-  // Compute mean value of frame
-  stats->mean = stats->sum / stats->num_pixels;
-}
-
-bool VideoProcessing::ValidFrameStats(const FrameStats& stats) {
-  if (stats.num_pixels == 0) {
-    LOG(LS_WARNING) << "Invalid frame stats.";
-    return false;
-  }
-  return true;
-}
-
-void VideoProcessing::ClearFrameStats(FrameStats* stats) {
-  stats->mean = 0;
-  stats->sum = 0;
-  stats->num_pixels = 0;
-  stats->sub_sampling_factor = 0;
-  memset(stats->hist, 0, sizeof(stats->hist));
-}
-
-void VideoProcessing::Brighten(int delta, VideoFrame* frame) {
-  RTC_DCHECK(!frame->IsZeroSize());
-  RTC_DCHECK(frame->width() > 0);
-  RTC_DCHECK(frame->height() > 0);
-
-  int num_pixels = frame->width() * frame->height();
-
-  int look_up[256];
-  for (int i = 0; i < 256; i++) {
-    int val = i + delta;
-    look_up[i] = ((((val < 0) ? 0 : val) > 255) ? 255 : val);
-  }
-
-  uint8_t* temp_ptr = frame->buffer(kYPlane);
-  for (int i = 0; i < num_pixels; i++) {
-    *temp_ptr = static_cast<uint8_t>(look_up[*temp_ptr]);
-    temp_ptr++;
-  }
-}
-
-int32_t VideoProcessingImpl::Deflickering(VideoFrame* frame,
-                                          FrameStats* stats) {
-  rtc::CritScope mutex(&mutex_);
-  return deflickering_.ProcessFrame(frame, stats);
-}
-
-int32_t VideoProcessingImpl::BrightnessDetection(const VideoFrame& frame,
-                                                 const FrameStats& stats) {
-  rtc::CritScope mutex(&mutex_);
-  return brightness_detection_.ProcessFrame(frame, stats);
-}
-
 void VideoProcessingImpl::EnableTemporalDecimation(bool enable) {
   rtc::CritScope mutex(&mutex_);
   frame_pre_processor_.EnableTemporalDecimation(enable);
@@ -135,11 +43,6 @@ int32_t VideoProcessingImpl::SetTargetResolution(uint32_t width,
   return frame_pre_processor_.SetTargetResolution(width, height, frame_rate);
 }
 
-void VideoProcessingImpl::SetTargetFramerate(int frame_rate) {
-  rtc::CritScope cs(&mutex_);
-  frame_pre_processor_.SetTargetFramerate(frame_rate);
-}
-
 uint32_t VideoProcessingImpl::GetDecimatedFrameRate() {
   rtc::CritScope cs(&mutex_);
   return frame_pre_processor_.GetDecimatedFrameRate();
@@ -155,9 +58,9 @@ uint32_t VideoProcessingImpl::GetDecimatedHeight() const {
   return frame_pre_processor_.GetDecimatedHeight();
 }
 
-void VideoProcessingImpl::EnableDenosing(bool enable) {
+void VideoProcessingImpl::EnableDenoising(bool enable) {
   rtc::CritScope cs(&mutex_);
-  frame_pre_processor_.EnableDenosing(enable);
+  frame_pre_processor_.EnableDenoising(enable);
 }
 
 const VideoFrame* VideoProcessingImpl::PreprocessFrame(
@@ -166,14 +69,4 @@ const VideoFrame* VideoProcessingImpl::PreprocessFrame(
   return frame_pre_processor_.PreprocessFrame(frame);
 }
 
-VideoContentMetrics* VideoProcessingImpl::GetContentMetrics() const {
-  rtc::CritScope mutex(&mutex_);
-  return frame_pre_processor_.GetContentMetrics();
-}
-
-void VideoProcessingImpl::EnableContentAnalysis(bool enable) {
-  rtc::CritScope mutex(&mutex_);
-  frame_pre_processor_.EnableContentAnalysis(enable);
-}
-
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.h b/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.h
index 1d9a3775cf1..21e23c904dd 100644
--- a/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.h
+++ b/chromium/third_party/webrtc/modules/video_processing/video_processing_impl.h
@@ -13,8 +13,6 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/modules/video_processing/brightness_detection.h"
-#include "webrtc/modules/video_processing/deflickering.h"
 #include "webrtc/modules/video_processing/frame_preprocessor.h"
 
 namespace webrtc {
@@ -26,28 +24,20 @@ class VideoProcessingImpl : public VideoProcessing {
   ~VideoProcessingImpl() override;
 
   // Implements VideoProcessing.
-  int32_t Deflickering(VideoFrame* frame, FrameStats* stats) override;
-  int32_t BrightnessDetection(const VideoFrame& frame,
-                              const FrameStats& stats) override;
   void EnableTemporalDecimation(bool enable) override;
   void SetInputFrameResampleMode(VideoFrameResampling resampling_mode) override;
-  void EnableContentAnalysis(bool enable) override;
   int32_t SetTargetResolution(uint32_t width,
                               uint32_t height,
                               uint32_t frame_rate) override;
-  void SetTargetFramerate(int frame_rate) override;
   uint32_t GetDecimatedFrameRate() override;
   uint32_t GetDecimatedWidth() const override;
   uint32_t GetDecimatedHeight() const override;
-  void EnableDenosing(bool enable) override;
+  void EnableDenoising(bool enable) override;
   const VideoFrame* PreprocessFrame(const VideoFrame& frame) override;
-  VideoContentMetrics* GetContentMetrics() const override;
 
  private:
   rtc::CriticalSection mutex_;
-  VPMDeflickering deflickering_ GUARDED_BY(mutex_);
-  VPMBrightnessDetection brightness_detection_;
-  VPMFramePreprocessor frame_pre_processor_;
+  VPMFramePreprocessor frame_pre_processor_ GUARDED_BY(mutex_);
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/BUILD.gn b/chromium/third_party/webrtc/modules/video_render/BUILD.gn
deleted file mode 100644
index 0771bd7080c..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/BUILD.gn
+++ /dev/null
@@ -1,178 +0,0 @@
-# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-import("../../build/webrtc.gni")
-
-source_set("video_render_module") {
-  sources = [
-    "external/video_render_external_impl.cc",
-    "external/video_render_external_impl.h",
-    "i_video_render.h",
-    "video_render.h",
-    "video_render_defines.h",
-    "video_render_impl.h",
-  ]
-
-  deps = [
-    "../..:webrtc_common",
-    "../../common_video",
-    "../../system_wrappers",
-    "../utility",
-  ]
-
-  configs += [ "../..:common_config" ]
-  public_configs = [ "../..:common_inherited_config" ]
-
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-}
-
-source_set("video_render") {
-  sources = [
-    "video_render_impl.cc",
-  ]
-  deps = [
-    ":video_render_module",
-    "../../system_wrappers",
-  ]
-
-  configs += [ "../..:common_config" ]
-  public_configs = [ "../..:common_inherited_config" ]
-
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-}
-
-if (!build_with_chromium) {
-  config("video_render_internal_impl_config") {
-    if (is_ios) {
-      libs = [
-        "OpenGLES.framework",
-        "QuartzCore.framework",
-      ]
-    }
-  }
-
-  source_set("video_render_internal_impl") {
-    libs = []
-    sources = [
-      "video_render_internal_impl.cc",
-    ]
-    deps = [
-      ":video_render_module",
-      "../../system_wrappers",
-    ]
-
-    if (is_linux) {
-      sources += [
-        "linux/video_render_linux_impl.cc",
-        "linux/video_render_linux_impl.h",
-        "linux/video_x11_channel.cc",
-        "linux/video_x11_channel.h",
-        "linux/video_x11_render.cc",
-        "linux/video_x11_render.h",
-      ]
-
-      deps += [ "../..:webrtc_common" ]
-
-      libs += [ "Xext" ]
-    }
-    if (is_mac) {
-      sources += [
-        "mac/cocoa_full_screen_window.h",
-        "mac/cocoa_full_screen_window.mm",
-        "mac/cocoa_render_view.h",
-        "mac/cocoa_render_view.mm",
-        "mac/video_render_agl.cc",
-        "mac/video_render_agl.h",
-        "mac/video_render_mac_carbon_impl.cc",
-        "mac/video_render_mac_carbon_impl.h",
-        "mac/video_render_mac_cocoa_impl.h",
-        "mac/video_render_mac_cocoa_impl.mm",
-        "mac/video_render_nsopengl.h",
-        "mac/video_render_nsopengl.mm",
-      ]
-
-      libs += [
-        "CoreVideo.framework",
-        "QTKit.framework",
-      ]
-    }
-    if (is_win) {
-      sources += [
-        "windows/i_video_render_win.h",
-        "windows/video_render_direct3d9.cc",
-        "windows/video_render_direct3d9.h",
-        "windows/video_render_windows_impl.cc",
-        "windows/video_render_windows_impl.h",
-      ]
-
-      directxsdk_exists =
-          exec_script("//build/dir_exists.py",
-                      [ rebase_path("//third_party/directxsdk/files",
-                                    root_build_dir) ],
-                      "trim string") == "True"
-      if (directxsdk_exists) {
-        directxsdk_path = "//third_party/directxsdk/files"
-      } else {
-        directxsdk_path =
-            exec_script("../../build/find_directx_sdk.py", [], "trim string")
-      }
-      include_dirs = [ directxsdk_path + "/Include" ]
-    }
-    if (is_android) {
-      sources += [
-        "android/video_render_android_impl.cc",
-        "android/video_render_android_impl.h",
-        "android/video_render_android_native_opengl2.cc",
-        "android/video_render_android_native_opengl2.h",
-        "android/video_render_android_surface_view.cc",
-        "android/video_render_android_surface_view.h",
-        "android/video_render_opengles20.cc",
-        "android/video_render_opengles20.h",
-      ]
-
-      libs += [ "GLESv2" ]
-    }
-    if (is_ios) {
-      sources += [
-        "ios/open_gles20.h",
-        "ios/open_gles20.mm",
-        "ios/video_render_ios_channel.h",
-        "ios/video_render_ios_channel.mm",
-        "ios/video_render_ios_gles20.h",
-        "ios/video_render_ios_gles20.mm",
-        "ios/video_render_ios_impl.h",
-        "ios/video_render_ios_impl.mm",
-        "ios/video_render_ios_view.h",
-        "ios/video_render_ios_view.mm",
-      ]
-
-      deps += [ "../..:webrtc_common" ]
-
-      cflags = [ "-fobjc-arc" ]  # CLANG_ENABLE_OBJC_ARC = YES.
-    }
-
-    all_dependent_configs = [ ":video_render_internal_impl_config" ]
-
-    configs += [ "../..:common_config" ]
-    public_configs = [ "../..:common_inherited_config" ]
-
-    if (is_clang) {
-      # Suppress warnings from Chrome's Clang plugins.
-      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-      configs -= [ "//build/config/clang:find_bad_constructs" ]
-    }
-  }
-}
diff --git a/chromium/third_party/webrtc/modules/video_render/OWNERS b/chromium/third_party/webrtc/modules/video_render/OWNERS
deleted file mode 100644
index 3aaa5328f5c..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/OWNERS
+++ /dev/null
@@ -1,12 +0,0 @@
-mflodman@webrtc.org
-perkj@webrtc.org
-tkchin@webrtc.org
-
-per-file *.isolate=kjellander@webrtc.org
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
-
-per-file BUILD.gn=kjellander@webrtc.org
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_impl.cc b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_impl.cc
deleted file mode 100644
index 9affb23d99f..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_impl.cc
+++ /dev/null
@@ -1,316 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/android/video_render_android_impl.h"
-
-#include "webrtc/modules/video_render/video_render_internal.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#ifdef ANDROID
-#include <android/log.h>
-#include <stdio.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
-#else
-#include "webrtc/system_wrappers/include/trace.h"
-#endif
-
-namespace webrtc {
-
-JavaVM* VideoRenderAndroid::g_jvm = NULL;
-
-int32_t SetRenderAndroidVM(JavaVM* javaVM) {
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, -1, "%s", __FUNCTION__);
-  VideoRenderAndroid::g_jvm = javaVM;
-  return 0;
-}
-
-VideoRenderAndroid::VideoRenderAndroid(
-    const int32_t id,
-    const VideoRenderType videoRenderType,
-    void* window,
-    const bool /*fullscreen*/):
-    _id(id),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _renderType(videoRenderType),
-    _ptrWindow((jobject)(window)),
-    _javaShutDownFlag(false),
-    _javaShutdownEvent(*EventWrapper::Create()),
-    _javaRenderEvent(*EventWrapper::Create()),
-    _lastJavaRenderEvent(0),
-    _javaRenderJniEnv(NULL) {
-}
-
-VideoRenderAndroid::~VideoRenderAndroid() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-               "VideoRenderAndroid dtor");
-
-  if (_javaRenderThread)
-    StopRender();
-
-  for (AndroidStreamMap::iterator it = _streamsMap.begin();
-       it != _streamsMap.end();
-       ++it) {
-    delete it->second;
-  }
-  delete &_javaShutdownEvent;
-  delete &_javaRenderEvent;
-  delete &_critSect;
-}
-
-int32_t VideoRenderAndroid::ChangeWindow(void* /*window*/) {
-  return -1;
-}
-
-VideoRenderCallback*
-VideoRenderAndroid::AddIncomingRenderStream(const uint32_t streamId,
-                                            const uint32_t zOrder,
-                                            const float left, const float top,
-                                            const float right,
-                                            const float bottom) {
-  CriticalSectionScoped cs(&_critSect);
-
-  AndroidStream* renderStream = NULL;
-  AndroidStreamMap::iterator item = _streamsMap.find(streamId);
-  if (item != _streamsMap.end() && item->second != NULL) {
-    WEBRTC_TRACE(kTraceInfo,
-                 kTraceVideoRenderer,
-                 -1,
-                 "%s: Render stream already exists",
-                 __FUNCTION__);
-    return renderStream;
-  }
-
-  renderStream = CreateAndroidRenderChannel(streamId, zOrder, left, top,
-                                            right, bottom, *this);
-  if (renderStream) {
-    _streamsMap[streamId] = renderStream;
-  }
-  else {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "(%s:%d): renderStream is NULL", __FUNCTION__, __LINE__);
-    return NULL;
-  }
-  return renderStream;
-}
-
-int32_t VideoRenderAndroid::DeleteIncomingRenderStream(
-    const uint32_t streamId) {
-  CriticalSectionScoped cs(&_critSect);
-
-  AndroidStreamMap::iterator item = _streamsMap.find(streamId);
-  if (item == _streamsMap.end()) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "(%s:%d): renderStream is NULL", __FUNCTION__, __LINE__);
-    return -1;
-  }
-  delete item->second;
-  _streamsMap.erase(item);
-  return 0;
-}
-
-int32_t VideoRenderAndroid::GetIncomingRenderStreamProperties(
-    const uint32_t streamId,
-    uint32_t& zOrder,
-    float& left,
-    float& top,
-    float& right,
-    float& bottom) const {
-  return -1;
-}
-
-int32_t VideoRenderAndroid::StartRender() {
-  CriticalSectionScoped cs(&_critSect);
-
-  if (_javaRenderThread) {
-    // StartRender is called when this stream should start render.
-    // However StopRender is not called when the streams stop rendering.
-    // Thus the the thread  is only deleted when the renderer is removed.
-    WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
-                 "%s, Render thread already exist", __FUNCTION__);
-    return 0;
-  }
-
-  _javaRenderThread.reset(new rtc::PlatformThread(JavaRenderThreadFun, this,
-                                                  "AndroidRenderThread"));
-
-  _javaRenderThread->Start();
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: thread started",
-               __FUNCTION__);
-  _javaRenderThread->SetPriority(rtc::kRealtimePriority);
-  return 0;
-}
-
-int32_t VideoRenderAndroid::StopRender() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:", __FUNCTION__);
-  {
-    CriticalSectionScoped cs(&_critSect);
-    if (!_javaRenderThread)
-    {
-      return -1;
-    }
-    _javaShutDownFlag = true;
-    _javaRenderEvent.Set();
-  }
-
-  _javaShutdownEvent.Wait(3000);
-  CriticalSectionScoped cs(&_critSect);
-  _javaRenderThread->Stop();
-  _javaRenderThread.reset();
-
-  return 0;
-}
-
-void VideoRenderAndroid::ReDraw() {
-  CriticalSectionScoped cs(&_critSect);
-  // Allow redraw if it was more than 20ms since last.
-  if (_lastJavaRenderEvent < TickTime::MillisecondTimestamp() - 20) {
-    _lastJavaRenderEvent = TickTime::MillisecondTimestamp();
-    _javaRenderEvent.Set();
-  }
-}
-
-bool VideoRenderAndroid::JavaRenderThreadFun(void* obj) {
-  return static_cast<VideoRenderAndroid*> (obj)->JavaRenderThreadProcess();
-}
-
-bool VideoRenderAndroid::JavaRenderThreadProcess()
-{
-  _javaRenderEvent.Wait(1000);
-
-  CriticalSectionScoped cs(&_critSect);
-  if (!_javaRenderJniEnv) {
-    // try to attach the thread and get the env
-    // Attach this thread to JVM
-    jint res = g_jvm->AttachCurrentThread(&_javaRenderJniEnv, NULL);
-
-    // Get the JNI env for this thread
-    if ((res < 0) || !_javaRenderJniEnv) {
-      WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__, res, _javaRenderJniEnv);
-      return false;
-    }
-  }
-
-  for (AndroidStreamMap::iterator it = _streamsMap.begin();
-       it != _streamsMap.end();
-       ++it) {
-    it->second->DeliverFrame(_javaRenderJniEnv);
-  }
-
-  if (_javaShutDownFlag) {
-    if (g_jvm->DetachCurrentThread() < 0)
-      WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    else {
-      WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-                   "%s: Java thread detached", __FUNCTION__);
-    }
-    _javaRenderJniEnv = NULL;
-    _javaShutDownFlag = false;
-    _javaShutdownEvent.Set();
-    return false; // Do not run this thread again.
-  }
-  return true;
-}
-
-VideoRenderType VideoRenderAndroid::RenderType() {
-  return _renderType;
-}
-
-RawVideoType VideoRenderAndroid::PerferedVideoType() {
-  return kVideoI420;
-}
-
-bool VideoRenderAndroid::FullScreen() {
-  return false;
-}
-
-int32_t VideoRenderAndroid::GetGraphicsMemory(
-    uint64_t& /*totalGraphicsMemory*/,
-    uint64_t& /*availableGraphicsMemory*/) const {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-int32_t VideoRenderAndroid::GetScreenResolution(
-    uint32_t& /*screenWidth*/,
-    uint32_t& /*screenHeight*/) const {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-uint32_t VideoRenderAndroid::RenderFrameRate(
-    const uint32_t /*streamId*/) {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-int32_t VideoRenderAndroid::SetStreamCropping(
-    const uint32_t /*streamId*/,
-    const float /*left*/,
-    const float /*top*/,
-    const float /*right*/,
-    const float /*bottom*/) {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-int32_t VideoRenderAndroid::SetTransparentBackground(const bool enable) {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-int32_t VideoRenderAndroid::ConfigureRenderer(
-    const uint32_t streamId,
-    const unsigned int zOrder,
-    const float left,
-    const float top,
-    const float right,
-    const float bottom) {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-int32_t VideoRenderAndroid::SetText(
-    const uint8_t textId,
-    const uint8_t* text,
-    const int32_t textLength,
-    const uint32_t textColorRef,
-    const uint32_t backgroundColorRef,
-    const float left, const float top,
-    const float rigth, const float bottom) {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-int32_t VideoRenderAndroid::SetBitmap(const void* bitMap,
-                                      const uint8_t pictureId,
-                                      const void* colorKey,
-                                      const float left, const float top,
-                                      const float right,
-                                      const float bottom) {
-  WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-               "%s - not supported on Android", __FUNCTION__);
-  return -1;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_impl.h b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_impl.h
deleted file mode 100644
index 06fd7a1c7cd..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_impl.h
+++ /dev/null
@@ -1,154 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
-
-#include <jni.h>
-
-#include <map>
-#include <memory>
-
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/modules/video_render/i_video_render.h"
-
-
-namespace webrtc {
-
-//#define ANDROID_LOG
-
-class CriticalSectionWrapper;
-class EventWrapper;
-
-// The object a module user uses to send new frames to the java renderer
-// Base class for android render streams.
-
-class AndroidStream : public VideoRenderCallback {
- public:
-  // DeliverFrame is called from a thread connected to the Java VM.
-  // Used for Delivering frame for rendering.
-  virtual void DeliverFrame(JNIEnv* jniEnv)=0;
-
-  virtual ~AndroidStream() {};
-};
-
-class VideoRenderAndroid: IVideoRender {
- public:
-  VideoRenderAndroid(const int32_t id,
-                     const VideoRenderType videoRenderType,
-                     void* window,
-                     const bool fullscreen);
-
-  virtual ~VideoRenderAndroid();
-
-  virtual int32_t Init()=0;
-
-  virtual int32_t ChangeWindow(void* window);
-
-  virtual VideoRenderCallback* AddIncomingRenderStream(
-      const uint32_t streamId,
-      const uint32_t zOrder,
-      const float left, const float top,
-      const float right, const float bottom);
-
-  virtual int32_t DeleteIncomingRenderStream(
-      const uint32_t streamId);
-
-  virtual int32_t GetIncomingRenderStreamProperties(
-      const uint32_t streamId,
-      uint32_t& zOrder,
-      float& left, float& top,
-      float& right, float& bottom) const;
-
-  virtual int32_t StartRender();
-
-  virtual int32_t StopRender();
-
-  virtual void ReDraw();
-
-  // Properties
-
-  virtual VideoRenderType RenderType();
-
-  virtual RawVideoType PerferedVideoType();
-
-  virtual bool FullScreen();
-
-  virtual int32_t GetGraphicsMemory(
-      uint64_t& totalGraphicsMemory,
-      uint64_t& availableGraphicsMemory) const;
-
-  virtual int32_t GetScreenResolution(
-      uint32_t& screenWidth,
-      uint32_t& screenHeight) const;
-
-  virtual uint32_t RenderFrameRate(const uint32_t streamId);
-
-  virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                    const float left, const float top,
-                                    const float right, const float bottom);
-
-  virtual int32_t SetTransparentBackground(const bool enable);
-
-  virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                    const unsigned int zOrder,
-                                    const float left, const float top,
-                                    const float right, const float bottom);
-
-  virtual int32_t SetText(const uint8_t textId,
-                          const uint8_t* text,
-                          const int32_t textLength,
-                          const uint32_t textColorRef,
-                          const uint32_t backgroundColorRef,
-                          const float left, const float top,
-                          const float rigth, const float bottom);
-
-  virtual int32_t SetBitmap(const void* bitMap,
-                            const uint8_t pictureId,
-                            const void* colorKey, const float left,
-                            const float top, const float right,
-                            const float bottom);
-  static JavaVM* g_jvm;
-
- protected:
-  virtual AndroidStream* CreateAndroidRenderChannel(
-      int32_t streamId,
-      int32_t zOrder,
-      const float left,
-      const float top,
-      const float right,
-      const float bottom,
-      VideoRenderAndroid& renderer) = 0;
-
-  int32_t _id;
-  CriticalSectionWrapper& _critSect;
-  VideoRenderType _renderType;
-  jobject _ptrWindow;
-
- private:
-  static bool JavaRenderThreadFun(void* obj);
-  bool JavaRenderThreadProcess();
-
-  // Map with streams to render.
-  typedef std::map<int32_t, AndroidStream*> AndroidStreamMap;
-  AndroidStreamMap _streamsMap;
-  // True if the _javaRenderThread thread shall be detached from the JVM.
-  bool _javaShutDownFlag;
-  EventWrapper& _javaShutdownEvent;
-  EventWrapper& _javaRenderEvent;
-  int64_t _lastJavaRenderEvent;
-  JNIEnv* _javaRenderJniEnv; // JNIEnv for the java render thread.
-  // TODO(pbos): Remove unique_ptr and use the member directly.
-  std::unique_ptr<rtc::PlatformThread> _javaRenderThread;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc
deleted file mode 100644
index 286776e317b..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.cc
+++ /dev/null
@@ -1,450 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/android/video_render_android_native_opengl2.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#ifdef ANDROID_LOG
-#include <android/log.h>
-#include <stdio.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
-#else
-#include "webrtc/system_wrappers/include/trace.h"
-#endif
-
-namespace webrtc {
-
-AndroidNativeOpenGl2Renderer::AndroidNativeOpenGl2Renderer(
-    const int32_t id,
-    const VideoRenderType videoRenderType,
-    void* window,
-    const bool fullscreen) :
-    VideoRenderAndroid(id, videoRenderType, window, fullscreen),
-    _javaRenderObj(NULL),
-    _javaRenderClass(NULL) {
-}
-
-bool AndroidNativeOpenGl2Renderer::UseOpenGL2(void* window) {
-  if (!g_jvm) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
-                 "RendererAndroid():UseOpenGL No JVM set.");
-    return false;
-  }
-  bool isAttached = false;
-  JNIEnv* env = NULL;
-  if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-    // try to attach the thread and get the env
-    // Attach this thread to JVM
-    jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
-    // Get the JNI env for this thread
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(
-          kTraceError,
-          kTraceVideoRenderer,
-          -1,
-          "RendererAndroid(): Could not attach thread to JVM (%d, %p)",
-          res, env);
-      return false;
-    }
-    isAttached = true;
-  }
-
-  // get the renderer class
-  jclass javaRenderClassLocal =
-      env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
-  if (!javaRenderClassLocal) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
-                 "%s: could not find ViEAndroidRenderer class",
-                 __FUNCTION__);
-    return false;
-  }
-
-  // get the method ID for UseOpenGL
-  jmethodID cidUseOpenGL = env->GetStaticMethodID(javaRenderClassLocal,
-                                                  "UseOpenGL2",
-                                                  "(Ljava/lang/Object;)Z");
-  if (cidUseOpenGL == NULL) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
-                 "%s: could not get UseOpenGL ID", __FUNCTION__);
-    return false;
-  }
-  jboolean res = env->CallStaticBooleanMethod(javaRenderClassLocal,
-                                              cidUseOpenGL, (jobject) window);
-
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (g_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    }
-  }
-  return res;
-}
-
-AndroidNativeOpenGl2Renderer::~AndroidNativeOpenGl2Renderer() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-               "AndroidNativeOpenGl2Renderer dtor");
-  if (g_jvm) {
-    // get the JNI env for this thread
-    bool isAttached = false;
-    JNIEnv* env = NULL;
-    if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-      // try to attach the thread and get the env
-      // Attach this thread to JVM
-      jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
-      // Get the JNI env for this thread
-      if ((res < 0) || !env) {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Could not attach thread to JVM (%d, %p)",
-                     __FUNCTION__, res, env);
-        env = NULL;
-      }
-      else {
-        isAttached = true;
-      }
-    }
-
-    env->DeleteGlobalRef(_javaRenderObj);
-    env->DeleteGlobalRef(_javaRenderClass);
-
-    if (isAttached) {
-      if (g_jvm->DetachCurrentThread() < 0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
-                     "%s: Could not detach thread from JVM",
-                     __FUNCTION__);
-      }
-    }
-  }
-}
-
-int32_t AndroidNativeOpenGl2Renderer::Init() {
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
-  if (!g_jvm) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "(%s): Not a valid Java VM pointer.", __FUNCTION__);
-    return -1;
-  }
-  if (!_ptrWindow) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
-                 "(%s): No window have been provided.", __FUNCTION__);
-    return -1;
-  }
-
-  // get the JNI env for this thread
-  bool isAttached = false;
-  JNIEnv* env = NULL;
-  if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-    // try to attach the thread and get the env
-    // Attach this thread to JVM
-    jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
-    // Get the JNI env for this thread
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__, res, env);
-      return -1;
-    }
-    isAttached = true;
-  }
-
-  // get the ViEAndroidGLES20 class
-  jclass javaRenderClassLocal =
-      env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
-  if (!javaRenderClassLocal) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: could not find ViEAndroidGLES20", __FUNCTION__);
-    return -1;
-  }
-
-  // create a global reference to the class (to tell JNI that
-  // we are referencing it after this function has returned)
-  _javaRenderClass =
-      reinterpret_cast<jclass> (env->NewGlobalRef(javaRenderClassLocal));
-  if (!_javaRenderClass) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: could not create Java SurfaceHolder class reference",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  // Delete local class ref, we only use the global ref
-  env->DeleteLocalRef(javaRenderClassLocal);
-
-  // create a reference to the object (to tell JNI that we are referencing it
-  // after this function has returned)
-  _javaRenderObj = env->NewGlobalRef(_ptrWindow);
-  if (!_javaRenderObj) {
-    WEBRTC_TRACE(
-        kTraceError,
-        kTraceVideoRenderer,
-        _id,
-        "%s: could not create Java SurfaceRender object reference",
-        __FUNCTION__);
-    return -1;
-  }
-
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (g_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    }
-  }
-
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s done",
-               __FUNCTION__);
-  return 0;
-
-}
-AndroidStream*
-AndroidNativeOpenGl2Renderer::CreateAndroidRenderChannel(
-    int32_t streamId,
-    int32_t zOrder,
-    const float left,
-    const float top,
-    const float right,
-    const float bottom,
-    VideoRenderAndroid& renderer) {
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: Id %d",
-               __FUNCTION__, streamId);
-  AndroidNativeOpenGl2Channel* stream =
-      new AndroidNativeOpenGl2Channel(streamId, g_jvm, renderer,
-                                      _javaRenderObj);
-  if (stream && stream->Init(zOrder, left, top, right, bottom) == 0)
-    return stream;
-  else {
-    delete stream;
-  }
-  return NULL;
-}
-
-AndroidNativeOpenGl2Channel::AndroidNativeOpenGl2Channel(
-    uint32_t streamId,
-    JavaVM* jvm,
-    VideoRenderAndroid& renderer,jobject javaRenderObj):
-    _id(streamId),
-    _renderCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _renderer(renderer), _jvm(jvm), _javaRenderObj(javaRenderObj),
-    _registerNativeCID(NULL), _deRegisterNativeCID(NULL),
-    _openGLRenderer(streamId) {
-
-}
-AndroidNativeOpenGl2Channel::~AndroidNativeOpenGl2Channel() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-               "AndroidNativeOpenGl2Channel dtor");
-  if (_jvm) {
-    // get the JNI env for this thread
-    bool isAttached = false;
-    JNIEnv* env = NULL;
-    if (_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-      // try to attach the thread and get the env
-      // Attach this thread to JVM
-      jint res = _jvm->AttachCurrentThread(&env, NULL);
-
-      // Get the JNI env for this thread
-      if ((res < 0) || !env) {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Could not attach thread to JVM (%d, %p)",
-                     __FUNCTION__, res, env);
-        env = NULL;
-      } else {
-        isAttached = true;
-      }
-    }
-    if (env && _deRegisterNativeCID) {
-      env->CallVoidMethod(_javaRenderObj, _deRegisterNativeCID);
-    }
-
-    if (isAttached) {
-      if (_jvm->DetachCurrentThread() < 0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
-                     "%s: Could not detach thread from JVM",
-                     __FUNCTION__);
-      }
-    }
-  }
-
-  delete &_renderCritSect;
-}
-
-int32_t AndroidNativeOpenGl2Channel::Init(int32_t zOrder,
-                                          const float left,
-                                          const float top,
-                                          const float right,
-                                          const float bottom)
-{
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
-               "%s: AndroidNativeOpenGl2Channel", __FUNCTION__);
-  if (!_jvm) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Not a valid Java VM pointer", __FUNCTION__);
-    return -1;
-  }
-
-  // get the JNI env for this thread
-  bool isAttached = false;
-  JNIEnv* env = NULL;
-  if (_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-    // try to attach the thread and get the env
-    // Attach this thread to JVM
-    jint res = _jvm->AttachCurrentThread(&env, NULL);
-
-    // Get the JNI env for this thread
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__, res, env);
-      return -1;
-    }
-    isAttached = true;
-  }
-
-  jclass javaRenderClass =
-      env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
-  if (!javaRenderClass) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: could not find ViESurfaceRenderer", __FUNCTION__);
-    return -1;
-  }
-
-  // get the method ID for the ReDraw function
-  _redrawCid = env->GetMethodID(javaRenderClass, "ReDraw", "()V");
-  if (_redrawCid == NULL) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: could not get ReDraw ID", __FUNCTION__);
-    return -1;
-  }
-
-  _registerNativeCID = env->GetMethodID(javaRenderClass,
-                                        "RegisterNativeObject", "(J)V");
-  if (_registerNativeCID == NULL) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: could not get RegisterNativeObject ID", __FUNCTION__);
-    return -1;
-  }
-
-  _deRegisterNativeCID = env->GetMethodID(javaRenderClass,
-                                          "DeRegisterNativeObject", "()V");
-  if (_deRegisterNativeCID == NULL) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: could not get DeRegisterNativeObject ID",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  JNINativeMethod nativeFunctions[2] = {
-    { "DrawNative",
-      "(J)V",
-      (void*) &AndroidNativeOpenGl2Channel::DrawNativeStatic, },
-    { "CreateOpenGLNative",
-      "(JII)I",
-      (void*) &AndroidNativeOpenGl2Channel::CreateOpenGLNativeStatic },
-  };
-  if (env->RegisterNatives(javaRenderClass, nativeFunctions, 2) == 0) {
-    WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, -1,
-                 "%s: Registered native functions", __FUNCTION__);
-  }
-  else {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
-                 "%s: Failed to register native functions", __FUNCTION__);
-    return -1;
-  }
-
-  env->CallVoidMethod(_javaRenderObj, _registerNativeCID, (jlong) this);
-
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    }
-  }
-
-  if (_openGLRenderer.SetCoordinates(zOrder, left, top, right, bottom) != 0) {
-    return -1;
-  }
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
-               "%s: AndroidNativeOpenGl2Channel done", __FUNCTION__);
-  return 0;
-}
-
-int32_t AndroidNativeOpenGl2Channel::RenderFrame(const uint32_t /*streamId*/,
-                                                 const VideoFrame& videoFrame) {
-  //   WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
-  _renderCritSect.Enter();
-  _bufferToRender = videoFrame;
-  _renderCritSect.Leave();
-  _renderer.ReDraw();
-  return 0;
-}
-
-/*Implements AndroidStream
- * Calls the Java object and render the buffer in _bufferToRender
- */
-void AndroidNativeOpenGl2Channel::DeliverFrame(JNIEnv* jniEnv) {
-  //TickTime timeNow=TickTime::Now();
-
-  //Draw the Surface
-  jniEnv->CallVoidMethod(_javaRenderObj, _redrawCid);
-
-  // WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id,
-  // "%s: time to deliver %lld" ,__FUNCTION__,
-  // (TickTime::Now()-timeNow).Milliseconds());
-}
-
-/*
- * JNI callback from Java class. Called when the render
- * want to render a frame. Called from the GLRenderThread
- * Method:    DrawNative
- * Signature: (J)V
- */
-void JNICALL AndroidNativeOpenGl2Channel::DrawNativeStatic(
-    JNIEnv * env, jobject, jlong context) {
-  AndroidNativeOpenGl2Channel* renderChannel =
-      reinterpret_cast<AndroidNativeOpenGl2Channel*>(context);
-  renderChannel->DrawNative();
-}
-
-void AndroidNativeOpenGl2Channel::DrawNative() {
-  _renderCritSect.Enter();
-  _openGLRenderer.Render(_bufferToRender);
-  _renderCritSect.Leave();
-}
-
-/*
- * JNI callback from Java class. Called when the GLSurfaceview
- * have created a surface. Called from the GLRenderThread
- * Method:    CreateOpenGLNativeStatic
- * Signature: (JII)I
- */
-jint JNICALL AndroidNativeOpenGl2Channel::CreateOpenGLNativeStatic(
-    JNIEnv * env,
-    jobject,
-    jlong context,
-    jint width,
-    jint height) {
-  AndroidNativeOpenGl2Channel* renderChannel =
-      reinterpret_cast<AndroidNativeOpenGl2Channel*> (context);
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1, "%s:", __FUNCTION__);
-  return renderChannel->CreateOpenGLNative(width, height);
-}
-
-jint AndroidNativeOpenGl2Channel::CreateOpenGLNative(
-    int width, int height) {
-  return _openGLRenderer.Setup(width, height);
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.h b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
deleted file mode 100644
index 8be247b8342..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_native_opengl2.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
-
-#include <jni.h>
-
-#include "webrtc/modules/video_render/android/video_render_android_impl.h"
-#include "webrtc/modules/video_render/android/video_render_opengles20.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-
-class AndroidNativeOpenGl2Channel: public AndroidStream {
- public:
-  AndroidNativeOpenGl2Channel(
-      uint32_t streamId,
-      JavaVM* jvm,
-      VideoRenderAndroid& renderer,jobject javaRenderObj);
-  ~AndroidNativeOpenGl2Channel();
-
-  int32_t Init(int32_t zOrder, const float left, const float top,
-               const float right, const float bottom);
-
-  //Implement VideoRenderCallback
-  virtual int32_t RenderFrame(const uint32_t streamId,
-                              const VideoFrame& videoFrame);
-
-  //Implements AndroidStream
-  virtual void DeliverFrame(JNIEnv* jniEnv);
-
- private:
-  static jint JNICALL CreateOpenGLNativeStatic(
-      JNIEnv * env,
-      jobject,
-      jlong context,
-      jint width,
-      jint height);
-  jint CreateOpenGLNative(int width, int height);
-
-  static void JNICALL DrawNativeStatic(JNIEnv * env,jobject, jlong context);
-  void DrawNative();
-  uint32_t _id;
-  CriticalSectionWrapper& _renderCritSect;
-
-  VideoFrame _bufferToRender;
-  VideoRenderAndroid& _renderer;
-  JavaVM*     _jvm;
-  jobject     _javaRenderObj;
-
-  jmethodID      _redrawCid;
-  jmethodID      _registerNativeCID;
-  jmethodID      _deRegisterNativeCID;
-  VideoRenderOpenGles20 _openGLRenderer;
-};
-
-
-class AndroidNativeOpenGl2Renderer: private VideoRenderAndroid {
- public:
-  AndroidNativeOpenGl2Renderer(const int32_t id,
-                               const VideoRenderType videoRenderType,
-                               void* window,
-                               const bool fullscreen);
-
-  ~AndroidNativeOpenGl2Renderer();
-  static bool UseOpenGL2(void* window);
-
-  int32_t Init();
-  virtual AndroidStream* CreateAndroidRenderChannel(
-      int32_t streamId,
-      int32_t zOrder,
-      const float left,
-      const float top,
-      const float right,
-      const float bottom,
-      VideoRenderAndroid& renderer);
-
- private:
-  jobject _javaRenderObj;
-  jclass _javaRenderClass;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_surface_view.cc b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_surface_view.cc
deleted file mode 100644
index ea3b106b1ed..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_surface_view.cc
+++ /dev/null
@@ -1,474 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/android/video_render_android_surface_view.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#ifdef ANDROID_LOG
-#include <android/log.h>
-#include <stdio.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
-#else
-#include "webrtc/system_wrappers/include/trace.h"
-#endif
-
-namespace webrtc {
-
-AndroidSurfaceViewRenderer::AndroidSurfaceViewRenderer(
-    const int32_t id,
-    const VideoRenderType videoRenderType,
-    void* window,
-    const bool fullscreen) :
-    VideoRenderAndroid(id,videoRenderType,window,fullscreen),
-    _javaRenderObj(NULL),
-    _javaRenderClass(NULL) {
-}
-
-AndroidSurfaceViewRenderer::~AndroidSurfaceViewRenderer() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-               "AndroidSurfaceViewRenderer dtor");
-  if(g_jvm) {
-    // get the JNI env for this thread
-    bool isAttached = false;
-    JNIEnv* env = NULL;
-    if (g_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK) {
-      // try to attach the thread and get the env
-      // Attach this thread to JVM
-      jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
-      // Get the JNI env for this thread
-      if ((res < 0) || !env) {
-        WEBRTC_TRACE(kTraceError,
-                     kTraceVideoRenderer,
-                     _id,
-                     "%s: Could not attach thread to JVM (%d, %p)",
-                     __FUNCTION__,
-                     res,
-                     env);
-        env=NULL;
-      }
-      else {
-        isAttached = true;
-      }
-    }
-    env->DeleteGlobalRef(_javaRenderObj);
-    env->DeleteGlobalRef(_javaRenderClass);
-
-    if (isAttached) {
-      if (g_jvm->DetachCurrentThread() < 0) {
-        WEBRTC_TRACE(kTraceWarning,
-                     kTraceVideoRenderer,
-                     _id,
-                     "%s: Could not detach thread from JVM",
-                     __FUNCTION__);
-      }
-    }
-  }
-}
-
-int32_t AndroidSurfaceViewRenderer::Init() {
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
-  if (!g_jvm) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "(%s): Not a valid Java VM pointer.",
-                 __FUNCTION__);
-    return -1;
-  }
-  if(!_ptrWindow) {
-    WEBRTC_TRACE(kTraceWarning,
-                 kTraceVideoRenderer,
-                 _id,
-                 "(%s): No window have been provided.",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  // get the JNI env for this thread
-  bool isAttached = false;
-  JNIEnv* env = NULL;
-  if (g_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK) {
-    // try to attach the thread and get the env
-    // Attach this thread to JVM
-    jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
-    // Get the JNI env for this thread
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(kTraceError,
-                   kTraceVideoRenderer,
-                   _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__,
-                   res,
-                   env);
-      return -1;
-    }
-    isAttached = true;
-  }
-
-  // get the ViESurfaceRender class
-  jclass javaRenderClassLocal =
-      env->FindClass("org/webrtc/videoengine/ViESurfaceRenderer");
-  if (!javaRenderClassLocal) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not find ViESurfaceRenderer",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  // create a global reference to the class (to tell JNI that
-  // we are referencing it after this function has returned)
-  _javaRenderClass =
-      reinterpret_cast<jclass>(env->NewGlobalRef(javaRenderClassLocal));
-  if (!_javaRenderClass) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not create Java ViESurfaceRenderer class reference",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  // Delete local class ref, we only use the global ref
-  env->DeleteLocalRef(javaRenderClassLocal);
-
-  // get the method ID for the constructor
-  jmethodID cid = env->GetMethodID(_javaRenderClass,
-                                   "<init>",
-                                   "(Landroid/view/SurfaceView;)V");
-  if (cid == NULL) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not get constructor ID",
-                 __FUNCTION__);
-    return -1; /* exception thrown */
-  }
-
-  // construct the object
-  jobject javaRenderObjLocal = env->NewObject(_javaRenderClass,
-                                              cid,
-                                              _ptrWindow);
-  if (!javaRenderObjLocal) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not create Java Render",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  // create a reference to the object (to tell JNI that we are referencing it
-  // after this function has returned)
-  _javaRenderObj = env->NewGlobalRef(javaRenderObjLocal);
-  if (!_javaRenderObj) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not create Java SurfaceRender object reference",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (g_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(kTraceWarning,
-                   kTraceVideoRenderer,
-                   _id,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    }
-  }
-
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s done", __FUNCTION__);
-  return 0;
-}
-
-AndroidStream*
-AndroidSurfaceViewRenderer::CreateAndroidRenderChannel(
-    int32_t streamId,
-    int32_t zOrder,
-    const float left,
-    const float top,
-    const float right,
-    const float bottom,
-    VideoRenderAndroid& renderer) {
-  WEBRTC_TRACE(kTraceDebug,
-               kTraceVideoRenderer,
-               _id,
-               "%s: Id %d",
-               __FUNCTION__,
-               streamId);
-  AndroidSurfaceViewChannel* stream =
-      new AndroidSurfaceViewChannel(streamId, g_jvm, renderer, _javaRenderObj);
-  if(stream && stream->Init(zOrder, left, top, right, bottom) == 0)
-    return stream;
-  else
-    delete stream;
-  return NULL;
-}
-
-AndroidSurfaceViewChannel::AndroidSurfaceViewChannel(
-    uint32_t streamId,
-    JavaVM* jvm,
-    VideoRenderAndroid& renderer,
-    jobject javaRenderObj) :
-    _id(streamId),
-    _renderCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _renderer(renderer),
-    _jvm(jvm),
-    _javaRenderObj(javaRenderObj),
-#ifndef ANDROID_NDK_8_OR_ABOVE
-    _javaByteBufferObj(NULL),
-    _directBuffer(NULL),
-#endif
-    _bitmapWidth(0),
-    _bitmapHeight(0) {
-}
-
-AndroidSurfaceViewChannel::~AndroidSurfaceViewChannel() {
-  WEBRTC_TRACE(kTraceInfo,
-               kTraceVideoRenderer,
-               _id,
-               "AndroidSurfaceViewChannel dtor");
-  delete &_renderCritSect;
-  if(_jvm) {
-    // get the JNI env for this thread
-    bool isAttached = false;
-    JNIEnv* env = NULL;
-    if ( _jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK) {
-      // try to attach the thread and get the env
-      // Attach this thread to JVM
-      jint res = _jvm->AttachCurrentThread(&env, NULL);
-
-      // Get the JNI env for this thread
-      if ((res < 0) || !env) {
-        WEBRTC_TRACE(kTraceError,
-                     kTraceVideoRenderer,
-                     _id,
-                     "%s: Could not attach thread to JVM (%d, %p)",
-                     __FUNCTION__,
-                     res,
-                     env);
-        env=NULL;
-      }
-      else {
-        isAttached = true;
-      }
-    }
-
-    env->DeleteGlobalRef(_javaByteBufferObj);
-    if (isAttached) {
-      if (_jvm->DetachCurrentThread() < 0) {
-        WEBRTC_TRACE(kTraceWarning,
-                     kTraceVideoRenderer,
-                     _id,
-                     "%s: Could not detach thread from JVM",
-                     __FUNCTION__);
-      }
-    }
-  }
-}
-
-int32_t AndroidSurfaceViewChannel::Init(
-    int32_t /*zOrder*/,
-    const float left,
-    const float top,
-    const float right,
-    const float bottom) {
-
-  WEBRTC_TRACE(kTraceDebug,
-               kTraceVideoRenderer,
-               _id,
-               "%s: AndroidSurfaceViewChannel",
-               __FUNCTION__);
-  if (!_jvm) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: Not a valid Java VM pointer",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  if( (top > 1 || top < 0) ||
-      (right > 1 || right < 0) ||
-      (bottom > 1 || bottom < 0) ||
-      (left > 1 || left < 0)) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Wrong coordinates", __FUNCTION__);
-    return -1;
-  }
-
-  // get the JNI env for this thread
-  bool isAttached = false;
-  JNIEnv* env = NULL;
-  if (_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK) {
-    // try to attach the thread and get the env
-    // Attach this thread to JVM
-    jint res = _jvm->AttachCurrentThread(&env, NULL);
-
-    // Get the JNI env for this thread
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(kTraceError,
-                   kTraceVideoRenderer,
-                   _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__,
-                   res,
-                   env);
-      return -1;
-    }
-    isAttached = true;
-  }
-
-  jclass javaRenderClass =
-      env->FindClass("org/webrtc/videoengine/ViESurfaceRenderer");
-  if (!javaRenderClass) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not find ViESurfaceRenderer",
-                 __FUNCTION__);
-    return -1;
-  }
-
-  // get the method ID for the CreateIntArray
-  _createByteBufferCid =
-      env->GetMethodID(javaRenderClass,
-                       "CreateByteBuffer",
-                       "(II)Ljava/nio/ByteBuffer;");
-  if (_createByteBufferCid == NULL) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not get CreateByteBuffer ID",
-                 __FUNCTION__);
-    return -1; /* exception thrown */
-  }
-
-  // get the method ID for the DrawByteBuffer function
-  _drawByteBufferCid = env->GetMethodID(javaRenderClass,
-                                        "DrawByteBuffer",
-                                        "()V");
-  if (_drawByteBufferCid == NULL) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not get DrawByteBuffer ID",
-                 __FUNCTION__);
-    return -1; /* exception thrown */
-  }
-
-  // get the method ID for the SetCoordinates function
-  _setCoordinatesCid = env->GetMethodID(javaRenderClass,
-                                        "SetCoordinates",
-                                        "(FFFF)V");
-  if (_setCoordinatesCid == NULL) {
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: could not get SetCoordinates ID",
-                 __FUNCTION__);
-    return -1; /* exception thrown */
-  }
-
-  env->CallVoidMethod(_javaRenderObj, _setCoordinatesCid,
-                      left, top, right, bottom);
-
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(kTraceWarning,
-                   kTraceVideoRenderer,
-                   _id,
-                   "%s: Could not detach thread from JVM",
-                   __FUNCTION__);
-    }
-  }
-
-  WEBRTC_TRACE(kTraceDebug,
-               kTraceVideoRenderer,
-               _id,
-               "%s: AndroidSurfaceViewChannel done",
-               __FUNCTION__);
-  return 0;
-}
-
-int32_t AndroidSurfaceViewChannel::RenderFrame(const uint32_t /*streamId*/,
-                                               const VideoFrame& videoFrame) {
-  // WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
-  _renderCritSect.Enter();
-  _bufferToRender = videoFrame;
-  _renderCritSect.Leave();
-  _renderer.ReDraw();
-  return 0;
-}
-
-
-/*Implements AndroidStream
- * Calls the Java object and render the buffer in _bufferToRender
- */
-void AndroidSurfaceViewChannel::DeliverFrame(JNIEnv* jniEnv) {
-  _renderCritSect.Enter();
-
-  if (_bitmapWidth != _bufferToRender.width() ||
-      _bitmapHeight != _bufferToRender.height()) {
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: New render size %d "
-                 "%d",__FUNCTION__,
-                 _bufferToRender.width(), _bufferToRender.height());
-    if (_javaByteBufferObj) {
-      jniEnv->DeleteGlobalRef(_javaByteBufferObj);
-      _javaByteBufferObj = NULL;
-      _directBuffer = NULL;
-    }
-
-    jobject javaByteBufferObj =
-        jniEnv->CallObjectMethod(_javaRenderObj, _createByteBufferCid,
-                                 _bufferToRender.width(),
-                                 _bufferToRender.height());
-    _javaByteBufferObj = jniEnv->NewGlobalRef(javaByteBufferObj);
-    if (!_javaByteBufferObj) {
-      WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,  "%s: could not "
-                   "create Java ByteBuffer object reference", __FUNCTION__);
-      _renderCritSect.Leave();
-      return;
-    } else {
-      _directBuffer = static_cast<unsigned char*>
-          (jniEnv->GetDirectBufferAddress(_javaByteBufferObj));
-      _bitmapWidth = _bufferToRender.width();
-      _bitmapHeight = _bufferToRender.height();
-    }
-  }
-
-  if(_javaByteBufferObj && _bitmapWidth && _bitmapHeight) {
-    const int conversionResult =
-        ConvertFromI420(_bufferToRender, kRGB565, 0, _directBuffer);
-
-    if (conversionResult < 0)  {
-      WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Color conversion"
-                   " failed.", __FUNCTION__);
-      _renderCritSect.Leave();
-      return;
-    }
-  }
-  _renderCritSect.Leave();
-  // Draw the Surface
-  jniEnv->CallVoidMethod(_javaRenderObj, _drawByteBufferCid);
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_surface_view.h b/chromium/third_party/webrtc/modules/video_render/android/video_render_android_surface_view.h
deleted file mode 100644
index 0f029b54f34..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_android_surface_view.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
-
-#include <jni.h>
-
-#include "webrtc/modules/video_render/android/video_render_android_impl.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-
-class AndroidSurfaceViewChannel : public AndroidStream {
- public:
-  AndroidSurfaceViewChannel(uint32_t streamId,
-                            JavaVM* jvm,
-                            VideoRenderAndroid& renderer,
-                            jobject javaRenderObj);
-  ~AndroidSurfaceViewChannel();
-
-  int32_t Init(int32_t zOrder, const float left, const float top,
-               const float right, const float bottom);
-
-  //Implement VideoRenderCallback
-  virtual int32_t RenderFrame(const uint32_t streamId,
-                              const VideoFrame& videoFrame);
-
-  //Implements AndroidStream
-  virtual void DeliverFrame(JNIEnv* jniEnv);
-
- private:
-  uint32_t _id;
-  CriticalSectionWrapper& _renderCritSect;
-
-  VideoFrame _bufferToRender;
-  VideoRenderAndroid& _renderer;
-  JavaVM* _jvm;
-  jobject _javaRenderObj;
-
-  jobject _javaByteBufferObj;
-  unsigned char* _directBuffer;
-  jmethodID _createByteBufferCid;
-  jmethodID _drawByteBufferCid;
-
-  jmethodID _setCoordinatesCid;
-  int _bitmapWidth;
-  int _bitmapHeight;
-};
-
-class AndroidSurfaceViewRenderer : private VideoRenderAndroid {
- public:
-  AndroidSurfaceViewRenderer(const int32_t id,
-                             const VideoRenderType videoRenderType,
-                             void* window,
-                             const bool fullscreen);
-  ~AndroidSurfaceViewRenderer();
-  int32_t Init();
-  virtual AndroidStream* CreateAndroidRenderChannel(
-      int32_t streamId,
-      int32_t zOrder,
-      const float left,
-      const float top,
-      const float right,
-      const float bottom,
-      VideoRenderAndroid& renderer);
- private:
-  jobject _javaRenderObj;
-  jclass _javaRenderClass;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_opengles20.cc b/chromium/third_party/webrtc/modules/video_render/android/video_render_opengles20.cc
deleted file mode 100644
index 45db56a4f6e..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_opengles20.cc
+++ /dev/null
@@ -1,397 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <GLES2/gl2.h>
-#include <GLES2/gl2ext.h>
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "webrtc/modules/video_render/android/video_render_opengles20.h"
-
-//#define ANDROID_LOG
-
-#ifdef ANDROID_LOG
-#include <android/log.h>
-#include <stdio.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...)  __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
-#else
-#include "webrtc/system_wrappers/include/trace.h"
-#endif
-
-namespace webrtc {
-
-const char VideoRenderOpenGles20::g_indices[] = { 0, 3, 2, 0, 2, 1 };
-
-const char VideoRenderOpenGles20::g_vertextShader[] = {
-  "attribute vec4 aPosition;\n"
-  "attribute vec2 aTextureCoord;\n"
-  "varying vec2 vTextureCoord;\n"
-  "void main() {\n"
-  "  gl_Position = aPosition;\n"
-  "  vTextureCoord = aTextureCoord;\n"
-  "}\n" };
-
-// The fragment shader.
-// Do YUV to RGB565 conversion.
-const char VideoRenderOpenGles20::g_fragmentShader[] = {
-  "precision mediump float;\n"
-  "uniform sampler2D Ytex;\n"
-  "uniform sampler2D Utex,Vtex;\n"
-  "varying vec2 vTextureCoord;\n"
-  "void main(void) {\n"
-  "  float nx,ny,r,g,b,y,u,v;\n"
-  "  mediump vec4 txl,ux,vx;"
-  "  nx=vTextureCoord[0];\n"
-  "  ny=vTextureCoord[1];\n"
-  "  y=texture2D(Ytex,vec2(nx,ny)).r;\n"
-  "  u=texture2D(Utex,vec2(nx,ny)).r;\n"
-  "  v=texture2D(Vtex,vec2(nx,ny)).r;\n"
-
-  //"  y = v;\n"+
-  "  y=1.1643*(y-0.0625);\n"
-  "  u=u-0.5;\n"
-  "  v=v-0.5;\n"
-
-  "  r=y+1.5958*v;\n"
-  "  g=y-0.39173*u-0.81290*v;\n"
-  "  b=y+2.017*u;\n"
-  "  gl_FragColor=vec4(r,g,b,1.0);\n"
-  "}\n" };
-
-VideoRenderOpenGles20::VideoRenderOpenGles20(int32_t id) :
-    _id(id),
-    _textureWidth(-1),
-    _textureHeight(-1) {
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: id %d",
-               __FUNCTION__, (int) _id);
-
-  const GLfloat vertices[20] = {
-    // X, Y, Z, U, V
-    -1, -1, 0, 0, 1, // Bottom Left
-    1, -1, 0, 1, 1, //Bottom Right
-    1, 1, 0, 1, 0, //Top Right
-    -1, 1, 0, 0, 0 }; //Top Left
-
-  memcpy(_vertices, vertices, sizeof(_vertices));
-}
-
-VideoRenderOpenGles20::~VideoRenderOpenGles20() {
-}
-
-int32_t VideoRenderOpenGles20::Setup(int32_t width, int32_t height) {
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
-               "%s: width %d, height %d", __FUNCTION__, (int) width,
-               (int) height);
-
-  printGLString("Version", GL_VERSION);
-  printGLString("Vendor", GL_VENDOR);
-  printGLString("Renderer", GL_RENDERER);
-  printGLString("Extensions", GL_EXTENSIONS);
-
-  int maxTextureImageUnits[2];
-  int maxTextureSize[2];
-  glGetIntegerv(GL_MAX_TEXTURE_IMAGE_UNITS, maxTextureImageUnits);
-  glGetIntegerv(GL_MAX_TEXTURE_SIZE, maxTextureSize);
-
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
-               "%s: number of textures %d, size %d", __FUNCTION__,
-               (int) maxTextureImageUnits[0], (int) maxTextureSize[0]);
-
-  _program = createProgram(g_vertextShader, g_fragmentShader);
-  if (!_program) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Could not create program", __FUNCTION__);
-    return -1;
-  }
-
-  int positionHandle = glGetAttribLocation(_program, "aPosition");
-  checkGlError("glGetAttribLocation aPosition");
-  if (positionHandle == -1) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Could not get aPosition handle", __FUNCTION__);
-    return -1;
-  }
-
-  int textureHandle = glGetAttribLocation(_program, "aTextureCoord");
-  checkGlError("glGetAttribLocation aTextureCoord");
-  if (textureHandle == -1) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Could not get aTextureCoord handle", __FUNCTION__);
-    return -1;
-  }
-
-  // set the vertices array in the shader
-  // _vertices contains 4 vertices with 5 coordinates.
-  // 3 for (xyz) for the vertices and 2 for the texture
-  glVertexAttribPointer(positionHandle, 3, GL_FLOAT, false,
-                        5 * sizeof(GLfloat), _vertices);
-  checkGlError("glVertexAttribPointer aPosition");
-
-  glEnableVertexAttribArray(positionHandle);
-  checkGlError("glEnableVertexAttribArray positionHandle");
-
-  // set the texture coordinate array in the shader
-  // _vertices contains 4 vertices with 5 coordinates.
-  // 3 for (xyz) for the vertices and 2 for the texture
-  glVertexAttribPointer(textureHandle, 2, GL_FLOAT, false, 5
-                        * sizeof(GLfloat), &_vertices[3]);
-  checkGlError("glVertexAttribPointer maTextureHandle");
-  glEnableVertexAttribArray(textureHandle);
-  checkGlError("glEnableVertexAttribArray textureHandle");
-
-  glUseProgram(_program);
-  int i = glGetUniformLocation(_program, "Ytex");
-  checkGlError("glGetUniformLocation");
-  glUniform1i(i, 0); /* Bind Ytex to texture unit 0 */
-  checkGlError("glUniform1i Ytex");
-
-  i = glGetUniformLocation(_program, "Utex");
-  checkGlError("glGetUniformLocation Utex");
-  glUniform1i(i, 1); /* Bind Utex to texture unit 1 */
-  checkGlError("glUniform1i Utex");
-
-  i = glGetUniformLocation(_program, "Vtex");
-  checkGlError("glGetUniformLocation");
-  glUniform1i(i, 2); /* Bind Vtex to texture unit 2 */
-  checkGlError("glUniform1i");
-
-  glViewport(0, 0, width, height);
-  checkGlError("glViewport");
-  return 0;
-}
-
-// SetCoordinates
-// Sets the coordinates where the stream shall be rendered.
-// Values must be between 0 and 1.
-int32_t VideoRenderOpenGles20::SetCoordinates(int32_t zOrder,
-                                              const float left,
-                                              const float top,
-                                              const float right,
-                                              const float bottom) {
-  if ((top > 1 || top < 0) || (right > 1 || right < 0) ||
-      (bottom > 1 || bottom < 0) || (left > 1 || left < 0)) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: Wrong coordinates", __FUNCTION__);
-    return -1;
-  }
-
-  //  X, Y, Z, U, V
-  // -1, -1, 0, 0, 1, // Bottom Left
-  //  1, -1, 0, 1, 1, //Bottom Right
-  //  1,  1, 0, 1, 0, //Top Right
-  // -1,  1, 0, 0, 0  //Top Left
-
-  // Bottom Left
-  _vertices[0] = (left * 2) - 1;
-  _vertices[1] = -1 * (2 * bottom) + 1;
-  _vertices[2] = zOrder;
-
-  //Bottom Right
-  _vertices[5] = (right * 2) - 1;
-  _vertices[6] = -1 * (2 * bottom) + 1;
-  _vertices[7] = zOrder;
-
-  //Top Right
-  _vertices[10] = (right * 2) - 1;
-  _vertices[11] = -1 * (2 * top) + 1;
-  _vertices[12] = zOrder;
-
-  //Top Left
-  _vertices[15] = (left * 2) - 1;
-  _vertices[16] = -1 * (2 * top) + 1;
-  _vertices[17] = zOrder;
-
-  return 0;
-}
-
-int32_t VideoRenderOpenGles20::Render(const VideoFrame& frameToRender) {
-  if (frameToRender.IsZeroSize()) {
-    return -1;
-  }
-
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: id %d",
-               __FUNCTION__, (int) _id);
-
-  glUseProgram(_program);
-  checkGlError("glUseProgram");
-
-  if (_textureWidth != (GLsizei) frameToRender.width() ||
-      _textureHeight != (GLsizei) frameToRender.height()) {
-    SetupTextures(frameToRender);
-  }
-  UpdateTextures(frameToRender);
-
-  glDrawElements(GL_TRIANGLES, 6, GL_UNSIGNED_BYTE, g_indices);
-  checkGlError("glDrawArrays");
-
-  return 0;
-}
-
-GLuint VideoRenderOpenGles20::loadShader(GLenum shaderType,
-                                         const char* pSource) {
-  GLuint shader = glCreateShader(shaderType);
-  if (shader) {
-    glShaderSource(shader, 1, &pSource, NULL);
-    glCompileShader(shader);
-    GLint compiled = 0;
-    glGetShaderiv(shader, GL_COMPILE_STATUS, &compiled);
-    if (!compiled) {
-      GLint infoLen = 0;
-      glGetShaderiv(shader, GL_INFO_LOG_LENGTH, &infoLen);
-      if (infoLen) {
-        char* buf = (char*) malloc(infoLen);
-        if (buf) {
-          glGetShaderInfoLog(shader, infoLen, NULL, buf);
-          WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                       "%s: Could not compile shader %d: %s",
-                       __FUNCTION__, shaderType, buf);
-          free(buf);
-        }
-        glDeleteShader(shader);
-        shader = 0;
-      }
-    }
-  }
-  return shader;
-}
-
-GLuint VideoRenderOpenGles20::createProgram(const char* pVertexSource,
-                                            const char* pFragmentSource) {
-  GLuint vertexShader = loadShader(GL_VERTEX_SHADER, pVertexSource);
-  if (!vertexShader) {
-    return 0;
-  }
-
-  GLuint pixelShader = loadShader(GL_FRAGMENT_SHADER, pFragmentSource);
-  if (!pixelShader) {
-    return 0;
-  }
-
-  GLuint program = glCreateProgram();
-  if (program) {
-    glAttachShader(program, vertexShader);
-    checkGlError("glAttachShader");
-    glAttachShader(program, pixelShader);
-    checkGlError("glAttachShader");
-    glLinkProgram(program);
-    GLint linkStatus = GL_FALSE;
-    glGetProgramiv(program, GL_LINK_STATUS, &linkStatus);
-    if (linkStatus != GL_TRUE) {
-      GLint bufLength = 0;
-      glGetProgramiv(program, GL_INFO_LOG_LENGTH, &bufLength);
-      if (bufLength) {
-        char* buf = (char*) malloc(bufLength);
-        if (buf) {
-          glGetProgramInfoLog(program, bufLength, NULL, buf);
-          WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                       "%s: Could not link program: %s",
-                       __FUNCTION__, buf);
-          free(buf);
-        }
-      }
-      glDeleteProgram(program);
-      program = 0;
-    }
-  }
-  return program;
-}
-
-void VideoRenderOpenGles20::printGLString(const char *name, GLenum s) {
-  const char *v = (const char *) glGetString(s);
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "GL %s = %s\n",
-               name, v);
-}
-
-void VideoRenderOpenGles20::checkGlError(const char* op) {
-#ifdef ANDROID_LOG
-  for (GLint error = glGetError(); error; error = glGetError()) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "after %s() glError (0x%x)\n", op, error);
-  }
-#else
-  return;
-#endif
-}
-
-static void InitializeTexture(int name, int id, int width, int height) {
-  glActiveTexture(name);
-  glBindTexture(GL_TEXTURE_2D, id);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-  glTexImage2D(GL_TEXTURE_2D, 0, GL_LUMINANCE, width, height, 0,
-               GL_LUMINANCE, GL_UNSIGNED_BYTE, NULL);
-}
-
-void VideoRenderOpenGles20::SetupTextures(const VideoFrame& frameToRender) {
-  WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
-               "%s: width %d, height %d", __FUNCTION__,
-               frameToRender.width(), frameToRender.height());
-
-  const GLsizei width = frameToRender.width();
-  const GLsizei height = frameToRender.height();
-
-  glGenTextures(3, _textureIds); //Generate  the Y, U and V texture
-  InitializeTexture(GL_TEXTURE0, _textureIds[0], width, height);
-  InitializeTexture(GL_TEXTURE1, _textureIds[1], width / 2, height / 2);
-  InitializeTexture(GL_TEXTURE2, _textureIds[2], width / 2, height / 2);
-
-  checkGlError("SetupTextures");
-
-  _textureWidth = width;
-  _textureHeight = height;
-}
-
-// Uploads a plane of pixel data, accounting for stride != width*bpp.
-static void GlTexSubImage2D(GLsizei width, GLsizei height, int stride,
-                            const uint8_t* plane) {
-  if (stride == width) {
-    // Yay!  We can upload the entire plane in a single GL call.
-    glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width, height, GL_LUMINANCE,
-                    GL_UNSIGNED_BYTE,
-                    static_cast<const GLvoid*>(plane));
-  } else {
-    // Boo!  Since GLES2 doesn't have GL_UNPACK_ROW_LENGTH and Android doesn't
-    // have GL_EXT_unpack_subimage we have to upload a row at a time.  Ick.
-    for (int row = 0; row < height; ++row) {
-      glTexSubImage2D(GL_TEXTURE_2D, 0, 0, row, width, 1, GL_LUMINANCE,
-                      GL_UNSIGNED_BYTE,
-                      static_cast<const GLvoid*>(plane + (row * stride)));
-    }
-  }
-}
-
-void VideoRenderOpenGles20::UpdateTextures(const VideoFrame& frameToRender) {
-  const GLsizei width = frameToRender.width();
-  const GLsizei height = frameToRender.height();
-
-  glActiveTexture(GL_TEXTURE0);
-  glBindTexture(GL_TEXTURE_2D, _textureIds[0]);
-  GlTexSubImage2D(width, height, frameToRender.stride(kYPlane),
-                  frameToRender.buffer(kYPlane));
-
-  glActiveTexture(GL_TEXTURE1);
-  glBindTexture(GL_TEXTURE_2D, _textureIds[1]);
-  GlTexSubImage2D(width / 2, height / 2, frameToRender.stride(kUPlane),
-                  frameToRender.buffer(kUPlane));
-
-  glActiveTexture(GL_TEXTURE2);
-  glBindTexture(GL_TEXTURE_2D, _textureIds[2]);
-  GlTexSubImage2D(width / 2, height / 2, frameToRender.stride(kVPlane),
-                  frameToRender.buffer(kVPlane));
-
-  checkGlError("UpdateTextures");
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/android/video_render_opengles20.h b/chromium/third_party/webrtc/modules/video_render/android/video_render_opengles20.h
deleted file mode 100644
index 57e2a10d42e..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/android/video_render_opengles20.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
-
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-#include <GLES2/gl2.h>
-#include <GLES2/gl2ext.h>
-
-namespace webrtc
-{
-
-class VideoRenderOpenGles20 {
- public:
-  VideoRenderOpenGles20(int32_t id);
-  ~VideoRenderOpenGles20();
-
-  int32_t Setup(int32_t widht, int32_t height);
-  int32_t Render(const VideoFrame& frameToRender);
-  int32_t SetCoordinates(int32_t zOrder, const float left, const float top,
-                         const float right, const float bottom);
-
- private:
-  void printGLString(const char *name, GLenum s);
-  void checkGlError(const char* op);
-  GLuint loadShader(GLenum shaderType, const char* pSource);
-  GLuint createProgram(const char* pVertexSource,
-                       const char* pFragmentSource);
-  void SetupTextures(const VideoFrame& frameToRender);
-  void UpdateTextures(const VideoFrame& frameToRender);
-
-  int32_t _id;
-  GLuint _textureIds[3]; // Texture id of Y,U and V texture.
-  GLuint _program;
-  GLsizei _textureWidth;
-  GLsizei _textureHeight;
-
-  GLfloat _vertices[20];
-  static const char g_indices[];
-
-  static const char g_vertextShader[];
-  static const char g_fragmentShader[];
-
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/external/video_render_external_impl.cc b/chromium/third_party/webrtc/modules/video_render/external/video_render_external_impl.cc
deleted file mode 100644
index 58df07875ec..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/external/video_render_external_impl.cc
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/external/video_render_external_impl.h"
-
-namespace webrtc {
-
-VideoRenderExternalImpl::VideoRenderExternalImpl(
-                                                 const int32_t id,
-                                                 const VideoRenderType videoRenderType,
-                                                 void* window,
-                                                 const bool fullscreen) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _fullscreen(fullscreen)
-{
-}
-
-VideoRenderExternalImpl::~VideoRenderExternalImpl()
-{
-    delete &_critSect;
-}
-
-int32_t VideoRenderExternalImpl::Init()
-{
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::ChangeWindow(void* window)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-VideoRenderCallback*
-VideoRenderExternalImpl::AddIncomingRenderStream(const uint32_t streamId,
-                                                 const uint32_t zOrder,
-                                                 const float left,
-                                                 const float top,
-                                                 const float right,
-                                                 const float bottom)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return this;
-}
-
-int32_t VideoRenderExternalImpl::DeleteIncomingRenderStream(
-                                                                  const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::GetIncomingRenderStreamProperties(
-                                                                         const uint32_t streamId,
-                                                                         uint32_t& zOrder,
-                                                                         float& left,
-                                                                         float& top,
-                                                                         float& right,
-                                                                         float& bottom) const
-{
-    CriticalSectionScoped cs(&_critSect);
-
-    zOrder = 0;
-    left = 0;
-    top = 0;
-    right = 0;
-    bottom = 0;
-
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::StartRender()
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::StopRender()
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-VideoRenderType VideoRenderExternalImpl::RenderType()
-{
-    return kRenderExternal;
-}
-
-RawVideoType VideoRenderExternalImpl::PerferedVideoType()
-{
-    return kVideoI420;
-}
-
-bool VideoRenderExternalImpl::FullScreen()
-{
-    CriticalSectionScoped cs(&_critSect);
-    return _fullscreen;
-}
-
-int32_t VideoRenderExternalImpl::GetGraphicsMemory(
-                                                         uint64_t& totalGraphicsMemory,
-                                                         uint64_t& availableGraphicsMemory) const
-{
-    totalGraphicsMemory = 0;
-    availableGraphicsMemory = 0;
-    return -1;
-}
-
-int32_t VideoRenderExternalImpl::GetScreenResolution(
-                                                           uint32_t& screenWidth,
-                                                           uint32_t& screenHeight) const
-{
-    CriticalSectionScoped cs(&_critSect);
-    screenWidth = 0;
-    screenHeight = 0;
-    return 0;
-}
-
-uint32_t VideoRenderExternalImpl::RenderFrameRate(
-                                                        const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::SetStreamCropping(
-                                                         const uint32_t streamId,
-                                                         const float left,
-                                                         const float top,
-                                                         const float right,
-                                                         const float bottom)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::ConfigureRenderer(
-                                                         const uint32_t streamId,
-                                                         const unsigned int zOrder,
-                                                         const float left,
-                                                         const float top,
-                                                         const float right,
-                                                         const float bottom)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::SetTransparentBackground(
-                                                                const bool enable)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::SetText(
-                                               const uint8_t textId,
-                                               const uint8_t* text,
-                                               const int32_t textLength,
-                                               const uint32_t textColorRef,
-                                               const uint32_t backgroundColorRef,
-                                               const float left,
-                                               const float top,
-                                               const float right,
-                                               const float bottom)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-int32_t VideoRenderExternalImpl::SetBitmap(const void* bitMap,
-                                           const uint8_t pictureId,
-                                           const void* colorKey,
-                                           const float left,
-                                           const float top,
-                                           const float right,
-                                           const float bottom)
-{
-    CriticalSectionScoped cs(&_critSect);
-    return 0;
-}
-
-// VideoRenderCallback
-int32_t VideoRenderExternalImpl::RenderFrame(const uint32_t streamId,
-                                             const VideoFrame& videoFrame) {
-    return 0;
-}
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/external/video_render_external_impl.h b/chromium/third_party/webrtc/modules/video_render/external/video_render_external_impl.h
deleted file mode 100644
index a8b663fff7e..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/external/video_render_external_impl.h
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-
-namespace webrtc {
-
-// Class definitions
-class VideoRenderExternalImpl: IVideoRender, public VideoRenderCallback
-{
-public:
-    /*
-     *   Constructor/destructor
-     */
-
-    VideoRenderExternalImpl(const int32_t id,
-                            const VideoRenderType videoRenderType,
-                            void* window, const bool fullscreen);
-
-    virtual ~VideoRenderExternalImpl();
-
-    virtual int32_t Init();
-
-    virtual int32_t ChangeWindow(void* window);
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderCallback
-            * AddIncomingRenderStream(const uint32_t streamId,
-                                      const uint32_t zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t
-            DeleteIncomingRenderStream(const uint32_t streamId);
-
-    virtual int32_t
-            GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                              uint32_t& zOrder,
-                                              float& left, float& top,
-                                              float& right, float& bottom) const;
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender();
-
-    virtual int32_t StopRender();
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderType RenderType();
-
-    virtual RawVideoType PerferedVideoType();
-
-    virtual bool FullScreen();
-
-    virtual int32_t
-            GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-                              uint64_t& availableGraphicsMemory) const;
-
-    virtual int32_t
-            GetScreenResolution(uint32_t& screenWidth,
-                                uint32_t& screenHeight) const;
-
-    virtual uint32_t RenderFrameRate(const uint32_t streamId);
-
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t SetTransparentBackground(const bool enable);
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t textColorRef,
-                            const uint32_t backgroundColorRef,
-                            const float left, const float top,
-                            const float right, const float bottom);
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey, const float left,
-                              const float top, const float right,
-                              const float bottom);
-
-    // VideoRenderCallback
-    virtual int32_t RenderFrame(const uint32_t streamId,
-                                const VideoFrame& videoFrame);
-
-private:
-    CriticalSectionWrapper& _critSect;
-    bool _fullscreen;
-};
-
-}  // namespace webrtc
-
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/i_video_render.h b/chromium/third_party/webrtc/modules/video_render/i_video_render.h
deleted file mode 100644
index e6ec7a4680b..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/i_video_render.h
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
-
-#include "webrtc/modules/video_render/video_render.h"
-
-namespace webrtc {
-
-// Class definitions
-class IVideoRender
-{
-public:
-    /*
-     *   Constructor/destructor
-     */
-
-    virtual ~IVideoRender() {}
-
-    virtual int32_t Init() = 0;
-
-    virtual int32_t ChangeWindow(void* window) = 0;
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderCallback
-            * AddIncomingRenderStream(const uint32_t streamId,
-                                      const uint32_t zOrder,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom) = 0;
-
-    virtual int32_t
-            DeleteIncomingRenderStream(const uint32_t streamId) = 0;
-
-    virtual int32_t
-            GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                              uint32_t& zOrder,
-                                              float& left,
-                                              float& top,
-                                              float& right,
-                                              float& bottom) const = 0;
-    // Implemented in common code?
-    //virtual uint32_t GetNumIncomingRenderStreams() const = 0;
-    //virtual bool HasIncomingRenderStream(const uint16_t stramId) const = 0;
-
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender() = 0;
-
-    virtual int32_t StopRender() = 0;
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-    virtual VideoRenderType RenderType() = 0;
-
-    virtual RawVideoType PerferedVideoType() = 0;
-
-    virtual bool FullScreen() = 0;
-
-    // TODO: This should be treated in platform specific code only
-    virtual int32_t
-            GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-                              uint64_t& availableGraphicsMemory) const = 0;
-
-    virtual int32_t
-            GetScreenResolution(uint32_t& screenWidth,
-                                uint32_t& screenHeight) const = 0;
-
-    virtual uint32_t RenderFrameRate(const uint32_t streamId) = 0;
-
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom) = 0;
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom) = 0;
-
-    virtual int32_t SetTransparentBackground(const bool enable) = 0;
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t textColorRef,
-                            const uint32_t backgroundColorRef,
-                            const float left,
-                            const float top,
-                            const float rigth,
-                            const float bottom) = 0;
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left,
-                              const float top,
-                              const float right,
-                              const float bottom) = 0;
-
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.h b/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.h
deleted file mode 100644
index 880ddb5231f..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_IOS_OPEN_GLES20_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_IOS_OPEN_GLES20_H_
-
-#include <OpenGLES/ES2/glext.h>
-
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-/*
- * This OpenGles20 is the class of renderer for VideoFrame into a GLES 2.0
- * windows used in the VideoRenderIosView class.
- */
-namespace webrtc {
-class OpenGles20 {
- public:
-  OpenGles20();
-  ~OpenGles20();
-
-  bool Setup(int32_t width, int32_t height);
-  bool Render(const VideoFrame& frame);
-
-  // SetCoordinates
-  // Sets the coordinates where the stream shall be rendered.
-  // Values must be between 0 and 1.
-  bool SetCoordinates(const float z_order,
-                      const float left,
-                      const float top,
-                      const float right,
-                      const float bottom);
-
- private:
-  // Compile and load the vertex and fragment shaders defined at the top of
-  // open_gles20.mm
-  GLuint LoadShader(GLenum shader_type, const char* shader_source);
-
-  GLuint CreateProgram(const char* vertex_source, const char* fragment_source);
-
-  // Initialize the textures by the frame width and height
-  void SetupTextures(const VideoFrame& frame);
-
-  // Update the textures by the YUV data from the frame
-  void UpdateTextures(const VideoFrame& frame);
-
-  GLuint texture_ids_[3];  // Texture id of Y,U and V texture.
-  GLuint program_;
-  GLsizei texture_width_;
-  GLsizei texture_height_;
-
-  GLfloat vertices_[20];
-  static const char indices_[];
-  static const char vertext_shader_[];
-  static const char fragment_shader_[];
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_IOS_OPEN_GLES20_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.mm b/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.mm
deleted file mode 100644
index d1735280f26..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/open_gles20.mm
+++ /dev/null
@@ -1,330 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if !defined(__has_feature) || !__has_feature(objc_arc)
-#error "This file requires ARC support."
-#endif
-
-// This files is mostly copied from
-// webrtc/modules/video_render/android/video_render_opengles20.h
-
-// TODO(sjlee): unify this copy with the android one.
-#include "webrtc/modules/video_render/ios/open_gles20.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-using namespace webrtc;
-
-const char OpenGles20::indices_[] = {0, 3, 2, 0, 2, 1};
-
-const char OpenGles20::vertext_shader_[] = {
-    "attribute vec4 aPosition;\n"
-    "attribute vec2 aTextureCoord;\n"
-    "varying vec2 vTextureCoord;\n"
-    "void main() {\n"
-    "  gl_Position = aPosition;\n"
-    "  vTextureCoord = aTextureCoord;\n"
-    "}\n"};
-
-// The fragment shader.
-// Do YUV to RGB565 conversion.
-const char OpenGles20::fragment_shader_[] = {
-    "precision mediump float;\n"
-    "uniform sampler2D Ytex;\n"
-    "uniform sampler2D Utex,Vtex;\n"
-    "varying vec2 vTextureCoord;\n"
-    "void main(void) {\n"
-    "  float nx,ny,r,g,b,y,u,v;\n"
-    "  mediump vec4 txl,ux,vx;"
-    "  nx=vTextureCoord[0];\n"
-    "  ny=vTextureCoord[1];\n"
-    "  y=texture2D(Ytex,vec2(nx,ny)).r;\n"
-    "  u=texture2D(Utex,vec2(nx,ny)).r;\n"
-    "  v=texture2D(Vtex,vec2(nx,ny)).r;\n"
-    "  y=1.1643*(y-0.0625);\n"
-    "  u=u-0.5;\n"
-    "  v=v-0.5;\n"
-    "  r=y+1.5958*v;\n"
-    "  g=y-0.39173*u-0.81290*v;\n"
-    "  b=y+2.017*u;\n"
-    "  gl_FragColor=vec4(r,g,b,1.0);\n"
-    "}\n"};
-
-OpenGles20::OpenGles20() : texture_width_(-1), texture_height_(-1) {
-  texture_ids_[0] = 0;
-  texture_ids_[1] = 0;
-  texture_ids_[2] = 0;
-
-  program_ = 0;
-
-  const GLfloat vertices[20] = {
-      // X, Y, Z, U, V
-      -1, -1, 0, 0, 1,   // Bottom Left
-      1,  -1, 0, 1, 1,   // Bottom Right
-      1,  1,  0, 1, 0,   // Top Right
-      -1, 1,  0, 0, 0};  // Top Left
-
-  memcpy(vertices_, vertices, sizeof(vertices_));
-}
-
-OpenGles20::~OpenGles20() {
-  if (program_) {
-    glDeleteTextures(3, texture_ids_);
-    glDeleteProgram(program_);
-  }
-}
-
-bool OpenGles20::Setup(int32_t width, int32_t height) {
-  program_ = CreateProgram(vertext_shader_, fragment_shader_);
-  if (!program_) {
-    return false;
-  }
-
-  int position_handle = glGetAttribLocation(program_, "aPosition");
-  int texture_handle = glGetAttribLocation(program_, "aTextureCoord");
-
-  // set the vertices array in the shader
-  // vertices_ contains 4 vertices with 5 coordinates.
-  // 3 for (xyz) for the vertices and 2 for the texture
-  glVertexAttribPointer(
-      position_handle, 3, GL_FLOAT, false, 5 * sizeof(GLfloat), vertices_);
-
-  glEnableVertexAttribArray(position_handle);
-
-  // set the texture coordinate array in the shader
-  // vertices_ contains 4 vertices with 5 coordinates.
-  // 3 for (xyz) for the vertices and 2 for the texture
-  glVertexAttribPointer(
-      texture_handle, 2, GL_FLOAT, false, 5 * sizeof(GLfloat), &vertices_[3]);
-  glEnableVertexAttribArray(texture_handle);
-
-  glUseProgram(program_);
-  int i = glGetUniformLocation(program_, "Ytex");
-  glUniform1i(i, 0); /* Bind Ytex to texture unit 0 */
-
-  i = glGetUniformLocation(program_, "Utex");
-  glUniform1i(i, 1); /* Bind Utex to texture unit 1 */
-
-  i = glGetUniformLocation(program_, "Vtex");
-  glUniform1i(i, 2); /* Bind Vtex to texture unit 2 */
-
-  glViewport(0, 0, width, height);
-  return true;
-}
-
-bool OpenGles20::SetCoordinates(const float z_order,
-                                const float left,
-                                const float top,
-                                const float right,
-                                const float bottom) {
-  if (top > 1 || top < 0 || right > 1 || right < 0 || bottom > 1 ||
-      bottom < 0 || left > 1 || left < 0) {
-    return false;
-  }
-
-  // Bottom Left
-  vertices_[0] = (left * 2) - 1;
-  vertices_[1] = -1 * (2 * bottom) + 1;
-  vertices_[2] = z_order;
-
-  // Bottom Right
-  vertices_[5] = (right * 2) - 1;
-  vertices_[6] = -1 * (2 * bottom) + 1;
-  vertices_[7] = z_order;
-
-  // Top Right
-  vertices_[10] = (right * 2) - 1;
-  vertices_[11] = -1 * (2 * top) + 1;
-  vertices_[12] = z_order;
-
-  // Top Left
-  vertices_[15] = (left * 2) - 1;
-  vertices_[16] = -1 * (2 * top) + 1;
-  vertices_[17] = z_order;
-
-  return true;
-}
-
-bool OpenGles20::Render(const VideoFrame& frame) {
-  if (texture_width_ != (GLsizei)frame.width() ||
-      texture_height_ != (GLsizei)frame.height()) {
-    SetupTextures(frame);
-  }
-  UpdateTextures(frame);
-
-  glDrawElements(GL_TRIANGLES, 6, GL_UNSIGNED_BYTE, indices_);
-
-  return true;
-}
-
-GLuint OpenGles20::LoadShader(GLenum shader_type, const char* shader_source) {
-  GLuint shader = glCreateShader(shader_type);
-  if (shader) {
-    glShaderSource(shader, 1, &shader_source, NULL);
-    glCompileShader(shader);
-
-    GLint compiled = 0;
-    glGetShaderiv(shader, GL_COMPILE_STATUS, &compiled);
-    if (!compiled) {
-      GLint info_len = 0;
-      glGetShaderiv(shader, GL_INFO_LOG_LENGTH, &info_len);
-      if (info_len) {
-        char* buf = (char*)malloc(info_len);
-        glGetShaderInfoLog(shader, info_len, NULL, buf);
-        WEBRTC_TRACE(kTraceError,
-                     kTraceVideoRenderer,
-                     0,
-                     "%s: Could not compile shader %d: %s",
-                     __FUNCTION__,
-                     shader_type,
-                     buf);
-        free(buf);
-      }
-      glDeleteShader(shader);
-      shader = 0;
-    }
-  }
-  return shader;
-}
-
-GLuint OpenGles20::CreateProgram(const char* vertex_source,
-                                 const char* fragment_source) {
-  GLuint vertex_shader = LoadShader(GL_VERTEX_SHADER, vertex_source);
-  if (!vertex_shader) {
-    return -1;
-  }
-
-  GLuint fragment_shader = LoadShader(GL_FRAGMENT_SHADER, fragment_source);
-  if (!fragment_shader) {
-    return -1;
-  }
-
-  GLuint program = glCreateProgram();
-  if (program) {
-    glAttachShader(program, vertex_shader);
-    glAttachShader(program, fragment_shader);
-    glLinkProgram(program);
-    GLint link_status = GL_FALSE;
-    glGetProgramiv(program, GL_LINK_STATUS, &link_status);
-    if (link_status != GL_TRUE) {
-      GLint info_len = 0;
-      glGetProgramiv(program, GL_INFO_LOG_LENGTH, &info_len);
-      if (info_len) {
-        char* buf = (char*)malloc(info_len);
-        glGetProgramInfoLog(program, info_len, NULL, buf);
-        WEBRTC_TRACE(kTraceError,
-                     kTraceVideoRenderer,
-                     0,
-                     "%s: Could not link program: %s",
-                     __FUNCTION__,
-                     buf);
-        free(buf);
-      }
-      glDeleteProgram(program);
-      program = 0;
-    }
-  }
-
-  if (vertex_shader) {
-    glDeleteShader(vertex_shader);
-  }
-
-  if (fragment_shader) {
-    glDeleteShader(fragment_shader);
-  }
-
-  return program;
-}
-
-static void InitializeTexture(int name, int id, int width, int height) {
-  glActiveTexture(name);
-  glBindTexture(GL_TEXTURE_2D, id);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
-  glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-  glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
-  glTexImage2D(GL_TEXTURE_2D,
-               0,
-               GL_LUMINANCE,
-               width,
-               height,
-               0,
-               GL_LUMINANCE,
-               GL_UNSIGNED_BYTE,
-               NULL);
-}
-
-void OpenGles20::SetupTextures(const VideoFrame& frame) {
-  const GLsizei width = frame.width();
-  const GLsizei height = frame.height();
-
-  if (!texture_ids_[0]) {
-    glGenTextures(3, texture_ids_);  // Generate  the Y, U and V texture
-  }
-
-  InitializeTexture(GL_TEXTURE0, texture_ids_[0], width, height);
-  InitializeTexture(GL_TEXTURE1, texture_ids_[1], width / 2, height / 2);
-  InitializeTexture(GL_TEXTURE2, texture_ids_[2], width / 2, height / 2);
-
-  texture_width_ = width;
-  texture_height_ = height;
-}
-
-// Uploads a plane of pixel data, accounting for stride != width*bpp.
-static void GlTexSubImage2D(GLsizei width,
-                            GLsizei height,
-                            int stride,
-                            const uint8_t* plane) {
-  if (stride == width) {
-    // Yay!  We can upload the entire plane in a single GL call.
-    glTexSubImage2D(GL_TEXTURE_2D,
-                    0,
-                    0,
-                    0,
-                    width,
-                    height,
-                    GL_LUMINANCE,
-                    GL_UNSIGNED_BYTE,
-                    static_cast<const GLvoid*>(plane));
-  } else {
-    // Boo!  Since GLES2 doesn't have GL_UNPACK_ROW_LENGTH and iOS doesn't
-    // have GL_EXT_unpack_subimage we have to upload a row at a time.  Ick.
-    for (int row = 0; row < height; ++row) {
-      glTexSubImage2D(GL_TEXTURE_2D,
-                      0,
-                      0,
-                      row,
-                      width,
-                      1,
-                      GL_LUMINANCE,
-                      GL_UNSIGNED_BYTE,
-                      static_cast<const GLvoid*>(plane + (row * stride)));
-    }
-  }
-}
-
-void OpenGles20::UpdateTextures(const VideoFrame& frame) {
-  const GLsizei width = frame.width();
-  const GLsizei height = frame.height();
-
-  glActiveTexture(GL_TEXTURE0);
-  glBindTexture(GL_TEXTURE_2D, texture_ids_[0]);
-  GlTexSubImage2D(width, height, frame.stride(kYPlane), frame.buffer(kYPlane));
-
-  glActiveTexture(GL_TEXTURE1);
-  glBindTexture(GL_TEXTURE_2D, texture_ids_[1]);
-  GlTexSubImage2D(
-      width / 2, height / 2, frame.stride(kUPlane), frame.buffer(kUPlane));
-
-  glActiveTexture(GL_TEXTURE2);
-  glBindTexture(GL_TEXTURE_2D, texture_ids_[2]);
-  GlTexSubImage2D(
-      width / 2, height / 2, frame.stride(kVPlane), frame.buffer(kVPlane));
-}
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.h b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.h
deleted file mode 100644
index a15ba393dc0..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_CHANNEL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_CHANNEL_H_
-
-#include "webrtc/modules/video_render/video_render_defines.h"
-#include "webrtc/modules/video_render/ios/video_render_ios_view.h"
-
-namespace webrtc {
-
-class VideoRenderIosGles20;
-
-class VideoRenderIosChannel : public VideoRenderCallback {
- public:
-  explicit VideoRenderIosChannel(VideoRenderIosView* view);
-  virtual ~VideoRenderIosChannel();
-
-  // Implementation of VideoRenderCallback.
-  int32_t RenderFrame(const uint32_t stream_id,
-                      const VideoFrame& video_frame) override;
-
-  int SetStreamSettings(const float z_order,
-                        const float left,
-                        const float top,
-                        const float right,
-                        const float bottom);
-  bool IsUpdated();
-  bool RenderOffScreenBuffer();
-
- private:
-  VideoRenderIosView* view_;
-  VideoFrame* current_frame_;
-  bool buffer_is_updated_;
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_CHANNEL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.mm
deleted file mode 100644
index b2b15857f93..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_channel.mm
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if !defined(__has_feature) || !__has_feature(objc_arc)
-#error "This file requires ARC support."
-#endif
-
-#include "webrtc/modules/video_render/ios/video_render_ios_channel.h"
-
-using namespace webrtc;
-
-VideoRenderIosChannel::VideoRenderIosChannel(VideoRenderIosView* view)
-    : view_(view), current_frame_(new VideoFrame()), buffer_is_updated_(false) {
-}
-
-VideoRenderIosChannel::~VideoRenderIosChannel() { delete current_frame_; }
-
-int32_t VideoRenderIosChannel::RenderFrame(const uint32_t stream_id,
-                                           const VideoFrame& video_frame) {
-  current_frame_->CopyFrame(video_frame);
-  current_frame_->set_render_time_ms(0);
-  buffer_is_updated_ = true;
-
-  return 0;
-}
-
-bool VideoRenderIosChannel::RenderOffScreenBuffer() {
-  if (![view_ renderFrame:current_frame_]) {
-    return false;
-  }
-
-  buffer_is_updated_ = false;
-
-  return true;
-}
-
-bool VideoRenderIosChannel::IsUpdated() { return buffer_is_updated_; }
-
-int VideoRenderIosChannel::SetStreamSettings(const float z_order,
-                                             const float left,
-                                             const float top,
-                                             const float right,
-                                             const float bottom) {
-  if (![view_ setCoordinatesForZOrder:z_order
-                                 Left:left
-                                  Top:bottom
-                                Right:right
-                               Bottom:top]) {
-
-    return -1;
-  }
-
-  return 0;
-}
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.h b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.h
deleted file mode 100644
index d4e04e79d73..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_GLES20_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_GLES20_H_
-
-#include <list>
-#include <map>
-#include <memory>
-
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/modules/video_render/ios/video_render_ios_channel.h"
-#include "webrtc/modules/video_render/ios/video_render_ios_view.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class EventTimerWrapper;
-
-class VideoRenderIosGles20 {
- public:
-  VideoRenderIosGles20(VideoRenderIosView* view,
-                       bool full_screen,
-                       int render_id);
-  virtual ~VideoRenderIosGles20();
-
-  int Init();
-  VideoRenderIosChannel* CreateEaglChannel(int channel,
-                                           int z_order,
-                                           float left,
-                                           float top,
-                                           float right,
-                                           float bottom);
-  int DeleteEaglChannel(int channel);
-  bool HasChannel(int channel);
-  bool ScreenUpdateProcess();
-  int GetWindowRect(Rect& rect);  // NOLINT
-
-  int GetScreenResolution(uint& screen_width, uint& screen_height);  // NOLINT
-  int SetStreamCropping(const uint stream_id,
-                        const float left,
-                        const float top,
-                        const float right,
-                        const float bottom);
-
-  int ChangeWindow(void* new_window);
-  int StartRender();
-  int StopRender();
-
- protected:
-  static bool ScreenUpdateThreadProc(void* obj);
-
- private:
-  bool RenderOffScreenBuffers();
-  int SwapAndDisplayBuffers();
-
- private:
-  std::unique_ptr<CriticalSectionWrapper> gles_crit_sec_;
-  EventTimerWrapper* screen_update_event_;
-  // TODO(pbos): Remove unique_ptr and use member directly.
-  std::unique_ptr<rtc::PlatformThread> screen_update_thread_;
-
-  VideoRenderIosView* view_;
-  Rect window_rect_;
-  int window_width_;
-  int window_height_;
-  bool is_full_screen_;
-  GLint backing_width_;
-  GLint backing_height_;
-  GLuint view_renderbuffer_;
-  GLuint view_framebuffer_;
-  GLuint depth_renderbuffer_;
-  std::map<int, VideoRenderIosChannel*> agl_channels_;
-  std::multimap<int, int> z_order_to_channel_;
-  EAGLContext* gles_context_;
-  bool is_rendering_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_GLES20_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
deleted file mode 100644
index 6ad5db8b8cb..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_gles20.mm
+++ /dev/null
@@ -1,285 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if !defined(__has_feature) || !__has_feature(objc_arc)
-#error "This file requires ARC support."
-#endif
-
-#include "webrtc/modules/video_render/ios/video_render_ios_gles20.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-
-using namespace webrtc;
-
-VideoRenderIosGles20::VideoRenderIosGles20(VideoRenderIosView* view,
-                                           bool full_screen,
-                                           int render_id)
-    : gles_crit_sec_(CriticalSectionWrapper::CreateCriticalSection()),
-      screen_update_event_(0),
-      view_(view),
-      window_rect_(),
-      window_width_(0),
-      window_height_(0),
-      is_full_screen_(full_screen),
-      agl_channels_(),
-      z_order_to_channel_(),
-      gles_context_([view context]),
-      is_rendering_(true) {
-  screen_update_thread_.reset(new rtc::PlatformThread(
-      ScreenUpdateThreadProc, this, "ScreenUpdateGles20"));
-  screen_update_event_ = EventTimerWrapper::Create();
-  GetWindowRect(window_rect_);
-}
-
-VideoRenderIosGles20::~VideoRenderIosGles20() {
-  // Signal event to exit thread, then delete it
-  rtc::PlatformThread* thread_wrapper = screen_update_thread_.release();
-
-  if (thread_wrapper) {
-    screen_update_event_->Set();
-    screen_update_event_->StopTimer();
-
-    thread_wrapper->Stop();
-    delete thread_wrapper;
-    delete screen_update_event_;
-    screen_update_event_ = NULL;
-    is_rendering_ = FALSE;
-  }
-
-  // Delete all channels
-  std::map<int, VideoRenderIosChannel*>::iterator it = agl_channels_.begin();
-  while (it != agl_channels_.end()) {
-    delete it->second;
-    agl_channels_.erase(it);
-    it = agl_channels_.begin();
-  }
-  agl_channels_.clear();
-
-  // Clean the zOrder map
-  std::multimap<int, int>::iterator z_it = z_order_to_channel_.begin();
-  while (z_it != z_order_to_channel_.end()) {
-    z_order_to_channel_.erase(z_it);
-    z_it = z_order_to_channel_.begin();
-  }
-  z_order_to_channel_.clear();
-}
-
-int VideoRenderIosGles20::Init() {
-  CriticalSectionScoped cs(gles_crit_sec_.get());
-
-  if (!view_) {
-    view_ = [[VideoRenderIosView alloc] init];
-  }
-
-  if (![view_ createContext]) {
-    return -1;
-  }
-
-  screen_update_thread_->Start();
-  screen_update_thread_->SetPriority(rtc::kRealtimePriority);
-
-  // Start the event triggering the render process
-  unsigned int monitor_freq = 60;
-  screen_update_event_->StartTimer(true, 1000 / monitor_freq);
-
-  window_width_ = window_rect_.right - window_rect_.left;
-  window_height_ = window_rect_.bottom - window_rect_.top;
-
-  return 0;
-}
-
-VideoRenderIosChannel* VideoRenderIosGles20::CreateEaglChannel(int channel,
-                                                               int z_order,
-                                                               float left,
-                                                               float top,
-                                                               float right,
-                                                               float bottom) {
-  CriticalSectionScoped cs(gles_crit_sec_.get());
-
-  if (HasChannel(channel)) {
-    return NULL;
-  }
-
-  VideoRenderIosChannel* new_eagl_channel = new VideoRenderIosChannel(view_);
-
-  if (new_eagl_channel->SetStreamSettings(z_order, left, top, right, bottom) ==
-      -1) {
-    return NULL;
-  }
-
-  agl_channels_[channel] = new_eagl_channel;
-  z_order_to_channel_.insert(std::pair<int, int>(z_order, channel));
-
-  return new_eagl_channel;
-}
-
-int VideoRenderIosGles20::DeleteEaglChannel(int channel) {
-  CriticalSectionScoped cs(gles_crit_sec_.get());
-
-  std::map<int, VideoRenderIosChannel*>::iterator it;
-  it = agl_channels_.find(channel);
-  if (it != agl_channels_.end()) {
-    delete it->second;
-    agl_channels_.erase(it);
-  } else {
-    return -1;
-  }
-
-  std::multimap<int, int>::iterator z_it = z_order_to_channel_.begin();
-  while (z_it != z_order_to_channel_.end()) {
-    if (z_it->second == channel) {
-      z_order_to_channel_.erase(z_it);
-      break;
-    }
-    z_it++;
-  }
-
-  return 0;
-}
-
-bool VideoRenderIosGles20::HasChannel(int channel) {
-  CriticalSectionScoped cs(gles_crit_sec_.get());
-
-  std::map<int, VideoRenderIosChannel*>::iterator it =
-      agl_channels_.find(channel);
-
-  if (it != agl_channels_.end()) {
-    return true;
-  }
-
-  return false;
-}
-
-// Rendering process
-bool VideoRenderIosGles20::ScreenUpdateThreadProc(void* obj) {
-  return static_cast<VideoRenderIosGles20*>(obj)->ScreenUpdateProcess();
-}
-
-bool VideoRenderIosGles20::ScreenUpdateProcess() {
-  screen_update_event_->Wait(100);
-
-  CriticalSectionScoped cs(gles_crit_sec_.get());
-
-  if (!is_rendering_) {
-    return false;
-  }
-
-  if (!screen_update_thread_) {
-    return false;
-  }
-
-  if (GetWindowRect(window_rect_) == -1) {
-    return true;
-  }
-
-  if (window_width_ != (window_rect_.right - window_rect_.left) ||
-      window_height_ != (window_rect_.bottom - window_rect_.top)) {
-    window_width_ = window_rect_.right - window_rect_.left;
-    window_height_ = window_rect_.bottom - window_rect_.top;
-  }
-
-  // Check if there are any updated buffers
-  bool updated = false;
-
-  std::map<int, VideoRenderIosChannel*>::iterator it = agl_channels_.begin();
-  while (it != agl_channels_.end()) {
-    VideoRenderIosChannel* agl_channel = it->second;
-
-    updated = agl_channel->IsUpdated();
-    if (updated) {
-      break;
-    }
-    it++;
-  }
-
-  if (updated) {
-    // At least one buffer has been updated, we need to repaint the texture
-    // Loop through all channels starting highest zOrder ending with lowest.
-    for (std::multimap<int, int>::reverse_iterator r_it =
-             z_order_to_channel_.rbegin();
-         r_it != z_order_to_channel_.rend();
-         r_it++) {
-      int channel_id = r_it->second;
-      std::map<int, VideoRenderIosChannel*>::iterator it =
-          agl_channels_.find(channel_id);
-
-      VideoRenderIosChannel* agl_channel = it->second;
-
-      agl_channel->RenderOffScreenBuffer();
-    }
-
-    [view_ presentFramebuffer];
-  }
-
-  return true;
-}
-
-int VideoRenderIosGles20::GetWindowRect(Rect& rect) {
-  CriticalSectionScoped cs(gles_crit_sec_.get());
-
-  if (!view_) {
-    return -1;
-  }
-
-  CGRect bounds = [view_ bounds];
-  rect.top = bounds.origin.y;
-  rect.left = bounds.origin.x;
-  rect.bottom = bounds.size.height + bounds.origin.y;
-  rect.right = bounds.size.width + bounds.origin.x;
-
-  return 0;
-}
-
-int VideoRenderIosGles20::ChangeWindow(void* new_window) {
-  CriticalSectionScoped cs(gles_crit_sec_.get());
-
-  view_ = (__bridge VideoRenderIosView*)new_window;
-
-  return 0;
-}
-
-int VideoRenderIosGles20::StartRender() {
-  is_rendering_ = true;
-  return 0;
-}
-
-int VideoRenderIosGles20::StopRender() {
-  is_rendering_ = false;
-  return 0;
-}
-
-int VideoRenderIosGles20::GetScreenResolution(uint& screen_width,
-                                              uint& screen_height) {
-  screen_width = [view_ bounds].size.width;
-  screen_height = [view_ bounds].size.height;
-  return 0;
-}
-
-int VideoRenderIosGles20::SetStreamCropping(const uint stream_id,
-                                            const float left,
-                                            const float top,
-                                            const float right,
-                                            const float bottom) {
-  // Check if there are any updated buffers
-  // bool updated = false;
-  uint counter = 0;
-
-  std::map<int, VideoRenderIosChannel*>::iterator it = agl_channels_.begin();
-  while (it != agl_channels_.end()) {
-    if (counter == stream_id) {
-      VideoRenderIosChannel* agl_channel = it->second;
-      agl_channel->SetStreamSettings(0, left, top, right, bottom);
-    }
-    counter++;
-    it++;
-  }
-
-  return 0;
-}
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.h b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.h
deleted file mode 100644
index 04a74933008..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.h
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_IMPL_H_
-
-#include <list>
-#include <map>
-#include <memory>
-
-#include "webrtc/modules/video_render/i_video_render.h"
-
-namespace webrtc {
-
-class VideoRenderIosGles20;
-class CriticalSectionWrapper;
-
-class VideoRenderIosImpl : IVideoRender {
- public:
-  explicit VideoRenderIosImpl(const int32_t id,
-                              void* window,
-                              const bool full_screen);
-
-  ~VideoRenderIosImpl();
-
-  // Implementation of IVideoRender.
-  int32_t Init() override;
-  int32_t ChangeWindow(void* window) override;
-
-  VideoRenderCallback* AddIncomingRenderStream(const uint32_t stream_id,
-                                               const uint32_t z_order,
-                                               const float left,
-                                               const float top,
-                                               const float right,
-                                               const float bottom) override;
-
-  int32_t DeleteIncomingRenderStream(const uint32_t stream_id) override;
-
-  int32_t GetIncomingRenderStreamProperties(const uint32_t stream_id,
-                                            uint32_t& z_order,
-                                            float& left,
-                                            float& top,
-                                            float& right,
-                                            float& bottom) const override;
-
-  int32_t StartRender() override;
-  int32_t StopRender() override;
-
-  VideoRenderType RenderType() override;
-  RawVideoType PerferedVideoType() override;
-  bool FullScreen() override;
-  int32_t GetGraphicsMemory(
-      uint64_t& total_graphics_memory,
-      uint64_t& available_graphics_memory) const override;  // NOLINT
-  int32_t GetScreenResolution(
-      uint32_t& screen_width,
-      uint32_t& screen_height) const override;  // NOLINT
-  uint32_t RenderFrameRate(const uint32_t stream_id);
-  int32_t SetStreamCropping(const uint32_t stream_id,
-                            const float left,
-                            const float top,
-                            const float right,
-                            const float bottom) override;
-  int32_t ConfigureRenderer(const uint32_t stream_id,
-                            const unsigned int z_order,
-                            const float left,
-                            const float top,
-                            const float right,
-                            const float bottom) override;
-  int32_t SetTransparentBackground(const bool enable) override;
-  int32_t SetText(const uint8_t text_id,
-                  const uint8_t* text,
-                  const int32_t text_length,
-                  const uint32_t text_color_ref,
-                  const uint32_t background_color_ref,
-                  const float left,
-                  const float top,
-                  const float right,
-                  const float bottom) override;
-  int32_t SetBitmap(const void* bit_map,
-                    const uint8_t picture_id,
-                    const void* color_key,
-                    const float left,
-                    const float top,
-                    const float right,
-                    const float bottom);
-  int32_t FullScreenRender(void* window, const bool enable);
-
- private:
-  int32_t id_;
-  void* ptr_window_;
-  bool full_screen_;
-
-  CriticalSectionWrapper* crit_sec_;
-  std::unique_ptr<VideoRenderIosGles20> ptr_ios_render_;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_IOS_VIDEO_RENDER_IOS_IMPL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.mm
deleted file mode 100644
index 0ef411d56f8..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_impl.mm
+++ /dev/null
@@ -1,170 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if !defined(__has_feature) || !__has_feature(objc_arc)
-#error "This file requires ARC support."
-#endif
-
-#include "webrtc/modules/video_render/ios/video_render_ios_impl.h"
-#include "webrtc/modules/video_render/ios/video_render_ios_gles20.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-using namespace webrtc;
-
-#define IOS_UNSUPPORTED()                                  \
-  WEBRTC_TRACE(kTraceError,                                \
-               kTraceVideoRenderer,                        \
-               id_,                                        \
-               "%s is not supported on the iOS platform.", \
-               __FUNCTION__);                              \
-  return -1;
-
-VideoRenderIosImpl::VideoRenderIosImpl(const int32_t id,
-                                       void* window,
-                                       const bool full_screen)
-    : id_(id),
-      ptr_window_(window),
-      full_screen_(full_screen),
-      crit_sec_(CriticalSectionWrapper::CreateCriticalSection()) {}
-
-VideoRenderIosImpl::~VideoRenderIosImpl() {
-  delete crit_sec_;
-}
-
-int32_t VideoRenderIosImpl::Init() {
-  CriticalSectionScoped cs(crit_sec_);
-
-  ptr_ios_render_.reset(new VideoRenderIosGles20(
-      (__bridge VideoRenderIosView*)ptr_window_, full_screen_, id_));
-
-  return ptr_ios_render_->Init();
-  ;
-}
-
-int32_t VideoRenderIosImpl::ChangeWindow(void* window) {
-  CriticalSectionScoped cs(crit_sec_);
-  if (window == NULL) {
-    return -1;
-  }
-
-  ptr_window_ = window;
-
-  return ptr_ios_render_->ChangeWindow(ptr_window_);
-}
-
-VideoRenderCallback* VideoRenderIosImpl::AddIncomingRenderStream(
-    const uint32_t stream_id,
-    const uint32_t z_order,
-    const float left,
-    const float top,
-    const float right,
-    const float bottom) {
-  CriticalSectionScoped cs(crit_sec_);
-  if (!ptr_window_) {
-    return NULL;
-  }
-
-  return ptr_ios_render_->CreateEaglChannel(
-      stream_id, z_order, left, top, right, bottom);
-}
-
-int32_t VideoRenderIosImpl::DeleteIncomingRenderStream(
-    const uint32_t stream_id) {
-  CriticalSectionScoped cs(crit_sec_);
-
-  return ptr_ios_render_->DeleteEaglChannel(stream_id);
-}
-
-int32_t VideoRenderIosImpl::GetIncomingRenderStreamProperties(
-    const uint32_t stream_id,
-    uint32_t& z_order,
-    float& left,
-    float& top,
-    float& right,
-    float& bottom) const {
-  IOS_UNSUPPORTED();
-}
-
-int32_t VideoRenderIosImpl::StartRender() {
-  return ptr_ios_render_->StartRender();
-}
-
-int32_t VideoRenderIosImpl::StopRender() {
-  return ptr_ios_render_->StopRender();
-}
-
-VideoRenderType VideoRenderIosImpl::RenderType() { return kRenderiOS; }
-
-RawVideoType VideoRenderIosImpl::PerferedVideoType() { return kVideoI420; }
-
-bool VideoRenderIosImpl::FullScreen() { IOS_UNSUPPORTED(); }
-
-int32_t VideoRenderIosImpl::GetGraphicsMemory(
-    uint64_t& totalGraphicsMemory,
-    uint64_t& availableGraphicsMemory) const {
-  IOS_UNSUPPORTED();
-}
-
-int32_t VideoRenderIosImpl::GetScreenResolution(uint32_t& screenWidth,
-                                                uint32_t& screenHeight) const {
-  return ptr_ios_render_->GetScreenResolution(screenWidth, screenHeight);
-}
-
-uint32_t VideoRenderIosImpl::RenderFrameRate(const uint32_t streamId) {
-  IOS_UNSUPPORTED();
-}
-
-int32_t VideoRenderIosImpl::SetStreamCropping(const uint32_t streamId,
-                                              const float left,
-                                              const float top,
-                                              const float right,
-                                              const float bottom) {
-  return ptr_ios_render_->SetStreamCropping(streamId, left, top, right, bottom);
-}
-
-int32_t VideoRenderIosImpl::ConfigureRenderer(const uint32_t streamId,
-                                              const unsigned int zOrder,
-                                              const float left,
-                                              const float top,
-                                              const float right,
-                                              const float bottom) {
-  IOS_UNSUPPORTED();
-}
-
-int32_t VideoRenderIosImpl::SetTransparentBackground(const bool enable) {
-  IOS_UNSUPPORTED();
-}
-
-int32_t VideoRenderIosImpl::SetText(const uint8_t textId,
-                                    const uint8_t* text,
-                                    const int32_t textLength,
-                                    const uint32_t textColorRef,
-                                    const uint32_t backgroundColorRef,
-                                    const float left,
-                                    const float top,
-                                    const float right,
-                                    const float bottom) {
-  IOS_UNSUPPORTED();
-}
-
-int32_t VideoRenderIosImpl::SetBitmap(const void* bitMap,
-                                      const uint8_t pictureId,
-                                      const void* colorKey,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom) {
-  IOS_UNSUPPORTED();
-}
-
-int32_t VideoRenderIosImpl::FullScreenRender(void* window, const bool enable) {
-  IOS_UNSUPPORTED();
-}
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.h b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.h
deleted file mode 100644
index d110bc78bd2..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_IOS_RENDER_VIEW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_IOS_RENDER_VIEW_H_
-
-#import <UIKit/UIKit.h>
-#import <QuartzCore/QuartzCore.h>
-
-#include "webrtc/modules/video_render/ios/open_gles20.h"
-
-@interface VideoRenderIosView : UIView
-
-- (BOOL)createContext;
-- (BOOL)presentFramebuffer;
-- (BOOL)renderFrame:(webrtc::VideoFrame*)frameToRender;
-- (BOOL)setCoordinatesForZOrder:(const float)zOrder
-                           Left:(const float)left
-                            Top:(const float)top
-                          Right:(const float)right
-                         Bottom:(const float)bottom;
-
-@property(nonatomic, retain) EAGLContext* context;
-
-@end
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_IOS_RENDER_VIEW_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.mm b/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.mm
deleted file mode 100644
index b106ffa5c4f..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/ios/video_render_ios_view.mm
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if !defined(__has_feature) || !__has_feature(objc_arc)
-#error "This file requires ARC support."
-#endif
-
-#include <memory>
-
-#include "webrtc/modules/video_render/ios/video_render_ios_view.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-using namespace webrtc;
-
-@implementation VideoRenderIosView {
-  EAGLContext* _context;
-  std::unique_ptr<webrtc::OpenGles20> _gles_renderer20;
-  int _frameBufferWidth;
-  int _frameBufferHeight;
-  unsigned int _defaultFrameBuffer;
-  unsigned int _colorRenderBuffer;
-}
-
-@synthesize context = context_;
-
-+ (Class)layerClass {
-  return [CAEAGLLayer class];
-}
-
-- (id)initWithCoder:(NSCoder*)coder {
-  // init super class
-  self = [super initWithCoder:coder];
-  if (self) {
-    _gles_renderer20.reset(new OpenGles20());
-  }
-  return self;
-}
-
-- (id)init {
-  // init super class
-  self = [super init];
-  if (self) {
-    _gles_renderer20.reset(new OpenGles20());
-  }
-  return self;
-}
-
-- (id)initWithFrame:(CGRect)frame {
-  // init super class
-  self = [super initWithFrame:frame];
-  if (self) {
-    _gles_renderer20.reset(new OpenGles20());
-  }
-  return self;
-}
-
-- (void)dealloc {
-  if (_defaultFrameBuffer) {
-    glDeleteFramebuffers(1, &_defaultFrameBuffer);
-    _defaultFrameBuffer = 0;
-  }
-
-  if (_colorRenderBuffer) {
-    glDeleteRenderbuffers(1, &_colorRenderBuffer);
-    _colorRenderBuffer = 0;
-  }
-
-  [EAGLContext setCurrentContext:nil];
-}
-
-- (NSString*)description {
-  return [NSString stringWithFormat:
-          @"A WebRTC implemented subclass of UIView."
-          "+Class method is overwritten, along with custom methods"];
-}
-
-- (BOOL)createContext {
-  // create OpenGLES context from self layer class
-  CAEAGLLayer* eagl_layer = (CAEAGLLayer*)self.layer;
-  eagl_layer.opaque = YES;
-  eagl_layer.drawableProperties =
-      [NSDictionary dictionaryWithObjectsAndKeys:[NSNumber numberWithBool:NO],
-          kEAGLDrawablePropertyRetainedBacking,
-          kEAGLColorFormatRGBA8,
-          kEAGLDrawablePropertyColorFormat,
-          nil];
-  _context = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2];
-
-  if (!_context) {
-    return NO;
-  }
-
-  if (![EAGLContext setCurrentContext:_context]) {
-    return NO;
-  }
-
-  // generates and binds the OpenGLES buffers
-  glGenFramebuffers(1, &_defaultFrameBuffer);
-  glBindFramebuffer(GL_FRAMEBUFFER, _defaultFrameBuffer);
-
-  // Create color render buffer and allocate backing store.
-  glGenRenderbuffers(1, &_colorRenderBuffer);
-  glBindRenderbuffer(GL_RENDERBUFFER, _colorRenderBuffer);
-  [_context renderbufferStorage:GL_RENDERBUFFER
-                   fromDrawable:(CAEAGLLayer*)self.layer];
-  glGetRenderbufferParameteriv(
-      GL_RENDERBUFFER, GL_RENDERBUFFER_WIDTH, &_frameBufferWidth);
-  glGetRenderbufferParameteriv(
-      GL_RENDERBUFFER, GL_RENDERBUFFER_HEIGHT, &_frameBufferHeight);
-  glFramebufferRenderbuffer(GL_FRAMEBUFFER,
-                            GL_COLOR_ATTACHMENT0,
-                            GL_RENDERBUFFER,
-                            _colorRenderBuffer);
-
-  if (glCheckFramebufferStatus(GL_FRAMEBUFFER) != GL_FRAMEBUFFER_COMPLETE) {
-    return NO;
-  }
-
-  // set the frame buffer
-  glBindFramebuffer(GL_FRAMEBUFFER, _defaultFrameBuffer);
-  glViewport(0, 0, self.frame.size.width, self.frame.size.height);
-
-  return _gles_renderer20->Setup([self bounds].size.width,
-                                 [self bounds].size.height);
-}
-
-- (BOOL)presentFramebuffer {
-  if (![_context presentRenderbuffer:GL_RENDERBUFFER]) {
-    WEBRTC_TRACE(kTraceWarning,
-                 kTraceVideoRenderer,
-                 0,
-                 "%s:%d [context present_renderbuffer] "
-                 "returned false",
-                 __FUNCTION__,
-                 __LINE__);
-  }
-  return YES;
-}
-
-- (BOOL)renderFrame:(VideoFrame*)frameToRender {
-  if (![EAGLContext setCurrentContext:_context]) {
-    return NO;
-  }
-
-  return _gles_renderer20->Render(*frameToRender);
-}
-
-- (BOOL)setCoordinatesForZOrder:(const float)zOrder
-                           Left:(const float)left
-                            Top:(const float)top
-                          Right:(const float)right
-                         Bottom:(const float)bottom {
-  return _gles_renderer20->SetCoordinates(zOrder, left, top, right, bottom);
-}
-
-@end
diff --git a/chromium/third_party/webrtc/modules/video_render/linux/video_render_linux_impl.cc b/chromium/third_party/webrtc/modules/video_render/linux/video_render_linux_impl.cc
deleted file mode 100644
index 7e53dfdf809..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/linux/video_render_linux_impl.cc
+++ /dev/null
@@ -1,261 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/linux/video_render_linux_impl.h"
-
-#include "webrtc/modules/video_render/linux/video_x11_render.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-#include <X11/Xlib.h>
-
-namespace webrtc {
-
-VideoRenderLinuxImpl::VideoRenderLinuxImpl(
-                                           const int32_t id,
-                                           const VideoRenderType videoRenderType,
-                                           void* window, const bool fullscreen) :
-            _id(id),
-            _renderLinuxCritsect(
-                                 *CriticalSectionWrapper::CreateCriticalSection()),
-            _ptrWindow(window), _ptrX11Render(NULL)
-{
-}
-
-VideoRenderLinuxImpl::~VideoRenderLinuxImpl()
-{
-    if (_ptrX11Render)
-        delete _ptrX11Render;
-
-    delete &_renderLinuxCritsect;
-}
-
-int32_t VideoRenderLinuxImpl::Init()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
-                 __FUNCTION__);
-
-    CriticalSectionScoped cs(&_renderLinuxCritsect);
-    _ptrX11Render = new VideoX11Render((Window) _ptrWindow);
-    if (!_ptrX11Render)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s",
-                     "Failed to create instance of VideoX11Render object");
-        return -1;
-    }
-    int retVal = _ptrX11Render->Init();
-    if (retVal == -1)
-    {
-        return -1;
-    }
-
-    return 0;
-
-}
-
-int32_t VideoRenderLinuxImpl::ChangeWindow(void* window)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
-                 __FUNCTION__);
-
-    CriticalSectionScoped cs(&_renderLinuxCritsect);
-    _ptrWindow = window;
-
-    if (_ptrX11Render)
-    {
-        return _ptrX11Render->ChangeWindow((Window) window);
-    }
-
-    return -1;
-}
-
-VideoRenderCallback* VideoRenderLinuxImpl::AddIncomingRenderStream(
-                                                                       const uint32_t streamId,
-                                                                       const uint32_t zOrder,
-                                                                       const float left,
-                                                                       const float top,
-                                                                       const float right,
-                                                                       const float bottom)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
-                 __FUNCTION__);
-    CriticalSectionScoped cs(&_renderLinuxCritsect);
-
-    VideoRenderCallback* renderCallback = NULL;
-    if (_ptrX11Render)
-    {
-        VideoX11Channel* renderChannel =
-                _ptrX11Render->CreateX11RenderChannel(streamId, zOrder, left,
-                                                      top, right, bottom);
-        if (!renderChannel)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                         "Render channel creation failed for stream id: %d",
-                         streamId);
-            return NULL;
-        }
-        renderCallback = (VideoRenderCallback *) renderChannel;
-    }
-    else
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "_ptrX11Render is NULL");
-        return NULL;
-    }
-    return renderCallback;
-}
-
-int32_t VideoRenderLinuxImpl::DeleteIncomingRenderStream(
-                                                               const uint32_t streamId)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
-                 __FUNCTION__);
-    CriticalSectionScoped cs(&_renderLinuxCritsect);
-
-    if (_ptrX11Render)
-    {
-        return _ptrX11Render->DeleteX11RenderChannel(streamId);
-    }
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::GetIncomingRenderStreamProperties(
-                                                                      const uint32_t streamId,
-                                                                      uint32_t& zOrder,
-                                                                      float& left,
-                                                                      float& top,
-                                                                      float& right,
-                                                                      float& bottom) const
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
-                 __FUNCTION__);
-    CriticalSectionScoped cs(&_renderLinuxCritsect);
-
-    if (_ptrX11Render)
-    {
-        return _ptrX11Render->GetIncomingStreamProperties(streamId, zOrder,
-                                                          left, top, right,
-                                                          bottom);
-    }
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::StartRender()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
-                 __FUNCTION__);
-    return 0;
-}
-
-int32_t VideoRenderLinuxImpl::StopRender()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
-                 __FUNCTION__);
-    return 0;
-}
-
-VideoRenderType VideoRenderLinuxImpl::RenderType()
-{
-    return kRenderX11;
-}
-
-RawVideoType VideoRenderLinuxImpl::PerferedVideoType()
-{
-    return kVideoI420;
-}
-
-bool VideoRenderLinuxImpl::FullScreen()
-{
-    return false;
-}
-
-int32_t VideoRenderLinuxImpl::GetGraphicsMemory(
-                                                      uint64_t& /*totalGraphicsMemory*/,
-                                                      uint64_t& /*availableGraphicsMemory*/) const
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s - not supported on Linux", __FUNCTION__);
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::GetScreenResolution(
-                                                        uint32_t& /*screenWidth*/,
-                                                        uint32_t& /*screenHeight*/) const
-{
-    return -1;
-}
-
-uint32_t VideoRenderLinuxImpl::RenderFrameRate(const uint32_t /*streamId*/)
-{
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::SetStreamCropping(
-                                                      const uint32_t /*streamId*/,
-                                                      const float /*left*/,
-                                                      const float /*top*/,
-                                                      const float /*right*/,
-                                                      const float /*bottom*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s - not supported on Linux", __FUNCTION__);
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::SetTransparentBackground(const bool /*enable*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s - not supported on Linux", __FUNCTION__);
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::ConfigureRenderer(
-                                                      const uint32_t streamId,
-                                                      const unsigned int zOrder,
-                                                      const float left,
-                                                      const float top,
-                                                      const float right,
-                                                      const float bottom)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s - not supported on Linux", __FUNCTION__);
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::SetText(
-                                            const uint8_t textId,
-                                            const uint8_t* text,
-                                            const int32_t textLength,
-                                            const uint32_t textColorRef,
-                                            const uint32_t backgroundColorRef,
-                                            const float left, const float top,
-                                            const float rigth,
-                                            const float bottom)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s - not supported on Linux", __FUNCTION__);
-    return -1;
-}
-
-int32_t VideoRenderLinuxImpl::SetBitmap(const void* bitMap,
-                                        const uint8_t pictureId,
-                                        const void* colorKey,
-                                        const float left,
-                                        const float top,
-                                        const float right,
-                                        const float bottom)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s - not supported on Linux", __FUNCTION__);
-    return -1;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/linux/video_render_linux_impl.h b/chromium/third_party/webrtc/modules/video_render/linux/video_render_linux_impl.h
deleted file mode 100644
index 0e9ae54c18f..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/linux/video_render_linux_impl.h
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
-
-#include "webrtc/modules/video_render/i_video_render.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class VideoX11Render;
-
-// Class definitions
-class VideoRenderLinuxImpl: IVideoRender
-{
-public:
-    /*
-     *   Constructor/destructor
-     */
-
-    VideoRenderLinuxImpl(const int32_t id,
-                         const VideoRenderType videoRenderType,
-                         void* window, const bool fullscreen);
-
-    virtual ~VideoRenderLinuxImpl();
-
-    virtual int32_t Init();
-
-    virtual int32_t ChangeWindow(void* window);
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderCallback
-            * AddIncomingRenderStream(const uint32_t streamId,
-                                      const uint32_t zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t
-            DeleteIncomingRenderStream(const uint32_t streamId);
-
-    virtual int32_t
-            GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                              uint32_t& zOrder,
-                                              float& left, float& top,
-                                              float& right, float& bottom) const;
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender();
-
-    virtual int32_t StopRender();
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderType RenderType();
-
-    virtual RawVideoType PerferedVideoType();
-
-    virtual bool FullScreen();
-
-    virtual int32_t
-            GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-                              uint64_t& availableGraphicsMemory) const;
-
-    virtual int32_t
-            GetScreenResolution(uint32_t& screenWidth,
-                                uint32_t& screenHeight) const;
-
-    virtual uint32_t RenderFrameRate(const uint32_t streamId);
-
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t SetTransparentBackground(const bool enable);
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t textColorRef,
-                            const uint32_t backgroundColorRef,
-                            const float left, const float top,
-                            const float rigth, const float bottom);
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left, const float top,
-                              const float right, const float bottom);
-
-private:
-    int32_t _id;
-    CriticalSectionWrapper& _renderLinuxCritsect;
-
-    void* _ptrWindow;
-
-    // X11 Render
-    VideoX11Render* _ptrX11Render;
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_channel.cc b/chromium/third_party/webrtc/modules/video_render/linux/video_x11_channel.cc
deleted file mode 100644
index 8d86b7c72ad..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_channel.cc
+++ /dev/null
@@ -1,315 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/linux/video_x11_channel.h"
-
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-#define DISP_MAX 128
-
-static Display *dispArray[DISP_MAX];
-static int dispCount = 0;
-
-
-VideoX11Channel::VideoX11Channel(int32_t id) :
-    _crit(*CriticalSectionWrapper::CreateCriticalSection()), _display(NULL),
-          _shminfo(), _image(NULL), _window(0L), _gc(NULL),
-          _width(DEFAULT_RENDER_FRAME_WIDTH),
-          _height(DEFAULT_RENDER_FRAME_HEIGHT), _outWidth(0), _outHeight(0),
-          _xPos(0), _yPos(0), _prepared(false), _dispCount(0), _buffer(NULL),
-          _top(0.0), _left(0.0), _right(0.0), _bottom(0.0),
-          _Id(id)
-{
-}
-
-VideoX11Channel::~VideoX11Channel()
-{
-    if (_prepared)
-    {
-        _crit.Enter();
-        ReleaseWindow();
-        _crit.Leave();
-    }
-    delete &_crit;
-}
-
-int32_t VideoX11Channel::RenderFrame(const uint32_t streamId,
-                                     const VideoFrame& videoFrame) {
-  CriticalSectionScoped cs(&_crit);
-  if (_width != videoFrame.width() || _height
-      != videoFrame.height()) {
-      if (FrameSizeChange(videoFrame.width(), videoFrame.height(), 1) == -1) {
-        return -1;
-    }
-  }
-  return DeliverFrame(videoFrame);
-}
-
-int32_t VideoX11Channel::FrameSizeChange(int32_t width,
-                                         int32_t height,
-                                         int32_t /*numberOfStreams */)
-{
-    CriticalSectionScoped cs(&_crit);
-    if (_prepared)
-    {
-        RemoveRenderer();
-    }
-    if (CreateLocalRenderer(width, height) == -1)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t VideoX11Channel::DeliverFrame(const VideoFrame& videoFrame) {
-  CriticalSectionScoped cs(&_crit);
-  if (!_prepared) {
-    return 0;
-  }
-
-  if (!dispArray[_dispCount]) {
-    return -1;
-  }
-
-  ConvertFromI420(videoFrame, kARGB, 0, _buffer);
-
-  // Put image in window.
-  XShmPutImage(_display, _window, _gc, _image, 0, 0, _xPos, _yPos, _width,
-               _height, True);
-
-  // Very important for the image to update properly!
-  XSync(_display, False);
-  return 0;
-}
-
-int32_t VideoX11Channel::GetFrameSize(int32_t& width, int32_t& height)
-{
-    width = _width;
-    height = _height;
-
-    return 0;
-}
-
-int32_t VideoX11Channel::Init(Window window, float left, float top,
-                              float right, float bottom)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
-                 __FUNCTION__);
-    CriticalSectionScoped cs(&_crit);
-
-    _window = window;
-    _left = left;
-    _right = right;
-    _top = top;
-    _bottom = bottom;
-
-    _display = XOpenDisplay(NULL); // Use default display
-    if (!_window || !_display)
-    {
-        return -1;
-    }
-
-    if (dispCount < DISP_MAX)
-    {
-        dispArray[dispCount] = _display;
-        _dispCount = dispCount;
-        dispCount++;
-    }
-    else
-    {
-        return -1;
-    }
-
-    if ((1 < left || left < 0) || (1 < top || top < 0) || (1 < right || right
-            < 0) || (1 < bottom || bottom < 0))
-    {
-        return -1;
-    }
-
-    // calculate position and size of rendered video
-    int x, y;
-    unsigned int winWidth, winHeight, borderwidth, depth;
-    Window rootret;
-    if (XGetGeometry(_display, _window, &rootret, &x, &y, &winWidth,
-                     &winHeight, &borderwidth, &depth) == 0)
-    {
-        return -1;
-    }
-
-    _xPos = (int32_t) (winWidth * left);
-    _yPos = (int32_t) (winHeight * top);
-    _outWidth = (int32_t) (winWidth * (right - left));
-    _outHeight = (int32_t) (winHeight * (bottom - top));
-    if (_outWidth % 2)
-        _outWidth++; // the renderer want's sizes that are multiples of two
-    if (_outHeight % 2)
-        _outHeight++;
-
-    _gc = XCreateGC(_display, _window, 0, 0);
-    if (!_gc) {
-      // Failed to create the graphics context.
-      assert(false);
-      return -1;
-    }
-
-    if (CreateLocalRenderer(winWidth, winHeight) == -1)
-    {
-        return -1;
-    }
-    return 0;
-
-}
-
-int32_t VideoX11Channel::ChangeWindow(Window window)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
-                 __FUNCTION__);
-    CriticalSectionScoped cs(&_crit);
-
-    // Stop the rendering, if we are rendering...
-    RemoveRenderer();
-    _window = window;
-
-    // calculate position and size of rendered video
-    int x, y;
-    unsigned int winWidth, winHeight, borderwidth, depth;
-    Window rootret;
-    if (XGetGeometry(_display, _window, &rootret, &x, &y, &winWidth,
-                     &winHeight, &borderwidth, &depth) == -1)
-    {
-        return -1;
-    }
-    _xPos = (int) (winWidth * _left);
-    _yPos = (int) (winHeight * _top);
-    _outWidth = (int) (winWidth * (_right - _left));
-    _outHeight = (int) (winHeight * (_bottom - _top));
-    if (_outWidth % 2)
-        _outWidth++; // the renderer want's sizes that are multiples of two
-    if (_outHeight % 2)
-        _outHeight++;
-
-    // Prepare rendering using the
-    if (CreateLocalRenderer(_width, _height) == -1)
-    {
-        return -1;
-    }
-    return 0;
-}
-
-int32_t VideoX11Channel::ReleaseWindow()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
-                 __FUNCTION__);
-    CriticalSectionScoped cs(&_crit);
-
-    RemoveRenderer();
-    if (_gc) {
-      XFreeGC(_display, _gc);
-      _gc = NULL;
-    }
-    if (_display)
-    {
-        XCloseDisplay(_display);
-        _display = NULL;
-    }
-    return 0;
-}
-
-int32_t VideoX11Channel::CreateLocalRenderer(int32_t width, int32_t height)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
-                 __FUNCTION__);
-    CriticalSectionScoped cs(&_crit);
-
-    if (!_window || !_display)
-    {
-        return -1;
-    }
-
-    if (_prepared)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _Id,
-                     "Renderer already prepared, exits.");
-        return -1;
-    }
-
-    _width = width;
-    _height = height;
-
-    // create shared memory image
-    _image = XShmCreateImage(_display, CopyFromParent, 24, ZPixmap, NULL,
-                             &_shminfo, _width, _height); // this parameter needs to be the same for some reason.
-    _shminfo.shmid = shmget(IPC_PRIVATE, (_image->bytes_per_line
-            * _image->height), IPC_CREAT | 0777);
-    _shminfo.shmaddr = _image->data = (char*) shmat(_shminfo.shmid, 0, 0);
-    if (_image->data == reinterpret_cast<char*>(-1))
-    {
-        return -1;
-    }
-    _buffer = (unsigned char*) _image->data;
-    _shminfo.readOnly = False;
-
-    // attach image to display
-    if (!XShmAttach(_display, &_shminfo))
-    {
-        //printf("XShmAttach failed !\n");
-        return -1;
-    }
-    XSync(_display, False);
-
-    _prepared = true;
-    return 0;
-}
-
-int32_t VideoX11Channel::RemoveRenderer()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
-                 __FUNCTION__);
-
-    if (!_prepared)
-    {
-        return 0;
-    }
-    _prepared = false;
-
-    // Free the memory.
-    XShmDetach(_display, &_shminfo);
-    XDestroyImage( _image );
-    _image = NULL;
-    shmdt(_shminfo.shmaddr);
-    _shminfo.shmaddr = NULL;
-    _buffer = NULL;
-    shmctl(_shminfo.shmid, IPC_RMID, 0);
-    _shminfo.shmid = 0;
-    return 0;
-}
-
-int32_t VideoX11Channel::GetStreamProperties(uint32_t& zOrder,
-                                             float& left, float& top,
-                                             float& right, float& bottom) const
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
-                 __FUNCTION__);
-
-    zOrder = 0; // no z-order support yet
-    left = _left;
-    top = _top;
-    right = _right;
-    bottom = _bottom;
-
-    return 0;
-}
-
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_channel.h b/chromium/third_party/webrtc/modules/video_render/linux/video_x11_channel.h
deleted file mode 100644
index 6eb402e12ee..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_channel.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
-
-#include <sys/shm.h>
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-#include <X11/Xlib.h>
-#include <X11/Xutil.h>
-#include <X11/extensions/XShm.h>
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-#define DEFAULT_RENDER_FRAME_WIDTH 352
-#define DEFAULT_RENDER_FRAME_HEIGHT 288
-
-
-class VideoX11Channel: public VideoRenderCallback
-{
-public:
-    VideoX11Channel(int32_t id);
-
-    virtual ~VideoX11Channel();
-
-    virtual int32_t RenderFrame(const uint32_t streamId,
-                                const VideoFrame& videoFrame);
-
-    int32_t FrameSizeChange(int32_t width, int32_t height,
-                            int32_t numberOfStreams);
-    int32_t DeliverFrame(const VideoFrame& videoFrame);
-    int32_t GetFrameSize(int32_t& width, int32_t& height);
-    int32_t Init(Window window, float left, float top, float right,
-                 float bottom);
-    int32_t ChangeWindow(Window window);
-    int32_t
-            GetStreamProperties(uint32_t& zOrder, float& left,
-                                float& top, float& right, float& bottom) const;
-    int32_t ReleaseWindow();
-
-    bool IsPrepared()
-    {
-        return _prepared;
-    }
-
-private:
-
-    int32_t
-            CreateLocalRenderer(int32_t width, int32_t height);
-    int32_t RemoveRenderer();
-
-    //FIXME a better place for this method? the GetWidthHeight no longer
-    // supported by common_video.
-    int GetWidthHeight(VideoType type, int bufferSize, int& width,
-                       int& height);
-
-    CriticalSectionWrapper& _crit;
-
-    Display* _display;
-    XShmSegmentInfo _shminfo;
-    XImage* _image;
-    Window _window;
-    GC _gc;
-    int32_t _width; // incoming frame width
-    int32_t _height; // incoming frame height
-    int32_t _outWidth; // render frame width
-    int32_t _outHeight; // render frame height
-    int32_t _xPos; // position within window
-    int32_t _yPos;
-    bool _prepared; // true if ready to use
-    int32_t _dispCount;
-
-    unsigned char* _buffer;
-    float _top;
-    float _left;
-    float _right;
-    float _bottom;
-
-    int32_t _Id;
-
-};
-
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_render.cc b/chromium/third_party/webrtc/modules/video_render/linux/video_x11_render.cc
deleted file mode 100644
index 5eb4f36f95f..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_render.cc
+++ /dev/null
@@ -1,153 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/linux/video_x11_channel.h"
-#include "webrtc/modules/video_render/linux/video_x11_render.h"
-
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-VideoX11Render::VideoX11Render(Window window) :
-    _window(window),
-            _critSect(*CriticalSectionWrapper::CreateCriticalSection())
-{
-}
-
-VideoX11Render::~VideoX11Render()
-{
-    delete &_critSect;
-}
-
-int32_t VideoX11Render::Init()
-{
-    CriticalSectionScoped cs(&_critSect);
-
-    _streamIdToX11ChannelMap.clear();
-
-    return 0;
-}
-
-int32_t VideoX11Render::ChangeWindow(Window window)
-{
-    CriticalSectionScoped cs(&_critSect);
-    VideoX11Channel* renderChannel = NULL;
-
-    std::map<int, VideoX11Channel*>::iterator iter =
-            _streamIdToX11ChannelMap.begin();
-
-    while (iter != _streamIdToX11ChannelMap.end())
-    {
-        renderChannel = iter->second;
-        if (renderChannel)
-        {
-            renderChannel->ChangeWindow(window);
-        }
-        iter++;
-    }
-
-    _window = window;
-
-    return 0;
-}
-
-VideoX11Channel* VideoX11Render::CreateX11RenderChannel(
-                                                                int32_t streamId,
-                                                                int32_t zOrder,
-                                                                const float left,
-                                                                const float top,
-                                                                const float right,
-                                                                const float bottom)
-{
-    CriticalSectionScoped cs(&_critSect);
-    VideoX11Channel* renderChannel = NULL;
-
-    std::map<int, VideoX11Channel*>::iterator iter =
-            _streamIdToX11ChannelMap.find(streamId);
-
-    if (iter == _streamIdToX11ChannelMap.end())
-    {
-        renderChannel = new VideoX11Channel(streamId);
-        if (!renderChannel)
-        {
-            WEBRTC_TRACE(
-                         kTraceError,
-                         kTraceVideoRenderer,
-                         -1,
-                         "Failed to create VideoX11Channel for streamId : %d",
-                         streamId);
-            return NULL;
-        }
-        renderChannel->Init(_window, left, top, right, bottom);
-        _streamIdToX11ChannelMap[streamId] = renderChannel;
-    }
-    else
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1,
-                     "Render Channel already exists for streamId: %d", streamId);
-        renderChannel = iter->second;
-    }
-
-    return renderChannel;
-}
-
-int32_t VideoX11Render::DeleteX11RenderChannel(int32_t streamId)
-{
-    CriticalSectionScoped cs(&_critSect);
-
-    std::map<int, VideoX11Channel*>::iterator iter =
-            _streamIdToX11ChannelMap.find(streamId);
-    if (iter != _streamIdToX11ChannelMap.end())
-    {
-        VideoX11Channel *renderChannel = iter->second;
-        if (renderChannel)
-        {
-            renderChannel->ReleaseWindow();
-            delete renderChannel;
-            renderChannel = NULL;
-        }
-        _streamIdToX11ChannelMap.erase(iter);
-    }
-
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
-                 "No VideoX11Channel object exists for stream id: %d",
-                 streamId);
-    return -1;
-}
-
-int32_t VideoX11Render::GetIncomingStreamProperties(
-                                                              int32_t streamId,
-                                                              uint32_t& zOrder,
-                                                              float& left,
-                                                              float& top,
-                                                              float& right,
-                                                              float& bottom)
-{
-    CriticalSectionScoped cs(&_critSect);
-
-    std::map<int, VideoX11Channel*>::iterator iter =
-            _streamIdToX11ChannelMap.find(streamId);
-    if (iter != _streamIdToX11ChannelMap.end())
-    {
-        VideoX11Channel *renderChannel = iter->second;
-        if (renderChannel)
-        {
-            renderChannel->GetStreamProperties(zOrder, left, top, right, bottom);
-        }
-    }
-
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
-                 "No VideoX11Channel object exists for stream id: %d",
-                 streamId);
-    return -1;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_render.h b/chromium/third_party/webrtc/modules/video_render/linux/video_x11_render.h
deleted file mode 100644
index 23b83bd67bb..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/linux/video_x11_render.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
-
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-#include <X11/Xlib.h>
-#include <map>
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class VideoX11Channel;
-
-class VideoX11Render
-{
-
-public:
-    VideoX11Render(Window window);
-    ~VideoX11Render();
-
-    int32_t Init();
-    int32_t ChangeWindow(Window window);
-
-    VideoX11Channel* CreateX11RenderChannel(int32_t streamId,
-                                            int32_t zOrder,
-                                            const float left,
-                                            const float top,
-                                            const float right,
-                                            const float bottom);
-
-    int32_t DeleteX11RenderChannel(int32_t streamId);
-
-    int32_t GetIncomingStreamProperties(int32_t streamId,
-                                        uint32_t& zOrder,
-                                        float& left, float& top,
-                                        float& right, float& bottom);
-
-private:
-    Window _window;
-    CriticalSectionWrapper& _critSect;
-    std::map<int, VideoX11Channel*> _streamIdToX11ChannelMap;
-
-};
-
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_full_screen_window.h b/chromium/third_party/webrtc/modules/video_render/mac/cocoa_full_screen_window.h
deleted file mode 100644
index c8e98bba674..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_full_screen_window.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-//  cocoa_full_screen_window.h
-//
-//
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
-
-#import <Cocoa/Cocoa.h>
-//#define GRAB_ALL_SCREENS 1
-
-@interface CocoaFullScreenWindow : NSObject {
-	NSWindow*			_window;
-}
-
--(id)init;
--(void)grabFullScreen;
--(void)releaseFullScreen;
--(NSWindow*)window;
-
-@end
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_full_screen_window.mm b/chromium/third_party/webrtc/modules/video_render/mac/cocoa_full_screen_window.mm
deleted file mode 100644
index b57223b4dfd..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_full_screen_window.mm
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/mac/cocoa_full_screen_window.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-using namespace webrtc;
-
-@implementation CocoaFullScreenWindow
-
--(id)init{	
-	
-	self = [super init];
-	if(!self){
-		WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d COULD NOT CREATE INSTANCE", __FUNCTION__, __LINE__); 
-		return nil;
-	}
-	
-	
-	WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, 0, "%s:%d Created instance", __FUNCTION__, __LINE__); 
-	return self;
-}
-
--(void)grabFullScreen{
-	
-#ifdef GRAB_ALL_SCREENS
-	if(CGCaptureAllDisplays() != kCGErrorSuccess)
-#else
-	if(CGDisplayCapture(kCGDirectMainDisplay) != kCGErrorSuccess)
-#endif
-	{
-		WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not capture main level", __FUNCTION__, __LINE__); 
-	}
-	
-	// get the shielding window level
-	int windowLevel = CGShieldingWindowLevel();
-	
-	// get the screen rect of main display
-	NSRect screenRect = [[NSScreen mainScreen]frame];
-	
-	_window = [[NSWindow alloc]initWithContentRect:screenRect 
-										   styleMask:NSBorderlessWindowMask
-											 backing:NSBackingStoreBuffered
-											   defer:NO
-											  screen:[NSScreen mainScreen]];
-	
-	[_window setLevel:windowLevel];
-	[_window setBackgroundColor:[NSColor blackColor]];
-	[_window makeKeyAndOrderFront:nil];
-
-}
- 
--(void)releaseFullScreen
-{
-	[_window orderOut:self];
-	
-#ifdef GRAB_ALL_SCREENS
-	if(CGReleaseAllDisplays() != kCGErrorSuccess)
-#else
-	if(CGDisplayRelease(kCGDirectMainDisplay) != kCGErrorSuccess)
-#endif
-	{
-		WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not release the displays", __FUNCTION__, __LINE__); 
-	}		
-}
-
-- (NSWindow*)window
-{
-  return _window;
-}
-
-- (void) dealloc
-{
-	[self releaseFullScreen];
-	[super dealloc];
-}	
-
-
-	
-@end
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_render_view.h b/chromium/third_party/webrtc/modules/video_render/mac/cocoa_render_view.h
deleted file mode 100644
index 15a8108dec7..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_render_view.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-//  cocoa_render_view.h
-//
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
-
-#import <Cocoa/Cocoa.h>
-#import <OpenGL/gl.h>
-#import <OpenGL/glu.h>
-#import <OpenGL/OpenGL.h>
-
-@interface CocoaRenderView : NSOpenGLView {
-  NSOpenGLContext* _nsOpenGLContext;
-}
-
--(void)initCocoaRenderView:(NSOpenGLPixelFormat*)fmt;
--(void)initCocoaRenderViewFullScreen:(NSOpenGLPixelFormat*)fmt;
--(NSOpenGLContext*)nsOpenGLContext;
-@end
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_render_view.mm b/chromium/third_party/webrtc/modules/video_render/mac/cocoa_render_view.mm
deleted file mode 100644
index 4631ff31a4f..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/cocoa_render_view.mm
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#import <Cocoa/Cocoa.h>
-#import <AppKit/AppKit.h>
-
-#include "webrtc/modules/video_render/mac/cocoa_render_view.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-using namespace webrtc;
-
-@implementation CocoaRenderView
-
--(void)initCocoaRenderView:(NSOpenGLPixelFormat*)fmt{
-	
-    self = [super initWithFrame:[self frame] pixelFormat:fmt];
-	if (self == nil){
-		
-		WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not create instance", __FUNCTION__, __LINE__); 
-	}
-	
-	
-	_nsOpenGLContext = [self openGLContext];
-
-}
-
--(NSOpenGLContext*)nsOpenGLContext {
-    return _nsOpenGLContext;
-}
-
--(void)initCocoaRenderViewFullScreen:(NSOpenGLPixelFormat*)fmt{
-	
-	NSRect screenRect = [[NSScreen mainScreen]frame];
-//	[_windowRef setFrame:screenRect];
-//	[_windowRef setBounds:screenRect];
-    self = [super initWithFrame:screenRect pixelFormat:fmt];
-	if (self == nil){
-		
-		WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not create instance", __FUNCTION__, __LINE__); 
-	}
-	
-	_nsOpenGLContext = [self openGLContext];
-
-}
-
-@end
-
-
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_agl.cc b/chromium/third_party/webrtc/modules/video_render/mac/video_render_agl.cc
deleted file mode 100644
index 3243563b2bf..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_agl.cc
+++ /dev/null
@@ -1,1987 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-
-#if defined(CARBON_RENDERING)
-
-#include "webrtc/modules/video_render/mac/video_render_agl.h"
-
-//  includes
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-/*
- *
- *    VideoChannelAGL
- *
- */
-
-#pragma mark VideoChannelAGL constructor
-
-VideoChannelAGL::VideoChannelAGL(AGLContext& aglContext, int iId, VideoRenderAGL* owner) :
-    _aglContext( aglContext),
-    _id( iId),
-    _owner( owner),
-    _width( 0),
-    _height( 0),
-    _stretchedWidth( 0),
-    _stretchedHeight( 0),
-    _startWidth( 0.0f),
-    _startHeight( 0.0f),
-    _stopWidth( 0.0f),
-    _stopHeight( 0.0f),
-    _xOldWidth( 0),
-    _yOldHeight( 0),
-    _oldStretchedHeight(0),
-    _oldStretchedWidth( 0),
-    _buffer( 0),
-    _bufferSize( 0),
-    _incomingBufferSize(0),
-    _bufferIsUpdated( false),
-    _sizeInitialized( false),
-    _numberOfStreams( 0),
-    _bVideoSizeStartedChanging(false),
-    _pixelFormat( GL_RGBA),
-    _pixelDataType( GL_UNSIGNED_INT_8_8_8_8),
-    _texture( 0)
-
-{
-    //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Constructor", __FUNCTION__, __LINE__);
-}
-
-VideoChannelAGL::~VideoChannelAGL()
-{
-    //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Destructor", __FUNCTION__, __LINE__);
-    if (_buffer)
-    {
-        delete [] _buffer;
-        _buffer = NULL;
-    }
-
-    aglSetCurrentContext(_aglContext);
-
-    if (_texture != 0)
-    {
-        glDeleteTextures(1, (const GLuint*) &_texture);
-        _texture = 0;
-    }
-}
-
-int32_t VideoChannelAGL::RenderFrame(const uint32_t streamId,
-                                     VideoFrame& videoFrame) {
-  _owner->LockAGLCntx();
-  if (_width != videoFrame.width() ||
-      _height != videoFrame.height()) {
-    if (FrameSizeChange(videoFrame.width(), videoFrame.height(), 1) == -1) {
-      WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d FrameSize
-                   Change returned an error", __FUNCTION__, __LINE__);
-      _owner->UnlockAGLCntx();
-      return -1;
-    }
-  }
-
-  _owner->UnlockAGLCntx();
-  return DeliverFrame(videoFrame);
-}
-
-int VideoChannelAGL::UpdateSize(int /*width*/, int /*height*/)
-{
-    _owner->LockAGLCntx();
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelAGL::UpdateStretchSize(int stretchHeight, int stretchWidth)
-{
-
-    _owner->LockAGLCntx();
-    _stretchedHeight = stretchHeight;
-    _stretchedWidth = stretchWidth;
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelAGL::FrameSizeChange(int width, int height, int numberOfStreams)
-{
-    //  We'll get a new frame size from VideoAPI, prepare the buffer
-
-    _owner->LockAGLCntx();
-
-    if (width == _width && _height == height)
-    {
-        // We already have a correct buffer size
-        _numberOfStreams = numberOfStreams;
-        _owner->UnlockAGLCntx();
-        return 0;
-    }
-
-    _width = width;
-    _height = height;
-
-    // Delete the old buffer, create a new one with correct size.
-    if (_buffer)
-    {
-        delete [] _buffer;
-        _bufferSize = 0;
-    }
-
-    _incomingBufferSize = CalcBufferSize(kI420, _width, _height);
-    _bufferSize = CalcBufferSize(kARGB, _width, _height);//_width * _height * bytesPerPixel;
-    _buffer = new unsigned char [_bufferSize];
-    memset(_buffer, 0, _bufferSize * sizeof(unsigned char));
-
-    if (aglSetCurrentContext(_aglContext) == false)
-    {
-        _owner->UnlockAGLCntx();
-        return -1;
-    }
-
-    // Delete a possible old texture
-    if (_texture != 0)
-    {
-        glDeleteTextures(1, (const GLuint*) &_texture);
-        _texture = 0;
-    }
-
-    // Create a new texture
-    glGenTextures(1, (GLuint *) &_texture);
-
-    GLenum glErr = glGetError();
-
-    if (glErr != GL_NO_ERROR)
-    {
-    }
-
-    // Do the setup for both textures
-    // Note: we setup two textures even if we're not running full screen
-    glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-
-    // Set texture parameters
-    glTexParameterf(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_PRIORITY, 1.0);
-
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
-    //glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
-    //glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
-
-    glTexEnvf(GL_TEXTURE_ENV, GL_TEXTURE_ENV_MODE, GL_MODULATE);
-
-    glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
-
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_STORAGE_HINT_APPLE, GL_STORAGE_SHARED_APPLE);
-
-    // Maximum width/height for a texture
-    GLint texSize;
-    glGetIntegerv(GL_MAX_TEXTURE_SIZE, &texSize);
-
-    if (texSize < _width || texSize < _height)
-    {
-        // Image too big for memory
-        _owner->UnlockAGLCntx();
-        return -1;
-    }
-
-    // Set up th texture type and size
-    glTexImage2D(GL_TEXTURE_RECTANGLE_EXT, // target
-            0, // level
-            GL_RGBA, // internal format
-            _width, // width
-            _height, // height
-            0, // border 0/1 = off/on
-            _pixelFormat, // format, GL_BGRA
-            _pixelDataType, // data type, GL_UNSIGNED_INT_8_8_8_8
-            _buffer); // pixel data
-
-    glErr = glGetError();
-    if (glErr != GL_NO_ERROR)
-    {
-        _owner->UnlockAGLCntx();
-        return -1;
-    }
-
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-// Called from video engine when a new frame should be rendered.
-int VideoChannelAGL::DeliverFrame(const VideoFrame& videoFrame) {
-  _owner->LockAGLCntx();
-
-  if (_texture == 0) {
-    _owner->UnlockAGLCntx();
-    return 0;
-  }
-
-  if (CalcBufferSize(kI420, videoFrame.width(), videoFrame.height()) !=
-      _incomingBufferSize) {
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  // Setting stride = width.
-  int rgbret = ConvertFromYV12(videoFrame, kBGRA, 0, _buffer);
-  if (rgbret < 0) {
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  aglSetCurrentContext(_aglContext);
-
-  // Put the new frame into the graphic card texture.
-  // Make sure this texture is the active one
-  glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-  GLenum glErr = glGetError();
-  if (glErr != GL_NO_ERROR) {
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  // Copy buffer to texture
-  glTexSubImage2D(GL_TEXTURE_RECTANGLE_EXT,
-                  0, // Level, not use
-                  0, // start point x, (low left of pic)
-                  0, // start point y,
-                  _width, // width
-                  _height, // height
-                  _pixelFormat, // pictue format for _buffer
-                  _pixelDataType, // data type of _buffer
-                  (const GLvoid*) _buffer); // the pixel data
-
-  if (glGetError() != GL_NO_ERROR) {
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  _bufferIsUpdated = true;
-  _owner->UnlockAGLCntx();
-
-  return 0;
-}
-
-int VideoChannelAGL::RenderOffScreenBuffer()
-{
-
-    _owner->LockAGLCntx();
-
-    if (_texture == 0)
-    {
-        _owner->UnlockAGLCntx();
-        return 0;
-    }
-
-    GLfloat xStart = 2.0f * _startWidth - 1.0f;
-    GLfloat xStop = 2.0f * _stopWidth - 1.0f;
-    GLfloat yStart = 1.0f - 2.0f * _stopHeight;
-    GLfloat yStop = 1.0f - 2.0f * _startHeight;
-
-    aglSetCurrentContext(_aglContext);
-    glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-
-    if(_stretchedWidth != _oldStretchedWidth || _stretchedHeight != _oldStretchedHeight)
-    {
-        glViewport(0, 0, _stretchedWidth, _stretchedHeight);
-    }
-    _oldStretchedHeight = _stretchedHeight;
-    _oldStretchedWidth = _stretchedWidth;
-
-    // Now really put the texture into the framebuffer
-    glLoadIdentity();
-
-    glEnable(GL_TEXTURE_RECTANGLE_EXT);
-
-    glBegin(GL_POLYGON);
-    {
-        glTexCoord2f(0.0, 0.0); glVertex2f(xStart, yStop);
-        glTexCoord2f(_width, 0.0); glVertex2f(xStop, yStop);
-        glTexCoord2f(_width, _height); glVertex2f(xStop, yStart);
-        glTexCoord2f(0.0, _height); glVertex2f(xStart, yStart);
-    }
-    glEnd();
-
-    glDisable(GL_TEXTURE_RECTANGLE_EXT);
-
-    _bufferIsUpdated = false;
-
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelAGL::IsUpdated(bool& isUpdated)
-{
-    _owner->LockAGLCntx();
-    isUpdated = _bufferIsUpdated;
-    _owner->UnlockAGLCntx();
-
-    return 0;
-}
-
-int VideoChannelAGL::SetStreamSettings(int /*streamId*/, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
-    _owner->LockAGLCntx();
-
-    _startWidth = startWidth;
-    _stopWidth = stopWidth;
-    _startHeight = startHeight;
-    _stopHeight = stopHeight;
-
-    int oldWidth = _width;
-    int oldHeight = _height;
-    int oldNumberOfStreams = _numberOfStreams;
-
-    _width = 0;
-    _height = 0;
-
-    int retVal = FrameSizeChange(oldWidth, oldHeight, oldNumberOfStreams);
-
-    _owner->UnlockAGLCntx();
-
-    return retVal;
-}
-
-int VideoChannelAGL::SetStreamCropSettings(int /*streamId*/, float /*startWidth*/, float /*startHeight*/, float /*stopWidth*/, float /*stopHeight*/)
-{
-    return -1;
-}
-
-#pragma mark VideoRenderAGL WindowRef constructor
-
-VideoRenderAGL::VideoRenderAGL(WindowRef windowRef, bool fullscreen, int iId) :
-_hiviewRef( 0),
-_windowRef( windowRef),
-_fullScreen( fullscreen),
-_id( iId),
-_renderCritSec(*CriticalSectionWrapper::CreateCriticalSection()),
-_screenUpdateEvent( 0),
-_isHIViewRef( false),
-_aglContext( 0),
-_windowWidth( 0),
-_windowHeight( 0),
-_lastWindowWidth( -1),
-_lastWindowHeight( -1),
-_lastHiViewWidth( -1),
-_lastHiViewHeight( -1),
-_currentParentWindowHeight( 0),
-_currentParentWindowWidth( 0),
-_currentParentWindowBounds( ),
-_windowHasResized( false),
-_lastParentWindowBounds( ),
-_currentHIViewBounds( ),
-_lastHIViewBounds( ),
-_windowRect( ),
-_aglChannels( ),
-_zOrderToChannel( ),
-_hiviewEventHandlerRef( NULL),
-_windowEventHandlerRef( NULL),
-_currentViewBounds( ),
-_lastViewBounds( ),
-_renderingIsPaused( false),
-
-{
-    //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s");
-
-    _screenUpdateThread.reset(
-        new rtc::PlatformThread(ScreenUpdateThreadProc, this, "ScreenUpdate"));
-    _screenUpdateEvent = EventWrapper::Create();
-
-    if(!IsValidWindowPtr(_windowRef))
-    {
-        //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Invalid WindowRef:0x%x", __FUNCTION__, __LINE__, _windowRef);
-    }
-    else
-    {
-        //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d WindowRef 0x%x is valid", __FUNCTION__, __LINE__, _windowRef);
-    }
-
-    GetWindowRect(_windowRect);
-
-    _lastViewBounds.origin.x = 0;
-    _lastViewBounds.origin.y = 0;
-    _lastViewBounds.size.width = 0;
-    _lastViewBounds.size.height = 0;
-
-}
-
-// this is a static function. It has been registered (in class constructor) to be called on various window redrawing or resizing.
-// Since it is a static method, I have passed in "this" as the userData (one and only allowed) parameter, then calling member methods on it.
-#pragma mark WindowRef Event Handler
-pascal OSStatus VideoRenderAGL::sHandleWindowResized (EventHandlerCallRef /*nextHandler*/,
-        EventRef theEvent,
-        void* userData)
-{
-    WindowRef windowRef = NULL;
-
-    int eventType = GetEventKind(theEvent);
-
-    // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/i686-win32/include/quicktime/CarbonEvents.h for a list of codes
-    GetEventParameter (theEvent,
-            kEventParamDirectObject,
-            typeWindowRef,
-            NULL,
-            sizeof (WindowRef),
-            NULL,
-            &windowRef);
-
-    VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
-
-    bool updateUI = true;
-    if(kEventWindowBoundsChanged == eventType)
-    {
-    }
-    else if(kEventWindowBoundsChanging == eventType)
-    {
-    }
-    else if(kEventWindowZoomed == eventType)
-    {
-    }
-    else if(kEventWindowExpanding == eventType)
-    {
-    }
-    else if(kEventWindowExpanded == eventType)
-    {
-    }
-    else if(kEventWindowClickResizeRgn == eventType)
-    {
-    }
-    else if(kEventWindowClickDragRgn == eventType)
-    {
-    }
-    else
-    {
-        updateUI = false;
-    }
-
-    if(true == updateUI)
-    {
-        obj->ParentWindowResized(windowRef);
-        obj->UpdateClipping();
-        obj->RenderOffScreenBuffers();
-    }
-
-    return noErr;
-}
-
-#pragma mark VideoRenderAGL HIViewRef constructor
-
-VideoRenderAGL::VideoRenderAGL(HIViewRef windowRef, bool fullscreen, int iId) :
-_hiviewRef( windowRef),
-_windowRef( 0),
-_fullScreen( fullscreen),
-_id( iId),
-_renderCritSec(*CriticalSectionWrapper::CreateCriticalSection()),
-_screenUpdateEvent( 0),
-_isHIViewRef( false),
-_aglContext( 0),
-_windowWidth( 0),
-_windowHeight( 0),
-_lastWindowWidth( -1),
-_lastWindowHeight( -1),
-_lastHiViewWidth( -1),
-_lastHiViewHeight( -1),
-_currentParentWindowHeight( 0),
-_currentParentWindowWidth( 0),
-_currentParentWindowBounds( ),
-_windowHasResized( false),
-_lastParentWindowBounds( ),
-_currentHIViewBounds( ),
-_lastHIViewBounds( ),
-_windowRect( ),
-_aglChannels( ),
-_zOrderToChannel( ),
-_hiviewEventHandlerRef( NULL),
-_windowEventHandlerRef( NULL),
-_currentViewBounds( ),
-_lastViewBounds( ),
-_renderingIsPaused( false),
-{
-    //WEBRTC_TRACE(kTraceDebug, "%s:%d Constructor", __FUNCTION__, __LINE__);
-    //    _renderCritSec = CriticalSectionWrapper::CreateCriticalSection();
-
-    _screenUpdateThread.reset(new rtc::PlatformThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdateThread"));
-    _screenUpdateEvent = EventWrapper::Create();
-
-    GetWindowRect(_windowRect);
-
-    _lastViewBounds.origin.x = 0;
-    _lastViewBounds.origin.y = 0;
-    _lastViewBounds.size.width = 0;
-    _lastViewBounds.size.height = 0;
-
-#ifdef NEW_HIVIEW_PARENT_EVENT_HANDLER
-    // This gets the parent window of the HIViewRef that's passed in and installs a WindowRef event handler on it
-    // The event handler looks for window resize events and adjusts the offset of the controls.
-
-    //WEBRTC_TRACE(kTraceDebug, "%s:%d Installing Eventhandler for hiviewRef's parent window", __FUNCTION__, __LINE__);
-
-
-    static const EventTypeSpec windowEventTypes[] =
-    {
-        kEventClassWindow, kEventWindowBoundsChanged,
-        kEventClassWindow, kEventWindowBoundsChanging,
-        kEventClassWindow, kEventWindowZoomed,
-        kEventClassWindow, kEventWindowExpanded,
-        kEventClassWindow, kEventWindowClickResizeRgn,
-        kEventClassWindow, kEventWindowClickDragRgn
-    };
-
-    WindowRef parentWindow = HIViewGetWindow(windowRef);
-
-    InstallWindowEventHandler (parentWindow,
-            NewEventHandlerUPP (sHandleWindowResized),
-            GetEventTypeCount(windowEventTypes),
-            windowEventTypes,
-            (void *) this, // this is an arbitrary parameter that will be passed on to your event handler when it is called later
-            &_windowEventHandlerRef);
-
-#endif
-
-#ifdef NEW_HIVIEW_EVENT_HANDLER
-    //WEBRTC_TRACE(kTraceDebug, "%s:%d Installing Eventhandler for hiviewRef", __FUNCTION__, __LINE__);
-
-    static const EventTypeSpec hiviewEventTypes[] =
-    {
-        kEventClassControl, kEventControlBoundsChanged,
-        kEventClassControl, kEventControlDraw
-        //			kEventControlDragLeave
-        //			kEventControlDragReceive
-        //			kEventControlGetFocusPart
-        //			kEventControlApplyBackground
-        //			kEventControlDraw
-        //			kEventControlHit
-
-    };
-
-    HIViewInstallEventHandler(_hiviewRef,
-            NewEventHandlerUPP(sHandleHiViewResized),
-            GetEventTypeCount(hiviewEventTypes),
-            hiviewEventTypes,
-            (void *) this,
-            &_hiviewEventHandlerRef);
-
-#endif
-}
-
-// this is a static function. It has been registered (in constructor) to be called on various window redrawing or resizing.
-// Since it is a static method, I have passed in "this" as the userData (one and only allowed) parameter, then calling member methods on it.
-#pragma mark HIViewRef Event Handler
-pascal OSStatus VideoRenderAGL::sHandleHiViewResized (EventHandlerCallRef nextHandler, EventRef theEvent, void* userData)
-{
-    //static int      callbackCounter = 1;
-    HIViewRef hiviewRef = NULL;
-
-    // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/i686-win32/include/quicktime/CarbonEvents.h for a list of codes
-    int eventType = GetEventKind(theEvent);
-    OSStatus status = noErr;
-    status = GetEventParameter (theEvent,
-            kEventParamDirectObject,
-            typeControlRef,
-            NULL,
-            sizeof (ControlRef),
-            NULL,
-            &hiviewRef);
-
-    VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
-    WindowRef parentWindow = HIViewGetWindow(hiviewRef);
-    bool updateUI = true;
-
-    if(kEventControlBoundsChanged == eventType)
-    {
-    }
-    else if(kEventControlDraw == eventType)
-    {
-    }
-    else
-    {
-        updateUI = false;
-    }
-
-    if(true == updateUI)
-    {
-        obj->ParentWindowResized(parentWindow);
-        obj->UpdateClipping();
-        obj->RenderOffScreenBuffers();
-    }
-
-    return status;
-}
-
-VideoRenderAGL::~VideoRenderAGL()
-{
-
-    //WEBRTC_TRACE(kTraceDebug, "%s:%d Destructor", __FUNCTION__, __LINE__);
-
-
-#ifdef USE_EVENT_HANDLERS
-    // remove event handlers
-    OSStatus status;
-    if(_isHIViewRef)
-    {
-        status = RemoveEventHandler(_hiviewEventHandlerRef);
-    }
-    else
-    {
-        status = RemoveEventHandler(_windowEventHandlerRef);
-    }
-    if(noErr != status)
-    {
-        if(_isHIViewRef)
-        {
-
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove hiview event handler: %d", __FUNCTION__, __LINE__, (int)_hiviewEventHandlerRef);
-        }
-        else
-        {
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove window event handler %d", __FUNCTION__, __LINE__, (int)_windowEventHandlerRef);
-        }
-    }
-
-#endif
-
-    OSStatus status;
-#ifdef NEW_HIVIEW_PARENT_EVENT_HANDLER
-    if(_windowEventHandlerRef)
-    {
-        status = RemoveEventHandler(_windowEventHandlerRef);
-        if(status != noErr)
-        {
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d failed to remove window event handler %d", __FUNCTION__, __LINE__, (int)_windowEventHandlerRef);
-        }
-    }
-#endif
-
-#ifdef NEW_HIVIEW_EVENT_HANDLER
-    if(_hiviewEventHandlerRef)
-    {
-        status = RemoveEventHandler(_hiviewEventHandlerRef);
-        if(status != noErr)
-        {
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove hiview event handler: %d", __FUNCTION__, __LINE__, (int)_hiviewEventHandlerRef);
-        }
-    }
-#endif
-
-    // Signal event to exit thread, then delete it
-    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
-
-    if (tmpPtr)
-    {
-        _screenUpdateEvent->Set();
-        _screenUpdateEvent->StopTimer();
-
-        tmpPtr->Stop();
-        delete tmpPtr;
-        delete _screenUpdateEvent;
-        _screenUpdateEvent = NULL;
-    }
-
-    if (_aglContext != 0)
-    {
-        aglSetCurrentContext(_aglContext);
-        aglDestroyContext(_aglContext);
-        _aglContext = 0;
-    }
-
-    // Delete all channels
-    std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
-    while (it!= _aglChannels.end())
-    {
-        delete it->second;
-        _aglChannels.erase(it);
-        it = _aglChannels.begin();
-    }
-    _aglChannels.clear();
-
-    // Clean the zOrder map
-    std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
-    while(zIt != _zOrderToChannel.end())
-    {
-        _zOrderToChannel.erase(zIt);
-        zIt = _zOrderToChannel.begin();
-    }
-    _zOrderToChannel.clear();
-
-    //delete _renderCritSec;
-
-
-}
-
-int VideoRenderAGL::GetOpenGLVersion(int& aglMajor, int& aglMinor)
-{
-    aglGetVersion((GLint *) &aglMajor, (GLint *) &aglMinor);
-    return 0;
-}
-
-int VideoRenderAGL::Init()
-{
-    LockAGLCntx();
-
-    // Start rendering thread...
-    if (!_screenUpdateThread)
-    {
-        UnlockAGLCntx();
-        //WEBRTC_TRACE(kTraceError, "%s:%d Thread not created", __FUNCTION__, __LINE__);
-        return -1;
-    }
-    _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
-
-    // Start the event triggering the render process
-    unsigned int monitorFreq = 60;
-    _screenUpdateEvent->StartTimer(true, 1000/monitorFreq);
-
-    // Create mixing textures
-    if (CreateMixingContext() == -1)
-    {
-        //WEBRTC_TRACE(kTraceError, "%s:%d Could not create a mixing context", __FUNCTION__, __LINE__);
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-VideoChannelAGL* VideoRenderAGL::CreateAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
-    LockAGLCntx();
-
-    //WEBRTC_TRACE(kTraceInfo, "%s:%d Creating AGL channel: %d", __FUNCTION__, __LINE__, channel);
-
-    if (HasChannel(channel))
-    {
-        //WEBRTC_TRACE(kTraceError, "%s:%d Channel already exists", __FUNCTION__, __LINE__);
-        UnlockAGLCntx();k
-        return NULL;
-    }
-
-    if (_zOrderToChannel.find(zOrder) != _zOrderToChannel.end())
-    {
-        // There are already one channel using this zOrder
-        // TODO: Allow multiple channels with same zOrder
-    }
-
-    VideoChannelAGL* newAGLChannel = new VideoChannelAGL(_aglContext, _id, this);
-
-    if (newAGLChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
-    {
-        if (newAGLChannel)
-        {
-            delete newAGLChannel;
-            newAGLChannel = NULL;
-        }
-        //WEBRTC_LOG(kTraceError, "Could not create AGL channel");
-        //WEBRTC_TRACE(kTraceError, "%s:%d Could not create AGL channel", __FUNCTION__, __LINE__);
-        UnlockAGLCntx();
-        return NULL;
-    }
-k
-    _aglChannels[channel] = newAGLChannel;
-    _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
-
-    UnlockAGLCntx();
-    return newAGLChannel;
-}
-
-int VideoRenderAGL::DeleteAllAGLChannels()
-{
-    CriticalSectionScoped cs(&_renderCritSec);
-
-    //WEBRTC_TRACE(kTraceInfo, "%s:%d Deleting all AGL channels", __FUNCTION__, __LINE__);
-    //int i = 0 ;
-    std::map<int, VideoChannelAGL*>::iterator it;
-    it = _aglChannels.begin();
-
-    while (it != _aglChannels.end())
-    {
-        VideoChannelAGL* channel = it->second;
-        if (channel)
-        delete channel;
-
-        _aglChannels.erase(it);
-        it = _aglChannels.begin();
-    }
-    _aglChannels.clear();
-    return 0;
-}
-
-int VideoRenderAGL::DeleteAGLChannel(int channel)
-{
-    CriticalSectionScoped cs(&_renderCritSec);
-    //WEBRTC_TRACE(kTraceDebug, "%s:%d Deleting AGL channel %d", __FUNCTION__, __LINE__, channel);
-
-    std::map<int, VideoChannelAGL*>::iterator it;
-    it = _aglChannels.find(channel);
-    if (it != _aglChannels.end())
-    {
-        delete it->second;
-        _aglChannels.erase(it);
-    }
-    else
-    {
-        //WEBRTC_TRACE(kTraceWarning, "%s:%d Channel not found", __FUNCTION__, __LINE__);
-        return -1;
-    }
-
-    std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
-    while( zIt != _zOrderToChannel.end())
-    {
-        if (zIt->second == channel)
-        {
-            _zOrderToChannel.erase(zIt);
-            break;
-        }
-        zIt++;// = _zOrderToChannel.begin();
-    }
-
-    return 0;
-}
-
-int VideoRenderAGL::StopThread()
-{
-    CriticalSectionScoped cs(&_renderCritSec);
-    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
-
-    if (tmpPtr)
-    {
-        _screenUpdateEvent->Set();
-        _renderCritSec.Leave();
-        tmpPtr->Stop();
-        delete tmpPtr;
-        _renderCritSec.Enter();
-    }
-
-    delete _screenUpdateEvent;
-    _screenUpdateEvent = NULL;
-
-    return 0;
-}
-
-bool VideoRenderAGL::IsFullScreen()
-{
-    CriticalSectionScoped cs(&_renderCritSec);
-    return _fullScreen;
-}
-
-bool VideoRenderAGL::HasChannels()
-{
-
-    CriticalSectionScoped cs(&_renderCritSec);
-
-    if (_aglChannels.begin() != _aglChannels.end())
-    {
-        return true;
-    }
-
-    return false;
-}
-
-bool VideoRenderAGL::HasChannel(int channel)
-{
-    CriticalSectionScoped cs(&_renderCritSec);
-
-    std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channel);
-    if (it != _aglChannels.end())
-    {
-        return true;
-    }
-
-    return false;
-}
-
-int VideoRenderAGL::GetChannels(std::list<int>& channelList)
-{
-
-    CriticalSectionScoped cs(&_renderCritSec);
-    std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
-
-    while (it != _aglChannels.end())
-    {
-        channelList.push_back(it->first);
-        it++;
-    }
-
-    return 0;
-}
-
-VideoChannelAGL* VideoRenderAGL::ConfigureAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
-    CriticalSectionScoped cs(&_renderCritSec);
-
-    std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channel);
-
-    if (it != _aglChannels.end())
-    {
-        VideoChannelAGL* aglChannel = it->second;
-        if (aglChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
-        {
-            return NULL;
-        }
-
-        std::multimap<int, int>::iterator it = _zOrderToChannel.begin();
-        while(it != _zOrderToChannel.end())
-        {
-            if (it->second == channel)
-            {
-                if (it->first != zOrder)
-                {
-                    _zOrderToChannel.erase(it);
-                    _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
-                }
-                break;
-            }
-            it++;
-        }
-        return aglChannel;
-    }
-
-    return NULL;
-}
-
-bool VideoRenderAGL::ScreenUpdateThreadProc(void* obj)
-{
-    return static_cast<VideoRenderAGL*>(obj)->ScreenUpdateProcess();
-}
-
-bool VideoRenderAGL::ScreenUpdateProcess()
-{
-    _screenUpdateEvent->Wait(100);
-
-    LockAGLCntx();
-
-    if (!_screenUpdateThread)
-    {
-        UnlockAGLCntx();
-        return false;
-    }
-
-    if (aglSetCurrentContext(_aglContext) == GL_FALSE)
-    {
-        UnlockAGLCntx();
-        return true;
-    }
-
-    if (GetWindowRect(_windowRect) == -1)
-    {
-        UnlockAGLCntx();
-        return true;
-    }
-
-    if (_windowWidth != (_windowRect.right - _windowRect.left)
-            || _windowHeight != (_windowRect.bottom - _windowRect.top))
-    {
-        // We have a new window size, update the context.
-        if (aglUpdateContext(_aglContext) == GL_FALSE)
-        {
-            UnlockAGLCntx();
-            return true;
-        }
-        _windowWidth = _windowRect.right - _windowRect.left;
-        _windowHeight = _windowRect.bottom - _windowRect.top;
-    }
-
-    // this section will poll to see if the window size has changed
-    // this is causing problem w/invalid windowRef
-    // this code has been modified and exists now in the window event handler
-#ifndef NEW_HIVIEW_PARENT_EVENT_HANDLER
-    if (_isHIViewRef)
-    {
-
-        if(FALSE == HIViewIsValid(_hiviewRef))
-        {
-
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d Invalid windowRef", __FUNCTION__, __LINE__);
-            UnlockAGLCntx();
-            return true;
-        }
-        WindowRef window = HIViewGetWindow(_hiviewRef);
-
-        if(FALSE == IsValidWindowPtr(window))
-        {
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d Invalide hiviewRef", __FUNCTION__, __LINE__);
-            UnlockAGLCntx();
-            return true;
-        }
-        if (window == NULL)
-        {
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d WindowRef = NULL", __FUNCTION__, __LINE__);
-            UnlockAGLCntx();
-            return true;
-        }
-
-        if(FALSE == MacIsWindowVisible(window))
-        {
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d MacIsWindowVisible == FALSE. Returning early", __FUNCTION__, __LINE__);
-            UnlockAGLCntx();
-            return true;
-        }
-
-        HIRect viewBounds; // Placement and size for HIView
-        int windowWidth = 0; // Parent window width
-        int windowHeight = 0; // Parent window height
-
-        // NOTE: Calling GetWindowBounds with kWindowStructureRgn will crash intermittentaly if the OS decides it needs to push it into the back for a moment.
-        // To counter this, we get the titlebar height on class construction and then add it to the content region here. Content regions seems not to crash
-        Rect contentBounds =
-        {   0, 0, 0, 0}; // The bounds for the parent window
-
-#if		defined(USE_CONTENT_RGN)
-        GetWindowBounds(window, kWindowContentRgn, &contentBounds);
-#elif	defined(USE_STRUCT_RGN)
-        GetWindowBounds(window, kWindowStructureRgn, &contentBounds);
-#endif
-
-        Rect globalBounds =
-        {   0, 0, 0, 0}; // The bounds for the parent window
-        globalBounds.top = contentBounds.top;
-        globalBounds.right = contentBounds.right;
-        globalBounds.bottom = contentBounds.bottom;
-        globalBounds.left = contentBounds.left;
-
-        windowHeight = globalBounds.bottom - globalBounds.top;
-        windowWidth = globalBounds.right - globalBounds.left;
-
-        // Get the size of the HIViewRef
-        HIViewGetBounds(_hiviewRef, &viewBounds);
-        HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
-
-        // Check if this is the first call..
-        if (_lastWindowHeight == -1 &&
-                _lastWindowWidth == -1)
-        {
-            _lastWindowWidth = windowWidth;
-            _lastWindowHeight = windowHeight;
-
-            _lastViewBounds.origin.x = viewBounds.origin.x;
-            _lastViewBounds.origin.y = viewBounds.origin.y;
-            _lastViewBounds.size.width = viewBounds.size.width;
-            _lastViewBounds.size.height = viewBounds.size.height;
-        }
-        sfasdfasdf
-
-        bool resized = false;
-
-        // Check if parent window size has changed
-        if (windowHeight != _lastWindowHeight ||
-                windowWidth != _lastWindowWidth)
-        {
-            resized = true;
-        }
-
-        // Check if the HIView has new size or is moved in the parent window
-        if (_lastViewBounds.origin.x != viewBounds.origin.x ||
-                _lastViewBounds.origin.y != viewBounds.origin.y ||
-                _lastViewBounds.size.width != viewBounds.size.width ||
-                _lastViewBounds.size.height != viewBounds.size.height)
-        {
-            // The HiView is resized or has moved.
-            resized = true;
-        }
-
-        if (resized)
-        {
-
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d Window has resized", __FUNCTION__, __LINE__);
-
-            // Calculate offset between the windows
-            // {x, y, widht, height}, x,y = lower left corner
-            const GLint offs[4] =
-            {   (int)(0.5f + viewBounds.origin.x),
-                (int)(0.5f + windowHeight - (viewBounds.origin.y + viewBounds.size.height)),
-                viewBounds.size.width, viewBounds.size.height};
-
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d contentBounds	t:%d r:%d b:%d l:%d", __FUNCTION__, __LINE__,
-            contentBounds.top, contentBounds.right, contentBounds.bottom, contentBounds.left);
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d windowHeight=%d", __FUNCTION__, __LINE__, windowHeight);
-            //WEBRTC_TRACE(kTraceDebug, "%s:%d offs[4] = %d, %d, %d, %d", __FUNCTION__, __LINE__, offs[0], offs[1], offs[2], offs[3]);
-
-            aglSetDrawable (_aglContext, GetWindowPort(window));
-            aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
-            aglEnable(_aglContext, AGL_BUFFER_RECT);
-
-            // We need to change the viewport too if the HIView size has changed
-            glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
-
-        }
-        _lastWindowWidth = windowWidth;
-        _lastWindowHeight = windowHeight;
-
-        _lastViewBounds.origin.x = viewBounds.origin.x;
-        _lastViewBounds.origin.y = viewBounds.origin.y;
-        _lastViewBounds.size.width = viewBounds.size.width;
-        _lastViewBounds.size.height = viewBounds.size.height;
-
-    }
-#endif
-    if (_fullScreen)
-    {
-        // TODO
-        // We use double buffers, must always update
-        //RenderOffScreenBuffersToBackBuffer();
-    }
-    else
-    {
-        // Check if there are any updated buffers
-        bool updated = false;
-
-        // TODO: check if window size is updated!
-        // TODO Improvement: Walk through the zOrder Map to only render the ones in need of update
-        std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
-        while (it != _aglChannels.end())
-        {
-
-            VideoChannelAGL* aglChannel = it->second;
-            aglChannel->UpdateStretchSize(_windowHeight, _windowWidth);
-            aglChannel->IsUpdated(updated);
-            if (updated)
-            {
-                break;
-            }
-            it++;
-        }
-
-        if (updated)
-        {
-            // At least on buffers is updated, we need to repaint the texture
-            if (RenderOffScreenBuffers() != -1)
-            {
-                // MF
-                //SwapAndDisplayBuffers();
-            }
-            else
-            {
-                // Error updating the mixing texture, don't swap.
-            }
-        }
-    }
-
-    UnlockAGLCntx();
-
-    //WEBRTC_LOG(kTraceDebug, "Leaving ScreenUpdateProcess()");
-    return true;
-}
-
-void VideoRenderAGL::ParentWindowResized(WindowRef window)
-{
-    //WEBRTC_LOG(kTraceDebug, "%s HIViewRef:%d owner window has resized", __FUNCTION__, (int)_hiviewRef);
-
-    LockAGLCntx();
-k
-    // set flag
-    _windowHasResized = false;
-
-    if(FALSE == HIViewIsValid(_hiviewRef))
-    {
-        //WEBRTC_LOG(kTraceDebug, "invalid windowRef");
-        UnlockAGLCntx();
-        return;
-    }
-
-    if(FALSE == IsValidWindowPtr(window))
-    {
-        //WEBRTC_LOG(kTraceError, "invalid windowRef");
-        UnlockAGLCntx();
-        return;
-    }
-
-    if (window == NULL)
-    {
-        //WEBRTC_LOG(kTraceError, "windowRef = NULL");
-        UnlockAGLCntx();
-        return;
-    }
-
-    if(FALSE == MacIsWindowVisible(window))
-    {
-        //WEBRTC_LOG(kTraceDebug, "MacIsWindowVisible = FALSE. Returning early.");
-        UnlockAGLCntx();
-        return;
-    }
-
-    Rect contentBounds =
-    {   0, 0, 0, 0};
-
-#if		defined(USE_CONTENT_RGN)
-    GetWindowBounds(window, kWindowContentRgn, &contentBounds);
-#elif	defined(USE_STRUCT_RGN)
-    GetWindowBounds(window, kWindowStructureRgn, &contentBounds);
-#endif
-
-    //WEBRTC_LOG(kTraceDebug, "%s contentBounds	t:%d r:%d b:%d l:%d", __FUNCTION__, contentBounds.top, contentBounds.right, contentBounds.bottom, contentBounds.left);
-
-    // update global vars
-    _currentParentWindowBounds.top = contentBounds.top;
-    _currentParentWindowBounds.left = contentBounds.left;
-    _currentParentWindowBounds.bottom = contentBounds.bottom;
-    _currentParentWindowBounds.right = contentBounds.right;
-
-    _currentParentWindowWidth = _currentParentWindowBounds.right - _currentParentWindowBounds.left;
-    _currentParentWindowHeight = _currentParentWindowBounds.bottom - _currentParentWindowBounds.top;
-
-    _windowHasResized = true;
-
-    // ********* update AGL offsets
-    HIRect viewBounds;
-    HIViewGetBounds(_hiviewRef, &viewBounds);
-    HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
-
-    const GLint offs[4] =
-    {   (int)(0.5f + viewBounds.origin.x),
-        (int)(0.5f + _currentParentWindowHeight - (viewBounds.origin.y + viewBounds.size.height)),
-        viewBounds.size.width, viewBounds.size.height};
-    //WEBRTC_LOG(kTraceDebug, "%s _currentParentWindowHeight=%d", __FUNCTION__, _currentParentWindowHeight);
-    //WEBRTC_LOG(kTraceDebug, "%s offs[4] = %d, %d, %d, %d", __FUNCTION__, offs[0], offs[1], offs[2], offs[3]);
-
-    aglSetCurrentContext(_aglContext);
-    aglSetDrawable (_aglContext, GetWindowPort(window));
-    aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
-    aglEnable(_aglContext, AGL_BUFFER_RECT);
-
-    // We need to change the viewport too if the HIView size has changed
-    glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
-
-    UnlockAGLCntx();
-
-    return;
-}
-
-int VideoRenderAGL::CreateMixingContext()
-{
-
-    LockAGLCntx();
-
-    //WEBRTC_LOG(kTraceDebug, "Entering CreateMixingContext()");
-
-    // Use both AGL_ACCELERATED and AGL_NO_RECOVERY to make sure
-    // a hardware renderer is used and not a software renderer.
-
-    GLint attributes[] =
-    {
-        AGL_DOUBLEBUFFER,
-        AGL_WINDOW,
-        AGL_RGBA,
-        AGL_NO_RECOVERY,
-        AGL_ACCELERATED,
-        AGL_RED_SIZE, 8,
-        AGL_GREEN_SIZE, 8,
-        AGL_BLUE_SIZE, 8,
-        AGL_ALPHA_SIZE, 8,
-        AGL_DEPTH_SIZE, 24,
-        AGL_NONE,
-    };
-
-    AGLPixelFormat aglPixelFormat;
-
-    // ***** Set up the OpenGL Context *****
-
-    // Get a pixel format for the attributes above
-    aglPixelFormat = aglChoosePixelFormat(NULL, 0, attributes);
-    if (NULL == aglPixelFormat)
-    {
-        //WEBRTC_LOG(kTraceError, "Could not create pixel format");
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    // Create an AGL context
-    _aglContext = aglCreateContext(aglPixelFormat, NULL);
-    if (_aglContext == NULL)
-    {
-        //WEBRTC_LOG(kTraceError, "Could no create AGL context");
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    // Release the pixel format memory
-    aglDestroyPixelFormat(aglPixelFormat);
-
-    // Set the current AGL context for the rest of the settings
-    if (aglSetCurrentContext(_aglContext) == false)
-    {
-        //WEBRTC_LOG(kTraceError, "Could not set current context: %d", aglGetError());
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    if (_isHIViewRef)
-    {
-        //---------------------------
-        // BEGIN: new test code
-#if 0
-        // Don't use this one!
-        // There seems to be an OS X bug that can't handle
-        // movements and resizing of the parent window
-        // and or the HIView
-        if (aglSetHIViewRef(_aglContext,_hiviewRef) == false)
-        {
-            //WEBRTC_LOG(kTraceError, "Could not set WindowRef: %d", aglGetError());
-            UnlockAGLCntx();
-            return -1;
-        }
-#else
-
-        // Get the parent window for this control
-        WindowRef window = GetControlOwner(_hiviewRef);
-
-        Rect globalBounds =
-        {   0,0,0,0}; // The bounds for the parent window
-        HIRect viewBounds; // Placemnt in the parent window and size.
-        int windowHeight = 0;
-
-        //		Rect titleBounds = {0,0,0,0};
-        //		GetWindowBounds(window, kWindowTitleBarRgn, &titleBounds);
-        //		_titleBarHeight = titleBounds.top - titleBounds.bottom;
-        //		if(0 == _titleBarHeight)
-        //		{
-        //            //WEBRTC_LOG(kTraceError, "Titlebar height = 0");
-        //            //return -1;
-        //		}
-
-
-        // Get the bounds for the parent window
-#if		defined(USE_CONTENT_RGN)
-        GetWindowBounds(window, kWindowContentRgn, &globalBounds);
-#elif	defined(USE_STRUCT_RGN)
-        GetWindowBounds(window, kWindowStructureRgn, &globalBounds);
-#endif
-        windowHeight = globalBounds.bottom - globalBounds.top;
-
-        // Get the bounds for the HIView
-        HIViewGetBounds(_hiviewRef, &viewBounds);
-
-        HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
-
-        const GLint offs[4] =
-        {   (int)(0.5f + viewBounds.origin.x),
-            (int)(0.5f + windowHeight - (viewBounds.origin.y + viewBounds.size.height)),
-            viewBounds.size.width, viewBounds.size.height};
-
-        //WEBRTC_LOG(kTraceDebug, "%s offs[4] = %d, %d, %d, %d", __FUNCTION__, offs[0], offs[1], offs[2], offs[3]);
-
-
-        aglSetDrawable (_aglContext, GetWindowPort(window));
-        aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
-        aglEnable(_aglContext, AGL_BUFFER_RECT);
-
-        GLint surfaceOrder = 1; // 1: above window, -1 below.
-        //OSStatus status = aglSetInteger(_aglContext, AGL_SURFACE_ORDER, &surfaceOrder);
-        aglSetInteger(_aglContext, AGL_SURFACE_ORDER, &surfaceOrder);
-
-        glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
-#endif
-
-    }
-    else
-    {
-        if(GL_FALSE == aglSetDrawable (_aglContext, GetWindowPort(_windowRef)))
-        {
-            //WEBRTC_LOG(kTraceError, "Could not set WindowRef: %d", aglGetError());
-            UnlockAGLCntx();
-            return -1;
-        }
-    }
-
-    _windowWidth = _windowRect.right - _windowRect.left;
-    _windowHeight = _windowRect.bottom - _windowRect.top;
-
-    // opaque surface
-    int surfaceOpacity = 1;
-    if (aglSetInteger(_aglContext, AGL_SURFACE_OPACITY, (const GLint *) &surfaceOpacity) == false)
-    {
-        //WEBRTC_LOG(kTraceError, "Could not set surface opacity: %d", aglGetError());
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    // 1 -> sync to screen rat, slow...
-    //int swapInterval = 0;  // 0 don't sync with vertical trace
-    int swapInterval = 0; // 1 sync with vertical trace
-    if (aglSetInteger(_aglContext, AGL_SWAP_INTERVAL, (const GLint *) &swapInterval) == false)
-    {
-        //WEBRTC_LOG(kTraceError, "Could not set swap interval: %d", aglGetError());
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    // Update the rect with the current size
-    if (GetWindowRect(_windowRect) == -1)
-    {
-        //WEBRTC_LOG(kTraceError, "Could not get window size");
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    // Disable not needed functionality to increase performance
-    glDisable(GL_DITHER);
-    glDisable(GL_ALPHA_TEST);
-    glDisable(GL_STENCIL_TEST);
-    glDisable(GL_FOG);
-    glDisable(GL_TEXTURE_2D);
-    glPixelZoom(1.0, 1.0);
-
-    glDisable(GL_BLEND);
-    glDisable(GL_DEPTH_TEST);
-    glDepthMask(GL_FALSE);
-    glDisable(GL_CULL_FACE);
-
-    glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
-    glClear(GL_COLOR_BUFFER_BIT);
-
-    GLenum glErr = glGetError();
-
-    if (glErr)
-    {
-    }
-
-    UpdateClipping();
-
-    //WEBRTC_LOG(kTraceDebug, "Leaving CreateMixingContext()");
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int VideoRenderAGL::RenderOffScreenBuffers()
-{
-    LockAGLCntx();
-
-    // Get the current window size, it might have changed since last render.
-    if (GetWindowRect(_windowRect) == -1)
-    {
-        //WEBRTC_LOG(kTraceError, "Could not get window rect");
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    if (aglSetCurrentContext(_aglContext) == false)
-    {
-        //WEBRTC_LOG(kTraceError, "Could not set current context for rendering");
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    // HERE - onl if updated!
-    glClear(GL_COLOR_BUFFER_BIT);
-
-    // Loop through all channels starting highest zOrder ending with lowest.
-    for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
-    rIt != _zOrderToChannel.rend();
-    rIt++)
-    {
-        int channelId = rIt->second;
-        std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channelId);
-
-        VideoChannelAGL* aglChannel = it->second;
-
-        aglChannel->RenderOffScreenBuffer();
-    }
-
-    SwapAndDisplayBuffers();
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int VideoRenderAGL::SwapAndDisplayBuffers()
-{
-
-    LockAGLCntx();
-    if (_fullScreen)
-    {
-        // TODO:
-        // Swap front and back buffers, rendering taking care of in the same call
-        //aglSwapBuffers(_aglContext);
-        // Update buffer index to the idx for the next rendering!
-        //_textureIdx = (_textureIdx + 1) & 1;
-    }
-    else
-    {
-        // Single buffer rendering, only update context.
-        glFlush();
-        aglSwapBuffers(_aglContext);
-        HIViewSetNeedsDisplay(_hiviewRef, true);
-    }
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int VideoRenderAGL::GetWindowRect(Rect& rect)
-{
-
-    LockAGLCntx();
-
-    if (_isHIViewRef)
-    {
-        if (_hiviewRef)
-        {
-            HIRect HIViewRect1;
-            if(FALSE == HIViewIsValid(_hiviewRef))
-            {
-                rect.top = 0;
-                rect.left = 0;
-                rect.right = 0;
-                rect.bottom = 0;
-                //WEBRTC_LOG(kTraceError,"GetWindowRect() HIViewIsValid() returned false");
-                UnlockAGLCntx();
-            }
-            HIViewGetBounds(_hiviewRef,&HIViewRect1);
-            HIRectConvert(&HIViewRect1, 1, NULL, 2, NULL);
-            if(HIViewRect1.origin.x < 0)
-            {
-                rect.top = 0;
-                //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.top = 0");
-            }
-            else
-            {
-                rect.top = HIViewRect1.origin.x;
-            }
-
-            if(HIViewRect1.origin.y < 0)
-            {
-                rect.left = 0;
-                //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.left = 0");
-            }
-            else
-            {
-                rect.left = HIViewRect1.origin.y;
-            }
-
-            if(HIViewRect1.size.width < 0)
-            {
-                rect.right = 0;
-                //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.right = 0");
-            }
-            else
-            {
-                rect.right = HIViewRect1.size.width;
-            }
-
-            if(HIViewRect1.size.height < 0)
-            {
-                rect.bottom = 0;
-                //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.bottom = 0");
-            }
-            else
-            {
-                rect.bottom = HIViewRect1.size.height;
-            }
-
-            ////WEBRTC_LOG(kTraceDebug,"GetWindowRect() HIViewRef: rect.top = %d, rect.left = %d, rect.right = %d, rect.bottom =%d in GetWindowRect", rect.top,rect.left,rect.right,rect.bottom);
-            UnlockAGLCntx();
-        }
-        else
-        {
-            //WEBRTC_LOG(kTraceError, "invalid HIViewRef");
-            UnlockAGLCntx();
-        }
-    }
-    else
-    {
-        if (_windowRef)
-        {
-            GetWindowBounds(_windowRef, kWindowContentRgn, &rect);
-            UnlockAGLCntx();
-        }
-        else
-        {
-            //WEBRTC_LOG(kTraceError, "No WindowRef");
-            UnlockAGLCntx();
-        }
-    }
-}
-
-int VideoRenderAGL::UpdateClipping()
-{
-    //WEBRTC_LOG(kTraceDebug, "Entering UpdateClipping()");
-    LockAGLCntx();
-
-    if(_isHIViewRef)
-    {
-        if(FALSE == HIViewIsValid(_hiviewRef))
-        {
-            //WEBRTC_LOG(kTraceError, "UpdateClipping() _isHIViewRef is invalid. Returning -1");
-            UnlockAGLCntx();
-            return -1;
-        }
-
-        RgnHandle visibleRgn = NewRgn();
-        SetEmptyRgn (visibleRgn);
-
-        if(-1 == CalculateVisibleRegion((ControlRef)_hiviewRef, visibleRgn, true))
-        {
-        }
-
-        if(GL_FALSE == aglSetCurrentContext(_aglContext))
-        {
-            GLenum glErr = aglGetError();
-            //WEBRTC_LOG(kTraceError, "aglSetCurrentContext returned FALSE with error code %d at line %d", glErr, __LINE__);
-        }
-
-        if(GL_FALSE == aglEnable(_aglContext, AGL_CLIP_REGION))
-        {
-            GLenum glErr = aglGetError();
-            //WEBRTC_LOG(kTraceError, "aglEnable returned FALSE with error code %d at line %d\n", glErr, __LINE__);
-        }
-
-        if(GL_FALSE == aglSetInteger(_aglContext, AGL_CLIP_REGION, (const GLint*)visibleRgn))
-        {
-            GLenum glErr = aglGetError();
-            //WEBRTC_LOG(kTraceError, "aglSetInteger returned FALSE with error code %d at line %d\n", glErr, __LINE__);
-        }
-
-        DisposeRgn(visibleRgn);
-    }
-    else
-    {
-        //WEBRTC_LOG(kTraceDebug, "Not using a hiviewref!\n");
-    }
-
-    //WEBRTC_LOG(kTraceDebug, "Leaving UpdateClipping()");
-    UnlockAGLCntx();
-    return true;
-}
-
-int VideoRenderAGL::CalculateVisibleRegion(ControlRef control, RgnHandle &visibleRgn, bool clipChildren)
-{
-
-    //	LockAGLCntx();
-
-    //WEBRTC_LOG(kTraceDebug, "Entering CalculateVisibleRegion()");
-    OSStatus osStatus = 0;
-    OSErr osErr = 0;
-
-    RgnHandle tempRgn = NewRgn();
-    if (IsControlVisible(control))
-    {
-        RgnHandle childRgn = NewRgn();
-        WindowRef window = GetControlOwner(control);
-        ControlRef rootControl;
-        GetRootControl(window, &rootControl); // 'wvnc'
-        ControlRef masterControl;
-        osStatus = GetSuperControl(rootControl, &masterControl);
-        // //WEBRTC_LOG(kTraceDebug, "IBM GetSuperControl=%d", osStatus);
-
-        if (masterControl != NULL)
-        {
-            CheckValidRegion(visibleRgn);
-            // init visibleRgn with region of 'wvnc'
-            osStatus = GetControlRegion(rootControl, kControlStructureMetaPart, visibleRgn);
-            // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d : %d", osStatus, __LINE__);
-            //GetSuperControl(rootControl, &rootControl);
-            ControlRef tempControl = control, lastControl = 0;
-            while (tempControl != masterControl) // current control != master
-
-            {
-                CheckValidRegion(tempRgn);
-
-                // //WEBRTC_LOG(kTraceDebug, "IBM tempControl=%d masterControl=%d", tempControl, masterControl);
-                ControlRef subControl;
-
-                osStatus = GetControlRegion(tempControl, kControlStructureMetaPart, tempRgn); // intersect the region of the current control with visibleRgn
-                // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d : %d", osStatus, __LINE__);
-                CheckValidRegion(tempRgn);
-
-                osErr = HIViewConvertRegion(tempRgn, tempControl, rootControl);
-                // //WEBRTC_LOG(kTraceDebug, "IBM HIViewConvertRegion=%d : %d", osErr, __LINE__);
-                CheckValidRegion(tempRgn);
-
-                SectRgn(tempRgn, visibleRgn, visibleRgn);
-                CheckValidRegion(tempRgn);
-                CheckValidRegion(visibleRgn);
-                if (EmptyRgn(visibleRgn)) // if the region is empty, bail
-                break;
-
-                if (clipChildren || tempControl != control) // clip children if true, cut out the tempControl if it's not one passed to this function
-
-                {
-                    UInt16 numChildren;
-                    osStatus = CountSubControls(tempControl, &numChildren); // count the subcontrols
-                    // //WEBRTC_LOG(kTraceDebug, "IBM CountSubControls=%d : %d", osStatus, __LINE__);
-
-                    // //WEBRTC_LOG(kTraceDebug, "IBM numChildren=%d", numChildren);
-                    for (int i = 0; i < numChildren; i++)
-                    {
-                        osErr = GetIndexedSubControl(tempControl, numChildren - i, &subControl); // retrieve the subcontrol in order by zorder
-                        // //WEBRTC_LOG(kTraceDebug, "IBM GetIndexedSubControls=%d : %d", osErr, __LINE__);
-                        if ( subControl == lastControl ) // break because of zorder
-
-                        {
-                            // //WEBRTC_LOG(kTraceDebug, "IBM breaking because of zorder %d", __LINE__);
-                            break;
-                        }
-
-                        if (!IsControlVisible(subControl)) // dont' clip invisible controls
-
-                        {
-                            // //WEBRTC_LOG(kTraceDebug, "IBM continue. Control is not visible %d", __LINE__);
-                            continue;
-                        }
-
-                        if(!subControl) continue;
-
-                        osStatus = GetControlRegion(subControl, kControlStructureMetaPart, tempRgn); //get the region of the current control and union to childrg
-                        // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d %d", osStatus, __LINE__);
-                        CheckValidRegion(tempRgn);
-                        if(osStatus != 0)
-                        {
-                            // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! osStatus=%d. Continuing. %d", osStatus, __LINE__);
-                            continue;
-                        }
-                        if(!tempRgn)
-                        {
-                            // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !tempRgn %d", osStatus, __LINE__);
-                            continue;
-                        }
-
-                        osStatus = HIViewConvertRegion(tempRgn, subControl, rootControl);
-                        CheckValidRegion(tempRgn);
-                        // //WEBRTC_LOG(kTraceDebug, "IBM HIViewConvertRegion=%d %d", osStatus, __LINE__);
-                        if(osStatus != 0)
-                        {
-                            // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! osStatus=%d. Continuing. %d", osStatus, __LINE__);
-                            continue;
-                        }
-                        if(!rootControl)
-                        {
-                            // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !rootControl %d", osStatus, __LINE__);
-                            continue;
-                        }
-
-                        UnionRgn(tempRgn, childRgn, childRgn);
-                        CheckValidRegion(tempRgn);
-                        CheckValidRegion(childRgn);
-                        CheckValidRegion(visibleRgn);
-                        if(!childRgn)
-                        {
-                            // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !childRgn %d", osStatus, __LINE__);
-                            continue;
-                        }
-
-                    }  // next child control
-                }
-                lastControl = tempControl;
-                GetSuperControl(tempControl, &subControl);
-                tempControl = subControl;
-            }
-
-            DiffRgn(visibleRgn, childRgn, visibleRgn);
-            CheckValidRegion(visibleRgn);
-            CheckValidRegion(childRgn);
-            DisposeRgn(childRgn);
-        }
-        else
-        {
-            CopyRgn(tempRgn, visibleRgn);
-            CheckValidRegion(tempRgn);
-            CheckValidRegion(visibleRgn);
-        }
-        DisposeRgn(tempRgn);
-    }
-
-    //WEBRTC_LOG(kTraceDebug, "Leaving CalculateVisibleRegion()");
-    //_aglCritPtr->Leave();
-    return 0;
-}
-
-bool VideoRenderAGL::CheckValidRegion(RgnHandle rHandle)
-{
-
-    Handle hndSize = (Handle)rHandle;
-    long size = GetHandleSize(hndSize);
-    if(0 == size)
-    {
-
-        OSErr memErr = MemError();
-        if(noErr != memErr)
-        {
-            // //WEBRTC_LOG(kTraceError, "IBM ERROR Could not get size of handle. MemError() returned %d", memErr);
-        }
-        else
-        {
-            // //WEBRTC_LOG(kTraceError, "IBM ERROR Could not get size of handle yet MemError() returned noErr");
-        }
-
-    }
-    else
-    {
-        // //WEBRTC_LOG(kTraceDebug, "IBM handleSize = %d", size);
-    }
-
-    if(false == IsValidRgnHandle(rHandle))
-    {
-        // //WEBRTC_LOG(kTraceError, "IBM ERROR Invalid Region found : $%d", rHandle);
-        assert(false);
-    }
-
-    int err = QDError();
-    switch(err)
-    {
-        case 0:
-        break;
-        case -147:
-        //WEBRTC_LOG(kTraceError, "ERROR region too big");
-        assert(false);
-        break;
-
-        case -149:
-        //WEBRTC_LOG(kTraceError, "ERROR not enough stack");
-        assert(false);
-        break;
-
-        default:
-        //WEBRTC_LOG(kTraceError, "ERROR Unknown QDError %d", err);
-        assert(false);
-        break;
-    }
-
-    return true;
-}
-
-int VideoRenderAGL::ChangeWindow(void* newWindowRef)
-{
-
-    LockAGLCntx();
-
-    UnlockAGLCntx();
-    return -1;
-}
-
-int32_t VideoRenderAGL::StartRender()
-{
-
-    LockAGLCntx();
-    const unsigned int MONITOR_FREQ = 60;
-    if(TRUE == _renderingIsPaused)
-    {
-        //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Rendering is paused. Restarting now", __FUNCTION__, __LINE__);
-
-        // we already have the thread. Most likely StopRender() was called and they were paused
-        if(FALSE == _screenUpdateThread->Start())
-        {
-            //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateThread", __FUNCTION__, __LINE__);
-            UnlockAGLCntx();
-            return -1;
-        }
-        _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
-        if(FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
-        {
-            //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateEvent", __FUNCTION__, __LINE__);
-            UnlockAGLCntx();
-            return -1;
-        }
-
-        return 0;
-    }
-
-    _screenUpdateThread.reset(
-        new rtc::PlatformThread(ScreenUpdateThreadProc, this, "ScreenUpdate"));
-    _screenUpdateEvent = EventWrapper::Create();
-
-    if (!_screenUpdateThread)
-    {
-        //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateThread", __FUNCTION__, __LINE__);
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
-    _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ);
-
-    //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Started screenUpdateThread", __FUNCTION__, __LINE__);
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int32_t VideoRenderAGL::StopRender()
-{
-    LockAGLCntx();
-
-    if(!_screenUpdateThread || !_screenUpdateEvent)
-    {
-        _renderingIsPaused = TRUE;
-        UnlockAGLCntx();
-        return 0;
-    }
-
-    if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
-    {
-        _renderingIsPaused = FALSE;
-        //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Could not stop either: screenUpdateThread or screenUpdateEvent", __FUNCTION__, __LINE__);
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    _renderingIsPaused = TRUE;
-
-    //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Stopped screenUpdateThread", __FUNCTION__, __LINE__);
-    UnlockAGLCntx();
-    return 0;
-}
-
-int32_t VideoRenderAGL::DeleteAGLChannel(const uint32_t streamID)
-{
-
-    LockAGLCntx();
-
-    std::map<int, VideoChannelAGL*>::iterator it;
-    it = _aglChannels.begin();
-
-    while (it != _aglChannels.end())
-    {
-        VideoChannelAGL* channel = it->second;
-        //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Deleting channel %d", __FUNCTION__, __LINE__, streamID);
-        delete channel;
-        it++;
-    }
-    _aglChannels.clear();
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int32_t VideoRenderAGL::GetChannelProperties(const uint16_t streamId,
-                                             uint32_t& zOrder,
-                                             float& left,
-                                             float& top,
-                                             float& right,
-                                             float& bottom)
-{
-
-    LockAGLCntx();
-    UnlockAGLCntx();
-    return -1;
-
-}
-
-void VideoRenderAGL::LockAGLCntx()
-{
-    _renderCritSec.Enter();
-}
-void VideoRenderAGL::UnlockAGLCntx()
-{
-    _renderCritSec.Leave();
-}
-
-}  // namespace webrtc
-
-#endif   // CARBON_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_agl.h b/chromium/third_party/webrtc/modules/video_render/mac/video_render_agl.h
deleted file mode 100644
index c0a60597e98..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_agl.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-
-#if defined(CARBON_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
-
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-#define NEW_HIVIEW_PARENT_EVENT_HANDLER 1
-#define NEW_HIVIEW_EVENT_HANDLER 1
-#define USE_STRUCT_RGN
-
-#include <AGL/agl.h>
-#include <Carbon/Carbon.h>
-#include <OpenGL/OpenGL.h>
-#include <OpenGL/glext.h>
-#include <OpenGL/glu.h>
-#include <list>
-#include <map>
-#include <memory>
-
-class VideoRenderAGL;
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-
-class VideoChannelAGL : public VideoRenderCallback {
- public:
-
-  VideoChannelAGL(AGLContext& aglContext, int iId, VideoRenderAGL* owner);
-  virtual ~VideoChannelAGL();
-  virtual int FrameSizeChange(int width, int height, int numberOfStreams);
-  virtual int DeliverFrame(const VideoFrame& videoFrame);
-  virtual int UpdateSize(int width, int height);
-  int SetStreamSettings(int streamId, float startWidth, float startHeight,
-                        float stopWidth, float stopHeight);
-  int SetStreamCropSettings(int streamId, float startWidth, float startHeight,
-                            float stopWidth, float stopHeight);
-  int RenderOffScreenBuffer();
-  int IsUpdated(bool& isUpdated);
-  virtual int UpdateStretchSize(int stretchHeight, int stretchWidth);
-  virtual int32_t RenderFrame(const uint32_t streamId, VideoFrame& videoFrame);
-
- private:
-
-  AGLContext _aglContext;
-  int _id;
-  VideoRenderAGL* _owner;
-  int _width;
-  int _height;
-  int _stretchedWidth;
-  int _stretchedHeight;
-  float _startHeight;
-  float _startWidth;
-  float _stopWidth;
-  float _stopHeight;
-  int _xOldWidth;
-  int _yOldHeight;
-  int _oldStretchedHeight;
-  int _oldStretchedWidth;
-  unsigned char* _buffer;
-  size_t _bufferSize;
-  size_t _incomingBufferSize;
-  bool _bufferIsUpdated;
-  bool _sizeInitialized;
-  int _numberOfStreams;
-  bool _bVideoSizeStartedChanging;
-  GLenum _pixelFormat;
-  GLenum _pixelDataType;
-  unsigned int _texture;
-};
-
-class VideoRenderAGL {
- public:
-  VideoRenderAGL(WindowRef windowRef, bool fullscreen, int iId);
-  VideoRenderAGL(HIViewRef windowRef, bool fullscreen, int iId);
-  ~VideoRenderAGL();
-
-  int Init();
-  VideoChannelAGL* CreateAGLChannel(int channel, int zOrder, float startWidth,
-                                    float startHeight, float stopWidth,
-                                    float stopHeight);
-  VideoChannelAGL* ConfigureAGLChannel(int channel, int zOrder,
-                                       float startWidth, float startHeight,
-                                       float stopWidth, float stopHeight);
-  int DeleteAGLChannel(int channel);
-  int DeleteAllAGLChannels();
-  int StopThread();
-  bool IsFullScreen();
-  bool HasChannels();
-  bool HasChannel(int channel);
-  int GetChannels(std::list<int>& channelList);
-  void LockAGLCntx();
-  void UnlockAGLCntx();
-
-  static int GetOpenGLVersion(int& aglMajor, int& aglMinor);
-
-  // ********** new module functions ************ //
-  int ChangeWindow(void* newWindowRef);
-  int32_t StartRender();
-  int32_t StopRender();
-  int32_t DeleteAGLChannel(const uint32_t streamID);
-  int32_t GetChannelProperties(const uint16_t streamId, uint32_t& zOrder,
-                               float& left, float& top, float& right,
-                               float& bottom);
-
- protected:
-  static bool ScreenUpdateThreadProc(void* obj);
-  bool ScreenUpdateProcess();
-  int GetWindowRect(Rect& rect);
-
- private:
-  int CreateMixingContext();
-  int RenderOffScreenBuffers();
-  int SwapAndDisplayBuffers();
-  int UpdateClipping();
-  int CalculateVisibleRegion(ControlRef control, RgnHandle& visibleRgn,
-                             bool clipChildren);
-  bool CheckValidRegion(RgnHandle rHandle);
-  void ParentWindowResized(WindowRef window);
-
-  // Carbon GUI event handlers
-  static pascal OSStatus sHandleWindowResized(
-      EventHandlerCallRef nextHandler, EventRef theEvent, void* userData);
-  static pascal OSStatus sHandleHiViewResized(
-      EventHandlerCallRef nextHandler, EventRef theEvent, void* userData);
-
-  HIViewRef _hiviewRef;
-  WindowRef _windowRef;
-  bool _fullScreen;
-  int _id;
-  webrtc::CriticalSectionWrapper& _renderCritSec;
-  // TODO(pbos): Remove unique_ptr and use PlatformThread directly.
-  std::unique_ptr<rtc::PlatformThread> _screenUpdateThread;
-  webrtc::EventWrapper* _screenUpdateEvent;
-  bool _isHIViewRef;
-  AGLContext _aglContext;
-  int _windowWidth;
-  int _windowHeight;
-  int _lastWindowWidth;
-  int _lastWindowHeight;
-  int _lastHiViewWidth;
-  int _lastHiViewHeight;
-  int _currentParentWindowHeight;
-  int _currentParentWindowWidth;
-  Rect _currentParentWindowBounds;
-  bool _windowHasResized;
-  Rect _lastParentWindowBounds;
-  Rect _currentHIViewBounds;
-  Rect _lastHIViewBounds;
-  Rect _windowRect;
-  std::map<int, VideoChannelAGL*> _aglChannels;
-  std::multimap<int, int> _zOrderToChannel;
-  EventHandlerRef _hiviewEventHandlerRef;
-  EventHandlerRef _windowEventHandlerRef;
-  HIRect _currentViewBounds;
-  HIRect _lastViewBounds;
-  bool _renderingIsPaused;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
-
-#endif  // CARBON_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.cc b/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.cc
deleted file mode 100644
index f85be5fb5e2..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.cc
+++ /dev/null
@@ -1,280 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-#if defined(CARBON_RENDERING)
-
-#include <AGL/agl.h>
-#include "webrtc/modules/video_render/mac/video_render_agl.h"
-#include "webrtc/modules/video_render/mac/video_render_mac_carbon_impl.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-VideoRenderMacCarbonImpl::VideoRenderMacCarbonImpl(const int32_t id,
-        const VideoRenderType videoRenderType,
-        void* window,
-        const bool fullscreen) :
-_id(id),
-_renderMacCarbonCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
-_fullScreen(fullscreen),
-_ptrWindow(window)
-{
-
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
-
-}
-
-VideoRenderMacCarbonImpl::~VideoRenderMacCarbonImpl()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Destructor %s:%d", __FUNCTION__, __LINE__);
-    delete &_renderMacCarbonCritsect;
-}
-
-int32_t
-VideoRenderMacCarbonImpl::Init()
-{
-    CriticalSectionScoped cs(&_renderMacCarbonCritsect);
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
-
-    if (!_ptrWindow)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
-        return -1;
-    }
-
-    // We don't know if the user passed us a WindowRef or a HIViewRef, so test.
-    bool referenceIsValid = false;
-
-    // Check if it's a valid WindowRef
-    //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d _ptrWindowRef before WindowRef cast: %x", __FUNCTION__, __LINE__, _ptrWindowRef);
-    WindowRef* windowRef = static_cast<WindowRef*>(_ptrWindow);
-    //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d _ptrWindowRef after cast: %x", __FUNCTION__, __LINE__, _ptrWindowRef);
-    if (IsValidWindowPtr(*windowRef))
-    {
-        _ptrCarbonRender = new VideoRenderAGL(*windowRef, _fullScreen, _id);
-        referenceIsValid = true;
-        WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Successfully initialized CarbonRenderer with WindowRef:%x", __FUNCTION__, __LINE__, *windowRef);
-    }
-    else
-    {
-        HIViewRef* hiviewRef = static_cast<HIViewRef*>(_ptrWindow);
-        if (HIViewIsValid(*hiviewRef))
-        {
-            _ptrCarbonRender = new VideoRenderAGL(*hiviewRef, _fullScreen, _id);
-            referenceIsValid = true;
-            WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Successfully initialized CarbonRenderer with HIViewRef:%x", __FUNCTION__, __LINE__, hiviewRef);
-        }
-    }
-
-    if(!referenceIsValid)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Invalid WindowRef/HIViewRef Returning -1", __FUNCTION__, __LINE__);
-        return -1;
-    }
-
-    if(!_ptrCarbonRender)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to create an instance of VideoRenderAGL. Returning -1", __FUNCTION__, __LINE__);
-    }
-
-    int retVal = _ptrCarbonRender->Init();
-    if (retVal == -1)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to init CarbonRenderer", __FUNCTION__, __LINE__);
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t
-VideoRenderMacCarbonImpl::ChangeWindow(void* window)
-{
-    return -1;
-    CriticalSectionScoped cs(&_renderMacCarbonCritsect);
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s changing ID to ", __FUNCTION__, window);
-
-    if (window == NULL)
-    {
-        return -1;
-    }
-    _ptrWindow = window;
-
-
-    _ptrWindow = window;
-
-    return 0;
-}
-
-VideoRenderCallback*
-VideoRenderMacCarbonImpl::AddIncomingRenderStream(const uint32_t streamId,
-        const uint32_t zOrder,
-        const float left,
-        const float top,
-        const float right,
-        const float bottom)
-{
-
-    CriticalSectionScoped cs(&_renderMacCarbonCritsect);
-    WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
-    VideoChannelAGL* AGLChannel = NULL;
-
-    if(!_ptrWindow)
-    {
-    }
-
-    if(!AGLChannel)
-    {
-        AGLChannel = _ptrCocoaRender->CreateNSGLChannel(streamId, zOrder, left, top, right, bottom);
-    }
-
-    return AGLChannel;
-
-}
-
-int32_t
-VideoRenderMacCarbonImpl::DeleteIncomingRenderStream(const uint32_t streamId)
-{
-
-    WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
-    CriticalSectionScoped cs(&_renderMacCarbonCritsect);
-    _ptrCarbonRender->DeleteAGLChannel(streamId);
-
-    return 0;
-}
-
-int32_t
-VideoRenderMacCarbonImpl::GetIncomingRenderStreamProperties(const uint32_t streamId,
-        uint32_t& zOrder,
-        float& left,
-        float& top,
-        float& right,
-        float& bottom) const
-{
-    return -1;
-    return _ptrCarbonRender->GetChannelProperties(streamId, zOrder, left, top, right, bottom);
-}
-
-int32_t
-VideoRenderMacCarbonImpl::StartRender()
-{
-    return _ptrCarbonRender->StartRender();
-}
-
-int32_t
-VideoRenderMacCarbonImpl::StopRender()
-{
-    return _ptrCarbonRender->StopRender();
-}
-
-VideoRenderType
-VideoRenderMacCarbonImpl::RenderType()
-{
-    return kRenderCarbon;
-}
-
-RawVideoType
-VideoRenderMacCarbonImpl::PerferedVideoType()
-{
-    return kVideoI420;
-}
-
-bool
-VideoRenderMacCarbonImpl::FullScreen()
-{
-    return false;
-}
-
-int32_t
-VideoRenderMacCarbonImpl::GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-        uint64_t& availableGraphicsMemory) const
-{
-    totalGraphicsMemory = 0;
-    availableGraphicsMemory = 0;
-    return 0;
-}
-
-int32_t
-VideoRenderMacCarbonImpl::GetScreenResolution(uint32_t& screenWidth,
-        uint32_t& screenHeight) const
-{
-    CriticalSectionScoped cs(&_renderMacCarbonCritsect);
-    //NSScreen* mainScreen = [NSScreen mainScreen];
-
-    //NSRect frame = [mainScreen frame];
-
-    //screenWidth = frame.size.width;
-    //screenHeight = frame.size.height;
-    return 0;
-}
-
-uint32_t
-VideoRenderMacCarbonImpl::RenderFrameRate(const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_renderMacCarbonCritsect);
-    return 0;
-}
-
-int32_t
-VideoRenderMacCarbonImpl::SetStreamCropping(const uint32_t streamId,
-        const float left,
-        const float top,
-        const float right,
-        const float bottom)
-{
-    return 0;
-}
-
-int32_t VideoRenderMacCarbonImpl::ConfigureRenderer(const uint32_t streamId,
-                                                    const unsigned int zOrder,
-                                                    const float left,
-                                                    const float top,
-                                                    const float right,
-                                                    const float bottom)
-{
-    return 0;
-}
-
-int32_t
-VideoRenderMacCarbonImpl::SetTransparentBackground(const bool enable)
-{
-    return 0;
-}
-
-int32_t VideoRenderMacCarbonImpl::SetText(const uint8_t textId,
-                                          const uint8_t* text,
-                                          const int32_t textLength,
-                                          const uint32_t textColorRef,
-                                          const uint32_t backgroundColorRef,
-                                          const float left,
-                                          const float top,
-                                          const float right,
-                                          const float bottom)
-{
-    return 0;
-}
-
-int32_t VideoRenderMacCarbonImpl::SetBitmap(const void* bitMap,
-                                            const uint8_t pictureId,
-                                            const void* colorKey,
-                                            const float left,
-                                            const float top,
-                                            const float right,
-                                            const float bottom)
-{
-    return 0;
-}
-
-
-}  // namespace webrtc
-
-#endif // CARBON_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.h b/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.h
deleted file mode 100644
index 9ad3a6cdd18..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_carbon_impl.h
+++ /dev/null
@@ -1,146 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-#if defined(CARBON_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
-
-#include "webrtc/modules/video_render/i_video_render.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class VideoRenderAGL;
-
-// Class definitions
-class VideoRenderMacCarbonImpl : IVideoRender
-{
-public:
-    /*
-     *   Constructor/destructor
-     */
-
-    VideoRenderMacCarbonImpl(const int32_t id,
-            const VideoRenderType videoRenderType,
-            void* window,
-            const bool fullscreen);
-
-    virtual ~VideoRenderMacCarbonImpl();
-
-    virtual int32_t Init();
-
-    virtual int32_t ChangeWindow(void* window);
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-    virtual VideoRenderCallback* AddIncomingRenderStream(const uint32_t streamId,
-            const uint32_t zOrder,
-            const float left,
-            const float top,
-            const float right,
-            const float bottom);
-
-    virtual int32_t DeleteIncomingRenderStream(const uint32_t streamId);
-
-    virtual int32_t GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                                      uint32_t& zOrder,
-                                                      float& left,
-                                                      float& top,
-                                                      float& right,
-                                                      float& bottom) const;
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender();
-
-    virtual int32_t StopRender();
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderType RenderType();
-
-    virtual RawVideoType PerferedVideoType();
-
-    virtual bool FullScreen();
-
-    virtual int32_t GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-            uint64_t& availableGraphicsMemory) const;
-
-    virtual int32_t GetScreenResolution(uint32_t& screenWidth,
-            uint32_t& screenHeight) const;
-
-    virtual uint32_t RenderFrameRate(const uint32_t streamId);
-
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-            const float left,
-            const float top,
-            const float right,
-            const float bottom);
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-            const unsigned int zOrder,
-            const float left,
-            const float top,
-            const float right,
-            const float bottom);
-
-    virtual int32_t SetTransparentBackground(const bool enable);
-
-    virtual int32_t SetText(const uint8_t textId,
-            const uint8_t* text,
-            const int32_t textLength,
-            const uint32_t textColorRef,
-            const uint32_t backgroundColorRef,
-            const float left,
-            const float top,
-            const float right,
-            const float bottom);
-
-    virtual int32_t SetBitmap(const void* bitMap,
-            const uint8_t pictureId,
-            const void* colorKey,
-            const float left,
-            const float top,
-            const float right,
-            const float bottom);
-
-    virtual int32_t FullScreenRender(void* window, const bool enable)
-    {
-        // not supported in Carbon at this time
-        return -1;
-    }
-
-private:
-    int32_t _id;
-    CriticalSectionWrapper& _renderMacCarbonCritsect;
-    bool _fullScreen;
-    void* _ptrWindow;
-    VideoRenderAGL* _ptrCarbonRender;
-
-};
-
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
-#endif // CARBON_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.h b/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.h
deleted file mode 100644
index 21add272bbd..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.h
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-
-#if defined(COCOA_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
-
-#include "webrtc/modules/video_render/i_video_render.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class VideoRenderNSOpenGL;
-
-// Class definitions
-class VideoRenderMacCocoaImpl : IVideoRender
-{
-public:
-    /*
-     *   Constructor/destructor
-     */
-
-    VideoRenderMacCocoaImpl(const int32_t id,
-            const VideoRenderType videoRenderType,
-            void* window,
-            const bool fullscreen);
-
-    virtual ~VideoRenderMacCocoaImpl();
-
-    virtual int32_t Init();
-
-    virtual int32_t ChangeWindow(void* window);
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-    virtual VideoRenderCallback* AddIncomingRenderStream(const uint32_t streamId,
-            const uint32_t zOrder,
-            const float left,
-            const float top,
-            const float right,
-            const float bottom);
-
-    virtual int32_t DeleteIncomingRenderStream(const uint32_t streamId);
-
-    virtual int32_t GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                                      uint32_t& zOrder,
-                                                      float& left,
-                                                      float& top,
-                                                      float& right,
-                                                      float& bottom) const;
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender();
-
-    virtual int32_t StopRender();
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderType RenderType();
-
-    virtual RawVideoType PerferedVideoType();
-
-    virtual bool FullScreen();
-
-    virtual int32_t GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-                                      uint64_t& availableGraphicsMemory) const;
-
-    virtual int32_t GetScreenResolution(uint32_t& screenWidth,
-                                        uint32_t& screenHeight) const;
-
-    virtual uint32_t RenderFrameRate(const uint32_t streamId);
-
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom);
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom);
-
-    virtual int32_t SetTransparentBackground(const bool enable);
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t textColorRef,
-                            const uint32_t backgroundColorRef,
-                            const float left,
-                            const float top,
-                            const float right,
-                            const float bottom);
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left,
-                              const float top,
-                              const float right,
-                              const float bottom);
-
-    virtual int32_t FullScreenRender(void* window, const bool enable);
-
-private:
-    int32_t _id;
-    CriticalSectionWrapper& _renderMacCocoaCritsect;
-    bool _fullScreen;
-    void* _ptrWindow;
-    VideoRenderNSOpenGL* _ptrCocoaRender;
-
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
-#endif	// COCOA_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.mm b/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.mm
deleted file mode 100644
index 5b017fecc0c..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.mm
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-#if defined(COCOA_RENDERING)
-
-#include "webrtc/modules/video_render/mac/cocoa_render_view.h"
-#include "webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.h"
-#include "webrtc/modules/video_render/mac/video_render_nsopengl.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-VideoRenderMacCocoaImpl::VideoRenderMacCocoaImpl(const int32_t id,
-        const VideoRenderType videoRenderType,
-        void* window,
-        const bool fullscreen) :
-_id(id),
-_renderMacCocoaCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
-_fullScreen(fullscreen),
-_ptrWindow(window)
-{
-
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
-}
-
-VideoRenderMacCocoaImpl::~VideoRenderMacCocoaImpl()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Destructor %s:%d", __FUNCTION__, __LINE__);
-    delete &_renderMacCocoaCritsect;
-    if (_ptrCocoaRender)
-    {
-        delete _ptrCocoaRender;
-        _ptrCocoaRender = NULL;
-    }
-}
-
-int32_t
-VideoRenderMacCocoaImpl::Init()
-{
-
-    CriticalSectionScoped cs(&_renderMacCocoaCritsect);
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
-
-    // cast ptrWindow from void* to CocoaRenderer. Void* was once NSOpenGLView, and CocoaRenderer is NSOpenGLView.
-    _ptrCocoaRender = new VideoRenderNSOpenGL((CocoaRenderView*)_ptrWindow, _fullScreen, _id);
-    if (!_ptrWindow)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
-        return -1;
-    }
-    int retVal = _ptrCocoaRender->Init();
-    if (retVal == -1)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Failed to init %s:%d", __FUNCTION__, __LINE__);
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t
-VideoRenderMacCocoaImpl::ChangeWindow(void* window)
-{
-
-    CriticalSectionScoped cs(&_renderMacCocoaCritsect);
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s changing ID to ", __FUNCTION__, window);
-
-    if (window == NULL)
-    {
-        return -1;
-    }
-    _ptrWindow = window;
-
-
-    _ptrWindow = window;
-    _ptrCocoaRender->ChangeWindow((CocoaRenderView*)_ptrWindow);
-
-    return 0;
-}
-
-VideoRenderCallback*
-VideoRenderMacCocoaImpl::AddIncomingRenderStream(const uint32_t streamId,
-        const uint32_t zOrder,
-        const float left,
-        const float top,
-        const float right,
-        const float bottom)
-{
-    CriticalSectionScoped cs(&_renderMacCocoaCritsect);
-    WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
-    VideoChannelNSOpenGL* nsOpenGLChannel = NULL;
-
-    if(!_ptrWindow)
-    {
-    }
-
-    if(!nsOpenGLChannel)
-    {
-        nsOpenGLChannel = _ptrCocoaRender->CreateNSGLChannel(streamId, zOrder, left, top, right, bottom);
-    }
-
-    return nsOpenGLChannel;
-
-}
-
-int32_t
-VideoRenderMacCocoaImpl::DeleteIncomingRenderStream(const uint32_t streamId)
-{
-    WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
-    CriticalSectionScoped cs(&_renderMacCocoaCritsect);
-    _ptrCocoaRender->DeleteNSGLChannel(streamId);
-
-    return 0;
-}
-
-int32_t
-VideoRenderMacCocoaImpl::GetIncomingRenderStreamProperties(const uint32_t streamId,
-        uint32_t& zOrder,
-        float& left,
-        float& top,
-        float& right,
-        float& bottom) const
-{
-    return _ptrCocoaRender->GetChannelProperties(streamId, zOrder, left, top, right, bottom);
-}
-
-int32_t
-VideoRenderMacCocoaImpl::StartRender()
-{
-    return _ptrCocoaRender->StartRender();
-}
-
-int32_t
-VideoRenderMacCocoaImpl::StopRender()
-{
-    return _ptrCocoaRender->StopRender();
-}
-
-VideoRenderType
-VideoRenderMacCocoaImpl::RenderType()
-{
-    return kRenderCocoa;
-}
-
-RawVideoType
-VideoRenderMacCocoaImpl::PerferedVideoType()
-{
-    return kVideoI420;
-}
-
-bool
-VideoRenderMacCocoaImpl::FullScreen()
-{
-    return false;
-}
-
-int32_t
-VideoRenderMacCocoaImpl::GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-        uint64_t& availableGraphicsMemory) const
-{
-    totalGraphicsMemory = 0;
-    availableGraphicsMemory = 0;
-    return 0;
-}
-
-int32_t
-VideoRenderMacCocoaImpl::GetScreenResolution(uint32_t& screenWidth,
-        uint32_t& screenHeight) const
-{
-    CriticalSectionScoped cs(&_renderMacCocoaCritsect);
-    NSScreen* mainScreen = [NSScreen mainScreen];
-
-    NSRect frame = [mainScreen frame];
-
-    screenWidth = frame.size.width;
-    screenHeight = frame.size.height;
-    return 0;
-}
-
-uint32_t
-VideoRenderMacCocoaImpl::RenderFrameRate(const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_renderMacCocoaCritsect);
-    return 0;
-}
-
-int32_t
-VideoRenderMacCocoaImpl::SetStreamCropping(const uint32_t streamId,
-        const float left,
-        const float top,
-        const float right,
-        const float bottom)
-{
-    return 0;
-}
-
-int32_t VideoRenderMacCocoaImpl::ConfigureRenderer(const uint32_t streamId,
-                                                   const unsigned int zOrder,
-                                                   const float left,
-                                                   const float top,
-                                                   const float right,
-                                                   const float bottom)
-{
-    return 0;
-}
-
-int32_t
-VideoRenderMacCocoaImpl::SetTransparentBackground(const bool enable)
-{
-    return 0;
-}
-
-int32_t VideoRenderMacCocoaImpl::SetText(const uint8_t textId,
-                                         const uint8_t* text,
-                                         const int32_t textLength,
-                                         const uint32_t textColorRef,
-                                         const uint32_t backgroundColorRef,
-                                         const float left,
-                                         const float top,
-                                         const float right,
-                                         const float bottom)
-{
-    return _ptrCocoaRender->SetText(textId, text, textLength, textColorRef, backgroundColorRef, left, top, right, bottom);
-}
-
-int32_t VideoRenderMacCocoaImpl::SetBitmap(const void* bitMap,
-                                           const uint8_t pictureId,
-                                           const void* colorKey,
-                                           const float left,
-                                           const float top,
-                                           const float right,
-                                           const float bottom)
-{
-    return 0;
-}
-
-int32_t VideoRenderMacCocoaImpl::FullScreenRender(void* window, const bool enable)
-{
-    return -1;
-}
-
-}  // namespace webrtc
-
-#endif // COCOA_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_nsopengl.h b/chromium/third_party/webrtc/modules/video_render/mac/video_render_nsopengl.h
deleted file mode 100644
index 457557dad64..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_nsopengl.h
+++ /dev/null
@@ -1,192 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-#if defined(COCOA_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
-
-#import <Cocoa/Cocoa.h>
-#import <OpenGL/OpenGL.h>
-#import <OpenGL/glext.h>
-#import <OpenGL/glu.h>
-#include <QuickTime/QuickTime.h>
-#include <list>
-#include <map>
-#include <memory>
-
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-#import "webrtc/modules/video_render/mac/cocoa_full_screen_window.h"
-#import "webrtc/modules/video_render/mac/cocoa_render_view.h"
-
-class Trace;
-
-namespace rtc {
-class PlatformThread;
-}  // namespace rtc
-
-namespace webrtc {
-class EventTimerWrapper;
-class VideoRenderNSOpenGL;
-class CriticalSectionWrapper;
-
-class VideoChannelNSOpenGL : public VideoRenderCallback {
-public:
-    VideoChannelNSOpenGL(NSOpenGLContext *nsglContext, int iId, VideoRenderNSOpenGL* owner);
-    virtual ~VideoChannelNSOpenGL();
-
-    // A new frame is delivered
-    virtual int DeliverFrame(const VideoFrame& videoFrame);
-
-    // Called when the incoming frame size and/or number of streams in mix
-    // changes.
-    virtual int FrameSizeChange(int width, int height, int numberOfStreams);
-
-    virtual int UpdateSize(int width, int height);
-
-    // Setup
-    int SetStreamSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
-    int SetStreamCropSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
-
-    // Called when it's time to render the last frame for the channel
-    int RenderOffScreenBuffer();
-
-    // Returns true if a new buffer has been delivered to the texture
-    int IsUpdated(bool& isUpdated);
-    virtual int UpdateStretchSize(int stretchHeight, int stretchWidth);
-
-    // ********** new module functions ************ //
-    virtual int32_t RenderFrame(const uint32_t streamId,
-                                const VideoFrame& videoFrame);
-
-    // ********** new module helper functions ***** //
-    int ChangeContext(NSOpenGLContext *nsglContext);
-    int32_t GetChannelProperties(float& left,
-                                 float& top,
-                                 float& right,
-                                 float& bottom);
-
-private:
-
-    NSOpenGLContext* _nsglContext;
-    const int _id;
-    VideoRenderNSOpenGL* _owner;
-    int32_t _width;
-    int32_t _height;
-    float _startWidth;
-    float _startHeight;
-    float _stopWidth;
-    float _stopHeight;
-    int _stretchedWidth;
-    int _stretchedHeight;
-    int _oldStretchedHeight;
-    int _oldStretchedWidth;
-    unsigned char* _buffer;
-    size_t _bufferSize;
-    size_t _incomingBufferSize;
-    bool _bufferIsUpdated;
-    int _numberOfStreams;
-    GLenum _pixelFormat;
-    GLenum _pixelDataType;
-    unsigned int _texture;
-};
-
-class VideoRenderNSOpenGL
-{
-
-public: // methods
-    VideoRenderNSOpenGL(CocoaRenderView *windowRef, bool fullScreen, int iId);
-    ~VideoRenderNSOpenGL();
-
-    static int GetOpenGLVersion(int& nsglMajor, int& nsglMinor);
-
-    // Allocates textures
-    int Init();
-    VideoChannelNSOpenGL* CreateNSGLChannel(int streamID, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
-    VideoChannelNSOpenGL* ConfigureNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
-    int DeleteNSGLChannel(int channel);
-    int DeleteAllNSGLChannels();
-    int StopThread();
-    bool IsFullScreen();
-    bool HasChannels();
-    bool HasChannel(int channel);
-    int GetChannels(std::list<int>& channelList);
-    void LockAGLCntx() EXCLUSIVE_LOCK_FUNCTION(_nsglContextCritSec);
-    void UnlockAGLCntx() UNLOCK_FUNCTION(_nsglContextCritSec);
-
-    // ********** new module functions ************ //
-    int ChangeWindow(CocoaRenderView* newWindowRef);
-    int32_t StartRender();
-    int32_t StopRender();
-    int32_t DeleteNSGLChannel(const uint32_t streamID);
-    int32_t GetChannelProperties(const uint16_t streamId,
-                                 uint32_t& zOrder,
-                                 float& left,
-                                 float& top,
-                                 float& right,
-                                 float& bottom);
-
-    int32_t SetText(const uint8_t textId,
-                    const uint8_t* text,
-                    const int32_t textLength,
-                    const uint32_t textColorRef,
-                    const uint32_t backgroundColorRef,
-                    const float left,
-                    const float top,
-                    const float right,
-                    const float bottom);
-
-    // ********** new module helper functions ***** //
-    int configureNSOpenGLEngine();
-    int configureNSOpenGLView();
-    int setRenderTargetWindow();
-    int setRenderTargetFullScreen();
-
-protected: // methods
-    static bool ScreenUpdateThreadProc(void* obj);
-    bool ScreenUpdateProcess();
-    int GetWindowRect(Rect& rect);
-
-private: // methods
-
-    int CreateMixingContext();
-    int RenderOffScreenBuffers();
-    int DisplayBuffers();
-
-private: // variables
-
-
-    CocoaRenderView* _windowRef;
-    bool _fullScreen;
-    int _id;
-    CriticalSectionWrapper& _nsglContextCritSec;
-    // TODO(pbos): Remove unique_ptr and use PlatformThread directly.
-    std::unique_ptr<rtc::PlatformThread> _screenUpdateThread;
-    EventTimerWrapper* _screenUpdateEvent;
-    NSOpenGLContext* _nsglContext;
-    NSOpenGLContext* _nsglFullScreenContext;
-    CocoaFullScreenWindow* _fullScreenWindow;
-    Rect _windowRect; // The size of the window
-    int _windowWidth;
-    int _windowHeight;
-    std::map<int, VideoChannelNSOpenGL*> _nsglChannels;
-    std::multimap<int, int> _zOrderToChannel;
-    bool _renderingIsPaused;
-    NSView* _windowRefSuperView;
-    NSRect _windowRefSuperViewFrame;
-};
-
-}  // namespace webrtc
-
-#endif   // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
-#endif	 // COCOA_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/mac/video_render_nsopengl.mm b/chromium/third_party/webrtc/modules/video_render/mac/video_render_nsopengl.mm
deleted file mode 100644
index b7683a96af4..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/mac/video_render_nsopengl.mm
+++ /dev/null
@@ -1,1247 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-#if defined(COCOA_RENDERING)
-
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/mac/video_render_nsopengl.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-VideoChannelNSOpenGL::VideoChannelNSOpenGL(NSOpenGLContext *nsglContext, int iId, VideoRenderNSOpenGL* owner) :
-_nsglContext( nsglContext),
-_id( iId),
-_owner( owner),
-_width( 0),
-_height( 0),
-_startWidth( 0.0f),
-_startHeight( 0.0f),
-_stopWidth( 0.0f),
-_stopHeight( 0.0f),
-_stretchedWidth( 0),
-_stretchedHeight( 0),
-_oldStretchedHeight( 0),
-_oldStretchedWidth( 0),
-_buffer( 0),
-_bufferSize( 0),
-_incomingBufferSize( 0),
-_bufferIsUpdated( false),
-_numberOfStreams( 0),
-_pixelFormat( GL_RGBA),
-_pixelDataType( GL_UNSIGNED_INT_8_8_8_8),
-_texture( 0)
-{
-
-}
-
-VideoChannelNSOpenGL::~VideoChannelNSOpenGL()
-{
-    if (_buffer)
-    {
-        delete [] _buffer;
-        _buffer = NULL;
-    }
-
-    if (_texture != 0)
-    {
-        [_nsglContext makeCurrentContext];
-        glDeleteTextures(1, (const GLuint*) &_texture);
-        _texture = 0;
-    }
-}
-
-int VideoChannelNSOpenGL::ChangeContext(NSOpenGLContext *nsglContext)
-{
-    _owner->LockAGLCntx();
-
-    _nsglContext = nsglContext;
-    [_nsglContext makeCurrentContext];
-
-    _owner->UnlockAGLCntx();
-    return 0;
-
-}
-
-int32_t VideoChannelNSOpenGL::GetChannelProperties(float& left, float& top,
-                                                   float& right, float& bottom)
-{
-
-    _owner->LockAGLCntx();
-
-    left = _startWidth;
-    top = _startHeight;
-    right = _stopWidth;
-    bottom = _stopHeight;
-
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int32_t VideoChannelNSOpenGL::RenderFrame(const uint32_t /*streamId*/,
-                                          const VideoFrame& videoFrame) {
-  _owner->LockAGLCntx();
-
-  if(_width != videoFrame.width() ||
-     _height != videoFrame.height()) {
-      if(FrameSizeChange(videoFrame.width(), videoFrame.height(), 1) == -1) {
-        _owner->UnlockAGLCntx();
-        return -1;
-      }
-  }
-  int ret = DeliverFrame(videoFrame);
-
-  _owner->UnlockAGLCntx();
-  return ret;
-}
-
-int VideoChannelNSOpenGL::UpdateSize(int width, int height)
-{
-    _owner->LockAGLCntx();
-    _width = width;
-    _height = height;
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelNSOpenGL::UpdateStretchSize(int stretchHeight, int stretchWidth)
-{
-
-    _owner->LockAGLCntx();
-    _stretchedHeight = stretchHeight;
-    _stretchedWidth = stretchWidth;
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelNSOpenGL::FrameSizeChange(int width, int height, int numberOfStreams)
-{
-    //  We got a new frame size from VideoAPI, prepare the buffer
-
-    _owner->LockAGLCntx();
-
-    if (width == _width && _height == height)
-    {
-        // We already have a correct buffer size
-        _numberOfStreams = numberOfStreams;
-        _owner->UnlockAGLCntx();
-        return 0;
-    }
-
-    _width = width;
-    _height = height;
-
-    // Delete the old buffer, create a new one with correct size.
-    if (_buffer)
-    {
-        delete [] _buffer;
-        _bufferSize = 0;
-    }
-
-    _incomingBufferSize = CalcBufferSize(kI420, _width, _height);
-    _bufferSize = CalcBufferSize(kARGB, _width, _height);
-    _buffer = new unsigned char [_bufferSize];
-    memset(_buffer, 0, _bufferSize * sizeof(unsigned char));
-
-    [_nsglContext makeCurrentContext];
-
-    if(glIsTexture(_texture))
-    {
-        glDeleteTextures(1, (const GLuint*) &_texture);
-        _texture = 0;
-    }
-
-    // Create a new texture
-    glGenTextures(1, (GLuint *) &_texture);
-
-    GLenum glErr = glGetError();
-
-    if (glErr != GL_NO_ERROR)
-    {
-
-    }
-
-    glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-
-    GLint texSize;
-    glGetIntegerv(GL_MAX_TEXTURE_SIZE, &texSize);
-
-    if (texSize < _width || texSize < _height)
-    {
-        _owner->UnlockAGLCntx();
-        return -1;
-    }
-
-    // Set up th texture type and size
-    glTexImage2D(GL_TEXTURE_RECTANGLE_EXT, // target
-            0, // level
-            GL_RGBA, // internal format
-            _width, // width
-            _height, // height
-            0, // border 0/1 = off/on
-            _pixelFormat, // format, GL_RGBA
-            _pixelDataType, // data type, GL_UNSIGNED_INT_8_8_8_8
-            _buffer); // pixel data
-
-    glErr = glGetError();
-    if (glErr != GL_NO_ERROR)
-    {
-        _owner->UnlockAGLCntx();
-        return -1;
-    }
-
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelNSOpenGL::DeliverFrame(const VideoFrame& videoFrame) {
-  _owner->LockAGLCntx();
-
-  if (_texture == 0) {
-    _owner->UnlockAGLCntx();
-    return 0;
-  }
-
-  if (CalcBufferSize(kI420, videoFrame.width(), videoFrame.height()) !=
-      _incomingBufferSize) {
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  // Using the VideoFrame for YV12: YV12 is YVU; I420 assumes
-  // YUV.
-  // TODO(mikhal) : Use appropriate functionality.
-  // TODO(wu): See if we are using glTexSubImage2D correctly.
-  int rgbRet = ConvertFromYV12(videoFrame, kBGRA, 0, _buffer);
-  if (rgbRet < 0) {
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  [_nsglContext makeCurrentContext];
-
-  // Make sure this texture is the active one
-  glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-  GLenum glErr = glGetError();
-  if (glErr != GL_NO_ERROR) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-    "ERROR %d while calling glBindTexture", glErr);
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  glTexSubImage2D(GL_TEXTURE_RECTANGLE_EXT,
-                  0, // Level, not use
-                  0, // start point x, (low left of pic)
-                  0, // start point y,
-                  _width, // width
-                  _height, // height
-                  _pixelFormat, // pictue format for _buffer
-                  _pixelDataType, // data type of _buffer
-                  (const GLvoid*) _buffer); // the pixel data
-
-  glErr = glGetError();
-  if (glErr != GL_NO_ERROR) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-    "ERROR %d while calling glTexSubImage2d", glErr);
-    _owner->UnlockAGLCntx();
-    return -1;
-  }
-
-  _bufferIsUpdated = true;
-
-  _owner->UnlockAGLCntx();
-  return 0;
-}
-
-int VideoChannelNSOpenGL::RenderOffScreenBuffer()
-{
-
-    _owner->LockAGLCntx();
-
-    if (_texture == 0)
-    {
-        _owner->UnlockAGLCntx();
-        return 0;
-    }
-
-    //	if(_fullscreen)
-    //	{
-    // NSRect mainDisplayRect = [[NSScreen mainScreen] frame];
-    //		_width = mainDisplayRect.size.width;
-    //		_height = mainDisplayRect.size.height;
-    //		glViewport(0, 0, mainDisplayRect.size.width, mainDisplayRect.size.height);
-    //		float newX = mainDisplayRect.size.width/_width;
-    //		float newY = mainDisplayRect.size.height/_height;
-
-    // convert from 0.0 <= size <= 1.0 to
-    // open gl world -1.0 < size < 1.0
-    GLfloat xStart = 2.0f * _startWidth - 1.0f;
-    GLfloat xStop = 2.0f * _stopWidth - 1.0f;
-    GLfloat yStart = 1.0f - 2.0f * _stopHeight;
-    GLfloat yStop = 1.0f - 2.0f * _startHeight;
-
-    [_nsglContext makeCurrentContext];
-
-    glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-    _oldStretchedHeight = _stretchedHeight;
-    _oldStretchedWidth = _stretchedWidth;
-
-    glLoadIdentity();
-    glEnable(GL_TEXTURE_RECTANGLE_EXT);
-    glBegin(GL_POLYGON);
-    {
-        glTexCoord2f(0.0, 0.0); glVertex2f(xStart, yStop);
-        glTexCoord2f(_width, 0.0); glVertex2f(xStop, yStop);
-        glTexCoord2f(_width, _height); glVertex2f(xStop, yStart);
-        glTexCoord2f(0.0, _height); glVertex2f(xStart, yStart);
-    }
-    glEnd();
-
-    glDisable(GL_TEXTURE_RECTANGLE_EXT);
-
-    _bufferIsUpdated = false;
-
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelNSOpenGL::IsUpdated(bool& isUpdated)
-{
-    _owner->LockAGLCntx();
-
-    isUpdated = _bufferIsUpdated;
-
-    _owner->UnlockAGLCntx();
-    return 0;
-}
-
-int VideoChannelNSOpenGL::SetStreamSettings(int /*streamId*/, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-    _owner->LockAGLCntx();
-
-    _startWidth = startWidth;
-    _stopWidth = stopWidth;
-    _startHeight = startHeight;
-    _stopHeight = stopHeight;
-
-    int oldWidth = _width;
-    int oldHeight = _height;
-    int oldNumberOfStreams = _numberOfStreams;
-
-    _width = 0;
-    _height = 0;
-
-    int retVal = FrameSizeChange(oldWidth, oldHeight, oldNumberOfStreams);
-
-    _owner->UnlockAGLCntx();
-    return retVal;
-}
-
-int VideoChannelNSOpenGL::SetStreamCropSettings(int /*streamId*/, float /*startWidth*/, float /*startHeight*/, float /*stopWidth*/, float /*stopHeight*/)
-{
-    return -1;
-}
-
-/*
- *
- *    VideoRenderNSOpenGL
- *
- */
-
-VideoRenderNSOpenGL::VideoRenderNSOpenGL(CocoaRenderView *windowRef, bool fullScreen, int iId) :
-_windowRef( (CocoaRenderView*)windowRef),
-_fullScreen( fullScreen),
-_id( iId),
-_nsglContextCritSec( *CriticalSectionWrapper::CreateCriticalSection()),
-_screenUpdateEvent(EventTimerWrapper::Create()),
-_nsglContext( 0),
-_nsglFullScreenContext( 0),
-_fullScreenWindow( nil),
-_windowRect( ),
-_windowWidth( 0),
-_windowHeight( 0),
-_nsglChannels( ),
-_zOrderToChannel( ),
-_renderingIsPaused (FALSE),
-_windowRefSuperView(NULL),
-_windowRefSuperViewFrame(NSMakeRect(0,0,0,0))
-{
-  _screenUpdateThread.reset(new rtc::PlatformThread(
-      ScreenUpdateThreadProc, this, "ScreenUpdateNSOpenGL"));
-}
-
-int VideoRenderNSOpenGL::ChangeWindow(CocoaRenderView* newWindowRef)
-{
-
-    LockAGLCntx();
-
-    _windowRef = newWindowRef;
-
-    if(CreateMixingContext() == -1)
-    {
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    int error = 0;
-    std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
-    while (it!= _nsglChannels.end())
-    {
-        error |= (it->second)->ChangeContext(_nsglContext);
-        it++;
-    }
-    if(error != 0)
-    {
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-/* Check if the thread and event already exist.
- * If so then they will simply be restarted
- * If not then create them and continue
- */
-int32_t VideoRenderNSOpenGL::StartRender()
-{
-
-    LockAGLCntx();
-
-    const unsigned int MONITOR_FREQ = 60;
-    if(TRUE == _renderingIsPaused)
-    {
-        WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Restarting screenUpdateThread");
-
-        // we already have the thread. Most likely StopRender() was called and they were paused
-        _screenUpdateThread->Start();
-        if (FALSE ==
-            _screenUpdateEvent->StartTimer(true, 1000 / MONITOR_FREQ)) {
-            WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "Failed to restart screenUpdateThread or screenUpdateEvent");
-            UnlockAGLCntx();
-            return -1;
-        }
-
-        _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
-
-        UnlockAGLCntx();
-        return 0;
-    }
-
-
-    if (!_screenUpdateThread)
-    {
-        WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "failed start screenUpdateThread");
-        UnlockAGLCntx();
-        return -1;
-    }
-
-
-    UnlockAGLCntx();
-    return 0;
-}
-int32_t VideoRenderNSOpenGL::StopRender()
-{
-
-    LockAGLCntx();
-
-    /* The code below is functional
-     * but it pauses for several seconds
-     */
-
-    // pause the update thread and the event timer
-    if(!_screenUpdateThread || !_screenUpdateEvent)
-    {
-        _renderingIsPaused = TRUE;
-
-        UnlockAGLCntx();
-        return 0;
-    }
-
-    _screenUpdateThread->Stop();
-    if (FALSE == _screenUpdateEvent->StopTimer()) {
-        _renderingIsPaused = FALSE;
-
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    _renderingIsPaused = TRUE;
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int VideoRenderNSOpenGL::configureNSOpenGLView()
-{
-    return 0;
-
-}
-
-int VideoRenderNSOpenGL::configureNSOpenGLEngine()
-{
-
-    LockAGLCntx();
-
-    // Disable not needed functionality to increase performance
-    glDisable(GL_DITHER);
-    glDisable(GL_ALPHA_TEST);
-    glDisable(GL_STENCIL_TEST);
-    glDisable(GL_FOG);
-    glDisable(GL_TEXTURE_2D);
-    glPixelZoom(1.0, 1.0);
-    glDisable(GL_BLEND);
-    glDisable(GL_DEPTH_TEST);
-    glDepthMask(GL_FALSE);
-    glDisable(GL_CULL_FACE);
-
-    // Set texture parameters
-    glTexParameterf(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_PRIORITY, 1.0);
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
-    glTexEnvf(GL_TEXTURE_ENV, GL_TEXTURE_ENV_MODE, GL_MODULATE);
-    glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
-    glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_STORAGE_HINT_APPLE, GL_STORAGE_SHARED_APPLE);
-
-    if (GetWindowRect(_windowRect) == -1)
-    {
-        UnlockAGLCntx();
-        return true;
-    }
-
-    if (_windowWidth != (_windowRect.right - _windowRect.left)
-            || _windowHeight != (_windowRect.bottom - _windowRect.top))
-    {
-        _windowWidth = _windowRect.right - _windowRect.left;
-        _windowHeight = _windowRect.bottom - _windowRect.top;
-    }
-    glViewport(0, 0, _windowWidth, _windowHeight);
-
-    // Synchronize buffer swaps with vertical refresh rate
-    GLint swapInt = 1;
-    [_nsglContext setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int VideoRenderNSOpenGL::setRenderTargetWindow()
-{
-    LockAGLCntx();
-
-
-    GLuint attribs[] =
-    {
-        NSOpenGLPFAColorSize, 24,
-        NSOpenGLPFAAlphaSize, 8,
-        NSOpenGLPFADepthSize, 16,
-        NSOpenGLPFAAccelerated,
-        0
-    };
-
-    NSOpenGLPixelFormat* fmt = [[[NSOpenGLPixelFormat alloc] initWithAttributes:
-                          (NSOpenGLPixelFormatAttribute*) attribs] autorelease];
-
-    if(_windowRef)
-    {
-        [_windowRef initCocoaRenderView:fmt];
-    }
-    else
-    {
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    _nsglContext = [_windowRef nsOpenGLContext];
-    [_nsglContext makeCurrentContext];
-
-    glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
-    glClear(GL_COLOR_BUFFER_BIT);
-
-
-    DisplayBuffers();
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int VideoRenderNSOpenGL::setRenderTargetFullScreen()
-{
-    LockAGLCntx();
-
-
-    GLuint attribs[] =
-    {
-        NSOpenGLPFAColorSize, 24,
-        NSOpenGLPFAAlphaSize, 8,
-        NSOpenGLPFADepthSize, 16,
-        NSOpenGLPFAAccelerated,
-        0
-    };
-
-    NSOpenGLPixelFormat* fmt = [[[NSOpenGLPixelFormat alloc] initWithAttributes:
-                          (NSOpenGLPixelFormatAttribute*) attribs] autorelease];
-
-    // Store original superview and frame for use when exiting full screens
-    _windowRefSuperViewFrame = [_windowRef frame];
-    _windowRefSuperView = [_windowRef superview];
-
-
-    // create new fullscreen window
-    NSRect screenRect = [[NSScreen mainScreen]frame];
-    [_windowRef setFrame:screenRect];
-    [_windowRef setBounds:screenRect];
-
-
-    _fullScreenWindow = [[CocoaFullScreenWindow alloc]init];
-    [_fullScreenWindow grabFullScreen];
-    [[[_fullScreenWindow window] contentView] addSubview:_windowRef];
-
-    if(_windowRef)
-    {
-        [_windowRef initCocoaRenderViewFullScreen:fmt];
-    }
-    else
-    {
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    _nsglContext = [_windowRef nsOpenGLContext];
-    [_nsglContext makeCurrentContext];
-
-    glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
-    glClear(GL_COLOR_BUFFER_BIT);
-
-    DisplayBuffers();
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-VideoRenderNSOpenGL::~VideoRenderNSOpenGL()
-{
-
-    if(_fullScreen)
-    {
-        if(_fullScreenWindow)
-        {
-            // Detach CocoaRenderView from full screen view back to
-            // it's original parent.
-            [_windowRef removeFromSuperview];
-            if(_windowRefSuperView)
-            {
-              [_windowRefSuperView addSubview:_windowRef];
-              [_windowRef setFrame:_windowRefSuperViewFrame];
-            }
-
-            WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, 0, "%s:%d Attempting to release fullscreen window", __FUNCTION__, __LINE__);
-            [_fullScreenWindow releaseFullScreen];
-
-        }
-    }
-
-    // Signal event to exit thread, then delete it
-    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
-
-    if (tmpPtr)
-    {
-        _screenUpdateEvent->Set();
-        _screenUpdateEvent->StopTimer();
-
-        tmpPtr->Stop();
-        delete tmpPtr;
-        delete _screenUpdateEvent;
-        _screenUpdateEvent = NULL;
-    }
-
-    if (_nsglContext != 0)
-    {
-        [_nsglContext makeCurrentContext];
-        _nsglContext = nil;
-    }
-
-    // Delete all channels
-    std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
-    while (it!= _nsglChannels.end())
-    {
-        delete it->second;
-        _nsglChannels.erase(it);
-        it = _nsglChannels.begin();
-    }
-    _nsglChannels.clear();
-
-    // Clean the zOrder map
-    std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
-    while(zIt != _zOrderToChannel.end())
-    {
-        _zOrderToChannel.erase(zIt);
-        zIt = _zOrderToChannel.begin();
-    }
-    _zOrderToChannel.clear();
-
-}
-
-/* static */
-int VideoRenderNSOpenGL::GetOpenGLVersion(int& /*nsglMajor*/, int& /*nsglMinor*/)
-{
-    return -1;
-}
-
-int VideoRenderNSOpenGL::Init()
-{
-
-    LockAGLCntx();
-    if (!_screenUpdateThread)
-    {
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
-
-    // Start the event triggering the render process
-    unsigned int monitorFreq = 60;
-    _screenUpdateEvent->StartTimer(true, 1000/monitorFreq);
-
-    if (CreateMixingContext() == -1)
-    {
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-VideoChannelNSOpenGL* VideoRenderNSOpenGL::CreateNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    if (HasChannel(channel))
-    {
-        return NULL;
-    }
-
-    if (_zOrderToChannel.find(zOrder) != _zOrderToChannel.end())
-    {
-
-    }
-
-    VideoChannelNSOpenGL* newAGLChannel = new VideoChannelNSOpenGL(_nsglContext, _id, this);
-    if (newAGLChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
-    {
-        if (newAGLChannel)
-        {
-            delete newAGLChannel;
-            newAGLChannel = NULL;
-        }
-
-        return NULL;
-    }
-
-    _nsglChannels[channel] = newAGLChannel;
-    _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
-
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s successfully created NSGL channel number %d", __FUNCTION__, channel);
-
-    return newAGLChannel;
-}
-
-int VideoRenderNSOpenGL::DeleteAllNSGLChannels()
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    std::map<int, VideoChannelNSOpenGL*>::iterator it;
-    it = _nsglChannels.begin();
-
-    while (it != _nsglChannels.end())
-    {
-        VideoChannelNSOpenGL* channel = it->second;
-        WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Deleting channel %d", __FUNCTION__, channel);
-        delete channel;
-        it++;
-    }
-    _nsglChannels.clear();
-    return 0;
-}
-
-int32_t VideoRenderNSOpenGL::DeleteNSGLChannel(const uint32_t channel)
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    std::map<int, VideoChannelNSOpenGL*>::iterator it;
-    it = _nsglChannels.find(channel);
-    if (it != _nsglChannels.end())
-    {
-        delete it->second;
-        _nsglChannels.erase(it);
-    }
-    else
-    {
-        return -1;
-    }
-
-    std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
-    while( zIt != _zOrderToChannel.end())
-    {
-        if (zIt->second == (int)channel)
-        {
-            _zOrderToChannel.erase(zIt);
-            break;
-        }
-        zIt++;
-    }
-
-    return 0;
-}
-
-int32_t VideoRenderNSOpenGL::GetChannelProperties(const uint16_t streamId,
-                                                  uint32_t& zOrder,
-                                                  float& left,
-                                                  float& top,
-                                                  float& right,
-                                                  float& bottom)
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    bool channelFound = false;
-
-    // Loop through all channels until we find a match.
-    // From that, get zorder.
-    // From that, get T, L, R, B
-    for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
-            rIt != _zOrderToChannel.rend();
-            rIt++)
-    {
-        if(streamId == rIt->second)
-        {
-            channelFound = true;
-
-            zOrder = rIt->second;
-
-            std::map<int, VideoChannelNSOpenGL*>::iterator rIt = _nsglChannels.find(streamId);
-            VideoChannelNSOpenGL* tempChannel = rIt->second;
-
-            if(-1 == tempChannel->GetChannelProperties(left, top, right, bottom) )
-            {
-                return -1;
-            }
-            break;
-        }
-    }
-
-    if(false == channelFound)
-    {
-
-        return -1;
-    }
-
-    return 0;
-}
-
-int VideoRenderNSOpenGL::StopThread()
-{
-
-    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
-    WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
-                 "%s Stopping thread ", __FUNCTION__, tmpPtr);
-
-    if (tmpPtr)
-    {
-        _screenUpdateEvent->Set();
-        tmpPtr->Stop();
-        delete tmpPtr;
-    }
-
-    delete _screenUpdateEvent;
-    _screenUpdateEvent = NULL;
-
-    return 0;
-}
-
-bool VideoRenderNSOpenGL::IsFullScreen()
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-    return _fullScreen;
-}
-
-bool VideoRenderNSOpenGL::HasChannels()
-{
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    if (_nsglChannels.begin() != _nsglChannels.end())
-    {
-        return true;
-    }
-    return false;
-}
-
-bool VideoRenderNSOpenGL::HasChannel(int channel)
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channel);
-
-    if (it != _nsglChannels.end())
-    {
-        return true;
-    }
-    return false;
-}
-
-int VideoRenderNSOpenGL::GetChannels(std::list<int>& channelList)
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
-
-    while (it != _nsglChannels.end())
-    {
-        channelList.push_back(it->first);
-        it++;
-    }
-
-    return 0;
-}
-
-VideoChannelNSOpenGL* VideoRenderNSOpenGL::ConfigureNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channel);
-
-    if (it != _nsglChannels.end())
-    {
-        VideoChannelNSOpenGL* aglChannel = it->second;
-        if (aglChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s failed to set stream settings: channel %d. channel=%d zOrder=%d startWidth=%d startHeight=%d stopWidth=%d stopHeight=%d",
-                    __FUNCTION__, channel, zOrder, startWidth, startHeight, stopWidth, stopHeight);
-            return NULL;
-        }
-        WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Configuring channel %d. channel=%d zOrder=%d startWidth=%d startHeight=%d stopWidth=%d stopHeight=%d",
-                __FUNCTION__, channel, zOrder, startWidth, startHeight, stopWidth, stopHeight);
-
-        std::multimap<int, int>::iterator it = _zOrderToChannel.begin();
-        while(it != _zOrderToChannel.end())
-        {
-            if (it->second == channel)
-            {
-                if (it->first != zOrder)
-                {
-                    _zOrderToChannel.erase(it);
-                    _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
-                }
-                break;
-            }
-            it++;
-        }
-        return aglChannel;
-    }
-
-    return NULL;
-}
-
-/*
- *
- *    Rendering process
- *
- */
-
-bool VideoRenderNSOpenGL::ScreenUpdateThreadProc(void* obj)
-{
-    return static_cast<VideoRenderNSOpenGL*>(obj)->ScreenUpdateProcess();
-}
-
-bool VideoRenderNSOpenGL::ScreenUpdateProcess()
-{
-
-    _screenUpdateEvent->Wait(10);
-    LockAGLCntx();
-
-    if (!_screenUpdateThread)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s no screen update thread", __FUNCTION__);
-        UnlockAGLCntx();
-        return false;
-    }
-
-    [_nsglContext makeCurrentContext];
-
-    if (GetWindowRect(_windowRect) == -1)
-    {
-        UnlockAGLCntx();
-        return true;
-    }
-
-    if (_windowWidth != (_windowRect.right - _windowRect.left)
-            || _windowHeight != (_windowRect.bottom - _windowRect.top))
-    {
-        _windowWidth = _windowRect.right - _windowRect.left;
-        _windowHeight = _windowRect.bottom - _windowRect.top;
-        glViewport(0, 0, _windowWidth, _windowHeight);
-    }
-
-    // Check if there are any updated buffers
-    bool updated = false;
-    std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
-    while (it != _nsglChannels.end())
-    {
-
-        VideoChannelNSOpenGL* aglChannel = it->second;
-        aglChannel->UpdateStretchSize(_windowHeight, _windowWidth);
-        aglChannel->IsUpdated(updated);
-        if (updated)
-        {
-            break;
-        }
-        it++;
-    }
-
-    if (updated)
-    {
-
-        // At least on buffers is updated, we need to repaint the texture
-        if (RenderOffScreenBuffers() != -1)
-        {
-            UnlockAGLCntx();
-            return true;
-        }
-    }
-    //    }
-    UnlockAGLCntx();
-    return true;
-}
-
-/*
- *
- *    Functions for creating mixing buffers and screen settings
- *
- */
-
-int VideoRenderNSOpenGL::CreateMixingContext()
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    if(_fullScreen)
-    {
-        if(-1 == setRenderTargetFullScreen())
-        {
-            return -1;
-        }
-    }
-    else
-    {
-
-        if(-1 == setRenderTargetWindow())
-        {
-            return -1;
-        }
-    }
-
-    configureNSOpenGLEngine();
-
-    DisplayBuffers();
-
-    GLenum glErr = glGetError();
-    if (glErr)
-    {
-    }
-
-    return 0;
-}
-
-/*
- *
- *    Rendering functions
- *
- */
-
-int VideoRenderNSOpenGL::RenderOffScreenBuffers()
-{
-    LockAGLCntx();
-
-    // Get the current window size, it might have changed since last render.
-    if (GetWindowRect(_windowRect) == -1)
-    {
-        UnlockAGLCntx();
-        return -1;
-    }
-
-    [_nsglContext makeCurrentContext];
-    glClear(GL_COLOR_BUFFER_BIT);
-
-    // Loop through all channels starting highest zOrder ending with lowest.
-    for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
-            rIt != _zOrderToChannel.rend();
-            rIt++)
-    {
-        int channelId = rIt->second;
-        std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channelId);
-
-        VideoChannelNSOpenGL* aglChannel = it->second;
-
-        aglChannel->RenderOffScreenBuffer();
-    }
-
-    DisplayBuffers();
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-/*
- *
- * Help functions
- *
- * All help functions assumes external protections
- *
- */
-
-int VideoRenderNSOpenGL::DisplayBuffers()
-{
-
-    LockAGLCntx();
-
-    glFinish();
-    [_nsglContext flushBuffer];
-
-    WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s glFinish and [_nsglContext flushBuffer]", __FUNCTION__);
-
-    UnlockAGLCntx();
-    return 0;
-}
-
-int VideoRenderNSOpenGL::GetWindowRect(Rect& rect)
-{
-
-    CriticalSectionScoped cs(&_nsglContextCritSec);
-
-    if (_windowRef)
-    {
-        if(_fullScreen)
-        {
-            NSRect mainDisplayRect = [[NSScreen mainScreen] frame];
-            rect.bottom = 0;
-            rect.left = 0;
-            rect.right = mainDisplayRect.size.width;
-            rect.top = mainDisplayRect.size.height;
-        }
-        else
-        {
-            rect.top = [_windowRef frame].origin.y;
-            rect.left = [_windowRef frame].origin.x;
-            rect.bottom = [_windowRef frame].origin.y + [_windowRef frame].size.height;
-            rect.right = [_windowRef frame].origin.x + [_windowRef frame].size.width;
-        }
-
-        return 0;
-    }
-    else
-    {
-        return -1;
-    }
-}
-
-int32_t VideoRenderNSOpenGL::SetText(const uint8_t /*textId*/,
-                                     const uint8_t* /*text*/,
-                                     const int32_t /*textLength*/,
-                                     const uint32_t /*textColorRef*/,
-                                     const uint32_t /*backgroundColorRef*/,
-                                     const float /*left*/,
-                                     const float /*top*/,
-                                     const float /*right*/,
-                                     const float /*bottom*/)
-{
-
-    return 0;
-
-}
-
-void VideoRenderNSOpenGL::LockAGLCntx()
-{
-    _nsglContextCritSec.Enter();
-}
-void VideoRenderNSOpenGL::UnlockAGLCntx()
-{
-    _nsglContextCritSec.Leave();
-}
-
-/*
-
- bool VideoRenderNSOpenGL::SetFullScreen(bool fullscreen)
- {
- NSRect mainDisplayRect, viewRect;
-
- // Create a screen-sized window on the display you want to take over
- // Note, mainDisplayRect has a non-zero origin if the key window is on a secondary display
- mainDisplayRect = [[NSScreen mainScreen] frame];
- fullScreenWindow = [[NSWindow alloc] initWithContentRect:mainDisplayRect styleMask:NSBorderlessWindowMask
- backing:NSBackingStoreBuffered defer:YES];
-
- // Set the window level to be above the menu bar
- [fullScreenWindow setLevel:NSMainMenuWindowLevel+1];
-
- // Perform any other window configuration you desire
- [fullScreenWindow setOpaque:YES];
- [fullScreenWindow setHidesOnDeactivate:YES];
-
- // Create a view with a double-buffered OpenGL context and attach it to the window
- // By specifying the non-fullscreen context as the shareContext, we automatically inherit the OpenGL objects (textures, etc) it has defined
- viewRect = NSMakeRect(0.0, 0.0, mainDisplayRect.size.width, mainDisplayRect.size.height);
- fullScreenView = [[MyOpenGLView alloc] initWithFrame:viewRect shareContext:[openGLView openGLContext]];
- [fullScreenWindow setContentView:fullScreenView];
-
- // Show the window
- [fullScreenWindow makeKeyAndOrderFront:self];
-
- // Set the scene with the full-screen viewport and viewing transformation
- [scene setViewportRect:viewRect];
-
- // Assign the view's MainController to self
- [fullScreenView setMainController:self];
-
- if (!isAnimating) {
- // Mark the view as needing drawing to initalize its contents
- [fullScreenView setNeedsDisplay:YES];
- }
- else {
- // Start playing the animation
- [fullScreenView startAnimation];
- }
-
- }
-
-
-
- */
-
-
-}  // namespace webrtc
-
-#endif // COCOA_RENDERING
diff --git a/chromium/third_party/webrtc/modules/video_render/test/testAPI/renderStartImage.bmp b/chromium/third_party/webrtc/modules/video_render/test/testAPI/renderStartImage.bmp
deleted file mode 100644
index c443a58f6cb..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/test/testAPI/renderStartImage.bmp
+++ /dev/null
diff --git a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI.cc b/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI.cc
deleted file mode 100644
index cea2f6b56fe..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI.cc
+++ /dev/null
@@ -1,645 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/video_render/test/testAPI/testAPI.h"
-
-#include <stdio.h>
-
-#if defined(_WIN32)
-#include <tchar.h>
-#include <windows.h>
-#include <assert.h>
-#include <fstream>
-#include <iostream>
-#include <string>
-#include <windows.h>
-#include <ddraw.h>
-
-#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
-
-#include <X11/Xlib.h>
-#include <X11/Xutil.h>
-#include <iostream>
-#include <sys/time.h>
-
-#endif
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/modules/video_render/video_render.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-using namespace webrtc;
-
-void GetTestVideoFrame(VideoFrame* frame, uint8_t startColor);
-int TestSingleStream(VideoRender* renderModule);
-int TestFullscreenStream(VideoRender* &renderModule,
-                         void* window,
-                         const VideoRenderType videoRenderType);
-int TestBitmapText(VideoRender* renderModule);
-int TestMultipleStreams(VideoRender* renderModule);
-int TestExternalRender(VideoRender* renderModule);
-
-#define TEST_FRAME_RATE 30
-#define TEST_TIME_SECOND 5
-#define TEST_FRAME_NUM (TEST_FRAME_RATE*TEST_TIME_SECOND)
-#define TEST_STREAM0_START_COLOR 0
-#define TEST_STREAM1_START_COLOR 64
-#define TEST_STREAM2_START_COLOR 128
-#define TEST_STREAM3_START_COLOR 192
-
-#if defined(WEBRTC_LINUX)
-
-#define GET_TIME_IN_MS timeGetTime()
-
-unsigned long timeGetTime()
-{
-    struct timeval tv;
-    struct timezone tz;
-    unsigned long val;
-
-    gettimeofday(&tv, &tz);
-    val= tv.tv_sec*1000+ tv.tv_usec/1000;
-    return(val);
-}
-
-#elif defined(WEBRTC_MAC)
-
-#include <unistd.h>
-
-#define GET_TIME_IN_MS timeGetTime()
-
-unsigned long timeGetTime()
-{
-    return 0;
-}
-
-#else
-
-#define GET_TIME_IN_MS ::timeGetTime()
-
-#endif
-
-using namespace std;
-
-#if defined(_WIN32)
-LRESULT CALLBACK WebRtcWinProc( HWND hWnd,UINT uMsg,WPARAM wParam,LPARAM lParam)
-{
-    switch(uMsg)
-    {
-        case WM_DESTROY:
-        break;
-        case WM_COMMAND:
-        break;
-    }
-    return DefWindowProc(hWnd,uMsg,wParam,lParam);
-}
-
-int WebRtcCreateWindow(HWND &hwndMain,int winNum, int width, int height)
-{
-    HINSTANCE hinst = GetModuleHandle(0);
-    WNDCLASSEX wcx;
-    wcx.hInstance = hinst;
-    wcx.lpszClassName = TEXT("VideoRenderTest");
-    wcx.lpfnWndProc = (WNDPROC)WebRtcWinProc;
-    wcx.style = CS_DBLCLKS;
-    wcx.hIcon = LoadIcon (NULL, IDI_APPLICATION);
-    wcx.hIconSm = LoadIcon (NULL, IDI_APPLICATION);
-    wcx.hCursor = LoadCursor (NULL, IDC_ARROW);
-    wcx.lpszMenuName = NULL;
-    wcx.cbSize = sizeof (WNDCLASSEX);
-    wcx.cbClsExtra = 0;
-    wcx.cbWndExtra = 0;
-    wcx.hbrBackground = GetSysColorBrush(COLOR_3DFACE);
-
-    // Register our window class with the operating system.
-    // If there is an error, exit program.
-    if ( !RegisterClassEx (&wcx) )
-    {
-        MessageBox( 0, TEXT("Failed to register window class!"),TEXT("Error!"), MB_OK|MB_ICONERROR );
-        return 0;
-    }
-
-    // Create the main window.
-    hwndMain = CreateWindowEx(
-            0, // no extended styles
-            TEXT("VideoRenderTest"), // class name
-            TEXT("VideoRenderTest Window"), // window name
-            WS_OVERLAPPED |WS_THICKFRAME, // overlapped window
-            800, // horizontal position
-            0, // vertical position
-            width, // width
-            height, // height
-            (HWND) NULL, // no parent or owner window
-            (HMENU) NULL, // class menu used
-            hinst, // instance handle
-            NULL); // no window creation data
-
-    if (!hwndMain)
-        return -1;
-
-    // Show the window using the flag specified by the program
-    // that started the application, and send the application
-    // a WM_PAINT message.
-
-    ShowWindow(hwndMain, SW_SHOWDEFAULT);
-    UpdateWindow(hwndMain);
-    return 0;
-}
-
-#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
-
-int WebRtcCreateWindow(Window *outWindow, Display **outDisplay, int winNum, int width, int height) // unsigned char* title, int titleLength)
-
-{
-    int screen, xpos = 10, ypos = 10;
-    XEvent evnt;
-    XSetWindowAttributes xswa; // window attribute struct
-    XVisualInfo vinfo; // screen visual info struct
-    unsigned long mask; // attribute mask
-
-    // get connection handle to xserver
-    Display* _display = XOpenDisplay( NULL );
-
-    // get screen number
-    screen = DefaultScreen(_display);
-
-    // put desired visual info for the screen in vinfo
-    if( XMatchVisualInfo(_display, screen, 24, TrueColor, &vinfo) != 0 )
-    {
-        //printf( "Screen visual info match!\n" );
-    }
-
-    // set window attributes
-    xswa.colormap = XCreateColormap(_display, DefaultRootWindow(_display), vinfo.visual, AllocNone);
-    xswa.event_mask = StructureNotifyMask | ExposureMask;
-    xswa.background_pixel = 0;
-    xswa.border_pixel = 0;
-
-    // value mask for attributes
-    mask = CWBackPixel | CWBorderPixel | CWColormap | CWEventMask;
-
-    switch( winNum )
-    {
-        case 0:
-        xpos = 200;
-        ypos = 200;
-        break;
-        case 1:
-        xpos = 300;
-        ypos = 200;
-        break;
-        default:
-        break;
-    }
-
-    // create a subwindow for parent (defroot)
-    Window _window = XCreateWindow(_display, DefaultRootWindow(_display),
-            xpos, ypos,
-            width,
-            height,
-            0, vinfo.depth,
-            InputOutput,
-            vinfo.visual,
-            mask, &xswa);
-
-    // Set window name
-    if( winNum == 0 )
-    {
-        XStoreName(_display, _window, "VE MM Local Window");
-        XSetIconName(_display, _window, "VE MM Local Window");
-    }
-    else if( winNum == 1 )
-    {
-        XStoreName(_display, _window, "VE MM Remote Window");
-        XSetIconName(_display, _window, "VE MM Remote Window");
-    }
-
-    // make x report events for mask
-    XSelectInput(_display, _window, StructureNotifyMask);
-
-    // map the window to the display
-    XMapWindow(_display, _window);
-
-    // wait for map event
-    do
-    {
-        XNextEvent(_display, &evnt);
-    }
-    while (evnt.type != MapNotify || evnt.xmap.event != _window);
-
-    *outWindow = _window;
-    *outDisplay = _display;
-
-    return 0;
-}
-#endif  // WEBRTC_LINUX
-
-// Note: Mac code is in testApi_mac.mm.
-
-class MyRenderCallback: public VideoRenderCallback
-{
-public:
-    MyRenderCallback() :
-        _cnt(0)
-    {
-    }
-    ;
-    ~MyRenderCallback()
-    {
-    }
-    ;
-    virtual int32_t RenderFrame(const uint32_t streamId,
-                                const VideoFrame& videoFrame) {
-        _cnt++;
-        if (_cnt % 100 == 0)
-        {
-            printf("Render callback %d \n",_cnt);
-        }
-        return 0;
-    }
-    int32_t _cnt;
-};
-
-void GetTestVideoFrame(VideoFrame* frame, uint8_t startColor) {
-    // changing color
-    static uint8_t color = startColor;
-
-    memset(frame->buffer(kYPlane), color, frame->allocated_size(kYPlane));
-    memset(frame->buffer(kUPlane), color, frame->allocated_size(kUPlane));
-    memset(frame->buffer(kVPlane), color, frame->allocated_size(kVPlane));
-
-    ++color;
-}
-
-int TestSingleStream(VideoRender* renderModule) {
-    int error = 0;
-    // Add settings for a stream to render
-    printf("Add stream 0 to entire window\n");
-    const int streamId0 = 0;
-    VideoRenderCallback* renderCallback0 = renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 1.0f, 1.0f);
-    assert(renderCallback0 != NULL);
-
-    printf("Start render\n");
-    error = renderModule->StartRender(streamId0);
-    if (error != 0) {
-      // TODO(phoglund): This test will not work if compiled in release mode.
-      // This rather silly construct here is to avoid compilation errors when
-      // compiling in release. Release => no asserts => unused 'error' variable.
-      assert(false);
-    }
-
-    // Loop through an I420 file and render each frame
-    const int width = 352;
-    const int half_width = (width + 1) / 2;
-    const int height = 288;
-
-    VideoFrame videoFrame0;
-    videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
-
-    const uint32_t renderDelayMs = 500;
-
-    for (int i=0; i<TEST_FRAME_NUM; i++) {
-        GetTestVideoFrame(&videoFrame0, TEST_STREAM0_START_COLOR);
-        // Render this frame with the specified delay
-        videoFrame0.set_render_time_ms(TickTime::MillisecondTimestamp()
-                                       + renderDelayMs);
-        renderCallback0->RenderFrame(streamId0, videoFrame0);
-        SleepMs(1000/TEST_FRAME_RATE);
-    }
-
-
-    // Shut down
-    printf("Closing...\n");
-    error = renderModule->StopRender(streamId0);
-    assert(error == 0);
-
-    error = renderModule->DeleteIncomingRenderStream(streamId0);
-    assert(error == 0);
-
-    return 0;
-}
-
-int TestFullscreenStream(VideoRender* &renderModule,
-                         void* window,
-                         const VideoRenderType videoRenderType) {
-    VideoRender::DestroyVideoRender(renderModule);
-    renderModule = VideoRender::CreateVideoRender(12345, window, true, videoRenderType);
-
-    TestSingleStream(renderModule);
-
-    VideoRender::DestroyVideoRender(renderModule);
-    renderModule = VideoRender::CreateVideoRender(12345, window, false, videoRenderType);
-
-    return 0;
-}
-
-int TestBitmapText(VideoRender* renderModule) {
-#if defined(WIN32)
-
-    int error = 0;
-    // Add settings for a stream to render
-    printf("Add stream 0 to entire window\n");
-    const int streamId0 = 0;
-    VideoRenderCallback* renderCallback0 = renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 1.0f, 1.0f);
-    assert(renderCallback0 != NULL);
-
-    printf("Adding Bitmap\n");
-    DDCOLORKEY ColorKey; // black
-    ColorKey.dwColorSpaceHighValue = RGB(0, 0, 0);
-    ColorKey.dwColorSpaceLowValue = RGB(0, 0, 0);
-    HBITMAP hbm = (HBITMAP)LoadImage(NULL,
-                                     (LPCTSTR)_T("renderStartImage.bmp"),
-                                     IMAGE_BITMAP, 0, 0, LR_LOADFROMFILE);
-    renderModule->SetBitmap(hbm, 0, &ColorKey, 0.0f, 0.0f, 0.3f,
-                             0.3f);
-
-    printf("Adding Text\n");
-    renderModule->SetText(1, (uint8_t*) "WebRtc Render Demo App", 20,
-                           RGB(255, 0, 0), RGB(0, 0, 0), 0.25f, 0.1f, 1.0f,
-                           1.0f);
-
-    printf("Start render\n");
-    error = renderModule->StartRender(streamId0);
-    assert(error == 0);
-
-    // Loop through an I420 file and render each frame
-    const int width = 352;
-    const int half_width = (width + 1) / 2;
-    const int height = 288;
-
-    VideoFrame videoFrame0;
-    videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
-
-    const uint32_t renderDelayMs = 500;
-
-    for (int i=0; i<TEST_FRAME_NUM; i++) {
-        GetTestVideoFrame(&videoFrame0, TEST_STREAM0_START_COLOR);
-        // Render this frame with the specified delay
-        videoFrame0.set_render_time_ms(TickTime::MillisecondTimestamp() +
-                                       renderDelayMs);
-        renderCallback0->RenderFrame(streamId0, videoFrame0);
-        SleepMs(1000/TEST_FRAME_RATE);
-    }
-    // Sleep and let all frames be rendered before closing
-    SleepMs(renderDelayMs*2);
-
-
-    // Shut down
-    printf("Closing...\n");
-    ColorKey.dwColorSpaceHighValue = RGB(0,0,0);
-    ColorKey.dwColorSpaceLowValue = RGB(0,0,0);
-    renderModule->SetBitmap(NULL, 0, &ColorKey, 0.0f, 0.0f, 0.0f, 0.0f);
-    renderModule->SetText(1, NULL, 20, RGB(255,255,255),
-                    RGB(0,0,0), 0.0f, 0.0f, 0.0f, 0.0f);
-
-    error = renderModule->StopRender(streamId0);
-    assert(error == 0);
-
-    error = renderModule->DeleteIncomingRenderStream(streamId0);
-    assert(error == 0);
-#endif
-
-    return 0;
-}
-
-int TestMultipleStreams(VideoRender* renderModule) {
-    int error = 0;
-
-    // Add settings for a stream to render
-    printf("Add stream 0\n");
-    const int streamId0 = 0;
-    VideoRenderCallback* renderCallback0 =
-        renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 0.45f, 0.45f);
-    assert(renderCallback0 != NULL);
-    printf("Add stream 1\n");
-    const int streamId1 = 1;
-    VideoRenderCallback* renderCallback1 =
-        renderModule->AddIncomingRenderStream(streamId1, 0, 0.55f, 0.0f, 1.0f, 0.45f);
-    assert(renderCallback1 != NULL);
-    printf("Add stream 2\n");
-    const int streamId2 = 2;
-    VideoRenderCallback* renderCallback2 =
-        renderModule->AddIncomingRenderStream(streamId2, 0, 0.0f, 0.55f, 0.45f, 1.0f);
-    assert(renderCallback2 != NULL);
-    printf("Add stream 3\n");
-    const int streamId3 = 3;
-    VideoRenderCallback* renderCallback3 =
-        renderModule->AddIncomingRenderStream(streamId3, 0, 0.55f, 0.55f, 1.0f, 1.0f);
-    assert(renderCallback3 != NULL);
-    error = renderModule->StartRender(streamId0);
-    if (error != 0) {
-      // TODO(phoglund): This test will not work if compiled in release mode.
-      // This rather silly construct here is to avoid compilation errors when
-      // compiling in release. Release => no asserts => unused 'error' variable.
-      assert(false);
-    }
-    error = renderModule->StartRender(streamId1);
-    assert(error == 0);
-    error = renderModule->StartRender(streamId2);
-    assert(error == 0);
-    error = renderModule->StartRender(streamId3);
-    assert(error == 0);
-
-    // Loop through an I420 file and render each frame
-    const int width = 352;
-    const int half_width = (width + 1) / 2;
-    const int height = 288;
-
-    VideoFrame videoFrame0;
-    videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
-    VideoFrame videoFrame1;
-    videoFrame1.CreateEmptyFrame(width, height, width, half_width, half_width);
-    VideoFrame videoFrame2;
-    videoFrame2.CreateEmptyFrame(width, height, width, half_width, half_width);
-    VideoFrame videoFrame3;
-    videoFrame3.CreateEmptyFrame(width, height, width, half_width, half_width);
-
-    const uint32_t renderDelayMs = 500;
-
-    // Render frames with the specified delay.
-    for (int i=0; i<TEST_FRAME_NUM; i++) {
-      GetTestVideoFrame(&videoFrame0, TEST_STREAM0_START_COLOR);
-
-      videoFrame0.set_render_time_ms(TickTime::MillisecondTimestamp() +
-                                     renderDelayMs);
-      renderCallback0->RenderFrame(streamId0, videoFrame0);
-
-      GetTestVideoFrame(&videoFrame1, TEST_STREAM1_START_COLOR);
-      videoFrame1.set_render_time_ms(TickTime::MillisecondTimestamp() +
-                                     renderDelayMs);
-      renderCallback1->RenderFrame(streamId1, videoFrame1);
-
-      GetTestVideoFrame(&videoFrame2,  TEST_STREAM2_START_COLOR);
-      videoFrame2.set_render_time_ms(TickTime::MillisecondTimestamp() +
-                                     renderDelayMs);
-      renderCallback2->RenderFrame(streamId2, videoFrame2);
-
-      GetTestVideoFrame(&videoFrame3, TEST_STREAM3_START_COLOR);
-      videoFrame3.set_render_time_ms(TickTime::MillisecondTimestamp() +
-                                     renderDelayMs);
-      renderCallback3->RenderFrame(streamId3, videoFrame3);
-
-      SleepMs(1000/TEST_FRAME_RATE);
-    }
-
-    // Shut down
-    printf("Closing...\n");
-    error = renderModule->StopRender(streamId0);
-    assert(error == 0);
-    error = renderModule->DeleteIncomingRenderStream(streamId0);
-    assert(error == 0);
-    error = renderModule->StopRender(streamId1);
-    assert(error == 0);
-    error = renderModule->DeleteIncomingRenderStream(streamId1);
-    assert(error == 0);
-    error = renderModule->StopRender(streamId2);
-    assert(error == 0);
-    error = renderModule->DeleteIncomingRenderStream(streamId2);
-    assert(error == 0);
-    error = renderModule->StopRender(streamId3);
-    assert(error == 0);
-    error = renderModule->DeleteIncomingRenderStream(streamId3);
-    assert(error == 0);
-
-    return 0;
-}
-
-int TestExternalRender(VideoRender* renderModule) {
-    int error = 0;
-    MyRenderCallback *externalRender = new MyRenderCallback();
-
-    const int streamId0 = 0;
-    VideoRenderCallback* renderCallback0 =
-        renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f,
-                                                   1.0f, 1.0f);
-    assert(renderCallback0 != NULL);
-    error = renderModule->AddExternalRenderCallback(streamId0, externalRender);
-    if (error != 0) {
-      // TODO(phoglund): This test will not work if compiled in release mode.
-      // This rather silly construct here is to avoid compilation errors when
-      // compiling in release. Release => no asserts => unused 'error' variable.
-      assert(false);
-    }
-
-    error = renderModule->StartRender(streamId0);
-    assert(error == 0);
-
-    const int width = 352;
-    const int half_width = (width + 1) / 2;
-    const int height = 288;
-    VideoFrame videoFrame0;
-    videoFrame0.CreateEmptyFrame(width, height, width, half_width, half_width);
-
-    const uint32_t renderDelayMs = 500;
-    int frameCount = TEST_FRAME_NUM;
-    for (int i=0; i<frameCount; i++) {
-        videoFrame0.set_render_time_ms(TickTime::MillisecondTimestamp() +
-                                       renderDelayMs);
-        renderCallback0->RenderFrame(streamId0, videoFrame0);
-        SleepMs(33);
-    }
-
-    // Sleep and let all frames be rendered before closing
-    SleepMs(2*renderDelayMs);
-
-    // Shut down
-    printf("Closing...\n");
-    error = renderModule->StopRender(streamId0);
-    assert(error == 0);
-    error = renderModule->DeleteIncomingRenderStream(streamId0);
-    assert(error == 0);
-    assert(frameCount == externalRender->_cnt);
-
-    delete externalRender;
-    externalRender = NULL;
-
-    return 0;
-}
-
-void RunVideoRenderTests(void* window, VideoRenderType windowType) {
-    int myId = 12345;
-
-    // Create the render module
-    printf("Create render module\n");
-    VideoRender* renderModule = NULL;
-    renderModule = VideoRender::CreateVideoRender(myId,
-                                                  window,
-                                                  false,
-                                                  windowType);
-    assert(renderModule != NULL);
-
-    // ##### Test single stream rendering ####
-    printf("#### TestSingleStream ####\n");
-    if (TestSingleStream(renderModule) != 0) {
-        printf ("TestSingleStream failed\n");
-    }
-
-    // ##### Test fullscreen rendering ####
-    printf("#### TestFullscreenStream ####\n");
-    if (TestFullscreenStream(renderModule, window, windowType) != 0) {
-        printf ("TestFullscreenStream failed\n");
-    }
-
-    // ##### Test bitmap and text ####
-    printf("#### TestBitmapText ####\n");
-    if (TestBitmapText(renderModule) != 0) {
-        printf ("TestBitmapText failed\n");
-    }
-
-    // ##### Test multiple streams ####
-    printf("#### TestMultipleStreams ####\n");
-    if (TestMultipleStreams(renderModule) != 0) {
-        printf ("TestMultipleStreams failed\n");
-    }
-
-    // ##### Test multiple streams ####
-    printf("#### TestExternalRender ####\n");
-    if (TestExternalRender(renderModule) != 0) {
-        printf ("TestExternalRender failed\n");
-    }
-
-    delete renderModule;
-    renderModule = NULL;
-
-    printf("VideoRender unit tests passed.\n");
-}
-
-// Note: The Mac main is implemented in testApi_mac.mm.
-#if defined(_WIN32)
-int _tmain(int argc, _TCHAR* argv[])
-#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
-int main(int argc, char* argv[])
-#endif
-#if !defined(WEBRTC_MAC) && !defined(WEBRTC_ANDROID)
-{
-    // Create a window for testing.
-    void* window = NULL;
-#if defined (_WIN32)
-    HWND testHwnd;
-    WebRtcCreateWindow(testHwnd, 0, 352, 288);
-    window = (void*)testHwnd;
-    VideoRenderType windowType = kRenderWindows;
-#elif defined(WEBRTC_LINUX)
-    Window testWindow;
-    Display* display;
-    WebRtcCreateWindow(&testWindow, &display, 0, 352, 288);
-    VideoRenderType windowType = kRenderX11;
-    window = (void*)testWindow;
-#endif // WEBRTC_LINUX
-
-    RunVideoRenderTests(window, windowType);
-    return 0;
-}
-#endif  // !WEBRTC_MAC
diff --git a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI.h b/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI.h
deleted file mode 100644
index 0655a5b4343..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
-
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-void RunVideoRenderTests(void* window, webrtc::VideoRenderType windowType);
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
diff --git a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm b/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
deleted file mode 100644
index dfee4c72983..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testAPI.h"
-
-#include <iostream>
-
-#import <Foundation/Foundation.h>
-#import <Cocoa/Cocoa.h>
-#import <AppKit/AppKit.h>
-#import <QTKit/QTKit.h>
-#include <sys/time.h>
-
-#import "webrtc/modules/video_render/mac/cocoa_render_view.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/modules/video_render/video_render.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-using namespace webrtc;
-
-int WebRtcCreateWindow(CocoaRenderView*& cocoaRenderer, int winNum, int width, int height)
-{
-    // In Cocoa, rendering is not done directly to a window like in Windows and Linux.
-    // It is rendererd to a Subclass of NSOpenGLView
-
-    // create cocoa container window
-    NSRect outWindowFrame = NSMakeRect(200, 800, width + 20, height + 20);
-    NSWindow* outWindow = [[NSWindow alloc] initWithContentRect:outWindowFrame 
-                                                      styleMask:NSTitledWindowMask 
-                                                        backing:NSBackingStoreBuffered 
-                                                          defer:NO];
-    [outWindow orderOut:nil];
-    [outWindow setTitle:@"Cocoa Renderer"];
-    [outWindow setBackgroundColor:[NSColor blueColor]];
-
-    // create renderer and attach to window
-    NSRect cocoaRendererFrame = NSMakeRect(10, 10, width, height);
-    cocoaRenderer = [[CocoaRenderView alloc] initWithFrame:cocoaRendererFrame];
-    [[outWindow contentView] addSubview:(NSView*)cocoaRenderer];
-
-    [outWindow makeKeyAndOrderFront:NSApp];
-
-    return 0;
-}
-
-int main (int argc, const char * argv[]) {
-    NSAutoreleasePool * pool = [[NSAutoreleasePool alloc] init];
-    [NSApplication sharedApplication];
-
-    CocoaRenderView* testWindow;
-    WebRtcCreateWindow(testWindow, 0, 352, 288);
-    VideoRenderType windowType = kRenderCocoa;
-    void* window = (void*)testWindow;
-
-    RunVideoRenderTests(window, windowType);
-
-    [pool release];
-}
diff --git a/chromium/third_party/webrtc/modules/video_render/video_render.gypi b/chromium/third_party/webrtc/modules/video_render/video_render.gypi
deleted file mode 100644
index e8cc03a4b02..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/video_render.gypi
+++ /dev/null
@@ -1,218 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      # Note this library is missing an implementation for the video render.
-      # For that targets must link with 'video_render' or
-      # 'video_render_module_internal_impl' if they want to compile and use
-      # the internal render as the default renderer.
-      'target_name': 'video_render_module',
-      'type': 'static_library',
-      'dependencies': [
-        'webrtc_utility',
-        '<(webrtc_root)/common.gyp:webrtc_common',
-        '<(webrtc_root)/common_video/common_video.gyp:common_video',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      ],
-      'sources': [
-        'external/video_render_external_impl.cc',
-        'external/video_render_external_impl.h',
-        'i_video_render.h',
-        'video_render.h',
-        'video_render_defines.h',
-        'video_render_impl.h',
-      ],
-    },
-    {
-      # Default video_render_module implementation that only supports external
-      # renders.
-      'target_name': 'video_render',
-      'type': 'static_library',
-      'dependencies': [
-        'video_render_module',
-      ],
-      'sources': [
-        'video_render_impl.cc',
-      ],
-    },
-  ], # targets
-
-  'conditions': [
-    ['build_with_chromium==0', {
-      'targets': [
-        {
-          # video_render_module implementation that supports the internal
-          # video_render implementation.
-          'target_name': 'video_render_module_internal_impl',
-          'type': 'static_library',
-          'dependencies': [
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            'video_render_module',
-          ],
-          'sources': [
-            'video_render_internal_impl.cc',
-          ],
-           # TODO(andrew): with the proper suffix, these files will be excluded
-           # automatically.
-          'conditions': [
-            ['OS=="android"', {
-              'sources': [
-                'android/video_render_android_impl.h',
-                'android/video_render_android_native_opengl2.h',
-                'android/video_render_android_surface_view.h',
-                'android/video_render_opengles20.h',
-                'android/video_render_android_impl.cc',
-                'android/video_render_android_native_opengl2.cc',
-                'android/video_render_android_surface_view.cc',
-                'android/video_render_opengles20.cc',
-              ],
-              'link_settings': {
-                'libraries': [
-                  '-lGLESv2',
-                ],
-              },
-            }],
-            ['OS=="ios"', {
-              'sources': [
-                # iOS
-                'ios/open_gles20.h',
-                'ios/open_gles20.mm',
-                'ios/video_render_ios_channel.h',
-                'ios/video_render_ios_channel.mm',
-                'ios/video_render_ios_gles20.h',
-                'ios/video_render_ios_gles20.mm',
-                'ios/video_render_ios_impl.h',
-                'ios/video_render_ios_impl.mm',
-                'ios/video_render_ios_view.h',
-                'ios/video_render_ios_view.mm',
-              ],
-              'xcode_settings': {
-                'CLANG_ENABLE_OBJC_ARC': 'YES',
-              },
-              'all_dependent_settings': {
-                'xcode_settings': {
-                  'OTHER_LDFLAGS': [
-                    '-framework OpenGLES',
-                    '-framework QuartzCore',
-                    '-framework UIKit',
-                  ],
-                },
-              },
-            }],
-            ['OS=="linux"', {
-              'sources': [
-                'linux/video_render_linux_impl.h',
-                'linux/video_x11_channel.h',
-                'linux/video_x11_render.h',
-                'linux/video_render_linux_impl.cc',
-                'linux/video_x11_channel.cc',
-                'linux/video_x11_render.cc',
-              ],
-              'link_settings': {
-                'libraries': [
-                  '-lXext',
-                ],
-              },
-            }],
-            ['OS=="mac"', {
-              'sources': [
-                'mac/cocoa_full_screen_window.h',
-                'mac/cocoa_render_view.h',
-                'mac/video_render_agl.h',
-                'mac/video_render_mac_carbon_impl.h',
-                'mac/video_render_mac_cocoa_impl.h',
-                'mac/video_render_nsopengl.h',
-                'mac/video_render_nsopengl.mm',
-                'mac/video_render_mac_cocoa_impl.mm',
-                'mac/video_render_agl.cc',
-                'mac/video_render_mac_carbon_impl.cc',
-                'mac/cocoa_render_view.mm',
-                'mac/cocoa_full_screen_window.mm',
-              ],
-            }],
-            ['OS=="win"', {
-              'sources': [
-                'windows/i_video_render_win.h',
-                'windows/video_render_direct3d9.h',
-                'windows/video_render_windows_impl.h',
-                'windows/video_render_direct3d9.cc',
-                'windows/video_render_windows_impl.cc',
-              ],
-              'include_dirs': [
-                '<(directx_sdk_path)/Include',
-              ],
-            }],
-            ['OS=="win" and clang==1', {
-              'msvs_settings': {
-                'VCCLCompilerTool': {
-                  'AdditionalOptions': [
-                    # Disable warnings failing when compiling with Clang on Windows.
-                    # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
-                    '-Wno-comment',
-                    '-Wno-reorder',
-                    '-Wno-unused-value',
-                    '-Wno-unused-private-field',
-                  ],
-                },
-              },
-            }],
-          ] # conditions
-        },
-      ],
-    }], # build_with_chromium==0
-    ['include_tests==1 and OS!="ios"', {
-      'targets': [
-        {
-          # Does not compile on iOS: webrtc:4755.
-          'target_name': 'video_render_tests',
-          'type': 'executable',
-          'dependencies': [
-            'video_render_module_internal_impl',
-            'webrtc_utility',
-            '<(webrtc_root)/common.gyp:webrtc_common',
-            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/common_video/common_video.gyp:common_video',
-          ],
-          'sources': [
-            'test/testAPI/testAPI.cc',
-            'test/testAPI/testAPI.h',
-            'test/testAPI/testAPI_android.cc',
-            'test/testAPI/testAPI_mac.mm',
-          ],
-          'conditions': [
-            ['OS=="mac" or OS=="linux"', {
-              'cflags': [
-                '-Wno-write-strings',
-              ],
-              'ldflags': [
-                '-lpthread -lm',
-              ],
-            }],
-            ['OS=="linux"', {
-              'link_settings': {
-                'libraries': [
-                  '-lX11',
-                ],
-              },
-            }],
-            ['OS=="mac"', {
-              'xcode_settings': {
-                'OTHER_LDFLAGS': [
-                  '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL',
-                ],
-              },
-            }],
-          ] # conditions
-        }, # video_render_module_test
-      ], # targets
-    }], # include_tests==1 and OS!=ios
-  ], # conditions
-}
-
diff --git a/chromium/third_party/webrtc/modules/video_render/video_render.h b/chromium/third_party/webrtc/modules/video_render/video_render.h
deleted file mode 100644
index 84c9536e35c..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/video_render.h
+++ /dev/null
@@ -1,268 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
-
-/*
- * video_render.h
- *
- * This header file together with module.h and module_common_types.h
- * contains all of the APIs that are needed for using the video render
- * module class.
- *
- */
-
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-namespace webrtc {
-
-// Class definitions
-class VideoRender: public Module
-{
-public:
-    /*
-     *   Create a video render module object
-     *
-     *   id              - unique identifier of this video render module object
-     *   window          - pointer to the window to render to
-     *   fullscreen      - true if this is a fullscreen renderer
-     *   videoRenderType - type of renderer to create
-     */
-    static VideoRender
-            * CreateVideoRender(
-                                          const int32_t id,
-                                          void* window,
-                                          const bool fullscreen,
-                                          const VideoRenderType videoRenderType =
-                                                  kRenderDefault);
-
-    /*
-     *   Destroy a video render module object
-     *
-     *   module  - object to destroy
-     */
-    static void DestroyVideoRender(VideoRender* module);
-
-    int64_t TimeUntilNextProcess() override = 0;
-    void Process() override = 0;
-
-    /**************************************************************************
-     *
-     *   Window functions
-     *
-     ***************************************************************************/
-
-    /*
-     *   Get window for this renderer
-     */
-    virtual void* Window() = 0;
-
-    /*
-     *   Change render window
-     *
-     *   window      - the new render window, assuming same type as originally created.
-     */
-    virtual int32_t ChangeWindow(void* window) = 0;
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-
-    /*
-     *   Add incoming render stream
-     *
-     *   streamID    - id of the stream to add
-     *   zOrder      - relative render order for the streams, 0 = on top
-     *   left        - position of the stream in the window, [0.0f, 1.0f]
-     *   top         - position of the stream in the window, [0.0f, 1.0f]
-     *   right       - position of the stream in the window, [0.0f, 1.0f]
-     *   bottom      - position of the stream in the window, [0.0f, 1.0f]
-     *
-     *   Return      - callback class to use for delivering new frames to render.
-     */
-    virtual VideoRenderCallback
-            * AddIncomingRenderStream(const uint32_t streamId,
-                                      const uint32_t zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom) = 0;
-    /*
-     *   Delete incoming render stream
-     *
-     *   streamID    - id of the stream to add
-     */
-    virtual int32_t
-            DeleteIncomingRenderStream(const uint32_t streamId) = 0;
-
-    /*
-     *   Add incoming render callback, used for external rendering
-     *
-     *   streamID     - id of the stream the callback is used for
-     *   renderObject - the VideoRenderCallback to use for this stream, NULL to remove
-     *
-     *   Return      - callback class to use for delivering new frames to render.
-     */
-    virtual int32_t
-            AddExternalRenderCallback(const uint32_t streamId,
-                                      VideoRenderCallback* renderObject) = 0;
-
-    /*
-     *   Get the porperties for an incoming render stream
-     *
-     *   streamID    - [in] id of the stream to get properties for
-     *   zOrder      - [out] relative render order for the streams, 0 = on top
-     *   left        - [out] position of the stream in the window, [0.0f, 1.0f]
-     *   top         - [out] position of the stream in the window, [0.0f, 1.0f]
-     *   right       - [out] position of the stream in the window, [0.0f, 1.0f]
-     *   bottom      - [out] position of the stream in the window, [0.0f, 1.0f]
-     */
-    virtual int32_t
-            GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                              uint32_t& zOrder,
-                                              float& left, float& top,
-                                              float& right, float& bottom) const = 0;
-    /*
-     *   The incoming frame rate to the module, not the rate rendered in the window.
-     */
-    virtual uint32_t
-            GetIncomingFrameRate(const uint32_t streamId) = 0;
-
-    /*
-     *   Returns the number of incoming streams added to this render module
-     */
-    virtual uint32_t GetNumIncomingRenderStreams() const = 0;
-
-    /*
-     *   Returns true if this render module has the streamId added, false otherwise.
-     */
-    virtual bool
-            HasIncomingRenderStream(const uint32_t streamId) const = 0;
-
-    /*
-     *   Registers a callback to get raw images in the same time as sent
-     *   to the renderer. To be used for external rendering.
-     */
-    virtual int32_t
-            RegisterRawFrameCallback(const uint32_t streamId,
-                                     VideoRenderCallback* callbackObj) = 0;
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    /*
-     *   Starts rendering the specified stream
-     */
-    virtual int32_t StartRender(const uint32_t streamId) = 0;
-
-    /*
-     *   Stops the renderer
-     */
-    virtual int32_t StopRender(const uint32_t streamId) = 0;
-
-    /*
-     *   Resets the renderer
-     *   No streams are removed. The state should be as after AddStream was called.
-     */
-    virtual int32_t ResetRender() = 0;
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    /*
-     *   Returns the preferred render video type
-     */
-    virtual RawVideoType PreferredVideoType() const = 0;
-
-    /*
-     *   Returns true if the renderer is in fullscreen mode, otherwise false.
-     */
-    virtual bool IsFullScreen() = 0;
-
-    /*
-     *   Gets screen resolution in pixels
-     */
-    virtual int32_t
-            GetScreenResolution(uint32_t& screenWidth,
-                                uint32_t& screenHeight) const = 0;
-
-    /*
-     *   Get the actual render rate for this stream. I.e rendered frame rate,
-     *   not frames delivered to the renderer.
-     */
-    virtual uint32_t RenderFrameRate(const uint32_t streamId) = 0;
-
-    /*
-     *   Set cropping of incoming stream
-     */
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom) = 0;
-
-    /*
-     * re-configure renderer
-     */
-
-    // Set the expected time needed by the graphics card or external renderer,
-    // i.e. frames will be released for rendering |delay_ms| before set render
-    // time in the video frame.
-    virtual int32_t SetExpectedRenderDelay(uint32_t stream_id,
-                                           int32_t delay_ms) = 0;
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom) = 0;
-
-    virtual int32_t SetTransparentBackground(const bool enable) = 0;
-
-    virtual int32_t FullScreenRender(void* window, const bool enable) = 0;
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left, const float top,
-                              const float right, const float bottom) = 0;
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t textColorRef,
-                            const uint32_t backgroundColorRef,
-                            const float left, const float top,
-                            const float right, const float bottom) = 0;
-
-    /*
-     * Set a start image. The image is rendered before the first image has been delivered
-     */
-    virtual int32_t SetStartImage(const uint32_t streamId,
-                                  const VideoFrame& videoFrame) = 0;
-
-    /*
-     * Set a timout image. The image is rendered if no videoframe has been delivered
-     */
-    virtual int32_t SetTimeoutImage(const uint32_t streamId,
-                                    const VideoFrame& videoFrame,
-                                    const uint32_t timeout) = 0;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_defines.h b/chromium/third_party/webrtc/modules/video_render/video_render_defines.h
deleted file mode 100644
index 999707cb6e5..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/video_render_defines.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
-
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/include/incoming_video_stream.h"
-#include "webrtc/modules/include/module_common_types.h"
-
-namespace webrtc
-{
-// Defines
-#ifndef NULL
-#define NULL    0
-#endif
-
-// Enums
-enum VideoRenderType
-{
-    kRenderExternal = 0, // External
-    kRenderWindows = 1, // Windows
-    kRenderCocoa = 2, // Mac
-    kRenderCarbon = 3,
-    kRenderiOS = 4, // iPhone
-    kRenderAndroid = 5, // Android
-    kRenderX11 = 6, // Linux
-    kRenderDefault
-};
-
-// Runtime errors
-enum VideoRenderError
-{
-    kRenderShutDown = 0,
-    kRenderPerformanceAlarm = 1
-};
-
-// Feedback class to be implemented by module user
-class VideoRenderFeedback
-{
-public:
-    virtual void OnRenderError(const int32_t streamId,
-                               const VideoRenderError error) = 0;
-
-protected:
-    virtual ~VideoRenderFeedback()
-    {
-    }
-};
-
-// Mobile enums
-enum StretchMode
-{
-    kStretchToInsideEdge = 1,
-    kStretchToOutsideEdge = 2,
-    kStretchMatchWidth = 3,
-    kStretchMatchHeight = 4,
-    kStretchNone = 5
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_DEFINES_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_impl.cc b/chromium/third_party/webrtc/modules/video_render/video_render_impl.cc
deleted file mode 100644
index 75403f8dd53..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/video_render_impl.cc
+++ /dev/null
@@ -1,602 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-
-#include "webrtc/common_video/include/incoming_video_stream.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/external/video_render_external_impl.h"
-#include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-#include "webrtc/modules/video_render/video_render_impl.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-VideoRender*
-VideoRender::CreateVideoRender(const int32_t id,
-                               void* window,
-                               const bool fullscreen,
-                               const VideoRenderType videoRenderType/*=kRenderDefault*/)
-{
-    VideoRenderType resultVideoRenderType = videoRenderType;
-    if (videoRenderType == kRenderDefault)
-    {
-        resultVideoRenderType = kRenderExternal;
-    }
-    return new ModuleVideoRenderImpl(id, resultVideoRenderType, window,
-                                     fullscreen);
-}
-
-void VideoRender::DestroyVideoRender(
-                                                         VideoRender* module)
-{
-    if (module)
-    {
-        delete module;
-    }
-}
-
-ModuleVideoRenderImpl::ModuleVideoRenderImpl(
-                                             const int32_t id,
-                                             const VideoRenderType videoRenderType,
-                                             void* window,
-                                             const bool fullscreen) :
-    _id(id), _moduleCrit(*CriticalSectionWrapper::CreateCriticalSection()),
-    _ptrWindow(window), _fullScreen(fullscreen), _ptrRenderer(NULL)
-{
-
-    // Create platform specific renderer
-    switch (videoRenderType)
-    {
-        case kRenderExternal:
-        {
-            VideoRenderExternalImpl* ptrRenderer(NULL);
-            ptrRenderer = new VideoRenderExternalImpl(_id, videoRenderType,
-                                                      window, _fullScreen);
-            if (ptrRenderer)
-            {
-                _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
-            }
-        }
-            break;
-        default:
-            // Error...
-            break;
-    }
-    if (_ptrRenderer)
-    {
-        if (_ptrRenderer->Init() == -1)
-        {
-        }
-    }
-}
-
-ModuleVideoRenderImpl::~ModuleVideoRenderImpl()
-{
-    delete &_moduleCrit;
-
-    for (IncomingVideoStreamMap::iterator it = _streamRenderMap.begin();
-         it != _streamRenderMap.end();
-         ++it) {
-      delete it->second;
-    }
-
-    // Delete platform specific renderer
-    if (_ptrRenderer)
-    {
-        VideoRenderType videoRenderType = _ptrRenderer->RenderType();
-
-        switch (videoRenderType)
-        {
-            case kRenderExternal:
-            {
-                VideoRenderExternalImpl
-                        * ptrRenderer =
-                                reinterpret_cast<VideoRenderExternalImpl*> (_ptrRenderer);
-                _ptrRenderer = NULL;
-                delete ptrRenderer;
-            }
-            break;
-
-            default:
-                // Error...
-                break;
-        }
-    }
-}
-
-int64_t ModuleVideoRenderImpl::TimeUntilNextProcess()
-{
-    // Not used
-    return 50;
-}
-void ModuleVideoRenderImpl::Process() {}
-
-void*
-ModuleVideoRenderImpl::Window()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-    return _ptrWindow;
-}
-
-int32_t ModuleVideoRenderImpl::ChangeWindow(void* window)
-{
-    return -1;
-}
-
-int32_t ModuleVideoRenderImpl::Id()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-    return _id;
-}
-
-uint32_t ModuleVideoRenderImpl::GetIncomingFrameRate(const uint32_t streamId) {
-  CriticalSectionScoped cs(&_moduleCrit);
-
-  IncomingVideoStreamMap::iterator it = _streamRenderMap.find(streamId);
-
-  if (it == _streamRenderMap.end()) {
-    // This stream doesn't exist
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: stream doesn't exist",
-                 __FUNCTION__);
-    return 0;
-  }
-  assert(it->second != NULL);
-  return it->second->IncomingRate();
-}
-
-VideoRenderCallback*
-ModuleVideoRenderImpl::AddIncomingRenderStream(const uint32_t streamId,
-                                               const uint32_t zOrder,
-                                               const float left,
-                                               const float top,
-                                               const float right,
-                                               const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return NULL;
-    }
-
-    if (_streamRenderMap.find(streamId) != _streamRenderMap.end()) {
-        // The stream already exists...
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream already exists", __FUNCTION__);
-        return NULL;
-    }
-
-    VideoRenderCallback* ptrRenderCallback =
-            _ptrRenderer->AddIncomingRenderStream(streamId, zOrder, left, top,
-                                                  right, bottom);
-    if (ptrRenderCallback == NULL)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Can't create incoming stream in renderer",
-                     __FUNCTION__);
-        return NULL;
-    }
-
-    // Create platform independant code
-    IncomingVideoStream* ptrIncomingStream =
-        new IncomingVideoStream(streamId, false);
-    ptrIncomingStream->SetRenderCallback(ptrRenderCallback);
-    VideoRenderCallback* moduleCallback = ptrIncomingStream->ModuleCallback();
-
-    // Store the stream
-    _streamRenderMap[streamId] = ptrIncomingStream;
-
-    return moduleCallback;
-}
-
-int32_t ModuleVideoRenderImpl::DeleteIncomingRenderStream(
-                                                                const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-    if (item == _streamRenderMap.end())
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-
-    delete item->second;
-
-    _ptrRenderer->DeleteIncomingRenderStream(streamId);
-
-    _streamRenderMap.erase(item);
-
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::AddExternalRenderCallback(
-    const uint32_t streamId,
-    VideoRenderCallback* renderObject) {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-
-    if (item == _streamRenderMap.end())
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-
-    if (item->second == NULL) {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: could not get stream", __FUNCTION__);
-        return -1;
-    }
-    item->second->SetExternalCallback(renderObject);
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::GetIncomingRenderStreamProperties(
-    const uint32_t streamId,
-    uint32_t& zOrder,
-    float& left,
-    float& top,
-    float& right,
-    float& bottom) const {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    return _ptrRenderer->GetIncomingRenderStreamProperties(streamId, zOrder,
-                                                           left, top, right,
-                                                           bottom);
-}
-
-uint32_t ModuleVideoRenderImpl::GetNumIncomingRenderStreams() const
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    return static_cast<uint32_t>(_streamRenderMap.size());
-}
-
-bool ModuleVideoRenderImpl::HasIncomingRenderStream(
-    const uint32_t streamId) const {
-  CriticalSectionScoped cs(&_moduleCrit);
-
-  return _streamRenderMap.find(streamId) != _streamRenderMap.end();
-}
-
-int32_t ModuleVideoRenderImpl::RegisterRawFrameCallback(
-    const uint32_t streamId,
-    VideoRenderCallback* callbackObj) {
-  return -1;
-}
-
-int32_t ModuleVideoRenderImpl::StartRender(const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    // Start the stream
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-
-    if (item == _streamRenderMap.end())
-    {
-        return -1;
-    }
-
-    if (item->second->Start() == -1)
-    {
-        return -1;
-    }
-
-    // Start the HW renderer
-    if (_ptrRenderer->StartRender() == -1)
-    {
-        return -1;
-    }
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::StopRender(const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s(%d): No renderer", __FUNCTION__, streamId);
-        return -1;
-    }
-
-    // Stop the incoming stream
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-
-    if (item == _streamRenderMap.end())
-    {
-        return -1;
-    }
-
-    if (item->second->Stop() == -1)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::ResetRender()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    int32_t ret = 0;
-    // Loop through all incoming streams and reset them
-    for (IncomingVideoStreamMap::iterator it = _streamRenderMap.begin();
-         it != _streamRenderMap.end();
-         ++it) {
-      if (it->second->Reset() == -1)
-        ret = -1;
-    }
-    return ret;
-}
-
-RawVideoType ModuleVideoRenderImpl::PreferredVideoType() const
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (_ptrRenderer == NULL)
-    {
-        return kVideoI420;
-    }
-
-    return _ptrRenderer->PerferedVideoType();
-}
-
-bool ModuleVideoRenderImpl::IsFullScreen()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->FullScreen();
-}
-
-int32_t ModuleVideoRenderImpl::GetScreenResolution(
-                                                         uint32_t& screenWidth,
-                                                         uint32_t& screenHeight) const
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->GetScreenResolution(screenWidth, screenHeight);
-}
-
-uint32_t ModuleVideoRenderImpl::RenderFrameRate(
-                                                      const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->RenderFrameRate(streamId);
-}
-
-int32_t ModuleVideoRenderImpl::SetStreamCropping(
-                                                       const uint32_t streamId,
-                                                       const float left,
-                                                       const float top,
-                                                       const float right,
-                                                       const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->SetStreamCropping(streamId, left, top, right, bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetTransparentBackground(const bool enable)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->SetTransparentBackground(enable);
-}
-
-int32_t ModuleVideoRenderImpl::FullScreenRender(void* window, const bool enable)
-{
-    return -1;
-}
-
-int32_t ModuleVideoRenderImpl::SetText(
-                                             const uint8_t textId,
-                                             const uint8_t* text,
-                                             const int32_t textLength,
-                                             const uint32_t textColorRef,
-                                             const uint32_t backgroundColorRef,
-                                             const float left, const float top,
-                                             const float right,
-                                             const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-    return _ptrRenderer->SetText(textId, text, textLength, textColorRef,
-                                 backgroundColorRef, left, top, right, bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetBitmap(const void* bitMap,
-                                         const uint8_t pictureId,
-                                         const void* colorKey,
-                                         const float left,
-                                         const float top,
-                                         const float right,
-                                         const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-    return _ptrRenderer->SetBitmap(bitMap, pictureId, colorKey, left, top,
-                                   right, bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetExpectedRenderDelay(
-    uint32_t stream_id, int32_t delay_ms) {
-  CriticalSectionScoped cs(&_moduleCrit);
-
-  if (!_ptrRenderer) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: No renderer", __FUNCTION__);
-    return false;
-  }
-
-  IncomingVideoStreamMap::const_iterator item =
-      _streamRenderMap.find(stream_id);
-  if (item == _streamRenderMap.end()) {
-    // This stream doesn't exist
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s(%u, %d): stream doesn't exist", __FUNCTION__, stream_id,
-                 delay_ms);
-    return -1;
-  }
-
-  assert(item->second != NULL);
-  return item->second->SetExpectedRenderDelay(delay_ms);
-}
-
-int32_t ModuleVideoRenderImpl::ConfigureRenderer(
-                                                       const uint32_t streamId,
-                                                       const unsigned int zOrder,
-                                                       const float left,
-                                                       const float top,
-                                                       const float right,
-                                                       const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->ConfigureRenderer(streamId, zOrder, left, top, right,
-                                           bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetStartImage(const uint32_t streamId,
-                                             const VideoFrame& videoFrame) {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    IncomingVideoStreamMap::const_iterator item =
-        _streamRenderMap.find(streamId);
-    if (item == _streamRenderMap.end())
-    {
-        // This stream doesn't exist
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-    assert (item->second != NULL);
-    item->second->SetStartImage(videoFrame);
-    return 0;
-
-}
-
-int32_t ModuleVideoRenderImpl::SetTimeoutImage(const uint32_t streamId,
-                                               const VideoFrame& videoFrame,
-                                               const uint32_t timeout) {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    IncomingVideoStreamMap::const_iterator item =
-        _streamRenderMap.find(streamId);
-    if (item == _streamRenderMap.end())
-    {
-        // This stream doesn't exist
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-    assert(item->second != NULL);
-    item->second->SetTimeoutImage(videoFrame, timeout);
-    return 0;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_impl.h b/chromium/third_party/webrtc/modules/video_render/video_render_impl.h
deleted file mode 100644
index 12244a60b82..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/video_render_impl.h
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
-
-#include <map>
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/video_render.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class IncomingVideoStream;
-class IVideoRender;
-
-// Class definitions
-class ModuleVideoRenderImpl: public VideoRender
-{
-public:
-    /*
-     *   VideoRenderer constructor/destructor
-     */
-    ModuleVideoRenderImpl(const int32_t id,
-                          const VideoRenderType videoRenderType,
-                          void* window, const bool fullscreen);
-
-    virtual ~ModuleVideoRenderImpl();
-
-    virtual int64_t TimeUntilNextProcess();
-    virtual void Process();
-
-    /*
-     *   Returns the render window
-     */
-    virtual void* Window();
-
-    /*
-     *   Change render window
-     */
-    virtual int32_t ChangeWindow(void* window);
-
-    /*
-     *   Returns module id
-     */
-    int32_t Id();
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-
-    /*
-     *   Add incoming render stream
-     */
-    virtual VideoRenderCallback
-            * AddIncomingRenderStream(const uint32_t streamId,
-                                      const uint32_t zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-    /*
-     *   Delete incoming render stream
-     */
-    virtual int32_t
-            DeleteIncomingRenderStream(const uint32_t streamId);
-
-    /*
-     *   Add incoming render callback, used for external rendering
-     */
-    virtual int32_t
-            AddExternalRenderCallback(const uint32_t streamId,
-                                      VideoRenderCallback* renderObject);
-
-    /*
-     *   Get the porperties for an incoming render stream
-     */
-    virtual int32_t
-            GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                              uint32_t& zOrder,
-                                              float& left, float& top,
-                                              float& right, float& bottom) const;
-    /*
-     *   Incoming frame rate for the specified stream.
-     */
-    virtual uint32_t GetIncomingFrameRate(const uint32_t streamId);
-
-    /*
-     *   Returns the number of incoming streams added to this render module
-     */
-    virtual uint32_t GetNumIncomingRenderStreams() const;
-
-    /*
-     *   Returns true if this render module has the streamId added, false otherwise.
-     */
-    virtual bool HasIncomingRenderStream(const uint32_t streamId) const;
-
-    /*
-     *
-     */
-    virtual int32_t
-            RegisterRawFrameCallback(const uint32_t streamId,
-                                     VideoRenderCallback* callbackObj);
-
-    virtual int32_t SetExpectedRenderDelay(uint32_t stream_id,
-                                           int32_t delay_ms);
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    /*
-     *   Starts rendering the specified stream
-     */
-    virtual int32_t StartRender(const uint32_t streamId);
-
-    /*
-     *   Stops the renderer
-     */
-    virtual int32_t StopRender(const uint32_t streamId);
-
-    /*
-     *   Sets the renderer in start state, no streams removed.
-     */
-    virtual int32_t ResetRender();
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    /*
-     *   Returns the prefered render video type
-     */
-    virtual RawVideoType PreferredVideoType() const;
-
-    /*
-     *   Returns true if the renderer is in fullscreen mode, otherwise false.
-     */
-    virtual bool IsFullScreen();
-
-    /*
-     *   Gets screen resolution in pixels
-     */
-    virtual int32_t
-            GetScreenResolution(uint32_t& screenWidth,
-                                uint32_t& screenHeight) const;
-
-    /*
-     *   Get the actual render rate for this stream. I.e rendered frame rate,
-     *   not frames delivered to the renderer.
-     */
-    virtual uint32_t RenderFrameRate(const uint32_t streamId);
-
-    /*
-     *   Set cropping of incoming stream
-     */
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t SetTransparentBackground(const bool enable);
-
-    virtual int32_t FullScreenRender(void* window, const bool enable);
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left, const float top,
-                              const float right, const float bottom);
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t textColorRef,
-                            const uint32_t backgroundColorRef,
-                            const float left, const float top,
-                            const float right, const float bottom);
-
-    virtual int32_t SetStartImage(const uint32_t streamId,
-                                  const VideoFrame& videoFrame);
-
-    virtual int32_t SetTimeoutImage(const uint32_t streamId,
-                                    const VideoFrame& videoFrame,
-                                    const uint32_t timeout);
-
-private:
-    int32_t _id;
-    CriticalSectionWrapper& _moduleCrit;
-    void* _ptrWindow;
-    bool _fullScreen;
-
-    IVideoRender* _ptrRenderer;
-    typedef std::map<uint32_t, IncomingVideoStream*> IncomingVideoStreamMap;
-    IncomingVideoStreamMap _streamRenderMap;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_internal.h b/chromium/third_party/webrtc/modules/video_render/video_render_internal.h
deleted file mode 100644
index 0508c1a7087..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/video_render_internal.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_INTERNAL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_INTERNAL_H_
-
-#ifdef ANDROID
-#include <jni.h>
-
-namespace webrtc {
-
-// In order to be able to use the internal webrtc video render
-// for android, the jvm objects must be set via this method.
-int32_t SetRenderAndroidVM(JavaVM* javaVM);
-
-}  // namespace webrtc
-
-#endif  // ANDROID
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_VIDEO_RENDER_INTERNAL_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/video_render_internal_impl.cc b/chromium/third_party/webrtc/modules/video_render/video_render_internal_impl.cc
deleted file mode 100644
index 2090fce5f86..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/video_render_internal_impl.cc
+++ /dev/null
@@ -1,825 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-
-#include "webrtc/common_video/include/incoming_video_stream.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-#include "webrtc/modules/video_render/video_render_impl.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-#if defined (_WIN32)
-#include "webrtc/modules/video_render/windows/video_render_windows_impl.h"
-#define STANDARD_RENDERING kRenderWindows
-
-// WEBRTC_IOS should go before WEBRTC_MAC because WEBRTC_MAC
-// gets defined if WEBRTC_IOS is defined
-#elif defined(WEBRTC_IOS)
-#define STANDARD_RENDERING kRenderiOS
-#include "webrtc/modules/video_render/ios/video_render_ios_impl.h"
-#elif defined(WEBRTC_MAC)
-#if defined(COCOA_RENDERING)
-#define STANDARD_RENDERING kRenderCocoa
-#include "webrtc/modules/video_render/mac/video_render_mac_cocoa_impl.h"
-#elif defined(CARBON_RENDERING)
-#define STANDARD_RENDERING kRenderCarbon
-#include "webrtc/modules/video_render/mac/video_render_mac_carbon_impl.h"
-#endif
-
-#elif defined(WEBRTC_ANDROID)
-#include "webrtc/modules/video_render/android/video_render_android_impl.h"
-#include "webrtc/modules/video_render/android/video_render_android_native_opengl2.h"
-#include "webrtc/modules/video_render/android/video_render_android_surface_view.h"
-#define STANDARD_RENDERING kRenderAndroid
-
-#elif defined(WEBRTC_LINUX)
-#include "webrtc/modules/video_render/linux/video_render_linux_impl.h"
-#define STANDARD_RENDERING kRenderX11
-
-#else
-//Other platforms
-#endif
-
-// For external rendering
-#include "webrtc/modules/video_render/external/video_render_external_impl.h"
-#ifndef STANDARD_RENDERING
-#define STANDARD_RENDERING kRenderExternal
-#endif  // STANDARD_RENDERING
-
-namespace webrtc {
-
-VideoRender*
-VideoRender::CreateVideoRender(const int32_t id,
-                               void* window,
-                               const bool fullscreen,
-                               const VideoRenderType videoRenderType/*=kRenderDefault*/)
-{
-    VideoRenderType resultVideoRenderType = videoRenderType;
-    if (videoRenderType == kRenderDefault)
-    {
-        resultVideoRenderType = STANDARD_RENDERING;
-    }
-    return new ModuleVideoRenderImpl(id, resultVideoRenderType, window,
-                                     fullscreen);
-}
-
-void VideoRender::DestroyVideoRender(
-                                                         VideoRender* module)
-{
-    if (module)
-    {
-        delete module;
-    }
-}
-
-ModuleVideoRenderImpl::ModuleVideoRenderImpl(
-                                             const int32_t id,
-                                             const VideoRenderType videoRenderType,
-                                             void* window,
-                                             const bool fullscreen) :
-    _id(id), _moduleCrit(*CriticalSectionWrapper::CreateCriticalSection()),
-    _ptrWindow(window), _fullScreen(fullscreen), _ptrRenderer(NULL)
-{
-
-    // Create platform specific renderer
-    switch (videoRenderType)
-    {
-#if defined(_WIN32)
-        case kRenderWindows:
-        {
-            VideoRenderWindowsImpl* ptrRenderer;
-            ptrRenderer = new VideoRenderWindowsImpl(_id, videoRenderType, window, _fullScreen);
-            if (ptrRenderer)
-            {
-                _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
-            }
-        }
-        break;
-
-#elif defined(WEBRTC_IOS)
-        case kRenderiOS:
-        {
-            VideoRenderIosImpl* ptrRenderer = new VideoRenderIosImpl(_id, window, _fullScreen);
-            if(ptrRenderer)
-            {
-                _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
-            }
-        }
-        break;
-
-#elif defined(WEBRTC_MAC)
-
-#if defined(COCOA_RENDERING)
-        case kRenderCocoa:
-        {
-            VideoRenderMacCocoaImpl* ptrRenderer = new VideoRenderMacCocoaImpl(_id, videoRenderType, window, _fullScreen);
-            if(ptrRenderer)
-            {
-                _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
-            }
-        }
-
-        break;
-#elif defined(CARBON_RENDERING)
-        case kRenderCarbon:
-        {
-            VideoRenderMacCarbonImpl* ptrRenderer = new VideoRenderMacCarbonImpl(_id, videoRenderType, window, _fullScreen);
-            if(ptrRenderer)
-            {
-                _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
-            }
-        }
-        break;
-#endif
-
-#elif defined(WEBRTC_ANDROID)
-        case kRenderAndroid:
-        {
-            if(AndroidNativeOpenGl2Renderer::UseOpenGL2(window))
-            {
-                AndroidNativeOpenGl2Renderer* ptrRenderer = NULL;
-                ptrRenderer = new AndroidNativeOpenGl2Renderer(_id, videoRenderType, window, _fullScreen);
-                if (ptrRenderer)
-                {
-                    _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
-                }
-            }
-            else
-            {
-                AndroidSurfaceViewRenderer* ptrRenderer = NULL;
-                ptrRenderer = new AndroidSurfaceViewRenderer(_id, videoRenderType, window, _fullScreen);
-                if (ptrRenderer)
-                {
-                    _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
-                }
-            }
-
-        }
-        break;
-#elif defined(WEBRTC_LINUX)
-        case kRenderX11:
-        {
-            VideoRenderLinuxImpl* ptrRenderer = NULL;
-            ptrRenderer = new VideoRenderLinuxImpl(_id, videoRenderType, window, _fullScreen);
-            if ( ptrRenderer )
-            {
-                _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
-            }
-        }
-        break;
-
-#else
-        // Other platforms
-#endif
-        case kRenderExternal:
-        {
-            VideoRenderExternalImpl* ptrRenderer(NULL);
-            ptrRenderer = new VideoRenderExternalImpl(_id, videoRenderType,
-                                                      window, _fullScreen);
-            if (ptrRenderer)
-            {
-                _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
-            }
-        }
-            break;
-        default:
-            // Error...
-            break;
-    }
-    if (_ptrRenderer)
-    {
-        if (_ptrRenderer->Init() == -1)
-        {
-        }
-    }
-}
-
-ModuleVideoRenderImpl::~ModuleVideoRenderImpl()
-{
-    delete &_moduleCrit;
-
-    for (IncomingVideoStreamMap::iterator it = _streamRenderMap.begin();
-         it != _streamRenderMap.end();
-         ++it) {
-      delete it->second;
-    }
-
-    // Delete platform specific renderer
-    if (_ptrRenderer)
-    {
-        VideoRenderType videoRenderType = _ptrRenderer->RenderType();
-
-        switch (videoRenderType)
-        {
-            case kRenderExternal:
-            {
-                VideoRenderExternalImpl
-                        * ptrRenderer =
-                                reinterpret_cast<VideoRenderExternalImpl*> (_ptrRenderer);
-                _ptrRenderer = NULL;
-                delete ptrRenderer;
-            }
-            break;
-#if defined(_WIN32)
-            case kRenderWindows:
-            {
-                VideoRenderWindowsImpl* ptrRenderer = reinterpret_cast<VideoRenderWindowsImpl*>(_ptrRenderer);
-                _ptrRenderer = NULL;
-                delete ptrRenderer;
-            }
-            break;
-#elif defined(WEBRTC_IOS)
-            case kRenderiOS:
-            {
-              VideoRenderIosImpl* ptrRenderer = reinterpret_cast<VideoRenderIosImpl*> (_ptrRenderer);
-              _ptrRenderer = NULL;
-              delete ptrRenderer;
-            }
-            break;
-#elif defined(WEBRTC_MAC)
-
-#if defined(COCOA_RENDERING)
-            case kRenderCocoa:
-            {
-                VideoRenderMacCocoaImpl* ptrRenderer = reinterpret_cast<VideoRenderMacCocoaImpl*> (_ptrRenderer);
-                _ptrRenderer = NULL;
-                delete ptrRenderer;
-            }
-            break;
-#elif defined(CARBON_RENDERING)
-            case kRenderCarbon:
-            {
-                VideoRenderMacCarbonImpl* ptrRenderer = reinterpret_cast<VideoRenderMacCarbonImpl*> (_ptrRenderer);
-                _ptrRenderer = NULL;
-                delete ptrRenderer;
-            }
-            break;
-#endif
-
-#elif defined(WEBRTC_ANDROID)
-            case kRenderAndroid:
-            {
-                VideoRenderAndroid* ptrRenderer = reinterpret_cast<VideoRenderAndroid*> (_ptrRenderer);
-                _ptrRenderer = NULL;
-                delete ptrRenderer;
-            }
-            break;
-
-#elif defined(WEBRTC_LINUX)
-            case kRenderX11:
-            {
-                VideoRenderLinuxImpl* ptrRenderer = reinterpret_cast<VideoRenderLinuxImpl*> (_ptrRenderer);
-                _ptrRenderer = NULL;
-                delete ptrRenderer;
-            }
-            break;
-#else
-            //other platforms
-#endif
-
-            default:
-                // Error...
-                break;
-        }
-    }
-}
-
-int64_t ModuleVideoRenderImpl::TimeUntilNextProcess()
-{
-    // Not used
-    return 50;
-}
-void ModuleVideoRenderImpl::Process() {}
-
-void*
-ModuleVideoRenderImpl::Window()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-    return _ptrWindow;
-}
-
-int32_t ModuleVideoRenderImpl::ChangeWindow(void* window)
-{
-
-    CriticalSectionScoped cs(&_moduleCrit);
-
-#if defined(WEBRTC_IOS) // WEBRTC_IOS must go before WEBRTC_MAC
-    _ptrRenderer = NULL;
-    delete _ptrRenderer;
-
-    VideoRenderIosImpl* ptrRenderer;
-    ptrRenderer = new VideoRenderIosImpl(_id, window, _fullScreen);
-    if (!ptrRenderer)
-    {
-        return -1;
-    }
-    _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
-    return _ptrRenderer->ChangeWindow(window);
-#elif defined(WEBRTC_MAC)
-
-    _ptrRenderer = NULL;
-    delete _ptrRenderer;
-
-#if defined(COCOA_RENDERING)
-    VideoRenderMacCocoaImpl* ptrRenderer;
-    ptrRenderer = new VideoRenderMacCocoaImpl(_id, kRenderCocoa, window, _fullScreen);
-#elif defined(CARBON_RENDERING)
-    VideoRenderMacCarbonImpl* ptrRenderer;
-    ptrRenderer = new VideoRenderMacCarbonImpl(_id, kRenderCarbon, window, _fullScreen);
-#endif
-    if (!ptrRenderer)
-    {
-        return -1;
-    }
-    _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
-    return _ptrRenderer->ChangeWindow(window);
-
-#else
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-    return _ptrRenderer->ChangeWindow(window);
-
-#endif
-}
-
-int32_t ModuleVideoRenderImpl::Id()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-    return _id;
-}
-
-uint32_t ModuleVideoRenderImpl::GetIncomingFrameRate(const uint32_t streamId) {
-  CriticalSectionScoped cs(&_moduleCrit);
-
-  IncomingVideoStreamMap::iterator it = _streamRenderMap.find(streamId);
-
-  if (it == _streamRenderMap.end()) {
-    // This stream doesn't exist
-    WEBRTC_TRACE(kTraceError,
-                 kTraceVideoRenderer,
-                 _id,
-                 "%s: stream doesn't exist",
-                 __FUNCTION__);
-    return 0;
-  }
-  assert(it->second != NULL);
-  return it->second->IncomingRate();
-}
-
-VideoRenderCallback*
-ModuleVideoRenderImpl::AddIncomingRenderStream(const uint32_t streamId,
-                                               const uint32_t zOrder,
-                                               const float left,
-                                               const float top,
-                                               const float right,
-                                               const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return NULL;
-    }
-
-    if (_streamRenderMap.find(streamId) != _streamRenderMap.end()) {
-        // The stream already exists...
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream already exists", __FUNCTION__);
-        return NULL;
-    }
-
-    VideoRenderCallback* ptrRenderCallback =
-            _ptrRenderer->AddIncomingRenderStream(streamId, zOrder, left, top,
-                                                  right, bottom);
-    if (ptrRenderCallback == NULL)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: Can't create incoming stream in renderer",
-                     __FUNCTION__);
-        return NULL;
-    }
-
-    // Create platform independant code
-    IncomingVideoStream* ptrIncomingStream =
-        new IncomingVideoStream(streamId, false);
-    ptrIncomingStream->SetRenderCallback(ptrRenderCallback);
-    VideoRenderCallback* moduleCallback = ptrIncomingStream->ModuleCallback();
-
-    // Store the stream
-    _streamRenderMap[streamId] = ptrIncomingStream;
-
-    return moduleCallback;
-}
-
-int32_t ModuleVideoRenderImpl::DeleteIncomingRenderStream(
-                                                                const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-    if (item == _streamRenderMap.end())
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-
-    delete item->second;
-
-    _ptrRenderer->DeleteIncomingRenderStream(streamId);
-
-    _streamRenderMap.erase(item);
-
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::AddExternalRenderCallback(
-    const uint32_t streamId,
-    VideoRenderCallback* renderObject) {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-
-    if (item == _streamRenderMap.end())
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-
-    if (item->second == NULL) {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: could not get stream", __FUNCTION__);
-        return -1;
-    }
-    item->second->SetExternalCallback(renderObject);
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::GetIncomingRenderStreamProperties(
-    const uint32_t streamId,
-    uint32_t& zOrder,
-    float& left,
-    float& top,
-    float& right,
-    float& bottom) const {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    return _ptrRenderer->GetIncomingRenderStreamProperties(streamId, zOrder,
-                                                           left, top, right,
-                                                           bottom);
-}
-
-uint32_t ModuleVideoRenderImpl::GetNumIncomingRenderStreams() const
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    return static_cast<uint32_t>(_streamRenderMap.size());
-}
-
-bool ModuleVideoRenderImpl::HasIncomingRenderStream(
-    const uint32_t streamId) const {
-  CriticalSectionScoped cs(&_moduleCrit);
-
-  return _streamRenderMap.find(streamId) != _streamRenderMap.end();
-}
-
-int32_t ModuleVideoRenderImpl::RegisterRawFrameCallback(
-    const uint32_t streamId,
-    VideoRenderCallback* callbackObj) {
-  return -1;
-}
-
-int32_t ModuleVideoRenderImpl::StartRender(const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    // Start the stream
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-
-    if (item == _streamRenderMap.end())
-    {
-        return -1;
-    }
-
-    if (item->second->Start() == -1)
-    {
-        return -1;
-    }
-
-    // Start the HW renderer
-    if (_ptrRenderer->StartRender() == -1)
-    {
-        return -1;
-    }
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::StopRender(const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s(%d): No renderer", __FUNCTION__, streamId);
-        return -1;
-    }
-
-    // Stop the incoming stream
-    IncomingVideoStreamMap::iterator item = _streamRenderMap.find(streamId);
-
-    if (item == _streamRenderMap.end())
-    {
-        return -1;
-    }
-
-    if (item->second->Stop() == -1)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t ModuleVideoRenderImpl::ResetRender()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    int32_t ret = 0;
-    // Loop through all incoming streams and reset them
-    for (IncomingVideoStreamMap::iterator it = _streamRenderMap.begin();
-         it != _streamRenderMap.end();
-         ++it) {
-      if (it->second->Reset() == -1)
-        ret = -1;
-    }
-    return ret;
-}
-
-RawVideoType ModuleVideoRenderImpl::PreferredVideoType() const
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (_ptrRenderer == NULL)
-    {
-        return kVideoI420;
-    }
-
-    return _ptrRenderer->PerferedVideoType();
-}
-
-bool ModuleVideoRenderImpl::IsFullScreen()
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->FullScreen();
-}
-
-int32_t ModuleVideoRenderImpl::GetScreenResolution(
-                                                         uint32_t& screenWidth,
-                                                         uint32_t& screenHeight) const
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->GetScreenResolution(screenWidth, screenHeight);
-}
-
-uint32_t ModuleVideoRenderImpl::RenderFrameRate(
-                                                      const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->RenderFrameRate(streamId);
-}
-
-int32_t ModuleVideoRenderImpl::SetStreamCropping(
-                                                       const uint32_t streamId,
-                                                       const float left,
-                                                       const float top,
-                                                       const float right,
-                                                       const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->SetStreamCropping(streamId, left, top, right, bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetTransparentBackground(const bool enable)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->SetTransparentBackground(enable);
-}
-
-int32_t ModuleVideoRenderImpl::FullScreenRender(void* window, const bool enable)
-{
-    return -1;
-}
-
-int32_t ModuleVideoRenderImpl::SetText(
-                                             const uint8_t textId,
-                                             const uint8_t* text,
-                                             const int32_t textLength,
-                                             const uint32_t textColorRef,
-                                             const uint32_t backgroundColorRef,
-                                             const float left, const float top,
-                                             const float right,
-                                             const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-    return _ptrRenderer->SetText(textId, text, textLength, textColorRef,
-                                 backgroundColorRef, left, top, right, bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetBitmap(const void* bitMap,
-                                         const uint8_t pictureId,
-                                         const void* colorKey,
-                                         const float left,
-                                         const float top,
-                                         const float right,
-                                         const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-    return _ptrRenderer->SetBitmap(bitMap, pictureId, colorKey, left, top,
-                                   right, bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetExpectedRenderDelay(
-    uint32_t stream_id, int32_t delay_ms) {
-  CriticalSectionScoped cs(&_moduleCrit);
-
-  if (!_ptrRenderer) {
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s: No renderer", __FUNCTION__);
-    return false;
-  }
-
-  IncomingVideoStreamMap::const_iterator item =
-      _streamRenderMap.find(stream_id);
-  if (item == _streamRenderMap.end()) {
-    // This stream doesn't exist
-    WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                 "%s(%u, %d): stream doesn't exist", __FUNCTION__, stream_id,
-                 delay_ms);
-    return -1;
-  }
-
-  assert(item->second != NULL);
-  return item->second->SetExpectedRenderDelay(delay_ms);
-}
-
-int32_t ModuleVideoRenderImpl::ConfigureRenderer(
-                                                       const uint32_t streamId,
-                                                       const unsigned int zOrder,
-                                                       const float left,
-                                                       const float top,
-                                                       const float right,
-                                                       const float bottom)
-{
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return false;
-    }
-    return _ptrRenderer->ConfigureRenderer(streamId, zOrder, left, top, right,
-                                           bottom);
-}
-
-int32_t ModuleVideoRenderImpl::SetStartImage(const uint32_t streamId,
-                                             const VideoFrame& videoFrame) {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    IncomingVideoStreamMap::const_iterator item =
-        _streamRenderMap.find(streamId);
-    if (item == _streamRenderMap.end())
-    {
-        // This stream doesn't exist
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-    assert (item->second != NULL);
-    item->second->SetStartImage(videoFrame);
-    return 0;
-
-}
-
-int32_t ModuleVideoRenderImpl::SetTimeoutImage(const uint32_t streamId,
-                                               const VideoFrame& videoFrame,
-                                               const uint32_t timeout) {
-    CriticalSectionScoped cs(&_moduleCrit);
-
-    if (!_ptrRenderer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: No renderer", __FUNCTION__);
-        return -1;
-    }
-
-    IncomingVideoStreamMap::const_iterator item =
-        _streamRenderMap.find(streamId);
-    if (item == _streamRenderMap.end())
-    {
-        // This stream doesn't exist
-        WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
-                     "%s: stream doesn't exist", __FUNCTION__);
-        return -1;
-    }
-    assert(item->second != NULL);
-    item->second->SetTimeoutImage(videoFrame, timeout);
-    return 0;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/windows/i_video_render_win.h b/chromium/third_party/webrtc/modules/video_render/windows/i_video_render_win.h
deleted file mode 100644
index 6dbb4fd3cb9..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/windows/i_video_render_win.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
-
-#include "webrtc/modules/video_render/video_render.h"
-
-namespace webrtc {
-
-// Class definitions
-class IVideoRenderWin
-{
-public:
-    /**************************************************************************
-     *
-     *   Constructor/destructor
-     *
-     ***************************************************************************/
-    virtual ~IVideoRenderWin()
-    {
-    };
-
-    virtual int32_t Init() = 0;
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderCallback
-            * CreateChannel(const uint32_t streamId,
-                            const uint32_t zOrder,
-                            const float left,
-                            const float top,
-                            const float right,
-                            const float bottom) = 0;
-
-    virtual int32_t DeleteChannel(const uint32_t streamId) = 0;
-
-    virtual int32_t GetStreamSettings(const uint32_t channel,
-                                      const uint16_t streamId,
-                                      uint32_t& zOrder,
-                                      float& left, float& top,
-                                      float& right, float& bottom) = 0;
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender() = 0;
-
-    virtual int32_t StopRender() = 0;
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    virtual bool IsFullScreen() = 0;
-
-    virtual int32_t SetCropping(const uint32_t channel,
-                                const uint16_t streamId,
-                                const float left, const float top,
-                                const float right, const float bottom) = 0;
-
-    virtual int32_t ConfigureRenderer(const uint32_t channel,
-                                      const uint16_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left,
-                                      const float top,
-                                      const float right,
-                                      const float bottom) = 0;
-
-    virtual int32_t SetTransparentBackground(const bool enable) = 0;
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t colorText,
-                            const uint32_t colorBg,
-                            const float left, const float top,
-                            const float rigth, const float bottom) = 0;
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left, const float top,
-                              const float right, const float bottom) = 0;
-
-    virtual int32_t ChangeWindow(void* window) = 0;
-
-    virtual int32_t GetGraphicsMemory(uint64_t& totalMemory,
-                                      uint64_t& availableMemory) = 0;
-
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/windows/video_render_direct3d9.cc b/chromium/third_party/webrtc/modules/video_render/windows/video_render_direct3d9.cc
deleted file mode 100644
index b59b944e483..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/windows/video_render_direct3d9.cc
+++ /dev/null
@@ -1,1160 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Own include file
-#include "webrtc/modules/video_render/windows/video_render_direct3d9.h"
-
-// System include files
-#include <windows.h>
-
-// WebRtc include files
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-// A structure for our custom vertex type
-struct CUSTOMVERTEX
-{
-    FLOAT x, y, z;
-    DWORD color; // The vertex color
-    FLOAT u, v;
-};
-
-// Our custom FVF, which describes our custom vertex structure
-#define D3DFVF_CUSTOMVERTEX (D3DFVF_XYZ|D3DFVF_DIFFUSE|D3DFVF_TEX1)
-
-/*
- *
- *    D3D9Channel
- *
- */
-D3D9Channel::D3D9Channel(LPDIRECT3DDEVICE9 pd3DDevice,
-                                 CriticalSectionWrapper* critSect,
-                                 Trace* trace) :
-    _width(0),
-    _height(0),
-    _pd3dDevice(pd3DDevice),
-    _pTexture(NULL),
-    _bufferIsUpdated(false),
-    _critSect(critSect),
-    _streamId(0),
-    _zOrder(0),
-    _startWidth(0),
-    _startHeight(0),
-    _stopWidth(0),
-    _stopHeight(0)
-{
-
-}
-
-D3D9Channel::~D3D9Channel()
-{
-    //release the texture
-    if (_pTexture != NULL)
-    {
-        _pTexture->Release();
-        _pTexture = NULL;
-    }
-}
-
-void D3D9Channel::SetStreamSettings(uint16_t streamId,
-                                        uint32_t zOrder,
-                                        float startWidth,
-                                        float startHeight,
-                                        float stopWidth,
-                                        float stopHeight)
-{
-    _streamId = streamId;
-    _zOrder = zOrder;
-    _startWidth = startWidth;
-    _startHeight = startHeight;
-    _stopWidth = stopWidth;
-    _stopHeight = stopHeight;
-}
-
-int D3D9Channel::GetStreamSettings(uint16_t streamId,
-                                       uint32_t& zOrder,
-                                       float& startWidth,
-                                       float& startHeight,
-                                       float& stopWidth,
-                                       float& stopHeight)
-{
-    streamId = _streamId;
-    zOrder = _zOrder;
-    startWidth = _startWidth;
-    startHeight = _startHeight;
-    stopWidth = _stopWidth;
-    stopHeight = _stopHeight;
-    return 0;
-}
-
-int D3D9Channel::GetTextureWidth()
-{
-    return _width;
-}
-
-int D3D9Channel::GetTextureHeight()
-{
-    return _height;
-}
-
-// Called from video engine when a the frame size changed
-int D3D9Channel::FrameSizeChange(int width, int height, int numberOfStreams)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
-                 "FrameSizeChange, wifth: %d, height: %d, streams: %d", width,
-                 height, numberOfStreams);
-
-    CriticalSectionScoped cs(_critSect);
-    _width = width;
-    _height = height;
-
-    //clean the previous texture
-    if (_pTexture != NULL)
-    {
-        _pTexture->Release();
-        _pTexture = NULL;
-    }
-
-    HRESULT ret = E_POINTER;
-
-    if (_pd3dDevice)
-      ret = _pd3dDevice->CreateTexture(_width, _height, 1, 0, D3DFMT_A8R8G8B8,
-                                       D3DPOOL_MANAGED, &_pTexture, NULL);
-
-    if (FAILED(ret))
-    {
-        _pTexture = NULL;
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t D3D9Channel::RenderFrame(const uint32_t streamId,
-                                 const VideoFrame& videoFrame) {
-    CriticalSectionScoped cs(_critSect);
-    if (_width != videoFrame.width() || _height != videoFrame.height())
-    {
-        if (FrameSizeChange(videoFrame.width(), videoFrame.height(), 1) == -1)
-        {
-            return -1;
-        }
-    }
-    return DeliverFrame(videoFrame);
-}
-
-// Called from video engine when a new frame should be rendered.
-int D3D9Channel::DeliverFrame(const VideoFrame& videoFrame) {
-  WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
-               "DeliverFrame to D3D9Channel");
-
-  CriticalSectionScoped cs(_critSect);
-
-  // FIXME if _bufferIsUpdated is still true (not be renderred), do we want to
-  // update the texture? probably not
-  if (_bufferIsUpdated) {
-    WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
-                 "Last frame hasn't been rendered yet. Drop this frame.");
-    return -1;
-  }
-
-  if (!_pd3dDevice) {
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                 "D3D for rendering not initialized.");
-    return -1;
-  }
-
-  if (!_pTexture) {
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                 "Texture for rendering not initialized.");
-    return -1;
-  }
-
-  D3DLOCKED_RECT lr;
-
-  if (FAILED(_pTexture->LockRect(0, &lr, NULL, 0))) {
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                 "Failed to lock a texture in D3D9 Channel.");
-    return -1;
-  }
-  UCHAR* pRect = (UCHAR*) lr.pBits;
-
-  ConvertFromI420(videoFrame, kARGB, 0, pRect);
-
-  if (FAILED(_pTexture->UnlockRect(0))) {
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                 "Failed to unlock a texture in D3D9 Channel.");
-    return -1;
-  }
-
-  _bufferIsUpdated = true;
-  return 0;
-}
-
-// Called by d3d channel owner to indicate the frame/texture has been rendered off
-int D3D9Channel::RenderOffFrame()
-{
-    WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
-                 "Frame has been rendered to the screen.");
-    CriticalSectionScoped cs(_critSect);
-    _bufferIsUpdated = false;
-    return 0;
-}
-
-// Called by d3d channel owner to check if the texture is updated
-int D3D9Channel::IsUpdated(bool& isUpdated)
-{
-    CriticalSectionScoped cs(_critSect);
-    isUpdated = _bufferIsUpdated;
-    return 0;
-}
-
-// Called by d3d channel owner to get the texture
-LPDIRECT3DTEXTURE9 D3D9Channel::GetTexture()
-{
-    CriticalSectionScoped cs(_critSect);
-    return _pTexture;
-}
-
-int D3D9Channel::ReleaseTexture()
-{
-    CriticalSectionScoped cs(_critSect);
-
-    //release the texture
-    if (_pTexture != NULL)
-    {
-        _pTexture->Release();
-        _pTexture = NULL;
-    }
-    _pd3dDevice = NULL;
-    return 0;
-}
-
-int D3D9Channel::RecreateTexture(LPDIRECT3DDEVICE9 pd3DDevice)
-{
-    CriticalSectionScoped cs(_critSect);
-
-    _pd3dDevice = pd3DDevice;
-
-    if (_pTexture != NULL)
-    {
-        _pTexture->Release();
-        _pTexture = NULL;
-    }
-
-    HRESULT ret;
-
-    ret = _pd3dDevice->CreateTexture(_width, _height, 1, 0, D3DFMT_A8R8G8B8,
-                                     D3DPOOL_MANAGED, &_pTexture, NULL);
-
-    if (FAILED(ret))
-    {
-        _pTexture = NULL;
-        return -1;
-    }
-
-    return 0;
-}
-
-/*
- *
- *    VideoRenderDirect3D9
- *
- */
-VideoRenderDirect3D9::VideoRenderDirect3D9(Trace* trace,
-                                                   HWND hWnd,
-                                                   bool fullScreen) :
-    _refD3DCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _trace(trace),
-    _hWnd(hWnd),
-    _fullScreen(fullScreen),
-    _pTextureLogo(NULL),
-    _pVB(NULL),
-    _pd3dDevice(NULL),
-    _pD3D(NULL),
-    _d3dChannels(),
-    _d3dZorder(),
-    _screenUpdateEvent(NULL),
-    _logoLeft(0),
-    _logoTop(0),
-    _logoRight(0),
-    _logoBottom(0),
-    _pd3dSurface(NULL),
-    _totalMemory(0),
-    _availableMemory(0)
-{
-    _screenUpdateThread.reset(new rtc::PlatformThread(
-        ScreenUpdateThreadProc, this, "ScreenUpdateThread"));
-    _screenUpdateEvent = EventTimerWrapper::Create();
-    SetRect(&_originalHwndRect, 0, 0, 0, 0);
-}
-
-VideoRenderDirect3D9::~VideoRenderDirect3D9()
-{
-    //NOTE: we should not enter CriticalSection in here!
-
-    // Signal event to exit thread, then delete it
-    rtc::PlatformThread* tmpPtr = _screenUpdateThread.release();
-    if (tmpPtr)
-    {
-        _screenUpdateEvent->Set();
-        _screenUpdateEvent->StopTimer();
-
-        tmpPtr->Stop();
-        delete tmpPtr;
-    }
-    delete _screenUpdateEvent;
-
-    //close d3d device
-    CloseDevice();
-
-    // Delete all channels
-    std::map<int, D3D9Channel*>::iterator it = _d3dChannels.begin();
-    while (it != _d3dChannels.end())
-    {
-        delete it->second;
-        it = _d3dChannels.erase(it);
-    }
-    // Clean the zOrder map
-    _d3dZorder.clear();
-
-    if (_fullScreen)
-    {
-        // restore hwnd to original size and position
-        ::SetWindowPos(_hWnd, HWND_NOTOPMOST, _originalHwndRect.left,
-                       _originalHwndRect.top, _originalHwndRect.right
-                               - _originalHwndRect.left,
-                       _originalHwndRect.bottom - _originalHwndRect.top,
-                       SWP_FRAMECHANGED);
-        ::RedrawWindow(_hWnd, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
-                | RDW_ERASE);
-        ::RedrawWindow(NULL, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
-                | RDW_ERASE);
-    }
-
-    delete &_refD3DCritsect;
-}
-
-DWORD VideoRenderDirect3D9::GetVertexProcessingCaps()
-{
-    D3DCAPS9 caps;
-    DWORD dwVertexProcessing = D3DCREATE_SOFTWARE_VERTEXPROCESSING;
-    if (SUCCEEDED(_pD3D->GetDeviceCaps(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL,
-                                       &caps)))
-    {
-        if ((caps.DevCaps & D3DDEVCAPS_HWTRANSFORMANDLIGHT)
-                == D3DDEVCAPS_HWTRANSFORMANDLIGHT)
-        {
-            dwVertexProcessing = D3DCREATE_HARDWARE_VERTEXPROCESSING;
-        }
-    }
-    return dwVertexProcessing;
-}
-
-int VideoRenderDirect3D9::InitializeD3D(HWND hWnd,
-                                            D3DPRESENT_PARAMETERS* pd3dpp)
-{
-    // initialize Direct3D
-    if (NULL == (_pD3D = Direct3DCreate9(D3D_SDK_VERSION)))
-    {
-        return -1;
-    }
-
-    // determine what type of vertex processing to use based on the device capabilities
-    DWORD dwVertexProcessing = GetVertexProcessingCaps();
-
-    // get the display mode
-    D3DDISPLAYMODE d3ddm;
-    _pD3D->GetAdapterDisplayMode(D3DADAPTER_DEFAULT, &d3ddm);
-    pd3dpp->BackBufferFormat = d3ddm.Format;
-
-    // create the D3D device
-    if (FAILED(_pD3D->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL, hWnd,
-                                   dwVertexProcessing | D3DCREATE_MULTITHREADED
-                                           | D3DCREATE_FPU_PRESERVE, pd3dpp,
-                                   &_pd3dDevice)))
-    {
-        //try the ref device
-        if (FAILED(_pD3D->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_REF,
-                                       hWnd, dwVertexProcessing
-                                               | D3DCREATE_MULTITHREADED
-                                               | D3DCREATE_FPU_PRESERVE,
-                                       pd3dpp, &_pd3dDevice)))
-        {
-            return -1;
-        }
-    }
-
-    return 0;
-}
-
-int VideoRenderDirect3D9::ResetDevice()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
-                 "VideoRenderDirect3D9::ResetDevice");
-
-    CriticalSectionScoped cs(&_refD3DCritsect);
-
-    //release the channel texture
-    std::map<int, D3D9Channel*>::iterator it;
-    it = _d3dChannels.begin();
-    while (it != _d3dChannels.end())
-    {
-        if (it->second)
-        {
-            it->second->ReleaseTexture();
-        }
-        it++;
-    }
-
-    //close d3d device
-    if (CloseDevice() != 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "VideoRenderDirect3D9::ResetDevice failed to CloseDevice");
-        return -1;
-    }
-
-    //reinit d3d device
-    if (InitDevice() != 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "VideoRenderDirect3D9::ResetDevice failed to InitDevice");
-        return -1;
-    }
-
-    //recreate channel texture
-    it = _d3dChannels.begin();
-    while (it != _d3dChannels.end())
-    {
-        if (it->second)
-        {
-            it->second->RecreateTexture(_pd3dDevice);
-        }
-        it++;
-    }
-
-    return 0;
-}
-
-int VideoRenderDirect3D9::InitDevice()
-{
-    // Set up the structure used to create the D3DDevice
-    ZeroMemory(&_d3dpp, sizeof(_d3dpp));
-    _d3dpp.SwapEffect = D3DSWAPEFFECT_DISCARD;
-    _d3dpp.BackBufferFormat = D3DFMT_A8R8G8B8;
-    if (GetWindowRect(_hWnd, &_originalHwndRect) == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "VideoRenderDirect3D9::InitDevice Could not get window size");
-        return -1;
-    }
-    if (!_fullScreen)
-    {
-        _winWidth = _originalHwndRect.right - _originalHwndRect.left;
-        _winHeight = _originalHwndRect.bottom - _originalHwndRect.top;
-        _d3dpp.Windowed = TRUE;
-        _d3dpp.BackBufferHeight = 0;
-        _d3dpp.BackBufferWidth = 0;
-    }
-    else
-    {
-        _winWidth = (LONG) ::GetSystemMetrics(SM_CXSCREEN);
-        _winHeight = (LONG) ::GetSystemMetrics(SM_CYSCREEN);
-        _d3dpp.Windowed = FALSE;
-        _d3dpp.BackBufferWidth = _winWidth;
-        _d3dpp.BackBufferHeight = _winHeight;
-        _d3dpp.PresentationInterval = D3DPRESENT_INTERVAL_IMMEDIATE;
-    }
-
-    if (InitializeD3D(_hWnd, &_d3dpp) == -1)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "VideoRenderDirect3D9::InitDevice failed in InitializeD3D");
-        return -1;
-    }
-
-    // Turn off culling, so we see the front and back of the triangle
-    _pd3dDevice->SetRenderState(D3DRS_CULLMODE, D3DCULL_NONE);
-
-    // Turn off D3D lighting, since we are providing our own vertex colors
-    _pd3dDevice->SetRenderState(D3DRS_LIGHTING, FALSE);
-
-    // Settings for alpha blending
-    _pd3dDevice->SetRenderState(D3DRS_ALPHABLENDENABLE, TRUE);
-    _pd3dDevice->SetRenderState(D3DRS_SRCBLEND, D3DBLEND_SRCALPHA);
-    _pd3dDevice->SetRenderState(D3DRS_DESTBLEND, D3DBLEND_INVSRCALPHA);
-
-    _pd3dDevice->SetSamplerState( 0, D3DSAMP_MINFILTER, D3DTEXF_LINEAR );
-    _pd3dDevice->SetSamplerState( 0, D3DSAMP_MAGFILTER, D3DTEXF_LINEAR );
-    _pd3dDevice->SetSamplerState( 0, D3DSAMP_MIPFILTER, D3DTEXF_LINEAR );
-
-    // Initialize Vertices
-    CUSTOMVERTEX Vertices[] = {
-            //front
-            { -1.0f, -1.0f, 0.0f, 0xffffffff, 0, 1 }, { -1.0f, 1.0f, 0.0f,
-                    0xffffffff, 0, 0 },
-            { 1.0f, -1.0f, 0.0f, 0xffffffff, 1, 1 }, { 1.0f, 1.0f, 0.0f,
-                    0xffffffff, 1, 0 } };
-
-    // Create the vertex buffer.
-    if (FAILED(_pd3dDevice->CreateVertexBuffer(sizeof(Vertices), 0,
-                                               D3DFVF_CUSTOMVERTEX,
-                                               D3DPOOL_DEFAULT, &_pVB, NULL )))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Failed to create the vertex buffer.");
-        return -1;
-    }
-
-    // Now we fill the vertex buffer.
-    VOID* pVertices;
-    if (FAILED(_pVB->Lock(0, sizeof(Vertices), (void**) &pVertices, 0)))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Failed to lock the vertex buffer.");
-        return -1;
-    }
-    memcpy(pVertices, Vertices, sizeof(Vertices));
-    _pVB->Unlock();
-
-    return 0;
-}
-
-int32_t VideoRenderDirect3D9::Init()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
-                 "VideoRenderDirect3D9::Init");
-
-    CriticalSectionScoped cs(&_refD3DCritsect);
-
-    // Start rendering thread...
-    if (!_screenUpdateThread)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Thread not created");
-        return -1;
-    }
-    _screenUpdateThread->Start();
-    _screenUpdateThread->SetPriority(rtc::kRealtimePriority);
-
-    // Start the event triggering the render process
-    unsigned int monitorFreq = 60;
-    DEVMODE dm;
-    // initialize the DEVMODE structure
-    ZeroMemory(&dm, sizeof(dm));
-    dm.dmSize = sizeof(dm);
-    if (0 != EnumDisplaySettings(NULL, ENUM_CURRENT_SETTINGS, &dm))
-    {
-        monitorFreq = dm.dmDisplayFrequency;
-    }
-    _screenUpdateEvent->StartTimer(true, 1000 / monitorFreq);
-
-    return InitDevice();
-}
-
-int32_t VideoRenderDirect3D9::ChangeWindow(void* window)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
-    return -1;
-}
-
-int VideoRenderDirect3D9::UpdateRenderSurface()
-{
-    CriticalSectionScoped cs(&_refD3DCritsect);
-
-    // Check if there are any updated buffers
-    bool updated = false;
-    std::map<int, D3D9Channel*>::iterator it;
-    it = _d3dChannels.begin();
-    while (it != _d3dChannels.end())
-    {
-
-        D3D9Channel* channel = it->second;
-        channel->IsUpdated(updated);
-        if (updated)
-        {
-            break;
-        }
-        it++;
-    }
-    //nothing is updated, continue
-    if (!updated)
-        return -1;
-
-    // Clear the backbuffer to a black color
-    _pd3dDevice->Clear(0, NULL, D3DCLEAR_TARGET, D3DCOLOR_XRGB(0, 0, 0), 1.0f,
-                       0);
-
-    // Begin the scene
-    if (SUCCEEDED(_pd3dDevice->BeginScene()))
-    {
-        _pd3dDevice->SetStreamSource(0, _pVB, 0, sizeof(CUSTOMVERTEX));
-        _pd3dDevice->SetFVF(D3DFVF_CUSTOMVERTEX);
-
-        //draw all the channels
-        //get texture from the channels
-        LPDIRECT3DTEXTURE9 textureFromChannel = NULL;
-        DWORD textureWidth, textureHeight;
-
-        std::multimap<int, unsigned int>::reverse_iterator it;
-        it = _d3dZorder.rbegin();
-        while (it != _d3dZorder.rend())
-        {
-            // loop through all channels and streams in Z order
-            int channel = it->second & 0x0000ffff;
-
-            std::map<int, D3D9Channel*>::iterator ddIt;
-            ddIt = _d3dChannels.find(channel);
-            if (ddIt != _d3dChannels.end())
-            {
-                // found the channel
-                D3D9Channel* channelObj = ddIt->second;
-                if (channelObj)
-                {
-                    textureFromChannel = channelObj->GetTexture();
-                    textureWidth = channelObj->GetTextureWidth();
-                    textureHeight = channelObj->GetTextureHeight();
-
-                    uint32_t zOrder;
-                    float startWidth, startHeight, stopWidth, stopHeight;
-                    channelObj->GetStreamSettings(0, zOrder, startWidth,
-                                                  startHeight, stopWidth,
-                                                  stopHeight);
-
-                    //draw the video stream
-                    UpdateVerticeBuffer(_pVB, 0, startWidth, startHeight,
-                                        stopWidth, stopHeight);
-                    _pd3dDevice->SetTexture(0, textureFromChannel);
-                    _pd3dDevice->DrawPrimitive(D3DPT_TRIANGLESTRIP, 0, 2);
-
-                    //Notice channel that this frame as been rendered
-                    channelObj->RenderOffFrame();
-                }
-            }
-            it++;
-        }
-
-        //draw the logo
-        if (_pTextureLogo)
-        {
-            UpdateVerticeBuffer(_pVB, 0, _logoLeft, _logoTop, _logoRight,
-                                _logoBottom);
-            _pd3dDevice->SetTexture(0, _pTextureLogo);
-            _pd3dDevice->DrawPrimitive(D3DPT_TRIANGLESTRIP, 0, 2);
-        }
-
-        // End the scene
-        _pd3dDevice->EndScene();
-    }
-
-    // Present the backbuffer contents to the display
-    _pd3dDevice->Present(NULL, NULL, NULL, NULL );
-
-    return 0;
-}
-
-//set the  alpha value of the pixal with a particular colorkey as 0
-int VideoRenderDirect3D9::SetTransparentColor(LPDIRECT3DTEXTURE9 pTexture,
-                                                  DDCOLORKEY* transparentColorKey,
-                                                  DWORD width,
-                                                  DWORD height)
-{
-    D3DLOCKED_RECT lr;
-    if (!pTexture)
-        return -1;
-
-    CriticalSectionScoped cs(&_refD3DCritsect);
-    if (SUCCEEDED(pTexture->LockRect(0, &lr, NULL, D3DLOCK_DISCARD)))
-    {
-        for (DWORD y = 0; y < height; y++)
-        {
-            DWORD dwOffset = y * width;
-
-            for (DWORD x = 0; x < width; x)
-            {
-                DWORD temp = ((DWORD*) lr.pBits)[dwOffset + x];
-                if ((temp & 0x00FFFFFF)
-                        == transparentColorKey->dwColorSpaceLowValue)
-                {
-                    temp &= 0x00FFFFFF;
-                }
-                else
-                {
-                    temp |= 0xFF000000;
-                }
-                ((DWORD*) lr.pBits)[dwOffset + x] = temp;
-                x++;
-            }
-        }
-        pTexture->UnlockRect(0);
-        return 0;
-    }
-    return -1;
-}
-
-/*
- *
- *    Rendering process
- *
- */
-bool VideoRenderDirect3D9::ScreenUpdateThreadProc(void* obj)
-{
-    return static_cast<VideoRenderDirect3D9*> (obj)->ScreenUpdateProcess();
-}
-
-bool VideoRenderDirect3D9::ScreenUpdateProcess()
-{
-    _screenUpdateEvent->Wait(100);
-
-    if (!_screenUpdateThread)
-    {
-        //stop the thread
-        return false;
-    }
-    if (!_pd3dDevice)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "d3dDevice not created.");
-        return true;
-    }
-
-    HRESULT hr = _pd3dDevice->TestCooperativeLevel();
-
-    if (SUCCEEDED(hr))
-    {
-        UpdateRenderSurface();
-    }
-
-    if (hr == D3DERR_DEVICELOST)
-    {
-        //Device is lost and cannot be reset yet
-
-    }
-    else if (hr == D3DERR_DEVICENOTRESET)
-    {
-        //Lost but we can reset it now
-        //Note: the standard way is to call Reset, however for some reason doesn't work here.
-        //so we will release the device and create it again.
-        ResetDevice();
-    }
-
-    return true;
-}
-
-int VideoRenderDirect3D9::CloseDevice()
-{
-    CriticalSectionScoped cs(&_refD3DCritsect);
-    WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
-                 "VideoRenderDirect3D9::CloseDevice");
-
-    if (_pTextureLogo != NULL)
-    {
-        _pTextureLogo->Release();
-        _pTextureLogo = NULL;
-    }
-
-    if (_pVB != NULL)
-    {
-        _pVB->Release();
-        _pVB = NULL;
-    }
-
-    if (_pd3dDevice != NULL)
-    {
-        _pd3dDevice->Release();
-        _pd3dDevice = NULL;
-    }
-
-    if (_pD3D != NULL)
-    {
-        _pD3D->Release();
-        _pD3D = NULL;
-    }
-
-    if (_pd3dSurface != NULL)
-        _pd3dSurface->Release();
-    return 0;
-}
-
-D3D9Channel* VideoRenderDirect3D9::GetD3DChannel(int channel)
-{
-    std::map<int, D3D9Channel*>::iterator ddIt;
-    ddIt = _d3dChannels.find(channel & 0x0000ffff);
-    D3D9Channel* ddobj = NULL;
-    if (ddIt != _d3dChannels.end())
-    {
-        ddobj = ddIt->second;
-    }
-    if (ddobj == NULL)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Direct3D render failed to find channel");
-        return NULL;
-    }
-    return ddobj;
-}
-
-int32_t VideoRenderDirect3D9::DeleteChannel(const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_refD3DCritsect);
-
-
-    std::multimap<int, unsigned int>::iterator it;
-    it = _d3dZorder.begin();
-    while (it != _d3dZorder.end())
-    {
-        if ((streamId & 0x0000ffff) == (it->second & 0x0000ffff))
-        {
-            it = _d3dZorder.erase(it);
-            break;
-        }
-        it++;
-    }
-
-    std::map<int, D3D9Channel*>::iterator ddIt;
-    ddIt = _d3dChannels.find(streamId & 0x0000ffff);
-    if (ddIt != _d3dChannels.end())
-    {
-        delete ddIt->second;
-        _d3dChannels.erase(ddIt);
-        return 0;
-    }
-    return -1;
-}
-
-VideoRenderCallback* VideoRenderDirect3D9::CreateChannel(const uint32_t channel,
-                                                                 const uint32_t zOrder,
-                                                                 const float left,
-                                                                 const float top,
-                                                                 const float right,
-                                                                 const float bottom)
-{
-    CriticalSectionScoped cs(&_refD3DCritsect);
-
-    //FIXME this should be done in VideoAPIWindows? stop the frame deliver first
-    //remove the old channel
-    DeleteChannel(channel);
-
-    D3D9Channel* d3dChannel = new D3D9Channel(_pd3dDevice,
-                                                      &_refD3DCritsect, _trace);
-    d3dChannel->SetStreamSettings(0, zOrder, left, top, right, bottom);
-
-    // store channel
-    _d3dChannels[channel & 0x0000ffff] = d3dChannel;
-
-    // store Z order
-    // default streamID is 0
-    _d3dZorder.insert(
-                      std::pair<int, unsigned int>(zOrder, channel & 0x0000ffff));
-
-    return d3dChannel;
-}
-
-int32_t VideoRenderDirect3D9::GetStreamSettings(const uint32_t channel,
-                                                const uint16_t streamId,
-                                                uint32_t& zOrder,
-                                                float& left, float& top,
-                                                float& right, float& bottom)
-{
-    std::map<int, D3D9Channel*>::iterator ddIt;
-    ddIt = _d3dChannels.find(channel & 0x0000ffff);
-    D3D9Channel* ddobj = NULL;
-    if (ddIt != _d3dChannels.end())
-    {
-        ddobj = ddIt->second;
-    }
-    if (ddobj == NULL)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Direct3D render failed to find channel");
-        return -1;
-    }
-    // Only allow one stream per channel, demuxing is
-    return ddobj->GetStreamSettings(0, zOrder, left, top, right, bottom);
-}
-
-int VideoRenderDirect3D9::UpdateVerticeBuffer(LPDIRECT3DVERTEXBUFFER9 pVB,
-                                                  int offset,
-                                                  float startWidth,
-                                                  float startHeight,
-                                                  float stopWidth,
-                                                  float stopHeight)
-{
-    if (pVB == NULL)
-        return -1;
-
-    float left, right, top, bottom;
-
-    //update the vertice buffer
-    //0,1 => -1,1
-    left = startWidth * 2 - 1;
-    right = stopWidth * 2 - 1;
-
-    //0,1 => 1,-1
-    top = 1 - startHeight * 2;
-    bottom = 1 - stopHeight * 2;
-
-    CUSTOMVERTEX newVertices[] = {
-            //logo
-            { left, bottom, 0.0f, 0xffffffff, 0, 1 }, { left, top, 0.0f,
-                    0xffffffff, 0, 0 },
-            { right, bottom, 0.0f, 0xffffffff, 1, 1 }, { right, top, 0.0f,
-                    0xffffffff, 1, 0 }, };
-    // Now we fill the vertex buffer.
-    VOID* pVertices;
-    if (FAILED(pVB->Lock(sizeof(CUSTOMVERTEX) * offset, sizeof(newVertices),
-                         (void**) &pVertices, 0)))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Failed to lock the vertex buffer.");
-        return -1;
-    }
-    memcpy(pVertices, newVertices, sizeof(newVertices));
-    pVB->Unlock();
-
-    return 0;
-}
-
-int32_t VideoRenderDirect3D9::StartRender()
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
-    return 0;
-}
-
-int32_t VideoRenderDirect3D9::StopRender()
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
-    return 0;
-}
-
-bool VideoRenderDirect3D9::IsFullScreen()
-{
-    return _fullScreen;
-}
-
-int32_t VideoRenderDirect3D9::SetCropping(const uint32_t channel,
-                                          const uint16_t streamId,
-                                          const float left, const float top,
-                                          const float right, const float bottom)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
-    return 0;
-}
-
-int32_t VideoRenderDirect3D9::SetTransparentBackground(
-                                                                 const bool enable)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
-    return 0;
-}
-
-int32_t VideoRenderDirect3D9::SetText(const uint8_t textId,
-                                      const uint8_t* text,
-                                      const int32_t textLength,
-                                      const uint32_t colorText,
-                                      const uint32_t colorBg,
-                                      const float left, const float top,
-                                      const float rigth, const float bottom)
-{
-    WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
-    return 0;
-}
-
-int32_t VideoRenderDirect3D9::SetBitmap(const void* bitMap,
-                                        const uint8_t pictureId,
-                                        const void* colorKey,
-                                        const float left, const float top,
-                                        const float right, const float bottom)
-{
-    if (!bitMap)
-    {
-        if (_pTextureLogo != NULL)
-        {
-            _pTextureLogo->Release();
-            _pTextureLogo = NULL;
-        }
-        WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "Remove bitmap.");
-        return 0;
-    }
-
-    // sanity
-    if (left > 1.0f || left < 0.0f ||
-        top > 1.0f || top < 0.0f ||
-        right > 1.0f || right < 0.0f ||
-        bottom > 1.0f || bottom < 0.0f)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Direct3D SetBitmap invalid parameter");
-        return -1;
-    }
-
-    if ((bottom <= top) || (right <= left))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Direct3D SetBitmap invalid parameter");
-        return -1;
-    }
-
-    CriticalSectionScoped cs(&_refD3DCritsect);
-
-    unsigned char* srcPtr;
-    HGDIOBJ oldhand;
-    BITMAPINFO pbi;
-    BITMAP bmap;
-    HDC hdcNew;
-    hdcNew = CreateCompatibleDC(0);
-    // Fill out the BITMAP structure.
-    GetObject((HBITMAP)bitMap, sizeof(bmap), &bmap);
-    //Select the bitmap handle into the new device context.
-    oldhand = SelectObject(hdcNew, (HGDIOBJ) bitMap);
-    // we are done with this object
-    DeleteObject(oldhand);
-    pbi.bmiHeader.biSize = 40;
-    pbi.bmiHeader.biWidth = bmap.bmWidth;
-    pbi.bmiHeader.biHeight = bmap.bmHeight;
-    pbi.bmiHeader.biPlanes = 1;
-    pbi.bmiHeader.biBitCount = bmap.bmBitsPixel;
-    pbi.bmiHeader.biCompression = BI_RGB;
-    pbi.bmiHeader.biSizeImage = bmap.bmWidth * bmap.bmHeight * 3;
-    srcPtr = new unsigned char[bmap.bmWidth * bmap.bmHeight * 4];
-    // the original un-stretched image in RGB24
-    int pixelHeight = GetDIBits(hdcNew, (HBITMAP)bitMap, 0, bmap.bmHeight, srcPtr, &pbi,
-                                DIB_RGB_COLORS);
-    if (pixelHeight == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Direct3D failed to GetDIBits in SetBitmap");
-        delete[] srcPtr;
-        return -1;
-    }
-    DeleteDC(hdcNew);
-    if (pbi.bmiHeader.biBitCount != 24 && pbi.bmiHeader.biBitCount != 32)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Direct3D failed to SetBitmap invalid bit depth");
-        delete[] srcPtr;
-        return -1;
-    }
-
-    HRESULT ret;
-    //release the previous logo texture
-    if (_pTextureLogo != NULL)
-    {
-        _pTextureLogo->Release();
-        _pTextureLogo = NULL;
-    }
-    ret = _pd3dDevice->CreateTexture(bmap.bmWidth, bmap.bmHeight, 1, 0,
-                                     D3DFMT_A8R8G8B8, D3DPOOL_MANAGED,
-                                     &_pTextureLogo, NULL);
-    if (FAILED(ret))
-    {
-        _pTextureLogo = NULL;
-        delete[] srcPtr;
-        return -1;
-    }
-    if (!_pTextureLogo)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Texture for rendering not initialized.");
-        delete[] srcPtr;
-        return -1;
-    }
-
-    D3DLOCKED_RECT lr;
-    if (FAILED(_pTextureLogo->LockRect(0, &lr, NULL, 0)))
-    {
-        delete[] srcPtr;
-        return -1;
-    }
-    unsigned char* dstPtr = (UCHAR*) lr.pBits;
-    int pitch = bmap.bmWidth * 4;
-
-    if (pbi.bmiHeader.biBitCount == 24)
-    {
-        ConvertRGB24ToARGB(srcPtr, dstPtr, bmap.bmWidth, bmap.bmHeight, 0);
-    }
-    else
-    {
-        unsigned char* srcTmp = srcPtr + (bmap.bmWidth * 4) * (bmap.bmHeight - 1);
-        for (int i = 0; i < bmap.bmHeight; ++i)
-        {
-            memcpy(dstPtr, srcTmp, bmap.bmWidth * 4);
-            srcTmp -= bmap.bmWidth * 4;
-            dstPtr += pitch;
-        }
-    }
-
-    delete[] srcPtr;
-    if (FAILED(_pTextureLogo->UnlockRect(0)))
-    {
-        return -1;
-    }
-
-    if (colorKey)
-    {
-        DDCOLORKEY* ddColorKey =
-                static_cast<DDCOLORKEY*> (const_cast<void*> (colorKey));
-        SetTransparentColor(_pTextureLogo, ddColorKey, bmap.bmWidth,
-                            bmap.bmHeight);
-    }
-
-    //update the vertice buffer
-    //0,1 => -1,1
-    _logoLeft = left;
-    _logoRight = right;
-
-    //0,1 => 1,-1
-    _logoTop = top;
-    _logoBottom = bottom;
-
-    return 0;
-
-}
-
-int32_t VideoRenderDirect3D9::GetGraphicsMemory(uint64_t& totalMemory,
-                                                uint64_t& availableMemory)
-{
-    totalMemory = _totalMemory;
-    availableMemory = _availableMemory;
-    return 0;
-}
-
-int32_t VideoRenderDirect3D9::ConfigureRenderer(const uint32_t channel,
-                                                const uint16_t streamId,
-                                                const unsigned int zOrder,
-                                                const float left,
-                                                const float top,
-                                                const float right,
-                                                const float bottom)
-{
-    std::map<int, D3D9Channel*>::iterator ddIt;
-    ddIt = _d3dChannels.find(channel & 0x0000ffff);
-    D3D9Channel* ddobj = NULL;
-    if (ddIt != _d3dChannels.end())
-    {
-        ddobj = ddIt->second;
-    }
-    if (ddobj == NULL)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
-                     "Direct3D render failed to find channel");
-        return -1;
-    }
-    // Only allow one stream per channel, demuxing is
-    ddobj->SetStreamSettings(0, zOrder, left, top, right, bottom);
-
-    return 0;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/windows/video_render_direct3d9.h b/chromium/third_party/webrtc/modules/video_render/windows/video_render_direct3d9.h
deleted file mode 100644
index eaa8c147e2e..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/windows/video_render_direct3d9.h
+++ /dev/null
@@ -1,256 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
-
-#include <memory>
-
-#include "webrtc/modules/video_render/windows/i_video_render_win.h"
-
-#include <d3d9.h>
-#include <ddraw.h>
-
-#include <Map>
-
-// Added
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-
-#pragma comment(lib, "d3d9.lib")       // located in DirectX SDK
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventTimerWrapper;
-class Trace;
-
-class D3D9Channel: public VideoRenderCallback
-{
-public:
-    D3D9Channel(LPDIRECT3DDEVICE9 pd3DDevice,
-                    CriticalSectionWrapper* critSect, Trace* trace);
-
-    virtual ~D3D9Channel();
-
-    // Inherited from VideoRencerCallback, called from VideoAPI class.
-    // Called when the incomming frame size and/or number of streams in mix changes
-    virtual int FrameSizeChange(int width, int height, int numberOfStreams);
-
-    // A new frame is delivered.
-    virtual int DeliverFrame(const VideoFrame& videoFrame);
-    virtual int32_t RenderFrame(const uint32_t streamId,
-                                const VideoFrame& videoFrame);
-
-    // Called to check if the video frame is updated.
-    int IsUpdated(bool& isUpdated);
-    // Called after the video frame has been render to the screen
-    int RenderOffFrame();
-    // Called to get the texture that contains the video frame
-    LPDIRECT3DTEXTURE9 GetTexture();
-    // Called to get the texture(video frame) size
-    int GetTextureWidth();
-    int GetTextureHeight();
-    //
-    void SetStreamSettings(uint16_t streamId,
-                           uint32_t zOrder,
-                           float startWidth,
-                           float startHeight,
-                           float stopWidth,
-                           float stopHeight);
-    int GetStreamSettings(uint16_t streamId,
-                          uint32_t& zOrder,
-                          float& startWidth,
-                          float& startHeight,
-                          float& stopWidth,
-                          float& stopHeight);
-
-    int ReleaseTexture();
-    int RecreateTexture(LPDIRECT3DDEVICE9 pd3DDevice);
-
-protected:
-
-private:
-    //critical section passed from the owner
-    CriticalSectionWrapper* _critSect;
-    LPDIRECT3DDEVICE9 _pd3dDevice;
-    LPDIRECT3DTEXTURE9 _pTexture;
-
-    bool _bufferIsUpdated;
-    // the frame size
-    int _width;
-    int _height;
-    //sream settings
-    //TODO support multiple streams in one channel
-    uint16_t _streamId;
-    uint32_t _zOrder;
-    float _startWidth;
-    float _startHeight;
-    float _stopWidth;
-    float _stopHeight;
-};
-
-class VideoRenderDirect3D9: IVideoRenderWin
-{
-public:
-    VideoRenderDirect3D9(Trace* trace, HWND hWnd, bool fullScreen);
-    ~VideoRenderDirect3D9();
-
-public:
-    //IVideoRenderWin
-
-    /**************************************************************************
-     *
-     *   Init
-     *
-     ***************************************************************************/
-    virtual int32_t Init();
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-    virtual VideoRenderCallback
-            * CreateChannel(const uint32_t streamId,
-                            const uint32_t zOrder,
-                            const float left,
-                            const float top,
-                            const float right,
-                            const float bottom);
-
-    virtual int32_t DeleteChannel(const uint32_t streamId);
-
-    virtual int32_t GetStreamSettings(const uint32_t channel,
-                                      const uint16_t streamId,
-                                      uint32_t& zOrder,
-                                      float& left, float& top,
-                                      float& right, float& bottom);
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender();
-    virtual int32_t StopRender();
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    virtual bool IsFullScreen();
-
-    virtual int32_t SetCropping(const uint32_t channel,
-                                const uint16_t streamId,
-                                const float left, const float top,
-                                const float right, const float bottom);
-
-    virtual int32_t ConfigureRenderer(const uint32_t channel,
-                                      const uint16_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t SetTransparentBackground(const bool enable);
-
-    virtual int32_t ChangeWindow(void* window);
-
-    virtual int32_t GetGraphicsMemory(uint64_t& totalMemory,
-                                      uint64_t& availableMemory);
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t colorText,
-                            const uint32_t colorBg,
-                            const float left, const float top,
-                            const float rigth, const float bottom);
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left, const float top,
-                              const float right, const float bottom);
-
-public:
-    // Get a channel by channel id
-    D3D9Channel* GetD3DChannel(int channel);
-    int UpdateRenderSurface();
-
-protected:
-    // The thread rendering the screen
-    static bool ScreenUpdateThreadProc(void* obj);
-    bool ScreenUpdateProcess();
-
-private:
-    // Init/close the d3d device
-    int InitDevice();
-    int CloseDevice();
-
-    // Transparent related functions
-    int SetTransparentColor(LPDIRECT3DTEXTURE9 pTexture,
-                            DDCOLORKEY* transparentColorKey,
-                            DWORD width,
-                            DWORD height);
-
-    CriticalSectionWrapper& _refD3DCritsect;
-    Trace* _trace;
-    // TODO(pbos): Remove unique_ptr and use PlatformThread directly.
-    std::unique_ptr<rtc::PlatformThread> _screenUpdateThread;
-    EventTimerWrapper* _screenUpdateEvent;
-
-    HWND _hWnd;
-    bool _fullScreen;
-    RECT _originalHwndRect;
-    //FIXME we probably don't need this since all the information can be get from _d3dChannels
-    int _channel;
-    //Window size
-    UINT _winWidth;
-    UINT _winHeight;
-
-    // Device
-    LPDIRECT3D9 _pD3D; // Used to create the D3DDevice
-    LPDIRECT3DDEVICE9 _pd3dDevice; // Our rendering device
-    LPDIRECT3DVERTEXBUFFER9 _pVB; // Buffer to hold Vertices
-    LPDIRECT3DTEXTURE9 _pTextureLogo;
-
-    std::map<int, D3D9Channel*> _d3dChannels;
-    std::multimap<int, unsigned int> _d3dZorder;
-
-    // The position where the logo will be placed
-    float _logoLeft;
-    float _logoTop;
-    float _logoRight;
-    float _logoBottom;
-
-    typedef HRESULT (WINAPI *DIRECT3DCREATE9EX)(UINT SDKVersion, IDirect3D9Ex**);
-    LPDIRECT3DSURFACE9 _pd3dSurface;
-
-    DWORD GetVertexProcessingCaps();
-    int InitializeD3D(HWND hWnd, D3DPRESENT_PARAMETERS* pd3dpp);
-
-    D3DPRESENT_PARAMETERS _d3dpp;
-    int ResetDevice();
-
-    int UpdateVerticeBuffer(LPDIRECT3DVERTEXBUFFER9 pVB, int offset,
-                            float startWidth, float startHeight,
-                            float stopWidth, float stopHeight);
-
-    //code for providing graphics settings
-    DWORD _totalMemory;
-    DWORD _availableMemory;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
diff --git a/chromium/third_party/webrtc/modules/video_render/windows/video_render_windows_impl.cc b/chromium/third_party/webrtc/modules/video_render/windows/video_render_windows_impl.cc
deleted file mode 100644
index 042d7fdfa33..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/windows/video_render_windows_impl.cc
+++ /dev/null
@@ -1,337 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/video_render/windows/video_render_windows_impl.h"
-
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#ifdef DIRECT3D9_RENDERING
-#include "webrtc/modules/video_render/windows/video_render_direct3d9.h"
-#endif
-
-#include <tchar.h>
-
-namespace webrtc {
-
-VideoRenderWindowsImpl::VideoRenderWindowsImpl(const int32_t id,
-    const VideoRenderType videoRenderType, void* window, const bool fullscreen)
-    : _renderWindowsCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
-      _prtWindow(window),
-      _fullscreen(fullscreen),
-      _renderMethod(kVideoRenderWinD3D9),
-      _ptrRendererWin(NULL) {
-}
-
-VideoRenderWindowsImpl::~VideoRenderWindowsImpl()
-{
-    delete &_renderWindowsCritsect;
-    if (_ptrRendererWin)
-    {
-        delete _ptrRendererWin;
-        _ptrRendererWin = NULL;
-    }
-}
-
-int32_t VideoRenderWindowsImpl::Init()
-{
-    // Create the win renderer
-    switch (_renderMethod)
-    {
-        case kVideoRenderWinD3D9:
-        {
-#ifdef DIRECT3D9_RENDERING
-            VideoRenderDirect3D9* ptrRenderer;
-            ptrRenderer = new VideoRenderDirect3D9(NULL, (HWND) _prtWindow, _fullscreen);
-            if (ptrRenderer == NULL)
-            {
-                break;
-            }
-            _ptrRendererWin = reinterpret_cast<IVideoRenderWin*>(ptrRenderer);
-#else
-            return NULL;
-#endif  //DIRECT3D9_RENDERING
-        }
-            break;
-        default:
-            break;
-    }
-
-    //Init renderer
-    if (_ptrRendererWin)
-        return _ptrRendererWin->Init();
-    else
-        return -1;
-}
-
-int32_t VideoRenderWindowsImpl::ChangeWindow(void* window)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    if (!_ptrRendererWin)
-    {
-        return -1;
-    }
-    else
-    {
-        return _ptrRendererWin->ChangeWindow(window);
-    }
-}
-
-VideoRenderCallback*
-VideoRenderWindowsImpl::AddIncomingRenderStream(const uint32_t streamId,
-                                                const uint32_t zOrder,
-                                                const float left,
-                                                const float top,
-                                                const float right,
-                                                const float bottom)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    VideoRenderCallback* renderCallback = NULL;
-
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        renderCallback = _ptrRendererWin->CreateChannel(streamId, zOrder, left,
-                                                        top, right, bottom);
-    }
-
-    return renderCallback;
-}
-
-int32_t VideoRenderWindowsImpl::DeleteIncomingRenderStream(
-                                                                 const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->DeleteChannel(streamId);
-    }
-    return error;
-}
-
-int32_t VideoRenderWindowsImpl::GetIncomingRenderStreamProperties(
-                                                                        const uint32_t streamId,
-                                                                        uint32_t& zOrder,
-                                                                        float& left,
-                                                                        float& top,
-                                                                        float& right,
-                                                                        float& bottom) const
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    zOrder = 0;
-    left = 0;
-    top = 0;
-    right = 0;
-    bottom = 0;
-
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->GetStreamSettings(streamId, 0, zOrder, left,
-                                                   top, right, bottom);
-    }
-    return error;
-}
-
-int32_t VideoRenderWindowsImpl::StartRender()
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->StartRender();
-    }
-    return error;
-}
-
-int32_t VideoRenderWindowsImpl::StopRender()
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->StopRender();
-    }
-    return error;
-}
-
-VideoRenderType VideoRenderWindowsImpl::RenderType()
-{
-    return kRenderWindows;
-}
-
-RawVideoType VideoRenderWindowsImpl::PerferedVideoType()
-{
-    return kVideoI420;
-}
-
-bool VideoRenderWindowsImpl::FullScreen()
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    bool fullscreen = false;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        fullscreen = _ptrRendererWin->IsFullScreen();
-    }
-    return fullscreen;
-}
-
-int32_t VideoRenderWindowsImpl::GetGraphicsMemory(
-                                                        uint64_t& totalGraphicsMemory,
-                                                        uint64_t& availableGraphicsMemory) const
-{
-    if (_ptrRendererWin)
-    {
-        return _ptrRendererWin->GetGraphicsMemory(totalGraphicsMemory,
-                                                  availableGraphicsMemory);
-    }
-
-    totalGraphicsMemory = 0;
-    availableGraphicsMemory = 0;
-    return -1;
-}
-
-int32_t VideoRenderWindowsImpl::GetScreenResolution(
-                                                          uint32_t& screenWidth,
-                                                          uint32_t& screenHeight) const
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    screenWidth = 0;
-    screenHeight = 0;
-    return 0;
-}
-
-uint32_t VideoRenderWindowsImpl::RenderFrameRate(
-                                                       const uint32_t streamId)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    return 0;
-}
-
-int32_t VideoRenderWindowsImpl::SetStreamCropping(
-                                                        const uint32_t streamId,
-                                                        const float left,
-                                                        const float top,
-                                                        const float right,
-                                                        const float bottom)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->SetCropping(streamId, 0, left, top, right,
-                                             bottom);
-    }
-    return error;
-}
-
-int32_t VideoRenderWindowsImpl::ConfigureRenderer(
-                                                        const uint32_t streamId,
-                                                        const unsigned int zOrder,
-                                                        const float left,
-                                                        const float top,
-                                                        const float right,
-                                                        const float bottom)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->ConfigureRenderer(streamId, 0, zOrder, left,
-                                                   top, right, bottom);
-    }
-
-    return error;
-}
-
-int32_t VideoRenderWindowsImpl::SetTransparentBackground(
-                                                               const bool enable)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->SetTransparentBackground(enable);
-    }
-    return error;
-}
-
-int32_t VideoRenderWindowsImpl::SetText(
-                                              const uint8_t textId,
-                                              const uint8_t* text,
-                                              const int32_t textLength,
-                                              const uint32_t textColorRef,
-                                              const uint32_t backgroundColorRef,
-                                              const float left,
-                                              const float top,
-                                              const float right,
-                                              const float bottom)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->SetText(textId, text, textLength,
-                                         textColorRef, backgroundColorRef,
-                                         left, top, right, bottom);
-    }
-    return error;
-}
-
-int32_t VideoRenderWindowsImpl::SetBitmap(const void* bitMap,
-                                          const uint8_t pictureId,
-                                          const void* colorKey,
-                                          const float left, const float top,
-                                          const float right, const float bottom)
-{
-    CriticalSectionScoped cs(&_renderWindowsCritsect);
-    int32_t error = -1;
-    if (!_ptrRendererWin)
-    {
-    }
-    else
-    {
-        error = _ptrRendererWin->SetBitmap(bitMap, pictureId, colorKey, left,
-                                           top, right, bottom);
-    }
-    return error;
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/modules/video_render/windows/video_render_windows_impl.h b/chromium/third_party/webrtc/modules/video_render/windows/video_render_windows_impl.h
deleted file mode 100644
index aaa3f81fc7e..00000000000
--- a/chromium/third_party/webrtc/modules/video_render/windows/video_render_windows_impl.h
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
-
-#include <Winerror.h>
-#include <dxdiag.h>
-
-#include "webrtc/modules/video_render/i_video_render.h"
-#include "webrtc/modules/video_render/windows/i_video_render_win.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-#define EXPAND(x)            x, sizeof(x)/sizeof(TCHAR)
-
-enum VideoRenderWinMethod {
-  kVideoRenderWinD3D9 = 0,
-};
-
-// Class definitions
-class VideoRenderWindowsImpl: IVideoRender
-{
-public:
-    /*
-     *   Constructor/destructor
-     */
-
-    VideoRenderWindowsImpl(const int32_t id,
-                           const VideoRenderType videoRenderType,
-                           void* window, const bool fullscreen);
-
-    virtual ~VideoRenderWindowsImpl();
-
-    virtual int32_t Init();
-
-    virtual int32_t ChangeWindow(void* window);
-
-    /**************************************************************************
-     *
-     *   Incoming Streams
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderCallback
-            * AddIncomingRenderStream(const uint32_t streamId,
-                                      const uint32_t zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t
-            DeleteIncomingRenderStream(const uint32_t streamId);
-
-    virtual int32_t
-            GetIncomingRenderStreamProperties(const uint32_t streamId,
-                                              uint32_t& zOrder,
-                                              float& left, float& top,
-                                              float& right, float& bottom) const;
-
-    /**************************************************************************
-     *
-     *   Start/Stop
-     *
-     ***************************************************************************/
-
-    virtual int32_t StartRender();
-
-    virtual int32_t StopRender();
-
-    /**************************************************************************
-     *
-     *   Properties
-     *
-     ***************************************************************************/
-
-    virtual VideoRenderType RenderType();
-
-    virtual RawVideoType PerferedVideoType();
-
-    virtual bool FullScreen();
-
-    virtual int32_t
-            GetGraphicsMemory(uint64_t& totalGraphicsMemory,
-                              uint64_t& availableGraphicsMemory) const;
-
-    virtual int32_t
-            GetScreenResolution(uint32_t& screenWidth,
-                                uint32_t& screenHeight) const;
-
-    virtual uint32_t RenderFrameRate(const uint32_t streamId);
-
-    virtual int32_t SetStreamCropping(const uint32_t streamId,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t ConfigureRenderer(const uint32_t streamId,
-                                      const unsigned int zOrder,
-                                      const float left, const float top,
-                                      const float right, const float bottom);
-
-    virtual int32_t SetTransparentBackground(const bool enable);
-
-    virtual int32_t SetText(const uint8_t textId,
-                            const uint8_t* text,
-                            const int32_t textLength,
-                            const uint32_t textColorRef,
-                            const uint32_t backgroundColorRef,
-                            const float left, const float top,
-                            const float right, const float bottom);
-
-    virtual int32_t SetBitmap(const void* bitMap,
-                              const uint8_t pictureId,
-                              const void* colorKey,
-                              const float left, const float top,
-                              const float right, const float bottom);
-
-private:
-    CriticalSectionWrapper& _renderWindowsCritsect;
-
-    void* _prtWindow;
-    bool _fullscreen;
-
-    VideoRenderWinMethod _renderMethod;
-    IVideoRenderWin* _ptrRendererWin;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
diff --git a/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket.h b/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket.h
index 3a15d4a399f..065b4c259b8 100644
--- a/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket.h
+++ b/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket.h
@@ -12,7 +12,7 @@
 #define WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
 
 #include "webrtc/base/asynctcpsocket.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/socketfactory.h"
 
 namespace cricket {
diff --git a/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc b/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
index 22c1b26903d..5929d1f84da 100644
--- a/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/p2p/base/asyncstuntcpsocket.h"
 #include "webrtc/base/asyncsocket.h"
 #include "webrtc/base/gunit.h"
@@ -122,11 +124,11 @@ class AsyncStunTCPSocketTest : public testing::Test,
     return ret;
   }
 
-  rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
   rtc::SocketServerScope ss_scope_;
-  rtc::scoped_ptr<AsyncStunTCPSocket> send_socket_;
-  rtc::scoped_ptr<AsyncStunTCPSocket> recv_socket_;
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> listen_socket_;
+  std::unique_ptr<AsyncStunTCPSocket> send_socket_;
+  std::unique_ptr<AsyncStunTCPSocket> recv_socket_;
+  std::unique_ptr<rtc::AsyncPacketSocket> listen_socket_;
   std::list<std::string> recv_packets_;
 };
 
diff --git a/chromium/third_party/webrtc/p2p/base/basicpacketsocketfactory.cc b/chromium/third_party/webrtc/p2p/base/basicpacketsocketfactory.cc
index 697518da9d0..a05f9df8dfc 100644
--- a/chromium/third_party/webrtc/p2p/base/basicpacketsocketfactory.cc
+++ b/chromium/third_party/webrtc/p2p/base/basicpacketsocketfactory.cc
@@ -17,7 +17,6 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/nethelpers.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketadapters.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/thread.h"
diff --git a/chromium/third_party/webrtc/p2p/base/candidate.h b/chromium/third_party/webrtc/p2p/base/candidate.h
index 4ab1b6f2356..b918344740a 100644
--- a/chromium/third_party/webrtc/p2p/base/candidate.h
+++ b/chromium/third_party/webrtc/p2p/base/candidate.h
@@ -241,6 +241,19 @@ class Candidate {
            (256 - component_);
   }
 
+  bool operator==(const Candidate& o) const {
+    return id_ == o.id_ && component_ == o.component_ &&
+           protocol_ == o.protocol_ && relay_protocol_ == o.relay_protocol_ &&
+           address_ == o.address_ && priority_ == o.priority_ &&
+           username_ == o.username_ && password_ == o.password_ &&
+           type_ == o.type_ && network_name_ == o.network_name_ &&
+           network_type_ == o.network_type_ && generation_ == o.generation_ &&
+           foundation_ == o.foundation_ &&
+           related_address_ == o.related_address_ && tcptype_ == o.tcptype_ &&
+           transport_name_ == o.transport_name_ && network_id_ == o.network_id_;
+  }
+  bool operator!=(const Candidate& o) const { return !(*this == o); }
+
  private:
   std::string ToStringInternal(bool sensitive) const {
     std::ostringstream ost;
@@ -249,7 +262,8 @@ class Candidate {
     ost << "Cand[" << transport_name_ << ":" << foundation_ << ":" << component_
         << ":" << protocol_ << ":" << priority_ << ":" << address << ":"
         << type_ << ":" << related_address_ << ":" << username_ << ":"
-        << password_ << ":" << network_id_ << ":" << network_cost_ << "]";
+        << password_ << ":" << network_id_ << ":" << network_cost_ << ":"
+        << generation_ << "]";
     return ost.str();
   }
 
diff --git a/chromium/third_party/webrtc/p2p/base/dtlstransport.h b/chromium/third_party/webrtc/p2p/base/dtlstransport.h
index 276b05f786f..a4bf383f0a6 100644
--- a/chromium/third_party/webrtc/p2p/base/dtlstransport.h
+++ b/chromium/third_party/webrtc/p2p/base/dtlstransport.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_P2P_BASE_DTLSTRANSPORT_H_
 #define WEBRTC_P2P_BASE_DTLSTRANSPORT_H_
 
+#include <memory>
+
 #include "webrtc/p2p/base/dtlstransportchannel.h"
 #include "webrtc/p2p/base/transport.h"
 
@@ -64,27 +66,11 @@ class DtlsTransport : public Base {
     rtc::SSLFingerprint* local_fp =
         Base::local_description()->identity_fingerprint.get();
 
-    if (local_fp) {
-      // Sanity check local fingerprint.
-      if (certificate_) {
-        rtc::scoped_ptr<rtc::SSLFingerprint> local_fp_tmp(
-            rtc::SSLFingerprint::Create(local_fp->algorithm,
-                                        certificate_->identity()));
-        ASSERT(local_fp_tmp.get() != NULL);
-        if (!(*local_fp_tmp == *local_fp)) {
-          std::ostringstream desc;
-          desc << "Local fingerprint does not match identity. Expected: ";
-          desc << local_fp_tmp->ToString();
-          desc << " Got: " << local_fp->ToString();
-          return BadTransportDescription(desc.str(), error_desc);
-        }
-      } else {
-        return BadTransportDescription(
-            "Local fingerprint provided but no identity available.",
-            error_desc);
-      }
-    } else {
+    if (!local_fp) {
       certificate_ = nullptr;
+    } else if (!Base::VerifyCertificateFingerprint(certificate_.get(), local_fp,
+                                                   error_desc)) {
+      return false;
     }
 
     if (!channel->SetLocalCertificate(certificate_)) {
@@ -103,96 +89,23 @@ class DtlsTransport : public Base {
                               "transport descriptions are negotiated";
       return BadTransportDescription(msg, error_desc);
     }
-
     rtc::SSLFingerprint* local_fp =
         Base::local_description()->identity_fingerprint.get();
     rtc::SSLFingerprint* remote_fp =
         Base::remote_description()->identity_fingerprint.get();
-
     if (remote_fp && local_fp) {
       remote_fingerprint_.reset(new rtc::SSLFingerprint(*remote_fp));
-
-      // From RFC 4145, section-4.1, The following are the values that the
-      // 'setup' attribute can take in an offer/answer exchange:
-      //       Offer      Answer
-      //      ________________
-      //      active     passive / holdconn
-      //      passive    active / holdconn
-      //      actpass    active / passive / holdconn
-      //      holdconn   holdconn
-      //
-      // Set the role that is most conformant with RFC 5763, Section 5, bullet 1
-      // The endpoint MUST use the setup attribute defined in [RFC4145].
-      // The endpoint that is the offerer MUST use the setup attribute
-      // value of setup:actpass and be prepared to receive a client_hello
-      // before it receives the answer.  The answerer MUST use either a
-      // setup attribute value of setup:active or setup:passive.  Note that
-      // if the answerer uses setup:passive, then the DTLS handshake will
-      // not begin until the answerer is received, which adds additional
-      // latency. setup:active allows the answer and the DTLS handshake to
-      // occur in parallel.  Thus, setup:active is RECOMMENDED.  Whichever
-      // party is active MUST initiate a DTLS handshake by sending a
-      // ClientHello over each flow (host/port quartet).
-      // IOW - actpass and passive modes should be treated as server and
-      // active as client.
-      ConnectionRole local_connection_role =
-          Base::local_description()->connection_role;
-      ConnectionRole remote_connection_role =
-          Base::remote_description()->connection_role;
-
-      bool is_remote_server = false;
-      if (local_role == CA_OFFER) {
-        if (local_connection_role != CONNECTIONROLE_ACTPASS) {
-          return BadTransportDescription(
-              "Offerer must use actpass value for setup attribute.",
-              error_desc);
-        }
-
-        if (remote_connection_role == CONNECTIONROLE_ACTIVE ||
-            remote_connection_role == CONNECTIONROLE_PASSIVE ||
-            remote_connection_role == CONNECTIONROLE_NONE) {
-          is_remote_server = (remote_connection_role == CONNECTIONROLE_PASSIVE);
-        } else {
-          const std::string msg =
-              "Answerer must use either active or passive value "
-              "for setup attribute.";
-          return BadTransportDescription(msg, error_desc);
-        }
-        // If remote is NONE or ACTIVE it will act as client.
-      } else {
-        if (remote_connection_role != CONNECTIONROLE_ACTPASS &&
-            remote_connection_role != CONNECTIONROLE_NONE) {
-          return BadTransportDescription(
-              "Offerer must use actpass value for setup attribute.",
-              error_desc);
-        }
-
-        if (local_connection_role == CONNECTIONROLE_ACTIVE ||
-            local_connection_role == CONNECTIONROLE_PASSIVE) {
-          is_remote_server = (local_connection_role == CONNECTIONROLE_ACTIVE);
-        } else {
-          const std::string msg =
-              "Answerer must use either active or passive value "
-              "for setup attribute.";
-          return BadTransportDescription(msg, error_desc);
-        }
-
-        // If local is passive, local will act as server.
+      if (!Base::NegotiateRole(local_role, &secure_role_, error_desc)) {
+        return false;
       }
-
-      secure_role_ = is_remote_server ? rtc::SSL_CLIENT :
-                                        rtc::SSL_SERVER;
-
     } else if (local_fp && (local_role == CA_ANSWER)) {
       return BadTransportDescription(
           "Local fingerprint supplied when caller didn't offer DTLS.",
           error_desc);
     } else {
       // We are not doing DTLS
-      remote_fingerprint_.reset(new rtc::SSLFingerprint(
-          "", NULL, 0));
+      remote_fingerprint_.reset(new rtc::SSLFingerprint("", nullptr, 0));
     }
-
     // Now run the negotiation for the base class.
     return Base::NegotiateTransportDescription(local_role, error_desc);
   }
@@ -242,7 +155,7 @@ class DtlsTransport : public Base {
   rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
   rtc::SSLRole secure_role_;
   rtc::SSLProtocolVersion ssl_max_version_;
-  rtc::scoped_ptr<rtc::SSLFingerprint> remote_fingerprint_;
+  std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint_;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.cc b/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.cc
index 88a11928cc2..1e3df13493f 100644
--- a/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.cc
+++ b/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <utility>
 
 #include "webrtc/p2p/base/dtlstransportchannel.h"
@@ -36,6 +37,13 @@ static bool IsDtlsPacket(const char* data, size_t len) {
   const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
   return (len >= kDtlsRecordHeaderLen && (u[0] > 19 && u[0] < 64));
 }
+static bool IsDtlsClientHelloPacket(const char* data, size_t len) {
+  if (!IsDtlsPacket(data, len)) {
+    return false;
+  }
+  const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
+  return len > 17 && u[0] == 22 && u[13] == 1;
+}
 static bool IsRtpPacket(const char* data, size_t len) {
   const uint8_t* u = reinterpret_cast<const uint8_t*>(data);
   return (len >= kMinRtpPacketLen && (u[0] & 0xC0) == 0x80);
@@ -250,7 +258,7 @@ bool DtlsTransportChannelWrapper::SetRemoteFingerprint(
   return true;
 }
 
-rtc::scoped_ptr<rtc::SSLCertificate>
+std::unique_ptr<rtc::SSLCertificate>
 DtlsTransportChannelWrapper::GetRemoteSSLCertificate() const {
   if (!dtls_) {
     return nullptr;
@@ -469,15 +477,18 @@ void DtlsTransportChannelWrapper::OnReadPacket(
   switch (dtls_state()) {
     case DTLS_TRANSPORT_NEW:
       if (dtls_) {
-        // Drop packets received before DTLS has actually started.
-        LOG_J(LS_INFO, this) << "Dropping packet received before DTLS started.";
+        LOG_J(LS_INFO, this) << "Packet received before DTLS started.";
+      } else {
+        LOG_J(LS_WARNING, this) << "Packet received before we know if we are "
+                                << "doing DTLS or not.";
+      }
+      // Cache a client hello packet received before DTLS has actually started.
+      if (IsDtlsClientHelloPacket(data, size)) {
+        LOG_J(LS_INFO, this) << "Caching DTLS ClientHello packet until DTLS is "
+                             << "started.";
+        cached_client_hello_.SetData(data, size);
       } else {
-        // Currently drop the packet, but we might in future
-        // decide to take this as evidence that the other
-        // side is ready to do DTLS and start the handshake
-        // on our end.
-        LOG_J(LS_WARNING, this) << "Received packet before we know if we are "
-                                << "doing DTLS or not; dropping.";
+        LOG_J(LS_INFO, this) << "Not a DTLS ClientHello packet; dropping.";
       }
       break;
 
@@ -576,6 +587,21 @@ bool DtlsTransportChannelWrapper::MaybeStartDtls() {
     LOG_J(LS_INFO, this)
       << "DtlsTransportChannelWrapper: Started DTLS handshake";
     set_dtls_state(DTLS_TRANSPORT_CONNECTING);
+    // Now that the handshake has started, we can process a cached ClientHello
+    // (if one exists).
+    if (cached_client_hello_.size()) {
+      if (ssl_role_ == rtc::SSL_SERVER) {
+        LOG_J(LS_INFO, this) << "Handling cached DTLS ClientHello packet.";
+        if (!HandleDtlsPacket(cached_client_hello_.data<char>(),
+                              cached_client_hello_.size())) {
+          LOG_J(LS_ERROR, this) << "Failed to handle DTLS packet.";
+        }
+      } else {
+        LOG_J(LS_WARNING, this) << "Discarding cached DTLS ClientHello packet "
+                                << "because we don't have the server role.";
+      }
+      cached_client_hello_.Clear();
+    }
   }
   return true;
 }
diff --git a/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.h b/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.h
index b6c3cfd2510..c5f55469f29 100644
--- a/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.h
+++ b/chromium/third_party/webrtc/p2p/base/dtlstransportchannel.h
@@ -11,13 +11,14 @@
 #ifndef WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
 #define WEBRTC_P2P_BASE_DTLSTRANSPORTCHANNEL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/p2p/base/transportchannelimpl.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/bufferqueue.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stream.h"
 
@@ -137,7 +138,7 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
 
   // Once DTLS has been established, this method retrieves the certificate in
   // use by the remote peer, for use in external identity verification.
-  rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate() const override;
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate() const override;
 
   // Once DTLS has established (i.e., this channel is writable), this method
   // extracts the keys negotiated during the DTLS handshake, for use in external
@@ -226,7 +227,7 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
   rtc::Thread* worker_thread_;  // Everything should occur on this thread.
   // Underlying channel, not owned by this class.
   TransportChannelImpl* const channel_;
-  rtc::scoped_ptr<rtc::SSLStreamAdapter> dtls_;  // The DTLS stream
+  std::unique_ptr<rtc::SSLStreamAdapter> dtls_;  // The DTLS stream
   StreamInterfaceChannel* downward_;  // Wrapper for channel_, owned by dtls_.
   std::vector<int> srtp_ciphers_;     // SRTP ciphers to use with DTLS.
   bool dtls_active_ = false;
@@ -236,6 +237,12 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
   rtc::Buffer remote_fingerprint_value_;
   std::string remote_fingerprint_algorithm_;
 
+  // Cached DTLS ClientHello packet that was received before we started the
+  // DTLS handshake. This could happen if the hello was received before the
+  // transport channel became writable, or before a remote fingerprint was
+  // received.
+  rtc::Buffer cached_client_hello_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(DtlsTransportChannelWrapper);
 };
 
diff --git a/chromium/third_party/webrtc/p2p/base/dtlstransportchannel_unittest.cc b/chromium/third_party/webrtc/p2p/base/dtlstransportchannel_unittest.cc
index 7643016b6a5..486b51aec97 100644
--- a/chromium/third_party/webrtc/p2p/base/dtlstransportchannel_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/dtlstransportchannel_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <set>
 
 #include "webrtc/p2p/base/dtlstransport.h"
@@ -16,7 +17,6 @@
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/sslstreamadapter.h"
@@ -33,28 +33,37 @@ static const char kIcePwd1[] = "TESTICEPWD00000000000001";
 static const size_t kPacketNumOffset = 8;
 static const size_t kPacketHeaderLen = 12;
 static const int kFakePacketId = 0x1234;
+static const int kTimeout = 10000;
 
 static bool IsRtpLeadByte(uint8_t b) {
   return ((b & 0xC0) == 0x80);
 }
 
+cricket::TransportDescription MakeTransportDescription(
+    const rtc::scoped_refptr<rtc::RTCCertificate>& cert,
+    cricket::ConnectionRole role) {
+  std::unique_ptr<rtc::SSLFingerprint> fingerprint;
+  if (cert) {
+    std::string digest_algorithm;
+    cert->ssl_certificate().GetSignatureDigestAlgorithm(&digest_algorithm);
+    fingerprint.reset(
+        rtc::SSLFingerprint::Create(digest_algorithm, cert->identity()));
+  }
+  return cricket::TransportDescription(std::vector<std::string>(), kIceUfrag1,
+                                       kIcePwd1, cricket::ICEMODE_FULL, role,
+                                       fingerprint.get());
+}
+
 using cricket::ConnectionRole;
 
 enum Flags { NF_REOFFER = 0x1, NF_EXPECT_FAILURE = 0x2 };
 
 class DtlsTestClient : public sigslot::has_slots<> {
  public:
-  DtlsTestClient(const std::string& name)
-      : name_(name),
-        packet_size_(0),
-        use_dtls_srtp_(false),
-        ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
-        negotiated_dtls_(false),
-        received_dtls_client_hello_(false),
-        received_dtls_server_hello_(false) {}
+  DtlsTestClient(const std::string& name) : name_(name) {}
   void CreateCertificate(rtc::KeyType key_type) {
     certificate_ =
-        rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+        rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
             rtc::SSLIdentity::Generate(name_, key_type)));
   }
   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate() {
@@ -122,8 +131,8 @@ class DtlsTestClient : public sigslot::has_slots<> {
                  ConnectionRole local_role,
                  ConnectionRole remote_role,
                  int flags) {
-    rtc::scoped_ptr<rtc::SSLFingerprint> local_fingerprint;
-    rtc::scoped_ptr<rtc::SSLFingerprint> remote_fingerprint;
+    std::unique_ptr<rtc::SSLFingerprint> local_fingerprint;
+    std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint;
     if (local_cert) {
       std::string digest_algorithm;
       ASSERT_TRUE(local_cert->ssl_certificate().GetSignatureDigestAlgorithm(
@@ -185,9 +194,8 @@ class DtlsTestClient : public sigslot::has_slots<> {
     negotiated_dtls_ = (local_cert && remote_cert);
   }
 
-  bool Connect(DtlsTestClient* peer) {
-    transport_->ConnectChannels();
-    transport_->SetDestination(peer->transport_.get());
+  bool Connect(DtlsTestClient* peer, bool asymmetric) {
+    transport_->SetDestination(peer->transport_.get(), asymmetric);
     return true;
   }
 
@@ -203,13 +211,29 @@ class DtlsTestClient : public sigslot::has_slots<> {
     return true;
   }
 
+  bool all_raw_channels_writable() const {
+    if (channels_.empty()) {
+      return false;
+    }
+    for (cricket::DtlsTransportChannelWrapper* channel : channels_) {
+      if (!channel->channel()->writable()) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  int received_dtls_client_hellos() const {
+    return received_dtls_client_hellos_;
+  }
+
   void CheckRole(rtc::SSLRole role) {
     if (role == rtc::SSL_CLIENT) {
-      ASSERT_FALSE(received_dtls_client_hello_);
-      ASSERT_TRUE(received_dtls_server_hello_);
+      ASSERT_EQ(0, received_dtls_client_hellos_);
+      ASSERT_GT(received_dtls_server_hellos_, 0);
     } else {
-      ASSERT_TRUE(received_dtls_client_hello_);
-      ASSERT_FALSE(received_dtls_server_hello_);
+      ASSERT_GT(received_dtls_client_hellos_, 0);
+      ASSERT_EQ(0, received_dtls_server_hellos_);
     }
   }
 
@@ -248,7 +272,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
 
   void SendPackets(size_t channel, size_t size, size_t count, bool srtp) {
     ASSERT(channel < channels_.size());
-    rtc::scoped_ptr<char[]> packet(new char[size]);
+    std::unique_ptr<char[]> packet(new char[size]);
     size_t sent = 0;
     do {
       // Fill the packet with a known value and a sequence number to check
@@ -272,7 +296,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
 
   int SendInvalidSrtpPacket(size_t channel, size_t size) {
     ASSERT(channel < channels_.size());
-    rtc::scoped_ptr<char[]> packet(new char[size]);
+    std::unique_ptr<char[]> packet(new char[size]);
     // Fill the packet with 0 to form an invalid SRTP packet.
     memset(packet.get(), 0, size);
 
@@ -358,20 +382,18 @@ class DtlsTestClient : public sigslot::has_slots<> {
 
     // Look at the handshake packets to see what role we played.
     // Check that non-handshake packets are DTLS data or SRTP bypass.
-    if (negotiated_dtls_) {
-      if (data[0] == 22 && size > 17) {
-        if (data[13] == 1) {
-          received_dtls_client_hello_ = true;
-        } else if (data[13] == 2) {
-          received_dtls_server_hello_ = true;
-        }
-      } else if (!(data[0] >= 20 && data[0] <= 22)) {
-        ASSERT_TRUE(data[0] == 23 || IsRtpLeadByte(data[0]));
-        if (data[0] == 23) {
-          ASSERT_TRUE(VerifyEncryptedPacket(data, size));
-        } else if (IsRtpLeadByte(data[0])) {
-          ASSERT_TRUE(VerifyPacket(data, size, NULL));
-        }
+    if (data[0] == 22 && size > 17) {
+      if (data[13] == 1) {
+        ++received_dtls_client_hellos_;
+      } else if (data[13] == 2) {
+        ++received_dtls_server_hellos_;
+      }
+    } else if (negotiated_dtls_ && !(data[0] >= 20 && data[0] <= 22)) {
+      ASSERT_TRUE(data[0] == 23 || IsRtpLeadByte(data[0]));
+      if (data[0] == 23) {
+        ASSERT_TRUE(VerifyEncryptedPacket(data, size));
+      } else if (IsRtpLeadByte(data[0])) {
+        ASSERT_TRUE(VerifyPacket(data, size, NULL));
       }
     }
   }
@@ -379,15 +401,15 @@ class DtlsTestClient : public sigslot::has_slots<> {
  private:
   std::string name_;
   rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
-  rtc::scoped_ptr<cricket::FakeTransport> transport_;
+  std::unique_ptr<cricket::FakeTransport> transport_;
   std::vector<cricket::DtlsTransportChannelWrapper*> channels_;
-  size_t packet_size_;
+  size_t packet_size_ = 0u;
   std::set<int> received_;
-  bool use_dtls_srtp_;
-  rtc::SSLProtocolVersion ssl_max_version_;
-  bool negotiated_dtls_;
-  bool received_dtls_client_hello_;
-  bool received_dtls_server_hello_;
+  bool use_dtls_srtp_ = false;
+  rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
+  bool negotiated_dtls_ = false;
+  int received_dtls_client_hellos_ = 0;
+  int received_dtls_server_hellos_ = 0;
   rtc::SentPacket sent_packet_;
 };
 
@@ -437,14 +459,14 @@ class DtlsTransportChannelTest : public testing::Test {
   bool Connect(ConnectionRole client1_role, ConnectionRole client2_role) {
     Negotiate(client1_role, client2_role);
 
-    bool rv = client1_.Connect(&client2_);
+    bool rv = client1_.Connect(&client2_, false);
     EXPECT_TRUE(rv);
     if (!rv)
       return false;
 
     EXPECT_TRUE_WAIT(
         client1_.all_channels_writable() && client2_.all_channels_writable(),
-        10000);
+        kTimeout);
     if (!client1_.all_channels_writable() || !client2_.all_channels_writable())
       return false;
 
@@ -535,7 +557,7 @@ class DtlsTransportChannelTest : public testing::Test {
     LOG(LS_INFO) << "Expect packets, size=" << size;
     client2_.ExpectPackets(channel, size);
     client1_.SendPackets(channel, size, count, srtp);
-    EXPECT_EQ_WAIT(count, client2_.NumPacketsReceived(), 10000);
+    EXPECT_EQ_WAIT(count, client2_.NumPacketsReceived(), kTimeout);
   }
 
  protected:
@@ -828,11 +850,11 @@ TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
 
   Renegotiate(&client1_, cricket::CONNECTIONROLE_ACTPASS,
               cricket::CONNECTIONROLE_ACTIVE, NF_REOFFER);
-  bool rv = client1_.Connect(&client2_);
+  bool rv = client1_.Connect(&client2_, false);
   EXPECT_TRUE(rv);
   EXPECT_TRUE_WAIT(
       client1_.all_channels_writable() && client2_.all_channels_writable(),
-      10000);
+      kTimeout);
 
   TestTransfer(0, 1000, 100, true);
   TestTransfer(1, 1000, 100, true);
@@ -846,8 +868,8 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) {
 
   rtc::scoped_refptr<rtc::RTCCertificate> certificate1;
   rtc::scoped_refptr<rtc::RTCCertificate> certificate2;
-  rtc::scoped_ptr<rtc::SSLCertificate> remote_cert1;
-  rtc::scoped_ptr<rtc::SSLCertificate> remote_cert2;
+  std::unique_ptr<rtc::SSLCertificate> remote_cert1;
+  std::unique_ptr<rtc::SSLCertificate> remote_cert2;
 
   // After negotiation, each side has a distinct local certificate, but still no
   // remote certificate, because connection has not yet occurred.
@@ -875,14 +897,80 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) {
             certificate2->ssl_certificate().ToPEMString());
 
   // Each side's remote certificate is the other side's local certificate.
-  rtc::scoped_ptr<rtc::SSLCertificate> remote_cert1 =
+  std::unique_ptr<rtc::SSLCertificate> remote_cert1 =
       client1_.transport()->GetRemoteSSLCertificate();
   ASSERT_TRUE(remote_cert1);
   ASSERT_EQ(remote_cert1->ToPEMString(),
             certificate2->ssl_certificate().ToPEMString());
-  rtc::scoped_ptr<rtc::SSLCertificate> remote_cert2 =
+  std::unique_ptr<rtc::SSLCertificate> remote_cert2 =
       client2_.transport()->GetRemoteSSLCertificate();
   ASSERT_TRUE(remote_cert2);
   ASSERT_EQ(remote_cert2->ToPEMString(),
             certificate1->ssl_certificate().ToPEMString());
 }
+
+// Test that DTLS completes promptly if a ClientHello is received before the
+// transport channel is writable (allowing a ServerHello to be sent).
+TEST_F(DtlsTransportChannelTest, TestReceiveClientHelloBeforeWritable) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
+  // Exchange transport descriptions.
+  Negotiate(cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTIVE);
+
+  // Make client2_ writable, but not client1_.
+  EXPECT_TRUE(client2_.Connect(&client1_, true));
+  EXPECT_TRUE_WAIT(client2_.all_raw_channels_writable(), kTimeout);
+
+  // Expect a DTLS ClientHello to be sent even while client1_ isn't writable.
+  EXPECT_EQ_WAIT(1, client1_.received_dtls_client_hellos(), kTimeout);
+  EXPECT_FALSE(client1_.all_raw_channels_writable());
+
+  // Now make client1_ writable and expect the handshake to complete
+  // without client2_ needing to retransmit the ClientHello.
+  EXPECT_TRUE(client1_.Connect(&client2_, true));
+  EXPECT_TRUE_WAIT(
+      client1_.all_channels_writable() && client2_.all_channels_writable(),
+      kTimeout);
+  EXPECT_EQ(1, client1_.received_dtls_client_hellos());
+}
+
+// Test that DTLS completes promptly if a ClientHello is received before the
+// transport channel has a remote fingerprint (allowing a ServerHello to be
+// sent).
+TEST_F(DtlsTransportChannelTest,
+       TestReceiveClientHelloBeforeRemoteFingerprint) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
+  client1_.SetupChannels(channel_ct_, cricket::ICEROLE_CONTROLLING);
+  client2_.SetupChannels(channel_ct_, cricket::ICEROLE_CONTROLLED);
+
+  // Make client2_ writable and give it local/remote certs, but don't yet give
+  // client1_ a remote fingerprint.
+  client1_.transport()->SetLocalTransportDescription(
+      MakeTransportDescription(client1_.certificate(),
+                               cricket::CONNECTIONROLE_ACTPASS),
+      cricket::CA_OFFER, nullptr);
+  client2_.Negotiate(&client1_, cricket::CA_ANSWER,
+                     cricket::CONNECTIONROLE_ACTIVE,
+                     cricket::CONNECTIONROLE_ACTPASS, 0);
+  EXPECT_TRUE(client2_.Connect(&client1_, true));
+  EXPECT_TRUE_WAIT(client2_.all_raw_channels_writable(), kTimeout);
+
+  // Expect a DTLS ClientHello to be sent even while client1_ doesn't have a
+  // remote fingerprint.
+  EXPECT_EQ_WAIT(1, client1_.received_dtls_client_hellos(), kTimeout);
+  EXPECT_FALSE(client1_.all_raw_channels_writable());
+
+  // Now make give client1_ its remote fingerprint and make it writable, and
+  // expect the handshake to complete without client2_ needing to retransmit
+  // the ClientHello.
+  client1_.transport()->SetRemoteTransportDescription(
+      MakeTransportDescription(client2_.certificate(),
+                               cricket::CONNECTIONROLE_ACTIVE),
+      cricket::CA_ANSWER, nullptr);
+  EXPECT_TRUE(client1_.Connect(&client2_, true));
+  EXPECT_TRUE_WAIT(
+      client1_.all_channels_writable() && client2_.all_channels_writable(),
+      kTimeout);
+  EXPECT_EQ(1, client1_.received_dtls_client_hellos());
+}
diff --git a/chromium/third_party/webrtc/p2p/client/fakeportallocator.h b/chromium/third_party/webrtc/p2p/base/fakeportallocator.h
index fb188261a27..6730d1ebb82 100644
--- a/chromium/third_party/webrtc/p2p/client/fakeportallocator.h
+++ b/chromium/third_party/webrtc/p2p/base/fakeportallocator.h
@@ -8,14 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_
-#define WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_
+#ifndef WEBRTC_P2P_BASE_FAKEPORTALLOCATOR_H_
+#define WEBRTC_P2P_BASE_FAKEPORTALLOCATOR_H_
 
+#include <memory>
 #include <string>
+#include <vector>
+
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/udpport.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 class SocketFactory;
@@ -82,24 +84,31 @@ class TestUDPPort : public UDPPort {
 
 class FakePortAllocatorSession : public PortAllocatorSession {
  public:
-  FakePortAllocatorSession(rtc::Thread* worker_thread,
+  FakePortAllocatorSession(PortAllocator* allocator,
+                           rtc::Thread* worker_thread,
                            rtc::PacketSocketFactory* factory,
                            const std::string& content_name,
                            int component,
                            const std::string& ice_ufrag,
                            const std::string& ice_pwd)
-      : PortAllocatorSession(content_name, component, ice_ufrag, ice_pwd,
-                             cricket::kDefaultPortAllocatorFlags),
+      : PortAllocatorSession(content_name,
+                             component,
+                             ice_ufrag,
+                             ice_pwd,
+                             allocator->flags()),
         worker_thread_(worker_thread),
         factory_(factory),
-        network_("network", "unittest",
-                 rtc::IPAddress(INADDR_LOOPBACK), 8),
-        port_(), running_(false),
-        port_config_count_(0) {
+        network_("network", "unittest", rtc::IPAddress(INADDR_LOOPBACK), 8),
+        port_(),
+        running_(false),
+        port_config_count_(0),
+        stun_servers_(allocator->stun_servers()),
+        turn_servers_(allocator->turn_servers()),
+        candidate_filter_(allocator->candidate_filter()) {
     network_.AddIP(rtc::IPAddress(INADDR_LOOPBACK));
   }
 
-  virtual void StartGettingPorts() {
+  void StartGettingPorts() override {
     if (!port_) {
       port_.reset(TestUDPPort::Create(worker_thread_, factory_, &network_,
                                       network_.GetBestIP(), 0, 0, username(),
@@ -110,32 +119,70 @@ class FakePortAllocatorSession : public PortAllocatorSession {
     running_ = true;
   }
 
-  virtual void StopGettingPorts() { running_ = false; }
-  virtual bool IsGettingPorts() { return running_; }
-  virtual void ClearGettingPorts() {}
+  void StopGettingPorts() override { running_ = false; }
+  bool IsGettingPorts() override { return running_; }
+  void ClearGettingPorts() override {}
+  std::vector<PortInterface*> ReadyPorts() const override {
+    return ready_ports_;
+  }
+  std::vector<Candidate> ReadyCandidates() const override {
+    return candidates_;
+  }
+  bool CandidatesAllocationDone() const override { return allocation_done_; }
 
   int port_config_count() { return port_config_count_; }
 
+  const ServerAddresses& stun_servers() const { return stun_servers_; }
+
+  const std::vector<RelayServerConfig>& turn_servers() const {
+    return turn_servers_;
+  }
+
+  uint32_t candidate_filter() const { return candidate_filter_; }
+
   void AddPort(cricket::Port* port) {
-    port->set_component(component_);
-    port->set_generation(0);
-    port->SignalPortComplete.connect(
-        this, &FakePortAllocatorSession::OnPortComplete);
+    port->set_component(component());
+    port->set_generation(generation());
+    port->SignalPortComplete.connect(this,
+                                     &FakePortAllocatorSession::OnPortComplete);
     port->PrepareAddress();
+    ready_ports_.push_back(port);
     SignalPortReady(this, port);
   }
   void OnPortComplete(cricket::Port* port) {
-    SignalCandidatesReady(this, port->Candidates());
+    const std::vector<Candidate>& candidates = port->Candidates();
+    candidates_.insert(candidates_.end(), candidates.begin(), candidates.end());
+    SignalCandidatesReady(this, candidates);
+
+    allocation_done_ = true;
     SignalCandidatesAllocationDone(this);
   }
 
+  int transport_info_update_count() const {
+    return transport_info_update_count_;
+  }
+
+ protected:
+  void UpdateIceParametersInternal() override {
+    // Since this class is a fake and this method only is overridden for tests,
+    // we don't need to actually update the transport info.
+    ++transport_info_update_count_;
+  }
+
  private:
   rtc::Thread* worker_thread_;
   rtc::PacketSocketFactory* factory_;
   rtc::Network network_;
-  rtc::scoped_ptr<cricket::Port> port_;
+  std::unique_ptr<cricket::Port> port_;
   bool running_;
   int port_config_count_;
+  std::vector<Candidate> candidates_;
+  std::vector<PortInterface*> ready_ports_;
+  bool allocation_done_ = false;
+  ServerAddresses stun_servers_;
+  std::vector<RelayServerConfig> turn_servers_;
+  uint32_t candidate_filter_;
+  int transport_info_update_count_ = 0;
 };
 
 class FakePortAllocator : public cricket::PortAllocator {
@@ -144,44 +191,29 @@ class FakePortAllocator : public cricket::PortAllocator {
                     rtc::PacketSocketFactory* factory)
       : worker_thread_(worker_thread), factory_(factory) {
     if (factory_ == NULL) {
-      owned_factory_.reset(new rtc::BasicPacketSocketFactory(
-          worker_thread_));
+      owned_factory_.reset(new rtc::BasicPacketSocketFactory(worker_thread_));
       factory_ = owned_factory_.get();
     }
   }
 
-  void SetIceServers(
-      const ServerAddresses& stun_servers,
-      const std::vector<RelayServerConfig>& turn_servers) override {
-    stun_servers_ = stun_servers;
-    turn_servers_ = turn_servers;
-  }
-
   void SetNetworkIgnoreMask(int network_ignore_mask) override {}
 
-  const ServerAddresses& stun_servers() const { return stun_servers_; }
-
-  const std::vector<RelayServerConfig>& turn_servers() const {
-    return turn_servers_;
-  }
-
-  virtual cricket::PortAllocatorSession* CreateSessionInternal(
+  cricket::PortAllocatorSession* CreateSessionInternal(
       const std::string& content_name,
       int component,
       const std::string& ice_ufrag,
       const std::string& ice_pwd) override {
-    return new FakePortAllocatorSession(
-        worker_thread_, factory_, content_name, component, ice_ufrag, ice_pwd);
+    return new FakePortAllocatorSession(this, worker_thread_, factory_,
+                                        content_name, component, ice_ufrag,
+                                        ice_pwd);
   }
 
  private:
   rtc::Thread* worker_thread_;
   rtc::PacketSocketFactory* factory_;
-  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> owned_factory_;
-  ServerAddresses stun_servers_;
-  std::vector<RelayServerConfig> turn_servers_;
+  std::unique_ptr<rtc::BasicPacketSocketFactory> owned_factory_;
 };
 
 }  // namespace cricket
 
-#endif  // WEBRTC_P2P_CLIENT_FAKEPORTALLOCATOR_H_
+#endif  // WEBRTC_P2P_BASE_FAKEPORTALLOCATOR_H_
diff --git a/chromium/third_party/webrtc/p2p/base/faketransportcontroller.h b/chromium/third_party/webrtc/p2p/base/faketransportcontroller.h
index 2e0c9a97fad..321537dd59b 100644
--- a/chromium/third_party/webrtc/p2p/base/faketransportcontroller.h
+++ b/chromium/third_party/webrtc/p2p/base/faketransportcontroller.h
@@ -12,6 +12,7 @@
 #define WEBRTC_P2P_BASE_FAKETRANSPORTCONTROLLER_H_
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -140,20 +141,22 @@ class FakeTransportChannel : public TransportChannelImpl,
 
   void SetWritable(bool writable) { set_writable(writable); }
 
-  void SetDestination(FakeTransportChannel* dest) {
+  // Simulates the two transport channels connecting to each other.
+  // If |asymmetric| is true this method only affects this FakeTransportChannel.
+  // If false, it affects |dest| as well.
+  void SetDestination(FakeTransportChannel* dest, bool asymmetric = false) {
     if (state_ == STATE_CONNECTING && dest) {
       // This simulates the delivery of candidates.
       dest_ = dest;
-      dest_->dest_ = this;
       if (local_cert_ && dest_->local_cert_) {
         do_dtls_ = true;
-        dest_->do_dtls_ = true;
         NegotiateSrtpCiphers();
       }
       state_ = STATE_CONNECTED;
-      dest_->state_ = STATE_CONNECTED;
       set_writable(true);
-      dest_->set_writable(true);
+      if (!asymmetric) {
+        dest->SetDestination(this, true);
+      }
     } else if (state_ == STATE_CONNECTED && !dest) {
       // Simulates loss of connectivity, by asymmetrically forgetting dest_.
       dest_ = nullptr;
@@ -206,7 +209,7 @@ class FakeTransportChannel : public TransportChannelImpl,
     } else {
       rtc::Thread::Current()->Send(this, 0, packet);
     }
-    rtc::SentPacket sent_packet(options.packet_id, rtc::Time64());
+    rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
     SignalSentPacket(this, sent_packet);
     return static_cast<int>(len);
   }
@@ -230,7 +233,7 @@ class FakeTransportChannel : public TransportChannelImpl,
   }
 
   bool SetLocalCertificate(
-      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override {
     local_cert_ = certificate;
     return true;
   }
@@ -256,13 +259,13 @@ class FakeTransportChannel : public TransportChannelImpl,
 
   bool GetSslCipherSuite(int* cipher_suite) override { return false; }
 
-  rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const {
+  rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override {
     return local_cert_;
   }
 
-  rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
       const override {
-    return remote_cert_ ? rtc::scoped_ptr<rtc::SSLCertificate>(
+    return remote_cert_ ? std::unique_ptr<rtc::SSLCertificate>(
                               remote_cert_->GetReference())
                         : nullptr;
   }
@@ -281,20 +284,6 @@ class FakeTransportChannel : public TransportChannelImpl,
     return false;
   }
 
-  void NegotiateSrtpCiphers() {
-    for (std::vector<int>::const_iterator it1 = srtp_ciphers_.begin();
-         it1 != srtp_ciphers_.end(); ++it1) {
-      for (std::vector<int>::const_iterator it2 = dest_->srtp_ciphers_.begin();
-           it2 != dest_->srtp_ciphers_.end(); ++it2) {
-        if (*it1 == *it2) {
-          chosen_crypto_suite_ = *it1;
-          dest_->chosen_crypto_suite_ = *it2;
-          return;
-        }
-      }
-    }
-  }
-
   bool GetStats(ConnectionInfos* infos) override {
     ConnectionInfo info;
     infos->clear();
@@ -310,6 +299,19 @@ class FakeTransportChannel : public TransportChannelImpl,
   }
 
  private:
+  void NegotiateSrtpCiphers() {
+    for (std::vector<int>::const_iterator it1 = srtp_ciphers_.begin();
+         it1 != srtp_ciphers_.end(); ++it1) {
+      for (std::vector<int>::const_iterator it2 = dest_->srtp_ciphers_.begin();
+           it2 != dest_->srtp_ciphers_.end(); ++it2) {
+        if (*it1 == *it2) {
+          chosen_crypto_suite_ = *it1;
+          return;
+        }
+      }
+    }
+  }
+
   enum State { STATE_INIT, STATE_CONNECTING, STATE_CONNECTED };
   FakeTransportChannel* dest_ = nullptr;
   State state_ = STATE_INIT;
@@ -358,11 +360,14 @@ class FakeTransport : public Transport {
   // If async, will send packets by "Post"-ing to message queue instead of
   // synchronously "Send"-ing.
   void SetAsync(bool async) { async_ = async; }
-  void SetDestination(FakeTransport* dest) {
+
+  // If |asymmetric| is true, only set the destination for this transport, and
+  // not |dest|.
+  void SetDestination(FakeTransport* dest, bool asymmetric = false) {
     dest_ = dest;
     for (const auto& kv : channels_) {
       kv.second->SetLocalCertificate(certificate_);
-      SetChannelDestination(kv.first, kv.second);
+      SetChannelDestination(kv.first, kv.second, asymmetric);
     }
   }
 
@@ -405,6 +410,8 @@ class FakeTransport : public Transport {
 
   using Transport::local_description;
   using Transport::remote_description;
+  using Transport::VerifyCertificateFingerprint;
+  using Transport::NegotiateRole;
 
  protected:
   TransportChannelImpl* CreateTransportChannel(int component) override {
@@ -414,7 +421,7 @@ class FakeTransport : public Transport {
     FakeTransportChannel* channel = new FakeTransportChannel(name(), component);
     channel->set_ssl_max_protocol_version(ssl_max_version_);
     channel->SetAsync(async_);
-    SetChannelDestination(component, channel);
+    SetChannelDestination(component, channel, false);
     channels_[component] = channel;
     return channel;
   }
@@ -430,15 +437,17 @@ class FakeTransport : public Transport {
     return (it != channels_.end()) ? it->second : nullptr;
   }
 
-  void SetChannelDestination(int component, FakeTransportChannel* channel) {
+  void SetChannelDestination(int component,
+                             FakeTransportChannel* channel,
+                             bool asymmetric) {
     FakeTransportChannel* dest_channel = nullptr;
     if (dest_) {
       dest_channel = dest_->GetFakeChannel(component);
-      if (dest_channel) {
+      if (dest_channel && !asymmetric) {
         dest_channel->SetLocalCertificate(dest_->certificate_);
       }
     }
-    channel->SetDestination(dest_channel);
+    channel->SetDestination(dest_channel, asymmetric);
   }
 
   // Note, this is distinct from the Channel map owned by Transport.
@@ -502,22 +511,22 @@ class FakeTransportController : public TransportController {
     SetIceRole(role);
   }
 
-  FakeTransport* GetTransport_w(const std::string& transport_name) {
+  FakeTransport* GetTransport_n(const std::string& transport_name) {
     return static_cast<FakeTransport*>(
-        TransportController::GetTransport_w(transport_name));
+        TransportController::GetTransport_n(transport_name));
   }
 
   void Connect(FakeTransportController* dest) {
-    worker_thread()->Invoke<void>(
-        rtc::Bind(&FakeTransportController::Connect_w, this, dest));
+    network_thread()->Invoke<void>(
+        rtc::Bind(&FakeTransportController::Connect_n, this, dest));
   }
 
-  TransportChannel* CreateTransportChannel_w(const std::string& transport_name,
+  TransportChannel* CreateTransportChannel_n(const std::string& transport_name,
                                              int component) override {
     if (fail_create_channel_) {
       return nullptr;
     }
-    return TransportController::CreateTransportChannel_w(transport_name,
+    return TransportController::CreateTransportChannel_n(transport_name,
                                                          component);
   }
 
@@ -538,23 +547,34 @@ class FakeTransportController : public TransportController {
   }
 
  protected:
-  Transport* CreateTransport_w(const std::string& transport_name) override {
+  Transport* CreateTransport_n(const std::string& transport_name) override {
     return new FakeTransport(transport_name);
   }
 
-  void Connect_w(FakeTransportController* dest) {
+  void Connect_n(FakeTransportController* dest) {
     // Simulate the exchange of candidates.
-    ConnectChannels_w();
-    dest->ConnectChannels_w();
+    ConnectChannels_n();
+    dest->ConnectChannels_n();
     for (auto& kv : transports()) {
       FakeTransport* transport = static_cast<FakeTransport*>(kv.second);
-      transport->SetDestination(dest->GetTransport_w(kv.first));
+      transport->SetDestination(dest->GetTransport_n(kv.first));
     }
   }
 
-  void ConnectChannels_w() {
+  void ConnectChannels_n() {
+    TransportDescription faketransport_desc(
+        std::vector<std::string>(),
+        rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH),
+        rtc::CreateRandomString(cricket::ICE_PWD_LENGTH), cricket::ICEMODE_FULL,
+        cricket::CONNECTIONROLE_NONE, nullptr);
     for (auto& kv : transports()) {
       FakeTransport* transport = static_cast<FakeTransport*>(kv.second);
+      // Set local transport description for FakeTransport before connecting.
+      // Otherwise, the RTC_CHECK in Transport.ConnectChannel will fail.
+      if (!transport->local_description()) {
+        transport->SetLocalTransportDescription(faketransport_desc,
+                                                cricket::CA_OFFER, nullptr);
+      }
       transport->ConnectChannels();
       transport->MaybeStartGathering();
     }
diff --git a/chromium/third_party/webrtc/p2p/base/p2ptransport.h b/chromium/third_party/webrtc/p2p/base/p2ptransport.h
index 0f965b4cdc1..d4da224f979 100644
--- a/chromium/third_party/webrtc/p2p/base/p2ptransport.h
+++ b/chromium/third_party/webrtc/p2p/base/p2ptransport.h
@@ -12,6 +12,8 @@
 #define WEBRTC_P2P_BASE_P2PTRANSPORT_H_
 
 #include <string>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/p2p/base/transport.h"
 
 namespace cricket {
diff --git a/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.cc b/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.cc
index 66532411b29..2801c446461 100644
--- a/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.cc
@@ -261,30 +261,27 @@ P2PTransportChannel::P2PTransportChannel(const std::string& transport_name,
 
 P2PTransportChannel::~P2PTransportChannel() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
-
-  for (size_t i = 0; i < allocator_sessions_.size(); ++i)
-    delete allocator_sessions_[i];
 }
 
 // Add the allocator session to our list so that we know which sessions
 // are still active.
-void P2PTransportChannel::AddAllocatorSession(PortAllocatorSession* session) {
+void P2PTransportChannel::AddAllocatorSession(
+    std::unique_ptr<PortAllocatorSession> session) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
   session->set_generation(static_cast<uint32_t>(allocator_sessions_.size()));
-  allocator_sessions_.push_back(session);
+  session->SignalPortReady.connect(this, &P2PTransportChannel::OnPortReady);
+  session->SignalCandidatesReady.connect(
+      this, &P2PTransportChannel::OnCandidatesReady);
+  session->SignalCandidatesAllocationDone.connect(
+      this, &P2PTransportChannel::OnCandidatesAllocationDone);
 
   // We now only want to apply new candidates that we receive to the ports
   // created by this new session because these are replacing those of the
   // previous sessions.
   ports_.clear();
 
-  session->SignalPortReady.connect(this, &P2PTransportChannel::OnPortReady);
-  session->SignalCandidatesReady.connect(
-      this, &P2PTransportChannel::OnCandidatesReady);
-  session->SignalCandidatesAllocationDone.connect(
-      this, &P2PTransportChannel::OnCandidatesAllocationDone);
-  session->StartGettingPorts();
+  allocator_sessions_.push_back(std::move(session));
 }
 
 void P2PTransportChannel::AddConnection(Connection* connection) {
@@ -390,10 +387,15 @@ void P2PTransportChannel::SetRemoteIceCredentials(const std::string& ice_ufrag,
       candidate.set_password(ice_pwd);
     }
   }
-  // We need to update the credentials for any peer reflexive candidates.
+  // We need to update the credentials and generation for any peer reflexive
+  // candidates.
   for (Connection* conn : connections_) {
-    conn->MaybeSetRemoteIceCredentials(ice_ufrag, ice_pwd);
+    conn->MaybeSetRemoteIceCredentialsAndGeneration(
+        ice_ufrag, ice_pwd,
+        static_cast<int>(remote_ice_parameters_.size() - 1));
   }
+  // Updating the remote ICE candidate generation could change the sort order.
+  RequestSort();
 }
 
 void P2PTransportChannel::SetRemoteIceMode(IceMode mode) {
@@ -467,9 +469,28 @@ void P2PTransportChannel::MaybeStartGathering() {
       gathering_state_ = kIceGatheringGathering;
       SignalGatheringState(this);
     }
-    // Time for a new allocator
-    AddAllocatorSession(allocator_->CreateSession(
-        SessionId(), transport_name(), component(), ice_ufrag_, ice_pwd_));
+    // Time for a new allocator.
+    std::unique_ptr<PortAllocatorSession> pooled_session =
+        allocator_->TakePooledSession(transport_name(), component(), ice_ufrag_,
+                                      ice_pwd_);
+    if (pooled_session) {
+      AddAllocatorSession(std::move(pooled_session));
+      PortAllocatorSession* raw_pooled_session =
+          allocator_sessions_.back().get();
+      // Process the pooled session's existing candidates/ports, if they exist.
+      OnCandidatesReady(raw_pooled_session,
+                        raw_pooled_session->ReadyCandidates());
+      for (PortInterface* port : allocator_sessions_.back()->ReadyPorts()) {
+        OnPortReady(raw_pooled_session, port);
+      }
+      if (allocator_sessions_.back()->CandidatesAllocationDone()) {
+        OnCandidatesAllocationDone(raw_pooled_session);
+      }
+    } else {
+      AddAllocatorSession(allocator_->CreateSession(
+          SessionId(), transport_name(), component(), ice_ufrag_, ice_pwd_));
+      allocator_sessions_.back()->StartGettingPorts();
+    }
   }
 }
 
@@ -1034,7 +1055,7 @@ rtc::DiffServCodePoint P2PTransportChannel::DefaultDscpValue() const {
 
 // Monitor connection states.
 void P2PTransportChannel::UpdateConnectionStates() {
-  int64_t now = rtc::Time64();
+  int64_t now = rtc::TimeMillis();
 
   // We need to copy the list of connections since some may delete themselves
   // when we call UpdateState.
@@ -1208,7 +1229,7 @@ void P2PTransportChannel::MaybeStopPortAllocatorSessions() {
     return;
   }
 
-  for (PortAllocatorSession* session : allocator_sessions_) {
+  for (const auto& session : allocator_sessions_) {
     if (!session->IsGettingPorts()) {
       continue;
     }
@@ -1279,7 +1300,7 @@ void P2PTransportChannel::OnCheckAndPing() {
   // When the best connection is either not receiving or not writable,
   // switch to weak ping interval.
   int ping_interval = weak() ? weak_ping_interval_ : STRONG_PING_INTERVAL;
-  if (rtc::Time64() >= last_ping_sent_ms_ + ping_interval) {
+  if (rtc::TimeMillis() >= last_ping_sent_ms_ + ping_interval) {
     Connection* conn = FindNextPingableConnection();
     if (conn) {
       PingConnection(conn);
@@ -1338,7 +1359,7 @@ bool P2PTransportChannel::IsPingable(Connection* conn, int64_t now) {
 // ping target to become writable instead. See the big comment in
 // CompareConnections.
 Connection* P2PTransportChannel::FindNextPingableConnection() {
-  int64_t now = rtc::Time64();
+  int64_t now = rtc::TimeMillis();
   Connection* conn_to_ping = nullptr;
   if (best_connection_ && best_connection_->connected() &&
       best_connection_->writable() &&
@@ -1379,7 +1400,7 @@ void P2PTransportChannel::PingConnection(Connection* conn) {
     use_candidate = best_connection_->writable();
   }
   conn->set_use_candidate_attr(use_candidate);
-  last_ping_sent_ms_ = rtc::Time64();
+  last_ping_sent_ms_ = rtc::TimeMillis();
   conn->Ping(last_ping_sent_ms_);
 }
 
@@ -1400,10 +1421,14 @@ void P2PTransportChannel::OnConnectionStateChange(Connection* connection) {
   }
 
   // May stop the allocator session when at least one connection becomes
-  // strongly connected after starting to get ports. It is not enough to check
+  // strongly connected after starting to get ports and the local candidate of
+  // the connection is at the latest generation. It is not enough to check
   // that the connection becomes weakly connected because the connection may be
   // changing from (writable, receiving) to (writable, not receiving).
-  if (!connection->weak()) {
+  bool strongly_connected = !connection->weak();
+  bool latest_generation = connection->local_candidate().generation() >=
+                           allocator_session()->generation();
+  if (strongly_connected && latest_generation) {
     MaybeStopPortAllocatorSessions();
   }
 
diff --git a/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.h b/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.h
index 4a53d755af5..b002c8bb9f3 100644
--- a/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.h
+++ b/chromium/third_party/webrtc/p2p/base/p2ptransportchannel.h
@@ -21,9 +21,12 @@
 #define WEBRTC_P2P_BASE_P2PTRANSPORTCHANNEL_H_
 
 #include <map>
+#include <memory>
 #include <set>
 #include <string>
 #include <vector>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/candidatepairinterface.h"
 #include "webrtc/p2p/base/p2ptransport.h"
@@ -144,7 +147,7 @@ class P2PTransportChannel : public TransportChannelImpl,
     return nullptr;
   }
 
-  rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
       const override {
     return nullptr;
   }
@@ -186,7 +189,7 @@ class P2PTransportChannel : public TransportChannelImpl,
 
   // Public for unit tests.
   PortAllocatorSession* allocator_session() {
-    return allocator_sessions_.back();
+    return allocator_sessions_.back().get();
   }
 
   // Public for unit tests.
@@ -224,7 +227,7 @@ class P2PTransportChannel : public TransportChannelImpl,
                                PortInterface* origin_port);
   bool IsPingable(Connection* conn, int64_t now);
   void PingConnection(Connection* conn);
-  void AddAllocatorSession(PortAllocatorSession* session);
+  void AddAllocatorSession(std::unique_ptr<PortAllocatorSession> session);
   void AddConnection(Connection* connection);
 
   void OnPortReady(PortAllocatorSession *session, PortInterface* port);
@@ -292,7 +295,7 @@ class P2PTransportChannel : public TransportChannelImpl,
   rtc::Thread* worker_thread_;
   bool incoming_only_;
   int error_;
-  std::vector<PortAllocatorSession*> allocator_sessions_;
+  std::vector<std::unique_ptr<PortAllocatorSession>> allocator_sessions_;
   std::vector<PortInterface *> ports_;
 
   // |connections_| is a sorted list with the first one always be the
diff --git a/chromium/third_party/webrtc/p2p/base/p2ptransportchannel_unittest.cc b/chromium/third_party/webrtc/p2p/base/p2ptransportchannel_unittest.cc
index 87e92a5b30d..54ab3196cbc 100644
--- a/chromium/third_party/webrtc/p2p/base/p2ptransportchannel_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/p2ptransportchannel_unittest.cc
@@ -8,12 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
+#include <memory>
+
+#include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/p2p/base/p2ptransportchannel.h"
 #include "webrtc/p2p/base/testrelayserver.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/fakenetwork.h"
 #include "webrtc/base/firewallsocketserver.h"
@@ -211,7 +214,7 @@ class P2PTransportChannelTestBase : public testing::Test,
 
     std::string name_;  // TODO - Currently not used.
     std::list<std::string> ch_packets_;
-    rtc::scoped_ptr<cricket::P2PTransportChannel> ch_;
+    std::unique_ptr<cricket::P2PTransportChannel> ch_;
   };
 
   struct CandidatesData : public rtc::MessageData {
@@ -255,7 +258,7 @@ class P2PTransportChannelTestBase : public testing::Test,
     }
 
     rtc::FakeNetworkManager network_manager_;
-    rtc::scoped_ptr<cricket::BasicPortAllocator> allocator_;
+    std::unique_ptr<cricket::BasicPortAllocator> allocator_;
     ChannelData cd1_;
     ChannelData cd2_;
     cricket::IceRole role_;
@@ -285,6 +288,8 @@ class P2PTransportChannelTestBase : public testing::Test,
         1, cricket::ICE_CANDIDATE_COMPONENT_DEFAULT,
         ice_ufrag_ep2_cd1_ch, ice_pwd_ep2_cd1_ch,
         ice_ufrag_ep1_cd1_ch, ice_pwd_ep1_cd1_ch));
+    ep1_.cd1_.ch_->MaybeStartGathering();
+    ep2_.cd1_.ch_->MaybeStartGathering();
     if (num == 2) {
       std::string ice_ufrag_ep1_cd2_ch = kIceUfrag[2];
       std::string ice_pwd_ep1_cd2_ch = kIcePwd[2];
@@ -298,6 +303,8 @@ class P2PTransportChannelTestBase : public testing::Test,
           1, cricket::ICE_CANDIDATE_COMPONENT_DEFAULT,
           ice_ufrag_ep2_cd2_ch, ice_pwd_ep2_cd2_ch,
           ice_ufrag_ep1_cd2_ch, ice_pwd_ep1_cd2_ch));
+      ep1_.cd2_.ch_->MaybeStartGathering();
+      ep2_.cd2_.ch_->MaybeStartGathering();
     }
   }
   cricket::P2PTransportChannel* CreateChannel(
@@ -326,7 +333,6 @@ class P2PTransportChannelTestBase : public testing::Test,
     channel->SetIceRole(GetEndpoint(endpoint)->ice_role());
     channel->SetIceTiebreaker(GetEndpoint(endpoint)->GetIceTiebreaker());
     channel->Connect();
-    channel->MaybeStartGathering();
     return channel;
   }
   void DestroyChannels() {
@@ -504,7 +510,7 @@ class P2PTransportChannelTestBase : public testing::Test,
   }
 
   void Test(const Result& expected) {
-    int64_t connect_start = rtc::Time64();
+    int64_t connect_start = rtc::TimeMillis();
     int64_t connect_time;
 
     // Create the channels and wait for them to connect.
@@ -517,7 +523,7 @@ class P2PTransportChannelTestBase : public testing::Test,
                             ep2_ch1()->writable(),
                             expected.connect_wait,
                             1000);
-    connect_time = rtc::Time64() - connect_start;
+    connect_time = rtc::TimeMillis() - connect_start;
     if (connect_time < expected.connect_wait) {
       LOG(LS_INFO) << "Connect time: " << connect_time << " ms";
     } else {
@@ -529,7 +535,7 @@ class P2PTransportChannelTestBase : public testing::Test,
     // This may take up to 2 seconds.
     if (ep1_ch1()->best_connection() &&
         ep2_ch1()->best_connection()) {
-      int64_t converge_start = rtc::Time64();
+      int64_t converge_start = rtc::TimeMillis();
       int64_t converge_time;
       int64_t converge_wait = 2000;
       EXPECT_TRUE_WAIT_MARGIN(CheckCandidate1(expected), converge_wait,
@@ -547,7 +553,7 @@ class P2PTransportChannelTestBase : public testing::Test,
       // For verbose
       ExpectCandidate2(expected);
 
-      converge_time = rtc::Time64() - converge_start;
+      converge_time = rtc::TimeMillis() - converge_start;
       if (converge_time < converge_wait) {
         LOG(LS_INFO) << "Converge time: " << converge_time << " ms";
       } else {
@@ -702,7 +708,7 @@ class P2PTransportChannelTestBase : public testing::Test,
   void OnMessage(rtc::Message* msg) {
     switch (msg->message_id) {
       case MSG_ADD_CANDIDATES: {
-        rtc::scoped_ptr<CandidatesData> data(
+        std::unique_ptr<CandidatesData> data(
             static_cast<CandidatesData*>(msg->pdata));
         cricket::P2PTransportChannel* rch = GetRemoteChannel(data->channel);
         for (auto& c : data->candidates) {
@@ -717,7 +723,7 @@ class P2PTransportChannelTestBase : public testing::Test,
         break;
       }
       case MSG_REMOVE_CANDIDATES: {
-        rtc::scoped_ptr<CandidatesData> data(
+        std::unique_ptr<CandidatesData> data(
             static_cast<CandidatesData*>(msg->pdata));
         cricket::P2PTransportChannel* rch = GetRemoteChannel(data->channel);
         for (cricket::Candidate& c : data->candidates) {
@@ -797,12 +803,12 @@ class P2PTransportChannelTestBase : public testing::Test,
 
  private:
   rtc::Thread* main_;
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
-  rtc::scoped_ptr<rtc::NATSocketServer> nss_;
-  rtc::scoped_ptr<rtc::FirewallSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
+  std::unique_ptr<rtc::NATSocketServer> nss_;
+  std::unique_ptr<rtc::FirewallSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
-  rtc::scoped_ptr<cricket::TestStunServer> stun_server_;
+  std::unique_ptr<cricket::TestStunServer> stun_server_;
   cricket::TestTurnServer turn_server_;
   cricket::TestRelayServer relay_server_;
   rtc::SocksProxyServer socks_server1_;
@@ -1193,7 +1199,7 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
   set_clear_remote_candidates_ufrag_pwd(false);
   CreateChannels(1);
   // Only have remote credentials come in for ep2, not ep1.
-  ep2_ch1()->SetRemoteIceCredentials(kIceUfrag[3], kIcePwd[3]);
+  ep2_ch1()->SetRemoteIceCredentials(kIceUfrag[0], kIcePwd[0]);
 
   // Pause sending ep2's candidates to ep1 until ep1 receives the peer reflexive
   // candidate.
@@ -1209,15 +1215,23 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
   EXPECT_EQ(kIceUfrag[1],
             ep1_ch1()->best_connection()->remote_candidate().username());
   EXPECT_EQ("", ep1_ch1()->best_connection()->remote_candidate().password());
+  // Because we don't have ICE credentials yet, we don't know the generation.
+  EXPECT_EQ(0u, ep1_ch1()->best_connection()->remote_candidate().generation());
   EXPECT_TRUE(nullptr == ep1_ch1()->FindNextPingableConnection());
 
+  // Add two sets of remote ICE credentials, so that the ones used by the
+  // candidate will be generation 1 instead of 0.
+  ep1_ch1()->SetRemoteIceCredentials(kIceUfrag[3], kIcePwd[3]);
   ep1_ch1()->SetRemoteIceCredentials(kIceUfrag[1], kIcePwd[1]);
-  ResumeCandidates(1);
-
+  // After setting the remote ICE credentials, the password and generation
+  // of the peer reflexive candidate should be updated.
   EXPECT_EQ(kIcePwd[1],
             ep1_ch1()->best_connection()->remote_candidate().password());
+  EXPECT_EQ(1u, ep1_ch1()->best_connection()->remote_candidate().generation());
   EXPECT_TRUE(nullptr != ep1_ch1()->FindNextPingableConnection());
 
+  ResumeCandidates(1);
+
   WAIT(ep2_ch1()->best_connection() != NULL, 2000);
   // Verify ep1's best connection is updated to use the 'local' candidate.
   EXPECT_EQ_WAIT(
@@ -1237,7 +1251,7 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
   set_clear_remote_candidates_ufrag_pwd(false);
   CreateChannels(1);
   // Only have remote credentials come in for ep2, not ep1.
-  ep2_ch1()->SetRemoteIceCredentials(kIceUfrag[3], kIcePwd[3]);
+  ep2_ch1()->SetRemoteIceCredentials(kIceUfrag[0], kIcePwd[0]);
   // Pause sending ep2's candidates to ep1 until ep1 receives the peer reflexive
   // candidate.
   PauseCandidates(1);
@@ -1251,14 +1265,21 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
   EXPECT_EQ(kIceUfrag[1],
             ep1_ch1()->best_connection()->remote_candidate().username());
   EXPECT_EQ("", ep1_ch1()->best_connection()->remote_candidate().password());
+  // Because we don't have ICE credentials yet, we don't know the generation.
+  EXPECT_EQ(0u, ep1_ch1()->best_connection()->remote_candidate().generation());
   EXPECT_TRUE(nullptr == ep1_ch1()->FindNextPingableConnection());
 
+  // Add two sets of remote ICE credentials, so that the ones used by the
+  // candidate will be generation 1 instead of 0.
+  ep1_ch1()->SetRemoteIceCredentials(kIceUfrag[3], kIcePwd[3]);
   ep1_ch1()->SetRemoteIceCredentials(kIceUfrag[1], kIcePwd[1]);
-  ResumeCandidates(1);
-
+  // After setting the remote ICE credentials, the password and generation
+  // of the peer reflexive candidate should be updated.
   EXPECT_EQ(kIcePwd[1],
             ep1_ch1()->best_connection()->remote_candidate().password());
-  EXPECT_TRUE(nullptr != ep1_ch1()->FindNextPingableConnection());
+  EXPECT_EQ(1u, ep1_ch1()->best_connection()->remote_candidate().generation());
+
+  ResumeCandidates(1);
 
   const cricket::Connection* best_connection = NULL;
   WAIT((best_connection = ep2_ch1()->best_connection()) != NULL, 2000);
@@ -1530,6 +1551,92 @@ TEST_F(P2PTransportChannelTest, TestContinualGathering) {
   DestroyChannels();
 }
 
+// Test that a connection succeeds when the P2PTransportChannel uses a pooled
+// PortAllocatorSession that has not yet finished gathering candidates.
+TEST_F(P2PTransportChannelTest, TestUsingPooledSessionBeforeDoneGathering) {
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
+  // First create a pooled session for each endpoint.
+  auto& allocator_1 = GetEndpoint(0)->allocator_;
+  auto& allocator_2 = GetEndpoint(1)->allocator_;
+  int pool_size = 1;
+  allocator_1->SetConfiguration(allocator_1->stun_servers(),
+                                allocator_1->turn_servers(), pool_size);
+  allocator_2->SetConfiguration(allocator_2->stun_servers(),
+                                allocator_2->turn_servers(), pool_size);
+  const cricket::PortAllocatorSession* pooled_session_1 =
+      allocator_1->GetPooledSession();
+  const cricket::PortAllocatorSession* pooled_session_2 =
+      allocator_2->GetPooledSession();
+  ASSERT_NE(nullptr, pooled_session_1);
+  ASSERT_NE(nullptr, pooled_session_2);
+  // Sanity check that pooled sessions haven't gathered anything yet.
+  EXPECT_TRUE(pooled_session_1->ReadyPorts().empty());
+  EXPECT_TRUE(pooled_session_1->ReadyCandidates().empty());
+  EXPECT_TRUE(pooled_session_2->ReadyPorts().empty());
+  EXPECT_TRUE(pooled_session_2->ReadyCandidates().empty());
+  // Now let the endpoints connect and try exchanging some data.
+  CreateChannels(1);
+  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1() != NULL && ep2_ch1() != NULL &&
+                              ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                              ep2_ch1()->receiving() && ep2_ch1()->writable(),
+                          1000, 1000);
+  TestSendRecv(1);
+  // Make sure the P2PTransportChannels are actually using ports from the
+  // pooled sessions.
+  auto pooled_ports_1 = pooled_session_1->ReadyPorts();
+  auto pooled_ports_2 = pooled_session_2->ReadyPorts();
+  EXPECT_NE(pooled_ports_1.end(),
+            std::find(pooled_ports_1.begin(), pooled_ports_1.end(),
+                      ep1_ch1()->best_connection()->port()));
+  EXPECT_NE(pooled_ports_2.end(),
+            std::find(pooled_ports_2.begin(), pooled_ports_2.end(),
+                      ep2_ch1()->best_connection()->port()));
+}
+
+// Test that a connection succeeds when the P2PTransportChannel uses a pooled
+// PortAllocatorSession that already finished gathering candidates.
+TEST_F(P2PTransportChannelTest, TestUsingPooledSessionAfterDoneGathering) {
+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
+                     kDefaultPortAllocatorFlags);
+  // First create a pooled session for each endpoint.
+  auto& allocator_1 = GetEndpoint(0)->allocator_;
+  auto& allocator_2 = GetEndpoint(1)->allocator_;
+  int pool_size = 1;
+  allocator_1->SetConfiguration(allocator_1->stun_servers(),
+                                allocator_1->turn_servers(), pool_size);
+  allocator_2->SetConfiguration(allocator_2->stun_servers(),
+                                allocator_2->turn_servers(), pool_size);
+  const cricket::PortAllocatorSession* pooled_session_1 =
+      allocator_1->GetPooledSession();
+  const cricket::PortAllocatorSession* pooled_session_2 =
+      allocator_2->GetPooledSession();
+  ASSERT_NE(nullptr, pooled_session_1);
+  ASSERT_NE(nullptr, pooled_session_2);
+  // Wait for the pooled sessions to finish gathering before the
+  // P2PTransportChannels try to use them.
+  EXPECT_TRUE_WAIT(pooled_session_1->CandidatesAllocationDone() &&
+                       pooled_session_2->CandidatesAllocationDone(),
+                   kDefaultTimeout);
+  // Now let the endpoints connect and try exchanging some data.
+  CreateChannels(1);
+  EXPECT_TRUE_WAIT_MARGIN(ep1_ch1() != NULL && ep2_ch1() != NULL &&
+                              ep1_ch1()->receiving() && ep1_ch1()->writable() &&
+                              ep2_ch1()->receiving() && ep2_ch1()->writable(),
+                          1000, 1000);
+  TestSendRecv(1);
+  // Make sure the P2PTransportChannels are actually using ports from the
+  // pooled sessions.
+  auto pooled_ports_1 = pooled_session_1->ReadyPorts();
+  auto pooled_ports_2 = pooled_session_2->ReadyPorts();
+  EXPECT_NE(pooled_ports_1.end(),
+            std::find(pooled_ports_1.begin(), pooled_ports_1.end(),
+                      ep1_ch1()->best_connection()->port()));
+  EXPECT_NE(pooled_ports_2.end(),
+            std::find(pooled_ports_2.begin(), pooled_ports_2.end(),
+                      ep2_ch1()->best_connection()->port()));
+}
+
 // Test what happens when we have 2 users behind the same NAT. This can lead
 // to interesting behavior because the STUN server will only give out the
 // address of the outermost NAT.
@@ -1979,8 +2086,8 @@ class P2PTransportChannelPingTest : public testing::Test,
   void reset_channel_ready_to_send() { channel_ready_to_send_ = false; }
 
  private:
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
   rtc::SocketServerScope ss_scope_;
   cricket::CandidatePairInterface* last_selected_candidate_pair_ = nullptr;
   int last_sent_packet_id_ = -1;
@@ -2532,9 +2639,10 @@ TEST_F(P2PTransportChannelPingTest, TestDeleteConnectionsIfAllWriteTimedout) {
   EXPECT_TRUE_WAIT(ch.connections().empty(), 1000);
 }
 
-// Test that after a port allocator session is started, it will be stopped
-// when a new connection becomes writable and receiving. Also test that this
-// holds even if the transport channel did not lose the writability.
+// Tests that after a port allocator session is started, it will be stopped
+// when a new connection becomes writable and receiving. Also tests that if a
+// connection belonging to an old session becomes writable, it won't stop
+// the current port allocator session.
 TEST_F(P2PTransportChannelPingTest, TestStopPortAllocatorSessions) {
   cricket::FakePortAllocator pa(rtc::Thread::Current(), nullptr);
   cricket::P2PTransportChannel ch("test channel", 1, &pa);
@@ -2548,11 +2656,17 @@ TEST_F(P2PTransportChannelPingTest, TestStopPortAllocatorSessions) {
   conn1->ReceivedPingResponse();  // Becomes writable and receiving
   EXPECT_TRUE(!ch.allocator_session()->IsGettingPorts());
 
-  // Restart gathering even if the transport channel is still writable.
-  // It should stop getting ports after a new connection becomes strongly
-  // connected.
+  // Start a new session. Even though conn1, which belongs to an older
+  // session, becomes unwritable and writable again, it should not stop the
+  // current session.
   ch.SetIceCredentials(kIceUfrag[1], kIcePwd[1]);
   ch.MaybeStartGathering();
+  conn1->Prune();
+  conn1->ReceivedPingResponse();
+  EXPECT_TRUE(ch.allocator_session()->IsGettingPorts());
+
+  // But if a new connection created from the new session becomes writable,
+  // it will stop the current session.
   ch.AddRemoteCandidate(CreateHostCandidate("2.2.2.2", 2, 100));
   cricket::Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
   ASSERT_TRUE(conn2 != nullptr);
@@ -2625,10 +2739,10 @@ class P2PTransportChannelMostLikelyToWorkFirstTest
   }
 
  private:
-  rtc::scoped_ptr<cricket::BasicPortAllocator> allocator_;
+  std::unique_ptr<cricket::BasicPortAllocator> allocator_;
   rtc::FakeNetworkManager network_manager_;
   cricket::TestTurnServer turn_server_;
-  rtc::scoped_ptr<cricket::P2PTransportChannel> channel_;
+  std::unique_ptr<cricket::P2PTransportChannel> channel_;
 };
 
 // Test that Relay/Relay connections will be pinged first when no other
diff --git a/chromium/third_party/webrtc/p2p/base/packetsocketfactory.h b/chromium/third_party/webrtc/p2p/base/packetsocketfactory.h
index 54037241b0a..290d9ca844b 100644
--- a/chromium/third_party/webrtc/p2p/base/packetsocketfactory.h
+++ b/chromium/third_party/webrtc/p2p/base/packetsocketfactory.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_
 #define WEBRTC_P2P_BASE_PACKETSOCKETFACTORY_H_
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/proxyinfo.h"
 
 namespace rtc {
diff --git a/chromium/third_party/webrtc/p2p/base/port.cc b/chromium/third_party/webrtc/p2p/base/port.cc
index d26a9302a8a..fbc64f2e0a7 100644
--- a/chromium/third_party/webrtc/p2p/base/port.cc
+++ b/chromium/third_party/webrtc/p2p/base/port.cc
@@ -21,7 +21,6 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagedigest.h"
 #include "webrtc/base/network.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 
@@ -220,6 +219,19 @@ Port::~Port() {
     delete list[i];
 }
 
+void Port::SetIceParameters(int component,
+                            const std::string& username_fragment,
+                            const std::string& password) {
+  component_ = component;
+  ice_username_fragment_ = username_fragment;
+  password_ = password;
+  for (Candidate& c : candidates_) {
+    c.set_component(component);
+    c.set_username(username_fragment);
+    c.set_password(password);
+  }
+}
+
 Connection* Port::GetConnection(const rtc::SocketAddress& remote_addr) {
   AddressMap::const_iterator iter = connections_.find(remote_addr);
   if (iter != connections_.end())
@@ -278,7 +290,7 @@ void Port::OnReadPacket(
 
   // If this is an authenticated STUN request, then signal unknown address and
   // send back a proper binding response.
-  rtc::scoped_ptr<IceMessage> msg;
+  std::unique_ptr<IceMessage> msg;
   std::string remote_username;
   if (!GetStunMessage(data, size, addr, &msg, &remote_username)) {
     LOG_J(LS_ERROR, this) << "Received non-STUN packet from unknown address ("
@@ -325,7 +337,7 @@ size_t Port::AddPrflxCandidate(const Candidate& local) {
 bool Port::GetStunMessage(const char* data,
                           size_t size,
                           const rtc::SocketAddress& addr,
-                          rtc::scoped_ptr<IceMessage>* out_msg,
+                          std::unique_ptr<IceMessage>* out_msg,
                           std::string* out_username) {
   // NOTE: This could clearly be optimized to avoid allocating any memory.
   //       However, at the data rates we'll be looking at on the client side,
@@ -342,7 +354,7 @@ bool Port::GetStunMessage(const char* data,
 
   // Parse the request message.  If the packet is not a complete and correct
   // STUN message, then ignore it.
-  rtc::scoped_ptr<IceMessage> stun_msg(new IceMessage());
+  std::unique_ptr<IceMessage> stun_msg(new IceMessage());
   rtc::ByteBufferReader buf(data, size);
   if (!stun_msg->Read(&buf) || (buf.Length() > 0)) {
     return false;
@@ -790,14 +802,14 @@ Connection::Connection(Port* port,
       last_ping_received_(0),
       last_data_received_(0),
       last_ping_response_received_(0),
-      recv_rate_tracker_(100u, 10u),
-      send_rate_tracker_(100u, 10u),
+      recv_rate_tracker_(100, 10u),
+      send_rate_tracker_(100, 10u),
       sent_packets_discarded_(0),
       sent_packets_total_(0),
       reported_(false),
       state_(STATE_WAITING),
       receiving_timeout_(WEAK_CONNECTION_RECEIVE_TIMEOUT),
-      time_created_ms_(rtc::Time64()) {
+      time_created_ms_(rtc::TimeMillis()) {
   // All of our connections start in WAITING state.
   // TODO(mallinath) - Start connections from STATE_FROZEN.
   // Wire up to send stun packets
@@ -895,14 +907,14 @@ void Connection::OnSendStunPacket(const void* data, size_t size,
 
 void Connection::OnReadPacket(
   const char* data, size_t size, const rtc::PacketTime& packet_time) {
-  rtc::scoped_ptr<IceMessage> msg;
+  std::unique_ptr<IceMessage> msg;
   std::string remote_ufrag;
   const rtc::SocketAddress& addr(remote_candidate_.address());
   if (!port_->GetStunMessage(data, size, addr, &msg, &remote_ufrag)) {
     // The packet did not parse as a valid STUN message
     // This is a data packet, pass it along.
     set_receiving(true);
-    last_data_received_ = rtc::Time64();
+    last_data_received_ = rtc::TimeMillis();
     recv_rate_tracker_.AddSamples(size);
     SignalReadPacket(this, data, size, packet_time);
 
@@ -1117,7 +1129,7 @@ void Connection::Ping(int64_t now) {
 
 void Connection::ReceivedPing() {
   set_receiving(true);
-  last_ping_received_ = rtc::Time64();
+  last_ping_received_ = rtc::TimeMillis();
 }
 
 void Connection::ReceivedPingResponse() {
@@ -1130,7 +1142,7 @@ void Connection::ReceivedPingResponse() {
   set_write_state(STATE_WRITABLE);
   set_state(STATE_SUCCEEDED);
   pings_since_last_response_.clear();
-  last_ping_response_received_ = rtc::Time64();
+  last_ping_response_received_ = rtc::TimeMillis();
 }
 
 bool Connection::dead(int64_t now) const {
@@ -1298,12 +1310,22 @@ void Connection::HandleRoleConflictFromPeer() {
   port_->SignalRoleConflict(port_);
 }
 
-void Connection::MaybeSetRemoteIceCredentials(const std::string& ice_ufrag,
-                                              const std::string& ice_pwd) {
+void Connection::MaybeSetRemoteIceCredentialsAndGeneration(
+    const std::string& ice_ufrag,
+    const std::string& ice_pwd,
+    int generation) {
   if (remote_candidate_.username() == ice_ufrag &&
       remote_candidate_.password().empty()) {
     remote_candidate_.set_password(ice_pwd);
   }
+  // TODO(deadbeef): A value of '0' for the generation is used for both
+  // generation 0 and "generation unknown". It should be changed to an
+  // rtc::Optional to fix this.
+  if (remote_candidate_.username() == ice_ufrag &&
+      remote_candidate_.password() == ice_pwd &&
+      remote_candidate_.generation() == 0) {
+    remote_candidate_.set_generation(generation);
+  }
 }
 
 void Connection::MaybeUpdatePeerReflexiveCandidate(
diff --git a/chromium/third_party/webrtc/p2p/base/port.h b/chromium/third_party/webrtc/p2p/base/port.h
index 7c4468d2e6b..937f3cd0284 100644
--- a/chromium/third_party/webrtc/p2p/base/port.h
+++ b/chromium/third_party/webrtc/p2p/base/port.h
@@ -12,6 +12,7 @@
 #define WEBRTC_P2P_BASE_PORT_H_
 
 #include <map>
+#include <memory>
 #include <set>
 #include <string>
 #include <vector>
@@ -106,6 +107,11 @@ struct ProtocolAddress {
       : address(a), proto(p), secure(false) { }
   ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p, bool sec)
       : address(a), proto(p), secure(sec) { }
+
+  bool operator==(const ProtocolAddress& o) const {
+    return address == o.address && proto == o.proto && secure == o.secure;
+  }
+  bool operator!=(const ProtocolAddress& o) const { return !(*this == o); }
 };
 
 typedef std::set<rtc::SocketAddress> ServerAddresses;
@@ -175,23 +181,16 @@ class Port : public PortInterface, public rtc::MessageHandler,
   uint32_t generation() { return generation_; }
   void set_generation(uint32_t generation) { generation_ = generation; }
 
-  // ICE requires a single username/password per content/media line. So the
-  // |ice_username_fragment_| of the ports that belongs to the same content will
-  // be the same. However this causes a small complication with our relay
-  // server, which expects different username for RTP and RTCP.
-  //
-  // To resolve this problem, we implemented the username_fragment(),
-  // which returns a different username (calculated from
-  // |ice_username_fragment_|) for RTCP in the case of ICEPROTO_GOOGLE. And the
-  // username_fragment() simply returns |ice_username_fragment_| when running
-  // in ICEPROTO_RFC5245.
-  //
-  // As a result the ICEPROTO_GOOGLE will use different usernames for RTP and
-  // RTCP. And the ICEPROTO_RFC5245 will use same username for both RTP and
-  // RTCP.
   const std::string username_fragment() const;
   const std::string& password() const { return password_; }
 
+  // May be called when this port was initially created by a pooled
+  // PortAllocatorSession, and is now being assigned to an ICE transport.
+  // Updates the information for candidates as well.
+  void SetIceParameters(int component,
+                        const std::string& username_fragment,
+                        const std::string& password);
+
   // Fired when candidates are discovered by the port. When all candidates
   // are discovered that belong to port SignalAddressReady is fired.
   sigslot::signal2<Port*, const Candidate&> SignalCandidateReady;
@@ -336,7 +335,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
   bool GetStunMessage(const char* data,
                       size_t size,
                       const rtc::SocketAddress& addr,
-                      rtc::scoped_ptr<IceMessage>* out_msg,
+                      std::unique_ptr<IceMessage>* out_msg,
                       std::string* out_username);
 
   // Checks if the address in addr is compatible with the port's ip.
@@ -574,10 +573,14 @@ class Connection : public CandidatePairInterface,
 
   uint32_t ComputeNetworkCost() const;
 
-  // Update the ICE password of the remote candidate if |ice_ufrag| matches
-  // the candidate's ufrag, and the candidate's passwrod has not been set.
-  void MaybeSetRemoteIceCredentials(const std::string& ice_ufrag,
-                                    const std::string& ice_pwd);
+  // Update the ICE password and/or generation of the remote candidate if a
+  // ufrag in |remote_ice_parameters| matches the candidate's ufrag, and the
+  // candidate's password and/or ufrag has not been set.
+  // |remote_ice_parameters| should be a list of known ICE parameters ordered
+  // by generation.
+  void MaybeSetRemoteIceCredentialsAndGeneration(const std::string& ice_ufrag,
+                                                 const std::string& ice_pwd,
+                                                 int generation);
 
   // If |remote_candidate_| is peer reflexive and is equivalent to
   // |new_candidate| except the type, update |remote_candidate_| to
diff --git a/chromium/third_party/webrtc/p2p/base/port_unittest.cc b/chromium/third_party/webrtc/p2p/base/port_unittest.cc
index fc49f20a5d7..efc26094017 100644
--- a/chromium/third_party/webrtc/p2p/base/port_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/port_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/relayport.h"
 #include "webrtc/p2p/base/stunport.h"
@@ -26,7 +28,6 @@
 #include "webrtc/base/natserver.h"
 #include "webrtc/base/natsocketfactory.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/stringutils.h"
@@ -43,7 +44,6 @@ using rtc::NAT_ADDR_RESTRICTED;
 using rtc::NAT_PORT_RESTRICTED;
 using rtc::NAT_SYMMETRIC;
 using rtc::PacketSocketFactory;
-using rtc::scoped_ptr;
 using rtc::Socket;
 using rtc::SocketAddress;
 using namespace cricket;
@@ -212,8 +212,8 @@ class TestPort : public Port {
                     const rtc::SentPacket& sent_packet) {
     PortInterface::SignalSentPacket(sent_packet);
   }
-  rtc::scoped_ptr<Buffer> last_stun_buf_;
-  rtc::scoped_ptr<IceMessage> last_stun_msg_;
+  std::unique_ptr<Buffer> last_stun_buf_;
+  std::unique_ptr<IceMessage> last_stun_msg_;
   int type_preference_ = 0;
 };
 
@@ -344,12 +344,12 @@ class TestChannel : public sigslot::has_slots<> {
   }
 
   IceMode ice_mode_;
-  rtc::scoped_ptr<Port> port_;
+  std::unique_ptr<Port> port_;
 
   int complete_count_;
   Connection* conn_;
   SocketAddress remote_address_;
-  rtc::scoped_ptr<StunMessage> remote_request_;
+  std::unique_ptr<StunMessage> remote_request_;
   std::string remote_frag_;
   bool nominated_;
   bool connection_ready_to_send_ = false;
@@ -764,18 +764,18 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
 
  private:
   rtc::Thread* main_;
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
   rtc::Network network_;
   rtc::BasicPacketSocketFactory socket_factory_;
-  rtc::scoped_ptr<rtc::NATServer> nat_server1_;
-  rtc::scoped_ptr<rtc::NATServer> nat_server2_;
+  std::unique_ptr<rtc::NATServer> nat_server1_;
+  std::unique_ptr<rtc::NATServer> nat_server2_;
   rtc::NATSocketFactory nat_factory1_;
   rtc::NATSocketFactory nat_factory2_;
   rtc::BasicPacketSocketFactory nat_socket_factory1_;
   rtc::BasicPacketSocketFactory nat_socket_factory2_;
-  scoped_ptr<TestStunServer> stun_server_;
+  std::unique_ptr<TestStunServer> stun_server_;
   TestTurnServer turn_server_;
   TestRelayServer relay_server_;
   std::string username_;
@@ -965,7 +965,7 @@ class FakePacketSocketFactory : public rtc::PacketSocketFactory {
   void set_next_client_tcp_socket(AsyncPacketSocket* next_client_tcp_socket) {
     next_client_tcp_socket_ = next_client_tcp_socket;
   }
-  rtc::AsyncResolverInterface* CreateAsyncResolver() {
+  rtc::AsyncResolverInterface* CreateAsyncResolver() override {
     return NULL;
   }
 
@@ -1217,7 +1217,7 @@ TEST_F(PortTest, TestTcpNeverConnect) {
   ch1.Start();
   ASSERT_EQ_WAIT(1, ch1.complete_count(), kTimeout);
 
-  rtc::scoped_ptr<rtc::AsyncSocket> server(
+  std::unique_ptr<rtc::AsyncSocket> server(
       vss()->CreateAsyncSocket(kLocalAddr2.family(), SOCK_STREAM));
   // Bind but not listen.
   EXPECT_EQ(0, server->Bind(kLocalAddr2));
@@ -1268,9 +1268,9 @@ TEST_F(PortTest, TestConnectionDead) {
   ASSERT_EQ_WAIT(1, ch2.complete_count(), kTimeout);
 
   // Test case that the connection has never received anything.
-  int64_t before_created = rtc::Time64();
+  int64_t before_created = rtc::TimeMillis();
   ch1.CreateConnection(GetCandidate(port2));
-  int64_t after_created = rtc::Time64();
+  int64_t after_created = rtc::TimeMillis();
   Connection* conn = ch1.conn();
   ASSERT(conn != nullptr);
   // It is not dead if it is after MIN_CONNECTION_LIFETIME but not pruned.
@@ -1291,9 +1291,9 @@ TEST_F(PortTest, TestConnectionDead) {
   ch1.CreateConnection(GetCandidate(port2));
   conn = ch1.conn();
   ASSERT(conn != nullptr);
-  int64_t before_last_receiving = rtc::Time64();
+  int64_t before_last_receiving = rtc::TimeMillis();
   conn->ReceivedPing();
-  int64_t after_last_receiving = rtc::Time64();
+  int64_t after_last_receiving = rtc::TimeMillis();
   // The connection will be dead after DEAD_CONNECTION_RECEIVE_TIMEOUT
   conn->UpdateState(
       before_last_receiving + DEAD_CONNECTION_RECEIVE_TIMEOUT - 1);
@@ -1322,7 +1322,7 @@ TEST_F(PortTest, TestLocalToLocalStandard) {
 // should remain equal to the request generated by the port and role of port
 // must be in controlling.
 TEST_F(PortTest, TestLoopbackCal) {
-  rtc::scoped_ptr<TestPort> lport(
+  std::unique_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
@@ -1354,7 +1354,7 @@ TEST_F(PortTest, TestLoopbackCal) {
   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, 1000);
   msg = lport->last_stun_msg();
   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
-  rtc::scoped_ptr<IceMessage> modified_req(
+  std::unique_ptr<IceMessage> modified_req(
       CreateStunMessage(STUN_BINDING_REQUEST));
   const StunByteStringAttribute* username_attr = msg->GetByteString(
       STUN_ATTR_USERNAME);
@@ -1368,7 +1368,7 @@ TEST_F(PortTest, TestLoopbackCal) {
   modified_req->AddFingerprint();
 
   lport->Reset();
-  rtc::scoped_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
+  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
   WriteStunMessage(modified_req.get(), buf.get());
   conn1->OnReadPacket(buf->Data(), buf->Length(), rtc::PacketTime());
   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, 1000);
@@ -1382,11 +1382,11 @@ TEST_F(PortTest, TestLoopbackCal) {
 // value of tiebreaker, when it receives ping request from |rport| it will
 // send role conflict signal.
 TEST_F(PortTest, TestIceRoleConflict) {
-  rtc::scoped_ptr<TestPort> lport(
+  std::unique_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
-  rtc::scoped_ptr<TestPort> rport(
+  std::unique_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
   rport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   rport->SetIceTiebreaker(kTiebreaker2);
@@ -1430,8 +1430,7 @@ TEST_F(PortTest, TestDelayedBindingUdp) {
   FakePacketSocketFactory socket_factory;
 
   socket_factory.set_next_udp_socket(socket);
-  scoped_ptr<UDPPort> port(
-      CreateUdpPort(kLocalAddr1, &socket_factory));
+  std::unique_ptr<UDPPort> port(CreateUdpPort(kLocalAddr1, &socket_factory));
 
   socket->set_state(AsyncPacketSocket::STATE_BINDING);
   port->PrepareAddress();
@@ -1447,8 +1446,7 @@ TEST_F(PortTest, TestDelayedBindingTcp) {
   FakePacketSocketFactory socket_factory;
 
   socket_factory.set_next_server_tcp_socket(socket);
-  scoped_ptr<TCPPort> port(
-      CreateTcpPort(kLocalAddr1, &socket_factory));
+  std::unique_ptr<TCPPort> port(CreateTcpPort(kLocalAddr1, &socket_factory));
 
   socket->set_state(AsyncPacketSocket::STATE_BINDING);
   port->PrepareAddress();
@@ -1461,7 +1459,7 @@ TEST_F(PortTest, TestDelayedBindingTcp) {
 
 void PortTest::TestCrossFamilyPorts(int type) {
   FakePacketSocketFactory factory;
-  scoped_ptr<Port> ports[4];
+  std::unique_ptr<Port> ports[4];
   SocketAddress addresses[4] = {SocketAddress("192.168.1.3", 0),
                                 SocketAddress("192.168.1.4", 0),
                                 SocketAddress("2001:db8::1", 0),
@@ -1531,7 +1529,7 @@ void PortTest::ExpectPortsCanConnect(bool can_connect, Port* p1, Port* p2) {
 
 TEST_F(PortTest, TestUdpV6CrossTypePorts) {
   FakePacketSocketFactory factory;
-  scoped_ptr<Port> ports[4];
+  std::unique_ptr<Port> ports[4];
   SocketAddress addresses[4] = {SocketAddress("2001:db8::1", 0),
                                 SocketAddress("fe80::1", 0),
                                 SocketAddress("fe80::2", 0),
@@ -1564,23 +1562,23 @@ TEST_F(PortTest, TestUdpV6CrossTypePorts) {
 // get through DefaultDscpValue.
 TEST_F(PortTest, TestDefaultDscpValue) {
   int dscp;
-  rtc::scoped_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
+  std::unique_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
   EXPECT_EQ(0, udpport->SetOption(rtc::Socket::OPT_DSCP,
                                   rtc::DSCP_CS6));
   EXPECT_EQ(0, udpport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
-  rtc::scoped_ptr<TCPPort> tcpport(CreateTcpPort(kLocalAddr1));
+  std::unique_ptr<TCPPort> tcpport(CreateTcpPort(kLocalAddr1));
   EXPECT_EQ(0, tcpport->SetOption(rtc::Socket::OPT_DSCP,
                                  rtc::DSCP_AF31));
   EXPECT_EQ(0, tcpport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
   EXPECT_EQ(rtc::DSCP_AF31, dscp);
-  rtc::scoped_ptr<StunPort> stunport(
+  std::unique_ptr<StunPort> stunport(
       CreateStunPort(kLocalAddr1, nat_socket_factory1()));
   EXPECT_EQ(0, stunport->SetOption(rtc::Socket::OPT_DSCP,
                                   rtc::DSCP_AF41));
   EXPECT_EQ(0, stunport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
   EXPECT_EQ(rtc::DSCP_AF41, dscp);
-  rtc::scoped_ptr<TurnPort> turnport1(CreateTurnPort(
-      kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
+  std::unique_ptr<TurnPort> turnport1(
+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
   // Socket is created in PrepareAddress.
   turnport1->PrepareAddress();
   EXPECT_EQ(0, turnport1->SetOption(rtc::Socket::OPT_DSCP,
@@ -1588,8 +1586,8 @@ TEST_F(PortTest, TestDefaultDscpValue) {
   EXPECT_EQ(0, turnport1->GetOption(rtc::Socket::OPT_DSCP, &dscp));
   EXPECT_EQ(rtc::DSCP_CS7, dscp);
   // This will verify correct value returned without the socket.
-  rtc::scoped_ptr<TurnPort> turnport2(CreateTurnPort(
-      kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
+  std::unique_ptr<TurnPort> turnport2(
+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
   EXPECT_EQ(0, turnport2->SetOption(rtc::Socket::OPT_DSCP,
                                   rtc::DSCP_CS6));
   EXPECT_EQ(0, turnport2->GetOption(rtc::Socket::OPT_DSCP, &dscp));
@@ -1598,9 +1596,9 @@ TEST_F(PortTest, TestDefaultDscpValue) {
 
 // Test sending STUN messages.
 TEST_F(PortTest, TestSendStunMessage) {
-  rtc::scoped_ptr<TestPort> lport(
+  std::unique_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
-  rtc::scoped_ptr<TestPort> rport(
+  std::unique_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
@@ -1647,7 +1645,7 @@ TEST_F(PortTest, TestSendStunMessage) {
   ASSERT_TRUE(msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT) == NULL);
 
   // Save a copy of the BINDING-REQUEST for use below.
-  rtc::scoped_ptr<IceMessage> request(CopyStunMessage(msg));
+  std::unique_ptr<IceMessage> request(CopyStunMessage(msg));
 
   // Respond with a BINDING-RESPONSE.
   rport->SendBindingResponse(request.get(), lport->Candidates()[0].address());
@@ -1738,9 +1736,9 @@ TEST_F(PortTest, TestSendStunMessage) {
 }
 
 TEST_F(PortTest, TestUseCandidateAttribute) {
-  rtc::scoped_ptr<TestPort> lport(
+  std::unique_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
-  rtc::scoped_ptr<TestPort> rport(
+  std::unique_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
@@ -1765,11 +1763,11 @@ TEST_F(PortTest, TestUseCandidateAttribute) {
 }
 
 TEST_F(PortTest, TestNetworkInfoAttribute) {
-  rtc::scoped_ptr<TestPort> lport(
+  std::unique_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
   // Set the network type for rport to be cellular so its cost will be 999.
   SetNetworkType(rtc::ADAPTER_TYPE_CELLULAR);
-  rtc::scoped_ptr<TestPort> rport(
+  std::unique_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
@@ -1812,11 +1810,10 @@ TEST_F(PortTest, TestNetworkInfoAttribute) {
 // Test handling STUN messages.
 TEST_F(PortTest, TestHandleStunMessage) {
   // Our port will act as the "remote" port.
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
+  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
 
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
+  std::unique_ptr<IceMessage> in_msg, out_msg;
+  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
   rtc::SocketAddress addr(kLocalAddr1);
   std::string username;
 
@@ -1862,11 +1859,10 @@ TEST_F(PortTest, TestHandleStunMessage) {
 
 // Tests handling of ICE binding requests with missing or incorrect usernames.
 TEST_F(PortTest, TestHandleStunMessageBadUsername) {
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
+  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
 
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
+  std::unique_ptr<IceMessage> in_msg, out_msg;
+  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
   rtc::SocketAddress addr(kLocalAddr1);
   std::string username;
 
@@ -1931,11 +1927,10 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
 // Test handling STUN messages with missing or malformed M-I.
 TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
   // Our port will act as the "remote" port.
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
+  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
 
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
+  std::unique_ptr<IceMessage> in_msg, out_msg;
+  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
   rtc::SocketAddress addr(kLocalAddr1);
   std::string username;
 
@@ -1972,11 +1967,10 @@ TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
 // Test handling STUN messages with missing or malformed FINGERPRINT.
 TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
   // Our port will act as the "remote" port.
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
+  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
 
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
+  std::unique_ptr<IceMessage> in_msg, out_msg;
+  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
   rtc::SocketAddress addr(kLocalAddr1);
   std::string username;
 
@@ -2038,14 +2032,14 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
 // Test handling of STUN binding indication messages . STUN binding
 // indications are allowed only to the connection which is in read mode.
 TEST_F(PortTest, TestHandleStunBindingIndication) {
-  rtc::scoped_ptr<TestPort> lport(
+  std::unique_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr2, "lfrag", "lpass"));
   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
   lport->SetIceTiebreaker(kTiebreaker1);
 
   // Verifying encoding and decoding STUN indication message.
-  rtc::scoped_ptr<IceMessage> in_msg, out_msg;
-  rtc::scoped_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
+  std::unique_ptr<IceMessage> in_msg, out_msg;
+  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
   rtc::SocketAddress addr(kLocalAddr1);
   std::string username;
 
@@ -2060,7 +2054,7 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
 
   // Verify connection can handle STUN indication and updates
   // last_ping_received.
-  rtc::scoped_ptr<TestPort> rport(
+  std::unique_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
   rport->SetIceTiebreaker(kTiebreaker2);
@@ -2096,8 +2090,7 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
 }
 
 TEST_F(PortTest, TestComputeCandidatePriority) {
-  rtc::scoped_ptr<TestPort> port(
-      CreateTestPort(kLocalAddr1, "name", "pass"));
+  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr1, "name", "pass"));
   port->set_type_preference(90);
   port->set_component(177);
   port->AddCandidateAddress(SocketAddress("192.168.1.4", 1234));
@@ -2134,7 +2127,7 @@ TEST_F(PortTest, TestComputeCandidatePriority) {
 // In the case of shared socket, one port may be shared by local and stun.
 // Test that candidates with different types will have different foundation.
 TEST_F(PortTest, TestFoundation) {
-  rtc::scoped_ptr<TestPort> testport(
+  std::unique_ptr<TestPort> testport(
       CreateTestPort(kLocalAddr1, "name", "pass"));
   testport->AddCandidateAddress(kLocalAddr1, kLocalAddr1,
                                 LOCAL_PORT_TYPE,
@@ -2148,21 +2141,21 @@ TEST_F(PortTest, TestFoundation) {
 
 // This test verifies the foundation of different types of ICE candidates.
 TEST_F(PortTest, TestCandidateFoundation) {
-  rtc::scoped_ptr<rtc::NATServer> nat_server(
+  std::unique_ptr<rtc::NATServer> nat_server(
       CreateNatServer(kNatAddr1, NAT_OPEN_CONE));
-  rtc::scoped_ptr<UDPPort> udpport1(CreateUdpPort(kLocalAddr1));
+  std::unique_ptr<UDPPort> udpport1(CreateUdpPort(kLocalAddr1));
   udpport1->PrepareAddress();
-  rtc::scoped_ptr<UDPPort> udpport2(CreateUdpPort(kLocalAddr1));
+  std::unique_ptr<UDPPort> udpport2(CreateUdpPort(kLocalAddr1));
   udpport2->PrepareAddress();
   EXPECT_EQ(udpport1->Candidates()[0].foundation(),
             udpport2->Candidates()[0].foundation());
-  rtc::scoped_ptr<TCPPort> tcpport1(CreateTcpPort(kLocalAddr1));
+  std::unique_ptr<TCPPort> tcpport1(CreateTcpPort(kLocalAddr1));
   tcpport1->PrepareAddress();
-  rtc::scoped_ptr<TCPPort> tcpport2(CreateTcpPort(kLocalAddr1));
+  std::unique_ptr<TCPPort> tcpport2(CreateTcpPort(kLocalAddr1));
   tcpport2->PrepareAddress();
   EXPECT_EQ(tcpport1->Candidates()[0].foundation(),
             tcpport2->Candidates()[0].foundation());
-  rtc::scoped_ptr<Port> stunport(
+  std::unique_ptr<Port> stunport(
       CreateStunPort(kLocalAddr1, nat_socket_factory1()));
   stunport->PrepareAddress();
   ASSERT_EQ_WAIT(1U, stunport->Candidates().size(), kTimeout);
@@ -2175,8 +2168,7 @@ TEST_F(PortTest, TestCandidateFoundation) {
   EXPECT_NE(udpport2->Candidates()[0].foundation(),
             stunport->Candidates()[0].foundation());
   // Verify GTURN candidate foundation.
-  rtc::scoped_ptr<RelayPort> relayport(
-      CreateGturnPort(kLocalAddr1));
+  std::unique_ptr<RelayPort> relayport(CreateGturnPort(kLocalAddr1));
   relayport->AddServerAddress(
       cricket::ProtocolAddress(kRelayUdpIntAddr, cricket::PROTO_UDP));
   relayport->PrepareAddress();
@@ -2186,8 +2178,8 @@ TEST_F(PortTest, TestCandidateFoundation) {
   EXPECT_NE(udpport2->Candidates()[0].foundation(),
             relayport->Candidates()[0].foundation());
   // Verifying TURN candidate foundation.
-  rtc::scoped_ptr<Port> turnport1(CreateTurnPort(
-      kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
+  std::unique_ptr<Port> turnport1(
+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
   turnport1->PrepareAddress();
   ASSERT_EQ_WAIT(1U, turnport1->Candidates().size(), kTimeout);
   EXPECT_NE(udpport1->Candidates()[0].foundation(),
@@ -2196,8 +2188,8 @@ TEST_F(PortTest, TestCandidateFoundation) {
             turnport1->Candidates()[0].foundation());
   EXPECT_NE(stunport->Candidates()[0].foundation(),
             turnport1->Candidates()[0].foundation());
-  rtc::scoped_ptr<Port> turnport2(CreateTurnPort(
-      kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
+  std::unique_ptr<Port> turnport2(
+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
   turnport2->PrepareAddress();
   ASSERT_EQ_WAIT(1U, turnport2->Candidates().size(), kTimeout);
   EXPECT_EQ(turnport1->Candidates()[0].foundation(),
@@ -2208,9 +2200,9 @@ TEST_F(PortTest, TestCandidateFoundation) {
   SocketAddress kTurnUdpExtAddr2("99.99.98.5", 0);
   TestTurnServer turn_server2(
       rtc::Thread::Current(), kTurnUdpIntAddr2, kTurnUdpExtAddr2);
-  rtc::scoped_ptr<Port> turnport3(CreateTurnPort(
-      kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP,
-      kTurnUdpIntAddr2));
+  std::unique_ptr<Port> turnport3(
+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP,
+                     kTurnUdpIntAddr2));
   turnport3->PrepareAddress();
   ASSERT_EQ_WAIT(1U, turnport3->Candidates().size(), kTimeout);
   EXPECT_NE(turnport3->Candidates()[0].foundation(),
@@ -2220,7 +2212,7 @@ TEST_F(PortTest, TestCandidateFoundation) {
   // different foundations if their relay protocols are different.
   TestTurnServer turn_server3(rtc::Thread::Current(), kTurnTcpIntAddr,
                               kTurnUdpExtAddr, PROTO_TCP);
-  rtc::scoped_ptr<Port> turnport4(
+  std::unique_ptr<Port> turnport4(
       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_TCP, PROTO_UDP));
   turnport4->PrepareAddress();
   ASSERT_EQ_WAIT(1U, turnport4->Candidates().size(), kTimeout);
@@ -2231,16 +2223,16 @@ TEST_F(PortTest, TestCandidateFoundation) {
 // This test verifies the related addresses of different types of
 // ICE candiates.
 TEST_F(PortTest, TestCandidateRelatedAddress) {
-  rtc::scoped_ptr<rtc::NATServer> nat_server(
+  std::unique_ptr<rtc::NATServer> nat_server(
       CreateNatServer(kNatAddr1, NAT_OPEN_CONE));
-  rtc::scoped_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
+  std::unique_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
   udpport->PrepareAddress();
   // For UDPPort, related address will be empty.
   EXPECT_TRUE(udpport->Candidates()[0].related_address().IsNil());
   // Testing related address for stun candidates.
   // For stun candidate related address must be equal to the base
   // socket address.
-  rtc::scoped_ptr<StunPort> stunport(
+  std::unique_ptr<StunPort> stunport(
       CreateStunPort(kLocalAddr1, nat_socket_factory1()));
   stunport->PrepareAddress();
   ASSERT_EQ_WAIT(1U, stunport->Candidates().size(), kTimeout);
@@ -2253,8 +2245,7 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
   // Verifying the related address for the GTURN candidates.
   // NOTE: In case of GTURN related address will be equal to the mapped
   // address, but address(mapped) will not be XOR.
-  rtc::scoped_ptr<RelayPort> relayport(
-      CreateGturnPort(kLocalAddr1));
+  std::unique_ptr<RelayPort> relayport(CreateGturnPort(kLocalAddr1));
   relayport->AddServerAddress(
       cricket::ProtocolAddress(kRelayUdpIntAddr, cricket::PROTO_UDP));
   relayport->PrepareAddress();
@@ -2264,8 +2255,8 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
             relayport->Candidates()[0].related_address());
   // Verifying the related address for TURN candidate.
   // For TURN related address must be equal to the mapped address.
-  rtc::scoped_ptr<Port> turnport(CreateTurnPort(
-      kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
+  std::unique_ptr<Port> turnport(
+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
   turnport->PrepareAddress();
   ASSERT_EQ_WAIT(1U, turnport->Candidates().size(), kTimeout);
   EXPECT_EQ(kTurnUdpExtAddr.ipaddr(),
@@ -2285,10 +2276,10 @@ TEST_F(PortTest, TestCandidatePriority) {
 
 // Test the Connection priority is calculated correctly.
 TEST_F(PortTest, TestConnectionPriority) {
-  rtc::scoped_ptr<TestPort> lport(
+  std::unique_ptr<TestPort> lport(
       CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
   lport->set_type_preference(cricket::ICE_TYPE_PREFERENCE_HOST);
-  rtc::scoped_ptr<TestPort> rport(
+  std::unique_ptr<TestPort> rport(
       CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
   rport->set_type_preference(cricket::ICE_TYPE_PREFERENCE_RELAY);
   lport->set_component(123);
@@ -2428,9 +2419,9 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
       kLocalAddr1, "lfrag", "lpass",
       cricket::ICEROLE_CONTROLLING, kTiebreaker1);
 
-  rtc::scoped_ptr<TestPort> ice_lite_port(CreateTestPort(
-      kLocalAddr2, "rfrag", "rpass",
-      cricket::ICEROLE_CONTROLLED, kTiebreaker2));
+  std::unique_ptr<TestPort> ice_lite_port(
+      CreateTestPort(kLocalAddr2, "rfrag", "rpass", cricket::ICEROLE_CONTROLLED,
+                     kTiebreaker2));
   // Setup TestChannel. This behaves like FULL mode client.
   TestChannel ch1(ice_full_port);
   ch1.SetIceMode(ICEMODE_FULL);
@@ -2462,7 +2453,7 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
   // But we need a connection to send a response message.
   ice_lite_port->CreateConnection(
       ice_full_port->Candidates()[0], cricket::Port::ORIGIN_MESSAGE);
-  rtc::scoped_ptr<IceMessage> request(CopyStunMessage(msg));
+  std::unique_ptr<IceMessage> request(CopyStunMessage(msg));
   ice_lite_port->SendBindingResponse(
       request.get(), ice_full_port->Candidates()[0].address());
 
@@ -2576,22 +2567,39 @@ TEST_F(PortTest, TestControlledToControllingNotDestroyed) {
 }
 
 TEST_F(PortTest, TestSupportsProtocol) {
-  rtc::scoped_ptr<Port> udp_port(CreateUdpPort(kLocalAddr1));
+  std::unique_ptr<Port> udp_port(CreateUdpPort(kLocalAddr1));
   EXPECT_TRUE(udp_port->SupportsProtocol(UDP_PROTOCOL_NAME));
   EXPECT_FALSE(udp_port->SupportsProtocol(TCP_PROTOCOL_NAME));
 
-  rtc::scoped_ptr<Port> stun_port(
+  std::unique_ptr<Port> stun_port(
       CreateStunPort(kLocalAddr1, nat_socket_factory1()));
   EXPECT_TRUE(stun_port->SupportsProtocol(UDP_PROTOCOL_NAME));
   EXPECT_FALSE(stun_port->SupportsProtocol(TCP_PROTOCOL_NAME));
 
-  rtc::scoped_ptr<Port> tcp_port(CreateTcpPort(kLocalAddr1));
+  std::unique_ptr<Port> tcp_port(CreateTcpPort(kLocalAddr1));
   EXPECT_TRUE(tcp_port->SupportsProtocol(TCP_PROTOCOL_NAME));
   EXPECT_TRUE(tcp_port->SupportsProtocol(SSLTCP_PROTOCOL_NAME));
   EXPECT_FALSE(tcp_port->SupportsProtocol(UDP_PROTOCOL_NAME));
 
-  rtc::scoped_ptr<Port> turn_port(
+  std::unique_ptr<Port> turn_port(
       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
   EXPECT_TRUE(turn_port->SupportsProtocol(UDP_PROTOCOL_NAME));
   EXPECT_FALSE(turn_port->SupportsProtocol(TCP_PROTOCOL_NAME));
 }
+
+// Test that SetIceParameters updates the component, ufrag and password
+// on both the port itself and its candidates.
+TEST_F(PortTest, TestSetIceParameters) {
+  std::unique_ptr<TestPort> port(
+      CreateTestPort(kLocalAddr1, "ufrag1", "password1"));
+  port->PrepareAddress();
+  EXPECT_EQ(1UL, port->Candidates().size());
+  port->SetIceParameters(1, "ufrag2", "password2");
+  EXPECT_EQ(1, port->component());
+  EXPECT_EQ("ufrag2", port->username_fragment());
+  EXPECT_EQ("password2", port->password());
+  const Candidate& candidate = port->Candidates()[0];
+  EXPECT_EQ(1, candidate.component());
+  EXPECT_EQ("ufrag2", candidate.username());
+  EXPECT_EQ("password2", candidate.password());
+}
diff --git a/chromium/third_party/webrtc/p2p/base/portallocator.cc b/chromium/third_party/webrtc/p2p/base/portallocator.cc
index 5c4243abf6b..fdf213b3119 100644
--- a/chromium/third_party/webrtc/p2p/base/portallocator.cc
+++ b/chromium/third_party/webrtc/p2p/base/portallocator.cc
@@ -18,23 +18,85 @@ PortAllocatorSession::PortAllocatorSession(const std::string& content_name,
                                            const std::string& ice_ufrag,
                                            const std::string& ice_pwd,
                                            uint32_t flags)
-    : content_name_(content_name),
-      component_(component),
-      flags_(flags),
+    : flags_(flags),
       generation_(0),
+      content_name_(content_name),
+      component_(component),
       ice_ufrag_(ice_ufrag),
       ice_pwd_(ice_pwd) {
-  RTC_DCHECK(!ice_ufrag.empty());
-  RTC_DCHECK(!ice_pwd.empty());
+  // Pooled sessions are allowed to be created with empty content name,
+  // component, ufrag and password.
+  RTC_DCHECK(ice_ufrag.empty() == ice_pwd.empty());
 }
 
-PortAllocatorSession* PortAllocator::CreateSession(
+void PortAllocator::SetConfiguration(
+    const ServerAddresses& stun_servers,
+    const std::vector<RelayServerConfig>& turn_servers,
+    int candidate_pool_size) {
+  bool ice_servers_changed =
+      (stun_servers != stun_servers_ || turn_servers != turn_servers_);
+  stun_servers_ = stun_servers;
+  turn_servers_ = turn_servers;
+
+  // If ICE servers changed, throw away any existing pooled sessions and create
+  // new ones.
+  if (ice_servers_changed) {
+    pooled_sessions_.clear();
+    allocated_pooled_session_count_ = 0;
+  }
+
+  // If |size| is less than the number of allocated sessions, get rid of the
+  // extras.
+  while (allocated_pooled_session_count_ > candidate_pool_size &&
+         !pooled_sessions_.empty()) {
+    pooled_sessions_.front().reset(nullptr);
+    pooled_sessions_.pop_front();
+    --allocated_pooled_session_count_;
+  }
+  // If |size| is greater than the number of allocated sessions, create new
+  // sessions.
+  while (allocated_pooled_session_count_ < candidate_pool_size) {
+    PortAllocatorSession* pooled_session = CreateSessionInternal("", 0, "", "");
+    pooled_session->StartGettingPorts();
+    pooled_sessions_.push_back(
+        std::unique_ptr<PortAllocatorSession>(pooled_session));
+    ++allocated_pooled_session_count_;
+  }
+  target_pooled_session_count_ = candidate_pool_size;
+}
+
+std::unique_ptr<PortAllocatorSession> PortAllocator::CreateSession(
     const std::string& sid,
     const std::string& content_name,
     int component,
     const std::string& ice_ufrag,
     const std::string& ice_pwd) {
-  return CreateSessionInternal(content_name, component, ice_ufrag, ice_pwd);
+  return std::unique_ptr<PortAllocatorSession>(
+      CreateSessionInternal(content_name, component, ice_ufrag, ice_pwd));
+}
+
+std::unique_ptr<PortAllocatorSession> PortAllocator::TakePooledSession(
+    const std::string& content_name,
+    int component,
+    const std::string& ice_ufrag,
+    const std::string& ice_pwd) {
+  RTC_DCHECK(!ice_ufrag.empty());
+  RTC_DCHECK(!ice_pwd.empty());
+  if (pooled_sessions_.empty()) {
+    return nullptr;
+  }
+  std::unique_ptr<PortAllocatorSession> ret =
+      std::move(pooled_sessions_.front());
+  ret->SetIceParameters(content_name, component, ice_ufrag, ice_pwd);
+  pooled_sessions_.pop_front();
+  return ret;
+}
+
+const PortAllocatorSession* PortAllocator::GetPooledSession() const {
+  if (pooled_sessions_.empty()) {
+    return nullptr;
+  }
+  return pooled_sessions_.front().get();
 }
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/portallocator.h b/chromium/third_party/webrtc/p2p/base/portallocator.h
index 6fb79b065e6..879657081a5 100644
--- a/chromium/third_party/webrtc/p2p/base/portallocator.h
+++ b/chromium/third_party/webrtc/p2p/base/portallocator.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_P2P_BASE_PORTALLOCATOR_H_
 #define WEBRTC_P2P_BASE_PORTALLOCATOR_H_
 
+#include <deque>
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -19,6 +21,7 @@
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/proxyinfo.h"
 #include "webrtc/base/sigslot.h"
+#include "webrtc/base/thread.h"
 
 namespace cricket {
 
@@ -82,6 +85,11 @@ struct RelayCredentials {
   RelayCredentials(const std::string& username, const std::string& password)
       : username(username), password(password) {}
 
+  bool operator==(const RelayCredentials& o) const {
+    return username == o.username && password == o.password;
+  }
+  bool operator!=(const RelayCredentials& o) const { return !(*this == o); }
+
   std::string username;
   std::string password;
 };
@@ -89,7 +97,7 @@ struct RelayCredentials {
 typedef std::vector<ProtocolAddress> PortList;
 // TODO(deadbeef): Rename to TurnServerConfig.
 struct RelayServerConfig {
-  RelayServerConfig(RelayType type) : type(type), priority(0) {}
+  RelayServerConfig(RelayType type) : type(type) {}
 
   RelayServerConfig(const std::string& address,
                     int port,
@@ -102,10 +110,16 @@ struct RelayServerConfig {
         ProtocolAddress(rtc::SocketAddress(address, port), proto, secure));
   }
 
+  bool operator==(const RelayServerConfig& o) const {
+    return type == o.type && ports == o.ports && credentials == o.credentials &&
+           priority == o.priority;
+  }
+  bool operator!=(const RelayServerConfig& o) const { return !(*this == o); }
+
   RelayType type;
   PortList ports;
   RelayCredentials credentials;
-  int priority;
+  int priority = 0;
 };
 
 class PortAllocatorSession : public sigslot::has_slots<> {
@@ -124,6 +138,9 @@ class PortAllocatorSession : public sigslot::has_slots<> {
   void set_flags(uint32_t flags) { flags_ = flags; }
   std::string content_name() const { return content_name_; }
   int component() const { return component_; }
+  const std::string& ice_ufrag() const { return ice_ufrag_; }
+  const std::string& ice_pwd() const { return ice_pwd_; }
+  bool pooled() const { return ice_ufrag_.empty(); }
 
   // Starts gathering STUN and Relay configurations.
   virtual void StartGettingPorts() = 0;
@@ -133,6 +150,14 @@ class PortAllocatorSession : public sigslot::has_slots<> {
   // Whether the process of getting ports has been stopped.
   virtual bool IsGettingPorts() = 0;
 
+  // Another way of getting the information provided by the signals below.
+  //
+  // Ports and candidates are not guaranteed to be in the same order as the
+  // signals were emitted in.
+  virtual std::vector<PortInterface*> ReadyPorts() const = 0;
+  virtual std::vector<Candidate> ReadyCandidates() const = 0;
+  virtual bool CandidatesAllocationDone() const = 0;
+
   sigslot::signal2<PortAllocatorSession*, PortInterface*> SignalPortReady;
   sigslot::signal2<PortAllocatorSession*,
                    const std::vector<Candidate>&> SignalCandidatesReady;
@@ -142,25 +167,46 @@ class PortAllocatorSession : public sigslot::has_slots<> {
   virtual void set_generation(uint32_t generation) { generation_ = generation; }
   sigslot::signal1<PortAllocatorSession*> SignalDestroyed;
 
-  const std::string& ice_ufrag() const { return ice_ufrag_; }
-  const std::string& ice_pwd() const { return ice_pwd_; }
-
  protected:
+  // This method is called when a pooled session (which doesn't have these
+  // properties initially) is returned by PortAllocator::TakePooledSession,
+  // and the content name, component, and ICE ufrag/pwd are updated.
+  //
+  // A subclass may need to override this method to perform additional actions,
+  // such as applying the updated information to ports and candidates.
+  virtual void UpdateIceParametersInternal() {}
+
   // TODO(deadbeef): Get rid of these when everyone switches to ice_ufrag and
   // ice_pwd.
   const std::string& username() const { return ice_ufrag_; }
   const std::string& password() const { return ice_pwd_; }
 
-  std::string content_name_;
-  int component_;
-
  private:
+  void SetIceParameters(const std::string& content_name,
+                        int component,
+                        const std::string& ice_ufrag,
+                        const std::string& ice_pwd) {
+    content_name_ = content_name;
+    component_ = component;
+    ice_ufrag_ = ice_ufrag;
+    ice_pwd_ = ice_pwd;
+    UpdateIceParametersInternal();
+  }
+
   uint32_t flags_;
   uint32_t generation_;
+  std::string content_name_;
+  int component_;
   std::string ice_ufrag_;
   std::string ice_pwd_;
+
+  // SetIceParameters is an implementation detail which only PortAllocator
+  // should be able to call.
+  friend class PortAllocator;
 };
 
+// Note that this class should only be used on one thread.
+// This includes calling the destructor.
 class PortAllocator : public sigslot::has_slots<> {
  public:
   PortAllocator() :
@@ -174,10 +220,25 @@ class PortAllocator : public sigslot::has_slots<> {
   }
   virtual ~PortAllocator() {}
 
-  // Set STUN and TURN servers to be used in future sessions.
-  virtual void SetIceServers(
-      const ServerAddresses& stun_servers,
-      const std::vector<RelayServerConfig>& turn_servers) = 0;
+  // Set STUN and TURN servers to be used in future sessions, and set
+  // candidate pool size, as described in JSEP.
+  //
+  // If the servers are changing and the candidate pool size is nonzero,
+  // existing pooled sessions will be destroyed and new ones created.
+  //
+  // If the servers are not changing but the candidate pool size is,
+  // pooled sessions will be either created or destroyed as necessary.
+  void SetConfiguration(const ServerAddresses& stun_servers,
+                        const std::vector<RelayServerConfig>& turn_servers,
+                        int candidate_pool_size);
+
+  const ServerAddresses& stun_servers() const { return stun_servers_; }
+
+  const std::vector<RelayServerConfig>& turn_servers() const {
+    return turn_servers_;
+  }
+
+  int candidate_pool_size() const { return target_pooled_session_count_; }
 
   // Sets the network types to ignore.
   // Values are defined by the AdapterType enum.
@@ -186,13 +247,27 @@ class PortAllocator : public sigslot::has_slots<> {
   // loopback interfaces.
   virtual void SetNetworkIgnoreMask(int network_ignore_mask) = 0;
 
-  PortAllocatorSession* CreateSession(
+  std::unique_ptr<PortAllocatorSession> CreateSession(
       const std::string& sid,
       const std::string& content_name,
       int component,
       const std::string& ice_ufrag,
       const std::string& ice_pwd);
 
+  // Get an available pooled session and set the transport information on it.
+  //
+  // Caller takes ownership of the returned session.
+  //
+  // If no pooled sessions are available, returns null.
+  std::unique_ptr<PortAllocatorSession> TakePooledSession(
+      const std::string& content_name,
+      int component,
+      const std::string& ice_ufrag,
+      const std::string& ice_pwd);
+
+  // Returns the next session that would be returned by TakePooledSession.
+  const PortAllocatorSession* GetPooledSession() const;
+
   uint32_t flags() const { return flags_; }
   void set_flags(uint32_t flags) { flags_ = flags; }
 
@@ -225,10 +300,9 @@ class PortAllocator : public sigslot::has_slots<> {
   }
 
   uint32_t candidate_filter() { return candidate_filter_; }
-  bool set_candidate_filter(uint32_t filter) {
+  void set_candidate_filter(uint32_t filter) {
     // TODO(mallinath) - Do transition check?
     candidate_filter_ = filter;
-    return true;
   }
 
   // Gets/Sets the Origin value used for WebRTC STUN requests.
@@ -251,6 +325,16 @@ class PortAllocator : public sigslot::has_slots<> {
   bool allow_tcp_listen_;
   uint32_t candidate_filter_;
   std::string origin_;
+
+ private:
+  ServerAddresses stun_servers_;
+  std::vector<RelayServerConfig> turn_servers_;
+  // The last size passed into SetConfiguration.
+  int target_pooled_session_count_ = 0;
+  // This variable represents the total number of pooled sessions
+  // both owned by this class and taken by TakePooledSession.
+  int allocated_pooled_session_count_ = 0;
+  std::deque<std::unique_ptr<PortAllocatorSession>> pooled_sessions_;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/portallocator_unittest.cc b/chromium/third_party/webrtc/p2p/base/portallocator_unittest.cc
new file mode 100644
index 00000000000..2e16725de48
--- /dev/null
+++ b/chromium/third_party/webrtc/p2p/base/portallocator_unittest.cc
@@ -0,0 +1,205 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/p2p/base/fakeportallocator.h"
+#include "webrtc/p2p/base/portallocator.h"
+
+static const char kContentName[] = "test content";
+// Based on ICE_UFRAG_LENGTH
+static const char kIceUfrag[] = "TESTICEUFRAG0000";
+// Based on ICE_PWD_LENGTH
+static const char kIcePwd[] = "TESTICEPWD00000000000000";
+static const char kTurnUsername[] = "test";
+static const char kTurnPassword[] = "test";
+
+class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
+ public:
+  PortAllocatorTest() {
+    allocator_.reset(
+        new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
+  }
+
+ protected:
+  void SetConfigurationWithPoolSize(int candidate_pool_size) {
+    allocator_->SetConfiguration(cricket::ServerAddresses(),
+                                 std::vector<cricket::RelayServerConfig>(),
+                                 candidate_pool_size);
+  }
+
+  const cricket::FakePortAllocatorSession* GetPooledSession() const {
+    return static_cast<const cricket::FakePortAllocatorSession*>(
+        allocator_->GetPooledSession());
+  }
+
+  std::unique_ptr<cricket::FakePortAllocatorSession> TakePooledSession() {
+    return std::unique_ptr<cricket::FakePortAllocatorSession>(
+        static_cast<cricket::FakePortAllocatorSession*>(
+            allocator_->TakePooledSession(kContentName, 0, kIceUfrag, kIcePwd)
+                .release()));
+  }
+
+  int GetAllPooledSessionsReturnCount() {
+    int count = 0;
+    while (GetPooledSession()) {
+      TakePooledSession();
+      ++count;
+    }
+    return count;
+  }
+
+  std::unique_ptr<cricket::FakePortAllocator> allocator_;
+  rtc::SocketAddress stun_server_1{"11.11.11.11", 3478};
+  rtc::SocketAddress stun_server_2{"22.22.22.22", 3478};
+  cricket::RelayServerConfig turn_server_1{"11.11.11.11",      3478,
+                                           kTurnUsername,      kTurnPassword,
+                                           cricket::PROTO_UDP, false};
+  cricket::RelayServerConfig turn_server_2{"22.22.22.22",      3478,
+                                           kTurnUsername,      kTurnPassword,
+                                           cricket::PROTO_UDP, false};
+};
+
+TEST_F(PortAllocatorTest, TestDefaults) {
+  EXPECT_EQ(0UL, allocator_->stun_servers().size());
+  EXPECT_EQ(0UL, allocator_->turn_servers().size());
+  EXPECT_EQ(0, allocator_->candidate_pool_size());
+  EXPECT_EQ(0, GetAllPooledSessionsReturnCount());
+}
+
+TEST_F(PortAllocatorTest, SetConfigurationUpdatesIceServers) {
+  cricket::ServerAddresses stun_servers_1 = {stun_server_1};
+  std::vector<cricket::RelayServerConfig> turn_servers_1 = {turn_server_1};
+  allocator_->SetConfiguration(stun_servers_1, turn_servers_1, 0);
+  EXPECT_EQ(stun_servers_1, allocator_->stun_servers());
+  EXPECT_EQ(turn_servers_1, allocator_->turn_servers());
+
+  // Update with a different set of servers.
+  cricket::ServerAddresses stun_servers_2 = {stun_server_2};
+  std::vector<cricket::RelayServerConfig> turn_servers_2 = {turn_server_2};
+  allocator_->SetConfiguration(stun_servers_2, turn_servers_2, 0);
+  EXPECT_EQ(stun_servers_2, allocator_->stun_servers());
+  EXPECT_EQ(turn_servers_2, allocator_->turn_servers());
+}
+
+TEST_F(PortAllocatorTest, SetConfigurationUpdatesCandidatePoolSize) {
+  SetConfigurationWithPoolSize(2);
+  EXPECT_EQ(2, allocator_->candidate_pool_size());
+  SetConfigurationWithPoolSize(3);
+  EXPECT_EQ(3, allocator_->candidate_pool_size());
+  SetConfigurationWithPoolSize(1);
+  EXPECT_EQ(1, allocator_->candidate_pool_size());
+  SetConfigurationWithPoolSize(4);
+  EXPECT_EQ(4, allocator_->candidate_pool_size());
+}
+
+// A negative pool size should just be treated as zero.
+TEST_F(PortAllocatorTest, SetConfigurationWithNegativePoolSizeDoesntCrash) {
+  SetConfigurationWithPoolSize(-1);
+  // No asserts; we're just testing that this doesn't crash.
+}
+
+// Test that if the candidate pool size is nonzero, pooled sessions are
+// created, and StartGettingPorts is called on them.
+TEST_F(PortAllocatorTest, SetConfigurationCreatesPooledSessions) {
+  SetConfigurationWithPoolSize(2);
+  auto session_1 = TakePooledSession();
+  auto session_2 = TakePooledSession();
+  ASSERT_NE(nullptr, session_1.get());
+  ASSERT_NE(nullptr, session_2.get());
+  EXPECT_EQ(1, session_1->port_config_count());
+  EXPECT_EQ(1, session_2->port_config_count());
+  EXPECT_EQ(0, GetAllPooledSessionsReturnCount());
+}
+
+// Test that if the candidate pool size is increased, pooled sessions are
+// created as necessary.
+TEST_F(PortAllocatorTest, SetConfigurationCreatesMorePooledSessions) {
+  SetConfigurationWithPoolSize(1);
+  SetConfigurationWithPoolSize(2);
+  EXPECT_EQ(2, GetAllPooledSessionsReturnCount());
+}
+
+// Test that if the candidate pool size is reduced, extra sessions are
+// destroyed.
+TEST_F(PortAllocatorTest, SetConfigurationDestroysPooledSessions) {
+  SetConfigurationWithPoolSize(2);
+  SetConfigurationWithPoolSize(1);
+  EXPECT_EQ(1, GetAllPooledSessionsReturnCount());
+}
+
+// Test that if the candidate pool size is reduced and increased, but reducing
+// didn't actually destroy any sessions (because they were already given away),
+// increasing the size to its initial value doesn't create a new session.
+TEST_F(PortAllocatorTest, SetConfigurationDoesntCreateExtraSessions) {
+  SetConfigurationWithPoolSize(1);
+  TakePooledSession();
+  SetConfigurationWithPoolSize(0);
+  SetConfigurationWithPoolSize(1);
+  EXPECT_EQ(0, GetAllPooledSessionsReturnCount());
+}
+
+// According to JSEP, exising pooled sessions should be destroyed and new
+// ones created when the ICE servers change.
+TEST_F(PortAllocatorTest,
+       SetConfigurationRecreatesPooledSessionsWhenIceServersChange) {
+  cricket::ServerAddresses stun_servers_1 = {stun_server_1};
+  std::vector<cricket::RelayServerConfig> turn_servers_1 = {turn_server_1};
+  allocator_->SetConfiguration(stun_servers_1, turn_servers_1, 1);
+  EXPECT_EQ(stun_servers_1, allocator_->stun_servers());
+  EXPECT_EQ(turn_servers_1, allocator_->turn_servers());
+
+  // Update with a different set of servers (and also change pool size).
+  cricket::ServerAddresses stun_servers_2 = {stun_server_2};
+  std::vector<cricket::RelayServerConfig> turn_servers_2 = {turn_server_2};
+  allocator_->SetConfiguration(stun_servers_2, turn_servers_2, 2);
+  EXPECT_EQ(stun_servers_2, allocator_->stun_servers());
+  EXPECT_EQ(turn_servers_2, allocator_->turn_servers());
+  auto session_1 = TakePooledSession();
+  auto session_2 = TakePooledSession();
+  ASSERT_NE(nullptr, session_1.get());
+  ASSERT_NE(nullptr, session_2.get());
+  EXPECT_EQ(stun_servers_2, session_1->stun_servers());
+  EXPECT_EQ(turn_servers_2, session_1->turn_servers());
+  EXPECT_EQ(stun_servers_2, session_2->stun_servers());
+  EXPECT_EQ(turn_servers_2, session_2->turn_servers());
+  EXPECT_EQ(0, GetAllPooledSessionsReturnCount());
+}
+
+TEST_F(PortAllocatorTest, GetPooledSessionReturnsNextSession) {
+  SetConfigurationWithPoolSize(2);
+  auto peeked_session_1 = GetPooledSession();
+  auto session_1 = TakePooledSession();
+  EXPECT_EQ(session_1.get(), peeked_session_1);
+  auto peeked_session_2 = GetPooledSession();
+  auto session_2 = TakePooledSession();
+  EXPECT_EQ(session_2.get(), peeked_session_2);
+}
+
+// Verify that subclasses of PortAllocatorSession are given a chance to update
+// ICE parameters when TakePooledSession is called, and the base class updates
+// the info itself.
+TEST_F(PortAllocatorTest, TakePooledSessionUpdatesIceParameters) {
+  SetConfigurationWithPoolSize(1);
+  auto peeked_session = GetPooledSession();
+  ASSERT_NE(nullptr, peeked_session);
+  EXPECT_EQ(0, peeked_session->transport_info_update_count());
+  std::unique_ptr<cricket::FakePortAllocatorSession> session(
+      static_cast<cricket::FakePortAllocatorSession*>(
+          allocator_->TakePooledSession(kContentName, 1, kIceUfrag, kIcePwd)
+              .release()));
+  EXPECT_EQ(1, session->transport_info_update_count());
+  EXPECT_EQ(kContentName, session->content_name());
+  EXPECT_EQ(1, session->component());
+  EXPECT_EQ(kIceUfrag, session->ice_ufrag());
+  EXPECT_EQ(kIcePwd, session->ice_pwd());
+}
diff --git a/chromium/third_party/webrtc/p2p/base/pseudotcp.cc b/chromium/third_party/webrtc/p2p/base/pseudotcp.cc
index 1dfdcbb8944..44893747cce 100644
--- a/chromium/third_party/webrtc/p2p/base/pseudotcp.cc
+++ b/chromium/third_party/webrtc/p2p/base/pseudotcp.cc
@@ -14,6 +14,7 @@
 #include <stdlib.h>
 
 #include <algorithm>
+#include <memory>
 #include <set>
 
 #include "webrtc/base/arraysize.h"
@@ -22,7 +23,6 @@
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
@@ -203,9 +203,9 @@ void ReportStats() {
 
 uint32_t PseudoTcp::Now() {
 #if 0  // Use this to synchronize timers with logging timestamps (easier debug)
-  return rtc::TimeSince(StartTime());
+  return static_cast<uint32_t>(rtc::TimeSince(StartTime()));
 #else
-  return rtc::Time();
+  return rtc::Time32();
 #endif
 }
 
@@ -289,7 +289,7 @@ void PseudoTcp::NotifyClock(uint32_t now) {
     return;
 
     // Check if it's time to retransmit a segment
-  if (m_rto_base && (rtc::TimeDiff(m_rto_base + m_rx_rto, now) <= 0)) {
+  if (m_rto_base && (rtc::TimeDiff32(m_rto_base + m_rx_rto, now) <= 0)) {
     if (m_slist.empty()) {
       ASSERT(false);
     } else {
@@ -320,9 +320,8 @@ void PseudoTcp::NotifyClock(uint32_t now) {
   }
 
   // Check if it's time to probe closed windows
-  if ((m_snd_wnd == 0)
-        && (rtc::TimeDiff(m_lastsend + m_rx_rto, now) <= 0)) {
-    if (rtc::TimeDiff(now, m_lastrecv) >= 15000) {
+  if ((m_snd_wnd == 0) && (rtc::TimeDiff32(m_lastsend + m_rx_rto, now) <= 0)) {
+    if (rtc::TimeDiff32(now, m_lastrecv) >= 15000) {
       closedown(ECONNABORTED);
       return;
     }
@@ -336,19 +335,22 @@ void PseudoTcp::NotifyClock(uint32_t now) {
   }
 
   // Check if it's time to send delayed acks
-  if (m_t_ack && (rtc::TimeDiff(m_t_ack + m_ack_delay, now) <= 0)) {
+  if (m_t_ack && (rtc::TimeDiff32(m_t_ack + m_ack_delay, now) <= 0)) {
     packet(m_snd_nxt, 0, 0, 0);
   }
 
 #if PSEUDO_KEEPALIVE
   // Check for idle timeout
-  if ((m_state == TCP_ESTABLISHED) && (TimeDiff(m_lastrecv + IDLE_TIMEOUT, now) <= 0)) {
+  if ((m_state == TCP_ESTABLISHED) &&
+      (TimeDiff32(m_lastrecv + IDLE_TIMEOUT, now) <= 0)) {
     closedown(ECONNABORTED);
     return;
   }
 
   // Check for ping timeout (to keep udp mapping open)
-  if ((m_state == TCP_ESTABLISHED) && (TimeDiff(m_lasttraffic + (m_bOutgoing ? IDLE_PING * 3/2 : IDLE_PING), now) <= 0)) {
+  if ((m_state == TCP_ESTABLISHED) &&
+      (TimeDiff32(m_lasttraffic + (m_bOutgoing ? IDLE_PING * 3 / 2 : IDLE_PING),
+                  now) <= 0)) {
     packet(m_snd_nxt, 0, 0, 0);
   }
 #endif // PSEUDO_KEEPALIVE
@@ -518,7 +520,7 @@ IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32_t seq,
 
   uint32_t now = Now();
 
-  rtc::scoped_ptr<uint8_t[]> buffer(new uint8_t[MAX_PACKET]);
+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[MAX_PACKET]);
   long_to_bytes(m_conv, buffer.get());
   long_to_bytes(seq, buffer.get() + 4);
   long_to_bytes(m_rcv_nxt, buffer.get() + 8);
@@ -571,7 +573,7 @@ IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32_t seq,
 }
 
 bool PseudoTcp::parse(const uint8_t* buffer, uint32_t size) {
-  if (size < 12)
+  if (size < HEADER_SIZE)
     return false;
 
   Segment seg;
@@ -621,23 +623,24 @@ bool PseudoTcp::clock_check(uint32_t now, long& nTimeout) {
   nTimeout = DEFAULT_TIMEOUT;
 
   if (m_t_ack) {
-    nTimeout =
-        std::min<int32_t>(nTimeout, rtc::TimeDiff(m_t_ack + m_ack_delay, now));
+    nTimeout = std::min<int32_t>(nTimeout,
+                                 rtc::TimeDiff32(m_t_ack + m_ack_delay, now));
   }
   if (m_rto_base) {
-    nTimeout =
-        std::min<int32_t>(nTimeout, rtc::TimeDiff(m_rto_base + m_rx_rto, now));
+    nTimeout = std::min<int32_t>(nTimeout,
+                                 rtc::TimeDiff32(m_rto_base + m_rx_rto, now));
   }
   if (m_snd_wnd == 0) {
-    nTimeout =
-        std::min<int32_t>(nTimeout, rtc::TimeDiff(m_lastsend + m_rx_rto, now));
+    nTimeout = std::min<int32_t>(nTimeout,
+                                 rtc::TimeDiff32(m_lastsend + m_rx_rto, now));
   }
 #if PSEUDO_KEEPALIVE
   if (m_state == TCP_ESTABLISHED) {
     nTimeout = std::min<int32_t>(
-        nTimeout, rtc::TimeDiff(m_lasttraffic + (m_bOutgoing ? IDLE_PING * 3 / 2
-                                                             : IDLE_PING),
-                                now));
+        nTimeout,
+        rtc::TimeDiff32(
+            m_lasttraffic + (m_bOutgoing ? IDLE_PING * 3 / 2 : IDLE_PING),
+            now));
   }
 #endif // PSEUDO_KEEPALIVE
   return true;
@@ -710,7 +713,7 @@ bool PseudoTcp::process(Segment& seg) {
   if ((seg.ack > m_snd_una) && (seg.ack <= m_snd_nxt)) {
     // Calculate round-trip time
     if (seg.tsecr) {
-      int32_t rtt = rtc::TimeDiff(now, seg.tsecr);
+      int32_t rtt = rtc::TimeDiff32(now, seg.tsecr);
       if (rtt >= 0) {
         if (m_rx_srtt == 0) {
           m_rx_srtt = rtt;
@@ -1033,7 +1036,7 @@ bool PseudoTcp::transmit(const SList::iterator& seg, uint32_t now) {
 void PseudoTcp::attemptSend(SendFlags sflags) {
   uint32_t now = Now();
 
-  if (rtc::TimeDiff(now, m_lastsend) > static_cast<long>(m_rx_rto)) {
+  if (rtc::TimeDiff32(now, m_lastsend) > static_cast<long>(m_rx_rto)) {
     m_cwnd = m_mss;
   }
 
diff --git a/chromium/third_party/webrtc/p2p/base/pseudotcp_unittest.cc b/chromium/third_party/webrtc/p2p/base/pseudotcp_unittest.cc
index c9ccbca1d95..a635bcef169 100644
--- a/chromium/third_party/webrtc/p2p/base/pseudotcp_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/pseudotcp_unittest.cc
@@ -208,7 +208,8 @@ class PseudoTcpTestBase : public testing::Test,
 class PseudoTcpTest : public PseudoTcpTestBase {
  public:
   void TestTransfer(int size) {
-    uint32_t start, elapsed;
+    uint32_t start;
+    int32_t elapsed;
     size_t received;
     // Create some dummy data to send.
     send_stream_.ReserveSize(size);
@@ -220,13 +221,13 @@ class PseudoTcpTest : public PseudoTcpTestBase {
     // Prepare the receive stream.
     recv_stream_.ReserveSize(size);
     // Connect and wait until connected.
-    start = rtc::Time();
+    start = rtc::Time32();
     EXPECT_EQ(0, Connect());
     EXPECT_TRUE_WAIT(have_connected_, kConnectTimeoutMs);
     // Sending will start from OnTcpWriteable and complete when all data has
     // been received.
     EXPECT_TRUE_WAIT(have_disconnected_, kTransferTimeoutMs);
-    elapsed = rtc::TimeSince(start);
+    elapsed = rtc::Time32() - start;
     recv_stream_.GetSize(&received);
     // Ensure we closed down OK and we got the right data.
     // TODO: Ensure the errors are cleared properly.
@@ -339,7 +340,7 @@ class PseudoTcpTestPingPong : public PseudoTcpTestBase {
     // Prepare the receive stream.
     recv_stream_.ReserveSize(size);
     // Connect and wait until connected.
-    start = rtc::Time();
+    start = rtc::Time32();
     EXPECT_EQ(0, Connect());
     EXPECT_TRUE_WAIT(have_connected_, kConnectTimeoutMs);
     // Sending will start from OnTcpWriteable and stop when the required
diff --git a/chromium/third_party/webrtc/p2p/base/relayport.cc b/chromium/third_party/webrtc/p2p/base/relayport.cc
index 719604374e0..7f05e1b1ef7 100644
--- a/chromium/third_party/webrtc/p2p/base/relayport.cc
+++ b/chromium/third_party/webrtc/p2p/base/relayport.cc
@@ -779,7 +779,7 @@ AllocateRequest::AllocateRequest(RelayEntry* entry,
     : StunRequest(new RelayMessage()),
       entry_(entry),
       connection_(connection) {
-  start_time_ = rtc::Time64();
+  start_time_ = rtc::TimeMillis();
 }
 
 void AllocateRequest::Prepare(StunMessage* request) {
@@ -834,7 +834,7 @@ void AllocateRequest::OnErrorResponse(StunMessage* response) {
               << " reason='" << attr->reason() << "'";
   }
 
-  if (rtc::Time64() - start_time_ <= kRetryTimeout)
+  if (rtc::TimeMillis() - start_time_ <= kRetryTimeout)
     entry_->ScheduleKeepAlive();
 }
 
diff --git a/chromium/third_party/webrtc/p2p/base/relayport_unittest.cc b/chromium/third_party/webrtc/p2p/base/relayport_unittest.cc
index d644d67c4f3..738ea9abb21 100644
--- a/chromium/third_party/webrtc/p2p/base/relayport_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/relayport_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/relayport.h"
 #include "webrtc/p2p/base/relayserver.h"
@@ -15,7 +17,6 @@
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketadapters.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/ssladapter.h"
@@ -179,7 +180,7 @@ class RelayPortTest : public testing::Test,
 
     // Create a tcp server socket that listens on the fake address so
     // the relay port can attempt to connect to it.
-    rtc::scoped_ptr<rtc::AsyncSocket> tcp_server_socket(
+    std::unique_ptr<rtc::AsyncSocket> tcp_server_socket(
         CreateServerSocket(kRelayTcpAddr));
 
     // Add server addresses to the relay port and let it start.
@@ -244,16 +245,15 @@ class RelayPortTest : public testing::Test,
   typedef std::map<rtc::AsyncPacketSocket*, int> PacketMap;
 
   rtc::Thread* main_;
-  rtc::scoped_ptr<rtc::PhysicalSocketServer>
-      physical_socket_server_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> virtual_socket_server_;
+  std::unique_ptr<rtc::PhysicalSocketServer> physical_socket_server_;
+  std::unique_ptr<rtc::VirtualSocketServer> virtual_socket_server_;
   rtc::SocketServerScope ss_scope_;
   rtc::Network network_;
   rtc::BasicPacketSocketFactory socket_factory_;
   std::string username_;
   std::string password_;
-  rtc::scoped_ptr<cricket::RelayPort> relay_port_;
-  rtc::scoped_ptr<cricket::RelayServer> relay_server_;
+  std::unique_ptr<cricket::RelayPort> relay_port_;
+  std::unique_ptr<cricket::RelayServer> relay_server_;
   std::vector<cricket::ProtocolAddress> failed_connections_;
   std::vector<cricket::ProtocolAddress> soft_timedout_connections_;
   PacketMap received_packet_count_;
diff --git a/chromium/third_party/webrtc/p2p/base/relayserver.cc b/chromium/third_party/webrtc/p2p/base/relayserver.cc
index e098cbcd843..ebe46c48c6a 100644
--- a/chromium/third_party/webrtc/p2p/base/relayserver.cc
+++ b/chromium/third_party/webrtc/p2p/base/relayserver.cc
@@ -694,7 +694,7 @@ void RelayServerBinding::AddExternalConnection(RelayServerConnection* conn) {
 }
 
 void RelayServerBinding::NoteUsed() {
-  last_used_ = rtc::Time64();
+  last_used_ = rtc::TimeMillis();
 }
 
 bool RelayServerBinding::HasMagicCookie(const char* bytes, size_t size) const {
@@ -735,7 +735,7 @@ void RelayServerBinding::OnMessage(rtc::Message *pmsg) {
 
     // If the lifetime timeout has been exceeded, then send a signal.
     // Otherwise, just keep waiting.
-    if (rtc::Time64() >= last_used_ + lifetime_) {
+    if (rtc::TimeMillis() >= last_used_ + lifetime_) {
       LOG(LS_INFO) << "Expiring binding " << username_;
       SignalTimeout(this);
     } else {
diff --git a/chromium/third_party/webrtc/p2p/base/relayserver_unittest.cc b/chromium/third_party/webrtc/p2p/base/relayserver_unittest.cc
index 3581f7153ba..1d07a07c7fb 100644
--- a/chromium/third_party/webrtc/p2p/base/relayserver_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/relayserver_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/p2p/base/relayserver.h"
@@ -60,16 +61,14 @@ class RelayServerTest : public testing::Test {
   }
 
   void Allocate() {
-    rtc::scoped_ptr<StunMessage> req(
-        CreateStunMessage(STUN_ALLOCATE_REQUEST));
+    std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_ALLOCATE_REQUEST));
     AddUsernameAttr(req.get(), username_);
     AddLifetimeAttr(req.get(), LIFETIME);
     Send1(req.get());
     delete Receive1();
   }
   void Bind() {
-    rtc::scoped_ptr<StunMessage> req(
-        CreateStunMessage(STUN_BINDING_REQUEST));
+    std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_BINDING_REQUEST));
     AddUsernameAttr(req.get(), username_);
     Send2(req.get());
     delete Receive1();
@@ -172,12 +171,12 @@ class RelayServerTest : public testing::Test {
     msg->AddAttribute(attr);
   }
 
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
-  rtc::scoped_ptr<RelayServer> server_;
-  rtc::scoped_ptr<rtc::TestClient> client1_;
-  rtc::scoped_ptr<rtc::TestClient> client2_;
+  std::unique_ptr<RelayServer> server_;
+  std::unique_ptr<rtc::TestClient> client1_;
+  std::unique_ptr<rtc::TestClient> client2_;
   std::string username_;
   std::string password_;
 };
@@ -190,8 +189,8 @@ TEST_F(RelayServerTest, TestBadRequest) {
 
 // Send an allocate request without a username and verify it is rejected.
 TEST_F(RelayServerTest, TestAllocateNoUsername) {
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_ALLOCATE_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_ALLOCATE_REQUEST)),
+      res;
 
   Send1(req.get());
   res.reset(Receive1());
@@ -209,8 +208,8 @@ TEST_F(RelayServerTest, TestAllocateNoUsername) {
 
 // Send a binding request and verify that it is rejected.
 TEST_F(RelayServerTest, TestBindingRequest) {
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_BINDING_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_BINDING_REQUEST)),
+      res;
   AddUsernameAttr(req.get(), username_);
 
   Send1(req.get());
@@ -229,8 +228,8 @@ TEST_F(RelayServerTest, TestBindingRequest) {
 
 // Send an allocate request and verify that it is accepted.
 TEST_F(RelayServerTest, TestAllocate) {
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_ALLOCATE_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_ALLOCATE_REQUEST)),
+      res;
   AddUsernameAttr(req.get(), username_);
   AddLifetimeAttr(req.get(), LIFETIME);
 
@@ -259,8 +258,8 @@ TEST_F(RelayServerTest, TestAllocate) {
 TEST_F(RelayServerTest, TestReallocate) {
   Allocate();
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_ALLOCATE_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_ALLOCATE_REQUEST)),
+      res;
   AddMagicCookieAttr(req.get());
   AddUsernameAttr(req.get(), username_);
 
@@ -289,8 +288,8 @@ TEST_F(RelayServerTest, TestReallocate) {
 TEST_F(RelayServerTest, TestRemoteBind) {
   Allocate();
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_BINDING_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_BINDING_REQUEST)),
+      res;
   AddUsernameAttr(req.get(), username_);
 
   Send2(req.get());
@@ -304,7 +303,7 @@ TEST_F(RelayServerTest, TestRemoteBind) {
   ASSERT_TRUE(recv_data != NULL);
 
   rtc::ByteBufferReader buf(recv_data->bytes(), recv_data->length());
-  rtc::scoped_ptr<StunMessage> res2(new StunMessage());
+  std::unique_ptr<StunMessage> res2(new StunMessage());
   EXPECT_TRUE(res2->Read(&buf));
   EXPECT_EQ(STUN_BINDING_REQUEST, res2->type());
   EXPECT_EQ(req->transaction_id(), res2->transaction_id());
@@ -335,8 +334,7 @@ TEST_F(RelayServerTest, TestSendRequestMissingUsername) {
   Allocate();
   Bind();
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_SEND_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_SEND_REQUEST)), res;
   AddMagicCookieAttr(req.get());
 
   Send1(req.get());
@@ -358,8 +356,7 @@ TEST_F(RelayServerTest, TestSendRequestBadUsername) {
   Allocate();
   Bind();
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_SEND_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_SEND_REQUEST)), res;
   AddMagicCookieAttr(req.get());
   AddUsernameAttr(req.get(), "foobarbizbaz");
 
@@ -383,8 +380,7 @@ TEST_F(RelayServerTest, TestSendRequestNoDestinationAddress) {
   Allocate();
   Bind();
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_SEND_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_SEND_REQUEST)), res;
   AddMagicCookieAttr(req.get());
   AddUsernameAttr(req.get(), username_);
 
@@ -407,8 +403,7 @@ TEST_F(RelayServerTest, TestSendRequestNoData) {
   Allocate();
   Bind();
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_SEND_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_SEND_REQUEST)), res;
   AddMagicCookieAttr(req.get());
   AddUsernameAttr(req.get(), username_);
   AddDestinationAttr(req.get(), client2_addr);
@@ -432,8 +427,8 @@ TEST_F(RelayServerTest, TestSendRequestWrongType) {
   Allocate();
   Bind();
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_BINDING_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_BINDING_REQUEST)),
+      res;
   AddMagicCookieAttr(req.get());
   AddUsernameAttr(req.get(), username_);
 
@@ -458,8 +453,7 @@ TEST_F(RelayServerTest, TestSendRaw) {
   Bind();
 
   for (int i = 0; i < 10; i++) {
-    rtc::scoped_ptr<StunMessage> req(
-        CreateStunMessage(STUN_SEND_REQUEST)), res;
+    std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_SEND_REQUEST)), res;
     AddMagicCookieAttr(req.get());
     AddUsernameAttr(req.get(), username_);
     AddDestinationAttr(req.get(), client2_addr);
@@ -501,8 +495,7 @@ TEST_F(RelayServerTest, DISABLED_TestExpiration) {
   // Wait twice the lifetime to make sure the server has expired the binding.
   rtc::Thread::Current()->ProcessMessages((LIFETIME * 2) * 1000);
 
-  rtc::scoped_ptr<StunMessage> req(
-      CreateStunMessage(STUN_SEND_REQUEST)), res;
+  std::unique_ptr<StunMessage> req(CreateStunMessage(STUN_SEND_REQUEST)), res;
   AddMagicCookieAttr(req.get());
   AddUsernameAttr(req.get(), username_);
   AddDestinationAttr(req.get(), client2_addr);
diff --git a/chromium/third_party/webrtc/p2p/base/stun.cc b/chromium/third_party/webrtc/p2p/base/stun.cc
index d0f001485de..ac3fd5f9368 100644
--- a/chromium/third_party/webrtc/p2p/base/stun.cc
+++ b/chromium/third_party/webrtc/p2p/base/stun.cc
@@ -12,12 +12,13 @@
 
 #include <string.h>
 
+#include <memory>
+
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/crc32.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagedigest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringencode.h"
 
 using rtc::ByteBufferReader;
@@ -173,7 +174,7 @@ bool StunMessage::ValidateMessageIntegrity(const char* data, size_t size,
 
   // Getting length of the message to calculate Message Integrity.
   size_t mi_pos = current_pos;
-  rtc::scoped_ptr<char[]> temp_data(new char[current_pos]);
+  std::unique_ptr<char[]> temp_data(new char[current_pos]);
   memcpy(temp_data.get(), data, current_pos);
   if (size > mi_pos + kStunAttributeHeaderSize + kStunMessageIntegritySize) {
     // Stun message has other attributes after message integrity.
diff --git a/chromium/third_party/webrtc/p2p/base/stun_unittest.cc b/chromium/third_party/webrtc/p2p/base/stun_unittest.cc
index 2213397f2e4..d7ca9991f85 100644
--- a/chromium/third_party/webrtc/p2p/base/stun_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/stun_unittest.cc
@@ -16,7 +16,6 @@
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagedigest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketaddress.h"
 
 namespace cricket {
diff --git a/chromium/third_party/webrtc/p2p/base/stunport.cc b/chromium/third_party/webrtc/p2p/base/stunport.cc
index 16546faaf1b..8ed8c448872 100644
--- a/chromium/third_party/webrtc/p2p/base/stunport.cc
+++ b/chromium/third_party/webrtc/p2p/base/stunport.cc
@@ -65,7 +65,7 @@ class StunBindingRequest : public StunRequest {
     }
 
     // The keep-alive requests will be stopped after its lifetime has passed.
-    if (WithinLifetime(rtc::Time64())) {
+    if (WithinLifetime(rtc::TimeMillis())) {
       port_->requests_.SendDelayed(
           new StunBindingRequest(port_, server_addr_, start_time_, lifetime_),
           port_->stun_keepalive_delay());
@@ -85,9 +85,9 @@ class StunBindingRequest : public StunRequest {
 
     port_->OnStunBindingOrResolveRequestFailed(server_addr_);
 
-    int64_t now = rtc::Time64();
+    int64_t now = rtc::TimeMillis();
     if (WithinLifetime(now) &&
-        rtc::TimeDiff64(now, start_time_) < RETRY_TIMEOUT) {
+        rtc::TimeDiff(now, start_time_) < RETRY_TIMEOUT) {
       port_->requests_.SendDelayed(
           new StunBindingRequest(port_, server_addr_, start_time_, lifetime_),
           port_->stun_keepalive_delay());
@@ -105,7 +105,7 @@ class StunBindingRequest : public StunRequest {
   // Returns true if |now| is within the lifetime of the request (a negative
   // lifetime means infinite).
   bool WithinLifetime(int64_t now) const {
-    return lifetime_ < 0 || rtc::TimeDiff64(now, start_time_) <= lifetime_;
+    return lifetime_ < 0 || rtc::TimeDiff(now, start_time_) <= lifetime_;
   }
   UDPPort* port_;
   const rtc::SocketAddress server_addr_;
@@ -411,7 +411,7 @@ void UDPPort::SendStunBindingRequest(const rtc::SocketAddress& stun_addr) {
   } else if (socket_->GetState() == rtc::AsyncPacketSocket::STATE_BOUND) {
     // Check if |server_addr_| is compatible with the port's ip.
     if (IsCompatibleAddress(stun_addr)) {
-      requests_.Send(new StunBindingRequest(this, stun_addr, rtc::Time64(),
+      requests_.Send(new StunBindingRequest(this, stun_addr, rtc::TimeMillis(),
                                             stun_keepalive_lifetime_));
     } else {
       // Since we can't send stun messages to the server, we should mark this
diff --git a/chromium/third_party/webrtc/p2p/base/stunport.h b/chromium/third_party/webrtc/p2p/base/stunport.h
index a0eba51cd43..cd844aa273e 100644
--- a/chromium/third_party/webrtc/p2p/base/stunport.h
+++ b/chromium/third_party/webrtc/p2p/base/stunport.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_P2P_BASE_STUNPORT_H_
 #define WEBRTC_P2P_BASE_STUNPORT_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/p2p/base/port.h"
@@ -224,7 +225,7 @@ class UDPPort : public Port {
   StunRequestManager requests_;
   rtc::AsyncPacketSocket* socket_;
   int error_;
-  rtc::scoped_ptr<AddressResolver> resolver_;
+  std::unique_ptr<AddressResolver> resolver_;
   bool ready_;
   int stun_keepalive_delay_;
   int stun_keepalive_lifetime_;
diff --git a/chromium/third_party/webrtc/p2p/base/stunport_unittest.cc b/chromium/third_party/webrtc/p2p/base/stunport_unittest.cc
index 1926b932799..702ec2504e8 100644
--- a/chromium/third_party/webrtc/p2p/base/stunport_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/stunport_unittest.cc
@@ -8,13 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/stunport.h"
 #include "webrtc/p2p/base/teststunserver.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/virtualsocketserver.h"
@@ -155,15 +156,15 @@ class StunPortTest : public testing::Test,
   }
 
  private:
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
   rtc::Network network_;
   rtc::BasicPacketSocketFactory socket_factory_;
-  rtc::scoped_ptr<cricket::UDPPort> stun_port_;
-  rtc::scoped_ptr<cricket::TestStunServer> stun_server_1_;
-  rtc::scoped_ptr<cricket::TestStunServer> stun_server_2_;
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
+  std::unique_ptr<cricket::UDPPort> stun_port_;
+  std::unique_ptr<cricket::TestStunServer> stun_server_1_;
+  std::unique_ptr<cricket::TestStunServer> stun_server_2_;
+  std::unique_ptr<rtc::AsyncPacketSocket> socket_;
   bool done_;
   bool error_;
   int stun_keepalive_delay_;
diff --git a/chromium/third_party/webrtc/p2p/base/stunrequest.cc b/chromium/third_party/webrtc/p2p/base/stunrequest.cc
index 546dd157d99..b75bcf67cc5 100644
--- a/chromium/third_party/webrtc/p2p/base/stunrequest.cc
+++ b/chromium/third_party/webrtc/p2p/base/stunrequest.cc
@@ -11,6 +11,8 @@
 #include "webrtc/p2p/base/stunrequest.h"
 
 #include <algorithm>
+#include <memory>
+
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
@@ -139,7 +141,7 @@ bool StunRequestManager::CheckResponse(const char* data, size_t size) {
   // Parse the STUN message and continue processing as usual.
 
   rtc::ByteBufferReader buf(data, size);
-  rtc::scoped_ptr<StunMessage> response(iter->second->msg_->CreateNew());
+  std::unique_ptr<StunMessage> response(iter->second->msg_->CreateNew());
   if (!response->Read(&buf)) {
     LOG(LS_WARNING) << "Failed to read STUN response " << rtc::hex_encode(id);
     return false;
@@ -192,7 +194,7 @@ const StunMessage* StunRequest::msg() const {
 }
 
 int StunRequest::Elapsed() const {
-  return static_cast<int>(rtc::Time64() - tstamp_);
+  return static_cast<int>(rtc::TimeMillis() - tstamp_);
 }
 
 
@@ -211,7 +213,7 @@ void StunRequest::OnMessage(rtc::Message* pmsg) {
     return;
   }
 
-  tstamp_ = rtc::Time64();
+  tstamp_ = rtc::TimeMillis();
 
   rtc::ByteBufferWriter buf;
   msg_->Write(&buf);
diff --git a/chromium/third_party/webrtc/p2p/base/stunrequest_unittest.cc b/chromium/third_party/webrtc/p2p/base/stunrequest_unittest.cc
index 5e4d2560021..5845a0ecbbb 100644
--- a/chromium/third_party/webrtc/p2p/base/stunrequest_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/stunrequest_unittest.cc
@@ -146,13 +146,13 @@ TEST_F(StunRequestTest, TestUnexpected) {
 TEST_F(StunRequestTest, TestBackoff) {
   StunMessage* req = CreateStunMessage(STUN_BINDING_REQUEST, NULL);
 
-  int64_t start = rtc::Time64();
+  int64_t start = rtc::TimeMillis();
   manager_.Send(new StunRequestThunker(req, this));
   StunMessage* res = CreateStunMessage(STUN_BINDING_RESPONSE, req);
   for (int i = 0; i < 9; ++i) {
     while (request_count_ == i)
       rtc::Thread::Current()->ProcessMessages(1);
-    int64_t elapsed = rtc::Time64() - start;
+    int64_t elapsed = rtc::TimeMillis() - start;
     LOG(LS_INFO) << "STUN request #" << (i + 1)
                  << " sent at " << elapsed << " ms";
     EXPECT_GE(TotalDelay(i + 1), elapsed);
diff --git a/chromium/third_party/webrtc/p2p/base/stunserver.h b/chromium/third_party/webrtc/p2p/base/stunserver.h
index a7eeab15445..9d1c169a507 100644
--- a/chromium/third_party/webrtc/p2p/base/stunserver.h
+++ b/chromium/third_party/webrtc/p2p/base/stunserver.h
@@ -11,9 +11,10 @@
 #ifndef WEBRTC_P2P_BASE_STUNSERVER_H_
 #define WEBRTC_P2P_BASE_STUNSERVER_H_
 
+#include <memory>
+
 #include "webrtc/p2p/base/stun.h"
 #include "webrtc/base/asyncudpsocket.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace cricket {
 
@@ -58,7 +59,7 @@ class StunServer : public sigslot::has_slots<> {
                            StunMessage* response) const;
 
  private:
-  rtc::scoped_ptr<rtc::AsyncUDPSocket> socket_;
+  std::unique_ptr<rtc::AsyncUDPSocket> socket_;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/stunserver_unittest.cc b/chromium/third_party/webrtc/p2p/base/stunserver_unittest.cc
index 973ab2adfc7..e468447005a 100644
--- a/chromium/third_party/webrtc/p2p/base/stunserver_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/stunserver_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 
 #include "webrtc/p2p/base/stunserver.h"
@@ -62,11 +63,11 @@ class StunServerTest : public testing::Test {
     return msg;
   }
  private:
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::Thread worker_;
-  rtc::scoped_ptr<StunServer> server_;
-  rtc::scoped_ptr<rtc::TestClient> client_;
+  std::unique_ptr<StunServer> server_;
+  std::unique_ptr<rtc::TestClient> client_;
 };
 
 // Disable for TSan v2, see
diff --git a/chromium/third_party/webrtc/p2p/base/tcpport.cc b/chromium/third_party/webrtc/p2p/base/tcpport.cc
index 59c4f317d4b..5ccb8108a05 100644
--- a/chromium/third_party/webrtc/p2p/base/tcpport.cc
+++ b/chromium/third_party/webrtc/p2p/base/tcpport.cc
@@ -387,28 +387,35 @@ void TCPConnection::OnConnect(rtc::AsyncPacketSocket* socket) {
   // the one we asked for. This is seen in Chrome, where TCP sockets cannot be
   // given a binding address, and the platform is expected to pick the
   // correct local address.
-  const rtc::IPAddress& socket_ip = socket->GetLocalAddress().ipaddr();
-  if (socket_ip == port()->ip() || IPIsAny(port()->ip())) {
-    if (socket_ip == port()->ip()) {
-      LOG_J(LS_VERBOSE, this) << "Connection established to "
-                              << socket->GetRemoteAddress().ToSensitiveString();
-    } else {
-      LOG(LS_WARNING) << "Socket is bound to a different address:"
-                      << socket->GetLocalAddress().ipaddr().ToString()
-                      << ", rather then the local port:"
-                      << port()->ip().ToString()
-                      << ". Still allowing it since it's any address"
-                      << ", possibly caused by multi-routes being disabled.";
-    }
-    set_connected(true);
-    connection_pending_ = false;
+  const rtc::SocketAddress& socket_addr = socket->GetLocalAddress();
+  if (socket_addr.ipaddr() == port()->ip()) {
+    LOG_J(LS_VERBOSE, this) << "Connection established to "
+                            << socket->GetRemoteAddress().ToSensitiveString();
+  } else if (IPIsAny(port()->ip())) {
+    LOG(LS_WARNING) << "Socket is bound to a different address:"
+                    << socket_addr.ipaddr().ToString()
+                    << ", rather then the local port:"
+                    << port()->ip().ToString()
+                    << ". Still allowing it since it's any address"
+                    << ", possibly caused by multi-routes being disabled.";
+  } else if (socket_addr.IsLoopbackIP()) {
+    LOG(LS_WARNING) << "Socket is bound to a different address:"
+                    << socket_addr.ipaddr().ToString()
+                    << ", rather then the local port:"
+                    << port()->ip().ToString()
+                    << ". Still allowing it since it's localhost.";
   } else {
     LOG_J(LS_WARNING, this) << "Dropping connection as TCP socket bound to IP "
-                            << socket_ip.ToSensitiveString()
+                            << socket_addr.ipaddr().ToSensitiveString()
                             << ", different from the local candidate IP "
                             << port()->ip().ToSensitiveString();
     OnClose(socket, 0);
+    return;
   }
+
+  // Connection is established successfully.
+  set_connected(true);
+  connection_pending_ = false;
 }
 
 void TCPConnection::OnClose(rtc::AsyncPacketSocket* socket, int error) {
diff --git a/chromium/third_party/webrtc/p2p/base/tcpport.h b/chromium/third_party/webrtc/p2p/base/tcpport.h
index cfc6245601e..77bbd09eab0 100644
--- a/chromium/third_party/webrtc/p2p/base/tcpport.h
+++ b/chromium/third_party/webrtc/p2p/base/tcpport.h
@@ -12,7 +12,9 @@
 #define WEBRTC_P2P_BASE_TCPPORT_H_
 
 #include <list>
+#include <memory>
 #include <string>
+
 #include "webrtc/p2p/base/port.h"
 #include "webrtc/base/asyncpacketsocket.h"
 
@@ -164,7 +166,7 @@ class TCPConnection : public Connection {
                     const rtc::PacketTime& packet_time);
   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
 
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
+  std::unique_ptr<rtc::AsyncPacketSocket> socket_;
   int error_;
   bool outgoing_;
 
diff --git a/chromium/third_party/webrtc/p2p/base/tcpport_unittest.cc b/chromium/third_party/webrtc/p2p/base/tcpport_unittest.cc
new file mode 100644
index 00000000000..e33dd433a7d
--- /dev/null
+++ b/chromium/third_party/webrtc/p2p/base/tcpport_unittest.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/physicalsocketserver.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/base/tcpport.h"
+
+using rtc::SocketAddress;
+using cricket::Connection;
+using cricket::Port;
+using cricket::TCPPort;
+using cricket::ICE_UFRAG_LENGTH;
+using cricket::ICE_PWD_LENGTH;
+
+static int kTimeout = 1000;
+static const SocketAddress kLocalAddr("11.11.11.11", 1);
+static const SocketAddress kRemoteAddr("22.22.22.22", 2);
+
+class TCPPortTest : public testing::Test, public sigslot::has_slots<> {
+ public:
+  TCPPortTest()
+      : main_(rtc::Thread::Current()),
+        pss_(new rtc::PhysicalSocketServer),
+        ss_(new rtc::VirtualSocketServer(pss_.get())),
+        ss_scope_(ss_.get()),
+        network_("unittest", "unittest", rtc::IPAddress(INADDR_ANY), 32),
+        socket_factory_(rtc::Thread::Current()),
+        username_(rtc::CreateRandomString(ICE_UFRAG_LENGTH)),
+        password_(rtc::CreateRandomString(ICE_PWD_LENGTH)) {
+    network_.AddIP(rtc::IPAddress(INADDR_ANY));
+  }
+
+  void ConnectSignalSocketCreated() {
+    ss_->SignalSocketCreated.connect(this, &TCPPortTest::OnSocketCreated);
+  }
+
+  void OnSocketCreated(rtc::VirtualSocket* socket) {
+    LOG(LS_INFO) << "socket created ";
+    socket->SignalAddressReady.connect(
+        this, &TCPPortTest::SetLocalhostAsAlternativeLocalAddress);
+  }
+
+  void SetLocalhostAsAlternativeLocalAddress(rtc::VirtualSocket* socket,
+                                             const SocketAddress& address) {
+    SocketAddress local_address("127.0.0.1", 2000);
+    socket->SetAlternativeLocalAddress(local_address);
+  }
+
+  TCPPort* CreateTCPPort(const SocketAddress& addr) {
+    return TCPPort::Create(main_, &socket_factory_, &network_, addr.ipaddr(), 0,
+                           0, username_, password_, true);
+  }
+
+ protected:
+  rtc::Thread* main_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
+  rtc::SocketServerScope ss_scope_;
+  rtc::Network network_;
+  rtc::BasicPacketSocketFactory socket_factory_;
+  std::string username_;
+  std::string password_;
+};
+
+TEST_F(TCPPortTest, TestTCPPortWithLocalhostAddress) {
+  std::unique_ptr<TCPPort> lport(CreateTCPPort(kLocalAddr));
+  std::unique_ptr<TCPPort> rport(CreateTCPPort(kRemoteAddr));
+  lport->PrepareAddress();
+  rport->PrepareAddress();
+  // Start to listen to new socket creation event.
+  ConnectSignalSocketCreated();
+  Connection* conn =
+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
+  EXPECT_TRUE_WAIT(conn->connected(), kTimeout);
+}
diff --git a/chromium/third_party/webrtc/p2p/base/testrelayserver.h b/chromium/third_party/webrtc/p2p/base/testrelayserver.h
index 87cb9e5dc32..7bc0beead23 100644
--- a/chromium/third_party/webrtc/p2p/base/testrelayserver.h
+++ b/chromium/third_party/webrtc/p2p/base/testrelayserver.h
@@ -11,9 +11,10 @@
 #ifndef WEBRTC_P2P_BASE_TESTRELAYSERVER_H_
 #define WEBRTC_P2P_BASE_TESTRELAYSERVER_H_
 
+#include <memory>
+
 #include "webrtc/p2p/base/relayserver.h"
 #include "webrtc/base/asynctcpsocket.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socketadapters.h"
 #include "webrtc/base/thread.h"
@@ -90,10 +91,10 @@ class TestRelayServer : public sigslot::has_slots<> {
   }
  private:
   cricket::RelayServer server_;
-  rtc::scoped_ptr<rtc::AsyncSocket> tcp_int_socket_;
-  rtc::scoped_ptr<rtc::AsyncSocket> tcp_ext_socket_;
-  rtc::scoped_ptr<rtc::AsyncSocket> ssl_int_socket_;
-  rtc::scoped_ptr<rtc::AsyncSocket> ssl_ext_socket_;
+  std::unique_ptr<rtc::AsyncSocket> tcp_int_socket_;
+  std::unique_ptr<rtc::AsyncSocket> tcp_ext_socket_;
+  std::unique_ptr<rtc::AsyncSocket> ssl_int_socket_;
+  std::unique_ptr<rtc::AsyncSocket> ssl_ext_socket_;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/transport.cc b/chromium/third_party/webrtc/p2p/base/transport.cc
index a1988820ed3..9688e4aedcd 100644
--- a/chromium/third_party/webrtc/p2p/base/transport.cc
+++ b/chromium/third_party/webrtc/p2p/base/transport.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <utility>  // for std::pair
 
 #include "webrtc/p2p/base/transport.h"
@@ -77,7 +78,7 @@ void Transport::SetIceRole(IceRole role) {
   }
 }
 
-rtc::scoped_ptr<rtc::SSLCertificate> Transport::GetRemoteSSLCertificate() {
+std::unique_ptr<rtc::SSLCertificate> Transport::GetRemoteSSLCertificate() {
   if (channels_.empty()) {
     return nullptr;
   }
@@ -227,21 +228,7 @@ void Transport::ConnectChannels() {
 
   connect_requested_ = true;
 
-  if (!local_description_) {
-    // TOOD(mallinath) : TransportDescription(TD) shouldn't be generated here.
-    // As Transport must know TD is offer or answer and cricket::Transport
-    // doesn't have the capability to decide it. This should be set by the
-    // Session.
-    // Session must generate local TD before remote candidates pushed when
-    // initiate request initiated by the remote.
-    LOG(LS_INFO) << "Transport::ConnectChannels: No local description has "
-                 << "been set. Will generate one.";
-    TransportDescription desc(std::vector<std::string>(),
-                              rtc::CreateRandomString(ICE_UFRAG_LENGTH),
-                              rtc::CreateRandomString(ICE_PWD_LENGTH),
-                              ICEMODE_FULL, CONNECTIONROLE_NONE, nullptr);
-    SetLocalTransportDescription(desc, CA_OFFER, nullptr);
-  }
+  RTC_DCHECK(local_description_);
 
   CallChannels(&TransportChannelImpl::Connect);
 }
@@ -414,4 +401,107 @@ bool Transport::NegotiateTransportDescription(ContentAction local_role,
   return true;
 }
 
+bool Transport::VerifyCertificateFingerprint(
+    const rtc::RTCCertificate* certificate,
+    const rtc::SSLFingerprint* fingerprint,
+    std::string* error_desc) const {
+  if (!fingerprint) {
+    return BadTransportDescription("No fingerprint.", error_desc);
+  }
+  if (!certificate) {
+    return BadTransportDescription(
+        "Fingerprint provided but no identity available.", error_desc);
+  }
+  std::unique_ptr<rtc::SSLFingerprint> fp_tmp(rtc::SSLFingerprint::Create(
+      fingerprint->algorithm, certificate->identity()));
+  ASSERT(fp_tmp.get() != NULL);
+  if (*fp_tmp == *fingerprint) {
+    return true;
+  }
+  std::ostringstream desc;
+  desc << "Local fingerprint does not match identity. Expected: ";
+  desc << fp_tmp->ToString();
+  desc << " Got: " << fingerprint->ToString();
+  return BadTransportDescription(desc.str(), error_desc);
+}
+
+bool Transport::NegotiateRole(ContentAction local_role,
+                              rtc::SSLRole* ssl_role,
+                              std::string* error_desc) const {
+  RTC_DCHECK(ssl_role);
+  if (!local_description() || !remote_description()) {
+    const std::string msg =
+        "Local and Remote description must be set before "
+        "transport descriptions are negotiated";
+    return BadTransportDescription(msg, error_desc);
+  }
+
+  // From RFC 4145, section-4.1, The following are the values that the
+  // 'setup' attribute can take in an offer/answer exchange:
+  //       Offer      Answer
+  //      ________________
+  //      active     passive / holdconn
+  //      passive    active / holdconn
+  //      actpass    active / passive / holdconn
+  //      holdconn   holdconn
+  //
+  // Set the role that is most conformant with RFC 5763, Section 5, bullet 1
+  // The endpoint MUST use the setup attribute defined in [RFC4145].
+  // The endpoint that is the offerer MUST use the setup attribute
+  // value of setup:actpass and be prepared to receive a client_hello
+  // before it receives the answer.  The answerer MUST use either a
+  // setup attribute value of setup:active or setup:passive.  Note that
+  // if the answerer uses setup:passive, then the DTLS handshake will
+  // not begin until the answerer is received, which adds additional
+  // latency. setup:active allows the answer and the DTLS handshake to
+  // occur in parallel.  Thus, setup:active is RECOMMENDED.  Whichever
+  // party is active MUST initiate a DTLS handshake by sending a
+  // ClientHello over each flow (host/port quartet).
+  // IOW - actpass and passive modes should be treated as server and
+  // active as client.
+  ConnectionRole local_connection_role = local_description()->connection_role;
+  ConnectionRole remote_connection_role = remote_description()->connection_role;
+
+  bool is_remote_server = false;
+  if (local_role == CA_OFFER) {
+    if (local_connection_role != CONNECTIONROLE_ACTPASS) {
+      return BadTransportDescription(
+          "Offerer must use actpass value for setup attribute.", error_desc);
+    }
+
+    if (remote_connection_role == CONNECTIONROLE_ACTIVE ||
+        remote_connection_role == CONNECTIONROLE_PASSIVE ||
+        remote_connection_role == CONNECTIONROLE_NONE) {
+      is_remote_server = (remote_connection_role == CONNECTIONROLE_PASSIVE);
+    } else {
+      const std::string msg =
+          "Answerer must use either active or passive value "
+          "for setup attribute.";
+      return BadTransportDescription(msg, error_desc);
+    }
+    // If remote is NONE or ACTIVE it will act as client.
+  } else {
+    if (remote_connection_role != CONNECTIONROLE_ACTPASS &&
+        remote_connection_role != CONNECTIONROLE_NONE) {
+      return BadTransportDescription(
+          "Offerer must use actpass value for setup attribute.", error_desc);
+    }
+
+    if (local_connection_role == CONNECTIONROLE_ACTIVE ||
+        local_connection_role == CONNECTIONROLE_PASSIVE) {
+      is_remote_server = (local_connection_role == CONNECTIONROLE_ACTIVE);
+    } else {
+      const std::string msg =
+          "Answerer must use either active or passive value "
+          "for setup attribute.";
+      return BadTransportDescription(msg, error_desc);
+    }
+
+    // If local is passive, local will act as server.
+  }
+
+  *ssl_role = is_remote_server ? rtc::SSL_CLIENT : rtc::SSL_SERVER;
+  return true;
+}
+
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/transport.h b/chromium/third_party/webrtc/p2p/base/transport.h
index 8b30127b7f9..e31d37a6f6a 100644
--- a/chromium/third_party/webrtc/p2p/base/transport.h
+++ b/chromium/third_party/webrtc/p2p/base/transport.h
@@ -26,8 +26,11 @@
 #define WEBRTC_P2P_BASE_TRANSPORT_H_
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/sessiondescription.h"
@@ -214,7 +217,7 @@ class Transport : public sigslot::has_slots<> {
   }
 
   // Get a copy of the remote certificate in use by the specified channel.
-  rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate();
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate();
 
   // Create, destroy, and lookup the channels of this type by their components.
   TransportChannelImpl* CreateChannel(int component);
@@ -312,6 +315,20 @@ class Transport : public sigslot::has_slots<> {
       TransportChannelImpl* channel,
       std::string* error_desc);
 
+  // Returns false if the certificate's identity does not match the fingerprint,
+  // or either is NULL.
+  virtual bool VerifyCertificateFingerprint(
+      const rtc::RTCCertificate* certificate,
+      const rtc::SSLFingerprint* fingerprint,
+      std::string* error_desc) const;
+
+  // Negotiates the SSL role based off the offer and answer as specified by
+  // RFC 4145, section-4.1. Returns false if the SSL role cannot be determined
+  // from the local description and remote description.
+  virtual bool NegotiateRole(ContentAction local_role,
+                             rtc::SSLRole* ssl_role,
+                             std::string* error_desc) const;
+
  private:
   // If a candidate is not acceptable, returns false and sets error.
   // Call this before calling OnRemoteCandidates.
@@ -333,8 +350,8 @@ class Transport : public sigslot::has_slots<> {
   uint64_t tiebreaker_ = 0;
   IceMode remote_ice_mode_ = ICEMODE_FULL;
   IceConfig ice_config_;
-  rtc::scoped_ptr<TransportDescription> local_description_;
-  rtc::scoped_ptr<TransportDescription> remote_description_;
+  std::unique_ptr<TransportDescription> local_description_;
+  std::unique_ptr<TransportDescription> remote_description_;
   bool local_description_set_ = false;
   bool remote_description_set_ = false;
 
diff --git a/chromium/third_party/webrtc/p2p/base/transport_unittest.cc b/chromium/third_party/webrtc/p2p/base/transport_unittest.cc
index 96ebc5eb506..cde60ce964a 100644
--- a/chromium/third_party/webrtc/p2p/base/transport_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/transport_unittest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/network.h"
@@ -50,7 +52,7 @@ class TransportTest : public testing::Test,
   }
 
  protected:
-  rtc::scoped_ptr<FakeTransport> transport_;
+  std::unique_ptr<FakeTransport> transport_;
   FakeTransportChannel* channel_;
 };
 
@@ -219,9 +221,194 @@ TEST_F(TransportTest, TestGetStats) {
   // Note that this tests the behavior of a FakeTransportChannel.
   ASSERT_EQ(1U, stats.channel_stats.size());
   EXPECT_EQ(1, stats.channel_stats[0].component);
+  // Set local transport description for FakeTransport before connecting.
+  TransportDescription faketransport_desc(
+      std::vector<std::string>(),
+      rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH),
+      rtc::CreateRandomString(cricket::ICE_PWD_LENGTH), cricket::ICEMODE_FULL,
+      cricket::CONNECTIONROLE_NONE, nullptr);
+  transport_->SetLocalTransportDescription(faketransport_desc,
+                                           cricket::CA_OFFER, nullptr);
   transport_->ConnectChannels();
   EXPECT_TRUE(transport_->GetStats(&stats));
   ASSERT_EQ(1U, stats.channel_stats.size());
   EXPECT_EQ(1, stats.channel_stats[0].component);
 }
 
+// Tests that VerifyCertificateFingerprint only returns true when the
+// certificate matches the fingerprint.
+TEST_F(TransportTest, TestVerifyCertificateFingerprint) {
+  std::string error_desc;
+  EXPECT_FALSE(
+      transport_->VerifyCertificateFingerprint(nullptr, nullptr, &error_desc));
+  rtc::KeyType key_types[] = {rtc::KT_RSA, rtc::KT_ECDSA};
+
+  for (auto& key_type : key_types) {
+    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
+        rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
+            rtc::SSLIdentity::Generate("testing", key_type)));
+    ASSERT_NE(nullptr, certificate);
+
+    std::string digest_algorithm;
+    ASSERT_TRUE(certificate->ssl_certificate().GetSignatureDigestAlgorithm(
+        &digest_algorithm));
+    ASSERT_FALSE(digest_algorithm.empty());
+    std::unique_ptr<rtc::SSLFingerprint> good_fingerprint(
+        rtc::SSLFingerprint::Create(digest_algorithm, certificate->identity()));
+    ASSERT_NE(nullptr, good_fingerprint);
+
+    EXPECT_TRUE(transport_->VerifyCertificateFingerprint(
+        certificate.get(), good_fingerprint.get(), &error_desc));
+    EXPECT_FALSE(transport_->VerifyCertificateFingerprint(
+        certificate.get(), nullptr, &error_desc));
+    EXPECT_FALSE(transport_->VerifyCertificateFingerprint(
+        nullptr, good_fingerprint.get(), &error_desc));
+
+    rtc::SSLFingerprint bad_fingerprint = *good_fingerprint;
+    bad_fingerprint.digest.AppendData("0", 1);
+    EXPECT_FALSE(transport_->VerifyCertificateFingerprint(
+        certificate.get(), &bad_fingerprint, &error_desc));
+  }
+}
+
+// Tests that NegotiateRole sets the SSL role correctly.
+TEST_F(TransportTest, TestNegotiateRole) {
+  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
+  TransportDescription remote_desc(kIceUfrag2, kIcePwd2);
+
+  struct NegotiateRoleParams {
+    cricket::ConnectionRole local_role;
+    cricket::ConnectionRole remote_role;
+    cricket::ContentAction local_action;
+    cricket::ContentAction remote_action;
+  };
+
+  rtc::SSLRole ssl_role;
+  std::string error_desc;
+
+  // Parameters which set the SSL role to SSL_CLIENT.
+  NegotiateRoleParams valid_client_params[] = {
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_OFFER, cricket::CA_PRANSWER}};
+
+  for (auto& param : valid_client_params) {
+    local_desc.connection_role = param.local_role;
+    remote_desc.connection_role = param.remote_role;
+
+    ASSERT_TRUE(transport_->SetRemoteTransportDescription(
+        remote_desc, param.remote_action, nullptr));
+    ASSERT_TRUE(transport_->SetLocalTransportDescription(
+        local_desc, param.local_action, nullptr));
+    EXPECT_TRUE(
+        transport_->NegotiateRole(param.local_action, &ssl_role, &error_desc));
+    EXPECT_EQ(rtc::SSL_CLIENT, ssl_role);
+  }
+
+  // Parameters which set the SSL role to SSL_SERVER.
+  NegotiateRoleParams valid_server_params[] = {
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_OFFER, cricket::CA_PRANSWER}};
+
+  for (auto& param : valid_server_params) {
+    local_desc.connection_role = param.local_role;
+    remote_desc.connection_role = param.remote_role;
+
+    ASSERT_TRUE(transport_->SetRemoteTransportDescription(
+        remote_desc, param.remote_action, nullptr));
+    ASSERT_TRUE(transport_->SetLocalTransportDescription(
+        local_desc, param.local_action, nullptr));
+    EXPECT_TRUE(
+        transport_->NegotiateRole(param.local_action, &ssl_role, &error_desc));
+    EXPECT_EQ(rtc::SSL_SERVER, ssl_role);
+  }
+
+  // Invalid parameters due to both peers having a duplicate role.
+  NegotiateRoleParams duplicate_params[] = {
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_OFFER, cricket::CA_PRANSWER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_OFFER, cricket::CA_PRANSWER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_OFFER, cricket::CA_PRANSWER}};
+
+  for (auto& param : duplicate_params) {
+    local_desc.connection_role = param.local_role;
+    remote_desc.connection_role = param.remote_role;
+
+    ASSERT_TRUE(transport_->SetRemoteTransportDescription(
+        remote_desc, param.remote_action, nullptr));
+    ASSERT_TRUE(transport_->SetLocalTransportDescription(
+        local_desc, param.local_action, nullptr));
+    EXPECT_FALSE(
+        transport_->NegotiateRole(param.local_action, &ssl_role, &error_desc));
+  }
+
+  // Invalid parameters due to the offerer not using ACTPASS.
+  NegotiateRoleParams offerer_without_actpass_params[] = {
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_ANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_PRANSWER, cricket::CA_OFFER},
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_OFFER, cricket::CA_ANSWER},
+      {cricket::CONNECTIONROLE_ACTIVE, cricket::CONNECTIONROLE_PASSIVE,
+       cricket::CA_OFFER, cricket::CA_PRANSWER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTIVE,
+       cricket::CA_OFFER, cricket::CA_PRANSWER},
+      {cricket::CONNECTIONROLE_PASSIVE, cricket::CONNECTIONROLE_ACTPASS,
+       cricket::CA_OFFER, cricket::CA_PRANSWER}};
+
+  for (auto& param : offerer_without_actpass_params) {
+    local_desc.connection_role = param.local_role;
+    remote_desc.connection_role = param.remote_role;
+
+    ASSERT_TRUE(transport_->SetRemoteTransportDescription(
+        remote_desc, param.remote_action, nullptr));
+    ASSERT_TRUE(transport_->SetLocalTransportDescription(
+        local_desc, param.local_action, nullptr));
+    EXPECT_FALSE(
+        transport_->NegotiateRole(param.local_action, &ssl_role, &error_desc));
+  }
+}
diff --git a/chromium/third_party/webrtc/p2p/base/transportchannel.h b/chromium/third_party/webrtc/p2p/base/transportchannel.h
index 24f90e3cf6f..87ed9fdea31 100644
--- a/chromium/third_party/webrtc/p2p/base/transportchannel.h
+++ b/chromium/third_party/webrtc/p2p/base/transportchannel.h
@@ -11,9 +11,11 @@
 #ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
 #define WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/candidatepairinterface.h"
 #include "webrtc/p2p/base/transport.h"
@@ -130,7 +132,7 @@ class TransportChannel : public sigslot::has_slots<> {
   GetLocalCertificate() const = 0;
 
   // Gets a copy of the remote side's SSL certificate.
-  virtual rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
+  virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
       const = 0;
 
   // Allows key material to be extracted for external encryption.
diff --git a/chromium/third_party/webrtc/p2p/base/transportchannelimpl.h b/chromium/third_party/webrtc/p2p/base/transportchannelimpl.h
index 904ebf563fd..f548aa012f8 100644
--- a/chromium/third_party/webrtc/p2p/base/transportchannelimpl.h
+++ b/chromium/third_party/webrtc/p2p/base/transportchannelimpl.h
@@ -12,6 +12,8 @@
 #define WEBRTC_P2P_BASE_TRANSPORTCHANNELIMPL_H_
 
 #include <string>
+
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/p2p/base/transportchannel.h"
 
 namespace buzz { class XmlElement; }
diff --git a/chromium/third_party/webrtc/p2p/base/transportcontroller.cc b/chromium/third_party/webrtc/p2p/base/transportcontroller.cc
index 708c60f9df5..a961541dc3e 100644
--- a/chromium/third_party/webrtc/p2p/base/transportcontroller.cc
+++ b/chromium/third_party/webrtc/p2p/base/transportcontroller.cc
@@ -11,6 +11,7 @@
 #include "webrtc/p2p/base/transportcontroller.h"
 
 #include <algorithm>
+#include <memory>
 
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
@@ -19,6 +20,10 @@
 #include "webrtc/p2p/base/p2ptransport.h"
 #include "webrtc/p2p/base/port.h"
 
+#ifdef HAVE_QUIC
+#include "webrtc/p2p/quic/quictransport.h"
+#endif  // HAVE_QUIC
+
 namespace cricket {
 
 enum {
@@ -38,59 +43,60 @@ struct CandidatesData : public rtc::MessageData {
 };
 
 TransportController::TransportController(rtc::Thread* signaling_thread,
-                                         rtc::Thread* worker_thread,
+                                         rtc::Thread* network_thread,
                                          PortAllocator* port_allocator)
     : signaling_thread_(signaling_thread),
-      worker_thread_(worker_thread),
+      network_thread_(network_thread),
       port_allocator_(port_allocator) {}
 
 TransportController::~TransportController() {
-  worker_thread_->Invoke<void>(
-      rtc::Bind(&TransportController::DestroyAllTransports_w, this));
+  network_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::DestroyAllTransports_n, this));
   signaling_thread_->Clear(this);
 }
 
 bool TransportController::SetSslMaxProtocolVersion(
     rtc::SSLProtocolVersion version) {
-  return worker_thread_->Invoke<bool>(rtc::Bind(
-      &TransportController::SetSslMaxProtocolVersion_w, this, version));
+  return network_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::SetSslMaxProtocolVersion_n, this, version));
 }
 
 void TransportController::SetIceConfig(const IceConfig& config) {
-  worker_thread_->Invoke<void>(
-      rtc::Bind(&TransportController::SetIceConfig_w, this, config));
+  network_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::SetIceConfig_n, this, config));
 }
 
 void TransportController::SetIceRole(IceRole ice_role) {
-  worker_thread_->Invoke<void>(
-      rtc::Bind(&TransportController::SetIceRole_w, this, ice_role));
+  network_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::SetIceRole_n, this, ice_role));
 }
 
 bool TransportController::GetSslRole(const std::string& transport_name,
                                      rtc::SSLRole* role) {
-  return worker_thread_->Invoke<bool>(rtc::Bind(
-      &TransportController::GetSslRole_w, this, transport_name, role));
+  return network_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::GetSslRole_n, this, transport_name, role));
 }
 
 bool TransportController::SetLocalCertificate(
     const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
-  return worker_thread_->Invoke<bool>(rtc::Bind(
-      &TransportController::SetLocalCertificate_w, this, certificate));
+  return network_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::SetLocalCertificate_n, this, certificate));
 }
 
 bool TransportController::GetLocalCertificate(
     const std::string& transport_name,
     rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
-  return worker_thread_->Invoke<bool>(
-      rtc::Bind(&TransportController::GetLocalCertificate_w, this,
+  return network_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::GetLocalCertificate_n, this,
                 transport_name, certificate));
 }
 
-rtc::scoped_ptr<rtc::SSLCertificate>
+std::unique_ptr<rtc::SSLCertificate>
 TransportController::GetRemoteSSLCertificate(
     const std::string& transport_name) {
-  return worker_thread_->Invoke<rtc::scoped_ptr<rtc::SSLCertificate>>(rtc::Bind(
-      &TransportController::GetRemoteSSLCertificate_w, this, transport_name));
+  return network_thread_->Invoke<std::unique_ptr<rtc::SSLCertificate>>(
+      rtc::Bind(&TransportController::GetRemoteSSLCertificate_n, this,
+                transport_name));
 }
 
 bool TransportController::SetLocalTransportDescription(
@@ -98,8 +104,8 @@ bool TransportController::SetLocalTransportDescription(
     const TransportDescription& tdesc,
     ContentAction action,
     std::string* err) {
-  return worker_thread_->Invoke<bool>(
-      rtc::Bind(&TransportController::SetLocalTransportDescription_w, this,
+  return network_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::SetLocalTransportDescription_n, this,
                 transport_name, tdesc, action, err));
 }
 
@@ -108,48 +114,48 @@ bool TransportController::SetRemoteTransportDescription(
     const TransportDescription& tdesc,
     ContentAction action,
     std::string* err) {
-  return worker_thread_->Invoke<bool>(
-      rtc::Bind(&TransportController::SetRemoteTransportDescription_w, this,
+  return network_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::SetRemoteTransportDescription_n, this,
                 transport_name, tdesc, action, err));
 }
 
 void TransportController::MaybeStartGathering() {
-  worker_thread_->Invoke<void>(
-      rtc::Bind(&TransportController::MaybeStartGathering_w, this));
+  network_thread_->Invoke<void>(
+      rtc::Bind(&TransportController::MaybeStartGathering_n, this));
 }
 
 bool TransportController::AddRemoteCandidates(const std::string& transport_name,
                                               const Candidates& candidates,
                                               std::string* err) {
-  return worker_thread_->Invoke<bool>(
-      rtc::Bind(&TransportController::AddRemoteCandidates_w, this,
+  return network_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::AddRemoteCandidates_n, this,
                 transport_name, candidates, err));
 }
 
 bool TransportController::RemoveRemoteCandidates(const Candidates& candidates,
                                                  std::string* err) {
-  return worker_thread_->Invoke<bool>(rtc::Bind(
-      &TransportController::RemoveRemoteCandidates_w, this, candidates, err));
+  return network_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::RemoveRemoteCandidates_n, this, candidates, err));
 }
 
 bool TransportController::ReadyForRemoteCandidates(
     const std::string& transport_name) {
-  return worker_thread_->Invoke<bool>(rtc::Bind(
-      &TransportController::ReadyForRemoteCandidates_w, this, transport_name));
+  return network_thread_->Invoke<bool>(rtc::Bind(
+      &TransportController::ReadyForRemoteCandidates_n, this, transport_name));
 }
 
 bool TransportController::GetStats(const std::string& transport_name,
                                    TransportStats* stats) {
-  return worker_thread_->Invoke<bool>(
-      rtc::Bind(&TransportController::GetStats_w, this, transport_name, stats));
+  return network_thread_->Invoke<bool>(
+      rtc::Bind(&TransportController::GetStats_n, this, transport_name, stats));
 }
 
-TransportChannel* TransportController::CreateTransportChannel_w(
+TransportChannel* TransportController::CreateTransportChannel_n(
     const std::string& transport_name,
     int component) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  auto it = FindChannel_w(transport_name, component);
+  auto it = FindChannel_n(transport_name, component);
   if (it != channels_.end()) {
     // Channel already exists; increment reference count and return.
     it->AddRef();
@@ -157,34 +163,34 @@ TransportChannel* TransportController::CreateTransportChannel_w(
   }
 
   // Need to create a new channel.
-  Transport* transport = GetOrCreateTransport_w(transport_name);
+  Transport* transport = GetOrCreateTransport_n(transport_name);
   TransportChannelImpl* channel = transport->CreateChannel(component);
   channel->SignalWritableState.connect(
-      this, &TransportController::OnChannelWritableState_w);
+      this, &TransportController::OnChannelWritableState_n);
   channel->SignalReceivingState.connect(
-      this, &TransportController::OnChannelReceivingState_w);
+      this, &TransportController::OnChannelReceivingState_n);
   channel->SignalGatheringState.connect(
-      this, &TransportController::OnChannelGatheringState_w);
+      this, &TransportController::OnChannelGatheringState_n);
   channel->SignalCandidateGathered.connect(
-      this, &TransportController::OnChannelCandidateGathered_w);
+      this, &TransportController::OnChannelCandidateGathered_n);
   channel->SignalCandidatesRemoved.connect(
-      this, &TransportController::OnChannelCandidatesRemoved_w);
+      this, &TransportController::OnChannelCandidatesRemoved_n);
   channel->SignalRoleConflict.connect(
-      this, &TransportController::OnChannelRoleConflict_w);
+      this, &TransportController::OnChannelRoleConflict_n);
   channel->SignalConnectionRemoved.connect(
-      this, &TransportController::OnChannelConnectionRemoved_w);
+      this, &TransportController::OnChannelConnectionRemoved_n);
   channels_.insert(channels_.end(), RefCountedChannel(channel))->AddRef();
   // Adding a channel could cause aggregate state to change.
-  UpdateAggregateStates_w();
+  UpdateAggregateStates_n();
   return channel;
 }
 
-void TransportController::DestroyTransportChannel_w(
+void TransportController::DestroyTransportChannel_n(
     const std::string& transport_name,
     int component) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  auto it = FindChannel_w(transport_name, component);
+  auto it = FindChannel_n(transport_name, component);
   if (it == channels_.end()) {
     LOG(LS_WARNING) << "Attempting to delete " << transport_name
                     << " TransportChannel " << component
@@ -198,15 +204,15 @@ void TransportController::DestroyTransportChannel_w(
   }
 
   channels_.erase(it);
-  Transport* transport = GetTransport_w(transport_name);
+  Transport* transport = GetTransport_n(transport_name);
   transport->DestroyChannel(component);
   // Just as we create a Transport when its first channel is created,
   // we delete it when its last channel is deleted.
   if (!transport->HasChannels()) {
-    DestroyTransport_w(transport_name);
+    DestroyTransport_n(transport_name);
   }
   // Removing a channel could cause aggregate state to change.
-  UpdateAggregateStates_w();
+  UpdateAggregateStates_n();
 }
 
 const rtc::scoped_refptr<rtc::RTCCertificate>&
@@ -214,18 +220,23 @@ TransportController::certificate_for_testing() {
   return certificate_;
 }
 
-Transport* TransportController::CreateTransport_w(
+Transport* TransportController::CreateTransport_n(
     const std::string& transport_name) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
+#ifdef HAVE_QUIC
+  if (quic_) {
+    return new QuicTransport(transport_name, port_allocator(), certificate_);
+  }
+#endif  // HAVE_QUIC
   Transport* transport = new DtlsTransport<P2PTransport>(
       transport_name, port_allocator(), certificate_);
   return transport;
 }
 
-Transport* TransportController::GetTransport_w(
+Transport* TransportController::GetTransport_n(
     const std::string& transport_name) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   auto iter = transports_.find(transport_name);
   return (iter != transports_.end()) ? iter->second : nullptr;
@@ -268,7 +279,7 @@ void TransportController::OnMessage(rtc::Message* pmsg) {
 }
 
 std::vector<TransportController::RefCountedChannel>::iterator
-TransportController::FindChannel_w(const std::string& transport_name,
+TransportController::FindChannel_n(const std::string& transport_name,
                                    int component) {
   return std::find_if(
       channels_.begin(), channels_.end(),
@@ -278,16 +289,16 @@ TransportController::FindChannel_w(const std::string& transport_name,
       });
 }
 
-Transport* TransportController::GetOrCreateTransport_w(
+Transport* TransportController::GetOrCreateTransport_n(
     const std::string& transport_name) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* transport = GetTransport_w(transport_name);
+  Transport* transport = GetTransport_n(transport_name);
   if (transport) {
     return transport;
   }
 
-  transport = CreateTransport_w(transport_name);
+  transport = CreateTransport_n(transport_name);
   // The stuff below happens outside of CreateTransport_w so that unit tests
   // can override CreateTransport_w to return a different type of transport.
   transport->SetSslMaxProtocolVersion(ssl_max_version_);
@@ -302,9 +313,9 @@ Transport* TransportController::GetOrCreateTransport_w(
   return transport;
 }
 
-void TransportController::DestroyTransport_w(
+void TransportController::DestroyTransport_n(
     const std::string& transport_name) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   auto iter = transports_.find(transport_name);
   if (iter != transports_.end()) {
@@ -313,8 +324,8 @@ void TransportController::DestroyTransport_w(
   }
 }
 
-void TransportController::DestroyAllTransports_w() {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+void TransportController::DestroyAllTransports_n() {
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   for (const auto& kv : transports_) {
     delete kv.second;
@@ -322,9 +333,9 @@ void TransportController::DestroyAllTransports_w() {
   transports_.clear();
 }
 
-bool TransportController::SetSslMaxProtocolVersion_w(
+bool TransportController::SetSslMaxProtocolVersion_n(
     rtc::SSLProtocolVersion version) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   // Max SSL version can only be set before transports are created.
   if (!transports_.empty()) {
@@ -335,27 +346,27 @@ bool TransportController::SetSslMaxProtocolVersion_w(
   return true;
 }
 
-void TransportController::SetIceConfig_w(const IceConfig& config) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+void TransportController::SetIceConfig_n(const IceConfig& config) {
+  RTC_DCHECK(network_thread_->IsCurrent());
   ice_config_ = config;
   for (const auto& kv : transports_) {
     kv.second->SetIceConfig(ice_config_);
   }
 }
 
-void TransportController::SetIceRole_w(IceRole ice_role) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+void TransportController::SetIceRole_n(IceRole ice_role) {
+  RTC_DCHECK(network_thread_->IsCurrent());
   ice_role_ = ice_role;
   for (const auto& kv : transports_) {
     kv.second->SetIceRole(ice_role_);
   }
 }
 
-bool TransportController::GetSslRole_w(const std::string& transport_name,
+bool TransportController::GetSslRole_n(const std::string& transport_name,
                                        rtc::SSLRole* role) {
-  RTC_DCHECK(worker_thread()->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* t = GetTransport_w(transport_name);
+  Transport* t = GetTransport_n(transport_name);
   if (!t) {
     return false;
   }
@@ -363,9 +374,9 @@ bool TransportController::GetSslRole_w(const std::string& transport_name,
   return t->GetSslRole(role);
 }
 
-bool TransportController::SetLocalCertificate_w(
+bool TransportController::SetLocalCertificate_n(
     const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   if (certificate_) {
     return false;
@@ -381,12 +392,12 @@ bool TransportController::SetLocalCertificate_w(
   return true;
 }
 
-bool TransportController::GetLocalCertificate_w(
+bool TransportController::GetLocalCertificate_n(
     const std::string& transport_name,
     rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* t = GetTransport_w(transport_name);
+  Transport* t = GetTransport_n(transport_name);
   if (!t) {
     return false;
   }
@@ -394,12 +405,12 @@ bool TransportController::GetLocalCertificate_w(
   return t->GetLocalCertificate(certificate);
 }
 
-rtc::scoped_ptr<rtc::SSLCertificate>
-TransportController::GetRemoteSSLCertificate_w(
+std::unique_ptr<rtc::SSLCertificate>
+TransportController::GetRemoteSSLCertificate_n(
     const std::string& transport_name) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* t = GetTransport_w(transport_name);
+  Transport* t = GetTransport_n(transport_name);
   if (!t) {
     return nullptr;
   }
@@ -407,14 +418,14 @@ TransportController::GetRemoteSSLCertificate_w(
   return t->GetRemoteSSLCertificate();
 }
 
-bool TransportController::SetLocalTransportDescription_w(
+bool TransportController::SetLocalTransportDescription_n(
     const std::string& transport_name,
     const TransportDescription& tdesc,
     ContentAction action,
     std::string* err) {
-  RTC_DCHECK(worker_thread()->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* transport = GetTransport_w(transport_name);
+  Transport* transport = GetTransport_n(transport_name);
   if (!transport) {
     // If we didn't find a transport, that's not an error;
     // it could have been deleted as a result of bundling.
@@ -426,14 +437,14 @@ bool TransportController::SetLocalTransportDescription_w(
   return transport->SetLocalTransportDescription(tdesc, action, err);
 }
 
-bool TransportController::SetRemoteTransportDescription_w(
+bool TransportController::SetRemoteTransportDescription_n(
     const std::string& transport_name,
     const TransportDescription& tdesc,
     ContentAction action,
     std::string* err) {
-  RTC_DCHECK(worker_thread()->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* transport = GetTransport_w(transport_name);
+  Transport* transport = GetTransport_n(transport_name);
   if (!transport) {
     // If we didn't find a transport, that's not an error;
     // it could have been deleted as a result of bundling.
@@ -445,19 +456,19 @@ bool TransportController::SetRemoteTransportDescription_w(
   return transport->SetRemoteTransportDescription(tdesc, action, err);
 }
 
-void TransportController::MaybeStartGathering_w() {
+void TransportController::MaybeStartGathering_n() {
   for (const auto& kv : transports_) {
     kv.second->MaybeStartGathering();
   }
 }
 
-bool TransportController::AddRemoteCandidates_w(
+bool TransportController::AddRemoteCandidates_n(
     const std::string& transport_name,
     const Candidates& candidates,
     std::string* err) {
-  RTC_DCHECK(worker_thread()->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* transport = GetTransport_w(transport_name);
+  Transport* transport = GetTransport_n(transport_name);
   if (!transport) {
     // If we didn't find a transport, that's not an error;
     // it could have been deleted as a result of bundling.
@@ -467,9 +478,9 @@ bool TransportController::AddRemoteCandidates_w(
   return transport->AddRemoteCandidates(candidates, err);
 }
 
-bool TransportController::RemoveRemoteCandidates_w(const Candidates& candidates,
+bool TransportController::RemoveRemoteCandidates_n(const Candidates& candidates,
                                                    std::string* err) {
-  RTC_DCHECK(worker_thread()->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
   std::map<std::string, Candidates> candidates_by_transport_name;
   for (const Candidate& cand : candidates) {
     RTC_DCHECK(!cand.transport_name().empty());
@@ -478,7 +489,7 @@ bool TransportController::RemoveRemoteCandidates_w(const Candidates& candidates,
 
   bool result = true;
   for (auto kv : candidates_by_transport_name) {
-    Transport* transport = GetTransport_w(kv.first);
+    Transport* transport = GetTransport_n(kv.first);
     if (!transport) {
       // If we didn't find a transport, that's not an error;
       // it could have been deleted as a result of bundling.
@@ -489,51 +500,51 @@ bool TransportController::RemoveRemoteCandidates_w(const Candidates& candidates,
   return result;
 }
 
-bool TransportController::ReadyForRemoteCandidates_w(
+bool TransportController::ReadyForRemoteCandidates_n(
     const std::string& transport_name) {
-  RTC_DCHECK(worker_thread()->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* transport = GetTransport_w(transport_name);
+  Transport* transport = GetTransport_n(transport_name);
   if (!transport) {
     return false;
   }
   return transport->ready_for_remote_candidates();
 }
 
-bool TransportController::GetStats_w(const std::string& transport_name,
+bool TransportController::GetStats_n(const std::string& transport_name,
                                      TransportStats* stats) {
-  RTC_DCHECK(worker_thread()->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
-  Transport* transport = GetTransport_w(transport_name);
+  Transport* transport = GetTransport_n(transport_name);
   if (!transport) {
     return false;
   }
   return transport->GetStats(stats);
 }
 
-void TransportController::OnChannelWritableState_w(TransportChannel* channel) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+void TransportController::OnChannelWritableState_n(TransportChannel* channel) {
+  RTC_DCHECK(network_thread_->IsCurrent());
   LOG(LS_INFO) << channel->transport_name() << " TransportChannel "
                << channel->component() << " writability changed to "
                << channel->writable() << ".";
-  UpdateAggregateStates_w();
+  UpdateAggregateStates_n();
 }
 
-void TransportController::OnChannelReceivingState_w(TransportChannel* channel) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
-  UpdateAggregateStates_w();
+void TransportController::OnChannelReceivingState_n(TransportChannel* channel) {
+  RTC_DCHECK(network_thread_->IsCurrent());
+  UpdateAggregateStates_n();
 }
 
-void TransportController::OnChannelGatheringState_w(
+void TransportController::OnChannelGatheringState_n(
     TransportChannelImpl* channel) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
-  UpdateAggregateStates_w();
+  RTC_DCHECK(network_thread_->IsCurrent());
+  UpdateAggregateStates_n();
 }
 
-void TransportController::OnChannelCandidateGathered_w(
+void TransportController::OnChannelCandidateGathered_n(
     TransportChannelImpl* channel,
     const Candidate& candidate) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   // We should never signal peer-reflexive candidates.
   if (candidate.type() == PRFLX_PORT_TYPE) {
@@ -547,7 +558,7 @@ void TransportController::OnChannelCandidateGathered_w(
   signaling_thread_->Post(this, MSG_CANDIDATESGATHERED, data);
 }
 
-void TransportController::OnChannelCandidatesRemoved_w(
+void TransportController::OnChannelCandidatesRemoved_n(
     TransportChannelImpl* channel,
     const Candidates& candidates) {
   invoker_.AsyncInvoke<void>(
@@ -562,9 +573,9 @@ void TransportController::OnChannelCandidatesRemoved(
   SignalCandidatesRemoved(candidates);
 }
 
-void TransportController::OnChannelRoleConflict_w(
+void TransportController::OnChannelRoleConflict_n(
     TransportChannelImpl* channel) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   if (ice_role_switch_) {
     LOG(LS_WARNING)
@@ -581,17 +592,17 @@ void TransportController::OnChannelRoleConflict_w(
   }
 }
 
-void TransportController::OnChannelConnectionRemoved_w(
+void TransportController::OnChannelConnectionRemoved_n(
     TransportChannelImpl* channel) {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+  RTC_DCHECK(network_thread_->IsCurrent());
   LOG(LS_INFO) << channel->transport_name() << " TransportChannel "
                << channel->component()
                << " connection removed. Check if state is complete.";
-  UpdateAggregateStates_w();
+  UpdateAggregateStates_n();
 }
 
-void TransportController::UpdateAggregateStates_w() {
-  RTC_DCHECK(worker_thread_->IsCurrent());
+void TransportController::UpdateAggregateStates_n() {
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   IceConnectionState new_connection_state = kIceConnectionConnecting;
   IceGatheringState new_gathering_state = kIceGatheringNew;
diff --git a/chromium/third_party/webrtc/p2p/base/transportcontroller.h b/chromium/third_party/webrtc/p2p/base/transportcontroller.h
index 9d06823920f..cc16a768ab9 100644
--- a/chromium/third_party/webrtc/p2p/base/transportcontroller.h
+++ b/chromium/third_party/webrtc/p2p/base/transportcontroller.h
@@ -12,6 +12,7 @@
 #define WEBRTC_P2P_BASE_TRANSPORTCONTROLLER_H_
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -31,13 +32,13 @@ class TransportController : public sigslot::has_slots<>,
                             public rtc::MessageHandler {
  public:
   TransportController(rtc::Thread* signaling_thread,
-                      rtc::Thread* worker_thread,
+                      rtc::Thread* network_thread,
                       PortAllocator* port_allocator);
 
   virtual ~TransportController();
 
   rtc::Thread* signaling_thread() const { return signaling_thread_; }
-  rtc::Thread* worker_thread() const { return worker_thread_; }
+  rtc::Thread* network_thread() const { return network_thread_; }
 
   PortAllocator* port_allocator() const { return port_allocator_; }
 
@@ -59,7 +60,7 @@ class TransportController : public sigslot::has_slots<>,
       const std::string& transport_name,
       rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
   // Caller owns returned certificate
-  rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
       const std::string& transport_name);
   bool SetLocalTransportDescription(const std::string& transport_name,
                                     const TransportDescription& tdesc,
@@ -81,15 +82,18 @@ class TransportController : public sigslot::has_slots<>,
 
   // Creates a channel if it doesn't exist. Otherwise, increments a reference
   // count and returns an existing channel.
-  virtual TransportChannel* CreateTransportChannel_w(
+  virtual TransportChannel* CreateTransportChannel_n(
       const std::string& transport_name,
       int component);
 
   // Decrements a channel's reference count, and destroys the channel if
   // nothing is referencing it.
-  virtual void DestroyTransportChannel_w(const std::string& transport_name,
+  virtual void DestroyTransportChannel_n(const std::string& transport_name,
                                          int component);
 
+  void use_quic() { quic_ = true; }
+  bool quic() const { return quic_; }
+
   // All of these signals are fired on the signalling thread.
 
   // If any transport failed => failed,
@@ -117,11 +121,11 @@ class TransportController : public sigslot::has_slots<>,
 
  protected:
   // Protected and virtual so we can override it in unit tests.
-  virtual Transport* CreateTransport_w(const std::string& transport_name);
+  virtual Transport* CreateTransport_n(const std::string& transport_name);
 
   // For unit tests
   const std::map<std::string, Transport*>& transports() { return transports_; }
-  Transport* GetTransport_w(const std::string& transport_name);
+  Transport* GetTransport_n(const std::string& transport_name);
 
  private:
   void OnMessage(rtc::Message* pmsg) override;
@@ -149,57 +153,57 @@ class TransportController : public sigslot::has_slots<>,
     int ref_;
   };
 
-  std::vector<RefCountedChannel>::iterator FindChannel_w(
+  std::vector<RefCountedChannel>::iterator FindChannel_n(
       const std::string& transport_name,
       int component);
 
-  Transport* GetOrCreateTransport_w(const std::string& transport_name);
-  void DestroyTransport_w(const std::string& transport_name);
-  void DestroyAllTransports_w();
+  Transport* GetOrCreateTransport_n(const std::string& transport_name);
+  void DestroyTransport_n(const std::string& transport_name);
+  void DestroyAllTransports_n();
 
-  bool SetSslMaxProtocolVersion_w(rtc::SSLProtocolVersion version);
-  void SetIceConfig_w(const IceConfig& config);
-  void SetIceRole_w(IceRole ice_role);
-  bool GetSslRole_w(const std::string& transport_name, rtc::SSLRole* role);
-  bool SetLocalCertificate_w(
+  bool SetSslMaxProtocolVersion_n(rtc::SSLProtocolVersion version);
+  void SetIceConfig_n(const IceConfig& config);
+  void SetIceRole_n(IceRole ice_role);
+  bool GetSslRole_n(const std::string& transport_name, rtc::SSLRole* role);
+  bool SetLocalCertificate_n(
       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
-  bool GetLocalCertificate_w(
+  bool GetLocalCertificate_n(
       const std::string& transport_name,
       rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
-  rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate_w(
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate_n(
       const std::string& transport_name);
-  bool SetLocalTransportDescription_w(const std::string& transport_name,
+  bool SetLocalTransportDescription_n(const std::string& transport_name,
                                       const TransportDescription& tdesc,
                                       ContentAction action,
                                       std::string* err);
-  bool SetRemoteTransportDescription_w(const std::string& transport_name,
+  bool SetRemoteTransportDescription_n(const std::string& transport_name,
                                        const TransportDescription& tdesc,
                                        ContentAction action,
                                        std::string* err);
-  void MaybeStartGathering_w();
-  bool AddRemoteCandidates_w(const std::string& transport_name,
+  void MaybeStartGathering_n();
+  bool AddRemoteCandidates_n(const std::string& transport_name,
                              const Candidates& candidates,
                              std::string* err);
-  bool RemoveRemoteCandidates_w(const Candidates& candidates, std::string* err);
-  bool ReadyForRemoteCandidates_w(const std::string& transport_name);
-  bool GetStats_w(const std::string& transport_name, TransportStats* stats);
+  bool RemoveRemoteCandidates_n(const Candidates& candidates, std::string* err);
+  bool ReadyForRemoteCandidates_n(const std::string& transport_name);
+  bool GetStats_n(const std::string& transport_name, TransportStats* stats);
 
   // Handlers for signals from Transport.
-  void OnChannelWritableState_w(TransportChannel* channel);
-  void OnChannelReceivingState_w(TransportChannel* channel);
-  void OnChannelGatheringState_w(TransportChannelImpl* channel);
-  void OnChannelCandidateGathered_w(TransportChannelImpl* channel,
+  void OnChannelWritableState_n(TransportChannel* channel);
+  void OnChannelReceivingState_n(TransportChannel* channel);
+  void OnChannelGatheringState_n(TransportChannelImpl* channel);
+  void OnChannelCandidateGathered_n(TransportChannelImpl* channel,
                                     const Candidate& candidate);
   void OnChannelCandidatesRemoved(const Candidates& candidates);
-  void OnChannelCandidatesRemoved_w(TransportChannelImpl* channel,
+  void OnChannelCandidatesRemoved_n(TransportChannelImpl* channel,
                                     const Candidates& candidates);
-  void OnChannelRoleConflict_w(TransportChannelImpl* channel);
-  void OnChannelConnectionRemoved_w(TransportChannelImpl* channel);
+  void OnChannelRoleConflict_n(TransportChannelImpl* channel);
+  void OnChannelConnectionRemoved_n(TransportChannelImpl* channel);
 
-  void UpdateAggregateStates_w();
+  void UpdateAggregateStates_n();
 
   rtc::Thread* const signaling_thread_ = nullptr;
-  rtc::Thread* const worker_thread_ = nullptr;
+  rtc::Thread* const network_thread_ = nullptr;
   typedef std::map<std::string, Transport*> TransportMap;
   TransportMap transports_;
 
@@ -221,6 +225,8 @@ class TransportController : public sigslot::has_slots<>,
   uint64_t ice_tiebreaker_ = rtc::CreateRandomId64();
   rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
   rtc::AsyncInvoker invoker_;
+  // True if QUIC is used instead of DTLS.
+  bool quic_ = false;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/transportcontroller_unittest.cc b/chromium/third_party/webrtc/p2p/base/transportcontroller_unittest.cc
index c90fd700ce3..627975c90ca 100644
--- a/chromium/third_party/webrtc/p2p/base/transportcontroller_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/transportcontroller_unittest.cc
@@ -9,19 +9,19 @@
  */
 
 #include <map>
+#include <memory>
 
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/p2p/base/dtlstransportchannel.h"
+#include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/p2p/base/p2ptransportchannel.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/transportcontroller.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
 
 static const int kTimeout = 100;
 static const char kIceUfrag1[] = "TESTICEUFRAG0001";
@@ -81,12 +81,12 @@ class TransportControllerTest : public testing::Test,
   FakeTransportChannel* CreateChannel(const std::string& content,
                                       int component) {
     TransportChannel* channel =
-        transport_controller_->CreateTransportChannel_w(content, component);
+        transport_controller_->CreateTransportChannel_n(content, component);
     return static_cast<FakeTransportChannel*>(channel);
   }
 
   void DestroyChannel(const std::string& content, int component) {
-    transport_controller_->DestroyTransportChannel_w(content, component);
+    transport_controller_->DestroyTransportChannel_n(content, component);
   }
 
   Candidate CreateCandidate(int component) {
@@ -177,8 +177,8 @@ class TransportControllerTest : public testing::Test,
     ++candidates_signal_count_;
   }
 
-  rtc::scoped_ptr<rtc::Thread> worker_thread_;  // Not used for most tests.
-  rtc::scoped_ptr<TransportControllerForTest> transport_controller_;
+  std::unique_ptr<rtc::Thread> worker_thread_;  // Not used for most tests.
+  std::unique_ptr<TransportControllerForTest> transport_controller_;
 
   // Information received from signals from transport controller.
   IceConnectionState connection_state_ = cricket::kIceConnectionConnecting;
@@ -269,10 +269,10 @@ TEST_F(TransportControllerTest, TestGetSslRole) {
 
 TEST_F(TransportControllerTest, TestSetAndGetLocalCertificate) {
   rtc::scoped_refptr<rtc::RTCCertificate> certificate1 =
-      rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
           rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
   rtc::scoped_refptr<rtc::RTCCertificate> certificate2 =
-      rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
           rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT)));
   rtc::scoped_refptr<rtc::RTCCertificate> returned_certificate;
 
@@ -308,7 +308,7 @@ TEST_F(TransportControllerTest, TestGetRemoteSSLCertificate) {
   ASSERT_NE(nullptr, channel);
 
   channel->SetRemoteSSLCertificate(&fake_certificate);
-  rtc::scoped_ptr<rtc::SSLCertificate> returned_certificate =
+  std::unique_ptr<rtc::SSLCertificate> returned_certificate =
       transport_controller_->GetRemoteSSLCertificate("audio");
   EXPECT_TRUE(returned_certificate);
   EXPECT_EQ(fake_certificate.ToPEMString(),
diff --git a/chromium/third_party/webrtc/p2p/base/transportdescription.h b/chromium/third_party/webrtc/p2p/base/transportdescription.h
index 003780b7679..42e45a670f9 100644
--- a/chromium/third_party/webrtc/p2p/base/transportdescription.h
+++ b/chromium/third_party/webrtc/p2p/base/transportdescription.h
@@ -12,11 +12,11 @@
 #define WEBRTC_P2P_BASE_TRANSPORTDESCRIPTION_H_
 
 #include <algorithm>
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/p2p/base/p2pconstants.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslfingerprint.h"
 
 namespace cricket {
@@ -139,7 +139,7 @@ struct TransportDescription {
   IceMode ice_mode;
   ConnectionRole connection_role;
 
-  rtc::scoped_ptr<rtc::SSLFingerprint> identity_fingerprint;
+  std::unique_ptr<rtc::SSLFingerprint> identity_fingerprint;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory.cc b/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory.cc
index 1ddf55d4a13..e57b7e3efae 100644
--- a/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory.cc
+++ b/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/p2p/base/transportdescriptionfactory.h"
 
+#include <memory>
+
 #include "webrtc/p2p/base/transportdescription.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
@@ -25,7 +27,7 @@ TransportDescriptionFactory::TransportDescriptionFactory()
 TransportDescription* TransportDescriptionFactory::CreateOffer(
     const TransportOptions& options,
     const TransportDescription* current_description) const {
-  rtc::scoped_ptr<TransportDescription> desc(new TransportDescription());
+  std::unique_ptr<TransportDescription> desc(new TransportDescription());
 
   // Generate the ICE credentials if we don't already have them.
   if (!current_description || options.ice_restart) {
@@ -59,7 +61,7 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
     return NULL;
   }
 
-  rtc::scoped_ptr<TransportDescription> desc(new TransportDescription());
+  std::unique_ptr<TransportDescription> desc(new TransportDescription());
   // Generate the ICE credentials if we don't already have them or ice is
   // being restarted.
   if (!current_description || options.ice_restart) {
diff --git a/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory_unittest.cc b/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
index 765c607a6d6..38675ba401e 100644
--- a/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -18,7 +19,6 @@
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/ssladapter.h"
 
-using rtc::scoped_ptr;
 using cricket::TransportDescriptionFactory;
 using cricket::TransportDescription;
 using cricket::TransportOptions;
@@ -26,10 +26,10 @@ using cricket::TransportOptions;
 class TransportDescriptionFactoryTest : public testing::Test {
  public:
   TransportDescriptionFactoryTest()
-      : cert1_(rtc::RTCCertificate::Create(
-            scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("User1")))),
-        cert2_(rtc::RTCCertificate::Create(
-            scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("User2")))) {}
+      : cert1_(rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
+            new rtc::FakeSSLIdentity("User1")))),
+        cert2_(rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
+            new rtc::FakeSSLIdentity("User2")))) {}
 
   void CheckDesc(const TransportDescription* desc,
                  const std::string& opt, const std::string& ice_ufrag,
@@ -71,22 +71,20 @@ class TransportDescriptionFactoryTest : public testing::Test {
 
     cricket::TransportOptions options;
     // The initial offer / answer exchange.
-    rtc::scoped_ptr<TransportDescription> offer(f1_.CreateOffer(
-        options, NULL));
-    rtc::scoped_ptr<TransportDescription> answer(
-        f2_.CreateAnswer(offer.get(),
-                         options, NULL));
+    std::unique_ptr<TransportDescription> offer(f1_.CreateOffer(options, NULL));
+    std::unique_ptr<TransportDescription> answer(
+        f2_.CreateAnswer(offer.get(), options, NULL));
 
     // Create an updated offer where we restart ice.
     options.ice_restart = true;
-    rtc::scoped_ptr<TransportDescription> restart_offer(f1_.CreateOffer(
-        options, offer.get()));
+    std::unique_ptr<TransportDescription> restart_offer(
+        f1_.CreateOffer(options, offer.get()));
 
     VerifyUfragAndPasswordChanged(dtls, offer.get(), restart_offer.get());
 
     // Create a new answer. The transport ufrag and password is changed since
     // |options.ice_restart == true|
-    rtc::scoped_ptr<TransportDescription> restart_answer(
+    std::unique_ptr<TransportDescription> restart_answer(
         f2_.CreateAnswer(restart_offer.get(), options, answer.get()));
     ASSERT_TRUE(restart_answer.get() != NULL);
 
@@ -120,8 +118,8 @@ class TransportDescriptionFactoryTest : public testing::Test {
 };
 
 TEST_F(TransportDescriptionFactoryTest, TestOfferDefault) {
-  scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
-      TransportOptions(), NULL));
+  std::unique_ptr<TransportDescription> desc(
+      f1_.CreateOffer(TransportOptions(), NULL));
   CheckDesc(desc.get(), "", "", "", "");
 }
 
@@ -131,8 +129,8 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) {
   std::string digest_alg;
   ASSERT_TRUE(cert1_->ssl_certificate().GetSignatureDigestAlgorithm(
       &digest_alg));
-  scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
-      TransportOptions(), NULL));
+  std::unique_ptr<TransportDescription> desc(
+      f1_.CreateOffer(TransportOptions(), NULL));
   CheckDesc(desc.get(), "", "", "", digest_alg);
   // Ensure it also works with SEC_REQUIRED.
   f1_.set_secure(cricket::SEC_REQUIRED);
@@ -143,8 +141,8 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) {
 // Test generating an offer with DTLS fails with no identity.
 TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsWithNoIdentity) {
   f1_.set_secure(cricket::SEC_ENABLED);
-  scoped_ptr<TransportDescription> desc(f1_.CreateOffer(
-      TransportOptions(), NULL));
+  std::unique_ptr<TransportDescription> desc(
+      f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(desc.get() == NULL);
 }
 
@@ -156,21 +154,21 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsReofferDtls) {
   std::string digest_alg;
   ASSERT_TRUE(cert1_->ssl_certificate().GetSignatureDigestAlgorithm(
       &digest_alg));
-  scoped_ptr<TransportDescription> old_desc(f1_.CreateOffer(
-      TransportOptions(), NULL));
+  std::unique_ptr<TransportDescription> old_desc(
+      f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(old_desc.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
+  std::unique_ptr<TransportDescription> desc(
       f1_.CreateOffer(TransportOptions(), old_desc.get()));
   CheckDesc(desc.get(), "",
             old_desc->ice_ufrag, old_desc->ice_pwd, digest_alg);
 }
 
 TEST_F(TransportDescriptionFactoryTest, TestAnswerDefault) {
-  scoped_ptr<TransportDescription> offer(f1_.CreateOffer(
-      TransportOptions(), NULL));
+  std::unique_ptr<TransportDescription> offer(
+      f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(f2_.CreateAnswer(
-      offer.get(), TransportOptions(), NULL));
+  std::unique_ptr<TransportDescription> desc(
+      f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
   CheckDesc(desc.get(), "", "", "", "");
   desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(),
                               NULL));
@@ -179,15 +177,14 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDefault) {
 
 // Test that we can update an answer properly; ICE credentials shouldn't change.
 TEST_F(TransportDescriptionFactoryTest, TestReanswer) {
-  scoped_ptr<TransportDescription> offer(
+  std::unique_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> old_desc(
+  std::unique_ptr<TransportDescription> old_desc(
       f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
   ASSERT_TRUE(old_desc.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
-      f2_.CreateAnswer(offer.get(), TransportOptions(),
-                       old_desc.get()));
+  std::unique_ptr<TransportDescription> desc(
+      f2_.CreateAnswer(offer.get(), TransportOptions(), old_desc.get()));
   ASSERT_TRUE(desc.get() != NULL);
   CheckDesc(desc.get(), "",
             old_desc->ice_ufrag, old_desc->ice_pwd, "");
@@ -197,10 +194,10 @@ TEST_F(TransportDescriptionFactoryTest, TestReanswer) {
 TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToNoDtls) {
   f1_.set_secure(cricket::SEC_ENABLED);
   f1_.set_certificate(cert1_);
-  scoped_ptr<TransportDescription> offer(
+  std::unique_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
+  std::unique_ptr<TransportDescription> desc(
       f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
   CheckDesc(desc.get(), "", "", "", "");
 }
@@ -210,10 +207,10 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToNoDtls) {
 TEST_F(TransportDescriptionFactoryTest, TestAnswerNoDtlsToDtls) {
   f2_.set_secure(cricket::SEC_ENABLED);
   f2_.set_certificate(cert2_);
-  scoped_ptr<TransportDescription> offer(
+  std::unique_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
+  std::unique_ptr<TransportDescription> desc(
       f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
   CheckDesc(desc.get(), "", "", "", "");
   f2_.set_secure(cricket::SEC_REQUIRED);
@@ -236,10 +233,10 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToDtls) {
   ASSERT_TRUE(cert2_->ssl_certificate().GetSignatureDigestAlgorithm(
       &digest_alg2));
 
-  scoped_ptr<TransportDescription> offer(
+  std::unique_ptr<TransportDescription> offer(
       f1_.CreateOffer(TransportOptions(), NULL));
   ASSERT_TRUE(offer.get() != NULL);
-  scoped_ptr<TransportDescription> desc(
+  std::unique_ptr<TransportDescription> desc(
       f2_.CreateAnswer(offer.get(), TransportOptions(), NULL));
   CheckDesc(desc.get(), "", "", "", digest_alg2);
   f2_.set_secure(cricket::SEC_REQUIRED);
diff --git a/chromium/third_party/webrtc/p2p/base/turnport.cc b/chromium/third_party/webrtc/p2p/base/turnport.cc
index 9ea354c42cf..02553950129 100644
--- a/chromium/third_party/webrtc/p2p/base/turnport.cc
+++ b/chromium/third_party/webrtc/p2p/base/turnport.cc
@@ -533,11 +533,15 @@ int TurnPort::SendTo(const void* data, size_t size,
   return static_cast<int>(size);
 }
 
-void TurnPort::OnReadPacket(
-    rtc::AsyncPacketSocket* socket, const char* data, size_t size,
-    const rtc::SocketAddress& remote_addr,
-    const rtc::PacketTime& packet_time) {
-  ASSERT(socket == socket_);
+bool TurnPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
+                                    const char* data, size_t size,
+                                    const rtc::SocketAddress& remote_addr,
+                                    const rtc::PacketTime& packet_time) {
+  if (socket != socket_) {
+    // The packet was received on a shared socket after we've allocated a new
+    // socket for this TURN port.
+    return false;
+  }
 
   // This is to guard against a STUN response from previous server after
   // alternative server redirection. TODO(guoweis): add a unit test for this
@@ -547,19 +551,19 @@ void TurnPort::OnReadPacket(
                             << remote_addr.ToString()
                             << ", server_address_:"
                             << server_address_.address.ToString();
-    return;
+    return false;
   }
 
   // The message must be at least the size of a channel header.
   if (size < TURN_CHANNEL_HEADER_SIZE) {
     LOG_J(LS_WARNING, this) << "Received TURN message that was too short";
-    return;
+    return false;
   }
 
   if (state_ == STATE_DISCONNECTED) {
     LOG_J(LS_WARNING, this)
         << "Received TURN message while the Turn port is disconnected";
-    return;
+    return false;
   }
 
   // Check the message type, to see if is a Channel Data message.
@@ -568,27 +572,41 @@ void TurnPort::OnReadPacket(
   uint16_t msg_type = rtc::GetBE16(data);
   if (IsTurnChannelData(msg_type)) {
     HandleChannelData(msg_type, data, size, packet_time);
-  } else if (msg_type == TURN_DATA_INDICATION) {
+    return true;
+
+  }
+
+  if (msg_type == TURN_DATA_INDICATION) {
     HandleDataIndication(data, size, packet_time);
-  } else {
-    if (SharedSocket() &&
-        (msg_type == STUN_BINDING_RESPONSE ||
-         msg_type == STUN_BINDING_ERROR_RESPONSE)) {
-      LOG_J(LS_VERBOSE, this) <<
-          "Ignoring STUN binding response message on shared socket.";
-      return;
-    }
+    return true;
+  }
 
-    // This must be a response for one of our requests.
-    // Check success responses, but not errors, for MESSAGE-INTEGRITY.
-    if (IsStunSuccessResponseType(msg_type) &&
-        !StunMessage::ValidateMessageIntegrity(data, size, hash())) {
-      LOG_J(LS_WARNING, this) << "Received TURN message with invalid "
-                              << "message integrity, msg_type=" << msg_type;
-      return;
-    }
-    request_manager_.CheckResponse(data, size);
+  if (SharedSocket() && (msg_type == STUN_BINDING_RESPONSE ||
+                         msg_type == STUN_BINDING_ERROR_RESPONSE)) {
+    LOG_J(LS_VERBOSE, this) <<
+        "Ignoring STUN binding response message on shared socket.";
+    return false;
   }
+
+  // This must be a response for one of our requests.
+  // Check success responses, but not errors, for MESSAGE-INTEGRITY.
+  if (IsStunSuccessResponseType(msg_type) &&
+      !StunMessage::ValidateMessageIntegrity(data, size, hash())) {
+    LOG_J(LS_WARNING, this) << "Received TURN message with invalid "
+                            << "message integrity, msg_type=" << msg_type;
+    return true;
+  }
+  request_manager_.CheckResponse(data, size);
+
+  return true;
+}
+
+void TurnPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
+                            const char* data,
+                            size_t size,
+                            const rtc::SocketAddress& remote_addr,
+                            const rtc::PacketTime& packet_time) {
+  HandleIncomingPacket(socket, data, size, remote_addr, packet_time);
 }
 
 void TurnPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
@@ -1011,7 +1029,7 @@ void TurnPort::OnConnectionDestroyed(Connection* conn) {
 
 void TurnPort::ScheduleEntryDestruction(TurnEntry* entry) {
   ASSERT(entry->destruction_timestamp() == 0);
-  int64_t timestamp = rtc::Time64();
+  int64_t timestamp = rtc::TimeMillis();
   entry->set_destruction_timestamp(timestamp);
   invoker_.AsyncInvokeDelayed<void>(
       thread(),
diff --git a/chromium/third_party/webrtc/p2p/base/turnport.h b/chromium/third_party/webrtc/p2p/base/turnport.h
index 797fa3f94f1..461fc1304db 100644
--- a/chromium/third_party/webrtc/p2p/base/turnport.h
+++ b/chromium/third_party/webrtc/p2p/base/turnport.h
@@ -94,13 +94,10 @@ class TurnPort : public Port {
   virtual int GetOption(rtc::Socket::Option opt, int* value);
   virtual int GetError();
 
-  virtual bool HandleIncomingPacket(
-      rtc::AsyncPacketSocket* socket, const char* data, size_t size,
-      const rtc::SocketAddress& remote_addr,
-      const rtc::PacketTime& packet_time) {
-    OnReadPacket(socket, data, size, remote_addr, packet_time);
-    return true;
-  }
+  virtual bool HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
+                                    const char* data, size_t size,
+                                    const rtc::SocketAddress& remote_addr,
+                                    const rtc::PacketTime& packet_time);
   virtual void OnReadPacket(rtc::AsyncPacketSocket* socket,
                             const char* data, size_t size,
                             const rtc::SocketAddress& remote_addr,
diff --git a/chromium/third_party/webrtc/p2p/base/turnport_unittest.cc b/chromium/third_party/webrtc/p2p/base/turnport_unittest.cc
index 2c93ead9dfa..15a7954d6dc 100644
--- a/chromium/third_party/webrtc/p2p/base/turnport_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/base/turnport_unittest.cc
@@ -11,6 +11,8 @@
 #include <dirent.h>
 #endif
 
+#include <memory>
+
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/portallocator.h"
@@ -26,7 +28,6 @@
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/thread.h"
@@ -507,15 +508,15 @@ class TurnPortTest : public testing::Test,
 
  protected:
   rtc::Thread* main_;
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<TurnPortTestVirtualSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<TurnPortTestVirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
   rtc::Network network_;
   rtc::BasicPacketSocketFactory socket_factory_;
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
+  std::unique_ptr<rtc::AsyncPacketSocket> socket_;
   cricket::TestTurnServer turn_server_;
-  rtc::scoped_ptr<TurnPort> turn_port_;
-  rtc::scoped_ptr<UDPPort> udp_port_;
+  std::unique_ptr<TurnPort> turn_port_;
+  std::unique_ptr<UDPPort> udp_port_;
   bool turn_ready_;
   bool turn_error_;
   bool turn_unknown_address_;
@@ -629,7 +630,7 @@ TEST_F(TurnPortTest, TestTurnAllocateNonceResetAfterAllocateMismatch) {
   // using timestamp |ts_before| but then get an allocate mismatch error and
   // receive an even newer nonce based on the system clock. |ts_before| is
   // chosen so that the two NONCEs generated by the server will be different.
-  int64_t ts_before = rtc::Time64() - 1;
+  int64_t ts_before = rtc::TimeMillis() - 1;
   std::string first_nonce =
       turn_server_.server()->SetTimestampForNextNonce(ts_before);
   turn_port_->PrepareAddress();
@@ -665,6 +666,13 @@ TEST_F(TurnPortTest, TestTurnAllocateMismatch) {
 
   // Verifies that the new port has a different address now.
   EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress());
+
+  // Verify that all packets received from the shared socket are ignored.
+  std::string test_packet = "Test packet";
+  EXPECT_FALSE(turn_port_->HandleIncomingPacket(
+      socket_.get(), test_packet.data(), test_packet.size(),
+      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
+      rtc::CreatePacketTime(0)));
 }
 
 // Tests that a shared-socket-TurnPort creates its own socket after
diff --git a/chromium/third_party/webrtc/p2p/base/turnserver.cc b/chromium/third_party/webrtc/p2p/base/turnserver.cc
index 13974a0bd91..16df6f944ba 100644
--- a/chromium/third_party/webrtc/p2p/base/turnserver.cc
+++ b/chromium/third_party/webrtc/p2p/base/turnserver.cc
@@ -424,7 +424,7 @@ bool TurnServer::ValidateNonce(const std::string& nonce) const {
   }
 
   // Validate the timestamp.
-  return rtc::Time64() - then < kNonceTimeout;
+  return rtc::TimeMillis() - then < kNonceTimeout;
 }
 
 TurnServerAllocation* TurnServer::FindAllocation(TurnServerConnection* conn) {
@@ -465,7 +465,7 @@ void TurnServer::SendErrorResponseWithRealmAndNonce(
   TurnMessage resp;
   InitErrorResponse(msg, code, reason, &resp);
 
-  int64_t timestamp = rtc::Time64();
+  int64_t timestamp = rtc::TimeMillis();
   if (ts_for_next_nonce_) {
     timestamp = ts_for_next_nonce_;
     ts_for_next_nonce_ = 0;
diff --git a/chromium/third_party/webrtc/p2p/base/turnserver.h b/chromium/third_party/webrtc/p2p/base/turnserver.h
index e520a9ef654..2bc3650f091 100644
--- a/chromium/third_party/webrtc/p2p/base/turnserver.h
+++ b/chromium/third_party/webrtc/p2p/base/turnserver.h
@@ -13,6 +13,7 @@
 
 #include <list>
 #include <map>
+#include <memory>
 #include <set>
 #include <string>
 
@@ -125,7 +126,7 @@ class TurnServerAllocation : public rtc::MessageHandler,
   TurnServer* server_;
   rtc::Thread* thread_;
   TurnServerConnection conn_;
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> external_socket_;
+  std::unique_ptr<rtc::AsyncPacketSocket> external_socket_;
   std::string key_;
   std::string transaction_id_;
   std::string username_;
@@ -143,6 +144,7 @@ class TurnAuthInterface {
   // Return true if the given username and realm are valid, or false if not.
   virtual bool GetKey(const std::string& username, const std::string& realm,
                       std::string* key) = 0;
+  virtual ~TurnAuthInterface() = default;
 };
 
 // An interface enables Turn Server to control redirection behavior.
@@ -269,8 +271,7 @@ class TurnServer : public sigslot::has_slots<> {
 
   InternalSocketMap server_sockets_;
   ServerSocketMap server_listen_sockets_;
-  rtc::scoped_ptr<rtc::PacketSocketFactory>
-      external_socket_factory_;
+  std::unique_ptr<rtc::PacketSocketFactory> external_socket_factory_;
   rtc::SocketAddress external_addr_;
 
   AllocationMap allocations_;
diff --git a/chromium/third_party/webrtc/p2p/client/basicportallocator.cc b/chromium/third_party/webrtc/p2p/client/basicportallocator.cc
index 22abf33e936..143b0361837 100644
--- a/chromium/third_party/webrtc/p2p/client/basicportallocator.cc
+++ b/chromium/third_party/webrtc/p2p/client/basicportallocator.cc
@@ -64,33 +64,26 @@ const uint32_t DISABLE_ALL_PHASES =
     PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY;
 
 // BasicPortAllocator
-BasicPortAllocator::BasicPortAllocator(
-    rtc::NetworkManager* network_manager,
-    rtc::PacketSocketFactory* socket_factory)
-    : network_manager_(network_manager),
-      socket_factory_(socket_factory),
-      stun_servers_() {
+BasicPortAllocator::BasicPortAllocator(rtc::NetworkManager* network_manager,
+                                       rtc::PacketSocketFactory* socket_factory)
+    : network_manager_(network_manager), socket_factory_(socket_factory) {
   ASSERT(network_manager_ != nullptr);
   ASSERT(socket_factory_ != nullptr);
   Construct();
 }
 
 BasicPortAllocator::BasicPortAllocator(rtc::NetworkManager* network_manager)
-    : network_manager_(network_manager),
-      socket_factory_(nullptr),
-      stun_servers_() {
+    : network_manager_(network_manager), socket_factory_(nullptr) {
   ASSERT(network_manager_ != nullptr);
   Construct();
 }
 
-BasicPortAllocator::BasicPortAllocator(
-    rtc::NetworkManager* network_manager,
-    rtc::PacketSocketFactory* socket_factory,
-    const ServerAddresses& stun_servers)
-    : network_manager_(network_manager),
-      socket_factory_(socket_factory),
-      stun_servers_(stun_servers) {
+BasicPortAllocator::BasicPortAllocator(rtc::NetworkManager* network_manager,
+                                       rtc::PacketSocketFactory* socket_factory,
+                                       const ServerAddresses& stun_servers)
+    : network_manager_(network_manager), socket_factory_(socket_factory) {
   ASSERT(socket_factory_ != NULL);
+  SetConfiguration(stun_servers, std::vector<RelayServerConfig>(), 0);
   Construct();
 }
 
@@ -100,10 +93,8 @@ BasicPortAllocator::BasicPortAllocator(
     const rtc::SocketAddress& relay_address_udp,
     const rtc::SocketAddress& relay_address_tcp,
     const rtc::SocketAddress& relay_address_ssl)
-    : network_manager_(network_manager),
-      socket_factory_(NULL),
-      stun_servers_(stun_servers) {
-
+    : network_manager_(network_manager), socket_factory_(NULL) {
+  std::vector<RelayServerConfig> turn_servers;
   RelayServerConfig config(RELAY_GTURN);
   if (!relay_address_udp.IsNil()) {
     config.ports.push_back(ProtocolAddress(relay_address_udp, PROTO_UDP));
@@ -116,9 +107,10 @@ BasicPortAllocator::BasicPortAllocator(
   }
 
   if (!config.ports.empty()) {
-    AddTurnServer(config);
+    turn_servers.push_back(config);
   }
 
+  SetConfiguration(stun_servers, turn_servers, 0);
   Construct();
 }
 
@@ -136,6 +128,11 @@ PortAllocatorSession* BasicPortAllocator::CreateSessionInternal(
       this, content_name, component, ice_ufrag, ice_pwd);
 }
 
+void BasicPortAllocator::AddTurnServer(const RelayServerConfig& turn_server) {
+  std::vector<RelayServerConfig> new_turn_servers = turn_servers();
+  new_turn_servers.push_back(turn_server);
+  SetConfiguration(stun_servers(), new_turn_servers, candidate_pool_size());
+}
 
 // BasicPortAllocatorSession
 BasicPortAllocatorSession::BasicPortAllocatorSession(
@@ -207,6 +204,61 @@ void BasicPortAllocatorSession::ClearGettingPorts() {
     sequences_[i]->Stop();
 }
 
+std::vector<PortInterface*> BasicPortAllocatorSession::ReadyPorts() const {
+  std::vector<PortInterface*> ret;
+  for (const PortData& port : ports_) {
+    if (port.ready() || port.complete()) {
+      ret.push_back(port.port());
+    }
+  }
+  return ret;
+}
+
+std::vector<Candidate> BasicPortAllocatorSession::ReadyCandidates() const {
+  std::vector<Candidate> candidates;
+  for (const PortData& data : ports_) {
+    for (const Candidate& candidate : data.port()->Candidates()) {
+      if (!CheckCandidateFilter(candidate)) {
+        continue;
+      }
+      ProtocolType pvalue;
+      if (!StringToProto(candidate.protocol().c_str(), &pvalue) ||
+          !data.sequence()->ProtocolEnabled(pvalue)) {
+        continue;
+      }
+      candidates.push_back(candidate);
+    }
+  }
+  return candidates;
+}
+
+bool BasicPortAllocatorSession::CandidatesAllocationDone() const {
+  // Done only if all required AllocationSequence objects
+  // are created.
+  if (!allocation_sequences_created_) {
+    return false;
+  }
+
+  // Check that all port allocation sequences are complete (not running).
+  if (std::any_of(sequences_.begin(), sequences_.end(),
+                  [](const AllocationSequence* sequence) {
+                    return sequence->state() == AllocationSequence::kRunning;
+                  })) {
+    return false;
+  }
+
+  // If all allocated ports are in complete state, session must have got all
+  // expected candidates. Session will trigger candidates allocation complete
+  // signal.
+  if (!std::all_of(ports_.begin(), ports_.end(), [](const PortData& port) {
+        return (port.complete() || port.error());
+      })) {
+    return false;
+  }
+
+  return true;
+}
+
 void BasicPortAllocatorSession::OnMessage(rtc::Message *message) {
   switch (message->message_id) {
   case MSG_CONFIG_START:
@@ -241,6 +293,13 @@ void BasicPortAllocatorSession::OnMessage(rtc::Message *message) {
   }
 }
 
+void BasicPortAllocatorSession::UpdateIceParametersInternal() {
+  for (PortData& port : ports_) {
+    port.port()->set_content_name(content_name());
+    port.port()->SetIceParameters(component(), ice_ufrag(), ice_pwd());
+  }
+}
+
 void BasicPortAllocatorSession::GetPortConfigurations() {
   PortConfiguration* config = new PortConfiguration(allocator_->stun_servers(),
                                                     username(),
@@ -274,7 +333,7 @@ void BasicPortAllocatorSession::OnConfigStop() {
   bool send_signal = false;
   for (std::vector<PortData>::iterator it = ports_.begin();
        it != ports_.end(); ++it) {
-    if (!it->complete()) {
+    if (!it->complete() && !it->error()) {
       // Updating port state to error, which didn't finish allocating candidates
       // yet.
       it->set_error();
@@ -436,12 +495,12 @@ void BasicPortAllocatorSession::AddAllocatedPort(Port* port,
 
   LOG(LS_INFO) << "Adding allocated port for " << content_name();
   port->set_content_name(content_name());
-  port->set_component(component_);
+  port->set_component(component());
   port->set_generation(generation());
   if (allocator_->proxy().type != rtc::PROXY_NONE)
     port->set_proxy(allocator_->user_agent(), allocator_->proxy());
-  port->set_send_retransmit_count_attribute((allocator_->flags() &
-      PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE) != 0);
+  port->set_send_retransmit_count_attribute(
+      (flags() & PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE) != 0);
 
   // Push down the candidate_filter to individual port.
   uint32_t candidate_filter = allocator_->candidate_filter();
@@ -484,8 +543,9 @@ void BasicPortAllocatorSession::OnCandidateReady(
   ASSERT(data != NULL);
   // Discarding any candidate signal if port allocation status is
   // already in completed state.
-  if (data->complete())
+  if (data->complete() || data->error()) {
     return;
+  }
 
   ProtocolType pvalue;
   bool candidate_signalable = CheckCandidateFilter(c);
@@ -536,8 +596,9 @@ void BasicPortAllocatorSession::OnPortComplete(Port* port) {
   ASSERT(data != NULL);
 
   // Ignore any late signals.
-  if (data->complete())
+  if (data->complete() || data->error()) {
     return;
+  }
 
   // Moving to COMPLETE state.
   data->set_complete();
@@ -550,8 +611,9 @@ void BasicPortAllocatorSession::OnPortError(Port* port) {
   PortData* data = FindPort(port);
   ASSERT(data != NULL);
   // We might have already given up on this port and stopped it.
-  if (data->complete())
+  if (data->complete() || data->error()) {
     return;
+  }
 
   // SignalAddressError is currently sent from StunPort/TurnPort.
   // But this signal itself is generic.
@@ -587,7 +649,7 @@ void BasicPortAllocatorSession::OnProtocolEnabled(AllocationSequence* seq,
   }
 }
 
-bool BasicPortAllocatorSession::CheckCandidateFilter(const Candidate& c) {
+bool BasicPortAllocatorSession::CheckCandidateFilter(const Candidate& c) const {
   uint32_t filter = allocator_->candidate_filter();
 
   // When binding to any address, before sending packets out, the getsockname
@@ -625,29 +687,15 @@ void BasicPortAllocatorSession::OnPortAllocationComplete(
 }
 
 void BasicPortAllocatorSession::MaybeSignalCandidatesAllocationDone() {
-  // Send signal only if all required AllocationSequence objects
-  // are created.
-  if (!allocation_sequences_created_)
-    return;
-
-  // Check that all port allocation sequences are complete.
-  for (std::vector<AllocationSequence*>::iterator it = sequences_.begin();
-       it != sequences_.end(); ++it) {
-    if ((*it)->state() == AllocationSequence::kRunning)
-      return;
-  }
-
-  // If all allocated ports are in complete state, session must have got all
-  // expected candidates. Session will trigger candidates allocation complete
-  // signal.
-  for (std::vector<PortData>::iterator it = ports_.begin();
-       it != ports_.end(); ++it) {
-    if (!it->complete())
-      return;
+  if (CandidatesAllocationDone()) {
+    if (pooled()) {
+      LOG(LS_INFO) << "All candidates gathered for pooled session.";
+    } else {
+      LOG(LS_INFO) << "All candidates gathered for " << content_name() << ":"
+                   << component() << ":" << generation();
+    }
+    SignalCandidatesAllocationDone(this);
   }
-  LOG(LS_INFO) << "All candidates gathered for " << content_name_ << ":"
-               << component_ << ":" << generation();
-  SignalCandidatesAllocationDone(this);
 }
 
 void BasicPortAllocatorSession::OnPortDestroyed(
@@ -1080,13 +1128,13 @@ void AllocationSequence::OnReadPacket(
   // a STUN binding response, so we pass the message to TurnPort regardless of
   // the message type. The TurnPort will just ignore the message since it will
   // not find any request by transaction ID.
-  for (std::vector<TurnPort*>::const_iterator it = turn_ports_.begin();
-       it != turn_ports_.end(); ++it) {
-    TurnPort* port = *it;
+  for (TurnPort* port : turn_ports_) {
     if (port->server_address().address == remote_addr) {
-      port->HandleIncomingPacket(socket, data, size, remote_addr, packet_time);
+      if (port->HandleIncomingPacket(socket, data, size, remote_addr,
+                                     packet_time)) {
+        return;
+      }
       turn_port_found = true;
-      break;
     }
   }
 
@@ -1097,8 +1145,9 @@ void AllocationSequence::OnReadPacket(
     // the TURN server is also a STUN server.
     if (!turn_port_found ||
         stun_servers.find(remote_addr) != stun_servers.end()) {
-      udp_port_->HandleIncomingPacket(
-          socket, data, size, remote_addr, packet_time);
+      RTC_DCHECK(udp_port_->SharedSocket());
+      udp_port_->HandleIncomingPacket(socket, data, size, remote_addr,
+                                      packet_time);
     }
   }
 }
diff --git a/chromium/third_party/webrtc/p2p/client/basicportallocator.h b/chromium/third_party/webrtc/p2p/client/basicportallocator.h
index ca1a23aaf2c..fd189c14e9b 100644
--- a/chromium/third_party/webrtc/p2p/client/basicportallocator.h
+++ b/chromium/third_party/webrtc/p2p/client/basicportallocator.h
@@ -11,13 +11,13 @@
 #ifndef WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_
 #define WEBRTC_P2P_CLIENT_BASICPORTALLOCATOR_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/base/messagequeue.h"
 #include "webrtc/base/network.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 
 namespace cricket {
@@ -37,13 +37,6 @@ class BasicPortAllocator : public PortAllocator {
                      const rtc::SocketAddress& relay_server_ssl);
   virtual ~BasicPortAllocator();
 
-  void SetIceServers(
-      const ServerAddresses& stun_servers,
-      const std::vector<RelayServerConfig>& turn_servers) override {
-    stun_servers_ = stun_servers;
-    turn_servers_ = turn_servers;
-  }
-
   // Set to kDefaultNetworkIgnoreMask by default.
   void SetNetworkIgnoreMask(int network_ignore_mask) override {
     // TODO(phoglund): implement support for other types than loopback.
@@ -60,30 +53,20 @@ class BasicPortAllocator : public PortAllocator {
   // creates its own socket factory.
   rtc::PacketSocketFactory* socket_factory() { return socket_factory_; }
 
-  const ServerAddresses& stun_servers() const {
-    return stun_servers_;
-  }
-
-  const std::vector<RelayServerConfig>& turn_servers() const {
-    return turn_servers_;
-  }
-  virtual void AddTurnServer(const RelayServerConfig& turn_server) {
-    turn_servers_.push_back(turn_server);
-  }
-
   PortAllocatorSession* CreateSessionInternal(
       const std::string& content_name,
       int component,
       const std::string& ice_ufrag,
       const std::string& ice_pwd) override;
 
+  // Convenience method that adds a TURN server to the configuration.
+  void AddTurnServer(const RelayServerConfig& turn_server);
+
  private:
   void Construct();
 
   rtc::NetworkManager* network_manager_;
   rtc::PacketSocketFactory* socket_factory_;
-  ServerAddresses stun_servers_;
-  std::vector<RelayServerConfig> turn_servers_;
   bool allow_tcp_listen_;
   int network_ignore_mask_ = rtc::kDefaultNetworkIgnoreMask;
 };
@@ -109,8 +92,14 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
   void StopGettingPorts() override;
   void ClearGettingPorts() override;
   bool IsGettingPorts() override { return running_; }
+  // These will all be cricket::Ports.
+  std::vector<PortInterface*> ReadyPorts() const override;
+  std::vector<Candidate> ReadyCandidates() const override;
+  bool CandidatesAllocationDone() const override;
 
  protected:
+  void UpdateIceParametersInternal() override;
+
   // Starts the process of getting the port configurations.
   virtual void GetPortConfigurations();
 
@@ -129,13 +118,11 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
     : port_(port), sequence_(seq), state_(STATE_INIT) {
     }
 
-    Port* port() { return port_; }
-    AllocationSequence* sequence() { return sequence_; }
+    Port* port() const { return port_; }
+    AllocationSequence* sequence() const { return sequence_; }
     bool ready() const { return state_ == STATE_READY; }
-    bool complete() const {
-      // Returns true if candidate allocation has completed one way or another.
-      return ((state_ == STATE_COMPLETE) || (state_ == STATE_ERROR));
-    }
+    bool complete() const { return state_ == STATE_COMPLETE; }
+    bool error() const { return state_ == STATE_ERROR; }
 
     void set_ready() { ASSERT(state_ == STATE_INIT); state_ = STATE_READY; }
     void set_complete() {
@@ -181,11 +168,11 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
   PortData* FindPort(Port* port);
   void GetNetworks(std::vector<rtc::Network*>* networks);
 
-  bool CheckCandidateFilter(const Candidate& c);
+  bool CheckCandidateFilter(const Candidate& c) const;
 
   BasicPortAllocator* allocator_;
   rtc::Thread* network_thread_;
-  rtc::scoped_ptr<rtc::PacketSocketFactory> owned_socket_factory_;
+  std::unique_ptr<rtc::PacketSocketFactory> owned_socket_factory_;
   rtc::PacketSocketFactory* socket_factory_;
   bool allocation_started_;
   bool network_manager_started_;
@@ -320,7 +307,7 @@ class AllocationSequence : public rtc::MessageHandler,
   State state_;
   uint32_t flags_;
   ProtocolList protocols_;
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> udp_socket_;
+  std::unique_ptr<rtc::AsyncPacketSocket> udp_socket_;
   // There will be only one udp port per AllocationSequence.
   UDPPort* udp_port_;
   std::vector<TurnPort*> turn_ports_;
diff --git a/chromium/third_party/webrtc/p2p/client/portallocator_unittest.cc b/chromium/third_party/webrtc/p2p/client/basicportallocator_unittest.cc
index a76900ab6a8..83d904f1814 100644
--- a/chromium/third_party/webrtc/p2p/client/portallocator_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/client/basicportallocator_unittest.cc
@@ -8,6 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <algorithm>
+#include <memory>
+
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/p2ptransportchannel.h"
@@ -40,8 +43,8 @@ static const SocketAddress kClientAddr("11.11.11.11", 0);
 static const SocketAddress kLoopbackAddr("127.0.0.1", 0);
 static const SocketAddress kPrivateAddr("192.168.1.11", 0);
 static const SocketAddress kPrivateAddr2("192.168.1.12", 0);
-static const SocketAddress kClientIPv6Addr(
-    "2401:fa00:4:1000:be30:5bff:fee5:c3", 0);
+static const SocketAddress kClientIPv6Addr("2401:fa00:4:1000:be30:5bff:fee5:c3",
+                                           0);
 static const SocketAddress kClientAddr2("22.22.22.22", 0);
 static const SocketAddress kNatUdpAddr("77.77.77.77", rtc::NAT_SERVER_UDP_PORT);
 static const SocketAddress kNatTcpAddr("77.77.77.77", rtc::NAT_SERVER_TCP_PORT);
@@ -82,20 +85,25 @@ std::ostream& operator<<(std::ostream& os, const cricket::Candidate& c) {
 
 }  // namespace cricket
 
-class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
+class BasicPortAllocatorTest : public testing::Test,
+                               public sigslot::has_slots<> {
  public:
-  PortAllocatorTest()
+  BasicPortAllocatorTest()
       : pss_(new rtc::PhysicalSocketServer),
         vss_(new rtc::VirtualSocketServer(pss_.get())),
         fss_(new rtc::FirewallSocketServer(vss_.get())),
         ss_scope_(fss_.get()),
         nat_factory_(vss_.get(), kNatUdpAddr, kNatTcpAddr),
         nat_socket_factory_(new rtc::BasicPacketSocketFactory(&nat_factory_)),
-        stun_server_(cricket::TestStunServer::Create(Thread::Current(),
-                                                     kStunAddr)),
-        relay_server_(Thread::Current(), kRelayUdpIntAddr, kRelayUdpExtAddr,
-                      kRelayTcpIntAddr, kRelayTcpExtAddr,
-                      kRelaySslTcpIntAddr, kRelaySslTcpExtAddr),
+        stun_server_(
+            cricket::TestStunServer::Create(Thread::Current(), kStunAddr)),
+        relay_server_(Thread::Current(),
+                      kRelayUdpIntAddr,
+                      kRelayUdpExtAddr,
+                      kRelayTcpIntAddr,
+                      kRelayTcpExtAddr,
+                      kRelaySslTcpIntAddr,
+                      kRelaySslTcpExtAddr),
         turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
         candidate_allocation_done_(false) {
     cricket::ServerAddresses stun_servers;
@@ -103,9 +111,8 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
     // Passing the addresses of GTURN servers will enable GTURN in
     // Basicportallocator.
     allocator_.reset(new cricket::BasicPortAllocator(
-        &network_manager_,
-        stun_servers,
-        kRelayUdpIntAddr, kRelayTcpIntAddr, kRelaySslTcpIntAddr));
+        &network_manager_, stun_servers, kRelayUdpIntAddr, kRelayTcpIntAddr,
+        kRelaySslTcpIntAddr));
     allocator_->set_step_delay(cricket::kMinimumStepDelay);
   }
 
@@ -176,55 +183,65 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
   }
 
   bool CreateSession(int component) {
-    session_.reset(CreateSession("session", component));
-    if (!session_)
+    session_ = CreateSession("session", component);
+    if (!session_) {
       return false;
+    }
     return true;
   }
 
   bool CreateSession(int component, const std::string& content_name) {
-    session_.reset(CreateSession("session", content_name, component));
-    if (!session_)
+    session_ = CreateSession("session", content_name, component);
+    if (!session_) {
       return false;
+    }
     return true;
   }
 
-  cricket::PortAllocatorSession* CreateSession(
-      const std::string& sid, int component) {
+  std::unique_ptr<cricket::PortAllocatorSession> CreateSession(
+      const std::string& sid,
+      int component) {
     return CreateSession(sid, kContentName, component);
   }
 
-  cricket::PortAllocatorSession* CreateSession(
-      const std::string& sid, const std::string& content_name, int component) {
+  std::unique_ptr<cricket::PortAllocatorSession> CreateSession(
+      const std::string& sid,
+      const std::string& content_name,
+      int component) {
     return CreateSession(sid, content_name, component, kIceUfrag0, kIcePwd0);
   }
 
-  cricket::PortAllocatorSession* CreateSession(
-      const std::string& sid, const std::string& content_name, int component,
-      const std::string& ice_ufrag, const std::string& ice_pwd) {
-    cricket::PortAllocatorSession* session =
-        allocator_->CreateSession(
-            sid, content_name, component, ice_ufrag, ice_pwd);
+  std::unique_ptr<cricket::PortAllocatorSession> CreateSession(
+      const std::string& sid,
+      const std::string& content_name,
+      int component,
+      const std::string& ice_ufrag,
+      const std::string& ice_pwd) {
+    std::unique_ptr<cricket::PortAllocatorSession> session =
+        allocator_->CreateSession(sid, content_name, component, ice_ufrag,
+                                  ice_pwd);
     session->SignalPortReady.connect(this,
-            &PortAllocatorTest::OnPortReady);
-    session->SignalCandidatesReady.connect(this,
-        &PortAllocatorTest::OnCandidatesReady);
-    session->SignalCandidatesAllocationDone.connect(this,
-        &PortAllocatorTest::OnCandidatesAllocationDone);
+                                     &BasicPortAllocatorTest::OnPortReady);
+    session->SignalCandidatesReady.connect(
+        this, &BasicPortAllocatorTest::OnCandidatesReady);
+    session->SignalCandidatesAllocationDone.connect(
+        this, &BasicPortAllocatorTest::OnCandidatesAllocationDone);
     return session;
   }
 
   static bool CheckCandidate(const cricket::Candidate& c,
-                             int component, const std::string& type,
+                             int component,
+                             const std::string& type,
                              const std::string& proto,
                              const SocketAddress& addr) {
     return (c.component() == component && c.type() == type &&
-        c.protocol() == proto && c.address().ipaddr() == addr.ipaddr() &&
-        ((addr.port() == 0 && (c.address().port() != 0)) ||
-        (c.address().port() == addr.port())));
+            c.protocol() == proto && c.address().ipaddr() == addr.ipaddr() &&
+            ((addr.port() == 0 && (c.address().port() != 0)) ||
+             (c.address().port() == addr.port())));
   }
   static bool CheckPort(const rtc::SocketAddress& addr,
-                        int min_port, int max_port) {
+                        int min_port,
+                        int max_port) {
     return (addr.port() >= min_port && addr.port() <= max_port);
   }
 
@@ -235,6 +252,7 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
       ASSERT_FALSE(candidate_allocation_done_);
       candidate_allocation_done_ = true;
     }
+    EXPECT_TRUE(session->CandidatesAllocationDone());
   }
 
   // Check if all ports allocated have send-buffer size |expected|. If
@@ -245,11 +263,10 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
       int send_buffer_size;
       if (expected == -1) {
         EXPECT_EQ(SOCKET_ERROR,
-                  (*it)->GetOption(rtc::Socket::OPT_SNDBUF,
-                                   &send_buffer_size));
+                  (*it)->GetOption(rtc::Socket::OPT_SNDBUF, &send_buffer_size));
       } else {
-        EXPECT_EQ(0, (*it)->GetOption(rtc::Socket::OPT_SNDBUF,
-                                      &send_buffer_size));
+        EXPECT_EQ(0,
+                  (*it)->GetOption(rtc::Socket::OPT_SNDBUF, &send_buffer_size));
         ASSERT_EQ(expected, send_buffer_size);
       }
     }
@@ -320,14 +337,16 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
   }
 
  protected:
-  cricket::BasicPortAllocator& allocator() {
-    return *allocator_;
-  }
+  cricket::BasicPortAllocator& allocator() { return *allocator_; }
 
   void OnPortReady(cricket::PortAllocatorSession* ses,
                    cricket::PortInterface* port) {
     LOG(LS_INFO) << "OnPortReady: " << port->ToString();
     ports_.push_back(port);
+    // Make sure the new port is added to ReadyPorts.
+    auto ready_ports = ses->ReadyPorts();
+    EXPECT_NE(ready_ports.end(),
+              std::find(ready_ports.begin(), ready_ports.end(), port));
   }
   void OnCandidatesReady(cricket::PortAllocatorSession* ses,
                          const std::vector<cricket::Candidate>& candidates) {
@@ -335,6 +354,13 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
       LOG(LS_INFO) << "OnCandidatesReady: " << candidates[i].ToString();
       candidates_.push_back(candidates[i]);
     }
+    // Make sure the new candidates are added to Candidates.
+    auto ses_candidates = ses->ReadyCandidates();
+    for (const cricket::Candidate& candidate : candidates) {
+      EXPECT_NE(
+          ses_candidates.end(),
+          std::find(ses_candidates.begin(), ses_candidates.end(), candidate));
+    }
   }
 
   bool HasRelayAddress(const cricket::ProtocolAddress& proto_addr) {
@@ -342,7 +368,7 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
       cricket::RelayServerConfig server_config = allocator_->turn_servers()[i];
       cricket::PortList::const_iterator relay_port;
       for (relay_port = server_config.ports.begin();
-          relay_port != server_config.ports.end(); ++relay_port) {
+           relay_port != server_config.ports.end(); ++relay_port) {
         if (proto_addr.address == relay_port->address &&
             proto_addr.proto == relay_port->proto)
           return true;
@@ -370,26 +396,26 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
     allocator().set_step_delay(cricket::kMinimumStepDelay);
   }
 
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> vss_;
-  rtc::scoped_ptr<rtc::FirewallSocketServer> fss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
+  std::unique_ptr<rtc::FirewallSocketServer> fss_;
   rtc::SocketServerScope ss_scope_;
-  rtc::scoped_ptr<rtc::NATServer> nat_server_;
+  std::unique_ptr<rtc::NATServer> nat_server_;
   rtc::NATSocketFactory nat_factory_;
-  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> nat_socket_factory_;
-  rtc::scoped_ptr<cricket::TestStunServer> stun_server_;
+  std::unique_ptr<rtc::BasicPacketSocketFactory> nat_socket_factory_;
+  std::unique_ptr<cricket::TestStunServer> stun_server_;
   cricket::TestRelayServer relay_server_;
   cricket::TestTurnServer turn_server_;
   rtc::FakeNetworkManager network_manager_;
-  rtc::scoped_ptr<cricket::BasicPortAllocator> allocator_;
-  rtc::scoped_ptr<cricket::PortAllocatorSession> session_;
+  std::unique_ptr<cricket::BasicPortAllocator> allocator_;
+  std::unique_ptr<cricket::PortAllocatorSession> session_;
   std::vector<cricket::PortInterface*> ports_;
   std::vector<cricket::Candidate> candidates_;
   bool candidate_allocation_done_;
 };
 
 // Tests that we can init the port allocator and create a session.
-TEST_F(PortAllocatorTest, TestBasic) {
+TEST_F(BasicPortAllocatorTest, TestBasic) {
   EXPECT_EQ(&network_manager_, allocator().network_manager());
   EXPECT_EQ(kStunAddr, *allocator().stun_servers().begin());
   ASSERT_EQ(1u, allocator().turn_servers().size());
@@ -397,17 +423,18 @@ TEST_F(PortAllocatorTest, TestBasic) {
   // Empty relay credentials are used for GTURN.
   EXPECT_TRUE(allocator().turn_servers()[0].credentials.username.empty());
   EXPECT_TRUE(allocator().turn_servers()[0].credentials.password.empty());
-  EXPECT_TRUE(HasRelayAddress(cricket::ProtocolAddress(
-      kRelayUdpIntAddr, cricket::PROTO_UDP)));
-  EXPECT_TRUE(HasRelayAddress(cricket::ProtocolAddress(
-      kRelayTcpIntAddr, cricket::PROTO_TCP)));
-  EXPECT_TRUE(HasRelayAddress(cricket::ProtocolAddress(
-      kRelaySslTcpIntAddr, cricket::PROTO_SSLTCP)));
+  EXPECT_TRUE(HasRelayAddress(
+      cricket::ProtocolAddress(kRelayUdpIntAddr, cricket::PROTO_UDP)));
+  EXPECT_TRUE(HasRelayAddress(
+      cricket::ProtocolAddress(kRelayTcpIntAddr, cricket::PROTO_TCP)));
+  EXPECT_TRUE(HasRelayAddress(
+      cricket::ProtocolAddress(kRelaySslTcpIntAddr, cricket::PROTO_SSLTCP)));
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
+  EXPECT_FALSE(session_->CandidatesAllocationDone());
 }
 
 // Tests that our network filtering works properly.
-TEST_F(PortAllocatorTest, TestIgnoreOnlyLoopbackNetworkByDefault) {
+TEST_F(BasicPortAllocatorTest, TestIgnoreOnlyLoopbackNetworkByDefault) {
   AddInterface(SocketAddress(IPAddress(0x12345600U), 0), "test_eth0",
                rtc::ADAPTER_TYPE_ETHERNET);
   AddInterface(SocketAddress(IPAddress(0x12345601U), 0), "test_wlan0",
@@ -430,7 +457,7 @@ TEST_F(PortAllocatorTest, TestIgnoreOnlyLoopbackNetworkByDefault) {
   }
 }
 
-TEST_F(PortAllocatorTest, TestIgnoreNetworksAccordingToIgnoreMask) {
+TEST_F(BasicPortAllocatorTest, TestIgnoreNetworksAccordingToIgnoreMask) {
   AddInterface(SocketAddress(IPAddress(0x12345600U), 0), "test_eth0",
                rtc::ADAPTER_TYPE_ETHERNET);
   AddInterface(SocketAddress(IPAddress(0x12345601U), 0), "test_wlan0",
@@ -451,7 +478,7 @@ TEST_F(PortAllocatorTest, TestIgnoreNetworksAccordingToIgnoreMask) {
 }
 
 // Tests that we allocator session not trying to allocate ports for every 250ms.
-TEST_F(PortAllocatorTest, TestNoNetworkInterface) {
+TEST_F(BasicPortAllocatorTest, TestNoNetworkInterface) {
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   // Waiting for one second to make sure BasicPortAllocatorSession has not
@@ -464,7 +491,7 @@ TEST_F(PortAllocatorTest, TestNoNetworkInterface) {
 }
 
 // Test that we could use loopback interface as host candidate.
-TEST_F(PortAllocatorTest, TestLoopbackNetworkInterface) {
+TEST_F(BasicPortAllocatorTest, TestLoopbackNetworkInterface) {
   AddInterface(kLoopbackAddr, "test_loopback", rtc::ADAPTER_TYPE_LOOPBACK);
   allocator_->SetNetworkIgnoreMask(0);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -477,34 +504,40 @@ TEST_F(PortAllocatorTest, TestLoopbackNetworkInterface) {
 }
 
 // Tests that we can get all the desired addresses successfully.
-TEST_F(PortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) {
   AddInterface(kClientAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_EQ(4U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[3],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpExtAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpExtAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[4],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp", kRelayTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp",
+               kRelayTcpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[5],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[6],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP,
-      "relay", "ssltcp", kRelaySslTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "ssltcp",
+               kRelaySslTcpIntAddr);
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
 // Test that when the same network interface is brought down and up, the
 // port allocator session will restart a new allocation sequence if
 // it is not stopped.
-TEST_F(PortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
+TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
   std::string if_name("test_net0");
   AddInterface(kClientAddr, if_name);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -532,7 +565,7 @@ TEST_F(PortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
 // Test that when the same network interface is brought down and up, the
 // port allocator session will not restart a new allocation sequence if
 // it is stopped.
-TEST_F(PortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) {
+TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) {
   std::string if_name("test_net0");
   AddInterface(kClientAddr, if_name);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -557,7 +590,7 @@ TEST_F(PortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) {
 }
 
 // Verify candidates with default step delay of 1sec.
-TEST_F(PortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) {
   AddInterface(kClientAddr);
   allocator_->set_step_delay(cricket::kDefaultStepDelay);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -567,29 +600,33 @@ TEST_F(PortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) {
   ASSERT_EQ_WAIT(4U, candidates_.size(), 2000);
   EXPECT_EQ(3U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[3],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpExtAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpExtAddr);
   ASSERT_EQ_WAIT(6U, candidates_.size(), 1500);
   EXPECT_PRED5(CheckCandidate, candidates_[4],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp", kRelayTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp",
+               kRelayTcpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[5],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientAddr);
   EXPECT_EQ(4U, ports_.size());
   ASSERT_EQ_WAIT(7U, candidates_.size(), 2000);
   EXPECT_PRED5(CheckCandidate, candidates_[6],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP,
-               "relay", "ssltcp", kRelaySslTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "ssltcp",
+               kRelaySslTcpIntAddr);
   EXPECT_EQ(4U, ports_.size());
   EXPECT_TRUE(candidate_allocation_done_);
   // If we Stop gathering now, we shouldn't get a second "done" callback.
   session_->StopGettingPorts();
 }
 
-TEST_F(PortAllocatorTest, TestSetupVideoRtpPortsWithNormalSendBuffers) {
+TEST_F(BasicPortAllocatorTest, TestSetupVideoRtpPortsWithNormalSendBuffers) {
   AddInterface(kClientAddr);
-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP,
-                            cricket::CN_VIDEO));
+  EXPECT_TRUE(
+      CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP, cricket::CN_VIDEO));
   session_->StartGettingPorts();
   ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_TRUE(candidate_allocation_done_);
@@ -601,7 +638,7 @@ TEST_F(PortAllocatorTest, TestSetupVideoRtpPortsWithNormalSendBuffers) {
 }
 
 // Tests that we can get callback after StopGetAllPorts.
-TEST_F(PortAllocatorTest, TestStopGetAllPorts) {
+TEST_F(BasicPortAllocatorTest, TestStopGetAllPorts) {
   AddInterface(kClientAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
@@ -614,7 +651,7 @@ TEST_F(PortAllocatorTest, TestStopGetAllPorts) {
 // Test that we restrict client ports appropriately when a port range is set.
 // We check the candidates for udp/stun/tcp ports, and the from address
 // for relay ports.
-TEST_F(PortAllocatorTest, TestGetAllPortsPortRange) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsPortRange) {
   AddInterface(kClientAddr);
   // Check that an invalid port range fails.
   EXPECT_FALSE(SetPortRange(kMaxPort, kMinPort));
@@ -631,15 +668,15 @@ TEST_F(PortAllocatorTest, TestGetAllPortsPortRange) {
   // Check the port number for the STUN port object.
   EXPECT_PRED3(CheckPort, candidates_[1].address(), kMinPort, kMaxPort);
   // Check the port number used to connect to the relay server.
-  EXPECT_PRED3(CheckPort, relay_server_.GetConnection(0).source(),
-               kMinPort, kMaxPort);
+  EXPECT_PRED3(CheckPort, relay_server_.GetConnection(0).source(), kMinPort,
+               kMaxPort);
   // Check the port number for the TCP port object.
   EXPECT_PRED3(CheckPort, candidates_[5].address(), kMinPort, kMaxPort);
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
 // Test that we don't crash or malfunction if we have no network adapters.
-TEST_F(PortAllocatorTest, TestGetAllPortsNoAdapters) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoAdapters) {
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   rtc::Thread::Current()->ProcessMessages(100);
@@ -650,7 +687,7 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoAdapters) {
 
 // Test that when enumeration is disabled, we should not have any ports when
 // candidate_filter() is set to CF_RELAY and no relay is specified.
-TEST_F(PortAllocatorTest,
+TEST_F(BasicPortAllocatorTest,
        TestDisableAdapterEnumerationWithoutNatRelayTransportOnly) {
   ResetWithStunServerNoNat(kStunAddr);
   allocator().set_candidate_filter(cricket::CF_RELAY);
@@ -662,7 +699,7 @@ TEST_F(PortAllocatorTest,
 // Test that even with multiple interfaces, the result should still be a single
 // default private, one STUN and one TURN candidate since we bind to any address
 // (i.e. all 0s).
-TEST_F(PortAllocatorTest,
+TEST_F(BasicPortAllocatorTest,
        TestDisableAdapterEnumerationBehindNatMultipleInterfaces) {
   AddInterface(kPrivateAddr);
   AddInterface(kPrivateAddr2);
@@ -685,7 +722,7 @@ TEST_F(PortAllocatorTest,
 
 // Test that we should get a default private, STUN, TURN/UDP and TURN/TCP
 // candidates when both TURN/UDP and TURN/TCP servers are specified.
-TEST_F(PortAllocatorTest, TestDisableAdapterEnumerationBehindNatWithTcp) {
+TEST_F(BasicPortAllocatorTest, TestDisableAdapterEnumerationBehindNatWithTcp) {
   turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
   AddInterface(kPrivateAddr);
   ResetWithStunServerAndNat(kStunAddr);
@@ -699,7 +736,8 @@ TEST_F(PortAllocatorTest, TestDisableAdapterEnumerationBehindNatWithTcp) {
 
 // Test that when adapter enumeration is disabled, for endpoints without
 // STUN/TURN specified, a default private candidate is still generated.
-TEST_F(PortAllocatorTest, TestDisableAdapterEnumerationWithoutNatOrServers) {
+TEST_F(BasicPortAllocatorTest,
+       TestDisableAdapterEnumerationWithoutNatOrServers) {
   ResetWithNoServersOrNat();
   // Expect to see 2 ports: STUN and TCP ports, one default private candidate.
   CheckDisableAdapterEnumeration(2U, kPrivateAddr.ipaddr(), rtc::IPAddress(),
@@ -709,7 +747,7 @@ TEST_F(PortAllocatorTest, TestDisableAdapterEnumerationWithoutNatOrServers) {
 // Test that when adapter enumeration is disabled, with
 // PORTALLOCATOR_DISABLE_LOCALHOST_CANDIDATE specified, for endpoints not behind
 // a NAT, there is no local candidate.
-TEST_F(PortAllocatorTest,
+TEST_F(BasicPortAllocatorTest,
        TestDisableAdapterEnumerationWithoutNatLocalhostCandidateDisabled) {
   ResetWithStunServerNoNat(kStunAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -726,7 +764,7 @@ TEST_F(PortAllocatorTest,
 // (kClientAddr) which was discovered when sending STUN requests, will become
 // the srflx addresses.
 TEST_F(
-    PortAllocatorTest,
+    BasicPortAllocatorTest,
     TestDisableAdapterEnumerationWithoutNatLocalhostCandidateDisabledWithDifferentDefaultRoute) {
   ResetWithStunServerNoNat(kStunAddr);
   AddInterfaceAsDefaultRoute(kClientAddr);
@@ -741,7 +779,7 @@ TEST_F(
 // Test that when adapter enumeration is disabled, with
 // PORTALLOCATOR_DISABLE_LOCALHOST_CANDIDATE specified, for endpoints behind a
 // NAT, there is only one STUN candidate.
-TEST_F(PortAllocatorTest,
+TEST_F(BasicPortAllocatorTest,
        TestDisableAdapterEnumerationWithNatLocalhostCandidateDisabled) {
   ResetWithStunServerAndNat(kStunAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -753,7 +791,7 @@ TEST_F(PortAllocatorTest,
 
 // Test that we disable relay over UDP, and only TCP is used when connecting to
 // the relay server.
-TEST_F(PortAllocatorTest, TestDisableUdpTurn) {
+TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) {
   turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
   AddInterface(kClientAddr);
   ResetWithStunServerAndNat(kStunAddr);
@@ -787,7 +825,7 @@ TEST_F(PortAllocatorTest, TestDisableUdpTurn) {
 
 // Test that we can get OnCandidatesAllocationDone callback when all the ports
 // are disabled.
-TEST_F(PortAllocatorTest, TestDisableAllPorts) {
+TEST_F(BasicPortAllocatorTest, TestDisableAllPorts) {
   AddInterface(kClientAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->set_flags(cricket::PORTALLOCATOR_DISABLE_UDP |
@@ -801,7 +839,7 @@ TEST_F(PortAllocatorTest, TestDisableAllPorts) {
 }
 
 // Test that we don't crash or malfunction if we can't create UDP sockets.
-TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpSockets) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSockets) {
   AddInterface(kClientAddr);
   fss_->set_udp_sockets_enabled(false);
   EXPECT_TRUE(CreateSession(1));
@@ -809,25 +847,29 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpSockets) {
   ASSERT_EQ_WAIT(5U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_EQ(2U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpExtAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpExtAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp", kRelayTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp",
+               kRelayTcpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[3],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[4],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP,
-      "relay", "ssltcp", kRelaySslTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "ssltcp",
+               kRelaySslTcpIntAddr);
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
-#endif // if !defined(ADDRESS_SANITIZER)
+#endif  // if !defined(ADDRESS_SANITIZER)
 
 // Test that we don't crash or malfunction if we can't create UDP sockets or
 // listen on TCP sockets. We still give out a local TCP address, since
 // apparently this is needed for the remote side to accept our connection.
-TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpSocketsNoTcpListen) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSocketsNoTcpListen) {
   AddInterface(kClientAddr);
   fss_->set_udp_sockets_enabled(false);
   fss_->set_tcp_listen_enabled(false);
@@ -835,35 +877,34 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpSocketsNoTcpListen) {
   session_->StartGettingPorts();
   ASSERT_EQ_WAIT(5U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_EQ(2U, ports_.size());
-  EXPECT_PRED5(CheckCandidate, candidates_[0],
-      1, "relay", "udp", kRelayUdpIntAddr);
-  EXPECT_PRED5(CheckCandidate, candidates_[1],
-      1, "relay", "udp", kRelayUdpExtAddr);
-  EXPECT_PRED5(CheckCandidate, candidates_[2],
-      1, "relay", "tcp", kRelayTcpIntAddr);
-  EXPECT_PRED5(CheckCandidate, candidates_[3],
-      1, "local", "tcp", kClientAddr);
-  EXPECT_PRED5(CheckCandidate, candidates_[4],
-      1, "relay", "ssltcp", kRelaySslTcpIntAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[0], 1, "relay", "udp",
+               kRelayUdpIntAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[1], 1, "relay", "udp",
+               kRelayUdpExtAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[2], 1, "relay", "tcp",
+               kRelayTcpIntAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[3], 1, "local", "tcp", kClientAddr);
+  EXPECT_PRED5(CheckCandidate, candidates_[4], 1, "relay", "ssltcp",
+               kRelaySslTcpIntAddr);
   EXPECT_TRUE(candidate_allocation_done_);
 }
 
 // Test that we don't crash or malfunction if we can't create any sockets.
-// TODO: Find a way to exit early here.
-TEST_F(PortAllocatorTest, TestGetAllPortsNoSockets) {
+// TODO(deadbeef): Find a way to exit early here.
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoSockets) {
   AddInterface(kClientAddr);
   fss_->set_tcp_sockets_enabled(false);
   fss_->set_udp_sockets_enabled(false);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   WAIT(candidates_.size() > 0, 2000);
-  // TODO - Check candidate_allocation_done signal.
+  // TODO(deadbeef): Check candidate_allocation_done signal.
   // In case of Relay, ports creation will succeed but sockets will fail.
   // There is no error reporting from RelayEntry to handle this failure.
 }
 
 // Testing STUN timeout.
-TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
   fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kClientAddr);
   AddInterface(kClientAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -871,27 +912,32 @@ TEST_F(PortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
   EXPECT_EQ_WAIT(2U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_EQ(2U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientAddr);
   // RelayPort connection timeout is 3sec. TCP connection with RelayServer
   // will be tried after 3 seconds.
   EXPECT_EQ_WAIT(6U, candidates_.size(), 4000);
   EXPECT_EQ(3U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[3],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp", kRelayTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "tcp",
+               kRelayTcpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[4],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "ssltcp",
-      kRelaySslTcpIntAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "ssltcp",
+               kRelaySslTcpIntAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[5],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp", kRelayUdpExtAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               kRelayUdpExtAddr);
   // Stun Timeout is 9sec.
   EXPECT_TRUE_WAIT(candidate_allocation_done_, 9000);
 }
 
-TEST_F(PortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) {
+TEST_F(BasicPortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) {
   AddInterface(kClientAddr);
   AddInterface(kClientAddr2);
   // Allocating only host UDP ports. This is done purely for testing
@@ -909,14 +955,14 @@ TEST_F(PortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) {
 }
 
 // Test to verify ICE restart process.
-TEST_F(PortAllocatorTest, TestGetAllPortsRestarts) {
+TEST_F(BasicPortAllocatorTest, TestGetAllPortsRestarts) {
   AddInterface(kClientAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   EXPECT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_EQ(4U, ports_.size());
   EXPECT_TRUE(candidate_allocation_done_);
-  // TODO - Extend this to verify ICE restart.
+  // TODO(deadbeef): Extend this to verify ICE restart.
 }
 
 // Test ICE candidate filter mechanism with options Relay/Host/Reflexive.
@@ -924,7 +970,7 @@ TEST_F(PortAllocatorTest, TestGetAllPortsRestarts) {
 // relay (i.e. IceTransportsType is relay), the raddr is an empty
 // address with the correct family. This is to prevent any local
 // reflective address leakage in the sdp line.
-TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) {
+TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithRelayOnly) {
   AddInterface(kClientAddr);
   // GTURN is not configured here.
   ResetWithTurnServersNoNat(kTurnUdpIntAddr, rtc::SocketAddress());
@@ -932,11 +978,8 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) {
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
-  EXPECT_PRED5(CheckCandidate,
-               candidates_[0],
-               cricket::ICE_CANDIDATE_COMPONENT_RTP,
-               "relay",
-               "udp",
+  EXPECT_PRED5(CheckCandidate, candidates_[0],
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
                rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
 
   EXPECT_EQ(1U, candidates_.size());
@@ -949,22 +992,22 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithRelayOnly) {
   }
 }
 
-TEST_F(PortAllocatorTest, TestCandidateFilterWithHostOnly) {
+TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithHostOnly) {
   AddInterface(kClientAddr);
   allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   allocator().set_candidate_filter(cricket::CF_HOST);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
-  EXPECT_EQ(2U, candidates_.size()); // Host UDP/TCP candidates only.
-  EXPECT_EQ(2U, ports_.size()); // UDP/TCP ports only.
+  EXPECT_EQ(2U, candidates_.size());  // Host UDP/TCP candidates only.
+  EXPECT_EQ(2U, ports_.size());       // UDP/TCP ports only.
   for (size_t i = 0; i < candidates_.size(); ++i) {
     EXPECT_EQ(std::string(cricket::LOCAL_PORT_TYPE), candidates_[i].type());
   }
 }
 
 // Host is behind the NAT.
-TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
+TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
   AddInterface(kPrivateAddr);
   ResetWithStunServerAndNat(kStunAddr);
 
@@ -975,8 +1018,8 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
   // Host is behind NAT, no private address will be exposed. Hence only UDP
   // port with STUN candidate will be sent outside.
-  EXPECT_EQ(1U, candidates_.size()); // Only STUN candidate.
-  EXPECT_EQ(1U, ports_.size());  // Only UDP port will be in ready state.
+  EXPECT_EQ(1U, candidates_.size());  // Only STUN candidate.
+  EXPECT_EQ(1U, ports_.size());       // Only UDP port will be in ready state.
   for (size_t i = 0; i < candidates_.size(); ++i) {
     EXPECT_EQ(std::string(cricket::STUN_PORT_TYPE), candidates_[i].type());
     EXPECT_EQ(
@@ -986,7 +1029,7 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
 }
 
 // Host is not behind the NAT.
-TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
+TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
   AddInterface(kClientAddr);
   allocator().set_flags(cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
   allocator().set_candidate_filter(cricket::CF_REFLEXIVE);
@@ -994,7 +1037,7 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
   session_->StartGettingPorts();
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
   // Host has a public address, both UDP and TCP candidates will be exposed.
-  EXPECT_EQ(2U, candidates_.size()); // Local UDP + TCP candidate.
+  EXPECT_EQ(2U, candidates_.size());  // Local UDP + TCP candidate.
   EXPECT_EQ(2U, ports_.size());  //  UDP and TCP ports will be in ready state.
   for (size_t i = 0; i < candidates_.size(); ++i) {
     EXPECT_EQ(std::string(cricket::LOCAL_PORT_TYPE), candidates_[i].type());
@@ -1002,17 +1045,20 @@ TEST_F(PortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
 }
 
 // Test that we get the same ufrag and pwd for all candidates.
-TEST_F(PortAllocatorTest, TestEnableSharedUfrag) {
+TEST_F(BasicPortAllocatorTest, TestEnableSharedUfrag) {
   AddInterface(kClientAddr);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
   session_->StartGettingPorts();
   ASSERT_EQ_WAIT(7U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[5],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientAddr);
   EXPECT_EQ(4U, ports_.size());
   EXPECT_EQ(kIceUfrag0, candidates_[0].username());
   EXPECT_EQ(kIceUfrag0, candidates_[1].username());
@@ -1026,7 +1072,7 @@ TEST_F(PortAllocatorTest, TestEnableSharedUfrag) {
 // is allocated for udp and stun. Also verify there is only one candidate
 // (local) if stun candidate is same as local candidate, which will be the case
 // in a public network like the below test.
-TEST_F(PortAllocatorTest, TestSharedSocketWithoutNat) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNat) {
   AddInterface(kClientAddr);
   allocator_->set_flags(allocator().flags() |
                         cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET);
@@ -1035,14 +1081,15 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithoutNat) {
   ASSERT_EQ_WAIT(6U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_EQ(3U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
 }
 
 // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled only one port
 // is allocated for udp and stun. In this test we should expect both stun and
 // local candidates as client behind a nat.
-TEST_F(PortAllocatorTest, TestSharedSocketWithNat) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNat) {
   AddInterface(kClientAddr);
   ResetWithStunServerAndNat(kStunAddr);
 
@@ -1053,16 +1100,17 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNat) {
   ASSERT_EQ_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout);
   ASSERT_EQ(2U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
-      rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
   EXPECT_EQ(3U, candidates_.size());
 }
 
 // Test TURN port in shared socket mode with UDP and TCP TURN server addresses.
-TEST_F(PortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
   turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
   AddInterface(kClientAddr);
   allocator_.reset(new cricket::BasicPortAllocator(&network_manager_));
@@ -1080,20 +1128,21 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
   ASSERT_EQ_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout);
   ASSERT_EQ(3U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
-      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
-      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
   EXPECT_EQ(3U, candidates_.size());
 }
 
 // Testing DNS resolve for the TURN server, this will test AllocationSequence
 // handling the unresolved address signal from TurnPort.
-TEST_F(PortAllocatorTest, TestSharedSocketWithServerAddressResolve) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithServerAddressResolve) {
   turn_server_.AddInternalSocket(rtc::SocketAddress("127.0.0.1", 3478),
                                  cricket::PROTO_UDP);
   AddInterface(kClientAddr);
@@ -1119,7 +1168,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithServerAddressResolve) {
 // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled only one port
 // is allocated for udp/stun/turn. In this test we should expect all local,
 // stun and turn candidates.
-TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
   AddInterface(kClientAddr);
   ResetWithStunServerAndNat(kStunAddr);
 
@@ -1135,13 +1184,14 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
   ASSERT_EQ_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout);
   ASSERT_EQ(2U, ports_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
-      rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
-      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
   EXPECT_EQ(3U, candidates_.size());
   // Local port will be created first and then TURN port.
@@ -1152,7 +1202,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
 // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled and the TURN
 // server is also used as the STUN server, we should get 'local', 'stun', and
 // 'relay' candidates.
-TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
   AddInterface(kClientAddr);
   // Use an empty SocketAddress to add a NAT without STUN server.
   ResetWithStunServerAndNat(SocketAddress());
@@ -1172,13 +1222,14 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
 
   ASSERT_EQ_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout);
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
-      rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "stun", "udp",
+               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
-      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "relay", "udp",
+               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
   EXPECT_EQ(candidates_[2].related_address(), candidates_[1].address());
 
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
@@ -1191,7 +1242,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
 // Test that when only a TCP TURN server is available, we do NOT use it as
 // a UDP STUN server, as this could leak our IP address. Thus we should only
 // expect two ports, a UDPPort and TurnPort.
-TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnTcpOnly) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnTcpOnly) {
   turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
   AddInterface(kClientAddr);
   ResetWithStunServerAndNat(rtc::SocketAddress());
@@ -1223,7 +1274,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnTcpOnly) {
 // 'relay' candidates.
 // TODO(deadbeef): Remove this test when support for non-shared socket mode
 // is removed.
-TEST_F(PortAllocatorTest, TestNonSharedSocketWithNatUsingTurnAsStun) {
+TEST_F(BasicPortAllocatorTest, TestNonSharedSocketWithNatUsingTurnAsStun) {
   AddInterface(kClientAddr);
   // Use an empty SocketAddress to add a NAT without STUN server.
   ResetWithStunServerAndNat(SocketAddress());
@@ -1259,7 +1310,7 @@ TEST_F(PortAllocatorTest, TestNonSharedSocketWithNatUsingTurnAsStun) {
 
 // Test that even when both a STUN and TURN server are configured, the TURN
 // server is used as a STUN server and we get a 'stun' candidate.
-TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) {
   AddInterface(kClientAddr);
   // Configure with STUN server but destroy it, so we can ensure that it's
   // the TURN server actually being used as a STUN server.
@@ -1293,7 +1344,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) {
 // This test verifies when PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is enabled
 // and fail to generate STUN candidate, local UDP candidate is generated
 // properly.
-TEST_F(PortAllocatorTest, TestSharedSocketNoUdpAllowed) {
+TEST_F(BasicPortAllocatorTest, TestSharedSocketNoUdpAllowed) {
   allocator().set_flags(allocator().flags() |
                         cricket::PORTALLOCATOR_DISABLE_RELAY |
                         cricket::PORTALLOCATOR_DISABLE_TCP |
@@ -1305,7 +1356,8 @@ TEST_F(PortAllocatorTest, TestSharedSocketNoUdpAllowed) {
   ASSERT_EQ_WAIT(1U, ports_.size(), kDefaultAllocationTimeout);
   EXPECT_EQ(1U, candidates_.size());
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp", kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   // STUN timeout is 9sec. We need to wait to get candidate done signal.
   EXPECT_TRUE_WAIT(candidate_allocation_done_, 10000);
   EXPECT_EQ(1U, candidates_.size());
@@ -1314,7 +1366,7 @@ TEST_F(PortAllocatorTest, TestSharedSocketNoUdpAllowed) {
 // Test that when the NetworkManager doesn't have permission to enumerate
 // adapters, the PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION is specified
 // automatically.
-TEST_F(PortAllocatorTest, TestNetworkPermissionBlocked) {
+TEST_F(BasicPortAllocatorTest, TestNetworkPermissionBlocked) {
   network_manager_.set_default_local_addresses(kPrivateAddr.ipaddr(),
                                                rtc::IPAddress());
   network_manager_.set_enumeration_permission(
@@ -1334,12 +1386,12 @@ TEST_F(PortAllocatorTest, TestNetworkPermissionBlocked) {
   EXPECT_PRED5(CheckCandidate, candidates_[0],
                cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
                kPrivateAddr);
-  EXPECT_TRUE((session_->flags() &
-               cricket::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) != 0);
+  EXPECT_NE(0U, session_->flags() &
+                    cricket::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
 }
 
 // This test verifies allocator can use IPv6 addresses along with IPv4.
-TEST_F(PortAllocatorTest, TestEnableIPv6Addresses) {
+TEST_F(BasicPortAllocatorTest, TestEnableIPv6Addresses) {
   allocator().set_flags(allocator().flags() |
                         cricket::PORTALLOCATOR_DISABLE_RELAY |
                         cricket::PORTALLOCATOR_ENABLE_IPV6 |
@@ -1353,21 +1405,21 @@ TEST_F(PortAllocatorTest, TestEnableIPv6Addresses) {
   EXPECT_EQ(4U, candidates_.size());
   EXPECT_TRUE_WAIT(candidate_allocation_done_, kDefaultAllocationTimeout);
   EXPECT_PRED5(CheckCandidate, candidates_[0],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
-      kClientIPv6Addr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientIPv6Addr);
   EXPECT_PRED5(CheckCandidate, candidates_[1],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
-      kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "udp",
+               kClientAddr);
   EXPECT_PRED5(CheckCandidate, candidates_[2],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
-      kClientIPv6Addr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientIPv6Addr);
   EXPECT_PRED5(CheckCandidate, candidates_[3],
-      cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
-      kClientAddr);
+               cricket::ICE_CANDIDATE_COMPONENT_RTP, "local", "tcp",
+               kClientAddr);
   EXPECT_EQ(4U, candidates_.size());
 }
 
-TEST_F(PortAllocatorTest, TestStopGettingPorts) {
+TEST_F(BasicPortAllocatorTest, TestStopGettingPorts) {
   AddInterface(kClientAddr);
   allocator_->set_step_delay(cricket::kDefaultStepDelay);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -1388,7 +1440,7 @@ TEST_F(PortAllocatorTest, TestStopGettingPorts) {
   EXPECT_EQ(0U, ports_.size());
 }
 
-TEST_F(PortAllocatorTest, TestClearGettingPorts) {
+TEST_F(BasicPortAllocatorTest, TestClearGettingPorts) {
   AddInterface(kClientAddr);
   allocator_->set_step_delay(cricket::kDefaultStepDelay);
   EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
@@ -1408,3 +1460,41 @@ TEST_F(PortAllocatorTest, TestClearGettingPorts) {
   ASSERT_EQ_WAIT(2U, candidates_.size(), 1000);
   EXPECT_EQ(2U, ports_.size());
 }
+
+// Test that the ports and candidates are updated with new ufrag/pwd/etc. when
+// a pooled session is taken out of the pool.
+TEST_F(BasicPortAllocatorTest, TestTransportInformationUpdated) {
+  AddInterface(kClientAddr);
+  int pool_size = 1;
+  allocator_->SetConfiguration(allocator_->stun_servers(),
+                               allocator_->turn_servers(), pool_size);
+  const cricket::PortAllocatorSession* peeked_session =
+      allocator_->GetPooledSession();
+  ASSERT_NE(nullptr, peeked_session);
+  EXPECT_EQ_WAIT(true, peeked_session->CandidatesAllocationDone(),
+                 kDefaultAllocationTimeout);
+  // Expect that when TakePooledSession is called,
+  // UpdateTransportInformationInternal will be called and the
+  // BasicPortAllocatorSession will update the ufrag/pwd of ports and
+  // candidates.
+  session_ =
+      allocator_->TakePooledSession(kContentName, 1, kIceUfrag0, kIcePwd0);
+  ASSERT_NE(nullptr, session_.get());
+  auto ready_ports = session_->ReadyPorts();
+  auto candidates = session_->ReadyCandidates();
+  EXPECT_FALSE(ready_ports.empty());
+  EXPECT_FALSE(candidates.empty());
+  for (const cricket::PortInterface* port_interface : ready_ports) {
+    const cricket::Port* port =
+        static_cast<const cricket::Port*>(port_interface);
+    EXPECT_EQ(kContentName, port->content_name());
+    EXPECT_EQ(1, port->component());
+    EXPECT_EQ(kIceUfrag0, port->username_fragment());
+    EXPECT_EQ(kIcePwd0, port->password());
+  }
+  for (const cricket::Candidate& candidate : candidates) {
+    EXPECT_EQ(1, candidate.component());
+    EXPECT_EQ(kIceUfrag0, candidate.username());
+    EXPECT_EQ(kIcePwd0, candidate.password());
+  }
+}
diff --git a/chromium/third_party/webrtc/p2p/p2p.gyp b/chromium/third_party/webrtc/p2p/p2p.gyp
index 5c9a575f8f2..7ec1814bc6b 100644
--- a/chromium/third_party/webrtc/p2p/p2p.gyp
+++ b/chromium/third_party/webrtc/p2p/p2p.gyp
@@ -16,9 +16,6 @@
         '<(webrtc_root)/base/base.gyp:rtc_base',
         '<(webrtc_root)/common.gyp:webrtc_common',
       ],
-      'cflags_cc!': [
-        '-Wnon-virtual-dtor',
-      ],
       'sources': [
         'base/asyncstuntcpsocket.cc',
         'base/asyncstuntcpsocket.h',
@@ -84,9 +81,6 @@
         'client/socketmonitor.h',
       ],
       'direct_dependent_settings': {
-        'cflags_cc!': [
-          '-Wnon-virtual-dtor',
-        ],
         'defines': [
           'FEATURE_ENABLE_VOICEMAIL',
         ],
@@ -107,6 +101,8 @@
             'quic/quicconnectionhelper.h',
             'quic/quicsession.cc',
             'quic/quicsession.h',
+            'quic/quictransport.cc',
+            'quic/quictransport.h',
             'quic/quictransportchannel.cc',
             'quic/quictransportchannel.h',
             'quic/reliablequicstream.cc',
@@ -125,9 +121,6 @@
         '<(webrtc_root)/base/base.gyp:rtc_base',
         '<(webrtc_root)/common.gyp:webrtc_common',
       ],
-      'cflags_cc!': [
-        '-Wnon-virtual-dtor',
-      ],
       'sources': [
         'stunprober/stunprober.cc',
       ],
@@ -139,9 +132,6 @@
         'libstunprober',
         'rtc_p2p'
       ],
-      'cflags_cc!': [
-        '-Wnon-virtual-dtor',
-      ],
       'sources': [
         'stunprober/main.cc',
       ],
@@ -156,9 +146,11 @@
           'direct_dependent_settings': {
             'sources': [
               'base/dtlstransportchannel_unittest.cc',
+              'base/fakeportallocator.h',
               'base/faketransportcontroller.h',
               'base/p2ptransportchannel_unittest.cc',
               'base/port_unittest.cc',
+              'base/portallocator_unittest.cc',
               'base/pseudotcp_unittest.cc',
               'base/relayport_unittest.cc',
               'base/relayserver_unittest.cc',
@@ -172,9 +164,9 @@
               'base/transport_unittest.cc',
               'base/transportcontroller_unittest.cc',
               'base/transportdescriptionfactory_unittest.cc',
+              'base/tcpport_unittest.cc',
               'base/turnport_unittest.cc',
-              'client/fakeportallocator.h',
-              'client/portallocator_unittest.cc',
+              'client/basicportallocator_unittest.cc',
               'stunprober/stunprober_unittest.cc',
             ],
             'conditions': [
@@ -182,6 +174,7 @@
                 'sources': [
                   'quic/quicconnectionhelper_unittest.cc',
                   'quic/quicsession_unittest.cc',
+		  'quic/quictransport_unittest.cc',
                   'quic/quictransportchannel_unittest.cc',
                   'quic/reliablequicstream_unittest.cc',
                 ],
diff --git a/chromium/third_party/webrtc/p2p/quic/quicconnectionhelper_unittest.cc b/chromium/third_party/webrtc/p2p/quic/quicconnectionhelper_unittest.cc
index 1a7313c2457..5f16b52c62e 100644
--- a/chromium/third_party/webrtc/p2p/quic/quicconnectionhelper_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/quic/quicconnectionhelper_unittest.cc
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/p2p/quic/quicconnectionhelper.h"
 
 #include "net/quic/quic_time.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 
 using cricket::QuicAlarm;
 using cricket::QuicConnectionHelper;
@@ -52,10 +53,7 @@ class MockAlarmDelegate : public QuicAlarm::Delegate {
  public:
   MockAlarmDelegate() : fired_(false) {}
 
-  QuicTime OnAlarm() override {
-    fired_ = true;
-    return QuicTime::Zero();
-  }
+  void OnAlarm() override { fired_ = true; }
 
   bool fired() const { return fired_; }
   void Clear() { fired_ = false; }
@@ -89,7 +87,7 @@ class QuicAlarmTest : public ::testing::Test {
   // Used for setting clock time relative to alarm.
   MockClock clock_;
 
-  rtc::scoped_ptr<QuicAlarm> alarm_;
+  std::unique_ptr<QuicAlarm> alarm_;
 };
 
 // Test that the alarm is fired.
diff --git a/chromium/third_party/webrtc/p2p/quic/quicsession.cc b/chromium/third_party/webrtc/p2p/quic/quicsession.cc
index 281af5e3a74..65ed1357545 100644
--- a/chromium/third_party/webrtc/p2p/quic/quicsession.cc
+++ b/chromium/third_party/webrtc/p2p/quic/quicsession.cc
@@ -20,7 +20,11 @@
 
 namespace cricket {
 
-QuicSession::QuicSession(rtc::scoped_ptr<net::QuicConnection> connection,
+// Default priority for incoming QUIC streams.
+// TODO(mikescarlett): Determine if this value is correct.
+static const net::SpdyPriority kDefaultPriority = 3;
+
+QuicSession::QuicSession(std::unique_ptr<net::QuicConnection> connection,
                          const net::QuicConfig& config)
     : net::QuicSession(connection.release(), config) {}
 
@@ -46,7 +50,7 @@ void QuicSession::SetCryptoStream(net::QuicCryptoStream* crypto_stream) {
 bool QuicSession::ExportKeyingMaterial(base::StringPiece label,
                                        base::StringPiece context,
                                        size_t result_len,
-                                       string* result) {
+                                       std::string* result) {
   return crypto_stream_->ExportKeyingMaterial(label, context, result_len,
                                               result);
 }
@@ -62,9 +66,19 @@ void QuicSession::OnCryptoHandshakeEvent(CryptoHandshakeEvent event) {
   }
 }
 
+void QuicSession::CloseStream(net::QuicStreamId stream_id) {
+  if (IsClosedStream(stream_id)) {
+    // When CloseStream has been called recursively (via
+    // ReliableQuicStream::OnClose), the stream is already closed so return.
+    return;
+  }
+  write_blocked_streams()->UnregisterStream(stream_id);
+  net::QuicSession::CloseStream(stream_id);
+}
+
 ReliableQuicStream* QuicSession::CreateIncomingDynamicStream(
     net::QuicStreamId id) {
-  ReliableQuicStream* stream = CreateDataStream(id);
+  ReliableQuicStream* stream = CreateDataStream(id, kDefaultPriority);
   if (stream) {
     SignalIncomingStream(stream);
   }
@@ -73,32 +87,39 @@ ReliableQuicStream* QuicSession::CreateIncomingDynamicStream(
 
 ReliableQuicStream* QuicSession::CreateOutgoingDynamicStream(
     net::SpdyPriority priority) {
-  ReliableQuicStream* stream = CreateDataStream(GetNextOutgoingStreamId());
-  if (stream) {
-    ActivateStream(stream);
-  }
-  return stream;
+  return CreateDataStream(GetNextOutgoingStreamId(), priority);
 }
 
-ReliableQuicStream* QuicSession::CreateDataStream(net::QuicStreamId id) {
+ReliableQuicStream* QuicSession::CreateDataStream(net::QuicStreamId id,
+                                                  net::SpdyPriority priority) {
   if (crypto_stream_ == nullptr || !crypto_stream_->encryption_established()) {
     // Encryption not active so no stream created
     return nullptr;
   }
-  return new ReliableQuicStream(id, this);
+  ReliableQuicStream* stream = new ReliableQuicStream(id, this);
+  if (stream) {
+    // Make QuicSession take ownership of the stream.
+    ActivateStream(stream);
+    // Register the stream to the QuicWriteBlockedList. |priority| is clamped
+    // between 0 and 7, with 0 being the highest priority and 7 the lowest
+    // priority.
+    write_blocked_streams()->RegisterStream(stream->id(), priority);
+  }
+  return stream;
 }
 
 void QuicSession::OnConnectionClosed(net::QuicErrorCode error,
+                                     const std::string& error_details,
                                      net::ConnectionCloseSource source) {
-  net::QuicSession::OnConnectionClosed(error, source);
+  net::QuicSession::OnConnectionClosed(error, error_details, source);
   SignalConnectionClosed(error,
                          source == net::ConnectionCloseSource::FROM_PEER);
 }
 
 bool QuicSession::OnReadPacket(const char* data, size_t data_len) {
-  net::QuicEncryptedPacket packet(data, data_len);
-  connection()->ProcessUdpPacket(connection()->self_address(),
-                                 connection()->peer_address(), packet);
+  net::QuicReceivedPacket packet(data, data_len, clock_.Now());
+  ProcessUdpPacket(connection()->self_address(), connection()->peer_address(),
+                   packet);
   return true;
 }
 
diff --git a/chromium/third_party/webrtc/p2p/quic/quicsession.h b/chromium/third_party/webrtc/p2p/quic/quicsession.h
index e0ea296b59c..f8b27e055e5 100644
--- a/chromium/third_party/webrtc/p2p/quic/quicsession.h
+++ b/chromium/third_party/webrtc/p2p/quic/quicsession.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_P2P_QUIC_QUICSESSION_H_
 #define WEBRTC_P2P_QUIC_QUICSESSION_H_
 
+#include <memory>
 #include <string>
 
 #include "net/quic/quic_crypto_client_stream.h"
@@ -29,7 +30,7 @@ namespace cricket {
 // reading/writing of data using QUIC packets.
 class QuicSession : public net::QuicSession, public sigslot::has_slots<> {
  public:
-  QuicSession(rtc::scoped_ptr<net::QuicConnection> connection,
+  QuicSession(std::unique_ptr<net::QuicConnection> connection,
               const net::QuicConfig& config);
   ~QuicSession() override;
 
@@ -48,16 +49,18 @@ class QuicSession : public net::QuicSession, public sigslot::has_slots<> {
 
   // QuicSession optional overrides.
   void OnCryptoHandshakeEvent(CryptoHandshakeEvent event) override;
+  void CloseStream(net::QuicStreamId stream_id) override;
 
   // QuicConnectionVisitorInterface overrides.
   void OnConnectionClosed(net::QuicErrorCode error,
+                          const std::string& error_details,
                           net::ConnectionCloseSource source) override;
 
   // Exports keying material for SRTP.
   bool ExportKeyingMaterial(base::StringPiece label,
                             base::StringPiece context,
                             size_t result_len,
-                            string* result);
+                            std::string* result);
 
   // Decrypts an incoming QUIC packet to a data stream.
   bool OnReadPacket(const char* data, size_t data_len);
@@ -79,10 +82,12 @@ class QuicSession : public net::QuicSession, public sigslot::has_slots<> {
   ReliableQuicStream* CreateIncomingDynamicStream(
       net::QuicStreamId id) override;
 
-  virtual ReliableQuicStream* CreateDataStream(net::QuicStreamId id);
+  virtual ReliableQuicStream* CreateDataStream(net::QuicStreamId id,
+                                               net::SpdyPriority priority);
 
  private:
-  rtc::scoped_ptr<net::QuicCryptoStream> crypto_stream_;
+  std::unique_ptr<net::QuicCryptoStream> crypto_stream_;
+  net::QuicClock clock_;  // For recording packet receipt time
 
   RTC_DISALLOW_COPY_AND_ASSIGN(QuicSession);
 };
diff --git a/chromium/third_party/webrtc/p2p/quic/quicsession_unittest.cc b/chromium/third_party/webrtc/p2p/quic/quicsession_unittest.cc
index 2e2d6ae0a6a..2f3aaae3322 100644
--- a/chromium/third_party/webrtc/p2p/quic/quicsession_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/quic/quicsession_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/p2p/quic/quicsession.h"
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -36,6 +37,7 @@ using net::ProofVerifyContext;
 using net::ProofVerifyDetails;
 using net::QuicByteCount;
 using net::QuicClock;
+using net::QuicCompressedCertsCache;
 using net::QuicConfig;
 using net::QuicConnection;
 using net::QuicCryptoClientConfig;
@@ -111,13 +113,16 @@ class FakeProofVerifier : public net::ProofVerifier {
   // ProofVerifier override
   net::QuicAsyncStatus VerifyProof(
       const std::string& hostname,
+      const uint16_t port,
       const std::string& server_config,
+      net::QuicVersion quic_version,
+      base::StringPiece chlo_hash,
       const std::vector<std::string>& certs,
       const std::string& cert_sct,
       const std::string& signature,
-      const net::ProofVerifyContext* verify_context,
+      const ProofVerifyContext* context,
       std::string* error_details,
-      scoped_ptr<net::ProofVerifyDetails>* verify_details,
+      std::unique_ptr<net::ProofVerifyDetails>* verify_details,
       net::ProofVerifierCallback* callback) override {
     return success_ ? net::QUIC_SUCCESS : net::QUIC_FAILURE;
   }
@@ -179,9 +184,9 @@ class FakeQuicPacketWriter : public QuicPacketWriter {
 // Wrapper for QuicSession and transport channel that stores incoming data.
 class QuicSessionForTest : public QuicSession {
  public:
-  QuicSessionForTest(rtc::scoped_ptr<net::QuicConnection> connection,
+  QuicSessionForTest(std::unique_ptr<net::QuicConnection> connection,
                      const net::QuicConfig& config,
-                     rtc::scoped_ptr<FakeTransportChannel> channel)
+                     std::unique_ptr<FakeTransportChannel> channel)
       : QuicSession(std::move(connection), config),
         channel_(std::move(channel)) {
     channel_->SignalReadPacket.connect(
@@ -219,7 +224,7 @@ class QuicSessionForTest : public QuicSession {
 
  private:
   // Transports QUIC packets to/from peer.
-  rtc::scoped_ptr<FakeTransportChannel> channel_;
+  std::unique_ptr<FakeTransportChannel> channel_;
   // Stores data received by peer once it is sent from the other peer.
   std::string last_received_data_;
   // Handles incoming streams from sender.
@@ -230,13 +235,16 @@ class QuicSessionForTest : public QuicSession {
 class QuicSessionTest : public ::testing::Test,
                         public QuicCryptoClientStream::ProofHandler {
  public:
-  QuicSessionTest() : quic_helper_(rtc::Thread::Current()) {}
+  QuicSessionTest()
+      : quic_helper_(rtc::Thread::Current()),
+        quic_compressed_certs_cache_(
+            QuicCompressedCertsCache::kQuicCompressedCertsCacheSize) {}
 
   // Instantiates |client_peer_| and |server_peer_|.
   void CreateClientAndServerSessions();
 
-  rtc::scoped_ptr<QuicSessionForTest> CreateSession(
-      rtc::scoped_ptr<FakeTransportChannel> channel,
+  std::unique_ptr<QuicSessionForTest> CreateSession(
+      std::unique_ptr<FakeTransportChannel> channel,
       Perspective perspective);
 
   QuicCryptoClientStream* CreateCryptoClientStream(QuicSessionForTest* session,
@@ -244,7 +252,7 @@ class QuicSessionTest : public ::testing::Test,
   QuicCryptoServerStream* CreateCryptoServerStream(QuicSessionForTest* session,
                                                    bool handshake_success);
 
-  rtc::scoped_ptr<QuicConnection> CreateConnection(
+  std::unique_ptr<QuicConnection> CreateConnection(
       FakeTransportChannel* channel,
       Perspective perspective);
 
@@ -267,17 +275,18 @@ class QuicSessionTest : public ::testing::Test,
   QuicConnectionHelper quic_helper_;
   QuicConfig config_;
   QuicClock clock_;
+  QuicCompressedCertsCache quic_compressed_certs_cache_;
 
-  rtc::scoped_ptr<QuicSessionForTest> client_peer_;
-  rtc::scoped_ptr<QuicSessionForTest> server_peer_;
+  std::unique_ptr<QuicSessionForTest> client_peer_;
+  std::unique_ptr<QuicSessionForTest> server_peer_;
 };
 
 // Initializes "client peer" who begins crypto handshake and "server peer" who
 // establishes encryption with client.
 void QuicSessionTest::CreateClientAndServerSessions() {
-  rtc::scoped_ptr<FakeTransportChannel> channel1(
+  std::unique_ptr<FakeTransportChannel> channel1(
       new FakeTransportChannel("channel1", 0));
-  rtc::scoped_ptr<FakeTransportChannel> channel2(
+  std::unique_ptr<FakeTransportChannel> channel2(
       new FakeTransportChannel("channel2", 0));
 
   // Prevent channel1->OnReadPacket and channel2->OnReadPacket from calling
@@ -294,12 +303,12 @@ void QuicSessionTest::CreateClientAndServerSessions() {
   server_peer_ = CreateSession(std::move(channel2), Perspective::IS_SERVER);
 }
 
-rtc::scoped_ptr<QuicSessionForTest> QuicSessionTest::CreateSession(
-    rtc::scoped_ptr<FakeTransportChannel> channel,
+std::unique_ptr<QuicSessionForTest> QuicSessionTest::CreateSession(
+    std::unique_ptr<FakeTransportChannel> channel,
     Perspective perspective) {
-  rtc::scoped_ptr<QuicConnection> quic_connection =
+  std::unique_ptr<QuicConnection> quic_connection =
       CreateConnection(channel.get(), perspective);
-  return rtc::scoped_ptr<QuicSessionForTest>(new QuicSessionForTest(
+  return std::unique_ptr<QuicSessionForTest>(new QuicSessionForTest(
       std::move(quic_connection), config_, std::move(channel)));
 }
 
@@ -323,10 +332,13 @@ QuicCryptoServerStream* QuicSessionTest::CreateCryptoServerStream(
   QuicServerConfigProtobuf* primary_config = server_config->GenerateConfig(
       QuicRandom::GetInstance(), &clock_, options);
   server_config->AddConfig(primary_config, clock_.WallNow());
-  return new QuicCryptoServerStream(server_config, session);
+  bool use_stateless_rejects_if_peer_supported = false;
+  return new QuicCryptoServerStream(
+      server_config, &quic_compressed_certs_cache_,
+      use_stateless_rejects_if_peer_supported, session);
 }
 
-rtc::scoped_ptr<QuicConnection> QuicSessionTest::CreateConnection(
+std::unique_ptr<QuicConnection> QuicSessionTest::CreateConnection(
     FakeTransportChannel* channel,
     Perspective perspective) {
   FakeQuicPacketWriter* writer = new FakeQuicPacketWriter(channel);
@@ -334,7 +346,7 @@ rtc::scoped_ptr<QuicConnection> QuicSessionTest::CreateConnection(
   IPAddress ip(0, 0, 0, 0);
   bool owns_writer = true;
 
-  return rtc::scoped_ptr<QuicConnection>(new QuicConnection(
+  return std::unique_ptr<QuicConnection>(new QuicConnection(
       0, net::IPEndPoint(ip, 0), &quic_helper_, writer, owns_writer,
       perspective, net::QuicSupportedVersions()));
 }
@@ -357,8 +369,8 @@ void QuicSessionTest::TestStreamConnection(QuicSessionForTest* from_session,
   ASSERT_TRUE(from_session->IsEncryptionEstablished());
   ASSERT_TRUE(to_session->IsEncryptionEstablished());
 
-  string from_key;
-  string to_key;
+  std::string from_key;
+  std::string to_key;
 
   bool from_success = from_session->ExportKeyingMaterial(
       kExporterLabel, kExporterContext, kExporterContextLen, &from_key);
@@ -450,3 +462,17 @@ TEST_F(QuicSessionTest, CannotCreateDataStreamBeforeHandshake) {
   EXPECT_EQ(nullptr, server_peer_->CreateOutgoingDynamicStream(5));
   EXPECT_EQ(nullptr, client_peer_->CreateOutgoingDynamicStream(5));
 }
+
+// Test that closing a QUIC stream causes the QuicSession to remove it.
+TEST_F(QuicSessionTest, CloseQuicStream) {
+  CreateClientAndServerSessions();
+  StartHandshake(true, true);
+  ASSERT_TRUE_WAIT(client_peer_->IsCryptoHandshakeConfirmed() &&
+                       server_peer_->IsCryptoHandshakeConfirmed(),
+                   kTimeoutMs);
+  ReliableQuicStream* stream = client_peer_->CreateOutgoingDynamicStream(5);
+  ASSERT_NE(nullptr, stream);
+  EXPECT_FALSE(client_peer_->IsClosedStream(stream->id()));
+  stream->Close();
+  EXPECT_TRUE(client_peer_->IsClosedStream(stream->id()));
+}
diff --git a/chromium/third_party/webrtc/p2p/quic/quictransport.cc b/chromium/third_party/webrtc/p2p/quic/quictransport.cc
new file mode 100644
index 00000000000..51f9a2b0a3f
--- /dev/null
+++ b/chromium/third_party/webrtc/p2p/quic/quictransport.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/p2p/quic/quictransport.h"
+
+#include "webrtc/p2p/base/p2ptransportchannel.h"
+
+namespace cricket {
+
+QuicTransport::QuicTransport(
+    const std::string& name,
+    PortAllocator* allocator,
+    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate)
+    : Transport(name, allocator), local_certificate_(certificate) {}
+
+QuicTransport::~QuicTransport() {
+  DestroyAllChannels();
+}
+
+void QuicTransport::SetLocalCertificate(
+    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+  local_certificate_ = certificate;
+}
+bool QuicTransport::GetLocalCertificate(
+    rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
+  if (!local_certificate_) {
+    return false;
+  }
+  *certificate = local_certificate_;
+  return true;
+}
+
+bool QuicTransport::ApplyLocalTransportDescription(
+    TransportChannelImpl* channel,
+    std::string* error_desc) {
+  rtc::SSLFingerprint* local_fp =
+      local_description()->identity_fingerprint.get();
+  if (!VerifyCertificateFingerprint(local_certificate_.get(), local_fp,
+                                    error_desc)) {
+    return false;
+  }
+  if (!channel->SetLocalCertificate(local_certificate_)) {
+    return BadTransportDescription("Failed to set local identity.", error_desc);
+  }
+  return Transport::ApplyLocalTransportDescription(channel, error_desc);
+}
+
+bool QuicTransport::NegotiateTransportDescription(ContentAction action,
+                                                  std::string* error_desc) {
+  if (!local_description() || !remote_description()) {
+    const std::string msg =
+        "Local and Remote description must be set before "
+        "transport descriptions are negotiated";
+    return BadTransportDescription(msg, error_desc);
+  }
+  rtc::SSLFingerprint* local_fp =
+      local_description()->identity_fingerprint.get();
+  rtc::SSLFingerprint* remote_fp =
+      remote_description()->identity_fingerprint.get();
+  if (!local_fp || !remote_fp) {
+    return BadTransportDescription("Fingerprints must be supplied for QUIC.",
+                                   error_desc);
+  }
+  remote_fingerprint_.reset(new rtc::SSLFingerprint(*remote_fp));
+  if (!NegotiateRole(action, &local_role_, error_desc)) {
+    return false;
+  }
+  // Now run the negotiation for the Transport class.
+  return Transport::NegotiateTransportDescription(action, error_desc);
+}
+
+QuicTransportChannel* QuicTransport::CreateTransportChannel(int component) {
+  P2PTransportChannel* ice_channel =
+      new P2PTransportChannel(name(), component, port_allocator());
+  return new QuicTransportChannel(ice_channel);
+}
+
+void QuicTransport::DestroyTransportChannel(TransportChannelImpl* channel) {
+  delete channel;
+}
+
+bool QuicTransport::GetSslRole(rtc::SSLRole* ssl_role) const {
+  ASSERT(ssl_role != NULL);
+  *ssl_role = local_role_;
+  return true;
+}
+
+bool QuicTransport::ApplyNegotiatedTransportDescription(
+    TransportChannelImpl* channel,
+    std::string* error_desc) {
+  // Set ssl role and remote fingerprint. These are required for QUIC setup.
+  if (!channel->SetSslRole(local_role_)) {
+    return BadTransportDescription("Failed to set ssl role for the channel.",
+                                   error_desc);
+  }
+  // Apply remote fingerprint.
+  if (!channel->SetRemoteFingerprint(
+          remote_fingerprint_->algorithm,
+          reinterpret_cast<const uint8_t*>(remote_fingerprint_->digest.data()),
+          remote_fingerprint_->digest.size())) {
+    return BadTransportDescription("Failed to apply remote fingerprint.",
+                                   error_desc);
+  }
+  return Transport::ApplyNegotiatedTransportDescription(channel, error_desc);
+}
+
+}  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/quic/quictransport.h b/chromium/third_party/webrtc/p2p/quic/quictransport.h
new file mode 100644
index 00000000000..14bd13f3b2b
--- /dev/null
+++ b/chromium/third_party/webrtc/p2p/quic/quictransport.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_P2P_QUIC_QUICTRANSPORT_H_
+#define WEBRTC_P2P_QUIC_QUICTRANSPORT_H_
+
+#include <string>
+#include <map>
+#include <memory>
+
+#include "webrtc/p2p/base/transport.h"
+#include "webrtc/p2p/quic/quictransportchannel.h"
+
+namespace cricket {
+
+class P2PTransportChannel;
+class PortAllocator;
+
+// TODO(mikescarlett): Refactor to avoid code duplication with DtlsTransport.
+class QuicTransport : public Transport {
+ public:
+  QuicTransport(const std::string& name,
+                PortAllocator* allocator,
+                const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
+
+  ~QuicTransport() override;
+
+  // Transport overrides.
+  void SetLocalCertificate(
+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override;
+  bool GetLocalCertificate(
+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override;
+  bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) override {
+    return true;  // Not needed by QUIC
+  }
+  bool GetSslRole(rtc::SSLRole* ssl_role) const override;
+
+ protected:
+  // Transport overrides.
+  QuicTransportChannel* CreateTransportChannel(int component) override;
+  void DestroyTransportChannel(TransportChannelImpl* channel) override;
+  bool ApplyLocalTransportDescription(TransportChannelImpl* channel,
+                                      std::string* error_desc) override;
+  bool NegotiateTransportDescription(ContentAction action,
+                                     std::string* error_desc) override;
+  bool ApplyNegotiatedTransportDescription(TransportChannelImpl* channel,
+                                           std::string* error_desc) override;
+
+ private:
+  rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
+  rtc::SSLRole local_role_ = rtc::SSL_CLIENT;
+  std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint_;
+};
+
+}  // namespace cricket
+
+#endif  // WEBRTC_P2P_QUIC_QUICTRANSPORT_H_
diff --git a/chromium/third_party/webrtc/p2p/quic/quictransport_unittest.cc b/chromium/third_party/webrtc/p2p/quic/quictransport_unittest.cc
new file mode 100644
index 00000000000..1fd48f7ec41
--- /dev/null
+++ b/chromium/third_party/webrtc/p2p/quic/quictransport_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/p2p/quic/quictransport.h"
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/rtccertificate.h"
+#include "webrtc/base/sslidentity.h"
+
+using cricket::TransportChannelImpl;
+using cricket::QuicTransport;
+using cricket::Transport;
+using cricket::TransportDescription;
+
+static const char kIceUfrag1[] = "TESTICEUFRAG0001";
+static const char kIcePwd1[] = "TESTICEPWD00000000000001";
+
+static const char kIceUfrag2[] = "TESTICEUFRAG0002";
+static const char kIcePwd2[] = "TESTICEPWD00000000000002";
+
+static rtc::scoped_refptr<rtc::RTCCertificate> CreateCertificate(
+    std::string name) {
+  return rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
+      rtc::SSLIdentity::Generate(name, rtc::KT_DEFAULT)));
+}
+
+static std::unique_ptr<rtc::SSLFingerprint> CreateFingerprint(
+    rtc::RTCCertificate* cert) {
+  std::string digest_algorithm;
+  cert->ssl_certificate().GetSignatureDigestAlgorithm(&digest_algorithm);
+  return std::unique_ptr<rtc::SSLFingerprint>(
+      rtc::SSLFingerprint::Create(digest_algorithm, cert->identity()));
+}
+
+class QuicTransportTest : public testing::Test {
+ public:
+  QuicTransportTest() : transport_("testing", nullptr, nullptr) {}
+
+  void SetTransportDescription(cricket::ConnectionRole local_role,
+                               cricket::ConnectionRole remote_role,
+                               cricket::ContentAction local_action,
+                               cricket::ContentAction remote_action,
+                               rtc::SSLRole expected_ssl_role) {
+    TransportChannelImpl* channel = transport_.CreateChannel(1);
+    ASSERT_NE(nullptr, channel);
+
+    rtc::scoped_refptr<rtc::RTCCertificate> local_certificate(
+        CreateCertificate("local"));
+    ASSERT_NE(nullptr, local_certificate);
+    transport_.SetLocalCertificate(local_certificate);
+
+    std::unique_ptr<rtc::SSLFingerprint> local_fingerprint =
+        CreateFingerprint(local_certificate.get());
+    ASSERT_NE(nullptr, local_fingerprint);
+    TransportDescription local_desc(std::vector<std::string>(), kIceUfrag1,
+                                    kIcePwd1, cricket::ICEMODE_FULL, local_role,
+                                    local_fingerprint.get());
+    ASSERT_TRUE(transport_.SetLocalTransportDescription(local_desc,
+                                                        local_action, nullptr));
+    // The certificate is applied to QuicTransportChannel when the local
+    // description is set.
+    rtc::scoped_refptr<rtc::RTCCertificate> channel_local_certificate =
+        channel->GetLocalCertificate();
+    ASSERT_NE(nullptr, channel_local_certificate);
+    EXPECT_EQ(local_certificate, channel_local_certificate);
+    std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint =
+        CreateFingerprint(CreateCertificate("remote").get());
+    // NegotiateTransportDescription was not called yet. The SSL role should
+    // not be set and neither should the remote fingerprint.
+    std::unique_ptr<rtc::SSLRole> role(new rtc::SSLRole());
+    EXPECT_FALSE(channel->GetSslRole(role.get()));
+    // Setting the remote description should set the SSL role.
+    ASSERT_NE(nullptr, remote_fingerprint);
+    TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag2,
+                                     kIcePwd2, cricket::ICEMODE_FULL,
+                                     remote_role, remote_fingerprint.get());
+    ASSERT_TRUE(transport_.SetRemoteTransportDescription(
+        remote_desc, remote_action, nullptr));
+    ASSERT_TRUE(channel->GetSslRole(role.get()));
+    // SSL role should be client because the remote description is an ANSWER.
+    EXPECT_EQ(expected_ssl_role, *role);
+  }
+
+ protected:
+  QuicTransport transport_;
+};
+
+// Test setting the local certificate.
+TEST_F(QuicTransportTest, SetLocalCertificate) {
+  rtc::scoped_refptr<rtc::RTCCertificate> local_certificate(
+      CreateCertificate("local"));
+  ASSERT_NE(nullptr, local_certificate);
+  rtc::scoped_refptr<rtc::RTCCertificate> transport_local_certificate;
+  EXPECT_FALSE(transport_.GetLocalCertificate(&transport_local_certificate));
+  transport_.SetLocalCertificate(local_certificate);
+  ASSERT_TRUE(transport_.GetLocalCertificate(&transport_local_certificate));
+  ASSERT_NE(nullptr, transport_local_certificate);
+  EXPECT_EQ(local_certificate, transport_local_certificate);
+}
+
+// Test setting the ICE role.
+TEST_F(QuicTransportTest, SetIceRole) {
+  TransportChannelImpl* channel1 = transport_.CreateChannel(1);
+  ASSERT_NE(nullptr, channel1);
+  transport_.SetIceRole(cricket::ICEROLE_CONTROLLING);
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING, transport_.ice_role());
+  TransportChannelImpl* channel2 = transport_.CreateChannel(2);
+  ASSERT_NE(nullptr, channel2);
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel1->GetIceRole());
+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel2->GetIceRole());
+}
+
+// Test setting the ICE tie breaker.
+TEST_F(QuicTransportTest, SetIceTiebreaker) {
+  transport_.SetIceTiebreaker(1u);
+  EXPECT_EQ(1u, transport_.IceTiebreaker());
+}
+
+// Test setting the local and remote descriptions for a SSL client.
+TEST_F(QuicTransportTest, SetLocalAndRemoteTransportDescriptionClient) {
+  SetTransportDescription(cricket::CONNECTIONROLE_ACTPASS,
+                          cricket::CONNECTIONROLE_PASSIVE, cricket::CA_OFFER,
+                          cricket::CA_ANSWER, rtc::SSL_CLIENT);
+}
+
+// Test setting the local and remote descriptions for a SSL server.
+TEST_F(QuicTransportTest, SetLocalAndRemoteTransportDescriptionServer) {
+  SetTransportDescription(cricket::CONNECTIONROLE_ACTPASS,
+                          cricket::CONNECTIONROLE_ACTIVE, cricket::CA_OFFER,
+                          cricket::CA_ANSWER, rtc::SSL_SERVER);
+}
+
+// Test creation and destruction of channels.
+TEST_F(QuicTransportTest, CreateAndDestroyChannels) {
+  TransportChannelImpl* channel1 = transport_.CreateChannel(1);
+  ASSERT_NE(nullptr, channel1);
+  EXPECT_TRUE(transport_.HasChannel(1));
+  EXPECT_EQ(channel1, transport_.GetChannel(1));
+  TransportChannelImpl* channel2 = transport_.CreateChannel(2);
+  ASSERT_NE(nullptr, channel2);
+  EXPECT_TRUE(transport_.HasChannel(2));
+  EXPECT_EQ(channel2, transport_.GetChannel(2));
+  transport_.DestroyChannel(1);
+  EXPECT_FALSE(transport_.HasChannel(1));
+  EXPECT_EQ(nullptr, transport_.GetChannel(1));
+  transport_.DestroyChannel(2);
+  EXPECT_FALSE(transport_.HasChannel(2));
+  EXPECT_EQ(nullptr, transport_.GetChannel(2));
+}
diff --git a/chromium/third_party/webrtc/p2p/quic/quictransportchannel.cc b/chromium/third_party/webrtc/p2p/quic/quictransportchannel.cc
index cc0576ddb23..968faee7bd1 100644
--- a/chromium/third_party/webrtc/p2p/quic/quictransportchannel.cc
+++ b/chromium/third_party/webrtc/p2p/quic/quictransportchannel.cc
@@ -104,13 +104,16 @@ class InsecureProofVerifier : public net::ProofVerifier {
   // ProofVerifier override.
   net::QuicAsyncStatus VerifyProof(
       const std::string& hostname,
+      const uint16_t port,
       const std::string& server_config,
+      net::QuicVersion quic_version,
+      base::StringPiece chlo_hash,
       const std::vector<std::string>& certs,
       const std::string& cert_sct,
       const std::string& signature,
-      const net::ProofVerifyContext* verify_context,
+      const net::ProofVerifyContext* context,
       std::string* error_details,
-      scoped_ptr<net::ProofVerifyDetails>* verify_details,
+      std::unique_ptr<net::ProofVerifyDetails>* verify_details,
       net::ProofVerifierCallback* callback) override {
     LOG(LS_INFO) << "VerifyProof() ignoring credentials and returning success";
     return net::QUIC_SUCCESS;
@@ -271,7 +274,7 @@ int QuicTransportChannel::SendPacket(const char* data,
 //       |channel_| again.
 void QuicTransportChannel::OnWritableState(TransportChannel* channel) {
   ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   LOG_J(LS_VERBOSE, this)
       << "QuicTransportChannel: channel writable state changed to "
       << channel_->writable();
@@ -305,7 +308,7 @@ void QuicTransportChannel::OnWritableState(TransportChannel* channel) {
 
 void QuicTransportChannel::OnReceivingState(TransportChannel* channel) {
   ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   LOG_J(LS_VERBOSE, this)
       << "QuicTransportChannel: channel receiving state changed to "
       << channel_->receiving();
@@ -321,7 +324,7 @@ void QuicTransportChannel::OnReadPacket(TransportChannel* channel,
                                         const rtc::PacketTime& packet_time,
                                         int flags) {
   ASSERT(rtc::Thread::Current() == worker_thread_);
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   ASSERT(flags == 0);
 
   switch (quic_state_) {
@@ -368,24 +371,24 @@ void QuicTransportChannel::OnReadyToSend(TransportChannel* channel) {
 }
 
 void QuicTransportChannel::OnGatheringState(TransportChannelImpl* channel) {
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   SignalGatheringState(this);
 }
 
 void QuicTransportChannel::OnCandidateGathered(TransportChannelImpl* channel,
                                                const Candidate& c) {
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   SignalCandidateGathered(this, c);
 }
 
 void QuicTransportChannel::OnRoleConflict(TransportChannelImpl* channel) {
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   SignalRoleConflict(this);
 }
 
 void QuicTransportChannel::OnRouteChange(TransportChannel* channel,
                                          const Candidate& candidate) {
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   SignalRouteChange(this, candidate);
 }
 
@@ -393,13 +396,13 @@ void QuicTransportChannel::OnSelectedCandidatePairChanged(
     TransportChannel* channel,
     CandidatePairInterface* selected_candidate_pair,
     int last_sent_packet_id) {
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   SignalSelectedCandidatePairChanged(this, selected_candidate_pair,
                                      last_sent_packet_id);
 }
 
 void QuicTransportChannel::OnConnectionRemoved(TransportChannelImpl* channel) {
-  ASSERT(channel == channel_);
+  ASSERT(channel == channel_.get());
   SignalConnectionRemoved(this);
 }
 
@@ -435,7 +438,7 @@ bool QuicTransportChannel::CreateQuicSession() {
                                      ? net::Perspective::IS_CLIENT
                                      : net::Perspective::IS_SERVER;
   bool owns_writer = false;
-  rtc::scoped_ptr<net::QuicConnection> connection(new net::QuicConnection(
+  std::unique_ptr<net::QuicConnection> connection(new net::QuicConnection(
       kConnectionId, kConnectionIpEndpoint, &helper_, this, owns_writer,
       perspective, net::QuicSupportedVersions()));
   quic_.reset(new QuicSession(std::move(connection), config_));
@@ -443,6 +446,8 @@ bool QuicTransportChannel::CreateQuicSession() {
       this, &QuicTransportChannel::OnHandshakeComplete);
   quic_->SignalConnectionClosed.connect(
       this, &QuicTransportChannel::OnConnectionClosed);
+  quic_->SignalIncomingStream.connect(this,
+                                      &QuicTransportChannel::OnIncomingStream);
   return true;
 }
 
@@ -483,8 +488,14 @@ bool QuicTransportChannel::StartQuicHandshake() {
     net::QuicCryptoServerConfig::ConfigOptions options;
     quic_crypto_server_config_->AddDefaultConfig(helper_.GetRandomGenerator(),
                                                  helper_.GetClock(), options);
+    quic_compressed_certs_cache_.reset(new net::QuicCompressedCertsCache(
+        net::QuicCompressedCertsCache::kQuicCompressedCertsCacheSize));
+    // TODO(mikescarlett): Add support for stateless rejects.
+    bool use_stateless_rejects_if_peer_supported = false;
     net::QuicCryptoServerStream* crypto_stream =
         new net::QuicCryptoServerStream(quic_crypto_server_config_.get(),
+                                        quic_compressed_certs_cache_.get(),
+                                        use_stateless_rejects_if_peer_supported,
                                         quic_.get());
     quic_->StartServerHandshake(crypto_stream);
     LOG_J(LS_INFO, this) << "QuicTransportChannel: Started server handshake.";
@@ -541,6 +552,7 @@ void QuicTransportChannel::OnConnectionClosed(net::QuicErrorCode error,
   // does not close due to failure.
   set_quic_state(QUIC_TRANSPORT_CLOSED);
   set_writable(false);
+  SignalClosed();
 }
 
 void QuicTransportChannel::OnProofValid(
@@ -569,4 +581,16 @@ void QuicTransportChannel::set_quic_state(QuicTransportState state) {
   quic_state_ = state;
 }
 
+ReliableQuicStream* QuicTransportChannel::CreateQuicStream() {
+  if (quic_) {
+    net::SpdyPriority priority = 0;  // Priority of the QUIC stream
+    return quic_->CreateOutgoingDynamicStream(priority);
+  }
+  return nullptr;
+}
+
+void QuicTransportChannel::OnIncomingStream(ReliableQuicStream* stream) {
+  SignalIncomingStream(stream);
+}
+
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/quic/quictransportchannel.h b/chromium/third_party/webrtc/p2p/quic/quictransportchannel.h
index ab02c77b952..dec24d21543 100644
--- a/chromium/third_party/webrtc/p2p/quic/quictransportchannel.h
+++ b/chromium/third_party/webrtc/p2p/quic/quictransportchannel.h
@@ -11,13 +11,14 @@
 #ifndef WEBRTC_P2P_QUIC_QUICTRANSPORTCHANNEL_H_
 #define WEBRTC_P2P_QUIC_QUICTRANSPORTCHANNEL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "net/quic/quic_crypto_client_stream.h"
 #include "net/quic/quic_packet_writer.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/p2p/base/transportchannelimpl.h"
 #include "webrtc/p2p/quic/quicconnectionhelper.h"
 #include "webrtc/p2p/quic/quicsession.h"
@@ -48,7 +49,7 @@ enum QuicTransportState {
 //       TransportChannelImpl* channel_;
 //   }
 //
-//   - Data written to SendPacket() is passed directly to |channel_| if  it is
+//   - Data written to SendPacket() is passed directly to |channel_| if it is
 //     an SRTP packet with the PF_SRTP_BYPASS flag.
 //
 //   - |quic_| passes outgoing packets to WritePacket(), which transfers them
@@ -61,8 +62,11 @@ enum QuicTransportState {
 //   - When the QUIC handshake is completed, quic_state() returns
 //     QUIC_TRANSPORT_CONNECTED and SRTP keying material can be exported.
 //
-// TODO(mikescarlett): Implement secure QUIC handshake, 0-RTT handshakes, and
-// QUIC data streams.
+//   - CreateQuicStream() creates an outgoing QUIC stream. Once the local peer
+//     sends data from this stream, the remote peer emits SignalIncomingStream
+//     with a QUIC stream of the same id to handle received data.
+//
+// TODO(mikescarlett): Implement secure QUIC handshake and 0-RTT handshakes.
 class QuicTransportChannel : public TransportChannelImpl,
                              public net::QuicPacketWriter,
                              public net::QuicCryptoClientStream::ProofHandler {
@@ -113,8 +117,9 @@ class QuicTransportChannel : public TransportChannelImpl,
                             size_t result_len) override;
   // TODO(mikescarlett): Remove this method once TransportChannel does not
   // require defining it.
-  bool GetRemoteSSLCertificate(rtc::SSLCertificate** cert) const override {
-    return false;
+  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
+      const override {
+    return nullptr;
   }
 
   // TransportChannelImpl overrides that we forward to the wrapped transport.
@@ -206,6 +211,14 @@ class QuicTransportChannel : public TransportChannelImpl,
   void OnCanWrite();
   // Connectivity state of QuicTransportChannel.
   QuicTransportState quic_state() const { return quic_state_; }
+  // Creates a new QUIC stream that can send data.
+  ReliableQuicStream* CreateQuicStream();
+
+  // Emitted when |quic_| creates a QUIC stream to receive data from the remote
+  // peer, when the stream did not exist previously.
+  sigslot::signal1<ReliableQuicStream*> SignalIncomingStream;
+  // Emitted when the QuicTransportChannel state becomes QUIC_TRANSPORT_CLOSED.
+  sigslot::signal0<> SignalClosed;
 
  private:
   // Fingerprint of remote peer.
@@ -241,6 +254,8 @@ class QuicTransportChannel : public TransportChannelImpl,
   void OnHandshakeComplete();
   // Called when |quic_| has closed the connection.
   void OnConnectionClosed(net::QuicErrorCode error, bool from_peer);
+  // Called when |quic_| has created a new QUIC stream for incoming data.
+  void OnIncomingStream(ReliableQuicStream* stream);
 
   // Called by OnReadPacket() when a QUIC packet is received.
   bool HandleQuicPacket(const char* data, size_t size);
@@ -257,12 +272,12 @@ class QuicTransportChannel : public TransportChannelImpl,
   rtc::Thread* worker_thread_;
   // Underlying channel which is responsible for connecting with the remote peer
   // and sending/receiving packets across the network.
-  TransportChannelImpl* const channel_;
+  std::unique_ptr<TransportChannelImpl> channel_;
   // Connectivity state of QuicTransportChannel.
   QuicTransportState quic_state_ = QUIC_TRANSPORT_NEW;
   // QUIC session which establishes the crypto handshake and converts data
   // to/from QUIC packets.
-  rtc::scoped_ptr<QuicSession> quic_;
+  std::unique_ptr<QuicSession> quic_;
   // Non-crypto config for |quic_|.
   net::QuicConfig config_;
   // Helper for net::QuicConnection that provides timing and
@@ -273,9 +288,11 @@ class QuicTransportChannel : public TransportChannelImpl,
   // the handshake. This must be set before we start QUIC.
   rtc::Optional<rtc::SSLRole> ssl_role_;
   // Config for QUIC crypto client stream, used when |ssl_role_| is SSL_CLIENT.
-  rtc::scoped_ptr<net::QuicCryptoClientConfig> quic_crypto_client_config_;
+  std::unique_ptr<net::QuicCryptoClientConfig> quic_crypto_client_config_;
   // Config for QUIC crypto server stream, used when |ssl_role_| is SSL_SERVER.
-  rtc::scoped_ptr<net::QuicCryptoServerConfig> quic_crypto_server_config_;
+  std::unique_ptr<net::QuicCryptoServerConfig> quic_crypto_server_config_;
+  // Used by QUIC crypto server stream to track most recently compressed certs.
+  std::unique_ptr<net::QuicCompressedCertsCache> quic_compressed_certs_cache_;
   // This peer's certificate.
   rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
   // Fingerprint of the remote peer. This must be set before we start QUIC.
diff --git a/chromium/third_party/webrtc/p2p/quic/quictransportchannel_unittest.cc b/chromium/third_party/webrtc/p2p/quic/quictransportchannel_unittest.cc
index 7b5b51848aa..0e16390a89b 100644
--- a/chromium/third_party/webrtc/p2p/quic/quictransportchannel_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/quic/quictransportchannel_unittest.cc
@@ -10,19 +10,20 @@
 
 #include "webrtc/p2p/quic/quictransportchannel.h"
 
+#include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/common.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 
 using cricket::ConnectionRole;
 using cricket::IceRole;
 using cricket::QuicTransportChannel;
+using cricket::ReliableQuicStream;
 using cricket::TransportChannel;
 using cricket::TransportDescription;
 
@@ -93,13 +94,17 @@ class QuicTestPeer : public sigslot::has_slots<> {
   explicit QuicTestPeer(const std::string& name)
       : name_(name),
         bytes_sent_(0),
-        ice_channel_(name_, 0),
-        quic_channel_(&ice_channel_) {
+        ice_channel_(new FailableTransportChannel(name_, 0)),
+        quic_channel_(ice_channel_),
+        incoming_stream_count_(0) {
     quic_channel_.SignalReadPacket.connect(
         this, &QuicTestPeer::OnTransportChannelReadPacket);
-    ice_channel_.SetAsync(true);
+    quic_channel_.SignalIncomingStream.connect(this,
+                                               &QuicTestPeer::OnIncomingStream);
+    quic_channel_.SignalClosed.connect(this, &QuicTestPeer::OnClosed);
+    ice_channel_->SetAsync(true);
     rtc::scoped_refptr<rtc::RTCCertificate> local_cert =
-        rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+        rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
             rtc::SSLIdentity::Generate(name_, rtc::KT_DEFAULT)));
     quic_channel_.SetLocalCertificate(local_cert);
     local_fingerprint_.reset(CreateFingerprint(local_cert.get()));
@@ -107,13 +112,13 @@ class QuicTestPeer : public sigslot::has_slots<> {
 
   // Connects |ice_channel_| to that of the other peer.
   void Connect(QuicTestPeer* other_peer) {
-    ice_channel_.Connect();
-    other_peer->ice_channel_.Connect();
-    ice_channel_.SetDestination(&other_peer->ice_channel_);
+    ice_channel_->Connect();
+    other_peer->ice_channel_->Connect();
+    ice_channel_->SetDestination(other_peer->ice_channel_);
   }
 
   // Disconnects |ice_channel_|.
-  void Disconnect() { ice_channel_.SetDestination(nullptr); }
+  void Disconnect() { ice_channel_->SetDestination(nullptr); }
 
   // Generates ICE credentials and passes them to |quic_channel_|.
   void SetIceParameters(IceRole local_ice_role,
@@ -144,7 +149,7 @@ class QuicTestPeer : public sigslot::has_slots<> {
     if (!get_digest_algorithm || digest_algorithm.empty()) {
       return nullptr;
     }
-    rtc::scoped_ptr<rtc::SSLFingerprint> fingerprint(
+    std::unique_ptr<rtc::SSLFingerprint> fingerprint(
         rtc::SSLFingerprint::Create(digest_algorithm, cert->identity()));
     if (digest_algorithm != rtc::DIGEST_SHA_256) {
       return nullptr;
@@ -184,22 +189,28 @@ class QuicTestPeer : public sigslot::has_slots<> {
 
   void ClearBytesReceived() { bytes_received_ = 0; }
 
-  void SetWriteError(int error) { ice_channel_.SetError(error); }
+  void SetWriteError(int error) { ice_channel_->SetError(error); }
 
   size_t bytes_received() const { return bytes_received_; }
 
   size_t bytes_sent() const { return bytes_sent_; }
 
-  FailableTransportChannel* ice_channel() { return &ice_channel_; }
+  FailableTransportChannel* ice_channel() { return ice_channel_; }
 
   QuicTransportChannel* quic_channel() { return &quic_channel_; }
 
-  rtc::scoped_ptr<rtc::SSLFingerprint>& local_fingerprint() {
+  std::unique_ptr<rtc::SSLFingerprint>& local_fingerprint() {
     return local_fingerprint_;
   }
 
+  ReliableQuicStream* incoming_quic_stream() { return incoming_quic_stream_; }
+
+  size_t incoming_stream_count() const { return incoming_stream_count_; }
+
+  bool signal_closed_emitted() const { return signal_closed_emitted_; }
+
  private:
-  // QUIC channel callback.
+  // QuicTransportChannel callbacks.
   void OnTransportChannelReadPacket(TransportChannel* channel,
                                     const char* data,
                                     size_t size,
@@ -210,13 +221,21 @@ class QuicTestPeer : public sigslot::has_slots<> {
     int expected_flags = IsRtpLeadByte(data[0]) ? cricket::PF_SRTP_BYPASS : 0;
     ASSERT_EQ(expected_flags, flags);
   }
+  void OnIncomingStream(ReliableQuicStream* stream) {
+    incoming_quic_stream_ = stream;
+    ++incoming_stream_count_;
+  }
+  void OnClosed() { signal_closed_emitted_ = true; }
 
   std::string name_;                      // Channel name.
   size_t bytes_sent_;                     // Bytes sent by QUIC channel.
   size_t bytes_received_;                 // Bytes received by QUIC channel.
-  FailableTransportChannel ice_channel_;  // Simulates an ICE channel.
+  FailableTransportChannel* ice_channel_;  // Simulates an ICE channel.
   QuicTransportChannel quic_channel_;     // QUIC channel to test.
-  rtc::scoped_ptr<rtc::SSLFingerprint> local_fingerprint_;
+  std::unique_ptr<rtc::SSLFingerprint> local_fingerprint_;
+  ReliableQuicStream* incoming_quic_stream_ = nullptr;
+  size_t incoming_stream_count_;
+  bool signal_closed_emitted_ = false;
 };
 
 class QuicTransportChannelTest : public testing::Test {
@@ -244,9 +263,9 @@ class QuicTransportChannelTest : public testing::Test {
     peer1_.quic_channel()->SetSslRole(peer1_ssl_role);
     peer2_.quic_channel()->SetSslRole(peer2_ssl_role);
 
-    rtc::scoped_ptr<rtc::SSLFingerprint>& peer1_fingerprint =
+    std::unique_ptr<rtc::SSLFingerprint>& peer1_fingerprint =
         peer1_.local_fingerprint();
-    rtc::scoped_ptr<rtc::SSLFingerprint>& peer2_fingerprint =
+    std::unique_ptr<rtc::SSLFingerprint>& peer2_fingerprint =
         peer2_.local_fingerprint();
 
     peer1_.quic_channel()->SetRemoteFingerprint(
@@ -450,7 +469,7 @@ TEST_F(QuicTransportChannelTest, QuicRoleReversalAfterQuic) {
 // Set the SSL role, then test that GetSslRole returns the same value.
 TEST_F(QuicTransportChannelTest, SetGetSslRole) {
   ASSERT_TRUE(peer1_.quic_channel()->SetSslRole(rtc::SSL_SERVER));
-  rtc::scoped_ptr<rtc::SSLRole> role(new rtc::SSLRole());
+  std::unique_ptr<rtc::SSLRole> role(new rtc::SSLRole());
   ASSERT_TRUE(peer1_.quic_channel()->GetSslRole(role.get()));
   EXPECT_EQ(rtc::SSL_SERVER, *role);
 }
@@ -486,3 +505,52 @@ TEST_F(QuicTransportChannelTest, IceReceivingBeforeConnected) {
   ASSERT_TRUE_WAIT(quic_connected(), kTimeoutMs);
   EXPECT_TRUE(peer1_.quic_channel()->receiving());
 }
+
+// Test that when peer 1 creates an outgoing stream, peer 2 creates an incoming
+// QUIC stream with the same ID and fires OnIncomingStream.
+TEST_F(QuicTransportChannelTest, CreateOutgoingAndIncomingQuicStream) {
+  Connect();
+  EXPECT_EQ(nullptr, peer1_.quic_channel()->CreateQuicStream());
+  ASSERT_TRUE_WAIT(quic_connected(), kTimeoutMs);
+  ReliableQuicStream* stream = peer1_.quic_channel()->CreateQuicStream();
+  ASSERT_NE(nullptr, stream);
+  stream->Write("Hi", 2);
+  EXPECT_TRUE_WAIT(peer2_.incoming_quic_stream() != nullptr, kTimeoutMs);
+  EXPECT_EQ(stream->id(), peer2_.incoming_quic_stream()->id());
+}
+
+// Test that if the QuicTransportChannel is unwritable, then all outgoing QUIC
+// streams can send data once the QuicTransprotChannel becomes writable again.
+TEST_F(QuicTransportChannelTest, OutgoingQuicStreamSendsDataAfterReconnect) {
+  Connect();
+  ASSERT_TRUE_WAIT(quic_connected(), kTimeoutMs);
+  ReliableQuicStream* stream1 = peer1_.quic_channel()->CreateQuicStream();
+  ASSERT_NE(nullptr, stream1);
+  ReliableQuicStream* stream2 = peer1_.quic_channel()->CreateQuicStream();
+  ASSERT_NE(nullptr, stream2);
+
+  peer1_.ice_channel()->SetWritable(false);
+  stream1->Write("First", 5);
+  EXPECT_EQ(5u, stream1->queued_data_bytes());
+  stream2->Write("Second", 6);
+  EXPECT_EQ(6u, stream2->queued_data_bytes());
+  EXPECT_EQ(0u, peer2_.incoming_stream_count());
+
+  peer1_.ice_channel()->SetWritable(true);
+  EXPECT_EQ_WAIT(0u, stream1->queued_data_bytes(), kTimeoutMs);
+  EXPECT_EQ_WAIT(0u, stream2->queued_data_bytes(), kTimeoutMs);
+  EXPECT_EQ_WAIT(2u, peer2_.incoming_stream_count(), kTimeoutMs);
+}
+
+// Test that SignalClosed is emitted when the QuicConnection closes.
+TEST_F(QuicTransportChannelTest, SignalClosedEmitted) {
+  Connect();
+  ASSERT_TRUE_WAIT(quic_connected(), kTimeoutMs);
+  ASSERT_FALSE(peer1_.signal_closed_emitted());
+  ReliableQuicStream* stream = peer1_.quic_channel()->CreateQuicStream();
+  ASSERT_NE(nullptr, stream);
+  stream->CloseConnectionWithDetails(net::QuicErrorCode::QUIC_NO_ERROR,
+                                     "Closing QUIC for testing");
+  EXPECT_TRUE(peer1_.signal_closed_emitted());
+  EXPECT_TRUE_WAIT(peer2_.signal_closed_emitted(), kTimeoutMs);
+}
diff --git a/chromium/third_party/webrtc/p2p/quic/reliablequicstream.cc b/chromium/third_party/webrtc/p2p/quic/reliablequicstream.cc
index ca2e3f0d3be..5b65f1e7c75 100644
--- a/chromium/third_party/webrtc/p2p/quic/reliablequicstream.cc
+++ b/chromium/third_party/webrtc/p2p/quic/reliablequicstream.cc
@@ -12,6 +12,7 @@
 
 #include <string>
 
+#include "net/quic/quic_session.h"
 #include "webrtc/base/checks.h"
 
 namespace cricket {
@@ -38,14 +39,26 @@ void ReliableQuicStream::OnClose() {
   SignalClosed(id(), connection_error());
 }
 
-rtc::StreamResult ReliableQuicStream::Write(const char* data, size_t len) {
+rtc::StreamResult ReliableQuicStream::Write(const char* data,
+                                            size_t len,
+                                            bool fin) {
   // Writes the data, or buffers it.
-  WriteOrBufferData(std::string(data, len), false, nullptr);
+  WriteOrBufferData(base::StringPiece(data, len), fin, nullptr);
   if (HasBufferedData()) {
     return rtc::StreamResult(rtc::SR_BLOCK);
   }
-
   return rtc::StreamResult(rtc::SR_SUCCESS);
 }
 
+void ReliableQuicStream::Close() {
+  net::ReliableQuicStream::session()->CloseStream(id());
+}
+
+void ReliableQuicStream::OnCanWrite() {
+  uint64_t prev_queued_bytes = queued_data_bytes();
+  net::ReliableQuicStream::OnCanWrite();
+  uint64_t queued_bytes_written = prev_queued_bytes - queued_data_bytes();
+  SignalQueuedBytesWritten(id(), queued_bytes_written);
+}
+
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/p2p/quic/reliablequicstream.h b/chromium/third_party/webrtc/p2p/quic/reliablequicstream.h
index 61d060f850b..9493a3fcc18 100644
--- a/chromium/third_party/webrtc/p2p/quic/reliablequicstream.h
+++ b/chromium/third_party/webrtc/p2p/quic/reliablequicstream.h
@@ -29,16 +29,23 @@ class ReliableQuicStream : public net::ReliableQuicStream,
   // ReliableQuicStream overrides.
   void OnDataAvailable() override;
   void OnClose() override;
+  void OnCanWrite() override;
 
   // Process decrypted data into encrypted QUIC packets, which get sent to the
   // QuicPacketWriter. rtc::SR_BLOCK is returned if the operation blocks instead
   // of writing, in which case the data is queued until OnCanWrite() is called.
-  rtc::StreamResult Write(const char* data, size_t len);
+  // If |fin| == true, then this stream closes after sending data.
+  rtc::StreamResult Write(const char* data, size_t len, bool fin = false);
+  // Removes this stream from the QuicSession's stream map.
+  void Close();
 
   // Called when decrypted data is ready to be read.
   sigslot::signal3<net::QuicStreamId, const char*, size_t> SignalDataReceived;
-  // Called when stream closed.
-  sigslot::signal2<net::QuicStreamId, net::QuicErrorCode> SignalClosed;
+  // Called when the stream is closed.
+  sigslot::signal2<net::QuicStreamId, int> SignalClosed;
+  // Emits the number of queued bytes that were written by OnCanWrite(), after
+  // the stream was previously write blocked.
+  sigslot::signal2<net::QuicStreamId, uint64_t> SignalQueuedBytesWritten;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(ReliableQuicStream);
diff --git a/chromium/third_party/webrtc/p2p/quic/reliablequicstream_unittest.cc b/chromium/third_party/webrtc/p2p/quic/reliablequicstream_unittest.cc
index aeb3e1af87c..cf9f5e92ddf 100644
--- a/chromium/third_party/webrtc/p2p/quic/reliablequicstream_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/quic/reliablequicstream_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/p2p/quic/reliablequicstream.h"
 
+#include <memory>
 #include <string>
 
 #include "net/base/ip_address_number.h"
@@ -25,7 +26,6 @@
 using cricket::QuicConnectionHelper;
 using cricket::ReliableQuicStream;
 
-using net::FecProtection;
 using net::IPAddress;
 using net::IPEndPoint;
 using net::PerPacketOptions;
@@ -65,7 +65,6 @@ class MockQuicSession : public QuicSession {
       QuicIOVector iovector,
       QuicStreamOffset offset,
       bool fin,
-      FecProtection fec_protection,
       QuicAckListenerInterface* ack_notifier_delegate) override {
     if (!writable_) {
       return QuicConsumedData(0, false);
@@ -117,11 +116,11 @@ class DummyPacketWriter : public QuicPacketWriter {
   DummyPacketWriter() {}
 
   // QuicPacketWriter overrides.
-  virtual net::WriteResult WritePacket(const char* buffer,
-                                       size_t buf_len,
-                                       const IPAddress& self_address,
-                                       const IPEndPoint& peer_address,
-                                       PerPacketOptions* options) {
+  net::WriteResult WritePacket(const char* buffer,
+                               size_t buf_len,
+                               const IPAddress& self_address,
+                               const IPEndPoint& peer_address,
+                               PerPacketOptions* options) override {
     return net::WriteResult(net::WRITE_STATUS_ERROR, 0);
   }
 
@@ -163,6 +162,8 @@ class ReliableQuicStreamTest : public ::testing::Test,
     stream_->SignalDataReceived.connect(
         this, &ReliableQuicStreamTest::OnDataReceived);
     stream_->SignalClosed.connect(this, &ReliableQuicStreamTest::OnClosed);
+    stream_->SignalQueuedBytesWritten.connect(
+        this, &ReliableQuicStreamTest::OnQueuedBytesWritten);
 
     session_->register_write_blocked_stream(stream_->id(), kDefaultPriority);
   }
@@ -172,11 +173,15 @@ class ReliableQuicStreamTest : public ::testing::Test,
     read_buffer_.append(data, length);
   }
 
-  void OnClosed(QuicStreamId id, QuicErrorCode err) { closed_ = true; }
+  void OnClosed(QuicStreamId id, int err) { closed_ = true; }
+
+  void OnQueuedBytesWritten(QuicStreamId id, uint64_t queued_bytes_written) {
+    queued_bytes_written_ = queued_bytes_written;
+  }
 
  protected:
-  rtc::scoped_ptr<ReliableQuicStream> stream_;
-  rtc::scoped_ptr<MockQuicSession> session_;
+  std::unique_ptr<ReliableQuicStream> stream_;
+  std::unique_ptr<MockQuicSession> session_;
 
   // Data written by the ReliableQuicStream.
   std::string write_buffer_;
@@ -184,6 +189,8 @@ class ReliableQuicStreamTest : public ::testing::Test,
   std::string read_buffer_;
   // Whether the ReliableQuicStream is closed.
   bool closed_ = false;
+  // Bytes written by OnCanWrite().
+  uint64_t queued_bytes_written_;
 };
 
 // Write an entire string.
@@ -213,6 +220,7 @@ TEST_F(ReliableQuicStreamTest, BufferData) {
 
   session_->set_writable(true);
   stream_->OnCanWrite();
+  EXPECT_EQ(7ul, queued_bytes_written_);
 
   EXPECT_FALSE(stream_->HasBufferedData());
   EXPECT_EQ("Foo bar", write_buffer_);
diff --git a/chromium/third_party/webrtc/p2p/stunprober/main.cc b/chromium/third_party/webrtc/p2p/stunprober/main.cc
index 9ef91e071b5..9d0fff87322 100644
--- a/chromium/third_party/webrtc/p2p/stunprober/main.cc
+++ b/chromium/third_party/webrtc/p2p/stunprober/main.cc
@@ -14,13 +14,14 @@
 
 #include <iostream>
 #include <map>
+#include <memory>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/flags.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/nethelpers.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
@@ -119,9 +120,9 @@ int main(int argc, char** argv) {
   rtc::InitializeSSL();
   rtc::InitRandom(rtc::Time32());
   rtc::Thread* thread = rtc::ThreadManager::Instance()->WrapCurrentThread();
-  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> socket_factory(
+  std::unique_ptr<rtc::BasicPacketSocketFactory> socket_factory(
       new rtc::BasicPacketSocketFactory());
-  rtc::scoped_ptr<rtc::BasicNetworkManager> network_manager(
+  std::unique_ptr<rtc::BasicNetworkManager> network_manager(
       new rtc::BasicNetworkManager());
   rtc::NetworkManager::NetworkList networks;
   network_manager->GetNetworks(&networks);
diff --git a/chromium/third_party/webrtc/p2p/stunprober/stunprober.cc b/chromium/third_party/webrtc/p2p/stunprober/stunprober.cc
index eb9a7a7d3b7..90dc60e1514 100644
--- a/chromium/third_party/webrtc/p2p/stunprober/stunprober.cc
+++ b/chromium/third_party/webrtc/p2p/stunprober/stunprober.cc
@@ -9,6 +9,7 @@
  */
 
 #include <map>
+#include <memory>
 #include <set>
 #include <string>
 
@@ -16,6 +17,7 @@
 #include "webrtc/base/asyncresolverinterface.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
@@ -89,11 +91,11 @@ class StunProber::Requester : public sigslot::has_slots<> {
   StunProber* prober_;
 
   // The socket for this session.
-  rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
+  std::unique_ptr<rtc::AsyncPacketSocket> socket_;
 
   // Temporary SocketAddress and buffer for RecvFrom.
   rtc::SocketAddress addr_;
-  rtc::scoped_ptr<rtc::ByteBufferWriter> response_packet_;
+  std::unique_ptr<rtc::ByteBufferWriter> response_packet_;
 
   std::vector<Request*> requests_;
   std::vector<rtc::SocketAddress> server_ips_;
@@ -140,7 +142,7 @@ void StunProber::Requester::SendStunRequest() {
       rtc::CreateRandomString(cricket::kStunTransactionIdLength));
   message.SetType(cricket::STUN_BINDING_REQUEST);
 
-  rtc::scoped_ptr<rtc::ByteBufferWriter> request_packet(
+  std::unique_ptr<rtc::ByteBufferWriter> request_packet(
       new rtc::ByteBufferWriter(nullptr, kMaxUdpBufferSize));
   if (!message.Write(request_packet.get())) {
     prober_->ReportOnFinished(WRITE_FAILED);
@@ -161,7 +163,7 @@ void StunProber::Requester::SendStunRequest() {
     return;
   }
 
-  request.sent_time_ms = rtc::Time64();
+  request.sent_time_ms = rtc::TimeMillis();
 
   num_request_sent_++;
   RTC_DCHECK(static_cast<size_t>(num_request_sent_) <= server_ips_.size());
@@ -169,7 +171,7 @@ void StunProber::Requester::SendStunRequest() {
 
 void StunProber::Requester::Request::ProcessResponse(const char* buf,
                                                      size_t buf_len) {
-  int64_t now = rtc::Time64();
+  int64_t now = rtc::TimeMillis();
   rtc::ByteBufferReader message(buf, buf_len);
   cricket::StunMessage stun_response;
   if (!stun_response.Read(&message)) {
@@ -344,7 +346,7 @@ void StunProber::OnServerResolved(rtc::AsyncResolverInterface* resolver) {
 
   // Prepare all the sockets beforehand. All of them will bind to "any" address.
   while (sockets_.size() < total_socket_required()) {
-    rtc::scoped_ptr<rtc::AsyncPacketSocket> socket(
+    std::unique_ptr<rtc::AsyncPacketSocket> socket(
         socket_factory_->CreateUdpSocket(rtc::SocketAddress(INADDR_ANY, 0), 0,
                                          0));
     if (!socket) {
@@ -412,7 +414,7 @@ int StunProber::get_wake_up_interval_ms() {
 
 void StunProber::MaybeScheduleStunRequests() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  int64_t now = rtc::Time64();
+  int64_t now = rtc::TimeMillis();
 
   if (Done()) {
     invoker_.AsyncInvokeDelayed<void>(
diff --git a/chromium/third_party/webrtc/p2p/stunprober/stunprober.h b/chromium/third_party/webrtc/p2p/stunprober/stunprober.h
index 44999a201ed..dbb67c61674 100644
--- a/chromium/third_party/webrtc/p2p/stunprober/stunprober.h
+++ b/chromium/third_party/webrtc/p2p/stunprober/stunprober.h
@@ -19,9 +19,9 @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/bytebuffer.h"
 #include "webrtc/base/callback.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/ipaddress.h"
 #include "webrtc/base/network.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"
diff --git a/chromium/third_party/webrtc/p2p/stunprober/stunprober_unittest.cc b/chromium/third_party/webrtc/p2p/stunprober/stunprober_unittest.cc
index cdcc14a36f2..f5ee4bffeff 100644
--- a/chromium/third_party/webrtc/p2p/stunprober/stunprober_unittest.cc
+++ b/chromium/third_party/webrtc/p2p/stunprober/stunprober_unittest.cc
@@ -8,13 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/asyncresolverinterface.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
@@ -83,7 +84,7 @@ class StunProberTest : public testing::Test {
     rtc::NetworkManager::NetworkList networks;
     networks.push_back(&ipv4_network1);
 
-    rtc::scoped_ptr<rtc::BasicPacketSocketFactory> socket_factory(
+    std::unique_ptr<rtc::BasicPacketSocketFactory> socket_factory(
         new rtc::BasicPacketSocketFactory());
 
     // Set up the expected results for verification.
@@ -119,14 +120,14 @@ class StunProberTest : public testing::Test {
   }
 
   rtc::Thread* main_;
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
-  rtc::scoped_ptr<StunProber> prober;
+  std::unique_ptr<StunProber> prober;
   int result_ = 0;
   bool stopped_ = false;
-  rtc::scoped_ptr<cricket::TestStunServer> stun_server_1_;
-  rtc::scoped_ptr<cricket::TestStunServer> stun_server_2_;
+  std::unique_ptr<cricket::TestStunServer> stun_server_1_;
+  std::unique_ptr<cricket::TestStunServer> stun_server_2_;
 };
 
 TEST_F(StunProberTest, NonSharedMode) {
diff --git a/chromium/third_party/webrtc/pc/audiomonitor.cc b/chromium/third_party/webrtc/pc/audiomonitor.cc
index 952ee9fbda8..085ab931913 100644
--- a/chromium/third_party/webrtc/pc/audiomonitor.cc
+++ b/chromium/third_party/webrtc/pc/audiomonitor.cc
@@ -101,4 +101,4 @@ rtc::Thread *AudioMonitor::monitor_thread() {
   return monitoring_thread_;
 }
 
-}
+}  // namespace cricket
diff --git a/chromium/third_party/webrtc/pc/audiomonitor.h b/chromium/third_party/webrtc/pc/audiomonitor.h
index 741709b60eb..885c60f08b5 100644
--- a/chromium/third_party/webrtc/pc/audiomonitor.h
+++ b/chromium/third_party/webrtc/pc/audiomonitor.h
@@ -8,10 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TALK_SESSION_MEDIA_AUDIOMONITOR_H_
-#define TALK_SESSION_MEDIA_AUDIOMONITOR_H_
+#ifndef WEBRTC_PC_AUDIOMONITOR_H_
+#define WEBRTC_PC_AUDIOMONITOR_H_
 
 #include <vector>
+#include <utility>
+
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/p2p/base/port.h"
@@ -24,7 +26,7 @@ struct AudioInfo {
   int input_level;
   int output_level;
   typedef std::vector<std::pair<uint32_t, int> > StreamList;
-  StreamList active_streams; // ssrcs contributing to output_level
+  StreamList active_streams;  // ssrcs contributing to output_level
 };
 
 class AudioMonitor : public rtc::MessageHandler,
@@ -53,6 +55,6 @@ class AudioMonitor : public rtc::MessageHandler,
   bool monitoring_;
 };
 
-}
+}  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_AUDIOMONITOR_H_
+#endif  // WEBRTC_PC_AUDIOMONITOR_H_
diff --git a/chromium/third_party/webrtc/pc/bundlefilter.h b/chromium/third_party/webrtc/pc/bundlefilter.h
index a69868171de..44a45da5280 100644
--- a/chromium/third_party/webrtc/pc/bundlefilter.h
+++ b/chromium/third_party/webrtc/pc/bundlefilter.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TALK_SESSION_MEDIA_BUNDLEFILTER_H_
-#define TALK_SESSION_MEDIA_BUNDLEFILTER_H_
+#ifndef WEBRTC_PC_BUNDLEFILTER_H_
+#define WEBRTC_PC_BUNDLEFILTER_H_
 
 #include <stdint.h>
 
@@ -51,4 +51,4 @@ class BundleFilter {
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_BUNDLEFILTER_H_
+#endif  // WEBRTC_PC_BUNDLEFILTER_H_
diff --git a/chromium/third_party/webrtc/pc/channel.cc b/chromium/third_party/webrtc/pc/channel.cc
index b76d7bdb1d2..97c1d669101 100644
--- a/chromium/third_party/webrtc/pc/channel.cc
+++ b/chromium/third_party/webrtc/pc/channel.cc
@@ -37,12 +37,18 @@ bool SetRawAudioSink_w(VoiceMediaChannel* channel,
   channel->SetRawAudioSink(ssrc, std::move(*sink));
   return true;
 }
+
+struct SendPacketMessageData : public rtc::MessageData {
+  rtc::CopyOnWriteBuffer packet;
+  rtc::PacketOptions options;
+};
+
 }  // namespace
 
 enum {
   MSG_EARLYMEDIATIMEOUT = 1,
-  MSG_RTPPACKET,
-  MSG_RTCPPACKET,
+  MSG_SEND_RTP_PACKET,
+  MSG_SEND_RTCP_PACKET,
   MSG_CHANNEL_ERROR,
   MSG_READYTOSENDDATA,
   MSG_DATARECEIVED,
@@ -61,11 +67,6 @@ static void SafeSetError(const std::string& message, std::string* error_desc) {
   }
 }
 
-struct PacketMessageData : public rtc::MessageData {
-  rtc::CopyOnWriteBuffer packet;
-  rtc::PacketOptions options;
-};
-
 struct VoiceChannelErrorMessageData : public rtc::MessageData {
   VoiceChannelErrorMessageData(uint32_t in_ssrc,
                                VoiceMediaChannel::Error in_error)
@@ -142,30 +143,38 @@ void RtpSendParametersFromMediaDescription(
   send_params->max_bandwidth_bps = desc->bandwidth();
 }
 
-BaseChannel::BaseChannel(rtc::Thread* thread,
+BaseChannel::BaseChannel(rtc::Thread* worker_thread,
+                         rtc::Thread* network_thread,
                          MediaChannel* media_channel,
                          TransportController* transport_controller,
                          const std::string& content_name,
                          bool rtcp)
-    : worker_thread_(thread),
-      transport_controller_(transport_controller),
-      media_channel_(media_channel),
+    : worker_thread_(worker_thread),
+      network_thread_(network_thread),
+
       content_name_(content_name),
+
+      transport_controller_(transport_controller),
       rtcp_transport_enabled_(rtcp),
       transport_channel_(nullptr),
       rtcp_transport_channel_(nullptr),
-      enabled_(false),
-      writable_(false),
       rtp_ready_to_send_(false),
       rtcp_ready_to_send_(false),
+      writable_(false),
       was_ever_writable_(false),
-      local_content_direction_(MD_INACTIVE),
-      remote_content_direction_(MD_INACTIVE),
       has_received_packet_(false),
       dtls_keyed_(false),
       secure_required_(false),
-      rtp_abs_sendtime_extn_id_(-1) {
+      rtp_abs_sendtime_extn_id_(-1),
+
+      media_channel_(media_channel),
+      enabled_(false),
+      local_content_direction_(MD_INACTIVE),
+      remote_content_direction_(MD_INACTIVE) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
+  if (transport_controller) {
+    RTC_DCHECK_EQ(network_thread, transport_controller->network_thread());
+  }
   LOG(LS_INFO) << "Created channel for " << content_name;
 }
 
@@ -174,57 +183,100 @@ BaseChannel::~BaseChannel() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   Deinit();
   StopConnectionMonitor();
-  FlushRtcpMessages();  // Send any outstanding RTCP packets.
-  worker_thread_->Clear(this);  // eats any outstanding messages or packets
+  // Eats any outstanding messages or packets.
+  worker_thread_->Clear(&invoker_);
+  worker_thread_->Clear(this);
   // We must destroy the media channel before the transport channel, otherwise
   // the media channel may try to send on the dead transport channel. NULLing
   // is not an effective strategy since the sends will come on another thread.
   delete media_channel_;
-  // Note that we don't just call set_transport_channel(nullptr) because that
+  // Note that we don't just call SetTransportChannel_n(nullptr) because that
   // would call a pure virtual method which we can't do from a destructor.
+  network_thread_->Invoke<void>(
+      Bind(&BaseChannel::DestroyTransportChannels_n, this));
+  LOG(LS_INFO) << "Destroyed channel";
+}
+
+void BaseChannel::DisconnectTransportChannels_n() {
+  // Send any outstanding RTCP packets.
+  FlushRtcpMessages_n();
+
+  // Stop signals from transport channels, but keep them alive because
+  // media_channel may use them from a different thread.
   if (transport_channel_) {
     DisconnectFromTransportChannel(transport_channel_);
-    transport_controller_->DestroyTransportChannel_w(
-        transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
   }
   if (rtcp_transport_channel_) {
     DisconnectFromTransportChannel(rtcp_transport_channel_);
-    transport_controller_->DestroyTransportChannel_w(
+  }
+
+  // Clear pending read packets/messages.
+  network_thread_->Clear(&invoker_);
+  network_thread_->Clear(this);
+}
+
+void BaseChannel::DestroyTransportChannels_n() {
+  if (transport_channel_) {
+    transport_controller_->DestroyTransportChannel_n(
+        transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
+  }
+  if (rtcp_transport_channel_) {
+    transport_controller_->DestroyTransportChannel_n(
         transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   }
-  LOG(LS_INFO) << "Destroyed channel";
+  // Clear pending send packets/messages.
+  network_thread_->Clear(&invoker_);
+  network_thread_->Clear(this);
+}
+
+bool BaseChannel::Init_w(const std::string* bundle_transport_name) {
+  if (!network_thread_->Invoke<bool>(
+          Bind(&BaseChannel::InitNetwork_n, this, bundle_transport_name))) {
+    return false;
+  }
+
+  // Both RTP and RTCP channels are set, we can call SetInterface on
+  // media channel and it can set network options.
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  media_channel_->SetInterface(this);
+  return true;
 }
 
-bool BaseChannel::Init() {
-  if (!SetTransport(content_name())) {
+bool BaseChannel::InitNetwork_n(const std::string* bundle_transport_name) {
+  RTC_DCHECK(network_thread_->IsCurrent());
+  const std::string& transport_name =
+      (bundle_transport_name ? *bundle_transport_name : content_name());
+  if (!SetTransport_n(transport_name)) {
     return false;
   }
 
-  if (!SetDtlsSrtpCryptoSuites(transport_channel(), false)) {
+  if (!SetDtlsSrtpCryptoSuites_n(transport_channel_, false)) {
     return false;
   }
   if (rtcp_transport_enabled() &&
-      !SetDtlsSrtpCryptoSuites(rtcp_transport_channel(), true)) {
+      !SetDtlsSrtpCryptoSuites_n(rtcp_transport_channel_, true)) {
     return false;
   }
-
-  // Both RTP and RTCP channels are set, we can call SetInterface on
-  // media channel and it can set network options.
-  media_channel_->SetInterface(this);
   return true;
 }
 
 void BaseChannel::Deinit() {
+  RTC_DCHECK(worker_thread_->IsCurrent());
   media_channel_->SetInterface(NULL);
+  // Packets arrive on the network thread, processing packets calls virtual
+  // functions, so need to stop this process in Deinit that is called in
+  // derived classes destructor.
+  network_thread_->Invoke<void>(
+      Bind(&BaseChannel::DisconnectTransportChannels_n, this));
 }
 
 bool BaseChannel::SetTransport(const std::string& transport_name) {
-  return worker_thread_->Invoke<bool>(
-      Bind(&BaseChannel::SetTransport_w, this, transport_name));
+  return network_thread_->Invoke<bool>(
+      Bind(&BaseChannel::SetTransport_n, this, transport_name));
 }
 
-bool BaseChannel::SetTransport_w(const std::string& transport_name) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+bool BaseChannel::SetTransport_n(const std::string& transport_name) {
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   if (transport_name == transport_name_) {
     // Nothing to do if transport name isn't changing
@@ -234,7 +286,7 @@ bool BaseChannel::SetTransport_w(const std::string& transport_name) {
   // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport
   // changes and wait until the DTLS handshake is complete to set the newly
   // negotiated parameters.
-  if (ShouldSetupDtlsSrtp()) {
+  if (ShouldSetupDtlsSrtp_n()) {
     // Set |writable_| to false such that UpdateWritableState_w can set up
     // DTLS-SRTP when the writable_ becomes true again.
     writable_ = false;
@@ -245,19 +297,19 @@ bool BaseChannel::SetTransport_w(const std::string& transport_name) {
   if (rtcp_transport_enabled()) {
     LOG(LS_INFO) << "Create RTCP TransportChannel for " << content_name()
                  << " on " << transport_name << " transport ";
-    set_rtcp_transport_channel(
-        transport_controller_->CreateTransportChannel_w(
+    SetRtcpTransportChannel_n(
+        transport_controller_->CreateTransportChannel_n(
             transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP),
         false /* update_writablity */);
-    if (!rtcp_transport_channel()) {
+    if (!rtcp_transport_channel_) {
       return false;
     }
   }
 
   // We're not updating the writablity during the transition state.
-  set_transport_channel(transport_controller_->CreateTransportChannel_w(
+  SetTransportChannel_n(transport_controller_->CreateTransportChannel_n(
       transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP));
-  if (!transport_channel()) {
+  if (!transport_channel_) {
     return false;
   }
 
@@ -266,14 +318,14 @@ bool BaseChannel::SetTransport_w(const std::string& transport_name) {
     // We can only update the RTCP ready to send after set_transport_channel has
     // handled channel writability.
     SetReadyToSend(
-        true, rtcp_transport_channel() && rtcp_transport_channel()->writable());
+        true, rtcp_transport_channel_ && rtcp_transport_channel_->writable());
   }
   transport_name_ = transport_name;
   return true;
 }
 
-void BaseChannel::set_transport_channel(TransportChannel* new_tc) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+void BaseChannel::SetTransportChannel_n(TransportChannel* new_tc) {
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   TransportChannel* old_tc = transport_channel_;
   if (!old_tc && !new_tc) {
@@ -284,7 +336,7 @@ void BaseChannel::set_transport_channel(TransportChannel* new_tc) {
 
   if (old_tc) {
     DisconnectFromTransportChannel(old_tc);
-    transport_controller_->DestroyTransportChannel_w(
+    transport_controller_->DestroyTransportChannel_n(
         transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
   }
 
@@ -299,13 +351,13 @@ void BaseChannel::set_transport_channel(TransportChannel* new_tc) {
 
   // Update aggregate writable/ready-to-send state between RTP and RTCP upon
   // setting new channel
-  UpdateWritableState_w();
+  UpdateWritableState_n();
   SetReadyToSend(false, new_tc && new_tc->writable());
 }
 
-void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc,
-                                             bool update_writablity) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+void BaseChannel::SetRtcpTransportChannel_n(TransportChannel* new_tc,
+                                            bool update_writablity) {
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   TransportChannel* old_tc = rtcp_transport_channel_;
   if (!old_tc && !new_tc) {
@@ -316,14 +368,14 @@ void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc,
 
   if (old_tc) {
     DisconnectFromTransportChannel(old_tc);
-    transport_controller_->DestroyTransportChannel_w(
+    transport_controller_->DestroyTransportChannel_n(
         transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   }
 
   rtcp_transport_channel_ = new_tc;
 
   if (new_tc) {
-    RTC_CHECK(!(ShouldSetupDtlsSrtp() && srtp_filter_.IsActive()))
+    RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_filter_.IsActive()))
         << "Setting RTCP for DTLS/SRTP after SrtpFilter is active "
         << "should never happen.";
     ConnectToTransportChannel(new_tc);
@@ -335,13 +387,13 @@ void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc,
   if (update_writablity) {
     // Update aggregate writable/ready-to-send state between RTP and RTCP upon
     // setting new channel
-    UpdateWritableState_w();
+    UpdateWritableState_n();
     SetReadyToSend(true, new_tc && new_tc->writable());
   }
 }
 
 void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
   tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead);
@@ -349,15 +401,18 @@ void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {
   tc->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState);
   tc->SignalSelectedCandidatePairChanged.connect(
       this, &BaseChannel::OnSelectedCandidatePairChanged);
+  tc->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n);
 }
 
 void BaseChannel::DisconnectFromTransportChannel(TransportChannel* tc) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   tc->SignalWritableState.disconnect(this);
   tc->SignalReadPacket.disconnect(this);
   tc->SignalReadyToSend.disconnect(this);
   tc->SignalDtlsState.disconnect(this);
+  tc->SignalSelectedCandidatePairChanged.disconnect(this);
+  tc->SignalSentPacket.disconnect(this);
 }
 
 bool BaseChannel::Enable(bool enable) {
@@ -405,8 +460,11 @@ void BaseChannel::StartConnectionMonitor(int cms) {
   // We pass in the BaseChannel instead of the transport_channel_
   // because if the transport_channel_ changes, the ConnectionMonitor
   // would be pointing to the wrong TransportChannel.
-  connection_monitor_.reset(new ConnectionMonitor(
-      this, worker_thread(), rtc::Thread::Current()));
+  // We pass in the network thread because on that thread connection monitor
+  // will call BaseChannel::GetConnectionStats which must be called on the
+  // network thread.
+  connection_monitor_.reset(
+      new ConnectionMonitor(this, network_thread(), rtc::Thread::Current()));
   connection_monitor_->SignalUpdate.connect(
       this, &BaseChannel::OnConnectionMonitorUpdate);
   connection_monitor_->Start(cms);
@@ -420,22 +478,27 @@ void BaseChannel::StopConnectionMonitor() {
 }
 
 bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(network_thread_->IsCurrent());
   return transport_channel_->GetStats(infos);
 }
 
-bool BaseChannel::IsReadyToReceive() const {
+bool BaseChannel::IsReadyToReceive_w() const {
   // Receive data if we are enabled and have local content,
   return enabled() && IsReceiveContentDirection(local_content_direction_);
 }
 
-bool BaseChannel::IsReadyToSend() const {
+bool BaseChannel::IsReadyToSend_w() const {
   // Send outgoing data if we are enabled, have local and remote content,
   // and we have had some form of connectivity.
   return enabled() && IsReceiveContentDirection(remote_content_direction_) &&
          IsSendContentDirection(local_content_direction_) &&
-         was_ever_writable() &&
-         (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp());
+         network_thread_->Invoke<bool>(
+             Bind(&BaseChannel::IsTransportReadyToSend_n, this));
+}
+
+bool BaseChannel::IsTransportReadyToSend_n() const {
+  return was_ever_writable() &&
+         (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp_n());
 }
 
 bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet,
@@ -450,7 +513,15 @@ bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet,
 
 int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
                            int value) {
-  TransportChannel* channel = NULL;
+  return network_thread_->Invoke<int>(
+      Bind(&BaseChannel::SetOption_n, this, type, opt, value));
+}
+
+int BaseChannel::SetOption_n(SocketType type,
+                             rtc::Socket::Option opt,
+                             int value) {
+  RTC_DCHECK(network_thread_->IsCurrent());
+  TransportChannel* channel = nullptr;
   switch (type) {
     case ST_RTP:
       channel = transport_channel_;
@@ -467,8 +538,10 @@ int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
 }
 
 void BaseChannel::OnWritableState(TransportChannel* channel) {
-  ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
-  UpdateWritableState_w();
+  RTC_DCHECK(channel == transport_channel_ ||
+             channel == rtcp_transport_channel_);
+  RTC_DCHECK(network_thread_->IsCurrent());
+  UpdateWritableState_n();
 }
 
 void BaseChannel::OnChannelRead(TransportChannel* channel,
@@ -477,7 +550,7 @@ void BaseChannel::OnChannelRead(TransportChannel* channel,
                                 int flags) {
   TRACE_EVENT0("webrtc", "BaseChannel::OnChannelRead");
   // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+  RTC_DCHECK(network_thread_->IsCurrent());
 
   // When using RTCP multiplexing we might get RTCP packets on the RTP
   // transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
@@ -493,7 +566,7 @@ void BaseChannel::OnReadyToSend(TransportChannel* channel) {
 
 void BaseChannel::OnDtlsState(TransportChannel* channel,
                               DtlsTransportState state) {
-  if (!ShouldSetupDtlsSrtp()) {
+  if (!ShouldSetupDtlsSrtp_n()) {
     return;
   }
 
@@ -512,33 +585,36 @@ void BaseChannel::OnSelectedCandidatePairChanged(
     CandidatePairInterface* selected_candidate_pair,
     int last_sent_packet_id) {
   ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
-  NetworkRoute network_route;
+  RTC_DCHECK(network_thread_->IsCurrent());
+  std::string transport_name = channel->transport_name();
+  rtc::NetworkRoute network_route;
   if (selected_candidate_pair) {
-    network_route =
-        NetworkRoute(selected_candidate_pair->local_candidate().network_id(),
-                     selected_candidate_pair->remote_candidate().network_id(),
-                     last_sent_packet_id);
+    network_route = rtc::NetworkRoute(
+        selected_candidate_pair->local_candidate().network_id(),
+        selected_candidate_pair->remote_candidate().network_id(),
+        last_sent_packet_id);
   }
-  media_channel()->OnNetworkRouteChanged(channel->transport_name(),
-                                         network_route);
+  invoker_.AsyncInvoke<void>(
+      worker_thread_, Bind(&MediaChannel::OnNetworkRouteChanged, media_channel_,
+                           transport_name, network_route));
 }
 
 void BaseChannel::SetReadyToSend(bool rtcp, bool ready) {
+  RTC_DCHECK(network_thread_->IsCurrent());
   if (rtcp) {
     rtcp_ready_to_send_ = ready;
   } else {
     rtp_ready_to_send_ = ready;
   }
 
-  if (rtp_ready_to_send_ &&
-      // In the case of rtcp mux |rtcp_transport_channel_| will be null.
-      (rtcp_ready_to_send_ || !rtcp_transport_channel_)) {
-    // Notify the MediaChannel when both rtp and rtcp channel can send.
-    media_channel_->OnReadyToSend(true);
-  } else {
-    // Notify the MediaChannel when either rtp or rtcp channel can't send.
-    media_channel_->OnReadyToSend(false);
-  }
+  bool ready_to_send =
+      (rtp_ready_to_send_ &&
+       // In the case of rtcp mux |rtcp_transport_channel_| will be null.
+       (rtcp_ready_to_send_ || !rtcp_transport_channel_));
+
+  invoker_.AsyncInvoke<void>(
+      worker_thread_,
+      Bind(&MediaChannel::OnReadyToSend, media_channel_, ready_to_send));
 }
 
 bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
@@ -550,22 +626,23 @@ bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
 bool BaseChannel::SendPacket(bool rtcp,
                              rtc::CopyOnWriteBuffer* packet,
                              const rtc::PacketOptions& options) {
-  // SendPacket gets called from MediaEngine, typically on an encoder thread.
-  // If the thread is not our worker thread, we will post to our worker
-  // so that the real work happens on our worker. This avoids us having to
+  // SendPacket gets called from MediaEngine, on a pacer or an encoder thread.
+  // If the thread is not our network thread, we will post to our network
+  // so that the real work happens on our network. This avoids us having to
   // synchronize access to all the pieces of the send path, including
   // SRTP and the inner workings of the transport channels.
   // The only downside is that we can't return a proper failure code if
   // needed. Since UDP is unreliable anyway, this should be a non-issue.
-  if (rtc::Thread::Current() != worker_thread_) {
+  if (!network_thread_->IsCurrent()) {
     // Avoid a copy by transferring the ownership of the packet data.
-    int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
-    PacketMessageData* data = new PacketMessageData;
+    int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET;
+    SendPacketMessageData* data = new SendPacketMessageData;
     data->packet = std::move(*packet);
     data->options = options;
-    worker_thread_->Post(this, message_id, data);
+    network_thread_->Post(this, message_id, data);
     return true;
   }
+  TRACE_EVENT0("webrtc", "BaseChannel::SendPacket");
 
   // Now that we are on the correct thread, ensure we have a place to send this
   // packet before doing anything. (We might get RTCP packets that we don't
@@ -589,6 +666,7 @@ bool BaseChannel::SendPacket(bool rtcp,
   updated_options = options;
   // Protect if needed.
   if (srtp_filter_.IsActive()) {
+    TRACE_EVENT0("webrtc", "SRTP Encode");
     bool res;
     uint8_t* data = packet->data();
     int len = static_cast<int>(packet->size());
@@ -656,9 +734,9 @@ bool BaseChannel::SendPacket(bool rtcp,
   }
 
   // Bon voyage.
-  int ret =
-      channel->SendPacket(packet->data<char>(), packet->size(), updated_options,
-                          (secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
+  int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL;
+  int ret = channel->SendPacket(packet->data<char>(), packet->size(),
+                                updated_options, flags);
   if (ret != static_cast<int>(packet->size())) {
     if (channel->GetError() == EWOULDBLOCK) {
       LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
@@ -687,6 +765,7 @@ bool BaseChannel::WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) {
 
 void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
                                const rtc::PacketTime& packet_time) {
+  RTC_DCHECK(network_thread_->IsCurrent());
   if (!WantsPacket(rtcp, packet)) {
     return;
   }
@@ -700,6 +779,7 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
 
   // Unprotect the packet, if needed.
   if (srtp_filter_.IsActive()) {
+    TRACE_EVENT0("webrtc", "SRTP Decode");
     char* data = packet->data<char>();
     int len = static_cast<int>(packet->size());
     bool res;
@@ -743,11 +823,22 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
     return;
   }
 
-  // Push it down to the media channel.
-  if (!rtcp) {
-    media_channel_->OnPacketReceived(packet, packet_time);
+  invoker_.AsyncInvoke<void>(
+      worker_thread_,
+      Bind(&BaseChannel::OnPacketReceived, this, rtcp, *packet, packet_time));
+}
+
+void BaseChannel::OnPacketReceived(bool rtcp,
+                                   const rtc::CopyOnWriteBuffer& packet,
+                                   const rtc::PacketTime& packet_time) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  // Need to copy variable because OnRtcpReceived/OnPacketReceived
+  // requires non-const pointer to buffer. This doesn't memcpy the actual data.
+  rtc::CopyOnWriteBuffer data(packet);
+  if (rtcp) {
+    media_channel_->OnRtcpReceived(&data, packet_time);
   } else {
-    media_channel_->OnRtcpReceived(packet, packet_time);
+    media_channel_->OnPacketReceived(&data, packet_time);
   }
 }
 
@@ -786,7 +877,7 @@ void BaseChannel::EnableMedia_w() {
 
   LOG(LS_INFO) << "Channel enabled";
   enabled_ = true;
-  ChangeState();
+  ChangeState_w();
 }
 
 void BaseChannel::DisableMedia_w() {
@@ -796,20 +887,20 @@ void BaseChannel::DisableMedia_w() {
 
   LOG(LS_INFO) << "Channel disabled";
   enabled_ = false;
-  ChangeState();
+  ChangeState_w();
 }
 
-void BaseChannel::UpdateWritableState_w() {
+void BaseChannel::UpdateWritableState_n() {
   if (transport_channel_ && transport_channel_->writable() &&
       (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) {
-    ChannelWritable_w();
+    ChannelWritable_n();
   } else {
-    ChannelNotWritable_w();
+    ChannelNotWritable_n();
   }
 }
 
-void BaseChannel::ChannelWritable_w() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+void BaseChannel::ChannelWritable_n() {
+  RTC_DCHECK(network_thread_->IsCurrent());
   if (writable_) {
     return;
   }
@@ -829,15 +920,16 @@ void BaseChannel::ChannelWritable_w() {
   }
 
   was_ever_writable_ = true;
-  MaybeSetupDtlsSrtp_w();
+  MaybeSetupDtlsSrtp_n();
   writable_ = true;
   ChangeState();
 }
 
-void BaseChannel::SignalDtlsSetupFailure_w(bool rtcp) {
-  ASSERT(worker_thread() == rtc::Thread::Current());
-  signaling_thread()->Invoke<void>(Bind(
-      &BaseChannel::SignalDtlsSetupFailure_s, this, rtcp));
+void BaseChannel::SignalDtlsSetupFailure_n(bool rtcp) {
+  RTC_DCHECK(network_thread_->IsCurrent());
+  invoker_.AsyncInvoke<void>(
+      signaling_thread(),
+      Bind(&BaseChannel::SignalDtlsSetupFailure_s, this, rtcp));
 }
 
 void BaseChannel::SignalDtlsSetupFailure_s(bool rtcp) {
@@ -845,26 +937,27 @@ void BaseChannel::SignalDtlsSetupFailure_s(bool rtcp) {
   SignalDtlsSetupFailure(this, rtcp);
 }
 
-bool BaseChannel::SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp) {
+bool BaseChannel::SetDtlsSrtpCryptoSuites_n(TransportChannel* tc, bool rtcp) {
   std::vector<int> crypto_suites;
   // We always use the default SRTP crypto suites for RTCP, but we may use
   // different crypto suites for RTP depending on the media type.
   if (!rtcp) {
-    GetSrtpCryptoSuites(&crypto_suites);
+    GetSrtpCryptoSuites_n(&crypto_suites);
   } else {
     GetDefaultSrtpCryptoSuites(&crypto_suites);
   }
   return tc->SetSrtpCryptoSuites(crypto_suites);
 }
 
-bool BaseChannel::ShouldSetupDtlsSrtp() const {
+bool BaseChannel::ShouldSetupDtlsSrtp_n() const {
   // Since DTLS is applied to all channels, checking RTP should be enough.
   return transport_channel_ && transport_channel_->IsDtlsActive();
 }
 
 // This function returns true if either DTLS-SRTP is not in use
 // *or* DTLS-SRTP is successfully set up.
-bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) {
+bool BaseChannel::SetupDtlsSrtp_n(bool rtcp_channel) {
+  RTC_DCHECK(network_thread_->IsCurrent());
   bool ret = false;
 
   TransportChannel* channel =
@@ -950,30 +1043,30 @@ bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) {
   return ret;
 }
 
-void BaseChannel::MaybeSetupDtlsSrtp_w() {
+void BaseChannel::MaybeSetupDtlsSrtp_n() {
   if (srtp_filter_.IsActive()) {
     return;
   }
 
-  if (!ShouldSetupDtlsSrtp()) {
+  if (!ShouldSetupDtlsSrtp_n()) {
     return;
   }
 
-  if (!SetupDtlsSrtp(false)) {
-    SignalDtlsSetupFailure_w(false);
+  if (!SetupDtlsSrtp_n(false)) {
+    SignalDtlsSetupFailure_n(false);
     return;
   }
 
   if (rtcp_transport_channel_) {
-    if (!SetupDtlsSrtp(true)) {
-      SignalDtlsSetupFailure_w(true);
+    if (!SetupDtlsSrtp_n(true)) {
+      SignalDtlsSetupFailure_n(true);
       return;
     }
   }
 }
 
-void BaseChannel::ChannelNotWritable_w() {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
+void BaseChannel::ChannelNotWritable_n() {
+  RTC_DCHECK(network_thread_->IsCurrent());
   if (!writable_)
     return;
 
@@ -982,7 +1075,7 @@ void BaseChannel::ChannelNotWritable_w() {
   ChangeState();
 }
 
-bool BaseChannel::SetRtpTransportParameters_w(
+bool BaseChannel::SetRtpTransportParameters(
     const MediaContentDescription* content,
     ContentAction action,
     ContentSource src,
@@ -993,15 +1086,27 @@ bool BaseChannel::SetRtpTransportParameters_w(
   }
 
   // Cache secure_required_ for belt and suspenders check on SendPacket
+  return network_thread_->Invoke<bool>(
+      Bind(&BaseChannel::SetRtpTransportParameters_n, this, content, action,
+           src, error_desc));
+}
+
+bool BaseChannel::SetRtpTransportParameters_n(
+    const MediaContentDescription* content,
+    ContentAction action,
+    ContentSource src,
+    std::string* error_desc) {
+  RTC_DCHECK(network_thread_->IsCurrent());
+
   if (src == CS_LOCAL) {
     set_secure_required(content->crypto_required() != CT_NONE);
   }
 
-  if (!SetSrtp_w(content->cryptos(), action, src, error_desc)) {
+  if (!SetSrtp_n(content->cryptos(), action, src, error_desc)) {
     return false;
   }
 
-  if (!SetRtcpMux_w(content->rtcp_mux(), action, src, error_desc)) {
+  if (!SetRtcpMux_n(content->rtcp_mux(), action, src, error_desc)) {
     return false;
   }
 
@@ -1010,19 +1115,18 @@ bool BaseChannel::SetRtpTransportParameters_w(
 
 // |dtls| will be set to true if DTLS is active for transport channel and
 // crypto is empty.
-bool BaseChannel::CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
-                                  bool* dtls,
-                                  std::string* error_desc) {
+bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
+                                    bool* dtls,
+                                    std::string* error_desc) {
   *dtls = transport_channel_->IsDtlsActive();
   if (*dtls && !cryptos.empty()) {
-    SafeSetError("Cryptos must be empty when DTLS is active.",
-                 error_desc);
+    SafeSetError("Cryptos must be empty when DTLS is active.", error_desc);
     return false;
   }
   return true;
 }
 
-bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
+bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
                             ContentAction action,
                             ContentSource src,
                             std::string* error_desc) {
@@ -1033,7 +1137,7 @@ bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
   }
   bool ret = false;
   bool dtls = false;
-  ret = CheckSrtpConfig(cryptos, &dtls, error_desc);
+  ret = CheckSrtpConfig_n(cryptos, &dtls, error_desc);
   if (!ret) {
     return false;
   }
@@ -1070,19 +1174,19 @@ bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
 }
 
 void BaseChannel::ActivateRtcpMux() {
-  worker_thread_->Invoke<void>(Bind(
-      &BaseChannel::ActivateRtcpMux_w, this));
+  network_thread_->Invoke<void>(Bind(&BaseChannel::ActivateRtcpMux_n, this));
 }
 
-void BaseChannel::ActivateRtcpMux_w() {
+void BaseChannel::ActivateRtcpMux_n() {
   if (!rtcp_mux_filter_.IsActive()) {
     rtcp_mux_filter_.SetActive();
-    set_rtcp_transport_channel(nullptr, true);
+    SetRtcpTransportChannel_n(nullptr, true);
     rtcp_transport_enabled_ = false;
   }
 }
 
-bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
+bool BaseChannel::SetRtcpMux_n(bool enable,
+                               ContentAction action,
                                ContentSource src,
                                std::string* error_desc) {
   bool ret = false;
@@ -1100,7 +1204,7 @@ bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
         LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name()
                      << " by destroying RTCP transport channel for "
                      << transport_name();
-        set_rtcp_transport_channel(nullptr, true);
+        SetRtcpTransportChannel_n(nullptr, true);
         rtcp_transport_enabled_ = false;
       }
       break;
@@ -1121,7 +1225,7 @@ bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
   if (rtcp_mux_filter_.IsActive()) {
     // If the RTP transport is already writable, then so are we.
     if (transport_channel_->writable()) {
-      ChannelWritable_w();
+      ChannelWritable_n();
     }
   }
 
@@ -1285,23 +1389,38 @@ bool BaseChannel::UpdateRemoteStreams_w(
   return ret;
 }
 
-void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension(
+void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w(
     const std::vector<RtpHeaderExtension>& extensions) {
+// Absolute Send Time extension id is used only with external auth,
+// so do not bother searching for it and making asyncronious call to set
+// something that is not used.
+#if defined(ENABLE_EXTERNAL_AUTH)
   const RtpHeaderExtension* send_time_extension =
       FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
-  rtp_abs_sendtime_extn_id_ =
+  int rtp_abs_sendtime_extn_id =
       send_time_extension ? send_time_extension->id : -1;
+  invoker_.AsyncInvoke<void>(
+      network_thread_, Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n,
+                            this, rtp_abs_sendtime_extn_id));
+#endif
+}
+
+void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n(
+    int rtp_abs_sendtime_extn_id) {
+  rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
 }
 
 void BaseChannel::OnMessage(rtc::Message *pmsg) {
   TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
   switch (pmsg->message_id) {
-    case MSG_RTPPACKET:
-    case MSG_RTCPPACKET: {
-      PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
-      SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet,
-                 data->options);
-      delete data;  // because it is Posted
+    case MSG_SEND_RTP_PACKET:
+    case MSG_SEND_RTCP_PACKET: {
+      RTC_DCHECK(network_thread_->IsCurrent());
+      SendPacketMessageData* data =
+          static_cast<SendPacketMessageData*>(pmsg->pdata);
+      bool rtcp = pmsg->message_id == MSG_SEND_RTCP_PACKET;
+      SendPacket(rtcp, &data->packet, data->options);
+      delete data;
       break;
     }
     case MSG_FIRSTPACKETRECEIVED: {
@@ -1311,25 +1430,39 @@ void BaseChannel::OnMessage(rtc::Message *pmsg) {
   }
 }
 
-void BaseChannel::FlushRtcpMessages() {
+void BaseChannel::FlushRtcpMessages_n() {
   // Flush all remaining RTCP messages. This should only be called in
   // destructor.
-  ASSERT(rtc::Thread::Current() == worker_thread_);
+  RTC_DCHECK(network_thread_->IsCurrent());
   rtc::MessageList rtcp_messages;
-  worker_thread_->Clear(this, MSG_RTCPPACKET, &rtcp_messages);
-  for (rtc::MessageList::iterator it = rtcp_messages.begin();
-       it != rtcp_messages.end(); ++it) {
-    worker_thread_->Send(this, MSG_RTCPPACKET, it->pdata);
+  network_thread_->Clear(this, MSG_SEND_RTCP_PACKET, &rtcp_messages);
+  for (const auto& message : rtcp_messages) {
+    network_thread_->Send(this, MSG_SEND_RTCP_PACKET, message.pdata);
   }
 }
 
-VoiceChannel::VoiceChannel(rtc::Thread* thread,
+void BaseChannel::SignalSentPacket_n(TransportChannel* /* channel */,
+                                     const rtc::SentPacket& sent_packet) {
+  RTC_DCHECK(network_thread_->IsCurrent());
+  invoker_.AsyncInvoke<void>(
+      worker_thread_,
+      rtc::Bind(&BaseChannel::SignalSentPacket_w, this, sent_packet));
+}
+
+void BaseChannel::SignalSentPacket_w(const rtc::SentPacket& sent_packet) {
+  RTC_DCHECK(worker_thread_->IsCurrent());
+  SignalSentPacket(sent_packet);
+}
+
+VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
+                           rtc::Thread* network_thread,
                            MediaEngineInterface* media_engine,
                            VoiceMediaChannel* media_channel,
                            TransportController* transport_controller,
                            const std::string& content_name,
                            bool rtcp)
-    : BaseChannel(thread,
+    : BaseChannel(worker_thread,
+                  network_thread,
                   media_channel,
                   transport_controller,
                   content_name,
@@ -1346,8 +1479,8 @@ VoiceChannel::~VoiceChannel() {
   Deinit();
 }
 
-bool VoiceChannel::Init() {
-  if (!BaseChannel::Init()) {
+bool VoiceChannel::Init_w(const std::string* bundle_transport_name) {
+  if (!BaseChannel::Init_w(bundle_transport_name)) {
     return false;
   }
   return true;
@@ -1403,28 +1536,49 @@ void VoiceChannel::SetRawAudioSink(
   InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
 }
 
-webrtc::RtpParameters VoiceChannel::GetRtpParameters(uint32_t ssrc) const {
+webrtc::RtpParameters VoiceChannel::GetRtpSendParameters(uint32_t ssrc) const {
   return worker_thread()->Invoke<webrtc::RtpParameters>(
-      Bind(&VoiceChannel::GetRtpParameters_w, this, ssrc));
+      Bind(&VoiceChannel::GetRtpSendParameters_w, this, ssrc));
 }
 
-webrtc::RtpParameters VoiceChannel::GetRtpParameters_w(uint32_t ssrc) const {
-  // Not yet implemented.
-  // TODO(skvlad): Add support for limiting send bitrate for audio channels.
-  return webrtc::RtpParameters();
+webrtc::RtpParameters VoiceChannel::GetRtpSendParameters_w(
+    uint32_t ssrc) const {
+  return media_channel()->GetRtpSendParameters(ssrc);
 }
 
-bool VoiceChannel::SetRtpParameters(uint32_t ssrc,
-                                    const webrtc::RtpParameters& parameters) {
+bool VoiceChannel::SetRtpSendParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
   return InvokeOnWorker(
-      Bind(&VoiceChannel::SetRtpParameters_w, this, ssrc, parameters));
+      Bind(&VoiceChannel::SetRtpSendParameters_w, this, ssrc, parameters));
 }
 
-bool VoiceChannel::SetRtpParameters_w(uint32_t ssrc,
-                                      webrtc::RtpParameters parameters) {
-  // Not yet implemented.
-  // TODO(skvlad): Add support for limiting send bitrate for audio channels.
-  return false;
+bool VoiceChannel::SetRtpSendParameters_w(uint32_t ssrc,
+                                          webrtc::RtpParameters parameters) {
+  return media_channel()->SetRtpSendParameters(ssrc, parameters);
+}
+
+webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters(
+    uint32_t ssrc) const {
+  return worker_thread()->Invoke<webrtc::RtpParameters>(
+      Bind(&VoiceChannel::GetRtpReceiveParameters_w, this, ssrc));
+}
+
+webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters_w(
+    uint32_t ssrc) const {
+  return media_channel()->GetRtpReceiveParameters(ssrc);
+}
+
+bool VoiceChannel::SetRtpReceiveParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
+  return InvokeOnWorker(
+      Bind(&VoiceChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
+}
+
+bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
+                                             webrtc::RtpParameters parameters) {
+  return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
@@ -1491,15 +1645,21 @@ void VoiceChannel::OnChannelRead(TransportChannel* channel,
   }
 }
 
-void VoiceChannel::ChangeState() {
+void BaseChannel::ChangeState() {
+  RTC_DCHECK(network_thread_->IsCurrent());
+  invoker_.AsyncInvoke<void>(worker_thread_,
+                             Bind(&BaseChannel::ChangeState_w, this));
+}
+
+void VoiceChannel::ChangeState_w() {
   // Render incoming data if we're the active call, and we have the local
   // content. We receive data on the default channel and multiplexed streams.
-  bool recv = IsReadyToReceive();
+  bool recv = IsReadyToReceive_w();
   media_channel()->SetPlayout(recv);
 
   // Send outgoing data if we're the active call, we have the remote content,
   // and we have had some form of connectivity.
-  bool send = IsReadyToSend();
+  bool send = IsReadyToSend_w();
   media_channel()->SetSend(send);
 
   LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
@@ -1525,7 +1685,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
     return false;
   }
 
@@ -1551,7 +1711,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
   }
 
   set_local_content_direction(content->direction());
-  ChangeState();
+  ChangeState_w();
   return true;
 }
 
@@ -1570,7 +1730,7 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
     return false;
   }
 
@@ -1598,11 +1758,11 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
   }
 
   if (audio->rtp_header_extensions_set()) {
-    MaybeCacheRtpAbsSendTimeHeaderExtension(audio->rtp_header_extensions());
+    MaybeCacheRtpAbsSendTimeHeaderExtension_w(audio->rtp_header_extensions());
   }
 
   set_remote_content_direction(content->direction());
-  ChangeState();
+  ChangeState_w();
   return true;
 }
 
@@ -1656,23 +1816,26 @@ void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
   SignalAudioMonitor(this, info);
 }
 
-void VoiceChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
+void VoiceChannel::GetSrtpCryptoSuites_n(
+    std::vector<int>* crypto_suites) const {
   GetSupportedAudioCryptoSuites(crypto_suites);
 }
 
-VideoChannel::VideoChannel(rtc::Thread* thread,
+VideoChannel::VideoChannel(rtc::Thread* worker_thread,
+                           rtc::Thread* network_thread,
                            VideoMediaChannel* media_channel,
                            TransportController* transport_controller,
                            const std::string& content_name,
                            bool rtcp)
-    : BaseChannel(thread,
+    : BaseChannel(worker_thread,
+                  network_thread,
                   media_channel,
                   transport_controller,
                   content_name,
                   rtcp) {}
 
-bool VideoChannel::Init() {
-  if (!BaseChannel::Init()) {
+bool VideoChannel::Init_w(const std::string* bundle_transport_name) {
+  if (!BaseChannel::Init_w(bundle_transport_name)) {
     return false;
   }
   return true;
@@ -1694,9 +1857,11 @@ bool VideoChannel::SetSink(uint32_t ssrc,
   return true;
 }
 
-bool VideoChannel::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
-  return InvokeOnWorker(Bind(&VideoMediaChannel::SetCapturer,
-                             media_channel(), ssrc, capturer));
+void VideoChannel::SetSource(
+    uint32_t ssrc,
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
+  worker_thread()->Invoke<void>(
+      Bind(&VideoMediaChannel::SetSource, media_channel(), ssrc, source));
 }
 
 bool VideoChannel::SetVideoSend(uint32_t ssrc,
@@ -1706,29 +1871,55 @@ bool VideoChannel::SetVideoSend(uint32_t ssrc,
                              ssrc, mute, options));
 }
 
-webrtc::RtpParameters VideoChannel::GetRtpParameters(uint32_t ssrc) const {
+webrtc::RtpParameters VideoChannel::GetRtpSendParameters(uint32_t ssrc) const {
+  return worker_thread()->Invoke<webrtc::RtpParameters>(
+      Bind(&VideoChannel::GetRtpSendParameters_w, this, ssrc));
+}
+
+webrtc::RtpParameters VideoChannel::GetRtpSendParameters_w(
+    uint32_t ssrc) const {
+  return media_channel()->GetRtpSendParameters(ssrc);
+}
+
+bool VideoChannel::SetRtpSendParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
+  return InvokeOnWorker(
+      Bind(&VideoChannel::SetRtpSendParameters_w, this, ssrc, parameters));
+}
+
+bool VideoChannel::SetRtpSendParameters_w(uint32_t ssrc,
+                                          webrtc::RtpParameters parameters) {
+  return media_channel()->SetRtpSendParameters(ssrc, parameters);
+}
+
+webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters(
+    uint32_t ssrc) const {
   return worker_thread()->Invoke<webrtc::RtpParameters>(
-      Bind(&VideoChannel::GetRtpParameters_w, this, ssrc));
+      Bind(&VideoChannel::GetRtpReceiveParameters_w, this, ssrc));
 }
 
-webrtc::RtpParameters VideoChannel::GetRtpParameters_w(uint32_t ssrc) const {
-  return media_channel()->GetRtpParameters(ssrc);
+webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters_w(
+    uint32_t ssrc) const {
+  return media_channel()->GetRtpReceiveParameters(ssrc);
 }
 
-bool VideoChannel::SetRtpParameters(uint32_t ssrc,
-                                    const webrtc::RtpParameters& parameters) {
+bool VideoChannel::SetRtpReceiveParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
   return InvokeOnWorker(
-      Bind(&VideoChannel::SetRtpParameters_w, this, ssrc, parameters));
+      Bind(&VideoChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
 }
 
-bool VideoChannel::SetRtpParameters_w(uint32_t ssrc,
-                                      webrtc::RtpParameters parameters) {
-  return media_channel()->SetRtpParameters(ssrc, parameters);
+bool VideoChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
+                                             webrtc::RtpParameters parameters) {
+  return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
 }
-void VideoChannel::ChangeState() {
+
+void VideoChannel::ChangeState_w() {
   // Send outgoing data if we're the active call, we have the remote content,
   // and we have had some form of connectivity.
-  bool send = IsReadyToSend();
+  bool send = IsReadyToSend_w();
   if (!media_channel()->SetSend(send)) {
     LOG(LS_ERROR) << "Failed to SetSend on video channel";
     // TODO(gangji): Report error back to server.
@@ -1777,7 +1968,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
     return false;
   }
 
-  if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
     return false;
   }
 
@@ -1803,7 +1994,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
   }
 
   set_local_content_direction(content->direction());
-  ChangeState();
+  ChangeState_w();
   return true;
 }
 
@@ -1822,8 +2013,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
     return false;
   }
 
-
-  if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
+  if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
     return false;
   }
 
@@ -1852,11 +2042,11 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   }
 
   if (video->rtp_header_extensions_set()) {
-    MaybeCacheRtpAbsSendTimeHeaderExtension(video->rtp_header_extensions());
+    MaybeCacheRtpAbsSendTimeHeaderExtension_w(video->rtp_header_extensions());
   }
 
   set_remote_content_direction(content->direction());
-  ChangeState();
+  ChangeState_w();
   return true;
 }
 
@@ -1887,16 +2077,19 @@ void VideoChannel::OnMediaMonitorUpdate(
   SignalMediaMonitor(this, info);
 }
 
-void VideoChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
+void VideoChannel::GetSrtpCryptoSuites_n(
+    std::vector<int>* crypto_suites) const {
   GetSupportedVideoCryptoSuites(crypto_suites);
 }
 
-DataChannel::DataChannel(rtc::Thread* thread,
+DataChannel::DataChannel(rtc::Thread* worker_thread,
+                         rtc::Thread* network_thread,
                          DataMediaChannel* media_channel,
                          TransportController* transport_controller,
                          const std::string& content_name,
                          bool rtcp)
-    : BaseChannel(thread,
+    : BaseChannel(worker_thread,
+                  network_thread,
                   media_channel,
                   transport_controller,
                   content_name,
@@ -1913,8 +2106,8 @@ DataChannel::~DataChannel() {
   Deinit();
 }
 
-bool DataChannel::Init() {
-  if (!BaseChannel::Init()) {
+bool DataChannel::Init_w(const std::string* bundle_transport_name) {
+  if (!BaseChannel::Init_w(bundle_transport_name)) {
     return false;
   }
   media_channel()->SignalDataReceived.connect(
@@ -2000,7 +2193,7 @@ bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
   }
 
   if (data_channel_type_ == DCT_RTP) {
-    if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
+    if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
       return false;
     }
   }
@@ -2032,7 +2225,7 @@ bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
   }
 
   set_local_content_direction(content->direction());
-  ChangeState();
+  ChangeState_w();
   return true;
 }
 
@@ -2062,7 +2255,7 @@ bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
 
   LOG(LS_INFO) << "Setting remote data description";
   if (data_channel_type_ == DCT_RTP &&
-      !SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
+      !SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
     return false;
   }
 
@@ -2087,21 +2280,21 @@ bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
   }
 
   set_remote_content_direction(content->direction());
-  ChangeState();
+  ChangeState_w();
   return true;
 }
 
-void DataChannel::ChangeState() {
+void DataChannel::ChangeState_w() {
   // Render incoming data if we're the active call, and we have the local
   // content. We receive data on the default channel and multiplexed streams.
-  bool recv = IsReadyToReceive();
+  bool recv = IsReadyToReceive_w();
   if (!media_channel()->SetReceive(recv)) {
     LOG(LS_ERROR) << "Failed to SetReceive on data channel";
   }
 
   // Send outgoing data if we're the active call, we have the remote content,
   // and we have had some form of connectivity.
-  bool send = IsReadyToSend();
+  bool send = IsReadyToSend_w();
   if (!media_channel()->SetSend(send)) {
     LOG(LS_ERROR) << "Failed to SetSend on data channel";
   }
@@ -2197,12 +2390,12 @@ void DataChannel::OnDataChannelReadyToSend(bool writable) {
                            new DataChannelReadyToSendMessageData(writable));
 }
 
-void DataChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
+void DataChannel::GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const {
   GetSupportedDataCryptoSuites(crypto_suites);
 }
 
-bool DataChannel::ShouldSetupDtlsSrtp() const {
-  return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
+bool DataChannel::ShouldSetupDtlsSrtp_n() const {
+  return data_channel_type_ == DCT_RTP && BaseChannel::ShouldSetupDtlsSrtp_n();
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
diff --git a/chromium/third_party/webrtc/pc/channel.h b/chromium/third_party/webrtc/pc/channel.h
index e05b6e5f3f0..d9f5fd6d279 100644
--- a/chromium/third_party/webrtc/pc/channel.h
+++ b/chromium/third_party/webrtc/pc/channel.h
@@ -19,6 +19,7 @@
 #include <vector>
 
 #include "webrtc/audio_sink.h"
+#include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/network.h"
@@ -27,8 +28,8 @@
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/base/streamparams.h"
-#include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/base/videosinkinterface.h"
+#include "webrtc/media/base/videosourceinterface.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"
 #include "webrtc/pc/audiomonitor.h"
@@ -47,14 +48,17 @@ namespace cricket {
 struct CryptoParams;
 class MediaContentDescription;
 
-enum SinkType {
-  SINK_PRE_CRYPTO,  // Sink packets before encryption or after decryption.
-  SINK_POST_CRYPTO  // Sink packets after encryption or before decryption.
-};
-
 // BaseChannel contains logic common to voice and video, including
-// enable, marshaling calls to a worker thread, and
+// enable, marshaling calls to a worker and network threads, and
 // connection and media monitors.
+// BaseChannel assumes signaling and other threads are allowed to make
+// synchronous calls to the worker thread, the worker thread makes synchronous
+// calls only to the network thread, and the network thread can't be blocked by
+// other threads.
+// All methods with _n suffix must be called on network thread,
+//     methods with _w suffix - on worker thread
+// and methods with _s suffix on signaling thread.
+// Network and worker threads may be the same thread.
 //
 // WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
 // This is required to avoid a data race between the destructor modifying the
@@ -66,26 +70,22 @@ class BaseChannel
       public MediaChannel::NetworkInterface,
       public ConnectionStatsGetter {
  public:
-  BaseChannel(rtc::Thread* thread,
+  BaseChannel(rtc::Thread* worker_thread,
+              rtc::Thread* network_thread,
               MediaChannel* channel,
               TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp);
   virtual ~BaseChannel();
-  bool Init();
-  // Deinit may be called multiple times and is simply ignored if it's alreay
+  bool Init_w(const std::string* bundle_transport_name);
+  // Deinit may be called multiple times and is simply ignored if it's already
   // done.
   void Deinit();
 
   rtc::Thread* worker_thread() const { return worker_thread_; }
+  rtc::Thread* network_thread() const { return network_thread_; }
   const std::string& content_name() const { return content_name_; }
   const std::string& transport_name() const { return transport_name_; }
-  TransportChannel* transport_channel() const {
-    return transport_channel_;
-  }
-  TransportChannel* rtcp_transport_channel() const {
-    return rtcp_transport_channel_;
-  }
   bool enabled() const { return enabled_; }
 
   // This function returns true if we are using SRTP.
@@ -143,18 +143,28 @@ class BaseChannel
   }
 
   sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure;
-  void SignalDtlsSetupFailure_w(bool rtcp);
+  void SignalDtlsSetupFailure_n(bool rtcp);
   void SignalDtlsSetupFailure_s(bool rtcp);
 
   // Used for latency measurements.
   sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
 
+  // Forward TransportChannel SignalSentPacket to worker thread.
+  sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
+
+  // Only public for unit tests.  Otherwise, consider private.
+  TransportChannel* transport_channel() const { return transport_channel_; }
+  TransportChannel* rtcp_transport_channel() const {
+    return rtcp_transport_channel_;
+  }
+
   // Made public for easier testing.
   void SetReadyToSend(bool rtcp, bool ready);
 
   // Only public for unit tests.  Otherwise, consider protected.
   int SetOption(SocketType type, rtc::Socket::Option o, int val)
       override;
+  int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
 
   SrtpFilter* srtp_filter() { return &srtp_filter_; }
 
@@ -162,11 +172,11 @@ class BaseChannel
   virtual MediaChannel* media_channel() const { return media_channel_; }
   // Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
   // true). Gets the transport channels from |transport_controller_|.
-  bool SetTransport_w(const std::string& transport_name);
+  bool SetTransport_n(const std::string& transport_name);
 
-  void set_transport_channel(TransportChannel* transport);
-  void set_rtcp_transport_channel(TransportChannel* transport,
-                                  bool update_writablity);
+  void SetTransportChannel_n(TransportChannel* transport);
+  void SetRtcpTransportChannel_n(TransportChannel* transport,
+                                 bool update_writablity);
 
   bool was_ever_writable() const { return was_ever_writable_; }
   void set_local_content_direction(MediaContentDirection direction) {
@@ -178,8 +188,8 @@ class BaseChannel
   void set_secure_required(bool secure_required) {
     secure_required_ = secure_required;
   }
-  bool IsReadyToReceive() const;
-  bool IsReadyToSend() const;
+  bool IsReadyToReceive_w() const;
+  bool IsReadyToSend_w() const;
   rtc::Thread* signaling_thread() {
     return transport_controller_->signaling_thread();
   }
@@ -188,7 +198,7 @@ class BaseChannel
   void ConnectToTransportChannel(TransportChannel* tc);
   void DisconnectFromTransportChannel(TransportChannel* tc);
 
-  void FlushRtcpMessages();
+  void FlushRtcpMessages_n();
 
   // NetworkInterface implementation, called by MediaEngine
   bool SendPacket(rtc::CopyOnWriteBuffer* packet,
@@ -217,28 +227,33 @@ class BaseChannel
   bool SendPacket(bool rtcp,
                   rtc::CopyOnWriteBuffer* packet,
                   const rtc::PacketOptions& options);
+
   virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
   void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
                     const rtc::PacketTime& packet_time);
+  void OnPacketReceived(bool rtcp,
+                        const rtc::CopyOnWriteBuffer& packet,
+                        const rtc::PacketTime& packet_time);
 
   void EnableMedia_w();
   void DisableMedia_w();
-  void UpdateWritableState_w();
-  void ChannelWritable_w();
-  void ChannelNotWritable_w();
+  void UpdateWritableState_n();
+  void ChannelWritable_n();
+  void ChannelNotWritable_n();
   bool AddRecvStream_w(const StreamParams& sp);
   bool RemoveRecvStream_w(uint32_t ssrc);
   bool AddSendStream_w(const StreamParams& sp);
   bool RemoveSendStream_w(uint32_t ssrc);
-  virtual bool ShouldSetupDtlsSrtp() const;
+  virtual bool ShouldSetupDtlsSrtp_n() const;
   // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
   // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
-  bool SetupDtlsSrtp(bool rtcp_channel);
-  void MaybeSetupDtlsSrtp_w();
+  bool SetupDtlsSrtp_n(bool rtcp_channel);
+  void MaybeSetupDtlsSrtp_n();
   // Set the DTLS-SRTP cipher policy on this channel as appropriate.
-  bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp);
+  bool SetDtlsSrtpCryptoSuites_n(TransportChannel* tc, bool rtcp);
 
-  virtual void ChangeState() = 0;
+  void ChangeState();
+  virtual void ChangeState_w() = 0;
 
   // Gets the content info appropriate to the channel (audio or video).
   virtual const ContentInfo* GetFirstContent(
@@ -255,25 +270,29 @@ class BaseChannel
   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc) = 0;
-  bool SetRtpTransportParameters_w(const MediaContentDescription* content,
+  bool SetRtpTransportParameters(const MediaContentDescription* content,
+                                 ContentAction action,
+                                 ContentSource src,
+                                 std::string* error_desc);
+  bool SetRtpTransportParameters_n(const MediaContentDescription* content,
                                    ContentAction action,
                                    ContentSource src,
                                    std::string* error_desc);
 
   // Helper method to get RTP Absoulute SendTime extension header id if
   // present in remote supported extensions list.
-  void MaybeCacheRtpAbsSendTimeHeaderExtension(
+  void MaybeCacheRtpAbsSendTimeHeaderExtension_w(
       const std::vector<RtpHeaderExtension>& extensions);
 
-  bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
-                       bool* dtls,
-                       std::string* error_desc);
-  bool SetSrtp_w(const std::vector<CryptoParams>& params,
+  bool CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
+                         bool* dtls,
+                         std::string* error_desc);
+  bool SetSrtp_n(const std::vector<CryptoParams>& params,
                  ContentAction action,
                  ContentSource src,
                  std::string* error_desc);
-  void ActivateRtcpMux_w();
-  bool SetRtcpMux_w(bool enable,
+  void ActivateRtcpMux_n();
+  bool SetRtcpMux_n(bool enable,
                     ContentAction action,
                     ContentSource src,
                     std::string* error_desc);
@@ -283,7 +302,7 @@ class BaseChannel
 
   // Handled in derived classes
   // Get the SRTP crypto suites to use for RTP media
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0;
+  virtual void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const = 0;
   virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
       const std::vector<ConnectionInfo>& infos) = 0;
 
@@ -294,13 +313,24 @@ class BaseChannel
   }
 
  private:
-  rtc::Thread* worker_thread_;
-  TransportController* transport_controller_;
-  MediaChannel* media_channel_;
-  std::vector<StreamParams> local_streams_;
-  std::vector<StreamParams> remote_streams_;
+  bool InitNetwork_n(const std::string* bundle_transport_name);
+  void DisconnectTransportChannels_n();
+  void DestroyTransportChannels_n();
+  void SignalSentPacket_n(TransportChannel* channel,
+                          const rtc::SentPacket& sent_packet);
+  void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
+  bool IsTransportReadyToSend_n() const;
+  void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
+
+  rtc::Thread* const worker_thread_;
+  rtc::Thread* const network_thread_;
+  rtc::AsyncInvoker invoker_;
 
   const std::string content_name_;
+  std::unique_ptr<ConnectionMonitor> connection_monitor_;
+
+  // Transport related members that should be accessed from network thread.
+  TransportController* const transport_controller_;
   std::string transport_name_;
   bool rtcp_transport_enabled_;
   TransportChannel* transport_channel_;
@@ -310,32 +340,40 @@ class BaseChannel
   SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
   BundleFilter bundle_filter_;
-  std::unique_ptr<ConnectionMonitor> connection_monitor_;
-  bool enabled_;
-  bool writable_;
   bool rtp_ready_to_send_;
   bool rtcp_ready_to_send_;
+  bool writable_;
   bool was_ever_writable_;
-  MediaContentDirection local_content_direction_;
-  MediaContentDirection remote_content_direction_;
   bool has_received_packet_;
   bool dtls_keyed_;
   bool secure_required_;
   int rtp_abs_sendtime_extn_id_;
+
+  // MediaChannel related members that should be access from worker thread.
+  MediaChannel* const media_channel_;
+  // Currently enabled_ flag accessed from signaling thread too, but it can
+  // be changed only when signaling thread does sunchronious call to worker
+  // thread, so it should be safe.
+  bool enabled_;
+  std::vector<StreamParams> local_streams_;
+  std::vector<StreamParams> remote_streams_;
+  MediaContentDirection local_content_direction_;
+  MediaContentDirection remote_content_direction_;
 };
 
 // VoiceChannel is a specialization that adds support for early media, DTMF,
 // and input/output level monitoring.
 class VoiceChannel : public BaseChannel {
  public:
-  VoiceChannel(rtc::Thread* thread,
+  VoiceChannel(rtc::Thread* worker_thread,
+               rtc::Thread* network_thread,
                MediaEngineInterface* media_engine,
                VoiceMediaChannel* channel,
                TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp);
   ~VoiceChannel();
-  bool Init();
+  bool Init_w(const std::string* bundle_transport_name);
 
   // Configure sending media on the stream with SSRC |ssrc|
   // If there is only one sending stream SSRC 0 can be used.
@@ -345,7 +383,7 @@ class VoiceChannel : public BaseChannel {
                     AudioSource* source);
 
   // downcasts a MediaChannel
-  virtual VoiceMediaChannel* media_channel() const {
+  VoiceMediaChannel* media_channel() const override {
     return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
   }
 
@@ -366,8 +404,12 @@ class VoiceChannel : public BaseChannel {
   bool SetOutputVolume(uint32_t ssrc, double volume);
   void SetRawAudioSink(uint32_t ssrc,
                        std::unique_ptr<webrtc::AudioSinkInterface> sink);
-  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const;
-  bool SetRtpParameters(uint32_t ssrc, const webrtc::RtpParameters& parameters);
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const;
+  bool SetRtpSendParameters(uint32_t ssrc,
+                            const webrtc::RtpParameters& parameters);
+  webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const;
+  bool SetRtpReceiveParameters(uint32_t ssrc,
+                               const webrtc::RtpParameters& parameters);
 
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
@@ -388,34 +430,39 @@ class VoiceChannel : public BaseChannel {
   int GetInputLevel_w();
   int GetOutputLevel_w();
   void GetActiveStreams_w(AudioInfo::StreamList* actives);
-  webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
-  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
+  webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
+  bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
+  webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const;
+  bool SetRtpReceiveParameters_w(uint32_t ssrc,
+                                 webrtc::RtpParameters parameters);
 
  private:
   // overrides from BaseChannel
-  virtual void OnChannelRead(TransportChannel* channel,
-                             const char* data, size_t len,
-                             const rtc::PacketTime& packet_time,
-                             int flags);
-  virtual void ChangeState();
-  virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
-  virtual bool SetLocalContent_w(const MediaContentDescription* content,
-                                 ContentAction action,
-                                 std::string* error_desc);
-  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
-                                  ContentAction action,
-                                  std::string* error_desc);
+  void OnChannelRead(TransportChannel* channel,
+                     const char* data,
+                     size_t len,
+                     const rtc::PacketTime& packet_time,
+                     int flags) override;
+  void ChangeState_w() override;
+  const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
+  bool SetLocalContent_w(const MediaContentDescription* content,
+                         ContentAction action,
+                         std::string* error_desc) override;
+  bool SetRemoteContent_w(const MediaContentDescription* content,
+                          ContentAction action,
+                          std::string* error_desc) override;
   void HandleEarlyMediaTimeout();
   bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
   bool SetOutputVolume_w(uint32_t ssrc, double volume);
   bool GetStats_w(VoiceMediaInfo* stats);
 
-  virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
-  virtual void OnConnectionMonitorUpdate(
-      ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
-  virtual void OnMediaMonitorUpdate(
-      VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
+  void OnMessage(rtc::Message* pmsg) override;
+  void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
+  void OnConnectionMonitorUpdate(
+      ConnectionMonitor* monitor,
+      const std::vector<ConnectionInfo>& infos) override;
+  void OnMediaMonitorUpdate(VoiceMediaChannel* media_channel,
+                            const VoiceMediaInfo& info);
   void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
 
   static const int kEarlyMediaTimeout = 1000;
@@ -435,21 +482,25 @@ class VoiceChannel : public BaseChannel {
 // VideoChannel is a specialization for video.
 class VideoChannel : public BaseChannel {
  public:
-  VideoChannel(rtc::Thread* thread,
+  VideoChannel(rtc::Thread* worker_thread,
+               rtc::Thread* netwokr_thread,
                VideoMediaChannel* channel,
                TransportController* transport_controller,
                const std::string& content_name,
                bool rtcp);
   ~VideoChannel();
-  bool Init();
+  bool Init_w(const std::string* bundle_transport_name);
 
   // downcasts a MediaChannel
-  virtual VideoMediaChannel* media_channel() const {
+  VideoMediaChannel* media_channel() const override {
     return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
   }
 
   bool SetSink(uint32_t ssrc, rtc::VideoSinkInterface<VideoFrame>* sink);
-  bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer);
+  // Register a source. The |ssrc| must correspond to a registered
+  // send stream.
+  void SetSource(uint32_t ssrc,
+                 rtc::VideoSourceInterface<cricket::VideoFrame>* source);
   // Get statistics about the current media session.
   bool GetStats(VideoMediaInfo* stats);
 
@@ -461,29 +512,37 @@ class VideoChannel : public BaseChannel {
   sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
 
   bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
-  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const;
-  bool SetRtpParameters(uint32_t ssrc, const webrtc::RtpParameters& parameters);
+  webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const;
+  bool SetRtpSendParameters(uint32_t ssrc,
+                            const webrtc::RtpParameters& parameters);
+  webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const;
+  bool SetRtpReceiveParameters(uint32_t ssrc,
+                               const webrtc::RtpParameters& parameters);
 
  private:
   // overrides from BaseChannel
-  virtual void ChangeState();
-  virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
-  virtual bool SetLocalContent_w(const MediaContentDescription* content,
-                                 ContentAction action,
-                                 std::string* error_desc);
-  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
-                                  ContentAction action,
-                                  std::string* error_desc);
+  void ChangeState_w() override;
+  const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
+  bool SetLocalContent_w(const MediaContentDescription* content,
+                         ContentAction action,
+                         std::string* error_desc) override;
+  bool SetRemoteContent_w(const MediaContentDescription* content,
+                          ContentAction action,
+                          std::string* error_desc) override;
   bool GetStats_w(VideoMediaInfo* stats);
-  webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
-  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
+  webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
+  bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
+  webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const;
+  bool SetRtpReceiveParameters_w(uint32_t ssrc,
+                                 webrtc::RtpParameters parameters);
 
-  virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
-  virtual void OnConnectionMonitorUpdate(
-      ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
-  virtual void OnMediaMonitorUpdate(
-      VideoMediaChannel* media_channel, const VideoMediaInfo& info);
+  void OnMessage(rtc::Message* pmsg) override;
+  void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
+  void OnConnectionMonitorUpdate(
+      ConnectionMonitor* monitor,
+      const std::vector<ConnectionInfo>& infos) override;
+  void OnMediaMonitorUpdate(VideoMediaChannel* media_channel,
+                            const VideoMediaInfo& info);
 
   std::unique_ptr<VideoMediaMonitor> media_monitor_;
 
@@ -498,13 +557,14 @@ class VideoChannel : public BaseChannel {
 // DataChannel is a specialization for data.
 class DataChannel : public BaseChannel {
  public:
-  DataChannel(rtc::Thread* thread,
+  DataChannel(rtc::Thread* worker_thread,
+              rtc::Thread* network_thread,
               DataMediaChannel* media_channel,
               TransportController* transport_controller,
               const std::string& content_name,
               bool rtcp);
   ~DataChannel();
-  bool Init();
+  bool Init_w(const std::string* bundle_transport_name);
 
   virtual bool SendData(const SendDataParams& params,
                         const rtc::CopyOnWriteBuffer& payload,
@@ -532,7 +592,7 @@ class DataChannel : public BaseChannel {
 
  protected:
   // downcasts a MediaChannel.
-  virtual DataMediaChannel* media_channel() const {
+  DataMediaChannel* media_channel() const override {
     return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
   }
 
@@ -569,7 +629,7 @@ class DataChannel : public BaseChannel {
   typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData;
 
   // overrides from BaseChannel
-  virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
+  const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
   // If data_channel_type_ is DCT_NONE, set it.  Otherwise, check that
   // it's the same as what was set previously.  Returns false if it's
   // set to one type one type and changed to another type later.
@@ -579,22 +639,23 @@ class DataChannel : public BaseChannel {
   // DataContentDescription.
   bool SetDataChannelTypeFromContent(const DataContentDescription* content,
                                      std::string* error_desc);
-  virtual bool SetLocalContent_w(const MediaContentDescription* content,
-                                 ContentAction action,
-                                 std::string* error_desc);
-  virtual bool SetRemoteContent_w(const MediaContentDescription* content,
-                                  ContentAction action,
-                                  std::string* error_desc);
-  virtual void ChangeState();
-  virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
+  bool SetLocalContent_w(const MediaContentDescription* content,
+                         ContentAction action,
+                         std::string* error_desc) override;
+  bool SetRemoteContent_w(const MediaContentDescription* content,
+                          ContentAction action,
+                          std::string* error_desc) override;
+  void ChangeState_w() override;
+  bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) override;
 
-  virtual void OnMessage(rtc::Message* pmsg);
-  virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
-  virtual void OnConnectionMonitorUpdate(
-      ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
-  virtual void OnMediaMonitorUpdate(
-      DataMediaChannel* media_channel, const DataMediaInfo& info);
-  virtual bool ShouldSetupDtlsSrtp() const;
+  void OnMessage(rtc::Message* pmsg) override;
+  void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
+  void OnConnectionMonitorUpdate(
+      ConnectionMonitor* monitor,
+      const std::vector<ConnectionInfo>& infos) override;
+  void OnMediaMonitorUpdate(DataMediaChannel* media_channel,
+                            const DataMediaInfo& info);
+  bool ShouldSetupDtlsSrtp_n() const override;
   void OnDataReceived(
       const ReceiveDataParams& params, const char* data, size_t len);
   void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
diff --git a/chromium/third_party/webrtc/pc/channel_unittest.cc b/chromium/third_party/webrtc/pc/channel_unittest.cc
index a423842e00d..6c6a3c10e43 100644
--- a/chromium/third_party/webrtc/pc/channel_unittest.cc
+++ b/chromium/third_party/webrtc/pc/channel_unittest.cc
@@ -10,24 +10,13 @@
 
 #include <memory>
 
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/fileutils.h"
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/signalthread.h"
-#include "webrtc/base/ssladapter.h"
-#include "webrtc/base/sslidentity.h"
-#include "webrtc/base/window.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/fakertp.h"
-#include "webrtc/media/base/fakescreencapturerfactory.h"
-#include "webrtc/media/base/fakevideocapturer.h"
 #include "webrtc/media/base/mediachannel.h"
-#include "webrtc/media/base/rtpdump.h"
-#include "webrtc/media/base/screencastid.h"
 #include "webrtc/media/base/testutils.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/channel.h"
@@ -46,19 +35,21 @@ using cricket::FakeVoiceMediaChannel;
 using cricket::ScreencastId;
 using cricket::StreamParams;
 using cricket::TransportChannel;
-using rtc::WindowId;
-
-static const cricket::AudioCodec kPcmuCodec(0, "PCMU", 64000, 8000, 1, 0);
-static const cricket::AudioCodec kPcmaCodec(8, "PCMA", 64000, 8000, 1, 0);
-static const cricket::AudioCodec kIsacCodec(103, "ISAC", 40000, 16000, 1, 0);
-static const cricket::VideoCodec kH264Codec(97, "H264", 640, 400, 30, 0);
-static const cricket::VideoCodec kH264SvcCodec(99, "H264-SVC", 320, 200, 15, 0);
-static const cricket::DataCodec kGoogleDataCodec(101, "google-data", 0);
-static const uint32_t kSsrc1 = 0x1111;
-static const uint32_t kSsrc2 = 0x2222;
-static const uint32_t kSsrc3 = 0x3333;
-static const int kAudioPts[] = {0, 8};
-static const int kVideoPts[] = {97, 99};
+
+namespace {
+const cricket::AudioCodec kPcmuCodec(0, "PCMU", 64000, 8000, 1);
+const cricket::AudioCodec kPcmaCodec(8, "PCMA", 64000, 8000, 1);
+const cricket::AudioCodec kIsacCodec(103, "ISAC", 40000, 16000, 1);
+const cricket::VideoCodec kH264Codec(97, "H264", 640, 400, 30);
+const cricket::VideoCodec kH264SvcCodec(99, "H264-SVC", 320, 200, 15);
+const cricket::DataCodec kGoogleDataCodec(101, "google-data");
+const uint32_t kSsrc1 = 0x1111;
+const uint32_t kSsrc2 = 0x2222;
+const uint32_t kSsrc3 = 0x3333;
+const int kAudioPts[] = {0, 8};
+const int kVideoPts[] = {97, 99};
+enum class NetworkIsWorker { Yes, No };
+}  // namespace
 
 template <class ChannelT,
           class MediaChannelT,
@@ -76,10 +67,6 @@ class Traits {
   typedef OptionsT Options;
 };
 
-// Controls how long we wait for a session to send messages that we
-// expect, in milliseconds.  We put it high to avoid flaky tests.
-static const int kEventTimeout = 5000;
-
 class VoiceTraits : public Traits<cricket::VoiceChannel,
                                   cricket::FakeVoiceMediaChannel,
                                   cricket::AudioContentDescription,
@@ -101,12 +88,7 @@ class DataTraits : public Traits<cricket::DataChannel,
                                  cricket::DataMediaInfo,
                                  cricket::DataOptions> {};
 
-rtc::StreamInterface* Open(const std::string& path) {
-  return rtc::Filesystem::OpenFile(
-      rtc::Pathname(path), "wb");
-}
-
-// Base class for Voice/VideoChannel tests
+// Base class for Voice/Video/DataChannel tests
 template<class T>
 class ChannelTest : public testing::Test, public sigslot::has_slots<> {
  public:
@@ -114,40 +96,52 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
                DTLS = 0x10 };
 
   ChannelTest(bool verify_playout,
-              const uint8_t* rtp_data,
-              int rtp_len,
-              const uint8_t* rtcp_data,
-              int rtcp_len)
+              rtc::ArrayView<const uint8_t> rtp_data,
+              rtc::ArrayView<const uint8_t> rtcp_data,
+              NetworkIsWorker network_is_worker)
       : verify_playout_(verify_playout),
-        transport_controller1_(cricket::ICEROLE_CONTROLLING),
-        transport_controller2_(cricket::ICEROLE_CONTROLLED),
         media_channel1_(NULL),
         media_channel2_(NULL),
-        rtp_packet_(reinterpret_cast<const char*>(rtp_data), rtp_len),
-        rtcp_packet_(reinterpret_cast<const char*>(rtcp_data), rtcp_len),
+        rtp_packet_(rtp_data.data(), rtp_data.size()),
+        rtcp_packet_(rtcp_data.data(), rtcp_data.size()),
         media_info_callbacks1_(),
-        media_info_callbacks2_() {}
+        media_info_callbacks2_() {
+    if (network_is_worker == NetworkIsWorker::Yes) {
+      network_thread_ = rtc::Thread::Current();
+    } else {
+      network_thread_keeper_ = rtc::Thread::Create();
+      network_thread_keeper_->SetName("Network", nullptr);
+      network_thread_keeper_->Start();
+      network_thread_ = network_thread_keeper_.get();
+    }
+    transport_controller1_.reset(new cricket::FakeTransportController(
+        network_thread_, cricket::ICEROLE_CONTROLLING));
+    transport_controller2_.reset(new cricket::FakeTransportController(
+        network_thread_, cricket::ICEROLE_CONTROLLED));
+  }
 
   void CreateChannels(int flags1, int flags2) {
     CreateChannels(new typename T::MediaChannel(NULL, typename T::Options()),
                    new typename T::MediaChannel(NULL, typename T::Options()),
-                   flags1, flags2, rtc::Thread::Current());
+                   flags1, flags2);
   }
-  void CreateChannels(
-      typename T::MediaChannel* ch1, typename T::MediaChannel* ch2,
-      int flags1, int flags2, rtc::Thread* thread) {
+  void CreateChannels(typename T::MediaChannel* ch1,
+                      typename T::MediaChannel* ch2,
+                      int flags1,
+                      int flags2) {
+    rtc::Thread* worker_thread = rtc::Thread::Current();
     media_channel1_ = ch1;
     media_channel2_ = ch2;
-    channel1_.reset(CreateChannel(thread, &media_engine_, ch1,
-                                  &transport_controller1_,
-                                  (flags1 & RTCP) != 0));
-    channel2_.reset(CreateChannel(thread, &media_engine_, ch2,
-                                  &transport_controller2_,
-                                  (flags2 & RTCP) != 0));
-    channel1_->SignalMediaMonitor.connect(
-        this, &ChannelTest<T>::OnMediaMonitor);
-    channel2_->SignalMediaMonitor.connect(
-        this, &ChannelTest<T>::OnMediaMonitor);
+    channel1_.reset(
+        CreateChannel(worker_thread, network_thread_, &media_engine_, ch1,
+                      transport_controller1_.get(), (flags1 & RTCP) != 0));
+    channel2_.reset(
+        CreateChannel(worker_thread, network_thread_, &media_engine_, ch2,
+                      transport_controller2_.get(), (flags2 & RTCP) != 0));
+    channel1_->SignalMediaMonitor.connect(this,
+                                          &ChannelTest<T>::OnMediaMonitor1);
+    channel2_->SignalMediaMonitor.connect(this,
+                                          &ChannelTest<T>::OnMediaMonitor2);
     if ((flags1 & DTLS) && (flags2 & DTLS)) {
       flags1 = (flags1 & ~SECURE);
       flags2 = (flags2 & ~SECURE);
@@ -161,14 +155,14 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
 
     if (flags1 & DTLS) {
       // Confirmed to work with KT_RSA and KT_ECDSA.
-      transport_controller1_.SetLocalCertificate(
-          rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+      transport_controller1_->SetLocalCertificate(
+          rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
               rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT))));
     }
     if (flags2 & DTLS) {
       // Confirmed to work with KT_RSA and KT_ECDSA.
-      transport_controller2_.SetLocalCertificate(
-          rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+      transport_controller2_->SetLocalCertificate(
+          rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
               rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT))));
     }
 
@@ -187,14 +181,16 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     }
   }
   typename T::Channel* CreateChannel(
-      rtc::Thread* thread,
+      rtc::Thread* worker_thread,
+      rtc::Thread* network_thread,
       cricket::MediaEngineInterface* engine,
       typename T::MediaChannel* ch,
       cricket::TransportController* transport_controller,
       bool rtcp) {
-    typename T::Channel* channel = new typename T::Channel(
-        thread, engine, ch, transport_controller, cricket::CN_AUDIO, rtcp);
-    if (!channel->Init()) {
+    typename T::Channel* channel =
+        new typename T::Channel(worker_thread, network_thread, engine, ch,
+                                transport_controller, cricket::CN_AUDIO, rtcp);
+    if (!channel->Init_w(nullptr)) {
       delete channel;
       channel = NULL;
     }
@@ -209,7 +205,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
       result = channel2_->SetRemoteContent(&remote_media_content1_,
                                            CA_OFFER, NULL);
       if (result) {
-        transport_controller1_.Connect(&transport_controller2_);
+        transport_controller1_->Connect(transport_controller2_.get());
 
         result = channel2_->SetLocalContent(&local_media_content2_,
                                             CA_ANSWER, NULL);
@@ -242,7 +238,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
       channel2_->Enable(true);
       result = channel1_->SetRemoteContent(&remote_media_content2_,
                                            CA_PRANSWER, NULL);
-      transport_controller1_.Connect(&transport_controller2_);
+      transport_controller1_->Connect(transport_controller2_.get());
     }
     return result;
   }
@@ -269,105 +265,92 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     return channel1_->RemoveRecvStream(id);
   }
 
-  // Calling "_w" method here is ok since we only use one thread for this test
   cricket::FakeTransport* GetTransport1() {
-    return transport_controller1_.GetTransport_w(channel1_->content_name());
+    std::string name = channel1_->content_name();
+    return network_thread_->Invoke<cricket::FakeTransport*>(
+        [this, name] { return transport_controller1_->GetTransport_n(name); });
   }
   cricket::FakeTransport* GetTransport2() {
-    return transport_controller2_.GetTransport_w(channel2_->content_name());
+    std::string name = channel2_->content_name();
+    return network_thread_->Invoke<cricket::FakeTransport*>(
+        [this, name] { return transport_controller2_->GetTransport_n(name); });
   }
 
-  bool SendRtp1() {
-    return media_channel1_->SendRtp(rtp_packet_.c_str(),
-                                    static_cast<int>(rtp_packet_.size()),
-                                    rtc::PacketOptions());
+  void SendRtp1() {
+    media_channel1_->SendRtp(rtp_packet_.data(), rtp_packet_.size(),
+                             rtc::PacketOptions());
   }
-  bool SendRtp2() {
-    return media_channel2_->SendRtp(rtp_packet_.c_str(),
-                                    static_cast<int>(rtp_packet_.size()),
-                                    rtc::PacketOptions());
+  void SendRtp2() {
+    media_channel2_->SendRtp(rtp_packet_.data(), rtp_packet_.size(),
+                             rtc::PacketOptions());
   }
-  bool SendRtcp1() {
-    return media_channel1_->SendRtcp(rtcp_packet_.c_str(),
-                                     static_cast<int>(rtcp_packet_.size()));
+  void SendRtcp1() {
+    media_channel1_->SendRtcp(rtcp_packet_.data(), rtcp_packet_.size());
   }
-  bool SendRtcp2() {
-    return media_channel2_->SendRtcp(rtcp_packet_.c_str(),
-                                     static_cast<int>(rtcp_packet_.size()));
+  void SendRtcp2() {
+    media_channel2_->SendRtcp(rtcp_packet_.data(), rtcp_packet_.size());
   }
   // Methods to send custom data.
-  bool SendCustomRtp1(uint32_t ssrc, int sequence_number, int pl_type = -1) {
-    std::string data(CreateRtpData(ssrc, sequence_number, pl_type));
-    return media_channel1_->SendRtp(data.c_str(), static_cast<int>(data.size()),
-                                    rtc::PacketOptions());
-  }
-  bool SendCustomRtp2(uint32_t ssrc, int sequence_number, int pl_type = -1) {
-    std::string data(CreateRtpData(ssrc, sequence_number, pl_type));
-    return media_channel2_->SendRtp(data.c_str(), static_cast<int>(data.size()),
-                                    rtc::PacketOptions());
-  }
-  bool SendCustomRtcp1(uint32_t ssrc) {
-    std::string data(CreateRtcpData(ssrc));
-    return media_channel1_->SendRtcp(data.c_str(),
-                                     static_cast<int>(data.size()));
-  }
-  bool SendCustomRtcp2(uint32_t ssrc) {
-    std::string data(CreateRtcpData(ssrc));
-    return media_channel2_->SendRtcp(data.c_str(),
-                                     static_cast<int>(data.size()));
+  void SendCustomRtp1(uint32_t ssrc, int sequence_number, int pl_type = -1) {
+    rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
+    media_channel1_->SendRtp(data.data(), data.size(), rtc::PacketOptions());
+  }
+  void SendCustomRtp2(uint32_t ssrc, int sequence_number, int pl_type = -1) {
+    rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
+    media_channel2_->SendRtp(data.data(), data.size(), rtc::PacketOptions());
+  }
+  void SendCustomRtcp1(uint32_t ssrc) {
+    rtc::Buffer data = CreateRtcpData(ssrc);
+    media_channel1_->SendRtcp(data.data(), data.size());
   }
+  void SendCustomRtcp2(uint32_t ssrc) {
+    rtc::Buffer data = CreateRtcpData(ssrc);
+    media_channel2_->SendRtcp(data.data(), data.size());
+  }
+
   bool CheckRtp1() {
-    return media_channel1_->CheckRtp(rtp_packet_.c_str(),
-                                     static_cast<int>(rtp_packet_.size()));
+    return media_channel1_->CheckRtp(rtp_packet_.data(), rtp_packet_.size());
   }
   bool CheckRtp2() {
-    return media_channel2_->CheckRtp(rtp_packet_.c_str(),
-                                     static_cast<int>(rtp_packet_.size()));
+    return media_channel2_->CheckRtp(rtp_packet_.data(), rtp_packet_.size());
   }
   bool CheckRtcp1() {
-    return media_channel1_->CheckRtcp(rtcp_packet_.c_str(),
-                                      static_cast<int>(rtcp_packet_.size()));
+    return media_channel1_->CheckRtcp(rtcp_packet_.data(), rtcp_packet_.size());
   }
   bool CheckRtcp2() {
-    return media_channel2_->CheckRtcp(rtcp_packet_.c_str(),
-                                      static_cast<int>(rtcp_packet_.size()));
+    return media_channel2_->CheckRtcp(rtcp_packet_.data(), rtcp_packet_.size());
   }
   // Methods to check custom data.
   bool CheckCustomRtp1(uint32_t ssrc, int sequence_number, int pl_type = -1) {
-    std::string data(CreateRtpData(ssrc, sequence_number, pl_type));
-    return media_channel1_->CheckRtp(data.c_str(),
-                                     static_cast<int>(data.size()));
+    rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
+    return media_channel1_->CheckRtp(data.data(), data.size());
   }
   bool CheckCustomRtp2(uint32_t ssrc, int sequence_number, int pl_type = -1) {
-    std::string data(CreateRtpData(ssrc, sequence_number, pl_type));
-    return media_channel2_->CheckRtp(data.c_str(),
-                                     static_cast<int>(data.size()));
+    rtc::Buffer data = CreateRtpData(ssrc, sequence_number, pl_type);
+    return media_channel2_->CheckRtp(data.data(), data.size());
   }
   bool CheckCustomRtcp1(uint32_t ssrc) {
-    std::string data(CreateRtcpData(ssrc));
-    return media_channel1_->CheckRtcp(data.c_str(),
-                                      static_cast<int>(data.size()));
+    rtc::Buffer data = CreateRtcpData(ssrc);
+    return media_channel1_->CheckRtcp(data.data(), data.size());
   }
   bool CheckCustomRtcp2(uint32_t ssrc) {
-    std::string data(CreateRtcpData(ssrc));
-    return media_channel2_->CheckRtcp(data.c_str(),
-                                      static_cast<int>(data.size()));
+    rtc::Buffer data = CreateRtcpData(ssrc);
+    return media_channel2_->CheckRtcp(data.data(), data.size());
   }
-  std::string CreateRtpData(uint32_t ssrc, int sequence_number, int pl_type) {
-    std::string data(rtp_packet_);
+  rtc::Buffer CreateRtpData(uint32_t ssrc, int sequence_number, int pl_type) {
+    rtc::Buffer data(rtp_packet_.data(), rtp_packet_.size());
     // Set SSRC in the rtp packet copy.
-    rtc::SetBE32(const_cast<char*>(data.c_str()) + 8, ssrc);
-    rtc::SetBE16(const_cast<char*>(data.c_str()) + 2, sequence_number);
+    rtc::SetBE32(data.data() + 8, ssrc);
+    rtc::SetBE16(data.data() + 2, sequence_number);
     if (pl_type >= 0) {
-      rtc::Set8(const_cast<char*>(data.c_str()), 1,
-                static_cast<uint8_t>(pl_type));
+      rtc::Set8(data.data(), 1, static_cast<uint8_t>(pl_type));
     }
     return data;
   }
-  std::string CreateRtcpData(uint32_t ssrc) {
-    std::string data(rtcp_packet_);
+  rtc::Buffer CreateRtcpData(uint32_t ssrc) {
+    rtc::Buffer data(rtcp_packet_.data(), rtcp_packet_.size());
     // Set SSRC in the rtcp packet copy.
-    rtc::SetBE32(const_cast<char*>(data.c_str()) + 4, ssrc);
+    rtc::SetBE32(data.data() + 4, ssrc);
     return data;
   }
 
@@ -408,86 +391,40 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
      return sdesc;
   }
 
-  class CallThread : public rtc::SignalThread {
-   public:
-    typedef bool (ChannelTest<T>::*Method)();
-    CallThread(ChannelTest<T>* obj, Method method, bool* result = nullptr)
-        : obj_(obj),
-          method_(method),
-          result_(false),
-          result_ptr_(result) {
-      if (result_ptr_)
-        *result_ptr_ = false;
-    }
-
-    ~CallThread() {
-      if (result_ptr_) {
-        rtc::CritScope cs(&result_lock_);
-        *result_ptr_ = result_;
-      }
-    }
-
-    virtual void DoWork() {
-      SetResult((*obj_.*method_)());
-    }
-
-    bool result() {
-      rtc::CritScope cs(&result_lock_);
-      return result_;
-    }
-
-   private:
-    void SetResult(const bool& result) {
-      rtc::CritScope cs(&result_lock_);
-      result_ = result;
-    }
-
-    ChannelTest<T>* obj_;
-    Method method_;
-    rtc::CriticalSection result_lock_;
-    bool result_ GUARDED_BY(result_lock_);
-    bool* result_ptr_;
-  };
-
   // Will manage the lifetime of a CallThread, making sure it's
   // destroyed before this object goes out of scope.
   class ScopedCallThread {
    public:
-    using Method = typename CallThread::Method;
-
-    ScopedCallThread(ChannelTest<T>* obj, Method method)
-        : thread_(new CallThread(obj, method)) {
+    template <class FunctorT>
+    ScopedCallThread(const FunctorT& functor)
+        : thread_(rtc::Thread::Create()),
+          task_(new rtc::FunctorMessageHandler<void, FunctorT>(functor)) {
       thread_->Start();
+      thread_->Post(task_.get());
     }
 
-    ~ScopedCallThread() {
-      thread_->Destroy(true);
-    }
+    ~ScopedCallThread() { thread_->Stop(); }
 
-    bool result() const { return thread_->result(); }
+    rtc::Thread* thread() { return thread_.get(); }
 
    private:
-    CallThread* thread_;
+    std::unique_ptr<rtc::Thread> thread_;
+    std::unique_ptr<rtc::MessageHandler> task_;
   };
 
-  void CallOnThreadAndWaitForDone(typename CallThread::Method method,
-                                  bool* result) {
-    CallThread* thread = new CallThread(this, method, result);
-    thread->Start();
-    thread->Destroy(true);
-  }
-
   bool CodecMatches(const typename T::Codec& c1, const typename T::Codec& c2) {
     return false;  // overridden in specialized classes
   }
 
-  void OnMediaMonitor(typename T::Channel* channel,
-                      const typename T::MediaInfo& info) {
-    if (channel == channel1_.get()) {
-      media_info_callbacks1_++;
-    } else if (channel == channel2_.get()) {
-      media_info_callbacks2_++;
-    }
+  void OnMediaMonitor1(typename T::Channel* channel,
+                       const typename T::MediaInfo& info) {
+    RTC_DCHECK_EQ(channel, channel1_.get());
+    media_info_callbacks1_++;
+  }
+  void OnMediaMonitor2(typename T::Channel* channel,
+                       const typename T::MediaInfo& info) {
+    RTC_DCHECK_EQ(channel, channel2_.get());
+    media_info_callbacks2_++;
   }
 
   cricket::CandidatePairInterface* last_selected_candidate_pair() {
@@ -792,7 +729,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
 
     EXPECT_TRUE(channel2_->SetRemoteContent(&content1, CA_OFFER, NULL));
     EXPECT_EQ(1u, media_channel2_->recv_streams().size());
-    transport_controller1_.Connect(&transport_controller2_);
+    transport_controller1_->Connect(transport_controller2_.get());
 
     // Channel 2 do not send anything.
     typename T::Content content2;
@@ -803,7 +740,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_TRUE(channel2_->Enable(true));
     EXPECT_EQ(0u, media_channel2_->send_streams().size());
 
-    EXPECT_TRUE(SendCustomRtp1(kSsrc1, 0));
+    SendCustomRtp1(kSsrc1, 0);
+    WaitForThreads();
     EXPECT_TRUE(CheckCustomRtp2(kSsrc1, 0));
 
     // Let channel 2 update the content by sending |stream2| and enable SRTP.
@@ -829,7 +767,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
 
     EXPECT_TRUE(channel1_->secure());
     EXPECT_TRUE(channel2_->secure());
-    EXPECT_TRUE(SendCustomRtp2(kSsrc2, 0));
+    SendCustomRtp2(kSsrc2, 0);
+    WaitForThreads();
     EXPECT_TRUE(CheckCustomRtp1(kSsrc2, 0));
   }
 
@@ -867,7 +806,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
       EXPECT_FALSE(media_channel2_->playout());
     }
     EXPECT_FALSE(media_channel2_->sending());
-    transport_controller1_.Connect(&transport_controller2_);
+    transport_controller1_->Connect(transport_controller2_.get());
     if (verify_playout_) {
       EXPECT_TRUE(media_channel1_->playout());
     }
@@ -915,7 +854,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_TRUE(channel2_->SetRemoteContent(&content1, CA_OFFER, NULL));
     EXPECT_TRUE(channel2_->SetLocalContent(&content2, CA_PRANSWER, NULL));
     EXPECT_TRUE(channel1_->SetRemoteContent(&content2, CA_PRANSWER, NULL));
-    transport_controller1_.Connect(&transport_controller2_);
+    transport_controller1_->Connect(transport_controller2_.get());
 
     if (verify_playout_) {
       EXPECT_TRUE(media_channel1_->playout());
@@ -958,39 +897,47 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
   // Tests that when the transport channel signals a candidate pair change
   // event, the media channel will receive a call on the network route change.
   void TestNetworkRouteChanges() {
+    constexpr uint16_t kLocalNetId = 1;
+    constexpr uint16_t kRemoteNetId = 2;
+    constexpr int kLastPacketId = 100;
+
     CreateChannels(0, 0);
 
     cricket::TransportChannel* transport_channel1 =
         channel1_->transport_channel();
-    ASSERT_TRUE(transport_channel1 != nullptr);
+    ASSERT_TRUE(transport_channel1);
     typename T::MediaChannel* media_channel1 =
         static_cast<typename T::MediaChannel*>(channel1_->media_channel());
-    ASSERT_TRUE(media_channel1 != nullptr);
-
-    media_channel1_->set_num_network_route_changes(0);
-    // The transport channel becomes disconnected.
-    transport_channel1->SignalSelectedCandidatePairChanged(transport_channel1,
-                                                           nullptr, -1);
-    EXPECT_EQ(1, media_channel1_->num_network_route_changes());
+    ASSERT_TRUE(media_channel1);
+
+    media_channel1->set_num_network_route_changes(0);
+    network_thread_->Invoke<void>([transport_channel1] {
+      // The transport channel becomes disconnected.
+      transport_channel1->SignalSelectedCandidatePairChanged(transport_channel1,
+                                                             nullptr, -1);
+    });
+    WaitForThreads();
+    EXPECT_EQ(1, media_channel1->num_network_route_changes());
     EXPECT_FALSE(media_channel1->last_network_route().connected);
-
-    media_channel1_->set_num_network_route_changes(0);
-    // The transport channel becomes connected.
-    rtc::SocketAddress local_address("192.168.1.1", 1000 /* port number */);
-    rtc::SocketAddress remote_address("192.168.1.2", 2000 /* port number */);
-    uint16_t local_net_id = 1;
-    uint16_t remote_net_id = 2;
-    int last_packet_id = 100;
-    rtc::scoped_ptr<cricket::CandidatePairInterface> candidate_pair(
-        transport_controller1_.CreateFakeCandidatePair(
-            local_address, local_net_id, remote_address, remote_net_id));
-    transport_channel1->SignalSelectedCandidatePairChanged(
-        transport_channel1, candidate_pair.get(), last_packet_id);
-    EXPECT_EQ(1, media_channel1_->num_network_route_changes());
-    cricket::NetworkRoute expected_network_route(local_net_id, remote_net_id,
-                                                 last_packet_id);
+    media_channel1->set_num_network_route_changes(0);
+
+    network_thread_->Invoke<void>([this, transport_channel1, media_channel1,
+                                   kLocalNetId, kRemoteNetId, kLastPacketId] {
+      // The transport channel becomes connected.
+      rtc::SocketAddress local_address("192.168.1.1", 1000 /* port number */);
+      rtc::SocketAddress remote_address("192.168.1.2", 2000 /* port number */);
+      std::unique_ptr<cricket::CandidatePairInterface> candidate_pair(
+          transport_controller1_->CreateFakeCandidatePair(
+              local_address, kLocalNetId, remote_address, kRemoteNetId));
+      transport_channel1->SignalSelectedCandidatePairChanged(
+          transport_channel1, candidate_pair.get(), kLastPacketId);
+    });
+    WaitForThreads();
+    EXPECT_EQ(1, media_channel1->num_network_route_changes());
+    rtc::NetworkRoute expected_network_route(kLocalNetId, kRemoteNetId,
+                                             kLastPacketId);
     EXPECT_EQ(expected_network_route, media_channel1->last_network_route());
-    EXPECT_EQ(last_packet_id,
+    EXPECT_EQ(kLastPacketId,
               media_channel1->last_network_route().last_sent_packet_id);
   }
 
@@ -1029,8 +976,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
       }
     };
     CreateChannels(new LastWordMediaChannel(), new LastWordMediaChannel(),
-                   RTCP | RTCP_MUX, RTCP | RTCP_MUX,
-                   rtc::Thread::Current());
+                   RTCP | RTCP_MUX, RTCP | RTCP_MUX);
     EXPECT_TRUE(SendInitiate());
     EXPECT_TRUE(SendAccept());
     EXPECT_TRUE(SendTerminate());
@@ -1045,14 +991,28 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     ASSERT_TRUE(GetTransport2());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
     EXPECT_EQ(1U, GetTransport2()->channels().size());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
+    SendRtp1();
+    SendRtp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckNoRtp2());
   }
 
+  void TestDeinit() {
+    CreateChannels(RTCP, RTCP);
+    EXPECT_TRUE(SendInitiate());
+    EXPECT_TRUE(SendAccept());
+    SendRtp1();
+    SendRtp2();
+    SendRtcp1();
+    SendRtcp2();
+    // Do not wait, destroy channels.
+    channel1_.reset(nullptr);
+    channel2_.reset(nullptr);
+  }
+
   // Check that RTCP is not transmitted if both sides don't support RTCP.
   void SendNoRtcpToNoRtcp() {
     CreateChannels(0, 0);
@@ -1062,8 +1022,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     ASSERT_TRUE(GetTransport2());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
     EXPECT_EQ(1U, GetTransport2()->channels().size());
-    EXPECT_FALSE(SendRtcp1());
-    EXPECT_FALSE(SendRtcp2());
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckNoRtcp1());
     EXPECT_TRUE(CheckNoRtcp2());
   }
@@ -1077,8 +1038,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     ASSERT_TRUE(GetTransport2());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
     EXPECT_EQ(2U, GetTransport2()->channels().size());
-    EXPECT_FALSE(SendRtcp1());
-    EXPECT_FALSE(SendRtcp2());
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckNoRtcp1());
     EXPECT_TRUE(CheckNoRtcp2());
   }
@@ -1092,8 +1054,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     ASSERT_TRUE(GetTransport2());
     EXPECT_EQ(2U, GetTransport1()->channels().size());
     EXPECT_EQ(1U, GetTransport2()->channels().size());
-    EXPECT_FALSE(SendRtcp1());
-    EXPECT_FALSE(SendRtcp2());
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckNoRtcp1());
     EXPECT_TRUE(CheckNoRtcp2());
   }
@@ -1107,8 +1070,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     ASSERT_TRUE(GetTransport2());
     EXPECT_EQ(2U, GetTransport1()->channels().size());
     EXPECT_EQ(2U, GetTransport2()->channels().size());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtcp1());
     EXPECT_TRUE(CheckRtcp2());
     EXPECT_TRUE(CheckNoRtcp1());
@@ -1124,8 +1088,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     ASSERT_TRUE(GetTransport2());
     EXPECT_EQ(2U, GetTransport1()->channels().size());
     EXPECT_EQ(2U, GetTransport2()->channels().size());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtcp1());
     EXPECT_TRUE(CheckRtcp2());
     EXPECT_TRUE(CheckNoRtcp1());
@@ -1142,10 +1107,11 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_EQ(1U, GetTransport2()->channels().size());
     EXPECT_TRUE(SendAccept());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtp1();
+    SendRtp2();
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
@@ -1167,10 +1133,11 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_EQ(1U, GetTransport1()->channels().size());
     EXPECT_EQ(1U, GetTransport2()->channels().size());
     EXPECT_TRUE(SendAccept());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtp1();
+    SendRtp2();
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
@@ -1193,10 +1160,11 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_EQ(1U, GetTransport2()->channels().size());
     EXPECT_TRUE(SendAccept());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtp1();
+    SendRtp2();
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
@@ -1220,10 +1188,11 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_EQ(1U, GetTransport2()->channels().size());
     EXPECT_TRUE(SendAccept());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtp1();
+    SendRtp2();
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
@@ -1258,21 +1227,24 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
 
     // RTCP can be sent before the call is accepted, if the transport is ready.
     // It should not be muxed though, as the remote side doesn't support mux.
-    EXPECT_TRUE(SendRtcp1());
+    SendRtcp1();
+    WaitForThreads();
     EXPECT_TRUE(CheckNoRtp2());
     EXPECT_TRUE(CheckRtcp2());
 
     // Send RTCP packet from callee and verify that it is received.
-    EXPECT_TRUE(SendRtcp2());
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckRtcp1());
 
     // Complete call setup and ensure everything is still OK.
     EXPECT_TRUE(SendAccept());
     EXPECT_EQ(2U, GetTransport1()->channels().size());
-    EXPECT_TRUE(SendRtcp1());
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtcp2());
-    EXPECT_TRUE(SendRtcp2());
     EXPECT_TRUE(CheckRtcp1());
   }
 
@@ -1290,19 +1262,23 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
 
     // RTCP can't be sent yet, since the RTCP transport isn't writable, and
     // we haven't yet received the accept that says we should mux.
-    EXPECT_FALSE(SendRtcp1());
+    SendRtcp1();
+    WaitForThreads();
+    EXPECT_TRUE(CheckNoRtcp2());
 
     // Send muxed RTCP packet from callee and verify that it is received.
-    EXPECT_TRUE(SendRtcp2());
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckRtcp1());
 
     // Complete call setup and ensure everything is still OK.
     EXPECT_TRUE(SendAccept());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
-    EXPECT_TRUE(SendRtcp1());
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtcp2());
-    EXPECT_TRUE(SendRtcp2());
     EXPECT_TRUE(CheckRtcp1());
   }
 
@@ -1320,17 +1296,19 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_FALSE(channel1_->secure());
     EXPECT_FALSE(channel2_->secure());
     EXPECT_TRUE(SendInitiate());
-    EXPECT_TRUE_WAIT(channel1_->writable(), kEventTimeout);
-    EXPECT_TRUE_WAIT(channel2_->writable(), kEventTimeout);
+    WaitForThreads();
+    EXPECT_TRUE(channel1_->writable());
+    EXPECT_TRUE(channel2_->writable());
     EXPECT_TRUE(SendAccept());
     EXPECT_TRUE(channel1_->secure());
     EXPECT_TRUE(channel2_->secure());
     EXPECT_EQ(dtls1 && dtls2, channel1_->secure_dtls());
     EXPECT_EQ(dtls1 && dtls2, channel2_->secure_dtls());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtp1();
+    SendRtp2();
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
@@ -1350,10 +1328,11 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_TRUE(SendAccept());
     EXPECT_FALSE(channel1_->secure());
     EXPECT_FALSE(channel2_->secure());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
-    EXPECT_TRUE(SendRtcp1());
-    EXPECT_TRUE(SendRtcp2());
+    SendRtp1();
+    SendRtp2();
+    SendRtcp1();
+    SendRtcp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
@@ -1379,15 +1358,18 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
       ASSERT_TRUE(GetTransport2());
       EXPECT_EQ(2U, GetTransport1()->channels().size());
       EXPECT_EQ(2U, GetTransport2()->channels().size());
-      EXPECT_TRUE(SendCustomRtcp1(kSsrc1));
+      WaitForThreads();  // Wait for 'sending' flag go through network thread.
+      SendCustomRtcp1(kSsrc1);
+      SendCustomRtp1(kSsrc1, ++sequence_number1_1);
+      WaitForThreads();
       EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
-      EXPECT_TRUE(SendCustomRtp1(kSsrc1, ++sequence_number1_1));
       EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
 
       // Send packets from callee and verify that it is received.
-      EXPECT_TRUE(SendCustomRtcp2(kSsrc2));
+      SendCustomRtcp2(kSsrc2);
+      SendCustomRtp2(kSsrc2, ++sequence_number2_2);
+      WaitForThreads();
       EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
-      EXPECT_TRUE(SendCustomRtp2(kSsrc2, ++sequence_number2_2));
       EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
 
       // Complete call setup and ensure everything is still OK.
@@ -1396,13 +1378,14 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
       EXPECT_EQ(1U, GetTransport2()->channels().size());
       EXPECT_TRUE(channel1_->secure());
       EXPECT_TRUE(channel2_->secure());
-      EXPECT_TRUE(SendCustomRtcp1(kSsrc1));
+      SendCustomRtcp1(kSsrc1);
+      SendCustomRtp1(kSsrc1, ++sequence_number1_1);
+      SendCustomRtcp2(kSsrc2);
+      SendCustomRtp2(kSsrc2, ++sequence_number2_2);
+      WaitForThreads();
       EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
-      EXPECT_TRUE(SendCustomRtp1(kSsrc1, ++sequence_number1_1));
       EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
-      EXPECT_TRUE(SendCustomRtcp2(kSsrc2));
       EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
-      EXPECT_TRUE(SendCustomRtp2(kSsrc2, ++sequence_number2_2));
       EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
   }
 
@@ -1411,20 +1394,20 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     CreateChannels(RTCP, RTCP);
     EXPECT_TRUE(SendInitiate());
     EXPECT_TRUE(SendAccept());
-    ScopedCallThread send_rtp1(this, &ChannelTest<T>::SendRtp1);
-    ScopedCallThread send_rtp2(this, &ChannelTest<T>::SendRtp2);
-    ScopedCallThread send_rtcp1(this, &ChannelTest<T>::SendRtcp1);
-    ScopedCallThread send_rtcp2(this, &ChannelTest<T>::SendRtcp2);
-    EXPECT_TRUE_WAIT(CheckRtp1(), 1000);
-    EXPECT_TRUE_WAIT(CheckRtp2(), 1000);
-    EXPECT_TRUE_WAIT(send_rtp1.result(), 1000);
-    EXPECT_TRUE_WAIT(send_rtp2.result(), 1000);
+    ScopedCallThread send_rtp1([this] { SendRtp1(); });
+    ScopedCallThread send_rtp2([this] { SendRtp2(); });
+    ScopedCallThread send_rtcp1([this] { SendRtcp1(); });
+    ScopedCallThread send_rtcp2([this] { SendRtcp2(); });
+    rtc::Thread* involved_threads[] = {send_rtp1.thread(), send_rtp2.thread(),
+                                       send_rtcp1.thread(),
+                                       send_rtcp2.thread()};
+    WaitForThreads(involved_threads);
+    EXPECT_TRUE(CheckRtp1());
+    EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckNoRtp2());
-    EXPECT_TRUE_WAIT(CheckRtcp1(), 1000);
-    EXPECT_TRUE_WAIT(CheckRtcp2(), 1000);
-    EXPECT_TRUE_WAIT(send_rtcp1.result(), 1000);
-    EXPECT_TRUE_WAIT(send_rtcp2.result(), 1000);
+    EXPECT_TRUE(CheckRtcp1());
+    EXPECT_TRUE(CheckRtcp2());
     EXPECT_TRUE(CheckNoRtcp1());
     EXPECT_TRUE(CheckNoRtcp2());
   }
@@ -1434,20 +1417,20 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     CreateChannels(RTCP | SECURE, RTCP | SECURE);
     EXPECT_TRUE(SendInitiate());
     EXPECT_TRUE(SendAccept());
-    ScopedCallThread send_rtp1(this, &ChannelTest<T>::SendRtp1);
-    ScopedCallThread send_rtp2(this, &ChannelTest<T>::SendRtp2);
-    ScopedCallThread send_rtcp1(this, &ChannelTest<T>::SendRtcp1);
-    ScopedCallThread send_rtcp2(this, &ChannelTest<T>::SendRtcp2);
-    EXPECT_TRUE_WAIT(CheckRtp1(), 1000);
-    EXPECT_TRUE_WAIT(CheckRtp2(), 1000);
-    EXPECT_TRUE_WAIT(send_rtp1.result(), 1000);
-    EXPECT_TRUE_WAIT(send_rtp2.result(), 1000);
+    ScopedCallThread send_rtp1([this] { SendRtp1(); });
+    ScopedCallThread send_rtp2([this] { SendRtp2(); });
+    ScopedCallThread send_rtcp1([this] { SendRtcp1(); });
+    ScopedCallThread send_rtcp2([this] { SendRtcp2(); });
+    rtc::Thread* involved_threads[] = {send_rtp1.thread(), send_rtp2.thread(),
+                                       send_rtcp1.thread(),
+                                       send_rtcp2.thread()};
+    WaitForThreads(involved_threads);
+    EXPECT_TRUE(CheckRtp1());
+    EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckNoRtp2());
-    EXPECT_TRUE_WAIT(CheckRtcp1(), 1000);
-    EXPECT_TRUE_WAIT(CheckRtcp2(), 1000);
-    EXPECT_TRUE_WAIT(send_rtcp1.result(), 1000);
-    EXPECT_TRUE_WAIT(send_rtcp2.result(), 1000);
+    EXPECT_TRUE(CheckRtcp1());
+    EXPECT_TRUE(CheckRtcp2());
     EXPECT_TRUE(CheckNoRtcp1());
     EXPECT_TRUE(CheckNoRtcp2());
   }
@@ -1462,45 +1445,54 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     ASSERT_TRUE(GetTransport2());
     EXPECT_EQ(1U, GetTransport1()->channels().size());
     EXPECT_EQ(1U, GetTransport2()->channels().size());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
+    SendRtp1();
+    SendRtp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckNoRtp2());
 
     // Lose writability, which should fail.
-    GetTransport1()->SetWritable(false);
-    EXPECT_FALSE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
+    network_thread_->Invoke<void>(
+        [this] { GetTransport1()->SetWritable(false); });
+    SendRtp1();
+    SendRtp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckNoRtp2());
 
     // Regain writability
-    GetTransport1()->SetWritable(true);
+    network_thread_->Invoke<void>(
+        [this] { GetTransport1()->SetWritable(true); });
     EXPECT_TRUE(media_channel1_->sending());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
+    SendRtp1();
+    SendRtp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckNoRtp2());
 
     // Lose writability completely
-    GetTransport1()->SetDestination(NULL);
+    network_thread_->Invoke<void>(
+        [this] { GetTransport1()->SetDestination(NULL); });
     EXPECT_TRUE(media_channel1_->sending());
 
     // Should fail also.
-    EXPECT_FALSE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
+    SendRtp1();
+    SendRtp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckNoRtp2());
 
     // Gain writability back
-    GetTransport1()->SetDestination(GetTransport2());
+    network_thread_->Invoke<void>(
+        [this] { GetTransport1()->SetDestination(GetTransport2()); });
     EXPECT_TRUE(media_channel1_->sending());
-    EXPECT_TRUE(SendRtp1());
-    EXPECT_TRUE(SendRtp2());
+    SendRtp1();
+    SendRtp2();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtp1());
     EXPECT_TRUE(CheckRtp2());
     EXPECT_TRUE(CheckNoRtp1());
@@ -1537,28 +1529,32 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_FALSE(channel2_->bundle_filter()->FindPayloadType(pl_type2));
 
     // Both channels can receive pl_type1 only.
-    EXPECT_TRUE(SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type1));
+    SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type1);
+    SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type1);
+    WaitForThreads();
     EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1, pl_type1));
-    EXPECT_TRUE(SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type1));
     EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2, pl_type1));
     EXPECT_TRUE(CheckNoRtp1());
     EXPECT_TRUE(CheckNoRtp2());
 
     // RTCP test
-    EXPECT_TRUE(SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type2));
+    SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type2);
+    SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type2);
+    WaitForThreads();
     EXPECT_FALSE(CheckCustomRtp2(kSsrc1, sequence_number1_1, pl_type2));
-    EXPECT_TRUE(SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type2));
     EXPECT_FALSE(CheckCustomRtp1(kSsrc2, sequence_number2_2, pl_type2));
 
-    EXPECT_TRUE(SendCustomRtcp1(kSsrc1));
-    EXPECT_TRUE(SendCustomRtcp2(kSsrc2));
+    SendCustomRtcp1(kSsrc1);
+    SendCustomRtcp2(kSsrc2);
+    WaitForThreads();
     EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
     EXPECT_TRUE(CheckNoRtcp1());
     EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
     EXPECT_TRUE(CheckNoRtcp2());
 
-    EXPECT_TRUE(SendCustomRtcp1(kSsrc2));
-    EXPECT_TRUE(SendCustomRtcp2(kSsrc1));
+    SendCustomRtcp1(kSsrc2);
+    SendCustomRtcp2(kSsrc1);
+    WaitForThreads();
     // Bundle filter shouldn't filter out any RTCP.
     EXPECT_TRUE(CheckCustomRtcp1(kSsrc1));
     EXPECT_TRUE(CheckCustomRtcp2(kSsrc2));
@@ -1704,7 +1700,6 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
   }
 
   void TestFlushRtcp() {
-    bool send_rtcp1;
     CreateChannels(RTCP, RTCP);
     EXPECT_TRUE(SendInitiate());
     EXPECT_TRUE(SendAccept());
@@ -1714,14 +1709,16 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_EQ(2U, GetTransport2()->channels().size());
 
     // Send RTCP1 from a different thread.
-    CallOnThreadAndWaitForDone(&ChannelTest<T>::SendRtcp1, &send_rtcp1);
-    EXPECT_TRUE(send_rtcp1);
+    ScopedCallThread send_rtcp([this] { SendRtcp1(); });
     // The sending message is only posted.  channel2_ should be empty.
     EXPECT_TRUE(CheckNoRtcp2());
+    rtc::Thread* wait_for[] = {send_rtcp.thread()};
+    WaitForThreads(wait_for);  // Ensure rtcp was posted
 
     // When channel1_ is deleted, the RTCP packet should be sent out to
     // channel2_.
     channel1_.reset();
+    WaitForThreads();
     EXPECT_TRUE(CheckRtcp2());
   }
 
@@ -1744,18 +1741,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     // So we need to pass in pl_type so that the packet can pass through
     // the bundle filter before it can be processed by the srtp filter.
     // The packet is not a valid srtp packet because it is too short.
-    unsigned const char kBadPacket[] = {0x84,
-                                        static_cast<unsigned char>(pl_type),
-                                        0x00,
-                                        0x01,
-                                        0x00,
-                                        0x00,
-                                        0x00,
-                                        0x00,
-                                        0x00,
-                                        0x00,
-                                        0x00,
-                                        0x01};
+    static unsigned const char kBadPacket[] = {
+        0x84, static_cast<unsigned char>(pl_type),
+        0x00, 0x01,
+        0x00, 0x00,
+        0x00, 0x00,
+        0x00, 0x00,
+        0x00, 0x01};
     CreateChannels(RTCP | SECURE, RTCP | SECURE);
     EXPECT_FALSE(channel1_->secure());
     EXPECT_FALSE(channel2_->secure());
@@ -1768,37 +1760,42 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
         &error_handler, &SrtpErrorHandler::OnSrtpError);
 
     // Testing failures in sending packets.
-    EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
-                                          rtc::PacketOptions()));
+    media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
+                             rtc::PacketOptions());
+    WaitForThreads();
     // The first failure will trigger an error.
-    EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
+    EXPECT_EQ(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_);
     EXPECT_EQ(cricket::SrtpFilter::PROTECT, error_handler.mode_);
     error_handler.error_ = cricket::SrtpFilter::ERROR_NONE;
     error_handler.mode_ = cricket::SrtpFilter::UNPROTECT;
     // The next 250 ms failures will not trigger an error.
-    EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
-                                          rtc::PacketOptions()));
+    media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
+                             rtc::PacketOptions());
     // Wait for a while to ensure no message comes in.
+    WaitForThreads();
     rtc::Thread::Current()->ProcessMessages(200);
     EXPECT_EQ(cricket::SrtpFilter::ERROR_NONE, error_handler.error_);
     EXPECT_EQ(cricket::SrtpFilter::UNPROTECT, error_handler.mode_);
     // Wait for a little more - the error will be triggered again.
     rtc::Thread::Current()->ProcessMessages(200);
-    EXPECT_FALSE(media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
-                                          rtc::PacketOptions()));
-    EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
+    media_channel2_->SendRtp(kBadPacket, sizeof(kBadPacket),
+                             rtc::PacketOptions());
+    WaitForThreads();
+    EXPECT_EQ(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_);
     EXPECT_EQ(cricket::SrtpFilter::PROTECT, error_handler.mode_);
 
     // Testing failures in receiving packets.
     error_handler.error_ = cricket::SrtpFilter::ERROR_NONE;
     error_handler.mode_ = cricket::SrtpFilter::UNPROTECT;
 
-    cricket::TransportChannel* transport_channel =
-        channel2_->transport_channel();
-    transport_channel->SignalReadPacket(
-        transport_channel, reinterpret_cast<const char*>(kBadPacket),
-        sizeof(kBadPacket), rtc::PacketTime(), 0);
-    EXPECT_EQ_WAIT(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_, 500);
+    network_thread_->Invoke<void>([this] {
+      cricket::TransportChannel* transport_channel =
+          channel2_->transport_channel();
+      transport_channel->SignalReadPacket(
+          transport_channel, reinterpret_cast<const char*>(kBadPacket),
+          sizeof(kBadPacket), rtc::PacketTime(), 0);
+    });
+    EXPECT_EQ(cricket::SrtpFilter::ERROR_FAIL, error_handler.error_);
     EXPECT_EQ(cricket::SrtpFilter::UNPROTECT, error_handler.mode_);
   }
 
@@ -1807,23 +1804,37 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     TransportChannel* rtp = channel1_->transport_channel();
     TransportChannel* rtcp = channel1_->rtcp_transport_channel();
     EXPECT_FALSE(media_channel1_->ready_to_send());
-    rtp->SignalReadyToSend(rtp);
+
+    network_thread_->Invoke<void>([rtp] { rtp->SignalReadyToSend(rtp); });
+    WaitForThreads();
     EXPECT_FALSE(media_channel1_->ready_to_send());
-    rtcp->SignalReadyToSend(rtcp);
+
+    network_thread_->Invoke<void>([rtcp] { rtcp->SignalReadyToSend(rtcp); });
+    WaitForThreads();
     // MediaChannel::OnReadyToSend only be called when both rtp and rtcp
     // channel are ready to send.
     EXPECT_TRUE(media_channel1_->ready_to_send());
 
     // rtp channel becomes not ready to send will be propagated to mediachannel
-    channel1_->SetReadyToSend(false, false);
+    network_thread_->Invoke<void>(
+        [this] { channel1_->SetReadyToSend(false, false); });
+    WaitForThreads();
     EXPECT_FALSE(media_channel1_->ready_to_send());
-    channel1_->SetReadyToSend(false, true);
+
+    network_thread_->Invoke<void>(
+        [this] { channel1_->SetReadyToSend(false, true); });
+    WaitForThreads();
     EXPECT_TRUE(media_channel1_->ready_to_send());
 
     // rtcp channel becomes not ready to send will be propagated to mediachannel
-    channel1_->SetReadyToSend(true, false);
+    network_thread_->Invoke<void>(
+        [this] { channel1_->SetReadyToSend(true, false); });
+    WaitForThreads();
     EXPECT_FALSE(media_channel1_->ready_to_send());
-    channel1_->SetReadyToSend(true, true);
+
+    network_thread_->Invoke<void>(
+        [this] { channel1_->SetReadyToSend(true, true); });
+    WaitForThreads();
     EXPECT_TRUE(media_channel1_->ready_to_send());
   }
 
@@ -1840,9 +1851,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_FALSE(media_channel1_->ready_to_send());
     // In the case of rtcp mux, the SignalReadyToSend() from rtp channel
     // should trigger the MediaChannel's OnReadyToSend.
-    rtp->SignalReadyToSend(rtp);
+    network_thread_->Invoke<void>([rtp] { rtp->SignalReadyToSend(rtp); });
+    WaitForThreads();
     EXPECT_TRUE(media_channel1_->ready_to_send());
-    channel1_->SetReadyToSend(false, false);
+
+    network_thread_->Invoke<void>(
+        [this] { channel1_->SetReadyToSend(false, false); });
+    WaitForThreads();
     EXPECT_FALSE(media_channel1_->ready_to_send());
   }
 
@@ -1872,7 +1887,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_TRUE(
         channel1_->SetLocalContent(&local_media_content1_, CA_OFFER, NULL));
     EXPECT_EQ(media_channel1_->max_bps(), -1);
-    VerifyMaxBitrate(media_channel1_->GetRtpParameters(kSsrc1), -1);
+    VerifyMaxBitrate(media_channel1_->GetRtpSendParameters(kSsrc1), -1);
   }
 
   void CanChangeMaxBitrate() {
@@ -1880,25 +1895,48 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_TRUE(
         channel1_->SetLocalContent(&local_media_content1_, CA_OFFER, NULL));
 
-    EXPECT_TRUE(
-        channel1_->SetRtpParameters(kSsrc1, BitrateLimitedParameters(1000)));
-    VerifyMaxBitrate(channel1_->GetRtpParameters(kSsrc1), 1000);
-    VerifyMaxBitrate(media_channel1_->GetRtpParameters(kSsrc1), 1000);
+    EXPECT_TRUE(channel1_->SetRtpSendParameters(
+        kSsrc1, BitrateLimitedParameters(1000)));
+    VerifyMaxBitrate(channel1_->GetRtpSendParameters(kSsrc1), 1000);
+    VerifyMaxBitrate(media_channel1_->GetRtpSendParameters(kSsrc1), 1000);
     EXPECT_EQ(-1, media_channel1_->max_bps());
 
     EXPECT_TRUE(
-        channel1_->SetRtpParameters(kSsrc1, BitrateLimitedParameters(-1)));
-    VerifyMaxBitrate(channel1_->GetRtpParameters(kSsrc1), -1);
-    VerifyMaxBitrate(media_channel1_->GetRtpParameters(kSsrc1), -1);
+        channel1_->SetRtpSendParameters(kSsrc1, BitrateLimitedParameters(-1)));
+    VerifyMaxBitrate(channel1_->GetRtpSendParameters(kSsrc1), -1);
+    VerifyMaxBitrate(media_channel1_->GetRtpSendParameters(kSsrc1), -1);
     EXPECT_EQ(-1, media_channel1_->max_bps());
   }
 
  protected:
+  void WaitForThreads() { WaitForThreads(rtc::ArrayView<rtc::Thread*>()); }
+  static void ProcessThreadQueue(rtc::Thread* thread) {
+    RTC_DCHECK(thread->IsCurrent());
+    while (!thread->empty()) {
+      thread->ProcessMessages(0);
+    }
+  }
+  void WaitForThreads(rtc::ArrayView<rtc::Thread*> threads) {
+    // |threads| and current thread post packets to network thread.
+    for (rtc::Thread* thread : threads) {
+      thread->Invoke<void>([thread] { ProcessThreadQueue(thread); });
+    }
+    ProcessThreadQueue(rtc::Thread::Current());
+    // Network thread move them around and post back to worker = current thread.
+    if (!network_thread_->IsCurrent()) {
+      network_thread_->Invoke<void>(
+          [this] { ProcessThreadQueue(network_thread_); });
+    }
+    // Worker thread = current Thread process received messages.
+    ProcessThreadQueue(rtc::Thread::Current());
+  }
   // TODO(pbos): Remove playout from all media channels and let renderers mute
   // themselves.
   const bool verify_playout_;
-  cricket::FakeTransportController transport_controller1_;
-  cricket::FakeTransportController transport_controller2_;
+  std::unique_ptr<rtc::Thread> network_thread_keeper_;
+  rtc::Thread* network_thread_;
+  std::unique_ptr<cricket::FakeTransportController> transport_controller1_;
+  std::unique_ptr<cricket::FakeTransportController> transport_controller2_;
   cricket::FakeMediaEngine media_engine_;
   // The media channels are owned by the voice channel objects below.
   typename T::MediaChannel* media_channel1_;
@@ -1910,8 +1948,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
   typename T::Content remote_media_content1_;
   typename T::Content remote_media_content2_;
   // The RTP and RTCP packets to send in the tests.
-  std::string rtp_packet_;
-  std::string rtcp_packet_;
+  rtc::Buffer rtp_packet_;
+  rtc::Buffer rtcp_packet_;
   int media_info_callbacks1_;
   int media_info_callbacks2_;
   cricket::CandidatePairInterface* last_selected_candidate_pair_;
@@ -1954,29 +1992,33 @@ void ChannelTest<VoiceTraits>::AddLegacyStreamInContent(
   audio->AddLegacyStream(ssrc);
 }
 
-class VoiceChannelTest
-    : public ChannelTest<VoiceTraits> {
+class VoiceChannelSingleThreadTest : public ChannelTest<VoiceTraits> {
+ public:
+  typedef ChannelTest<VoiceTraits> Base;
+  VoiceChannelSingleThreadTest()
+      : Base(true, kPcmuFrame, kRtcpReport, NetworkIsWorker::Yes) {}
+};
+
+class VoiceChannelDoubleThreadTest : public ChannelTest<VoiceTraits> {
  public:
   typedef ChannelTest<VoiceTraits> Base;
-  VoiceChannelTest()
-      : Base(true,
-             kPcmuFrame,
-             sizeof(kPcmuFrame),
-             kRtcpReport,
-             sizeof(kRtcpReport)) {}
+  VoiceChannelDoubleThreadTest()
+      : Base(true, kPcmuFrame, kRtcpReport, NetworkIsWorker::No) {}
 };
 
 // override to add NULL parameter
 template <>
 cricket::VideoChannel* ChannelTest<VideoTraits>::CreateChannel(
-    rtc::Thread* thread,
+    rtc::Thread* worker_thread,
+    rtc::Thread* network_thread,
     cricket::MediaEngineInterface* engine,
     cricket::FakeVideoMediaChannel* ch,
     cricket::TransportController* transport_controller,
     bool rtcp) {
-  cricket::VideoChannel* channel = new cricket::VideoChannel(
-      thread, ch, transport_controller, cricket::CN_VIDEO, rtcp);
-  if (!channel->Init()) {
+  cricket::VideoChannel* channel =
+      new cricket::VideoChannel(worker_thread, network_thread, ch,
+                                transport_controller, cricket::CN_VIDEO, rtcp);
+  if (!channel->Init_w(nullptr)) {
     delete channel;
     channel = NULL;
   }
@@ -2026,67 +2068,72 @@ void ChannelTest<VideoTraits>::AddLegacyStreamInContent(
   video->AddLegacyStream(ssrc);
 }
 
-class VideoChannelTest
-    : public ChannelTest<VideoTraits> {
+class VideoChannelSingleThreadTest : public ChannelTest<VideoTraits> {
  public:
   typedef ChannelTest<VideoTraits> Base;
-  VideoChannelTest()
-      : Base(false,
-             kH264Packet,
-             sizeof(kH264Packet),
-             kRtcpReport,
-             sizeof(kRtcpReport)) {}
+  VideoChannelSingleThreadTest()
+      : Base(false, kH264Packet, kRtcpReport, NetworkIsWorker::Yes) {}
 };
 
+class VideoChannelDoubleThreadTest : public ChannelTest<VideoTraits> {
+ public:
+  typedef ChannelTest<VideoTraits> Base;
+  VideoChannelDoubleThreadTest()
+      : Base(false, kH264Packet, kRtcpReport, NetworkIsWorker::No) {}
+};
 
-// VoiceChannelTest
-TEST_F(VoiceChannelTest, TestInit) {
+// VoiceChannelSingleThreadTest
+TEST_F(VoiceChannelSingleThreadTest, TestInit) {
   Base::TestInit();
   EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
   EXPECT_TRUE(media_channel1_->dtmf_info_queue().empty());
 }
 
-TEST_F(VoiceChannelTest, TestSetContents) {
+TEST_F(VoiceChannelSingleThreadTest, TestDeinit) {
+  Base::TestDeinit();
+}
+
+TEST_F(VoiceChannelSingleThreadTest, TestSetContents) {
   Base::TestSetContents();
 }
 
-TEST_F(VoiceChannelTest, TestSetContentsNullOffer) {
+TEST_F(VoiceChannelSingleThreadTest, TestSetContentsNullOffer) {
   Base::TestSetContentsNullOffer();
 }
 
-TEST_F(VoiceChannelTest, TestSetContentsRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, TestSetContentsRtcpMux) {
   Base::TestSetContentsRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, TestSetContentsRtcpMuxWithPrAnswer) {
+TEST_F(VoiceChannelSingleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
   Base::TestSetContentsRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, TestSetRemoteContentUpdate) {
+TEST_F(VoiceChannelSingleThreadTest, TestSetRemoteContentUpdate) {
   Base::TestSetRemoteContentUpdate();
 }
 
-TEST_F(VoiceChannelTest, TestStreams) {
+TEST_F(VoiceChannelSingleThreadTest, TestStreams) {
   Base::TestStreams();
 }
 
-TEST_F(VoiceChannelTest, TestUpdateStreamsInLocalContent) {
+TEST_F(VoiceChannelSingleThreadTest, TestUpdateStreamsInLocalContent) {
   Base::TestUpdateStreamsInLocalContent();
 }
 
-TEST_F(VoiceChannelTest, TestUpdateRemoteStreamsInContent) {
+TEST_F(VoiceChannelSingleThreadTest, TestUpdateRemoteStreamsInContent) {
   Base::TestUpdateStreamsInRemoteContent();
 }
 
-TEST_F(VoiceChannelTest, TestChangeStreamParamsInContent) {
+TEST_F(VoiceChannelSingleThreadTest, TestChangeStreamParamsInContent) {
   Base::TestChangeStreamParamsInContent();
 }
 
-TEST_F(VoiceChannelTest, TestPlayoutAndSendingStates) {
+TEST_F(VoiceChannelSingleThreadTest, TestPlayoutAndSendingStates) {
   Base::TestPlayoutAndSendingStates();
 }
 
-TEST_F(VoiceChannelTest, TestMuteStream) {
+TEST_F(VoiceChannelSingleThreadTest, TestMuteStream) {
   CreateChannels(0, 0);
   // Test that we can Mute the default channel even though the sending SSRC
   // is unknown.
@@ -2108,123 +2155,123 @@ TEST_F(VoiceChannelTest, TestMuteStream) {
   EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
 }
 
-TEST_F(VoiceChannelTest, TestMediaContentDirection) {
+TEST_F(VoiceChannelSingleThreadTest, TestMediaContentDirection) {
   Base::TestMediaContentDirection();
 }
 
-TEST_F(VoiceChannelTest, TestNetworkRouteChanges) {
+TEST_F(VoiceChannelSingleThreadTest, TestNetworkRouteChanges) {
   Base::TestNetworkRouteChanges();
 }
 
-TEST_F(VoiceChannelTest, TestCallSetup) {
+TEST_F(VoiceChannelSingleThreadTest, TestCallSetup) {
   Base::TestCallSetup();
 }
 
-TEST_F(VoiceChannelTest, TestCallTeardownRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, TestCallTeardownRtcpMux) {
   Base::TestCallTeardownRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, SendRtpToRtp) {
+TEST_F(VoiceChannelSingleThreadTest, SendRtpToRtp) {
   Base::SendRtpToRtp();
 }
 
-TEST_F(VoiceChannelTest, SendNoRtcpToNoRtcp) {
+TEST_F(VoiceChannelSingleThreadTest, SendNoRtcpToNoRtcp) {
   Base::SendNoRtcpToNoRtcp();
 }
 
-TEST_F(VoiceChannelTest, SendNoRtcpToRtcp) {
+TEST_F(VoiceChannelSingleThreadTest, SendNoRtcpToRtcp) {
   Base::SendNoRtcpToRtcp();
 }
 
-TEST_F(VoiceChannelTest, SendRtcpToNoRtcp) {
+TEST_F(VoiceChannelSingleThreadTest, SendRtcpToNoRtcp) {
   Base::SendRtcpToNoRtcp();
 }
 
-TEST_F(VoiceChannelTest, SendRtcpToRtcp) {
+TEST_F(VoiceChannelSingleThreadTest, SendRtcpToRtcp) {
   Base::SendRtcpToRtcp();
 }
 
-TEST_F(VoiceChannelTest, SendRtcpMuxToRtcp) {
+TEST_F(VoiceChannelSingleThreadTest, SendRtcpMuxToRtcp) {
   Base::SendRtcpMuxToRtcp();
 }
 
-TEST_F(VoiceChannelTest, SendRtcpMuxToRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendRtcpMuxToRtcpMux) {
   Base::SendRtcpMuxToRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, SendRequireRtcpMuxToRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendRequireRtcpMuxToRtcpMux) {
   Base::SendRequireRtcpMuxToRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, SendRtcpMuxToRequireRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendRtcpMuxToRequireRtcpMux) {
   Base::SendRtcpMuxToRequireRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, SendRequireRtcpMuxToRequireRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
   Base::SendRequireRtcpMuxToRequireRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, SendRequireRtcpMuxToNoRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
   Base::SendRequireRtcpMuxToNoRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, SendEarlyRtcpMuxToRtcp) {
+TEST_F(VoiceChannelSingleThreadTest, SendEarlyRtcpMuxToRtcp) {
   Base::SendEarlyRtcpMuxToRtcp();
 }
 
-TEST_F(VoiceChannelTest, SendEarlyRtcpMuxToRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
   Base::SendEarlyRtcpMuxToRtcpMux();
 }
 
-TEST_F(VoiceChannelTest, SendSrtpToSrtpRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendSrtpToSrtpRtcpMux) {
   Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
 }
 
-TEST_F(VoiceChannelTest, SendSrtpToRtp) {
+TEST_F(VoiceChannelSingleThreadTest, SendSrtpToRtp) {
   Base::SendSrtpToSrtp();
 }
 
-TEST_F(VoiceChannelTest, SendSrtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendSrtcpMux) {
   Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
 }
 
-TEST_F(VoiceChannelTest, SendDtlsSrtpToSrtp) {
+TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToSrtp) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   Base::SendSrtpToSrtp(DTLS, 0);
 }
 
-TEST_F(VoiceChannelTest, SendDtlsSrtpToDtlsSrtp) {
+TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   Base::SendSrtpToSrtp(DTLS, DTLS);
 }
 
-TEST_F(VoiceChannelTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
 }
 
-TEST_F(VoiceChannelTest, SendEarlyMediaUsingRtcpMuxSrtp) {
+TEST_F(VoiceChannelSingleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
   Base::SendEarlyMediaUsingRtcpMuxSrtp();
 }
 
-TEST_F(VoiceChannelTest, SendRtpToRtpOnThread) {
+TEST_F(VoiceChannelSingleThreadTest, SendRtpToRtpOnThread) {
   Base::SendRtpToRtpOnThread();
 }
 
-TEST_F(VoiceChannelTest, SendSrtpToSrtpOnThread) {
+TEST_F(VoiceChannelSingleThreadTest, SendSrtpToSrtpOnThread) {
   Base::SendSrtpToSrtpOnThread();
 }
 
-TEST_F(VoiceChannelTest, SendWithWritabilityLoss) {
+TEST_F(VoiceChannelSingleThreadTest, SendWithWritabilityLoss) {
   Base::SendWithWritabilityLoss();
 }
 
-TEST_F(VoiceChannelTest, TestMediaMonitor) {
+TEST_F(VoiceChannelSingleThreadTest, TestMediaMonitor) {
   Base::TestMediaMonitor();
 }
 
 // Test that InsertDtmf properly forwards to the media channel.
-TEST_F(VoiceChannelTest, TestInsertDtmf) {
+TEST_F(VoiceChannelSingleThreadTest, TestInsertDtmf) {
   CreateChannels(0, 0);
   EXPECT_TRUE(SendInitiate());
   EXPECT_TRUE(SendAccept());
@@ -2243,44 +2290,44 @@ TEST_F(VoiceChannelTest, TestInsertDtmf) {
                               3, 7, 120));
 }
 
-TEST_F(VoiceChannelTest, TestSetContentFailure) {
+TEST_F(VoiceChannelSingleThreadTest, TestSetContentFailure) {
   Base::TestSetContentFailure();
 }
 
-TEST_F(VoiceChannelTest, TestSendTwoOffers) {
+TEST_F(VoiceChannelSingleThreadTest, TestSendTwoOffers) {
   Base::TestSendTwoOffers();
 }
 
-TEST_F(VoiceChannelTest, TestReceiveTwoOffers) {
+TEST_F(VoiceChannelSingleThreadTest, TestReceiveTwoOffers) {
   Base::TestReceiveTwoOffers();
 }
 
-TEST_F(VoiceChannelTest, TestSendPrAnswer) {
+TEST_F(VoiceChannelSingleThreadTest, TestSendPrAnswer) {
   Base::TestSendPrAnswer();
 }
 
-TEST_F(VoiceChannelTest, TestReceivePrAnswer) {
+TEST_F(VoiceChannelSingleThreadTest, TestReceivePrAnswer) {
   Base::TestReceivePrAnswer();
 }
 
-TEST_F(VoiceChannelTest, TestFlushRtcp) {
+TEST_F(VoiceChannelSingleThreadTest, TestFlushRtcp) {
   Base::TestFlushRtcp();
 }
 
-TEST_F(VoiceChannelTest, TestSrtpError) {
+TEST_F(VoiceChannelSingleThreadTest, TestSrtpError) {
   Base::TestSrtpError(kAudioPts[0]);
 }
 
-TEST_F(VoiceChannelTest, TestOnReadyToSend) {
+TEST_F(VoiceChannelSingleThreadTest, TestOnReadyToSend) {
   Base::TestOnReadyToSend();
 }
 
-TEST_F(VoiceChannelTest, TestOnReadyToSendWithRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, TestOnReadyToSendWithRtcpMux) {
   Base::TestOnReadyToSendWithRtcpMux();
 }
 
 // Test that we can scale the output volume properly for 1:1 calls.
-TEST_F(VoiceChannelTest, TestScaleVolume1to1Call) {
+TEST_F(VoiceChannelSingleThreadTest, TestScaleVolume1to1Call) {
   CreateChannels(RTCP, RTCP);
   EXPECT_TRUE(SendInitiate());
   EXPECT_TRUE(SendAccept());
@@ -2304,7 +2351,7 @@ TEST_F(VoiceChannelTest, TestScaleVolume1to1Call) {
 }
 
 // Test that we can scale the output volume properly for multiway calls.
-TEST_F(VoiceChannelTest, TestScaleVolumeMultiwayCall) {
+TEST_F(VoiceChannelSingleThreadTest, TestScaleVolumeMultiwayCall) {
   CreateChannels(RTCP, RTCP);
   EXPECT_TRUE(SendInitiate());
   EXPECT_TRUE(SendAccept());
@@ -2341,88 +2388,654 @@ TEST_F(VoiceChannelTest, TestScaleVolumeMultiwayCall) {
   EXPECT_DOUBLE_EQ(0.0, volume);
 }
 
-TEST_F(VoiceChannelTest, SendBundleToBundle) {
+TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundle) {
   Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
 }
 
-TEST_F(VoiceChannelTest, SendBundleToBundleSecure) {
+TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundleSecure) {
   Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, true);
 }
 
-TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMux) {
+TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundleWithRtcpMux) {
   Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, false);
 }
 
-TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMuxSecure) {
+TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
   Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, true);
 }
 
-TEST_F(VoiceChannelTest, GetRtpParametersIsNotImplemented) {
-  // These tests verify that the Get/SetRtpParameters methods for VoiceChannel
-  // always fail as they are not implemented.
-  // TODO(skvlad): Replace with full tests when support for bitrate limiting
-  // for audio RtpSenders is added.
+TEST_F(VoiceChannelSingleThreadTest, DefaultMaxBitrateIsUnlimited) {
+  Base::DefaultMaxBitrateIsUnlimited();
+}
+
+TEST_F(VoiceChannelSingleThreadTest, CanChangeMaxBitrate) {
+  Base::CanChangeMaxBitrate();
+}
+
+// VoiceChannelDoubleThreadTest
+TEST_F(VoiceChannelDoubleThreadTest, TestInit) {
+  Base::TestInit();
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+  EXPECT_TRUE(media_channel1_->dtmf_info_queue().empty());
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestDeinit) {
+  Base::TestDeinit();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSetContents) {
+  Base::TestSetContents();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsNullOffer) {
+  Base::TestSetContentsNullOffer();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsRtcpMux) {
+  Base::TestSetContentsRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
+  Base::TestSetContentsRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSetRemoteContentUpdate) {
+  Base::TestSetRemoteContentUpdate();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestStreams) {
+  Base::TestStreams();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestUpdateStreamsInLocalContent) {
+  Base::TestUpdateStreamsInLocalContent();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestUpdateRemoteStreamsInContent) {
+  Base::TestUpdateStreamsInRemoteContent();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
+  Base::TestChangeStreamParamsInContent();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
+  Base::TestPlayoutAndSendingStates();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestMuteStream) {
   CreateChannels(0, 0);
-  EXPECT_TRUE(
-      channel1_->SetLocalContent(&local_media_content1_, CA_OFFER, NULL));
-  webrtc::RtpParameters voice_parameters = channel1_->GetRtpParameters(kSsrc1);
-  EXPECT_EQ(0UL, voice_parameters.encodings.size());
+  // Test that we can Mute the default channel even though the sending SSRC
+  // is unknown.
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+  EXPECT_TRUE(channel1_->SetAudioSend(0, false, nullptr, nullptr));
+  EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
+  EXPECT_TRUE(channel1_->SetAudioSend(0, true, nullptr, nullptr));
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+
+  // Test that we can not mute an unknown SSRC.
+  EXPECT_FALSE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
+
+  SendInitiate();
+  // After the local session description has been set, we can mute a stream
+  // with its SSRC.
+  EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
+  EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
+  EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestMediaContentDirection) {
+  Base::TestMediaContentDirection();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestNetworkRouteChanges) {
+  Base::TestNetworkRouteChanges();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestCallSetup) {
+  Base::TestCallSetup();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestCallTeardownRtcpMux) {
+  Base::TestCallTeardownRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRtpToRtp) {
+  Base::SendRtpToRtp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendNoRtcpToNoRtcp) {
+  Base::SendNoRtcpToNoRtcp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendNoRtcpToRtcp) {
+  Base::SendNoRtcpToRtcp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRtcpToNoRtcp) {
+  Base::SendRtcpToNoRtcp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRtcpToRtcp) {
+  Base::SendRtcpToRtcp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRtcpMuxToRtcp) {
+  Base::SendRtcpMuxToRtcp();
 }
 
-TEST_F(VoiceChannelTest, SetRtpParametersIsNotImplemented) {
+TEST_F(VoiceChannelDoubleThreadTest, SendRtcpMuxToRtcpMux) {
+  Base::SendRtcpMuxToRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRequireRtcpMuxToRtcpMux) {
+  Base::SendRequireRtcpMuxToRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRtcpMuxToRequireRtcpMux) {
+  Base::SendRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
+  Base::SendRequireRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
+  Base::SendRequireRtcpMuxToNoRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcp) {
+  Base::SendEarlyRtcpMuxToRtcp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
+  Base::SendEarlyRtcpMuxToRtcpMux();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendSrtpToSrtpRtcpMux) {
+  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendSrtpToRtp) {
+  Base::SendSrtpToSrtp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendSrtcpMux) {
+  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToSrtp) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  Base::SendSrtpToSrtp(DTLS, 0);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  Base::SendSrtpToSrtp(DTLS, DTLS);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
+  Base::SendEarlyMediaUsingRtcpMuxSrtp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendRtpToRtpOnThread) {
+  Base::SendRtpToRtpOnThread();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendSrtpToSrtpOnThread) {
+  Base::SendSrtpToSrtpOnThread();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendWithWritabilityLoss) {
+  Base::SendWithWritabilityLoss();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestMediaMonitor) {
+  Base::TestMediaMonitor();
+}
+
+// Test that InsertDtmf properly forwards to the media channel.
+TEST_F(VoiceChannelDoubleThreadTest, TestInsertDtmf) {
   CreateChannels(0, 0);
+  EXPECT_TRUE(SendInitiate());
+  EXPECT_TRUE(SendAccept());
+  EXPECT_EQ(0U, media_channel1_->dtmf_info_queue().size());
+
+  EXPECT_TRUE(channel1_->InsertDtmf(1, 3, 100));
+  EXPECT_TRUE(channel1_->InsertDtmf(2, 5, 110));
+  EXPECT_TRUE(channel1_->InsertDtmf(3, 7, 120));
+
+  ASSERT_EQ(3U, media_channel1_->dtmf_info_queue().size());
+  EXPECT_TRUE(
+      CompareDtmfInfo(media_channel1_->dtmf_info_queue()[0], 1, 3, 100));
   EXPECT_TRUE(
-      channel1_->SetLocalContent(&local_media_content1_, CA_OFFER, NULL));
-  EXPECT_FALSE(
-      channel1_->SetRtpParameters(kSsrc1, BitrateLimitedParameters(1000)));
+      CompareDtmfInfo(media_channel1_->dtmf_info_queue()[1], 2, 5, 110));
+  EXPECT_TRUE(
+      CompareDtmfInfo(media_channel1_->dtmf_info_queue()[2], 3, 7, 120));
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSetContentFailure) {
+  Base::TestSetContentFailure();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSendTwoOffers) {
+  Base::TestSendTwoOffers();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestReceiveTwoOffers) {
+  Base::TestReceiveTwoOffers();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSendPrAnswer) {
+  Base::TestSendPrAnswer();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestReceivePrAnswer) {
+  Base::TestReceivePrAnswer();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestFlushRtcp) {
+  Base::TestFlushRtcp();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestSrtpError) {
+  Base::TestSrtpError(kAudioPts[0]);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestOnReadyToSend) {
+  Base::TestOnReadyToSend();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, TestOnReadyToSendWithRtcpMux) {
+  Base::TestOnReadyToSendWithRtcpMux();
+}
+
+// Test that we can scale the output volume properly for 1:1 calls.
+TEST_F(VoiceChannelDoubleThreadTest, TestScaleVolume1to1Call) {
+  CreateChannels(RTCP, RTCP);
+  EXPECT_TRUE(SendInitiate());
+  EXPECT_TRUE(SendAccept());
+  double volume;
+
+  // Default is (1.0).
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
+  EXPECT_DOUBLE_EQ(1.0, volume);
+  // invalid ssrc.
+  EXPECT_FALSE(media_channel1_->GetOutputVolume(3, &volume));
+
+  // Set scale to (1.5).
+  EXPECT_TRUE(channel1_->SetOutputVolume(0, 1.5));
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
+  EXPECT_DOUBLE_EQ(1.5, volume);
+
+  // Set scale to (0).
+  EXPECT_TRUE(channel1_->SetOutputVolume(0, 0.0));
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
+  EXPECT_DOUBLE_EQ(0.0, volume);
+}
+
+// Test that we can scale the output volume properly for multiway calls.
+TEST_F(VoiceChannelDoubleThreadTest, TestScaleVolumeMultiwayCall) {
+  CreateChannels(RTCP, RTCP);
+  EXPECT_TRUE(SendInitiate());
+  EXPECT_TRUE(SendAccept());
+  EXPECT_TRUE(AddStream1(1));
+  EXPECT_TRUE(AddStream1(2));
+
+  double volume;
+  // Default is (1.0).
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
+  EXPECT_DOUBLE_EQ(1.0, volume);
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
+  EXPECT_DOUBLE_EQ(1.0, volume);
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
+  EXPECT_DOUBLE_EQ(1.0, volume);
+  // invalid ssrc.
+  EXPECT_FALSE(media_channel1_->GetOutputVolume(3, &volume));
+
+  // Set scale to (1.5) for ssrc = 1.
+  EXPECT_TRUE(channel1_->SetOutputVolume(1, 1.5));
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
+  EXPECT_DOUBLE_EQ(1.5, volume);
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
+  EXPECT_DOUBLE_EQ(1.0, volume);
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
+  EXPECT_DOUBLE_EQ(1.0, volume);
+
+  // Set scale to (0) for all ssrcs.
+  EXPECT_TRUE(channel1_->SetOutputVolume(0, 0.0));
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(0, &volume));
+  EXPECT_DOUBLE_EQ(0.0, volume);
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(1, &volume));
+  EXPECT_DOUBLE_EQ(0.0, volume);
+  EXPECT_TRUE(media_channel1_->GetOutputVolume(2, &volume));
+  EXPECT_DOUBLE_EQ(0.0, volume);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundle) {
+  Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundleSecure) {
+  Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, true);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundleWithRtcpMux) {
+  Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, false);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
+  Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, true);
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, DefaultMaxBitrateIsUnlimited) {
+  Base::DefaultMaxBitrateIsUnlimited();
+}
+
+TEST_F(VoiceChannelDoubleThreadTest, CanChangeMaxBitrate) {
+  Base::CanChangeMaxBitrate();
+}
+
+// VideoChannelSingleThreadTest
+TEST_F(VideoChannelSingleThreadTest, TestInit) {
+  Base::TestInit();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestDeinit) {
+  Base::TestDeinit();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSetContents) {
+  Base::TestSetContents();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSetContentsNullOffer) {
+  Base::TestSetContentsNullOffer();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSetContentsRtcpMux) {
+  Base::TestSetContentsRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
+  Base::TestSetContentsRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSetRemoteContentUpdate) {
+  Base::TestSetRemoteContentUpdate();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestStreams) {
+  Base::TestStreams();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestUpdateStreamsInLocalContent) {
+  Base::TestUpdateStreamsInLocalContent();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestUpdateRemoteStreamsInContent) {
+  Base::TestUpdateStreamsInRemoteContent();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestChangeStreamParamsInContent) {
+  Base::TestChangeStreamParamsInContent();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestPlayoutAndSendingStates) {
+  Base::TestPlayoutAndSendingStates();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestMuteStream) {
+  CreateChannels(0, 0);
+  // Test that we can Mute the default channel even though the sending SSRC
+  // is unknown.
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+  EXPECT_TRUE(channel1_->SetVideoSend(0, false, nullptr));
+  EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
+  EXPECT_TRUE(channel1_->SetVideoSend(0, true, nullptr));
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+  // Test that we can not mute an unknown SSRC.
+  EXPECT_FALSE(channel1_->SetVideoSend(kSsrc1, false, nullptr));
+  SendInitiate();
+  // After the local session description has been set, we can mute a stream
+  // with its SSRC.
+  EXPECT_TRUE(channel1_->SetVideoSend(kSsrc1, false, nullptr));
+  EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
+  EXPECT_TRUE(channel1_->SetVideoSend(kSsrc1, true, nullptr));
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestMediaContentDirection) {
+  Base::TestMediaContentDirection();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestNetworkRouteChanges) {
+  Base::TestNetworkRouteChanges();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestCallSetup) {
+  Base::TestCallSetup();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestCallTeardownRtcpMux) {
+  Base::TestCallTeardownRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRtpToRtp) {
+  Base::SendRtpToRtp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendNoRtcpToNoRtcp) {
+  Base::SendNoRtcpToNoRtcp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendNoRtcpToRtcp) {
+  Base::SendNoRtcpToRtcp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRtcpToNoRtcp) {
+  Base::SendRtcpToNoRtcp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRtcpToRtcp) {
+  Base::SendRtcpToRtcp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRtcpMuxToRtcp) {
+  Base::SendRtcpMuxToRtcp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRtcpMuxToRtcpMux) {
+  Base::SendRtcpMuxToRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRequireRtcpMuxToRtcpMux) {
+  Base::SendRequireRtcpMuxToRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRtcpMuxToRequireRtcpMux) {
+  Base::SendRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
+  Base::SendRequireRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
+  Base::SendRequireRtcpMuxToNoRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendEarlyRtcpMuxToRtcp) {
+  Base::SendEarlyRtcpMuxToRtcp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
+  Base::SendEarlyRtcpMuxToRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendSrtpToSrtp) {
+  Base::SendSrtpToSrtp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendSrtpToRtp) {
+  Base::SendSrtpToSrtp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToSrtp) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  Base::SendSrtpToSrtp(DTLS, 0);
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  Base::SendSrtpToSrtp(DTLS, DTLS);
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendSrtcpMux) {
+  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
+  Base::SendEarlyMediaUsingRtcpMuxSrtp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendRtpToRtpOnThread) {
+  Base::SendRtpToRtpOnThread();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendSrtpToSrtpOnThread) {
+  Base::SendSrtpToSrtpOnThread();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendWithWritabilityLoss) {
+  Base::SendWithWritabilityLoss();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestMediaMonitor) {
+  Base::TestMediaMonitor();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSetContentFailure) {
+  Base::TestSetContentFailure();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSendTwoOffers) {
+  Base::TestSendTwoOffers();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestReceiveTwoOffers) {
+  Base::TestReceiveTwoOffers();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSendPrAnswer) {
+  Base::TestSendPrAnswer();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestReceivePrAnswer) {
+  Base::TestReceivePrAnswer();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestFlushRtcp) {
+  Base::TestFlushRtcp();
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendBundleToBundle) {
+  Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, false);
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendBundleToBundleSecure) {
+  Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, true);
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendBundleToBundleWithRtcpMux) {
+  Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, false);
+}
+
+TEST_F(VideoChannelSingleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
+  Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, true);
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestSrtpError) {
+  Base::TestSrtpError(kVideoPts[0]);
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestOnReadyToSend) {
+  Base::TestOnReadyToSend();
+}
+
+TEST_F(VideoChannelSingleThreadTest, TestOnReadyToSendWithRtcpMux) {
+  Base::TestOnReadyToSendWithRtcpMux();
+}
+
+TEST_F(VideoChannelSingleThreadTest, DefaultMaxBitrateIsUnlimited) {
+  Base::DefaultMaxBitrateIsUnlimited();
+}
+
+TEST_F(VideoChannelSingleThreadTest, CanChangeMaxBitrate) {
+  Base::CanChangeMaxBitrate();
 }
 
-// VideoChannelTest
-TEST_F(VideoChannelTest, TestInit) {
+// VideoChannelDoubleThreadTest
+TEST_F(VideoChannelDoubleThreadTest, TestInit) {
   Base::TestInit();
 }
 
-TEST_F(VideoChannelTest, TestSetContents) {
+TEST_F(VideoChannelDoubleThreadTest, TestDeinit) {
+  Base::TestDeinit();
+}
+
+TEST_F(VideoChannelDoubleThreadTest, TestSetContents) {
   Base::TestSetContents();
 }
 
-TEST_F(VideoChannelTest, TestSetContentsNullOffer) {
+TEST_F(VideoChannelDoubleThreadTest, TestSetContentsNullOffer) {
   Base::TestSetContentsNullOffer();
 }
 
-TEST_F(VideoChannelTest, TestSetContentsRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, TestSetContentsRtcpMux) {
   Base::TestSetContentsRtcpMux();
 }
 
-TEST_F(VideoChannelTest, TestSetContentsRtcpMuxWithPrAnswer) {
+TEST_F(VideoChannelDoubleThreadTest, TestSetContentsRtcpMuxWithPrAnswer) {
   Base::TestSetContentsRtcpMux();
 }
 
-TEST_F(VideoChannelTest, TestSetRemoteContentUpdate) {
+TEST_F(VideoChannelDoubleThreadTest, TestSetRemoteContentUpdate) {
   Base::TestSetRemoteContentUpdate();
 }
 
-TEST_F(VideoChannelTest, TestStreams) {
+TEST_F(VideoChannelDoubleThreadTest, TestStreams) {
   Base::TestStreams();
 }
 
-TEST_F(VideoChannelTest, TestUpdateStreamsInLocalContent) {
+TEST_F(VideoChannelDoubleThreadTest, TestUpdateStreamsInLocalContent) {
   Base::TestUpdateStreamsInLocalContent();
 }
 
-TEST_F(VideoChannelTest, TestUpdateRemoteStreamsInContent) {
+TEST_F(VideoChannelDoubleThreadTest, TestUpdateRemoteStreamsInContent) {
   Base::TestUpdateStreamsInRemoteContent();
 }
 
-TEST_F(VideoChannelTest, TestChangeStreamParamsInContent) {
+TEST_F(VideoChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
   Base::TestChangeStreamParamsInContent();
 }
 
-TEST_F(VideoChannelTest, TestPlayoutAndSendingStates) {
+TEST_F(VideoChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
   Base::TestPlayoutAndSendingStates();
 }
 
-TEST_F(VideoChannelTest, TestMuteStream) {
+TEST_F(VideoChannelDoubleThreadTest, TestMuteStream) {
   CreateChannels(0, 0);
   // Test that we can Mute the default channel even though the sending SSRC
   // is unknown.
@@ -2442,214 +3055,217 @@ TEST_F(VideoChannelTest, TestMuteStream) {
   EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
 }
 
-TEST_F(VideoChannelTest, TestMediaContentDirection) {
+TEST_F(VideoChannelDoubleThreadTest, TestMediaContentDirection) {
   Base::TestMediaContentDirection();
 }
 
-TEST_F(VideoChannelTest, TestNetworkRouteChanges) {
+TEST_F(VideoChannelDoubleThreadTest, TestNetworkRouteChanges) {
   Base::TestNetworkRouteChanges();
 }
 
-TEST_F(VideoChannelTest, TestCallSetup) {
+TEST_F(VideoChannelDoubleThreadTest, TestCallSetup) {
   Base::TestCallSetup();
 }
 
-TEST_F(VideoChannelTest, TestCallTeardownRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, TestCallTeardownRtcpMux) {
   Base::TestCallTeardownRtcpMux();
 }
 
-TEST_F(VideoChannelTest, SendRtpToRtp) {
+TEST_F(VideoChannelDoubleThreadTest, SendRtpToRtp) {
   Base::SendRtpToRtp();
 }
 
-TEST_F(VideoChannelTest, SendNoRtcpToNoRtcp) {
+TEST_F(VideoChannelDoubleThreadTest, SendNoRtcpToNoRtcp) {
   Base::SendNoRtcpToNoRtcp();
 }
 
-TEST_F(VideoChannelTest, SendNoRtcpToRtcp) {
+TEST_F(VideoChannelDoubleThreadTest, SendNoRtcpToRtcp) {
   Base::SendNoRtcpToRtcp();
 }
 
-TEST_F(VideoChannelTest, SendRtcpToNoRtcp) {
+TEST_F(VideoChannelDoubleThreadTest, SendRtcpToNoRtcp) {
   Base::SendRtcpToNoRtcp();
 }
 
-TEST_F(VideoChannelTest, SendRtcpToRtcp) {
+TEST_F(VideoChannelDoubleThreadTest, SendRtcpToRtcp) {
   Base::SendRtcpToRtcp();
 }
 
-TEST_F(VideoChannelTest, SendRtcpMuxToRtcp) {
+TEST_F(VideoChannelDoubleThreadTest, SendRtcpMuxToRtcp) {
   Base::SendRtcpMuxToRtcp();
 }
 
-TEST_F(VideoChannelTest, SendRtcpMuxToRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendRtcpMuxToRtcpMux) {
   Base::SendRtcpMuxToRtcpMux();
 }
 
-TEST_F(VideoChannelTest, SendRequireRtcpMuxToRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendRequireRtcpMuxToRtcpMux) {
   Base::SendRequireRtcpMuxToRtcpMux();
 }
 
-TEST_F(VideoChannelTest, SendRtcpMuxToRequireRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendRtcpMuxToRequireRtcpMux) {
   Base::SendRtcpMuxToRequireRtcpMux();
 }
 
-TEST_F(VideoChannelTest, SendRequireRtcpMuxToRequireRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
   Base::SendRequireRtcpMuxToRequireRtcpMux();
 }
 
-TEST_F(VideoChannelTest, SendRequireRtcpMuxToNoRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
   Base::SendRequireRtcpMuxToNoRtcpMux();
 }
 
-TEST_F(VideoChannelTest, SendEarlyRtcpMuxToRtcp) {
+TEST_F(VideoChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcp) {
   Base::SendEarlyRtcpMuxToRtcp();
 }
 
-TEST_F(VideoChannelTest, SendEarlyRtcpMuxToRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
   Base::SendEarlyRtcpMuxToRtcpMux();
 }
 
-TEST_F(VideoChannelTest, SendSrtpToSrtp) {
+TEST_F(VideoChannelDoubleThreadTest, SendSrtpToSrtp) {
   Base::SendSrtpToSrtp();
 }
 
-TEST_F(VideoChannelTest, SendSrtpToRtp) {
+TEST_F(VideoChannelDoubleThreadTest, SendSrtpToRtp) {
   Base::SendSrtpToSrtp();
 }
 
-TEST_F(VideoChannelTest, SendDtlsSrtpToSrtp) {
+TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToSrtp) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   Base::SendSrtpToSrtp(DTLS, 0);
 }
 
-TEST_F(VideoChannelTest, SendDtlsSrtpToDtlsSrtp) {
+TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   Base::SendSrtpToSrtp(DTLS, DTLS);
 }
 
-TEST_F(VideoChannelTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
   MAYBE_SKIP_TEST(HaveDtlsSrtp);
   Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
 }
 
-TEST_F(VideoChannelTest, SendSrtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendSrtcpMux) {
   Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
 }
 
-TEST_F(VideoChannelTest, SendEarlyMediaUsingRtcpMuxSrtp) {
+TEST_F(VideoChannelDoubleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
   Base::SendEarlyMediaUsingRtcpMuxSrtp();
 }
 
-TEST_F(VideoChannelTest, SendRtpToRtpOnThread) {
+TEST_F(VideoChannelDoubleThreadTest, SendRtpToRtpOnThread) {
   Base::SendRtpToRtpOnThread();
 }
 
-TEST_F(VideoChannelTest, SendSrtpToSrtpOnThread) {
+TEST_F(VideoChannelDoubleThreadTest, SendSrtpToSrtpOnThread) {
   Base::SendSrtpToSrtpOnThread();
 }
 
-TEST_F(VideoChannelTest, SendWithWritabilityLoss) {
+TEST_F(VideoChannelDoubleThreadTest, SendWithWritabilityLoss) {
   Base::SendWithWritabilityLoss();
 }
 
-TEST_F(VideoChannelTest, TestMediaMonitor) {
+TEST_F(VideoChannelDoubleThreadTest, TestMediaMonitor) {
   Base::TestMediaMonitor();
 }
 
-TEST_F(VideoChannelTest, TestSetContentFailure) {
+TEST_F(VideoChannelDoubleThreadTest, TestSetContentFailure) {
   Base::TestSetContentFailure();
 }
 
-TEST_F(VideoChannelTest, TestSendTwoOffers) {
+TEST_F(VideoChannelDoubleThreadTest, TestSendTwoOffers) {
   Base::TestSendTwoOffers();
 }
 
-TEST_F(VideoChannelTest, TestReceiveTwoOffers) {
+TEST_F(VideoChannelDoubleThreadTest, TestReceiveTwoOffers) {
   Base::TestReceiveTwoOffers();
 }
 
-TEST_F(VideoChannelTest, TestSendPrAnswer) {
+TEST_F(VideoChannelDoubleThreadTest, TestSendPrAnswer) {
   Base::TestSendPrAnswer();
 }
 
-TEST_F(VideoChannelTest, TestReceivePrAnswer) {
+TEST_F(VideoChannelDoubleThreadTest, TestReceivePrAnswer) {
   Base::TestReceivePrAnswer();
 }
 
-TEST_F(VideoChannelTest, TestFlushRtcp) {
+TEST_F(VideoChannelDoubleThreadTest, TestFlushRtcp) {
   Base::TestFlushRtcp();
 }
 
-TEST_F(VideoChannelTest, SendBundleToBundle) {
+TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundle) {
   Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, false);
 }
 
-TEST_F(VideoChannelTest, SendBundleToBundleSecure) {
+TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundleSecure) {
   Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, true);
 }
 
-TEST_F(VideoChannelTest, SendBundleToBundleWithRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundleWithRtcpMux) {
   Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, false);
 }
 
-TEST_F(VideoChannelTest, SendBundleToBundleWithRtcpMuxSecure) {
+TEST_F(VideoChannelDoubleThreadTest, SendBundleToBundleWithRtcpMuxSecure) {
   Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, true);
 }
 
-TEST_F(VideoChannelTest, TestSrtpError) {
+TEST_F(VideoChannelDoubleThreadTest, TestSrtpError) {
   Base::TestSrtpError(kVideoPts[0]);
 }
 
-TEST_F(VideoChannelTest, TestOnReadyToSend) {
+TEST_F(VideoChannelDoubleThreadTest, TestOnReadyToSend) {
   Base::TestOnReadyToSend();
 }
 
-TEST_F(VideoChannelTest, TestOnReadyToSendWithRtcpMux) {
+TEST_F(VideoChannelDoubleThreadTest, TestOnReadyToSendWithRtcpMux) {
   Base::TestOnReadyToSendWithRtcpMux();
 }
 
-TEST_F(VideoChannelTest, DefaultMaxBitrateIsUnlimited) {
+TEST_F(VideoChannelDoubleThreadTest, DefaultMaxBitrateIsUnlimited) {
   Base::DefaultMaxBitrateIsUnlimited();
 }
 
-TEST_F(VideoChannelTest, CanChangeMaxBitrate) {
+TEST_F(VideoChannelDoubleThreadTest, CanChangeMaxBitrate) {
   Base::CanChangeMaxBitrate();
 }
 
-// DataChannelTest
+// DataChannelSingleThreadTest
+class DataChannelSingleThreadTest : public ChannelTest<DataTraits> {
+ public:
+  typedef ChannelTest<DataTraits> Base;
+  DataChannelSingleThreadTest()
+      : Base(true, kDataPacket, kRtcpReport, NetworkIsWorker::Yes) {}
+};
 
-class DataChannelTest
-    : public ChannelTest<DataTraits> {
+// DataChannelDoubleThreadTest
+class DataChannelDoubleThreadTest : public ChannelTest<DataTraits> {
  public:
-  typedef ChannelTest<DataTraits>
-  Base;
-  DataChannelTest()
-      : Base(true,
-             kDataPacket,
-             sizeof(kDataPacket),
-             kRtcpReport,
-             sizeof(kRtcpReport)) {}
+  typedef ChannelTest<DataTraits> Base;
+  DataChannelDoubleThreadTest()
+      : Base(true, kDataPacket, kRtcpReport, NetworkIsWorker::No) {}
 };
 
 // Override to avoid engine channel parameter.
 template <>
 cricket::DataChannel* ChannelTest<DataTraits>::CreateChannel(
-    rtc::Thread* thread,
+    rtc::Thread* worker_thread,
+    rtc::Thread* network_thread,
     cricket::MediaEngineInterface* engine,
     cricket::FakeDataMediaChannel* ch,
     cricket::TransportController* transport_controller,
     bool rtcp) {
-  cricket::DataChannel* channel = new cricket::DataChannel(
-      thread, ch, transport_controller, cricket::CN_DATA, rtcp);
-  if (!channel->Init()) {
+  cricket::DataChannel* channel =
+      new cricket::DataChannel(worker_thread, network_thread, ch,
+                               transport_controller, cricket::CN_DATA, rtcp);
+  if (!channel->Init_w(nullptr)) {
     delete channel;
     channel = NULL;
   }
   return channel;
 }
 
-template<>
+template <>
 void ChannelTest<DataTraits>::CreateContent(
     int flags,
     const cricket::AudioCodec& audio_codec,
@@ -2664,14 +3280,14 @@ void ChannelTest<DataTraits>::CreateContent(
   }
 }
 
-template<>
+template <>
 void ChannelTest<DataTraits>::CopyContent(
     const cricket::DataContentDescription& source,
     cricket::DataContentDescription* data) {
   *data = source;
 }
 
-template<>
+template <>
 bool ChannelTest<DataTraits>::CodecMatches(const cricket::DataCodec& c1,
                                            const cricket::DataCodec& c2) {
   return c1.name == c2.name;
@@ -2685,132 +3301,280 @@ void ChannelTest<DataTraits>::AddLegacyStreamInContent(
   data->AddLegacyStream(ssrc);
 }
 
-TEST_F(DataChannelTest, TestInit) {
+TEST_F(DataChannelSingleThreadTest, TestInit) {
+  Base::TestInit();
+  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+}
+
+TEST_F(DataChannelSingleThreadTest, TestDeinit) {
+  Base::TestDeinit();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestSetContents) {
+  Base::TestSetContents();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestSetContentsNullOffer) {
+  Base::TestSetContentsNullOffer();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestSetContentsRtcpMux) {
+  Base::TestSetContentsRtcpMux();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestSetRemoteContentUpdate) {
+  Base::TestSetRemoteContentUpdate();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestStreams) {
+  Base::TestStreams();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestUpdateStreamsInLocalContent) {
+  Base::TestUpdateStreamsInLocalContent();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestUpdateRemoteStreamsInContent) {
+  Base::TestUpdateStreamsInRemoteContent();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestChangeStreamParamsInContent) {
+  Base::TestChangeStreamParamsInContent();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestPlayoutAndSendingStates) {
+  Base::TestPlayoutAndSendingStates();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestMediaContentDirection) {
+  Base::TestMediaContentDirection();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestCallSetup) {
+  Base::TestCallSetup();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestCallTeardownRtcpMux) {
+  Base::TestCallTeardownRtcpMux();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestOnReadyToSend) {
+  Base::TestOnReadyToSend();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestOnReadyToSendWithRtcpMux) {
+  Base::TestOnReadyToSendWithRtcpMux();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendRtpToRtp) {
+  Base::SendRtpToRtp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendNoRtcpToNoRtcp) {
+  Base::SendNoRtcpToNoRtcp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendNoRtcpToRtcp) {
+  Base::SendNoRtcpToRtcp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendRtcpToNoRtcp) {
+  Base::SendRtcpToNoRtcp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendRtcpToRtcp) {
+  Base::SendRtcpToRtcp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendRtcpMuxToRtcp) {
+  Base::SendRtcpMuxToRtcp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendRtcpMuxToRtcpMux) {
+  Base::SendRtcpMuxToRtcpMux();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendEarlyRtcpMuxToRtcp) {
+  Base::SendEarlyRtcpMuxToRtcp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
+  Base::SendEarlyRtcpMuxToRtcpMux();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendSrtpToSrtp) {
+  Base::SendSrtpToSrtp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendSrtpToRtp) {
+  Base::SendSrtpToSrtp();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendSrtcpMux) {
+  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
+}
+
+TEST_F(DataChannelSingleThreadTest, SendRtpToRtpOnThread) {
+  Base::SendRtpToRtpOnThread();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendSrtpToSrtpOnThread) {
+  Base::SendSrtpToSrtpOnThread();
+}
+
+TEST_F(DataChannelSingleThreadTest, SendWithWritabilityLoss) {
+  Base::SendWithWritabilityLoss();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestMediaMonitor) {
+  Base::TestMediaMonitor();
+}
+
+TEST_F(DataChannelSingleThreadTest, TestSendData) {
+  CreateChannels(0, 0);
+  EXPECT_TRUE(SendInitiate());
+  EXPECT_TRUE(SendAccept());
+
+  cricket::SendDataParams params;
+  params.ssrc = 42;
+  unsigned char data[] = {'f', 'o', 'o'};
+  rtc::CopyOnWriteBuffer payload(data, 3);
+  cricket::SendDataResult result;
+  ASSERT_TRUE(media_channel1_->SendData(params, payload, &result));
+  EXPECT_EQ(params.ssrc, media_channel1_->last_sent_data_params().ssrc);
+  EXPECT_EQ("foo", media_channel1_->last_sent_data());
+}
+
+TEST_F(DataChannelDoubleThreadTest, TestInit) {
   Base::TestInit();
   EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
 }
 
-TEST_F(DataChannelTest, TestSetContents) {
+TEST_F(DataChannelDoubleThreadTest, TestDeinit) {
+  Base::TestDeinit();
+}
+
+TEST_F(DataChannelDoubleThreadTest, TestSetContents) {
   Base::TestSetContents();
 }
 
-TEST_F(DataChannelTest, TestSetContentsNullOffer) {
+TEST_F(DataChannelDoubleThreadTest, TestSetContentsNullOffer) {
   Base::TestSetContentsNullOffer();
 }
 
-TEST_F(DataChannelTest, TestSetContentsRtcpMux) {
+TEST_F(DataChannelDoubleThreadTest, TestSetContentsRtcpMux) {
   Base::TestSetContentsRtcpMux();
 }
 
-TEST_F(DataChannelTest, TestSetRemoteContentUpdate) {
+TEST_F(DataChannelDoubleThreadTest, TestSetRemoteContentUpdate) {
   Base::TestSetRemoteContentUpdate();
 }
 
-TEST_F(DataChannelTest, TestStreams) {
+TEST_F(DataChannelDoubleThreadTest, TestStreams) {
   Base::TestStreams();
 }
 
-TEST_F(DataChannelTest, TestUpdateStreamsInLocalContent) {
+TEST_F(DataChannelDoubleThreadTest, TestUpdateStreamsInLocalContent) {
   Base::TestUpdateStreamsInLocalContent();
 }
 
-TEST_F(DataChannelTest, TestUpdateRemoteStreamsInContent) {
+TEST_F(DataChannelDoubleThreadTest, TestUpdateRemoteStreamsInContent) {
   Base::TestUpdateStreamsInRemoteContent();
 }
 
-TEST_F(DataChannelTest, TestChangeStreamParamsInContent) {
+TEST_F(DataChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
   Base::TestChangeStreamParamsInContent();
 }
 
-TEST_F(DataChannelTest, TestPlayoutAndSendingStates) {
+TEST_F(DataChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
   Base::TestPlayoutAndSendingStates();
 }
 
-TEST_F(DataChannelTest, TestMediaContentDirection) {
+TEST_F(DataChannelDoubleThreadTest, TestMediaContentDirection) {
   Base::TestMediaContentDirection();
 }
 
-TEST_F(DataChannelTest, TestCallSetup) {
+TEST_F(DataChannelDoubleThreadTest, TestCallSetup) {
   Base::TestCallSetup();
 }
 
-TEST_F(DataChannelTest, TestCallTeardownRtcpMux) {
+TEST_F(DataChannelDoubleThreadTest, TestCallTeardownRtcpMux) {
   Base::TestCallTeardownRtcpMux();
 }
 
-TEST_F(DataChannelTest, TestOnReadyToSend) {
+TEST_F(DataChannelDoubleThreadTest, TestOnReadyToSend) {
   Base::TestOnReadyToSend();
 }
 
-TEST_F(DataChannelTest, TestOnReadyToSendWithRtcpMux) {
+TEST_F(DataChannelDoubleThreadTest, TestOnReadyToSendWithRtcpMux) {
   Base::TestOnReadyToSendWithRtcpMux();
 }
 
-TEST_F(DataChannelTest, SendRtpToRtp) {
+TEST_F(DataChannelDoubleThreadTest, SendRtpToRtp) {
   Base::SendRtpToRtp();
 }
 
-TEST_F(DataChannelTest, SendNoRtcpToNoRtcp) {
+TEST_F(DataChannelDoubleThreadTest, SendNoRtcpToNoRtcp) {
   Base::SendNoRtcpToNoRtcp();
 }
 
-TEST_F(DataChannelTest, SendNoRtcpToRtcp) {
+TEST_F(DataChannelDoubleThreadTest, SendNoRtcpToRtcp) {
   Base::SendNoRtcpToRtcp();
 }
 
-TEST_F(DataChannelTest, SendRtcpToNoRtcp) {
+TEST_F(DataChannelDoubleThreadTest, SendRtcpToNoRtcp) {
   Base::SendRtcpToNoRtcp();
 }
 
-TEST_F(DataChannelTest, SendRtcpToRtcp) {
+TEST_F(DataChannelDoubleThreadTest, SendRtcpToRtcp) {
   Base::SendRtcpToRtcp();
 }
 
-TEST_F(DataChannelTest, SendRtcpMuxToRtcp) {
+TEST_F(DataChannelDoubleThreadTest, SendRtcpMuxToRtcp) {
   Base::SendRtcpMuxToRtcp();
 }
 
-TEST_F(DataChannelTest, SendRtcpMuxToRtcpMux) {
+TEST_F(DataChannelDoubleThreadTest, SendRtcpMuxToRtcpMux) {
   Base::SendRtcpMuxToRtcpMux();
 }
 
-TEST_F(DataChannelTest, SendEarlyRtcpMuxToRtcp) {
+TEST_F(DataChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcp) {
   Base::SendEarlyRtcpMuxToRtcp();
 }
 
-TEST_F(DataChannelTest, SendEarlyRtcpMuxToRtcpMux) {
+TEST_F(DataChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
   Base::SendEarlyRtcpMuxToRtcpMux();
 }
 
-TEST_F(DataChannelTest, SendSrtpToSrtp) {
+TEST_F(DataChannelDoubleThreadTest, SendSrtpToSrtp) {
   Base::SendSrtpToSrtp();
 }
 
-TEST_F(DataChannelTest, SendSrtpToRtp) {
+TEST_F(DataChannelDoubleThreadTest, SendSrtpToRtp) {
   Base::SendSrtpToSrtp();
 }
 
-TEST_F(DataChannelTest, SendSrtcpMux) {
+TEST_F(DataChannelDoubleThreadTest, SendSrtcpMux) {
   Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
 }
 
-TEST_F(DataChannelTest, SendRtpToRtpOnThread) {
+TEST_F(DataChannelDoubleThreadTest, SendRtpToRtpOnThread) {
   Base::SendRtpToRtpOnThread();
 }
 
-TEST_F(DataChannelTest, SendSrtpToSrtpOnThread) {
+TEST_F(DataChannelDoubleThreadTest, SendSrtpToSrtpOnThread) {
   Base::SendSrtpToSrtpOnThread();
 }
 
-TEST_F(DataChannelTest, SendWithWritabilityLoss) {
+TEST_F(DataChannelDoubleThreadTest, SendWithWritabilityLoss) {
   Base::SendWithWritabilityLoss();
 }
 
-TEST_F(DataChannelTest, TestMediaMonitor) {
+TEST_F(DataChannelDoubleThreadTest, TestMediaMonitor) {
   Base::TestMediaMonitor();
 }
 
-TEST_F(DataChannelTest, TestSendData) {
+TEST_F(DataChannelDoubleThreadTest, TestSendData) {
   CreateChannels(0, 0);
   EXPECT_TRUE(SendInitiate());
   EXPECT_TRUE(SendAccept());
diff --git a/chromium/third_party/webrtc/pc/channelmanager.cc b/chromium/third_party/webrtc/pc/channelmanager.cc
index f59a3df9c72..56dd135549b 100644
--- a/chromium/third_party/webrtc/pc/channelmanager.cc
+++ b/chromium/third_party/webrtc/pc/channelmanager.cc
@@ -44,25 +44,26 @@ static DataEngineInterface* ConstructDataEngine() {
 
 ChannelManager::ChannelManager(MediaEngineInterface* me,
                                DataEngineInterface* dme,
-                               rtc::Thread* worker_thread) {
-  Construct(me, dme, worker_thread);
+                               rtc::Thread* thread) {
+  Construct(me, dme, thread, thread);
 }
 
 ChannelManager::ChannelManager(MediaEngineInterface* me,
-                               rtc::Thread* worker_thread) {
-  Construct(me,
-            ConstructDataEngine(),
-            worker_thread);
+                               rtc::Thread* worker_thread,
+                               rtc::Thread* network_thread) {
+  Construct(me, ConstructDataEngine(), worker_thread, network_thread);
 }
 
 void ChannelManager::Construct(MediaEngineInterface* me,
                                DataEngineInterface* dme,
-                               rtc::Thread* worker_thread) {
+                               rtc::Thread* worker_thread,
+                               rtc::Thread* network_thread) {
   media_engine_.reset(me);
   data_media_engine_.reset(dme);
   initialized_ = false;
   main_thread_ = rtc::Thread::Current();
   worker_thread_ = worker_thread;
+  network_thread_ = network_thread;
   audio_output_volume_ = kNotSetOutputVolume;
   capturing_ = false;
   enable_rtx_ = false;
@@ -144,18 +145,16 @@ bool ChannelManager::Init() {
   if (initialized_) {
     return false;
   }
-  ASSERT(worker_thread_ != NULL);
-  if (!worker_thread_) {
-    return false;
-  }
-  if (worker_thread_ != rtc::Thread::Current()) {
-    // Do not allow invoking calls to other threads on the worker thread.
-    worker_thread_->Invoke<bool>(rtc::Bind(
-        &rtc::Thread::SetAllowBlockingCalls, worker_thread_, false));
+  RTC_DCHECK(network_thread_);
+  RTC_DCHECK(worker_thread_);
+  if (!network_thread_->IsCurrent()) {
+    // Do not allow invoking calls to other threads on the network thread.
+    network_thread_->Invoke<bool>(
+        rtc::Bind(&rtc::Thread::SetAllowBlockingCalls, network_thread_, false));
   }
 
-  initialized_ = worker_thread_->Invoke<bool>(Bind(
-      &ChannelManager::InitMediaEngine_w, this));
+  initialized_ = worker_thread_->Invoke<bool>(
+      Bind(&ChannelManager::InitMediaEngine_w, this));
   ASSERT(initialized_);
   if (!initialized_) {
     return false;
@@ -206,17 +205,20 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
+    const std::string* bundle_transport_name,
     bool rtcp,
     const AudioOptions& options) {
   return worker_thread_->Invoke<VoiceChannel*>(
       Bind(&ChannelManager::CreateVoiceChannel_w, this, media_controller,
-           transport_controller, content_name, rtcp, options));
+           transport_controller, content_name, bundle_transport_name, rtcp,
+           options));
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
+    const std::string* bundle_transport_name,
     bool rtcp,
     const AudioOptions& options) {
   ASSERT(initialized_);
@@ -228,9 +230,9 @@ VoiceChannel* ChannelManager::CreateVoiceChannel_w(
     return nullptr;
 
   VoiceChannel* voice_channel =
-      new VoiceChannel(worker_thread_, media_engine_.get(), media_channel,
-                       transport_controller, content_name, rtcp);
-  if (!voice_channel->Init()) {
+      new VoiceChannel(worker_thread_, network_thread_, media_engine_.get(),
+                       media_channel, transport_controller, content_name, rtcp);
+  if (!voice_channel->Init_w(bundle_transport_name)) {
     delete voice_channel;
     return nullptr;
   }
@@ -264,17 +266,20 @@ VideoChannel* ChannelManager::CreateVideoChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
+    const std::string* bundle_transport_name,
     bool rtcp,
     const VideoOptions& options) {
   return worker_thread_->Invoke<VideoChannel*>(
       Bind(&ChannelManager::CreateVideoChannel_w, this, media_controller,
-           transport_controller, content_name, rtcp, options));
+           transport_controller, content_name, bundle_transport_name, rtcp,
+           options));
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportController* transport_controller,
     const std::string& content_name,
+    const std::string* bundle_transport_name,
     bool rtcp,
     const VideoOptions& options) {
   ASSERT(initialized_);
@@ -286,9 +291,10 @@ VideoChannel* ChannelManager::CreateVideoChannel_w(
     return NULL;
   }
 
-  VideoChannel* video_channel = new VideoChannel(
-      worker_thread_, media_channel, transport_controller, content_name, rtcp);
-  if (!video_channel->Init()) {
+  VideoChannel* video_channel =
+      new VideoChannel(worker_thread_, network_thread_, media_channel,
+                       transport_controller, content_name, rtcp);
+  if (!video_channel->Init_w(bundle_transport_name)) {
     delete video_channel;
     return NULL;
   }
@@ -322,16 +328,18 @@ void ChannelManager::DestroyVideoChannel_w(VideoChannel* video_channel) {
 DataChannel* ChannelManager::CreateDataChannel(
     TransportController* transport_controller,
     const std::string& content_name,
+    const std::string* bundle_transport_name,
     bool rtcp,
     DataChannelType channel_type) {
   return worker_thread_->Invoke<DataChannel*>(
       Bind(&ChannelManager::CreateDataChannel_w, this, transport_controller,
-           content_name, rtcp, channel_type));
+           content_name, bundle_transport_name, rtcp, channel_type));
 }
 
 DataChannel* ChannelManager::CreateDataChannel_w(
     TransportController* transport_controller,
     const std::string& content_name,
+    const std::string* bundle_transport_name,
     bool rtcp,
     DataChannelType data_channel_type) {
   // This is ok to alloc from a thread other than the worker thread.
@@ -344,9 +352,10 @@ DataChannel* ChannelManager::CreateDataChannel_w(
     return NULL;
   }
 
-  DataChannel* data_channel = new DataChannel(
-      worker_thread_, media_channel, transport_controller, content_name, rtcp);
-  if (!data_channel->Init()) {
+  DataChannel* data_channel =
+      new DataChannel(worker_thread_, network_thread_, media_channel,
+                      transport_controller, content_name, rtcp);
+  if (!data_channel->Init_w(bundle_transport_name)) {
     LOG(LS_WARNING) << "Failed to init data channel.";
     delete data_channel;
     return NULL;
@@ -413,9 +422,11 @@ void ChannelManager::StopAecDump() {
       Bind(&MediaEngineInterface::StopAecDump, media_engine_.get()));
 }
 
-bool ChannelManager::StartRtcEventLog(rtc::PlatformFile file) {
+bool ChannelManager::StartRtcEventLog(rtc::PlatformFile file,
+                                      int64_t max_size_bytes) {
   return worker_thread_->Invoke<bool>(
-      Bind(&MediaEngineInterface::StartRtcEventLog, media_engine_.get(), file));
+      Bind(&MediaEngineInterface::StartRtcEventLog, media_engine_.get(), file,
+           max_size_bytes));
 }
 
 void ChannelManager::StopRtcEventLog() {
diff --git a/chromium/third_party/webrtc/pc/channelmanager.h b/chromium/third_party/webrtc/pc/channelmanager.h
index 72a2f056b5a..3b0c0e967ea 100644
--- a/chromium/third_party/webrtc/pc/channelmanager.h
+++ b/chromium/third_party/webrtc/pc/channelmanager.h
@@ -42,20 +42,31 @@ class ChannelManager {
   // ownership of these objects.
   ChannelManager(MediaEngineInterface* me,
                  DataEngineInterface* dme,
-                 rtc::Thread* worker);
+                 rtc::Thread* worker_and_network);
   // Same as above, but gives an easier default DataEngine.
   ChannelManager(MediaEngineInterface* me,
-                 rtc::Thread* worker);
+                 rtc::Thread* worker,
+                 rtc::Thread* network);
   ~ChannelManager();
 
   // Accessors for the worker thread, allowing it to be set after construction,
   // but before Init. set_worker_thread will return false if called after Init.
   rtc::Thread* worker_thread() const { return worker_thread_; }
   bool set_worker_thread(rtc::Thread* thread) {
-    if (initialized_) return false;
+    if (initialized_) {
+      return false;
+    }
     worker_thread_ = thread;
     return true;
   }
+  rtc::Thread* network_thread() const { return network_thread_; }
+  bool set_network_thread(rtc::Thread* thread) {
+    if (initialized_) {
+      return false;
+    }
+    network_thread_ = thread;
+    return true;
+  }
 
   MediaEngineInterface* media_engine() { return media_engine_.get(); }
 
@@ -80,6 +91,7 @@ class ChannelManager {
       webrtc::MediaControllerInterface* media_controller,
       TransportController* transport_controller,
       const std::string& content_name,
+      const std::string* bundle_transport_name,
       bool rtcp,
       const AudioOptions& options);
   // Destroys a voice channel created with the Create API.
@@ -90,12 +102,14 @@ class ChannelManager {
       webrtc::MediaControllerInterface* media_controller,
       TransportController* transport_controller,
       const std::string& content_name,
+      const std::string* bundle_transport_name,
       bool rtcp,
       const VideoOptions& options);
   // Destroys a video channel created with the Create API.
   void DestroyVideoChannel(VideoChannel* video_channel);
   DataChannel* CreateDataChannel(TransportController* transport_controller,
                                  const std::string& content_name,
+                                 const std::string* bundle_transport_name,
                                  bool rtcp,
                                  DataChannelType data_channel_type);
   // Destroys a data channel created with the Create API.
@@ -126,7 +140,7 @@ class ChannelManager {
   void StopAecDump();
 
   // Starts RtcEventLog using existing file.
-  bool StartRtcEventLog(rtc::PlatformFile file);
+  bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops logging RtcEventLog.
   void StopRtcEventLog();
@@ -138,7 +152,8 @@ class ChannelManager {
 
   void Construct(MediaEngineInterface* me,
                  DataEngineInterface* dme,
-                 rtc::Thread* worker_thread);
+                 rtc::Thread* worker_thread,
+                 rtc::Thread* network_thread);
   bool InitMediaEngine_w();
   void DestructorDeletes_w();
   void Terminate_w();
@@ -146,6 +161,7 @@ class ChannelManager {
       webrtc::MediaControllerInterface* media_controller,
       TransportController* transport_controller,
       const std::string& content_name,
+      const std::string* bundle_transport_name,
       bool rtcp,
       const AudioOptions& options);
   void DestroyVoiceChannel_w(VoiceChannel* voice_channel);
@@ -153,11 +169,13 @@ class ChannelManager {
       webrtc::MediaControllerInterface* media_controller,
       TransportController* transport_controller,
       const std::string& content_name,
+      const std::string* bundle_transport_name,
       bool rtcp,
       const VideoOptions& options);
   void DestroyVideoChannel_w(VideoChannel* video_channel);
   DataChannel* CreateDataChannel_w(TransportController* transport_controller,
                                    const std::string& content_name,
+                                   const std::string* bundle_transport_name,
                                    bool rtcp,
                                    DataChannelType data_channel_type);
   void DestroyDataChannel_w(DataChannel* data_channel);
@@ -167,6 +185,7 @@ class ChannelManager {
   bool initialized_;
   rtc::Thread* main_thread_;
   rtc::Thread* worker_thread_;
+  rtc::Thread* network_thread_;
 
   VoiceChannels voice_channels_;
   VideoChannels video_channels_;
diff --git a/chromium/third_party/webrtc/pc/channelmanager_unittest.cc b/chromium/third_party/webrtc/pc/channelmanager_unittest.cc
index 1eb09198d9f..e5e7b4f7de4 100644
--- a/chromium/third_party/webrtc/pc/channelmanager_unittest.cc
+++ b/chromium/third_party/webrtc/pc/channelmanager_unittest.cc
@@ -22,14 +22,12 @@
 namespace cricket {
 
 static const AudioCodec kAudioCodecs[] = {
-  AudioCodec(97, "voice", 1, 2, 3, 0),
-  AudioCodec(111, "OPUS", 48000, 32000, 2, 0),
+    AudioCodec(97, "voice", 1, 2, 3), AudioCodec(111, "OPUS", 48000, 32000, 2),
 };
 
 static const VideoCodec kVideoCodecs[] = {
-  VideoCodec(99, "H264", 100, 200, 300, 0),
-  VideoCodec(100, "VP8", 100, 200, 300, 0),
-  VideoCodec(96, "rtx", 100, 200, 300, 0),
+    VideoCodec(99, "H264", 100, 200, 300),
+    VideoCodec(100, "VP8", 100, 200, 300), VideoCodec(96, "rtx", 100, 200, 300),
 };
 
 class ChannelManagerTest : public testing::Test {
@@ -58,6 +56,7 @@ class ChannelManagerTest : public testing::Test {
     fme_ = NULL;
   }
 
+  rtc::Thread network_;
   rtc::Thread worker_;
   cricket::FakeMediaEngine* fme_;
   cricket::FakeDataEngine* fdme_;
@@ -79,14 +78,18 @@ TEST_F(ChannelManagerTest, StartupShutdown) {
 
 // Test that we startup/shutdown properly with a worker thread.
 TEST_F(ChannelManagerTest, StartupShutdownOnThread) {
+  network_.Start();
   worker_.Start();
   EXPECT_FALSE(cm_->initialized());
   EXPECT_EQ(rtc::Thread::Current(), cm_->worker_thread());
+  EXPECT_TRUE(cm_->set_network_thread(&network_));
+  EXPECT_EQ(&network_, cm_->network_thread());
   EXPECT_TRUE(cm_->set_worker_thread(&worker_));
   EXPECT_EQ(&worker_, cm_->worker_thread());
   EXPECT_TRUE(cm_->Init());
   EXPECT_TRUE(cm_->initialized());
-  // Setting the worker thread while initialized should fail.
+  // Setting the network or worker thread while initialized should fail.
+  EXPECT_FALSE(cm_->set_network_thread(rtc::Thread::Current()));
   EXPECT_FALSE(cm_->set_worker_thread(rtc::Thread::Current()));
   cm_->Terminate();
   EXPECT_FALSE(cm_->initialized());
@@ -95,16 +98,17 @@ TEST_F(ChannelManagerTest, StartupShutdownOnThread) {
 // Test that we can create and destroy a voice and video channel.
 TEST_F(ChannelManagerTest, CreateDestroyChannels) {
   EXPECT_TRUE(cm_->Init());
-  cricket::VoiceChannel* voice_channel =
-      cm_->CreateVoiceChannel(&fake_mc_, transport_controller_,
-                              cricket::CN_AUDIO, false, AudioOptions());
+  cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
+      &fake_mc_, transport_controller_, cricket::CN_AUDIO, nullptr, false,
+      AudioOptions());
   EXPECT_TRUE(voice_channel != nullptr);
-  cricket::VideoChannel* video_channel =
-      cm_->CreateVideoChannel(&fake_mc_, transport_controller_,
-                              cricket::CN_VIDEO, false, VideoOptions());
+  cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
+      &fake_mc_, transport_controller_, cricket::CN_VIDEO, nullptr, false,
+      VideoOptions());
   EXPECT_TRUE(video_channel != nullptr);
-  cricket::DataChannel* data_channel = cm_->CreateDataChannel(
-      transport_controller_, cricket::CN_DATA, false, cricket::DCT_RTP);
+  cricket::DataChannel* data_channel =
+      cm_->CreateDataChannel(transport_controller_, cricket::CN_DATA, nullptr,
+                             false, cricket::DCT_RTP);
   EXPECT_TRUE(data_channel != nullptr);
   cm_->DestroyVideoChannel(video_channel);
   cm_->DestroyVoiceChannel(voice_channel);
@@ -114,22 +118,25 @@ TEST_F(ChannelManagerTest, CreateDestroyChannels) {
 
 // Test that we can create and destroy a voice and video channel with a worker.
 TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
+  network_.Start();
   worker_.Start();
   EXPECT_TRUE(cm_->set_worker_thread(&worker_));
+  EXPECT_TRUE(cm_->set_network_thread(&network_));
   EXPECT_TRUE(cm_->Init());
   delete transport_controller_;
   transport_controller_ =
-      new cricket::FakeTransportController(&worker_, ICEROLE_CONTROLLING);
-  cricket::VoiceChannel* voice_channel =
-      cm_->CreateVoiceChannel(&fake_mc_, transport_controller_,
-                              cricket::CN_AUDIO, false, AudioOptions());
+      new cricket::FakeTransportController(&network_, ICEROLE_CONTROLLING);
+  cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
+      &fake_mc_, transport_controller_, cricket::CN_AUDIO, nullptr, false,
+      AudioOptions());
   EXPECT_TRUE(voice_channel != nullptr);
-  cricket::VideoChannel* video_channel =
-      cm_->CreateVideoChannel(&fake_mc_, transport_controller_,
-                              cricket::CN_VIDEO, false, VideoOptions());
+  cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
+      &fake_mc_, transport_controller_, cricket::CN_VIDEO, nullptr, false,
+      VideoOptions());
   EXPECT_TRUE(video_channel != nullptr);
-  cricket::DataChannel* data_channel = cm_->CreateDataChannel(
-      transport_controller_, cricket::CN_DATA, false, cricket::DCT_RTP);
+  cricket::DataChannel* data_channel =
+      cm_->CreateDataChannel(transport_controller_, cricket::CN_DATA, nullptr,
+                             false, cricket::DCT_RTP);
   EXPECT_TRUE(data_channel != nullptr);
   cm_->DestroyVideoChannel(video_channel);
   cm_->DestroyVoiceChannel(voice_channel);
@@ -144,19 +151,20 @@ TEST_F(ChannelManagerTest, NoTransportChannelTest) {
   transport_controller_->set_fail_channel_creation(true);
   // The test is useless unless the session does not fail creating
   // cricket::TransportChannel.
-  ASSERT_TRUE(transport_controller_->CreateTransportChannel_w(
+  ASSERT_TRUE(transport_controller_->CreateTransportChannel_n(
                   "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP) == nullptr);
 
-  cricket::VoiceChannel* voice_channel =
-      cm_->CreateVoiceChannel(&fake_mc_, transport_controller_,
-                              cricket::CN_AUDIO, false, AudioOptions());
+  cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
+      &fake_mc_, transport_controller_, cricket::CN_AUDIO, nullptr, false,
+      AudioOptions());
   EXPECT_TRUE(voice_channel == nullptr);
-  cricket::VideoChannel* video_channel =
-      cm_->CreateVideoChannel(&fake_mc_, transport_controller_,
-                              cricket::CN_VIDEO, false, VideoOptions());
+  cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
+      &fake_mc_, transport_controller_, cricket::CN_VIDEO, nullptr, false,
+      VideoOptions());
   EXPECT_TRUE(video_channel == nullptr);
-  cricket::DataChannel* data_channel = cm_->CreateDataChannel(
-      transport_controller_, cricket::CN_DATA, false, cricket::DCT_RTP);
+  cricket::DataChannel* data_channel =
+      cm_->CreateDataChannel(transport_controller_, cricket::CN_DATA, nullptr,
+                             false, cricket::DCT_RTP);
   EXPECT_TRUE(data_channel == nullptr);
   cm_->Terminate();
 }
@@ -198,7 +206,7 @@ TEST_F(ChannelManagerTest, GetSetOutputVolume) {
 
 TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
   std::vector<VideoCodec> codecs;
-  const VideoCodec rtx_codec(96, "rtx", 0, 0, 0, 0);
+  const VideoCodec rtx_codec(96, "rtx", 0, 0, 0);
 
   // By default RTX is disabled.
   cm_->GetSupportedVideoCodecs(&codecs);
diff --git a/chromium/third_party/webrtc/pc/currentspeakermonitor.cc b/chromium/third_party/webrtc/pc/currentspeakermonitor.cc
index bbb33e3143c..ce0d579486a 100644
--- a/chromium/third_party/webrtc/pc/currentspeakermonitor.cc
+++ b/chromium/third_party/webrtc/pc/currentspeakermonitor.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/pc/currentspeakermonitor.h"
 
+#include <vector>
+
 #include "webrtc/base/logging.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/pc/audiomonitor.h"
@@ -61,7 +63,7 @@ void CurrentSpeakerMonitor::Stop() {
 }
 
 void CurrentSpeakerMonitor::set_min_time_between_switches(
-    uint32_t min_time_between_switches) {
+    int min_time_between_switches) {
   min_time_between_switches_ = min_time_between_switches;
 }
 
@@ -163,7 +165,7 @@ void CurrentSpeakerMonitor::OnAudioMonitor(
 
   // We avoid over-switching by disabling switching for a period of time after
   // a switch is done.
-  uint32_t now = rtc::Time();
+  int64_t now = rtc::TimeMillis();
   if (earliest_permitted_switch_time_ <= now &&
       current_speaker_ssrc_ != loudest_speaker_ssrc) {
     current_speaker_ssrc_ = loudest_speaker_ssrc;
diff --git a/chromium/third_party/webrtc/pc/currentspeakermonitor.h b/chromium/third_party/webrtc/pc/currentspeakermonitor.h
index 5ca68ae9416..730ded075b3 100644
--- a/chromium/third_party/webrtc/pc/currentspeakermonitor.h
+++ b/chromium/third_party/webrtc/pc/currentspeakermonitor.h
@@ -11,8 +11,8 @@
 // CurrentSpeakerMonitor monitors the audio levels for a session and determines
 // which participant is currently speaking.
 
-#ifndef TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
-#define TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
+#ifndef WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
+#define WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
 
 #include <map>
 
@@ -45,7 +45,7 @@ class AudioSourceContext {
 // It's recommended that the audio monitor be started with a 100 ms period.
 class CurrentSpeakerMonitor : public sigslot::has_slots<> {
  public:
-  CurrentSpeakerMonitor(AudioSourceContext* audio_source_context);
+  explicit CurrentSpeakerMonitor(AudioSourceContext* audio_source_context);
   ~CurrentSpeakerMonitor();
 
   void Start();
@@ -54,7 +54,7 @@ class CurrentSpeakerMonitor : public sigslot::has_slots<> {
   // Used by tests.  Note that the actual minimum time between switches
   // enforced by the monitor will be the given value plus or minus the
   // resolution of the system clock.
-  void set_min_time_between_switches(uint32_t min_time_between_switches);
+  void set_min_time_between_switches(int min_time_between_switches);
 
   // This is fired when the current speaker changes, and provides his audio
   // SSRC.  This only fires after the audio monitor on the underlying
@@ -86,10 +86,10 @@ class CurrentSpeakerMonitor : public sigslot::has_slots<> {
   uint32_t current_speaker_ssrc_;
   // To prevent overswitching, switching is disabled for some time after a
   // switch is made.  This gives us the earliest time a switch is permitted.
-  uint32_t earliest_permitted_switch_time_;
-  uint32_t min_time_between_switches_;
+  int64_t earliest_permitted_switch_time_;
+  int min_time_between_switches_;
 };
 
-}
+}  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
+#endif  // WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
diff --git a/chromium/third_party/webrtc/pc/externalhmac.h b/chromium/third_party/webrtc/pc/externalhmac.h
index 74652b0894f..edca74d4d47 100644
--- a/chromium/third_party/webrtc/pc/externalhmac.h
+++ b/chromium/third_party/webrtc/pc/externalhmac.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_
-#define TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_
+#ifndef WEBRTC_PC_EXTERNALHMAC_H_
+#define WEBRTC_PC_EXTERNALHMAC_H_
 
 // External libsrtp HMAC auth module which implements methods defined in
 // auth_type_t.
@@ -72,4 +72,4 @@ err_status_t external_hmac_compute(ExternalHmacContext* state,
 err_status_t external_crypto_init();
 
 #endif  // defined(HAVE_SRTP) && defined(ENABLE_EXTERNAL_AUTH)
-#endif  // TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_
+#endif  // WEBRTC_PC_EXTERNALHMAC_H_
diff --git a/chromium/third_party/webrtc/pc/mediamonitor.cc b/chromium/third_party/webrtc/pc/mediamonitor.cc
index c9ba2d39640..066094d159f 100644
--- a/chromium/third_party/webrtc/pc/mediamonitor.cc
+++ b/chromium/third_party/webrtc/pc/mediamonitor.cc
@@ -88,4 +88,4 @@ void MediaMonitor::PollMediaChannel() {
   worker_thread_->PostDelayed(rate_, this, MSG_MONITOR_POLL);
 }
 
-}
+}  // namespace cricket
diff --git a/chromium/third_party/webrtc/pc/mediamonitor.h b/chromium/third_party/webrtc/pc/mediamonitor.h
index c2846bed83d..d294cf13e71 100644
--- a/chromium/third_party/webrtc/pc/mediamonitor.h
+++ b/chromium/third_party/webrtc/pc/mediamonitor.h
@@ -10,8 +10,8 @@
 
 // Class to collect statistics from a media channel
 
-#ifndef TALK_SESSION_MEDIA_MEDIAMONITOR_H_
-#define TALK_SESSION_MEDIA_MEDIAMONITOR_H_
+#ifndef WEBRTC_PC_MEDIAMONITOR_H_
+#define WEBRTC_PC_MEDIAMONITOR_H_
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/sigslot.h"
@@ -79,4 +79,4 @@ typedef MediaMonitorT<DataMediaChannel, DataMediaInfo> DataMediaMonitor;
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_MEDIAMONITOR_H_
+#endif  // WEBRTC_PC_MEDIAMONITOR_H_
diff --git a/chromium/third_party/webrtc/pc/mediasession.cc b/chromium/third_party/webrtc/pc/mediasession.cc
index a9d1b95c22a..0fa20d8bd5d 100644
--- a/chromium/third_party/webrtc/pc/mediasession.cc
+++ b/chromium/third_party/webrtc/pc/mediasession.cc
@@ -10,11 +10,12 @@
 
 #include "webrtc/pc/mediasession.h"
 
-#include <algorithm>  // For std::find_if.
+#include <algorithm>  // For std::find_if, std::sort.
 #include <functional>
 #include <map>
 #include <memory>
 #include <set>
+#include <unordered_map>
 #include <utility>
 
 #include "webrtc/base/helpers.h"
@@ -45,7 +46,7 @@ void GetSupportedCryptoSuiteNames(void (*func)(std::vector<int>*),
   }
 #endif
 }
-}
+}  // namespace
 
 namespace cricket {
 
@@ -210,55 +211,6 @@ static bool SelectCrypto(const MediaContentDescription* offer,
   return false;
 }
 
-static const StreamParams* FindFirstStreamParamsByCname(
-    const StreamParamsVec& params_vec,
-    const std::string& cname) {
-  for (StreamParamsVec::const_iterator it = params_vec.begin();
-       it != params_vec.end(); ++it) {
-    if (cname == it->cname)
-      return &*it;
-  }
-  return NULL;
-}
-
-// Generates a new CNAME or the CNAME of an already existing StreamParams
-// if a StreamParams exist for another Stream in streams with sync_label
-// sync_label.
-static bool GenerateCname(const StreamParamsVec& params_vec,
-                          const MediaSessionOptions::Streams& streams,
-                          const std::string& synch_label,
-                          std::string* cname) {
-  ASSERT(cname != NULL);
-  if (!cname)
-    return false;
-
-  // Check if a CNAME exist for any of the other synched streams.
-  for (MediaSessionOptions::Streams::const_iterator stream_it = streams.begin();
-       stream_it != streams.end() ; ++stream_it) {
-    if (synch_label != stream_it->sync_label)
-      continue;
-
-    // groupid is empty for StreamParams generated using
-    // MediaSessionDescriptionFactory.
-    const StreamParams* param = GetStreamByIds(params_vec, "", stream_it->id);
-    if (param) {
-      *cname = param->cname;
-      return true;
-    }
-  }
-  // No other stream seems to exist that we should sync with.
-  // Generate a random string for the RTCP CNAME, as stated in RFC 6222.
-  // This string is only used for synchronization, and therefore is opaque.
-  do {
-    if (!rtc::CreateRandomString(16, cname)) {
-      ASSERT(false);
-      return false;
-    }
-  } while (FindFirstStreamParamsByCname(params_vec, *cname));
-
-  return true;
-}
-
 // Generate random SSRC values that are not already present in |params_vec|.
 // The generated values are added to |ssrcs|.
 // |num_ssrcs| is the number of the SSRC will be generated.
@@ -443,15 +395,15 @@ static bool IsSctp(const MediaContentDescription* desc) {
 // media_type to content_description.
 // |current_params| - All currently known StreamParams of any media type.
 template <class C>
-static bool AddStreamParams(
-    MediaType media_type,
-    const MediaSessionOptions::Streams& streams,
-    StreamParamsVec* current_streams,
-    MediaContentDescriptionImpl<C>* content_description,
-    const bool add_legacy_stream) {
+static bool AddStreamParams(MediaType media_type,
+                            const MediaSessionOptions& options,
+                            StreamParamsVec* current_streams,
+                            MediaContentDescriptionImpl<C>* content_description,
+                            const bool add_legacy_stream) {
   const bool include_rtx_streams =
       ContainsRtxCodec(content_description->codecs());
 
+  const MediaSessionOptions::Streams& streams = options.streams;
   if (streams.empty() && add_legacy_stream) {
     // TODO(perkj): Remove this legacy stream when all apps use StreamParams.
     std::vector<uint32_t> ssrcs;
@@ -482,13 +434,6 @@ static bool AddStreamParams(
     // MediaSessionDescriptionFactory.
     if (!param) {
       // This is a new stream.
-      // Get a CNAME. Either new or same as one of the other synched streams.
-      std::string cname;
-      if (!GenerateCname(*current_streams, streams, stream_it->sync_label,
-                         &cname)) {
-        return false;
-      }
-
       std::vector<uint32_t> ssrcs;
       if (IsSctp(content_description)) {
         GenerateSctpSids(*current_streams, &ssrcs);
@@ -516,7 +461,7 @@ static bool AddStreamParams(
         }
         content_description->set_multistream(true);
       }
-      stream_param.cname = cname;
+      stream_param.cname = options.rtcp_cname;
       stream_param.sync_label = stream_it->sync_label;
       content_description->AddStream(stream_param);
 
@@ -749,7 +694,6 @@ static bool CreateMediaContentOffer(
     StreamParamsVec* current_streams,
     MediaContentDescriptionImpl<C>* offer) {
   offer->AddCodecs(codecs);
-  offer->SortCodecs();
 
   if (secure_policy == SEC_REQUIRED) {
     offer->set_crypto_required(CT_SDES);
@@ -761,9 +705,8 @@ static bool CreateMediaContentOffer(
   offer->set_multistream(options.is_muc);
   offer->set_rtp_header_extensions(rtp_extensions);
 
-  if (!AddStreamParams(
-          offer->type(), options.streams, current_streams,
-          offer, add_legacy_stream)) {
+  if (!AddStreamParams(offer->type(), options, current_streams, offer,
+                       add_legacy_stream)) {
     return false;
   }
 
@@ -810,6 +753,8 @@ static void NegotiateCodecs(const std::vector<C>& local_codecs,
                             std::vector<C>* negotiated_codecs) {
   for (const C& ours : local_codecs) {
     C theirs;
+    // Note that we intentionally only find one matching codec for each of our
+    // local codecs, in case the remote offer contains duplicate codecs.
     if (FindMatchingCodec(local_codecs, offered_codecs, ours, &theirs)) {
       C negotiated = ours;
       negotiated.IntersectFeedbackParams(theirs);
@@ -822,14 +767,23 @@ static void NegotiateCodecs(const std::vector<C>& local_codecs,
                             offered_apt_value);
       }
       negotiated.id = theirs.id;
-      // RFC3264: Although the answerer MAY list the formats in their desired
-      // order of preference, it is RECOMMENDED that unless there is a
-      // specific reason, the answerer list formats in the same relative order
-      // they were present in the offer.
-      negotiated.preference = theirs.preference;
       negotiated_codecs->push_back(negotiated);
     }
   }
+  // RFC3264: Although the answerer MAY list the formats in their desired
+  // order of preference, it is RECOMMENDED that unless there is a
+  // specific reason, the answerer list formats in the same relative order
+  // they were present in the offer.
+  std::unordered_map<int, int> payload_type_preferences;
+  int preference = static_cast<int>(offered_codecs.size() + 1);
+  for (const C& codec : offered_codecs) {
+    payload_type_preferences[codec.id] = preference--;
+  }
+  std::sort(negotiated_codecs->begin(), negotiated_codecs->end(),
+            [&payload_type_preferences](const C& a, const C& b) {
+              return payload_type_preferences[a.id] >
+                     payload_type_preferences[b.id];
+            });
 }
 
 // Finds a codec in |codecs2| that matches |codec_to_match|, which is
@@ -1042,7 +996,6 @@ static bool CreateMediaContentAnswer(
   std::vector<C> negotiated_codecs;
   NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs);
   answer->AddCodecs(negotiated_codecs);
-  answer->SortCodecs();
   answer->set_protocol(offer->protocol());
   RtpHeaderExtensions negotiated_rtp_extensions;
   NegotiateRtpHeaderExtensions(local_rtp_extenstions,
@@ -1070,9 +1023,8 @@ static bool CreateMediaContentAnswer(
     return false;
   }
 
-  if (!AddStreamParams(
-          answer->type(), options.streams, current_streams,
-          answer, add_legacy_stream)) {
+  if (!AddStreamParams(answer->type(), options, current_streams, answer,
+                       add_legacy_stream)) {
     return false;  // Something went seriously wrong.
   }
 
@@ -1105,21 +1057,55 @@ static bool CreateMediaContentAnswer(
   return true;
 }
 
+static bool IsDtlsRtp(const std::string& protocol) {
+  // Most-likely values first.
+  return protocol == "UDP/TLS/RTP/SAVPF" || protocol == "TCP/TLS/RTP/SAVPF" ||
+         protocol == "UDP/TLS/RTP/SAVP" || protocol == "TCP/TLS/RTP/SAVP";
+}
+
+static bool IsPlainRtp(const std::string& protocol) {
+  // Most-likely values first.
+  return protocol == "RTP/SAVPF" || protocol == "RTP/AVPF" ||
+         protocol == "RTP/SAVP" || protocol == "RTP/AVP";
+}
+
+static bool IsDtlsSctp(const std::string& protocol) {
+  return protocol == "DTLS/SCTP";
+}
+
+static bool IsPlainSctp(const std::string& protocol) {
+  return protocol == "SCTP";
+}
+
 static bool IsMediaProtocolSupported(MediaType type,
                                      const std::string& protocol,
                                      bool secure_transport) {
-  // Data channels can have a protocol of SCTP or SCTP/DTLS.
-  if (type == MEDIA_TYPE_DATA &&
-      ((protocol == kMediaProtocolSctp && !secure_transport)||
-       (protocol == kMediaProtocolDtlsSctp && secure_transport))) {
+  // Since not all applications serialize and deserialize the media protocol,
+  // we will have to accept |protocol| to be empty.
+  if (protocol.empty()) {
     return true;
   }
 
-  // Since not all applications serialize and deserialize the media protocol,
-  // we will have to accept |protocol| to be empty.
-  return protocol == kMediaProtocolAvpf || protocol.empty() ||
-      protocol == kMediaProtocolSavpf ||
-      (protocol == kMediaProtocolDtlsSavpf && secure_transport);
+  if (type == MEDIA_TYPE_DATA) {
+    // Check for SCTP, but also for RTP for RTP-based data channels.
+    // TODO(pthatcher): Remove RTP once RTP-based data channels are gone.
+    if (secure_transport) {
+      // Most likely scenarios first.
+      return IsDtlsSctp(protocol) || IsDtlsRtp(protocol) ||
+             IsPlainRtp(protocol);
+    } else {
+      return IsPlainSctp(protocol) || IsPlainRtp(protocol);
+    }
+  }
+
+  // Allow for non-DTLS RTP protocol even when using DTLS because that's what
+  // JSEP specifies.
+  if (secure_transport) {
+    // Most likely scenarios first.
+    return IsDtlsRtp(protocol) || IsPlainRtp(protocol);
+  } else {
+    return IsPlainRtp(protocol);
+  }
 }
 
 static void SetMediaProtocol(bool secure_transport,
@@ -1365,8 +1351,8 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
     const SessionDescription* offer, const MediaSessionOptions& options,
     const SessionDescription* current_description) const {
   // The answer contains the intersection of the codecs in the offer with the
-  // codecs we support, ordered by our local preference. As indicated by
-  // XEP-0167, we retain the same payload ids from the offer in the answer.
+  // codecs we support. As indicated by XEP-0167, we retain the same payload ids
+  // from the offer in the answer.
   std::unique_ptr<SessionDescription> answer(new SessionDescription());
 
   StreamParamsVec current_streams;
@@ -1933,10 +1919,9 @@ bool IsDataContent(const ContentInfo* content) {
 
 const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
                                         MediaType media_type) {
-  for (ContentInfos::const_iterator content = contents.begin();
-       content != contents.end(); content++) {
-    if (IsMediaContentOfType(&*content, media_type)) {
-      return &*content;
+  for (const ContentInfo& content : contents) {
+    if (IsMediaContentOfType(&content, media_type)) {
+      return &content;
     }
   }
   return nullptr;
@@ -2000,4 +1985,77 @@ const DataContentDescription* GetFirstDataContentDescription(
       GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA));
 }
 
+//
+// Non-const versions of the above functions.
+//
+
+ContentInfo* GetFirstMediaContent(ContentInfos& contents,
+                                  MediaType media_type) {
+  for (ContentInfo& content : contents) {
+    if (IsMediaContentOfType(&content, media_type)) {
+      return &content;
+    }
+  }
+  return nullptr;
+}
+
+ContentInfo* GetFirstAudioContent(ContentInfos& contents) {
+  return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
+}
+
+ContentInfo* GetFirstVideoContent(ContentInfos& contents) {
+  return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
+}
+
+ContentInfo* GetFirstDataContent(ContentInfos& contents) {
+  return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
+}
+
+static ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
+                                         MediaType media_type) {
+  if (sdesc == nullptr) {
+    return nullptr;
+  }
+
+  return GetFirstMediaContent(sdesc->contents(), media_type);
+}
+
+ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {
+  return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
+}
+
+ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) {
+  return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
+}
+
+ContentInfo* GetFirstDataContent(SessionDescription* sdesc) {
+  return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
+}
+
+MediaContentDescription* GetFirstMediaContentDescription(
+    SessionDescription* sdesc,
+    MediaType media_type) {
+  ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
+  ContentDescription* description = content ? content->description : NULL;
+  return static_cast<MediaContentDescription*>(description);
+}
+
+AudioContentDescription* GetFirstAudioContentDescription(
+    SessionDescription* sdesc) {
+  return static_cast<AudioContentDescription*>(
+      GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO));
+}
+
+VideoContentDescription* GetFirstVideoContentDescription(
+    SessionDescription* sdesc) {
+  return static_cast<VideoContentDescription*>(
+      GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO));
+}
+
+DataContentDescription* GetFirstDataContentDescription(
+    SessionDescription* sdesc) {
+  return static_cast<DataContentDescription*>(
+      GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA));
+}
+
 }  // namespace cricket
diff --git a/chromium/third_party/webrtc/pc/mediasession.h b/chromium/third_party/webrtc/pc/mediasession.h
index ae221554e20..22291c42a64 100644
--- a/chromium/third_party/webrtc/pc/mediasession.h
+++ b/chromium/third_party/webrtc/pc/mediasession.h
@@ -10,8 +10,8 @@
 
 // Types and classes used in media session descriptions.
 
-#ifndef TALK_SESSION_MEDIA_MEDIASESSION_H_
-#define TALK_SESSION_MEDIA_MEDIASESSION_H_
+#ifndef WEBRTC_PC_MEDIASESSION_H_
+#define WEBRTC_PC_MEDIASESSION_H_
 
 #include <algorithm>
 #include <map>
@@ -76,18 +76,21 @@ extern const char kMediaProtocolTcpDtlsSctp[];
 const int kAutoBandwidth = -1;
 const int kBufferedModeDisabled = 0;
 
+// Default RTCP CNAME for unit tests.
+const char kDefaultRtcpCname[] = "DefaultRtcpCname";
+
 struct MediaSessionOptions {
-  MediaSessionOptions() :
-      recv_audio(true),
-      recv_video(false),
-      data_channel_type(DCT_NONE),
-      is_muc(false),
-      vad_enabled(true),  // When disabled, removes all CN codecs from SDP.
-      rtcp_mux_enabled(true),
-      bundle_enabled(false),
-      video_bandwidth(kAutoBandwidth),
-      data_bandwidth(kDataMaxBandwidth) {
-  }
+  MediaSessionOptions()
+      : recv_audio(true),
+        recv_video(false),
+        data_channel_type(DCT_NONE),
+        is_muc(false),
+        vad_enabled(true),  // When disabled, removes all CN codecs from SDP.
+        rtcp_mux_enabled(true),
+        bundle_enabled(false),
+        video_bandwidth(kAutoBandwidth),
+        data_bandwidth(kDataMaxBandwidth),
+        rtcp_cname(kDefaultRtcpCname) {}
 
   bool has_audio() const {
     return recv_audio || HasSendMediaStream(MEDIA_TYPE_AUDIO);
@@ -133,6 +136,7 @@ struct MediaSessionOptions {
   int data_bandwidth;
   // content name ("mid") => options.
   std::map<std::string, TransportOptions> transport_options;
+  std::string rtcp_cname;
 
   struct Stream {
     Stream(MediaType type,
@@ -293,10 +297,9 @@ class MediaContentDescription : public ContentDescription {
 template <class C>
 class MediaContentDescriptionImpl : public MediaContentDescription {
  public:
-  struct PreferenceSort {
-    bool operator()(C a, C b) { return a.preference > b.preference; }
-  };
+  typedef C CodecType;
 
+  // Codecs should be in preference order (most preferred codec first).
   const std::vector<C>& codecs() const { return codecs_; }
   void set_codecs(const std::vector<C>& codecs) { codecs_ = codecs; }
   virtual bool has_codecs() const { return !codecs_.empty(); }
@@ -330,9 +333,6 @@ class MediaContentDescriptionImpl : public MediaContentDescription {
       AddCodec(*codec);
     }
   }
-  void SortCodecs() {
-    std::sort(codecs_.begin(), codecs_.end(), PreferenceSort());
-  }
 
  private:
   std::vector<C> codecs_;
@@ -532,6 +532,21 @@ const VideoContentDescription* GetFirstVideoContentDescription(
     const SessionDescription* sdesc);
 const DataContentDescription* GetFirstDataContentDescription(
     const SessionDescription* sdesc);
+// Non-const versions of the above functions.
+// Useful when modifying an existing description.
+ContentInfo* GetFirstMediaContent(ContentInfos& contents, MediaType media_type);
+ContentInfo* GetFirstAudioContent(ContentInfos& contents);
+ContentInfo* GetFirstVideoContent(ContentInfos& contents);
+ContentInfo* GetFirstDataContent(ContentInfos& contents);
+ContentInfo* GetFirstAudioContent(SessionDescription* sdesc);
+ContentInfo* GetFirstVideoContent(SessionDescription* sdesc);
+ContentInfo* GetFirstDataContent(SessionDescription* sdesc);
+AudioContentDescription* GetFirstAudioContentDescription(
+    SessionDescription* sdesc);
+VideoContentDescription* GetFirstVideoContentDescription(
+    SessionDescription* sdesc);
+DataContentDescription* GetFirstDataContentDescription(
+    SessionDescription* sdesc);
 
 void GetSupportedAudioCryptoSuites(std::vector<int>* crypto_suites);
 void GetSupportedVideoCryptoSuites(std::vector<int>* crypto_suites);
@@ -548,4 +563,4 @@ void GetDefaultSrtpCryptoSuiteNames(
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_MEDIASESSION_H_
+#endif  // WEBRTC_PC_MEDIASESSION_H_
diff --git a/chromium/third_party/webrtc/pc/mediasession_unittest.cc b/chromium/third_party/webrtc/pc/mediasession_unittest.cc
index 6ea7aeb8ab9..b14bd801cbe 100644
--- a/chromium/third_party/webrtc/pc/mediasession_unittest.cc
+++ b/chromium/third_party/webrtc/pc/mediasession_unittest.cc
@@ -75,53 +75,43 @@ using rtc::CS_AES_CM_128_HMAC_SHA1_32;
 using rtc::CS_AES_CM_128_HMAC_SHA1_80;
 
 static const AudioCodec kAudioCodecs1[] = {
-  AudioCodec(103, "ISAC",   16000, -1,    1, 6),
-  AudioCodec(102, "iLBC",   8000,  13300, 1, 5),
-  AudioCodec(0,   "PCMU",   8000,  64000, 1, 4),
-  AudioCodec(8,   "PCMA",   8000,  64000, 1, 3),
-  AudioCodec(117, "red",    8000,  0,     1, 2),
-  AudioCodec(107, "CN",     48000, 0,     1, 1)
-};
+    AudioCodec(103, "ISAC", 16000, -1, 1),
+    AudioCodec(102, "iLBC", 8000, 13300, 1),
+    AudioCodec(0, "PCMU", 8000, 64000, 1),
+    AudioCodec(8, "PCMA", 8000, 64000, 1),
+    AudioCodec(117, "red", 8000, 0, 1),
+    AudioCodec(107, "CN", 48000, 0, 1)};
 
 static const AudioCodec kAudioCodecs2[] = {
-  AudioCodec(126, "speex",  16000, 22000, 1, 3),
-  AudioCodec(0,   "PCMU",   8000,  64000, 1, 2),
-  AudioCodec(127, "iLBC",   8000,  13300, 1, 1),
+    AudioCodec(126, "speex", 16000, 22000, 1),
+    AudioCodec(0, "PCMU", 8000, 64000, 1),
+    AudioCodec(127, "iLBC", 8000, 13300, 1),
 };
 
 static const AudioCodec kAudioCodecsAnswer[] = {
-  AudioCodec(102, "iLBC",   8000,  13300, 1, 5),
-  AudioCodec(0,   "PCMU",   8000,  64000, 1, 4),
+    AudioCodec(102, "iLBC", 8000, 13300, 1),
+    AudioCodec(0, "PCMU", 8000, 64000, 1),
 };
 
 static const VideoCodec kVideoCodecs1[] = {
-  VideoCodec(96, "H264-SVC", 320, 200, 30, 2),
-  VideoCodec(97, "H264", 320, 200, 30, 1)
-};
+    VideoCodec(96, "H264-SVC", 320, 200, 30),
+    VideoCodec(97, "H264", 320, 200, 30)};
 
 static const VideoCodec kVideoCodecs2[] = {
-  VideoCodec(126, "H264", 320, 200, 30, 2),
-  VideoCodec(127, "H263", 320, 200, 30, 1)
-};
+    VideoCodec(126, "H264", 320, 200, 30),
+    VideoCodec(127, "H263", 320, 200, 30)};
 
 static const VideoCodec kVideoCodecsAnswer[] = {
-  VideoCodec(97, "H264", 320, 200, 30, 1)
-};
+    VideoCodec(97, "H264", 320, 200, 30)};
 
-static const DataCodec kDataCodecs1[] = {
-  DataCodec(98, "binary-data", 2),
-  DataCodec(99, "utf8-text", 1)
-};
+static const DataCodec kDataCodecs1[] = {DataCodec(98, "binary-data"),
+                                         DataCodec(99, "utf8-text")};
 
-static const DataCodec kDataCodecs2[] = {
-  DataCodec(126, "binary-data", 2),
-  DataCodec(127, "utf8-text", 1)
-};
+static const DataCodec kDataCodecs2[] = {DataCodec(126, "binary-data"),
+                                         DataCodec(127, "utf8-text")};
 
-static const DataCodec kDataCodecsAnswer[] = {
-  DataCodec(98, "binary-data", 2),
-  DataCodec(99, "utf8-text", 1)
-};
+static const DataCodec kDataCodecsAnswer[] = {DataCodec(98, "binary-data"),
+                                              DataCodec(99, "utf8-text")};
 
 static const RtpHeaderExtension kAudioRtpExtension1[] = {
   RtpHeaderExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
@@ -180,6 +170,12 @@ static const char kDataTrack1[] = "data_1";
 static const char kDataTrack2[] = "data_2";
 static const char kDataTrack3[] = "data_3";
 
+static const char* kMediaProtocols[] = {"RTP/AVP", "RTP/SAVP", "RTP/AVPF",
+                                        "RTP/SAVPF"};
+static const char* kMediaProtocolsDtls[] = {
+    "TCP/TLS/RTP/SAVPF", "TCP/TLS/RTP/SAVP", "UDP/TLS/RTP/SAVPF",
+    "UDP/TLS/RTP/SAVP"};
+
 static bool IsMediaContentOfType(const ContentInfo* content,
                                  MediaType media_type) {
   const MediaContentDescription* mdesc =
@@ -222,9 +218,9 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
     f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2));
     f2_.set_data_codecs(MAKE_VECTOR(kDataCodecs2));
     tdf1_.set_certificate(rtc::RTCCertificate::Create(
-        rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
+        std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
     tdf2_.set_certificate(rtc::RTCCertificate::Create(
-        rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
+        std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
   }
 
   // Create a video StreamParamsVec object with:
@@ -420,12 +416,12 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
     opts.recv_video = true;
     std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
     ASSERT_TRUE(offer.get() != NULL);
-    ContentInfo* ac_offer= offer->GetContentByName("audio");
+    ContentInfo* ac_offer = offer->GetContentByName("audio");
     ASSERT_TRUE(ac_offer != NULL);
     AudioContentDescription* acd_offer =
         static_cast<AudioContentDescription*>(ac_offer->description);
     acd_offer->set_direction(direction_in_offer);
-    ContentInfo* vc_offer= offer->GetContentByName("video");
+    ContentInfo* vc_offer = offer->GetContentByName("video");
     ASSERT_TRUE(vc_offer != NULL);
     VideoContentDescription* vcd_offer =
         static_cast<VideoContentDescription*>(vc_offer->description);
@@ -899,7 +895,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   f1_.set_secure(SEC_ENABLED);
   f2_.set_secure(SEC_ENABLED);
   std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
-  ContentInfo* dc_offer= offer->GetContentByName("data");
+  ContentInfo* dc_offer = offer->GetContentByName("data");
   ASSERT_TRUE(dc_offer != NULL);
   DataContentDescription* dcd_offer =
       static_cast<DataContentDescription*>(dc_offer->description);
@@ -1316,7 +1312,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   ASSERT_EQ(2U, updated_video_streams.size());
   EXPECT_EQ(video_streams[0], updated_video_streams[0]);
   EXPECT_EQ(kVideoTrack2, updated_video_streams[1].id);
-  EXPECT_NE(updated_video_streams[1].cname, updated_video_streams[0].cname);
+  // All the media streams in one PeerConnection share one RTCP CNAME.
+  EXPECT_EQ(updated_video_streams[1].cname, updated_video_streams[0].cname);
 
   const StreamParamsVec& updated_data_streams = updated_dcd->streams();
   ASSERT_EQ(2U, updated_data_streams.size());
@@ -1325,6 +1322,10 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   ASSERT_EQ(1U, updated_data_streams[1].ssrcs.size());
   EXPECT_NE(0U, updated_data_streams[1].ssrcs[0]);
   EXPECT_EQ(updated_data_streams[0].cname, updated_data_streams[1].cname);
+  // The stream correctly got the CNAME from the MediaSessionOptions.
+  // The Expected RTCP CNAME is the default one as we are using the default
+  // MediaSessionOptions.
+  EXPECT_EQ(updated_data_streams[0].cname, cricket::kDefaultRtcpCname);
 }
 
 // Create an offer with simulcast video stream.
@@ -1435,7 +1436,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   EXPECT_TRUE(dcd->rtcp_mux());                 // rtcp-mux defaults on
 
   // Update the answer. Add a new video track that is not synched to the
-  // other traacks and remove 1 audio track.
+  // other tracks and remove 1 audio track.
   opts.AddSendStream(MEDIA_TYPE_VIDEO, kVideoTrack2, kMediaStream2);
   opts.RemoveSendStream(MEDIA_TYPE_AUDIO, kAudioTrack2);
   opts.RemoveSendStream(MEDIA_TYPE_DATA, kDataTrack2);
@@ -1478,7 +1479,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   ASSERT_EQ(2U, updated_video_streams.size());
   EXPECT_EQ(video_streams[0], updated_video_streams[0]);
   EXPECT_EQ(kVideoTrack2, updated_video_streams[1].id);
-  EXPECT_NE(updated_video_streams[1].cname, updated_video_streams[0].cname);
+  // All media streams in one PeerConnection share one CNAME.
+  EXPECT_EQ(updated_video_streams[1].cname, updated_video_streams[0].cname);
 
   const StreamParamsVec& updated_data_streams = updated_dcd->streams();
   ASSERT_EQ(1U, updated_data_streams.size());
@@ -1570,11 +1572,9 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   EXPECT_EQ(expected_codecs, vcd->codecs());
 
-  // Now, make sure we get same result, except for the preference order,
-  // if |f2_| creates an updated offer even though the default payload types
-  // are different from |f1_|.
-  expected_codecs[0].preference = f1_codecs[1].preference;
-
+  // Now, make sure we get same result (except for the order) if |f2_| creates
+  // an updated offer even though the default payload types between |f1_| and
+  // |f2_| are different.
   std::unique_ptr<SessionDescription> updated_offer(
       f2_.CreateOffer(opts, answer.get()));
   ASSERT_TRUE(updated_offer);
@@ -1694,7 +1694,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, RtxWithoutApt) {
   opts.recv_audio = false;
   std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
   // This creates RTX without associated payload type parameter.
-  AddRtxCodec(VideoCodec(126, cricket::kRtxCodecName, 0, 0, 0, 0), &f1_codecs);
+  AddRtxCodec(VideoCodec(126, cricket::kRtxCodecName, 0, 0, 0), &f1_codecs);
   f1_.set_video_codecs(f1_codecs);
 
   std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
@@ -1842,7 +1842,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
 
   // Use a single real codec, and then add RTX for it.
   std::vector<VideoCodec> f1_codecs;
-  f1_codecs.push_back(VideoCodec(97, "H264", 320, 200, 30, 1));
+  f1_codecs.push_back(VideoCodec(97, "H264", 320, 200, 30));
   AddRtxCodec(VideoCodec::CreateRtxCodec(125, 97), &f1_codecs);
   f1_.set_video_codecs(f1_codecs);
 
@@ -2391,3 +2391,62 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   EXPECT_EQ("video_modified", video_content->name);
   EXPECT_EQ("data_modified", data_content->name);
 }
+
+class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
+ public:
+  MediaProtocolTest() : f1_(&tdf1_), f2_(&tdf2_) {
+    f1_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs1));
+    f1_.set_video_codecs(MAKE_VECTOR(kVideoCodecs1));
+    f1_.set_data_codecs(MAKE_VECTOR(kDataCodecs1));
+    f2_.set_audio_codecs(MAKE_VECTOR(kAudioCodecs2));
+    f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2));
+    f2_.set_data_codecs(MAKE_VECTOR(kDataCodecs2));
+    f1_.set_secure(SEC_ENABLED);
+    f2_.set_secure(SEC_ENABLED);
+    tdf1_.set_certificate(rtc::RTCCertificate::Create(
+        std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
+    tdf2_.set_certificate(rtc::RTCCertificate::Create(
+        std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
+    tdf1_.set_secure(SEC_ENABLED);
+    tdf2_.set_secure(SEC_ENABLED);
+  }
+
+ protected:
+  MediaSessionDescriptionFactory f1_;
+  MediaSessionDescriptionFactory f2_;
+  TransportDescriptionFactory tdf1_;
+  TransportDescriptionFactory tdf2_;
+};
+
+TEST_P(MediaProtocolTest, TestAudioVideoAcceptance) {
+  MediaSessionOptions opts;
+  opts.recv_video = true;
+  std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, nullptr));
+  ASSERT_TRUE(offer.get() != nullptr);
+  // Set the protocol for all the contents.
+  for (auto content : offer.get()->contents()) {
+    static_cast<MediaContentDescription*>(content.description)
+        ->set_protocol(GetParam());
+  }
+  std::unique_ptr<SessionDescription> answer(
+      f2_.CreateAnswer(offer.get(), opts, nullptr));
+  const ContentInfo* ac = answer->GetContentByName("audio");
+  const ContentInfo* vc = answer->GetContentByName("video");
+  ASSERT_TRUE(ac != nullptr);
+  ASSERT_TRUE(vc != nullptr);
+  EXPECT_FALSE(ac->rejected);  // the offer is accepted
+  EXPECT_FALSE(vc->rejected);
+  const AudioContentDescription* acd =
+      static_cast<const AudioContentDescription*>(ac->description);
+  const VideoContentDescription* vcd =
+      static_cast<const VideoContentDescription*>(vc->description);
+  EXPECT_EQ(GetParam(), acd->protocol());
+  EXPECT_EQ(GetParam(), vcd->protocol());
+}
+
+INSTANTIATE_TEST_CASE_P(MediaProtocolPatternTest,
+                        MediaProtocolTest,
+                        ::testing::ValuesIn(kMediaProtocols));
+INSTANTIATE_TEST_CASE_P(MediaProtocolDtlsPatternTest,
+                        MediaProtocolTest,
+                        ::testing::ValuesIn(kMediaProtocolsDtls));
diff --git a/chromium/third_party/webrtc/pc/mediasink.h b/chromium/third_party/webrtc/pc/mediasink.h
index e97a390aedd..01bd3fa1739 100644
--- a/chromium/third_party/webrtc/pc/mediasink.h
+++ b/chromium/third_party/webrtc/pc/mediasink.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TALK_SESSION_MEDIA_MEDIASINK_H_
-#define TALK_SESSION_MEDIA_MEDIASINK_H_
+#ifndef WEBRTC_PC_MEDIASINK_H_
+#define WEBRTC_PC_MEDIASINK_H_
 
 namespace cricket {
 
@@ -28,4 +28,4 @@ class MediaSinkInterface {
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_MEDIASINK_H_
+#endif  // WEBRTC_PC_MEDIASINK_H_
diff --git a/chromium/third_party/webrtc/pc/pc.gyp b/chromium/third_party/webrtc/pc/pc.gyp
index b07f7b72638..25ebc5de78c 100755
--- a/chromium/third_party/webrtc/pc/pc.gyp
+++ b/chromium/third_party/webrtc/pc/pc.gyp
@@ -33,11 +33,6 @@
       'defines': [
         '<@(rtc_pc_defines)',
       ],
-      # TODO(kjellander): Make the code compile without disabling these flags.
-      # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
-      'cflags_cc!': [
-        '-Wnon-virtual-dtor',
-      ],
       'include_dirs': [
         '<(DEPTH)/testing/gtest/include',
       ],
@@ -97,11 +92,6 @@
             'rtcpmuxfilter_unittest.cc',
             'srtpfilter_unittest.cc',
           ],
-          # TODO(kjellander): Make the code compile without disabling these flags.
-          # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
-          'cflags_cc!': [
-            '-Wnon-virtual-dtor',
-          ],
           'conditions': [
             ['clang==0', {
               'cflags': [
diff --git a/chromium/third_party/webrtc/pc/rtcpmuxfilter.h b/chromium/third_party/webrtc/pc/rtcpmuxfilter.h
index 272c804bec1..94dc41d980e 100644
--- a/chromium/third_party/webrtc/pc/rtcpmuxfilter.h
+++ b/chromium/third_party/webrtc/pc/rtcpmuxfilter.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
-#define TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
+#ifndef WEBRTC_PC_RTCPMUXFILTER_H_
+#define WEBRTC_PC_RTCPMUXFILTER_H_
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/p2p/base/sessiondescription.h"
@@ -69,4 +69,4 @@ class RtcpMuxFilter {
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
+#endif  // WEBRTC_PC_RTCPMUXFILTER_H_
diff --git a/chromium/third_party/webrtc/pc/srtpfilter.cc b/chromium/third_party/webrtc/pc/srtpfilter.cc
index e8ea2890eda..e4796fd6720 100644
--- a/chromium/third_party/webrtc/pc/srtpfilter.cc
+++ b/chromium/third_party/webrtc/pc/srtpfilter.cc
@@ -259,7 +259,7 @@ bool SrtpFilter::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
   return send_session_->GetRtpAuthParams(key, key_len, tag_len);
 }
 
-void SrtpFilter::set_signal_silent_time(uint32_t signal_silent_time_in_ms) {
+void SrtpFilter::set_signal_silent_time(int signal_silent_time_in_ms) {
   signal_silent_time_in_ms_ = signal_silent_time_in_ms;
   if (IsActive()) {
     ASSERT(send_session_ != NULL);
@@ -641,7 +641,7 @@ bool SrtpSession::GetSendStreamPacketIndex(void* p,
   return true;
 }
 
-void SrtpSession::set_signal_silent_time(uint32_t signal_silent_time_in_ms) {
+void SrtpSession::set_signal_silent_time(int signal_silent_time_in_ms) {
   srtp_stat_->set_signal_silent_time(signal_silent_time_in_ms);
 }
 
@@ -891,10 +891,10 @@ void SrtpStat::HandleSrtpResult(const SrtpStat::FailureKey& key) {
   if (key.error != SrtpFilter::ERROR_NONE) {
     // For errors, signal first time and wait for 1 sec.
     FailureStat* stat = &(failures_[key]);
-    uint32_t current_time = rtc::Time();
+    int64_t current_time = rtc::TimeMillis();
     if (stat->last_signal_time == 0 ||
         rtc::TimeDiff(current_time, stat->last_signal_time) >
-        static_cast<int>(signal_silent_time_)) {
+            signal_silent_time_) {
       SignalSrtpError(key.ssrc, key.mode, key.error);
       stat->last_signal_time = current_time;
     }
diff --git a/chromium/third_party/webrtc/pc/srtpfilter.h b/chromium/third_party/webrtc/pc/srtpfilter.h
index f4c1d33dbfa..b54eb8bc866 100644
--- a/chromium/third_party/webrtc/pc/srtpfilter.h
+++ b/chromium/third_party/webrtc/pc/srtpfilter.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TALK_SESSION_MEDIA_SRTPFILTER_H_
-#define TALK_SESSION_MEDIA_SRTPFILTER_H_
+#ifndef WEBRTC_PC_SRTPFILTER_H_
+#define WEBRTC_PC_SRTPFILTER_H_
 
 #include <list>
 #include <map>
@@ -18,6 +18,7 @@
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/sigslotrepeater.h"
 #include "webrtc/base/sslstreamadapter.h"
@@ -119,7 +120,7 @@ class SrtpFilter {
   bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
 
   // Update the silent threshold (in ms) for signaling errors.
-  void set_signal_silent_time(uint32_t signal_silent_time_in_ms);
+  void set_signal_silent_time(int signal_silent_time_in_ms);
 
   bool ResetParams();
 
@@ -165,7 +166,7 @@ class SrtpFilter {
     ST_RECEIVEDPRANSWER
   };
   State state_;
-  uint32_t signal_silent_time_in_ms_;
+  int signal_silent_time_in_ms_;
   std::vector<CryptoParams> offer_params_;
   std::unique_ptr<SrtpSession> send_session_;
   std::unique_ptr<SrtpSession> recv_session_;
@@ -207,7 +208,7 @@ class SrtpSession {
   bool GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len);
 
   // Update the silent threshold (in ms) for signaling errors.
-  void set_signal_silent_time(uint32_t signal_silent_time_in_ms);
+  void set_signal_silent_time(int signal_silent_time_in_ms);
 
   // Calls srtp_shutdown if it's initialized.
   static void Terminate();
@@ -251,9 +252,9 @@ class SrtpStat {
   void AddUnprotectRtcpResult(int result);
 
   // Get silent time (in ms) for SRTP statistics handler.
-  uint32_t signal_silent_time() const { return signal_silent_time_; }
+  int signal_silent_time() const { return signal_silent_time_; }
   // Set silent time (in ms) for SRTP statistics handler.
-  void set_signal_silent_time(uint32_t signal_silent_time) {
+  void set_signal_silent_time(int signal_silent_time) {
     signal_silent_time_ = signal_silent_time;
   }
 
@@ -295,7 +296,7 @@ class SrtpStat {
     void Reset() {
       last_signal_time = 0;
     }
-    uint32_t last_signal_time;
+    int64_t last_signal_time;
   };
 
   // Inspect SRTP result and signal error if needed.
@@ -303,11 +304,11 @@ class SrtpStat {
 
   std::map<FailureKey, FailureStat> failures_;
   // Threshold in ms to silent the signaling errors.
-  uint32_t signal_silent_time_;
+  int signal_silent_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpStat);
 };
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_SRTPFILTER_H_
+#endif  // WEBRTC_PC_SRTPFILTER_H_
diff --git a/chromium/third_party/webrtc/pc/srtpfilter_unittest.cc b/chromium/third_party/webrtc/pc/srtpfilter_unittest.cc
index 2eee35c2023..cc5b3e5fb3c 100644
--- a/chromium/third_party/webrtc/pc/srtpfilter_unittest.cc
+++ b/chromium/third_party/webrtc/pc/srtpfilter_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "webrtc/base/byteorder.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/cryptoparams.h"
diff --git a/chromium/third_party/webrtc/pc/voicechannel.h b/chromium/third_party/webrtc/pc/voicechannel.h
index 9b6f16eaba3..78524ab555c 100644
--- a/chromium/third_party/webrtc/pc/voicechannel.h
+++ b/chromium/third_party/webrtc/pc/voicechannel.h
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef _VOICECHANNEL_H_
-#define _VOICECHANNEL_H_
+#ifndef WEBRTC_PC_VOICECHANNEL_H_
+#define WEBRTC_PC_VOICECHANNEL_H_
 
 #include "webrtc/pc/channel.h"
 
-#endif // _VOICECHANNEL_H_
+#endif  // WEBRTC_PC_VOICECHANNEL_H_
diff --git a/chromium/third_party/webrtc/pc/yuvscaler_unittest.cc b/chromium/third_party/webrtc/pc/yuvscaler_unittest.cc
index 9e0f8ed2c0c..a33d49523d7 100644
--- a/chromium/third_party/webrtc/pc/yuvscaler_unittest.cc
+++ b/chromium/third_party/webrtc/pc/yuvscaler_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <sstream>
 
 #include "libyuv/cpu_id.h"
diff --git a/chromium/third_party/webrtc/rtc_unittests_apk.isolate b/chromium/third_party/webrtc/rtc_unittests_apk.isolate
new file mode 100644
index 00000000000..14fd20a81d2
--- /dev/null
+++ b/chromium/third_party/webrtc/rtc_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../build/android/android.isolate',
+    'rtc_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_rtc_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../build/config/',
+      '../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/rtc_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_rtc_unittests',
+      'rtc_unittests.isolate',
+    ]
+  },
+}
diff --git a/chromium/third_party/webrtc/sdk/BUILD.gn b/chromium/third_party/webrtc/sdk/BUILD.gn
new file mode 100644
index 00000000000..91a4c8828ed
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/BUILD.gn
@@ -0,0 +1,155 @@
+# Copyright 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+import("//build_overrides/webrtc.gni")
+
+if (is_ios) {
+  source_set("rtc_sdk_common_objc") {
+    deps = [
+      "../base:rtc_base",
+    ]
+    cflags = [ "-fobjc-arc" ]
+    configs += [ "..:common_config" ]
+    public_configs = [ "..:common_inherited_config" ]
+    include_dirs = [
+      "objc/Framework/Classes",
+      "objc/Framework/Headers",
+    ]
+    sources = [
+      "objc/Framework/Classes/NSString+StdString.h",
+      "objc/Framework/Classes/NSString+StdString.mm",
+      "objc/Framework/Classes/RTCCameraPreviewView.m",
+      "objc/Framework/Classes/RTCDispatcher.m",
+      "objc/Framework/Classes/RTCFieldTrials.mm",
+      "objc/Framework/Classes/RTCLogging.mm",
+      "objc/Framework/Classes/RTCSSLAdapter.mm",
+      "objc/Framework/Classes/RTCTracing.mm",
+      "objc/Framework/Classes/RTCUIApplication.h",
+      "objc/Framework/Classes/RTCUIApplication.mm",
+      "objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h",
+      "objc/Framework/Headers/WebRTC/RTCDispatcher.h",
+      "objc/Framework/Headers/WebRTC/RTCFieldTrials.h",
+      "objc/Framework/Headers/WebRTC/RTCLogging.h",
+      "objc/Framework/Headers/WebRTC/RTCMacros.h",
+      "objc/Framework/Headers/WebRTC/RTCSSLAdapter.h",
+      "objc/Framework/Headers/WebRTC/RTCTracing.h",
+    ]
+    if (!build_with_chromium) {
+      sources += [
+        "objc/Framework/Classes/RTCFileLogger.mm",
+        "objc/Framework/Headers/WebRTC/RTCFileLogger.h",
+      ]
+    }
+    libs = [ "AVFoundation.framework" ]
+  }
+
+  source_set("rtc_sdk_peerconnection_objc") {
+    deps = [
+      ":rtc_sdk_common_objc",
+      #"../../talk/libjingle:libjingle_peerconnection",
+    ]
+    cflags = [
+      "-fobjc-arc",
+      "-Wobjc-missing-property-synthesis",
+    ]
+    include_dirs = [
+      "objc/Framework/Classes",
+      "objc/Framework/Headers",
+    ]
+    sources = [
+      # Add these when there's a BUILD.gn for peer connection APIs
+      #"objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h",
+      #"objc/Framework/Classes/RTCAVFoundationVideoSource.mm",
+      #"objc/Framework/Classes/RTCAudioTrack+Private.h",
+      #"objc/Framework/Classes/RTCAudioTrack.mm",
+      #"objc/Framework/Classes/RTCConfiguration+Private.h",
+      #"objc/Framework/Classes/RTCConfiguration.mm",
+      #"objc/Framework/Classes/RTCDataChannel+Private.h",
+      #"objc/Framework/Classes/RTCDataChannel.mm",
+      #"objc/Framework/Classes/RTCDataChannelConfiguration+Private.h",
+      #"objc/Framework/Classes/RTCDataChannelConfiguration.mm",
+      #"objc/Framework/Classes/RTCIceCandidate+Private.h",
+      #"objc/Framework/Classes/RTCIceCandidate.mm",
+      #"objc/Framework/Classes/RTCIceServer+Private.h",
+      #"objc/Framework/Classes/RTCIceServer.mm",
+      #"objc/Framework/Classes/RTCMediaConstraints+Private.h",
+      #"objc/Framework/Classes/RTCMediaConstraints.mm",
+      #"objc/Framework/Classes/RTCMediaStream+Private.h",
+      #"objc/Framework/Classes/RTCMediaStream.mm",
+      #"objc/Framework/Classes/RTCMediaStreamTrack+Private.h",
+      #"objc/Framework/Classes/RTCMediaStreamTrack.mm",
+      #"objc/Framework/Classes/RTCOpenGLVideoRenderer.mm",
+      #"objc/Framework/Classes/RTCPeerConnection+Private.h",
+      #"objc/Framework/Classes/RTCPeerConnection.mm",
+      #"objc/Framework/Classes/RTCPeerConnectionFactory+Private.h",
+      #"objc/Framework/Classes/RTCPeerConnectionFactory.mm",
+      #"objc/Framework/Classes/RTCRtpEncodingParameters+Private.h",
+      #"objc/Framework/Classes/RTCRtpEncodingParameters.mm",
+      #"objc/Framework/Classes/RTCRtpParameters+Private.h",
+      #"objc/Framework/Classes/RTCRtpParameters.mm",
+      #"objc/Framework/Classes/RTCRtpReceiver+Private.h",
+      #"objc/Framework/Classes/RTCRtpReceiver.mm",
+      #"objc/Framework/Classes/RTCRtpSender+Private.h",
+      #"objc/Framework/Classes/RTCRtpSender.mm",
+      #"objc/Framework/Classes/RTCSessionDescription+Private.h",
+      #"objc/Framework/Classes/RTCSessionDescription.mm",
+      #"objc/Framework/Classes/RTCStatsReport+Private.h",
+      #"objc/Framework/Classes/RTCStatsReport.mm",
+      #"objc/Framework/Classes/RTCVideoFrame+Private.h",
+      #"objc/Framework/Classes/RTCVideoFrame.mm",
+      #"objc/Framework/Classes/RTCVideoRendererAdapter+Private.h",
+      #"objc/Framework/Classes/RTCVideoRendererAdapter.h",
+      #"objc/Framework/Classes/RTCVideoRendererAdapter.mm",
+      #"objc/Framework/Classes/RTCVideoSource+Private.h",
+      #"objc/Framework/Classes/RTCVideoSource.mm",
+      #"objc/Framework/Classes/RTCVideoTrack+Private.h",
+      #"objc/Framework/Classes/RTCVideoTrack.mm",
+      #"objc/Framework/Classes/avfoundationvideocapturer.h",
+      #"objc/Framework/Classes/avfoundationvideocapturer.mm",
+      #"objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h",
+      #"objc/Framework/Headers/WebRTC/RTCAudioTrack.h",
+      #"objc/Framework/Headers/WebRTC/RTCConfiguration.h",
+      #"objc/Framework/Headers/WebRTC/RTCDataChannel.h",
+      #"objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h",
+      #"objc/Framework/Headers/WebRTC/RTCIceCandidate.h",
+      #"objc/Framework/Headers/WebRTC/RTCIceServer.h",
+      #"objc/Framework/Headers/WebRTC/RTCMediaConstraints.h",
+      #"objc/Framework/Headers/WebRTC/RTCMediaStream.h",
+      #"objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h",
+      #"objc/Framework/Headers/WebRTC/RTCOpenGLVideoRenderer.h",
+      #"objc/Framework/Headers/WebRTC/RTCPeerConnection.h",
+      #"objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h",
+      #"objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h",
+      #"objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h",
+      #"objc/Framework/Headers/WebRTC/RTCRtpParameters.h",
+      #"objc/Framework/Headers/WebRTC/RTCRtpReceiver.h",
+      #"objc/Framework/Headers/WebRTC/RTCRtpSender.h",
+      #"objc/Framework/Headers/WebRTC/RTCSessionDescription.h",
+      #"objc/Framework/Headers/WebRTC/RTCStatsReport.h",
+      #"objc/Framework/Headers/WebRTC/RTCVideoFrame.h",
+      #"objc/Framework/Headers/WebRTC/RTCVideoRenderer.h",
+      #"objc/Framework/Headers/WebRTC/RTCVideoSource.h",
+      #"objc/Framework/Headers/WebRTC/RTCVideoTrack.h",
+    ]
+
+    if (is_ios) {
+      sources += [
+        "objc/Framework/Classes/RTCEAGLVideoView.m",
+        "objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h",
+      ]
+    }
+
+    if (is_mac) {
+      sources += [
+        "objc/Framework/Classes/RTCNSGLVideoView.m",
+        "objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h",
+      ]
+    }
+  }
+}
diff --git a/chromium/third_party/webrtc/modules/video_render/DEPS b/chromium/third_party/webrtc/sdk/DEPS
index 58ae9fe714f..1bb7a2fe8c8 100644
--- a/chromium/third_party/webrtc/modules/video_render/DEPS
+++ b/chromium/third_party/webrtc/sdk/DEPS
@@ -1,5 +1,6 @@
 include_rules = [
-  "+webrtc/base",
-  "+webrtc/common_video",
+  "+WebRTC",
+  "+webrtc/api",
+  "+webrtc/media",
   "+webrtc/system_wrappers",
 ]
diff --git a/chromium/third_party/webrtc/api/objc/OWNERS b/chromium/third_party/webrtc/sdk/OWNERS
index cd06158b7fc..cd06158b7fc 100644
--- a/chromium/third_party/webrtc/api/objc/OWNERS
+++ b/chromium/third_party/webrtc/sdk/OWNERS
diff --git a/chromium/third_party/webrtc/base/objc/NSString+StdString.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/NSString+StdString.h
index 8bf6cc94be0..8bf6cc94be0 100644
--- a/chromium/third_party/webrtc/base/objc/NSString+StdString.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/NSString+StdString.h
diff --git a/chromium/third_party/webrtc/base/objc/NSString+StdString.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/NSString+StdString.mm
index 3210ff0b65d..3210ff0b65d 100644
--- a/chromium/third_party/webrtc/base/objc/NSString+StdString.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/NSString+StdString.mm
diff --git a/chromium/third_party/webrtc/api/objc/RTCAVFoundationVideoSource+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h
index 067e5067c05..7a4de08240b 100644
--- a/chromium/third_party/webrtc/api/objc/RTCAVFoundationVideoSource+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h
@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCAVFoundationVideoSource.h"
+#import "WebRTC/RTCAVFoundationVideoSource.h"
 
-#include "webrtc/api/objc/avfoundationvideocapturer.h"
+#include "avfoundationvideocapturer.h"
 
 NS_ASSUME_NONNULL_BEGIN
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCAVFoundationVideoSource.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAVFoundationVideoSource.mm
index 1005c7dfa12..528e8cb6244 100644
--- a/chromium/third_party/webrtc/api/objc/RTCAVFoundationVideoSource.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAVFoundationVideoSource.mm
@@ -10,20 +10,25 @@
 
 #import "RTCAVFoundationVideoSource+Private.h"
 
-#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
-#import "webrtc/api/objc/RTCPeerConnectionFactory+Private.h"
-#import "webrtc/api/objc/RTCVideoSource+Private.h"
+#import "RTCMediaConstraints+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoSource+Private.h"
 
-@implementation RTCAVFoundationVideoSource
+@implementation RTCAVFoundationVideoSource {
+  webrtc::AVFoundationVideoCapturer *_capturer;
+}
 
 - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
                     constraints:(RTCMediaConstraints *)constraints {
   NSParameterAssert(factory);
-  rtc::scoped_ptr<webrtc::AVFoundationVideoCapturer> capturer;
-  capturer.reset(new webrtc::AVFoundationVideoCapturer());
+  // We pass ownership of the capturer to the source, but since we own
+  // the source, it should be ok to keep a raw pointer to the
+  // capturer.
+  _capturer = new webrtc::AVFoundationVideoCapturer();
   rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
       factory.nativeFactory->CreateVideoSource(
-          capturer.release(), constraints.nativeConstraints.get());
+          _capturer, constraints.nativeConstraints.get());
+
   return [super initWithNativeVideoSource:source];
 }
 
@@ -44,12 +49,7 @@
 }
 
 - (webrtc::AVFoundationVideoCapturer *)capturer {
-  cricket::VideoCapturer *capturer = self.nativeVideoSource->GetVideoCapturer();
-  // This should be safe because no one should have changed the underlying video
-  // source.
-  webrtc::AVFoundationVideoCapturer *foundationCapturer =
-      static_cast<webrtc::AVFoundationVideoCapturer *>(capturer);
-  return foundationCapturer;
+  return _capturer;
 }
 
 @end
diff --git a/chromium/third_party/webrtc/api/objc/RTCAudioTrack+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAudioTrack+Private.h
index ea86b1feaa7..cb5f1865aee 100644
--- a/chromium/third_party/webrtc/api/objc/RTCAudioTrack+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAudioTrack+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCAudioTrack.h"
+#import "WebRTC/RTCAudioTrack.h"
 
 #include "webrtc/api/mediastreaminterface.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCAudioTrack.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAudioTrack.mm
index 158d1b362b8..42542b8b8f0 100644
--- a/chromium/third_party/webrtc/api/objc/RTCAudioTrack.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCAudioTrack.mm
@@ -8,12 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCAudioTrack.h"
+#import "RTCAudioTrack+Private.h"
 
-#import "webrtc/api/objc/RTCAudioTrack+Private.h"
-#import "webrtc/api/objc/RTCMediaStreamTrack+Private.h"
-#import "webrtc/api/objc/RTCPeerConnectionFactory+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
 
 @implementation RTCAudioTrack
 
diff --git a/chromium/third_party/webrtc/base/objc/RTCCameraPreviewView.m b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCCameraPreviewView.m
index 5a57483676f..659973ff180 100644
--- a/chromium/third_party/webrtc/base/objc/RTCCameraPreviewView.m
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCCameraPreviewView.m
@@ -8,15 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#if !defined(__has_feature) || !__has_feature(objc_arc)
-#error "This file requires ARC support."
-#endif
-
-#import "webrtc/base/objc/RTCCameraPreviewView.h"
+#import "WebRTC/RTCCameraPreviewView.h"
 
 #import <AVFoundation/AVFoundation.h>
 
-#import "webrtc/base/objc/RTCDispatcher.h"
+#import "RTCDispatcher+Private.h"
 
 @implementation RTCCameraPreviewView
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCConfiguration+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCConfiguration+Private.h
index 001dac60166..7f90b35246b 100644
--- a/chromium/third_party/webrtc/api/objc/RTCConfiguration+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCConfiguration+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCConfiguration.h"
+#import "WebRTC/RTCConfiguration.h"
 
 #include "webrtc/api/peerconnectioninterface.h"
 
@@ -16,13 +16,6 @@ NS_ASSUME_NONNULL_BEGIN
 
 @interface RTCConfiguration ()
 
-/**
- * RTCConfiguration struct representation of this RTCConfiguration. This is
- * needed to pass to the underlying C++ APIs.
- */
-@property(nonatomic, readonly)
-    webrtc::PeerConnectionInterface::RTCConfiguration nativeConfiguration;
-
 + (webrtc::PeerConnectionInterface::IceTransportsType)
     nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy;
 
@@ -55,6 +48,13 @@ NS_ASSUME_NONNULL_BEGIN
 
 + (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
 
+/**
+ * RTCConfiguration struct representation of this RTCConfiguration. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+- (webrtc::PeerConnectionInterface::RTCConfiguration *)
+    createNativeConfiguration;
+
 @end
 
 NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/api/objc/RTCConfiguration.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCConfiguration.mm
index 2d44d015675..0a63f69fd98 100644
--- a/chromium/third_party/webrtc/api/objc/RTCConfiguration.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCConfiguration.mm
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCConfiguration.h"
+#import "RTCConfiguration+Private.h"
 
-#include "webrtc/base/sslidentity.h"
+#include <memory>
+
+#import "RTCIceServer+Private.h"
+#import "WebRTC/RTCLogging.h"
 
-#import "webrtc/api/objc/RTCConfiguration+Private.h"
-#import "webrtc/api/objc/RTCIceServer+Private.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#include "webrtc/base/rtccertificategenerator.h"
+#include "webrtc/base/sslidentity.h"
 
 @implementation RTCConfiguration
 
@@ -23,6 +25,7 @@
 @synthesize bundlePolicy = _bundlePolicy;
 @synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
 @synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
+@synthesize continualGatheringPolicy = _continualGatheringPolicy;
 @synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
 @synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
 @synthesize iceBackupCandidatePairPingInterval =
@@ -42,6 +45,10 @@
         [[self class] rtcpMuxPolicyForNativePolicy:config.rtcp_mux_policy];
     _tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
         config.tcp_candidate_policy];
+    webrtc::PeerConnectionInterface::ContinualGatheringPolicy nativePolicy =
+        config.continual_gathering_policy;
+    _continualGatheringPolicy =
+        [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
     _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
     _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
     _iceBackupCandidatePairPingInterval =
@@ -53,12 +60,14 @@
 
 - (NSString *)description {
   return [NSString stringWithFormat:
-      @"RTCConfiguration: {\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n}\n",
+      @"RTCConfiguration: {\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n}\n",
       _iceServers,
       [[self class] stringForTransportPolicy:_iceTransportPolicy],
       [[self class] stringForBundlePolicy:_bundlePolicy],
       [[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
       [[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
+      [[self class]
+          stringForContinualGatheringPolicy:_continualGatheringPolicy],
       _audioJitterBufferMaxPackets,
       _iceConnectionReceivingTimeout,
       _iceBackupCandidatePairPingInterval];
@@ -66,37 +75,44 @@
 
 #pragma mark - Private
 
-- (webrtc::PeerConnectionInterface::RTCConfiguration)nativeConfiguration {
-  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig;
+- (webrtc::PeerConnectionInterface::RTCConfiguration *)
+    createNativeConfiguration {
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
+      nativeConfig(new webrtc::PeerConnectionInterface::RTCConfiguration());
 
   for (RTCIceServer *iceServer in _iceServers) {
-    nativeConfig.servers.push_back(iceServer.nativeServer);
+    nativeConfig->servers.push_back(iceServer.nativeServer);
   }
-  nativeConfig.type =
+  nativeConfig->type =
       [[self class] nativeTransportsTypeForTransportPolicy:_iceTransportPolicy];
-  nativeConfig.bundle_policy =
+  nativeConfig->bundle_policy =
       [[self class] nativeBundlePolicyForPolicy:_bundlePolicy];
-  nativeConfig.rtcp_mux_policy =
+  nativeConfig->rtcp_mux_policy =
       [[self class] nativeRtcpMuxPolicyForPolicy:_rtcpMuxPolicy];
-  nativeConfig.tcp_candidate_policy =
+  nativeConfig->tcp_candidate_policy =
       [[self class] nativeTcpCandidatePolicyForPolicy:_tcpCandidatePolicy];
-  nativeConfig.audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
-  nativeConfig.ice_connection_receiving_timeout =
+  nativeConfig->continual_gathering_policy = [[self class]
+      nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
+  nativeConfig->audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
+  nativeConfig->ice_connection_receiving_timeout =
       _iceConnectionReceivingTimeout;
-  nativeConfig.ice_backup_candidate_pair_ping_interval =
+  nativeConfig->ice_backup_candidate_pair_ping_interval =
       _iceBackupCandidatePairPingInterval;
-  if (_keyType == RTCEncryptionKeyTypeECDSA) {
-    rtc::scoped_ptr<rtc::SSLIdentity> identity(
-        rtc::SSLIdentity::Generate(webrtc::kIdentityName, rtc::KT_ECDSA));
-    if (identity) {
-      nativeConfig.certificates.push_back(
-          rtc::RTCCertificate::Create(std::move(identity)));
-    } else {
-      RTCLogWarning(@"Failed to generate ECDSA identity. RSA will be used.");
+  rtc::KeyType keyType =
+      [[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
+  // Generate non-default certificate.
+  if (keyType != rtc::KT_DEFAULT) {
+    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
+        rtc::RTCCertificateGenerator::GenerateCertificate(
+            rtc::KeyParams(keyType), rtc::Optional<uint64_t>());
+    if (!certificate) {
+      RTCLogError(@"Failed to generate certificate.");
+      return nullptr;
     }
+    nativeConfig->certificates.push_back(certificate);
   }
 
-  return nativeConfig;
+  return nativeConfig.release();
 }
 
 + (webrtc::PeerConnectionInterface::IceTransportsType)
@@ -214,6 +230,16 @@
   }
 }
 
++ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:
+    (RTCEncryptionKeyType)keyType {
+  switch (keyType) {
+    case RTCEncryptionKeyTypeRSA:
+      return rtc::KT_RSA;
+    case RTCEncryptionKeyTypeECDSA:
+      return rtc::KT_ECDSA;
+  }
+}
+
 + (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
     (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy {
   switch (nativePolicy) {
@@ -233,4 +259,35 @@
   }
 }
 
++ (webrtc::PeerConnectionInterface::ContinualGatheringPolicy)
+    nativeContinualGatheringPolicyForPolicy:
+        (RTCContinualGatheringPolicy)policy {
+  switch (policy) {
+    case RTCContinualGatheringPolicyGatherOnce:
+      return webrtc::PeerConnectionInterface::GATHER_ONCE;
+    case RTCContinualGatheringPolicyGatherContinually:
+      return webrtc::PeerConnectionInterface::GATHER_CONTINUALLY;
+  }
+}
+
++ (RTCContinualGatheringPolicy)continualGatheringPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::ContinualGatheringPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::GATHER_ONCE:
+      return RTCContinualGatheringPolicyGatherOnce;
+    case webrtc::PeerConnectionInterface::GATHER_CONTINUALLY:
+      return RTCContinualGatheringPolicyGatherContinually;
+  }
+}
+
++ (NSString *)stringForContinualGatheringPolicy:
+    (RTCContinualGatheringPolicy)policy {
+  switch (policy) {
+    case RTCContinualGatheringPolicyGatherOnce:
+      return @"GATHER_ONCE";
+    case RTCContinualGatheringPolicyGatherContinually:
+      return @"GATHER_CONTINUALLY";
+  }
+}
+
 @end
diff --git a/chromium/third_party/webrtc/api/objc/RTCDataChannel+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannel+Private.h
index 179192c83f4..82e132fa2e0 100644
--- a/chromium/third_party/webrtc/api/objc/RTCDataChannel+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannel+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCDataChannel.h"
+#import "WebRTC/RTCDataChannel.h"
 
 #include "webrtc/api/datachannelinterface.h"
 #include "webrtc/base/scoped_ref_ptr.h"
diff --git a/chromium/third_party/webrtc/api/objc/RTCDataChannel.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannel.mm
index a2c2e182ed1..706e43e2ed5 100644
--- a/chromium/third_party/webrtc/api/objc/RTCDataChannel.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannel.mm
@@ -8,12 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCDataChannel.h"
+#import "RTCDataChannel+Private.h"
 
-#import "webrtc/api/objc/RTCDataChannel+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
 
 namespace webrtc {
 
@@ -47,7 +46,7 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 
 
 @implementation RTCDataBuffer {
-  rtc::scoped_ptr<webrtc::DataBuffer> _dataBuffer;
+  std::unique_ptr<webrtc::DataBuffer> _dataBuffer;
 }
 
 - (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary {
@@ -87,7 +86,7 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 
 @implementation RTCDataChannel {
   rtc::scoped_refptr<webrtc::DataChannelInterface> _nativeDataChannel;
-  rtc::scoped_ptr<webrtc::DataChannelDelegateAdapter> _observer;
+  std::unique_ptr<webrtc::DataChannelDelegateAdapter> _observer;
   BOOL _isObserverRegistered;
 }
 
@@ -96,7 +95,7 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
 - (void)dealloc {
   // Handles unregistering the observer properly. We need to do this because
   // there may still be other references to the underlying data channel.
-  self.delegate = nil;
+  _nativeDataChannel->UnregisterObserver();
 }
 
 - (NSString *)label {
@@ -148,21 +147,6 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
   return _nativeDataChannel->buffered_amount();
 }
 
-- (void)setDelegate:(id<RTCDataChannelDelegate>)delegate {
-  if (_delegate == delegate) {
-    return;
-  }
-  if (_isObserverRegistered) {
-    _nativeDataChannel->UnregisterObserver();
-    _isObserverRegistered = NO;
-  }
-  _delegate = delegate;
-  if (_delegate) {
-    _nativeDataChannel->RegisterObserver(_observer.get());
-    _isObserverRegistered = YES;
-  }
-}
-
 - (void)close {
   _nativeDataChannel->Close();
 }
@@ -187,6 +171,7 @@ class DataChannelDelegateAdapter : public DataChannelObserver {
   if (self = [super init]) {
     _nativeDataChannel = nativeDataChannel;
     _observer.reset(new webrtc::DataChannelDelegateAdapter(self));
+    _nativeDataChannel->RegisterObserver(_observer.get());
   }
   return self;
 }
diff --git a/chromium/third_party/webrtc/api/objc/RTCDataChannelConfiguration+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannelConfiguration+Private.h
index 13478e78e8d..e9ea74705d4 100644
--- a/chromium/third_party/webrtc/api/objc/RTCDataChannelConfiguration+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannelConfiguration+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCDataChannelConfiguration.h"
+#import "WebRTC/RTCDataChannelConfiguration.h"
 
 #include "webrtc/api/datachannelinterface.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCDataChannelConfiguration.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannelConfiguration.mm
index 4a1dcb061c8..89c56de7863 100644
--- a/chromium/third_party/webrtc/api/objc/RTCDataChannelConfiguration.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDataChannelConfiguration.mm
@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCDataChannelConfiguration.h"
+#import "RTCDataChannelConfiguration+Private.h"
 
-#import "webrtc/api/objc/RTCDataChannelConfiguration+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
 
 @implementation RTCDataChannelConfiguration
 
diff --git a/chromium/third_party/webrtc/base/objc/RTCDispatcher+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDispatcher+Private.h
index 2ef6cf1ef35..3c114e553df 100644
--- a/chromium/third_party/webrtc/base/objc/RTCDispatcher+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDispatcher+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/base/objc/RTCDispatcher.h"
+#import "WebRTC/RTCDispatcher.h"
 
 @interface RTCDispatcher ()
 
diff --git a/chromium/third_party/webrtc/base/objc/RTCDispatcher.m b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDispatcher.m
index dc2a990dfc3..94176ac6d81 100644
--- a/chromium/third_party/webrtc/base/objc/RTCDispatcher.m
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCDispatcher.m
@@ -8,8 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCDispatcher.h"
-
 #import "RTCDispatcher+Private.h"
 
 static dispatch_queue_t kAudioSessionQueue = nil;
diff --git a/chromium/third_party/webrtc/api/objc/RTCEAGLVideoView.m b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCEAGLVideoView.m
index 58fd108c4c1..d215265d3b0 100644
--- a/chromium/third_party/webrtc/api/objc/RTCEAGLVideoView.m
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCEAGLVideoView.m
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCEAGLVideoView.h"
+#import "WebRTC/RTCEAGLVideoView.h"
 
 #import <GLKit/GLKit.h>
 
-#import "webrtc/api/objc/RTCOpenGLVideoRenderer.h"
-#import "webrtc/api/objc/RTCVideoFrame.h"
+#import "RTCOpenGLVideoRenderer.h"
+#import "WebRTC//RTCVideoFrame.h"
 
 // RTCDisplayLinkTimer wraps a CADisplayLink and is set to fire every two screen
 // refreshes, which should be 30fps. We wrap the display link in order to avoid
diff --git a/chromium/third_party/webrtc/base/objc/RTCFieldTrials.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCFieldTrials.mm
index 3072f394ee1..38c293f0ecd 100644
--- a/chromium/third_party/webrtc/base/objc/RTCFieldTrials.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCFieldTrials.mm
@@ -8,12 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/base/objc/RTCFieldTrials.h"
+#import "WebRTC/RTCFieldTrials.h"
 
 #include <memory>
-#include "webrtc/system_wrappers/include/field_trial_default.h"
 
-#import "webrtc/base/objc/RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/system_wrappers/include/field_trial_default.h"
 
 static NSString * const kRTCEnableSendSideBweString =
     @"WebRTC-SendSideBwe/Enabled/";
diff --git a/chromium/third_party/webrtc/base/objc/RTCFileLogger.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCFileLogger.mm
index bf61794bd14..c1fbd747c3c 100644
--- a/chromium/third_party/webrtc/base/objc/RTCFileLogger.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCFileLogger.mm
@@ -8,13 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCFileLogger.h"
+#import "WebRTC/RTCFileLogger.h"
+
+#include <memory>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/filerotatingstream.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/logsinks.h"
-#include "webrtc/base/scoped_ptr.h"
 
 NSString *const kDefaultLogDirName = @"webrtc_logs";
 NSUInteger const kDefaultMaxFileSize = 10 * 1024 * 1024; // 10MB.
@@ -24,7 +25,7 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
   BOOL _hasStarted;
   NSString *_dirPath;
   NSUInteger _maxFileSize;
-  rtc::scoped_ptr<rtc::FileRotatingLogSink> _logSink;
+  std::unique_ptr<rtc::FileRotatingLogSink> _logSink;
 }
 
 @synthesize severity = _severity;
@@ -129,7 +130,7 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
     return nil;
   }
   NSMutableData* logData = [NSMutableData data];
-  rtc::scoped_ptr<rtc::FileRotatingStream> stream;
+  std::unique_ptr<rtc::FileRotatingStream> stream;
   switch(_rotationType) {
     case RTCFileLoggerTypeApp:
       stream.reset(
@@ -150,7 +151,7 @@ const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
   size_t read = 0;
   // Allocate memory using malloc so we can pass it direcly to NSData without
   // copying.
-  rtc::scoped_ptr<uint8_t[]> buffer(static_cast<uint8_t*>(malloc(bufferSize)));
+  std::unique_ptr<uint8_t[]> buffer(static_cast<uint8_t*>(malloc(bufferSize)));
   stream->ReadAll(buffer.get(), bufferSize, &read, nullptr);
   logData = [[NSMutableData alloc] initWithBytesNoCopy:buffer.release()
                                                 length:read];
diff --git a/chromium/third_party/webrtc/api/objc/RTCIceCandidate+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h
index f98e04ba1c2..b00c8da3a3e 100644
--- a/chromium/third_party/webrtc/api/objc/RTCIceCandidate+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h
@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCIceCandidate.h"
+#import "WebRTC/RTCIceCandidate.h"
+
+#include <memory>
 
 #include "webrtc/api/jsep.h"
-#include "webrtc/base/scoped_ptr.h"
 
 NS_ASSUME_NONNULL_BEGIN
 
@@ -22,7 +23,7 @@ NS_ASSUME_NONNULL_BEGIN
  * object. This is needed to pass to the underlying C++ APIs.
  */
 @property(nonatomic, readonly)
-    rtc::scoped_ptr<webrtc::IceCandidateInterface> nativeCandidate;
+    std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
 
 /**
  * Initialize an RTCIceCandidate from a native IceCandidateInterface. No
diff --git a/chromium/third_party/webrtc/api/objc/RTCIceCandidate.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate.mm
index ecc128f8b29..193403d11d8 100644
--- a/chromium/third_party/webrtc/api/objc/RTCIceCandidate.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate.mm
@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCIceCandidate.h"
+#import "RTCIceCandidate+Private.h"
 
-#import "webrtc/api/objc/RTCIceCandidate+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#include <memory>
+
+#import "NSString+StdString.h"
+#import "WebRTC/RTCLogging.h"
 
 @implementation RTCIceCandidate
 
@@ -52,7 +53,7 @@
                     sdpMid:[NSString stringForStdString:candidate->sdp_mid()]];
 }
 
-- (rtc::scoped_ptr<webrtc::IceCandidateInterface>)nativeCandidate {
+- (std::unique_ptr<webrtc::IceCandidateInterface>)nativeCandidate {
   webrtc::SdpParseError error;
 
   webrtc::IceCandidateInterface *candidate = webrtc::CreateIceCandidate(
@@ -64,7 +65,7 @@
            error.line.c_str());
   }
 
-  return rtc::scoped_ptr<webrtc::IceCandidateInterface>(candidate);
+  return std::unique_ptr<webrtc::IceCandidateInterface>(candidate);
 }
 
 @end
diff --git a/chromium/third_party/webrtc/api/objc/RTCIceServer+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceServer+Private.h
index 8098bee8f8b..7efeda412fc 100644
--- a/chromium/third_party/webrtc/api/objc/RTCIceServer+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceServer+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCIceServer.h"
+#import "WebRTC/RTCIceServer.h"
 
 #include "webrtc/api/peerconnectioninterface.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCIceServer.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceServer.mm
index 95c380e0713..41084b93571 100644
--- a/chromium/third_party/webrtc/api/objc/RTCIceServer.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCIceServer.mm
@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCIceServer.h"
+#import "RTCIceServer+Private.h"
 
-#import "webrtc/api/objc/RTCIceServer+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
 
 @implementation RTCIceServer
 
diff --git a/chromium/third_party/webrtc/base/objc/RTCLogging.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm
index 598d52de664..ef62d1f6697 100644
--- a/chromium/third_party/webrtc/base/objc/RTCLogging.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCLogging.h"
+#import "WebRTC/RTCLogging.h"
 
 #include "webrtc/base/logging.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaConstraints+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h
index fa582ecae59..606a132da64 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaConstraints+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h
@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaConstraints.h"
+#import "WebRTC/RTCMediaConstraints.h"
+
+#include <memory>
 
 #include "webrtc/api/mediaconstraintsinterface.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -41,7 +42,7 @@ NS_ASSUME_NONNULL_BEGIN
  * A MediaConstraints representation of this RTCMediaConstraints object. This is
  * needed to pass to the underlying C++ APIs.
  */
-- (rtc::scoped_ptr<webrtc::MediaConstraints>)nativeConstraints;
+- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints;
 
 /** Return a native Constraints object representing these constraints */
 + (webrtc::MediaConstraintsInterface::Constraints)
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaConstraints.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints.mm
index bf50668e1d2..11be2ec026c 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaConstraints.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints.mm
@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaConstraints.h"
+#import "RTCMediaConstraints+Private.h"
 
-#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+
+#include <memory>
 
 namespace webrtc {
 
@@ -63,7 +64,7 @@ MediaConstraints::GetOptional() const {
 
 #pragma mark - Private
 
-- (rtc::scoped_ptr<webrtc::MediaConstraints>)nativeConstraints {
+- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints {
   webrtc::MediaConstraintsInterface::Constraints mandatory =
       [[self class] nativeConstraintsForConstraints:_mandatory];
   webrtc::MediaConstraintsInterface::Constraints optional =
@@ -71,7 +72,7 @@ MediaConstraints::GetOptional() const {
 
   webrtc::MediaConstraints *nativeConstraints =
       new webrtc::MediaConstraints(mandatory, optional);
-  return rtc::scoped_ptr<webrtc::MediaConstraints>(nativeConstraints);
+  return std::unique_ptr<webrtc::MediaConstraints>(nativeConstraints);
 }
 
 + (webrtc::MediaConstraintsInterface::Constraints)
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaStream+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStream+Private.h
index b03b091c11f..7f8707228d3 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaStream+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStream+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaStream.h"
+#import "WebRTC/RTCMediaStream.h"
 
 #include "webrtc/api/mediastreaminterface.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaStream.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStream.mm
index 666022301a1..0737dee8a93 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaStream.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStream.mm
@@ -8,16 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaStream.h"
+#import "RTCMediaStream+Private.h"
 
 #include <vector>
 
-#import "webrtc/api/objc/RTCAudioTrack+Private.h"
-#import "webrtc/api/objc/RTCMediaStream+Private.h"
-#import "webrtc/api/objc/RTCMediaStreamTrack+Private.h"
-#import "webrtc/api/objc/RTCPeerConnectionFactory+Private.h"
-#import "webrtc/api/objc/RTCVideoTrack+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoTrack+Private.h"
 
 @implementation RTCMediaStream {
   NSMutableArray *_audioTracks;
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaStreamTrack+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h
index 155e31228b3..d5261266b89 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaStreamTrack+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h
@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaStreamTrack.h"
+#import "WebRTC/RTCMediaStreamTrack.h"
 
 #include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/base/scoped_ptr.h"
 
 typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) {
   RTCMediaStreamTrackTypeAudio,
@@ -37,6 +36,11 @@ NS_ASSUME_NONNULL_BEGIN
                                type:(RTCMediaStreamTrackType)type
     NS_DESIGNATED_INITIALIZER;
 
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
+
+- (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track;
+
 + (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
     (RTCMediaStreamTrackState)state;
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaStreamTrack.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack.mm
index 25979b38d9a..208550f6c86 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaStreamTrack.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack.mm
@@ -8,10 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaStreamTrack.h"
+#import "RTCMediaStreamTrack+Private.h"
 
-#import "webrtc/api/objc/RTCMediaStreamTrack+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+
+NSString * const kRTCMediaStreamTrackKindAudio =
+    @(webrtc::MediaStreamTrackInterface::kAudioKind);
+NSString * const kRTCMediaStreamTrackKindVideo =
+    @(webrtc::MediaStreamTrackInterface::kVideoKind);
 
 @implementation RTCMediaStreamTrack {
   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _nativeTrack;
@@ -47,6 +51,20 @@
                                     readyState];
 }
 
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  return [self isEqualToTrack:(RTCMediaStreamTrack *)object];
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeTrack.get();
+}
+
 #pragma mark - Private
 
 - (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
@@ -64,6 +82,29 @@
   return self;
 }
 
+- (instancetype)initWithNativeTrack:
+    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+  NSParameterAssert(nativeTrack);
+  if (nativeTrack->kind() ==
+      std::string(webrtc::MediaStreamTrackInterface::kAudioKind)) {
+    return [self initWithNativeTrack:nativeTrack
+                                type:RTCMediaStreamTrackTypeAudio];
+  }
+  if (nativeTrack->kind() ==
+      std::string(webrtc::MediaStreamTrackInterface::kVideoKind)) {
+    return [self initWithNativeTrack:nativeTrack
+                                type:RTCMediaStreamTrackTypeVideo];
+  }
+  return nil;
+}
+
+- (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track {
+  if (!track) {
+    return NO;
+  }
+  return _nativeTrack == track.nativeTrack;
+}
+
 + (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
     (RTCMediaStreamTrackState)state {
   switch (state) {
diff --git a/chromium/third_party/webrtc/api/objc/RTCNSGLVideoView.m b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m
index 063e6f1330e..415efe82282 100644
--- a/chromium/third_party/webrtc/api/objc/RTCNSGLVideoView.m
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m
@@ -8,12 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCNSGLVideoView.h"
+#if !TARGET_OS_IPHONE
+
+#import "WebRTC/RTCNSGLVideoView.h"
 
 #import <CoreVideo/CVDisplayLink.h>
 #import <OpenGL/gl3.h>
-#import "RTCVideoFrame.h"
+
 #import "RTCOpenGLVideoRenderer.h"
+#import "WebRTC/RTCVideoFrame.h"
 
 @interface RTCNSGLVideoView ()
 // |videoFrame| is set when we receive a frame from a worker thread and is read
@@ -139,3 +142,5 @@ static CVReturn OnDisplayLinkFired(CVDisplayLinkRef displayLink,
 }
 
 @end
+
+#endif  // !TARGET_OS_IPHONE
diff --git a/chromium/third_party/webrtc/api/objc/RTCOpenGLVideoRenderer.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.h
index 729839c6a30..7041861014a 100644
--- a/chromium/third_party/webrtc/api/objc/RTCOpenGLVideoRenderer.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.h
@@ -15,6 +15,8 @@
 #import <AppKit/NSOpenGL.h>
 #endif
 
+#import "WebRTC/RTCMacros.h"
+
 NS_ASSUME_NONNULL_BEGIN
 
 @class RTCVideoFrame;
@@ -23,6 +25,7 @@ NS_ASSUME_NONNULL_BEGIN
 // the currently bound framebuffer. Supports OpenGL 3.2 and OpenGLES 2.0. OpenGL
 // framebuffer creation and management should be handled elsewhere using the
 // same context used to initialize this class.
+RTC_EXPORT
 @interface RTCOpenGLVideoRenderer : NSObject
 
 // The last successfully drawn frame. Used to avoid drawing frames unnecessarily
diff --git a/chromium/third_party/webrtc/api/objc/RTCOpenGLVideoRenderer.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.mm
index b4a7c7b13a0..7d7b416b888 100644
--- a/chromium/third_party/webrtc/api/objc/RTCOpenGLVideoRenderer.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.mm
@@ -10,17 +10,16 @@
 
 #import "RTCOpenGLVideoRenderer.h"
 
-#include <string.h>
-
-#include "webrtc/base/scoped_ptr.h"
-
 #if TARGET_OS_IPHONE
 #import <OpenGLES/ES3/gl.h>
 #else
 #import <OpenGL/gl3.h>
 #endif
+#include <string.h>
+#include <memory>
+
+#import "WebRTC/RTCVideoFrame.h"
 
-#import "RTCVideoFrame.h"
 
 // TODO(tkchin): check and log openGL errors. Methods here return BOOLs in
 // anticipation of that happening in the future.
@@ -163,7 +162,7 @@ static const GLsizei kNumTextures = 3 * kNumTextureSets;
   GLint _vSampler;
   // Used to create a non-padded plane for GPU upload when we receive padded
   // frames.
-  rtc::scoped_ptr<uint8_t[]> _planeBuffer;
+  std::unique_ptr<uint8_t[]> _planeBuffer;
 }
 
 @synthesize lastDrawnFrame = _lastDrawnFrame;
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnection+DataChannel.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection+DataChannel.mm
index cf646adda67..b3825d4e4eb 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnection+DataChannel.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection+DataChannel.mm
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCPeerConnection+Private.h"
+#import "RTCPeerConnection+Private.h"
 
-#import "webrtc/api/objc/RTCDataChannel+Private.h"
-#import "webrtc/api/objc/RTCDataChannelConfiguration+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCDataChannel+Private.h"
+#import "RTCDataChannelConfiguration+Private.h"
 
 @implementation RTCPeerConnection (DataChannel)
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnection+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection+Private.h
index 031631a37a9..cbae3607cf6 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnection+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCPeerConnection.h"
+#import "WebRTC/RTCPeerConnection.h"
 
 #include "webrtc/api/peerconnectioninterface.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnection+Stats.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection+Stats.mm
index 5032c84e54b..ccbd58f103d 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnection+Stats.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection+Stats.mm
@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCPeerConnection+Private.h"
+#import "RTCPeerConnection+Private.h"
 
-#include "webrtc/base/checks.h"
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCStatsReport+Private.h"
 
-#import "webrtc/api/objc/RTCMediaStreamTrack+Private.h"
-#import "webrtc/api/objc/RTCStatsReport+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnection.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm
index 50d05f1bc69..9a488fdf494 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnection.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm
@@ -8,22 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/api/objc/RTCPeerConnection.h"
+#import "RTCPeerConnection+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCConfiguration+Private.h"
+#import "RTCDataChannel+Private.h"
+#import "RTCIceCandidate+Private.h"
+#import "RTCMediaConstraints+Private.h"
+#import "RTCMediaStream+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCRtpReceiver+Private.h"
+#import "RTCRtpSender+Private.h"
+#import "RTCSessionDescription+Private.h"
+#import "RTCStatsReport+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include <memory>
 
 #include "webrtc/base/checks.h"
 
-#import "webrtc/api/objc/RTCPeerConnection+Private.h"
-#import "webrtc/api/objc/RTCConfiguration+Private.h"
-#import "webrtc/api/objc/RTCDataChannel+Private.h"
-#import "webrtc/api/objc/RTCIceCandidate+Private.h"
-#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
-#import "webrtc/api/objc/RTCMediaStream+Private.h"
-#import "webrtc/api/objc/RTCPeerConnectionFactory+Private.h"
-#import "webrtc/api/objc/RTCSessionDescription+Private.h"
-#import "webrtc/api/objc/RTCStatsReport+Private.h"
-#import "webrtc/base/objc/RTCLogging.h"
-#import "webrtc/base/objc/NSString+StdString.h"
-
 NSString * const kRTCPeerConnectionErrorDomain =
     @"org.webrtc.RTCPeerConnection";
 int const kRTCPeerConnnectionSessionDescriptionError = -1;
@@ -45,8 +48,8 @@ class CreateSessionDescriptionObserverAdapter
 
   void OnSuccess(SessionDescriptionInterface *desc) override {
     RTC_DCHECK(completion_handler_);
-    rtc::scoped_ptr<webrtc::SessionDescriptionInterface> description =
-        rtc::scoped_ptr<webrtc::SessionDescriptionInterface>(desc);
+    std::unique_ptr<webrtc::SessionDescriptionInterface> description =
+        std::unique_ptr<webrtc::SessionDescriptionInterface>(desc);
     RTCSessionDescription* session =
         [[RTCSessionDescription alloc] initWithNativeDescription:
             description.get()];
@@ -184,7 +187,7 @@ void PeerConnectionDelegateAdapter::OnIceCandidate(
 
 @implementation RTCPeerConnection {
   NSMutableArray *_localStreams;
-  rtc::scoped_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
+  std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection;
 }
 
@@ -195,14 +198,17 @@ void PeerConnectionDelegateAdapter::OnIceCandidate(
                     constraints:(RTCMediaConstraints *)constraints
                        delegate:(id<RTCPeerConnectionDelegate>)delegate {
   NSParameterAssert(factory);
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
+      [configuration createNativeConfiguration]);
+  if (!config) {
+    return nil;
+  }
   if (self = [super init]) {
     _observer.reset(new webrtc::PeerConnectionDelegateAdapter(self));
-    webrtc::PeerConnectionInterface::RTCConfiguration config =
-        configuration.nativeConfiguration;
-    rtc::scoped_ptr<webrtc::MediaConstraints> nativeConstraints =
+    std::unique_ptr<webrtc::MediaConstraints> nativeConstraints =
         constraints.nativeConstraints;
     _peerConnection =
-        factory.nativeFactory->CreatePeerConnection(config,
+        factory.nativeFactory->CreatePeerConnection(*config,
                                                     nativeConstraints.get(),
                                                     nullptr,
                                                     nullptr,
@@ -249,7 +255,12 @@ void PeerConnectionDelegateAdapter::OnIceCandidate(
 }
 
 - (BOOL)setConfiguration:(RTCConfiguration *)configuration {
-  return _peerConnection->SetConfiguration(configuration.nativeConfiguration);
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
+      [configuration createNativeConfiguration]);
+  if (!config) {
+    return NO;
+  }
+  return _peerConnection->SetConfiguration(*config);
 }
 
 - (void)close {
@@ -257,7 +268,7 @@ void PeerConnectionDelegateAdapter::OnIceCandidate(
 }
 
 - (void)addIceCandidate:(RTCIceCandidate *)candidate {
-  rtc::scoped_ptr<const webrtc::IceCandidateInterface> iceCandidate(
+  std::unique_ptr<const webrtc::IceCandidateInterface> iceCandidate(
       candidate.nativeCandidate);
   _peerConnection->AddIceCandidate(iceCandidate.get());
 }
@@ -311,6 +322,41 @@ void PeerConnectionDelegateAdapter::OnIceCandidate(
   _peerConnection->SetRemoteDescription(observer, sdp.nativeDescription);
 }
 
+- (RTCRtpSender *)senderWithKind:(NSString *)kind
+                        streamId:(NSString *)streamId {
+  std::string nativeKind = [NSString stdStringForString:kind];
+  std::string nativeStreamId = [NSString stdStringForString:streamId];
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
+      _peerConnection->CreateSender(nativeKind, nativeStreamId));
+  return nativeSender ?
+      [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender]
+      : nil;
+}
+
+- (NSArray<RTCRtpSender *> *)senders {
+  std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenders(
+      _peerConnection->GetSenders());
+  NSMutableArray *senders = [[NSMutableArray alloc] init];
+  for (const auto &nativeSender : nativeSenders) {
+    RTCRtpSender *sender =
+        [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender];
+    [senders addObject:sender];
+  }
+  return senders;
+}
+
+- (NSArray<RTCRtpReceiver *> *)receivers {
+  std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> nativeReceivers(
+      _peerConnection->GetReceivers());
+  NSMutableArray *receivers = [[NSMutableArray alloc] init];
+  for (const auto &nativeReceiver : nativeReceivers) {
+    RTCRtpReceiver *receiver =
+        [[RTCRtpReceiver alloc] initWithNativeRtpReceiver:nativeReceiver];
+    [receivers addObject:receiver];
+  }
+  return receivers;
+}
+
 #pragma mark - Private
 
 + (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnectionFactory+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory+Private.h
index 55a473b4cd1..a7e453c917f 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnectionFactory+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCPeerConnectionFactory.h"
+#import "WebRTC/RTCPeerConnectionFactory.h"
 
 #include "webrtc/api/peerconnectionfactory.h"
 #include "webrtc/base/scoped_ref_ptr.h"
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnectionFactory.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory.mm
index e2d60212310..2398ce5ddc4 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnectionFactory.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory.mm
@@ -8,43 +8,50 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCPeerConnectionFactory.h"
+#import "RTCPeerConnectionFactory+Private.h"
 
+#import "NSString+StdString.h"
 #if defined(WEBRTC_IOS)
-#import "webrtc/api/objc/RTCAVFoundationVideoSource+Private.h"
+#import "RTCAVFoundationVideoSource+Private.h"
 #endif
-#import "webrtc/api/objc/RTCAudioTrack+Private.h"
-#import "webrtc/api/objc/RTCMediaStream+Private.h"
-#import "webrtc/api/objc/RTCPeerConnection+Private.h"
-#import "webrtc/api/objc/RTCPeerConnectionFactory+Private.h"
-#import "webrtc/api/objc/RTCVideoSource+Private.h"
-#import "webrtc/api/objc/RTCVideoTrack+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaStream+Private.h"
+#import "RTCPeerConnection+Private.h"
+#import "RTCVideoSource+Private.h"
+#import "RTCVideoTrack+Private.h"
+
+#include <memory>
 
 @implementation RTCPeerConnectionFactory {
-  rtc::scoped_ptr<rtc::Thread> _signalingThread;
-  rtc::scoped_ptr<rtc::Thread> _workerThread;
+  std::unique_ptr<rtc::Thread> _networkThread;
+  std::unique_ptr<rtc::Thread> _workerThread;
+  std::unique_ptr<rtc::Thread> _signalingThread;
 }
 
 @synthesize nativeFactory = _nativeFactory;
 
 - (instancetype)init {
   if ((self = [super init])) {
-    _signalingThread.reset(new rtc::Thread());
-    BOOL result = _signalingThread->Start();
-    NSAssert(result, @"Failed to start signaling thread.");
-    _workerThread.reset(new rtc::Thread());
+    _networkThread = rtc::Thread::CreateWithSocketServer();
+    BOOL result = _networkThread->Start();
+    NSAssert(result, @"Failed to start network thread.");
+
+    _workerThread = rtc::Thread::Create();
     result = _workerThread->Start();
     NSAssert(result, @"Failed to start worker thread.");
 
+    _signalingThread = rtc::Thread::Create();
+    result = _signalingThread->Start();
+    NSAssert(result, @"Failed to start signaling thread.");
+
     _nativeFactory = webrtc::CreatePeerConnectionFactory(
-        _workerThread.get(), _signalingThread.get(), nullptr, nullptr, nullptr);
+        _networkThread.get(), _workerThread.get(), _signalingThread.get(),
+        nullptr, nullptr, nullptr);
     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
   }
   return self;
 }
 
-
 - (RTCAVFoundationVideoSource *)avFoundationVideoSourceWithConstraints:
     (nullable RTCMediaConstraints *)constraints {
 #if defined(WEBRTC_IOS)
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpCodecParameters+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpCodecParameters+Private.h
new file mode 100644
index 00000000000..fe33e9e9633
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpCodecParameters+Private.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "WebRTC/RTCRtpCodecParameters.h"
+
+#include "webrtc/api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpCodecParameters ()
+
+/** Returns the equivalent native RtpCodecParameters structure. */
+@property(nonatomic, readonly) webrtc::RtpCodecParameters nativeParameters;
+
+/** Initialize the object with a native RtpCodecParameters structure. */
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpCodecParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpCodecParameters.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpCodecParameters.mm
new file mode 100644
index 00000000000..77047694723
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpCodecParameters.mm
@@ -0,0 +1,65 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpCodecParameters+Private.h"
+
+#import "NSString+StdString.h"
+
+#include "webrtc/media/base/mediaconstants.h"
+
+const NSString * const kRTCRtxCodecMimeType = @(cricket::kRtxCodecName);
+const NSString * const kRTCRedCodecMimeType = @(cricket::kRedCodecName);
+const NSString * const kRTCUlpfecCodecMimeType = @(cricket::kUlpfecCodecName);
+const NSString * const kRTCOpusCodecMimeType = @(cricket::kOpusCodecName);
+const NSString * const kRTCIsacCodecMimeType = @(cricket::kIsacCodecName);
+const NSString * const kRTCL16CodecMimeType  = @(cricket::kL16CodecName);
+const NSString * const kRTCG722CodecMimeType = @(cricket::kG722CodecName);
+const NSString * const kRTCIlbcCodecMimeType = @(cricket::kIlbcCodecName);
+const NSString * const kRTCPcmuCodecMimeType = @(cricket::kPcmuCodecName);
+const NSString * const kRTCPcmaCodecMimeType = @(cricket::kPcmaCodecName);
+const NSString * const kRTCDtmfCodecMimeType = @(cricket::kDtmfCodecName);
+const NSString * const kRTCComfortNoiseCodecMimeType =
+    @(cricket::kComfortNoiseCodecName);
+const NSString * const kVp8CodecMimeType = @(cricket::kVp8CodecName);
+const NSString * const kVp9CodecMimeType = @(cricket::kVp9CodecName);
+const NSString * const kH264CodecMimeType = @(cricket::kH264CodecName);
+
+@implementation RTCRtpCodecParameters
+
+@synthesize payloadType = _payloadType;
+@synthesize mimeType = _mimeType;
+@synthesize clockRate = _clockRate;
+@synthesize channels = _channels;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpCodecParameters &)nativeParameters {
+  if (self = [self init]) {
+    _payloadType = nativeParameters.payload_type;
+    _mimeType = [NSString stringForStdString:nativeParameters.mime_type];
+    _clockRate = nativeParameters.clock_rate;
+    _channels = nativeParameters.channels;
+  }
+  return self;
+}
+
+- (webrtc::RtpCodecParameters)nativeParameters {
+  webrtc::RtpCodecParameters parameters;
+  parameters.payload_type = _payloadType;
+  parameters.mime_type = [NSString stdStringForString:_mimeType];
+  parameters.clock_rate = _clockRate;
+  parameters.channels = _channels;
+  return parameters;
+}
+
+@end
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpEncodingParameters+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpEncodingParameters+Private.h
new file mode 100644
index 00000000000..1d752684785
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpEncodingParameters+Private.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "WebRTC/RTCRtpEncodingParameters.h"
+
+#include "webrtc/api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpEncodingParameters ()
+
+/** Returns the equivalent native RtpEncodingParameters structure. */
+@property(nonatomic, readonly) webrtc::RtpEncodingParameters nativeParameters;
+
+/** Initialize the object with a native RtpEncodingParameters structure. */
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpEncodingParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpEncodingParameters.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpEncodingParameters.mm
new file mode 100644
index 00000000000..af07a0485d6
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpEncodingParameters.mm
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpEncodingParameters+Private.h"
+
+@implementation RTCRtpEncodingParameters
+
+@synthesize isActive = _isActive;
+@synthesize maxBitrateBps = _maxBitrateBps;
+
+static const int kBitrateUnlimited = -1;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpEncodingParameters &)nativeParameters {
+  if (self = [self init]) {
+    _isActive = nativeParameters.active;
+    // TODO(skvlad): Replace with rtc::Optional once the C++ code is updated.
+    if (nativeParameters.max_bitrate_bps != kBitrateUnlimited) {
+      _maxBitrateBps =
+          [NSNumber numberWithInt:nativeParameters.max_bitrate_bps];
+    }
+  }
+  return self;
+}
+
+- (webrtc::RtpEncodingParameters)nativeParameters {
+  webrtc::RtpEncodingParameters parameters;
+  parameters.active = _isActive;
+  if (_maxBitrateBps != nil) {
+    parameters.max_bitrate_bps = _maxBitrateBps.intValue;
+  }
+  return parameters;
+}
+
+@end
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpParameters+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpParameters+Private.h
new file mode 100644
index 00000000000..0cb1ffed814
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpParameters+Private.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "WebRTC/RTCRtpParameters.h"
+
+#include "webrtc/api/rtpparameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpParameters ()
+
+/** Returns the equivalent native RtpParameters structure. */
+@property(nonatomic, readonly) webrtc::RtpParameters nativeParameters;
+
+/** Initialize the object with a native RtpParameters structure. */
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpParameters &)nativeParameters;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpParameters.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpParameters.mm
new file mode 100644
index 00000000000..5e791066df1
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpParameters.mm
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpParameters+Private.h"
+
+#import "RTCRtpCodecParameters+Private.h"
+#import "RTCRtpEncodingParameters+Private.h"
+
+@implementation RTCRtpParameters
+
+@synthesize encodings = _encodings;
+@synthesize codecs = _codecs;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpParameters &)nativeParameters {
+  if (self = [self init]) {
+    NSMutableArray *encodings = [[NSMutableArray alloc] init];
+    for (const auto &encoding : nativeParameters.encodings) {
+      [encodings addObject:[[RTCRtpEncodingParameters alloc]
+                               initWithNativeParameters:encoding]];
+    }
+    _encodings = encodings;
+
+    NSMutableArray *codecs = [[NSMutableArray alloc] init];
+    for (const auto &codec : nativeParameters.codecs) {
+      [codecs addObject:[[RTCRtpCodecParameters alloc]
+                            initWithNativeParameters:codec]];
+    }
+    _codecs = codecs;
+  }
+  return self;
+}
+
+- (webrtc::RtpParameters)nativeParameters {
+    webrtc::RtpParameters parameters;
+  for (RTCRtpEncodingParameters *encoding in _encodings) {
+    parameters.encodings.push_back(encoding.nativeParameters);
+  }
+  for (RTCRtpCodecParameters *codec in _codecs) {
+    parameters.codecs.push_back(codec.nativeParameters);
+  }
+  return parameters;
+}
+
+@end
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpReceiver+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpReceiver+Private.h
new file mode 100644
index 00000000000..14b68fa0113
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpReceiver+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "WebRTC/RTCRtpReceiver.h"
+
+#include "webrtc/api/rtpreceiverinterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpReceiver ()
+
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
+
+/** Initialize an RTCRtpReceiver with a native RtpReceiverInterface. */
+- (instancetype)initWithNativeRtpReceiver:
+    (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpReceiver.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpReceiver.mm
new file mode 100644
index 00000000000..46c0e6544a4
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpReceiver.mm
@@ -0,0 +1,88 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpReceiver+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+@implementation RTCRtpReceiver {
+  rtc::scoped_refptr<webrtc::RtpReceiverInterface> _nativeRtpReceiver;
+}
+
+- (NSString *)receiverId {
+  return [NSString stringForStdString:_nativeRtpReceiver->id()];
+}
+
+- (RTCRtpParameters *)parameters {
+  return [[RTCRtpParameters alloc]
+      initWithNativeParameters:_nativeRtpReceiver->GetParameters()];
+}
+
+- (void)setParameters:(RTCRtpParameters *)parameters {
+  if (!_nativeRtpReceiver->SetParameters(parameters.nativeParameters)) {
+    RTCLogError(@"RTCRtpReceiver(%p): Failed to set parameters: %@", self,
+        parameters);
+  }
+}
+
+- (RTCMediaStreamTrack *)track {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
+    _nativeRtpReceiver->track());
+  if (nativeTrack) {
+    return [[RTCMediaStreamTrack alloc] initWithNativeTrack:nativeTrack];
+  }
+  return nil;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCRtpReceiver {\n  receiverId: %@\n}",
+      self.receiverId];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTCRtpReceiver *receiver = (RTCRtpReceiver *)object;
+  return _nativeRtpReceiver == receiver.nativeRtpReceiver;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpReceiver.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
+  return _nativeRtpReceiver;
+}
+
+- (instancetype)initWithNativeRtpReceiver:
+    (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
+  if (self = [super init]) {
+    _nativeRtpReceiver = nativeRtpReceiver;
+    RTCLogInfo(
+        @"RTCRtpReceiver(%p): created receiver: %@", self, self.description);
+  }
+  return self;
+}
+
+@end
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpSender+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpSender+Private.h
new file mode 100644
index 00000000000..e372c523325
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpSender+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "WebRTC/RTCRtpSender.h"
+
+#include "webrtc/api/rtpsenderinterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCRtpSender ()
+
+@property(nonatomic, readonly)
+    rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
+
+/** Initialize an RTCRtpSender with a native RtpSenderInterface. */
+- (instancetype)initWithNativeRtpSender:
+    (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpSender.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpSender.mm
new file mode 100644
index 00000000000..9ecf0ae5ae9
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCRtpSender.mm
@@ -0,0 +1,94 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpSender+Private.h"
+
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/api/mediastreaminterface.h"
+
+@implementation RTCRtpSender {
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> _nativeRtpSender;
+}
+
+- (NSString *)senderId {
+  return [NSString stringForStdString:_nativeRtpSender->id()];
+}
+
+- (RTCRtpParameters *)parameters {
+  return [[RTCRtpParameters alloc]
+      initWithNativeParameters:_nativeRtpSender->GetParameters()];
+}
+
+- (void)setParameters:(RTCRtpParameters *)parameters {
+  if (!_nativeRtpSender->SetParameters(parameters.nativeParameters)) {
+    RTCLogError(@"RTCRtpSender(%p): Failed to set parameters: %@", self,
+        parameters);
+  }
+}
+
+- (RTCMediaStreamTrack *)track {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
+    _nativeRtpSender->track());
+  if (nativeTrack) {
+    return [[RTCMediaStreamTrack alloc] initWithNativeTrack:nativeTrack];
+  }
+  return nil;
+}
+
+- (void)setTrack:(RTCMediaStreamTrack *)track {
+  if (!_nativeRtpSender->SetTrack(track.nativeTrack)) {
+    RTCLogError(@"RTCRtpSender(%p): Failed to set track %@", self, track);
+  }
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTCRtpSender {\n  senderId: %@\n}",
+      self.senderId];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTCRtpSender *sender = (RTCRtpSender *)object;
+  return _nativeRtpSender == sender.nativeRtpSender;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpSender.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  return _nativeRtpSender;
+}
+
+- (instancetype)initWithNativeRtpSender:
+    (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  NSParameterAssert(nativeRtpSender);
+  if (self = [super init]) {
+    _nativeRtpSender = nativeRtpSender;
+    RTCLogInfo(@"RTCRtpSender(%p): created sender: %@", self, self.description);
+  }
+  return self;
+}
+
+@end
diff --git a/chromium/third_party/webrtc/base/objc/RTCSSLAdapter.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCSSLAdapter.mm
index c881a4487ed..a0da105bc35 100644
--- a/chromium/third_party/webrtc/base/objc/RTCSSLAdapter.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCSSLAdapter.mm
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCSSLAdapter.h"
+#import "WebRTC/RTCSSLAdapter.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/ssladapter.h"
diff --git a/chromium/third_party/webrtc/api/objc/RTCSessionDescription+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription+Private.h
index 9de8f0e9430..04b6fbe8642 100644
--- a/chromium/third_party/webrtc/api/objc/RTCSessionDescription+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCSessionDescription.h"
+#import "WebRTC/RTCSessionDescription.h"
 
 #include "webrtc/api/jsep.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCSessionDescription.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription.mm
index 94c1a3f2d98..417ff7dfca2 100644
--- a/chromium/third_party/webrtc/api/objc/RTCSessionDescription.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription.mm
@@ -8,13 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCSessionDescription.h"
+#import "RTCSessionDescription+Private.h"
 
-#include "webrtc/base/checks.h"
+#import "NSString+StdString.h"
+#import "WebRTC/RTCLogging.h"
 
-#import "webrtc/api/objc/RTCSessionDescription+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#include "webrtc/base/checks.h"
 
 @implementation RTCSessionDescription
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCStatsReport+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCStatsReport+Private.h
index 5ce5801bee2..e1af6f75ea6 100644
--- a/chromium/third_party/webrtc/api/objc/RTCStatsReport+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCStatsReport+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCStatsReport.h"
+#import "WebRTC/RTCStatsReport.h"
 
 #include "webrtc/api/statstypes.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCStatsReport.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCStatsReport.mm
index 99cdd287b8a..1bc3eb9f0ee 100644
--- a/chromium/third_party/webrtc/api/objc/RTCStatsReport.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCStatsReport.mm
@@ -8,13 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCStatsReport.h"
+#import "RTCStatsReport+Private.h"
 
-#include "webrtc/base/checks.h"
+#import "NSString+StdString.h"
+#import "WebRTC/RTCLogging.h"
 
-#import "webrtc/api/objc/RTCStatsReport+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#include "webrtc/base/checks.h"
 
 @implementation RTCStatsReport
 
diff --git a/chromium/third_party/webrtc/base/objc/RTCTracing.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCTracing.mm
index a51c7033402..37755a6b9ab 100644
--- a/chromium/third_party/webrtc/base/objc/RTCTracing.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCTracing.mm
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "webrtc/base/objc/RTCTracing.h"
+#import "WebRTC/RTCTracing.h"
 
 #include "webrtc/base/event_tracer.h"
 
diff --git a/chromium/third_party/webrtc/base/objc/RTCUIApplication.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCUIApplication.h
index c06404f777b..fb11edef44e 100644
--- a/chromium/third_party/webrtc/base/objc/RTCUIApplication.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCUIApplication.h
@@ -11,11 +11,11 @@
 #ifndef WEBRTC_BASE_OBJC_RTC_UI_APPLICATION_H_
 #define WEBRTC_BASE_OBJC_RTC_UI_APPLICATION_H_
 
-#include "webrtc/base/objc/RTCMacros.h"
+#include "WebRTC/RTCMacros.h"
 
 #if defined(WEBRTC_IOS)
 /** Convenience function to get UIApplicationState from C++. */
-RTC_EXPORT bool RTCIsUIApplicationActive();
+RTC_EXTERN bool RTCIsUIApplicationActive();
 #endif  // WEBRTC_IOS
 
 #endif  // WEBRTC_BASE_OBJC_RTC_UI_APPLICATION_H_
diff --git a/chromium/third_party/webrtc/base/objc/RTCUIApplication.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCUIApplication.mm
index 85376b4e0e7..7e8aea62955 100644
--- a/chromium/third_party/webrtc/base/objc/RTCUIApplication.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCUIApplication.mm
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/base/objc/RTCUIApplication.h"
+#include "RTCUIApplication.h"
 
 #if defined(WEBRTC_IOS)
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoFrame+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoFrame+Private.h
index 52f532ce885..a4807127ca7 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoFrame+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoFrame+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoFrame.h"
+#import "WebRTC/RTCVideoFrame.h"
 
 #include "webrtc/media/base/videoframe.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoFrame.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoFrame.mm
index 95f4ac82bfe..1fb26954a39 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoFrame.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoFrame.mm
@@ -8,14 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoFrame.h"
+#import "RTCVideoFrame+Private.h"
 
-#include "webrtc/base/scoped_ptr.h"
-
-#import "webrtc/api/objc/RTCVideoFrame+Private.h"
+#include <memory>
 
 @implementation RTCVideoFrame {
-  rtc::scoped_ptr<cricket::VideoFrame> _videoFrame;
+  std::unique_ptr<cricket::VideoFrame> _videoFrame;
   rtc::scoped_refptr<webrtc::VideoFrameBuffer> _i420Buffer;
 }
 
@@ -92,10 +90,10 @@
   if (!_i420Buffer) {
     if (_videoFrame->GetNativeHandle()) {
       // Convert to I420.
-      _i420Buffer = _videoFrame->GetVideoFrameBuffer()->NativeToI420Buffer();
+      _i420Buffer = _videoFrame->video_frame_buffer()->NativeToI420Buffer();
     } else {
       // Should already be I420.
-      _i420Buffer = _videoFrame->GetVideoFrameBuffer();
+      _i420Buffer = _videoFrame->video_frame_buffer();
     }
   }
 }
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoRendererAdapter+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter+Private.h
index f0e0c6c0add..b413f7e3f6b 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoRendererAdapter+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter+Private.h
@@ -10,9 +10,9 @@
 
 #import "RTCVideoRendererAdapter.h"
 
-#include "webrtc/api/mediastreaminterface.h"
+#import "WebRTC/RTCVideoRenderer.h"
 
-#import "RTCVideoRenderer.h"
+#include "webrtc/api/mediastreaminterface.h"
 
 NS_ASSUME_NONNULL_BEGIN
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoRendererAdapter.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter.h
index b0b6f044884..b0b6f044884 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoRendererAdapter.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter.h
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoRendererAdapter.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter.mm
index 81e7888f3bc..4976ba9f1f1 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoRendererAdapter.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter.mm
@@ -8,10 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoRendererAdapter.h"
+#import "RTCVideoRendererAdapter+Private.h"
 
-#import "webrtc/api/objc/RTCVideoFrame+Private.h"
-#import "webrtc/api/objc/RTCVideoRendererAdapter+Private.h"
+#import "RTCVideoFrame+Private.h"
+
+#include <memory>
+
+#include "webrtc/media/engine/webrtcvideoframe.h"
 
 namespace webrtc {
 
@@ -24,15 +27,31 @@ class VideoRendererAdapter
   }
 
   void OnFrame(const cricket::VideoFrame& nativeVideoFrame) override {
-    const cricket::VideoFrame *frame =
-        nativeVideoFrame.GetCopyWithRotationApplied();
-    CGSize current_size = CGSizeMake(frame->width(), frame->height());
+    RTCVideoFrame *videoFrame = nil;
+    // Rotation of native handles is unsupported right now. Convert to CPU
+    // I420 buffer for rotation before calling the rotation method otherwise
+    // it will hit a DCHECK.
+    if (nativeVideoFrame.rotation() != webrtc::kVideoRotation_0 &&
+        nativeVideoFrame.GetNativeHandle()) {
+      rtc::scoped_refptr<webrtc::VideoFrameBuffer> i420Buffer =
+          nativeVideoFrame.video_frame_buffer()->NativeToI420Buffer();
+      std::unique_ptr<cricket::VideoFrame> cpuFrame(
+          new cricket::WebRtcVideoFrame(i420Buffer,
+                                        nativeVideoFrame.rotation(),
+                                        nativeVideoFrame.timestamp_us()));
+      const cricket::VideoFrame *rotatedFrame =
+          cpuFrame->GetCopyWithRotationApplied();
+      videoFrame = [[RTCVideoFrame alloc] initWithNativeFrame:rotatedFrame];
+    } else {
+      const cricket::VideoFrame *rotatedFrame =
+          nativeVideoFrame.GetCopyWithRotationApplied();
+      videoFrame = [[RTCVideoFrame alloc] initWithNativeFrame:rotatedFrame];
+    }
+    CGSize current_size = CGSizeMake(videoFrame.width, videoFrame.height);
     if (!CGSizeEqualToSize(size_, current_size)) {
       size_ = current_size;
       [adapter_.videoRenderer setSize:size_];
     }
-    RTCVideoFrame *videoFrame =
-        [[RTCVideoFrame alloc] initWithNativeFrame:frame];
     [adapter_.videoRenderer renderFrame:videoFrame];
   }
 
@@ -43,7 +62,7 @@ class VideoRendererAdapter
 }
 
 @implementation RTCVideoRendererAdapter {
-  rtc::scoped_ptr<webrtc::VideoRendererAdapter> _adapter;
+  std::unique_ptr<webrtc::VideoRendererAdapter> _adapter;
 }
 
 @synthesize videoRenderer = _videoRenderer;
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoSource+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoSource+Private.h
index 7746ba518ff..757c1746ef7 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoSource+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoSource+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoSource.h"
+#import "WebRTC/RTCVideoSource.h"
 
 #include "webrtc/api/mediastreaminterface.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoSource.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoSource.mm
index 8e9c39d3ddc..eddf5e0c684 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoSource.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoSource.mm
@@ -8,9 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoSource.h"
-
-#import "webrtc/api/objc/RTCVideoSource+Private.h"
+#import "RTCVideoSource+Private.h"
 
 @implementation RTCVideoSource {
   rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> _nativeVideoSource;
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoTrack+Private.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoTrack+Private.h
index be041246c3a..5199be3b26c 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoTrack+Private.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoTrack+Private.h
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoTrack.h"
+#import "WebRTC/RTCVideoTrack.h"
 
 #include "webrtc/api/mediastreaminterface.h"
 
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoTrack.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoTrack.mm
index a32ea7bc7c1..6691375fbff 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoTrack.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/RTCVideoTrack.mm
@@ -8,14 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoTrack.h"
+#import "RTCVideoTrack+Private.h"
 
-#import "webrtc/api/objc/RTCMediaStreamTrack+Private.h"
-#import "webrtc/api/objc/RTCPeerConnectionFactory+Private.h"
-#import "webrtc/api/objc/RTCVideoRendererAdapter+Private.h"
-#import "webrtc/api/objc/RTCVideoSource+Private.h"
-#import "webrtc/api/objc/RTCVideoTrack+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoRendererAdapter+Private.h"
+#import "RTCVideoSource+Private.h"
 
 @implementation RTCVideoTrack {
   NSMutableArray *_adapters;
diff --git a/chromium/third_party/webrtc/api/objc/avfoundationvideocapturer.h b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h
index 73cecd9d6c2..c523b527bbb 100644
--- a/chromium/third_party/webrtc/api/objc/avfoundationvideocapturer.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h
@@ -11,12 +11,11 @@
 #ifndef WEBRTC_API_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
 #define WEBRTC_API_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#import <AVFoundation/AVFoundation.h>
+
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/video_frame.h"
 
-#import <AVFoundation/AVFoundation.h>
-
 @class RTCAVFoundationVideoCapturerInternal;
 
 namespace rtc {
diff --git a/chromium/third_party/webrtc/api/objc/avfoundationvideocapturer.mm b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.mm
index 85b96586502..cb945f41cae 100644
--- a/chromium/third_party/webrtc/api/objc/avfoundationvideocapturer.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.mm
@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/api/objc/avfoundationvideocapturer.h"
-
-#include "webrtc/base/bind.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/thread.h"
+#include "avfoundationvideocapturer.h"
 
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 #import <UIKit/UIKit.h>
 
-#import "webrtc/base/objc/RTCDispatcher+Private.h"
-#import "webrtc/base/objc/RTCLogging.h"
+#import "RTCDispatcher+Private.h"
+#import "WebRTC/RTCLogging.h"
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/thread.h"
 
 // TODO(tkchin): support other formats.
 static NSString *const kDefaultPreset = AVCaptureSessionPreset640x480;
diff --git a/chromium/third_party/webrtc/api/objc/RTCAVFoundationVideoSource.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
index 6d224d3f8e5..2a732b9686b 100644
--- a/chromium/third_party/webrtc/api/objc/RTCAVFoundationVideoSource.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
@@ -8,7 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCVideoSource.h"
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCVideoSource.h>
 
 @class AVCaptureSession;
 @class RTCMediaConstraints;
@@ -23,6 +24,7 @@ NS_ASSUME_NONNULL_BEGIN
  * guarantee its lifetime. Instead, we expose its properties through the ref
  * counted video source interface.
  */
+RTC_EXPORT
 @interface RTCAVFoundationVideoSource : RTCVideoSource
 
 - (instancetype)init NS_UNAVAILABLE;
diff --git a/chromium/third_party/webrtc/api/objc/RTCAudioTrack.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
index c33a7b42bf6..a9f5fe22777 100644
--- a/chromium/third_party/webrtc/api/objc/RTCAudioTrack.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
@@ -8,10 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaStreamTrack.h"
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCAudioTrack : RTCMediaStreamTrack
 
 - (instancetype)init NS_UNAVAILABLE;
diff --git a/chromium/third_party/webrtc/base/objc/RTCCameraPreviewView.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
index 03e94c29aeb..9018aec3d18 100644
--- a/chromium/third_party/webrtc/base/objc/RTCCameraPreviewView.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
@@ -11,12 +11,15 @@
 #import <Foundation/Foundation.h>
 #import <UIKit/UIKit.h>
 
+#import <WebRTC/RTCMacros.h>
+
 @class AVCaptureSession;
 @class RTCAVFoundationVideoSource;
 
 /** RTCCameraPreviewView is a view that renders local video from an
  *  AVCaptureSession.
  */
+RTC_EXPORT
 @interface RTCCameraPreviewView : UIView
 
 /** The capture session being rendered in the view. Capture session
diff --git a/chromium/third_party/webrtc/api/objc/RTCConfiguration.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
index 144c8d3bd5e..74e7801ac05 100644
--- a/chromium/third_party/webrtc/api/objc/RTCConfiguration.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 @class RTCIceServer;
 
 /**
@@ -42,6 +44,12 @@ typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
   RTCTcpCandidatePolicyDisabled
 };
 
+/** Represents the continual gathering policy. */
+typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
+  RTCContinualGatheringPolicyGatherOnce,
+  RTCContinualGatheringPolicyGatherContinually
+};
+
 /** Represents the encryption key type. */
 typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
   RTCEncryptionKeyTypeRSA,
@@ -50,6 +58,7 @@ typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
 
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCConfiguration : NSObject
 
 /** An array of Ice Servers available to be used by ICE. */
@@ -65,6 +74,8 @@ NS_ASSUME_NONNULL_BEGIN
 /** The rtcp-mux policy to use when gathering ICE candidates. */
 @property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
 @property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign)
+    RTCContinualGatheringPolicy continualGatheringPolicy;
 @property(nonatomic, assign) int audioJitterBufferMaxPackets;
 @property(nonatomic, assign) int iceConnectionReceivingTimeout;
 @property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
diff --git a/chromium/third_party/webrtc/api/objc/RTCDataChannel.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
index 59ab2b8b26f..893bd0a3365 100644
--- a/chromium/third_party/webrtc/api/objc/RTCDataChannel.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
@@ -11,8 +11,11 @@
 #import <AvailabilityMacros.h>
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCDataBuffer : NSObject
 
 /** NSData representation of the underlying buffer. */
@@ -33,6 +36,7 @@ NS_ASSUME_NONNULL_BEGIN
 
 
 @class RTCDataChannel;
+RTC_EXPORT
 @protocol RTCDataChannelDelegate <NSObject>
 
 /** The data channel state changed. */
@@ -58,7 +62,7 @@ typedef NS_ENUM(NSInteger, RTCDataChannelState) {
   RTCDataChannelStateClosed,
 };
 
-
+RTC_EXPORT
 @interface RTCDataChannel : NSObject
 
 /**
diff --git a/chromium/third_party/webrtc/api/objc/RTCDataChannelConfiguration.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
index fbe342d2c86..65f59318447 100644
--- a/chromium/third_party/webrtc/api/objc/RTCDataChannelConfiguration.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
@@ -11,8 +11,11 @@
 #import <AvailabilityMacros.h>
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCDataChannelConfiguration : NSObject
 
 /** Set to YES if ordered delivery is required. */
diff --git a/chromium/third_party/webrtc/base/objc/RTCDispatcher.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
index a9b92a661fd..328beaf8728 100644
--- a/chromium/third_party/webrtc/base/objc/RTCDispatcher.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
   // Main dispatcher queue.
   RTCDispatcherTypeMain,
@@ -23,6 +25,7 @@ typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
 /** Dispatcher that asynchronously dispatches blocks to a specific
  *  shared dispatch queue.
  */
+RTC_EXPORT
 @interface RTCDispatcher : NSObject
 
 - (instancetype)init NS_UNAVAILABLE;
diff --git a/chromium/third_party/webrtc/api/objc/RTCEAGLVideoView.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
index 1a57df76bbe..5f8d73bc8d6 100644
--- a/chromium/third_party/webrtc/api/objc/RTCEAGLVideoView.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
@@ -11,11 +11,13 @@
 #import <Foundation/Foundation.h>
 #import <UIKit/UIKit.h>
 
-#import "RTCVideoRenderer.h"
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCVideoRenderer.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
 @class RTCEAGLVideoView;
+RTC_EXPORT
 @protocol RTCEAGLVideoViewDelegate
 
 - (void)videoView:(RTCEAGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
@@ -26,6 +28,7 @@ NS_ASSUME_NONNULL_BEGIN
  * RTCEAGLVideoView is an RTCVideoRenderer which renders video frames in its
  * bounds using OpenGLES 2.0.
  */
+RTC_EXPORT
 @interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
 
 @property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
diff --git a/chromium/third_party/webrtc/base/objc/RTCFieldTrials.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
index 837a8f7a609..7e7c1c4480a 100644
--- a/chromium/third_party/webrtc/base/objc/RTCFieldTrials.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/base/objc/RTCMacros.h"
-
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 typedef NS_OPTIONS(NSUInteger, RTCFieldTrialOptions) {
   RTCFieldTrialOptionsNone = 0,
   RTCFieldTrialOptionsSendSideBwe = 1 << 0,
@@ -20,4 +20,4 @@ typedef NS_OPTIONS(NSUInteger, RTCFieldTrialOptions) {
 /** Must be called before any other call into WebRTC. See:
  *  webrtc/system_wrappers/include/field_trial_default.h
  */
-RTC_EXPORT void RTCInitFieldTrials(RTCFieldTrialOptions options);
+RTC_EXTERN void RTCInitFieldTrials(RTCFieldTrialOptions options);
diff --git a/chromium/third_party/webrtc/base/objc/RTCFileLogger.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
index 281be4e31ff..5656b7bf43f 100644
--- a/chromium/third_party/webrtc/base/objc/RTCFileLogger.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
   RTCFileLoggerSeverityVerbose,
   RTCFileLoggerSeverityInfo,
@@ -31,6 +33,7 @@ NS_ASSUME_NONNULL_BEGIN
 // are preserved while the middle section is overwritten instead.
 // For kRTCFileLoggerTypeApp, the oldest log is overwritten.
 // This class is not threadsafe.
+RTC_EXPORT
 @interface RTCFileLogger : NSObject
 
 // The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
diff --git a/chromium/third_party/webrtc/api/objc/RTCIceCandidate.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
index e521ae04f6d..9f472b8df05 100644
--- a/chromium/third_party/webrtc/api/objc/RTCIceCandidate.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
@@ -10,8 +10,11 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCIceCandidate : NSObject
 
 /**
diff --git a/chromium/third_party/webrtc/api/objc/RTCIceServer.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
index 487588ebc48..ff5489cb412 100644
--- a/chromium/third_party/webrtc/api/objc/RTCIceServer.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
@@ -10,8 +10,11 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCIceServer : NSObject
 
 /** URI(s) for this server represented as NSStrings. */
diff --git a/chromium/third_party/webrtc/base/objc/RTCLogging.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
index fbc4ffade27..e4f19203829 100644
--- a/chromium/third_party/webrtc/base/objc/RTCLogging.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 // Subset of rtc::LoggingSeverity.
 typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
   RTCLoggingSeverityVerbose,
@@ -18,24 +20,16 @@ typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
   RTCLoggingSeverityError,
 };
 
-#if defined(__cplusplus)
-extern "C" void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
-extern "C" void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
-extern "C" NSString* RTCFileName(const char* filePath);
-#else
-
 // Wrapper for C++ LOG(sev) macros.
 // Logs the log string to the webrtc logstream for the given severity.
-extern void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
 
 // Wrapper for rtc::LogMessage::LogToDebug.
 // Sets the minimum severity to be logged to console.
-extern void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
 
 // Returns the filename with the path prefix removed.
-extern NSString* RTCFileName(const char* filePath);
-
-#endif
+RTC_EXTERN NSString* RTCFileName(const char* filePath);
 
 // Some convenience macros.
 
diff --git a/chromium/third_party/webrtc/base/objc/RTCMacros.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMacros.h
index 4979a2b87f5..f9f15c37d39 100644
--- a/chromium/third_party/webrtc/base/objc/RTCMacros.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMacros.h
@@ -11,10 +11,12 @@
 #ifndef WEBRTC_BASE_OBJC_RTC_MACROS_H_
 #define WEBRTC_BASE_OBJC_RTC_MACROS_H_
 
+#define RTC_EXPORT __attribute__((visibility("default")))
+
 #if defined(__cplusplus)
-#define RTC_EXPORT extern "C"
+#define RTC_EXTERN extern "C" RTC_EXPORT
 #else
-#define RTC_EXPORT extern
+#define RTC_EXTERN extern RTC_EXPORT
 #endif
 
 #ifdef __OBJC__
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaConstraints.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
index a8ad39142e3..adc73f2f432 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaConstraints.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
@@ -10,8 +10,11 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCMediaConstraints : NSObject
 
 - (instancetype)init NS_UNAVAILABLE;
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaStream.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
index 50ae7df925c..b97960d436c 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaStream.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
@@ -10,12 +10,15 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
 @class RTCAudioTrack;
 @class RTCPeerConnectionFactory;
 @class RTCVideoTrack;
 
+RTC_EXPORT
 @interface RTCMediaStream : NSObject
 
 /** The audio tracks in this stream. */
diff --git a/chromium/third_party/webrtc/api/objc/RTCMediaStreamTrack.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
index b8576bdf3bd..c42f0cfb1a0 100644
--- a/chromium/third_party/webrtc/api/objc/RTCMediaStreamTrack.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 /**
  * Represents the state of the track. This exposes the same states in C++.
  */
@@ -20,6 +22,10 @@ typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
 
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXTERN NSString * const kRTCMediaStreamTrackKindAudio;
+RTC_EXTERN NSString * const kRTCMediaStreamTrackKindVideo;
+
+RTC_EXPORT
 @interface RTCMediaStreamTrack : NSObject
 
 /**
diff --git a/chromium/third_party/webrtc/api/objc/RTCNSGLVideoView.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
index 27eb31e9af1..d9ed6fb9f26 100644
--- a/chromium/third_party/webrtc/api/objc/RTCNSGLVideoView.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
@@ -8,13 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#if TARGET_OS_IPHONE
-#error "This file targets OSX."
-#endif
+#if !TARGET_OS_IPHONE
 
 #import <AppKit/NSOpenGLView.h>
 
-#import "RTCVideoRenderer.h"
+#import <WebRTC/RTCVideoRenderer.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
@@ -32,3 +30,5 @@ NS_ASSUME_NONNULL_BEGIN
 @end
 
 NS_ASSUME_NONNULL_END
+
+#endif
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnection.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
index e0f9b78e8eb..2ba8661d13a 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnection.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 @class RTCConfiguration;
 @class RTCDataChannel;
 @class RTCDataChannelConfiguration;
@@ -18,6 +20,8 @@
 @class RTCMediaStream;
 @class RTCMediaStreamTrack;
 @class RTCPeerConnectionFactory;
+@class RTCRtpReceiver;
+@class RTCRtpSender;
 @class RTCSessionDescription;
 @class RTCStatsReport;
 
@@ -64,6 +68,7 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
 
 @class RTCPeerConnection;
 
+RTC_EXPORT
 @protocol RTCPeerConnectionDelegate <NSObject>
 
 /** Called when the SignalingState changed. */
@@ -99,7 +104,7 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
 
 @end
 
-
+RTC_EXPORT
 @interface RTCPeerConnection : NSObject
 
 /** The object that will be notifed about events such as state changes and
@@ -115,6 +120,18 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
 @property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
 @property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
 
+/** Gets all RTCRtpSenders associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpSender.
+ *  Use isEqual: instead of == to compare RTCRtpSender instances.
+ */
+@property(nonatomic, readonly) NSArray<RTCRtpSender *> *senders;
+
+/** Gets all RTCRtpReceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpReceiver.
+ *  Use isEqual: instead of == to compare RTCRtpReceiver instances.
+ */
+@property(nonatomic, readonly) NSArray<RTCRtpReceiver *> *receivers;
+
 - (instancetype)init NS_UNAVAILABLE;
 
 /** Sets the PeerConnection's global configuration to |configuration|.
@@ -161,6 +178,16 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
 
 @end
 
+@interface RTCPeerConnection (Media)
+
+/**
+ * Create an RTCRtpSender with the specified kind and media stream ID.
+ * See RTCMediaStreamTrack.h for available kinds.
+ */
+- (RTCRtpSender *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId;
+
+@end
+
 @interface RTCPeerConnection (DataChannel)
 
 /** Create a new data channel with the given label and configuration. */
diff --git a/chromium/third_party/webrtc/api/objc/RTCPeerConnectionFactory.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
index f98a551846b..f21c107581e 100644
--- a/chromium/third_party/webrtc/api/objc/RTCPeerConnectionFactory.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
 @class RTCAVFoundationVideoSource;
@@ -22,6 +24,7 @@ NS_ASSUME_NONNULL_BEGIN
 @class RTCVideoTrack;
 @protocol RTCPeerConnectionDelegate;
 
+RTC_EXPORT
 @interface RTCPeerConnectionFactory : NSObject
 
 - (instancetype)init NS_DESIGNATED_INITIALIZER;
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
new file mode 100644
index 00000000000..fef8429803c
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const NSString * const kRTCRtxCodecMimeType;
+RTC_EXTERN const NSString * const kRTCRedCodecMimeType;
+RTC_EXTERN const NSString * const kRTCUlpfecCodecMimeType;
+RTC_EXTERN const NSString * const kRTCOpusCodecMimeType;
+RTC_EXTERN const NSString * const kRTCIsacCodecMimeType;
+RTC_EXTERN const NSString * const kRTCL16CodecMimeType;
+RTC_EXTERN const NSString * const kRTCG722CodecMimeType;
+RTC_EXTERN const NSString * const kRTCIlbcCodecMimeType;
+RTC_EXTERN const NSString * const kRTCPcmuCodecMimeType;
+RTC_EXTERN const NSString * const kRTCPcmaCodecMimeType;
+RTC_EXTERN const NSString * const kRTCDtmfCodecMimeType;
+RTC_EXTERN const NSString * const kRTCComfortNoiseCodecMimeType;
+RTC_EXTERN const NSString * const kRTCVp8CodecMimeType;
+RTC_EXTERN const NSString * const kRTCVp9CodecMimeType;
+RTC_EXTERN const NSString * const kRTCH264CodecMimeType;
+
+/** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
+RTC_EXPORT
+@interface RTCRtpCodecParameters : NSObject
+
+/** The RTP payload type. */
+@property(nonatomic, assign) int payloadType;
+
+/**
+ * The codec MIME type. Valid types are listed in:
+ * http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
+ *
+ * Several supported types are represented by the constants above.
+ */
+@property(nonatomic, nonnull) NSString *mimeType;
+
+/** The codec clock rate expressed in Hertz. */
+@property(nonatomic, assign) int clockRate;
+
+/** The number of channels (mono=1, stereo=2). */
+@property(nonatomic, assign) int channels;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
new file mode 100644
index 00000000000..65b8deafc08
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@interface RTCRtpEncodingParameters : NSObject
+
+/** Controls whether the encoding is currently transmitted. */
+@property(nonatomic, assign) BOOL isActive;
+
+/** The maximum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
new file mode 100644
index 00000000000..bdebf84884c
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCRtpCodecParameters.h>
+#import <WebRTC/RTCRtpEncodingParameters.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@interface RTCRtpParameters : NSObject
+
+/** The currently active encodings in the order of preference. */
+@property(nonatomic, copy) NSArray<RTCRtpEncodingParameters *> *encodings;
+
+/** The negotiated set of send codecs in order of preference. */
+@property(nonatomic, copy) NSArray<RTCRtpCodecParameters *> *codecs;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
new file mode 100644
index 00000000000..e98a8f3bea1
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
+#import <WebRTC/RTCRtpParameters.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@protocol RTCRtpReceiver <NSObject>
+
+/** A unique identifier for this receiver. */
+@property(nonatomic, readonly) NSString *receiverId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ *
+ *  The WebRTC specification only defines RTCRtpParameters in terms of senders,
+ *  but this API also applies them to receivers, similar to ORTC:
+ *  http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+ */
+@property(nonatomic, readonly) RTCRtpParameters *parameters;
+
+/** The RTCMediaStreamTrack associated with the receiver.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, readonly) RTCMediaStreamTrack *track;
+
+@end
+
+RTC_EXPORT
+@interface RTCRtpReceiver : NSObject <RTCRtpReceiver>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
new file mode 100644
index 00000000000..d910c6ceb28
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/RTCMacros.h>
+#import <WebRTC/RTCMediaStreamTrack.h>
+#import <WebRTC/RTCRtpParameters.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXPORT
+@protocol RTCRtpSender <NSObject>
+
+/** A unique identifier for this sender. */
+@property(nonatomic, readonly) NSString *senderId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ */
+@property(nonatomic, copy) RTCRtpParameters *parameters;
+
+/** The RTCMediaStreamTrack associated with the sender.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, copy, nullable) RTCMediaStreamTrack *track;
+
+@end
+
+RTC_EXPORT
+@interface RTCRtpSender : NSObject <RTCRtpSender>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/chromium/third_party/webrtc/base/objc/RTCSSLAdapter.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
index 34db8432c2f..1341a5c003c 100644
--- a/chromium/third_party/webrtc/base/objc/RTCSSLAdapter.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
@@ -10,11 +10,11 @@
 
 #import <Foundation/Foundation.h>
 
-#import "webrtc/base/objc/RTCMacros.h"
+#import <WebRTC/RTCMacros.h>
 
 /**
  * Initialize and clean up the SSL library. Failure is fatal. These call the
  * corresponding functions in webrtc/base/ssladapter.h.
  */
-RTC_EXPORT BOOL RTCInitializeSSL();
-RTC_EXPORT BOOL RTCCleanupSSL();
+RTC_EXTERN BOOL RTCInitializeSSL();
+RTC_EXTERN BOOL RTCCleanupSSL();
diff --git a/chromium/third_party/webrtc/api/objc/RTCSessionDescription.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
index 2635633f39f..41439804a8a 100644
--- a/chromium/third_party/webrtc/api/objc/RTCSessionDescription.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 /**
  * Represents the session description type. This exposes the same types that are
  * in C++, which doesn't include the rollback type that is in the W3C spec.
@@ -22,6 +24,7 @@ typedef NS_ENUM(NSInteger, RTCSdpType) {
 
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCSessionDescription : NSObject
 
 /** The type of session description. */
diff --git a/chromium/third_party/webrtc/api/objc/RTCStatsReport.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCStatsReport.h
index f3f8907dfb3..5a559679991 100644
--- a/chromium/third_party/webrtc/api/objc/RTCStatsReport.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCStatsReport.h
@@ -10,9 +10,12 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
 /** This does not currently conform to the spec. */
+RTC_EXPORT
 @interface RTCStatsReport : NSObject
 
 /** Time since 1970-01-01T00:00:00Z in milliseconds. */
diff --git a/chromium/third_party/webrtc/base/objc/RTCTracing.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
index 5b8e2f6ac37..136479118b5 100644
--- a/chromium/third_party/webrtc/base/objc/RTCTracing.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
@@ -10,12 +10,12 @@
 
 #import <Foundation/Foundation.h>
 
-#import "webrtc/base/objc/RTCMacros.h"
+#import <WebRTC/RTCMacros.h>
 
-RTC_EXPORT void RTCSetupInternalTracer();
+RTC_EXTERN void RTCSetupInternalTracer();
 /** Starts capture to specified file. Must be a valid writable path.
  *  Returns YES if capture starts.
  */
-RTC_EXPORT BOOL RTCStartInternalCapture(NSString *filePath);
-RTC_EXPORT void RTCStopInternalCapture();
-RTC_EXPORT void RTCShutdownInternalTracer();
+RTC_EXTERN BOOL RTCStartInternalCapture(NSString *filePath);
+RTC_EXTERN void RTCStopInternalCapture();
+RTC_EXTERN void RTCShutdownInternalTracer();
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoFrame.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
index b44bf7331d1..efb666e4bf2 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoFrame.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
@@ -11,9 +11,12 @@
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
 // RTCVideoFrame is an ObjectiveC version of cricket::VideoFrame.
+RTC_EXPORT
 @interface RTCVideoFrame : NSObject
 
 /** Width without rotation applied. */
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoRenderer.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
index 2fe4efb45d6..5e2e820918a 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoRenderer.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
@@ -13,10 +13,13 @@
 #import <UIKit/UIKit.h>
 #endif
 
+#import <WebRTC/RTCMacros.h>
+
 NS_ASSUME_NONNULL_BEGIN
 
 @class RTCVideoFrame;
 
+RTC_EXPORT
 @protocol RTCVideoRenderer <NSObject>
 
 /** The size of the frame. */
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoSource.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
index a98fa6c6bba..e234c3e0896 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoSource.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
@@ -10,6 +10,8 @@
 
 #import <Foundation/Foundation.h>
 
+#import <WebRTC/RTCMacros.h>
+
 typedef NS_ENUM(NSInteger, RTCSourceState) {
   RTCSourceStateInitializing,
   RTCSourceStateLive,
@@ -19,6 +21,7 @@ typedef NS_ENUM(NSInteger, RTCSourceState) {
 
 NS_ASSUME_NONNULL_BEGIN
 
+RTC_EXPORT
 @interface RTCVideoSource : NSObject
 
 /** The current state of the RTCVideoSource. */
diff --git a/chromium/third_party/webrtc/api/objc/RTCVideoTrack.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
index 1d883760681..899d7c34780 100644
--- a/chromium/third_party/webrtc/api/objc/RTCVideoTrack.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
@@ -8,7 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import "RTCMediaStreamTrack.h"
+#import <WebRTC/RTCMediaStreamTrack.h>
+
+#import <WebRTC/RTCMacros.h>
 
 NS_ASSUME_NONNULL_BEGIN
 
@@ -16,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
 @class RTCPeerConnectionFactory;
 @class RTCVideoSource;
 
+RTC_EXPORT
 @interface RTCVideoTrack : RTCMediaStreamTrack
 
 /** The video source for this video track. */
diff --git a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC/WebRTC.h b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/WebRTC.h
index 9f2d8454e97..bf08a0d3478 100644
--- a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC/WebRTC.h
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Headers/WebRTC/WebRTC.h
@@ -8,16 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#import <Foundation/Foundation.h>
-
-//! Project version number for WebRTC.
-FOUNDATION_EXPORT double WebRTCVersionNumber;
-
-//! Project version string for WebRTC.
-FOUNDATION_EXPORT const unsigned char WebRTCVersionString[];
-
-#import <WebRTC/RTCAudioTrack.h>
 #import <WebRTC/RTCAVFoundationVideoSource.h>
+#import <WebRTC/RTCAudioTrack.h>
 #import <WebRTC/RTCCameraPreviewView.h>
 #import <WebRTC/RTCConfiguration.h>
 #import <WebRTC/RTCDataChannel.h>
@@ -33,15 +25,18 @@ FOUNDATION_EXPORT const unsigned char WebRTCVersionString[];
 #import <WebRTC/RTCMediaConstraints.h>
 #import <WebRTC/RTCMediaStream.h>
 #import <WebRTC/RTCMediaStreamTrack.h>
-#import <WebRTC/RTCOpenGLVideoRenderer.h>
 #import <WebRTC/RTCPeerConnection.h>
 #import <WebRTC/RTCPeerConnectionFactory.h>
-#import <WebRTC/RTCSessionDescription.h>
+#import <WebRTC/RTCRtpCodecParameters.h>
+#import <WebRTC/RTCRtpEncodingParameters.h>
+#import <WebRTC/RTCRtpParameters.h>
+#import <WebRTC/RTCRtpReceiver.h>
+#import <WebRTC/RTCRtpSender.h>
 #import <WebRTC/RTCSSLAdapter.h>
+#import <WebRTC/RTCSessionDescription.h>
 #import <WebRTC/RTCStatsReport.h>
 #import <WebRTC/RTCTracing.h>
 #import <WebRTC/RTCVideoFrame.h>
 #import <WebRTC/RTCVideoRenderer.h>
-#import <WebRTC/RTCVideoRendererAdapter.h>
 #import <WebRTC/RTCVideoSource.h>
 #import <WebRTC/RTCVideoTrack.h>
diff --git a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC/Info.plist b/chromium/third_party/webrtc/sdk/objc/Framework/Info.plist
index d3de8eefb69..38c437e7fed 100644
--- a/chromium/third_party/webrtc/build/ios/SDK/Framework/WebRTC/Info.plist
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Info.plist
@@ -5,13 +5,13 @@
 	<key>CFBundleDevelopmentRegion</key>
 	<string>en</string>
 	<key>CFBundleExecutable</key>
-	<string>$(EXECUTABLE_NAME)</string>
+	<string>WebRTC</string>
 	<key>CFBundleIdentifier</key>
-	<string>$(PRODUCT_BUNDLE_IDENTIFIER)</string>
+	<string>org.webrtc.WebRTC</string>
 	<key>CFBundleInfoDictionaryVersion</key>
 	<string>6.0</string>
 	<key>CFBundleName</key>
-	<string>$(PRODUCT_NAME)</string>
+	<string>WebRTC</string>
 	<key>CFBundlePackageType</key>
 	<string>FMWK</string>
 	<key>CFBundleShortVersionString</key>
@@ -19,7 +19,7 @@
 	<key>CFBundleSignature</key>
 	<string>????</string>
 	<key>CFBundleVersion</key>
-	<string>$(CURRENT_PROJECT_VERSION)</string>
+	<string>1.0</string>
 	<key>NSPrincipalClass</key>
 	<string></string>
 </dict>
diff --git a/chromium/third_party/webrtc/sdk/objc/Framework/Modules/module.modulemap b/chromium/third_party/webrtc/sdk/objc/Framework/Modules/module.modulemap
new file mode 100644
index 00000000000..cd485a4e81e
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/Modules/module.modulemap
@@ -0,0 +1,6 @@
+framework module WebRTC {
+  umbrella header "WebRTC.h"
+
+  export *
+  module * { export * }
+}
diff --git a/chromium/third_party/webrtc/api/objctests/RTCConfigurationTest.mm b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm
index 8cfa04d993c..f565adbefd0 100644
--- a/chromium/third_party/webrtc/api/objctests/RTCConfigurationTest.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm
@@ -14,10 +14,10 @@
 
 #include "webrtc/base/gunit.h"
 
-#import "webrtc/api/objc/RTCConfiguration.h"
-#import "webrtc/api/objc/RTCConfiguration+Private.h"
-#import "webrtc/api/objc/RTCIceServer.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCConfiguration+Private.h"
+#import "WebRTC/RTCConfiguration.h"
+#import "WebRTC/RTCIceServer.h"
 
 @interface RTCConfigurationTest : NSObject
 - (void)testConversionToNativeConfiguration;
@@ -41,25 +41,30 @@
   config.audioJitterBufferMaxPackets = maxPackets;
   config.iceConnectionReceivingTimeout = timeout;
   config.iceBackupCandidatePairPingInterval = interval;
+  config.continualGatheringPolicy =
+      RTCContinualGatheringPolicyGatherContinually;
 
-  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig =
-      config.nativeConfiguration;
-  EXPECT_EQ(1u, nativeConfig.servers.size());
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
+      nativeConfig([config createNativeConfiguration]);
+  EXPECT_TRUE(nativeConfig.get());
+  EXPECT_EQ(1u, nativeConfig->servers.size());
   webrtc::PeerConnectionInterface::IceServer nativeServer =
-      nativeConfig.servers.front();
+      nativeConfig->servers.front();
   EXPECT_EQ(1u, nativeServer.urls.size());
   EXPECT_EQ("stun:stun1.example.net", nativeServer.urls.front());
 
-  EXPECT_EQ(webrtc::PeerConnectionInterface::kRelay, nativeConfig.type);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kRelay, nativeConfig->type);
   EXPECT_EQ(webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle,
-            nativeConfig.bundle_policy);
+            nativeConfig->bundle_policy);
   EXPECT_EQ(webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate,
-            nativeConfig.rtcp_mux_policy);
+            nativeConfig->rtcp_mux_policy);
   EXPECT_EQ(webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled,
-            nativeConfig.tcp_candidate_policy);
-  EXPECT_EQ(maxPackets, nativeConfig.audio_jitter_buffer_max_packets);
-  EXPECT_EQ(timeout, nativeConfig.ice_connection_receiving_timeout);
-  EXPECT_EQ(interval, nativeConfig.ice_backup_candidate_pair_ping_interval);
+            nativeConfig->tcp_candidate_policy);
+  EXPECT_EQ(maxPackets, nativeConfig->audio_jitter_buffer_max_packets);
+  EXPECT_EQ(timeout, nativeConfig->ice_connection_receiving_timeout);
+  EXPECT_EQ(interval, nativeConfig->ice_backup_candidate_pair_ping_interval);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::GATHER_CONTINUALLY,
+            nativeConfig->continual_gathering_policy);
 }
 
 @end
diff --git a/chromium/third_party/webrtc/api/objctests/RTCDataChannelConfigurationTest.mm b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm
index 1b3f21b8f90..275898dba85 100644
--- a/chromium/third_party/webrtc/api/objctests/RTCDataChannelConfigurationTest.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm
@@ -12,9 +12,9 @@
 
 #include "webrtc/base/gunit.h"
 
-#import "webrtc/api/objc/RTCDataChannelConfiguration.h"
-#import "webrtc/api/objc/RTCDataChannelConfiguration+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCDataChannelConfiguration+Private.h"
+#import "WebRTC/RTCDataChannelConfiguration.h"
 
 @interface RTCDataChannelConfigurationTest : NSObject
 - (void)testConversionToNativeDataChannelInit;
diff --git a/chromium/third_party/webrtc/api/objctests/RTCIceCandidateTest.mm b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm
index 391db44ae10..20a9d739a96 100644
--- a/chromium/third_party/webrtc/api/objctests/RTCIceCandidateTest.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm
@@ -10,11 +10,13 @@
 
 #import <Foundation/Foundation.h>
 
+#include <memory>
+
 #include "webrtc/base/gunit.h"
 
-#import "webrtc/api/objc/RTCIceCandidate.h"
-#import "webrtc/api/objc/RTCIceCandidate+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCIceCandidate+Private.h"
+#import "WebRTC/RTCIceCandidate.h"
 
 @interface RTCIceCandidateTest : NSObject
 - (void)testCandidate;
@@ -32,7 +34,7 @@
                                                       sdpMLineIndex:0
                                                              sdpMid:@"audio"];
 
-  rtc::scoped_ptr<webrtc::IceCandidateInterface> nativeCandidate =
+  std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate =
       candidate.nativeCandidate;
   EXPECT_EQ("audio", nativeCandidate->sdp_mid());
   EXPECT_EQ(0, nativeCandidate->sdp_mline_index());
diff --git a/chromium/third_party/webrtc/api/objctests/RTCIceServerTest.mm b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCIceServerTest.mm
index 5116ba0c18c..a5159dcaad3 100644
--- a/chromium/third_party/webrtc/api/objctests/RTCIceServerTest.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCIceServerTest.mm
@@ -14,9 +14,9 @@
 
 #include "webrtc/base/gunit.h"
 
-#import "webrtc/api/objc/RTCIceServer.h"
-#import "webrtc/api/objc/RTCIceServer+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCIceServer+Private.h"
+#import "WebRTC/RTCIceServer.h"
 
 @interface RTCIceServerTest : NSObject
 - (void)testOneURLServer;
diff --git a/chromium/third_party/webrtc/api/objctests/RTCMediaConstraintsTest.mm b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm
index f73a6ca895d..3413dfc039a 100644
--- a/chromium/third_party/webrtc/api/objctests/RTCMediaConstraintsTest.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm
@@ -10,11 +10,13 @@
 
 #import <Foundation/Foundation.h>
 
+#include <memory>
+
 #include "webrtc/base/gunit.h"
 
-#import "webrtc/api/objc/RTCMediaConstraints.h"
-#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCMediaConstraints+Private.h"
+#import "WebRTC/RTCMediaConstraints.h"
 
 @interface RTCMediaConstraintsTest : NSObject
 - (void)testMediaConstraints;
@@ -29,7 +31,7 @@
   RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc]
       initWithMandatoryConstraints:mandatory
                optionalConstraints:optional];
-  rtc::scoped_ptr<webrtc::MediaConstraints> nativeConstraints =
+  std::unique_ptr<webrtc::MediaConstraints> nativeConstraints =
       [constraints nativeConstraints];
 
   webrtc::MediaConstraintsInterface::Constraints nativeMandatory =
diff --git a/chromium/third_party/webrtc/api/objctests/RTCSessionDescriptionTest.mm b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm
index 37ac65cf944..90f4698b301 100644
--- a/chromium/third_party/webrtc/api/objctests/RTCSessionDescriptionTest.mm
+++ b/chromium/third_party/webrtc/sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm
@@ -12,9 +12,9 @@
 
 #include "webrtc/base/gunit.h"
 
-#import "webrtc/api/objc/RTCSessionDescription.h"
-#import "webrtc/api/objc/RTCSessionDescription+Private.h"
-#import "webrtc/base/objc/NSString+StdString.h"
+#import "NSString+StdString.h"
+#import "RTCSessionDescription+Private.h"
+#import "WebRTC/RTCSessionDescription.h"
 
 @interface RTCSessionDescriptionTest : NSObject
 - (void)testSessionDescriptionConversion;
diff --git a/chromium/third_party/webrtc/build/WebRTC-Prefix.pch b/chromium/third_party/webrtc/sdk/objc/WebRTC-Prefix.pch
index 506d4fbb75e..c56cb48ab87 100644
--- a/chromium/third_party/webrtc/build/WebRTC-Prefix.pch
+++ b/chromium/third_party/webrtc/sdk/objc/WebRTC-Prefix.pch
@@ -18,10 +18,12 @@
 #error "This file requires nullability support."
 #endif
 
+#import <Foundation/Foundation.h>
+
 // The following nonnull macros were introduced in OSX SDK 10.10.3. However,
 // the bots appear to be running something older. We define them here if they
-// aren't already defined in NSObjCRuntime.h
-#include <Foundation/NSObjCRuntime.h>
+// aren't already defined in NSObjCRuntime.h, which is included by
+// Foundation/Foundation.h.
 
 #if !defined(NS_ASSUME_NONNULL_BEGIN)
 #define NS_ASSUME_NONNULL_BEGIN
diff --git a/chromium/third_party/webrtc/build/ios/SDK/WebRTC.podspec b/chromium/third_party/webrtc/sdk/objc/WebRTC.podspec
index 523b840a59d..37ab166b0c5 100644
--- a/chromium/third_party/webrtc/build/ios/SDK/WebRTC.podspec
+++ b/chromium/third_party/webrtc/sdk/objc/WebRTC.podspec
@@ -1,6 +1,6 @@
 Pod::Spec.new do |s|
   s.name         = "WebRTC"
-  s.version      = "0.0.1"
+  s.version      = "${FRAMEWORK_VERSION_NUMBER}"
   s.summary      = "WebRTC SDK for iOS"
   s.description  = <<-DESC
     WebRTC is a free, open project that provides browsers and mobile
diff --git a/chromium/third_party/webrtc/sdk/sdk.gyp b/chromium/third_party/webrtc/sdk/sdk.gyp
new file mode 100644
index 00000000000..5bfbb249ea6
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/sdk.gyp
@@ -0,0 +1,274 @@
+# Copyright 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [
+    '../build/common.gypi',
+    'sdk.gypi',
+  ],
+  'conditions': [
+    ['OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")', {
+      'targets': [
+        {
+          'target_name': 'rtc_sdk_common_objc',
+          'type': 'static_library',
+          'includes': [ '../build/objc_common.gypi' ],
+          'dependencies': [
+            '../base/base.gyp:rtc_base',
+          ],
+          'include_dirs': [
+            'objc/Framework/Classes',
+            'objc/Framework/Headers',
+          ],
+          'direct_dependent_settings': {
+            'include_dirs': [
+              'objc/Framework/Classes',
+              'objc/Framework/Headers',
+            ],
+          },
+          'sources': [
+            'objc/Framework/Classes/NSString+StdString.h',
+            'objc/Framework/Classes/NSString+StdString.mm',
+            'objc/Framework/Classes/RTCDispatcher.m',
+            'objc/Framework/Classes/RTCFieldTrials.mm',
+            'objc/Framework/Classes/RTCLogging.mm',
+            'objc/Framework/Classes/RTCSSLAdapter.mm',
+            'objc/Framework/Classes/RTCTracing.mm',
+            'objc/Framework/Headers/WebRTC/RTCDispatcher.h',
+            'objc/Framework/Headers/WebRTC/RTCFieldTrials.h',
+            'objc/Framework/Headers/WebRTC/RTCLogging.h',
+            'objc/Framework/Headers/WebRTC/RTCMacros.h',
+            'objc/Framework/Headers/WebRTC/RTCSSLAdapter.h',
+            'objc/Framework/Headers/WebRTC/RTCTracing.h',
+          ],
+          'conditions': [
+            ['OS=="ios"', {
+              'sources': [
+                'objc/Framework/Classes/RTCCameraPreviewView.m',
+                'objc/Framework/Classes/RTCUIApplication.h',
+                'objc/Framework/Classes/RTCUIApplication.mm',
+                'objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h',
+              ],
+              'link_settings': {
+                'xcode_settings': {
+                  'OTHER_LDFLAGS': [
+                    '-framework AVFoundation',
+                  ],
+                },
+              },
+            }],  # OS=="ios"
+            ['build_with_chromium==0', {
+              'sources': [
+                'objc/Framework/Classes/RTCFileLogger.mm',
+                'objc/Framework/Headers/WebRTC/RTCFileLogger.h',
+              ],
+            }],
+          ],
+        },
+        {
+          'target_name': 'rtc_sdk_peerconnection_objc',
+          'type': 'static_library',
+          'includes': [ '../build/objc_common.gypi' ],
+          'dependencies': [
+            '<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
+            'rtc_sdk_common_objc',
+          ],
+          'include_dirs': [
+            'objc/Framework/Classes',
+            'objc/Framework/Headers',
+          ],
+          'direct_dependent_settings': {
+            'include_dirs': [
+              'objc/Framework/Classes',
+              'objc/Framework/Headers',
+            ],
+          },
+          'link_settings': {
+            'libraries': [
+              '-lstdc++',
+            ],
+          }, # link_settings
+          'sources': [
+            'objc/Framework/Classes/RTCAudioTrack+Private.h',
+            'objc/Framework/Classes/RTCAudioTrack.mm',
+            'objc/Framework/Classes/RTCConfiguration+Private.h',
+            'objc/Framework/Classes/RTCConfiguration.mm',
+            'objc/Framework/Classes/RTCDataChannel+Private.h',
+            'objc/Framework/Classes/RTCDataChannel.mm',
+            'objc/Framework/Classes/RTCDataChannelConfiguration+Private.h',
+            'objc/Framework/Classes/RTCDataChannelConfiguration.mm',
+            'objc/Framework/Classes/RTCIceCandidate+Private.h',
+            'objc/Framework/Classes/RTCIceCandidate.mm',
+            'objc/Framework/Classes/RTCIceServer+Private.h',
+            'objc/Framework/Classes/RTCIceServer.mm',
+            'objc/Framework/Classes/RTCMediaConstraints+Private.h',
+            'objc/Framework/Classes/RTCMediaConstraints.mm',
+            'objc/Framework/Classes/RTCMediaStream+Private.h',
+            'objc/Framework/Classes/RTCMediaStream.mm',
+            'objc/Framework/Classes/RTCMediaStreamTrack+Private.h',
+            'objc/Framework/Classes/RTCMediaStreamTrack.mm',
+            'objc/Framework/Classes/RTCOpenGLVideoRenderer.h',
+            'objc/Framework/Classes/RTCOpenGLVideoRenderer.mm',
+            'objc/Framework/Classes/RTCPeerConnection+DataChannel.mm',
+            'objc/Framework/Classes/RTCPeerConnection+Private.h',
+            'objc/Framework/Classes/RTCPeerConnection+Stats.mm',
+            'objc/Framework/Classes/RTCPeerConnection.mm',
+            'objc/Framework/Classes/RTCPeerConnectionFactory+Private.h',
+            'objc/Framework/Classes/RTCPeerConnectionFactory.mm',
+            'objc/Framework/Classes/RTCRtpCodecParameters+Private.h',
+            'objc/Framework/Classes/RTCRtpCodecParameters.mm',
+            'objc/Framework/Classes/RTCRtpEncodingParameters+Private.h',
+            'objc/Framework/Classes/RTCRtpEncodingParameters.mm',
+            'objc/Framework/Classes/RTCRtpParameters+Private.h',
+            'objc/Framework/Classes/RTCRtpParameters.mm',
+            'objc/Framework/Classes/RTCRtpReceiver+Private.h',
+            'objc/Framework/Classes/RTCRtpReceiver.mm',
+            'objc/Framework/Classes/RTCRtpSender+Private.h',
+            'objc/Framework/Classes/RTCRtpSender.mm',
+            'objc/Framework/Classes/RTCSessionDescription+Private.h',
+            'objc/Framework/Classes/RTCSessionDescription.mm',
+            'objc/Framework/Classes/RTCStatsReport+Private.h',
+            'objc/Framework/Classes/RTCStatsReport.mm',
+            'objc/Framework/Classes/RTCVideoFrame+Private.h',
+            'objc/Framework/Classes/RTCVideoFrame.mm',
+            'objc/Framework/Classes/RTCVideoRendererAdapter+Private.h',
+            'objc/Framework/Classes/RTCVideoRendererAdapter.h',
+            'objc/Framework/Classes/RTCVideoRendererAdapter.mm',
+            'objc/Framework/Classes/RTCVideoSource+Private.h',
+            'objc/Framework/Classes/RTCVideoSource.mm',
+            'objc/Framework/Classes/RTCVideoTrack+Private.h',
+            'objc/Framework/Classes/RTCVideoTrack.mm',
+            'objc/Framework/Headers/WebRTC/RTCAudioTrack.h',
+            'objc/Framework/Headers/WebRTC/RTCConfiguration.h',
+            'objc/Framework/Headers/WebRTC/RTCDataChannel.h',
+            'objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h',
+            'objc/Framework/Headers/WebRTC/RTCIceCandidate.h',
+            'objc/Framework/Headers/WebRTC/RTCIceServer.h',
+            'objc/Framework/Headers/WebRTC/RTCMediaConstraints.h',
+            'objc/Framework/Headers/WebRTC/RTCMediaStream.h',
+            'objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h',
+            'objc/Framework/Headers/WebRTC/RTCPeerConnection.h',
+            'objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h',
+            'objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h',
+            'objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h',
+            'objc/Framework/Headers/WebRTC/RTCRtpParameters.h',
+            'objc/Framework/Headers/WebRTC/RTCRtpReceiver.h',
+            'objc/Framework/Headers/WebRTC/RTCRtpSender.h',
+            'objc/Framework/Headers/WebRTC/RTCSessionDescription.h',
+            'objc/Framework/Headers/WebRTC/RTCStatsReport.h',
+            'objc/Framework/Headers/WebRTC/RTCVideoFrame.h',
+            'objc/Framework/Headers/WebRTC/RTCVideoRenderer.h',
+            'objc/Framework/Headers/WebRTC/RTCVideoSource.h',
+            'objc/Framework/Headers/WebRTC/RTCVideoTrack.h',
+          ], # sources
+          'conditions': [
+            ['OS=="ios"', {
+              'sources': [
+                'objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h',
+                'objc/Framework/Classes/RTCAVFoundationVideoSource.mm',
+                'objc/Framework/Classes/RTCEAGLVideoView.m',
+                'objc/Framework/Classes/avfoundationvideocapturer.h',
+                'objc/Framework/Classes/avfoundationvideocapturer.mm',
+                'objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h',
+                'objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h',
+              ],
+              'link_settings': {
+                'xcode_settings': {
+                  'OTHER_LDFLAGS': [
+                    '-framework CoreGraphics',
+                    '-framework GLKit',
+                    '-framework OpenGLES',
+                    '-framework QuartzCore',
+                  ],
+                },
+              },  # link_settings
+            }],  # OS=="ios"
+            ['OS=="mac"', {
+              'sources': [
+                'objc/Framework/Classes/RTCNSGLVideoView.m',
+                'objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h',
+              ],
+              'link_settings': {
+                'xcode_settings': {
+                  'OTHER_LDFLAGS': [
+                    '-framework OpenGL',
+                  ],
+                },
+              },
+            }],
+          ],  # conditions
+        },  # rtc_sdk_peerconnection_objc
+        {
+          'target_name': 'rtc_sdk_framework_objc',
+          'type': 'shared_library',
+          'product_name': 'WebRTC',
+          'mac_bundle': 1,
+          'includes': [ '../build/objc_common.gypi' ],
+          # Slightly hacky, but we need to re-declare files here that are C
+          # interfaces because otherwise they will be dead-stripped during
+          # linking (ObjC classes cannot be dead-stripped). We might consider
+          # just only using ObjC interfaces.
+          'sources': [
+            'objc/Framework/Classes/RTCFieldTrials.mm',
+            'objc/Framework/Classes/RTCLogging.mm',
+            'objc/Framework/Classes/RTCSSLAdapter.mm',
+            'objc/Framework/Classes/RTCTracing.mm',
+            'objc/Framework/Headers/WebRTC/RTCFieldTrials.h',
+            'objc/Framework/Headers/WebRTC/RTCLogging.h',
+            'objc/Framework/Headers/WebRTC/RTCSSLAdapter.h',
+            'objc/Framework/Headers/WebRTC/RTCTracing.h',
+            'objc/Framework/Headers/WebRTC/WebRTC.h',
+            'objc/Framework/Modules/module.modulemap',
+          ],
+          'mac_framework_headers': [
+            '<!@(find -E objc/Framework/Headers -regex ".*(h)")',
+          ],
+          'dependencies': [
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
+            'rtc_sdk_peerconnection_objc',
+          ],
+          'xcode_settings': {
+            'CODE_SIGNING_REQUIRED': 'NO',
+            'CODE_SIGN_IDENTITY': '',
+            'DEFINES_MODULE': 'YES',
+            'INFOPLIST_FILE': 'objc/Framework/Info.plist',
+            'LD_DYLIB_INSTALL_NAME': '@rpath/WebRTC.framework/WebRTC',
+            'MODULEMAP_FILE': '<(webrtc_root)/sdk/Framework/Modules/module.modulemap',
+          },
+          'link_settings': {
+            'xcode_settings': {
+              'OTHER_LDFLAGS': [
+                '-framework AVFoundation',
+                '-framework AudioToolbox',
+                '-framework CoreGraphics',
+                '-framework CoreMedia',
+                '-framework GLKit',
+                '-framework VideoToolbox',
+              ],
+            },
+          },  # link_settings
+          'conditions': [
+            # TODO(tkchin): Generate WebRTC.h based off of
+            # mac_framework_headers instead of hard-coding. Ok for now since we
+            # only really care about dynamic lib on iOS outside of chromium.
+            ['OS!="mac"', {
+              'mac_framework_headers!': [
+                'objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h',
+              ],
+            }],
+            ['build_with_chromium==1', {
+              'mac_framework_headers!': [
+                'objc/Framework/Headers/WebRTC/RTCFileLogger.h',
+              ],
+            }],
+          ],  # conditions
+        }, # rtc_sdk_framework_objc
+      ],  # targets
+    }],  # OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")
+  ],
+}
diff --git a/chromium/third_party/webrtc/sdk/sdk.gypi b/chromium/third_party/webrtc/sdk/sdk.gypi
new file mode 100644
index 00000000000..8f8ee97a9d2
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/sdk.gypi
@@ -0,0 +1,26 @@
+# Copyright 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'target_defaults': {
+    'configurations': {
+      'Profile': {
+        'xcode_settings': {
+          'DEBUG_INFORMARTION_FORMAT': 'dwarf-with-dsym',
+          # We manually strip using strip -S and strip -x. We need to run
+          # dsymutil ourselves so we need symbols around before that.
+          'DEPLOYMENT_POSTPROCESSING': 'NO',
+          'GCC_OPTIMIZATION_LEVEL': 's',
+          'GCC_SYMBOLS_PRIVATE_EXTERN': 'YES',
+          'STRIP_INSTALLED_PRODUCT': 'NO',
+          'USE_HEADERMAP': 'YES',
+        },
+      },
+    },
+  },
+}
diff --git a/chromium/third_party/webrtc/sdk/sdk_tests.gyp b/chromium/third_party/webrtc/sdk/sdk_tests.gyp
new file mode 100644
index 00000000000..70e8e7b901f
--- /dev/null
+++ b/chromium/third_party/webrtc/sdk/sdk_tests.gyp
@@ -0,0 +1,42 @@
+# Copyright 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [ '../build/common.gypi', ],
+  'conditions': [
+    ['OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")', {
+      'targets': [
+        {
+          'target_name': 'rtc_sdk_peerconnection_objc_tests',
+          'type': 'executable',
+          'includes': [
+            '../build/objc_common.gypi',
+          ],
+          'dependencies': [
+            '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
+            '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_peerconnection_objc',
+          ],
+          'sources': [
+            'objc/Framework/UnitTests/RTCConfigurationTest.mm',
+            'objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm',
+            'objc/Framework/UnitTests/RTCIceCandidateTest.mm',
+            'objc/Framework/UnitTests/RTCIceServerTest.mm',
+            'objc/Framework/UnitTests/RTCMediaConstraintsTest.mm',
+            'objc/Framework/UnitTests/RTCSessionDescriptionTest.mm',
+          ],
+          'xcode_settings': {
+            # |-ObjC| flag needed to make sure category method implementations
+            # are included:
+            # https://developer.apple.com/library/mac/qa/qa1490/_index.html
+            'OTHER_LDFLAGS': ['-ObjC'],
+          },
+        },
+      ],
+    }],  # OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")
+  ],
+}
diff --git a/chromium/third_party/webrtc/stream.h b/chromium/third_party/webrtc/stream.h
deleted file mode 100644
index 5afab0f200c..00000000000
--- a/chromium/third_party/webrtc/stream.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_STREAM_H_
-#define WEBRTC_STREAM_H_
-
-#include "webrtc/common_types.h"
-
-namespace webrtc {
-
-enum NetworkState {
-  kNetworkUp,
-  kNetworkDown,
-};
-
-// Common base class for streams.
-class Stream {
- public:
-  // Starts stream activity.
-  // When a stream is active, it can receive, process and deliver packets.
-  virtual void Start() = 0;
-  // Stops stream activity.
-  // When a stream is stopped, it can't receive, process or deliver packets.
-  virtual void Stop() = 0;
-  // Called to notify that network state has changed, so that the stream can
-  // respond, e.g. by pausing or resuming activity.
-  virtual void SignalNetworkState(NetworkState state) = 0;
-  // Called when a RTCP packet is received.
-  virtual bool DeliverRtcp(const uint8_t* packet, size_t length) = 0;
-
- protected:
-  virtual ~Stream() {}
-};
-
-// Common base class for receive streams.
-class ReceiveStream : public Stream {
- public:
-  // Called when a RTP packet is received.
-  virtual bool DeliverRtp(const uint8_t* packet,
-                          size_t length,
-                          const PacketTime& packet_time) = 0;
-};
-
-// Common base class for send streams.
-// A tag class that denotes send stream type.
-class SendStream : public Stream {};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_STREAM_H_
diff --git a/chromium/third_party/webrtc/supplement.gypi b/chromium/third_party/webrtc/supplement.gypi
index 2cd5400f224..f1974d73d83 100644
--- a/chromium/third_party/webrtc/supplement.gypi
+++ b/chromium/third_party/webrtc/supplement.gypi
@@ -13,9 +13,6 @@
     'build_with_chromium': 0,
     'conditions': [
       ['OS=="ios"', {
-        # Default to using BoringSSL on iOS.
-        'use_openssl%': 1,
-
         # Set target_subarch for if not already set. This is needed because the
         # Chromium iOS toolchain relies on target_subarch being set.
         'conditions': [
@@ -27,6 +24,11 @@
           }],
         ],
       }],
+      ['OS=="android"', {
+        # MJPEG capture is not used on Android. Disable to reduce
+        # libjingle_peerconnection_so file size.
+        'libyuv_disable_jpeg%': 1,
+      }],
     ],
   },
   'target_defaults': {
diff --git a/chromium/third_party/webrtc/system_wrappers/BUILD.gn b/chromium/third_party/webrtc/system_wrappers/BUILD.gn
index aff29de502a..66e6df5cc1a 100644
--- a/chromium/third_party/webrtc/system_wrappers/BUILD.gn
+++ b/chromium/third_party/webrtc/system_wrappers/BUILD.gn
@@ -34,7 +34,6 @@ static_library("system_wrappers") {
     "include/static_instance.h",
     "include/stl_util.h",
     "include/stringize_macros.h",
-    "include/tick_util.h",
     "include/timestamp_extrapolator.h",
     "include/trace.h",
     "include/utf_util_win.h",
@@ -65,7 +64,6 @@ static_library("system_wrappers") {
     "source/rw_lock_winxp_win.h",
     "source/sleep.cc",
     "source/sort.cc",
-    "source/tick_util.cc",
     "source/timestamp_extrapolator.cc",
     "source/trace_impl.cc",
     "source/trace_impl.h",
diff --git a/chromium/third_party/webrtc/system_wrappers/include/aligned_malloc.h b/chromium/third_party/webrtc/system_wrappers/include/aligned_malloc.h
index 277abec020d..bdd82ccfd8c 100644
--- a/chromium/third_party/webrtc/system_wrappers/include/aligned_malloc.h
+++ b/chromium/third_party/webrtc/system_wrappers/include/aligned_malloc.h
@@ -46,8 +46,8 @@ T* AlignedMalloc(size_t size, size_t alignment) {
   return reinterpret_cast<T*>(AlignedMalloc(size, alignment));
 }
 
-// Deleter for use with scoped_ptr. E.g., use as
-//   scoped_ptr<Foo, AlignedFreeDeleter> foo;
+// Deleter for use with unique_ptr. E.g., use as
+//   std::unique_ptr<Foo, AlignedFreeDeleter> foo;
 struct AlignedFreeDeleter {
   inline void operator()(void* ptr) const {
     AlignedFree(ptr);
diff --git a/chromium/third_party/webrtc/system_wrappers/include/clock.h b/chromium/third_party/webrtc/system_wrappers/include/clock.h
index f443057bea3..a209770261f 100644
--- a/chromium/third_party/webrtc/system_wrappers/include/clock.h
+++ b/chromium/third_party/webrtc/system_wrappers/include/clock.h
@@ -11,7 +11,8 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
 #define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 
@@ -76,7 +77,7 @@ class SimulatedClock : public Clock {
 
  private:
   int64_t time_us_;
-  rtc::scoped_ptr<RWLockWrapper> lock_;
+  std::unique_ptr<RWLockWrapper> lock_;
 };
 
 };  // namespace webrtc
diff --git a/chromium/third_party/webrtc/system_wrappers/include/data_log_impl.h b/chromium/third_party/webrtc/system_wrappers/include/data_log_impl.h
index 35519609b99..6d59fa8c66b 100644
--- a/chromium/third_party/webrtc/system_wrappers/include/data_log_impl.h
+++ b/chromium/third_party/webrtc/system_wrappers/include/data_log_impl.h
@@ -18,12 +18,12 @@
 #define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_DATA_LOG_IMPL_H_
 
 #include <map>
+#include <memory>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -139,16 +139,16 @@ class DataLogImpl {
 
   // Collection of tables indexed by the table name as std::string.
   typedef std::map<std::string, LogTable*> TableMap;
-  typedef rtc::scoped_ptr<CriticalSectionWrapper> CritSectScopedPtr;
+  typedef std::unique_ptr<CriticalSectionWrapper> CritSectScopedPtr;
 
   static CritSectScopedPtr  crit_sect_;
   static DataLogImpl*       instance_;
   int                       counter_;
   TableMap                  tables_;
   EventWrapper*             flush_event_;
-  // This is a scoped_ptr so that we don't have to create threads in the no-op
+  // This is a unique_ptr so that we don't have to create threads in the no-op
   // impl.
-  rtc::scoped_ptr<rtc::PlatformThread> file_writer_thread_;
+  std::unique_ptr<rtc::PlatformThread> file_writer_thread_;
   RWLockWrapper*            tables_lock_;
 };
 
diff --git a/chromium/third_party/webrtc/system_wrappers/include/metrics.h b/chromium/third_party/webrtc/system_wrappers/include/metrics.h
index 1576f919107..d5e549285fa 100644
--- a/chromium/third_party/webrtc/system_wrappers/include/metrics.h
+++ b/chromium/third_party/webrtc/system_wrappers/include/metrics.h
@@ -69,6 +69,9 @@
 #define RTC_HISTOGRAM_COUNTS_200(name, sample) \
   RTC_HISTOGRAM_COUNTS(name, sample, 1, 200, 50)
 
+#define RTC_HISTOGRAM_COUNTS_500(name, sample) \
+  RTC_HISTOGRAM_COUNTS(name, sample, 1, 500, 50)
+
 #define RTC_HISTOGRAM_COUNTS_1000(name, sample) \
   RTC_HISTOGRAM_COUNTS(name, sample, 1, 1000, 50)
 
@@ -89,6 +92,9 @@
 #define RTC_LOGGED_HISTOGRAM_COUNTS_200(name, sample) \
   RTC_LOGGED_HISTOGRAM_COUNTS(name, sample, 1, 200, 50)
 
+#define RTC_LOGGED_HISTOGRAM_COUNTS_500(name, sample) \
+  RTC_LOGGED_HISTOGRAM_COUNTS(name, sample, 1, 500, 50)
+
 #define RTC_LOGGED_HISTOGRAM_COUNTS_1000(name, sample) \
   RTC_LOGGED_HISTOGRAM_COUNTS(name, sample, 1, 1000, 50)
 
@@ -175,6 +181,10 @@
   RTC_HISTOGRAMS_COMMON(index, name, sample, \
       RTC_HISTOGRAM_COUNTS(name, sample, 1, 200, 50))
 
+#define RTC_HISTOGRAMS_COUNTS_500(index, name, sample) \
+  RTC_HISTOGRAMS_COMMON(index, name, sample, \
+      RTC_HISTOGRAM_COUNTS(name, sample, 1, 500, 50))
+
 #define RTC_HISTOGRAMS_COUNTS_1000(index, name, sample) \
   RTC_HISTOGRAMS_COMMON(index, name, sample, \
       RTC_HISTOGRAM_COUNTS(name, sample, 1, 1000, 50))
@@ -204,6 +214,10 @@
   RTC_HISTOGRAMS_COMMON(index, name, sample, \
       RTC_LOGGED_HISTOGRAM_COUNTS(name, sample, 1, 200, 50))
 
+#define RTC_LOGGED_HISTOGRAMS_COUNTS_500(index, name, sample) \
+  RTC_HISTOGRAMS_COMMON(index, name, sample, \
+      RTC_LOGGED_HISTOGRAM_COUNTS(name, sample, 1, 500, 50))
+
 #define RTC_LOGGED_HISTOGRAMS_COUNTS_1000(index, name, sample) \
   RTC_HISTOGRAMS_COMMON(index, name, sample, \
       RTC_LOGGED_HISTOGRAM_COUNTS(name, sample, 1, 1000, 50))
diff --git a/chromium/third_party/webrtc/system_wrappers/include/tick_util.h b/chromium/third_party/webrtc/system_wrappers/include/tick_util.h
deleted file mode 100644
index 52f9b4ae4d3..00000000000
--- a/chromium/third_party/webrtc/system_wrappers/include/tick_util.h
+++ /dev/null
@@ -1,190 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for polling elapsed time in ms and us.
-// The implementation works in the tick domain which can be mapped over to the
-// time domain.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TICK_UTIL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TICK_UTIL_H_
-
-#if _WIN32
-// Note: The Windows header must always be included before mmsystem.h
-#include <windows.h>
-#include <mmsystem.h>
-#elif WEBRTC_LINUX
-#include <time.h>
-#elif WEBRTC_MAC
-#include <mach/mach_time.h>
-#include <string.h>
-#else
-#include <sys/time.h>
-#include <time.h>
-#endif
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class TickInterval;
-
-// Class representing the current time.
-class TickTime {
- public:
-  TickTime();
-  explicit TickTime(int64_t ticks);
-
-  // Current time in the tick domain.
-  static TickTime Now();
-
-  // Now in the time domain in ms.
-  static int64_t MillisecondTimestamp();
-
-  // Now in the time domain in us.
-  static int64_t MicrosecondTimestamp();
-
-  // Returns the number of ticks in the tick domain.
-  int64_t Ticks() const;
-
-  static int64_t MillisecondsToTicks(const int64_t ms);
-
-  static int64_t TicksToMilliseconds(const int64_t ticks);
-
-  static int64_t TicksToMicroseconds(const int64_t ticks);
-
-  // Returns a TickTime that is ticks later than the passed TickTime.
-  friend TickTime operator+(const TickTime lhs, const int64_t ticks);
-  TickTime& operator+=(const int64_t& ticks);
-
-  // Returns a TickInterval that is the difference in ticks beween rhs and lhs.
-  friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
-
- private:
-  static int64_t QueryOsForTicks();
-
-  int64_t ticks_;
-};
-
-// Represents a time delta in ticks.
-class TickInterval {
- public:
-  TickInterval();
-  explicit TickInterval(int64_t interval);
-
-  int64_t Milliseconds() const;
-  int64_t Microseconds() const;
-
-  // Returns the sum of two TickIntervals as a TickInterval.
-  friend TickInterval operator+(const TickInterval& lhs,
-                                const TickInterval& rhs);
-  TickInterval& operator+=(const TickInterval& rhs);
-
-  // Returns a TickInterval corresponding to rhs - lhs.
-  friend TickInterval operator-(const TickInterval& lhs,
-                                const TickInterval& rhs);
-  TickInterval& operator-=(const TickInterval& rhs);
-
-  friend bool operator>(const TickInterval& lhs, const TickInterval& rhs);
-  friend bool operator<=(const TickInterval& lhs, const TickInterval& rhs);
-  friend bool operator<(const TickInterval& lhs, const TickInterval& rhs);
-  friend bool operator>=(const TickInterval& lhs, const TickInterval& rhs);
-
- private:
-  friend class TickTime;
-  friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
-
- private:
-  int64_t interval_;
-};
-
-inline int64_t TickInterval::Milliseconds() const {
-  return TickTime::TicksToMilliseconds(interval_);
-}
-
-inline int64_t TickInterval::Microseconds() const {
-  return TickTime::TicksToMicroseconds(interval_);
-}
-
-inline TickInterval operator+(const TickInterval& lhs,
-                              const TickInterval& rhs) {
-  return TickInterval(lhs.interval_ + rhs.interval_);
-}
-
-inline TickInterval operator-(const TickInterval& lhs,
-                              const TickInterval& rhs) {
-  return TickInterval(lhs.interval_ - rhs.interval_);
-}
-
-inline TickInterval operator-(const TickTime& lhs, const TickTime& rhs) {
-  return TickInterval(lhs.ticks_ - rhs.ticks_);
-}
-
-inline TickTime operator+(const TickTime lhs, const int64_t ticks) {
-  TickTime time = lhs;
-  time.ticks_ += ticks;
-  return time;
-}
-
-inline bool operator>(const TickInterval& lhs, const TickInterval& rhs) {
-  return lhs.interval_ > rhs.interval_;
-}
-
-inline bool operator<=(const TickInterval& lhs, const TickInterval& rhs) {
-  return lhs.interval_ <= rhs.interval_;
-}
-
-inline bool operator<(const TickInterval& lhs, const TickInterval& rhs) {
-  return lhs.interval_ <= rhs.interval_;
-}
-
-inline bool operator>=(const TickInterval& lhs, const TickInterval& rhs) {
-  return lhs.interval_ >= rhs.interval_;
-}
-
-inline TickTime::TickTime()
-    : ticks_(0) {
-}
-
-inline TickTime::TickTime(int64_t ticks)
-    : ticks_(ticks) {
-}
-
-inline TickTime TickTime::Now() {
-  return TickTime(QueryOsForTicks());
-}
-
-inline int64_t TickTime::Ticks() const {
-  return ticks_;
-}
-
-inline TickTime& TickTime::operator+=(const int64_t& ticks) {
-  ticks_ += ticks;
-  return *this;
-}
-
-inline TickInterval::TickInterval() : interval_(0) {
-}
-
-inline TickInterval::TickInterval(const int64_t interval)
-  : interval_(interval) {
-}
-
-inline TickInterval& TickInterval::operator+=(const TickInterval& rhs) {
-  interval_ += rhs.interval_;
-  return *this;
-}
-
-inline TickInterval& TickInterval::operator-=(const TickInterval& rhs) {
-  interval_ -= rhs.interval_;
-  return *this;
-}
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_TICK_UTIL_H_
diff --git a/chromium/third_party/webrtc/system_wrappers/include/utf_util_win.h b/chromium/third_party/webrtc/system_wrappers/include/utf_util_win.h
index 0e3f2d01c60..ac91fe30fe1 100644
--- a/chromium/third_party/webrtc/system_wrappers/include/utf_util_win.h
+++ b/chromium/third_party/webrtc/system_wrappers/include/utf_util_win.h
@@ -15,16 +15,17 @@
 
 #ifdef WIN32
 #include <windows.h>
+
+#include <memory>
 #include <string>
 
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
 inline std::wstring ToUtf16(const char* utf8, size_t len) {
   int len16 = ::MultiByteToWideChar(CP_UTF8, 0, utf8, static_cast<int>(len),
                                     NULL, 0);
-  rtc::scoped_ptr<wchar_t[]> ws(new wchar_t[len16]);
+  std::unique_ptr<wchar_t[]> ws(new wchar_t[len16]);
   ::MultiByteToWideChar(CP_UTF8, 0, utf8, static_cast<int>(len), ws.get(),
                         len16);
   return std::wstring(ws.get(), len16);
@@ -37,7 +38,7 @@ inline std::wstring ToUtf16(const std::string& str) {
 inline std::string ToUtf8(const wchar_t* wide, size_t len) {
   int len8 = ::WideCharToMultiByte(CP_UTF8, 0, wide, static_cast<int>(len),
                                    NULL, 0, NULL, NULL);
-  rtc::scoped_ptr<char[]> ns(new char[len8]);
+  std::unique_ptr<char[]> ns(new char[len8]);
   ::WideCharToMultiByte(CP_UTF8, 0, wide, static_cast<int>(len), ns.get(), len8,
                         NULL, NULL);
   return std::string(ns.get(), len8);
diff --git a/chromium/third_party/webrtc/system_wrappers/source/aligned_malloc_unittest.cc b/chromium/third_party/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
index 3933c2ac05d..ed6cd424db9 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
+++ b/chromium/third_party/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/system_wrappers/include/aligned_malloc.h"
 
+#include <memory>
+
 #if _WIN32
 #include <windows.h>
 #else
@@ -17,14 +19,13 @@
 #endif
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Returns true if |size| and |alignment| are valid combinations.
 bool CorrectUsage(size_t size, size_t alignment) {
-  rtc::scoped_ptr<char, AlignedFreeDeleter> scoped(
+  std::unique_ptr<char, AlignedFreeDeleter> scoped(
       static_cast<char*>(AlignedMalloc(size, alignment)));
   if (scoped.get() == NULL) {
     return false;
@@ -37,7 +38,7 @@ TEST(AlignedMalloc, GetRightAlign) {
   const size_t size = 100;
   const size_t alignment = 32;
   const size_t left_misalignment = 1;
-  rtc::scoped_ptr<char, AlignedFreeDeleter> scoped(
+  std::unique_ptr<char, AlignedFreeDeleter> scoped(
       static_cast<char*>(AlignedMalloc(size, alignment)));
   EXPECT_TRUE(scoped.get() != NULL);
   const uintptr_t aligned_address = reinterpret_cast<uintptr_t> (scoped.get());
diff --git a/chromium/third_party/webrtc/system_wrappers/source/clock.cc b/chromium/third_party/webrtc/system_wrappers/source/clock.cc
index 926b95a3dfd..05dabd8538c 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/clock.cc
+++ b/chromium/third_party/webrtc/system_wrappers/source/clock.cc
@@ -20,8 +20,8 @@
 #endif
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ -37,13 +37,13 @@ class RealTimeClock : public Clock {
   // Return a timestamp in milliseconds relative to some arbitrary source; the
   // source is fixed for this clock.
   int64_t TimeInMilliseconds() const override {
-    return TickTime::MillisecondTimestamp();
+    return rtc::TimeMillis();
   }
 
   // Return a timestamp in microseconds relative to some arbitrary source; the
   // source is fixed for this clock.
   int64_t TimeInMicroseconds() const override {
-    return TickTime::MicrosecondTimestamp();
+    return rtc::TimeMicros();
   }
 
   // Retrieve an NTP absolute timestamp in seconds and fractions of a second.
diff --git a/chromium/third_party/webrtc/system_wrappers/source/condition_variable_unittest.cc b/chromium/third_party/webrtc/system_wrappers/source/condition_variable_unittest.cc
index 4b1b6cc6089..5fc6bc6b73e 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/condition_variable_unittest.cc
+++ b/chromium/third_party/webrtc/system_wrappers/source/condition_variable_unittest.cc
@@ -17,9 +17,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -192,9 +191,9 @@ TEST(CondVarWaitTest, WaitingWaits) {
   InitializeCriticalSection(&crit_sect);
   ConditionVariableEventWin cond_var;
   EnterCriticalSection(&crit_sect);
-  int64_t start_ms = TickTime::MillisecondTimestamp();
+  int64_t start_ms = rtc::TimeMillis();
   EXPECT_FALSE(cond_var.SleepCS(&crit_sect, kVeryShortWaitMs));
-  int64_t end_ms = TickTime::MillisecondTimestamp();
+  int64_t end_ms = rtc::TimeMillis();
   EXPECT_LE(start_ms + kVeryShortWaitMs, end_ms)
       << "actual elapsed:" << end_ms - start_ms;
   LeaveCriticalSection(&crit_sect);
diff --git a/chromium/third_party/webrtc/system_wrappers/source/event_timer_posix.h b/chromium/third_party/webrtc/system_wrappers/source/event_timer_posix.h
index af3715ee8b2..599eb55ee16 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/event_timer_posix.h
+++ b/chromium/third_party/webrtc/system_wrappers/source/event_timer_posix.h
@@ -13,6 +13,8 @@
 
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 
+#include <memory>
+
 #include <pthread.h>
 #include <time.h>
 
@@ -49,9 +51,9 @@ class EventTimerPosix : public EventTimerWrapper {
   pthread_mutex_t mutex_;
   bool event_set_;
 
-  // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
-  rtc::scoped_ptr<rtc::PlatformThread> timer_thread_;
-  rtc::scoped_ptr<EventTimerPosix> timer_event_;
+  // TODO(pbos): Remove unique_ptr and use PlatformThread directly.
+  std::unique_ptr<rtc::PlatformThread> timer_thread_;
+  std::unique_ptr<EventTimerPosix> timer_event_;
   timespec       created_at_;
 
   bool          periodic_;
diff --git a/chromium/third_party/webrtc/system_wrappers/source/file_impl.h b/chromium/third_party/webrtc/system_wrappers/source/file_impl.h
index 06ba58200bb..51103d648ba 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/file_impl.h
+++ b/chromium/third_party/webrtc/system_wrappers/source/file_impl.h
@@ -13,7 +13,8 @@
 
 #include <stdio.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 namespace webrtc {
@@ -52,7 +53,7 @@ class FileWrapperImpl : public FileWrapper {
   int CloseFileImpl();
   int FlushImpl();
 
-  rtc::scoped_ptr<RWLockWrapper> rw_lock_;
+  std::unique_ptr<RWLockWrapper> rw_lock_;
 
   FILE* id_;
   bool managed_file_handle_;
diff --git a/chromium/third_party/webrtc/system_wrappers/source/logging_unittest.cc b/chromium/third_party/webrtc/system_wrappers/source/logging_unittest.cc
index 695b03f93a0..118c3425bf2 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/logging_unittest.cc
+++ b/chromium/third_party/webrtc/system_wrappers/source/logging_unittest.cc
@@ -13,7 +13,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/event.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp.cc b/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp.cc
index 706d861b55c..6504737fd54 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp.cc
+++ b/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp.cc
@@ -71,8 +71,9 @@ bool UpdateRtcpList(uint32_t ntp_secs,
 
   for (RtcpList::iterator it = rtcp_list->begin();
        it != rtcp_list->end(); ++it) {
-    if (measurement.ntp_secs == (*it).ntp_secs &&
-        measurement.ntp_frac == (*it).ntp_frac) {
+    if ((measurement.ntp_secs == (*it).ntp_secs &&
+         measurement.ntp_frac == (*it).ntp_frac) ||
+        (measurement.rtp_timestamp == (*it).rtp_timestamp)) {
       // This RTCP has already been added to the list.
       return true;
     }
diff --git a/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc b/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
index d2929f5cbc0..78e4a52716d 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
+++ b/chromium/third_party/webrtc/system_wrappers/source/rtp_to_ntp_unittest.cc
@@ -135,4 +135,44 @@ TEST(WrapAroundTests, OldRtp_OldRtcpWrapped) {
   int64_t timestamp_in_ms = -1;
   EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, &timestamp_in_ms));
 }
+
+TEST(UpdateRtcpListTests, InjectRtcpSrWithEqualNtp) {
+  RtcpList rtcp;
+  uint32_t ntp_sec = 0;
+  uint32_t ntp_frac = 2;
+  uint32_t timestamp = 0x12345678;
+
+  bool new_sr;
+  EXPECT_TRUE(UpdateRtcpList(ntp_sec, ntp_frac, timestamp, &rtcp, &new_sr));
+  EXPECT_TRUE(new_sr);
+
+  ++timestamp;
+  EXPECT_TRUE(UpdateRtcpList(ntp_sec, ntp_frac, timestamp, &rtcp, &new_sr));
+  EXPECT_FALSE(new_sr);
+}
+
+TEST(UpdateRtcpListTests, InjectRtcpSrWithEqualTimestamp) {
+  RtcpList rtcp;
+  uint32_t ntp_sec = 0;
+  uint32_t ntp_frac = 2;
+  uint32_t timestamp = 0x12345678;
+
+  bool new_sr;
+  EXPECT_TRUE(UpdateRtcpList(ntp_sec, ntp_frac, timestamp, &rtcp, &new_sr));
+  EXPECT_TRUE(new_sr);
+
+  ++ntp_frac;
+  EXPECT_TRUE(UpdateRtcpList(ntp_sec, ntp_frac, timestamp, &rtcp, &new_sr));
+  EXPECT_FALSE(new_sr);
+}
+
+TEST(UpdateRtcpListTests, InjectRtcpSrWithZeroNtpFails) {
+  RtcpList rtcp;
+  uint32_t ntp_sec = 0;
+  uint32_t ntp_frac = 0;
+  uint32_t timestamp = 0x12345678;
+
+  bool new_sr;
+  EXPECT_FALSE(UpdateRtcpList(ntp_sec, ntp_frac, timestamp, &rtcp, &new_sr));
+}
 };  // namespace webrtc
diff --git a/chromium/third_party/webrtc/system_wrappers/source/tick_util.cc b/chromium/third_party/webrtc/system_wrappers/source/tick_util.cc
deleted file mode 100644
index 0485e429211..00000000000
--- a/chromium/third_party/webrtc/system_wrappers/source/tick_util.cc
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/system_wrappers/include/tick_util.h"
-
-#include "webrtc/base/timeutils.h"
-
-namespace webrtc {
-
-int64_t TickTime::MillisecondTimestamp() {
-  return TicksToMilliseconds(TickTime::Now().Ticks());
-}
-
-int64_t TickTime::MicrosecondTimestamp() {
-  return TicksToMicroseconds(TickTime::Now().Ticks());
-}
-
-int64_t TickTime::MillisecondsToTicks(const int64_t ms) {
-  return ms * rtc::kNumNanosecsPerMillisec;
-}
-
-int64_t TickTime::TicksToMilliseconds(const int64_t ticks) {
-  return ticks / rtc::kNumNanosecsPerMillisec;
-}
-
-int64_t TickTime::TicksToMicroseconds(const int64_t ticks) {
-  return ticks / rtc::kNumNanosecsPerMicrosec;
-}
-
-// Gets the native system tick count, converted to nanoseconds.
-int64_t TickTime::QueryOsForTicks() {
-  return rtc::TimeNanos();
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/system_wrappers/source/trace_impl.h b/chromium/third_party/webrtc/system_wrappers/source/trace_impl.h
index c6d81d5b0bf..182f5809a50 100644
--- a/chromium/third_party/webrtc/system_wrappers/source/trace_impl.h
+++ b/chromium/third_party/webrtc/system_wrappers/source/trace_impl.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
 #define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
 
+#include <memory>
+
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/static_instance.h"
@@ -97,7 +98,7 @@ class TraceImpl : public Trace {
   uint32_t row_count_text_ GUARDED_BY(crit_);
   uint32_t file_count_text_ GUARDED_BY(crit_);
 
-  const rtc::scoped_ptr<FileWrapper> trace_file_ GUARDED_BY(crit_);
+  const std::unique_ptr<FileWrapper> trace_file_ GUARDED_BY(crit_);
   rtc::CriticalSection crit_;
 };
 
diff --git a/chromium/third_party/webrtc/system_wrappers/system_wrappers.gyp b/chromium/third_party/webrtc/system_wrappers/system_wrappers.gyp
index cd6d3ea6714..b61ecf28ee1 100644
--- a/chromium/third_party/webrtc/system_wrappers/system_wrappers.gyp
+++ b/chromium/third_party/webrtc/system_wrappers/system_wrappers.gyp
@@ -42,7 +42,6 @@
         'include/static_instance.h',
         'include/stl_util.h',
         'include/stringize_macros.h',
-        'include/tick_util.h',
         'include/timestamp_extrapolator.h',
         'include/trace.h',
         'include/utf_util_win.h',
@@ -77,7 +76,6 @@
         'source/rw_lock_winxp_win.h',
         'source/sleep.cc',
         'source/sort.cc',
-        'source/tick_util.cc',
         'source/timestamp_extrapolator.cc',
         'source/trace_impl.cc',
         'source/trace_impl.h',
diff --git a/chromium/third_party/webrtc/system_wrappers/system_wrappers_tests.gyp b/chromium/third_party/webrtc/system_wrappers/system_wrappers_tests.gyp
index 863818b1bf5..174c96c948f 100644
--- a/chromium/third_party/webrtc/system_wrappers/system_wrappers_tests.gyp
+++ b/chromium/third_party/webrtc/system_wrappers/system_wrappers_tests.gyp
@@ -56,7 +56,7 @@
     },
   ],
   'conditions': [
-    ['include_tests==1 and OS=="android"', {
+    ['OS=="android"', {
       'targets': [
         {
           'target_name': 'system_wrappers_unittests_apk_target',
@@ -66,7 +66,28 @@
           ],
         },
       ],
-    }],
+      'conditions': [
+        ['test_isolation_mode != "noop"',
+          {
+            'targets': [
+              {
+                'target_name': 'system_wrappers_unittests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):system_wrappers_unittests_apk',
+                ],
+                'includes': [
+                  '../build/isolate.gypi',
+                ],
+                'sources': [
+                  'system_wrappers_unittests_apk.isolate',
+                ],
+              },
+            ],
+          },
+        ],
+      ],
+    }],  # OS=="android"
     ['test_isolation_mode != "noop"', {
       'targets': [
         {
diff --git a/chromium/third_party/webrtc/system_wrappers/system_wrappers_unittests_apk.isolate b/chromium/third_party/webrtc/system_wrappers/system_wrappers_unittests_apk.isolate
new file mode 100644
index 00000000000..974064a71f8
--- /dev/null
+++ b/chromium/third_party/webrtc/system_wrappers/system_wrappers_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'system_wrappers_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_system_wrappers_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/system_wrappers_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_system_wrappers_unittests',
+      'system_wrappers_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/system_wrappers/test/TestSort/TestSort.cc b/chromium/third_party/webrtc/system_wrappers/test/TestSort/TestSort.cc
index b2b9f857553..8c585abb674 100644
--- a/chromium/third_party/webrtc/system_wrappers/test/TestSort/TestSort.cc
+++ b/chromium/third_party/webrtc/system_wrappers/test/TestSort/TestSort.cc
@@ -13,8 +13,8 @@
 
 #include <algorithm>
 
+#include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/include/sort.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 // Excellent work polluting the global namespace Visual Studio...
 #undef max
@@ -144,7 +144,7 @@ void RunSortTest(webrtc::Type sortType, bool keySort)
         printf("Running %s Sort() tests...\n", TypeEnumToString(sortType));
     }
 
-    TickInterval accTicks;
+    int64_t accTicks;
     for (int i = 0; i < NumOfTests; i++)
     {
         for (int j = 0; j < DataLength; j++)
@@ -159,7 +159,7 @@ void RunSortTest(webrtc::Type sortType, bool keySort)
         memcpy(keyRef, key, sizeof(key));
 
         retVal = 0;
-        TickTime t0 = TickTime::Now();
+        int64_t t0 = rtc::TimeNanos();
         if (keySort)
         {
             retVal = webrtc::KeySort(data, key, DataLength, sizeof(LotsOfData<KeyType>),
@@ -176,7 +176,7 @@ void RunSortTest(webrtc::Type sortType, bool keySort)
             //std::sort(key, key + DataLength);
             //qsort(key, DataLength, sizeof(KeyType), Compare<KeyType>);
         }
-        TickTime t1 = TickTime::Now();
+        int64_t t1 = rtc::TimeNanos();
         accTicks += (t1 - t0);
 
         if (retVal != 0)
@@ -236,7 +236,7 @@ void RunSortTest(webrtc::Type sortType, bool keySort)
 
     printf("Compliance test passed over %d iterations\n", NumOfTests);
 
-    int64_t executeTime = accTicks.Milliseconds();
+    int64_t executeTime = accTicks / rtc::kNumNanosecsPerMillisec;
     printf("Execute time: %.2f s\n\n", (float)executeTime / 1000);
 }
 
diff --git a/chromium/third_party/webrtc/test/call_test.cc b/chromium/third_party/webrtc/test/call_test.cc
index a9a502d92c5..768c007c3cb 100644
--- a/chromium/third_party/webrtc/test/call_test.cc
+++ b/chromium/third_party/webrtc/test/call_test.cc
@@ -15,7 +15,6 @@
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_network.h"
 
 namespace webrtc {
 namespace test {
@@ -78,9 +77,6 @@ void CallTest::RunBaseTest(BaseTest* test) {
   if (test->ShouldCreateReceivers()) {
     CreateMatchingReceiveConfigs(receive_transport_.get());
   }
-  if (num_audio_streams_ > 0)
-    SetupVoiceEngineTransports(send_transport_.get(), receive_transport_.get());
-
   if (num_video_streams_ > 0) {
     test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
                              &video_encoder_config_);
@@ -275,6 +271,10 @@ void CallTest::CreateVideoStreams() {
   }
 }
 
+void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
+  frame_generator_capturer_->SetFakeRotation(rotation);
+}
+
 void CallTest::CreateAudioStreams() {
   audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
   for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
@@ -306,7 +306,6 @@ void CallTest::CreateVoiceEngines() {
   CreateFakeAudioDevices();
   voe_send_.voice_engine = VoiceEngine::Create();
   voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
-  voe_send_.network = VoENetwork::GetInterface(voe_send_.voice_engine);
   voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
   EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr));
   Config voe_config;
@@ -316,35 +315,17 @@ void CallTest::CreateVoiceEngines() {
 
   voe_recv_.voice_engine = VoiceEngine::Create();
   voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
-  voe_recv_.network = VoENetwork::GetInterface(voe_recv_.voice_engine);
   voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
   EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr));
   voe_recv_.channel_id = voe_recv_.base->CreateChannel();
   EXPECT_GE(voe_recv_.channel_id, 0);
 }
 
-void CallTest::SetupVoiceEngineTransports(PacketTransport* send_transport,
-                                          PacketTransport* recv_transport) {
-  voe_send_.transport_adapter.reset(
-      new internal::TransportAdapter(send_transport));
-  voe_send_.transport_adapter->Enable();
-  EXPECT_EQ(0, voe_send_.network->RegisterExternalTransport(
-                   voe_send_.channel_id, *voe_send_.transport_adapter.get()));
-
-  voe_recv_.transport_adapter.reset(
-      new internal::TransportAdapter(recv_transport));
-  voe_recv_.transport_adapter->Enable();
-  EXPECT_EQ(0, voe_recv_.network->RegisterExternalTransport(
-                   voe_recv_.channel_id, *voe_recv_.transport_adapter.get()));
-}
-
 void CallTest::DestroyVoiceEngines() {
   voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
   voe_recv_.channel_id = -1;
   voe_recv_.base->Release();
   voe_recv_.base = nullptr;
-  voe_recv_.network->Release();
-  voe_recv_.network = nullptr;
   voe_recv_.codec->Release();
   voe_recv_.codec = nullptr;
 
@@ -352,8 +333,6 @@ void CallTest::DestroyVoiceEngines() {
   voe_send_.channel_id = -1;
   voe_send_.base->Release();
   voe_send_.base = nullptr;
-  voe_send_.network->Release();
-  voe_send_.network = nullptr;
   voe_send_.codec->Release();
   voe_send_.codec = nullptr;
 
diff --git a/chromium/third_party/webrtc/test/call_test.h b/chromium/third_party/webrtc/test/call_test.h
index eb33c0f1c40..ebc2bb2a544 100644
--- a/chromium/third_party/webrtc/test/call_test.h
+++ b/chromium/third_party/webrtc/test/call_test.h
@@ -14,7 +14,6 @@
 #include <vector>
 
 #include "webrtc/call.h"
-#include "webrtc/call/transport_adapter.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
@@ -25,7 +24,6 @@ namespace webrtc {
 
 class VoEBase;
 class VoECodec;
-class VoENetwork;
 
 namespace test {
 
@@ -80,25 +78,26 @@ class CallTest : public ::testing::Test {
   void Start();
   void Stop();
   void DestroyStreams();
+  void SetFakeVideoCaptureRotation(VideoRotation rotation);
 
   Clock* const clock_;
 
-  rtc::scoped_ptr<Call> sender_call_;
-  rtc::scoped_ptr<PacketTransport> send_transport_;
+  std::unique_ptr<Call> sender_call_;
+  std::unique_ptr<PacketTransport> send_transport_;
   VideoSendStream::Config video_send_config_;
   VideoEncoderConfig video_encoder_config_;
   VideoSendStream* video_send_stream_;
   AudioSendStream::Config audio_send_config_;
   AudioSendStream* audio_send_stream_;
 
-  rtc::scoped_ptr<Call> receiver_call_;
-  rtc::scoped_ptr<PacketTransport> receive_transport_;
+  std::unique_ptr<Call> receiver_call_;
+  std::unique_ptr<PacketTransport> receive_transport_;
   std::vector<VideoReceiveStream::Config> video_receive_configs_;
   std::vector<VideoReceiveStream*> video_receive_streams_;
   std::vector<AudioReceiveStream::Config> audio_receive_configs_;
   std::vector<AudioReceiveStream*> audio_receive_streams_;
 
-  rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
+  std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
   test::FakeEncoder fake_encoder_;
   std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
   size_t num_video_streams_;
@@ -112,30 +111,24 @@ class CallTest : public ::testing::Test {
     VoiceEngineState()
         : voice_engine(nullptr),
           base(nullptr),
-          network(nullptr),
           codec(nullptr),
-          channel_id(-1),
-          transport_adapter(nullptr) {}
+          channel_id(-1) {}
 
     VoiceEngine* voice_engine;
     VoEBase* base;
-    VoENetwork* network;
     VoECodec* codec;
     int channel_id;
-    rtc::scoped_ptr<internal::TransportAdapter> transport_adapter;
   };
 
   void CreateVoiceEngines();
-  void SetupVoiceEngineTransports(PacketTransport* send_transport,
-                                  PacketTransport* recv_transport);
   void DestroyVoiceEngines();
 
   VoiceEngineState voe_send_;
   VoiceEngineState voe_recv_;
 
   // The audio devices must outlive the voice engines.
-  rtc::scoped_ptr<test::FakeAudioDevice> fake_send_audio_device_;
-  rtc::scoped_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
+  std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
+  std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
 };
 
 class BaseTest : public RtpRtcpObserver {
diff --git a/chromium/third_party/webrtc/test/channel_transport/udp_transport_impl.cc b/chromium/third_party/webrtc/test/channel_transport/udp_transport_impl.cc
index c7049aa8a2d..897c8085226 100644
--- a/chromium/third_party/webrtc/test/channel_transport/udp_transport_impl.cc
+++ b/chromium/third_party/webrtc/test/channel_transport/udp_transport_impl.cc
@@ -2922,6 +2922,10 @@ bool UdpTransport::IsIpAddressValid(const char* ipadr, const bool ipV6)
                 // Store index of dots and count number of dots.
                 iDotPos[nDots++] = i;
             }
+            else if (isdigit(ipadr[i]) == 0)
+            {
+                return false;
+            }
         }
 
         bool allUnder256 = false;
@@ -2942,7 +2946,7 @@ bool UdpTransport::IsIpAddressValid(const char* ipadr, const bool ipV6)
                 memset(nr,0,4);
                 strncpy(nr,&ipadr[0],iDotPos[0]);
                 int32_t num = atoi(nr);
-                if (num > 255)
+                if (num > 255 || num < 0)
                 {
                     break;
                 }
@@ -2956,7 +2960,7 @@ bool UdpTransport::IsIpAddressValid(const char* ipadr, const bool ipV6)
                 memset(nr,0,4);
                 strncpy(nr,&ipadr[iDotPos[0]+1], iDotPos[1] - iDotPos[0] - 1);
                 int32_t num = atoi(nr);
-                if (num > 255)
+                if (num > 255 || num < 0)
                     break;
             } else {
                 break;
@@ -2966,20 +2970,27 @@ bool UdpTransport::IsIpAddressValid(const char* ipadr, const bool ipV6)
             {
                 char nr[4];
                 memset(nr,0,4);
-                strncpy(nr,&ipadr[iDotPos[1]+1], iDotPos[1] - iDotPos[0] - 1);
+                strncpy(nr,&ipadr[iDotPos[1]+1], iDotPos[2] - iDotPos[1] - 1);
                 int32_t num = atoi(nr);
-                if (num > 255)
+                if (num > 255 || num < 0)
                     break;
+            } else {
+                break;
+            }
 
+            if (len - iDotPos[2] <= 4)
+            {
+                char nr[4];
                 memset(nr,0,4);
                 strncpy(nr,&ipadr[iDotPos[2]+1], len - iDotPos[2] -1);
-                num = atoi(nr);
-                if (num > 255)
+                int32_t num = atoi(nr);
+                if (num > 255 || num < 0)
                     break;
                 else
                     allUnder256 = true;
-            } else
+            } else {
                 break;
+            }
         } while(false);
 
         if (nDots != 3 || !allUnder256)
diff --git a/chromium/third_party/webrtc/test/configurable_frame_size_encoder.h b/chromium/third_party/webrtc/test/configurable_frame_size_encoder.h
index 3794e8db088..d269441667b 100644
--- a/chromium/third_party/webrtc/test/configurable_frame_size_encoder.h
+++ b/chromium/third_party/webrtc/test/configurable_frame_size_encoder.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_
 #define WEBRTC_TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_
 
+#include <memory>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/video_encoder.h"
 
 namespace webrtc {
@@ -49,7 +49,7 @@ class ConfigurableFrameSizeEncoder : public VideoEncoder {
   EncodedImageCallback* callback_;
   const size_t max_frame_size_;
   size_t current_frame_size_;
-  rtc::scoped_ptr<uint8_t[]> buffer_;
+  std::unique_ptr<uint8_t[]> buffer_;
 };
 
 }  // namespace test
diff --git a/chromium/third_party/webrtc/test/direct_transport.h b/chromium/third_party/webrtc/test/direct_transport.h
index d68bc7184eb..e1844763d95 100644
--- a/chromium/third_party/webrtc/test/direct_transport.h
+++ b/chromium/third_party/webrtc/test/direct_transport.h
@@ -17,7 +17,6 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/fake_network_pipe.h"
 #include "webrtc/transport.h"
 
diff --git a/chromium/third_party/webrtc/test/fake_audio_device.h b/chromium/third_party/webrtc/test/fake_audio_device.h
index 180abf6c924..77a74bac8f6 100644
--- a/chromium/third_party/webrtc/test/fake_audio_device.h
+++ b/chromium/third_party/webrtc/test/fake_audio_device.h
@@ -10,11 +10,11 @@
 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
 #define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/test/drifting_clock.h"
 #include "webrtc/typedefs.h"
@@ -59,11 +59,11 @@ class FakeAudioDevice : public FakeAudioDeviceModule {
   int64_t last_playout_ms_;
 
   DriftingClock clock_;
-  rtc::scoped_ptr<EventTimerWrapper> tick_;
+  std::unique_ptr<EventTimerWrapper> tick_;
   rtc::CriticalSection lock_;
   rtc::PlatformThread thread_;
-  rtc::scoped_ptr<ModuleFileUtility> file_utility_;
-  rtc::scoped_ptr<FileWrapper> input_stream_;
+  std::unique_ptr<ModuleFileUtility> file_utility_;
+  std::unique_ptr<FileWrapper> input_stream_;
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/test/fake_network_pipe.h b/chromium/third_party/webrtc/test/fake_network_pipe.h
index d488d492c8d..608ff008c99 100644
--- a/chromium/third_party/webrtc/test/fake_network_pipe.h
+++ b/chromium/third_party/webrtc/test/fake_network_pipe.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
 #define WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
 
+#include <memory>
 #include <set>
 #include <string.h>
 #include <queue>
@@ -18,7 +19,6 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -50,7 +50,7 @@ class NetworkPacket {
 
  private:
   // The packet data.
-  rtc::scoped_ptr<uint8_t[]> data_;
+  std::unique_ptr<uint8_t[]> data_;
   // Length of data_.
   size_t data_length_;
   // The time the packet was sent out on the network.
diff --git a/chromium/third_party/webrtc/test/fake_network_pipe_unittest.cc b/chromium/third_party/webrtc/test/fake_network_pipe_unittest.cc
index 233c5972b3e..0bd46df7fcf 100644
--- a/chromium/third_party/webrtc/test/fake_network_pipe_unittest.cc
+++ b/chromium/third_party/webrtc/test/fake_network_pipe_unittest.cc
@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/fake_network_pipe.h"
@@ -71,7 +72,7 @@ class FakeNetworkPipeTest : public ::testing::Test {
 
   void SendPackets(FakeNetworkPipe* pipe, int number_packets, int packet_size) {
     RTC_DCHECK_GE(packet_size, static_cast<int>(sizeof(int)));
-    rtc::scoped_ptr<uint8_t[]> packet(new uint8_t[packet_size]);
+    std::unique_ptr<uint8_t[]> packet(new uint8_t[packet_size]);
     for (int i = 0; i < number_packets; ++i) {
       // Set a sequence number for the packets by
       // using the first bytes in the packet.
@@ -85,7 +86,7 @@ class FakeNetworkPipeTest : public ::testing::Test {
   }
 
   SimulatedClock fake_clock_;
-  rtc::scoped_ptr<TestReceiver> receiver_;
+  std::unique_ptr<TestReceiver> receiver_;
 };
 
 void DeleteMemory(uint8_t* data, int length) { delete [] data; }
@@ -95,7 +96,7 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+  std::unique_ptr<FakeNetworkPipe> pipe(
       new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
@@ -135,7 +136,7 @@ TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
   config.queue_length_packets = 20;
   config.queue_delay_ms = 100;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+  std::unique_ptr<FakeNetworkPipe> pipe(
       new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
@@ -169,7 +170,7 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 2;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+  std::unique_ptr<FakeNetworkPipe> pipe(
       new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
@@ -193,7 +194,7 @@ TEST_F(FakeNetworkPipeTest, StatisticsTest) {
   config.queue_length_packets = 2;
   config.queue_delay_ms = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+  std::unique_ptr<FakeNetworkPipe> pipe(
       new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
@@ -223,7 +224,7 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+  std::unique_ptr<FakeNetworkPipe> pipe(
       new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
@@ -282,7 +283,7 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
   FakeNetworkPipe::Config config;
   config.queue_length_packets = 20;
   config.link_capacity_kbps = 80;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+  std::unique_ptr<FakeNetworkPipe> pipe(
       new FakeNetworkPipe(&fake_clock_, config));
   pipe->SetReceiver(receiver_.get());
 
@@ -337,7 +338,7 @@ TEST_F(FakeNetworkPipeTest, DisallowReorderingThenAllowReordering) {
   config.link_capacity_kbps = 800;
   config.queue_delay_ms = 100;
   config.delay_standard_deviation_ms = 10;
-  rtc::scoped_ptr<FakeNetworkPipe> pipe(
+  std::unique_ptr<FakeNetworkPipe> pipe(
       new FakeNetworkPipe(&fake_clock_, config));
   ReorderTestReceiver* receiver = new ReorderTestReceiver();
   receiver_.reset(receiver);
diff --git a/chromium/third_party/webrtc/test/fake_texture_frame.h b/chromium/third_party/webrtc/test/fake_texture_frame.h
index 9575fae469d..15b70d8f7b5 100644
--- a/chromium/third_party/webrtc/test/fake_texture_frame.h
+++ b/chromium/third_party/webrtc/test/fake_texture_frame.h
@@ -42,9 +42,9 @@ class FakeNativeHandleBuffer : public NativeHandleBuffer {
         new rtc::RefCountedObject<I420Buffer>(width_, height_));
     int half_height = (height_ + 1) / 2;
     int half_width = (width_ + 1) / 2;
-    memset(buffer->MutableData(kYPlane), 0, height_ * width_);
-    memset(buffer->MutableData(kUPlane), 0, half_height * half_width);
-    memset(buffer->MutableData(kVPlane), 0, half_height * half_width);
+    memset(buffer->MutableDataY(), 0, height_ * width_);
+    memset(buffer->MutableDataU(), 0, half_height * half_width);
+    memset(buffer->MutableDataV(), 0, half_height * half_width);
     return buffer;
   }
 };
diff --git a/chromium/third_party/webrtc/test/frame_generator.cc b/chromium/third_party/webrtc/test/frame_generator.cc
index 3287abab816..ed7e95a1267 100644
--- a/chromium/third_party/webrtc/test/frame_generator.cc
+++ b/chromium/third_party/webrtc/test/frame_generator.cc
@@ -13,6 +13,8 @@
 #include <stdio.h>
 #include <string.h>
 
+#include <memory>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/include/clock.h"
@@ -39,9 +41,12 @@ class ChromaGenerator : public FrameGenerator {
     uint8_t u = fabs(sin(angle_)) * 0xFF;
     uint8_t v = fabs(cos(angle_)) * 0xFF;
 
-    memset(frame_.buffer(kYPlane), 0x80, frame_.allocated_size(kYPlane));
-    memset(frame_.buffer(kUPlane), u, frame_.allocated_size(kUPlane));
-    memset(frame_.buffer(kVPlane), v, frame_.allocated_size(kVPlane));
+    memset(frame_.video_frame_buffer()->MutableDataY(), 0x80,
+           frame_.allocated_size(kYPlane));
+    memset(frame_.video_frame_buffer()->MutableDataU(), u,
+           frame_.allocated_size(kUPlane));
+    memset(frame_.video_frame_buffer()->MutableDataV(), v,
+           frame_.allocated_size(kVPlane));
     return &frame_;
   }
 
@@ -121,7 +126,7 @@ class YuvFileGenerator : public FrameGenerator {
   const size_t width_;
   const size_t height_;
   const size_t frame_size_;
-  const rtc::scoped_ptr<uint8_t[]> frame_buffer_;
+  const std::unique_ptr<uint8_t[]> frame_buffer_;
   const int frame_display_count_;
   int current_display_count_;
   VideoFrame last_read_frame_;
@@ -200,24 +205,24 @@ class ScrollingImageFrameGenerator : public FrameGenerator {
     int pixels_scrolled_y =
         static_cast<int>(scroll_margin_y * scroll_factor + 0.5);
 
-    int offset_y = (current_source_frame_->stride(PlaneType::kYPlane) *
+    int offset_y = (current_source_frame_->video_frame_buffer()->StrideY() *
                     pixels_scrolled_y) +
                    pixels_scrolled_x;
-    int offset_u = (current_source_frame_->stride(PlaneType::kUPlane) *
+    int offset_u = (current_source_frame_->video_frame_buffer()->StrideU() *
                     (pixels_scrolled_y / 2)) +
                    (pixels_scrolled_x / 2);
-    int offset_v = (current_source_frame_->stride(PlaneType::kVPlane) *
+    int offset_v = (current_source_frame_->video_frame_buffer()->StrideV() *
                     (pixels_scrolled_y / 2)) +
                    (pixels_scrolled_x / 2);
 
     current_frame_.CreateFrame(
-        &current_source_frame_->buffer(PlaneType::kYPlane)[offset_y],
-        &current_source_frame_->buffer(PlaneType::kUPlane)[offset_u],
-        &current_source_frame_->buffer(PlaneType::kVPlane)[offset_v],
+        &current_source_frame_->video_frame_buffer()->DataY()[offset_y],
+        &current_source_frame_->video_frame_buffer()->DataU()[offset_u],
+        &current_source_frame_->video_frame_buffer()->DataV()[offset_v],
         kTargetWidth, kTargetHeight,
-        current_source_frame_->stride(PlaneType::kYPlane),
-        current_source_frame_->stride(PlaneType::kUPlane),
-        current_source_frame_->stride(PlaneType::kVPlane),
+        current_source_frame_->video_frame_buffer()->StrideY(),
+        current_source_frame_->video_frame_buffer()->StrideU(),
+        current_source_frame_->video_frame_buffer()->StrideV(),
         kVideoRotation_0);
   }
 
diff --git a/chromium/third_party/webrtc/test/frame_generator_capturer.cc b/chromium/third_party/webrtc/test/frame_generator_capturer.cc
index 35ce6168a2e..95ac624c421 100644
--- a/chromium/third_party/webrtc/test/frame_generator_capturer.cc
+++ b/chromium/third_party/webrtc/test/frame_generator_capturer.cc
@@ -80,6 +80,11 @@ FrameGeneratorCapturer::~FrameGeneratorCapturer() {
   thread_.Stop();
 }
 
+void FrameGeneratorCapturer::SetFakeRotation(VideoRotation rotation) {
+  rtc::CritScope cs(&lock_);
+  fake_rotation_ = rotation;
+}
+
 bool FrameGeneratorCapturer::Init() {
   // This check is added because frame_generator_ might be file based and should
   // not crash because a file moved.
@@ -104,6 +109,7 @@ void FrameGeneratorCapturer::InsertFrame() {
     if (sending_) {
       VideoFrame* frame = frame_generator_->NextFrame();
       frame->set_ntp_time_ms(clock_->CurrentNtpInMilliseconds());
+      frame->set_rotation(fake_rotation_);
       if (first_frame_capture_time_ == -1) {
         first_frame_capture_time_ = frame->ntp_time_ms();
       }
diff --git a/chromium/third_party/webrtc/test/frame_generator_capturer.h b/chromium/third_party/webrtc/test/frame_generator_capturer.h
index 6bd0e0b3272..1d6fb626640 100644
--- a/chromium/third_party/webrtc/test/frame_generator_capturer.h
+++ b/chromium/third_party/webrtc/test/frame_generator_capturer.h
@@ -10,11 +10,12 @@
 #ifndef WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_
 #define WEBRTC_TEST_FRAME_GENERATOR_CAPTURER_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_video/rotation.h"
 #include "webrtc/test/video_capturer.h"
 #include "webrtc/typedefs.h"
 
@@ -46,6 +47,7 @@ class FrameGeneratorCapturer : public VideoCapturer {
   void Start() override;
   void Stop() override;
   void ForceFrame();
+  void SetFakeRotation(VideoRotation rotation);
 
   int64_t first_frame_capture_time() const { return first_frame_capture_time_; }
 
@@ -62,12 +64,13 @@ class FrameGeneratorCapturer : public VideoCapturer {
   Clock* const clock_;
   bool sending_;
 
-  rtc::scoped_ptr<EventTimerWrapper> tick_;
+  std::unique_ptr<EventTimerWrapper> tick_;
   rtc::CriticalSection lock_;
   rtc::PlatformThread thread_;
-  rtc::scoped_ptr<FrameGenerator> frame_generator_;
+  std::unique_ptr<FrameGenerator> frame_generator_;
 
   int target_fps_;
+  VideoRotation fake_rotation_ = kVideoRotation_0;
 
   int64_t first_frame_capture_time_;
 };
diff --git a/chromium/third_party/webrtc/test/frame_generator_unittest.cc b/chromium/third_party/webrtc/test/frame_generator_unittest.cc
index 6376e2c221c..2cffdcaa8c3 100644
--- a/chromium/third_party/webrtc/test/frame_generator_unittest.cc
+++ b/chromium/third_party/webrtc/test/frame_generator_unittest.cc
@@ -9,10 +9,11 @@
  */
 
 #include <stdio.h>
+
+#include <memory>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -46,7 +47,7 @@ class FrameGeneratorTest : public ::testing::Test {
  protected:
   void WriteYuvFile(FILE* file, uint8_t y, uint8_t u, uint8_t v) {
     assert(file);
-    rtc::scoped_ptr<uint8_t[]> plane_buffer(new uint8_t[y_size]);
+    std::unique_ptr<uint8_t[]> plane_buffer(new uint8_t[y_size]);
     memset(plane_buffer.get(), y, y_size);
     fwrite(plane_buffer.get(), 1, y_size, file);
     memset(plane_buffer.get(), u, uv_size);
@@ -58,17 +59,17 @@ class FrameGeneratorTest : public ::testing::Test {
   void CheckFrameAndMutate(VideoFrame* frame, uint8_t y, uint8_t u, uint8_t v) {
     // Check that frame is valid, has the correct color and timestamp are clean.
     ASSERT_NE(nullptr, frame);
-    uint8_t* buffer;
+    const uint8_t* buffer;
     ASSERT_EQ(y_size, frame->allocated_size(PlaneType::kYPlane));
-    buffer = frame->buffer(PlaneType::kYPlane);
+    buffer = frame->video_frame_buffer()->DataY();
     for (int i = 0; i < y_size; ++i)
       ASSERT_EQ(y, buffer[i]);
     ASSERT_EQ(uv_size, frame->allocated_size(PlaneType::kUPlane));
-    buffer = frame->buffer(PlaneType::kUPlane);
+    buffer = frame->video_frame_buffer()->DataU();
     for (int i = 0; i < uv_size; ++i)
       ASSERT_EQ(u, buffer[i]);
     ASSERT_EQ(uv_size, frame->allocated_size(PlaneType::kVPlane));
-    buffer = frame->buffer(PlaneType::kVPlane);
+    buffer = frame->video_frame_buffer()->DataV();
     for (int i = 0; i < uv_size; ++i)
       ASSERT_EQ(v, buffer[i]);
     EXPECT_EQ(0, frame->ntp_time_ms());
@@ -88,7 +89,7 @@ class FrameGeneratorTest : public ::testing::Test {
 };
 
 TEST_F(FrameGeneratorTest, SingleFrameFile) {
-  rtc::scoped_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
+  std::unique_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
       std::vector<std::string>(1, one_frame_filename_), kFrameWidth,
       kFrameHeight, 1));
   CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
@@ -96,7 +97,7 @@ TEST_F(FrameGeneratorTest, SingleFrameFile) {
 }
 
 TEST_F(FrameGeneratorTest, TwoFrameFile) {
-  rtc::scoped_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
+  std::unique_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
       std::vector<std::string>(1, two_frame_filename_), kFrameWidth,
       kFrameHeight, 1));
   CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
@@ -109,7 +110,7 @@ TEST_F(FrameGeneratorTest, MultipleFrameFiles) {
   files.push_back(two_frame_filename_);
   files.push_back(one_frame_filename_);
 
-  rtc::scoped_ptr<FrameGenerator> generator(
+  std::unique_ptr<FrameGenerator> generator(
       FrameGenerator::CreateFromYuvFile(files, kFrameWidth, kFrameHeight, 1));
   CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
   CheckFrameAndMutate(generator->NextFrame(), 127, 127, 127);
@@ -119,7 +120,7 @@ TEST_F(FrameGeneratorTest, MultipleFrameFiles) {
 
 TEST_F(FrameGeneratorTest, TwoFrameFileWithRepeat) {
   const int kRepeatCount = 3;
-  rtc::scoped_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
+  std::unique_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
       std::vector<std::string>(1, two_frame_filename_), kFrameWidth,
       kFrameHeight, kRepeatCount));
   for (int i = 0; i < kRepeatCount; ++i)
@@ -134,7 +135,7 @@ TEST_F(FrameGeneratorTest, MultipleFrameFilesWithRepeat) {
   std::vector<std::string> files;
   files.push_back(two_frame_filename_);
   files.push_back(one_frame_filename_);
-  rtc::scoped_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
+  std::unique_ptr<FrameGenerator> generator(FrameGenerator::CreateFromYuvFile(
       files, kFrameWidth, kFrameHeight, kRepeatCount));
   for (int i = 0; i < kRepeatCount; ++i)
     CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
diff --git a/chromium/third_party/webrtc/test/frame_utils.cc b/chromium/third_party/webrtc/test/frame_utils.cc
index 0f411447458..0fad3adec85 100644
--- a/chromium/third_party/webrtc/test/frame_utils.cc
+++ b/chromium/third_party/webrtc/test/frame_utils.cc
@@ -16,53 +16,60 @@ namespace test {
 
 bool EqualPlane(const uint8_t* data1,
                 const uint8_t* data2,
-                int stride,
+                int stride1,
+                int stride2,
                 int width,
                 int height) {
   for (int y = 0; y < height; ++y) {
     if (memcmp(data1, data2, width) != 0)
       return false;
-    data1 += stride;
-    data2 += stride;
+    data1 += stride1;
+    data2 += stride2;
   }
   return true;
 }
+
 bool FramesEqual(const webrtc::VideoFrame& f1, const webrtc::VideoFrame& f2) {
-  if (f1.width() != f2.width() || f1.height() != f2.height() ||
-      f1.stride(webrtc::kYPlane) != f2.stride(webrtc::kYPlane) ||
-      f1.stride(webrtc::kUPlane) != f2.stride(webrtc::kUPlane) ||
-      f1.stride(webrtc::kVPlane) != f2.stride(webrtc::kVPlane) ||
-      f1.timestamp() != f2.timestamp() ||
+  if (f1.timestamp() != f2.timestamp() ||
       f1.ntp_time_ms() != f2.ntp_time_ms() ||
       f1.render_time_ms() != f2.render_time_ms()) {
     return false;
   }
-  const int half_width = (f1.width() + 1) / 2;
-  const int half_height = (f1.height() + 1) / 2;
-  return EqualPlane(f1.buffer(webrtc::kYPlane), f2.buffer(webrtc::kYPlane),
-                    f1.stride(webrtc::kYPlane), f1.width(), f1.height()) &&
-         EqualPlane(f1.buffer(webrtc::kUPlane), f2.buffer(webrtc::kUPlane),
-                    f1.stride(webrtc::kUPlane), half_width, half_height) &&
-         EqualPlane(f1.buffer(webrtc::kVPlane), f2.buffer(webrtc::kVPlane),
-                    f1.stride(webrtc::kVPlane), half_width, half_height);
+  return FrameBufsEqual(f1.video_frame_buffer(), f2.video_frame_buffer());
 }
 
 bool FrameBufsEqual(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f1,
                     const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f2) {
-  if (f1->width() != f2->width() || f1->height() != f2->height() ||
-      f1->stride(webrtc::kYPlane) != f2->stride(webrtc::kYPlane) ||
-      f1->stride(webrtc::kUPlane) != f2->stride(webrtc::kUPlane) ||
-      f1->stride(webrtc::kVPlane) != f2->stride(webrtc::kVPlane)) {
+  if (f1 == f2) {
+    return true;
+  }
+  // Exlude nullptr (except if both are nullptr, as above)
+  if (!f1 || !f2) {
+    return false;
+  }
+
+  if (f1->width() != f2->width() || f1->height() != f2->height()) {
+    return false;
+  }
+  // Exclude native handle
+  if (f1->native_handle()) {
+    return f1->native_handle() == f2->native_handle();
+  }
+
+  if (f2->native_handle()) {
     return false;
   }
   const int half_width = (f1->width() + 1) / 2;
   const int half_height = (f1->height() + 1) / 2;
-  return EqualPlane(f1->data(webrtc::kYPlane), f2->data(webrtc::kYPlane),
-                    f1->stride(webrtc::kYPlane), f1->width(), f1->height()) &&
-         EqualPlane(f1->data(webrtc::kUPlane), f2->data(webrtc::kUPlane),
-                    f1->stride(webrtc::kUPlane), half_width, half_height) &&
-         EqualPlane(f1->data(webrtc::kVPlane), f2->data(webrtc::kVPlane),
-                    f1->stride(webrtc::kVPlane), half_width, half_height);
+  return EqualPlane(f1->DataY(), f2->DataY(),
+                    f1->StrideY(), f2->StrideY(),
+                    f1->width(), f1->height()) &&
+         EqualPlane(f1->DataU(), f2->DataU(),
+                    f1->StrideU(), f2->StrideU(),
+                    half_width, half_height) &&
+         EqualPlane(f1->DataV(), f2->DataV(),
+                    f1->StrideV(), f2->StrideV(),
+                    half_width, half_height);
 }
 
 }  // namespace test
diff --git a/chromium/third_party/webrtc/test/frame_utils.h b/chromium/third_party/webrtc/test/frame_utils.h
index 668d9994ab7..aef3c9ff2af 100644
--- a/chromium/third_party/webrtc/test/frame_utils.h
+++ b/chromium/third_party/webrtc/test/frame_utils.h
@@ -20,10 +20,19 @@ namespace test {
 
 bool EqualPlane(const uint8_t* data1,
                 const uint8_t* data2,
-                int stride,
+                int stride1,
+                int stride2,
                 int width,
                 int height);
 
+static inline bool EqualPlane(const uint8_t* data1,
+                              const uint8_t* data2,
+                              int stride,
+                              int width,
+                              int height) {
+  return EqualPlane(data1, data2, stride, stride, width, height);
+}
+
 bool FramesEqual(const webrtc::VideoFrame& f1, const webrtc::VideoFrame& f2);
 
 bool FrameBufsEqual(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f1,
diff --git a/chromium/third_party/webrtc/test/fuzzers/BUILD.gn b/chromium/third_party/webrtc/test/fuzzers/BUILD.gn
index c46cc1e2c0a..3e59339299f 100644
--- a/chromium/third_party/webrtc/test/fuzzers/BUILD.gn
+++ b/chromium/third_party/webrtc/test/fuzzers/BUILD.gn
@@ -75,6 +75,15 @@ webrtc_fuzzer_test("rtcp_receiver_fuzzer") {
   ]
 }
 
+webrtc_fuzzer_test("rtp_packet_fuzzer") {
+  sources = [
+    "rtp_packet_fuzzer.cc",
+  ]
+  deps = [
+    "../../modules/rtp_rtcp/",
+  ]
+}
+
 source_set("audio_decoder_fuzzer") {
   public_configs = [ "../..:common_inherited_config" ]
   sources = [
diff --git a/chromium/third_party/webrtc/test/fuzzers/producer_fec_fuzzer.cc b/chromium/third_party/webrtc/test/fuzzers/producer_fec_fuzzer.cc
index 77336e6f1e0..53f74938699 100644
--- a/chromium/third_party/webrtc/test/fuzzers/producer_fec_fuzzer.cc
+++ b/chromium/third_party/webrtc/test/fuzzers/producer_fec_fuzzer.cc
@@ -7,8 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+
+#include <memory>
+
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 
@@ -20,9 +22,8 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
   size_t i = 0;
   if (size < 4)
     return;
-  FecProtectionParams params = {data[i++] % 128, data[i++] % 2 == 1,
-                                static_cast<int>(data[i++] % 10),
-                                kFecMaskBursty};
+  FecProtectionParams params = {
+      data[i++] % 128, static_cast<int>(data[i++] % 10), kFecMaskBursty};
   producer.SetFecParameters(&params, 0);
   uint16_t seq_num = data[i++];
 
@@ -31,14 +32,14 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
     size_t payload_size = data[i++] % 10;
     if (i + payload_size + rtp_header_length + 2 > size)
       break;
-    rtc::scoped_ptr<uint8_t[]> packet(
+    std::unique_ptr<uint8_t[]> packet(
         new uint8_t[payload_size + rtp_header_length]);
     memcpy(packet.get(), &data[i], payload_size + rtp_header_length);
     ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++);
     i += payload_size + rtp_header_length;
     // Make sure sequence numbers are increasing.
     const int kRedPayloadType = 98;
-    rtc::scoped_ptr<RedPacket> red_packet(producer.BuildRedPacket(
+    std::unique_ptr<RedPacket> red_packet(producer.BuildRedPacket(
         packet.get(), payload_size, rtp_header_length, kRedPayloadType));
     const bool protect = data[i++] % 2 == 1;
     if (protect) {
diff --git a/chromium/third_party/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc b/chromium/third_party/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
index 944c79d7d3b..b7a4fdfdf8f 100644
--- a/chromium/third_party/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
+++ b/chromium/third_party/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
diff --git a/chromium/third_party/webrtc/test/fuzzers/rtp_packet_fuzzer.cc b/chromium/third_party/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
new file mode 100644
index 00000000000..a9efdb96ecf
--- /dev/null
+++ b/chromium/third_party/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  RtpPacketReceived packet;
+
+  packet.Parse(data, size);
+
+  // Call packet accessors because they have extra checks.
+  packet.Marker();
+  packet.PayloadType();
+  packet.SequenceNumber();
+  packet.Timestamp();
+  packet.Ssrc();
+  packet.Csrcs();
+}
+
+}  // namespace webrtc
+
diff --git a/chromium/third_party/webrtc/test/layer_filtering_transport.cc b/chromium/third_party/webrtc/test/layer_filtering_transport.cc
index 41d63ad6e7d..cba1a57eb45 100644
--- a/chromium/third_party/webrtc/test/layer_filtering_transport.cc
+++ b/chromium/third_party/webrtc/test/layer_filtering_transport.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
@@ -62,7 +64,7 @@ bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
     const size_t payload_data_length = payload_length - header.paddingLength;
 
     const bool is_vp8 = header.payloadType == vp8_video_payload_type_;
-    rtc::scoped_ptr<RtpDepacketizer> depacketizer(
+    std::unique_ptr<RtpDepacketizer> depacketizer(
         RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
     RtpDepacketizer::ParsedPayload parsed_payload;
     if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) {
diff --git a/chromium/third_party/webrtc/test/mock_voe_channel_proxy.h b/chromium/third_party/webrtc/test/mock_voe_channel_proxy.h
index c2211f85547..a27a739006e 100644
--- a/chromium/third_party/webrtc/test/mock_voe_channel_proxy.h
+++ b/chromium/third_party/webrtc/test/mock_voe_channel_proxy.h
@@ -44,6 +44,12 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
   MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
   MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
+  MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
+  MOCK_METHOD0(DeRegisterExternalTransport, void());
+  MOCK_METHOD3(ReceivedRTPPacket, bool(const uint8_t* packet,
+                                       size_t length,
+                                       const PacketTime& packet_time));
+  MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/test/mock_voice_engine.h b/chromium/third_party/webrtc/test/mock_voice_engine.h
index fac088b0e17..b9eb05fb8be 100644
--- a/chromium/third_party/webrtc/test/mock_voice_engine.h
+++ b/chromium/third_party/webrtc/test/mock_voice_engine.h
@@ -24,6 +24,10 @@ namespace test {
 // able to get the various interfaces as usual, via T::GetInterface().
 class MockVoiceEngine : public VoiceEngineImpl {
  public:
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
   MockVoiceEngine() : VoiceEngineImpl(new Config(), true) {
     // Increase ref count so this object isn't automatically deleted whenever
     // interfaces are Release():d.
@@ -36,7 +40,7 @@ class MockVoiceEngine : public VoiceEngineImpl {
               return new testing::NiceMock<MockVoEChannelProxy>();
             }));
   }
-  ~MockVoiceEngine() override {
+  ~MockVoiceEngine() /* override */ {
     // Decrease ref count before base class d-tor is called; otherwise it will
     // trigger an assertion.
     --_ref_count;
@@ -45,7 +49,8 @@ class MockVoiceEngine : public VoiceEngineImpl {
   MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id));
 
   // VoiceEngineImpl
-  std::unique_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id) override {
+  std::unique_ptr<voe::ChannelProxy> GetChannelProxy(
+      int channel_id) /* override */ {
     return std::unique_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id));
   }
 
diff --git a/chromium/third_party/webrtc/test/rtp_file_reader.cc b/chromium/third_party/webrtc/test/rtp_file_reader.cc
index 1413f007979..d437d419248 100644
--- a/chromium/third_party/webrtc/test/rtp_file_reader.cc
+++ b/chromium/third_party/webrtc/test/rtp_file_reader.cc
@@ -17,8 +17,8 @@
 #include <vector>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/format_macros.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
@@ -130,7 +130,7 @@ class RtpDumpReader : public RtpFileReaderImpl {
   }
 
   bool Init(const std::string& filename,
-            const std::set<uint32_t>& ssrc_filter) {
+            const std::set<uint32_t>& ssrc_filter) override {
     file_ = fopen(filename.c_str(), "rb");
     if (file_ == NULL) {
       printf("ERROR: Can't open file: %s\n", filename.c_str());
diff --git a/chromium/third_party/webrtc/test/rtp_file_reader_unittest.cc b/chromium/third_party/webrtc/test/rtp_file_reader_unittest.cc
index 15a456ccf61..fceac3836d6 100644
--- a/chromium/third_party/webrtc/test/rtp_file_reader_unittest.cc
+++ b/chromium/third_party/webrtc/test/rtp_file_reader_unittest.cc
@@ -9,9 +9,9 @@
  */
 
 #include <map>
+#include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -43,7 +43,7 @@ class TestRtpFileReader : public ::testing::Test {
   }
 
  private:
-  rtc::scoped_ptr<test::RtpFileReader> rtp_packet_source_;
+  std::unique_ptr<test::RtpFileReader> rtp_packet_source_;
   bool headers_only_file_;
 };
 
@@ -94,7 +94,7 @@ class TestPcapFileReader : public ::testing::Test {
   }
 
  private:
-  rtc::scoped_ptr<test::RtpFileReader> rtp_packet_source_;
+  std::unique_ptr<test::RtpFileReader> rtp_packet_source_;
 };
 
 TEST_F(TestPcapFileReader, TestEthernetIIFrame) {
diff --git a/chromium/third_party/webrtc/test/rtp_file_writer_unittest.cc b/chromium/third_party/webrtc/test/rtp_file_writer_unittest.cc
index 2c7c88cc3f3..3287f976c6c 100644
--- a/chromium/third_party/webrtc/test/rtp_file_writer_unittest.cc
+++ b/chromium/third_party/webrtc/test/rtp_file_writer_unittest.cc
@@ -10,8 +10,9 @@
 
 #include <string.h>
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/rtp_file_writer.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -43,7 +44,7 @@ class RtpFileWriterTest : public ::testing::Test {
   void VerifyFileContents(int expected_packets) {
     ASSERT_TRUE(rtp_writer_.get() == NULL)
         << "Must call CloseOutputFile before VerifyFileContents";
-    rtc::scoped_ptr<test::RtpFileReader> rtp_reader(
+    std::unique_ptr<test::RtpFileReader> rtp_reader(
         test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filename_));
     ASSERT_TRUE(rtp_reader.get() != NULL);
     test::RtpPacket packet;
@@ -61,7 +62,7 @@ class RtpFileWriterTest : public ::testing::Test {
   }
 
  private:
-  rtc::scoped_ptr<test::RtpFileWriter> rtp_writer_;
+  std::unique_ptr<test::RtpFileWriter> rtp_writer_;
   std::string filename_;
 };
 
diff --git a/chromium/third_party/webrtc/test/rtp_rtcp_observer.h b/chromium/third_party/webrtc/test/rtp_rtcp_observer.h
index 5eb88d3f0d3..ab865611673 100644
--- a/chromium/third_party/webrtc/test/rtp_rtcp_observer.h
+++ b/chromium/third_party/webrtc/test/rtp_rtcp_observer.h
@@ -11,6 +11,7 @@
 #define WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
 
 #include <map>
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
@@ -69,7 +70,7 @@ class RtpRtcpObserver {
   }
 
   rtc::Event observation_complete_;
-  const rtc::scoped_ptr<RtpHeaderParser> parser_;
+  const std::unique_ptr<RtpHeaderParser> parser_;
 
  private:
   const int timeout_ms_;
diff --git a/chromium/third_party/webrtc/test/test.gyp b/chromium/third_party/webrtc/test/test.gyp
index ee3283ea6ad..a04a128a0eb 100644
--- a/chromium/third_party/webrtc/test/test.gyp
+++ b/chromium/third_party/webrtc/test/test.gyp
@@ -301,10 +301,9 @@
        '<(DEPTH)/testing/gmock.gyp:gmock',
        '<(DEPTH)/testing/gtest.gyp:gtest',
        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-       '<(webrtc_root)/base/base.gyp:rtc_base',
+       '<(webrtc_root)/base/base.gyp:rtc_base_approved',
        '<(webrtc_root)/common.gyp:webrtc_common',
        '<(webrtc_root)/modules/modules.gyp:media_file',
-       '<(webrtc_root)/modules/modules.gyp:video_render',
        '<(webrtc_root)/webrtc.gyp:webrtc',
        'rtp_test_utils',
        'test_support',
@@ -401,7 +400,7 @@
     },
   ],
   'conditions': [
-    ['include_tests==1 and OS=="android"', {
+    ['OS=="android"', {
       'targets': [
         {
           'target_name': 'test_support_unittests_apk_target',
@@ -411,7 +410,28 @@
           ],
         },
       ],
-    }],
+      'conditions': [
+        ['test_isolation_mode != "noop"',
+          {
+            'targets': [
+              {
+                'target_name': 'test_support_unittests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):test_support_unittests_apk',
+                ],
+                'includes': [
+                  '../build/isolate.gypi',
+                ],
+                'sources': [
+                  'test_support_unittests_apk.isolate',
+                ],
+              },
+            ],
+          },
+        ],
+      ],
+    }],  # OS=="android"
     ['test_isolation_mode != "noop"', {
       'targets': [
         {
diff --git a/chromium/third_party/webrtc/test/test_suite.h b/chromium/third_party/webrtc/test/test_suite.h
index dab2acd3881..4b27f9f2c32 100644
--- a/chromium/third_party/webrtc/test/test_suite.h
+++ b/chromium/third_party/webrtc/test/test_suite.h
@@ -17,8 +17,9 @@
 // instantiate this class in your main function and call its Run method to run
 // any gtest based tests that are linked into your executable.
 
+#include <memory>
+
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 namespace test {
@@ -41,7 +42,7 @@ class TestSuite {
   RTC_DISALLOW_COPY_AND_ASSIGN(TestSuite);
 
  private:
-  rtc::scoped_ptr<TraceToStderr> trace_to_stderr_;
+  std::unique_ptr<TraceToStderr> trace_to_stderr_;
 };
 
 }  // namespace test
diff --git a/chromium/third_party/webrtc/test/test_support_unittests_apk.isolate b/chromium/third_party/webrtc/test/test_support_unittests_apk.isolate
new file mode 100644
index 00000000000..e52980b3270
--- /dev/null
+++ b/chromium/third_party/webrtc/test/test_support_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'test_support_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_test_support_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/test_support_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_test_support_unittests',
+      'test_support_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/test/testsupport/fileutils.cc b/chromium/third_party/webrtc/test/testsupport/fileutils.cc
index 2fab425a316..d99b990918c 100644
--- a/chromium/third_party/webrtc/test/testsupport/fileutils.cc
+++ b/chromium/third_party/webrtc/test/testsupport/fileutils.cc
@@ -23,7 +23,6 @@
 #else
 #include <unistd.h>
 
-#include "webrtc/base/scoped_ptr.h"
 #define GET_CURRENT_DIR getcwd
 #endif
 
@@ -36,6 +35,8 @@
 #include <stdlib.h>
 #include <string.h>
 
+#include <memory>
+
 #include "webrtc/typedefs.h"  // For architecture defines
 
 namespace webrtc {
@@ -183,7 +184,7 @@ std::string TempFilename(const std::string &dir, const std::string &prefix) {
   return "";
 #else
   int len = dir.size() + prefix.size() + 2 + 6;
-  rtc::scoped_ptr<char[]> tempname(new char[len]);
+  std::unique_ptr<char[]> tempname(new char[len]);
 
   snprintf(tempname.get(), len, "%s/%sXXXXXX", dir.c_str(),
            prefix.c_str());
diff --git a/chromium/third_party/webrtc/tools/BUILD.gn b/chromium/third_party/webrtc/tools/BUILD.gn
index 603e99944c6..ebeacc24602 100644
--- a/chromium/third_party/webrtc/tools/BUILD.gn
+++ b/chromium/third_party/webrtc/tools/BUILD.gn
@@ -46,6 +46,7 @@ executable("frame_analyzer") {
   deps = [
     ":command_line_parser",
     ":video_quality_analysis",
+    "//build/win:default_exe_manifest",
   ]
 }
 
@@ -58,6 +59,7 @@ executable("rgba_to_i420_converter") {
   deps = [
     ":command_line_parser",
     "../common_video",
+    "//build/win:default_exe_manifest",
   ]
 }
 
@@ -77,6 +79,7 @@ if (!build_with_chromium) {
     deps = [
       ":command_line_parser",
       "../test:test_support_main",
+      "//build/win:default_exe_manifest",
       "//testing/gtest",
     ]
   }
diff --git a/chromium/third_party/webrtc/tools/agc/activity_metric.cc b/chromium/third_party/webrtc/tools/agc/activity_metric.cc
index 258d02377ee..6c1f756ec58 100644
--- a/chromium/third_party/webrtc/tools/agc/activity_metric.cc
+++ b/chromium/third_party/webrtc/tools/agc/activity_metric.cc
@@ -14,10 +14,10 @@
 #include <stdlib.h>
 
 #include <algorithm>
+#include <memory>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/agc/agc.h"
 #include "webrtc/modules/audio_processing/agc/histogram.h"
 #include "webrtc/modules/audio_processing/agc/utility.h"
@@ -155,10 +155,10 @@ class AgcStat {
   int video_index_;
   double activity_threshold_;
   double video_vad_[kMaxNumFrames];
-  rtc::scoped_ptr<Histogram> audio_content_;
-  rtc::scoped_ptr<VadAudioProc> audio_processing_;
-  rtc::scoped_ptr<PitchBasedVad> vad_;
-  rtc::scoped_ptr<StandaloneVad> standalone_vad_;
+  std::unique_ptr<Histogram> audio_content_;
+  std::unique_ptr<VadAudioProc> audio_processing_;
+  std::unique_ptr<PitchBasedVad> vad_;
+  std::unique_ptr<StandaloneVad> standalone_vad_;
 
   FILE* audio_content_fid_;
 };
diff --git a/chromium/third_party/webrtc/tools/agc/agc_harness.cc b/chromium/third_party/webrtc/tools/agc/agc_harness.cc
index 0d35d4b56aa..17919629b91 100644
--- a/chromium/third_party/webrtc/tools/agc/agc_harness.cc
+++ b/chromium/third_party/webrtc/tools/agc/agc_harness.cc
@@ -10,10 +10,11 @@
 
 // Refer to kUsage below for a description.
 
+#include <memory>
+
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/channel_transport/channel_transport.h"
@@ -217,13 +218,13 @@ class AgcVoiceEngine {
   int channel_;
   int capture_idx_;
   int render_idx_;
-  rtc::scoped_ptr<test::VoiceChannelTransport> channel_transport_;
+  std::unique_ptr<test::VoiceChannelTransport> channel_transport_;
 };
 
 void RunHarness() {
-  rtc::scoped_ptr<AgcVoiceEngine> voe1(new AgcVoiceEngine(
+  std::unique_ptr<AgcVoiceEngine> voe1(new AgcVoiceEngine(
       FLAGS_legacy_agc, 2000, 2000, FLAGS_capture1, FLAGS_render1));
-  rtc::scoped_ptr<AgcVoiceEngine> voe2;
+  std::unique_ptr<AgcVoiceEngine> voe2;
   if (FLAGS_parallel) {
     voe2.reset(new AgcVoiceEngine(!FLAGS_legacy_agc, 3000, 3000, FLAGS_capture2,
                                   FLAGS_render2));
diff --git a/chromium/third_party/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc b/chromium/third_party/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
index 2594fd1317b..fb07b569faf 100644
--- a/chromium/third_party/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
+++ b/chromium/third_party/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
@@ -12,10 +12,11 @@
 // and runs forever. Some parameters can be configured through command-line
 // flags.
 
+#include <memory>
+
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
@@ -48,7 +49,7 @@ void RunHarness() {
   int channel = base->CreateChannel();
   ASSERT_NE(-1, channel);
 
-  rtc::scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+  std::unique_ptr<VoiceChannelTransport> voice_channel_transport(
       new VoiceChannelTransport(network, channel));
 
   ASSERT_EQ(0, voice_channel_transport->SetSendDestination("127.0.0.1", 1234));
diff --git a/chromium/third_party/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc b/chromium/third_party/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
index b6b15968665..2bab2881bbe 100644
--- a/chromium/third_party/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
+++ b/chromium/third_party/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
@@ -12,7 +12,6 @@
 
 #include <stdio.h>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
diff --git a/chromium/third_party/webrtc/tools/frame_editing/frame_editing_lib.cc b/chromium/third_party/webrtc/tools/frame_editing/frame_editing_lib.cc
index 90855a354c0..bb6a75edaab 100644
--- a/chromium/third_party/webrtc/tools/frame_editing/frame_editing_lib.cc
+++ b/chromium/third_party/webrtc/tools/frame_editing/frame_editing_lib.cc
@@ -11,9 +11,9 @@
 #include <stdio.h>
 #include <stdlib.h>
 
+#include <memory>
 #include <string>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/tools/frame_editing/frame_editing_lib.h"
 #include "webrtc/typedefs.h"
@@ -39,7 +39,7 @@ int EditFrames(const string& in_path, int width, int height,
   // Frame size of I420.
   size_t frame_length = CalcBufferSize(kI420, width, height);
 
-  rtc::scoped_ptr<uint8_t[]> temp_buffer(new uint8_t[frame_length]);
+  std::unique_ptr<uint8_t[]> temp_buffer(new uint8_t[frame_length]);
 
   FILE* out_fid = fopen(out_path.c_str(), "wb");
 
diff --git a/chromium/third_party/webrtc/tools/frame_editing/frame_editing_unittest.cc b/chromium/third_party/webrtc/tools/frame_editing/frame_editing_unittest.cc
index 31991b71757..e1ba1de0fcb 100644
--- a/chromium/third_party/webrtc/tools/frame_editing/frame_editing_unittest.cc
+++ b/chromium/third_party/webrtc/tools/frame_editing/frame_editing_unittest.cc
@@ -12,9 +12,9 @@
 #include <stdlib.h>
 
 #include <fstream>
+#include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/tools/frame_editing/frame_editing_lib.h"
@@ -56,8 +56,8 @@ class FrameEditingTest : public ::testing::Test {
   // Compares the frames in both streams to the end of one of the streams.
   void CompareToTheEnd(FILE* test_video_fid,
                        FILE* ref_video_fid,
-                       rtc::scoped_ptr<int[]>* ref_buffer,
-                       rtc::scoped_ptr<int[]>* test_buffer) {
+                       std::unique_ptr<int[]>* ref_buffer,
+                       std::unique_ptr<int[]>* test_buffer) {
     while (!feof(test_video_fid) && !feof(ref_video_fid)) {
       num_bytes_read_ = fread(ref_buffer->get(), 1, kFrameSize, ref_video_fid);
       if (!feof(ref_video_fid)) {
@@ -81,8 +81,8 @@ class FrameEditingTest : public ::testing::Test {
   FILE* original_fid_;
   FILE* edited_fid_;
   size_t num_bytes_read_;
-  rtc::scoped_ptr<int[]> original_buffer_;
-  rtc::scoped_ptr<int[]> edited_buffer_;
+  std::unique_ptr<int[]> original_buffer_;
+  std::unique_ptr<int[]> edited_buffer_;
   int num_frames_read_;
 };
 
diff --git a/chromium/third_party/webrtc/tools/tools.gyp b/chromium/third_party/webrtc/tools/tools.gyp
index b69f7cb5b86..5a08ab99bd6 100644
--- a/chromium/third_party/webrtc/tools/tools.gyp
+++ b/chromium/third_party/webrtc/tools/tools.gyp
@@ -189,6 +189,27 @@
               ],
             },
           ],
+          'conditions': [
+            ['test_isolation_mode != "noop"',
+              {
+                'targets': [
+                  {
+                    'target_name': 'tools_unittests_apk_run',
+                    'type': 'none',
+                    'dependencies': [
+                      '<(apk_tests_path):tools_unittests_apk',
+                    ],
+                    'includes': [
+                      '../build/isolate.gypi',
+                    ],
+                    'sources': [
+                      'tools_unittests_apk.isolate',
+                    ],
+                  },
+                ],
+              },
+            ],
+          ],
         }],
         ['test_isolation_mode != "noop"', {
           'targets': [
diff --git a/chromium/third_party/webrtc/tools/tools_unittests_apk.isolate b/chromium/third_party/webrtc/tools/tools_unittests_apk.isolate
new file mode 100644
index 00000000000..a4ecb07903e
--- /dev/null
+++ b/chromium/third_party/webrtc/tools/tools_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'tools_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_tools_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/tools_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_tools_unittests',
+      'tools_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/typedefs.h b/chromium/third_party/webrtc/typedefs.h
index 6a3f441e22b..e65b77a65f8 100644
--- a/chromium/third_party/webrtc/typedefs.h
+++ b/chromium/third_party/webrtc/typedefs.h
@@ -57,8 +57,7 @@
 
 // TODO(zhongwei.yao): WEBRTC_CPU_DETECTION is only used in one place; we should
 // probably just remove it.
-#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)) || \
-    defined(WEBRTC_DETECT_NEON)
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
 #define WEBRTC_CPU_DETECTION
 #endif
 
diff --git a/chromium/third_party/webrtc/video/BUILD.gn b/chromium/third_party/webrtc/video/BUILD.gn
index 4f1b7ae1976..9e350f0cdba 100644
--- a/chromium/third_party/webrtc/video/BUILD.gn
+++ b/chromium/third_party/webrtc/video/BUILD.gn
@@ -24,6 +24,10 @@ source_set("video") {
     "receive_statistics_proxy.h",
     "report_block_stats.cc",
     "report_block_stats.h",
+    "rtp_stream_receiver.cc",
+    "rtp_stream_receiver.h",
+    "send_delay_stats.cc",
+    "send_delay_stats.h",
     "send_statistics_proxy.cc",
     "send_statistics_proxy.h",
     "stream_synchronization.cc",
@@ -36,12 +40,10 @@ source_set("video") {
     "video_receive_stream.h",
     "video_send_stream.cc",
     "video_send_stream.h",
-    "vie_channel.cc",
-    "vie_channel.h",
+    "video_stream_decoder.cc",
+    "video_stream_decoder.h",
     "vie_encoder.cc",
     "vie_encoder.h",
-    "vie_receiver.cc",
-    "vie_receiver.h",
     "vie_remb.cc",
     "vie_remb.h",
     "vie_sync_module.cc",
@@ -71,7 +73,6 @@ source_set("video") {
     "../modules/video_capture:video_capture_module",
     "../modules/video_coding",
     "../modules/video_processing",
-    "../modules/video_render:video_render_module",
     "../system_wrappers",
     "../voice_engine",
   ]
diff --git a/chromium/third_party/webrtc/video/DEPS b/chromium/third_party/webrtc/video/DEPS
index 426f47c4236..7e53144d33d 100644
--- a/chromium/third_party/webrtc/video/DEPS
+++ b/chromium/third_party/webrtc/video/DEPS
@@ -2,6 +2,7 @@ include_rules = [
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_video",
+  "+webrtc/media/base",
   "+webrtc/modules/bitrate_controller",
   "+webrtc/modules/congestion_controller",
   "+webrtc/modules/pacing",
@@ -11,7 +12,6 @@ include_rules = [
   "+webrtc/modules/video_coding",
   "+webrtc/modules/video_capture",
   "+webrtc/modules/video_processing",
-  "+webrtc/modules/video_render",
   "+webrtc/system_wrappers",
   "+webrtc/voice_engine",
 ]
diff --git a/chromium/third_party/webrtc/video/call_stats.cc b/chromium/third_party/webrtc/video/call_stats.cc
index a1bf221d4e2..59efcfa13de 100644
--- a/chromium/third_party/webrtc/video/call_stats.cc
+++ b/chromium/third_party/webrtc/video/call_stats.cc
@@ -13,9 +13,9 @@
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 namespace {
diff --git a/chromium/third_party/webrtc/video/call_stats_unittest.cc b/chromium/third_party/webrtc/video/call_stats_unittest.cc
index 6e2e1bca786..8b3cde07a8e 100644
--- a/chromium/third_party/webrtc/video/call_stats_unittest.cc
+++ b/chromium/third_party/webrtc/video/call_stats_unittest.cc
@@ -15,7 +15,6 @@
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/test/histogram.h"
 #include "webrtc/video/call_stats.h"
 
diff --git a/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.cc b/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.cc
index 84283608a04..974149170ab 100644
--- a/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.cc
+++ b/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.cc
@@ -26,9 +26,11 @@ int32_t EncodedFrameCallbackAdapter::Encoded(
     const EncodedImage& encodedImage,
     const CodecSpecificInfo* codecSpecificInfo,
     const RTPFragmentationHeader* fragmentation) {
-  RTC_DCHECK(observer_);
+  if (!observer_)
+    return 0;
   const EncodedFrame frame(encodedImage._buffer, encodedImage._length,
                            encodedImage._frameType);
+
   observer_->EncodedFrameCallback(frame);
   return 0;
 }
diff --git a/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.h b/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.h
index b10c4f1645d..83fe4bd3187 100644
--- a/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.h
+++ b/chromium/third_party/webrtc/video/encoded_frame_callback_adapter.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_VIDEO_ENCODED_FRAME_CALLBACK_ADAPTER_H_
 #define WEBRTC_VIDEO_ENCODED_FRAME_CALLBACK_ADAPTER_H_
 
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
-#include "webrtc/frame_callback.h"
 
 namespace webrtc {
 namespace internal {
@@ -27,7 +27,7 @@ class EncodedFrameCallbackAdapter : public EncodedImageCallback {
                           const RTPFragmentationHeader* fragmentation);
 
  private:
-  EncodedFrameObserver* observer_;
+  EncodedFrameObserver* const observer_;
 };
 
 }  // namespace internal
diff --git a/chromium/third_party/webrtc/video/encoder_state_feedback.cc b/chromium/third_party/webrtc/video/encoder_state_feedback.cc
index 0240e487a6a..28508d09a15 100644
--- a/chromium/third_party/webrtc/video/encoder_state_feedback.cc
+++ b/chromium/third_party/webrtc/video/encoder_state_feedback.cc
@@ -13,23 +13,18 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/video/vie_encoder.h"
 
-namespace webrtc {
+static const int kMinKeyFrameRequestIntervalMs = 300;
 
-EncoderStateFeedback::EncoderStateFeedback() : vie_encoder_(nullptr) {}
+namespace webrtc {
 
-void EncoderStateFeedback::Init(const std::vector<uint32_t>& ssrcs,
-                                ViEEncoder* encoder) {
+EncoderStateFeedback::EncoderStateFeedback(Clock* clock,
+                                           const std::vector<uint32_t>& ssrcs,
+                                           ViEEncoder* encoder)
+    : clock_(clock),
+      ssrcs_(ssrcs),
+      vie_encoder_(encoder),
+      time_last_intra_request_ms_(ssrcs.size(), -1) {
   RTC_DCHECK(!ssrcs.empty());
-  rtc::CritScope lock(&crit_);
-  ssrcs_ = ssrcs;
-  vie_encoder_ = encoder;
-}
-
-void EncoderStateFeedback::TearDown() {
-  rtc::CritScope lock(&crit_);
-  RTC_DCHECK(vie_encoder_);
-  ssrcs_.clear();
-  vie_encoder_ = nullptr;
 }
 
 bool EncoderStateFeedback::HasSsrc(uint32_t ssrc) {
@@ -40,41 +35,59 @@ bool EncoderStateFeedback::HasSsrc(uint32_t ssrc) {
   return false;
 }
 
+size_t EncoderStateFeedback::GetStreamIndex(uint32_t ssrc) {
+  for (size_t i = 0; i < ssrcs_.size(); ++i) {
+    if (ssrcs_[i] == ssrc)
+      return i;
+  }
+  RTC_NOTREACHED() << "Unknown ssrc " << ssrc;
+  return 0;
+}
+
 void EncoderStateFeedback::OnReceivedIntraFrameRequest(uint32_t ssrc) {
-  rtc::CritScope lock(&crit_);
   if (!HasSsrc(ssrc))
     return;
-  RTC_DCHECK(vie_encoder_);
 
-  vie_encoder_->OnReceivedIntraFrameRequest(ssrc);
+  size_t index = GetStreamIndex(ssrc);
+  {
+    int64_t now_ms = clock_->TimeInMilliseconds();
+    rtc::CritScope lock(&crit_);
+    if (time_last_intra_request_ms_[index] + kMinKeyFrameRequestIntervalMs >
+        now_ms) {
+      return;
+    }
+    time_last_intra_request_ms_[index] = now_ms;
+  }
+
+  vie_encoder_->OnReceivedIntraFrameRequest(index);
 }
 
 void EncoderStateFeedback::OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) {
-  rtc::CritScope lock(&crit_);
   if (!HasSsrc(ssrc))
     return;
-  RTC_DCHECK(vie_encoder_);
 
-  vie_encoder_->OnReceivedSLI(ssrc, picture_id);
+  vie_encoder_->OnReceivedSLI(picture_id);
 }
 
 void EncoderStateFeedback::OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) {
-  rtc::CritScope lock(&crit_);
   if (!HasSsrc(ssrc))
     return;
-  RTC_DCHECK(vie_encoder_);
 
-  vie_encoder_->OnReceivedRPSI(ssrc, picture_id);
+  vie_encoder_->OnReceivedRPSI(picture_id);
 }
 
 // Sending SSRCs for this encoder should never change since they are configured
-// once and not reconfigured.
+// once and not reconfigured, however, OnLocalSsrcChanged is called when the
+// RtpModules are created with a different SSRC than what will be used in the
+// end.
+// TODO(perkj): Can we make sure the RTP module is created with the right SSRC
+// from the beginning so this method is not triggered during creation ?
 void EncoderStateFeedback::OnLocalSsrcChanged(uint32_t old_ssrc,
                                               uint32_t new_ssrc) {
   if (!RTC_DCHECK_IS_ON)
     return;
-  rtc::CritScope lock(&crit_);
-  if (ssrcs_.empty())  // Encoder not yet attached (or detached for teardown).
+
+  if (old_ssrc == 0)  // old_ssrc == 0 during creation.
     return;
   // SSRC shouldn't change to something we haven't already registered with the
   // encoder.
diff --git a/chromium/third_party/webrtc/video/encoder_state_feedback.h b/chromium/third_party/webrtc/video/encoder_state_feedback.h
index 6326bed9c13..c9fb9cc7476 100644
--- a/chromium/third_party/webrtc/video/encoder_state_feedback.h
+++ b/chromium/third_party/webrtc/video/encoder_state_feedback.h
@@ -7,10 +7,6 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-
-// TODO(mflodman) ViEEncoder has a time check to not send key frames too often,
-// move the logic to this class.
-
 #ifndef WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
 #define WEBRTC_VIDEO_ENCODER_STATE_FEEDBACK_H_
 
@@ -18,6 +14,7 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -26,28 +23,24 @@ class ViEEncoder;
 
 class EncoderStateFeedback : public RtcpIntraFrameObserver {
  public:
-  EncoderStateFeedback();
-
-  // Adds an encoder to receive feedback for a set of SSRCs.
-  void Init(const std::vector<uint32_t>& ssrc, ViEEncoder* encoder);
-
-  // Removes the registered encoder. Necessary since RTP modules outlive
-  // ViEEncoder.
-  // TODO(pbos): Make sure RTP modules are not running when tearing down
-  // ViEEncoder, then remove this function.
-  void TearDown();
-
+  EncoderStateFeedback(Clock* clock,
+                       const std::vector<uint32_t>& ssrcs,
+                       ViEEncoder* encoder);
   void OnReceivedIntraFrameRequest(uint32_t ssrc) override;
   void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) override;
   void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) override;
   void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) override;
 
  private:
-  bool HasSsrc(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  rtc::CriticalSection crit_;
+  bool HasSsrc(uint32_t ssrc);
+  size_t GetStreamIndex(uint32_t ssrc);
 
-  std::vector<uint32_t> ssrcs_ GUARDED_BY(crit_);
-  ViEEncoder* vie_encoder_ GUARDED_BY(crit_);
+  Clock* const clock_;
+  const std::vector<uint32_t> ssrcs_;
+  ViEEncoder* const vie_encoder_;
+
+  rtc::CriticalSection crit_;
+  std::vector<int64_t> time_last_intra_request_ms_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/encoder_state_feedback_unittest.cc b/chromium/third_party/webrtc/video/encoder_state_feedback_unittest.cc
index 3341cf06674..7e6ec71d6eb 100644
--- a/chromium/third_party/webrtc/video/encoder_state_feedback_unittest.cc
+++ b/chromium/third_party/webrtc/video/encoder_state_feedback_unittest.cc
@@ -8,16 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
-// This file includes unit tests for EncoderStateFeedback.
 #include "webrtc/video/encoder_state_feedback.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/pacing/paced_sender.h"
-#include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/utility/include/mock/mock_process_thread.h"
 #include "webrtc/video/vie_encoder.h"
 
@@ -27,52 +21,58 @@ namespace webrtc {
 
 class MockVieEncoder : public ViEEncoder {
  public:
-  explicit MockVieEncoder(ProcessThread* process_thread, PacedSender* pacer)
-      : ViEEncoder(1,
-                   std::vector<uint32_t>(),
-                   process_thread,
-                   nullptr,
-                   nullptr,
-                   nullptr,
-                   pacer,
-                   nullptr) {}
+  explicit MockVieEncoder(ProcessThread* process_thread)
+      : ViEEncoder(1, process_thread, nullptr, nullptr) {}
   ~MockVieEncoder() {}
 
-  MOCK_METHOD1(OnReceivedIntraFrameRequest,
-               void(uint32_t));
-  MOCK_METHOD2(OnReceivedSLI,
-               void(uint32_t ssrc, uint8_t picture_id));
-  MOCK_METHOD2(OnReceivedRPSI,
-               void(uint32_t ssrc, uint64_t picture_id));
+  MOCK_METHOD1(OnReceivedIntraFrameRequest, void(size_t));
+  MOCK_METHOD1(OnReceivedSLI, void(uint8_t picture_id));
+  MOCK_METHOD1(OnReceivedRPSI, void(uint64_t picture_id));
 };
 
-TEST(VieKeyRequestTest, CreateAndTriggerRequests) {
-  static const uint32_t kSsrc = 1234;
-  NiceMock<MockProcessThread> process_thread;
-  PacketRouter router;
-  PacedSender pacer(Clock::GetRealTimeClock(), &router,
-                     BitrateController::kDefaultStartBitrateKbps,
-                     PacedSender::kDefaultPaceMultiplier *
-                         BitrateController::kDefaultStartBitrateKbps,
-                     0);
-  MockVieEncoder encoder(&process_thread, &pacer);
+class VieKeyRequestTest : public ::testing::Test {
+ public:
+  VieKeyRequestTest()
+      : encoder_(&process_thread_),
+        simulated_clock_(123456789),
+        encoder_state_feedback_(
+            &simulated_clock_,
+            std::vector<uint32_t>(1, VieKeyRequestTest::kSsrc),
+            &encoder_) {}
 
-  EncoderStateFeedback encoder_state_feedback;
-  encoder_state_feedback.Init(std::vector<uint32_t>(1, kSsrc), &encoder);
+ protected:
+  const uint32_t kSsrc = 1234;
+  NiceMock<MockProcessThread> process_thread_;
+  MockVieEncoder encoder_;
+  SimulatedClock simulated_clock_;
+  EncoderStateFeedback encoder_state_feedback_;
+};
 
-  EXPECT_CALL(encoder, OnReceivedIntraFrameRequest(kSsrc))
-      .Times(1);
-  encoder_state_feedback.OnReceivedIntraFrameRequest(kSsrc);
+TEST_F(VieKeyRequestTest, CreateAndTriggerRequests) {
+  EXPECT_CALL(encoder_, OnReceivedIntraFrameRequest(0)).Times(1);
+  encoder_state_feedback_.OnReceivedIntraFrameRequest(kSsrc);
 
   const uint8_t sli_picture_id = 3;
-  EXPECT_CALL(encoder, OnReceivedSLI(kSsrc, sli_picture_id))
-      .Times(1);
-  encoder_state_feedback.OnReceivedSLI(kSsrc, sli_picture_id);
+  EXPECT_CALL(encoder_, OnReceivedSLI(sli_picture_id)).Times(1);
+  encoder_state_feedback_.OnReceivedSLI(kSsrc, sli_picture_id);
 
   const uint64_t rpsi_picture_id = 9;
-  EXPECT_CALL(encoder, OnReceivedRPSI(kSsrc, rpsi_picture_id))
-      .Times(1);
-  encoder_state_feedback.OnReceivedRPSI(kSsrc, rpsi_picture_id);
+  EXPECT_CALL(encoder_, OnReceivedRPSI(rpsi_picture_id)).Times(1);
+  encoder_state_feedback_.OnReceivedRPSI(kSsrc, rpsi_picture_id);
+}
+
+TEST_F(VieKeyRequestTest, TooManyOnReceivedIntraFrameRequest) {
+  EXPECT_CALL(encoder_, OnReceivedIntraFrameRequest(0)).Times(1);
+  encoder_state_feedback_.OnReceivedIntraFrameRequest(kSsrc);
+  encoder_state_feedback_.OnReceivedIntraFrameRequest(kSsrc);
+  simulated_clock_.AdvanceTimeMilliseconds(10);
+  encoder_state_feedback_.OnReceivedIntraFrameRequest(kSsrc);
+
+  EXPECT_CALL(encoder_, OnReceivedIntraFrameRequest(0)).Times(1);
+  simulated_clock_.AdvanceTimeMilliseconds(300);
+  encoder_state_feedback_.OnReceivedIntraFrameRequest(kSsrc);
+  encoder_state_feedback_.OnReceivedIntraFrameRequest(kSsrc);
+  encoder_state_feedback_.OnReceivedIntraFrameRequest(kSsrc);
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/end_to_end_tests.cc b/chromium/third_party/webrtc/video/end_to_end_tests.cc
index e000b3f8b5f..b60d2a629a4 100644
--- a/chromium/third_party/webrtc/video/end_to_end_tests.cc
+++ b/chromium/third_party/webrtc/video/end_to_end_tests.cc
@@ -19,10 +19,9 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
-#include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
@@ -266,107 +265,103 @@ TEST_F(EndToEndTest, TransmitsFirstFrame) {
   DestroyStreams();
 }
 
-TEST_F(EndToEndTest, SendsAndReceivesVP9) {
-  class VP9Observer : public test::EndToEndTest,
+class CodecObserver : public test::EndToEndTest,
                       public rtc::VideoSinkInterface<VideoFrame> {
-   public:
-    VP9Observer()
-        : EndToEndTest(2 * kDefaultTimeoutMs),
-          encoder_(VideoEncoder::Create(VideoEncoder::kVp9)),
-          decoder_(VP9Decoder::Create()),
-          frame_counter_(0) {}
+ public:
+  CodecObserver(int no_frames_to_wait_for,
+                VideoRotation rotation_to_test,
+                const std::string& payload_name,
+                webrtc::VideoEncoder* encoder,
+                webrtc::VideoDecoder* decoder)
+      : EndToEndTest(2 * webrtc::EndToEndTest::kDefaultTimeoutMs),
+        no_frames_to_wait_for_(no_frames_to_wait_for),
+        expected_rotation_(rotation_to_test),
+        payload_name_(payload_name),
+        encoder_(encoder),
+        decoder_(decoder),
+        frame_counter_(0) {}
 
-    void PerformTest() override {
-      EXPECT_TRUE(Wait())
-          << "Timed out while waiting for enough frames to be decoded.";
-    }
+  void PerformTest() override {
+    EXPECT_TRUE(Wait())
+        << "Timed out while waiting for enough frames to be decoded.";
+  }
 
-    void ModifyVideoConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector<VideoReceiveStream::Config>* receive_configs,
-        VideoEncoderConfig* encoder_config) override {
-      send_config->encoder_settings.encoder = encoder_.get();
-      send_config->encoder_settings.payload_name = "VP9";
-      send_config->encoder_settings.payload_type = 124;
-      encoder_config->streams[0].min_bitrate_bps = 50000;
-      encoder_config->streams[0].target_bitrate_bps =
-          encoder_config->streams[0].max_bitrate_bps = 2000000;
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override {
+    send_config->encoder_settings.encoder = encoder_.get();
+    send_config->encoder_settings.payload_name = payload_name_;
+    send_config->encoder_settings.payload_type = 126;
+    encoder_config->streams[0].min_bitrate_bps = 50000;
+    encoder_config->streams[0].target_bitrate_bps =
+        encoder_config->streams[0].max_bitrate_bps = 2000000;
+
+    (*receive_configs)[0].renderer = this;
+    (*receive_configs)[0].decoders.resize(1);
+    (*receive_configs)[0].decoders[0].payload_type =
+        send_config->encoder_settings.payload_type;
+    (*receive_configs)[0].decoders[0].payload_name =
+        send_config->encoder_settings.payload_name;
+    (*receive_configs)[0].decoders[0].decoder = decoder_.get();
+  }
 
-      (*receive_configs)[0].renderer = this;
-      (*receive_configs)[0].decoders.resize(1);
-      (*receive_configs)[0].decoders[0].payload_type =
-          send_config->encoder_settings.payload_type;
-      (*receive_configs)[0].decoders[0].payload_name =
-          send_config->encoder_settings.payload_name;
-      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
-    }
+  void OnFrame(const VideoFrame& video_frame) override {
+    EXPECT_EQ(expected_rotation_, video_frame.rotation());
+    if (++frame_counter_ == no_frames_to_wait_for_)
+      observation_complete_.Set();
+  }
 
-    void OnFrame(const VideoFrame& video_frame) override {
-      const int kRequiredFrames = 500;
-      if (++frame_counter_ == kRequiredFrames)
-        observation_complete_.Set();
-    }
+  void OnFrameGeneratorCapturerCreated(
+      test::FrameGeneratorCapturer* frame_generator_capturer) override {
+    frame_generator_capturer->SetFakeRotation(expected_rotation_);
+  }
 
-   private:
-    std::unique_ptr<webrtc::VideoEncoder> encoder_;
-    std::unique_ptr<webrtc::VideoDecoder> decoder_;
-    int frame_counter_;
-  } test;
+ private:
+  int no_frames_to_wait_for_;
+  VideoRotation expected_rotation_;
+  std::string payload_name_;
+  std::unique_ptr<webrtc::VideoEncoder> encoder_;
+  std::unique_ptr<webrtc::VideoDecoder> decoder_;
+  int frame_counter_;
+};
 
+TEST_F(EndToEndTest, SendsAndReceivesVP8Rotation90) {
+  CodecObserver test(5, kVideoRotation_90, "VP8",
+                     VideoEncoder::Create(VideoEncoder::kVp8),
+                     VP8Decoder::Create());
   RunBaseTest(&test);
 }
 
-#if defined(WEBRTC_END_TO_END_H264_TESTS)
-
-TEST_F(EndToEndTest, SendsAndReceivesH264) {
-  class H264Observer : public test::EndToEndTest,
-                       public rtc::VideoSinkInterface<VideoFrame> {
-   public:
-    H264Observer()
-        : EndToEndTest(2 * kDefaultTimeoutMs),
-          encoder_(VideoEncoder::Create(VideoEncoder::kH264)),
-          decoder_(H264Decoder::Create()),
-          frame_counter_(0) {}
-
-    void PerformTest() override {
-      EXPECT_TRUE(Wait())
-          << "Timed out while waiting for enough frames to be decoded.";
-    }
-
-    void ModifyVideoConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector<VideoReceiveStream::Config>* receive_configs,
-        VideoEncoderConfig* encoder_config) override {
-      send_config->rtp.nack.rtp_history_ms =
-          (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-      send_config->encoder_settings.encoder = encoder_.get();
-      send_config->encoder_settings.payload_name = "H264";
-      send_config->encoder_settings.payload_type = 126;
-      encoder_config->streams[0].min_bitrate_bps = 50000;
-      encoder_config->streams[0].target_bitrate_bps =
-          encoder_config->streams[0].max_bitrate_bps = 2000000;
+#if !defined(RTC_DISABLE_VP9)
+TEST_F(EndToEndTest, SendsAndReceivesVP9) {
+  CodecObserver test(500, kVideoRotation_0, "VP9",
+                     VideoEncoder::Create(VideoEncoder::kVp9),
+                     VP9Decoder::Create());
+  RunBaseTest(&test);
+}
 
-      (*receive_configs)[0].renderer = this;
-      (*receive_configs)[0].decoders.resize(1);
-      (*receive_configs)[0].decoders[0].payload_type =
-          send_config->encoder_settings.payload_type;
-      (*receive_configs)[0].decoders[0].payload_name =
-          send_config->encoder_settings.payload_name;
-      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
-    }
+TEST_F(EndToEndTest, SendsAndReceivesVP9VideoRotation90) {
+  CodecObserver test(5, kVideoRotation_90, "VP9",
+                     VideoEncoder::Create(VideoEncoder::kVp9),
+                     VP9Decoder::Create());
+  RunBaseTest(&test);
+}
+#endif  // !defined(RTC_DISABLE_VP9)
 
-    void OnFrame(const VideoFrame& video_frame) override {
-      const int kRequiredFrames = 500;
-      if (++frame_counter_ == kRequiredFrames)
-        observation_complete_.Set();
-    }
+#if defined(WEBRTC_END_TO_END_H264_TESTS)
 
-   private:
-    std::unique_ptr<webrtc::VideoEncoder> encoder_;
-    std::unique_ptr<webrtc::VideoDecoder> decoder_;
-    int frame_counter_;
-  } test;
+TEST_F(EndToEndTest, SendsAndReceivesH264) {
+  CodecObserver test(500, kVideoRotation_0, "H264",
+                     VideoEncoder::Create(VideoEncoder::kH264),
+                     H264Decoder::Create());
+  RunBaseTest(&test);
+}
 
+TEST_F(EndToEndTest, SendsAndReceivesH264VideoRotation90) {
+  CodecObserver test(5, kVideoRotation_90, "H264",
+                     VideoEncoder::Create(VideoEncoder::kH264),
+                     H264Decoder::Create());
   RunBaseTest(&test);
 }
 
@@ -864,107 +859,6 @@ TEST_F(EndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
   DecodesRetransmittedFrame(true, true);
 }
 
-TEST_F(EndToEndTest, UsesFrameCallbacks) {
-  static const int kWidth = 320;
-  static const int kHeight = 240;
-
-  class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
-   public:
-    Renderer() : event_(false, false) {}
-
-    void OnFrame(const VideoFrame& video_frame) override {
-      EXPECT_EQ(0, *video_frame.buffer(kYPlane))
-          << "Rendered frame should have zero luma which is applied by the "
-             "pre-render callback.";
-      event_.Set();
-    }
-
-    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
-    rtc::Event event_;
-  } renderer;
-
-  class TestFrameCallback : public I420FrameCallback {
-   public:
-    TestFrameCallback(int expected_luma_byte, int next_luma_byte)
-        : event_(false, false),
-          expected_luma_byte_(expected_luma_byte),
-          next_luma_byte_(next_luma_byte) {}
-
-    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
-
-   private:
-    virtual void FrameCallback(VideoFrame* frame) {
-      EXPECT_EQ(kWidth, frame->width())
-          << "Width not as expected, callback done before resize?";
-      EXPECT_EQ(kHeight, frame->height())
-          << "Height not as expected, callback done before resize?";
-
-      // Previous luma specified, observed luma should be fairly close.
-      if (expected_luma_byte_ != -1) {
-        EXPECT_NEAR(expected_luma_byte_, *frame->buffer(kYPlane), 10);
-      }
-
-      memset(frame->buffer(kYPlane),
-             next_luma_byte_,
-             frame->allocated_size(kYPlane));
-
-      event_.Set();
-    }
-
-    rtc::Event event_;
-    int expected_luma_byte_;
-    int next_luma_byte_;
-  };
-
-  TestFrameCallback pre_encode_callback(-1, 255);  // Changes luma to 255.
-  TestFrameCallback pre_render_callback(255, 0);  // Changes luma from 255 to 0.
-
-  CreateCalls(Call::Config(), Call::Config());
-
-  test::DirectTransport sender_transport(sender_call_.get());
-  test::DirectTransport receiver_transport(receiver_call_.get());
-  sender_transport.SetReceiver(receiver_call_->Receiver());
-  receiver_transport.SetReceiver(sender_call_->Receiver());
-
-  CreateSendConfig(1, 0, &sender_transport);
-  std::unique_ptr<VideoEncoder> encoder(
-      VideoEncoder::Create(VideoEncoder::kVp8));
-  video_send_config_.encoder_settings.encoder = encoder.get();
-  video_send_config_.encoder_settings.payload_name = "VP8";
-  ASSERT_EQ(1u, video_encoder_config_.streams.size()) << "Test setup error.";
-  video_encoder_config_.streams[0].width = kWidth;
-  video_encoder_config_.streams[0].height = kHeight;
-  video_send_config_.pre_encode_callback = &pre_encode_callback;
-
-  CreateMatchingReceiveConfigs(&receiver_transport);
-  video_receive_configs_[0].pre_render_callback = &pre_render_callback;
-  video_receive_configs_[0].renderer = &renderer;
-
-  CreateVideoStreams();
-  Start();
-
-  // Create frames that are smaller than the send width/height, this is done to
-  // check that the callbacks are done after processing video.
-  std::unique_ptr<test::FrameGenerator> frame_generator(
-      test::FrameGenerator::CreateChromaGenerator(kWidth / 2, kHeight / 2));
-  video_send_stream_->Input()->IncomingCapturedFrame(
-      *frame_generator->NextFrame());
-
-  EXPECT_TRUE(pre_encode_callback.Wait())
-      << "Timed out while waiting for pre-encode callback.";
-  EXPECT_TRUE(pre_render_callback.Wait())
-      << "Timed out while waiting for pre-render callback.";
-  EXPECT_TRUE(renderer.Wait())
-      << "Timed out while waiting for the frame to render.";
-
-  Stop();
-
-  sender_transport.StopSending();
-  receiver_transport.StopSending();
-
-  DestroyStreams();
-}
-
 void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
   static const int kPacketsToDrop = 1;
 
@@ -1872,7 +1766,7 @@ TEST_F(EndToEndTest, RembWithSendSideBwe) {
 
     ~BweObserver() {}
 
-    test::PacketTransport* CreateReceiveTransport() {
+    test::PacketTransport* CreateReceiveTransport() override {
       receive_transport_ = new test::PacketTransport(
           nullptr, this, test::PacketTransport::kReceiver,
           FakeNetworkPipe::Config());
@@ -2254,6 +2148,9 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx,
       "WebRTC.Video.DecodedFramesPerSecond"));
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.RenderFramesPerSecond"));
 
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.JitterBufferDelayInMs"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.TargetDelayInMs"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.CurrentDelayInMs"));
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.OnewayDelayInMs"));
   EXPECT_EQ(
       1, test::NumHistogramSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
@@ -2280,6 +2177,7 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx,
   EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SendSideDelayInMs"));
   EXPECT_EQ(1,
             test::NumHistogramSamples(video_prefix + "SendSideDelayMaxInMs"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.SendDelayInMs"));
 
   int num_rtx_samples = use_rtx ? 1 : 0;
   EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
@@ -2352,7 +2250,7 @@ void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
       return SEND_PACKET;
     }
     // Send stream should send SR packets (and DLRR packets if enabled).
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
@@ -2571,7 +2469,8 @@ TEST_F(EndToEndTest, ReportsSetEncoderRates) {
 TEST_F(EndToEndTest, GetStats) {
   static const int kStartBitrateBps = 3000000;
   static const int kExpectedRenderDelayMs = 20;
-  class StatsObserver : public test::EndToEndTest, public I420FrameCallback {
+  class StatsObserver : public test::EndToEndTest,
+                        public rtc::VideoSinkInterface<VideoFrame> {
    public:
     StatsObserver()
         : EndToEndTest(kLongTimeoutMs),
@@ -2601,11 +2500,9 @@ TEST_F(EndToEndTest, GetStats) {
       return SEND_PACKET;
     }
 
-    void FrameCallback(VideoFrame* video_frame) override {
+    void OnFrame(const VideoFrame& video_frame) override {
       // Ensure that we have at least 5ms send side delay.
-      int64_t render_time = video_frame->render_time_ms();
-      if (render_time > 0)
-        video_frame->set_render_time_ms(render_time - 5);
+      SleepMs(5);
     }
 
     bool CheckReceiveStats() {
@@ -3038,7 +2935,9 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
       } else {
         // Verify timestamps are reasonably close.
         uint32_t latest_observed = timestamp_it->second;
-        int32_t timestamp_gap = rtc::TimeDiff(timestamp, latest_observed);
+        // Wraparound handling is unnecessary here as long as an int variable
+        // is used to store the result.
+        int32_t timestamp_gap = timestamp - latest_observed;
         EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap)
             << "Gap in timestamps (" << latest_observed << " -> "
             << timestamp << ") too large for SSRC: " << ssrc << ".";
@@ -3442,6 +3341,13 @@ TEST_F(EndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
   class UnusedEncoder : public test::FakeEncoder {
    public:
     UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
+
+    int32_t InitEncode(const VideoCodec* config,
+                       int32_t number_of_cores,
+                       size_t max_payload_size) override {
+      EXPECT_GT(config->startBitrate, 0u);
+      return 0;
+    }
     int32_t Encode(const VideoFrame& input_image,
                    const CodecSpecificInfo* codec_specific_info,
                    const std::vector<FrameType>* frame_types) override {
diff --git a/chromium/third_party/webrtc/video/full_stack.cc b/chromium/third_party/webrtc/video/full_stack.cc
index 8a97ba8a9a1..0399ed2ed73 100644
--- a/chromium/third_party/webrtc/video/full_stack.cc
+++ b/chromium/third_party/webrtc/video/full_stack.cc
@@ -56,6 +56,7 @@ class FullStackTest : public VideoQualityTest {
 //   logs          // bool
 // };
 
+#if !defined(RTC_DISABLE_VP9)
 TEST_F(FullStackTest, ForemanCifWithoutPacketLossVp9) {
   ForemanCifWithoutPacketLoss("VP9");
 }
@@ -63,6 +64,7 @@ TEST_F(FullStackTest, ForemanCifWithoutPacketLossVp9) {
 TEST_F(FullStackTest, ForemanCifPlr5Vp9) {
   ForemanCifPlr5("VP9");
 }
+#endif  // !defined(RTC_DISABLE_VP9)
 
 TEST_F(FullStackTest, ParisQcifWithoutPacketLoss) {
   VideoQualityTest::Params paris_qcif = {
@@ -200,6 +202,7 @@ TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_VeryLossyNet) {
   RunTest(screenshare);
 }
 
+#if !defined(RTC_DISABLE_VP9)
 TEST_F(FullStackTest, ScreenshareSlidesVP9_2SL) {
   VideoQualityTest::Params screenshare = {
       {1850, 1110, 5, 50000, 200000, 2000000, "VP9", 1, 0, 400000},
@@ -211,4 +214,5 @@ TEST_F(FullStackTest, ScreenshareSlidesVP9_2SL) {
       {std::vector<VideoStream>(), 0, 2, 1}};
   RunTest(screenshare);
 }
+#endif  // !defined(RTC_DISABLE_VP9)
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/overuse_frame_detector.cc b/chromium/third_party/webrtc/video/overuse_frame_detector.cc
index 1ce97771ee1..8498008f974 100644
--- a/chromium/third_party/webrtc/video/overuse_frame_detector.cc
+++ b/chromium/third_party/webrtc/video/overuse_frame_detector.cc
@@ -20,13 +20,13 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/exp_filter.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video_frame.h"
 
-#if defined(WEBRTC_MAC)
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
 #include <mach/mach.h>
-#endif
+#endif  // defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
 
 namespace webrtc {
 
@@ -56,7 +56,7 @@ CpuOveruseOptions::CpuOveruseOptions()
       min_frame_samples(120),
       min_process_count(3),
       high_threshold_consecutive_count(2) {
-#if defined(WEBRTC_MAC)
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
   // This is proof-of-concept code for letting the physical core count affect
   // the interval into which we attempt to scale. For now, the code is Mac OS
   // specific, since that's the platform were we saw most problems.
@@ -90,8 +90,8 @@ CpuOveruseOptions::CpuOveruseOptions()
     high_encode_usage_threshold_percent = 20;  // Roughly 1/4 of 100%.
   else if (n_physical_cores == 2)
     high_encode_usage_threshold_percent = 40;  // Roughly 1/4 of 200%.
+#endif  // defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
 
-#endif  // WEBRTC_MAC
   // Note that we make the interval 2x+epsilon wide, since libyuv scaling steps
   // are close to that (when squared). This wide interval makes sure that
   // scaling up or down does not jump all the way across the interval.
diff --git a/chromium/third_party/webrtc/video/overuse_frame_detector_unittest.cc b/chromium/third_party/webrtc/video/overuse_frame_detector_unittest.cc
index 06cff38bf61..67d05325e3f 100644
--- a/chromium/third_party/webrtc/video/overuse_frame_detector_unittest.cc
+++ b/chromium/third_party/webrtc/video/overuse_frame_detector_unittest.cc
@@ -53,7 +53,7 @@ class CpuOveruseObserverImpl : public CpuOveruseObserver {
 class OveruseFrameDetectorTest : public ::testing::Test,
                                  public CpuOveruseMetricsObserver {
  protected:
-  virtual void SetUp() {
+  void SetUp() override {
     clock_.reset(new SimulatedClock(1234));
     observer_.reset(new MockCpuOveruseObserver());
     options_.min_process_count = 0;
diff --git a/chromium/third_party/webrtc/video/payload_router.cc b/chromium/third_party/webrtc/video/payload_router.cc
index 968d82df62a..3be5882cdbf 100644
--- a/chromium/third_party/webrtc/video/payload_router.cc
+++ b/chromium/third_party/webrtc/video/payload_router.cc
@@ -13,11 +13,88 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 
 namespace webrtc {
 
-PayloadRouter::PayloadRouter()
-    : active_(false), num_sending_modules_(0) {}
+namespace {
+// Map information from info into rtp.
+void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
+  RTC_DCHECK(info);
+  switch (info->codecType) {
+    case kVideoCodecVP8: {
+      rtp->codec = kRtpVideoVp8;
+      rtp->codecHeader.VP8.InitRTPVideoHeaderVP8();
+      rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
+      rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference;
+      rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
+      rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync;
+      rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx;
+      rtp->codecHeader.VP8.keyIdx = info->codecSpecific.VP8.keyIdx;
+      rtp->simulcastIdx = info->codecSpecific.VP8.simulcastIdx;
+      return;
+    }
+    case kVideoCodecVP9: {
+      rtp->codec = kRtpVideoVp9;
+      rtp->codecHeader.VP9.InitRTPVideoHeaderVP9();
+      rtp->codecHeader.VP9.inter_pic_predicted =
+          info->codecSpecific.VP9.inter_pic_predicted;
+      rtp->codecHeader.VP9.flexible_mode =
+          info->codecSpecific.VP9.flexible_mode;
+      rtp->codecHeader.VP9.ss_data_available =
+          info->codecSpecific.VP9.ss_data_available;
+      rtp->codecHeader.VP9.picture_id = info->codecSpecific.VP9.picture_id;
+      rtp->codecHeader.VP9.tl0_pic_idx = info->codecSpecific.VP9.tl0_pic_idx;
+      rtp->codecHeader.VP9.temporal_idx = info->codecSpecific.VP9.temporal_idx;
+      rtp->codecHeader.VP9.spatial_idx = info->codecSpecific.VP9.spatial_idx;
+      rtp->codecHeader.VP9.temporal_up_switch =
+          info->codecSpecific.VP9.temporal_up_switch;
+      rtp->codecHeader.VP9.inter_layer_predicted =
+          info->codecSpecific.VP9.inter_layer_predicted;
+      rtp->codecHeader.VP9.gof_idx = info->codecSpecific.VP9.gof_idx;
+      rtp->codecHeader.VP9.num_spatial_layers =
+          info->codecSpecific.VP9.num_spatial_layers;
+
+      if (info->codecSpecific.VP9.ss_data_available) {
+        rtp->codecHeader.VP9.spatial_layer_resolution_present =
+            info->codecSpecific.VP9.spatial_layer_resolution_present;
+        if (info->codecSpecific.VP9.spatial_layer_resolution_present) {
+          for (size_t i = 0; i < info->codecSpecific.VP9.num_spatial_layers;
+               ++i) {
+            rtp->codecHeader.VP9.width[i] = info->codecSpecific.VP9.width[i];
+            rtp->codecHeader.VP9.height[i] = info->codecSpecific.VP9.height[i];
+          }
+        }
+        rtp->codecHeader.VP9.gof.CopyGofInfoVP9(info->codecSpecific.VP9.gof);
+      }
+
+      rtp->codecHeader.VP9.num_ref_pics = info->codecSpecific.VP9.num_ref_pics;
+      for (int i = 0; i < info->codecSpecific.VP9.num_ref_pics; ++i)
+        rtp->codecHeader.VP9.pid_diff[i] = info->codecSpecific.VP9.p_diff[i];
+      return;
+    }
+    case kVideoCodecH264:
+      rtp->codec = kRtpVideoH264;
+      return;
+    case kVideoCodecGeneric:
+      rtp->codec = kRtpVideoGeneric;
+      rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
+      return;
+    default:
+      return;
+  }
+}
+
+}  // namespace
+
+PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
+                             int payload_type)
+    : active_(false),
+      num_sending_modules_(1),
+      rtp_modules_(rtp_modules),
+      payload_type_(payload_type) {
+  UpdateModuleSendingState();
+}
 
 PayloadRouter::~PayloadRouter() {}
 
@@ -26,12 +103,6 @@ size_t PayloadRouter::DefaultMaxPayloadLength() {
   return IP_PACKET_SIZE - kIpUdpSrtpLength;
 }
 
-void PayloadRouter::Init(
-    const std::vector<RtpRtcp*>& rtp_modules) {
-  RTC_DCHECK(rtp_modules_.empty());
-  rtp_modules_ = rtp_modules;
-}
-
 void PayloadRouter::set_active(bool active) {
   rtc::CritScope lock(&crit_);
   if (active_ == active)
@@ -45,10 +116,12 @@ bool PayloadRouter::active() {
   return active_ && !rtp_modules_.empty();
 }
 
-void PayloadRouter::SetSendingRtpModules(size_t num_sending_modules) {
-  RTC_DCHECK_LE(num_sending_modules, rtp_modules_.size());
+void PayloadRouter::SetSendStreams(const std::vector<VideoStream>& streams) {
+  RTC_DCHECK_LE(streams.size(), rtp_modules_.size());
   rtc::CritScope lock(&crit_);
-  num_sending_modules_ = num_sending_modules;
+  num_sending_modules_ = streams.size();
+  streams_ = streams;
+  // TODO(perkj): Should SetSendStreams also call SetTargetSendBitrate?
   UpdateModuleSendingState();
 }
 
@@ -64,39 +137,51 @@ void PayloadRouter::UpdateModuleSendingState() {
   }
 }
 
-bool PayloadRouter::RoutePayload(FrameType frame_type,
-                                 int8_t payload_type,
-                                 uint32_t time_stamp,
-                                 int64_t capture_time_ms,
-                                 const uint8_t* payload_data,
-                                 size_t payload_length,
-                                 const RTPFragmentationHeader* fragmentation,
-                                 const RTPVideoHeader* rtp_video_hdr) {
+int32_t PayloadRouter::Encoded(const EncodedImage& encoded_image,
+                               const CodecSpecificInfo* codec_specific_info,
+                               const RTPFragmentationHeader* fragmentation) {
   rtc::CritScope lock(&crit_);
   RTC_DCHECK(!rtp_modules_.empty());
   if (!active_ || num_sending_modules_ == 0)
-    return false;
+    return -1;
 
   int stream_idx = 0;
-  if (rtp_video_hdr) {
-    RTC_DCHECK_LT(rtp_video_hdr->simulcastIdx, rtp_modules_.size());
-    // The simulcast index might actually be larger than the number of modules
-    // in case the encoder was processing a frame during a codec reconfig.
-    if (rtp_video_hdr->simulcastIdx >= num_sending_modules_)
-      return false;
-    stream_idx = rtp_video_hdr->simulcastIdx;
-  }
+
+  RTPVideoHeader rtp_video_header;
+  memset(&rtp_video_header, 0, sizeof(RTPVideoHeader));
+  if (codec_specific_info)
+    CopyCodecSpecific(codec_specific_info, &rtp_video_header);
+  rtp_video_header.rotation = encoded_image.rotation_;
+
+  RTC_DCHECK_LT(rtp_video_header.simulcastIdx, rtp_modules_.size());
+  // The simulcast index might actually be larger than the number of modules
+  // in case the encoder was processing a frame during a codec reconfig.
+  if (rtp_video_header.simulcastIdx >= num_sending_modules_)
+    return -1;
+  stream_idx = rtp_video_header.simulcastIdx;
+
   return rtp_modules_[stream_idx]->SendOutgoingData(
-      frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
-      payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
+      encoded_image._frameType, payload_type_, encoded_image._timeStamp,
+      encoded_image.capture_time_ms_, encoded_image._buffer,
+      encoded_image._length, fragmentation, &rtp_video_header);
 }
 
-void PayloadRouter::SetTargetSendBitrates(
-    const std::vector<uint32_t>& stream_bitrates) {
+void PayloadRouter::SetTargetSendBitrate(uint32_t bitrate_bps) {
   rtc::CritScope lock(&crit_);
-  RTC_DCHECK_LE(stream_bitrates.size(), rtp_modules_.size());
-  for (size_t i = 0; i < stream_bitrates.size(); ++i) {
-    rtp_modules_[i]->SetTargetSendBitrate(stream_bitrates[i]);
+  RTC_DCHECK_LE(streams_.size(), rtp_modules_.size());
+
+  // TODO(sprang): Rebase https://codereview.webrtc.org/1913073002/ on top of
+  // this.
+  int bitrate_remainder = bitrate_bps;
+  for (size_t i = 0; i < streams_.size() && bitrate_remainder > 0; ++i) {
+    int stream_bitrate = 0;
+    if (streams_[i].max_bitrate_bps > bitrate_remainder) {
+      stream_bitrate = bitrate_remainder;
+    } else {
+      stream_bitrate = streams_[i].max_bitrate_bps;
+    }
+    bitrate_remainder -= stream_bitrate;
+    rtp_modules_[i]->SetTargetSendBitrate(stream_bitrate);
   }
 }
 
diff --git a/chromium/third_party/webrtc/video/payload_router.h b/chromium/third_party/webrtc/video/payload_router.h
index 9eaf7163224..ce65bae6f81 100644
--- a/chromium/third_party/webrtc/video/payload_router.h
+++ b/chromium/third_party/webrtc/video/payload_router.h
@@ -17,6 +17,8 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
+#include "webrtc/config.h"
+#include "webrtc/video_encoder.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 
 namespace webrtc {
@@ -27,37 +29,29 @@ struct RTPVideoHeader;
 
 // PayloadRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
-class PayloadRouter {
+class PayloadRouter : public EncodedImageCallback {
  public:
-  PayloadRouter();
+  // Rtp modules are assumed to be sorted in simulcast index order.
+  explicit PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
+                         int payload_type);
   ~PayloadRouter();
 
   static size_t DefaultMaxPayloadLength();
-
-  // Rtp modules are assumed to be sorted in simulcast index order.
-  void Init(const std::vector<RtpRtcp*>& rtp_modules);
-
-  void SetSendingRtpModules(size_t num_sending_modules);
+  void SetSendStreams(const std::vector<VideoStream>& streams);
 
   // PayloadRouter will only route packets if being active, all packets will be
   // dropped otherwise.
   void set_active(bool active);
   bool active();
 
-  // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
-  // Returns true if the packet was routed / sent, false otherwise.
-  bool RoutePayload(FrameType frame_type,
-                    int8_t payload_type,
-                    uint32_t time_stamp,
-                    int64_t capture_time_ms,
-                    const uint8_t* payload_data,
-                    size_t payload_size,
-                    const RTPFragmentationHeader* fragmentation,
-                    const RTPVideoHeader* rtp_video_hdr);
-
-  // Configures current target bitrate per module. 'stream_bitrates' is assumed
-  // to be in the same order as 'SetSendingRtpModules'.
-  void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
+  // Implements EncodedImageCallback.
+  // Returns 0 if the packet was routed / sent, -1 otherwise.
+  int32_t Encoded(const EncodedImage& encoded_image,
+                  const CodecSpecificInfo* codec_specific_info,
+                  const RTPFragmentationHeader* fragmentation) override;
+
+  // Configures current target bitrate.
+  void SetTargetSendBitrate(uint32_t bitrate_bps);
 
   // Returns the maximum allowed data payload length, given the configured MTU
   // and RTP headers.
@@ -66,13 +60,15 @@ class PayloadRouter {
  private:
   void UpdateModuleSendingState() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
-  // TODO(pbos): Set once and for all on construction and make const.
-  std::vector<RtpRtcp*> rtp_modules_;
-
   rtc::CriticalSection crit_;
   bool active_ GUARDED_BY(crit_);
+  std::vector<VideoStream> streams_ GUARDED_BY(crit_);
   size_t num_sending_modules_ GUARDED_BY(crit_);
 
+  // Rtp modules are assumed to be sorted in simulcast index order. Not owned.
+  const std::vector<RtpRtcp*> rtp_modules_;
+  const int payload_type_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
 };
 
diff --git a/chromium/third_party/webrtc/video/payload_router_unittest.cc b/chromium/third_party/webrtc/video/payload_router_unittest.cc
index 5b3dc9340a5..5b6612124c2 100644
--- a/chromium/third_party/webrtc/video/payload_router_unittest.cc
+++ b/chromium/third_party/webrtc/video/payload_router_unittest.cc
@@ -14,6 +14,7 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/video/payload_router.h"
 
 using ::testing::_;
@@ -23,135 +24,148 @@ using ::testing::Return;
 
 namespace webrtc {
 
-class PayloadRouterTest : public ::testing::Test {
- protected:
-  virtual void SetUp() {
-    payload_router_.reset(new PayloadRouter());
-  }
-  std::unique_ptr<PayloadRouter> payload_router_;
-};
-
-TEST_F(PayloadRouterTest, SendOnOneModule) {
-  MockRtpRtcp rtp;
+TEST(PayloadRouterTest, SendOnOneModule) {
+  NiceMock<MockRtpRtcp> rtp;
   std::vector<RtpRtcp*> modules(1, &rtp);
-
-  payload_router_->Init(modules);
-  payload_router_->SetSendingRtpModules(modules.size());
+  std::vector<VideoStream> streams(1);
 
   uint8_t payload = 'a';
-  FrameType frame_type = kVideoFrameKey;
   int8_t payload_type = 96;
-
-  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1,
-                                    nullptr, nullptr))
+  EncodedImage encoded_image;
+  encoded_image._timeStamp = 1;
+  encoded_image.capture_time_ms_ = 2;
+  encoded_image._frameType = kVideoFrameKey;
+  encoded_image._buffer = &payload;
+  encoded_image._length = 1;
+
+  PayloadRouter payload_router(modules, payload_type);
+  payload_router.SetSendStreams(streams);
+
+  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
+                                    encoded_image._timeStamp,
+                                    encoded_image.capture_time_ms_, &payload,
+                                    encoded_image._length, nullptr, _))
       .Times(0);
-  EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
-                                             &payload, 1, nullptr, nullptr));
+  EXPECT_EQ(-1, payload_router.Encoded(encoded_image, nullptr, nullptr));
 
-  payload_router_->set_active(true);
-  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1,
-                                    nullptr, nullptr))
+  payload_router.set_active(true);
+  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
+                                    encoded_image._timeStamp,
+                                    encoded_image.capture_time_ms_, &payload,
+                                    encoded_image._length, nullptr, _))
       .Times(1);
-  EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
-                                            &payload, 1, nullptr, nullptr));
+  EXPECT_EQ(0, payload_router.Encoded(encoded_image, nullptr, nullptr));
 
-  payload_router_->set_active(false);
-  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1,
-                                    nullptr, nullptr))
+  payload_router.set_active(false);
+  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
+                                    encoded_image._timeStamp,
+                                    encoded_image.capture_time_ms_, &payload,
+                                    encoded_image._length, nullptr, _))
       .Times(0);
-  EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
-                                             &payload, 1, nullptr, nullptr));
+  EXPECT_EQ(-1, payload_router.Encoded(encoded_image, nullptr, nullptr));
 
-  payload_router_->set_active(true);
-  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1,
-                                    nullptr, nullptr))
+  payload_router.set_active(true);
+  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
+                                    encoded_image._timeStamp,
+                                    encoded_image.capture_time_ms_, &payload,
+                                    encoded_image._length, nullptr, _))
       .Times(1);
-  EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
-                                            &payload, 1, nullptr, nullptr));
-
-  payload_router_->SetSendingRtpModules(0);
-  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1,
-                                    nullptr, nullptr))
+  EXPECT_EQ(0, payload_router.Encoded(encoded_image, nullptr, nullptr));
+
+  streams.clear();
+  payload_router.SetSendStreams(streams);
+  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type,
+                                    encoded_image._timeStamp,
+                                    encoded_image.capture_time_ms_, &payload,
+                                    encoded_image._length, nullptr, _))
       .Times(0);
-  EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
-                                             &payload, 1, nullptr, nullptr));
+  EXPECT_EQ(-1, payload_router.Encoded(encoded_image, nullptr, nullptr));
 }
 
-TEST_F(PayloadRouterTest, SendSimulcast) {
-  MockRtpRtcp rtp_1;
-  MockRtpRtcp rtp_2;
+TEST(PayloadRouterTest, SendSimulcast) {
+  NiceMock<MockRtpRtcp> rtp_1;
+  NiceMock<MockRtpRtcp> rtp_2;
   std::vector<RtpRtcp*> modules;
   modules.push_back(&rtp_1);
   modules.push_back(&rtp_2);
+  std::vector<VideoStream> streams(2);
 
-  payload_router_->Init(modules);
-  payload_router_->SetSendingRtpModules(modules.size());
-
-  uint8_t payload_1 = 'a';
-  FrameType frame_type_1 = kVideoFrameKey;
-  int8_t payload_type_1 = 96;
-  RTPVideoHeader rtp_hdr_1;
-  rtp_hdr_1.simulcastIdx = 0;
-
-  payload_router_->set_active(true);
-  EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1,
-                                      nullptr, &rtp_hdr_1))
+  int8_t payload_type = 96;
+  uint8_t payload = 'a';
+  EncodedImage encoded_image;
+  encoded_image._timeStamp = 1;
+  encoded_image.capture_time_ms_ = 2;
+  encoded_image._frameType = kVideoFrameKey;
+  encoded_image._buffer = &payload;
+  encoded_image._length = 1;
+
+  PayloadRouter payload_router(modules, payload_type);
+  payload_router.SetSendStreams(streams);
+
+  CodecSpecificInfo codec_info_1;
+  memset(&codec_info_1, 0, sizeof(CodecSpecificInfo));
+  codec_info_1.codecType = kVideoCodecVP8;
+  codec_info_1.codecSpecific.VP8.simulcastIdx = 0;
+
+  payload_router.set_active(true);
+  EXPECT_CALL(rtp_1, SendOutgoingData(encoded_image._frameType, payload_type,
+                                      encoded_image._timeStamp,
+                                      encoded_image.capture_time_ms_, &payload,
+                                      encoded_image._length, nullptr, _))
       .Times(1);
   EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
       .Times(0);
-  EXPECT_TRUE(payload_router_->RoutePayload(
-      frame_type_1, payload_type_1, 0, 0, &payload_1, 1, nullptr, &rtp_hdr_1));
-
-  uint8_t payload_2 = 'b';
-  FrameType frame_type_2 = kVideoFrameDelta;
-  int8_t payload_type_2 = 97;
-  RTPVideoHeader rtp_hdr_2;
-  rtp_hdr_2.simulcastIdx = 1;
-  EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1,
-                                      nullptr, &rtp_hdr_2))
+  EXPECT_EQ(0, payload_router.Encoded(encoded_image, &codec_info_1, nullptr));
+
+  CodecSpecificInfo codec_info_2;
+  memset(&codec_info_2, 0, sizeof(CodecSpecificInfo));
+  codec_info_2.codecType = kVideoCodecVP8;
+  codec_info_2.codecSpecific.VP8.simulcastIdx = 1;
+
+  EXPECT_CALL(rtp_2, SendOutgoingData(encoded_image._frameType, payload_type,
+                                      encoded_image._timeStamp,
+                                      encoded_image.capture_time_ms_, &payload,
+                                      encoded_image._length, nullptr, _))
       .Times(1);
   EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
       .Times(0);
-  EXPECT_TRUE(payload_router_->RoutePayload(
-      frame_type_2, payload_type_2, 0, 0, &payload_2, 1, nullptr, &rtp_hdr_2));
+  EXPECT_EQ(0, payload_router.Encoded(encoded_image, &codec_info_2, nullptr));
 
   // Inactive.
-  payload_router_->set_active(false);
+  payload_router.set_active(false);
   EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
       .Times(0);
   EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
       .Times(0);
-  EXPECT_FALSE(payload_router_->RoutePayload(
-      frame_type_1, payload_type_1, 0, 0, &payload_1, 1, nullptr, &rtp_hdr_1));
-  EXPECT_FALSE(payload_router_->RoutePayload(
-      frame_type_2, payload_type_2, 0, 0, &payload_2, 1, nullptr, &rtp_hdr_2));
+  EXPECT_EQ(-1, payload_router.Encoded(encoded_image, &codec_info_1, nullptr));
+  EXPECT_EQ(-1, payload_router.Encoded(encoded_image, &codec_info_2, nullptr));
 
   // Invalid simulcast index.
-  payload_router_->SetSendingRtpModules(1);
-  payload_router_->set_active(true);
+  streams.pop_back();  // Remove a stream.
+  payload_router.SetSendStreams(streams);
+  payload_router.set_active(true);
   EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
       .Times(0);
   EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
       .Times(0);
-  rtp_hdr_1.simulcastIdx = 1;
-  EXPECT_FALSE(payload_router_->RoutePayload(
-      frame_type_1, payload_type_1, 0, 0, &payload_1, 1, nullptr, &rtp_hdr_1));
+  codec_info_2.codecSpecific.VP8.simulcastIdx = 1;
+  EXPECT_EQ(-1, payload_router.Encoded(encoded_image, &codec_info_2, nullptr));
 }
 
-TEST_F(PayloadRouterTest, MaxPayloadLength) {
+TEST(PayloadRouterTest, MaxPayloadLength) {
   // Without any limitations from the modules, verify we get the max payload
   // length for IP/UDP/SRTP with a MTU of 150 bytes.
   const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
-  EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength());
-  EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
-
-  MockRtpRtcp rtp_1;
-  MockRtpRtcp rtp_2;
+  NiceMock<MockRtpRtcp> rtp_1;
+  NiceMock<MockRtpRtcp> rtp_2;
   std::vector<RtpRtcp*> modules;
   modules.push_back(&rtp_1);
   modules.push_back(&rtp_2);
-  payload_router_->Init(modules);
-  payload_router_->SetSendingRtpModules(modules.size());
+  PayloadRouter payload_router(modules, 42);
+
+  EXPECT_EQ(kDefaultMaxLength, PayloadRouter::DefaultMaxPayloadLength());
+  std::vector<VideoStream> streams(2);
+  payload_router.SetSendStreams(streams);
 
   // Modules return a higher length than the default value.
   EXPECT_CALL(rtp_1, MaxDataPayloadLength())
@@ -160,7 +174,7 @@ TEST_F(PayloadRouterTest, MaxPayloadLength) {
   EXPECT_CALL(rtp_2, MaxDataPayloadLength())
       .Times(1)
       .WillOnce(Return(kDefaultMaxLength + 10));
-  EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
+  EXPECT_EQ(kDefaultMaxLength, payload_router.MaxPayloadLength());
 
   // The modules return a value lower than default.
   const size_t kTestMinPayloadLength = 1001;
@@ -170,33 +184,27 @@ TEST_F(PayloadRouterTest, MaxPayloadLength) {
   EXPECT_CALL(rtp_2, MaxDataPayloadLength())
       .Times(1)
       .WillOnce(Return(kTestMinPayloadLength));
-  EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength());
+  EXPECT_EQ(kTestMinPayloadLength, payload_router.MaxPayloadLength());
 }
 
-TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
-  MockRtpRtcp rtp_1;
-  MockRtpRtcp rtp_2;
+TEST(PayloadRouterTest, SetTargetSendBitrates) {
+  NiceMock<MockRtpRtcp> rtp_1;
+  NiceMock<MockRtpRtcp> rtp_2;
   std::vector<RtpRtcp*> modules;
   modules.push_back(&rtp_1);
   modules.push_back(&rtp_2);
-  payload_router_->Init(modules);
-  payload_router_->SetSendingRtpModules(modules.size());
+  PayloadRouter payload_router(modules, 42);
+  std::vector<VideoStream> streams(2);
+  streams[0].max_bitrate_bps = 10000;
+  streams[1].max_bitrate_bps = 100000;
+  payload_router.SetSendStreams(streams);
 
   const uint32_t bitrate_1 = 10000;
   const uint32_t bitrate_2 = 76543;
-  std::vector<uint32_t> bitrates(2, bitrate_1);
-  bitrates[1] = bitrate_2;
   EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
       .Times(1);
   EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
       .Times(1);
-  payload_router_->SetTargetSendBitrates(bitrates);
-
-  bitrates.resize(1);
-  EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
-      .Times(1);
-  EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
-      .Times(0);
-  payload_router_->SetTargetSendBitrates(bitrates);
+  payload_router.SetTargetSendBitrate(bitrate_1 + bitrate_2);
 }
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/receive_statistics_proxy.cc b/chromium/third_party/webrtc/video/receive_statistics_proxy.cc
index ff419249d4f..4757efa81b6 100644
--- a/chromium/third_party/webrtc/video/receive_statistics_proxy.cc
+++ b/chromium/third_party/webrtc/video/receive_statistics_proxy.cc
@@ -27,8 +27,8 @@ ReceiveStatisticsProxy::ReceiveStatisticsProxy(
       // 1000ms window, scale 1000 for ms to s.
       decode_fps_estimator_(1000, 1000),
       renders_fps_estimator_(1000, 1000),
-      render_fps_tracker_(100u, 10u),
-      render_pixel_tracker_(100u, 10u) {
+      render_fps_tracker_(100, 10u),
+      render_pixel_tracker_(100, 10u) {
   stats_.ssrc = config.rtp.remote_ssrc;
   for (auto it : config.rtp.rtx)
     rtx_stats_[it.second.ssrc] = StreamDataCounters();
@@ -63,8 +63,10 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
                                       height);
   }
   int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
-  if (sync_offset_ms != -1)
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms);
+  if (sync_offset_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs",
+                                      sync_offset_ms);
+  }
 
   int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
   if (qp != -1)
@@ -77,6 +79,22 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
   if (decode_ms != -1)
     RTC_LOGGED_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
 
+  int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  if (jb_delay_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
+                                      jb_delay_ms);
+  }
+  int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  if (target_delay_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs",
+                                      target_delay_ms);
+  }
+  int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  if (current_delay_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
+                                      current_delay_ms);
+  }
+
   int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (delay_ms != -1)
     RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
@@ -170,6 +188,9 @@ void ReceiveStatisticsProxy::OnDecoderTiming(int decode_ms,
   stats_.min_playout_delay_ms = min_playout_delay_ms;
   stats_.render_delay_ms = render_delay_ms;
   decode_time_counter_.Add(decode_ms);
+  jitter_buffer_delay_counter_.Add(jitter_buffer_ms);
+  target_delay_counter_.Add(target_delay_ms);
+  current_delay_counter_.Add(current_delay_ms);
   // Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
   // render delay).
   delay_counter_.Add(target_delay_ms + rtt_ms / 2);
diff --git a/chromium/third_party/webrtc/video/receive_statistics_proxy.h b/chromium/third_party/webrtc/video/receive_statistics_proxy.h
index dc612750883..f88bbf51a27 100644
--- a/chromium/third_party/webrtc/video/receive_statistics_proxy.h
+++ b/chromium/third_party/webrtc/video/receive_statistics_proxy.h
@@ -19,10 +19,10 @@
 #include "webrtc/base/ratetracker.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/video/report_block_stats.h"
-#include "webrtc/video/vie_channel.h"
+#include "webrtc/video/video_stream_decoder.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
@@ -108,6 +108,9 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
   SampleCounter render_height_counter_ GUARDED_BY(crit_);
   SampleCounter sync_offset_counter_ GUARDED_BY(crit_);
   SampleCounter decode_time_counter_ GUARDED_BY(crit_);
+  SampleCounter jitter_buffer_delay_counter_ GUARDED_BY(crit_);
+  SampleCounter target_delay_counter_ GUARDED_BY(crit_);
+  SampleCounter current_delay_counter_ GUARDED_BY(crit_);
   SampleCounter delay_counter_ GUARDED_BY(crit_);
   ReportBlockStats report_block_stats_ GUARDED_BY(crit_);
   QpCounters qp_counters_;  // Only accessed on the decoding thread.
diff --git a/chromium/third_party/webrtc/video/rtp_stream_receiver.cc b/chromium/third_party/webrtc/video/rtp_stream_receiver.cc
new file mode 100644
index 00000000000..c305df54135
--- /dev/null
+++ b/chromium/third_party/webrtc/video/rtp_stream_receiver.cc
@@ -0,0 +1,542 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/rtp_stream_receiver.h"
+
+#include <vector>
+
+#include "webrtc/base/logging.h"
+#include "webrtc/common_types.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/video/receive_statistics_proxy.h"
+#include "webrtc/video/vie_remb.h"
+
+namespace webrtc {
+
+std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
+    ReceiveStatistics* receive_statistics,
+    Transport* outgoing_transport,
+    RtcpRttStats* rtt_stats,
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+    RemoteBitrateEstimator* remote_bitrate_estimator,
+    RtpPacketSender* paced_sender,
+    TransportSequenceNumberAllocator* transport_sequence_number_allocator) {
+  RtpRtcp::Configuration configuration;
+  configuration.audio = false;
+  configuration.receiver_only = true;
+  configuration.receive_statistics = receive_statistics;
+  configuration.outgoing_transport = outgoing_transport;
+  configuration.intra_frame_callback = nullptr;
+  configuration.rtt_stats = rtt_stats;
+  configuration.rtcp_packet_type_counter_observer =
+      rtcp_packet_type_counter_observer;
+  configuration.paced_sender = paced_sender;
+  configuration.transport_sequence_number_allocator =
+      transport_sequence_number_allocator;
+  configuration.send_bitrate_observer = nullptr;
+  configuration.send_frame_count_observer = nullptr;
+  configuration.send_side_delay_observer = nullptr;
+  configuration.send_packet_observer = nullptr;
+  configuration.bandwidth_callback = nullptr;
+  configuration.transport_feedback_callback = nullptr;
+
+  std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
+  rtp_rtcp->SetSendingStatus(false);
+  rtp_rtcp->SetSendingMediaStatus(false);
+  rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
+
+  return rtp_rtcp;
+}
+
+static const int kPacketLogIntervalMs = 10000;
+
+RtpStreamReceiver::RtpStreamReceiver(
+    vcm::VideoReceiver* video_receiver,
+    RemoteBitrateEstimator* remote_bitrate_estimator,
+    Transport* transport,
+    RtcpRttStats* rtt_stats,
+    PacedSender* paced_sender,
+    PacketRouter* packet_router,
+    VieRemb* remb,
+    const VideoReceiveStream::Config& config,
+    ReceiveStatisticsProxy* receive_stats_proxy,
+    ProcessThread* process_thread)
+    : clock_(Clock::GetRealTimeClock()),
+      config_(config),
+      video_receiver_(video_receiver),
+      remote_bitrate_estimator_(remote_bitrate_estimator),
+      packet_router_(packet_router),
+      remb_(remb),
+      process_thread_(process_thread),
+      ntp_estimator_(clock_),
+      rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
+      rtp_header_parser_(RtpHeaderParser::Create()),
+      rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
+                                                     this,
+                                                     this,
+                                                     &rtp_payload_registry_)),
+      rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
+      fec_receiver_(FecReceiver::Create(this)),
+      receiving_(false),
+      restored_packet_in_use_(false),
+      last_packet_log_ms_(-1),
+      rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
+                                    transport,
+                                    rtt_stats,
+                                    receive_stats_proxy,
+                                    remote_bitrate_estimator_,
+                                    paced_sender,
+                                    packet_router)) {
+  packet_router_->AddRtpModule(rtp_rtcp_.get());
+  rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy);
+  rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
+
+  RTC_DCHECK(config.rtp.rtcp_mode != RtcpMode::kOff)
+      << "A stream should not be configured with RTCP disabled. This value is "
+         "reserved for internal usage.";
+  RTC_DCHECK(config_.rtp.remote_ssrc != 0);
+  // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
+  RTC_DCHECK(config_.rtp.local_ssrc != 0);
+  RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
+
+  rtp_rtcp_->SetRTCPStatus(config.rtp.rtcp_mode);
+  rtp_rtcp_->SetSSRC(config.rtp.local_ssrc);
+  rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
+  if (config.rtp.remb) {
+    rtp_rtcp_->SetREMBStatus(true);
+    remb_->AddReceiveChannel(rtp_rtcp_.get());
+  }
+
+  for (size_t i = 0; i < config.rtp.extensions.size(); ++i) {
+    EnableReceiveRtpHeaderExtension(config.rtp.extensions[i].name,
+                                    config.rtp.extensions[i].id);
+  }
+
+  static const int kMaxPacketAgeToNack = 450;
+  const int max_reordering_threshold = (config.rtp.nack.rtp_history_ms > 0)
+      ? kMaxPacketAgeToNack : kDefaultMaxReorderingThreshold;
+  rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
+
+  // TODO(pbos): Support multiple RTX, per video payload.
+  for (const auto& kv : config_.rtp.rtx) {
+    RTC_DCHECK(kv.second.ssrc != 0);
+    RTC_DCHECK(kv.second.payload_type != 0);
+
+    rtp_payload_registry_.SetRtxSsrc(kv.second.ssrc);
+    rtp_payload_registry_.SetRtxPayloadType(kv.second.payload_type,
+                                            kv.first);
+  }
+
+  // If set to true, the RTX payload type mapping supplied in
+  // |SetRtxPayloadType| will be used when restoring RTX packets. Without it,
+  // RTX packets will always be restored to the last non-RTX packet payload type
+  // received.
+  // TODO(holmer): When Chrome no longer depends on this being false by default,
+  // always use the mapping and remove this whole codepath.
+  rtp_payload_registry_.set_use_rtx_payload_mapping_on_restore(
+      config_.rtp.use_rtx_payload_mapping_on_restore);
+
+  if (IsFecEnabled()) {
+    VideoCodec ulpfec_codec = {};
+    ulpfec_codec.codecType = kVideoCodecULPFEC;
+    strncpy(ulpfec_codec.plName, "ulpfec", sizeof(ulpfec_codec.plName));
+    ulpfec_codec.plType = config_.rtp.fec.ulpfec_payload_type;
+    RTC_CHECK(SetReceiveCodec(ulpfec_codec));
+
+    VideoCodec red_codec = {};
+    red_codec.codecType = kVideoCodecRED;
+    strncpy(red_codec.plName, "red", sizeof(red_codec.plName));
+    red_codec.plType = config_.rtp.fec.red_payload_type;
+    RTC_CHECK(SetReceiveCodec(red_codec));
+    if (config_.rtp.fec.red_rtx_payload_type != -1) {
+      rtp_payload_registry_.SetRtxPayloadType(
+          config_.rtp.fec.red_rtx_payload_type,
+          config_.rtp.fec.red_payload_type);
+    }
+
+    rtp_rtcp_->SetGenericFECStatus(true,
+                                   config_.rtp.fec.red_payload_type,
+                                   config_.rtp.fec.ulpfec_payload_type);
+  }
+
+  if (config.rtp.rtcp_xr.receiver_reference_time_report)
+    rtp_rtcp_->SetRtcpXrRrtrStatus(true);
+
+  // Stats callback for CNAME changes.
+  rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
+
+  process_thread_->RegisterModule(rtp_receive_statistics_.get());
+  process_thread_->RegisterModule(rtp_rtcp_.get());
+}
+
+RtpStreamReceiver::~RtpStreamReceiver() {
+  process_thread_->DeRegisterModule(rtp_receive_statistics_.get());
+  process_thread_->DeRegisterModule(rtp_rtcp_.get());
+
+  packet_router_->RemoveRtpModule(rtp_rtcp_.get());
+  rtp_rtcp_->SetREMBStatus(false);
+  remb_->RemoveReceiveChannel(rtp_rtcp_.get());
+  UpdateHistograms();
+}
+
+bool RtpStreamReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
+  int8_t old_pltype = -1;
+  if (rtp_payload_registry_.ReceivePayloadType(
+          video_codec.plName, kVideoPayloadTypeFrequency, 0,
+          video_codec.maxBitrate, &old_pltype) != -1) {
+    rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
+  }
+
+  return rtp_receiver_->RegisterReceivePayload(
+             video_codec.plName, video_codec.plType, kVideoPayloadTypeFrequency,
+             0, 0) == 0;
+}
+
+uint32_t RtpStreamReceiver::GetRemoteSsrc() const {
+  return rtp_receiver_->SSRC();
+}
+
+int RtpStreamReceiver::GetCsrcs(uint32_t* csrcs) const {
+  return rtp_receiver_->CSRCs(csrcs);
+}
+
+RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const {
+  return rtp_receiver_.get();
+}
+
+int32_t RtpStreamReceiver::OnReceivedPayloadData(
+    const uint8_t* payload_data,
+    const size_t payload_size,
+    const WebRtcRTPHeader* rtp_header) {
+  RTC_DCHECK(video_receiver_);
+  WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
+  rtp_header_with_ntp.ntp_time_ms =
+      ntp_estimator_.Estimate(rtp_header->header.timestamp);
+  if (video_receiver_->IncomingPacket(payload_data, payload_size,
+                                      rtp_header_with_ntp) != 0) {
+    // Check this...
+    return -1;
+  }
+  return 0;
+}
+
+bool RtpStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
+                                          size_t rtp_packet_length) {
+  RTPHeader header;
+  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
+    return false;
+  }
+  header.payload_type_frequency = kVideoPayloadTypeFrequency;
+  bool in_order = IsPacketInOrder(header);
+  return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
+}
+
+// TODO(pbos): Remove as soon as audio can handle a changing payload type
+// without this callback.
+int32_t RtpStreamReceiver::OnInitializeDecoder(
+    const int8_t payload_type,
+    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+    const int frequency,
+    const size_t channels,
+    const uint32_t rate) {
+  RTC_NOTREACHED();
+  return 0;
+}
+
+void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) {
+  rtp_rtcp_->SetRemoteSSRC(ssrc);
+}
+
+bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet,
+                                   size_t rtp_packet_length,
+                                   const PacketTime& packet_time) {
+  RTC_DCHECK(remote_bitrate_estimator_);
+  {
+    rtc::CritScope lock(&receive_cs_);
+    if (!receiving_) {
+      return false;
+    }
+  }
+
+  RTPHeader header;
+  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
+                                 &header)) {
+    return false;
+  }
+  size_t payload_length = rtp_packet_length - header.headerLength;
+  int64_t arrival_time_ms;
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (packet_time.timestamp != -1)
+    arrival_time_ms = (packet_time.timestamp + 500) / 1000;
+  else
+    arrival_time_ms = now_ms;
+
+  {
+    // Periodically log the RTP header of incoming packets.
+    rtc::CritScope lock(&receive_cs_);
+    if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
+      std::stringstream ss;
+      ss << "Packet received on SSRC: " << header.ssrc << " with payload type: "
+         << static_cast<int>(header.payloadType) << ", timestamp: "
+         << header.timestamp << ", sequence number: " << header.sequenceNumber
+         << ", arrival time: " << arrival_time_ms;
+      if (header.extension.hasTransmissionTimeOffset)
+        ss << ", toffset: " << header.extension.transmissionTimeOffset;
+      if (header.extension.hasAbsoluteSendTime)
+        ss << ", abs send time: " << header.extension.absoluteSendTime;
+      LOG(LS_INFO) << ss.str();
+      last_packet_log_ms_ = now_ms;
+    }
+  }
+
+  remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length,
+                                            header);
+  header.payload_type_frequency = kVideoPayloadTypeFrequency;
+
+  bool in_order = IsPacketInOrder(header);
+  rtp_payload_registry_.SetIncomingPayloadType(header);
+  bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
+  // Update receive statistics after ReceivePacket.
+  // Receive statistics will be reset if the payload type changes (make sure
+  // that the first packet is included in the stats).
+  rtp_receive_statistics_->IncomingPacket(
+      header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
+  return ret;
+}
+
+int32_t RtpStreamReceiver::RequestKeyFrame() {
+  return rtp_rtcp_->RequestKeyFrame();
+}
+
+int32_t RtpStreamReceiver::SliceLossIndicationRequest(
+    const uint64_t picture_id) {
+  return rtp_rtcp_->SendRTCPSliceLossIndication(
+      static_cast<uint8_t>(picture_id));
+}
+
+bool RtpStreamReceiver::IsFecEnabled() const {
+  return config_.rtp.fec.red_payload_type != -1 &&
+      config_.rtp.fec.ulpfec_payload_type != -1;
+}
+
+bool RtpStreamReceiver::IsRetransmissionsEnabled() const {
+  return config_.rtp.nack.rtp_history_ms > 0;
+}
+
+void RtpStreamReceiver::RequestPacketRetransmit(
+    const std::vector<uint16_t>& sequence_numbers) {
+  rtp_rtcp_->SendNack(sequence_numbers);
+}
+
+int32_t RtpStreamReceiver::ResendPackets(const uint16_t* sequence_numbers,
+                                         uint16_t length) {
+  return rtp_rtcp_->SendNACK(sequence_numbers, length);
+}
+
+bool RtpStreamReceiver::ReceivePacket(const uint8_t* packet,
+                                      size_t packet_length,
+                                      const RTPHeader& header,
+                                      bool in_order) {
+  if (rtp_payload_registry_.IsEncapsulated(header)) {
+    return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
+  }
+  const uint8_t* payload = packet + header.headerLength;
+  assert(packet_length >= header.headerLength);
+  size_t payload_length = packet_length - header.headerLength;
+  PayloadUnion payload_specific;
+  if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
+                                                 &payload_specific)) {
+    return false;
+  }
+  return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
+                                          payload_specific, in_order);
+}
+
+bool RtpStreamReceiver::ParseAndHandleEncapsulatingHeader(
+    const uint8_t* packet, size_t packet_length, const RTPHeader& header) {
+  if (rtp_payload_registry_.IsRed(header)) {
+    int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
+    if (packet[header.headerLength] == ulpfec_pt) {
+      rtp_receive_statistics_->FecPacketReceived(header, packet_length);
+      // Notify video_receiver about received FEC packets to avoid NACKing these
+      // packets.
+      NotifyReceiverOfFecPacket(header);
+    }
+    if (fec_receiver_->AddReceivedRedPacket(
+            header, packet, packet_length, ulpfec_pt) != 0) {
+      return false;
+    }
+    return fec_receiver_->ProcessReceivedFec() == 0;
+  } else if (rtp_payload_registry_.IsRtx(header)) {
+    if (header.headerLength + header.paddingLength == packet_length) {
+      // This is an empty packet and should be silently dropped before trying to
+      // parse the RTX header.
+      return true;
+    }
+    // Remove the RTX header and parse the original RTP header.
+    if (packet_length < header.headerLength)
+      return false;
+    if (packet_length > sizeof(restored_packet_))
+      return false;
+    rtc::CritScope lock(&receive_cs_);
+    if (restored_packet_in_use_) {
+      LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
+      return false;
+    }
+    if (!rtp_payload_registry_.RestoreOriginalPacket(
+            restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
+            header)) {
+      LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
+                      << header.ssrc << " payload type: "
+                      << static_cast<int>(header.payloadType);
+      return false;
+    }
+    restored_packet_in_use_ = true;
+    bool ret = OnRecoveredPacket(restored_packet_, packet_length);
+    restored_packet_in_use_ = false;
+    return ret;
+  }
+  return false;
+}
+
+void RtpStreamReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
+  int8_t last_media_payload_type =
+      rtp_payload_registry_.last_received_media_payload_type();
+  if (last_media_payload_type < 0) {
+    LOG(LS_WARNING) << "Failed to get last media payload type.";
+    return;
+  }
+  // Fake an empty media packet.
+  WebRtcRTPHeader rtp_header = {};
+  rtp_header.header = header;
+  rtp_header.header.payloadType = last_media_payload_type;
+  rtp_header.header.paddingLength = 0;
+  PayloadUnion payload_specific;
+  if (!rtp_payload_registry_.GetPayloadSpecifics(last_media_payload_type,
+                                                 &payload_specific)) {
+    LOG(LS_WARNING) << "Failed to get payload specifics.";
+    return;
+  }
+  rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
+  rtp_header.type.Video.rotation = kVideoRotation_0;
+  if (header.extension.hasVideoRotation) {
+    rtp_header.type.Video.rotation =
+        ConvertCVOByteToVideoRotation(header.extension.videoRotation);
+  }
+  OnReceivedPayloadData(nullptr, 0, &rtp_header);
+}
+
+bool RtpStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
+                                    size_t rtcp_packet_length) {
+  {
+    rtc::CritScope lock(&receive_cs_);
+    if (!receiving_) {
+      return false;
+    }
+  }
+
+  rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
+
+  int64_t rtt = 0;
+  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
+  if (rtt == 0) {
+    // Waiting for valid rtt.
+    return true;
+  }
+  uint32_t ntp_secs = 0;
+  uint32_t ntp_frac = 0;
+  uint32_t rtp_timestamp = 0;
+  if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
+                           &rtp_timestamp) != 0) {
+    // Waiting for RTCP.
+    return true;
+  }
+  ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
+
+  return true;
+}
+
+void RtpStreamReceiver::SignalNetworkState(NetworkState state) {
+  rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
+                                               : RtcpMode::kOff);
+}
+
+void RtpStreamReceiver::StartReceive() {
+  rtc::CritScope lock(&receive_cs_);
+  receiving_ = true;
+}
+
+void RtpStreamReceiver::StopReceive() {
+  rtc::CritScope lock(&receive_cs_);
+  receiving_ = false;
+}
+
+bool RtpStreamReceiver::IsPacketInOrder(const RTPHeader& header) const {
+  StreamStatistician* statistician =
+      rtp_receive_statistics_->GetStatistician(header.ssrc);
+  if (!statistician)
+    return false;
+  return statistician->IsPacketInOrder(header.sequenceNumber);
+}
+
+bool RtpStreamReceiver::IsPacketRetransmitted(const RTPHeader& header,
+                                              bool in_order) const {
+  // Retransmissions are handled separately if RTX is enabled.
+  if (rtp_payload_registry_.RtxEnabled())
+    return false;
+  StreamStatistician* statistician =
+      rtp_receive_statistics_->GetStatistician(header.ssrc);
+  if (!statistician)
+    return false;
+  // Check if this is a retransmission.
+  int64_t min_rtt = 0;
+  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr);
+  return !in_order &&
+      statistician->IsRetransmitOfOldPacket(header, min_rtt);
+}
+
+void RtpStreamReceiver::UpdateHistograms() {
+  FecPacketCounter counter = fec_receiver_->GetPacketCounter();
+  if (counter.num_packets > 0) {
+    RTC_LOGGED_HISTOGRAM_PERCENTAGE(
+        "WebRTC.Video.ReceivedFecPacketsInPercent",
+        static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
+  }
+  if (counter.num_fec_packets > 0) {
+    RTC_LOGGED_HISTOGRAM_PERCENTAGE(
+        "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
+        static_cast<int>(counter.num_recovered_packets * 100 /
+                         counter.num_fec_packets));
+  }
+}
+
+void RtpStreamReceiver::EnableReceiveRtpHeaderExtension(
+    const std::string& extension, int id) {
+  // One-byte-extension local identifiers are in the range 1-14 inclusive.
+  RTC_DCHECK_GE(id, 1);
+  RTC_DCHECK_LE(id, 14);
+  RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
+  RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
+      StringToRtpExtensionType(extension), id));
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/vie_receiver.h b/chromium/third_party/webrtc/video/rtp_stream_receiver.h
index 999e66d68b1..4b176f8845c 100644
--- a/chromium/third_party/webrtc/video/vie_receiver.h
+++ b/chromium/third_party/webrtc/video/rtp_stream_receiver.h
@@ -8,63 +8,70 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_VIDEO_VIE_RECEIVER_H_
-#define WEBRTC_VIDEO_VIE_RECEIVER_H_
+#ifndef WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
+#define WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
 
 #include <list>
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
 class FecReceiver;
+class PacedSender;
+class PacketRouter;
+class ProcessThread;
 class RemoteNtpTimeEstimator;
 class ReceiveStatistics;
+class ReceiveStatisticsProxy;
 class RemoteBitrateEstimator;
+class RtcpRttStats;
 class RtpHeaderParser;
 class RTPPayloadRegistry;
 class RtpReceiver;
-class RtpRtcp;
-class VideoCodingModule;
+class Transport;
+class VieRemb;
 
-class ViEReceiver : public RtpData {
+namespace vcm {
+class VideoReceiver;
+}  // namespace vcm
+
+class RtpStreamReceiver : public RtpData, public RtpFeedback,
+                          public VCMFrameTypeCallback,
+                          public VCMPacketRequestCallback {
  public:
-  ViEReceiver(VideoCodingModule* module_vcm,
-              RemoteBitrateEstimator* remote_bitrate_estimator,
-              RtpFeedback* rtp_feedback);
-  ~ViEReceiver();
+  RtpStreamReceiver(vcm::VideoReceiver* video_receiver,
+                    RemoteBitrateEstimator* remote_bitrate_estimator,
+                    Transport* transport,
+                    RtcpRttStats* rtt_stats,
+                    PacedSender* paced_sender,
+                    PacketRouter* packet_router,
+                    VieRemb* remb,
+                    const VideoReceiveStream::Config& config,
+                    ReceiveStatisticsProxy* receive_stats_proxy,
+                    ProcessThread* process_thread);
+  ~RtpStreamReceiver();
 
   bool SetReceiveCodec(const VideoCodec& video_codec);
 
-  void SetNackStatus(bool enable, int max_nack_reordering_threshold);
-  void SetRtxPayloadType(int payload_type, int associated_payload_type);
-  // If set to true, the RTX payload type mapping supplied in
-  // |SetRtxPayloadType| will be used when restoring RTX packets. Without it,
-  // RTX packets will always be restored to the last non-RTX packet payload type
-  // received.
-  void SetUseRtxPayloadMappingOnRestore(bool val);
-  void SetRtxSsrc(uint32_t ssrc);
-  bool GetRtxSsrc(uint32_t* ssrc) const;
-
-  bool IsFecEnabled() const;
-
   uint32_t GetRemoteSsrc() const;
   int GetCsrcs(uint32_t* csrcs) const;
 
-  void Init(const std::vector<RtpRtcp*>& modules);
-
   RtpReceiver* GetRtpReceiver() const;
-
-  void EnableReceiveRtpHeaderExtension(const std::string& extension, int id);
+  RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); }
 
   void StartReceive();
   void StopReceive();
@@ -74,13 +81,35 @@ class ViEReceiver : public RtpData {
                   const PacketTime& packet_time);
   bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
 
+  void SignalNetworkState(NetworkState state);
+
   // Implements RtpData.
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 const size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override;
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
-  ReceiveStatistics* GetReceiveStatistics() const;
+  // Implements RtpFeedback.
+  int32_t OnInitializeDecoder(const int8_t payload_type,
+                              const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+                              const int frequency,
+                              const size_t channels,
+                              const uint32_t rate) override;
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override;
+  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
+
+  // Implements VCMFrameTypeCallback.
+  int32_t RequestKeyFrame() override;
+  int32_t SliceLossIndicationRequest(const uint64_t picture_id) override;
+
+  bool IsFecEnabled() const;
+  bool IsRetransmissionsEnabled() const;
+  // Don't use, still experimental.
+  void RequestPacketRetransmit(const std::vector<uint16_t>& sequence_numbers);
+
+  // Implements VCMPacketRequestCallback.
+  int32_t ResendPackets(const uint16_t* sequenceNumbers,
+                        uint16_t length) override;
 
  private:
   bool ReceivePacket(const uint8_t* packet,
@@ -96,13 +125,15 @@ class ViEReceiver : public RtpData {
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
   void UpdateHistograms();
+  void EnableReceiveRtpHeaderExtension(const std::string& extension, int id);
 
   Clock* const clock_;
-  VideoCodingModule* const vcm_;
+  const VideoReceiveStream::Config config_;
+  vcm::VideoReceiver* const video_receiver_;
   RemoteBitrateEstimator* const remote_bitrate_estimator_;
-
-  // TODO(pbos): Make const and set on construction.
-  std::vector<RtpRtcp*> rtp_rtcp_;
+  PacketRouter* const packet_router_;
+  VieRemb* const remb_;
+  ProcessThread* const process_thread_;
 
   RemoteNtpTimeEstimator ntp_estimator_;
   RTPPayloadRegistry rtp_payload_registry_;
@@ -117,8 +148,10 @@ class ViEReceiver : public RtpData {
   uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
   bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
   int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
+
+  const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_VIE_RECEIVER_H_
+#endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
diff --git a/chromium/third_party/webrtc/video/screenshare_loopback.cc b/chromium/third_party/webrtc/video/screenshare_loopback.cc
index 91002ef9043..912aaaddc45 100644
--- a/chromium/third_party/webrtc/video/screenshare_loopback.cc
+++ b/chromium/third_party/webrtc/video/screenshare_loopback.cc
@@ -41,12 +41,14 @@ int MinBitrateKbps() {
   return static_cast<int>(FLAGS_min_bitrate);
 }
 
-DEFINE_int32(start_bitrate, 200, "Call start bitrate in kbps.");
+DEFINE_int32(start_bitrate,
+             Call::Config::kDefaultStartBitrateBps / 1000,
+             "Call start bitrate in kbps.");
 int StartBitrateKbps() {
   return static_cast<int>(FLAGS_start_bitrate);
 }
 
-DEFINE_int32(target_bitrate, 2000, "Stream target bitrate in kbps.");
+DEFINE_int32(target_bitrate, 200, "Stream target bitrate in kbps.");
 int TargetBitrateKbps() {
   return static_cast<int>(FLAGS_target_bitrate);
 }
diff --git a/chromium/third_party/webrtc/video/send_delay_stats.cc b/chromium/third_party/webrtc/video/send_delay_stats.cc
new file mode 100644
index 00000000000..87010665198
--- /dev/null
+++ b/chromium/third_party/webrtc/video/send_delay_stats.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/send_delay_stats.h"
+
+#include "webrtc/base/logging.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+// Packet with a larger delay are removed and excluded from the delay stats.
+// Set to larger than max histogram delay which is 10000.
+const int64_t kMaxSentPacketDelayMs = 11000;
+const size_t kMaxPacketMapSize = 2000;
+
+// Limit for the maximum number of streams to calculate stats for.
+const size_t kMaxSsrcMapSize = 50;
+const int kMinRequiredSamples = 200;
+}  // namespace
+
+SendDelayStats::SendDelayStats(Clock* clock)
+    : clock_(clock), num_old_packets_(0), num_skipped_packets_(0) {}
+
+SendDelayStats::~SendDelayStats() {
+  if (num_old_packets_ > 0 || num_skipped_packets_ > 0) {
+    LOG(LS_WARNING) << "Delay stats: number of old packets " << num_old_packets_
+                    << ", skipped packets " << num_skipped_packets_
+                    << ". Number of streams " << send_delay_counters_.size();
+  }
+  UpdateHistograms();
+}
+
+void SendDelayStats::UpdateHistograms() {
+  rtc::CritScope lock(&crit_);
+  for (const auto& it : send_delay_counters_) {
+    int send_delay_ms = it.second.Avg(kMinRequiredSamples);
+    if (send_delay_ms != -1) {
+      RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.SendDelayInMs",
+                                        send_delay_ms);
+    }
+  }
+}
+
+void SendDelayStats::AddSsrcs(const VideoSendStream::Config& config) {
+  rtc::CritScope lock(&crit_);
+  if (ssrcs_.size() > kMaxSsrcMapSize)
+    return;
+  for (const auto& ssrc : config.rtp.ssrcs)
+    ssrcs_.insert(ssrc);
+}
+
+void SendDelayStats::OnSendPacket(uint16_t packet_id,
+                                  int64_t capture_time_ms,
+                                  uint32_t ssrc) {
+  // Packet sent to transport.
+  rtc::CritScope lock(&crit_);
+  if (ssrcs_.find(ssrc) == ssrcs_.end())
+    return;
+
+  int64_t now = clock_->TimeInMilliseconds();
+  RemoveOld(now, &packets_);
+
+  if (packets_.size() > kMaxPacketMapSize) {
+    ++num_skipped_packets_;
+    return;
+  }
+  packets_.insert(
+      std::make_pair(packet_id, Packet(ssrc, capture_time_ms, now)));
+}
+
+bool SendDelayStats::OnSentPacket(int packet_id, int64_t time_ms) {
+  // Packet leaving socket.
+  if (packet_id == -1)
+    return false;
+
+  rtc::CritScope lock(&crit_);
+  auto it = packets_.find(packet_id);
+  if (it == packets_.end())
+    return false;
+
+  // TODO(asapersson): Remove SendSideDelayUpdated(), use capture -> sent.
+  // Elapsed time from send (to transport) -> sent (leaving socket).
+  int diff_ms = time_ms - it->second.send_time_ms;
+  send_delay_counters_[it->second.ssrc].Add(diff_ms);
+  packets_.erase(it);
+  return true;
+}
+
+void SendDelayStats::RemoveOld(int64_t now, PacketMap* packets) {
+  while (!packets->empty()) {
+    auto it = packets->begin();
+    if (now - it->second.capture_time_ms < kMaxSentPacketDelayMs)
+      break;
+
+    packets->erase(it);
+    ++num_old_packets_;
+  }
+}
+
+void SendDelayStats::SampleCounter::Add(int sample) {
+  sum += sample;
+  ++num_samples;
+}
+
+int SendDelayStats::SampleCounter::Avg(int min_required_samples) const {
+  if (num_samples < min_required_samples || num_samples == 0)
+    return -1;
+  return (sum + (num_samples / 2)) / num_samples;
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/send_delay_stats.h b/chromium/third_party/webrtc/video/send_delay_stats.h
new file mode 100644
index 00000000000..20a97814c72
--- /dev/null
+++ b/chromium/third_party/webrtc/video/send_delay_stats.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_SEND_DELAY_STATS_H_
+#define WEBRTC_VIDEO_SEND_DELAY_STATS_H_
+
+#include <map>
+#include <memory>
+#include <set>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/video_send_stream.h"
+
+namespace webrtc {
+
+class SendDelayStats : public SendPacketObserver {
+ public:
+  explicit SendDelayStats(Clock* clock);
+  virtual ~SendDelayStats();
+
+  // Adds the configured ssrcs for the rtp streams.
+  // Stats will be calculated for these streams.
+  void AddSsrcs(const VideoSendStream::Config& config);
+
+  // Called when a packet is sent (leaving socket).
+  bool OnSentPacket(int packet_id, int64_t time_ms);
+
+ protected:
+  // From SendPacketObserver.
+  // Called when a packet is sent to the transport.
+  void OnSendPacket(uint16_t packet_id,
+                    int64_t capture_time_ms,
+                    uint32_t ssrc) override;
+
+ private:
+  // Map holding sent packets (mapped by sequence number).
+  struct SequenceNumberOlderThan {
+    bool operator()(uint16_t seq1, uint16_t seq2) const {
+      return IsNewerSequenceNumber(seq2, seq1);
+    }
+  };
+  struct Packet {
+    Packet(uint32_t ssrc, int64_t capture_time_ms, int64_t send_time_ms)
+        : ssrc(ssrc),
+          capture_time_ms(capture_time_ms),
+          send_time_ms(send_time_ms) {}
+    uint32_t ssrc;
+    int64_t capture_time_ms;
+    int64_t send_time_ms;
+  };
+  typedef std::map<uint16_t, Packet, SequenceNumberOlderThan> PacketMap;
+
+  class SampleCounter {
+   public:
+    SampleCounter() : sum(0), num_samples(0) {}
+    ~SampleCounter() {}
+    void Add(int sample);
+    int Avg(int min_required_samples) const;
+
+   private:
+    int sum;
+    int num_samples;
+  };
+
+  void UpdateHistograms();
+  void RemoveOld(int64_t now, PacketMap* packets)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  Clock* const clock_;
+  rtc::CriticalSection crit_;
+
+  PacketMap packets_ GUARDED_BY(crit_);
+  size_t num_old_packets_ GUARDED_BY(crit_);
+  size_t num_skipped_packets_ GUARDED_BY(crit_);
+
+  std::set<uint32_t> ssrcs_ GUARDED_BY(crit_);
+  std::map<uint32_t, SampleCounter> send_delay_counters_
+      GUARDED_BY(crit_);  // Mapped by SSRC.
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_VIDEO_SEND_DELAY_STATS_H_
diff --git a/chromium/third_party/webrtc/video/send_delay_stats_unittest.cc b/chromium/third_party/webrtc/video/send_delay_stats_unittest.cc
new file mode 100644
index 00000000000..44c62c3a452
--- /dev/null
+++ b/chromium/third_party/webrtc/video/send_delay_stats_unittest.cc
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/send_delay_stats.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/test/histogram.h"
+
+namespace webrtc {
+namespace {
+const uint32_t kSsrc1 = 17;
+const uint32_t kSsrc2 = 42;
+const uint32_t kRtxSsrc1 = 18;
+const uint32_t kRtxSsrc2 = 43;
+const uint16_t kPacketId = 2345;
+const int64_t kMaxPacketDelayMs = 11000;
+const int kMinRequiredSamples = 200;
+}  // namespace
+
+class SendDelayStatsTest : public ::testing::Test {
+ public:
+  SendDelayStatsTest() : clock_(1234), config_(CreateConfig()) {}
+  virtual ~SendDelayStatsTest() {}
+
+ protected:
+  virtual void SetUp() {
+    stats_.reset(new SendDelayStats(&clock_));
+    stats_->AddSsrcs(config_);
+  }
+
+  VideoSendStream::Config CreateConfig() {
+    VideoSendStream::Config config(nullptr);
+    config.rtp.ssrcs.push_back(kSsrc1);
+    config.rtp.ssrcs.push_back(kSsrc2);
+    config.rtp.rtx.ssrcs.push_back(kRtxSsrc1);
+    config.rtp.rtx.ssrcs.push_back(kRtxSsrc2);
+    return config;
+  }
+
+  void OnSendPacket(uint16_t id, uint32_t ssrc) {
+    OnSendPacket(id, ssrc, clock_.TimeInMilliseconds());
+  }
+
+  void OnSendPacket(uint16_t id, uint32_t ssrc, int64_t capture_ms) {
+    SendPacketObserver* observer = stats_.get();
+    observer->OnSendPacket(id, capture_ms, ssrc);
+  }
+
+  bool OnSentPacket(uint16_t id) {
+    return stats_->OnSentPacket(id, clock_.TimeInMilliseconds());
+  }
+
+  SimulatedClock clock_;
+  VideoSendStream::Config config_;
+  std::unique_ptr<SendDelayStats> stats_;
+};
+
+TEST_F(SendDelayStatsTest, SentPacketFound) {
+  EXPECT_FALSE(OnSentPacket(kPacketId));
+  OnSendPacket(kPacketId, kSsrc1);
+  EXPECT_TRUE(OnSentPacket(kPacketId));   // Packet found.
+  EXPECT_FALSE(OnSentPacket(kPacketId));  // Packet removed when found.
+}
+
+TEST_F(SendDelayStatsTest, SentPacketNotFoundForNonRegisteredSsrc) {
+  OnSendPacket(kPacketId, kSsrc1);
+  EXPECT_TRUE(OnSentPacket(kPacketId));
+  OnSendPacket(kPacketId + 1, kSsrc2);
+  EXPECT_TRUE(OnSentPacket(kPacketId + 1));
+  OnSendPacket(kPacketId + 2, kRtxSsrc1);  // RTX SSRC not registered.
+  EXPECT_FALSE(OnSentPacket(kPacketId + 2));
+}
+
+TEST_F(SendDelayStatsTest, SentPacketFoundWithMaxSendDelay) {
+  OnSendPacket(kPacketId, kSsrc1);
+  clock_.AdvanceTimeMilliseconds(kMaxPacketDelayMs - 1);
+  OnSendPacket(kPacketId + 1, kSsrc1);       // kPacketId -> not old/removed.
+  EXPECT_TRUE(OnSentPacket(kPacketId));      // Packet found.
+  EXPECT_TRUE(OnSentPacket(kPacketId + 1));  // Packet found.
+}
+
+TEST_F(SendDelayStatsTest, OldPacketsRemoved) {
+  const int64_t kCaptureTimeMs = clock_.TimeInMilliseconds();
+  OnSendPacket(0xffffu, kSsrc1, kCaptureTimeMs);
+  OnSendPacket(0u, kSsrc1, kCaptureTimeMs);
+  OnSendPacket(1u, kSsrc1, kCaptureTimeMs + 1);
+  clock_.AdvanceTimeMilliseconds(kMaxPacketDelayMs);  // 0xffff, 0 -> old.
+  OnSendPacket(2u, kSsrc1, kCaptureTimeMs + 2);
+
+  EXPECT_FALSE(OnSentPacket(0xffffu));  // Old removed.
+  EXPECT_FALSE(OnSentPacket(0u));       // Old removed.
+  EXPECT_TRUE(OnSentPacket(1u));
+  EXPECT_TRUE(OnSentPacket(2u));
+}
+
+TEST_F(SendDelayStatsTest, HistogramsAreUpdated) {
+  test::ClearHistograms();
+  const int64_t kDelayMs1 = 5;
+  const int64_t kDelayMs2 = 10;
+  uint16_t id = 0;
+  for (int i = 0; i < kMinRequiredSamples; ++i) {
+    OnSendPacket(++id, kSsrc1);
+    clock_.AdvanceTimeMilliseconds(kDelayMs1);
+    EXPECT_TRUE(OnSentPacket(id));
+    OnSendPacket(++id, kSsrc2);
+    clock_.AdvanceTimeMilliseconds(kDelayMs2);
+    EXPECT_TRUE(OnSentPacket(id));
+  }
+  stats_.reset();
+  EXPECT_EQ(2, test::NumHistogramSamples("WebRTC.Video.SendDelayInMs"));
+  EXPECT_EQ(kDelayMs2, test::LastHistogramSample("WebRTC.Video.SendDelayInMs"));
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/send_statistics_proxy.cc b/chromium/third_party/webrtc/video/send_statistics_proxy.cc
index 86694c930e7..d8c11a8c358 100644
--- a/chromium/third_party/webrtc/video/send_statistics_proxy.cc
+++ b/chromium/third_party/webrtc/video/send_statistics_proxy.cc
@@ -17,6 +17,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
@@ -96,8 +97,8 @@ SendStatisticsProxy::UmaSamplesContainer::UmaSamplesContainer(
       clock_(clock),
       max_sent_width_per_timestamp_(0),
       max_sent_height_per_timestamp_(0),
-      input_frame_rate_tracker_(100u, 10u),
-      sent_frame_rate_tracker_(100u, 10u),
+      input_frame_rate_tracker_(100, 10u),
+      sent_frame_rate_tracker_(100, 10u),
       first_rtcp_stats_time_ms_(-1),
       first_rtp_stats_time_ms_(-1),
       start_stats_(stats) {}
@@ -194,26 +195,46 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
   }
 
   for (const auto& it : qp_counters_) {
-    int qp = it.second.vp8.Avg(kMinRequiredSamples);
-    if (qp != -1) {
+    int qp_vp8 = it.second.vp8.Avg(kMinRequiredSamples);
+    if (qp_vp8 != -1) {
       int spatial_idx = it.first;
       if (spatial_idx == -1) {
         RTC_LOGGED_HISTOGRAMS_COUNTS_200(kIndex, uma_prefix_ + "Encoded.Qp.Vp8",
-                                         qp);
+                                         qp_vp8);
       } else if (spatial_idx == 0) {
-        RTC_LOGGED_HISTOGRAMS_COUNTS_200(kIndex,
-                                         uma_prefix_ + "Encoded.Qp.Vp8.S0", qp);
+        RTC_LOGGED_HISTOGRAMS_COUNTS_200(
+            kIndex, uma_prefix_ + "Encoded.Qp.Vp8.S0", qp_vp8);
       } else if (spatial_idx == 1) {
-        RTC_LOGGED_HISTOGRAMS_COUNTS_200(kIndex,
-                                         uma_prefix_ + "Encoded.Qp.Vp8.S1", qp);
+        RTC_LOGGED_HISTOGRAMS_COUNTS_200(
+            kIndex, uma_prefix_ + "Encoded.Qp.Vp8.S1", qp_vp8);
       } else if (spatial_idx == 2) {
-        RTC_LOGGED_HISTOGRAMS_COUNTS_200(kIndex,
-                                         uma_prefix_ + "Encoded.Qp.Vp8.S2", qp);
+        RTC_LOGGED_HISTOGRAMS_COUNTS_200(
+            kIndex, uma_prefix_ + "Encoded.Qp.Vp8.S2", qp_vp8);
       } else {
         LOG(LS_WARNING) << "QP stats not recorded for VP8 spatial idx "
                         << spatial_idx;
       }
     }
+    int qp_vp9 = it.second.vp9.Avg(kMinRequiredSamples);
+    if (qp_vp9 != -1) {
+      int spatial_idx = it.first;
+      if (spatial_idx == -1) {
+        RTC_LOGGED_HISTOGRAMS_COUNTS_500(kIndex, uma_prefix_ + "Encoded.Qp.Vp9",
+                                         qp_vp9);
+      } else if (spatial_idx == 0) {
+        RTC_LOGGED_HISTOGRAMS_COUNTS_500(
+            kIndex, uma_prefix_ + "Encoded.Qp.Vp9.S0", qp_vp9);
+      } else if (spatial_idx == 1) {
+        RTC_LOGGED_HISTOGRAMS_COUNTS_500(
+            kIndex, uma_prefix_ + "Encoded.Qp.Vp9.S1", qp_vp9);
+      } else if (spatial_idx == 2) {
+        RTC_LOGGED_HISTOGRAMS_COUNTS_500(
+            kIndex, uma_prefix_ + "Encoded.Qp.Vp9.S2", qp_vp9);
+      } else {
+        LOG(LS_WARNING) << "QP stats not recorded for VP9 spatial layer "
+                        << spatial_idx;
+      }
+    }
   }
 
   if (first_rtcp_stats_time_ms_ != -1) {
@@ -321,16 +342,14 @@ void SendStatisticsProxy::SetContentType(
   }
 }
 
-void SendStatisticsProxy::OnEncoderImplementationName(
-    const char* implementation_name) {
-  rtc::CritScope lock(&crit_);
-  stats_.encoder_implementation_name = implementation_name;
-}
-
-void SendStatisticsProxy::OnOutgoingRate(uint32_t framerate, uint32_t bitrate) {
+void SendStatisticsProxy::OnEncoderStatsUpdate(
+    uint32_t framerate,
+    uint32_t bitrate,
+    const std::string& encoder_name) {
   rtc::CritScope lock(&crit_);
   stats_.encode_frame_rate = framerate;
   stats_.media_bitrate_bps = bitrate;
+  stats_.encoder_implementation_name = encoder_name;
 }
 
 void SendStatisticsProxy::OnEncodedFrameTimeMeasured(
@@ -406,8 +425,17 @@ void SendStatisticsProxy::OnSetRates(uint32_t bitrate_bps, int framerate) {
 
 void SendStatisticsProxy::OnSendEncodedImage(
     const EncodedImage& encoded_image,
-    const RTPVideoHeader* rtp_video_header) {
-  size_t simulcast_idx = rtp_video_header ? rtp_video_header->simulcastIdx : 0;
+    const CodecSpecificInfo* codec_info) {
+  size_t simulcast_idx = 0;
+
+  if (codec_info) {
+    if (codec_info->codecType == kVideoCodecVP8) {
+      simulcast_idx = codec_info->codecSpecific.VP8.simulcastIdx;
+    } else if (codec_info->codecType == kVideoCodecGeneric) {
+      simulcast_idx = codec_info->codecSpecific.generic.simulcast_idx;
+    }
+  }
+
   if (simulcast_idx >= config_.rtp.ssrcs.size()) {
     LOG(LS_ERROR) << "Encoded image outside simulcast range (" << simulcast_idx
                   << " >= " << config_.rtp.ssrcs.size() << ").";
@@ -449,11 +477,18 @@ void SendStatisticsProxy::OnSendEncodedImage(
     }
   }
 
-  if (encoded_image.qp_ != -1 && rtp_video_header &&
-      rtp_video_header->codec == kRtpVideoVp8) {
-    int spatial_idx =
-        (config_.rtp.ssrcs.size() == 1) ? -1 : static_cast<int>(simulcast_idx);
-    uma_container_->qp_counters_[spatial_idx].vp8.Add(encoded_image.qp_);
+  if (encoded_image.qp_ != -1 && codec_info) {
+    if (codec_info->codecType == kVideoCodecVP8) {
+      int spatial_idx = (config_.rtp.ssrcs.size() == 1)
+                            ? -1
+                            : static_cast<int>(simulcast_idx);
+      uma_container_->qp_counters_[spatial_idx].vp8.Add(encoded_image.qp_);
+    } else if (codec_info->codecType == kVideoCodecVP9) {
+      int spatial_idx = (codec_info->codecSpecific.VP9.num_spatial_layers == 1)
+                            ? -1
+                            : codec_info->codecSpecific.VP9.spatial_idx;
+      uma_container_->qp_counters_[spatial_idx].vp9.Add(encoded_image.qp_);
+    }
   }
 
   // TODO(asapersson): This is incorrect if simulcast layers are encoded on
diff --git a/chromium/third_party/webrtc/video/send_statistics_proxy.h b/chromium/third_party/webrtc/video/send_statistics_proxy.h
index 5a479a1e804..74d5261ebd0 100644
--- a/chromium/third_party/webrtc/video/send_statistics_proxy.h
+++ b/chromium/third_party/webrtc/video/send_statistics_proxy.h
@@ -49,12 +49,13 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
   VideoSendStream::Stats GetStats();
 
   virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
-                                  const RTPVideoHeader* rtp_video_header);
+                                  const CodecSpecificInfo* codec_info);
   // Used to update incoming frame rate.
   void OnIncomingFrame(int width, int height);
 
-  void OnEncoderImplementationName(const char* implementation_name);
-  void OnOutgoingRate(uint32_t framerate, uint32_t bitrate);
+  void OnEncoderStatsUpdate(uint32_t framerate,
+                            uint32_t bitrate,
+                            const std::string& encoder_name);
   void OnSuspendChange(bool is_suspended);
   void OnInactiveSsrc(uint32_t ssrc);
 
@@ -126,7 +127,8 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
     int64_t bitrate_update_ms;
   };
   struct QpCounters {
-    SampleCounter vp8;
+    SampleCounter vp8;  // QP range: 0-127
+    SampleCounter vp9;  // QP range: 0-255
   };
   void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
diff --git a/chromium/third_party/webrtc/video/send_statistics_proxy_unittest.cc b/chromium/third_party/webrtc/video/send_statistics_proxy_unittest.cc
index bb404fb8bed..636b9dc40cf 100644
--- a/chromium/third_party/webrtc/video/send_statistics_proxy_unittest.cc
+++ b/chromium/third_party/webrtc/video/send_statistics_proxy_unittest.cc
@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-// This file includes unit tests for SendStatisticsProxy.
 #include "webrtc/video/send_statistics_proxy.h"
 
 #include <map>
@@ -21,11 +20,16 @@
 #include "webrtc/test/histogram.h"
 
 namespace webrtc {
+namespace {
+const uint32_t kFirstSsrc = 17;
+const uint32_t kSecondSsrc = 42;
+const uint32_t kFirstRtxSsrc = 18;
+const uint32_t kSecondRtxSsrc = 43;
 
-static const uint32_t kFirstSsrc = 17;
-static const uint32_t kSecondSsrc = 42;
-static const uint32_t kFirstRtxSsrc = 18;
-static const uint32_t kSecondRtxSsrc = 43;
+const int kMinRequiredSamples = 200;
+const int kQpIdx0 = 21;
+const int kQpIdx1 = 39;
+}  // namespace
 
 class SendStatisticsProxyTest : public ::testing::Test {
  public:
@@ -106,10 +110,7 @@ class SendStatisticsProxyTest : public ::testing::Test {
 
 TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
   RtcpStatisticsCallback* callback = statistics_proxy_.get();
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
-       it != config_.rtp.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.ssrcs) {
     VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc];
 
     // Add statistics with some arbitrary, but unique, numbers.
@@ -120,10 +121,7 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
     ssrc_stats.rtcp_stats.jitter = offset + 3;
     callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc);
   }
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
-       it != config_.rtp.rtx.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.rtx.ssrcs) {
     VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc];
 
     // Add statistics with some arbitrary, but unique, numbers.
@@ -142,11 +140,13 @@ TEST_F(SendStatisticsProxyTest, EncodedBitrateAndFramerate) {
   int media_bitrate_bps = 500;
   int encode_fps = 29;
 
-  statistics_proxy_->OnOutgoingRate(encode_fps, media_bitrate_bps);
+  statistics_proxy_->OnEncoderStatsUpdate(encode_fps, media_bitrate_bps,
+                                          "encoder name");
 
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
   EXPECT_EQ(media_bitrate_bps, stats.media_bitrate_bps);
   EXPECT_EQ(encode_fps, stats.encode_frame_rate);
+  EXPECT_EQ("encoder name", stats.encoder_implementation_name);
 }
 
 TEST_F(SendStatisticsProxyTest, Suspended) {
@@ -164,10 +164,7 @@ TEST_F(SendStatisticsProxyTest, Suspended) {
 
 TEST_F(SendStatisticsProxyTest, FrameCounts) {
   FrameCountObserver* observer = statistics_proxy_.get();
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
-       it != config_.rtp.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.ssrcs) {
     // Add statistics with some arbitrary, but unique, numbers.
     VideoSendStream::StreamStats& stats = expected_.substreams[ssrc];
     uint32_t offset = ssrc * sizeof(VideoSendStream::StreamStats);
@@ -177,10 +174,7 @@ TEST_F(SendStatisticsProxyTest, FrameCounts) {
     stats.frame_counts = frame_counts;
     observer->FrameCountUpdated(frame_counts, ssrc);
   }
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
-       it != config_.rtp.rtx.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.rtx.ssrcs) {
     // Add statistics with some arbitrary, but unique, numbers.
     VideoSendStream::StreamStats& stats = expected_.substreams[ssrc];
     uint32_t offset = ssrc * sizeof(VideoSendStream::StreamStats);
@@ -197,10 +191,7 @@ TEST_F(SendStatisticsProxyTest, FrameCounts) {
 
 TEST_F(SendStatisticsProxyTest, DataCounters) {
   StreamDataCountersCallback* callback = statistics_proxy_.get();
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
-       it != config_.rtp.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.ssrcs) {
     StreamDataCounters& counters = expected_.substreams[ssrc].rtp_stats;
     // Add statistics with some arbitrary, but unique, numbers.
     size_t offset = ssrc * sizeof(StreamDataCounters);
@@ -213,10 +204,7 @@ TEST_F(SendStatisticsProxyTest, DataCounters) {
     counters.transmitted.packets = offset_uint32 + 5;
     callback->DataCountersUpdated(counters, ssrc);
   }
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
-       it != config_.rtp.rtx.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.rtx.ssrcs) {
     StreamDataCounters& counters = expected_.substreams[ssrc].rtp_stats;
     // Add statistics with some arbitrary, but unique, numbers.
     size_t offset = ssrc * sizeof(StreamDataCounters);
@@ -236,10 +224,7 @@ TEST_F(SendStatisticsProxyTest, DataCounters) {
 
 TEST_F(SendStatisticsProxyTest, Bitrate) {
   BitrateStatisticsObserver* observer = statistics_proxy_.get();
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
-       it != config_.rtp.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.ssrcs) {
     BitrateStatistics total;
     BitrateStatistics retransmit;
     // Use ssrc as bitrate_bps to get a unique value for each stream.
@@ -249,10 +234,7 @@ TEST_F(SendStatisticsProxyTest, Bitrate) {
     expected_.substreams[ssrc].total_bitrate_bps = total.bitrate_bps;
     expected_.substreams[ssrc].retransmit_bitrate_bps = retransmit.bitrate_bps;
   }
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
-       it != config_.rtp.rtx.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.rtx.ssrcs) {
     BitrateStatistics total;
     BitrateStatistics retransmit;
     // Use ssrc as bitrate_bps to get a unique value for each stream.
@@ -269,10 +251,7 @@ TEST_F(SendStatisticsProxyTest, Bitrate) {
 
 TEST_F(SendStatisticsProxyTest, SendSideDelay) {
   SendSideDelayObserver* observer = statistics_proxy_.get();
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
-       it != config_.rtp.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.ssrcs) {
     // Use ssrc as avg_delay_ms and max_delay_ms to get a unique value for each
     // stream.
     int avg_delay_ms = ssrc;
@@ -281,10 +260,7 @@ TEST_F(SendStatisticsProxyTest, SendSideDelay) {
     expected_.substreams[ssrc].avg_delay_ms = avg_delay_ms;
     expected_.substreams[ssrc].max_delay_ms = max_delay_ms;
   }
-  for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
-       it != config_.rtp.rtx.ssrcs.end();
-       ++it) {
-    const uint32_t ssrc = *it;
+  for (const auto& ssrc : config_.rtp.rtx.ssrcs) {
     // Use ssrc as avg_delay_ms and max_delay_ms to get a unique value for each
     // stream.
     int avg_delay_ms = ssrc;
@@ -310,7 +286,6 @@ TEST_F(SendStatisticsProxyTest, OnEncodedFrameTimeMeasured) {
 
 TEST_F(SendStatisticsProxyTest, SwitchContentTypeUpdatesHistograms) {
   test::ClearHistograms();
-  const int kMinRequiredSamples = 200;
   const int kWidth = 640;
   const int kHeight = 480;
 
@@ -329,21 +304,17 @@ TEST_F(SendStatisticsProxyTest, SwitchContentTypeUpdatesHistograms) {
 
 TEST_F(SendStatisticsProxyTest, VerifyQpHistogramStats_Vp8) {
   test::ClearHistograms();
-  const int kMinRequiredSamples = 200;
-  const int kQpIdx0 = 21;
-  const int kQpIdx1 = 39;
   EncodedImage encoded_image;
-
-  RTPVideoHeader rtp_video_header;
-  rtp_video_header.codec = kRtpVideoVp8;
+  CodecSpecificInfo codec_info;
+  codec_info.codecType = kVideoCodecVP8;
 
   for (int i = 0; i < kMinRequiredSamples; ++i) {
-    rtp_video_header.simulcastIdx = 0;
+    codec_info.codecSpecific.VP8.simulcastIdx = 0;
     encoded_image.qp_ = kQpIdx0;
-    statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
-    rtp_video_header.simulcastIdx = 1;
+    statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
+    codec_info.codecSpecific.VP8.simulcastIdx = 1;
     encoded_image.qp_ = kQpIdx1;
-    statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+    statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
   }
   statistics_proxy_.reset();
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.Encoded.Qp.Vp8.S0"));
@@ -361,23 +332,66 @@ TEST_F(SendStatisticsProxyTest, VerifyQpHistogramStats_Vp8OneSsrc) {
       &fake_clock_, config, VideoEncoderConfig::ContentType::kRealtimeVideo));
 
   test::ClearHistograms();
-  const int kMinRequiredSamples = 200;
-  const int kQpIdx0 = 21;
   EncodedImage encoded_image;
-
-  RTPVideoHeader rtp_video_header;
-  rtp_video_header.codec = kRtpVideoVp8;
+  CodecSpecificInfo codec_info;
+  codec_info.codecType = kVideoCodecVP8;
 
   for (int i = 0; i < kMinRequiredSamples; ++i) {
-    rtp_video_header.simulcastIdx = 0;
+    codec_info.codecSpecific.VP8.simulcastIdx = 0;
     encoded_image.qp_ = kQpIdx0;
-    statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+    statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
   }
   statistics_proxy_.reset();
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.Encoded.Qp.Vp8"));
   EXPECT_EQ(kQpIdx0, test::LastHistogramSample("WebRTC.Video.Encoded.Qp.Vp8"));
 }
 
+TEST_F(SendStatisticsProxyTest, VerifyQpHistogramStats_Vp9) {
+  test::ClearHistograms();
+  EncodedImage encoded_image;
+  CodecSpecificInfo codec_info;
+  codec_info.codecType = kVideoCodecVP9;
+  codec_info.codecSpecific.VP9.num_spatial_layers = 2;
+
+  for (int i = 0; i < kMinRequiredSamples; ++i) {
+    encoded_image.qp_ = kQpIdx0;
+    codec_info.codecSpecific.VP9.spatial_idx = 0;
+    statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
+    encoded_image.qp_ = kQpIdx1;
+    codec_info.codecSpecific.VP9.spatial_idx = 1;
+    statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
+  }
+  statistics_proxy_.reset();
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.Encoded.Qp.Vp9.S0"));
+  EXPECT_EQ(kQpIdx0,
+            test::LastHistogramSample("WebRTC.Video.Encoded.Qp.Vp9.S0"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.Encoded.Qp.Vp9.S1"));
+  EXPECT_EQ(kQpIdx1,
+            test::LastHistogramSample("WebRTC.Video.Encoded.Qp.Vp9.S1"));
+}
+
+TEST_F(SendStatisticsProxyTest, VerifyQpHistogramStats_Vp9OneSpatialLayer) {
+  VideoSendStream::Config config(nullptr);
+  config.rtp.ssrcs.push_back(kFirstSsrc);
+  statistics_proxy_.reset(new SendStatisticsProxy(
+      &fake_clock_, config, VideoEncoderConfig::ContentType::kRealtimeVideo));
+
+  test::ClearHistograms();
+  EncodedImage encoded_image;
+  CodecSpecificInfo codec_info;
+  codec_info.codecType = kVideoCodecVP9;
+  codec_info.codecSpecific.VP9.num_spatial_layers = 1;
+
+  for (int i = 0; i < kMinRequiredSamples; ++i) {
+    encoded_image.qp_ = kQpIdx0;
+    codec_info.codecSpecific.VP9.spatial_idx = 0;
+    statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
+  }
+  statistics_proxy_.reset();
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.Encoded.Qp.Vp9"));
+  EXPECT_EQ(kQpIdx0, test::LastHistogramSample("WebRTC.Video.Encoded.Qp.Vp9"));
+}
+
 TEST_F(SendStatisticsProxyTest, NoSubstreams) {
   uint32_t excluded_ssrc =
       std::max(
@@ -413,12 +427,13 @@ TEST_F(SendStatisticsProxyTest, EncodedResolutionTimesOut) {
   encoded_image._encodedWidth = kEncodedWidth;
   encoded_image._encodedHeight = kEncodedHeight;
 
-  RTPVideoHeader rtp_video_header;
+  CodecSpecificInfo codec_info;
+  codec_info.codecType = kVideoCodecVP8;
+  codec_info.codecSpecific.VP8.simulcastIdx = 0;
 
-  rtp_video_header.simulcastIdx = 0;
-  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
-  rtp_video_header.simulcastIdx = 1;
-  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
+  codec_info.codecSpecific.VP8.simulcastIdx = 1;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
 
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
   EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].width);
@@ -440,8 +455,8 @@ TEST_F(SendStatisticsProxyTest, EncodedResolutionTimesOut) {
 
   // Report stats for second SSRC to make sure it's not outdated along with the
   // first SSRC.
-  rtp_video_header.simulcastIdx = 1;
-  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+  codec_info.codecSpecific.VP8.simulcastIdx = 1;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
 
   // Forward 1 ms, reach timeout, substream 0 should have no resolution
   // reported, but substream 1 should.
@@ -460,12 +475,13 @@ TEST_F(SendStatisticsProxyTest, ClearsResolutionFromInactiveSsrcs) {
   encoded_image._encodedWidth = kEncodedWidth;
   encoded_image._encodedHeight = kEncodedHeight;
 
-  RTPVideoHeader rtp_video_header;
+  CodecSpecificInfo codec_info;
+  codec_info.codecType = kVideoCodecVP8;
+  codec_info.codecSpecific.VP8.simulcastIdx = 0;
 
-  rtp_video_header.simulcastIdx = 0;
-  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
-  rtp_video_header.simulcastIdx = 1;
-  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
+  codec_info.codecSpecific.VP8.simulcastIdx = 1;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
 
   statistics_proxy_->OnInactiveSsrc(config_.rtp.ssrcs[1]);
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
diff --git a/chromium/third_party/webrtc/video/video_capture_input.cc b/chromium/third_party/webrtc/video/video_capture_input.cc
index 16263ccaf35..8f574e21154 100644
--- a/chromium/third_party/webrtc/video/video_capture_input.cc
+++ b/chromium/third_party/webrtc/video/video_capture_input.cc
@@ -16,7 +16,6 @@
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_capture/video_capture_factory.h"
 #include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
 #include "webrtc/video/overuse_frame_detector.h"
 #include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video/vie_encoder.h"
@@ -84,10 +83,11 @@ void VideoCaptureInput::IncomingCapturedFrame(const VideoFrame& video_frame) {
     return;
   }
 
-  captured_frame_.ShallowCopy(incoming_frame);
+  captured_frame_.reset(new VideoFrame);
+  captured_frame_->ShallowCopy(incoming_frame);
   last_captured_timestamp_ = incoming_frame.ntp_time_ms();
 
-  overuse_detector_->FrameCaptured(captured_frame_);
+  overuse_detector_->FrameCaptured(*captured_frame_);
 
   TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(),
                            "render_time", video_frame.render_time_ms());
@@ -97,11 +97,11 @@ void VideoCaptureInput::IncomingCapturedFrame(const VideoFrame& video_frame) {
 
 bool VideoCaptureInput::GetVideoFrame(VideoFrame* video_frame) {
   rtc::CritScope lock(&crit_);
-  if (captured_frame_.IsZeroSize())
+  if (!captured_frame_)
     return false;
 
-  *video_frame = captured_frame_;
-  captured_frame_.Reset();
+  *video_frame = *captured_frame_;
+  captured_frame_.reset();
   return true;
 }
 
diff --git a/chromium/third_party/webrtc/video/video_capture_input.h b/chromium/third_party/webrtc/video/video_capture_input.h
index a47e8772c1d..5877f6c94fd 100644
--- a/chromium/third_party/webrtc/video/video_capture_input.h
+++ b/chromium/third_party/webrtc/video/video_capture_input.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VIDEO_VIDEO_CAPTURE_INPUT_H_
 #define WEBRTC_VIDEO_VIDEO_CAPTURE_INPUT_H_
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
@@ -53,7 +54,7 @@ class VideoCaptureInput : public webrtc::VideoCaptureInput {
   SendStatisticsProxy* const stats_proxy_;
   rtc::Event* const capture_event_;
 
-  VideoFrame captured_frame_ GUARDED_BY(crit_);
+  std::unique_ptr<VideoFrame> captured_frame_ GUARDED_BY(crit_);
   Clock* const clock_;
   // Used to make sure incoming time stamp is increasing for every frame.
   int64_t last_captured_timestamp_;
diff --git a/chromium/third_party/webrtc/video/video_capture_input_unittest.cc b/chromium/third_party/webrtc/video/video_capture_input_unittest.cc
index b36c2577f91..2da722b47d4 100644
--- a/chromium/third_party/webrtc/video/video_capture_input_unittest.cc
+++ b/chromium/third_party/webrtc/video/video_capture_input_unittest.cc
@@ -16,6 +16,7 @@
 #include "webrtc/base/event.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/test/fake_texture_frame.h"
+#include "webrtc/test/frame_utils.h"
 #include "webrtc/video/send_statistics_proxy.h"
 
 // If an output frame does not arrive in 500ms, the test will fail.
@@ -23,9 +24,6 @@
 
 namespace webrtc {
 
-bool EqualFrames(const VideoFrame& frame1, const VideoFrame& frame2);
-bool EqualTextureFrames(const VideoFrame& frame1, const VideoFrame& frame2);
-bool EqualBufferFrames(const VideoFrame& frame1, const VideoFrame& frame2);
 bool EqualFramesVector(const std::vector<std::unique_ptr<VideoFrame>>& frames1,
                        const std::vector<std::unique_ptr<VideoFrame>>& frames2);
 std::unique_ptr<VideoFrame> CreateVideoFrame(uint8_t length);
@@ -54,9 +52,9 @@ class VideoCaptureInputTest : public ::testing::Test {
     EXPECT_TRUE(capture_event_.Wait(FRAME_TIMEOUT_MS));
     VideoFrame frame;
     EXPECT_TRUE(input_->GetVideoFrame(&frame));
-    if (!frame.native_handle()) {
-      output_frame_ybuffers_.push_back(
-          static_cast<const VideoFrame*>(&frame)->buffer(kYPlane));
+    ASSERT_TRUE(frame.video_frame_buffer());
+    if (!frame.video_frame_buffer()->native_handle()) {
+      output_frame_ybuffers_.push_back(frame.video_frame_buffer()->DataY());
     }
     output_frames_.push_back(
         std::unique_ptr<VideoFrame>(new VideoFrame(frame)));
@@ -95,17 +93,18 @@ TEST_F(VideoCaptureInputTest, DoesNotRetainHandleNorCopyBuffer) {
     rtc::Event* const event_;
   };
 
-  VideoFrame frame(
-      new rtc::RefCountedObject<TestBuffer>(&frame_destroyed_event), 1, 1,
-      kVideoRotation_0);
+  {
+    VideoFrame frame(
+        new rtc::RefCountedObject<TestBuffer>(&frame_destroyed_event), 1, 1,
+        kVideoRotation_0);
 
-  AddInputFrame(&frame);
-  WaitOutputFrame();
+    AddInputFrame(&frame);
+    WaitOutputFrame();
 
-  EXPECT_EQ(output_frames_[0]->video_frame_buffer().get(),
-            frame.video_frame_buffer().get());
-  output_frames_.clear();
-  frame.Reset();
+    EXPECT_EQ(output_frames_[0]->video_frame_buffer().get(),
+              frame.video_frame_buffer().get());
+    output_frames_.clear();
+  }
   EXPECT_TRUE(frame_destroyed_event.Wait(FRAME_TIMEOUT_MS));
 }
 
@@ -168,7 +167,9 @@ TEST_F(VideoCaptureInputTest, TestTextureFrames) {
                                             i + 1, webrtc::kVideoRotation_0))));
     AddInputFrame(input_frames_[i].get());
     WaitOutputFrame();
-    EXPECT_EQ(dummy_handle, output_frames_[i]->native_handle());
+    ASSERT_TRUE(output_frames_[i]->video_frame_buffer());
+    EXPECT_EQ(dummy_handle,
+              output_frames_[i]->video_frame_buffer()->native_handle());
   }
 
   EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
@@ -179,8 +180,7 @@ TEST_F(VideoCaptureInputTest, TestI420Frames) {
   std::vector<const uint8_t*> ybuffer_pointers;
   for (int i = 0; i < kNumFrame; ++i) {
     input_frames_.push_back(CreateVideoFrame(static_cast<uint8_t>(i + 1)));
-    const VideoFrame* const_input_frame = input_frames_[i].get();
-    ybuffer_pointers.push_back(const_input_frame->buffer(kYPlane));
+    ybuffer_pointers.push_back(input_frames_[i]->video_frame_buffer()->DataY());
     AddInputFrame(input_frames_[i].get());
     WaitOutputFrame();
   }
@@ -198,7 +198,9 @@ TEST_F(VideoCaptureInputTest, TestI420FrameAfterTextureFrame) {
           dummy_handle, 1, 1, 1, 1, webrtc::kVideoRotation_0))));
   AddInputFrame(input_frames_[0].get());
   WaitOutputFrame();
-  EXPECT_EQ(dummy_handle, output_frames_[0]->native_handle());
+  ASSERT_TRUE(output_frames_[0]->video_frame_buffer());
+  EXPECT_EQ(dummy_handle,
+            output_frames_[0]->video_frame_buffer()->native_handle());
 
   input_frames_.push_back(CreateVideoFrame(2));
   AddInputFrame(input_frames_[1].get());
@@ -222,43 +224,17 @@ TEST_F(VideoCaptureInputTest, TestTextureFrameAfterI420Frame) {
   EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
 }
 
-bool EqualFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
-  if (frame1.native_handle() || frame2.native_handle())
-    return EqualTextureFrames(frame1, frame2);
-  return EqualBufferFrames(frame1, frame2);
-}
-
-bool EqualTextureFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
-  return ((frame1.native_handle() == frame2.native_handle()) &&
-          (frame1.width() == frame2.width()) &&
-          (frame1.height() == frame2.height()));
-}
-
-bool EqualBufferFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
-  return ((frame1.width() == frame2.width()) &&
-          (frame1.height() == frame2.height()) &&
-          (frame1.stride(kYPlane) == frame2.stride(kYPlane)) &&
-          (frame1.stride(kUPlane) == frame2.stride(kUPlane)) &&
-          (frame1.stride(kVPlane) == frame2.stride(kVPlane)) &&
-          (frame1.allocated_size(kYPlane) == frame2.allocated_size(kYPlane)) &&
-          (frame1.allocated_size(kUPlane) == frame2.allocated_size(kUPlane)) &&
-          (frame1.allocated_size(kVPlane) == frame2.allocated_size(kVPlane)) &&
-          (memcmp(frame1.buffer(kYPlane), frame2.buffer(kYPlane),
-                  frame1.allocated_size(kYPlane)) == 0) &&
-          (memcmp(frame1.buffer(kUPlane), frame2.buffer(kUPlane),
-                  frame1.allocated_size(kUPlane)) == 0) &&
-          (memcmp(frame1.buffer(kVPlane), frame2.buffer(kVPlane),
-                  frame1.allocated_size(kVPlane)) == 0));
-}
-
 bool EqualFramesVector(
     const std::vector<std::unique_ptr<VideoFrame>>& frames1,
     const std::vector<std::unique_ptr<VideoFrame>>& frames2) {
   if (frames1.size() != frames2.size())
     return false;
   for (size_t i = 0; i < frames1.size(); ++i) {
-    if (!EqualFrames(*frames1[i], *frames2[i]))
+    // Compare frame buffers, since we don't care about differing timestamps.
+    if (!test::FrameBufsEqual(frames1[i]->video_frame_buffer(),
+                              frames2[i]->video_frame_buffer())) {
       return false;
+    }
   }
   return true;
 }
diff --git a/chromium/third_party/webrtc/video/video_decoder.cc b/chromium/third_party/webrtc/video/video_decoder.cc
index 602799e075b..5bf503cd802 100644
--- a/chromium/third_party/webrtc/video/video_decoder.cc
+++ b/chromium/third_party/webrtc/video/video_decoder.cc
@@ -25,6 +25,7 @@ VideoDecoder* VideoDecoder::Create(VideoDecoder::DecoderType codec_type) {
     case kVp8:
       return VP8Decoder::Create();
     case kVp9:
+      RTC_DCHECK(VP9Decoder::IsSupported());
       return VP9Decoder::Create();
     case kUnsupportedCodec:
       LOG(LS_ERROR) << "Creating NullVideoDecoder for unsupported codec.";
diff --git a/chromium/third_party/webrtc/video/video_encoder.cc b/chromium/third_party/webrtc/video/video_encoder.cc
index e85e3d97a72..1534a97fdf4 100644
--- a/chromium/third_party/webrtc/video/video_encoder.cc
+++ b/chromium/third_party/webrtc/video/video_encoder.cc
@@ -25,6 +25,7 @@ VideoEncoder* VideoEncoder::Create(VideoEncoder::EncoderType codec_type) {
     case kVp8:
       return VP8Encoder::Create();
     case kVp9:
+      RTC_DCHECK(VP9Encoder::IsSupported());
       return VP9Encoder::Create();
     case kUnsupportedCodec:
       RTC_NOTREACHED();
@@ -191,10 +192,4 @@ const char* VideoEncoderSoftwareFallbackWrapper::ImplementationName() const {
   return encoder_->ImplementationName();
 }
 
-int VideoEncoderSoftwareFallbackWrapper::GetTargetFramerate() {
-  if (fallback_encoder_)
-    return fallback_encoder_->GetTargetFramerate();
-  return encoder_->GetTargetFramerate();
-}
-
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/video_encoder_unittest.cc b/chromium/third_party/webrtc/video/video_encoder_unittest.cc
index 0f28f891634..f2c3ea61c46 100644
--- a/chromium/third_party/webrtc/video/video_encoder_unittest.cc
+++ b/chromium/third_party/webrtc/video/video_encoder_unittest.cc
@@ -107,11 +107,11 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
 void VideoEncoderSoftwareFallbackWrapperTest::EncodeFrame() {
   frame_.CreateEmptyFrame(kWidth, kHeight, kWidth, (kWidth + 1) / 2,
                           (kWidth + 1) / 2);
-  memset(frame_.buffer(webrtc::kYPlane), 16,
+  memset(frame_.video_frame_buffer()->MutableDataY(), 16,
          frame_.allocated_size(webrtc::kYPlane));
-  memset(frame_.buffer(webrtc::kUPlane), 128,
+  memset(frame_.video_frame_buffer()->MutableDataU(), 128,
          frame_.allocated_size(webrtc::kUPlane));
-  memset(frame_.buffer(webrtc::kVPlane), 128,
+  memset(frame_.video_frame_buffer()->MutableDataV(), 128,
          frame_.allocated_size(webrtc::kVPlane));
 
   std::vector<FrameType> types(1, kVideoFrameKey);
diff --git a/chromium/third_party/webrtc/video/video_loopback.cc b/chromium/third_party/webrtc/video/video_loopback.cc
index 87aacc6755f..0974d5869f1 100644
--- a/chromium/third_party/webrtc/video/video_loopback.cc
+++ b/chromium/third_party/webrtc/video/video_loopback.cc
@@ -178,6 +178,8 @@ DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
 
 DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
 
+DEFINE_bool(use_fec, false, "Use forward error correction.");
+
 DEFINE_string(
     force_fieldtrials,
     "",
@@ -216,7 +218,9 @@ void Loopback() {
        flags::MaxBitrateKbps() * 1000, flags::Codec(),
        flags::NumTemporalLayers(), flags::SelectedTL(),
        0,  // No min transmit bitrate.
-       call_bitrate_config, flags::FLAGS_send_side_bwe},
+       call_bitrate_config,
+       flags::FLAGS_send_side_bwe,
+       flags::FLAGS_use_fec},
       {flags::Clip()},
       {},  // Screenshare specific.
       {"video", 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename(),
diff --git a/chromium/third_party/webrtc/video/video_quality_test.cc b/chromium/third_party/webrtc/video/video_quality_test.cc
index 60f957c456a..78d8845ffdb 100644
--- a/chromium/third_party/webrtc/video/video_quality_test.cc
+++ b/chromium/third_party/webrtc/video/video_quality_test.cc
@@ -21,6 +21,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/format_macros.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@@ -63,11 +64,14 @@ class VideoAnalyzer : public PacketReceiver,
         graph_data_output_file_(graph_data_output_file),
         graph_title_(graph_title),
         ssrc_to_analyze_(ssrc_to_analyze),
+        pre_encode_proxy_(this),
         encode_timing_proxy_(this),
         frames_to_process_(duration_frames),
         frames_recorded_(0),
         frames_processed_(0),
         dropped_frames_(0),
+        dropped_frames_before_first_encode_(0),
+        dropped_frames_before_rendering_(0),
         last_render_time_(0),
         rtp_timestamp_delta_(0),
         avg_psnr_threshold_(avg_psnr_threshold),
@@ -143,18 +147,26 @@ class VideoAnalyzer : public PacketReceiver,
   void IncomingCapturedFrame(const VideoFrame& video_frame) override {
     VideoFrame copy = video_frame;
     copy.set_timestamp(copy.ntp_time_ms() * 90);
-
     {
       rtc::CritScope lock(&crit_);
-      if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0)
-        first_send_frame_ = copy;
-
       frames_.push_back(copy);
     }
 
     input_->IncomingCapturedFrame(video_frame);
   }
 
+  void PreEncodeOnFrame(const VideoFrame& video_frame) {
+    rtc::CritScope lock(&crit_);
+    if (!first_send_timestamp_ && rtp_timestamp_delta_ == 0) {
+      while (frames_.front().timestamp() != video_frame.timestamp()) {
+        ++dropped_frames_before_first_encode_;
+        frames_.pop_front();
+        RTC_CHECK(!frames_.empty());
+      }
+      first_send_timestamp_ = rtc::Optional<uint32_t>(video_frame.timestamp());
+    }
+  }
+
   bool SendRtp(const uint8_t* packet,
                size_t length,
                const PacketOptions& options) override {
@@ -169,8 +181,8 @@ class VideoAnalyzer : public PacketReceiver,
       rtc::CritScope lock(&crit_);
 
       if (rtp_timestamp_delta_ == 0) {
-        rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
-        first_send_frame_.Reset();
+        rtp_timestamp_delta_ = header.timestamp - *first_send_timestamp_;
+        first_send_timestamp_ = rtc::Optional<uint32_t>();
       }
       int64_t timestamp =
           wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
@@ -203,9 +215,18 @@ class VideoAnalyzer : public PacketReceiver,
         wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_);
 
     while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) {
+      if (last_rendered_frame_.IsZeroSize()) {
+        // No previous frame rendered, this one was dropped after sending but
+        // before rendering.
+        ++dropped_frames_before_rendering_;
+        frames_.pop_front();
+        RTC_CHECK(!frames_.empty());
+        continue;
+      }
       AddFrameComparison(frames_.front(), last_rendered_frame_, true,
                          render_time_ms);
       frames_.pop_front();
+      RTC_DCHECK(!frames_.empty());
     }
 
     VideoFrame reference_frame = frames_.front();
@@ -267,6 +288,9 @@ class VideoAnalyzer : public PacketReceiver,
     stats_polling_thread_.Stop();
   }
 
+  rtc::VideoSinkInterface<VideoFrame>* pre_encode_proxy() {
+    return &pre_encode_proxy_;
+  }
   EncodedFrameObserver* encode_timing_proxy() { return &encode_timing_proxy_; }
 
   VideoCaptureInput* input_;
@@ -350,11 +374,26 @@ class VideoAnalyzer : public PacketReceiver,
     VideoAnalyzer* const parent_;
   };
 
+  // This class receives the send-side OnFrame callback and is provided to not
+  // conflict with the receiver-side renderer callback.
+  class PreEncodeProxy : public rtc::VideoSinkInterface<VideoFrame> {
+   public:
+    explicit PreEncodeProxy(VideoAnalyzer* parent) : parent_(parent) {}
+
+    void OnFrame(const VideoFrame& video_frame) override {
+      parent_->PreEncodeOnFrame(video_frame);
+    }
+
+   private:
+    VideoAnalyzer* const parent_;
+  };
+
   void AddFrameComparison(const VideoFrame& reference,
                           const VideoFrame& render,
                           bool dropped,
                           int64_t render_time_ms)
       EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+    RTC_DCHECK(!render.IsZeroSize());
     int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp());
     int64_t send_time_ms = send_times_[reference_timestamp];
     send_times_.erase(reference_timestamp);
@@ -487,8 +526,6 @@ class VideoAnalyzer : public PacketReceiver,
     PrintResult("psnr", psnr_, " dB");
     PrintResult("ssim", ssim_, " score");
     PrintResult("sender_time", sender_time_, " ms");
-    printf("RESULT dropped_frames: %s = %d frames\n", test_label_.c_str(),
-           dropped_frames_);
     PrintResult("receiver_time", receiver_time_, " ms");
     PrintResult("total_delay_incl_network", end_to_end_, " ms");
     PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
@@ -498,6 +535,13 @@ class VideoAnalyzer : public PacketReceiver,
     PrintResult("encode_usage_percent", encode_usage_percent, " percent");
     PrintResult("media_bitrate", media_bitrate_bps, " bps");
 
+    printf("RESULT dropped_frames: %s = %d frames\n", test_label_.c_str(),
+           dropped_frames_);
+    printf("RESULT dropped_frames_before_first_encode: %s = %d frames\n",
+           test_label_.c_str(), dropped_frames_before_first_encode_);
+    printf("RESULT dropped_frames_before_rendering: %s = %d frames\n",
+           test_label_.c_str(), dropped_frames_before_rendering_);
+
     EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
     EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
   }
@@ -593,6 +637,7 @@ class VideoAnalyzer : public PacketReceiver,
   FILE* const graph_data_output_file_;
   const std::string graph_title_;
   const uint32_t ssrc_to_analyze_;
+  PreEncodeProxy pre_encode_proxy_;
   OnEncodeTimingProxy encode_timing_proxy_;
   std::vector<Sample> samples_ GUARDED_BY(comparison_lock_);
   std::map<int64_t, int> samples_encode_time_ms_ GUARDED_BY(comparison_lock_);
@@ -612,6 +657,8 @@ class VideoAnalyzer : public PacketReceiver,
   int frames_recorded_;
   int frames_processed_;
   int dropped_frames_;
+  int dropped_frames_before_first_encode_;
+  int dropped_frames_before_rendering_;
   int64_t last_render_time_;
   uint32_t rtp_timestamp_delta_;
 
@@ -622,7 +669,7 @@ class VideoAnalyzer : public PacketReceiver,
   std::map<int64_t, int64_t> send_times_ GUARDED_BY(crit_);
   std::map<int64_t, int64_t> recv_times_ GUARDED_BY(crit_);
   std::map<int64_t, size_t> encoded_frame_sizes_ GUARDED_BY(crit_);
-  VideoFrame first_send_frame_ GUARDED_BY(crit_);
+  rtc::Optional<uint32_t> first_send_timestamp_ GUARDED_BY(crit_);
   const double avg_psnr_threshold_;
   const double avg_ssim_threshold_;
 
@@ -1005,6 +1052,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
 
   SetupCommon(&analyzer, &recv_transport);
   video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
+  video_send_config_.pre_encode_callback = analyzer.pre_encode_proxy();
   for (auto& config : video_receive_configs_)
     config.pre_decode_callback = &analyzer;
   RTC_DCHECK(!video_send_config_.post_encode_callback);
@@ -1079,6 +1127,15 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) {
   video_send_config_.local_renderer = local_preview.get();
   video_receive_configs_[stream_id].renderer = loopback_video.get();
 
+  if (params.common.fec) {
+    video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
+    video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+    video_receive_configs_[stream_id].rtp.fec.red_payload_type =
+      kRedPayloadType;
+    video_receive_configs_[stream_id].rtp.fec.ulpfec_payload_type =
+      kUlpfecPayloadType;
+  }
+
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
diff --git a/chromium/third_party/webrtc/video/video_quality_test.h b/chromium/third_party/webrtc/video/video_quality_test.h
index b476004aae8..e138a89a8c2 100644
--- a/chromium/third_party/webrtc/video/video_quality_test.h
+++ b/chromium/third_party/webrtc/video/video_quality_test.h
@@ -41,6 +41,7 @@ class VideoQualityTest : public test::CallTest {
 
       Call::Config::BitrateConfig call_bitrate_config;
       bool send_side_bwe;
+      bool fec;
     } common;
     struct {  // Video-specific settings.
       std::string clip_name;
diff --git a/chromium/third_party/webrtc/video/video_receive_stream.cc b/chromium/third_party/webrtc/video/video_receive_stream.cc
index c0c3dea5f7a..1753db5d48f 100644
--- a/chromium/third_party/webrtc/video/video_receive_stream.cc
+++ b/chromium/third_party/webrtc/video/video_receive_stream.cc
@@ -21,14 +21,16 @@
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video/vie_remb.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
+static const bool kEnableFrameRecording = false;
+
 static bool UseSendSideBwe(const VideoReceiveStream::Config& config) {
   if (!config.rtp.transport_cc)
     return false;
@@ -158,117 +160,37 @@ VideoReceiveStream::VideoReceiveStream(
       decode_thread_(DecodeThreadFunction, this, "DecodingThread"),
       congestion_controller_(congestion_controller),
       call_stats_(call_stats),
-      remb_(remb),
-      vcm_(VideoCodingModule::Create(clock_,
-                                     nullptr,
-                                     nullptr,
-                                     this,
-                                     this)),
-      incoming_video_stream_(0, config.disable_prerenderer_smoothing),
+      video_receiver_(clock_, nullptr, this, this, this),
+      incoming_video_stream_(config.disable_prerenderer_smoothing),
       stats_proxy_(config_, clock_),
-      vie_channel_(&transport_adapter_,
-                   process_thread,
-                   nullptr,
-                   vcm_.get(),
-                   nullptr,
-                   nullptr,
-                   nullptr,
-                   congestion_controller_->GetRemoteBitrateEstimator(
-                       UseSendSideBwe(config_)),
-                   call_stats_->rtcp_rtt_stats(),
-                   congestion_controller_->pacer(),
-                   congestion_controller_->packet_router(),
-                   1,
-                   false),
-      vie_receiver_(vie_channel_.vie_receiver()),
-      vie_sync_(vcm_.get()),
-      rtp_rtcp_(vie_channel_.rtp_rtcp().front()) {
+      rtp_stream_receiver_(&video_receiver_,
+                           congestion_controller_->GetRemoteBitrateEstimator(
+                               UseSendSideBwe(config_)),
+                           &transport_adapter_,
+                           call_stats_->rtcp_rtt_stats(),
+                           congestion_controller_->pacer(),
+                           congestion_controller_->packet_router(),
+                           remb,
+                           config,
+                           &stats_proxy_,
+                           process_thread_),
+      video_stream_decoder_(&video_receiver_,
+                            &rtp_stream_receiver_,
+                            &rtp_stream_receiver_,
+                            rtp_stream_receiver_.IsRetransmissionsEnabled(),
+                            rtp_stream_receiver_.IsFecEnabled(),
+                            &stats_proxy_,
+                            &incoming_video_stream_,
+                            this),
+      vie_sync_(&video_receiver_) {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
   RTC_DCHECK(process_thread_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
-  RTC_DCHECK(remb_);
-  RTC_CHECK(vie_channel_.Init() == 0);
 
   // Register the channel to receive stats updates.
-  call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
-
-  // TODO(pbos): This is not fine grained enough...
-  vie_channel_.SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, -1,
-                                 -1);
-  RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
-      << "A stream should not be configured with RTCP disabled. This value is "
-         "reserved for internal usage.";
-  rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
-
-  RTC_DCHECK(config_.rtp.remote_ssrc != 0);
-  // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
-  RTC_DCHECK(config_.rtp.local_ssrc != 0);
-  RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
-  rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
-
-  // TODO(pbos): Support multiple RTX, per video payload.
-  for (const auto& kv : config_.rtp.rtx) {
-    RTC_DCHECK(kv.second.ssrc != 0);
-    RTC_DCHECK(kv.second.payload_type != 0);
-
-    vie_receiver_->SetRtxSsrc(kv.second.ssrc);
-    vie_receiver_->SetRtxPayloadType(kv.second.payload_type, kv.first);
-  }
-  // TODO(holmer): When Chrome no longer depends on this being false by default,
-  // always use the mapping and remove this whole codepath.
-  vie_receiver_->SetUseRtxPayloadMappingOnRestore(
-      config_.rtp.use_rtx_payload_mapping_on_restore);
-
-  if (config_.rtp.remb) {
-    rtp_rtcp_->SetREMBStatus(true);
-    remb_->AddReceiveChannel(rtp_rtcp_);
-  }
-
-  for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
-    const std::string& extension = config_.rtp.extensions[i].name;
-    int id = config_.rtp.extensions[i].id;
-    // One-byte-extension local identifiers are in the range 1-14 inclusive.
-    RTC_DCHECK_GE(id, 1);
-    RTC_DCHECK_LE(id, 14);
-    vie_receiver_->EnableReceiveRtpHeaderExtension(extension, id);
-  }
-
-  if (config_.rtp.fec.ulpfec_payload_type != -1) {
-    // ULPFEC without RED doesn't make sense.
-    RTC_DCHECK(config_.rtp.fec.red_payload_type != -1);
-    VideoCodec codec;
-    memset(&codec, 0, sizeof(codec));
-    codec.codecType = kVideoCodecULPFEC;
-    strncpy(codec.plName, "ulpfec", sizeof(codec.plName));
-    codec.plType = config_.rtp.fec.ulpfec_payload_type;
-    RTC_CHECK(vie_receiver_->SetReceiveCodec(codec));
-  }
-  if (config_.rtp.fec.red_payload_type != -1) {
-    VideoCodec codec;
-    memset(&codec, 0, sizeof(codec));
-    codec.codecType = kVideoCodecRED;
-    strncpy(codec.plName, "red", sizeof(codec.plName));
-    codec.plType = config_.rtp.fec.red_payload_type;
-    RTC_CHECK(vie_receiver_->SetReceiveCodec(codec));
-    if (config_.rtp.fec.red_rtx_payload_type != -1) {
-      vie_receiver_->SetRtxPayloadType(config_.rtp.fec.red_rtx_payload_type,
-                                       config_.rtp.fec.red_payload_type);
-    }
-  }
-
-  if (config.rtp.rtcp_xr.receiver_reference_time_report)
-    rtp_rtcp_->SetRtcpXrRrtrStatus(true);
-
-  vie_channel_.RegisterReceiveStatisticsProxy(&stats_proxy_);
-  vie_receiver_->GetReceiveStatistics()->RegisterRtpStatisticsCallback(
-      &stats_proxy_);
-  vie_receiver_->GetReceiveStatistics()->RegisterRtcpStatisticsCallback(
-      &stats_proxy_);
-  // Stats callback for CNAME changes.
-  rtp_rtcp_->RegisterRtcpStatisticsCallback(&stats_proxy_);
-  vie_channel_.RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
+  call_stats_->RegisterStatsObserver(&video_stream_decoder_);
 
   RTC_DCHECK(!config_.decoders.empty());
   std::set<int> decoder_payload_types;
@@ -279,23 +201,20 @@ VideoReceiveStream::VideoReceiveStream(
         << "Duplicate payload type (" << decoder.payload_type
         << ") for different decoders.";
     decoder_payload_types.insert(decoder.payload_type);
-    vcm_->RegisterExternalDecoder(decoder.decoder, decoder.payload_type);
+    video_receiver_.RegisterExternalDecoder(decoder.decoder,
+                                            decoder.payload_type);
 
     VideoCodec codec = CreateDecoderVideoCodec(decoder);
-
-    RTC_CHECK(vie_receiver_->SetReceiveCodec(codec));
-    RTC_CHECK_EQ(VCM_OK,
-                 vcm_->RegisterReceiveCodec(&codec, num_cpu_cores, false));
+    RTC_CHECK(rtp_stream_receiver_.SetReceiveCodec(codec));
+    RTC_CHECK_EQ(VCM_OK, video_receiver_.RegisterReceiveCodec(
+                             &codec, num_cpu_cores, false));
   }
 
-  vcm_->SetRenderDelay(config.render_delay_ms);
+  video_receiver_.SetRenderDelay(config.render_delay_ms);
   incoming_video_stream_.SetExpectedRenderDelay(config.render_delay_ms);
-  vcm_->RegisterPreDecodeImageCallback(this);
   incoming_video_stream_.SetExternalCallback(this);
-  vie_channel_.SetIncomingVideoStream(&incoming_video_stream_);
-  vie_channel_.RegisterPreRenderCallback(this);
 
-  process_thread_->RegisterModule(vcm_.get());
+  process_thread_->RegisterModule(&video_receiver_);
   process_thread_->RegisterModule(&vie_sync_);
 }
 
@@ -304,24 +223,34 @@ VideoReceiveStream::~VideoReceiveStream() {
   Stop();
 
   process_thread_->DeRegisterModule(&vie_sync_);
-  process_thread_->DeRegisterModule(vcm_.get());
+  process_thread_->DeRegisterModule(&video_receiver_);
 
-  // Deregister external decoders so that they are no longer running during
+  // Deregister external decoders so they are no longer running during
   // destruction. This effectively stops the VCM since the decoder thread is
   // stopped, the VCM is deregistered and no asynchronous decoder threads are
   // running.
   for (const Decoder& decoder : config_.decoders)
-    vcm_->RegisterExternalDecoder(nullptr, decoder.payload_type);
-
-  vie_channel_.RegisterPreRenderCallback(nullptr);
-  vcm_->RegisterPreDecodeImageCallback(nullptr);
+    video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
 
-  call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver());
-  rtp_rtcp_->SetREMBStatus(false);
-  remb_->RemoveReceiveChannel(rtp_rtcp_);
+  call_stats_->DeregisterStatsObserver(&video_stream_decoder_);
 
   congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config_))
-      ->RemoveStream(vie_receiver_->GetRemoteSsrc());
+      ->RemoveStream(rtp_stream_receiver_.GetRemoteSsrc());
+}
+
+void VideoReceiveStream::SignalNetworkState(NetworkState state) {
+  rtp_stream_receiver_.SignalNetworkState(state);
+}
+
+
+bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  return rtp_stream_receiver_.DeliverRtcp(packet, length);
+}
+
+bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
+                                    size_t length,
+                                    const PacketTime& packet_time) {
+  return rtp_stream_receiver_.DeliverRtp(packet, length, packet_time);
 }
 
 void VideoReceiveStream::Start() {
@@ -332,13 +261,13 @@ void VideoReceiveStream::Start() {
   // Start the decode thread
   decode_thread_.Start();
   decode_thread_.SetPriority(rtc::kHighestPriority);
-  vie_receiver_->StartReceive();
+  rtp_stream_receiver_.StartReceive();
 }
 
 void VideoReceiveStream::Stop() {
   incoming_video_stream_.Stop();
-  vie_receiver_->StopReceive();
-  vcm_->TriggerDecoderShutdown();
+  rtp_stream_receiver_.StopReceive();
+  video_receiver_.TriggerDecoderShutdown();
   decode_thread_.Stop();
   transport_adapter_.Disable();
 }
@@ -347,41 +276,31 @@ void VideoReceiveStream::SetSyncChannel(VoiceEngine* voice_engine,
                                         int audio_channel_id) {
   if (voice_engine && audio_channel_id != -1) {
     VoEVideoSync* voe_sync_interface = VoEVideoSync::GetInterface(voice_engine);
-    vie_sync_.ConfigureSync(audio_channel_id, voe_sync_interface, rtp_rtcp_,
-                            vie_receiver_->GetRtpReceiver());
+    vie_sync_.ConfigureSync(audio_channel_id, voe_sync_interface,
+                            rtp_stream_receiver_.rtp_rtcp(),
+                            rtp_stream_receiver_.GetRtpReceiver());
     voe_sync_interface->Release();
-    return;
+  } else {
+    vie_sync_.ConfigureSync(-1, nullptr, rtp_stream_receiver_.rtp_rtcp(),
+                            rtp_stream_receiver_.GetRtpReceiver());
   }
-  vie_sync_.ConfigureSync(-1, nullptr, rtp_rtcp_,
-                          vie_receiver_->GetRtpReceiver());
 }
 
 VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
   return stats_proxy_.GetStats();
 }
 
-bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return vie_receiver_->DeliverRtcp(packet, length);
-}
-
-bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
-                                    size_t length,
-                                    const PacketTime& packet_time) {
-  return vie_receiver_->DeliverRtp(packet, length, packet_time);
-}
-
 void VideoReceiveStream::FrameCallback(VideoFrame* video_frame) {
   stats_proxy_.OnDecodedFrame();
 
   // Post processing is not supported if the frame is backed by a texture.
-  if (!video_frame->native_handle()) {
+  if (!video_frame->video_frame_buffer()->native_handle()) {
     if (config_.pre_render_callback)
       config_.pre_render_callback->FrameCallback(video_frame);
   }
 }
 
-int VideoReceiveStream::RenderFrame(const uint32_t /*stream_id*/,
-                                    const VideoFrame& video_frame) {
+void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) {
   int64_t sync_offset_ms;
   if (vie_sync_.GetStreamSyncOffsetInMs(video_frame, &sync_offset_ms))
     stats_proxy_.OnSyncOffsetUpdated(sync_offset_ms);
@@ -390,8 +309,6 @@ int VideoReceiveStream::RenderFrame(const uint32_t /*stream_id*/,
     config_.renderer->OnFrame(video_frame);
 
   stats_proxy_.OnRenderedFrame(video_frame.width(), video_frame.height());
-
-  return 0;
 }
 
 // TODO(asapersson): Consider moving callback from video_encoder.h or
@@ -406,12 +323,21 @@ int32_t VideoReceiveStream::Encoded(
     encoded_frame_proxy_.Encoded(
         encoded_image, codec_specific_info, fragmentation);
   }
-  return 0;
-}
+  if (kEnableFrameRecording) {
+    if (!ivf_writer_.get()) {
+      RTC_DCHECK(codec_specific_info);
+      std::ostringstream oss;
+      oss << "receive_bitstream_ssrc_" << config_.rtp.remote_ssrc << ".ivf";
+      ivf_writer_ =
+          IvfFileWriter::Open(oss.str(), codec_specific_info->codecType);
+    }
+    if (ivf_writer_.get()) {
+      bool ok = ivf_writer_->WriteFrame(encoded_image);
+      RTC_DCHECK(ok);
+    }
+  }
 
-void VideoReceiveStream::SignalNetworkState(NetworkState state) {
-  rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
-                                               : RtcpMode::kOff);
+  return 0;
 }
 
 bool VideoReceiveStream::DecodeThreadFunction(void* ptr) {
@@ -421,16 +347,16 @@ bool VideoReceiveStream::DecodeThreadFunction(void* ptr) {
 
 void VideoReceiveStream::Decode() {
   static const int kMaxDecodeWaitTimeMs = 50;
-  vcm_->Decode(kMaxDecodeWaitTimeMs);
+  video_receiver_.Decode(kMaxDecodeWaitTimeMs);
 }
 
 void VideoReceiveStream::SendNack(
     const std::vector<uint16_t>& sequence_numbers) {
-  rtp_rtcp_->SendNack(sequence_numbers);
+  rtp_stream_receiver_.RequestPacketRetransmit(sequence_numbers);
 }
 
 void VideoReceiveStream::RequestKeyFrame() {
-  rtp_rtcp_->RequestKeyFrame();
+  rtp_stream_receiver_.RequestKeyFrame();
 }
 
 }  // namespace internal
diff --git a/chromium/third_party/webrtc/video/video_receive_stream.h b/chromium/third_party/webrtc/video/video_receive_stream.h
index 3aca570f99e..0ea5385b319 100644
--- a/chromium/third_party/webrtc/video/video_receive_stream.h
+++ b/chromium/third_party/webrtc/video/video_receive_stream.h
@@ -18,19 +18,19 @@
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 #include "webrtc/video/receive_statistics_proxy.h"
-#include "webrtc/video/vie_channel.h"
-#include "webrtc/video/vie_encoder.h"
-#include "webrtc/video_encoder.h"
+#include "webrtc/video/rtp_stream_receiver.h"
+#include "webrtc/video/video_stream_decoder.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
 class CallStats;
 class CongestionController;
+class IvfFileWriter;
 class ProcessThread;
 class VoiceEngine;
 class VieRemb;
@@ -39,7 +39,7 @@ namespace internal {
 
 class VideoReceiveStream : public webrtc::VideoReceiveStream,
                            public I420FrameCallback,
-                           public VideoRenderCallback,
+                           public rtc::VideoSinkInterface<VideoFrame>,
                            public EncodedImageCallback,
                            public NackSender,
                            public KeyFrameRequestSender {
@@ -53,24 +53,23 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
                      VieRemb* remb);
   ~VideoReceiveStream() override;
 
-  // webrtc::ReceiveStream implementation.
-  void Start() override;
-  void Stop() override;
-  void SignalNetworkState(NetworkState state) override;
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
+  void SignalNetworkState(NetworkState state);
+  bool DeliverRtcp(const uint8_t* packet, size_t length);
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
-                  const PacketTime& packet_time) override;
+                  const PacketTime& packet_time);
 
   // webrtc::VideoReceiveStream implementation.
+  void Start() override;
+  void Stop() override;
+
   webrtc::VideoReceiveStream::Stats GetStats() const override;
 
   // Overrides I420FrameCallback.
   void FrameCallback(VideoFrame* video_frame) override;
 
-  // Overrides VideoRenderCallback.
-  int RenderFrame(const uint32_t /*stream_id*/,
-                  const VideoFrame& video_frame) override;
+  // Overrides rtc::VideoSinkInterface<VideoFrame>.
+  void OnFrame(const VideoFrame& video_frame) override;
 
   // Overrides EncodedImageCallback.
   int32_t Encoded(const EncodedImage& encoded_image,
@@ -81,10 +80,10 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
 
   void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id);
 
-  // NackSender
+  // Implements NackSender.
   void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
 
-  // KeyFrameRequestSender
+  // Implements KeyFrameRequestSender.
   void RequestKeyFrame() override;
 
  private:
@@ -101,15 +100,15 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
 
   CongestionController* const congestion_controller_;
   CallStats* const call_stats_;
-  VieRemb* const remb_;
 
-  std::unique_ptr<VideoCodingModule> vcm_;
+  vcm::VideoReceiver video_receiver_;
   IncomingVideoStream incoming_video_stream_;
   ReceiveStatisticsProxy stats_proxy_;
-  ViEChannel vie_channel_;
-  ViEReceiver* const vie_receiver_;
+  RtpStreamReceiver rtp_stream_receiver_;
+  VideoStreamDecoder video_stream_decoder_;
   ViESyncModule vie_sync_;
-  RtpRtcp* const rtp_rtcp_;
+
+  std::unique_ptr<IvfFileWriter> ivf_writer_;
 };
 }  // namespace internal
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/video_send_stream.cc b/chromium/third_party/webrtc/video/video_send_stream.cc
index 6b6b1af3464..ec8fbea3630 100644
--- a/chromium/third_party/webrtc/video/video_send_stream.cc
+++ b/chromium/third_party/webrtc/video/video_send_stream.cc
@@ -24,6 +24,7 @@
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/video_capture_input.h"
 #include "webrtc/video/vie_remb.h"
@@ -34,6 +35,54 @@ namespace webrtc {
 class RtcpIntraFrameObserver;
 class TransportFeedbackObserver;
 
+static const int kMinSendSidePacketHistorySize = 600;
+
+namespace {
+
+std::vector<RtpRtcp*> CreateRtpRtcpModules(
+    Transport* outgoing_transport,
+    RtcpIntraFrameObserver* intra_frame_callback,
+    RtcpBandwidthObserver* bandwidth_callback,
+    TransportFeedbackObserver* transport_feedback_callback,
+    RtcpRttStats* rtt_stats,
+    RtpPacketSender* paced_sender,
+    TransportSequenceNumberAllocator* transport_sequence_number_allocator,
+    SendStatisticsProxy* stats_proxy,
+    SendDelayStats* send_delay_stats,
+    size_t num_modules) {
+  RTC_DCHECK_GT(num_modules, 0u);
+  RtpRtcp::Configuration configuration;
+  ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
+  configuration.audio = false;
+  configuration.receiver_only = false;
+  configuration.receive_statistics = null_receive_statistics;
+  configuration.outgoing_transport = outgoing_transport;
+  configuration.intra_frame_callback = intra_frame_callback;
+  configuration.rtt_stats = rtt_stats;
+  configuration.rtcp_packet_type_counter_observer = stats_proxy;
+  configuration.paced_sender = paced_sender;
+  configuration.transport_sequence_number_allocator =
+      transport_sequence_number_allocator;
+  configuration.send_bitrate_observer = stats_proxy;
+  configuration.send_frame_count_observer = stats_proxy;
+  configuration.send_side_delay_observer = stats_proxy;
+  configuration.send_packet_observer = send_delay_stats;
+  configuration.bandwidth_callback = bandwidth_callback;
+  configuration.transport_feedback_callback = transport_feedback_callback;
+
+  std::vector<RtpRtcp*> modules;
+  for (size_t i = 0; i < num_modules; ++i) {
+    RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
+    rtp_rtcp->SetSendingStatus(false);
+    rtp_rtcp->SetSendingMediaStatus(false);
+    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
+    modules.push_back(rtp_rtcp);
+  }
+  return modules;
+}
+
+}  // namespace
+
 std::string
 VideoSendStream::Config::EncoderSettings::ToString() const {
   std::stringstream ss;
@@ -150,6 +199,149 @@ CpuOveruseOptions GetCpuOveruseOptions(bool full_overuse_time) {
   }
   return options;
 }
+
+VideoCodec VideoEncoderConfigToVideoCodec(const VideoEncoderConfig& config,
+                                          const std::string& payload_name,
+                                          int payload_type) {
+  const std::vector<VideoStream>& streams = config.streams;
+  static const int kEncoderMinBitrateKbps = 30;
+  RTC_DCHECK(!streams.empty());
+  RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
+
+  VideoCodec video_codec;
+  memset(&video_codec, 0, sizeof(video_codec));
+  video_codec.codecType = PayloadNameToCodecType(payload_name);
+
+  switch (config.content_type) {
+    case VideoEncoderConfig::ContentType::kRealtimeVideo:
+      video_codec.mode = kRealtimeVideo;
+      break;
+    case VideoEncoderConfig::ContentType::kScreen:
+      video_codec.mode = kScreensharing;
+      if (config.streams.size() == 1 &&
+          config.streams[0].temporal_layer_thresholds_bps.size() == 1) {
+        video_codec.targetBitrate =
+            config.streams[0].temporal_layer_thresholds_bps[0] / 1000;
+      }
+      break;
+  }
+
+  switch (video_codec.codecType) {
+    case kVideoCodecVP8: {
+      if (config.encoder_specific_settings) {
+        video_codec.codecSpecific.VP8 = *reinterpret_cast<const VideoCodecVP8*>(
+            config.encoder_specific_settings);
+      } else {
+        video_codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
+      }
+      video_codec.codecSpecific.VP8.numberOfTemporalLayers =
+          static_cast<unsigned char>(
+              streams.back().temporal_layer_thresholds_bps.size() + 1);
+      break;
+    }
+    case kVideoCodecVP9: {
+      if (config.encoder_specific_settings) {
+        video_codec.codecSpecific.VP9 = *reinterpret_cast<const VideoCodecVP9*>(
+            config.encoder_specific_settings);
+        if (video_codec.mode == kScreensharing) {
+          video_codec.codecSpecific.VP9.flexibleMode = true;
+          // For now VP9 screensharing use 1 temporal and 2 spatial layers.
+          RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfTemporalLayers,
+                        1);
+          RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfSpatialLayers, 2);
+        }
+      } else {
+        video_codec.codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings();
+      }
+      video_codec.codecSpecific.VP9.numberOfTemporalLayers =
+          static_cast<unsigned char>(
+              streams.back().temporal_layer_thresholds_bps.size() + 1);
+      break;
+    }
+    case kVideoCodecH264: {
+      if (config.encoder_specific_settings) {
+        video_codec.codecSpecific.H264 =
+            *reinterpret_cast<const VideoCodecH264*>(
+                config.encoder_specific_settings);
+      } else {
+        video_codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
+      }
+      break;
+    }
+    default:
+      // TODO(pbos): Support encoder_settings codec-agnostically.
+      RTC_DCHECK(!config.encoder_specific_settings)
+          << "Encoder-specific settings for codec type not wired up.";
+      break;
+  }
+
+  strncpy(video_codec.plName, payload_name.c_str(), kPayloadNameSize - 1);
+  video_codec.plName[kPayloadNameSize - 1] = '\0';
+  video_codec.plType = payload_type;
+  video_codec.numberOfSimulcastStreams =
+      static_cast<unsigned char>(streams.size());
+  video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
+  if (video_codec.minBitrate < kEncoderMinBitrateKbps)
+    video_codec.minBitrate = kEncoderMinBitrateKbps;
+  RTC_DCHECK_LE(streams.size(), static_cast<size_t>(kMaxSimulcastStreams));
+  if (video_codec.codecType == kVideoCodecVP9) {
+    // If the vector is empty, bitrates will be configured automatically.
+    RTC_DCHECK(config.spatial_layers.empty() ||
+               config.spatial_layers.size() ==
+                   video_codec.codecSpecific.VP9.numberOfSpatialLayers);
+    RTC_DCHECK_LE(video_codec.codecSpecific.VP9.numberOfSpatialLayers,
+                  kMaxSimulcastStreams);
+    for (size_t i = 0; i < config.spatial_layers.size(); ++i)
+      video_codec.spatialLayers[i] = config.spatial_layers[i];
+  }
+  for (size_t i = 0; i < streams.size(); ++i) {
+    SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
+    RTC_DCHECK_GT(streams[i].width, 0u);
+    RTC_DCHECK_GT(streams[i].height, 0u);
+    RTC_DCHECK_GT(streams[i].max_framerate, 0);
+    // Different framerates not supported per stream at the moment.
+    RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
+    RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0);
+    RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
+    RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
+    RTC_DCHECK_GE(streams[i].max_qp, 0);
+
+    sim_stream->width = static_cast<uint16_t>(streams[i].width);
+    sim_stream->height = static_cast<uint16_t>(streams[i].height);
+    sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
+    sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
+    sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
+    sim_stream->qpMax = streams[i].max_qp;
+    sim_stream->numberOfTemporalLayers = static_cast<unsigned char>(
+        streams[i].temporal_layer_thresholds_bps.size() + 1);
+
+    video_codec.width = std::max(video_codec.width,
+                                 static_cast<uint16_t>(streams[i].width));
+    video_codec.height = std::max(
+        video_codec.height, static_cast<uint16_t>(streams[i].height));
+    video_codec.minBitrate =
+        std::min(static_cast<uint16_t>(video_codec.minBitrate),
+                 static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
+    video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
+    video_codec.qpMax = std::max(video_codec.qpMax,
+                                 static_cast<unsigned int>(streams[i].max_qp));
+  }
+
+  if (video_codec.maxBitrate == 0) {
+    // Unset max bitrate -> cap to one bit per pixel.
+    video_codec.maxBitrate =
+        (video_codec.width * video_codec.height * video_codec.maxFramerate) /
+        1000;
+  }
+  if (video_codec.maxBitrate < kEncoderMinBitrateKbps)
+    video_codec.maxBitrate = kEncoderMinBitrateKbps;
+
+  RTC_DCHECK_GT(streams[0].max_framerate, 0);
+  video_codec.maxFramerate = streams[0].max_framerate;
+
+  return video_codec;
+}
+
 }  // namespace
 
 namespace internal {
@@ -159,6 +351,7 @@ VideoSendStream::VideoSendStream(
     CallStats* call_stats,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
+    SendDelayStats* send_delay_stats,
     VieRemb* remb,
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config,
@@ -183,31 +376,28 @@ VideoSendStream::VideoSendStream(
           this,
           config.post_encode_callback,
           &stats_proxy_),
-      vie_channel_(config.send_transport,
-                   module_process_thread_,
-                   &payload_router_,
-                   nullptr,
-                   &encoder_feedback_,
-                   congestion_controller_->GetBitrateController()
-                       ->CreateRtcpBandwidthObserver(),
-                   congestion_controller_->GetTransportFeedbackObserver(),
-                   nullptr,
-                   call_stats_->rtcp_rtt_stats(),
-                   congestion_controller_->pacer(),
-                   congestion_controller_->packet_router(),
-                   config_.rtp.ssrcs.size(),
-                   true),
-      vie_receiver_(vie_channel_.vie_receiver()),
       vie_encoder_(num_cpu_cores,
-                   config_.rtp.ssrcs,
                    module_process_thread_,
                    &stats_proxy_,
-                   config.pre_encode_callback,
-                   &overuse_detector_,
-                   congestion_controller_->pacer(),
-                   &payload_router_),
-      vcm_(vie_encoder_.vcm()),
-      rtp_rtcp_modules_(vie_channel_.rtp_rtcp()),
+                   &overuse_detector_),
+      encoder_feedback_(Clock::GetRealTimeClock(),
+                        config.rtp.ssrcs,
+                        &vie_encoder_),
+      video_sender_(vie_encoder_.video_sender()),
+      bandwidth_observer_(congestion_controller_->GetBitrateController()
+                              ->CreateRtcpBandwidthObserver()),
+      rtp_rtcp_modules_(CreateRtpRtcpModules(
+          config.send_transport,
+          &encoder_feedback_,
+          bandwidth_observer_.get(),
+          congestion_controller_->GetTransportFeedbackObserver(),
+          call_stats_->rtcp_rtt_stats(),
+          congestion_controller_->pacer(),
+          congestion_controller_->packet_router(),
+          &stats_proxy_,
+          send_delay_stats,
+          config_.rtp.ssrcs.size())),
+      payload_router_(rtp_rtcp_modules_, config.encoder_settings.payload_type),
       input_(&encoder_wakeup_event_,
              config_.local_renderer,
              &stats_proxy_,
@@ -220,14 +410,14 @@ VideoSendStream::VideoSendStream(
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(remb_);
 
-  payload_router_.Init(rtp_rtcp_modules_);
-  RTC_CHECK(vie_encoder_.Init());
-  encoder_feedback_.Init(config_.rtp.ssrcs, &vie_encoder_);
-  RTC_CHECK(vie_channel_.Init() == 0);
 
-  vcm_->RegisterProtectionCallback(vie_channel_.vcm_protection_callback());
+  // RTP/RTCP initialization.
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    module_process_thread_->RegisterModule(rtp_rtcp);
+    congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp);
+  }
 
-  call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
+  video_sender_->RegisterProtectionCallback(this);
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
@@ -245,28 +435,7 @@ VideoSendStream::VideoSendStream(
   remb_->AddRembSender(rtp_rtcp_modules_[0]);
   rtp_rtcp_modules_[0]->SetREMBStatus(true);
 
-  // Enable NACK, FEC or both.
-  const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
-  bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
-  // Payload types without picture ID cannot determine that a stream is complete
-  // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is
-  // a waste of bandwidth since FEC packets still have to be transmitted. Note
-  // that this is not the case with FLEXFEC.
-  if (enable_protection_nack &&
-      !PayloadTypeSupportsSkippingFecPackets(
-          config_.encoder_settings.payload_name)) {
-    LOG(LS_WARNING) << "Transmitting payload type without picture ID using"
-                       "NACK+FEC is a waste of bandwidth since FEC packets "
-                       "also have to be retransmitted. Disabling FEC.";
-    enable_protection_fec = false;
-  }
-  // TODO(changbin): Should set RTX for RED mapping in RTP sender in future.
-  vie_channel_.SetProtectionMode(enable_protection_nack, enable_protection_fec,
-                                 config_.rtp.fec.red_payload_type,
-                                 config_.rtp.fec.ulpfec_payload_type);
-  vie_encoder_.SetProtectionMethod(enable_protection_nack,
-                                   enable_protection_fec);
-
+  ConfigureProtection();
   ConfigureSsrcs();
 
   // TODO(pbos): Should we set CNAME on all RTP modules?
@@ -288,27 +457,8 @@ VideoSendStream::VideoSendStream(
   RTC_DCHECK(config.encoder_settings.encoder);
   RTC_DCHECK_GE(config.encoder_settings.payload_type, 0);
   RTC_DCHECK_LE(config.encoder_settings.payload_type, 127);
-  RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder(
-                      config.encoder_settings.encoder,
-                      config.encoder_settings.payload_type,
-                      config.encoder_settings.internal_source));
-
   ReconfigureVideoEncoder(encoder_config);
 
-  vie_channel_.RegisterSendSideDelayObserver(&stats_proxy_);
-
-  if (config_.post_encode_callback)
-    vie_encoder_.RegisterPostEncodeImageCallback(&encoded_frame_proxy_);
-
-  if (config_.suspend_below_min_bitrate) {
-    vcm_->SuspendBelowMinBitrate();
-    bitrate_allocator_->EnforceMinBitrate(false);
-  }
-
-  vie_channel_.RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
-  vie_channel_.RegisterSendBitrateObserver(&stats_proxy_);
-  vie_channel_.RegisterSendFrameCountObserver(&stats_proxy_);
-
   module_process_thread_->RegisterModule(&overuse_detector_);
 
   encoder_thread_.Start();
@@ -330,45 +480,43 @@ VideoSendStream::~VideoSendStream() {
   bitrate_allocator_->RemoveObserver(this);
 
   module_process_thread_->DeRegisterModule(&overuse_detector_);
-  vie_channel_.RegisterSendFrameCountObserver(nullptr);
-  vie_channel_.RegisterSendBitrateObserver(nullptr);
-  vie_channel_.RegisterRtcpPacketTypeCounterObserver(nullptr);
 
-  vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type);
-
-  call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver());
   rtp_rtcp_modules_[0]->SetREMBStatus(false);
   remb_->RemoveRembSender(rtp_rtcp_modules_[0]);
 
-  // ViEChannel outlives ViEEncoder so remove encoder from feedback before
-  // destruction.
-  encoder_feedback_.TearDown();
-
-  congestion_controller_->GetRemoteBitrateEstimator(false)->RemoveStream(
-      vie_receiver_->GetRemoteSsrc());
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    congestion_controller_->packet_router()->RemoveRtpModule(rtp_rtcp);
+    module_process_thread_->DeRegisterModule(rtp_rtcp);
+    delete rtp_rtcp;
+  }
 }
 
-VideoCaptureInput* VideoSendStream::Input() {
-  return &input_;
+bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->IncomingRtcpPacket(packet, length);
+  return true;
 }
 
 void VideoSendStream::Start() {
   if (payload_router_.active())
     return;
-  vie_encoder_.Pause();
+  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
   payload_router_.set_active(true);
   // Was not already started, trigger a keyframe.
   vie_encoder_.SendKeyFrame();
-  vie_encoder_.Restart();
-  vie_receiver_->StartReceive();
+  vie_encoder_.Start();
 }
 
 void VideoSendStream::Stop() {
   if (!payload_router_.active())
     return;
-  // TODO(pbos): Make sure the encoder stops here.
+  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
+  vie_encoder_.Pause();
   payload_router_.set_active(false);
-  vie_receiver_->StopReceive();
+}
+
+VideoCaptureInput* VideoSendStream::Input() {
+  return &input_;
 }
 
 bool VideoSendStream::EncoderThreadFunction(void* obj) {
@@ -378,176 +526,186 @@ bool VideoSendStream::EncoderThreadFunction(void* obj) {
 }
 
 void VideoSendStream::EncoderProcess() {
+  RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder(
+                      config_.encoder_settings.encoder,
+                      config_.encoder_settings.payload_type,
+                      config_.encoder_settings.internal_source));
+
   while (true) {
     encoder_wakeup_event_.Wait(rtc::Event::kForever);
     if (rtc::AtomicOps::AcquireLoad(&stop_encoder_thread_))
-      return;
+      break;
+    rtc::Optional<EncoderSettings> encoder_settings;
+    {
+      rtc::CritScope lock(&encoder_settings_crit_);
+      if (pending_encoder_settings_) {
+        encoder_settings = pending_encoder_settings_;
+        pending_encoder_settings_ = rtc::Optional<EncoderSettings>();
+      }
+    }
+    if (encoder_settings) {
+      encoder_settings->video_codec.startBitrate =
+          bitrate_allocator_->AddObserver(
+              this, encoder_settings->video_codec.minBitrate * 1000,
+              encoder_settings->video_codec.maxBitrate * 1000,
+              !config_.suspend_below_min_bitrate) /
+          1000;
+
+      payload_router_.SetSendStreams(encoder_settings->streams);
+      vie_encoder_.SetEncoder(encoder_settings->video_codec,
+                              encoder_settings->min_transmit_bitrate_bps,
+                              payload_router_.MaxPayloadLength(), this);
+
+      // vie_encoder_.SetEncoder must be called before this.
+      if (config_.suspend_below_min_bitrate)
+        video_sender_->SuspendBelowMinBitrate();
+
+      // Clear stats for disabled layers.
+      for (size_t i = encoder_settings->streams.size();
+           i < config_.rtp.ssrcs.size(); ++i) {
+        stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
+      }
+
+      // We might've gotten new settings while configuring the encoder settings,
+      // restart from the top to see if that's the case before trying to encode
+      // a frame (which might correspond to the last frame size).
+      encoder_wakeup_event_.Set();
+      continue;
+    }
 
     VideoFrame frame;
-    if (input_.GetVideoFrame(&frame))
+    if (input_.GetVideoFrame(&frame)) {
+      // TODO(perkj): |pre_encode_callback| is only used by tests. Tests should
+      // register as a sink to the VideoSource instead.
+      if (config_.pre_encode_callback) {
+        config_.pre_encode_callback->OnFrame(frame);
+      }
       vie_encoder_.EncodeVideoFrame(frame);
+    }
   }
+  vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type);
 }
 
 void VideoSendStream::ReconfigureVideoEncoder(
     const VideoEncoderConfig& config) {
   TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
   LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
-  const std::vector<VideoStream>& streams = config.streams;
-  static const int kEncoderMinBitrateKbps = 30;
-  RTC_DCHECK(!streams.empty());
-  RTC_DCHECK_GE(config_.rtp.ssrcs.size(), streams.size());
-  RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
+  RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size());
+  VideoCodec video_codec = VideoEncoderConfigToVideoCodec(
+      config, config_.encoder_settings.payload_name,
+      config_.encoder_settings.payload_type);
+  {
+    rtc::CritScope lock(&encoder_settings_crit_);
+    pending_encoder_settings_ = rtc::Optional<EncoderSettings>(
+        {video_codec, config.min_transmit_bitrate_bps, config.streams});
+  }
+  encoder_wakeup_event_.Set();
+}
 
-  VideoCodec video_codec;
-  memset(&video_codec, 0, sizeof(video_codec));
-  video_codec.codecType =
-      PayloadNameToCodecType(config_.encoder_settings.payload_name);
+VideoSendStream::Stats VideoSendStream::GetStats() {
+  return stats_proxy_.GetStats();
+}
 
-  switch (config.content_type) {
-    case VideoEncoderConfig::ContentType::kRealtimeVideo:
-      video_codec.mode = kRealtimeVideo;
-      break;
-    case VideoEncoderConfig::ContentType::kScreen:
-      video_codec.mode = kScreensharing;
-      if (config.streams.size() == 1 &&
-          config.streams[0].temporal_layer_thresholds_bps.size() == 1) {
-        video_codec.targetBitrate =
-            config.streams[0].temporal_layer_thresholds_bps[0] / 1000;
-      }
-      break;
-  }
+void VideoSendStream::OveruseDetected() {
+  if (config_.overuse_callback)
+    config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
+}
 
-  if (video_codec.codecType == kVideoCodecVP8) {
-    video_codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings();
-  } else if (video_codec.codecType == kVideoCodecVP9) {
-    video_codec.codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings();
-  } else if (video_codec.codecType == kVideoCodecH264) {
-    video_codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings();
-  }
+void VideoSendStream::NormalUsage() {
+  if (config_.overuse_callback)
+    config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
+}
 
-  if (video_codec.codecType == kVideoCodecVP8) {
-    if (config.encoder_specific_settings) {
-      video_codec.codecSpecific.VP8 = *reinterpret_cast<const VideoCodecVP8*>(
-                                          config.encoder_specific_settings);
-    }
-    video_codec.codecSpecific.VP8.numberOfTemporalLayers =
-        static_cast<unsigned char>(
-            streams.back().temporal_layer_thresholds_bps.size() + 1);
-  } else if (video_codec.codecType == kVideoCodecVP9) {
-    if (config.encoder_specific_settings) {
-      video_codec.codecSpecific.VP9 = *reinterpret_cast<const VideoCodecVP9*>(
-                                          config.encoder_specific_settings);
-      if (video_codec.mode == kScreensharing) {
-        video_codec.codecSpecific.VP9.flexibleMode = true;
-        // For now VP9 screensharing use 1 temporal and 2 spatial layers.
-        RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfTemporalLayers, 1);
-        RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfSpatialLayers, 2);
+int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image,
+                                 const CodecSpecificInfo* codec_specific_info,
+                                 const RTPFragmentationHeader* fragmentation) {
+  // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|;
+  encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info,
+                               fragmentation);
+  int32_t return_value = payload_router_.Encoded(
+      encoded_image, codec_specific_info, fragmentation);
+
+  if (kEnableFrameRecording) {
+    int layer = codec_specific_info->codecType == kVideoCodecVP8
+                    ? codec_specific_info->codecSpecific.VP8.simulcastIdx
+                    : 0;
+    IvfFileWriter* file_writer;
+    {
+      if (file_writers_[layer] == nullptr) {
+        std::ostringstream oss;
+        oss << "send_bitstream_ssrc";
+        for (uint32_t ssrc : config_.rtp.ssrcs)
+          oss << "_" << ssrc;
+        oss << "_layer" << layer << ".ivf";
+        file_writers_[layer] =
+            IvfFileWriter::Open(oss.str(), codec_specific_info->codecType);
       }
+      file_writer = file_writers_[layer].get();
     }
-    video_codec.codecSpecific.VP9.numberOfTemporalLayers =
-        static_cast<unsigned char>(
-            streams.back().temporal_layer_thresholds_bps.size() + 1);
-  } else if (video_codec.codecType == kVideoCodecH264) {
-    if (config.encoder_specific_settings) {
-      video_codec.codecSpecific.H264 = *reinterpret_cast<const VideoCodecH264*>(
-                                           config.encoder_specific_settings);
+    if (file_writer) {
+      bool ok = file_writer->WriteFrame(encoded_image);
+      RTC_DCHECK(ok);
     }
-  } else {
-    // TODO(pbos): Support encoder_settings codec-agnostically.
-    RTC_DCHECK(!config.encoder_specific_settings)
-        << "Encoder-specific settings for codec type not wired up.";
   }
 
-  strncpy(video_codec.plName,
-          config_.encoder_settings.payload_name.c_str(),
-          kPayloadNameSize - 1);
-  video_codec.plName[kPayloadNameSize - 1] = '\0';
-  video_codec.plType = config_.encoder_settings.payload_type;
-  video_codec.numberOfSimulcastStreams =
-      static_cast<unsigned char>(streams.size());
-  video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
-  if (video_codec.minBitrate < kEncoderMinBitrateKbps)
-    video_codec.minBitrate = kEncoderMinBitrateKbps;
-  RTC_DCHECK_LE(streams.size(), static_cast<size_t>(kMaxSimulcastStreams));
-  if (video_codec.codecType == kVideoCodecVP9) {
-    // If the vector is empty, bitrates will be configured automatically.
-    RTC_DCHECK(config.spatial_layers.empty() ||
-               config.spatial_layers.size() ==
-                   video_codec.codecSpecific.VP9.numberOfSpatialLayers);
-    RTC_DCHECK_LE(video_codec.codecSpecific.VP9.numberOfSpatialLayers,
-                  kMaxSimulcastStreams);
-    for (size_t i = 0; i < config.spatial_layers.size(); ++i)
-      video_codec.spatialLayers[i] = config.spatial_layers[i];
+  return return_value;
+}
+
+void VideoSendStream::ConfigureProtection() {
+  // Enable NACK, FEC or both.
+  const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
+  bool enable_protection_fec = config_.rtp.fec.ulpfec_payload_type != -1;
+  // Payload types without picture ID cannot determine that a stream is complete
+  // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is
+  // a waste of bandwidth since FEC packets still have to be transmitted. Note
+  // that this is not the case with FLEXFEC.
+  if (enable_protection_nack &&
+      !PayloadTypeSupportsSkippingFecPackets(
+          config_.encoder_settings.payload_name)) {
+    LOG(LS_WARNING) << "Transmitting payload type without picture ID using"
+                       "NACK+FEC is a waste of bandwidth since FEC packets "
+                       "also have to be retransmitted. Disabling FEC.";
+    enable_protection_fec = false;
   }
-  for (size_t i = 0; i < streams.size(); ++i) {
-    SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
-    RTC_DCHECK_GT(streams[i].width, 0u);
-    RTC_DCHECK_GT(streams[i].height, 0u);
-    RTC_DCHECK_GT(streams[i].max_framerate, 0);
-    // Different framerates not supported per stream at the moment.
-    RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
-    RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0);
-    RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
-    RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
-    RTC_DCHECK_GE(streams[i].max_qp, 0);
 
-    sim_stream->width = static_cast<uint16_t>(streams[i].width);
-    sim_stream->height = static_cast<uint16_t>(streams[i].height);
-    sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
-    sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
-    sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
-    sim_stream->qpMax = streams[i].max_qp;
-    sim_stream->numberOfTemporalLayers = static_cast<unsigned char>(
-        streams[i].temporal_layer_thresholds_bps.size() + 1);
+  // Set to valid uint8_ts to be castable later without signed overflows.
+  uint8_t payload_type_red = 0;
+  uint8_t payload_type_fec = 0;
 
-    video_codec.width = std::max(video_codec.width,
-                                 static_cast<uint16_t>(streams[i].width));
-    video_codec.height = std::max(
-        video_codec.height, static_cast<uint16_t>(streams[i].height));
-    video_codec.minBitrate =
-        std::min(static_cast<uint16_t>(video_codec.minBitrate),
-                 static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
-    video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
-    video_codec.qpMax = std::max(video_codec.qpMax,
-                                 static_cast<unsigned int>(streams[i].max_qp));
+  // TODO(changbin): Should set RTX for RED mapping in RTP sender in future.
+  // Validate payload types. If either RED or FEC payload types are set then
+  // both should be. If FEC is enabled then they both have to be set.
+  if (config_.rtp.fec.red_payload_type != -1) {
+    RTC_DCHECK_GE(config_.rtp.fec.red_payload_type, 0);
+    RTC_DCHECK_LE(config_.rtp.fec.red_payload_type, 127);
+    // TODO(holmer): We should only enable red if ulpfec is also enabled, but
+    // but due to an incompatibility issue with previous versions the receiver
+    // assumes rtx packets are containing red if it has been configured to
+    // receive red. Remove this in a few versions once the incompatibility
+    // issue is resolved (M53 timeframe).
+    payload_type_red = static_cast<uint8_t>(config_.rtp.fec.red_payload_type);
   }
-
-  if (video_codec.maxBitrate == 0) {
-    // Unset max bitrate -> cap to one bit per pixel.
-    video_codec.maxBitrate =
-        (video_codec.width * video_codec.height * video_codec.maxFramerate) /
-        1000;
+  if (config_.rtp.fec.ulpfec_payload_type != -1) {
+    RTC_DCHECK_GE(config_.rtp.fec.ulpfec_payload_type, 0);
+    RTC_DCHECK_LE(config_.rtp.fec.ulpfec_payload_type, 127);
+    payload_type_fec =
+        static_cast<uint8_t>(config_.rtp.fec.ulpfec_payload_type);
   }
-  if (video_codec.maxBitrate < kEncoderMinBitrateKbps)
-    video_codec.maxBitrate = kEncoderMinBitrateKbps;
-
-  RTC_DCHECK_GT(streams[0].max_framerate, 0);
-  video_codec.maxFramerate = streams[0].max_framerate;
-
-  video_codec.startBitrate =
-      bitrate_allocator_->AddObserver(this,
-                                      video_codec.minBitrate * 1000,
-                                      video_codec.maxBitrate * 1000) / 1000;
-  vie_encoder_.SetEncoder(video_codec, config.min_transmit_bitrate_bps);
-}
 
-bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return vie_receiver_->DeliverRtcp(packet, length);
-}
-
-VideoSendStream::Stats VideoSendStream::GetStats() {
-  return stats_proxy_.GetStats();
-}
-
-void VideoSendStream::OveruseDetected() {
-  if (config_.overuse_callback)
-    config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
-}
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    // Set NACK.
+    rtp_rtcp->SetStorePacketsStatus(
+        enable_protection_nack || congestion_controller_->pacer(),
+        kMinSendSidePacketHistorySize);
+    // Set FEC.
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+      rtp_rtcp->SetGenericFECStatus(enable_protection_fec, payload_type_red,
+                                    payload_type_fec);
+    }
+  }
 
-void VideoSendStream::NormalUsage() {
-  if (config_.overuse_callback)
-    config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
+  vie_encoder_.SetProtectionMethod(enable_protection_nack,
+                                   enable_protection_fec);
 }
 
 void VideoSendStream::ConfigureSsrcs() {
@@ -560,7 +718,7 @@ void VideoSendStream::ConfigureSsrcs() {
     // Restore RTP state if previous existed.
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
-      rtp_rtcp->SetRtpStateForSsrc(ssrc, it->second);
+      rtp_rtcp->SetRtpState(it->second);
   }
 
   // Set up RTX if available.
@@ -575,7 +733,7 @@ void VideoSendStream::ConfigureSsrcs() {
     rtp_rtcp->SetRtxSsrc(ssrc);
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
-      rtp_rtcp->SetRtpStateForSsrc(ssrc, it->second);
+      rtp_rtcp->SetRtxState(it->second);
   }
 
   // Configure RTX payload types.
@@ -598,29 +756,22 @@ std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
   std::map<uint32_t, RtpState> rtp_states;
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
-    rtp_states[ssrc] = vie_channel_.GetRtpStateForSsrc(ssrc);
+    RTC_DCHECK_EQ(ssrc, rtp_rtcp_modules_[i]->SSRC());
+    rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtpState();
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
-    rtp_states[ssrc] = vie_channel_.GetRtpStateForSsrc(ssrc);
+    rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState();
   }
 
   return rtp_states;
 }
 
 void VideoSendStream::SignalNetworkState(NetworkState state) {
-  // When network goes up, enable RTCP status before setting transmission state.
-  // When it goes down, disable RTCP afterwards. This ensures that any packets
-  // sent due to the network state changed will not be dropped.
-  if (state == kNetworkUp) {
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-      rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode);
-  }
-  vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
-  if (state == kNetworkDown) {
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-      rtp_rtcp->SetRTCPStatus(RtcpMode::kOff);
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
+                                                : RtcpMode::kOff);
   }
 }
 
@@ -631,8 +782,32 @@ int VideoSendStream::GetPaddingNeededBps() const {
 void VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                        uint8_t fraction_loss,
                                        int64_t rtt) {
+  payload_router_.SetTargetSendBitrate(bitrate_bps);
   vie_encoder_.OnBitrateUpdated(bitrate_bps, fraction_loss, rtt);
 }
 
+int VideoSendStream::ProtectionRequest(const FecProtectionParams* delta_params,
+                                       const FecProtectionParams* key_params,
+                                       uint32_t* sent_video_rate_bps,
+                                       uint32_t* sent_nack_rate_bps,
+                                       uint32_t* sent_fec_rate_bps) {
+  *sent_video_rate_bps = 0;
+  *sent_nack_rate_bps = 0;
+  *sent_fec_rate_bps = 0;
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+    uint32_t not_used = 0;
+    uint32_t module_video_rate = 0;
+    uint32_t module_fec_rate = 0;
+    uint32_t module_nack_rate = 0;
+    rtp_rtcp->SetFecParameters(delta_params, key_params);
+    rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
+                          &module_nack_rate);
+    *sent_video_rate_bps += module_video_rate;
+    *sent_nack_rate_bps += module_nack_rate;
+    *sent_fec_rate_bps += module_fec_rate;
+  }
+  return 0;
+}
+
 }  // namespace internal
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/video_send_stream.h b/chromium/third_party/webrtc/video/video_send_stream.h
index 8b3d064f3e3..fa6a7a7c16d 100644
--- a/chromium/third_party/webrtc/video/video_send_stream.h
+++ b/chromium/third_party/webrtc/video/video_send_stream.h
@@ -12,17 +12,19 @@
 #define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
 
 #include <map>
+#include <memory>
 #include <vector>
 
 #include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 #include "webrtc/video/encoder_state_feedback.h"
 #include "webrtc/video/payload_router.h"
+#include "webrtc/video/send_delay_stats.h"
 #include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video/video_capture_input.h"
-#include "webrtc/video/vie_channel.h"
 #include "webrtc/video/vie_encoder.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
@@ -32,23 +34,30 @@ namespace webrtc {
 class BitrateAllocator;
 class CallStats;
 class CongestionController;
+class IvfFileWriter;
 class ProcessThread;
 class RtpRtcp;
-class ViEChannel;
 class ViEEncoder;
 class VieRemb;
 
+namespace vcm {
+class VideoSender;
+}  // namespace vcm
+
 namespace internal {
 
 class VideoSendStream : public webrtc::VideoSendStream,
                         public webrtc::CpuOveruseObserver,
-                        public webrtc::BitrateAllocatorObserver {
+                        public webrtc::BitrateAllocatorObserver,
+                        public webrtc::VCMProtectionCallback,
+                        protected webrtc::EncodedImageCallback {
  public:
   VideoSendStream(int num_cpu_cores,
                   ProcessThread* module_process_thread,
                   CallStats* call_stats,
                   CongestionController* congestion_controller,
                   BitrateAllocator* bitrate_allocator,
+                  SendDelayStats* send_delay_stats,
                   VieRemb* remb,
                   const VideoSendStream::Config& config,
                   const VideoEncoderConfig& encoder_config,
@@ -56,13 +65,12 @@ class VideoSendStream : public webrtc::VideoSendStream,
 
   ~VideoSendStream() override;
 
-  // webrtc::SendStream implementation.
-  void Start() override;
-  void Stop() override;
-  void SignalNetworkState(NetworkState state) override;
-  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
+  void SignalNetworkState(NetworkState state);
+  bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   // webrtc::VideoSendStream implementation.
+  void Start() override;
+  void Stop() override;
   VideoCaptureInput* Input() override;
   void ReconfigureVideoEncoder(const VideoEncoderConfig& config) override;
   Stats GetStats() override;
@@ -81,10 +89,31 @@ class VideoSendStream : public webrtc::VideoSendStream,
                         uint8_t fraction_loss,
                         int64_t rtt) override;
 
+  // Implements webrtc::VCMProtectionCallback.
+  int ProtectionRequest(const FecProtectionParams* delta_params,
+                        const FecProtectionParams* key_params,
+                        uint32_t* sent_video_rate_bps,
+                        uint32_t* sent_nack_rate_bps,
+                        uint32_t* sent_fec_rate_bps) override;
+
  private:
+  struct EncoderSettings {
+    VideoCodec video_codec;
+    int min_transmit_bitrate_bps;
+    std::vector<VideoStream> streams;
+  };
+
+  // Implements EncodedImageCallback. The implementation routes encoded frames
+  // to the |payload_router_| and |config.pre_encode_callback| if set.
+  // Called on an arbitrary encoder callback thread.
+  int32_t Encoded(const EncodedImage& encoded_image,
+                  const CodecSpecificInfo* codec_specific_info,
+                  const RTPFragmentationHeader* fragmentation) override;
+
   static bool EncoderThreadFunction(void* obj);
   void EncoderProcess();
 
+  void ConfigureProtection();
   void ConfigureSsrcs();
 
   SendStatisticsProxy stats_proxy_;
@@ -98,21 +127,26 @@ class VideoSendStream : public webrtc::VideoSendStream,
   BitrateAllocator* const bitrate_allocator_;
   VieRemb* const remb_;
 
+  static const bool kEnableFrameRecording = false;
+  static const int kMaxLayers = 3;
+  std::unique_ptr<IvfFileWriter> file_writers_[kMaxLayers];
+
   rtc::PlatformThread encoder_thread_;
   rtc::Event encoder_wakeup_event_;
   volatile int stop_encoder_thread_;
+  rtc::CriticalSection encoder_settings_crit_;
+  rtc::Optional<EncoderSettings> pending_encoder_settings_
+      GUARDED_BY(encoder_settings_crit_);
 
   OveruseFrameDetector overuse_detector_;
-  PayloadRouter payload_router_;
-  EncoderStateFeedback encoder_feedback_;
-  ViEChannel vie_channel_;
-  ViEReceiver* const vie_receiver_;
   ViEEncoder vie_encoder_;
-  VideoCodingModule* const vcm_;
-  // TODO(pbos): Move RtpRtcp ownership to VideoSendStream.
-  // RtpRtcp modules, currently owned by ViEChannel but ownership should
-  // eventually move here.
+  EncoderStateFeedback encoder_feedback_;
+  vcm::VideoSender* const video_sender_;
+
+  const std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
+  // RtpRtcp modules, declared here as they use other members on construction.
   const std::vector<RtpRtcp*> rtp_rtcp_modules_;
+  PayloadRouter payload_router_;
   VideoCaptureInput input_;
 };
 }  // namespace internal
diff --git a/chromium/third_party/webrtc/video/video_send_stream_tests.cc b/chromium/third_party/webrtc/video/video_send_stream_tests.cc
index 53f72de5025..1d4acfcb04f 100644
--- a/chromium/third_party/webrtc/video/video_send_stream_tests.cc
+++ b/chromium/third_party/webrtc/video/video_send_stream_tests.cc
@@ -21,7 +21,7 @@
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
@@ -32,6 +32,7 @@
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/configurable_frame_size_encoder.h"
 #include "webrtc/test/fake_texture_frame.h"
+#include "webrtc/test/frame_utils.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/video/send_statistics_proxy.h"
@@ -42,11 +43,6 @@ namespace webrtc {
 
 enum VideoFormat { kGeneric, kVP8, };
 
-void ExpectEqualFrames(const VideoFrame& frame1, const VideoFrame& frame2);
-void ExpectEqualTextureFrames(const VideoFrame& frame1,
-                              const VideoFrame& frame2);
-void ExpectEqualBufferFrames(const VideoFrame& frame1,
-                             const VideoFrame& frame2);
 void ExpectEqualFramesVector(const std::vector<VideoFrame>& frames1,
                              const std::vector<VideoFrame>& frames2);
 VideoFrame CreateVideoFrame(int width, int height, uint8_t data);
@@ -312,11 +308,13 @@ class FecObserver : public test::EndToEndTest {
   FecObserver(bool header_extensions_enabled,
               bool use_nack,
               bool expect_red,
+              bool expect_fec,
               const std::string& codec)
       : EndToEndTest(VideoSendStreamTest::kDefaultTimeoutMs),
         payload_name_(codec),
         use_nack_(use_nack),
         expect_red_(expect_red),
+        expect_fec_(expect_fec),
         send_count_(0),
         received_media_(false),
         received_fec_(false),
@@ -379,6 +377,7 @@ class FecObserver : public test::EndToEndTest {
     if (encapsulated_payload_type != -1) {
       if (encapsulated_payload_type ==
           VideoSendStreamTest::kUlpfecPayloadType) {
+        EXPECT_TRUE(expect_fec_);
         received_fec_ = true;
       } else {
         received_media_ = true;
@@ -386,7 +385,7 @@ class FecObserver : public test::EndToEndTest {
     }
 
     if (send_count_ > 100 && received_media_) {
-      if (received_fec_ || !expect_red_)
+      if (received_fec_ || !expect_fec_)
         observation_complete_.Set();
     }
 
@@ -446,6 +445,7 @@ class FecObserver : public test::EndToEndTest {
   const std::string payload_name_;
   const bool use_nack_;
   const bool expect_red_;
+  const bool expect_fec_;
   int send_count_;
   bool received_media_;
   bool received_fec_;
@@ -454,12 +454,12 @@ class FecObserver : public test::EndToEndTest {
 };
 
 TEST_F(VideoSendStreamTest, SupportsFecWithExtensions) {
-  FecObserver test(true, false, true, "VP8");
+  FecObserver test(true, false, true, true, "VP8");
   RunBaseTest(&test);
 }
 
 TEST_F(VideoSendStreamTest, SupportsFecWithoutExtensions) {
-  FecObserver test(false, false, true, "VP8");
+  FecObserver test(false, false, true, true, "VP8");
   RunBaseTest(&test);
 }
 
@@ -467,26 +467,28 @@ TEST_F(VideoSendStreamTest, SupportsFecWithoutExtensions) {
 // since we'll still have to re-request FEC packets, effectively wasting
 // bandwidth since the receiver has to wait for FEC retransmissions to determine
 // that the received state is actually decodable.
-TEST_F(VideoSendStreamTest, DoesNotUtilizeRedForH264WithNackEnabled) {
-  FecObserver test(false, true, false, "H264");
+TEST_F(VideoSendStreamTest, DoesNotUtilizeFecForH264WithNackEnabled) {
+  FecObserver test(false, true, true, false, "H264");
   RunBaseTest(&test);
 }
 
 // Without retransmissions FEC for H264 is fine.
 TEST_F(VideoSendStreamTest, DoesUtilizeRedForH264WithoutNackEnabled) {
-  FecObserver test(false, false, true, "H264");
+  FecObserver test(false, false, true, true, "H264");
   RunBaseTest(&test);
 }
 
 TEST_F(VideoSendStreamTest, DoesUtilizeRedForVp8WithNackEnabled) {
-  FecObserver test(false, true, true, "VP8");
+  FecObserver test(false, true, true, true, "VP8");
   RunBaseTest(&test);
 }
 
+#if !defined(RTC_DISABLE_VP9)
 TEST_F(VideoSendStreamTest, DoesUtilizeRedForVp9WithNackEnabled) {
-  FecObserver test(false, true, true, "VP9");
+  FecObserver test(false, true, true, true, "VP9");
   RunBaseTest(&test);
 }
+#endif  // !defined(RTC_DISABLE_VP9)
 
 void VideoSendStreamTest::TestNackRetransmission(
     uint32_t retransmit_ssrc,
@@ -712,7 +714,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
       }
     }
 
-    virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
+    void EncodedFrameCallback(const EncodedFrame& encoded_frame) override {
       // Increase frame size for next encoded frame, in the context of the
       // encoder thread.
       if (!use_fec_ &&
@@ -811,7 +813,8 @@ TEST_F(VideoSendStreamTest, FragmentsVp8AccordingToMaxPacketSizeWithFec) {
 TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
   static const int kSuspendTimeFrames = 60;  // Suspend for 2 seconds @ 30 fps.
 
-  class RembObserver : public test::SendTest, public I420FrameCallback {
+  class RembObserver : public test::SendTest,
+                       public rtc::VideoSinkInterface<VideoFrame> {
    public:
     RembObserver()
         : SendTest(kDefaultTimeoutMs),
@@ -862,8 +865,8 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
       return SEND_PACKET;
     }
 
-    // This method implements the I420FrameCallback.
-    void FrameCallback(VideoFrame* video_frame) override {
+    // This method implements the rtc::VideoSinkInterface
+    void OnFrame(const VideoFrame& video_frame) override {
       rtc::CritScope lock(&crit_);
       if (test_state_ == kDuringSuspend &&
           ++suspended_frame_count_ > kSuspendTimeFrames) {
@@ -1002,8 +1005,8 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
 
     size_t GetNumVideoStreams() const override { return 3; }
 
-    virtual void OnFrameGeneratorCapturerCreated(
-        test::FrameGeneratorCapturer* frame_generator_capturer) {
+    void OnFrameGeneratorCapturerCreated(
+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
       rtc::CritScope lock(&crit_);
       capturer_ = frame_generator_capturer;
     }
@@ -1043,7 +1046,7 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (RtpHeaderParser::IsRtcp(packet, length))
         return DROP_PACKET;
 
@@ -1115,18 +1118,22 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
   class StartBitrateObserver : public test::FakeEncoder {
    public:
     StartBitrateObserver()
-        : FakeEncoder(Clock::GetRealTimeClock()), start_bitrate_kbps_(0) {}
+        : FakeEncoder(Clock::GetRealTimeClock()),
+          start_bitrate_changed_(false, false),
+          start_bitrate_kbps_(0) {}
     int32_t InitEncode(const VideoCodec* config,
                        int32_t number_of_cores,
                        size_t max_payload_size) override {
       rtc::CritScope lock(&crit_);
       start_bitrate_kbps_ = config->startBitrate;
+      start_bitrate_changed_.Set();
       return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
     }
 
     int32_t SetRates(uint32_t new_target_bitrate, uint32_t framerate) override {
       rtc::CritScope lock(&crit_);
       start_bitrate_kbps_ = new_target_bitrate;
+      start_bitrate_changed_.Set();
       return FakeEncoder::SetRates(new_target_bitrate, framerate);
     }
 
@@ -1135,8 +1142,14 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
       return start_bitrate_kbps_;
     }
 
+    bool WaitForStartBitrate() {
+      return start_bitrate_changed_.Wait(
+          VideoSendStreamTest::kDefaultTimeoutMs);
+    }
+
    private:
     rtc::CriticalSection crit_;
+    rtc::Event start_bitrate_changed_;
     int start_bitrate_kbps_ GUARDED_BY(crit_);
   };
 
@@ -1155,6 +1168,7 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
 
   CreateVideoStreams();
 
+  EXPECT_TRUE(encoder.WaitForStartBitrate());
   EXPECT_EQ(video_encoder_config_.streams[0].max_bitrate_bps / 1000,
             encoder.GetStartBitrateKbps());
 
@@ -1165,6 +1179,7 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
   // New bitrate should be reconfigured above the previous max. As there's no
   // network connection this shouldn't be flaky, as no bitrate should've been
   // reported in between.
+  EXPECT_TRUE(encoder.WaitForStartBitrate());
   EXPECT_EQ(bitrate_config.start_bitrate_bps / 1000,
             encoder.GetStartBitrateKbps());
 
@@ -1172,12 +1187,12 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
 }
 
 TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
-  class FrameObserver : public I420FrameCallback {
+  class FrameObserver : public rtc::VideoSinkInterface<VideoFrame> {
    public:
     FrameObserver() : output_frame_event_(false, false) {}
 
-    void FrameCallback(VideoFrame* video_frame) override {
-      output_frames_.push_back(*video_frame);
+    void OnFrame(const VideoFrame& video_frame) override {
+      output_frames_.push_back(video_frame);
       output_frame_event_.Set();
     }
 
@@ -1244,49 +1259,13 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
   DestroyStreams();
 }
 
-void ExpectEqualFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
-  if (frame1.native_handle() || frame2.native_handle())
-    ExpectEqualTextureFrames(frame1, frame2);
-  else
-    ExpectEqualBufferFrames(frame1, frame2);
-}
-
-void ExpectEqualTextureFrames(const VideoFrame& frame1,
-                              const VideoFrame& frame2) {
-  EXPECT_EQ(frame1.native_handle(), frame2.native_handle());
-  EXPECT_EQ(frame1.width(), frame2.width());
-  EXPECT_EQ(frame1.height(), frame2.height());
-}
-
-void ExpectEqualBufferFrames(const VideoFrame& frame1,
-                             const VideoFrame& frame2) {
-  EXPECT_EQ(frame1.width(), frame2.width());
-  EXPECT_EQ(frame1.height(), frame2.height());
-  EXPECT_EQ(frame1.stride(kYPlane), frame2.stride(kYPlane));
-  EXPECT_EQ(frame1.stride(kUPlane), frame2.stride(kUPlane));
-  EXPECT_EQ(frame1.stride(kVPlane), frame2.stride(kVPlane));
-  ASSERT_EQ(frame1.allocated_size(kYPlane), frame2.allocated_size(kYPlane));
-  EXPECT_EQ(0,
-            memcmp(frame1.buffer(kYPlane),
-                   frame2.buffer(kYPlane),
-                   frame1.allocated_size(kYPlane)));
-  ASSERT_EQ(frame1.allocated_size(kUPlane), frame2.allocated_size(kUPlane));
-  EXPECT_EQ(0,
-            memcmp(frame1.buffer(kUPlane),
-                   frame2.buffer(kUPlane),
-                   frame1.allocated_size(kUPlane)));
-  ASSERT_EQ(frame1.allocated_size(kVPlane), frame2.allocated_size(kVPlane));
-  EXPECT_EQ(0,
-            memcmp(frame1.buffer(kVPlane),
-                   frame2.buffer(kVPlane),
-                   frame1.allocated_size(kVPlane)));
-}
-
 void ExpectEqualFramesVector(const std::vector<VideoFrame>& frames1,
                              const std::vector<VideoFrame>& frames2) {
   EXPECT_EQ(frames1.size(), frames2.size());
   for (size_t i = 0; i < std::min(frames1.size(), frames2.size()); ++i)
-    ExpectEqualFrames(frames1[i], frames2[i]);
+    // Compare frame buffers, since we don't care about differing timestamps.
+    EXPECT_TRUE(test::FrameBufsEqual(frames1[i].video_frame_buffer(),
+                                     frames2[i].video_frame_buffer()));
 }
 
 VideoFrame CreateVideoFrame(int width, int height, uint8_t data) {
@@ -1378,9 +1357,6 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
-      // Encoder initialization should be done in stream construction before
-      // starting.
-      EXPECT_TRUE(IsReadyForEncode());
       stream_ = send_stream;
     }
 
@@ -1428,6 +1404,7 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
     VideoCodecConfigObserver()
         : SendTest(kDefaultTimeoutMs),
           FakeEncoder(Clock::GetRealTimeClock()),
+          init_encode_event_(false, false),
           num_initializations_(0) {}
 
    private:
@@ -1456,19 +1433,23 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
         EXPECT_EQ(kScreensharing, config->mode);
       }
       ++num_initializations_;
+      init_encode_event_.Set();
       return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
     }
 
     void PerformTest() override {
+      EXPECT_TRUE(init_encode_event_.Wait(kDefaultTimeoutMs));
       EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
 
       encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
       stream_->ReconfigureVideoEncoder(encoder_config_);
+      EXPECT_TRUE(init_encode_event_.Wait(kDefaultTimeoutMs));
       EXPECT_EQ(2u, num_initializations_)
           << "ReconfigureVideoEncoder did not reinitialize the encoder with "
              "new encoder settings.";
     }
 
+    rtc::Event init_encode_event_;
     size_t num_initializations_;
     VideoSendStream* stream_;
     VideoEncoderConfig encoder_config_;
@@ -1488,6 +1469,7 @@ class VideoCodecConfigObserver : public test::SendTest,
         FakeEncoder(Clock::GetRealTimeClock()),
         video_codec_type_(video_codec_type),
         codec_name_(codec_name),
+        init_encode_event_(false, false),
         num_initializations_(0) {
     memset(&encoder_settings_, 0, sizeof(encoder_settings_));
   }
@@ -1521,16 +1503,21 @@ class VideoCodecConfigObserver : public test::SendTest,
     EXPECT_EQ(video_codec_type_, config->codecType);
     VerifyCodecSpecifics(*config);
     ++num_initializations_;
+    init_encode_event_.Set();
     return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
   }
 
   void VerifyCodecSpecifics(const VideoCodec& config) const;
 
   void PerformTest() override {
-    EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
+    EXPECT_TRUE(
+        init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
+    ASSERT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
 
     encoder_settings_.frameDroppingOn = true;
     stream_->ReconfigureVideoEncoder(encoder_config_);
+    ASSERT_TRUE(
+        init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
     EXPECT_EQ(2u, num_initializations_)
         << "ReconfigureVideoEncoder did not reinitialize the encoder with "
            "new encoder settings.";
@@ -1546,6 +1533,7 @@ class VideoCodecConfigObserver : public test::SendTest,
   T encoder_settings_;
   const VideoCodecType video_codec_type_;
   const char* const codec_name_;
+  rtc::Event init_encode_event_;
   size_t num_initializations_;
   VideoSendStream* stream_;
   VideoEncoderConfig encoder_config_;
@@ -1708,18 +1696,7 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
   RunBaseTest(&test);
 }
 
-// Disabled on LinuxAsan:
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5382
-#if defined(ADDRESS_SANITIZER) && defined(WEBRTC_LINUX)
-#define MAYBE_ReconfigureBitratesSetsEncoderBitratesCorrectly \
-  DISABLED_ReconfigureBitratesSetsEncoderBitratesCorrectly
-#else
-#define MAYBE_ReconfigureBitratesSetsEncoderBitratesCorrectly \
-  ReconfigureBitratesSetsEncoderBitratesCorrectly
-#endif
-
-TEST_F(VideoSendStreamTest,
-       MAYBE_ReconfigureBitratesSetsEncoderBitratesCorrectly) {
+TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
   // These are chosen to be "kind of odd" to not be accidentally checked against
   // default values.
   static const int kMinBitrateKbps = 137;
@@ -1734,6 +1711,7 @@ TEST_F(VideoSendStreamTest,
     EncoderBitrateThresholdObserver()
         : SendTest(kDefaultTimeoutMs),
           FakeEncoder(Clock::GetRealTimeClock()),
+          init_encode_event_(false, false),
           num_initializations_(0) {}
 
    private:
@@ -1762,6 +1740,7 @@ TEST_F(VideoSendStreamTest,
                   codecSettings->startBitrate);
       }
       ++num_initializations_;
+      init_encode_event_.Set();
       return FakeEncoder::InitEncode(codecSettings, numberOfCores,
                                      maxPayloadSize);
     }
@@ -1797,6 +1776,9 @@ TEST_F(VideoSendStreamTest,
     }
 
     void PerformTest() override {
+      ASSERT_TRUE(
+          init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs))
+          << "Timed out while waiting encoder to be configured.";
       Call::Config::BitrateConfig bitrate_config;
       bitrate_config.start_bitrate_bps = kIncreasedStartBitrateKbps * 1000;
       bitrate_config.max_bitrate_bps = kIncreasedMaxBitrateKbps * 1000;
@@ -1806,6 +1788,8 @@ TEST_F(VideoSendStreamTest,
       encoder_config_.streams[0].min_bitrate_bps = 0;
       encoder_config_.streams[0].max_bitrate_bps = kLowerMaxBitrateKbps * 1000;
       send_stream_->ReconfigureVideoEncoder(encoder_config_);
+      ASSERT_TRUE(
+          init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
       EXPECT_EQ(2, num_initializations_)
           << "Encoder should have been reconfigured with the new value.";
       encoder_config_.streams[0].target_bitrate_bps =
@@ -1813,10 +1797,13 @@ TEST_F(VideoSendStreamTest,
       encoder_config_.streams[0].max_bitrate_bps =
           kIncreasedMaxBitrateKbps * 1000;
       send_stream_->ReconfigureVideoEncoder(encoder_config_);
+      ASSERT_TRUE(
+          init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
       EXPECT_EQ(3, num_initializations_)
           << "Encoder should have been reconfigured with the new value.";
     }
 
+    rtc::Event init_encode_event_;
     int num_initializations_;
     webrtc::Call* call_;
     webrtc::VideoSendStream* send_stream_;
@@ -1846,7 +1833,6 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
                    const CodecSpecificInfo* codecSpecificInfo,
                    const std::vector<FrameType>* frame_types) override {
       CodecSpecificInfo specifics;
-      memset(&specifics, 0, sizeof(specifics));
       specifics.codecType = kVideoCodecGeneric;
 
       uint8_t buffer[16] = {0};
@@ -1906,6 +1892,7 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
   RunBaseTest(&test);
 }
 
+#if !defined(RTC_DISABLE_VP9)
 class Vp9HeaderObserver : public test::SendTest {
  public:
   Vp9HeaderObserver()
@@ -2289,9 +2276,6 @@ TEST_F(VideoSendStreamTest, Vp9NonFlexModeSmallResolution) {
   RunBaseTest(&test);
 }
 
-#if !defined(MEMORY_SANITIZER)
-// Fails under MemorySanitizer:
-// See https://code.google.com/p/webrtc/issues/detail?id=5402.
 TEST_F(VideoSendStreamTest, Vp9FlexModeRefCount) {
   class FlexibleMode : public Vp9HeaderObserver {
     void ModifyVideoConfigsHook(
@@ -2316,6 +2300,6 @@ TEST_F(VideoSendStreamTest, Vp9FlexModeRefCount) {
 
   RunBaseTest(&test);
 }
-#endif
+#endif  // !defined(RTC_DISABLE_VP9)
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/video_stream_decoder.cc b/chromium/third_party/webrtc/video/video_stream_decoder.cc
new file mode 100644
index 00000000000..5aab7b246c7
--- /dev/null
+++ b/chromium/third_party/webrtc/video/video_stream_decoder.cc
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video/video_stream_decoder.h"
+
+#include <algorithm>
+#include <map>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/common_video/include/frame_callback.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/modules/video_processing/include/video_processing.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+#include "webrtc/video/call_stats.h"
+#include "webrtc/video/payload_router.h"
+#include "webrtc/video/receive_statistics_proxy.h"
+
+namespace webrtc {
+
+VideoStreamDecoder::VideoStreamDecoder(
+    vcm::VideoReceiver* video_receiver,
+    VCMFrameTypeCallback* vcm_frame_type_callback,
+    VCMPacketRequestCallback* vcm_packet_request_callback,
+    bool enable_nack,
+    bool enable_fec,  // TODO(philipel): Actually use this.
+    ReceiveStatisticsProxy* receive_statistics_proxy,
+    IncomingVideoStream* incoming_video_stream,
+    I420FrameCallback* pre_render_callback)
+    : video_receiver_(video_receiver),
+      receive_stats_callback_(receive_statistics_proxy),
+      incoming_video_stream_(incoming_video_stream),
+      pre_render_callback_(pre_render_callback),
+      last_rtt_ms_(0) {
+  RTC_DCHECK(video_receiver_);
+
+  static const int kMaxPacketAgeToNack = 450;
+  static const int kMaxNackListSize = 250;
+  video_receiver_->SetNackSettings(kMaxNackListSize,
+                                   kMaxPacketAgeToNack, 0);
+  video_receiver_->RegisterReceiveCallback(this);
+  video_receiver_->RegisterFrameTypeCallback(vcm_frame_type_callback);
+  video_receiver_->RegisterReceiveStatisticsCallback(this);
+  video_receiver_->RegisterDecoderTimingCallback(this);
+  static const int kDefaultRenderDelayMs = 10;
+  video_receiver_->SetRenderDelay(kDefaultRenderDelayMs);
+
+  VCMVideoProtection video_protection = kProtectionNone;
+  if (enable_nack) {
+    if (enable_fec)
+      video_protection = kProtectionNackFEC;
+    else
+      video_protection = kProtectionNack;
+  }
+
+  VCMDecodeErrorMode decode_error_mode = enable_nack ? kNoErrors : kWithErrors;
+  video_receiver_->SetVideoProtection(video_protection, true);
+  video_receiver_->SetDecodeErrorMode(decode_error_mode);
+  VCMPacketRequestCallback* packet_request_callback =
+      enable_nack ? vcm_packet_request_callback : nullptr;
+  video_receiver_->RegisterPacketRequestCallback(packet_request_callback);
+}
+
+VideoStreamDecoder::~VideoStreamDecoder() {}
+
+// Do not acquire the lock of |video_receiver_| in this function. Decode
+// callback won't necessarily be called from the decoding thread. The decoding
+// thread may have held the lock when calling VideoDecoder::Decode, Reset, or
+// Release. Acquiring the same lock in the path of decode callback can deadlock.
+int32_t VideoStreamDecoder::FrameToRender(VideoFrame& video_frame) {  // NOLINT
+  if (pre_render_callback_)
+    pre_render_callback_->FrameCallback(&video_frame);
+
+  incoming_video_stream_->OnFrame(video_frame);
+  return 0;
+}
+
+int32_t VideoStreamDecoder::ReceivedDecodedReferenceFrame(
+  const uint64_t picture_id) {
+  RTC_NOTREACHED();
+  return 0;
+}
+
+void VideoStreamDecoder::OnIncomingPayloadType(int payload_type) {
+  receive_stats_callback_->OnIncomingPayloadType(payload_type);
+}
+
+void VideoStreamDecoder::OnDecoderImplementationName(
+    const char* implementation_name) {
+  receive_stats_callback_->OnDecoderImplementationName(implementation_name);
+}
+
+void VideoStreamDecoder::OnReceiveRatesUpdated(uint32_t bit_rate,
+                                               uint32_t frame_rate) {
+  receive_stats_callback_->OnIncomingRate(frame_rate, bit_rate);
+}
+
+void VideoStreamDecoder::OnDiscardedPacketsUpdated(int discarded_packets) {
+  receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets);
+}
+
+void VideoStreamDecoder::OnFrameCountsUpdated(const FrameCounts& frame_counts) {
+  receive_stats_callback_->OnFrameCountsUpdated(frame_counts);
+}
+
+void VideoStreamDecoder::OnDecoderTiming(int decode_ms,
+                                         int max_decode_ms,
+                                         int current_delay_ms,
+                                         int target_delay_ms,
+                                         int jitter_buffer_ms,
+                                         int min_playout_delay_ms,
+                                         int render_delay_ms) {
+  int last_rtt = -1;
+  {
+    rtc::CritScope lock(&crit_);
+    last_rtt = last_rtt_ms_;
+  }
+
+  receive_stats_callback_->OnDecoderTiming(
+      decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
+      jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt);
+}
+
+void VideoStreamDecoder::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
+  video_receiver_->SetReceiveChannelParameters(max_rtt_ms);
+
+  rtc::CritScope lock(&crit_);
+  last_rtt_ms_ = avg_rtt_ms;
+}
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/video_stream_decoder.h b/chromium/third_party/webrtc/video/video_stream_decoder.h
new file mode 100644
index 00000000000..24a0ea3449b
--- /dev/null
+++ b/chromium/third_party/webrtc/video/video_stream_decoder.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_VIDEO_STREAM_DECODER_H_
+#define WEBRTC_VIDEO_VIDEO_STREAM_DECODER_H_
+
+#include <list>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video/vie_sync_module.h"
+
+namespace webrtc {
+
+class CallStatsObserver;
+class ChannelStatsObserver;
+class Config;
+class EncodedImageCallback;
+class I420FrameCallback;
+class IncomingVideoStream;
+class ReceiveStatisticsProxy;
+class VideoRenderCallback;
+class VoEVideoSync;
+
+namespace vcm {
+class VideoReceiver;
+}  // namespace vcm
+
+enum StreamType {
+  kViEStreamTypeNormal = 0,  // Normal media stream
+  kViEStreamTypeRtx = 1      // Retransmission media stream
+};
+
+class VideoStreamDecoder : public VCMReceiveCallback,
+                           public VCMReceiveStatisticsCallback,
+                           public VCMDecoderTimingCallback,
+                           public CallStatsObserver {
+ public:
+  friend class ChannelStatsObserver;
+
+  VideoStreamDecoder(vcm::VideoReceiver* video_receiver,
+                     VCMFrameTypeCallback* vcm_frame_type_callback,
+                     VCMPacketRequestCallback* vcm_packet_request_callback,
+                     bool enable_nack,
+                     bool enable_fec,
+                     ReceiveStatisticsProxy* receive_statistics_proxy,
+                     IncomingVideoStream* incoming_video_stream,
+                     I420FrameCallback* pre_render_callback);
+  ~VideoStreamDecoder();
+
+  // Implements VCMReceiveCallback.
+  int32_t FrameToRender(VideoFrame& video_frame) override;  // NOLINT
+  int32_t ReceivedDecodedReferenceFrame(const uint64_t picture_id) override;
+  void OnIncomingPayloadType(int payload_type) override;
+  void OnDecoderImplementationName(const char* implementation_name) override;
+
+  // Implements VCMReceiveStatisticsCallback.
+  void OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) override;
+  void OnDiscardedPacketsUpdated(int discarded_packets) override;
+  void OnFrameCountsUpdated(const FrameCounts& frame_counts) override;
+
+  // Implements VCMDecoderTimingCallback.
+  void OnDecoderTiming(int decode_ms,
+                       int max_decode_ms,
+                       int current_delay_ms,
+                       int target_delay_ms,
+                       int jitter_buffer_ms,
+                       int min_playout_delay_ms,
+                       int render_delay_ms) override;
+
+  void RegisterReceiveStatisticsProxy(
+      ReceiveStatisticsProxy* receive_statistics_proxy);
+
+  // Implements StatsObserver.
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+
+ private:
+  // Assumed to be protected.
+  void StartDecodeThread();
+  void StopDecodeThread();
+
+  // Used for all registered callbacks except rendering.
+  rtc::CriticalSection crit_;
+
+  vcm::VideoReceiver* const video_receiver_;
+
+  ReceiveStatisticsProxy* const receive_stats_callback_;
+  IncomingVideoStream* const incoming_video_stream_;
+
+  I420FrameCallback* const pre_render_callback_;
+
+  int64_t last_rtt_ms_ GUARDED_BY(crit_);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_VIDEO_STREAM_DECODER_H_
diff --git a/chromium/third_party/webrtc/video/vie_channel.cc b/chromium/third_party/webrtc/video/vie_channel.cc
deleted file mode 100644
index c0676bc83b1..00000000000
--- a/chromium/third_party/webrtc/video/vie_channel.cc
+++ /dev/null
@@ -1,515 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video/vie_channel.h"
-
-#include <algorithm>
-#include <map>
-#include <vector>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/common_video/include/incoming_video_stream.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/frame_callback.h"
-#include "webrtc/modules/pacing/paced_sender.h"
-#include "webrtc/modules/pacing/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/modules/video_coding/include/video_coding.h"
-#include "webrtc/modules/video_processing/include/video_processing.h"
-#include "webrtc/modules/video_render/video_render_defines.h"
-#include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/video/call_stats.h"
-#include "webrtc/video/payload_router.h"
-#include "webrtc/video/receive_statistics_proxy.h"
-
-namespace webrtc {
-
-static const int kMinSendSidePacketHistorySize = 600;
-static const int kMaxPacketAgeToNack = 450;
-static const int kMaxNackListSize = 250;
-
-// Helper class receiving statistics callbacks.
-class ChannelStatsObserver : public CallStatsObserver {
- public:
-  explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {}
-  virtual ~ChannelStatsObserver() {}
-
-  // Implements StatsObserver.
-  virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
-    owner_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
-  }
-
- private:
-  ViEChannel* const owner_;
-};
-
-class ViEChannelProtectionCallback : public VCMProtectionCallback {
- public:
-  explicit ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {}
-  ~ViEChannelProtectionCallback() {}
-
-
-  int ProtectionRequest(
-      const FecProtectionParams* delta_fec_params,
-      const FecProtectionParams* key_fec_params,
-      uint32_t* sent_video_rate_bps,
-      uint32_t* sent_nack_rate_bps,
-      uint32_t* sent_fec_rate_bps) override {
-    return owner_->ProtectionRequest(delta_fec_params, key_fec_params,
-                                     sent_video_rate_bps, sent_nack_rate_bps,
-                                     sent_fec_rate_bps);
-  }
- private:
-  ViEChannel* owner_;
-};
-
-ViEChannel::ViEChannel(Transport* transport,
-                       ProcessThread* module_process_thread,
-                       PayloadRouter* send_payload_router,
-                       VideoCodingModule* vcm,
-                       RtcpIntraFrameObserver* intra_frame_observer,
-                       RtcpBandwidthObserver* bandwidth_observer,
-                       TransportFeedbackObserver* transport_feedback_observer,
-                       RemoteBitrateEstimator* remote_bitrate_estimator,
-                       RtcpRttStats* rtt_stats,
-                       PacedSender* paced_sender,
-                       PacketRouter* packet_router,
-                       size_t max_rtp_streams,
-                       bool sender)
-    : sender_(sender),
-      module_process_thread_(module_process_thread),
-      send_payload_router_(send_payload_router),
-      vcm_protection_callback_(new ViEChannelProtectionCallback(this)),
-      vcm_(vcm),
-      vie_receiver_(vcm_, remote_bitrate_estimator, this),
-      stats_observer_(new ChannelStatsObserver(this)),
-      receive_stats_callback_(nullptr),
-      incoming_video_stream_(nullptr),
-      intra_frame_observer_(intra_frame_observer),
-      rtt_stats_(rtt_stats),
-      paced_sender_(paced_sender),
-      packet_router_(packet_router),
-      bandwidth_observer_(bandwidth_observer),
-      transport_feedback_observer_(transport_feedback_observer),
-      max_nack_reordering_threshold_(kMaxPacketAgeToNack),
-      pre_render_callback_(nullptr),
-      last_rtt_ms_(0),
-      rtp_rtcp_modules_(
-          CreateRtpRtcpModules(!sender,
-                               vie_receiver_.GetReceiveStatistics(),
-                               transport,
-                               intra_frame_observer_,
-                               bandwidth_observer_.get(),
-                               transport_feedback_observer_,
-                               rtt_stats_,
-                               &rtcp_packet_type_counter_observer_,
-                               remote_bitrate_estimator,
-                               paced_sender_,
-                               packet_router_,
-                               &send_bitrate_observer_,
-                               &send_frame_count_observer_,
-                               &send_side_delay_observer_,
-                               max_rtp_streams)) {
-  vie_receiver_.Init(rtp_rtcp_modules_);
-  if (sender_) {
-    RTC_DCHECK(send_payload_router_);
-    RTC_DCHECK(!vcm_);
-  } else {
-    RTC_DCHECK(!send_payload_router_);
-    RTC_DCHECK(vcm_);
-    vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
-  }
-}
-
-int32_t ViEChannel::Init() {
-  static const int kDefaultRenderDelayMs = 10;
-  module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics());
-
-  // RTP/RTCP initialization.
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    module_process_thread_->RegisterModule(rtp_rtcp);
-    packet_router_->AddRtpModule(rtp_rtcp);
-  }
-
-  rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
-  if (paced_sender_) {
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-      rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
-  }
-  if (sender_) {
-    send_payload_router_->SetSendingRtpModules(1);
-    RTC_DCHECK(!send_payload_router_->active());
-  } else {
-    if (vcm_->RegisterReceiveCallback(this) != 0) {
-      return -1;
-    }
-    vcm_->RegisterFrameTypeCallback(this);
-    vcm_->RegisterReceiveStatisticsCallback(this);
-    vcm_->RegisterDecoderTimingCallback(this);
-    vcm_->SetRenderDelay(kDefaultRenderDelayMs);
-  }
-  return 0;
-}
-
-ViEChannel::~ViEChannel() {
-  // Make sure we don't get more callbacks from the RTP module.
-  module_process_thread_->DeRegisterModule(
-      vie_receiver_.GetReceiveStatistics());
-  if (sender_) {
-    send_payload_router_->SetSendingRtpModules(0);
-  }
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    packet_router_->RemoveRtpModule(rtp_rtcp);
-    module_process_thread_->DeRegisterModule(rtp_rtcp);
-    delete rtp_rtcp;
-  }
-}
-
-void ViEChannel::SetProtectionMode(bool enable_nack,
-                                   bool enable_fec,
-                                   int payload_type_red,
-                                   int payload_type_fec) {
-  // Validate payload types. If either RED or FEC payload types are set then
-  // both should be. If FEC is enabled then they both have to be set.
-  if (enable_fec || payload_type_red != -1 || payload_type_fec != -1) {
-    RTC_DCHECK_GE(payload_type_red, 0);
-    RTC_DCHECK_GE(payload_type_fec, 0);
-    RTC_DCHECK_LE(payload_type_red, 127);
-    RTC_DCHECK_LE(payload_type_fec, 127);
-  } else {
-    // Payload types unset.
-    RTC_DCHECK_EQ(payload_type_red, -1);
-    RTC_DCHECK_EQ(payload_type_fec, -1);
-    // Set to valid uint8_ts to be castable later without signed overflows.
-    payload_type_red = 0;
-    payload_type_fec = 0;
-  }
-
-  VCMVideoProtection protection_method;
-  if (enable_nack) {
-    protection_method = enable_fec ? kProtectionNackFEC : kProtectionNack;
-  } else {
-    protection_method = kProtectionNone;
-  }
-
-  if (!sender_)
-    vcm_->SetVideoProtection(protection_method, true);
-
-  // Set NACK.
-  ProcessNACKRequest(enable_nack);
-
-  // Set FEC.
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    rtp_rtcp->SetGenericFECStatus(enable_fec,
-                                  static_cast<uint8_t>(payload_type_red),
-                                  static_cast<uint8_t>(payload_type_fec));
-  }
-}
-
-void ViEChannel::ProcessNACKRequest(const bool enable) {
-  if (enable) {
-    // Turn on NACK.
-    if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff)
-      return;
-    vie_receiver_.SetNackStatus(true, max_nack_reordering_threshold_);
-
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-      rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
-
-    if (!sender_) {
-      vcm_->RegisterPacketRequestCallback(this);
-      // Don't introduce errors when NACK is enabled.
-      vcm_->SetDecodeErrorMode(kNoErrors);
-    }
-  } else {
-    if (!sender_) {
-      vcm_->RegisterPacketRequestCallback(nullptr);
-      for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-        rtp_rtcp->SetStorePacketsStatus(false, 0);
-      // When NACK is off, allow decoding with errors. Otherwise, the video
-      // will freeze, and will only recover with a complete key frame.
-      vcm_->SetDecodeErrorMode(kWithErrors);
-    }
-    vie_receiver_.SetNackStatus(false, max_nack_reordering_threshold_);
-  }
-}
-
-int ViEChannel::GetRequiredNackListSize(int target_delay_ms) {
-  // The max size of the nack list should be large enough to accommodate the
-  // the number of packets (frames) resulting from the increased delay.
-  // Roughly estimating for ~40 packets per frame @ 30fps.
-  return target_delay_ms * 40 * 30 / 1000;
-}
-
-RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) const {
-  RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending());
-  RtpState rtp_state;
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    if (rtp_rtcp->GetRtpStateForSsrc(ssrc, &rtp_state))
-      return rtp_state;
-  }
-  LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc;
-  return rtp_state;
-}
-
-void ViEChannel::RegisterRtcpPacketTypeCounterObserver(
-    RtcpPacketTypeCounterObserver* observer) {
-  rtcp_packet_type_counter_observer_.Set(observer);
-}
-
-void ViEChannel::GetSendStreamDataCounters(
-    StreamDataCounters* rtp_counters,
-    StreamDataCounters* rtx_counters) const {
-  *rtp_counters = StreamDataCounters();
-  *rtx_counters = StreamDataCounters();
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    StreamDataCounters rtp_data;
-    StreamDataCounters rtx_data;
-    rtp_rtcp->GetSendStreamDataCounters(&rtp_data, &rtx_data);
-    rtp_counters->Add(rtp_data);
-    rtx_counters->Add(rtx_data);
-  }
-}
-
-void ViEChannel::RegisterSendSideDelayObserver(
-    SendSideDelayObserver* observer) {
-  send_side_delay_observer_.Set(observer);
-}
-
-void ViEChannel::RegisterSendBitrateObserver(
-    BitrateStatisticsObserver* observer) {
-  send_bitrate_observer_.Set(observer);
-}
-
-const std::vector<RtpRtcp*>& ViEChannel::rtp_rtcp() const {
-  return rtp_rtcp_modules_;
-}
-
-ViEReceiver* ViEChannel::vie_receiver() {
-  return &vie_receiver_;
-}
-
-VCMProtectionCallback* ViEChannel::vcm_protection_callback() {
-  return vcm_protection_callback_.get();
-}
-
-CallStatsObserver* ViEChannel::GetStatsObserver() {
-  return stats_observer_.get();
-}
-
-// Do not acquire the lock of |vcm_| in this function. Decode callback won't
-// necessarily be called from the decoding thread. The decoding thread may have
-// held the lock when calling VideoDecoder::Decode, Reset, or Release. Acquiring
-// the same lock in the path of decode callback can deadlock.
-int32_t ViEChannel::FrameToRender(VideoFrame& video_frame) {  // NOLINT
-  rtc::CritScope lock(&crit_);
-
-  if (pre_render_callback_)
-    pre_render_callback_->FrameCallback(&video_frame);
-
-  // TODO(pbos): Remove stream id argument.
-  incoming_video_stream_->RenderFrame(0xFFFFFFFF, video_frame);
-  return 0;
-}
-
-int32_t ViEChannel::ReceivedDecodedReferenceFrame(
-  const uint64_t picture_id) {
-  return rtp_rtcp_modules_[0]->SendRTCPReferencePictureSelection(picture_id);
-}
-
-void ViEChannel::OnIncomingPayloadType(int payload_type) {
-  rtc::CritScope lock(&crit_);
-  if (receive_stats_callback_)
-    receive_stats_callback_->OnIncomingPayloadType(payload_type);
-}
-
-void ViEChannel::OnDecoderImplementationName(const char* implementation_name) {
-  rtc::CritScope lock(&crit_);
-  if (receive_stats_callback_)
-    receive_stats_callback_->OnDecoderImplementationName(implementation_name);
-}
-
-void ViEChannel::OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) {
-  rtc::CritScope lock(&crit_);
-  if (receive_stats_callback_)
-    receive_stats_callback_->OnIncomingRate(frame_rate, bit_rate);
-}
-
-void ViEChannel::OnDiscardedPacketsUpdated(int discarded_packets) {
-  rtc::CritScope lock(&crit_);
-  if (receive_stats_callback_)
-    receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets);
-}
-
-void ViEChannel::OnFrameCountsUpdated(const FrameCounts& frame_counts) {
-  rtc::CritScope lock(&crit_);
-  receive_frame_counts_ = frame_counts;
-  if (receive_stats_callback_)
-    receive_stats_callback_->OnFrameCountsUpdated(frame_counts);
-}
-
-void ViEChannel::OnDecoderTiming(int decode_ms,
-                                 int max_decode_ms,
-                                 int current_delay_ms,
-                                 int target_delay_ms,
-                                 int jitter_buffer_ms,
-                                 int min_playout_delay_ms,
-                                 int render_delay_ms) {
-  rtc::CritScope lock(&crit_);
-  if (!receive_stats_callback_)
-    return;
-  receive_stats_callback_->OnDecoderTiming(
-      decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
-      jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt_ms_);
-}
-
-int32_t ViEChannel::RequestKeyFrame() {
-  return rtp_rtcp_modules_[0]->RequestKeyFrame();
-}
-
-int32_t ViEChannel::SliceLossIndicationRequest(
-  const uint64_t picture_id) {
-  return rtp_rtcp_modules_[0]->SendRTCPSliceLossIndication(
-      static_cast<uint8_t>(picture_id));
-}
-
-int32_t ViEChannel::ResendPackets(const uint16_t* sequence_numbers,
-                                  uint16_t length) {
-  return rtp_rtcp_modules_[0]->SendNACK(sequence_numbers, length);
-}
-
-void ViEChannel::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
-  if (!sender_)
-    vcm_->SetReceiveChannelParameters(max_rtt_ms);
-
-  rtc::CritScope lock(&crit_);
-  last_rtt_ms_ = avg_rtt_ms;
-}
-
-int ViEChannel::ProtectionRequest(const FecProtectionParams* delta_fec_params,
-                                  const FecProtectionParams* key_fec_params,
-                                  uint32_t* video_rate_bps,
-                                  uint32_t* nack_rate_bps,
-                                  uint32_t* fec_rate_bps) {
-  *video_rate_bps = 0;
-  *nack_rate_bps = 0;
-  *fec_rate_bps = 0;
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
-    uint32_t not_used = 0;
-    uint32_t module_video_rate = 0;
-    uint32_t module_fec_rate = 0;
-    uint32_t module_nack_rate = 0;
-    rtp_rtcp->SetFecParameters(delta_fec_params, key_fec_params);
-    rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
-                          &module_nack_rate);
-    *video_rate_bps += module_video_rate;
-    *nack_rate_bps += module_nack_rate;
-    *fec_rate_bps += module_fec_rate;
-  }
-  return 0;
-}
-
-std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules(
-    bool receiver_only,
-    ReceiveStatistics* receive_statistics,
-    Transport* outgoing_transport,
-    RtcpIntraFrameObserver* intra_frame_callback,
-    RtcpBandwidthObserver* bandwidth_callback,
-    TransportFeedbackObserver* transport_feedback_callback,
-    RtcpRttStats* rtt_stats,
-    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
-    RemoteBitrateEstimator* remote_bitrate_estimator,
-    RtpPacketSender* paced_sender,
-    TransportSequenceNumberAllocator* transport_sequence_number_allocator,
-    BitrateStatisticsObserver* send_bitrate_observer,
-    FrameCountObserver* send_frame_count_observer,
-    SendSideDelayObserver* send_side_delay_observer,
-    size_t num_modules) {
-  RTC_DCHECK_GT(num_modules, 0u);
-  RtpRtcp::Configuration configuration;
-  ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
-  configuration.audio = false;
-  configuration.receiver_only = receiver_only;
-  configuration.receive_statistics = receive_statistics;
-  configuration.outgoing_transport = outgoing_transport;
-  configuration.intra_frame_callback = intra_frame_callback;
-  configuration.rtt_stats = rtt_stats;
-  configuration.rtcp_packet_type_counter_observer =
-      rtcp_packet_type_counter_observer;
-  configuration.paced_sender = paced_sender;
-  configuration.transport_sequence_number_allocator =
-      transport_sequence_number_allocator;
-  configuration.send_bitrate_observer = send_bitrate_observer;
-  configuration.send_frame_count_observer = send_frame_count_observer;
-  configuration.send_side_delay_observer = send_side_delay_observer;
-  configuration.bandwidth_callback = bandwidth_callback;
-  configuration.transport_feedback_callback = transport_feedback_callback;
-
-  std::vector<RtpRtcp*> modules;
-  for (size_t i = 0; i < num_modules; ++i) {
-    RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
-    rtp_rtcp->SetSendingStatus(false);
-    rtp_rtcp->SetSendingMediaStatus(false);
-    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
-    modules.push_back(rtp_rtcp);
-    // Receive statistics and remote bitrate estimator should only be set for
-    // the primary (first) module.
-    configuration.receive_statistics = null_receive_statistics;
-    configuration.remote_bitrate_estimator = nullptr;
-  }
-  return modules;
-}
-
-void ViEChannel::RegisterPreRenderCallback(
-    I420FrameCallback* pre_render_callback) {
-  RTC_DCHECK(!sender_);
-  rtc::CritScope lock(&crit_);
-  pre_render_callback_ = pre_render_callback;
-}
-
-// TODO(pbos): Remove as soon as audio can handle a changing payload type
-// without this callback.
-int32_t ViEChannel::OnInitializeDecoder(
-    const int8_t payload_type,
-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-    const int frequency,
-    const size_t channels,
-    const uint32_t rate) {
-  RTC_NOTREACHED();
-  return 0;
-}
-
-void ViEChannel::OnIncomingSSRCChanged(const uint32_t ssrc) {
-  rtp_rtcp_modules_[0]->SetRemoteSSRC(ssrc);
-}
-
-void ViEChannel::OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) {}
-
-void ViEChannel::RegisterSendFrameCountObserver(
-    FrameCountObserver* observer) {
-  send_frame_count_observer_.Set(observer);
-}
-
-void ViEChannel::RegisterReceiveStatisticsProxy(
-    ReceiveStatisticsProxy* receive_statistics_proxy) {
-  rtc::CritScope lock(&crit_);
-  receive_stats_callback_ = receive_statistics_proxy;
-}
-
-void ViEChannel::SetIncomingVideoStream(
-    IncomingVideoStream* incoming_video_stream) {
-  rtc::CritScope lock(&crit_);
-  incoming_video_stream_ = incoming_video_stream;
-}
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/vie_channel.h b/chromium/third_party/webrtc/video/vie_channel.h
deleted file mode 100644
index 8a9d70e58ed..00000000000
--- a/chromium/third_party/webrtc/video/vie_channel.h
+++ /dev/null
@@ -1,311 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_VIE_CHANNEL_H_
-#define WEBRTC_VIDEO_VIE_CHANNEL_H_
-
-#include <list>
-#include <map>
-#include <memory>
-#include <vector>
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/include/video_coding_defines.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video/vie_receiver.h"
-#include "webrtc/video/vie_sync_module.h"
-
-namespace webrtc {
-
-class CallStatsObserver;
-class ChannelStatsObserver;
-class Config;
-class EncodedImageCallback;
-class I420FrameCallback;
-class IncomingVideoStream;
-class PacedSender;
-class PacketRouter;
-class PayloadRouter;
-class ProcessThread;
-class ReceiveStatisticsProxy;
-class RtcpRttStats;
-class ViEChannelProtectionCallback;
-class ViERTPObserver;
-class VideoCodingModule;
-class VideoRenderCallback;
-class VoEVideoSync;
-
-enum StreamType {
-  kViEStreamTypeNormal = 0,  // Normal media stream
-  kViEStreamTypeRtx = 1      // Retransmission media stream
-};
-
-class ViEChannel : public VCMFrameTypeCallback,
-                   public VCMReceiveCallback,
-                   public VCMReceiveStatisticsCallback,
-                   public VCMDecoderTimingCallback,
-                   public VCMPacketRequestCallback,
-                   public RtpFeedback {
- public:
-  friend class ChannelStatsObserver;
-  friend class ViEChannelProtectionCallback;
-
-  ViEChannel(Transport* transport,
-             ProcessThread* module_process_thread,
-             PayloadRouter* send_payload_router,
-             VideoCodingModule* vcm,
-             RtcpIntraFrameObserver* intra_frame_observer,
-             RtcpBandwidthObserver* bandwidth_observer,
-             TransportFeedbackObserver* transport_feedback_observer,
-             RemoteBitrateEstimator* remote_bitrate_estimator,
-             RtcpRttStats* rtt_stats,
-             PacedSender* paced_sender,
-             PacketRouter* packet_router,
-             size_t max_rtp_streams,
-             bool sender);
-  ~ViEChannel();
-
-  int32_t Init();
-
-  void SetProtectionMode(bool enable_nack,
-                         bool enable_fec,
-                         int payload_type_red,
-                         int payload_type_fec);
-
-  RtpState GetRtpStateForSsrc(uint32_t ssrc) const;
-
-  // Gets send statistics for the rtp and rtx stream.
-  void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
-                                 StreamDataCounters* rtx_counters) const;
-
-  void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
-
-  // Called on any new send bitrate estimate.
-  void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer);
-
-  // Implements RtpFeedback.
-  int32_t OnInitializeDecoder(const int8_t payload_type,
-                              const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-                              const int frequency,
-                              const size_t channels,
-                              const uint32_t rate) override;
-  void OnIncomingSSRCChanged(const uint32_t ssrc) override;
-  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override;
-
-  // Gets the modules used by the channel.
-  const std::vector<RtpRtcp*>& rtp_rtcp() const;
-  ViEReceiver* vie_receiver();
-  VCMProtectionCallback* vcm_protection_callback();
-
-
-  CallStatsObserver* GetStatsObserver();
-
-  // Implements VCMReceiveCallback.
-  virtual int32_t FrameToRender(VideoFrame& video_frame);  // NOLINT
-
-  // Implements VCMReceiveCallback.
-  virtual int32_t ReceivedDecodedReferenceFrame(
-      const uint64_t picture_id);
-
-  // Implements VCMReceiveCallback.
-  void OnIncomingPayloadType(int payload_type) override;
-  void OnDecoderImplementationName(const char* implementation_name) override;
-
-  // Implements VCMReceiveStatisticsCallback.
-  void OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) override;
-  void OnDiscardedPacketsUpdated(int discarded_packets) override;
-  void OnFrameCountsUpdated(const FrameCounts& frame_counts) override;
-
-  // Implements VCMDecoderTimingCallback.
-  virtual void OnDecoderTiming(int decode_ms,
-                               int max_decode_ms,
-                               int current_delay_ms,
-                               int target_delay_ms,
-                               int jitter_buffer_ms,
-                               int min_playout_delay_ms,
-                               int render_delay_ms);
-
-  // Implements FrameTypeCallback.
-  virtual int32_t RequestKeyFrame();
-
-  // Implements FrameTypeCallback.
-  virtual int32_t SliceLossIndicationRequest(
-      const uint64_t picture_id);
-
-  // Implements VideoPacketRequestCallback.
-  int32_t ResendPackets(const uint16_t* sequence_numbers,
-                        uint16_t length) override;
-
-  void RegisterPreRenderCallback(I420FrameCallback* pre_render_callback);
-
-  void RegisterSendFrameCountObserver(FrameCountObserver* observer);
-  void RegisterRtcpPacketTypeCounterObserver(
-      RtcpPacketTypeCounterObserver* observer);
-  void RegisterReceiveStatisticsProxy(
-      ReceiveStatisticsProxy* receive_statistics_proxy);
-  void SetIncomingVideoStream(IncomingVideoStream* incoming_video_stream);
-
- protected:
-  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms);
-
-  int ProtectionRequest(const FecProtectionParams* delta_fec_params,
-                        const FecProtectionParams* key_fec_params,
-                        uint32_t* sent_video_rate_bps,
-                        uint32_t* sent_nack_rate_bps,
-                        uint32_t* sent_fec_rate_bps);
-
- private:
-  static std::vector<RtpRtcp*> CreateRtpRtcpModules(
-      bool receiver_only,
-      ReceiveStatistics* receive_statistics,
-      Transport* outgoing_transport,
-      RtcpIntraFrameObserver* intra_frame_callback,
-      RtcpBandwidthObserver* bandwidth_callback,
-      TransportFeedbackObserver* transport_feedback_callback,
-      RtcpRttStats* rtt_stats,
-      RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
-      RemoteBitrateEstimator* remote_bitrate_estimator,
-      RtpPacketSender* paced_sender,
-      TransportSequenceNumberAllocator* transport_sequence_number_allocator,
-      BitrateStatisticsObserver* send_bitrate_observer,
-      FrameCountObserver* send_frame_count_observer,
-      SendSideDelayObserver* send_side_delay_observer,
-      size_t num_modules);
-
-  // Assumed to be protected.
-  void StartDecodeThread();
-  void StopDecodeThread();
-
-  void ProcessNACKRequest(const bool enable);
-  // Compute NACK list parameters for the buffering mode.
-  int GetRequiredNackListSize(int target_delay_ms);
-
-  // ViEChannel exposes methods that allow to modify observers and callbacks
-  // to be modified. Such an API-style is cumbersome to implement and maintain
-  // at all the levels when comparing to only setting them at construction. As
-  // so this class instantiates its children with a wrapper that can be modified
-  // at a later time.
-  template <class T>
-  class RegisterableCallback : public T {
-   public:
-    RegisterableCallback() : callback_(nullptr) {}
-
-    void Set(T* callback) {
-      rtc::CritScope lock(&critsect_);
-      callback_ = callback;
-    }
-
-   protected:
-    // Note: this should be implemented with a RW-lock to allow simultaneous
-    // calls into the callback. However that doesn't seem to be needed for the
-    // current type of callbacks covered by this class.
-    rtc::CriticalSection critsect_;
-    T* callback_ GUARDED_BY(critsect_);
-
-   private:
-    RTC_DISALLOW_COPY_AND_ASSIGN(RegisterableCallback);
-  };
-
-  class RegisterableBitrateStatisticsObserver:
-    public RegisterableCallback<BitrateStatisticsObserver> {
-    virtual void Notify(const BitrateStatistics& total_stats,
-                        const BitrateStatistics& retransmit_stats,
-                        uint32_t ssrc) {
-      rtc::CritScope lock(&critsect_);
-      if (callback_)
-        callback_->Notify(total_stats, retransmit_stats, ssrc);
-    }
-  } send_bitrate_observer_;
-
-  class RegisterableFrameCountObserver
-      : public RegisterableCallback<FrameCountObserver> {
-   public:
-    virtual void FrameCountUpdated(const FrameCounts& frame_counts,
-                                   uint32_t ssrc) {
-      rtc::CritScope lock(&critsect_);
-      if (callback_)
-        callback_->FrameCountUpdated(frame_counts, ssrc);
-    }
-
-   private:
-  } send_frame_count_observer_;
-
-  class RegisterableSendSideDelayObserver :
-      public RegisterableCallback<SendSideDelayObserver> {
-    void SendSideDelayUpdated(int avg_delay_ms,
-                              int max_delay_ms,
-                              uint32_t ssrc) override {
-      rtc::CritScope lock(&critsect_);
-      if (callback_)
-        callback_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
-    }
-  } send_side_delay_observer_;
-
-  class RegisterableRtcpPacketTypeCounterObserver
-      : public RegisterableCallback<RtcpPacketTypeCounterObserver> {
-   public:
-    void RtcpPacketTypesCounterUpdated(
-        uint32_t ssrc,
-        const RtcpPacketTypeCounter& packet_counter) override {
-      rtc::CritScope lock(&critsect_);
-      if (callback_)
-        callback_->RtcpPacketTypesCounterUpdated(ssrc, packet_counter);
-    }
-
-   private:
-  } rtcp_packet_type_counter_observer_;
-
-  const bool sender_;
-
-  ProcessThread* const module_process_thread_;
-  PayloadRouter* const send_payload_router_;
-
-  // Used for all registered callbacks except rendering.
-  rtc::CriticalSection crit_;
-
-  // Owned modules/classes.
-  std::unique_ptr<ViEChannelProtectionCallback> vcm_protection_callback_;
-
-  VideoCodingModule* const vcm_;
-  ViEReceiver vie_receiver_;
-
-  // Helper to report call statistics.
-  std::unique_ptr<ChannelStatsObserver> stats_observer_;
-
-  // Not owned.
-  ReceiveStatisticsProxy* receive_stats_callback_ GUARDED_BY(crit_);
-  FrameCounts receive_frame_counts_ GUARDED_BY(crit_);
-  IncomingVideoStream* incoming_video_stream_ GUARDED_BY(crit_);
-  RtcpIntraFrameObserver* const intra_frame_observer_;
-  RtcpRttStats* const rtt_stats_;
-  PacedSender* const paced_sender_;
-  PacketRouter* const packet_router_;
-
-  const std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
-  TransportFeedbackObserver* const transport_feedback_observer_;
-
-  int max_nack_reordering_threshold_;
-  I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_);
-
-  int64_t last_rtt_ms_ GUARDED_BY(crit_);
-
-  // RtpRtcp modules, declared last as they use other members on construction.
-  const std::vector<RtpRtcp*> rtp_rtcp_modules_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_VIE_CHANNEL_H_
diff --git a/chromium/third_party/webrtc/video/vie_encoder.cc b/chromium/third_party/webrtc/video/vie_encoder.cc
index 24b8ce3cd1c..2871c4045c9 100644
--- a/chromium/third_party/webrtc/video/vie_encoder.cc
+++ b/chromium/third_party/webrtc/video/vie_encoder.cc
@@ -17,132 +17,50 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/common_video/include/video_image.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/pacing/paced_sender.h"
-#include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
-#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video/overuse_frame_detector.h"
-#include "webrtc/video/payload_router.h"
 #include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video_frame.h"
 
 namespace webrtc {
 
 static const float kStopPaddingThresholdMs = 2000;
-static const int kMinKeyFrameRequestIntervalMs = 300;
-
-std::vector<uint32_t> AllocateStreamBitrates(
-    uint32_t total_bitrate,
-    const SimulcastStream* stream_configs,
-    size_t number_of_streams) {
-  if (number_of_streams == 0) {
-    std::vector<uint32_t> stream_bitrates(1, 0);
-    stream_bitrates[0] = total_bitrate;
-    return stream_bitrates;
-  }
-  std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
-  uint32_t bitrate_remainder = total_bitrate;
-  for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
-    if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
-      stream_bitrates[i] = bitrate_remainder;
-    } else {
-      stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
-    }
-    bitrate_remainder -= stream_bitrates[i];
-  }
-  return stream_bitrates;
-}
-
-class QMVideoSettingsCallback : public VCMQMSettingsCallback {
- public:
-  explicit QMVideoSettingsCallback(VideoProcessing* vpm);
-
-  ~QMVideoSettingsCallback();
-
-  // Update VPM with QM (quality modes: frame size & frame rate) settings.
-  int32_t SetVideoQMSettings(const uint32_t frame_rate,
-                             const uint32_t width,
-                             const uint32_t height);
-
-  // Update target frame rate.
-  void SetTargetFramerate(int frame_rate);
-
- private:
-  VideoProcessing* vp_;
-};
 
 ViEEncoder::ViEEncoder(uint32_t number_of_cores,
-                       const std::vector<uint32_t>& ssrcs,
                        ProcessThread* module_process_thread,
                        SendStatisticsProxy* stats_proxy,
-                       I420FrameCallback* pre_encode_callback,
-                       OveruseFrameDetector* overuse_detector,
-                       PacedSender* pacer,
-                       PayloadRouter* payload_router)
+                       OveruseFrameDetector* overuse_detector)
     : number_of_cores_(number_of_cores),
-      ssrcs_(ssrcs),
       vp_(VideoProcessing::Create()),
-      qm_callback_(new QMVideoSettingsCallback(vp_.get())),
-      vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(),
-                                     this,
-                                     qm_callback_.get())),
+      video_sender_(Clock::GetRealTimeClock(), this, this, this),
       stats_proxy_(stats_proxy),
-      pre_encode_callback_(pre_encode_callback),
       overuse_detector_(overuse_detector),
-      pacer_(pacer),
-      send_payload_router_(payload_router),
       time_of_last_frame_activity_ms_(0),
       encoder_config_(),
       min_transmit_bitrate_bps_(0),
       last_observed_bitrate_bps_(0),
-      network_is_transmitting_(true),
-      encoder_paused_(false),
+      encoder_paused_(true),
       encoder_paused_and_dropped_frame_(false),
-      time_last_intra_request_ms_(ssrcs.size(), -1),
       module_process_thread_(module_process_thread),
       has_received_sli_(false),
       picture_id_sli_(0),
       has_received_rpsi_(false),
       picture_id_rpsi_(0),
       video_suspended_(false) {
-  module_process_thread_->RegisterModule(vcm_.get());
-}
-
-bool ViEEncoder::Init() {
+  module_process_thread_->RegisterModule(&video_sender_);
   vp_->EnableTemporalDecimation(true);
-
-  // Enable/disable content analysis: off by default for now.
-  vp_->EnableContentAnalysis(false);
-
-  if (vcm_->RegisterTransportCallback(this) != 0) {
-    return false;
-  }
-  if (vcm_->RegisterSendStatisticsCallback(this) != 0) {
-    return false;
-  }
-  return true;
 }
 
-VideoCodingModule* ViEEncoder::vcm() const {
-  return vcm_.get();
+vcm::VideoSender* ViEEncoder::video_sender() {
+  return &video_sender_;
 }
 
 ViEEncoder::~ViEEncoder() {
-  module_process_thread_->DeRegisterModule(vcm_.get());
-}
-
-void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
-  {
-    rtc::CritScope lock(&data_cs_);
-    network_is_transmitting_ = is_transmitting;
-  }
+  module_process_thread_->DeRegisterModule(&video_sender_);
 }
 
 void ViEEncoder::Pause() {
@@ -150,7 +68,7 @@ void ViEEncoder::Pause() {
   encoder_paused_ = true;
 }
 
-void ViEEncoder::Restart() {
+void ViEEncoder::Start() {
   rtc::CritScope lock(&data_cs_);
   encoder_paused_ = false;
 }
@@ -158,22 +76,18 @@ void ViEEncoder::Restart() {
 int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
                                             uint8_t pl_type,
                                             bool internal_source) {
-  if (vcm_->RegisterExternalEncoder(encoder, pl_type, internal_source) !=
-      VCM_OK) {
-    return -1;
-  }
+  video_sender_.RegisterExternalEncoder(encoder, pl_type, internal_source);
   return 0;
 }
 
 int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
-  if (vcm_->RegisterExternalEncoder(nullptr, pl_type) != VCM_OK) {
-    return -1;
-  }
+  video_sender_.RegisterExternalEncoder(nullptr, pl_type, false);
   return 0;
 }
 void ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec,
-                            int min_transmit_bitrate_bps) {
-  RTC_DCHECK(send_payload_router_);
+                            int min_transmit_bitrate_bps,
+                            size_t max_data_payload_length,
+                            EncodedImageCallback* sink) {
   // Setting target width and height for VPM.
   RTC_CHECK_EQ(VPM_OK,
                vp_->SetTargetResolution(video_codec.width, video_codec.height,
@@ -184,12 +98,14 @@ void ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec,
   {
     rtc::CritScope lock(&data_cs_);
     encoder_config_ = video_codec;
-    encoder_paused_ = true;
     min_transmit_bitrate_bps_ = min_transmit_bitrate_bps;
   }
+  {
+    rtc::CritScope lock(&sink_cs_);
+    sink_ = sink;
+  }
 
-  size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
-  bool success = vcm_->RegisterSendCodec(
+  bool success = video_sender_.RegisterSendCodec(
                      &video_codec, number_of_cores_,
                      static_cast<uint32_t>(max_data_payload_length)) == VCM_OK;
   if (!success) {
@@ -197,17 +113,7 @@ void ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec,
     RTC_DCHECK(success);
   }
 
-  send_payload_router_->SetSendingRtpModules(
-      video_codec.numberOfSimulcastStreams);
-
-  // Restart the media flow
-  Restart();
   if (stats_proxy_) {
-    // Clear stats for disabled layers.
-    for (size_t i = video_codec.numberOfSimulcastStreams; i < ssrcs_.size();
-         ++i) {
-      stats_proxy_->OnInactiveSsrc(ssrcs_[i]);
-    }
     VideoEncoderConfig::ContentType content_type =
         VideoEncoderConfig::ContentType::kRealtimeVideo;
     switch (video_codec.mode) {
@@ -242,7 +148,7 @@ int ViEEncoder::GetPaddingNeededBps() const {
     send_codec = encoder_config_;
   }
 
-  bool video_is_suspended = vcm_->VideoSuspended();
+  bool video_is_suspended = video_sender_.VideoSuspended();
 
   // Find the max amount of padding we can allow ourselves to send at this
   // point, based on which streams are currently active and what our current
@@ -267,7 +173,7 @@ int ViEEncoder::GetPaddingNeededBps() const {
 
   // The amount of padding should decay to zero if no frames are being
   // captured/encoded unless a min-transmit bitrate is used.
-  int64_t now_ms = TickTime::MillisecondTimestamp();
+  int64_t now_ms = rtc::TimeMillis();
   if (now_ms - time_of_last_frame_activity_ms > kStopPaddingThresholdMs)
     pad_up_to_bitrate_bps = 0;
 
@@ -285,14 +191,9 @@ int ViEEncoder::GetPaddingNeededBps() const {
 bool ViEEncoder::EncoderPaused() const {
   // Pause video if paused by caller or as long as the network is down or the
   // pacer queue has grown too large in buffered mode.
-  if (encoder_paused_) {
-    return true;
-  }
-  if (pacer_->ExpectedQueueTimeMs() > PacedSender::kMaxQueueLengthMs) {
-    // Too much data in pacer queue, drop frame.
-    return true;
-  }
-  return !network_is_transmitting_;
+  // If the pacer queue has grown to large or the network is down,
+  // last_observed_bitrate_bps_ will be 0.
+  return encoder_paused_ || video_suspended_ || last_observed_bitrate_bps_ == 0;
 }
 
 void ViEEncoder::TraceFrameDropStart() {
@@ -313,15 +214,10 @@ void ViEEncoder::TraceFrameDropEnd() {
 }
 
 void ViEEncoder::EncodeVideoFrame(const VideoFrame& video_frame) {
-  if (!send_payload_router_->active()) {
-    // We've paused or we have no channels attached, don't waste resources on
-    // encoding.
-    return;
-  }
   VideoCodecType codec_type;
   {
     rtc::CritScope lock(&data_cs_);
-    time_of_last_frame_activity_ms_ = TickTime::MillisecondTimestamp();
+    time_of_last_frame_activity_ms_ = rtc::TimeMillis();
     if (EncoderPaused()) {
       TraceFrameDropStart();
       return;
@@ -334,7 +230,7 @@ void ViEEncoder::EncodeVideoFrame(const VideoFrame& video_frame) {
                           "Encode");
   const VideoFrame* frame_to_send = &video_frame;
   // TODO(wuchengli): support texture frames.
-  if (!video_frame.native_handle()) {
+  if (!video_frame.video_frame_buffer()->native_handle()) {
     // Pass frame via preprocessor.
     frame_to_send = vp_->PreprocessFrame(video_frame);
     if (!frame_to_send) {
@@ -343,15 +239,6 @@ void ViEEncoder::EncodeVideoFrame(const VideoFrame& video_frame) {
     }
   }
 
-  // If we haven't resampled the frame and we have a FrameCallback, we need to
-  // make a deep copy of |video_frame|.
-  VideoFrame copied_frame;
-  if (pre_encode_callback_) {
-    copied_frame.CopyFrame(*frame_to_send);
-    pre_encode_callback_->FrameCallback(&copied_frame);
-    frame_to_send = &copied_frame;
-  }
-
   if (codec_type == webrtc::kVideoCodecVP8) {
     webrtc::CodecSpecificInfo codec_specific_info;
     codec_specific_info.codecType = webrtc::kVideoCodecVP8;
@@ -369,26 +256,14 @@ void ViEEncoder::EncodeVideoFrame(const VideoFrame& video_frame) {
       has_received_rpsi_ = false;
     }
 
-    vcm_->AddVideoFrame(*frame_to_send, vp_->GetContentMetrics(),
-                        &codec_specific_info);
+    video_sender_.AddVideoFrame(*frame_to_send, &codec_specific_info);
     return;
   }
-  vcm_->AddVideoFrame(*frame_to_send);
+  video_sender_.AddVideoFrame(*frame_to_send, nullptr);
 }
 
 void ViEEncoder::SendKeyFrame() {
-  vcm_->IntraFrameRequest(0);
-}
-
-uint32_t ViEEncoder::LastObservedBitrateBps() const {
-  rtc::CritScope lock(&data_cs_);
-  return last_observed_bitrate_bps_;
-}
-
-int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
-  if (vcm_->Bitrate(bitrate) != 0)
-    return -1;
-  return 0;
+  video_sender_.IntraFrameRequest(0);
 }
 
 void ViEEncoder::SetProtectionMethod(bool nack, bool fec) {
@@ -400,7 +275,7 @@ void ViEEncoder::SetProtectionMethod(bool nack, bool fec) {
   } else {
     protection_mode = nack ? kProtectionNack : kProtectionNone;
   }
-  vcm_->SetVideoProtection(protection_mode, true);
+  video_sender_.SetVideoProtection(protection_mode);
 }
 
 void ViEEncoder::OnSetRates(uint32_t bitrate_bps, int framerate) {
@@ -408,132 +283,76 @@ void ViEEncoder::OnSetRates(uint32_t bitrate_bps, int framerate) {
     stats_proxy_->OnSetRates(bitrate_bps, framerate);
 }
 
-int32_t ViEEncoder::SendData(const uint8_t payload_type,
-                             const EncodedImage& encoded_image,
-                             const RTPFragmentationHeader* fragmentation_header,
-                             const RTPVideoHeader* rtp_video_hdr) {
-  RTC_DCHECK(send_payload_router_);
-
+int32_t ViEEncoder::Encoded(const EncodedImage& encoded_image,
+                            const CodecSpecificInfo* codec_specific_info,
+                            const RTPFragmentationHeader* fragmentation) {
   {
     rtc::CritScope lock(&data_cs_);
-    time_of_last_frame_activity_ms_ = TickTime::MillisecondTimestamp();
+    time_of_last_frame_activity_ms_ = rtc::TimeMillis();
+  }
+  if (stats_proxy_) {
+    stats_proxy_->OnSendEncodedImage(encoded_image, codec_specific_info);
   }
 
-  if (stats_proxy_)
-    stats_proxy_->OnSendEncodedImage(encoded_image, rtp_video_hdr);
+  int success = 0;
+  {
+    rtc::CritScope lock(&sink_cs_);
+    success = sink_->Encoded(encoded_image, codec_specific_info, fragmentation);
+  }
 
-  bool success = send_payload_router_->RoutePayload(
-      encoded_image._frameType, payload_type, encoded_image._timeStamp,
-      encoded_image.capture_time_ms_, encoded_image._buffer,
-      encoded_image._length, fragmentation_header, rtp_video_hdr);
   overuse_detector_->FrameSent(encoded_image._timeStamp);
-  return success ? 0 : -1;
-}
-
-void ViEEncoder::OnEncoderImplementationName(
-    const char* implementation_name) {
-  if (stats_proxy_)
-    stats_proxy_->OnEncoderImplementationName(implementation_name);
+  return success;
 }
 
-int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
-                                   const uint32_t frame_rate) {
+void ViEEncoder::SendStatistics(uint32_t bit_rate,
+                                uint32_t frame_rate,
+                                const std::string& encoder_name) {
   if (stats_proxy_)
-    stats_proxy_->OnOutgoingRate(frame_rate, bit_rate);
-  return 0;
+    stats_proxy_->OnEncoderStatsUpdate(frame_rate, bit_rate, encoder_name);
 }
 
-void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
-                               uint8_t picture_id) {
+void ViEEncoder::OnReceivedSLI(uint8_t picture_id) {
   rtc::CritScope lock(&data_cs_);
   picture_id_sli_ = picture_id;
   has_received_sli_ = true;
 }
 
-void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
-                                uint64_t picture_id) {
+void ViEEncoder::OnReceivedRPSI(uint64_t picture_id) {
   rtc::CritScope lock(&data_cs_);
   picture_id_rpsi_ = picture_id;
   has_received_rpsi_ = true;
 }
 
-void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
+void ViEEncoder::OnReceivedIntraFrameRequest(size_t stream_index) {
   // Key frame request from remote side, signal to VCM.
   TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
-
-  for (size_t i = 0; i < ssrcs_.size(); ++i) {
-    if (ssrcs_[i] != ssrc)
-      continue;
-    int64_t now_ms = TickTime::MillisecondTimestamp();
-    {
-      rtc::CritScope lock(&data_cs_);
-      if (time_last_intra_request_ms_[i] + kMinKeyFrameRequestIntervalMs >
-          now_ms) {
-        return;
-      }
-      time_last_intra_request_ms_[i] = now_ms;
-    }
-    vcm_->IntraFrameRequest(static_cast<int>(i));
-    return;
-  }
-  RTC_NOTREACHED() << "Should not receive keyframe requests on unknown SSRCs.";
+  video_sender_.IntraFrameRequest(stream_index);
 }
 
 void ViEEncoder::OnBitrateUpdated(uint32_t bitrate_bps,
                                   uint8_t fraction_lost,
                                   int64_t round_trip_time_ms) {
-  LOG(LS_VERBOSE) << "OnBitrateUpdated, bitrate" << bitrate_bps
+  LOG(LS_VERBOSE) << "OnBitrateUpdated, bitrate " << bitrate_bps
                   << " packet loss " << static_cast<int>(fraction_lost)
                   << " rtt " << round_trip_time_ms;
-  RTC_DCHECK(send_payload_router_);
-  vcm_->SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
-  bool video_is_suspended = vcm_->VideoSuspended();
+  video_sender_.SetChannelParameters(bitrate_bps, fraction_lost,
+                                     round_trip_time_ms);
+  bool video_is_suspended = video_sender_.VideoSuspended();
   bool video_suspension_changed;
-  VideoCodec send_codec;
   {
     rtc::CritScope lock(&data_cs_);
     last_observed_bitrate_bps_ = bitrate_bps;
     video_suspension_changed = video_suspended_ != video_is_suspended;
     video_suspended_ = video_is_suspended;
-    send_codec = encoder_config_;
   }
 
-  SimulcastStream* stream_configs = send_codec.simulcastStream;
-  // Allocate the bandwidth between the streams.
-  std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
-      bitrate_bps, stream_configs, send_codec.numberOfSimulcastStreams);
-  send_payload_router_->SetTargetSendBitrates(stream_bitrates);
-
   if (!video_suspension_changed)
     return;
   // Video suspend-state changed, inform codec observer.
-  LOG(LS_INFO) << "Video suspend state changed " << video_is_suspended
-               << " for ssrc " << ssrcs_[0];
+  LOG(LS_INFO) << "Video suspend state changed " << video_is_suspended;
+
   if (stats_proxy_)
     stats_proxy_->OnSuspendChange(video_is_suspended);
 }
 
-void ViEEncoder::RegisterPostEncodeImageCallback(
-      EncodedImageCallback* post_encode_callback) {
-  vcm_->RegisterPostEncodeImageCallback(post_encode_callback);
-}
-
-QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessing* vpm)
-    : vp_(vpm) {
-}
-
-QMVideoSettingsCallback::~QMVideoSettingsCallback() {
-}
-
-int32_t QMVideoSettingsCallback::SetVideoQMSettings(
-    const uint32_t frame_rate,
-    const uint32_t width,
-    const uint32_t height) {
-  return vp_->SetTargetResolution(width, height, frame_rate);
-}
-
-void QMVideoSettingsCallback::SetTargetFramerate(int frame_rate) {
-  vp_->SetTargetFramerate(frame_rate);
-}
-
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/vie_encoder.h b/chromium/third_party/webrtc/video/vie_encoder.h
index ce1e508eba9..556ed449955 100644
--- a/chromium/third_party/webrtc/video/vie_encoder.h
+++ b/chromium/third_party/webrtc/video/vie_encoder.h
@@ -12,15 +12,18 @@
 #define WEBRTC_VIDEO_VIE_ENCODER_H_
 
 #include <memory>
+#include <string>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/video_encoder.h"
+#include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/modules/video_processing/include/video_processing.h"
 #include "webrtc/typedefs.h"
 
@@ -30,84 +33,79 @@ class Config;
 class EncodedImageCallback;
 class OveruseFrameDetector;
 class PacedSender;
-class PayloadRouter;
 class ProcessThread;
-class QMVideoSettingsCallback;
 class SendStatisticsProxy;
 class ViEBitrateObserver;
 class ViEEffectFilter;
-class VideoCodingModule;
 class VideoEncoder;
 
+// VieEncoder represent a video encoder that accepts raw video frames as input
+// and produces an encoded bit stream.
+// Usage:
+// 1. Instantiate
+// 2. Call Init
+// 3. Call RegisterExternalEncoder if available.
+// 4. Call SetEncoder with the codec settings and the object that shall receive
+//    the encoded bit stream.
+// 5. Call Start.
+// 6. For each available raw video frame call EncodeVideoFrame.
 class ViEEncoder : public VideoEncoderRateObserver,
-                   public VCMPacketizationCallback,
+                   public EncodedImageCallback,
                    public VCMSendStatisticsCallback {
  public:
   friend class ViEBitrateObserver;
 
   ViEEncoder(uint32_t number_of_cores,
-             const std::vector<uint32_t>& ssrcs,
              ProcessThread* module_process_thread,
              SendStatisticsProxy* stats_proxy,
-             I420FrameCallback* pre_encode_callback,
-             OveruseFrameDetector* overuse_detector,
-             PacedSender* pacer,
-             PayloadRouter* payload_router);
+             OveruseFrameDetector* overuse_detector);
   ~ViEEncoder();
 
-  bool Init();
-
-  VideoCodingModule* vcm() const;
-
-  void SetNetworkTransmissionState(bool is_transmitting);
+  vcm::VideoSender* video_sender();
 
   // Returns the id of the owning channel.
   int Owner() const;
 
+  void Start();
   // Drops incoming packets before they get to the encoder.
   void Pause();
-  void Restart();
 
   // Codec settings.
   int32_t RegisterExternalEncoder(VideoEncoder* encoder,
                                   uint8_t pl_type,
                                   bool internal_source);
   int32_t DeRegisterExternalEncoder(uint8_t pl_type);
-  void SetEncoder(const VideoCodec& video_codec, int min_transmit_bitrate_bps);
+  void SetEncoder(const VideoCodec& video_codec,
+                  int min_transmit_bitrate_bps,
+                  size_t max_data_payload_length,
+                  EncodedImageCallback* sink);
 
   void EncodeVideoFrame(const VideoFrame& video_frame);
   void SendKeyFrame();
 
   uint32_t LastObservedBitrateBps() const;
-  int CodecTargetBitrate(uint32_t* bitrate) const;
   // Loss protection. Must be called before SetEncoder() to have max packet size
   // updated according to protection.
-  // TODO(pbos): Set protection method on construction or extract vcm_ outside
-  // this class and set it on construction there.
+  // TODO(pbos): Set protection method on construction.
   void SetProtectionMethod(bool nack, bool fec);
 
   // Implements VideoEncoderRateObserver.
   void OnSetRates(uint32_t bitrate_bps, int framerate) override;
 
-  // Implements VCMPacketizationCallback.
-  int32_t SendData(uint8_t payload_type,
-                   const EncodedImage& encoded_image,
-                   const RTPFragmentationHeader* fragmentation_header,
-                   const RTPVideoHeader* rtp_video_hdr) override;
-  void OnEncoderImplementationName(const char* implementation_name) override;
+  // Implements EncodedImageCallback.
+  int32_t Encoded(const EncodedImage& encoded_image,
+                  const CodecSpecificInfo* codec_specific_info,
+                  const RTPFragmentationHeader* fragmentation) override;
 
   // Implements VideoSendStatisticsCallback.
-  int32_t SendStatistics(const uint32_t bit_rate,
-                         const uint32_t frame_rate) override;
+  void SendStatistics(uint32_t bit_rate,
+                      uint32_t frame_rate,
+                      const std::string& encoder_name) override;
 
   // virtual to test EncoderStateFeedback with mocks.
-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc);
-  virtual void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id);
-  virtual void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id);
-
-  // New-style callbacks, used by VideoSendStream.
-  void RegisterPostEncodeImageCallback(
-        EncodedImageCallback* post_encode_callback);
+  virtual void OnReceivedIntraFrameRequest(size_t stream_index);
+  virtual void OnReceivedSLI(uint8_t picture_id);
+  virtual void OnReceivedRPSI(uint64_t picture_id);
 
   int GetPaddingNeededBps() const;
 
@@ -121,19 +119,14 @@ class ViEEncoder : public VideoEncoderRateObserver,
   void TraceFrameDropEnd() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
 
   const uint32_t number_of_cores_;
-  const std::vector<uint32_t> ssrcs_;
 
   const std::unique_ptr<VideoProcessing> vp_;
-  const std::unique_ptr<QMVideoSettingsCallback> qm_callback_;
-  const std::unique_ptr<VideoCodingModule> vcm_;
+  vcm::VideoSender video_sender_;
 
   rtc::CriticalSection data_cs_;
 
   SendStatisticsProxy* const stats_proxy_;
-  I420FrameCallback* const pre_encode_callback_;
   OveruseFrameDetector* const overuse_detector_;
-  PacedSender* const pacer_;
-  PayloadRouter* const send_payload_router_;
 
   // The time we last received an input frame or encoded frame. This is used to
   // track when video is stopped long enough that we also want to stop sending
@@ -142,10 +135,11 @@ class ViEEncoder : public VideoEncoderRateObserver,
   VideoCodec encoder_config_ GUARDED_BY(data_cs_);
   int min_transmit_bitrate_bps_ GUARDED_BY(data_cs_);
   uint32_t last_observed_bitrate_bps_ GUARDED_BY(data_cs_);
-  bool network_is_transmitting_ GUARDED_BY(data_cs_);
   bool encoder_paused_ GUARDED_BY(data_cs_);
   bool encoder_paused_and_dropped_frame_ GUARDED_BY(data_cs_);
-  std::vector<int64_t> time_last_intra_request_ms_ GUARDED_BY(data_cs_);
+
+  rtc::CriticalSection sink_cs_;
+  EncodedImageCallback* sink_ GUARDED_BY(sink_cs_);
 
   ProcessThread* module_process_thread_;
 
diff --git a/chromium/third_party/webrtc/video/vie_receiver.cc b/chromium/third_party/webrtc/video/vie_receiver.cc
deleted file mode 100644
index e5ec167fa7d..00000000000
--- a/chromium/third_party/webrtc/video/vie_receiver.cc
+++ /dev/null
@@ -1,389 +0,0 @@
-/*
- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/video/vie_receiver.h"
-
-#include <vector>
-
-#include "webrtc/base/logging.h"
-#include "webrtc/config.h"
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/include/video_coding.h"
-#include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
-#include "webrtc/system_wrappers/include/trace.h"
-
-namespace webrtc {
-
-static const int kPacketLogIntervalMs = 10000;
-
-ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm,
-                         RemoteBitrateEstimator* remote_bitrate_estimator,
-                         RtpFeedback* rtp_feedback)
-    : clock_(Clock::GetRealTimeClock()),
-      vcm_(module_vcm),
-      remote_bitrate_estimator_(remote_bitrate_estimator),
-      ntp_estimator_(clock_),
-      rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
-      rtp_header_parser_(RtpHeaderParser::Create()),
-      rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
-                                                     this,
-                                                     rtp_feedback,
-                                                     &rtp_payload_registry_)),
-      rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
-      fec_receiver_(FecReceiver::Create(this)),
-      receiving_(false),
-      restored_packet_in_use_(false),
-      last_packet_log_ms_(-1) {}
-
-ViEReceiver::~ViEReceiver() {
-  UpdateHistograms();
-}
-
-void ViEReceiver::UpdateHistograms() {
-  FecPacketCounter counter = fec_receiver_->GetPacketCounter();
-  if (counter.num_packets > 0) {
-    RTC_LOGGED_HISTOGRAM_PERCENTAGE(
-        "WebRTC.Video.ReceivedFecPacketsInPercent",
-        static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
-  }
-  if (counter.num_fec_packets > 0) {
-    RTC_LOGGED_HISTOGRAM_PERCENTAGE(
-        "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
-        static_cast<int>(counter.num_recovered_packets * 100 /
-                         counter.num_fec_packets));
-  }
-}
-
-bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
-  int8_t old_pltype = -1;
-  if (rtp_payload_registry_.ReceivePayloadType(
-          video_codec.plName, kVideoPayloadTypeFrequency, 0,
-          video_codec.maxBitrate, &old_pltype) != -1) {
-    rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
-  }
-
-  return rtp_receiver_->RegisterReceivePayload(
-             video_codec.plName, video_codec.plType, kVideoPayloadTypeFrequency,
-             0, 0) == 0;
-}
-
-void ViEReceiver::SetNackStatus(bool enable,
-                                int max_nack_reordering_threshold) {
-  if (!enable) {
-    // Reset the threshold back to the lower default threshold when NACK is
-    // disabled since we no longer will be receiving retransmissions.
-    max_nack_reordering_threshold = kDefaultMaxReorderingThreshold;
-  }
-  rtp_receive_statistics_->SetMaxReorderingThreshold(
-      max_nack_reordering_threshold);
-  rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
-}
-
-void ViEReceiver::SetRtxPayloadType(int payload_type,
-                                    int associated_payload_type) {
-  rtp_payload_registry_.SetRtxPayloadType(payload_type,
-                                          associated_payload_type);
-}
-
-void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) {
-  rtp_payload_registry_.set_use_rtx_payload_mapping_on_restore(val);
-}
-
-void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
-  rtp_payload_registry_.SetRtxSsrc(ssrc);
-}
-
-bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const {
-  return rtp_payload_registry_.GetRtxSsrc(ssrc);
-}
-
-bool ViEReceiver::IsFecEnabled() const {
-  return rtp_payload_registry_.ulpfec_payload_type() > -1;
-}
-
-uint32_t ViEReceiver::GetRemoteSsrc() const {
-  return rtp_receiver_->SSRC();
-}
-
-int ViEReceiver::GetCsrcs(uint32_t* csrcs) const {
-  return rtp_receiver_->CSRCs(csrcs);
-}
-
-void ViEReceiver::Init(const std::vector<RtpRtcp*>& modules) {
-  rtp_rtcp_ = modules;
-}
-
-RtpReceiver* ViEReceiver::GetRtpReceiver() const {
-  return rtp_receiver_.get();
-}
-
-void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension,
-                                                  int id) {
-  RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
-  RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
-      StringToRtpExtensionType(extension), id));
-}
-
-int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
-                                           const size_t payload_size,
-                                           const WebRtcRTPHeader* rtp_header) {
-  RTC_DCHECK(vcm_);
-  WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
-  rtp_header_with_ntp.ntp_time_ms =
-      ntp_estimator_.Estimate(rtp_header->header.timestamp);
-  if (vcm_->IncomingPacket(payload_data,
-                           payload_size,
-                           rtp_header_with_ntp) != 0) {
-    // Check this...
-    return -1;
-  }
-  return 0;
-}
-
-bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
-                                    size_t rtp_packet_length) {
-  RTPHeader header;
-  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
-    return false;
-  }
-  header.payload_type_frequency = kVideoPayloadTypeFrequency;
-  bool in_order = IsPacketInOrder(header);
-  return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
-}
-
-bool ViEReceiver::DeliverRtp(const uint8_t* rtp_packet,
-                             size_t rtp_packet_length,
-                             const PacketTime& packet_time) {
-  RTC_DCHECK(remote_bitrate_estimator_);
-  {
-    rtc::CritScope lock(&receive_cs_);
-    if (!receiving_) {
-      return false;
-    }
-  }
-
-  RTPHeader header;
-  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
-                                 &header)) {
-    return false;
-  }
-  size_t payload_length = rtp_packet_length - header.headerLength;
-  int64_t arrival_time_ms;
-  int64_t now_ms = clock_->TimeInMilliseconds();
-  if (packet_time.timestamp != -1)
-    arrival_time_ms = (packet_time.timestamp + 500) / 1000;
-  else
-    arrival_time_ms = now_ms;
-
-  {
-    // Periodically log the RTP header of incoming packets.
-    rtc::CritScope lock(&receive_cs_);
-    if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
-      std::stringstream ss;
-      ss << "Packet received on SSRC: " << header.ssrc << " with payload type: "
-         << static_cast<int>(header.payloadType) << ", timestamp: "
-         << header.timestamp << ", sequence number: " << header.sequenceNumber
-         << ", arrival time: " << arrival_time_ms;
-      if (header.extension.hasTransmissionTimeOffset)
-        ss << ", toffset: " << header.extension.transmissionTimeOffset;
-      if (header.extension.hasAbsoluteSendTime)
-        ss << ", abs send time: " << header.extension.absoluteSendTime;
-      LOG(LS_INFO) << ss.str();
-      last_packet_log_ms_ = now_ms;
-    }
-  }
-
-  remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length,
-                                            header, true);
-  header.payload_type_frequency = kVideoPayloadTypeFrequency;
-
-  bool in_order = IsPacketInOrder(header);
-  rtp_payload_registry_.SetIncomingPayloadType(header);
-  bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
-  // Update receive statistics after ReceivePacket.
-  // Receive statistics will be reset if the payload type changes (make sure
-  // that the first packet is included in the stats).
-  rtp_receive_statistics_->IncomingPacket(
-      header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
-  return ret;
-}
-
-bool ViEReceiver::ReceivePacket(const uint8_t* packet,
-                                size_t packet_length,
-                                const RTPHeader& header,
-                                bool in_order) {
-  if (rtp_payload_registry_.IsEncapsulated(header)) {
-    return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
-  }
-  const uint8_t* payload = packet + header.headerLength;
-  assert(packet_length >= header.headerLength);
-  size_t payload_length = packet_length - header.headerLength;
-  PayloadUnion payload_specific;
-  if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
-                                                 &payload_specific)) {
-    return false;
-  }
-  return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
-                                          payload_specific, in_order);
-}
-
-bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
-                                                    size_t packet_length,
-                                                    const RTPHeader& header) {
-  if (rtp_payload_registry_.IsRed(header)) {
-    int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
-    if (packet[header.headerLength] == ulpfec_pt) {
-      rtp_receive_statistics_->FecPacketReceived(header, packet_length);
-      // Notify vcm about received FEC packets to avoid NACKing these packets.
-      NotifyReceiverOfFecPacket(header);
-    }
-    if (fec_receiver_->AddReceivedRedPacket(
-            header, packet, packet_length, ulpfec_pt) != 0) {
-      return false;
-    }
-    return fec_receiver_->ProcessReceivedFec() == 0;
-  } else if (rtp_payload_registry_.IsRtx(header)) {
-    if (header.headerLength + header.paddingLength == packet_length) {
-      // This is an empty packet and should be silently dropped before trying to
-      // parse the RTX header.
-      return true;
-    }
-    // Remove the RTX header and parse the original RTP header.
-    if (packet_length < header.headerLength)
-      return false;
-    if (packet_length > sizeof(restored_packet_))
-      return false;
-    rtc::CritScope lock(&receive_cs_);
-    if (restored_packet_in_use_) {
-      LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
-      return false;
-    }
-    if (!rtp_payload_registry_.RestoreOriginalPacket(
-            restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
-            header)) {
-      LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
-                      << header.ssrc << " payload type: "
-                      << static_cast<int>(header.payloadType);
-      return false;
-    }
-    restored_packet_in_use_ = true;
-    bool ret = OnRecoveredPacket(restored_packet_, packet_length);
-    restored_packet_in_use_ = false;
-    return ret;
-  }
-  return false;
-}
-
-void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
-  int8_t last_media_payload_type =
-      rtp_payload_registry_.last_received_media_payload_type();
-  if (last_media_payload_type < 0) {
-    LOG(LS_WARNING) << "Failed to get last media payload type.";
-    return;
-  }
-  // Fake an empty media packet.
-  WebRtcRTPHeader rtp_header = {};
-  rtp_header.header = header;
-  rtp_header.header.payloadType = last_media_payload_type;
-  rtp_header.header.paddingLength = 0;
-  PayloadUnion payload_specific;
-  if (!rtp_payload_registry_.GetPayloadSpecifics(last_media_payload_type,
-                                                 &payload_specific)) {
-    LOG(LS_WARNING) << "Failed to get payload specifics.";
-    return;
-  }
-  rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
-  rtp_header.type.Video.rotation = kVideoRotation_0;
-  if (header.extension.hasVideoRotation) {
-    rtp_header.type.Video.rotation =
-        ConvertCVOByteToVideoRotation(header.extension.videoRotation);
-  }
-  OnReceivedPayloadData(nullptr, 0, &rtp_header);
-}
-
-bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
-                              size_t rtcp_packet_length) {
-  // Should be set by owner at construction time.
-  RTC_DCHECK(!rtp_rtcp_.empty());
-  {
-    rtc::CritScope lock(&receive_cs_);
-    if (!receiving_) {
-      return false;
-    }
-  }
-
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_)
-    rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
-
-  int64_t rtt = 0;
-  rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
-  if (rtt == 0) {
-    // Waiting for valid rtt.
-    return true;
-  }
-  uint32_t ntp_secs = 0;
-  uint32_t ntp_frac = 0;
-  uint32_t rtp_timestamp = 0;
-  if (rtp_rtcp_[0]->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
-                              &rtp_timestamp) != 0) {
-    // Waiting for RTCP.
-    return true;
-  }
-  ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
-
-  return true;
-}
-
-void ViEReceiver::StartReceive() {
-  rtc::CritScope lock(&receive_cs_);
-  receiving_ = true;
-}
-
-void ViEReceiver::StopReceive() {
-  rtc::CritScope lock(&receive_cs_);
-  receiving_ = false;
-}
-
-ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const {
-  return rtp_receive_statistics_.get();
-}
-
-bool ViEReceiver::IsPacketInOrder(const RTPHeader& header) const {
-  StreamStatistician* statistician =
-      rtp_receive_statistics_->GetStatistician(header.ssrc);
-  if (!statistician)
-    return false;
-  return statistician->IsPacketInOrder(header.sequenceNumber);
-}
-
-bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header,
-                                        bool in_order) const {
-  // Retransmissions are handled separately if RTX is enabled.
-  if (rtp_payload_registry_.RtxEnabled())
-    return false;
-  StreamStatistician* statistician =
-      rtp_receive_statistics_->GetStatistician(header.ssrc);
-  if (!statistician)
-    return false;
-  // Check if this is a retransmission.
-  int64_t min_rtt = 0;
-  rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr);
-  return !in_order &&
-      statistician->IsRetransmitOfOldPacket(header, min_rtt);
-}
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video/vie_remb.cc b/chromium/third_party/webrtc/video/vie_remb.cc
index b759f5b72c4..a5ccc52ba6d 100644
--- a/chromium/third_party/webrtc/video/vie_remb.cc
+++ b/chromium/third_party/webrtc/video/vie_remb.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/video/vie_remb_unittest.cc b/chromium/third_party/webrtc/video/vie_remb_unittest.cc
index 5f72b967cba..2c33401b45c 100644
--- a/chromium/third_party/webrtc/video/vie_remb_unittest.cc
+++ b/chromium/third_party/webrtc/video/vie_remb_unittest.cc
@@ -8,9 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-
-// This file includes unit tests for ViERemb.
-
 #include <memory>
 #include <vector>
 
@@ -19,7 +16,6 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/utility/include/mock/mock_process_thread.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video/vie_remb.h"
 
 using ::testing::_;
diff --git a/chromium/third_party/webrtc/video/vie_sync_module.cc b/chromium/third_party/webrtc/video/vie_sync_module.cc
index af57ab4b76d..2e62ff81439 100644
--- a/chromium/third_party/webrtc/video/vie_sync_module.cc
+++ b/chromium/third_party/webrtc/video/vie_sync_module.cc
@@ -12,10 +12,11 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/stream_synchronization.h"
 #include "webrtc/video_frame.h"
@@ -48,14 +49,14 @@ int UpdateMeasurements(StreamSynchronization::Measurements* stream,
 }
 }  // namespace
 
-ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
-    : vcm_(vcm),
+ViESyncModule::ViESyncModule(vcm::VideoReceiver* video_receiver)
+    : video_receiver_(video_receiver),
       clock_(Clock::GetRealTimeClock()),
-      video_receiver_(nullptr),
+      rtp_receiver_(nullptr),
       video_rtp_rtcp_(nullptr),
       voe_channel_id_(-1),
       voe_sync_interface_(nullptr),
-      last_sync_time_(TickTime::Now()),
+      last_sync_time_(rtc::TimeNanos()),
       sync_() {}
 
 ViESyncModule::~ViESyncModule() {
@@ -64,20 +65,19 @@ ViESyncModule::~ViESyncModule() {
 void ViESyncModule::ConfigureSync(int voe_channel_id,
                                   VoEVideoSync* voe_sync_interface,
                                   RtpRtcp* video_rtcp_module,
-                                  RtpReceiver* video_receiver) {
+                                  RtpReceiver* rtp_receiver) {
   if (voe_channel_id != -1)
     RTC_DCHECK(voe_sync_interface);
   rtc::CritScope lock(&data_cs_);
   // Prevent expensive no-ops.
   if (voe_channel_id_ == voe_channel_id &&
       voe_sync_interface_ == voe_sync_interface &&
-      video_receiver_ == video_receiver &&
-      video_rtp_rtcp_ == video_rtcp_module) {
+      rtp_receiver_ == rtp_receiver && video_rtp_rtcp_ == video_rtcp_module) {
     return;
   }
   voe_channel_id_ = voe_channel_id;
   voe_sync_interface_ = voe_sync_interface;
-  video_receiver_ = video_receiver;
+  rtp_receiver_ = rtp_receiver;
   video_rtp_rtcp_ = video_rtcp_module;
   sync_.reset(
       new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
@@ -85,14 +85,15 @@ void ViESyncModule::ConfigureSync(int voe_channel_id,
 
 int64_t ViESyncModule::TimeUntilNextProcess() {
   const int64_t kSyncIntervalMs = 1000;
-  return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
+  return kSyncIntervalMs -
+      (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
 }
 
 void ViESyncModule::Process() {
   rtc::CritScope lock(&data_cs_);
-  last_sync_time_ = TickTime::Now();
+  last_sync_time_ = rtc::TimeNanos();
 
-  const int current_video_delay_ms = vcm_->Delay();
+  const int current_video_delay_ms = video_receiver_->Delay();
 
   if (voe_channel_id_ == -1) {
     return;
@@ -120,7 +121,7 @@ void ViESyncModule::Process() {
   assert(voice_receiver);
 
   if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
-                         *video_receiver_) != 0) {
+                         *rtp_receiver_) != 0) {
     return;
   }
 
@@ -154,7 +155,7 @@ void ViESyncModule::Process() {
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
-  vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
+  video_receiver_->SetMinimumPlayoutDelay(target_video_delay_ms);
 }
 
 bool ViESyncModule::GetStreamSyncOffsetInMs(const VideoFrame& frame,
diff --git a/chromium/third_party/webrtc/video/vie_sync_module.h b/chromium/third_party/webrtc/video/vie_sync_module.h
index 2b499ff4d38..18b6c5d0948 100644
--- a/chromium/third_party/webrtc/video/vie_sync_module.h
+++ b/chromium/third_party/webrtc/video/vie_sync_module.h
@@ -18,7 +18,6 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/include/module.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
@@ -26,20 +25,23 @@ namespace webrtc {
 
 class Clock;
 class RtpRtcp;
-class VideoCodingModule;
 class VideoFrame;
 class ViEChannel;
 class VoEVideoSync;
 
+namespace vcm {
+class VideoReceiver;
+}  // namespace vcm
+
 class ViESyncModule : public Module {
  public:
-  explicit ViESyncModule(VideoCodingModule* vcm);
+  explicit ViESyncModule(vcm::VideoReceiver* vcm);
   ~ViESyncModule();
 
   void ConfigureSync(int voe_channel_id,
                      VoEVideoSync* voe_sync_interface,
                      RtpRtcp* video_rtcp_module,
-                     RtpReceiver* video_receiver);
+                     RtpReceiver* rtp_receiver);
 
   // Implements Module.
   int64_t TimeUntilNextProcess() override;
@@ -52,13 +54,13 @@ class ViESyncModule : public Module {
 
  private:
   rtc::CriticalSection data_cs_;
-  VideoCodingModule* const vcm_;
+  vcm::VideoReceiver* const video_receiver_;
   Clock* const clock_;
-  RtpReceiver* video_receiver_;
+  RtpReceiver* rtp_receiver_;
   RtpRtcp* video_rtp_rtcp_;
   int voe_channel_id_;
   VoEVideoSync* voe_sync_interface_;
-  TickTime last_sync_time_;
+  int64_t last_sync_time_;
   std::unique_ptr<StreamSynchronization> sync_;
   StreamSynchronization::Measurements audio_measurement_;
   StreamSynchronization::Measurements video_measurement_;
diff --git a/chromium/third_party/webrtc/video/webrtc_video.gypi b/chromium/third_party/webrtc/video/webrtc_video.gypi
index f11ce957270..3d6afdee071 100644
--- a/chromium/third_party/webrtc/video/webrtc_video.gypi
+++ b/chromium/third_party/webrtc/video/webrtc_video.gypi
@@ -17,7 +17,6 @@
       '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       '<(webrtc_root)/modules/modules.gyp:video_capture_module',
       '<(webrtc_root)/modules/modules.gyp:video_processing',
-      '<(webrtc_root)/modules/modules.gyp:video_render_module',
       '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
       '<(webrtc_root)/modules/modules.gyp:webrtc_video_coding',
       '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
@@ -39,6 +38,10 @@
       'video/receive_statistics_proxy.h',
       'video/report_block_stats.cc',
       'video/report_block_stats.h',
+      'video/rtp_stream_receiver.cc',
+      'video/rtp_stream_receiver.h',
+      'video/send_delay_stats.cc',
+      'video/send_delay_stats.h',
       'video/send_statistics_proxy.cc',
       'video/send_statistics_proxy.h',
       'video/stream_synchronization.cc',
@@ -51,12 +54,10 @@
       'video/video_receive_stream.h',
       'video/video_send_stream.cc',
       'video/video_send_stream.h',
-      'video/vie_channel.cc',
-      'video/vie_channel.h',
+      'video/video_stream_decoder.cc',
+      'video/video_stream_decoder.h',
       'video/vie_encoder.cc',
       'video/vie_encoder.h',
-      'video/vie_receiver.cc',
-      'video/vie_receiver.h',
       'video/vie_remb.cc',
       'video/vie_remb.h',
       'video/vie_sync_module.cc',
diff --git a/chromium/third_party/webrtc/video_encoder.h b/chromium/third_party/webrtc/video_encoder.h
index 89a6464b7c1..0100239e0a4 100644
--- a/chromium/third_party/webrtc/video_encoder.h
+++ b/chromium/third_party/webrtc/video_encoder.h
@@ -31,6 +31,7 @@ class EncodedImageCallback {
   virtual ~EncodedImageCallback() {}
 
   // Callback function which is called when an image has been encoded.
+  // TODO(perkj): Change this to return void.
   virtual int32_t Encoded(const EncodedImage& encoded_image,
                           const CodecSpecificInfo* codec_specific_info,
                           const RTPFragmentationHeader* fragmentation) = 0;
@@ -124,7 +125,6 @@ class VideoEncoder {
 
   virtual int32_t SetPeriodicKeyFrames(bool enable) { return -1; }
   virtual void OnDroppedFrame() {}
-  virtual int GetTargetFramerate() { return -1; }
   virtual bool SupportsNativeHandle() const { return false; }
   virtual const char* ImplementationName() const { return "unknown"; }
 };
@@ -152,7 +152,6 @@ class VideoEncoderSoftwareFallbackWrapper : public VideoEncoder {
 
   int32_t SetRates(uint32_t bitrate, uint32_t framerate) override;
   void OnDroppedFrame() override;
-  int GetTargetFramerate() override;
   bool SupportsNativeHandle() const override;
   const char* ImplementationName() const override;
 
diff --git a/chromium/third_party/webrtc/video_engine_tests_apk.isolate b/chromium/third_party/webrtc/video_engine_tests_apk.isolate
new file mode 100644
index 00000000000..6f4c3d41cc2
--- /dev/null
+++ b/chromium/third_party/webrtc/video_engine_tests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../build/android/android.isolate',
+    'video_engine_tests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_video_engine_tests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../build/config/',
+      '../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/video_engine_tests_apk/',
+      '<(PRODUCT_DIR)/bin/run_video_engine_tests',
+      'video_engine_tests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/video_frame.h b/chromium/third_party/webrtc/video_frame.h
index 28a6b8716d3..4dc3411e98a 100644
--- a/chromium/third_party/webrtc/video_frame.h
+++ b/chromium/third_party/webrtc/video_frame.h
@@ -65,20 +65,9 @@ class VideoFrame {
   // reference to the video buffer also retained by |videoFrame|.
   void ShallowCopy(const VideoFrame& videoFrame);
 
-  // Release frame buffer and reset time stamps.
-  void Reset();
-
-  // Get pointer to buffer per plane.
-  uint8_t* buffer(PlaneType type);
-  // Overloading with const.
-  const uint8_t* buffer(PlaneType type) const;
-
   // Get allocated size per plane.
   int allocated_size(PlaneType type) const;
 
-  // Get allocated stride per plane.
-  int stride(PlaneType type) const;
-
   // Get frame width.
   int width() const;
 
@@ -125,13 +114,12 @@ class VideoFrame {
   // Return true if underlying plane buffers are of zero size, false if not.
   bool IsZeroSize() const;
 
-  // Return the handle of the underlying video frame. This is used when the
-  // frame is backed by a texture. The object should be destroyed when it is no
-  // longer in use, so the underlying resource can be freed.
-  void* native_handle() const;
-
-  // Return the underlying buffer.
-  rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer() const;
+  // Return the underlying buffer. Never nullptr for a properly
+  // initialized VideoFrame.
+  // Creating a new reference breaks the HasOneRef and IsMutable
+  // logic. So return a const ref to our reference.
+  const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& video_frame_buffer()
+      const;
 
   // Set the underlying buffer.
   void set_video_frame_buffer(
@@ -161,6 +149,7 @@ class EncodedImage {
   static size_t GetBufferPaddingBytes(VideoCodecType codec_type);
 
   EncodedImage() : EncodedImage(nullptr, 0, 0) {}
+
   EncodedImage(uint8_t* buffer, size_t length, size_t size)
       : _buffer(buffer), _length(length), _size(size) {}
 
@@ -186,6 +175,7 @@ class EncodedImage {
   uint8_t* _buffer;
   size_t _length;
   size_t _size;
+  VideoRotation rotation_ = kVideoRotation_0;
   bool _completeFrame = false;
   AdaptReason adapt_reason_;
   int qp_ = -1;  // Quantizer value.
diff --git a/chromium/third_party/webrtc/video_receive_stream.h b/chromium/third_party/webrtc/video_receive_stream.h
index 109c2aa15b8..167df47dff4 100644
--- a/chromium/third_party/webrtc/video_receive_stream.h
+++ b/chromium/third_party/webrtc/video_receive_stream.h
@@ -17,17 +17,16 @@
 #include <vector>
 
 #include "webrtc/common_types.h"
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
-#include "webrtc/frame_callback.h"
-#include "webrtc/stream.h"
-#include "webrtc/transport.h"
 #include "webrtc/media/base/videosinkinterface.h"
+#include "webrtc/transport.h"
 
 namespace webrtc {
 
 class VideoDecoder;
 
-class VideoReceiveStream : public ReceiveStream {
+class VideoReceiveStream {
  public:
   // TODO(mflodman) Move all these settings to VideoDecoder and move the
   // declaration to common_types.h.
@@ -176,8 +175,18 @@ class VideoReceiveStream : public ReceiveStream {
     int target_delay_ms = 0;
   };
 
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
   // TODO(pbos): Add info on currently-received codec to Stats.
   virtual Stats GetStats() const = 0;
+
+ protected:
+  virtual ~VideoReceiveStream() {}
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/video_send_stream.h b/chromium/third_party/webrtc/video_send_stream.h
index 7815acfc806..886367f0a50 100644
--- a/chromium/third_party/webrtc/video_send_stream.h
+++ b/chromium/third_party/webrtc/video_send_stream.h
@@ -15,9 +15,9 @@
 #include <string>
 
 #include "webrtc/common_types.h"
+#include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
-#include "webrtc/frame_callback.h"
-#include "webrtc/stream.h"
+#include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/transport.h"
 #include "webrtc/media/base/videosinkinterface.h"
 
@@ -38,7 +38,7 @@ class VideoCaptureInput {
   virtual ~VideoCaptureInput() {}
 };
 
-class VideoSendStream : public SendStream {
+class VideoSendStream {
  public:
   struct StreamStats {
     FrameCounts frame_counts;
@@ -139,7 +139,7 @@ class VideoSendStream : public SendStream {
 
     // Called for each I420 frame before encoding the frame. Can be used for
     // effects, snapshots etc. 'nullptr' disables the callback.
-    I420FrameCallback* pre_encode_callback = nullptr;
+    rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
 
     // Called for each encoded frame, e.g. used for file storage. 'nullptr'
     // disables the callback. Also measures timing and passes the time
@@ -166,6 +166,13 @@ class VideoSendStream : public SendStream {
     bool suspend_below_min_bitrate = false;
   };
 
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
   // Gets interface used to insert captured frames. Valid as long as the
   // VideoSendStream is valid.
   virtual VideoCaptureInput* Input() = 0;
@@ -176,6 +183,9 @@ class VideoSendStream : public SendStream {
   virtual void ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
   virtual Stats GetStats() = 0;
+
+ protected:
+  virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/voice_engine/channel.cc b/chromium/third_party/webrtc/voice_engine/channel.cc
index b9adde7e3ec..d9184c828ed 100644
--- a/chromium/third_party/webrtc/voice_engine/channel.cc
+++ b/chromium/third_party/webrtc/voice_engine/channel.cc
@@ -72,13 +72,11 @@ class TransportFeedbackProxy : public TransportFeedbackObserver {
   }
 
   // Implements TransportFeedbackObserver.
-  void AddPacket(uint16_t sequence_number,
-                 size_t length,
-                 bool was_paced) override {
+  void AddPacket(uint16_t sequence_number, size_t length) override {
     RTC_DCHECK(pacer_thread_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
     if (feedback_observer_)
-      feedback_observer_->AddPacket(sequence_number, length, was_paced);
+      feedback_observer_->AddPacket(sequence_number, length);
   }
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
     RTC_DCHECK(network_thread_.CalledOnValidThread());
@@ -476,22 +474,32 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
   return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
 }
 
-int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
+MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
+    int32_t id,
+    AudioFrame* audioFrame) {
   if (event_log_) {
     unsigned int ssrc;
     RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
     event_log_->LogAudioPlayout(ssrc);
   }
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
-  if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame) ==
-      -1) {
+  bool muted;
+  if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
+                                     &muted) == -1) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
     // In all likelihood, the audio in this frame is garbage. We return an
     // error so that the audio mixer module doesn't add it to the mix. As
     // a result, it won't be played out and the actions skipped here are
     // irrelevant.
-    return -1;
+    return MixerParticipant::AudioFrameInfo::kError;
+  }
+
+  if (muted) {
+    // TODO(henrik.lundin): We should be able to do better than this. But we
+    // will have to go through all the cases below where the audio samples may
+    // be used, and handle the muted case in some way.
+    audioFrame->Mute();
   }
 
   if (_RxVadDetection) {
@@ -563,6 +571,7 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
   // Mix decoded PCM output with file if file mixing is enabled
   if (state.output_file_playing) {
     MixAudioWithFile(*audioFrame, audioFrame->sample_rate_hz_);
+    muted = false;  // We may have added non-zero samples.
   }
 
   // External media
@@ -587,6 +596,7 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
   }
 
   // Measure audio level (0-9)
+  // TODO(henrik.lundin) Use the |muted| information here too.
   _outputAudioLevel.ComputeLevel(*audioFrame);
 
   if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) {
@@ -619,7 +629,8 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
     }
   }
 
-  return 0;
+  return muted ? MixerParticipant::AudioFrameInfo::kMuted
+               : MixerParticipant::AudioFrameInfo::kNormal;
 }
 
 int32_t Channel::NeededFrequency(int32_t id) const {
@@ -811,6 +822,7 @@ Channel::Channel(int32_t channelId,
   }
   acm_config.neteq_config.enable_fast_accelerate =
       config.Get<NetEqFastAccelerate>().enabled;
+  acm_config.neteq_config.enable_muted_state = true;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _outputAudioLevel.Clear();
@@ -1449,12 +1461,11 @@ int Channel::SetOpusDtx(bool enable_dtx) {
   return 0;
 }
 
-int32_t Channel::RegisterExternalTransport(Transport& transport) {
+int32_t Channel::RegisterExternalTransport(Transport* transport) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalTransport()");
 
   rtc::CritScope cs(&_callbackCritSect);
-
   if (_externalTransport) {
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceError,
@@ -1462,7 +1473,7 @@ int32_t Channel::RegisterExternalTransport(Transport& transport) {
     return -1;
   }
   _externalTransport = true;
-  _transportPtr = &transport;
+  _transportPtr = transport;
   return 0;
 }
 
@@ -1471,22 +1482,21 @@ int32_t Channel::DeRegisterExternalTransport() {
                "Channel::DeRegisterExternalTransport()");
 
   rtc::CritScope cs(&_callbackCritSect);
-
-  if (!_transportPtr) {
+  if (_transportPtr) {
+    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+                 "DeRegisterExternalTransport() all transport is disabled");
+  } else {
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceWarning,
         "DeRegisterExternalTransport() external transport already "
         "disabled");
-    return 0;
   }
   _externalTransport = false;
   _transportPtr = NULL;
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "DeRegisterExternalTransport() all transport is disabled");
   return 0;
 }
 
-int32_t Channel::ReceivedRTPPacket(const int8_t* data,
+int32_t Channel::ReceivedRTPPacket(const uint8_t* received_packet,
                                    size_t length,
                                    const PacketTime& packet_time) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
@@ -1495,7 +1505,6 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data,
   // Store playout timestamp for the received RTP packet
   UpdatePlayoutTimestamp(false);
 
-  const uint8_t* received_packet = reinterpret_cast<const uint8_t*>(data);
   RTPHeader header;
   if (!rtp_header_parser_->Parse(received_packet, length, &header)) {
     WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
@@ -1585,14 +1594,14 @@ bool Channel::IsPacketRetransmitted(const RTPHeader& header,
   return !in_order && statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 
-int32_t Channel::ReceivedRTCPPacket(const int8_t* data, size_t length) {
+int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::ReceivedRTCPPacket()");
   // Store playout timestamp for the received RTCP packet
   UpdatePlayoutTimestamp(true);
 
   // Deliver RTCP packet to RTP/RTCP module for parsing
-  if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, length) == -1) {
+  if (_rtpRtcpModule->IncomingRtcpPacket(data, length) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
         "Channel::IncomingRTPPacket() RTCP packet is invalid");
@@ -2917,7 +2926,6 @@ void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
   if (!pacing_enabled_)
     _rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
   rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
-  rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
   if (enable)
     audio_coding_->EnableNack(maxNumberOfPackets);
   else
diff --git a/chromium/third_party/webrtc/voice_engine/channel.h b/chromium/third_party/webrtc/voice_engine/channel.h
index d22da74ae5f..cd31e7702dd 100644
--- a/chromium/third_party/webrtc/voice_engine/channel.h
+++ b/chromium/third_party/webrtc/voice_engine/channel.h
@@ -218,12 +218,12 @@ class Channel
   int SetOpusDtx(bool enable_dtx);
 
   // VoENetwork
-  int32_t RegisterExternalTransport(Transport& transport);
+  int32_t RegisterExternalTransport(Transport* transport);
   int32_t DeRegisterExternalTransport();
-  int32_t ReceivedRTPPacket(const int8_t* data,
+  int32_t ReceivedRTPPacket(const uint8_t* received_packet,
                             size_t length,
                             const PacketTime& packet_time);
-  int32_t ReceivedRTCPPacket(const int8_t* data, size_t length);
+  int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
 
   // VoEFile
   int StartPlayingFileLocally(const char* fileName,
@@ -394,7 +394,9 @@ class Channel
   bool SendRtcp(const uint8_t* data, size_t len) override;
 
   // From MixerParticipant
-  int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override;
+  MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted(
+      int32_t id,
+      AudioFrame* audioFrame) override;
   int32_t NeededFrequency(int32_t id) const override;
 
   // From FileCallback
diff --git a/chromium/third_party/webrtc/voice_engine/channel_manager.cc b/chromium/third_party/webrtc/voice_engine/channel_manager.cc
index eac2e50919d..6071f19548e 100644
--- a/chromium/third_party/webrtc/voice_engine/channel_manager.cc
+++ b/chromium/third_party/webrtc/voice_engine/channel_manager.cc
@@ -49,7 +49,7 @@ ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
     : instance_id_(instance_id),
       last_channel_id_(-1),
       config_(config),
-      event_log_(RtcEventLog::Create()) {}
+      event_log_(RtcEventLog::Create(Clock::GetRealTimeClock())) {}
 
 ChannelOwner ChannelManager::CreateChannel() {
   return CreateChannelInternal(config_);
diff --git a/chromium/third_party/webrtc/voice_engine/channel_proxy.cc b/chromium/third_party/webrtc/voice_engine/channel_proxy.cc
index 10c88212024..4cc7f5cbbcd 100644
--- a/chromium/third_party/webrtc/voice_engine/channel_proxy.cc
+++ b/chromium/third_party/webrtc/voice_engine/channel_proxy.cc
@@ -158,6 +158,29 @@ void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   channel()->SetSink(std::move(sink));
 }
 
+void ChannelProxy::RegisterExternalTransport(Transport* transport) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  int error = channel()->RegisterExternalTransport(transport);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::DeRegisterExternalTransport() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->DeRegisterExternalTransport();
+}
+
+bool ChannelProxy::ReceivedRTPPacket(const uint8_t* packet,
+                                     size_t length,
+                                     const PacketTime& packet_time) {
+  // May be called on either worker thread or network thread.
+  return channel()->ReceivedRTPPacket(packet, length, packet_time) == 0;
+}
+
+bool ChannelProxy::ReceivedRTCPPacket(const uint8_t* packet, size_t length) {
+  // May be called on either worker thread or network thread.
+  return channel()->ReceivedRTCPPacket(packet, length) == 0;
+}
+
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
diff --git a/chromium/third_party/webrtc/voice_engine/channel_proxy.h b/chromium/third_party/webrtc/voice_engine/channel_proxy.h
index 344769e1a91..df0c3f22ef6 100644
--- a/chromium/third_party/webrtc/voice_engine/channel_proxy.h
+++ b/chromium/third_party/webrtc/voice_engine/channel_proxy.h
@@ -25,6 +25,7 @@ namespace webrtc {
 class AudioSinkInterface;
 class PacketRouter;
 class RtpPacketSender;
+class Transport;
 class TransportFeedbackObserver;
 
 namespace voe {
@@ -70,9 +71,15 @@ class ChannelProxy {
 
   virtual bool SetSendTelephoneEventPayloadType(int payload_type);
   virtual bool SendTelephoneEventOutband(int event, int duration_ms);
-
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
+  virtual void RegisterExternalTransport(Transport* transport);
+  virtual void DeRegisterExternalTransport();
+  virtual bool ReceivedRTPPacket(const uint8_t* packet,
+                                 size_t length,
+                                 const PacketTime& packet_time);
+  virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
+
  private:
   Channel* channel() const;
 
diff --git a/chromium/third_party/webrtc/voice_engine/monitor_module.cc b/chromium/third_party/webrtc/voice_engine/monitor_module.cc
index fb3c2b49201..8a1865a95b4 100644
--- a/chromium/third_party/webrtc/voice_engine/monitor_module.cc
+++ b/chromium/third_party/webrtc/voice_engine/monitor_module.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/voice_engine/monitor_module.h"
 
 namespace webrtc  {
@@ -17,7 +17,7 @@ namespace voe  {
 
 MonitorModule::MonitorModule() :
     _observerPtr(NULL),
-    _lastProcessTime(TickTime::MillisecondTimestamp())
+    _lastProcessTime(rtc::TimeMillis())
 {
 }
 
@@ -52,7 +52,7 @@ MonitorModule::DeRegisterObserver()
 int64_t
 MonitorModule::TimeUntilNextProcess()
 {
-    int64_t now = TickTime::MillisecondTimestamp();
+    int64_t now = rtc::TimeMillis();
     const int64_t kAverageProcessUpdateTimeMs = 1000;
     return kAverageProcessUpdateTimeMs - (now - _lastProcessTime);
 }
@@ -60,7 +60,7 @@ MonitorModule::TimeUntilNextProcess()
 void
 MonitorModule::Process()
 {
-    _lastProcessTime = TickTime::MillisecondTimestamp();
+    _lastProcessTime = rtc::TimeMillis();
     rtc::CritScope lock(&_callbackCritSect);
     if (_observerPtr)
     {
diff --git a/chromium/third_party/webrtc/voice_engine/shared_data.cc b/chromium/third_party/webrtc/voice_engine/shared_data.cc
index 997f51b4396..7a67561d1ee 100644
--- a/chromium/third_party/webrtc/voice_engine/shared_data.cc
+++ b/chromium/third_party/webrtc/voice_engine/shared_data.cc
@@ -28,7 +28,7 @@ SharedData::SharedData(const Config& config)
       _engineStatistics(_gInstanceCounter),
       _audioDevicePtr(NULL),
       _moduleProcessThreadPtr(
-          rtc::ScopedToUnique(ProcessThread::Create("VoiceProcessThread"))) {
+          ProcessThread::Create("VoiceProcessThread")) {
     Trace::CreateTrace();
     if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
     {
diff --git a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index 086eeab7b1b..b15d72fecbb 100644
--- a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -131,7 +131,7 @@ void ConferenceTransport::StorePacket(Packet::Type type,
                                       size_t len) {
   {
     rtc::CritScope lock(&pq_crit_);
-    packet_queue_.push_back(Packet(type, data, len, rtc::Time()));
+    packet_queue_.push_back(Packet(type, data, len, rtc::TimeMillis()));
   }
   packet_event_->Set();
 }
diff --git a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 8fd74577112..bbdf5015d4a 100644
--- a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -108,7 +108,7 @@ class ConferenceTransport: public webrtc::Transport {
     enum Type { Rtp, Rtcp, } type_;
 
     Packet() : len_(0) {}
-    Packet(Type type, const void* data, size_t len, uint32_t time_ms)
+    Packet(Type type, const void* data, size_t len, int64_t time_ms)
         : type_(type), len_(len), send_time_ms_(time_ms) {
       EXPECT_LE(len_, kMaxPacketSizeByte);
       memcpy(data_, data, len_);
@@ -116,7 +116,7 @@ class ConferenceTransport: public webrtc::Transport {
 
     uint8_t data_[kMaxPacketSizeByte];
     size_t len_;
-    uint32_t send_time_ms_;
+    int64_t send_time_ms_;
   };
 
   static bool Run(void* transport) {
diff --git a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
index d4438a4e158..1787915356a 100644
--- a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
+++ b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
@@ -14,11 +14,10 @@
 
 namespace voetest {
 
-void LoudestFilter::RemoveTimeoutStreams(uint32_t time_ms) {
+void LoudestFilter::RemoveTimeoutStreams(int64_t time_ms) {
   auto it = stream_levels_.begin();
   while (it != stream_levels_.end()) {
-    if (rtc::TimeDiff(time_ms, it->second.last_time_ms) >
-        kStreamTimeOutMs) {
+    if (rtc::TimeDiff(time_ms, it->second.last_time_ms) > kStreamTimeOutMs) {
       stream_levels_.erase(it++);
     } else {
       ++it;
@@ -41,7 +40,7 @@ unsigned int LoudestFilter::FindQuietestStream() {
 }
 
 bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) {
-  uint32_t time_now_ms = rtc::Time();
+  int64_t time_now_ms = rtc::TimeMillis();
   RemoveTimeoutStreams(time_now_ms);
 
   int source_ssrc = rtp_header.ssrc;
diff --git a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h
index 73b801cc989..f862c818e72 100644
--- a/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h
+++ b/chromium/third_party/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h
@@ -29,15 +29,15 @@ class LoudestFilter {
 
  private:
   struct Status {
-    void Set(int audio_level, uint32_t last_time_ms) {
+    void Set(int audio_level, int64_t last_time_ms) {
       this->audio_level = audio_level;
       this->last_time_ms = last_time_ms;
     }
     int audio_level;
-    uint32_t last_time_ms;
+    int64_t last_time_ms;
   };
 
-  void RemoveTimeoutStreams(uint32_t time_ms);
+  void RemoveTimeoutStreams(int64_t time_ms);
   unsigned int FindQuietestStream();
 
   // Keeps the streams being forwarded in pair<SSRC, Status>.
diff --git a/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_conference_test.cc b/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
index c70d92a9462..946eb4e525f 100644
--- a/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
+++ b/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
@@ -72,15 +72,15 @@ TEST(VoeConferenceTest, RttAndStartNtpTime) {
   const int kStatsRequestIntervalMs = 1000;
   const int kStatsBufferSize = 3;
 
-  uint32_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
+  int64_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
   // Run the following up to |kMaxRunTimeMs| milliseconds.
   int successive_pass = 0;
   webrtc::CallStatistics stats_1;
   webrtc::CallStatistics stats_2;
   std::queue<Stats> stats_buffer;
 
-  while (rtc::TimeIsLater(rtc::Time(), deadline) &&
-      successive_pass < kNeedSuccessivePass) {
+  while (rtc::TimeMillis() < deadline &&
+         successive_pass < kNeedSuccessivePass) {
     webrtc::SleepMs(kStatsRequestIntervalMs);
 
     EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
diff --git a/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_output_test.cc b/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_output_test.cc
index d1bcf968b00..795dac5c0a9 100644
--- a/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_output_test.cc
+++ b/chromium/third_party/webrtc/voice_engine/test/auto_test/voe_output_test.cc
@@ -183,7 +183,7 @@ TEST(OutputTest, DISABLED_OpusDtxHasNoNoisePump) {
   OutputTest test(-kDtxBoundForSilence, kDtxBoundForSilence);
   Random random(1234ull);
 
-  uint32_t start_time = rtc::Time();
+  int64_t start_time = rtc::TimeMillis();
   test.Start();
   while (rtc::TimeSince(start_time) < kRuntimeMs) {
     webrtc::SleepMs(random.Rand(kUnmuteTimeMs - kUnmuteTimeMs / 10,
diff --git a/chromium/third_party/webrtc/voice_engine/voe_base_impl.cc b/chromium/third_party/webrtc/voice_engine/voe_base_impl.cc
index e3dee0f1a97..8e93778f7ed 100644
--- a/chromium/third_party/webrtc/voice_engine/voe_base_impl.cc
+++ b/chromium/third_party/webrtc/voice_engine/voe_base_impl.cc
@@ -229,7 +229,7 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
     return -1;
 #else
     // Create the internal ADM implementation.
-    shared_->set_audio_device(AudioDeviceModuleImpl::Create(
+    shared_->set_audio_device(AudioDeviceModule::Create(
         VoEId(shared_->instance_id(), -1), shared_->audio_device_layer()));
 
     if (shared_->audio_device() == nullptr) {
@@ -620,11 +620,14 @@ int32_t VoEBaseImpl::StopPlayout() {
 }
 
 int32_t VoEBaseImpl::StartSend() {
-  if (!shared_->audio_device()->Recording()) {
+  if (!shared_->audio_device()->RecordingIsInitialized() &&
+      !shared_->audio_device()->Recording()) {
     if (shared_->audio_device()->InitRecording() != 0) {
       LOG_F(LS_ERROR) << "Failed to initialize recording";
       return -1;
     }
+  }
+  if (!shared_->audio_device()->Recording()) {
     if (shared_->audio_device()->StartRecording() != 0) {
       LOG_F(LS_ERROR) << "Failed to start recording";
       return -1;
diff --git a/chromium/third_party/webrtc/voice_engine/voe_external_media_impl.cc b/chromium/third_party/webrtc/voice_engine/voe_external_media_impl.cc
index 7b2b6a00f1e..cae0715ced2 100644
--- a/chromium/third_party/webrtc/voice_engine/voe_external_media_impl.cc
+++ b/chromium/third_party/webrtc/voice_engine/voe_external_media_impl.cc
@@ -153,7 +153,11 @@ int VoEExternalMediaImpl::GetAudioFrame(int channel, int desired_sample_rate_hz,
   }
   frame->sample_rate_hz_ =
       desired_sample_rate_hz == 0 ? -1 : desired_sample_rate_hz;
-  return channelPtr->GetAudioFrame(channel, frame);
+  auto ret = channelPtr->GetAudioFrameWithMuted(channel, frame);
+  if (ret == MixerParticipant::AudioFrameInfo::kMuted) {
+    frame->Mute();
+  }
+  return ret == MixerParticipant::AudioFrameInfo::kError ? -1 : 0;
 }
 
 int VoEExternalMediaImpl::SetExternalMixing(int channel, bool enable) {
diff --git a/chromium/third_party/webrtc/voice_engine/voe_network_impl.cc b/chromium/third_party/webrtc/voice_engine/voe_network_impl.cc
index 55620482951..6941629d79d 100644
--- a/chromium/third_party/webrtc/voice_engine/voe_network_impl.cc
+++ b/chromium/third_party/webrtc/voice_engine/voe_network_impl.cc
@@ -43,7 +43,7 @@ int VoENetworkImpl::RegisterExternalTransport(int channel,
     LOG_F(LS_ERROR) << "Failed to locate channel: " << channel;
     return -1;
   }
-  return channelPtr->RegisterExternalTransport(transport);
+  return channelPtr->RegisterExternalTransport(&transport);
 }
 
 int VoENetworkImpl::DeRegisterExternalTransport(int channel) {
@@ -84,8 +84,8 @@ int VoENetworkImpl::ReceivedRTPPacket(int channel,
     LOG_F(LS_ERROR) << "No external transport for channel: " << channel;
     return -1;
   }
-  return channelPtr->ReceivedRTPPacket((const int8_t*)data, length,
-                                       packet_time);
+  return channelPtr->ReceivedRTPPacket(static_cast<const uint8_t*>(data),
+                                       length, packet_time);
 }
 
 int VoENetworkImpl::ReceivedRTCPPacket(int channel,
@@ -107,7 +107,8 @@ int VoENetworkImpl::ReceivedRTCPPacket(int channel,
     LOG_F(LS_ERROR) << "No external transport for channel: " << channel;
     return -1;
   }
-  return channelPtr->ReceivedRTCPPacket((const int8_t*)data, length);
+  return channelPtr->ReceivedRTCPPacket(static_cast<const uint8_t*>(data),
+                                        length);
 }
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/voice_engine/voice_engine.gyp b/chromium/third_party/webrtc/voice_engine/voice_engine.gyp
index b93c903d711..a728b1af3a6 100644
--- a/chromium/third_party/webrtc/voice_engine/voice_engine.gyp
+++ b/chromium/third_party/webrtc/voice_engine/voice_engine.gyp
@@ -255,6 +255,27 @@
               ],
             },
           ],
+          'conditions': [
+            ['test_isolation_mode != "noop"',
+              {
+                'targets': [
+                  {
+                    'target_name': 'voice_engine_unittests_apk_run',
+                    'type': 'none',
+                    'dependencies': [
+                      '<(apk_tests_path):voice_engine_unittests_apk',
+                    ],
+                    'includes': [
+                      '../build/isolate.gypi',
+                    ],
+                    'sources': [
+                      'voice_engine_unittests_apk.isolate',
+                    ],
+                  },
+                ],
+              },
+            ],
+          ],
         }],
         ['test_isolation_mode != "noop"', {
           'targets': [
diff --git a/chromium/third_party/webrtc/voice_engine/voice_engine_unittests_apk.isolate b/chromium/third_party/webrtc/voice_engine/voice_engine_unittests_apk.isolate
new file mode 100644
index 00000000000..6f0244c526d
--- /dev/null
+++ b/chromium/third_party/webrtc/voice_engine/voice_engine_unittests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../../build/android/android.isolate',
+    'voice_engine_unittests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_voice_engine_unittests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../../build/config/',
+      '../../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/voice_engine_unittests_apk/',
+      '<(PRODUCT_DIR)/bin/run_voice_engine_unittests',
+      'voice_engine_unittests.isolate',
+    ]
+  }
+}
diff --git a/chromium/third_party/webrtc/webrtc.gyp b/chromium/third_party/webrtc/webrtc.gyp
index ad4b3534d45..793bf96335e 100644
--- a/chromium/third_party/webrtc/webrtc.gyp
+++ b/chromium/third_party/webrtc/webrtc.gyp
@@ -35,6 +35,18 @@
         ],
       },
     }],
+    ['build_with_chromium==0 and'
+     '(OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7"))', {
+      # TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/
+      # has been moved to webrtc/. It can't be processed by Chromium since the
+      # reference to buid/java.gypi is using an absolute path (and includes
+      # entries cannot contain variables).
+      'variables': {
+        'webrtc_all_dependencies': [
+          'sdk/sdk.gyp:*',
+        ],
+      },
+    }],
     ['include_tests==1', {
       'includes': [
         'webrtc_tests.gypi',
@@ -55,6 +67,24 @@
         },
       ],
     }],
+    ['enable_protobuf==1', {
+      'targets': [
+        {
+          'target_name': 'rtc_event_log_parser',
+          'type': 'static_library',
+          'sources': [
+            'call/rtc_event_log_parser.cc',
+            'call/rtc_event_log_parser.h',
+          ],
+          'dependencies': [
+            'rtc_event_log_proto',
+          ],
+          'export_dependent_settings': [
+            'rtc_event_log_proto',
+          ],
+        },
+      ],
+    }],
     ['include_tests==1 and enable_protobuf==1', {
       'targets': [
         {
@@ -63,7 +93,7 @@
           'sources': ['call/rtc_event_log2rtp_dump.cc',],
           'dependencies': [
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            'rtc_event_log',
+            'rtc_event_log_parser',
             'rtc_event_log_proto',
             'test/test.gyp:rtp_test_utils'
           ],
@@ -97,6 +127,12 @@
             'webrtc_tests',
           ],
         }],
+        ['include_tests==1 and'
+         '(OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7"))', {
+           'dependencies': [
+             'sdk/sdk_tests.gyp:*',
+           ],
+        }],
       ],
     },
     {
@@ -108,8 +144,6 @@
         'audio_state.h',
         'call.h',
         'config.h',
-        'frame_callback.h',
-        'stream.h',
         'transport.h',
         'video_receive_stream.h',
         'video_send_stream.h',
@@ -131,7 +165,6 @@
         ['build_with_chromium==1', {
           'dependencies': [
             '<(webrtc_root)/modules/modules.gyp:video_capture',
-            '<(webrtc_root)/modules/modules.gyp:video_render',
           ],
         }],
       ],
@@ -142,6 +175,8 @@
       'sources': [
         'call/rtc_event_log.cc',
         'call/rtc_event_log.h',
+        'call/rtc_event_log_helper_thread.cc',
+        'call/rtc_event_log_helper_thread.h',
       ],
       'conditions': [
         # If enable_protobuf is defined, we want to compile the protobuf
diff --git a/chromium/third_party/webrtc/webrtc_examples.gyp b/chromium/third_party/webrtc/webrtc_examples.gyp
index d5d8adefd9e..6f07b4d7fe3 100755
--- a/chromium/third_party/webrtc/webrtc_examples.gyp
+++ b/chromium/third_party/webrtc/webrtc_examples.gyp
@@ -154,7 +154,7 @@
           'target_name': 'apprtc_common',
           'type': 'static_library',
           'dependencies': [
-            '<(webrtc_root)/base/base.gyp:rtc_base_objc',
+            '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
           ],
           'sources': [
@@ -201,8 +201,7 @@
           'target_name': 'apprtc_signaling',
           'type': 'static_library',
           'dependencies': [
-            '<(webrtc_root)/api/api.gyp:rtc_api_objc',
-            '<(webrtc_root)/base/base.gyp:rtc_base_objc',
+            '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_peerconnection_objc',
             'apprtc_common',
             'socketrocket',
           ],
@@ -251,7 +250,7 @@
             ],
           },
           'export_dependent_settings': [
-            '<(webrtc_root)/api/api.gyp:rtc_api_objc',
+            '<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_peerconnection_objc',
           ],
           'conditions': [
             ['OS=="ios"', {
@@ -409,6 +408,7 @@
           'type': 'none',
           'dependencies': [
             'api/api.gyp:libjingle_peerconnection_java',
+            '<(DEPTH)/third_party/android_tools/android_tools.gyp:android_support_design_javalib'
           ],
           'variables': {
             'apk_name': 'AppRTCDemo',
@@ -419,9 +419,11 @@
             'R_package_relpath': 'org/appspot/apprtc',
             'input_jars_paths': [
               'examples/androidapp/third_party/autobanh/autobanh.jar',
+              '<(DEPTH)/third_party/android_tools/sdk/extras/android/support/v4/android-support-v4.jar',
              ],
             'library_dexed_jars_paths': [
               'examples/androidapp/third_party/autobanh/autobanh.jar',
+              '<(DEPTH)/third_party/android_tools/sdk/extras/android/support/v4/android-support-v4.jar',
              ],
             'native_lib_target': 'libjingle_peerconnection_so',
             'add_to_dependents_classpaths':1,
@@ -454,8 +456,36 @@
             'apk_name': 'AppRTCDemoTest',
             'java_in_dir': 'examples/androidtests',
             'is_test_apk': 1,
+            'test_type': 'instrumentation',
+            'test_runner_path': '<(DEPTH)/webrtc/build/android/test_runner.py',
           },
-          'includes': [ '../build/java_apk.gypi' ],
+          'includes': [
+            '../build/java_apk.gypi',
+            '../build/android/test_runner.gypi',
+          ],
+        },
+
+        {
+          'target_name': 'AppRTCDemoJUnitTest',
+          'type': 'none',
+          'dependencies': [
+            'AppRTCDemo_apk',
+            '<(DEPTH)/base/base.gyp:base_java',
+            '<(DEPTH)/base/base.gyp:base_java_test_support',
+            '<(DEPTH)/base/base.gyp:base_junit_test_support',
+          ],
+          'variables': {
+            'main_class': 'org.chromium.testing.local.JunitTestMain',
+            'src_paths': [
+              'examples/androidjunit/',
+            ],
+            'test_type': 'junit',
+            'wrapper_script_name': 'helper/<(_target_name)',
+          },
+          'includes': [
+            '../build/host_jar.gypi',
+            '../build/android/test_runner.gypi',
+          ],
         },
       ],  # targets
     }],  # OS=="android"
diff --git a/chromium/third_party/webrtc/webrtc_nonparallel_tests_apk.isolate b/chromium/third_party/webrtc/webrtc_nonparallel_tests_apk.isolate
new file mode 100644
index 00000000000..ee83d108143
--- /dev/null
+++ b/chromium/third_party/webrtc/webrtc_nonparallel_tests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../build/android/android.isolate',
+    'webrtc_nonparallel_tests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_webrtc_nonparallel_tests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../build/config/',
+      '../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/webrtc_nonparallel_tests_apk/',
+      '<(PRODUCT_DIR)/bin/run_webrtc_nonparallel_tests',
+      'webrtc_nonparallel_tests.isolate',
+    ]
+  },
+}
diff --git a/chromium/third_party/webrtc/webrtc_perf_tests_apk.isolate b/chromium/third_party/webrtc/webrtc_perf_tests_apk.isolate
new file mode 100644
index 00000000000..d1be691a0b1
--- /dev/null
+++ b/chromium/third_party/webrtc/webrtc_perf_tests_apk.isolate
@@ -0,0 +1,26 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+{
+  'includes': [
+    '../build/android/android.isolate',
+    'webrtc_perf_tests.isolate',
+  ],
+  'variables': {
+    'command': [
+      '<(PRODUCT_DIR)/bin/run_webrtc_perf_tests',
+      '--logcat-output-dir', '${ISOLATED_OUTDIR}/logcats',
+    ],
+    'files': [
+      '../build/config/',
+      '../third_party/instrumented_libraries/instrumented_libraries.isolate',
+      '<(PRODUCT_DIR)/webrtc_perf_tests_apk/',
+      '<(PRODUCT_DIR)/bin/run_webrtc_perf_tests',
+      'webrtc_perf_tests.isolate',
+    ]
+  },
+}
diff --git a/chromium/third_party/webrtc/webrtc_tests.gypi b/chromium/third_party/webrtc/webrtc_tests.gypi
index 7840e80b5ed..ae31c6c33cd 100644
--- a/chromium/third_party/webrtc/webrtc_tests.gypi
+++ b/chromium/third_party/webrtc/webrtc_tests.gypi
@@ -30,11 +30,11 @@
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
           ],
         }],
-        ['OS=="ios"', {
+        ['OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")', {
           'dependencies': [
-            'api/api_tests.gyp:rtc_api_objc_tests',
-          ]
-        }]
+            'sdk/sdk_tests.gyp:rtc_sdk_peerconnection_objc_tests',
+          ],
+        }],
       ],
     },
     {
@@ -57,7 +57,6 @@
       ],
       'dependencies': [
         '<(DEPTH)/testing/gtest.gyp:gtest',
-        '<(webrtc_root)/modules/modules.gyp:video_render',
         '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         'webrtc',
@@ -144,7 +143,6 @@
         'test/test.gyp:test_common',
         'test/test.gyp:test_renderer',
         '<(webrtc_root)/modules/modules.gyp:video_capture',
-        '<(webrtc_root)/modules/modules.gyp:video_render',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
         'webrtc',
       ],
@@ -161,6 +159,7 @@
         'call/bitrate_estimator_tests.cc',
         'call/call_unittest.cc',
         'call/packet_injection_tests.cc',
+        'call/ringbuffer_unittest.cc',
         'test/common_unittest.cc',
         'test/testsupport/metrics/video_metrics_unittest.cc',
         'video/call_stats_unittest.cc',
@@ -169,6 +168,7 @@
         'video/overuse_frame_detector_unittest.cc',
         'video/payload_router_unittest.cc',
         'video/report_block_stats_unittest.cc',
+        'video/send_delay_stats_unittest.cc',
         'video/send_statistics_proxy_unittest.cc',
         'video/stream_synchronization_unittest.cc',
         'video/video_capture_input_unittest.cc',
@@ -183,7 +183,6 @@
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
         '<(webrtc_root)/modules/modules.gyp:video_capture',
-        '<(webrtc_root)/modules/modules.gyp:video_render',
         '<(webrtc_root)/test/test.gyp:channel_transport',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
         'test/metrics.gyp:metrics',
@@ -214,10 +213,12 @@
           ],
           'dependencies': [
             'webrtc.gyp:rtc_event_log',
+            'webrtc.gyp:rtc_event_log_parser',
             'webrtc.gyp:rtc_event_log_proto',
           ],
           'sources': [
             'call/rtc_event_log_unittest.cc',
+            'call/rtc_event_log_unittest_helper.cc'
           ],
         }],
       ],
@@ -343,6 +344,66 @@
           ],
         },
       ],
+      'conditions': [
+        ['test_isolation_mode != "noop"',
+          {
+            'targets': [
+              {
+                'target_name': 'rtc_unittests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):rtc_unittests_apk',
+                ],
+                'includes': [
+                  'build/isolate.gypi',
+                ],
+                'sources': [
+                  'rtc_unittests_apk.isolate',
+                ],
+              },
+              {
+                'target_name': 'video_engine_tests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):video_engine_tests_apk',
+                ],
+                'includes': [
+                  'build/isolate.gypi',
+                ],
+                'sources': [
+                  'video_engine_tests_apk.isolate',
+                ],
+              },
+              {
+                'target_name': 'webrtc_perf_tests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):webrtc_perf_tests_apk',
+                ],
+                'includes': [
+                  'build/isolate.gypi',
+                ],
+                'sources': [
+                  'webrtc_perf_tests_apk.isolate',
+                ],
+              },
+              {
+                'target_name': 'webrtc_nonparallel_tests_apk_run',
+                'type': 'none',
+                'dependencies': [
+                  '<(apk_tests_path):webrtc_nonparallel_tests_apk',
+                ],
+                'includes': [
+                  'build/isolate.gypi',
+                ],
+                'sources': [
+                  'webrtc_nonparallel_tests_apk.isolate',
+                ],
+              },
+            ],
+          },
+        ],
+      ],
     }],
     ['test_isolation_mode != "noop"', {
       'targets': [