BASELINE: Update chromium to 45.0.2454.40

Change-Id: Id2121d9f11a8fc633677236c65a3e41feef589e4
Reviewed-by: Andras Becsi <andras.becsi@theqtcompany.com>

144 files changed, 2906 insertions, 4872 deletions

diff --git a/chromium/third_party/libjingle/BUILD.gn b/chromium/third_party/libjingle/BUILD.gn
index ade2f66f00e..1d500d04eea 100644
--- a/chromium/third_party/libjingle/BUILD.gn
+++ b/chromium/third_party/libjingle/BUILD.gn
@@ -417,8 +417,6 @@ if (enable_webrtc) {
       "source/talk/media/base/constants.cc",
       "source/talk/media/base/constants.h",
       "source/talk/media/base/cryptoparams.h",
-      "source/talk/media/base/filemediaengine.cc",
-      "source/talk/media/base/filemediaengine.h",
       "source/talk/media/base/hybriddataengine.h",
       "source/talk/media/base/mediachannel.h",
       "source/talk/media/base/mediaengine.cc",
@@ -448,13 +446,10 @@ if (enable_webrtc) {
       "source/talk/media/webrtc/webrtccommon.h",
       "source/talk/media/webrtc/webrtcpassthroughrender.cc",
       "source/talk/media/webrtc/webrtcpassthroughrender.h",
-      "source/talk/media/webrtc/webrtcvideocapturer.cc",
-      "source/talk/media/webrtc/webrtcvideocapturer.h",
       "source/talk/media/webrtc/webrtcvideoframe.cc",
       "source/talk/media/webrtc/webrtcvideoframe.h",
       "source/talk/media/webrtc/webrtcvideoframefactory.cc",
       "source/talk/media/webrtc/webrtcvideoframefactory.h",
-      "source/talk/media/webrtc/webrtcvie.h",
       "source/talk/media/webrtc/webrtcvoe.h",
       "source/talk/session/media/audiomonitor.cc",
       "source/talk/session/media/audiomonitor.h",
@@ -517,8 +512,6 @@ if (enable_webrtc) {
       "source/talk/media/webrtc/simulcast.h",
       "source/talk/media/webrtc/webrtcmediaengine.cc",
       "source/talk/media/webrtc/webrtcmediaengine.h",
-      "source/talk/media/webrtc/webrtcvideoengine.cc",
-      "source/talk/media/webrtc/webrtcvideoengine.h",
       "source/talk/media/webrtc/webrtcvideoengine2.cc",
       "source/talk/media/webrtc/webrtcvideoengine2.h",
       "source/talk/media/webrtc/webrtcvoiceengine.cc",
@@ -539,5 +532,17 @@ if (enable_webrtc) {
       "//third_party/webrtc/voice_engine",
     ]
   }
+
+  source_set("libstunprober") {
+    p2p_dir = "../webrtc/p2p"
+    sources = [
+      "$p2p_dir/stunprober/stunprober.cc",
+    ]
+
+    deps = [
+      "//third_party/webrtc/base:rtc_base",
+      ":libjingle_webrtc_common",
+    ]
+  }
 }  # enable_webrtc
 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.
diff --git a/chromium/third_party/libjingle/README.chromium b/chromium/third_party/libjingle/README.chromium
index 6028000d3df..2078ca7792b 100644
--- a/chromium/third_party/libjingle/README.chromium
+++ b/chromium/third_party/libjingle/README.chromium
@@ -1,7 +1,7 @@
 Name: libjingle
 URL: http://code.google.com/p/webrtc/
 Version: unknown
-Revision: 9186
+Revision: 9564
 License: BSD
 License File: source/talk/COPYING
 Security Critical: yes
diff --git a/chromium/third_party/libjingle/libjingle.gyp b/chromium/third_party/libjingle/libjingle.gyp
index 7cda19660a0..aec101989c4 100644
--- a/chromium/third_party/libjingle/libjingle.gyp
+++ b/chromium/third_party/libjingle/libjingle.gyp
@@ -26,7 +26,6 @@
       'HAVE_SRTP',
       'HAVE_WEBRTC_VIDEO',
       'HAVE_WEBRTC_VOICE',
-      'LIBPEERCONNECTION_LIB=1',
       'LOGGING_INSIDE_WEBRTC',
       'NO_MAIN_THREAD_WRAPPING',
       'NO_SOUND_SYSTEM',
@@ -325,12 +324,7 @@
           # GN version: //third_party/libjingle:libjingle_webrtc_common
           'target_name': 'libjingle_webrtc_common',
           'type': 'static_library',
-          'all_dependent_settings': {
-            'defines': [ 'LIBPEERCONNECTION_LIB=1' ],
-          },
           'sources': [
-            'overrides/talk/media/webrtc/webrtcexport.h',
-
             '<(libjingle_source)/talk/app/webrtc/audiotrack.cc',
             '<(libjingle_source)/talk/app/webrtc/audiotrack.h',
             '<(libjingle_source)/talk/app/webrtc/audiotrackrenderer.cc',
@@ -407,8 +401,6 @@
             '<(libjingle_source)/talk/media/base/constants.cc',
             '<(libjingle_source)/talk/media/base/constants.h',
             '<(libjingle_source)/talk/media/base/cryptoparams.h',
-            '<(libjingle_source)/talk/media/base/filemediaengine.cc',
-            '<(libjingle_source)/talk/media/base/filemediaengine.h',
             '<(libjingle_source)/talk/media/base/hybriddataengine.h',
             '<(libjingle_source)/talk/media/base/mediachannel.h',
             '<(libjingle_source)/talk/media/base/mediaengine.cc',
@@ -438,13 +430,10 @@
             '<(libjingle_source)/talk/media/webrtc/webrtccommon.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcpassthroughrender.cc',
             '<(libjingle_source)/talk/media/webrtc/webrtcpassthroughrender.h',
-            '<(libjingle_source)/talk/media/webrtc/webrtcvideocapturer.cc',
-            '<(libjingle_source)/talk/media/webrtc/webrtcvideocapturer.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoframe.cc',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoframe.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoframefactory.cc',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoframefactory.h',
-            '<(libjingle_source)/talk/media/webrtc/webrtcvie.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcvoe.h',
             '<(libjingle_source)/talk/session/media/audiomonitor.cc',
             '<(libjingle_source)/talk/session/media/audiomonitor.h',
@@ -564,8 +553,6 @@
             '<(libjingle_source)/talk/media/webrtc/simulcast.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcmediaengine.cc',
             '<(libjingle_source)/talk/media/webrtc/webrtcmediaengine.h',
-            '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine.cc',
-            '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine2.cc',
             '<(libjingle_source)/talk/media/webrtc/webrtcvideoengine2.h',
             '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.cc',
diff --git a/chromium/third_party/libjingle/libjingle_nacl.gyp b/chromium/third_party/libjingle/libjingle_nacl.gyp
index edcbdd29223..1ca1e810d0c 100644
--- a/chromium/third_party/libjingle/libjingle_nacl.gyp
+++ b/chromium/third_party/libjingle/libjingle_nacl.gyp
@@ -156,6 +156,8 @@
         '<(webrtc_base)/pathutils.h',
         '<(webrtc_base)/physicalsocketserver.cc',
         '<(webrtc_base)/physicalsocketserver.h',
+        '<(webrtc_base)/platform_thread.cc',
+        '<(webrtc_base)/platform_thread.h',
         '<(webrtc_base)/proxydetect.cc',
         '<(webrtc_base)/proxydetect.h',
         '<(webrtc_base)/proxyinfo.cc',
diff --git a/chromium/third_party/libjingle/overrides/allocator_shim/allocator_proxy.cc b/chromium/third_party/libjingle/overrides/allocator_shim/allocator_proxy.cc
deleted file mode 100644
index 3a6772cc9df..00000000000
--- a/chromium/third_party/libjingle/overrides/allocator_shim/allocator_proxy.cc
+++ /dev/null
@@ -1,27 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "third_party/libjingle/overrides/allocator_shim/allocator_stub.h"
-
-#if !defined(LIBPEERCONNECTION_IMPLEMENTATION) || defined(LIBPEERCONNECTION_LIB)
-#error "Only compile the allocator proxy with the shared_library implementation"
-#endif
-
-#if defined(OS_MACOSX) || defined(OS_ANDROID)
-#error "The allocator proxy isn't supported (or needed) on mac or android."
-#endif
-
-extern AllocateFunction g_alloc;
-extern DellocateFunction g_dealloc;
-
-// Override the global new/delete routines and proxy them over to the allocator
-// routines handed to us via InitializeModule.
-
-void* operator new(std::size_t n) throw() {
-  return g_alloc(n);
-}
-
-void operator delete(void* p) throw() {
-  g_dealloc(p);
-}
diff --git a/chromium/third_party/libjingle/overrides/allocator_shim/allocator_stub.cc b/chromium/third_party/libjingle/overrides/allocator_shim/allocator_stub.cc
deleted file mode 100644
index 4664ed7cd09..00000000000
--- a/chromium/third_party/libjingle/overrides/allocator_shim/allocator_stub.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "third_party/libjingle/overrides/allocator_shim/allocator_stub.h"
-
-#if defined(OS_MACOSX) || defined(OS_ANDROID)
-#error "The allocator stub isn't supported (or needed) on mac or android."
-#endif
-
-void* Allocate(std::size_t n) {
-  return operator new(n);
-}
-
-void Dellocate(void* p) {
-  return operator delete(p);
-}
diff --git a/chromium/third_party/libjingle/overrides/allocator_shim/allocator_stub.h b/chromium/third_party/libjingle/overrides/allocator_shim/allocator_stub.h
deleted file mode 100644
index 20b9c7d42d6..00000000000
--- a/chromium/third_party/libjingle/overrides/allocator_shim/allocator_stub.h
+++ /dev/null
@@ -1,24 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#ifndef THIRD_PARTY_LIBJINGLE_OVERRIDES_ALLOCATOR_SHIM_ALLOCATOR_STUB_H_
-#define THIRD_PARTY_LIBJINGLE_OVERRIDES_ALLOCATOR_SHIM_ALLOCATOR_STUB_H_
-
-#include <new>
-
-#include "base/basictypes.h"
-
-#if !defined(OS_MACOSX) && !defined(OS_ANDROID)
-
-typedef void* (*AllocateFunction)(std::size_t);
-typedef void (*DellocateFunction)(void*);
-
-// The stub implementations that forward new / delete calls to the allocator
-// in the current binary (i.e. tcmalloc).
-void* Allocate(std::size_t n);
-void Dellocate(void* p);
-
-#endif  // OS_MACOSX && OS_ANDROID
-
-#endif  // THIRD_PARTY_LIBJINGLE_OVERRIDES_ALLOCATOR_SHIM_ALLOCATOR_STUB_H_
diff --git a/chromium/third_party/libjingle/overrides/init_webrtc.cc b/chromium/third_party/libjingle/overrides/init_webrtc.cc
index 3b2e4066a75..6ba7f04091b 100644
--- a/chromium/third_party/libjingle/overrides/init_webrtc.cc
+++ b/chromium/third_party/libjingle/overrides/init_webrtc.cc
@@ -7,6 +7,7 @@
 #include "base/command_line.h"
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
+#include "base/logging.h"
 #include "base/metrics/field_trial.h"
 #include "base/metrics/histogram.h"
 #include "base/native_library.h"
@@ -14,6 +15,7 @@
 #include "base/trace_event/trace_event.h"
 #include "third_party/webrtc/overrides/webrtc/base/basictypes.h"
 #include "third_party/webrtc/overrides/webrtc/base/logging.h"
+#include "third_party/webrtc/system_wrappers/interface/event_tracer.h"
 
 const unsigned char* GetCategoryGroupEnabled(const char* category_group) {
   return TRACE_EVENT_API_GET_CATEGORY_GROUP_ENABLED(category_group);
@@ -44,6 +46,12 @@ std::string FindFullName(const std::string& trial_name) {
 
 // Define webrtc::metrics functions to provide webrtc with implementations.
 namespace metrics {
+
+// This class doesn't actually exist, so don't go looking for it :)
+// This type is just fwd declared here in order to use it as an opaque type
+// between the Histogram functions in this file.
+class Histogram;
+
 Histogram* HistogramFactoryGetCounts(
     const std::string& name, int min, int max, int bucket_count) {
   return reinterpret_cast<Histogram*>(
@@ -69,8 +77,6 @@ void HistogramAdd(
 }  // namespace metrics
 }  // namespace webrtc
 
-#if defined(LIBPEERCONNECTION_LIB)
-
 // libpeerconnection is being compiled as a static lib.  In this case
 // we don't need to do any initializing but to keep things simple we
 // provide an empty intialization routine so that this #ifdef doesn't
@@ -79,115 +85,3 @@ bool InitializeWebRtcModule() {
   webrtc::SetupEventTracer(&GetCategoryGroupEnabled, &AddTraceEvent);
   return true;
 }
-
-#else  // !LIBPEERCONNECTION_LIB
-
-// When being compiled as a shared library, we need to bridge the gap between
-// the current module and the libpeerconnection module, so things get a tad
-// more complicated.
-
-// Global function pointers to the factory functions in the shared library.
-CreateWebRtcMediaEngineFunction g_create_webrtc_media_engine = NULL;
-DestroyWebRtcMediaEngineFunction g_destroy_webrtc_media_engine = NULL;
-
-// Returns the full or relative path to the libpeerconnection module depending
-// on what platform we're on.
-static base::FilePath GetLibPeerConnectionPath() {
-  base::FilePath path;
-  CHECK(PathService::Get(base::DIR_MODULE, &path));
-#if defined(OS_WIN)
-  path = path.Append(FILE_PATH_LITERAL("libpeerconnection.dll"));
-#elif defined(OS_MACOSX)
-  // Simulate '@loader_path/Libraries'.
-  path = path.Append(FILE_PATH_LITERAL("Libraries"))
-             .Append(FILE_PATH_LITERAL("libpeerconnection.so"));
-#elif defined(OS_ANDROID)
-  path = path.Append(FILE_PATH_LITERAL("libpeerconnection.so"));
-#else
-  path = path.Append(FILE_PATH_LITERAL("lib"))
-             .Append(FILE_PATH_LITERAL("libpeerconnection.so"));
-#endif
-  return path;
-}
-
-bool InitializeWebRtcModule() {
-  TRACE_EVENT0("webrtc", "InitializeWebRtcModule");
-
-  if (g_create_webrtc_media_engine)
-    return true;  // InitializeWebRtcModule has already been called.
-
-  base::FilePath path(GetLibPeerConnectionPath());
-  DVLOG(1) << "Loading WebRTC module: " << path.value();
-
-  base::NativeLibraryLoadError error;
-  static base::NativeLibrary lib = base::LoadNativeLibrary(path, &error);
-#if defined(OS_WIN)
-  // We've been seeing problems on Windows with loading the DLL and we're
-  // not sure exactly why.  It could be that AV programs are quarantining the
-  // file or disallowing loading the DLL. To get a better picture of the errors
-  // we're checking these specific error codes.
-  if (error.code == ERROR_MOD_NOT_FOUND) {
-    // It's possible that we get this error due to failure to load other
-    // dependencies, so check first that libpeerconnection actually exists.
-    CHECK(base::PathExists(path));  // libpeerconnection itself is missing.
-    CHECK(lib);  // If we hit this, a dependency is missing.
-  } else if (error.code == ERROR_ACCESS_DENIED) {
-    CHECK(lib);  // AV blocking access?
-  }
-#endif
-
-  // Catch-all error handler for all other sorts of errors.
-  CHECK(lib) << error.ToString();
-
-  InitializeModuleFunction initialize_module =
-      reinterpret_cast<InitializeModuleFunction>(
-          base::GetFunctionPointerFromNativeLibrary(
-              lib, "InitializeModule"));
-
-  // Initialize the proxy by supplying it with a pointer to our
-  // allocator/deallocator routines.
-  // On mac we use malloc zones, which are global, so we provide NULLs for
-  // the alloc/dealloc functions.
-  // PS: This function is actually implemented in allocator_proxy.cc with the
-  // new/delete overrides.
-  InitDiagnosticLoggingDelegateFunctionFunction init_diagnostic_logging = NULL;
-  bool init_ok = initialize_module(*base::CommandLine::ForCurrentProcess(),
-#if !defined(OS_MACOSX) && !defined(OS_ANDROID)
-      &Allocate,
-      &Dellocate,
-#endif
-      &webrtc::field_trial::FindFullName,
-      &webrtc::metrics::HistogramFactoryGetCounts,
-      &webrtc::metrics::HistogramFactoryGetEnumeration,
-      &webrtc::metrics::HistogramAdd,
-      logging::GetLogMessageHandler(),
-      &GetCategoryGroupEnabled,
-      &AddTraceEvent,
-      &g_create_webrtc_media_engine,
-      &g_destroy_webrtc_media_engine,
-      &init_diagnostic_logging);
-
-  if (init_ok)
-    rtc::SetExtraLoggingInit(init_diagnostic_logging);
-  return init_ok;
-}
-
-cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
-    webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
-    cricket::WebRtcVideoEncoderFactory* encoder_factory,
-    cricket::WebRtcVideoDecoderFactory* decoder_factory) {
-  // For convenience of tests etc, we call InitializeWebRtcModule here.
-  // For Chrome however, InitializeWebRtcModule must be called
-  // explicitly before the sandbox is initialized.  In that case, this call is
-  // effectively a noop.
-  InitializeWebRtcModule();
-  return g_create_webrtc_media_engine(adm, adm_sc, encoder_factory,
-      decoder_factory);
-}
-
-void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
-  g_destroy_webrtc_media_engine(media_engine);
-}
-
-#endif  // LIBPEERCONNECTION_LIB
diff --git a/chromium/third_party/libjingle/overrides/init_webrtc.h b/chromium/third_party/libjingle/overrides/init_webrtc.h
index 23a7f8a5533..4a7db383a15 100644
--- a/chromium/third_party/libjingle/overrides/init_webrtc.h
+++ b/chromium/third_party/libjingle/overrides/init_webrtc.h
@@ -5,82 +5,9 @@
 #ifndef THIRD_PARTY_LIBJINGLE_OVERRIDES_INIT_WEBRTC_H_
 #define THIRD_PARTY_LIBJINGLE_OVERRIDES_INIT_WEBRTC_H_
 
-#include <string>
-
-#include "third_party/libjingle/overrides/allocator_shim/allocator_stub.h"
-#include "base/logging.h"
-#include "third_party/webrtc/system_wrappers/interface/event_tracer.h"
-
-namespace base {
-class CommandLine;
-}
-
-namespace cricket {
-class MediaEngineInterface;
-class WebRtcVideoDecoderFactory;
-class WebRtcVideoEncoderFactory;
-}  // namespace cricket
-
-namespace webrtc {
-class AudioDeviceModule;
-namespace metrics {
-class Histogram;
-}  // namespace metrics
-}  // namespace webrtc
-
-typedef std::string (*FieldTrialFindFullName)(const std::string& trial_name);
-
-typedef webrtc::metrics::Histogram* (*RtcHistogramFactoryGetCounts)(
-    const std::string& name, int min, int max, int bucket_count);
-typedef webrtc::metrics::Histogram* (*RtcHistogramFactoryGetEnumeration)(
-    const std::string& name, int boundary);
-typedef void (*RtcHistogramAdd)(
-    webrtc::metrics::Histogram* histogram_pointer,
-    const std::string& name,
-    int sample);
-
-typedef cricket::MediaEngineInterface* (*CreateWebRtcMediaEngineFunction)(
-    webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
-    cricket::WebRtcVideoEncoderFactory* encoder_factory,
-    cricket::WebRtcVideoDecoderFactory* decoder_factory);
-
-typedef void (*DestroyWebRtcMediaEngineFunction)(
-    cricket::MediaEngineInterface* media_engine);
-
-typedef void (*InitDiagnosticLoggingDelegateFunctionFunction)(
-    void (*DelegateFunction)(const std::string&));
-
-// A typedef for the main initialize function in libpeerconnection.
-// This will initialize logging in the module with the proper arguments
-// as well as provide pointers back to a couple webrtc factory functions.
-// The reason we get pointers to these functions this way is to avoid having
-// to go through GetProcAddress et al and rely on specific name mangling.
-// TODO(tommi): The number of functions is growing. Use a struct.
-typedef bool (*InitializeModuleFunction)(
-    const base::CommandLine& command_line,
-#if !defined(OS_MACOSX) && !defined(OS_ANDROID)
-    AllocateFunction alloc,
-    DellocateFunction dealloc,
-#endif
-    FieldTrialFindFullName field_trial_find,
-    RtcHistogramFactoryGetCounts factory_get_counts,
-    RtcHistogramFactoryGetEnumeration factory_get_enumeration,
-    RtcHistogramAdd histogram_add,
-    logging::LogMessageHandlerFunction log_handler,
-    webrtc::GetCategoryEnabledPtr trace_get_category_enabled,
-    webrtc::AddTraceEventPtr trace_add_trace_event,
-    CreateWebRtcMediaEngineFunction* create_media_engine,
-    DestroyWebRtcMediaEngineFunction* destroy_media_engine,
-    InitDiagnosticLoggingDelegateFunctionFunction* init_diagnostic_logging);
-
-#if !defined(LIBPEERCONNECTION_IMPLEMENTATION)
-// Load and initialize the shared WebRTC module (libpeerconnection).
-// Call this explicitly to load and initialize the WebRTC module (e.g. before
-// initializing the sandbox in Chrome).
-// If not called explicitly, this function will still be called from the main
-// CreateWebRtcMediaEngine factory function the first time it is called.
+// Initialize WebRTC. Call this explicitly to initialize WebRTC module
+// (before initializing the sandbox in Chrome) and hook up Chrome+WebRTC
+// integration such as common logging and tracing.
 bool InitializeWebRtcModule();
-#endif
 
 #endif // THIRD_PARTY_LIBJINGLE_OVERRIDES_INIT_WEBRTC_H_
diff --git a/chromium/third_party/libjingle/overrides/initialize_module.cc b/chromium/third_party/libjingle/overrides/initialize_module.cc
deleted file mode 100644
index 89671c5252d..00000000000
--- a/chromium/third_party/libjingle/overrides/initialize_module.cc
+++ /dev/null
@@ -1,147 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#if defined(OS_WIN) && defined(ARCH_CPU_X86_64)
-#include <math.h> // needed for _set_FMA3_enable
-#endif  // WIN && ARCH_CPU_X86_64
-
-#include "base/command_line.h"
-#include "base/files/file_path.h"
-#include "base/logging.h"
-#include "third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.h"
-#include "third_party/libjingle/allocator_shim/allocator_stub.h"
-#include "third_party/libjingle/overrides/init_webrtc.h"
-#include "third_party/webrtc/base/basictypes.h"
-#include "third_party/webrtc/base/logging.h"
-
-#if !defined(LIBPEERCONNECTION_IMPLEMENTATION) || defined(LIBPEERCONNECTION_LIB)
-#error "Only compile the allocator proxy with the shared_library implementation"
-#endif
-
-#if defined(OS_WIN)
-#define ALLOC_EXPORT __declspec(dllexport)
-#else
-#define ALLOC_EXPORT __attribute__((visibility("default")))
-#endif
-
-#if !defined(OS_MACOSX) && !defined(OS_ANDROID)
-// These are used by our new/delete overrides in
-// allocator_shim/allocator_proxy.cc
-AllocateFunction g_alloc = NULL;
-DellocateFunction g_dealloc = NULL;
-#endif
-
-// Forward declare of the libjingle internal factory and destroy methods for the
-// WebRTC media engine.
-cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
-    webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
-    cricket::WebRtcVideoEncoderFactory* encoder_factory,
-    cricket::WebRtcVideoDecoderFactory* decoder_factory);
-
-void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine);
-
-namespace {
-// Provide webrtc with a field trial and metrics implementations.
-// The implementations are provided by the loader via the InitializeModule.
-
-// Defines webrtc::field_trial::FindFullName.
-FieldTrialFindFullName g_field_trial_find_ = NULL;
-// Defines webrtc::metrics::RtcFactoryGetCounts.
-RtcHistogramFactoryGetCounts g_factory_get_counts = NULL;
-// Defines webrtc::metrics::RtcFactoryGetEnumeration.
-RtcHistogramFactoryGetEnumeration g_factory_get_enumeration = NULL;
-// Defines webrtc::metrics::RtcAdd.
-RtcHistogramAdd g_histogram_add = NULL;
-}
-
-namespace webrtc {
-namespace field_trial {
-std::string FindFullName(const std::string& trial_name) {
-  return g_field_trial_find_(trial_name);
-}
-}  // namespace field_trial
-
-namespace metrics {
-Histogram* HistogramFactoryGetCounts(
-    const std::string& name, int min, int max, int bucket_count) {
-  return g_factory_get_counts(name, min, max, bucket_count);
-}
-
-Histogram* HistogramFactoryGetEnumeration(
-    const std::string& name, int boundary) {
-  return g_factory_get_enumeration(name, boundary);
-}
-
-void HistogramAdd(
-    Histogram* histogram_pointer, const std::string& name, int sample) {
-  g_histogram_add(histogram_pointer, name, sample);
-}
-}  // namespace metrics
-}  // namespace webrtc
-
-extern "C" {
-
-// Initialize logging, set the forward allocator functions (not on mac), and
-// return pointers to libjingle's WebRTC factory methods.
-// Called from init_webrtc.cc.
-ALLOC_EXPORT
-bool InitializeModule(const base::CommandLine& command_line,
-#if !defined(OS_MACOSX) && !defined(OS_ANDROID)
-                      AllocateFunction alloc,
-                      DellocateFunction dealloc,
-#endif
-                      FieldTrialFindFullName field_trial_find,
-                      RtcHistogramFactoryGetCounts factory_get_counts,
-                      RtcHistogramFactoryGetEnumeration factory_get_enumeration,
-                      RtcHistogramAdd histogram_add,
-                      logging::LogMessageHandlerFunction log_handler,
-                      webrtc::GetCategoryEnabledPtr trace_get_category_enabled,
-                      webrtc::AddTraceEventPtr trace_add_trace_event,
-                      CreateWebRtcMediaEngineFunction* create_media_engine,
-                      DestroyWebRtcMediaEngineFunction* destroy_media_engine,
-                      InitDiagnosticLoggingDelegateFunctionFunction*
-                          init_diagnostic_logging) {
-#if !defined(OS_MACOSX) && !defined(OS_ANDROID)
-  g_alloc = alloc;
-  g_dealloc = dealloc;
-#endif
-
-  g_field_trial_find_ = field_trial_find;
-  g_factory_get_counts = factory_get_counts;
-  g_factory_get_enumeration = factory_get_enumeration;
-  g_histogram_add = histogram_add;
-
-  *create_media_engine = &CreateWebRtcMediaEngine;
-  *destroy_media_engine = &DestroyWebRtcMediaEngine;
-  *init_diagnostic_logging = &rtc::InitDiagnosticLoggingDelegateFunction;
-
-#if defined(OS_WIN) && defined(ARCH_CPU_X86_64)
-  // VS2013 only checks the existence of FMA3 instructions, not the enabled-ness
-  // of them at the OS level (this is fixed in VS2015). We force off usage of
-  // FMA3 instructions in the CRT to avoid using that path and hitting illegal
-  // instructions when running on CPUs that support FMA3, but OSs that don't.
-  // See http://crbug.com/436603 and http://crbug.com/446983.
-  _set_FMA3_enable(0);
-#endif  // WIN && ARCH_CPU_X86_64
-
-  if (base::CommandLine::Init(0, NULL)) {
-#if !defined(OS_WIN)
-    // This is not needed on Windows since CommandLine::Init has already
-    // done the equivalent thing via the GetCommandLine() API.
-    base::CommandLine::ForCurrentProcess()->AppendArguments(command_line, true);
-#endif
-    logging::LoggingSettings settings;
-    settings.logging_dest = logging::LOG_TO_SYSTEM_DEBUG_LOG;
-    logging::InitLogging(settings);
-
-    // Override the log message handler to forward logs to chrome's handler.
-    logging::SetLogMessageHandler(log_handler);
-    webrtc::SetupEventTracer(trace_get_category_enabled,
-                             trace_add_trace_event);
-  }
-
-  return true;
-}
-}  // extern "C"
diff --git a/chromium/third_party/libjingle/overrides/talk/media/webrtc/webrtcexport.h b/chromium/third_party/libjingle/overrides/talk/media/webrtc/webrtcexport.h
deleted file mode 100644
index ffa60bb860c..00000000000
--- a/chromium/third_party/libjingle/overrides/talk/media/webrtc/webrtcexport.h
+++ /dev/null
@@ -1,21 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-// This file is overridden to disable exports and imports in libjingle
-// between the libpeerconnection and libjingle_webrtc targets.
-// TODO(tommi): Remove when a version of libjingle has been rolled in that
-// either removes this header file or offers an easy way to turn this off.
-
-#ifndef TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
-#define TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
-
-#ifndef NON_EXPORTED_BASE
-#define NON_EXPORTED_BASE(code) code
-#endif  // NON_EXPORTED_BASE
-
-#ifndef WRME_EXPORT
-#define WRME_EXPORT
-#endif
-
-#endif  // TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/androidtests/jni/Android.mk b/chromium/third_party/libjingle/source/talk/app/webrtc/androidtests/jni/Android.mk
deleted file mode 100644
index 8e80160039b..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/androidtests/jni/Android.mk
+++ /dev/null
@@ -1,2 +0,0 @@
-# This space intentionally left blank (required for Android build system).
-
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.cc
index 89ab4869699..65f883ef9d0 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.cc
@@ -89,10 +89,10 @@ class AndroidVideoCapturer::FrameFactory : public cricket::VideoFrameFactory {
     // |captured_frame_.data| is only guaranteed to be valid during the scope
     // of |AndroidVideoCapturer::OnIncomingFrame_w|.
     // Check that captured_frame is actually our frame.
-    DCHECK(captured_frame == &captured_frame_);
+    CHECK(captured_frame == &captured_frame_);
 
     if (!apply_rotation_ || captured_frame->rotation == kVideoRotation_0) {
-      DCHECK(captured_frame->fourcc == cricket::FOURCC_YV12);
+      CHECK(captured_frame->fourcc == cricket::FOURCC_YV12);
       const uint8_t* y_plane = static_cast<uint8_t*>(captured_frame_.data);
 
       // Android guarantees that the stride is a multiple of 16.
@@ -160,7 +160,7 @@ AndroidVideoCapturer::AndroidVideoCapturer(
   std::vector<cricket::VideoFormat> formats;
   for (Json::ArrayIndex i = 0; i < json_values.size(); ++i) {
       const Json::Value& json_value = json_values[i];
-      DCHECK(!json_value["width"].isNull() && !json_value["height"].isNull() &&
+      CHECK(!json_value["width"].isNull() && !json_value["height"].isNull() &&
              !json_value["framerate"].isNull());
       cricket::VideoFormat format(
           json_value["width"].asInt(),
@@ -173,16 +173,16 @@ AndroidVideoCapturer::AndroidVideoCapturer(
 }
 
 AndroidVideoCapturer::~AndroidVideoCapturer() {
-  DCHECK(!running_);
+  CHECK(!running_);
 }
 
 cricket::CaptureState AndroidVideoCapturer::Start(
     const cricket::VideoFormat& capture_format) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(!running_);
-
   LOG(LS_INFO) << " AndroidVideoCapturer::Start w = " << capture_format.width
                << " h = " << capture_format.height;
+  CHECK(thread_checker_.CalledOnValidThread());
+  CHECK(!running_);
+
   frame_factory_ = new AndroidVideoCapturer::FrameFactory(
       capture_format.width, capture_format.height, delegate_.get());
   set_frame_factory(frame_factory_);
@@ -197,9 +197,9 @@ cricket::CaptureState AndroidVideoCapturer::Start(
 }
 
 void AndroidVideoCapturer::Stop() {
-  DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << " AndroidVideoCapturer::Stop ";
-  DCHECK(running_);
+  CHECK(thread_checker_.CalledOnValidThread());
+  CHECK(running_);
   running_ = false;
   SetCaptureFormat(NULL);
 
@@ -209,18 +209,18 @@ void AndroidVideoCapturer::Stop() {
 }
 
 bool AndroidVideoCapturer::IsRunning() {
-  DCHECK(thread_checker_.CalledOnValidThread());
+  CHECK(thread_checker_.CalledOnValidThread());
   return running_;
 }
 
 bool AndroidVideoCapturer::GetPreferredFourccs(std::vector<uint32>* fourccs) {
-  DCHECK(thread_checker_.CalledOnValidThread());
+  CHECK(thread_checker_.CalledOnValidThread());
   fourccs->push_back(cricket::FOURCC_YV12);
   return true;
 }
 
 void AndroidVideoCapturer::OnCapturerStarted(bool success) {
-  DCHECK(thread_checker_.CalledOnValidThread());
+  CHECK(thread_checker_.CalledOnValidThread());
   cricket::CaptureState new_state =
       success ? cricket::CS_RUNNING : cricket::CS_FAILED;
   if (new_state == current_state_)
@@ -237,9 +237,18 @@ void AndroidVideoCapturer::OnIncomingFrame(void* frame_data,
                                            int length,
                                            int rotation,
                                            int64 time_stamp) {
-  DCHECK(thread_checker_.CalledOnValidThread());
+  CHECK(thread_checker_.CalledOnValidThread());
   frame_factory_->UpdateCapturedFrame(frame_data, length, rotation, time_stamp);
   SignalFrameCaptured(this, frame_factory_->GetCapturedFrame());
 }
 
+void AndroidVideoCapturer::OnOutputFormatRequest(
+    int width, int height, int fps) {
+  CHECK(thread_checker_.CalledOnValidThread());
+  const cricket::VideoFormat& current = video_adapter()->output_format();
+  cricket::VideoFormat format(
+      width, height, cricket::VideoFormat::FpsToInterval(fps), current.fourcc);
+  video_adapter()->OnOutputFormatRequest(format);
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.h b/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.h
index 2dffb9c1414..2cfbdd808a1 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/androidvideocapturer.h
@@ -74,6 +74,9 @@ class AndroidVideoCapturer : public cricket::VideoCapturer {
                        int rotation,
                        int64 time_stamp);
 
+  // Called from JNI to request a new video format.
+  void OnOutputFormatRequest(int width, int height, int fps);
+
   AndroidVideoCapturerDelegate* delegate() { return delegate_.get(); }
 
  private:
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc
index 1897b73fc88..690ee65d3b7 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc
@@ -109,24 +109,26 @@ DataChannel::DataChannel(
       state_(kConnecting),
       data_channel_type_(dct),
       provider_(provider),
-      waiting_for_open_ack_(false),
-      was_ever_writable_(false),
+      handshake_state_(kHandshakeInit),
       connected_to_provider_(false),
       send_ssrc_set_(false),
       receive_ssrc_set_(false),
+      writable_(false),
       send_ssrc_(0),
       receive_ssrc_(0) {
 }
 
 bool DataChannel::Init(const InternalDataChannelInit& config) {
-  if (data_channel_type_ == cricket::DCT_RTP &&
-      (config.reliable ||
-       config.id != -1 ||
-       config.maxRetransmits != -1 ||
-       config.maxRetransmitTime != -1)) {
-    LOG(LS_ERROR) << "Failed to initialize the RTP data channel due to "
-                  << "invalid DataChannelInit.";
-    return false;
+  if (data_channel_type_ == cricket::DCT_RTP) {
+    if (config.reliable ||
+        config.id != -1 ||
+        config.maxRetransmits != -1 ||
+        config.maxRetransmitTime != -1) {
+      LOG(LS_ERROR) << "Failed to initialize the RTP data channel due to "
+                    << "invalid DataChannelInit.";
+      return false;
+    }
+    handshake_state_ = kHandshakeReady;
   } else if (data_channel_type_ == cricket::DCT_SCTP) {
     if (config.id < -1 ||
         config.maxRetransmits < -1 ||
@@ -142,6 +144,18 @@ bool DataChannel::Init(const InternalDataChannelInit& config) {
     }
     config_ = config;
 
+    switch (config_.open_handshake_role) {
+    case webrtc::InternalDataChannelInit::kNone:  // pre-negotiated
+      handshake_state_ = kHandshakeReady;
+      break;
+    case webrtc::InternalDataChannelInit::kOpener:
+      handshake_state_ = kHandshakeShouldSendOpen;
+      break;
+    case webrtc::InternalDataChannelInit::kAcker:
+      handshake_state_ = kHandshakeShouldSendAck;
+      break;
+    };
+
     // Try to connect to the transport in case the transport channel already
     // exists.
     OnTransportChannelCreated();
@@ -298,7 +312,7 @@ void DataChannel::OnDataReceived(cricket::DataChannel* channel,
 
   if (params.type == cricket::DMT_CONTROL) {
     ASSERT(data_channel_type_ == cricket::DCT_SCTP);
-    if (!waiting_for_open_ack_) {
+    if (handshake_state_ != kHandshakeWaitingForAck) {
       // Ignore it if we are not expecting an ACK message.
       LOG(LS_WARNING) << "DataChannel received unexpected CONTROL message, "
                       << "sid = " << params.ssrc;
@@ -306,7 +320,7 @@ void DataChannel::OnDataReceived(cricket::DataChannel* channel,
     }
     if (ParseDataChannelOpenAckMessage(payload)) {
       // We can send unordered as soon as we receive the ACK message.
-      waiting_for_open_ack_ = false;
+      handshake_state_ = kHandshakeReady;
       LOG(LS_INFO) << "DataChannel received OPEN_ACK message, sid = "
                    << params.ssrc;
     } else {
@@ -323,11 +337,13 @@ void DataChannel::OnDataReceived(cricket::DataChannel* channel,
   // We can send unordered as soon as we receive any DATA message since the
   // remote side must have received the OPEN (and old clients do not send
   // OPEN_ACK).
-  waiting_for_open_ack_ = false;
+  if (handshake_state_ == kHandshakeWaitingForAck) {
+    handshake_state_ = kHandshakeReady;
+  }
 
   bool binary = (params.type == cricket::DMT_BINARY);
   rtc::scoped_ptr<DataBuffer> buffer(new DataBuffer(payload, binary));
-  if (was_ever_writable_ && observer_) {
+  if (state_ == kOpen && observer_) {
     observer_->OnMessage(*buffer.get());
   } else {
     if (queued_received_data_.byte_count() + payload.size() >
@@ -346,38 +362,14 @@ void DataChannel::OnDataReceived(cricket::DataChannel* channel,
 }
 
 void DataChannel::OnChannelReady(bool writable) {
+  writable_ = writable;
   if (!writable) {
     return;
   }
-  // Update the readyState and send the queued control message if the channel
-  // is writable for the first time; otherwise it means the channel was blocked
-  // for sending and now unblocked, so send the queued data now.
-  if (!was_ever_writable_) {
-    was_ever_writable_ = true;
-
-    if (data_channel_type_ == cricket::DCT_SCTP) {
-      rtc::Buffer payload;
-
-      if (config_.open_handshake_role == InternalDataChannelInit::kOpener) {
-        WriteDataChannelOpenMessage(label_, config_, &payload);
-        SendControlMessage(payload);
-      } else if (config_.open_handshake_role ==
-                     InternalDataChannelInit::kAcker) {
-        WriteDataChannelOpenAckMessage(&payload);
-        SendControlMessage(payload);
-      }
-    }
-
-    UpdateState();
-    ASSERT(queued_send_data_.Empty());
-  } else if (state_ == kOpen) {
-    // TODO(jiayl): Sending OPEN message here contradicts with the pre-condition
-    // that the readyState is open. According to the standard, the channel
-    // should not become open before the OPEN message is sent.
-    SendQueuedControlMessages();
 
-    SendQueuedDataMessages();
-  }
+  SendQueuedControlMessages();
+  SendQueuedDataMessages();
+  UpdateState();
 }
 
 void DataChannel::DoClose() {
@@ -391,20 +383,34 @@ void DataChannel::DoClose() {
 }
 
 void DataChannel::UpdateState() {
+  // UpdateState determines what to do from a few state variables.  Include
+  // all conditions required for each state transition here for
+  // clarity. OnChannelReady(true) will send any queued data and then invoke
+  // UpdateState().
   switch (state_) {
     case kConnecting: {
       if (send_ssrc_set_ == receive_ssrc_set_) {
         if (data_channel_type_ == cricket::DCT_RTP && !connected_to_provider_) {
           connected_to_provider_ = provider_->ConnectDataChannel(this);
         }
-        if (was_ever_writable_) {
-          // TODO(jiayl): Do not transition to kOpen if we failed to send the
-          // OPEN message.
-          SendQueuedControlMessages();
-          SetState(kOpen);
-          // If we have received buffers before the channel got writable.
-          // Deliver them now.
-          DeliverQueuedReceivedData();
+        if (connected_to_provider_) {
+          if (handshake_state_ == kHandshakeShouldSendOpen) {
+            rtc::Buffer payload;
+            WriteDataChannelOpenMessage(label_, config_, &payload);
+            SendControlMessage(payload);
+          } else if (handshake_state_ == kHandshakeShouldSendAck) {
+            rtc::Buffer payload;
+            WriteDataChannelOpenAckMessage(&payload);
+            SendControlMessage(payload);
+          }
+          if (writable_ &&
+              (handshake_state_ == kHandshakeReady ||
+               handshake_state_ == kHandshakeWaitingForAck)) {
+            SetState(kOpen);
+            // If we have received buffers before the channel got writable.
+            // Deliver them now.
+            DeliverQueuedReceivedData();
+          }
         }
       }
       break;
@@ -413,10 +419,14 @@ void DataChannel::UpdateState() {
       break;
     }
     case kClosing: {
-      DisconnectFromTransport();
+      if (queued_send_data_.Empty() && queued_control_data_.Empty()) {
+        if (connected_to_provider_) {
+          DisconnectFromProvider();
+        }
 
-      if (!send_ssrc_set_ && !receive_ssrc_set_) {
-        SetState(kClosed);
+        if (!connected_to_provider_ && !send_ssrc_set_ && !receive_ssrc_set_) {
+          SetState(kClosed);
+        }
       }
       break;
     }
@@ -435,7 +445,7 @@ void DataChannel::SetState(DataState state) {
   }
 }
 
-void DataChannel::DisconnectFromTransport() {
+void DataChannel::DisconnectFromProvider() {
   if (!connected_to_provider_)
     return;
 
@@ -448,7 +458,7 @@ void DataChannel::DisconnectFromTransport() {
 }
 
 void DataChannel::DeliverQueuedReceivedData() {
-  if (!was_ever_writable_ || !observer_) {
+  if (!observer_) {
     return;
   }
 
@@ -460,8 +470,13 @@ void DataChannel::DeliverQueuedReceivedData() {
 }
 
 void DataChannel::SendQueuedDataMessages() {
-  ASSERT(was_ever_writable_ && state_ == kOpen);
+  if (queued_send_data_.Empty()) {
+    return;
+  }
 
+  ASSERT(state_ == kOpen || state_ == kClosing);
+
+  uint64 start_buffered_amount = buffered_amount();
   while (!queued_send_data_.Empty()) {
     DataBuffer* buffer = queued_send_data_.Front();
     if (!SendDataMessage(*buffer, false)) {
@@ -471,6 +486,10 @@ void DataChannel::SendQueuedDataMessages() {
     queued_send_data_.Pop();
     delete buffer;
   }
+
+  if (observer_ && buffered_amount() < start_buffered_amount) {
+    observer_->OnBufferedAmountChange(start_buffered_amount);
+  }
 }
 
 bool DataChannel::SendDataMessage(const DataBuffer& buffer,
@@ -479,8 +498,8 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
 
   if (data_channel_type_ == cricket::DCT_SCTP) {
     send_params.ordered = config_.ordered;
-    // Send as ordered if it is waiting for the OPEN_ACK message.
-    if (waiting_for_open_ack_ && !config_.ordered) {
+    // Send as ordered if it is still going through OPEN/ACK signaling.
+    if (handshake_state_ != kHandshakeReady && !config_.ordered) {
       send_params.ordered = true;
       LOG(LS_VERBOSE) << "Sending data as ordered for unordered DataChannel "
                       << "because the OPEN_ACK message has not been received.";
@@ -520,17 +539,21 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
 }
 
 bool DataChannel::QueueSendDataMessage(const DataBuffer& buffer) {
-  if (queued_send_data_.byte_count() >= kMaxQueuedSendDataBytes) {
+  size_t start_buffered_amount = buffered_amount();
+  if (start_buffered_amount >= kMaxQueuedSendDataBytes) {
     LOG(LS_ERROR) << "Can't buffer any more data for the data channel.";
     return false;
   }
   queued_send_data_.Push(new DataBuffer(buffer));
+
+  // The buffer can have length zero, in which case there is no change.
+  if (observer_ && buffered_amount() > start_buffered_amount) {
+    observer_->OnBufferedAmountChange(start_buffered_amount);
+  }
   return true;
 }
 
 void DataChannel::SendQueuedControlMessages() {
-  ASSERT(was_ever_writable_);
-
   PacketQueue control_packets;
   control_packets.Swap(&queued_control_data_);
 
@@ -546,16 +569,18 @@ void DataChannel::QueueControlMessage(const rtc::Buffer& buffer) {
 }
 
 bool DataChannel::SendControlMessage(const rtc::Buffer& buffer) {
-  bool is_open_message =
-      (config_.open_handshake_role == InternalDataChannelInit::kOpener);
+  bool is_open_message = handshake_state_ == kHandshakeShouldSendOpen;
 
   ASSERT(data_channel_type_ == cricket::DCT_SCTP &&
-         was_ever_writable_ &&
+         writable_ &&
          config_.id >= 0 &&
          (!is_open_message || !config_.negotiated));
 
   cricket::SendDataParams send_params;
   send_params.ssrc = config_.id;
+  // Send data as ordered before we receive any message from the remote peer to
+  // make sure the remote peer will not receive any data before it receives the
+  // OPEN message.
   send_params.ordered = config_.ordered || is_open_message;
   send_params.type = cricket::DMT_CONTROL;
 
@@ -564,11 +589,10 @@ bool DataChannel::SendControlMessage(const rtc::Buffer& buffer) {
   if (retval) {
     LOG(LS_INFO) << "Sent CONTROL message on channel " << config_.id;
 
-    if (is_open_message) {
-      // Send data as ordered before we receive any message from the remote peer
-      // to make sure the remote peer will not receive any data before it
-      // receives the OPEN message.
-      waiting_for_open_ack_ = true;
+    if (handshake_state_ == kHandshakeShouldSendAck) {
+      handshake_state_ = kHandshakeReady;
+    } else if (handshake_state_ == kHandshakeShouldSendOpen) {
+      handshake_state_ = kHandshakeWaitingForAck;
     }
   } else if (send_result == cricket::SDR_BLOCK) {
     QueueControlMessage(buffer);
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h
index fe8fac1c67f..8e58d0664b2 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h
@@ -204,11 +204,20 @@ class DataChannel : public DataChannelInterface,
     size_t byte_count_;
   };
 
+  // The OPEN(_ACK) signaling state.
+  enum HandshakeState {
+    kHandshakeInit,
+    kHandshakeShouldSendOpen,
+    kHandshakeShouldSendAck,
+    kHandshakeWaitingForAck,
+    kHandshakeReady
+  };
+
   bool Init(const InternalDataChannelInit& config);
   void DoClose();
   void UpdateState();
   void SetState(DataState state);
-  void DisconnectFromTransport();
+  void DisconnectFromProvider();
 
   void DeliverQueuedReceivedData();
 
@@ -226,11 +235,11 @@ class DataChannel : public DataChannelInterface,
   DataState state_;
   cricket::DataChannelType data_channel_type_;
   DataChannelProviderInterface* provider_;
-  bool waiting_for_open_ack_;
-  bool was_ever_writable_;
+  HandshakeState handshake_state_;
   bool connected_to_provider_;
   bool send_ssrc_set_;
   bool receive_ssrc_set_;
+  bool writable_;
   uint32 send_ssrc_;
   uint32 receive_ssrc_;
   // Control messages that always have to get sent out before any queued
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc
index ab5dbe9a1b5..e3c290bd9b5 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc
@@ -35,12 +35,18 @@ using webrtc::DataChannel;
 class FakeDataChannelObserver : public webrtc::DataChannelObserver {
  public:
   FakeDataChannelObserver()
-      : messages_received_(0), on_state_change_count_(0) {}
+      : messages_received_(0),
+        on_state_change_count_(0),
+        on_buffered_amount_change_count_(0) {}
 
   void OnStateChange() {
     ++on_state_change_count_;
   }
 
+  void OnBufferedAmountChange(uint64 previous_amount) {
+    ++on_buffered_amount_change_count_;
+  }
+
   void OnMessage(const webrtc::DataBuffer& buffer) {
     ++messages_received_;
   }
@@ -53,13 +59,22 @@ class FakeDataChannelObserver : public webrtc::DataChannelObserver {
     on_state_change_count_ = 0;
   }
 
+  void ResetOnBufferedAmountChangeCount() {
+    on_buffered_amount_change_count_ = 0;
+  }
+
   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:
   size_t messages_received_;
   size_t on_state_change_count_;
+  size_t on_buffered_amount_change_count_;
 };
 
 class SctpDataChannelTest : public testing::Test {
@@ -133,11 +148,13 @@ TEST_F(SctpDataChannelTest, StateTransition) {
 // Tests that DataChannel::buffered_amount() is correct after the channel is
 // blocked.
 TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) {
+  AddObserver();
   SetChannelReady();
   webrtc::DataBuffer buffer("abcd");
   EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
 
   EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount());
+  EXPECT_EQ(0U, observer_->on_buffered_amount_change_count());
 
   provider_.set_send_blocked(true);
 
@@ -147,37 +164,46 @@ TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) {
   }
   EXPECT_EQ(buffer.data.size() * number_of_packets,
             webrtc_data_channel_->buffered_amount());
+  EXPECT_EQ(number_of_packets, observer_->on_buffered_amount_change_count());
 }
 
 // Tests that the queued data are sent when the channel transitions from blocked
 // to unblocked.
 TEST_F(SctpDataChannelTest, QueuedDataSentWhenUnblocked) {
+  AddObserver();
   SetChannelReady();
   webrtc::DataBuffer buffer("abcd");
   provider_.set_send_blocked(true);
   EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
 
+  EXPECT_EQ(1U, observer_->on_buffered_amount_change_count());
+
   provider_.set_send_blocked(false);
   SetChannelReady();
   EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount());
+  EXPECT_EQ(2U, observer_->on_buffered_amount_change_count());
 }
 
 // Tests that no crash when the channel is blocked right away while trying to
 // send queued data.
 TEST_F(SctpDataChannelTest, BlockedWhenSendQueuedDataNoCrash) {
+  AddObserver();
   SetChannelReady();
   webrtc::DataBuffer buffer("abcd");
   provider_.set_send_blocked(true);
   EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
+  EXPECT_EQ(1U, observer_->on_buffered_amount_change_count());
 
   // Set channel ready while it is still blocked.
   SetChannelReady();
   EXPECT_EQ(buffer.size(), webrtc_data_channel_->buffered_amount());
+  EXPECT_EQ(1U, observer_->on_buffered_amount_change_count());
 
   // Unblock the channel to send queued data again, there should be no crash.
   provider_.set_send_blocked(false);
   SetChannelReady();
   EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount());
+  EXPECT_EQ(2U, observer_->on_buffered_amount_change_count());
 }
 
 // Tests that the queued control message is sent when channel is ready.
@@ -269,6 +295,41 @@ TEST_F(SctpDataChannelTest, SendUnorderedAfterReceiveData) {
   EXPECT_FALSE(provider_.last_send_data_params().ordered);
 }
 
+// Tests that the channel can't open until it's successfully sent the OPEN
+// message.
+TEST_F(SctpDataChannelTest, OpenWaitsForOpenMesssage) {
+  webrtc::DataBuffer buffer("foo");
+
+  provider_.set_send_blocked(true);
+  SetChannelReady();
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting,
+            webrtc_data_channel_->state());
+  provider_.set_send_blocked(false);
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen,
+                 webrtc_data_channel_->state(), 1000);
+  EXPECT_EQ(cricket::DMT_CONTROL, provider_.last_send_data_params().type);
+}
+
+// Tests that close first makes sure all queued data gets sent.
+TEST_F(SctpDataChannelTest, QueuedCloseFlushes) {
+  webrtc::DataBuffer buffer("foo");
+
+  provider_.set_send_blocked(true);
+  SetChannelReady();
+  EXPECT_EQ(webrtc::DataChannelInterface::kConnecting,
+            webrtc_data_channel_->state());
+  provider_.set_send_blocked(false);
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen,
+                 webrtc_data_channel_->state(), 1000);
+  provider_.set_send_blocked(true);
+  webrtc_data_channel_->Send(buffer);
+  webrtc_data_channel_->Close();
+  provider_.set_send_blocked(false);
+  EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed,
+                 webrtc_data_channel_->state(), 1000);
+  EXPECT_EQ(cricket::DMT_TEXT, provider_.last_send_data_params().type);
+}
+
 // Tests that messages are sent with the right ssrc.
 TEST_F(SctpDataChannelTest, SendDataSsrc) {
   webrtc_data_channel_->SetSctpSid(1);
@@ -369,8 +430,9 @@ TEST_F(SctpDataChannelTest, ClosedWhenSendBufferFull) {
     EXPECT_TRUE(webrtc_data_channel_->Send(packet));
   }
 
-  EXPECT_EQ(webrtc::DataChannelInterface::kClosed,
-            webrtc_data_channel_->state());
+  EXPECT_TRUE(
+      webrtc::DataChannelInterface::kClosed == webrtc_data_channel_->state() ||
+      webrtc::DataChannelInterface::kClosing == webrtc_data_channel_->state());
 }
 
 // Tests that the DataChannel is closed on transport errors.
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h
index 63122629f56..90573ebbf3e 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h
@@ -91,6 +91,8 @@ class DataChannelObserver {
   virtual void OnStateChange() = 0;
   //  A data buffer was successfully received.
   virtual void OnMessage(const DataBuffer& buffer) = 0;
+  // The data channel's buffered_amount has changed.
+  virtual void OnBufferedAmountChange(uint64 previous_amount){};
 
  protected:
   virtual ~DataChannelObserver() {}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.cc
index dd9bc5d68b5..554a84a06f1 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.cc
@@ -31,7 +31,6 @@
 #include "webrtc/base/logging.h"
 
 using webrtc::DTLSIdentityRequestObserver;
-using webrtc::WebRtcSessionDescriptionFactory;
 
 namespace webrtc {
 
@@ -57,13 +56,14 @@ class DtlsIdentityStore::WorkerTask : public sigslot::has_slots<>,
   explicit WorkerTask(DtlsIdentityStore* store)
       : signaling_thread_(rtc::Thread::Current()), store_(store) {
     store_->SignalDestroyed.connect(this, &WorkerTask::OnStoreDestroyed);
-  };
+  }
 
   virtual ~WorkerTask() { DCHECK(rtc::Thread::Current() == signaling_thread_); }
 
-  void GenerateIdentity() {
+ private:
+  void GenerateIdentity_w() {
     rtc::scoped_ptr<rtc::SSLIdentity> identity(
-      rtc::SSLIdentity::Generate(DtlsIdentityStore::kIdentityName));
+        rtc::SSLIdentity::Generate(DtlsIdentityStore::kIdentityName));
 
     {
       rtc::CritScope cs(&cs_);
@@ -76,27 +76,29 @@ class DtlsIdentityStore::WorkerTask : public sigslot::has_slots<>,
   void OnMessage(rtc::Message* msg) override {
     switch (msg->message_id) {
       case MSG_GENERATE_IDENTITY:
-        GenerateIdentity();
+        // This message always runs on the worker thread.
+        GenerateIdentity_w();
 
         // Must delete |this|, owned by msg->pdata, on the signaling thread to
         // avoid races on disconnecting the signal.
         signaling_thread_->Post(this, MSG_DESTROY, msg->pdata);
         break;
       case MSG_DESTROY:
+        DCHECK(rtc::Thread::Current() == signaling_thread_);
         delete msg->pdata;
+        // |this| has now been deleted. Don't touch member variables.
         break;
       default:
         CHECK(false) << "Unexpected message type";
     }
   }
 
- private:
   void OnStoreDestroyed() {
     rtc::CritScope cs(&cs_);
     store_ = NULL;
   }
 
-  rtc::Thread* signaling_thread_;
+  rtc::Thread* const signaling_thread_;
   rtc::CriticalSection cs_;
   DtlsIdentityStore* store_;
 };
@@ -116,6 +118,7 @@ DtlsIdentityStore::~DtlsIdentityStore() {
 }
 
 void DtlsIdentityStore::Initialize() {
+  DCHECK(rtc::Thread::Current() == signaling_thread_);
   // Do not aggressively generate the free identity if the worker thread and the
   // signaling thread are the same.
   if (worker_thread_ != signaling_thread_) {
@@ -139,6 +142,7 @@ void DtlsIdentityStore::RequestIdentity(DTLSIdentityRequestObserver* observer) {
 }
 
 void DtlsIdentityStore::OnMessage(rtc::Message* msg) {
+  DCHECK(rtc::Thread::Current() == signaling_thread_);
   switch (msg->message_id) {
     case MSG_GENERATE_IDENTITY_RESULT: {
       rtc::scoped_ptr<IdentityResultMessageData> pdata(
@@ -156,10 +160,12 @@ void DtlsIdentityStore::OnMessage(rtc::Message* msg) {
 }
 
 bool DtlsIdentityStore::HasFreeIdentityForTesting() const {
-  return free_identity_.get();
+  DCHECK(rtc::Thread::Current() == signaling_thread_);
+  return free_identity_.get() != nullptr;
 }
 
 void DtlsIdentityStore::GenerateIdentity() {
+  DCHECK(rtc::Thread::Current() == signaling_thread_);
   pending_jobs_++;
   LOG(LS_VERBOSE) << "New DTLS identity generation is posted, "
                   << "pending_identities=" << pending_jobs_;
@@ -191,6 +197,7 @@ void DtlsIdentityStore::OnIdentityGenerated(
 
 void DtlsIdentityStore::ReturnIdentity(
     rtc::scoped_ptr<rtc::SSLIdentity> identity) {
+  DCHECK(rtc::Thread::Current() == signaling_thread_);
   DCHECK(!free_identity_.get());
   DCHECK(!pending_observers_.empty());
 
@@ -211,8 +218,8 @@ void DtlsIdentityStore::ReturnIdentity(
   if (worker_thread_ != signaling_thread_ &&
       pending_observers_.empty() &&
       pending_jobs_ == 0) {
-      // Generate a free identity in the background.
-      GenerateIdentity();
+    // Generate a free identity in the background.
+    GenerateIdentity();
   }
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.h b/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.h
index 1ceaa825885..b2a797462fb 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.h
@@ -79,8 +79,8 @@ class DtlsIdentityStore : public rtc::MessageHandler {
 
   void PostGenerateIdentityResult_w(rtc::scoped_ptr<rtc::SSLIdentity> identity);
 
-  rtc::Thread* signaling_thread_;
-  rtc::Thread* worker_thread_;
+  rtc::Thread* const signaling_thread_;
+  rtc::Thread* const worker_thread_;
 
   // These members should be accessed on the signaling thread only.
   int pending_jobs_;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore_unittest.cc
index c0b204a85f9..12f58feea56 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore_unittest.cc
@@ -33,7 +33,6 @@
 #include "webrtc/base/ssladapter.h"
 
 using webrtc::DtlsIdentityStore;
-using webrtc::WebRtcSessionDescriptionFactory;
 
 static const int kTimeoutMs = 10000;
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/fakemetricsobserver.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/fakemetricsobserver.cc
new file mode 100644
index 00000000000..66e1c51f2a4
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/fakemetricsobserver.cc
@@ -0,0 +1,81 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+#include "talk/app/webrtc/fakemetricsobserver.h"
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+FakeMetricsObserver::FakeMetricsObserver() {
+  Reset();
+}
+
+void FakeMetricsObserver::Reset() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  memset(counters_, 0, sizeof(counters_));
+  memset(int_histogram_samples_, 0, sizeof(int_histogram_samples_));
+  for (std::string& type : string_histogram_samples_) {
+    type.clear();
+  }
+}
+
+void FakeMetricsObserver::IncrementCounter(PeerConnectionMetricsCounter type) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  ++counters_[type];
+}
+
+void FakeMetricsObserver::AddHistogramSample(PeerConnectionMetricsName type,
+    int value) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK_EQ(int_histogram_samples_[type], 0);
+  int_histogram_samples_[type] = value;
+}
+
+void FakeMetricsObserver::AddHistogramSample(PeerConnectionMetricsName type,
+    const std::string& value) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  string_histogram_samples_[type].assign(value);
+}
+
+int FakeMetricsObserver::GetCounter(PeerConnectionMetricsCounter type) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return counters_[type];
+}
+
+int FakeMetricsObserver::GetIntHistogramSample(
+    PeerConnectionMetricsName type) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return int_histogram_samples_[type];
+}
+
+const std::string& FakeMetricsObserver::GetStringHistogramSample(
+    PeerConnectionMetricsName type) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return string_histogram_samples_[type];
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/fakemetricsobserver.h b/chromium/third_party/libjingle/source/talk/app/webrtc/fakemetricsobserver.h
new file mode 100644
index 00000000000..e9e49749bf1
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/fakemetricsobserver.h
@@ -0,0 +1,68 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+#ifndef TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
+#define TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
+
+#include <map>
+#include <string>
+
+#include "talk/app/webrtc/peerconnectioninterface.h"
+#include "webrtc/base/thread_checker.h"
+
+namespace webrtc {
+
+class FakeMetricsObserver : public MetricsObserverInterface {
+ public:
+  FakeMetricsObserver();
+  void Reset();
+
+  void IncrementCounter(PeerConnectionMetricsCounter type) override;
+  void AddHistogramSample(PeerConnectionMetricsName type,
+                          int value) override;
+  void AddHistogramSample(PeerConnectionMetricsName type,
+                          const std::string& value) override;
+
+  // Accessors to be used by the tests.
+  int GetCounter(PeerConnectionMetricsCounter type) const;
+  int GetIntHistogramSample(PeerConnectionMetricsName type) const;
+  const std::string& GetStringHistogramSample(
+      PeerConnectionMetricsName type) const;
+
+ protected:
+  ~FakeMetricsObserver() {}
+
+ private:
+  rtc::ThreadChecker thread_checker_;
+  int counters_[kPeerConnectionMetricsCounter_Max];
+  int int_histogram_samples_[kPeerConnectionMetricsCounter_Max];
+  std::string string_histogram_samples_[kPeerConnectionMetricsName_Max];
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/androidmediacodeccommon.h b/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/androidmediacodeccommon.h
index 23f6c527574..d9a3ebef988 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/androidmediacodeccommon.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/androidmediacodeccommon.h
@@ -92,6 +92,18 @@ static inline jobject JavaEnumFromIndex(
                            state_class, index);
 }
 
+// Checks for any Java exception, prints stack backtrace and clears
+// currently thrown exception.
+static inline bool CheckException(JNIEnv* jni) {
+  if (jni->ExceptionCheck()) {
+    ALOGE("Java JNI exception.");
+    jni->ExceptionDescribe();
+    jni->ExceptionClear();
+    return true;
+  }
+  return false;
+}
+
 }  // namespace webrtc_jni
 
 #endif  // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/classreferenceholder.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/classreferenceholder.cc
index 7ff48b972c5..2c4f1e6358b 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/classreferenceholder.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/classreferenceholder.cc
@@ -73,17 +73,18 @@ ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni) {
   LoadClass(jni, "android/graphics/SurfaceTexture");
   LoadClass(jni, "org/webrtc/VideoCapturerAndroid");
   LoadClass(jni, "org/webrtc/VideoCapturerAndroid$NativeObserver");
+  LoadClass(jni, "org/webrtc/EglBase");
   LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder");
   LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo");
   LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$VideoCodecType");
   LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder");
   LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecoderOutputBufferInfo");
   LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$VideoCodecType");
-  jclass j_decoder_class = GetClass("org/webrtc/MediaCodecVideoDecoder");
+  jclass j_egl_base_class = GetClass("org/webrtc/EglBase");
   jmethodID j_is_egl14_supported_method = jni->GetStaticMethodID(
-      j_decoder_class, "isEGL14Supported", "()Z");
+      j_egl_base_class, "isEGL14Supported", "()Z");
   bool is_egl14_supported = jni->CallStaticBooleanMethod(
-      j_decoder_class, j_is_egl14_supported_method);
+      j_egl_base_class, j_is_egl14_supported_method);
   CHECK_EXCEPTION(jni);
   if (is_egl14_supported) {
     LoadClass(jni, "android/opengl/EGLContext");
@@ -93,6 +94,7 @@ ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni) {
   LoadClass(jni, "org/webrtc/MediaStream");
   LoadClass(jni, "org/webrtc/MediaStreamTrack$State");
   LoadClass(jni, "org/webrtc/PeerConnection$BundlePolicy");
+  LoadClass(jni, "org/webrtc/PeerConnection$RtcpMuxPolicy");
   LoadClass(jni, "org/webrtc/PeerConnection$IceConnectionState");
   LoadClass(jni, "org/webrtc/PeerConnection$IceGatheringState");
   LoadClass(jni, "org/webrtc/PeerConnection$IceTransportsType");
@@ -143,4 +145,3 @@ jclass FindClass(JNIEnv* jni, const char* name) {
 }
 
 }  // namespace webrtc_jni
-
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/native_handle_impl.h b/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/native_handle_impl.h
index 8c876961187..cdb72ff4d59 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/native_handle_impl.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/native_handle_impl.h
@@ -29,6 +29,9 @@
 #ifndef TALK_APP_WEBRTC_JAVA_JNI_NATIVE_HANDLE_IMPL_H_
 #define TALK_APP_WEBRTC_JAVA_JNI_NATIVE_HANDLE_IMPL_H_
 
+#include "webrtc/base/checks.h"
+#include "webrtc/common_video/interface/video_frame_buffer.h"
+
 namespace webrtc_jni {
 
 // Wrapper for texture object.
@@ -52,6 +55,23 @@ class NativeHandleImpl {
   int32_t texture_id_;
 };
 
+class JniNativeHandleBuffer : public webrtc::NativeHandleBuffer {
+ public:
+  JniNativeHandleBuffer(void* native_handle, int width, int height)
+      : NativeHandleBuffer(native_handle, width, height) {}
+
+  // TODO(pbos): Override destructor to release native handle, at the moment the
+  // native handle is not released based on refcount.
+
+ private:
+  rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override {
+    // TODO(pbos): Implement before using this in the encoder pipeline (or
+    // remove the CHECK() in VideoCapture).
+    RTC_NOTREACHED();
+    return nullptr;
+  }
+};
+
 }  // namespace webrtc_jni
 
 #endif  // TALK_APP_WEBRTC_JAVA_JNI_NATIVE_HANDLE_IMPL_H_
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription.cc
index 60467ba620a..697f332c27c 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription.cc
@@ -62,10 +62,11 @@ const int JsepSessionDescription::kDefaultVideoCodecId = 100;
 const int JsepSessionDescription::kDefaultVideoCodecFramerate = 60;
 const char JsepSessionDescription::kDefaultVideoCodecName[] = "VP8";
 // Used as default max video codec size before we have it in signaling.
-#if defined(ANDROID)
+#if defined(ANDROID) || defined(WEBRTC_IOS)
 // Limit default max video codec size for Android to avoid
 // HW VP8 codec initialization failure for resolutions higher
 // than 1280x720 or 720x1280.
+// Same patch for iOS to support 720P in portrait mode.
 const int JsepSessionDescription::kMaxVideoCodecWidth = 1280;
 const int JsepSessionDescription::kMaxVideoCodecHeight = 1280;
 #else
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc
index b37d1e3e41f..63c6f13a3d7 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc
@@ -43,43 +43,43 @@ namespace {
 // invalid.
 void FromConstraints(const MediaConstraintsInterface::Constraints& constraints,
                      cricket::AudioOptions* options) {
-  MediaConstraintsInterface::Constraints::const_iterator iter;
-
   // This design relies on the fact that all the audio constraints are actually
   // "options", i.e. boolean-valued and always satisfiable.  If the constraints
   // are extended to include non-boolean values or actual format constraints,
   // a different algorithm will be required.
-  for (iter = constraints.begin(); iter != constraints.end(); ++iter) {
+  struct {
+    const char* name;
+    cricket::Settable<bool>& value;
+  } key_to_value[] = {
+      {MediaConstraintsInterface::kGoogEchoCancellation,
+       options->echo_cancellation},
+      {MediaConstraintsInterface::kExtendedFilterEchoCancellation,
+       options->extended_filter_aec},
+      {MediaConstraintsInterface::kDAEchoCancellation,
+       options->delay_agnostic_aec},
+      {MediaConstraintsInterface::kAutoGainControl, options->auto_gain_control},
+      {MediaConstraintsInterface::kExperimentalAutoGainControl,
+       options->experimental_agc},
+      {MediaConstraintsInterface::kNoiseSuppression,
+       options->noise_suppression},
+      {MediaConstraintsInterface::kExperimentalNoiseSuppression,
+       options->experimental_ns},
+      {MediaConstraintsInterface::kHighpassFilter, options->highpass_filter},
+      {MediaConstraintsInterface::kTypingNoiseDetection,
+       options->typing_detection},
+      {MediaConstraintsInterface::kAudioMirroring, options->stereo_swapping},
+      {MediaConstraintsInterface::kAecDump, options->aec_dump}
+  };
+
+  for (const auto& constraint : constraints) {
     bool value = false;
-
-    if (!rtc::FromString(iter->value, &value))
+    if (!rtc::FromString(constraint.value, &value))
       continue;
 
-    if (iter->key == MediaConstraintsInterface::kEchoCancellation)
-      options->echo_cancellation.Set(value);
-    else if (iter->key ==
-        MediaConstraintsInterface::kExperimentalEchoCancellation)
-      options->experimental_aec.Set(value);
-    else if (iter->key == MediaConstraintsInterface::kDAEchoCancellation)
-      options->delay_agnostic_aec.Set(value);
-    else if (iter->key == MediaConstraintsInterface::kAutoGainControl)
-      options->auto_gain_control.Set(value);
-    else if (iter->key ==
-        MediaConstraintsInterface::kExperimentalAutoGainControl)
-      options->experimental_agc.Set(value);
-    else if (iter->key == MediaConstraintsInterface::kNoiseSuppression)
-      options->noise_suppression.Set(value);
-    else if (iter->key ==
-          MediaConstraintsInterface::kExperimentalNoiseSuppression)
-      options->experimental_ns.Set(value);
-    else if (iter->key == MediaConstraintsInterface::kHighpassFilter)
-      options->highpass_filter.Set(value);
-    else if (iter->key == MediaConstraintsInterface::kTypingNoiseDetection)
-      options->typing_detection.Set(value);
-    else if (iter->key == MediaConstraintsInterface::kAudioMirroring)
-      options->stereo_swapping.Set(value);
-    else if (iter->key == MediaConstraintsInterface::kAecDump)
-      options->aec_dump.Set(value);
+    for (auto& entry : key_to_value) {
+      if (constraint.key.compare(entry.name) == 0)
+        entry.value.Set(value);
+    }
   }
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource_unittest.cc
index 569f814a52c..62620259b60 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource_unittest.cc
@@ -43,9 +43,10 @@ using webrtc::PeerConnectionFactoryInterface;
 
 TEST(LocalAudioSourceTest, SetValidOptions) {
   webrtc::FakeConstraints constraints;
-  constraints.AddMandatory(MediaConstraintsInterface::kEchoCancellation, false);
+  constraints.AddMandatory(
+      MediaConstraintsInterface::kGoogEchoCancellation, false);
   constraints.AddOptional(
-      MediaConstraintsInterface::kExperimentalEchoCancellation, true);
+      MediaConstraintsInterface::kExtendedFilterEchoCancellation, true);
   constraints.AddOptional(MediaConstraintsInterface::kDAEchoCancellation, true);
   constraints.AddOptional(MediaConstraintsInterface::kAutoGainControl, true);
   constraints.AddOptional(
@@ -61,7 +62,7 @@ TEST(LocalAudioSourceTest, SetValidOptions) {
   bool value;
   EXPECT_TRUE(source->options().echo_cancellation.Get(&value));
   EXPECT_FALSE(value);
-  EXPECT_TRUE(source->options().experimental_aec.Get(&value));
+  EXPECT_TRUE(source->options().extended_filter_aec.Get(&value));
   EXPECT_TRUE(value);
   EXPECT_TRUE(source->options().delay_agnostic_aec.Get(&value));
   EXPECT_TRUE(value);
@@ -88,8 +89,10 @@ TEST(LocalAudioSourceTest, OptionNotSet) {
 
 TEST(LocalAudioSourceTest, MandatoryOverridesOptional) {
   webrtc::FakeConstraints constraints;
-  constraints.AddMandatory(MediaConstraintsInterface::kEchoCancellation, false);
-  constraints.AddOptional(MediaConstraintsInterface::kEchoCancellation, true);
+  constraints.AddMandatory(
+      MediaConstraintsInterface::kGoogEchoCancellation, false);
+  constraints.AddOptional(
+      MediaConstraintsInterface::kGoogEchoCancellation, true);
 
   rtc::scoped_refptr<LocalAudioSource> source =
       LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.cc
index a84dde27b38..66a2ad2cba3 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.cc
@@ -47,8 +47,10 @@ const char MediaConstraintsInterface::kMinFrameRate[] = "minFrameRate";
 
 // Audio constraints.
 const char MediaConstraintsInterface::kEchoCancellation[] =
+    "echoCancellation";
+const char MediaConstraintsInterface::kGoogEchoCancellation[] =
     "googEchoCancellation";
-const char MediaConstraintsInterface::kExperimentalEchoCancellation[] =
+const char MediaConstraintsInterface::kExtendedFilterEchoCancellation[] =
     "googEchoCancellation2";
 const char MediaConstraintsInterface::kDAEchoCancellation[] =
     "googDAEchoCancellation";
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h
index cc682097f76..777c3cc5526 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h
@@ -72,9 +72,12 @@ class MediaConstraintsInterface {
   static const char kMinFrameRate[];  // minFrameRate
 
   // Constraint keys used by a local audio source.
+  static const char kEchoCancellation[];  // echoCancellation
+
   // These keys are google specific.
-  static const char kEchoCancellation[];  // googEchoCancellation
-  static const char kExperimentalEchoCancellation[];  // googEchoCancellation2
+  static const char kGoogEchoCancellation[];  // googEchoCancellation
+
+  static const char kExtendedFilterEchoCancellation[];  // googEchoCancellation2
   static const char kDAEchoCancellation[];  // googDAEchoCancellation
   static const char kAutoGainControl[];  // googAutoGainControl
   static const char kExperimentalAutoGainControl[];  // googAutoGainControl2
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/.clang-format b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/.clang-format
new file mode 100644
index 00000000000..34694e57dcf
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/.clang-format
@@ -0,0 +1,10 @@
+BasedOnStyle: Chromium
+ColumnLimit: 100
+BinPackParameters: false
+AllowAllParametersOfDeclarationOnNextLine: true
+DerivePointerAlignment: false
+PointerAlignment: Right
+SpacesBeforeTrailingComments: 1
+ObjCBlockIndentWidth: 2
+ObjCSpaceAfterProperty: false
+ObjCSpaceBeforeProtocolList: true
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCDataChannel.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCDataChannel.mm
index 94d22d25952..8a9b6b6095a 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCDataChannel.mm
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCDataChannel.mm
@@ -43,6 +43,15 @@ class RTCDataChannelObserver : public DataChannelObserver {
     [_channel.delegate channelDidChangeState:_channel];
   }
 
+  void OnBufferedAmountChange(uint64 previousAmount) override {
+    RTCDataChannel* channel = _channel;
+    id<RTCDataChannelDelegate> delegate = channel.delegate;
+    if ([delegate
+            respondsToSelector:@selector(channel:didChangeBufferedAmount:)]) {
+      [delegate channel:channel didChangeBufferedAmount:previousAmount];
+    }
+  }
+
   void OnMessage(const DataBuffer& buffer) override {
     if (!_channel.delegate) {
       return;
@@ -57,7 +66,8 @@ class RTCDataChannelObserver : public DataChannelObserver {
 };
 }
 
-// TODO(tkchin): move to shared location
+// TODO(henrika): move to shared location.
+// See https://code.google.com/p/webrtc/issues/detail?id=4773 for details.
 NSString* NSStringFromStdString(const std::string& stdString) {
   // std::string may contain null termination character so we construct
   // using length.
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.h b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.h
index 42ea45bdcfa..eb06c186453 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.h
@@ -25,14 +25,16 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#import <Foundation/Foundation.h>
-
+// TODO(tkchin): remove this in favor of having objc headers mirror their C++ counterparts.
+// TODO(tkchin): see if we can move C++ enums into their own file so we can avoid all this
+// conversion code.
 #import "RTCTypes.h"
 
-#include "talk/app/webrtc/peerconnectioninterface.h"
+#import "talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h"
 
 @interface RTCEnumConverter : NSObject
 
+// TODO(tkchin): rename these.
 + (RTCICEConnectionState)convertIceConnectionStateToObjC:
         (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
 
@@ -54,4 +56,28 @@
 + (RTCTrackState)convertTrackStateToObjC:
         (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
 
++ (RTCIceTransportsType)iceTransportsTypeForNativeEnum:
+        (webrtc::PeerConnectionInterface::IceTransportsType)nativeEnum;
+
++ (webrtc::PeerConnectionInterface::IceTransportsType)nativeEnumForIceTransportsType:
+        (RTCIceTransportsType)iceTransportsType;
+
++ (RTCBundlePolicy)bundlePolicyForNativeEnum:
+        (webrtc::PeerConnectionInterface::BundlePolicy)nativeEnum;
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeEnumForBundlePolicy:
+        (RTCBundlePolicy)bundlePolicy;
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativeEnum:
+        (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeEnum;
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeEnumForRtcpMuxPolicy:
+        (RTCRtcpMuxPolicy)rtcpMuxPolicy;
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativeEnum:
+        (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeEnum;
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeEnumForTcpCandidatePolicy:
+        (RTCTcpCandidatePolicy)tcpCandidatePolicy;
+
 @end
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.mm
index f893fc871f1..6254b95e807 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.mm
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCEnumConverter.mm
@@ -133,4 +133,96 @@
   }
 }
 
++ (RTCIceTransportsType)iceTransportsTypeForNativeEnum:
+        (webrtc::PeerConnectionInterface::IceTransportsType)nativeEnum {
+  switch (nativeEnum) {
+    case webrtc::PeerConnectionInterface::kNone:
+      return kRTCIceTransportsTypeNone;
+    case webrtc::PeerConnectionInterface::kRelay:
+      return kRTCIceTransportsTypeRelay;
+    case webrtc::PeerConnectionInterface::kNoHost:
+      return kRTCIceTransportsTypeNoHost;
+    case webrtc::PeerConnectionInterface::kAll:
+      return kRTCIceTransportsTypeAll;
+  }
+}
+
++ (webrtc::PeerConnectionInterface::IceTransportsType)nativeEnumForIceTransportsType:
+        (RTCIceTransportsType)iceTransportsType {
+  switch (iceTransportsType) {
+    case kRTCIceTransportsTypeNone:
+      return webrtc::PeerConnectionInterface::kNone;
+    case kRTCIceTransportsTypeRelay:
+      return webrtc::PeerConnectionInterface::kRelay;
+    case kRTCIceTransportsTypeNoHost:
+      return webrtc::PeerConnectionInterface::kNoHost;
+    case kRTCIceTransportsTypeAll:
+      return webrtc::PeerConnectionInterface::kAll;
+  }
+}
+
++ (RTCBundlePolicy)bundlePolicyForNativeEnum:
+        (webrtc::PeerConnectionInterface::BundlePolicy)nativeEnum {
+  switch (nativeEnum) {
+    case webrtc::PeerConnectionInterface::kBundlePolicyBalanced:
+      return kRTCBundlePolicyBalanced;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle:
+      return kRTCBundlePolicyMaxBundle;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat:
+      return kRTCBundlePolicyMaxCompat;
+  }
+}
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeEnumForBundlePolicy:
+        (RTCBundlePolicy)bundlePolicy {
+  switch (bundlePolicy) {
+    case kRTCBundlePolicyBalanced:
+      return webrtc::PeerConnectionInterface::kBundlePolicyBalanced;
+    case kRTCBundlePolicyMaxBundle:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
+    case kRTCBundlePolicyMaxCompat:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
+  }
+}
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativeEnum:
+        (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeEnum {
+  switch (nativeEnum) {
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate:
+      return kRTCRtcpMuxPolicyNegotiate;
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire:
+      return kRTCRtcpMuxPolicyRequire;
+  }
+}
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeEnumForRtcpMuxPolicy:
+        (RTCRtcpMuxPolicy)rtcpMuxPolicy {
+  switch (rtcpMuxPolicy) {
+    case kRTCRtcpMuxPolicyNegotiate:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
+    case kRTCRtcpMuxPolicyRequire:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  }
+}
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativeEnum:
+        (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeEnum {
+  switch (nativeEnum) {
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled:
+      return kRTCTcpCandidatePolicyEnabled;
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled:
+      return kRTCTcpCandidatePolicyDisabled;
+  }
+}
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeEnumForTcpCandidatePolicy:
+        (RTCTcpCandidatePolicy)tcpCandidatePolicy {
+  switch (tcpCandidatePolicy) {
+    case kRTCTcpCandidatePolicyEnabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled;
+    case kRTCTcpCandidatePolicyDisabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled;
+  }
+}
+
 @end
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection+Internal.h b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection+Internal.h
index 8d7201f811f..96d63ab4121 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection+Internal.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection+Internal.h
@@ -36,8 +36,13 @@
 @property(nonatomic, assign, readonly)
     rtc::scoped_refptr<webrtc::PeerConnectionInterface> peerConnection;
 
-- (instancetype)initWithFactory:(webrtc::PeerConnectionFactoryInterface*)factory
-     iceServers:(const webrtc::PeerConnectionInterface::IceServers&)iceServers
-    constraints:(const webrtc::MediaConstraintsInterface*)constraints;
+- (instancetype)initWithFactory:(webrtc::PeerConnectionFactoryInterface *)factory
+                     iceServers:(const webrtc::PeerConnectionInterface::IceServers &)iceServers
+                    constraints:(const webrtc::MediaConstraintsInterface *)constraints;
+
+- (instancetype)initWithFactory:(webrtc::PeerConnectionFactoryInterface *)factory
+                         config:(const webrtc::PeerConnectionInterface::RTCConfiguration &)config
+                    constraints:(const webrtc::MediaConstraintsInterface *)constraints
+                       delegate:(id<RTCPeerConnectionDelegate>)delegate;
 
 @end
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection.mm
index 769219e1eda..0d30acc0eb2 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection.mm
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnection.mm
@@ -285,6 +285,21 @@ class RTCStatsObserver : public StatsObserver {
   return self;
 }
 
+- (instancetype)initWithFactory:(webrtc::PeerConnectionFactoryInterface *)factory
+                         config:(const webrtc::PeerConnectionInterface::RTCConfiguration &)config
+                    constraints:(const webrtc::MediaConstraintsInterface *)constraints
+                       delegate:(id<RTCPeerConnectionDelegate>)delegate {
+  NSParameterAssert(factory);
+  if (self = [super init]) {
+    _observer.reset(new webrtc::RTCPeerConnectionObserver(self));
+    _peerConnection =
+        factory->CreatePeerConnection(config, constraints, nullptr, nullptr, _observer.get());
+    _localStreams = [[NSMutableArray alloc] init];
+    _delegate = delegate;
+  }
+  return self;
+}
+
 - (rtc::scoped_refptr<webrtc::PeerConnectionInterface>)peerConnection {
   return _peerConnection;
 }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm
index a4a12addd9f..b7b89662394 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm
@@ -41,6 +41,7 @@
 #import "RTCMediaStreamTrack+Internal.h"
 #import "RTCPeerConnection+Internal.h"
 #import "RTCPeerConnectionDelegate.h"
+#import "RTCPeerConnectionInterface+Internal.h"
 #import "RTCVideoCapturer+Internal.h"
 #import "RTCVideoSource+Internal.h"
 #import "RTCVideoTrack+Internal.h"
@@ -53,7 +54,6 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/ssladapter.h"
 
-
 @implementation RTCPeerConnectionFactory {
   rtc::scoped_ptr<rtc::Thread> _signalingThread;
   rtc::scoped_ptr<rtc::Thread> _workerThread;
@@ -79,8 +79,9 @@
     _workerThread.reset(new rtc::Thread());
     result = _workerThread->Start();
     NSAssert(result, @"Failed to start worker thread.");
+
     _nativeFactory = webrtc::CreatePeerConnectionFactory(
-        _signalingThread.get(), _workerThread.get(), NULL, NULL, NULL);
+        _signalingThread.get(), _workerThread.get(), nullptr, nullptr, nullptr);
     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
     // Uncomment to get sensitive logs emitted (to stderr or logcat).
     // rtc::LogMessage::LogToDebug(rtc::LS_SENSITIVE);
@@ -88,6 +89,15 @@
   return self;
 }
 
+- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
+                                           constraints:(RTCMediaConstraints *)constraints
+                                              delegate:(id<RTCPeerConnectionDelegate>)delegate {
+  return [[RTCPeerConnection alloc] initWithFactory:self.nativeFactory.get()
+                                             config:configuration.nativeConfiguration
+                                        constraints:constraints.constraints
+                                           delegate:delegate];
+}
+
 - (RTCPeerConnection*)
     peerConnectionWithICEServers:(NSArray*)servers
                      constraints:(RTCMediaConstraints*)constraints
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h
new file mode 100644
index 00000000000..5e8dbbf604e
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h
@@ -0,0 +1,37 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+#import "talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h"
+
+#include "talk/app/webrtc/peerconnectioninterface.h"
+
+@interface RTCConfiguration ()
+
+@property(nonatomic, readonly)
+    webrtc::PeerConnectionInterface::RTCConfiguration nativeConfiguration;
+
+@end
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm
new file mode 100644
index 00000000000..32298c8d725
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm
@@ -0,0 +1,87 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+#import "talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h"
+
+#import "talk/app/webrtc/objc/RTCEnumConverter.h"
+#import "talk/app/webrtc/objc/RTCICEServer+Internal.h"
+
+@implementation RTCConfiguration
+
+@synthesize iceTransportsType = _iceTransportsType;
+@synthesize iceServers = _iceServers;
+@synthesize bundlePolicy = _bundlePolicy;
+@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
+@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
+@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    // Copy defaults.
+    webrtc::PeerConnectionInterface::RTCConfiguration config;
+    _iceTransportsType = [RTCEnumConverter iceTransportsTypeForNativeEnum:config.type];
+    _bundlePolicy = [RTCEnumConverter bundlePolicyForNativeEnum:config.bundle_policy];
+    _rtcpMuxPolicy = [RTCEnumConverter rtcpMuxPolicyForNativeEnum:config.rtcp_mux_policy];
+    _tcpCandidatePolicy =
+        [RTCEnumConverter tcpCandidatePolicyForNativeEnum:config.tcp_candidate_policy];
+    _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
+  }
+  return self;
+}
+
+- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
+                             bundlePolicy:(RTCBundlePolicy)bundlePolicy
+                            rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
+                       tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
+              audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets {
+  if (self = [super init]) {
+    _iceTransportsType = iceTransportsType;
+    _bundlePolicy = bundlePolicy;
+    _rtcpMuxPolicy = rtcpMuxPolicy;
+    _tcpCandidatePolicy = tcpCandidatePolicy;
+    _audioJitterBufferMaxPackets = audioJitterBufferMaxPackets;
+  }
+  return self;
+}
+
+#pragma mark - Private
+
+- (webrtc::PeerConnectionInterface::RTCConfiguration)nativeConfiguration {
+  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig;
+  nativeConfig.type = [RTCEnumConverter nativeEnumForIceTransportsType:_iceTransportsType];
+  for (RTCICEServer *iceServer : _iceServers) {
+    nativeConfig.servers.push_back(iceServer.iceServer);
+  }
+  nativeConfig.bundle_policy = [RTCEnumConverter nativeEnumForBundlePolicy:_bundlePolicy];
+  nativeConfig.rtcp_mux_policy = [RTCEnumConverter nativeEnumForRtcpMuxPolicy:_rtcpMuxPolicy];
+  nativeConfig.tcp_candidate_policy =
+      [RTCEnumConverter nativeEnumForTcpCandidatePolicy:_tcpCandidatePolicy];
+  nativeConfig.audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
+  return nativeConfig;
+}
+
+@end
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/avfoundationvideocapturer.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/avfoundationvideocapturer.mm
index 9f5361a60b7..d68fdff79a0 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/avfoundationvideocapturer.mm
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/avfoundationvideocapturer.mm
@@ -142,6 +142,7 @@ static dispatch_queue_t kBackgroundQueue = nil;
   if (!_isRunning) {
     return;
   }
+  [_videoOutput setSampleBufferDelegate:nil queue:nullptr];
   AVCaptureSession* session = _captureSession;
   dispatch_async(kBackgroundQueue, ^{
     [session stopRunning];
@@ -262,10 +263,11 @@ static dispatch_queue_t kBackgroundQueue = nil;
   AVCaptureVideoOrientation orientation = AVCaptureVideoOrientationPortrait;
   switch ([UIDevice currentDevice].orientation) {
     case UIDeviceOrientationPortrait:
-      orientation = AVCaptureVideoOrientationPortraitUpsideDown;
-    case UIDeviceOrientationPortraitUpsideDown:
       orientation = AVCaptureVideoOrientationPortrait;
       break;
+    case UIDeviceOrientationPortraitUpsideDown:
+      orientation = AVCaptureVideoOrientationPortraitUpsideDown;
+      break;
     case UIDeviceOrientationLandscapeLeft:
       orientation = AVCaptureVideoOrientationLandscapeRight;
       break;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCDataChannel.h b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCDataChannel.h
index 7c225802455..24a46f655c8 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCDataChannel.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCDataChannel.h
@@ -82,6 +82,12 @@ typedef enum {
 - (void)channel:(RTCDataChannel*)channel
     didReceiveMessageWithBuffer:(RTCDataBuffer*)buffer;
 
+@optional
+
+// Called when the buffered amount has changed.
+- (void)channel:(RTCDataChannel*)channel
+    didChangeBufferedAmount:(NSUInteger)amount;
+
 @end
 
 // ObjectiveC wrapper for a DataChannel object.
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCPeerConnectionFactory.h b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCPeerConnectionFactory.h
index f0b2e3ad991..e1e69b4b7f1 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCPeerConnectionFactory.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCPeerConnectionFactory.h
@@ -28,6 +28,7 @@
 #import <Foundation/Foundation.h>
 
 @class RTCAudioTrack;
+@class RTCConfiguration;
 @class RTCMediaConstraints;
 @class RTCMediaStream;
 @class RTCPeerConnection;
@@ -44,7 +45,7 @@
 + (void)initializeSSL;
 + (void)deinitializeSSL;
 
-// Create an RTCPeerConnection object.   RTCPeerConnectionFactory will create
+// Create an RTCPeerConnection object. RTCPeerConnectionFactory will create
 // required libjingle threads, socket and network manager factory classes for
 // networking.
 - (RTCPeerConnection *)
@@ -52,6 +53,11 @@
                      constraints:(RTCMediaConstraints *)constraints
                         delegate:(id<RTCPeerConnectionDelegate>)delegate;
 
+// Creates a peer connection using the default port allocator factory and identity service.
+- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
+                                           constraints:(RTCMediaConstraints *)constraints
+                                              delegate:(id<RTCPeerConnectionDelegate>)delegate;
+
 // Create an RTCMediaStream named |label|.
 - (RTCMediaStream *)mediaStreamWithLabel:(NSString *)label;
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h
new file mode 100644
index 00000000000..fd7af620d9d
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h
@@ -0,0 +1,73 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+// See talk/app/webrtc/peerconnectioninterface.h.
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, RTCIceTransportsType) {
+  kRTCIceTransportsTypeNone,
+  kRTCIceTransportsTypeRelay,
+  kRTCIceTransportsTypeNoHost,
+  kRTCIceTransportsTypeAll,
+};
+
+// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
+typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
+  kRTCBundlePolicyBalanced,
+  kRTCBundlePolicyMaxBundle,
+  kRTCBundlePolicyMaxCompat,
+};
+
+// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) {
+  kRTCRtcpMuxPolicyNegotiate,
+  kRTCRtcpMuxPolicyRequire,
+};
+
+typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
+  kRTCTcpCandidatePolicyEnabled,
+  kRTCTcpCandidatePolicyDisabled,
+};
+
+// Configuration object used for creating a peer connection.
+@interface RTCConfiguration : NSObject
+
+@property(nonatomic, assign) RTCIceTransportsType iceTransportsType;
+@property(nonatomic, copy) NSArray *iceServers;
+@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
+@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
+@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign) int audioJitterBufferMaxPackets;
+
+- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
+                             bundlePolicy:(RTCBundlePolicy)bundlePolicy
+                            rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
+                       tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
+              audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets;
+
+@end
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m b/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m
index 5c766724142..5070b789b89 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m
@@ -231,6 +231,12 @@
 }
 
 - (void)channel:(RTCDataChannel*)channel
+    didChangeBufferedAmount:(NSUInteger)previousAmount {
+  NSAssert(channel.bufferedAmount != previousAmount,
+           @"Invalid bufferedAmount change");
+}
+
+- (void)channel:(RTCDataChannel*)channel
     didReceiveMessageWithBuffer:(RTCDataBuffer*)buffer {
   NSAssert([_expectedMessages count] > 0,
            @"Unexpected message received");
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc
index 52260413f7c..0a243077bcf 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc
@@ -166,9 +166,10 @@ typedef webrtc::PortAllocatorFactoryInterface::StunConfiguration
 typedef webrtc::PortAllocatorFactoryInterface::TurnConfiguration
     TurnConfiguration;
 
-bool ParseIceServers(const PeerConnectionInterface::IceServers& configuration,
-                     std::vector<StunConfiguration>* stun_config,
-                     std::vector<TurnConfiguration>* turn_config) {
+bool ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
+                       const std::string& url,
+                       std::vector<StunConfiguration>* stun_config,
+                       std::vector<TurnConfiguration>* turn_config) {
   // draft-nandakumar-rtcweb-stun-uri-01
   // stunURI       = scheme ":" stun-host [ ":" stun-port ]
   // scheme        = "stun" / "stuns"
@@ -183,103 +184,124 @@ bool ParseIceServers(const PeerConnectionInterface::IceServers& configuration,
   // transport-ext = 1*unreserved
   // turn-host     = IP-literal / IPv4address / reg-name
   // turn-port     = *DIGIT
-  for (size_t i = 0; i < configuration.size(); ++i) {
-    webrtc::PeerConnectionInterface::IceServer server = configuration[i];
-    if (server.uri.empty()) {
-      LOG(WARNING) << "Empty uri.";
-      continue;
-    }
-    std::vector<std::string> tokens;
-    std::string turn_transport_type = kUdpTransportType;
-    rtc::tokenize(server.uri, '?', &tokens);
-    std::string uri_without_transport = tokens[0];
-    // Let's look into transport= param, if it exists.
-    if (tokens.size() == kTurnTransportTokensNum) {  // ?transport= is present.
-      std::string uri_transport_param = tokens[1];
-      rtc::tokenize(uri_transport_param, '=', &tokens);
-      if (tokens[0] == kTransport) {
-        // As per above grammar transport param will be consist of lower case
-        // letters.
-        if (tokens[1] != kUdpTransportType && tokens[1] != kTcpTransportType) {
-          LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
-          continue;
-        }
-        turn_transport_type = tokens[1];
+  std::vector<std::string> tokens;
+  std::string turn_transport_type = kUdpTransportType;
+  ASSERT(!url.empty());
+  rtc::tokenize(url, '?', &tokens);
+  std::string uri_without_transport = tokens[0];
+  // Let's look into transport= param, if it exists.
+  if (tokens.size() == kTurnTransportTokensNum) {  // ?transport= is present.
+    std::string uri_transport_param = tokens[1];
+    rtc::tokenize(uri_transport_param, '=', &tokens);
+    if (tokens[0] == kTransport) {
+      // As per above grammar transport param will be consist of lower case
+      // letters.
+      if (tokens[1] != kUdpTransportType && tokens[1] != kTcpTransportType) {
+        LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
+        return true;
       }
+      turn_transport_type = tokens[1];
     }
+  }
 
-    std::string hoststring;
-    ServiceType service_type = INVALID;
-    if (!GetServiceTypeAndHostnameFromUri(uri_without_transport,
-                                         &service_type,
-                                         &hoststring)) {
-      LOG(LS_WARNING) << "Invalid transport parameter in ICE URI: "
-                      << uri_without_transport;
-      continue;
-    }
-
-    ASSERT(!hoststring.empty());
-
-    // Let's break hostname.
-    tokens.clear();
-    rtc::tokenize(hoststring, '@', &tokens);
-    ASSERT(!tokens.empty());
-    // TODO(pthatcher): What's the right thing to do if tokens.size() is >2?
-    // E.g. a string like "foo@bar@bat".
-    if (tokens.size() >= kTurnHostTokensNum) {
-      server.username = rtc::s_url_decode(tokens[0]);
-      hoststring = tokens[1];
-    } else {
-      hoststring = tokens[0];
-    }
+  std::string hoststring;
+  ServiceType service_type = INVALID;
+  if (!GetServiceTypeAndHostnameFromUri(uri_without_transport,
+                                       &service_type,
+                                       &hoststring)) {
+    LOG(LS_WARNING) << "Invalid transport parameter in ICE URI: "
+                    << uri_without_transport;
+    return true;
+  }
+
+  ASSERT(!hoststring.empty());
+
+  // Let's break hostname.
+  tokens.clear();
+  rtc::tokenize(hoststring, '@', &tokens);
+  ASSERT(!tokens.empty());
+  std::string username(server.username);
+  // TODO(pthatcher): What's the right thing to do if tokens.size() is >2?
+  // E.g. a string like "foo@bar@bat".
+  if (tokens.size() >= kTurnHostTokensNum) {
+    username.assign(rtc::s_url_decode(tokens[0]));
+    hoststring = tokens[1];
+  } else {
+    hoststring = tokens[0];
+  }
 
-    int port = kDefaultStunPort;
-    if (service_type == TURNS) {
-      port = kDefaultStunTlsPort;
-      turn_transport_type = kTcpTransportType;
-    }
+  int port = kDefaultStunPort;
+  if (service_type == TURNS) {
+    port = kDefaultStunTlsPort;
+    turn_transport_type = kTcpTransportType;
+  }
 
-    std::string address;
-    if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
-      LOG(WARNING) << "Invalid Hostname format: " << uri_without_transport;
-      continue;
-    }
+  std::string address;
+  if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
+    LOG(WARNING) << "Invalid Hostname format: " << uri_without_transport;
+    return true;
+  }
 
-    if (port <= 0 || port > 0xffff) {
-      LOG(WARNING) << "Invalid port: " << port;
-      continue;
-    }
+  if (port <= 0 || port > 0xffff) {
+    LOG(WARNING) << "Invalid port: " << port;
+    return true;
+  }
 
-    switch (service_type) {
-      case STUN:
-      case STUNS:
-        stun_config->push_back(StunConfiguration(address, port));
-        break;
-      case TURN:
-      case TURNS: {
-        if (server.username.empty()) {
-          // Turn url example from the spec |url:"turn:user@turn.example.org"|.
-          std::vector<std::string> turn_tokens;
-          rtc::tokenize(address, '@', &turn_tokens);
-          if (turn_tokens.size() == kTurnHostTokensNum) {
-            server.username = rtc::s_url_decode(turn_tokens[0]);
-            address = turn_tokens[1];
-          }
+  switch (service_type) {
+    case STUN:
+    case STUNS:
+      stun_config->push_back(StunConfiguration(address, port));
+      break;
+    case TURN:
+    case TURNS: {
+      if (username.empty()) {
+        // Turn url example from the spec |url:"turn:user@turn.example.org"|.
+        std::vector<std::string> turn_tokens;
+        rtc::tokenize(address, '@', &turn_tokens);
+        if (turn_tokens.size() == kTurnHostTokensNum) {
+          username.assign(rtc::s_url_decode(turn_tokens[0]));
+          address = turn_tokens[1];
         }
+      }
 
-        bool secure = (service_type == TURNS);
+      bool secure = (service_type == TURNS);
 
-        turn_config->push_back(TurnConfiguration(address, port,
-                                                 server.username,
-                                                 server.password,
-                                                 turn_transport_type,
-                                                 secure));
-        break;
+      turn_config->push_back(TurnConfiguration(address, port,
+                                               username,
+                                               server.password,
+                                               turn_transport_type,
+                                               secure));
+      break;
+    }
+    case INVALID:
+    default:
+      LOG(WARNING) << "Configuration not supported: " << url;
+      return false;
+  }
+  return true;
+}
+
+bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
+                     std::vector<StunConfiguration>* stun_config,
+                     std::vector<TurnConfiguration>* turn_config) {
+  for (const webrtc::PeerConnectionInterface::IceServer& server : servers) {
+    if (!server.urls.empty()) {
+      for (const std::string& url : server.urls) {
+        if (url.empty()) {
+          LOG(WARNING) << "Empty uri.";
+          continue;
+        }
+        if (!ParseIceServerUrl(server, url, stun_config, turn_config)) {
+          return false;
+        }
       }
-      case INVALID:
-      default:
-        LOG(WARNING) << "Configuration not supported: " << server.uri;
+    } else if (!server.uri.empty()) {
+      // Fallback to old .uri if new .urls isn't present.
+      if (!ParseIceServerUrl(server, server.uri, stun_config, turn_config)) {
         return false;
+      }
+    } else {
+      LOG(WARNING) << "Empty uri.";
     }
   }
   return true;
@@ -316,6 +338,7 @@ PeerConnection::PeerConnection(PeerConnectionFactory* factory)
 }
 
 PeerConnection::~PeerConnection() {
+  ASSERT(signaling_thread()->IsCurrent());
   if (mediastream_signaling_)
     mediastream_signaling_->TearDown();
   if (stream_handler_container_)
@@ -864,6 +887,11 @@ void PeerConnection::OnIceComplete() {
   observer_->OnIceComplete();
 }
 
+void PeerConnection::OnIceConnectionReceivingChange(bool receiving) {
+  ASSERT(signaling_thread()->IsCurrent());
+  observer_->OnIceConnectionReceivingChange(receiving);
+}
+
 void PeerConnection::ChangeSignalingState(
     PeerConnectionInterface::SignalingState signaling_state) {
   signaling_state_ = signaling_state;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.h b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.h
index bea9c2f61b6..1f6b59f832e 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.h
@@ -142,13 +142,14 @@ class PeerConnection : public PeerConnectionInterface,
                                uint32 ssrc) override;
   void OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
                                VideoTrackInterface* video_track) override;
-  virtual void OnRemoveLocalStream(MediaStreamInterface* stream);
+  void OnRemoveLocalStream(MediaStreamInterface* stream) override;
 
   // Implements IceObserver
-  virtual void OnIceConnectionChange(IceConnectionState new_state);
-  virtual void OnIceGatheringChange(IceGatheringState new_state);
-  virtual void OnIceCandidate(const IceCandidateInterface* candidate);
-  virtual void OnIceComplete();
+  void OnIceConnectionChange(IceConnectionState new_state) override;
+  void OnIceGatheringChange(IceGatheringState new_state) override;
+  void OnIceCandidate(const IceCandidateInterface* candidate) override;
+  void OnIceComplete() override;
+  void OnIceConnectionReceivingChange(bool receiving) override;
 
   // Signals from WebRtcSession.
   void OnSessionStateChange(cricket::BaseSession* session,
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
index 93793cfc3e3..fd58ecdff6f 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
@@ -33,6 +33,7 @@
 #include <vector>
 
 #include "talk/app/webrtc/dtmfsender.h"
+#include "talk/app/webrtc/fakemetricsobserver.h"
 #include "talk/app/webrtc/fakeportallocatorfactory.h"
 #include "talk/app/webrtc/localaudiosource.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
@@ -82,6 +83,7 @@ using webrtc::MockDataChannelObserver;
 using webrtc::MockSetSessionDescriptionObserver;
 using webrtc::MockStatsObserver;
 using webrtc::PeerConnectionInterface;
+using webrtc::PeerConnectionFactory;
 using webrtc::SessionDescriptionInterface;
 using webrtc::StreamCollectionInterface;
 
@@ -349,6 +351,7 @@ class PeerConnectionTestClientBase
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+    EXPECT_NE(0, observer->timestamp());
     return observer->AudioOutputLevel();
   }
 
@@ -358,6 +361,7 @@ class PeerConnectionTestClientBase
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+    EXPECT_NE(0, observer->timestamp());
     return observer->AudioInputLevel();
   }
 
@@ -367,6 +371,7 @@ class PeerConnectionTestClientBase
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+    EXPECT_NE(0, observer->timestamp());
     return observer->BytesReceived();
   }
 
@@ -376,6 +381,7 @@ class PeerConnectionTestClientBase
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+    EXPECT_NE(0, observer->timestamp());
     return observer->BytesSent();
   }
 
@@ -385,6 +391,7 @@ class PeerConnectionTestClientBase
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+    EXPECT_NE(0, observer->timestamp());
     int bw = observer->AvailableReceiveBandwidth();
     return bw;
   }
@@ -395,6 +402,7 @@ class PeerConnectionTestClientBase
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+    EXPECT_NE(0, observer->timestamp());
     return observer->DtlsCipher();
   }
 
@@ -404,6 +412,7 @@ class PeerConnectionTestClientBase
     EXPECT_TRUE(peer_connection_->GetStats(
         observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
+    EXPECT_NE(0, observer->timestamp());
     return observer->SrtpCipher();
   }
 
@@ -501,7 +510,8 @@ class PeerConnectionTestClientBase
         video_decoder_factory_enabled_(false),
         signaling_message_receiver_(NULL) {
   }
-  bool Init(const MediaConstraintsInterface* constraints) {
+  bool Init(const MediaConstraintsInterface* constraints,
+            const PeerConnectionFactory::Options* options) {
     EXPECT_TRUE(!peer_connection_);
     EXPECT_TRUE(!peer_connection_factory_);
     allocator_factory_ = webrtc::FakePortAllocatorFactory::Create();
@@ -523,6 +533,9 @@ class PeerConnectionTestClientBase
     if (!peer_connection_factory_) {
       return false;
     }
+    if (options) {
+      peer_connection_factory_->SetOptions(*options);
+    }
     peer_connection_ = CreatePeerConnection(allocator_factory_.get(),
                                             constraints);
     return peer_connection_.get() != NULL;
@@ -619,9 +632,10 @@ class JsepTestClient
  public:
   static JsepTestClient* CreateClient(
       const std::string& id,
-      const MediaConstraintsInterface* constraints) {
+      const MediaConstraintsInterface* constraints,
+      const PeerConnectionFactory::Options* options) {
     JsepTestClient* client(new JsepTestClient(id));
-    if (!client->Init(constraints)) {
+    if (!client->Init(constraints, options)) {
       delete client;
       return NULL;
     }
@@ -967,10 +981,19 @@ class P2PTestConductor : public testing::Test {
 
   bool CreateTestClients(MediaConstraintsInterface* init_constraints,
                          MediaConstraintsInterface* recv_constraints) {
+    return CreateTestClients(init_constraints, NULL, recv_constraints, NULL);
+  }
+
+  bool CreateTestClients(MediaConstraintsInterface* init_constraints,
+                         PeerConnectionFactory::Options* init_options,
+                         MediaConstraintsInterface* recv_constraints,
+                         PeerConnectionFactory::Options* recv_options) {
     initiating_client_.reset(SignalingClass::CreateClient("Caller: ",
-                                                          init_constraints));
+                                                          init_constraints,
+                                                          init_options));
     receiving_client_.reset(SignalingClass::CreateClient("Callee: ",
-                                                         recv_constraints));
+                                                         recv_constraints,
+                                                         recv_options));
     if (!initiating_client_ || !receiving_client_) {
       return false;
     }
@@ -1307,20 +1330,122 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) {
       kMaxWaitForStatsMs);
 }
 
-// Test that we can get negotiated ciphers.
-TEST_F(JsepPeerConnectionP2PTestClient, GetNegotiatedCiphersStats) {
-  ASSERT_TRUE(CreateTestClients());
+// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
+  PeerConnectionFactory::Options init_options;
+  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
+  PeerConnectionFactory::Options recv_options;
+  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
+  ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  initializing_client()->pc()->RegisterUMAObserver(init_observer);
   LocalP2PTest();
 
   EXPECT_EQ_WAIT(
-      rtc::SSLStreamAdapter::GetDefaultSslCipher(),
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
       initializing_client()->GetDtlsCipherStats(),
       kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
+      init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
 
   EXPECT_EQ_WAIT(
       kDefaultSrtpCipher,
       initializing_client()->GetSrtpCipherStats(),
       kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+}
+
+// Test that DTLS 1.2 is used if both ends support it.
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
+  PeerConnectionFactory::Options init_options;
+  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
+  PeerConnectionFactory::Options recv_options;
+  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
+  ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  initializing_client()->pc()->RegisterUMAObserver(init_observer);
+  LocalP2PTest();
+
+  EXPECT_EQ_WAIT(
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_12),
+      initializing_client()->GetDtlsCipherStats(),
+      kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_12),
+      init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
+
+  EXPECT_EQ_WAIT(
+      kDefaultSrtpCipher,
+      initializing_client()->GetSrtpCipherStats(),
+      kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+}
+
+// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
+// received supports 1.0.
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
+  PeerConnectionFactory::Options init_options;
+  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
+  PeerConnectionFactory::Options recv_options;
+  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
+  ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  initializing_client()->pc()->RegisterUMAObserver(init_observer);
+  LocalP2PTest();
+
+  EXPECT_EQ_WAIT(
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
+      initializing_client()->GetDtlsCipherStats(),
+      kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
+      init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
+
+  EXPECT_EQ_WAIT(
+      kDefaultSrtpCipher,
+      initializing_client()->GetSrtpCipherStats(),
+      kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
+}
+
+// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
+// received supports 1.2.
+TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
+  PeerConnectionFactory::Options init_options;
+  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
+  PeerConnectionFactory::Options recv_options;
+  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
+  ASSERT_TRUE(CreateTestClients(NULL, &init_options, NULL, &recv_options));
+  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
+      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+  initializing_client()->pc()->RegisterUMAObserver(init_observer);
+  LocalP2PTest();
+
+  EXPECT_EQ_WAIT(
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
+      initializing_client()->GetDtlsCipherStats(),
+      kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      rtc::SSLStreamAdapter::GetDefaultSslCipher(rtc::SSL_PROTOCOL_DTLS_10),
+      init_observer->GetStringHistogramSample(webrtc::kAudioSslCipher));
+
+  EXPECT_EQ_WAIT(
+      kDefaultSrtpCipher,
+      initializing_client()->GetSrtpCipherStats(),
+      kMaxWaitForStatsMs);
+  EXPECT_EQ(
+      kDefaultSrtpCipher,
+      init_observer->GetStringHistogramSample(webrtc::kAudioSrtpCipher));
 }
 
 // This test sets up a call between two parties with audio, video and data.
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc
index 1c933764ac7..3524af7932a 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc
@@ -276,7 +276,7 @@ rtc::Thread* PeerConnectionFactory::worker_thread() {
 cricket::MediaEngineInterface* PeerConnectionFactory::CreateMediaEngine_w() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   return cricket::WebRtcMediaEngineFactory::Create(
-      default_adm_.get(), NULL, video_encoder_factory_.get(),
+      default_adm_.get(), video_encoder_factory_.get(),
       video_decoder_factory_.get());
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory_unittest.cc
index 67a20331dac..80885888522 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory_unittest.cc
@@ -199,6 +199,38 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
 }
 
 // This test verifies creation of PeerConnection with valid STUN and TURN
+// configuration. Also verifies the list of URL's parsed correctly as expected.
+TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) {
+  PeerConnectionInterface::RTCConfiguration config;
+  webrtc::PeerConnectionInterface::IceServer ice_server;
+  ice_server.urls.push_back("");  // Empty URLs should be ignored.
+  ice_server.urls.push_back(kStunIceServer);
+  ice_server.urls.push_back(kTurnIceServer);
+  ice_server.urls.push_back(kTurnIceServerWithTransport);
+  ice_server.password = kTurnPassword;
+  config.servers.push_back(ice_server);
+  rtc::scoped_refptr<PeerConnectionInterface> pc(
+      factory_->CreatePeerConnection(config, NULL,
+                                     allocator_factory_.get(),
+                                     new FakeIdentityService(),
+                                     &observer_));
+  EXPECT_TRUE(pc.get() != NULL);
+  StunConfigurations stun_configs;
+  webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
+      "stun.l.google.com", 19302);
+  stun_configs.push_back(stun1);
+  VerifyStunConfigurations(stun_configs);
+  TurnConfigurations turn_configs;
+  webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn1(
+      "test.com", 1234, "test@hello.com", kTurnPassword, "udp", false);
+  turn_configs.push_back(turn1);
+  webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn2(
+      "hello.com", kDefaultStunPort, "test", kTurnPassword, "tcp", false);
+  turn_configs.push_back(turn2);
+  VerifyTurnConfigurations(turn_configs);
+}
+
+// This test verifies creation of PeerConnection with valid STUN and TURN
 // configuration. Also verifies the URL's parsed correctly as expected.
 // This version doesn't use RTCConfiguration.
 // TODO(mallinath) - Remove this method after clients start using RTCConfig.
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h
index e32676e259f..521dad26956 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h
@@ -79,6 +79,7 @@
 #include "talk/app/webrtc/umametrics.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/network.h"
+#include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/socketaddress.h"
 
 namespace rtc {
@@ -126,6 +127,9 @@ class MetricsObserverInterface : public rtc::RefCountInterface {
   virtual void IncrementCounter(PeerConnectionMetricsCounter type) = 0;
   virtual void AddHistogramSample(PeerConnectionMetricsName type,
                                   int value) = 0;
+  // TODO(jbauch): Make method abstract when it is implemented by Chromium.
+  virtual void AddHistogramSample(PeerConnectionMetricsName type,
+                                  const std::string& value) {}
 
  protected:
   virtual ~MetricsObserverInterface() {}
@@ -175,7 +179,9 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
   };
 
   struct IceServer {
+    // TODO(jbauch): Remove uri when all code using it has switched to urls.
     std::string uri;
+    std::vector<std::string> urls;
     std::string username;
     std::string password;
   };
@@ -197,6 +203,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
     kBundlePolicyMaxCompat
   };
 
+  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
+  enum RtcpMuxPolicy {
+    kRtcpMuxPolicyNegotiate,
+    kRtcpMuxPolicyRequire,
+  };
+
   enum TcpCandidatePolicy {
     kTcpCandidatePolicyEnabled,
     kTcpCandidatePolicyDisabled
@@ -210,14 +222,18 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
     // 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;
 
     RTCConfiguration()
         : type(kAll),
           bundle_policy(kBundlePolicyBalanced),
+          rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
           tcp_candidate_policy(kTcpCandidatePolicyEnabled),
-          audio_jitter_buffer_max_packets(50) {}
+          audio_jitter_buffer_max_packets(50),
+          audio_jitter_buffer_fast_accelerate(false) {}
   };
 
   struct RTCOfferAnswerOptions {
@@ -395,6 +411,9 @@ class PeerConnectionObserver {
   // All Ice candidates have been found.
   virtual void OnIceComplete() {}
 
+  // Called when the ICE connection receiving status changes.
+  virtual void OnIceConnectionReceivingChange(bool receiving) {}
+
  protected:
   // Dtor protected as objects shouldn't be deleted via this interface.
   ~PeerConnectionObserver() {}
@@ -508,7 +527,8 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
     Options() :
       disable_encryption(false),
       disable_sctp_data_channels(false),
-      network_ignore_mask(rtc::kDefaultNetworkIgnoreMask) {
+      network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
+      ssl_max_version(rtc::SSL_PROTOCOL_DTLS_10) {
     }
     bool disable_encryption;
     bool disable_sctp_data_channels;
@@ -517,6 +537,11 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
     // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
     // loopback interfaces.
     int network_ignore_mask;
+
+    // Sets the maximum supported protocol version. The highest version
+    // supported by both ends will be used for the connection, i.e. if one
+    // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
+    rtc::SSLProtocolVersion ssl_max_version;
   };
 
   virtual void SetOptions(const Options& options) = 0;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc
index fb1595236b1..ad64639eb93 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc
@@ -80,18 +80,25 @@ StatsReport::Id GetTransportIdFromProxy(const cricket::ProxyTransportMap& map,
       found->second, cricket::ICE_CANDIDATE_COMPONENT_RTP);
 }
 
-void AddTrackReport(StatsCollection* reports, const std::string& track_id) {
+StatsReport* AddTrackReport(StatsCollection* reports,
+                            const std::string& track_id) {
   // Adds an empty track report.
   StatsReport::Id id(
       StatsReport::NewTypedId(StatsReport::kStatsReportTypeTrack, track_id));
   StatsReport* report = reports->ReplaceOrAddNew(id);
   report->AddString(StatsReport::kStatsValueNameTrackId, track_id);
+  return report;
 }
 
 template <class TrackVector>
-void CreateTrackReports(const TrackVector& tracks, StatsCollection* reports) {
-  for (const auto& track : tracks)
-    AddTrackReport(reports, track->id());
+void CreateTrackReports(const TrackVector& tracks, StatsCollection* reports,
+                        TrackIdMap& track_ids) {
+  for (const auto& track : tracks) {
+    const std::string& track_id = track->id();
+    StatsReport* report = AddTrackReport(reports, track_id);
+    DCHECK(report != nullptr);
+    track_ids[track_id] = report;
+  }
 }
 
 void ExtractCommonSendProperties(const cricket::MediaSenderInfo& info,
@@ -129,6 +136,9 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
     { StatsReport::kStatsValueNameSecondaryDecodedRate,
       info.secondary_decoded_rate },
     { StatsReport::kStatsValueNameSpeechExpandRate, info.speech_expand_rate },
+    { StatsReport::kStatsValueNameAccelerateRate, info.accelerate_rate },
+    { StatsReport::kStatsValueNamePreemptiveExpandRate,
+      info.preemptive_expand_rate },
   };
 
   const IntForAdd ints[] = {
@@ -362,9 +372,9 @@ void StatsCollector::AddStream(MediaStreamInterface* stream) {
   DCHECK(stream != NULL);
 
   CreateTrackReports<AudioTrackVector>(stream->GetAudioTracks(),
-                                       &reports_);
+                                       &reports_, track_ids_);
   CreateTrackReports<VideoTrackVector>(stream->GetVideoTracks(),
-                                       &reports_);
+                                       &reports_, track_ids_);
 }
 
 void StatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track,
@@ -463,6 +473,7 @@ StatsCollector::UpdateStats(PeerConnectionInterface::StatsOutputLevel level) {
     ExtractVoiceInfo();
     ExtractVideoInfo(level);
     ExtractDataInfo();
+    UpdateTrackReports();
   }
 }
 
@@ -866,6 +877,7 @@ void StatsCollector::UpdateStatsFromExistingLocalAudioTracks() {
     if (!v || v->string_val() != track->id())
       continue;
 
+    report->set_timestamp(stats_gathering_started_);
     UpdateReportFromAudioTrack(track, report);
   }
 }
@@ -913,6 +925,18 @@ bool StatsCollector::GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
   return true;
 }
 
+void StatsCollector::UpdateTrackReports() {
+  DCHECK(session_->signaling_thread()->IsCurrent());
+
+  rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
+
+  for (const auto& entry : track_ids_) {
+    StatsReport* report = entry.second;
+    report->set_timestamp(stats_gathering_started_);
+  }
+
+}
+
 void StatsCollector::ClearUpdateStatsCacheForTest() {
   stats_gathering_started_ = 0;
 }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.h b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.h
index 3c0aaf9b8ec..99130a3f013 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.h
@@ -52,6 +52,9 @@ const char* IceCandidateTypeToStatsType(const std::string& candidate_type);
 // only used by stats collector.
 const char* AdapterTypeToStatsType(rtc::AdapterType type);
 
+// A mapping between track ids and their StatsReport.
+typedef std::map<std::string, StatsReport*> TrackIdMap;
+
 class StatsCollector {
  public:
   // The caller is responsible for ensuring that the session outlives the
@@ -139,8 +142,12 @@ class StatsCollector {
   bool GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
                         StatsReport::Direction direction);
 
+  // Helper method to update the timestamp of track records.
+  void UpdateTrackReports();
+
   // A collection for all of our stats reports.
   StatsCollection reports_;
+  TrackIdMap track_ids_;
   // Raw pointer to the session the statistics are gathered from.
   WebRtcSession* const session_;
   double stats_gathering_started_;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector_unittest.cc
index 4e2bbedf7c9..315366ceca3 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector_unittest.cc
@@ -313,6 +313,12 @@ void VerifyVoiceReceiverInfoReport(
   EXPECT_TRUE(GetValue(
       report, StatsReport::kStatsValueNameSpeechExpandRate, &value_in_report));
   EXPECT_EQ(rtc::ToString<float>(info.speech_expand_rate), value_in_report);
+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAccelerateRate,
+                       &value_in_report));
+  EXPECT_EQ(rtc::ToString<float>(info.accelerate_rate), value_in_report);
+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNamePreemptiveExpandRate,
+                       &value_in_report));
+  EXPECT_EQ(rtc::ToString<float>(info.preemptive_expand_rate), value_in_report);
   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameSecondaryDecodedRate,
                        &value_in_report));
   EXPECT_EQ(rtc::ToString<float>(info.secondary_decoded_rate), value_in_report);
@@ -451,6 +457,8 @@ void InitVoiceReceiverInfo(cricket::VoiceReceiverInfo* voice_receiver_info) {
   voice_receiver_info->expand_rate = 121;
   voice_receiver_info->speech_expand_rate = 122;
   voice_receiver_info->secondary_decoded_rate = 123;
+  voice_receiver_info->accelerate_rate = 124;
+  voice_receiver_info->preemptive_expand_rate = 125;
 }
 
 class StatsCollectorForTest : public webrtc::StatsCollector {
@@ -589,6 +597,7 @@ class StatsCollectorTest : public testing::Test {
     const StatsReport* report = FindNthReportByType(
         *reports, StatsReport::kStatsReportTypeSsrc, 1);
     EXPECT_FALSE(report == NULL);
+    EXPECT_EQ(stats->GetTimeNow(), report->timestamp());
     std::string track_id = ExtractSsrcStatsValue(
         *reports, StatsReport::kStatsValueNameTrackId);
     EXPECT_EQ(audio_track->id(), track_id);
@@ -611,6 +620,7 @@ class StatsCollectorTest : public testing::Test {
     const StatsReport* track_report = FindNthReportByType(
         track_reports, StatsReport::kStatsReportTypeSsrc, 1);
     EXPECT_TRUE(track_report);
+    EXPECT_EQ(stats->GetTimeNow(), track_report->timestamp());
     track_id = ExtractSsrcStatsValue(track_reports,
                                      StatsReport::kStatsValueNameTrackId);
     EXPECT_EQ(audio_track->id(), track_id);
@@ -921,6 +931,7 @@ TEST_F(StatsCollectorTest, TrackObjectExistsWithoutUpdateStats) {
   stats.GetStats(NULL, &reports);
   EXPECT_EQ((size_t)1, reports.size());
   EXPECT_EQ(StatsReport::kStatsReportTypeTrack, reports[0]->type());
+  EXPECT_EQ(0, reports[0]->timestamp());
 
   std::string trackValue =
       ExtractStatsValue(StatsReport::kStatsReportTypeTrack,
@@ -983,6 +994,7 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
   track_report = FindNthReportByType(
       reports, StatsReport::kStatsReportTypeTrack, 1);
   EXPECT_TRUE(track_report);
+  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
 
   std::string ssrc_id = ExtractSsrcStatsValue(
       reports, StatsReport::kStatsValueNameSsrc);
@@ -1171,6 +1183,7 @@ TEST_F(StatsCollectorTest, ReportsFromRemoteTrack) {
   const StatsReport* track_report = FindNthReportByType(
       reports, StatsReport::kStatsReportTypeTrack, 1);
   EXPECT_TRUE(track_report);
+  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
 
   std::string ssrc_id = ExtractSsrcStatsValue(
       reports, StatsReport::kStatsValueNameSsrc);
@@ -1545,6 +1558,7 @@ TEST_F(StatsCollectorTest, GetStatsAfterRemoveAudioStream) {
   const StatsReport* report = FindNthReportByType(
       reports, StatsReport::kStatsReportTypeSsrc, 1);
   EXPECT_FALSE(report == NULL);
+  EXPECT_EQ(stats.GetTimeNow(), report->timestamp());
   std::string track_id = ExtractSsrcStatsValue(
       reports, StatsReport::kStatsValueNameTrackId);
   EXPECT_EQ(kLocalTrackId, track_id);
@@ -1622,6 +1636,7 @@ TEST_F(StatsCollectorTest, LocalAndRemoteTracksWithSameSsrc) {
   const StatsReport* track_report = FindNthReportByType(
       reports, StatsReport::kStatsReportTypeSsrc, 1);
   EXPECT_TRUE(track_report);
+  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
   std::string track_id = ExtractSsrcStatsValue(
       reports, StatsReport::kStatsValueNameTrackId);
   EXPECT_EQ(kLocalTrackId, track_id);
@@ -1633,6 +1648,7 @@ TEST_F(StatsCollectorTest, LocalAndRemoteTracksWithSameSsrc) {
   track_report = FindNthReportByType(reports,
                                      StatsReport::kStatsReportTypeSsrc, 1);
   EXPECT_TRUE(track_report);
+  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
   track_id = ExtractSsrcStatsValue(reports,
                                    StatsReport::kStatsValueNameTrackId);
   EXPECT_EQ(kRemoteTrackId, track_id);
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.cc
index 582d2033612..a902210478c 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.cc
@@ -408,6 +408,8 @@ const char* StatsReport::Value::display_name() const {
       return "datachannelid";
 
     // 'goog' prefixed constants.
+    case kStatsValueNameAccelerateRate:
+      return "googAccelerateRate";
     case kStatsValueNameActiveConnection:
       return "googActiveConnection";
     case kStatsValueNameActualEncBitrate:
@@ -544,6 +546,8 @@ const char* StatsReport::Value::display_name() const {
       return "googNacksReceived";
     case kStatsValueNameNacksSent:
       return "googNacksSent";
+    case kStatsValueNamePreemptiveExpandRate:
+      return "googPreemptiveExpandRate";
     case kStatsValueNamePlisReceived:
       return "googPlisReceived";
     case kStatsValueNamePlisSent:
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h b/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h
index 9df1de795e6..c58f39058b1 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h
@@ -131,6 +131,7 @@ class StatsReport {
     kStatsValueNameTransportId,
 
     // Internal StatsValue names.
+    kStatsValueNameAccelerateRate,
     kStatsValueNameActualEncBitrate,
     kStatsValueNameAdaptationChanges,
     kStatsValueNameAvailableReceiveBandwidth,
@@ -197,6 +198,7 @@ class StatsReport {
     kStatsValueNameNacksSent,
     kStatsValueNamePlisReceived,
     kStatsValueNamePlisSent,
+    kStatsValueNamePreemptiveExpandRate,
     kStatsValueNamePreferredJitterBufferMs,
     kStatsValueNameRemoteAddress,
     kStatsValueNameRemoteCandidateId,
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.cc
index c6339d3c3f4..47f17a1739a 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.cc
@@ -45,7 +45,7 @@ static const uint32 kAdmMaxIdleTimeProcess = 1000;
 // Constants here are derived by running VoE using a real ADM.
 // The constants correspond to 10ms of mono audio at 44kHz.
 static const int kTimePerFrameMs = 10;
-static const int kNumberOfChannels = 1;
+static const uint8_t kNumberOfChannels = 1;
 static const int kSamplesPerSecond = 44000;
 static const int kTotalDelayMs = 0;
 static const int kClockDriftMs = 0;
@@ -623,8 +623,8 @@ bool FakeAudioCaptureModule::Initialize() {
 
 void FakeAudioCaptureModule::SetSendBuffer(int value) {
   Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_);
-  const int buffer_size_in_samples = sizeof(send_buffer_) /
-      kNumberBytesPerSample;
+  const int buffer_size_in_samples =
+      sizeof(send_buffer_) / kNumberBytesPerSample;
   for (int i = 0; i < buffer_size_in_samples; ++i) {
     buffer_ptr[i] = value;
   }
@@ -636,8 +636,8 @@ void FakeAudioCaptureModule::ResetRecBuffer() {
 
 bool FakeAudioCaptureModule::CheckRecBuffer(int value) {
   const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_);
-  const int buffer_size_in_samples = sizeof(rec_buffer_) /
-      kNumberBytesPerSample;
+  const int buffer_size_in_samples =
+      sizeof(rec_buffer_) / kNumberBytesPerSample;
   for (int i = 0; i < buffer_size_in_samples; ++i) {
     if (buffer_ptr[i] >= value) return true;
   }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.h b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.h
index 57c1e58aafd..8ff4aa19e75 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule.h
@@ -58,8 +58,8 @@ class FakeAudioCaptureModule
 
   // The value for the following constants have been derived by running VoE
   // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
-  enum{kNumberSamples = 440};
-  enum{kNumberBytesPerSample = sizeof(Sample)};
+  static const int kNumberSamples = 440;
+  static const int kNumberBytesPerSample = sizeof(Sample);
 
   // Creates a FakeAudioCaptureModule or returns NULL on failure.
   // |process_thread| is used to push and pull audio frames to and from the
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index 8fcbfd70123..56f1d070edd 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -55,18 +55,18 @@ class FakeAdmTest : public testing::Test,
 
   // Callbacks inherited from webrtc::AudioTransport.
   // ADM is pushing data.
-  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                          const uint32_t nSamples,
-                                          const uint8_t nBytesPerSample,
-                                          const uint8_t nChannels,
-                                          const uint32_t samplesPerSec,
-                                          const uint32_t totalDelayMS,
-                                          const int32_t clockDrift,
-                                          const uint32_t currentMicLevel,
-                                          const bool keyPressed,
-                                          uint32_t& newMicLevel) {
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  const uint32_t nSamples,
+                                  const uint8_t nBytesPerSample,
+                                  const uint8_t nChannels,
+                                  const uint32_t samplesPerSec,
+                                  const uint32_t totalDelayMS,
+                                  const int32_t clockDrift,
+                                  const uint32_t currentMicLevel,
+                                  const bool keyPressed,
+                                  uint32_t& newMicLevel) override {
     rec_buffer_bytes_ = nSamples * nBytesPerSample;
-    if ((rec_buffer_bytes_ <= 0) ||
+    if ((rec_buffer_bytes_ == 0) ||
         (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
          FakeAudioCaptureModule::kNumberBytesPerSample)) {
       ADD_FAILURE();
@@ -79,14 +79,14 @@ class FakeAdmTest : public testing::Test,
   }
 
   // ADM is pulling data.
-  virtual int32_t NeedMorePlayData(const uint32_t nSamples,
-                                   const uint8_t nBytesPerSample,
-                                   const uint8_t nChannels,
-                                   const uint32_t samplesPerSec,
-                                   void* audioSamples,
-                                   uint32_t& nSamplesOut,
-                                   int64_t* elapsed_time_ms,
-                                   int64_t* ntp_time_ms) {
+  int32_t NeedMorePlayData(const uint32_t nSamples,
+                           const uint8_t nBytesPerSample,
+                           const uint8_t nChannels,
+                           const uint32_t samplesPerSec,
+                           void* audioSamples,
+                           uint32_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
     ++pull_iterations_;
     const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
     const uint32_t bytes_out = RecordedDataReceived() ?
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakedatachannelprovider.h b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakedatachannelprovider.h
index bf64a94e45d..eb86873c904 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakedatachannelprovider.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/test/fakedatachannelprovider.h
@@ -91,11 +91,15 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
   void set_send_blocked(bool blocked) {
     send_blocked_ = blocked;
     if (!blocked) {
-      std::set<webrtc::DataChannel*>::iterator it;
-      for (it = connected_channels_.begin();
-           it != connected_channels_.end();
-           ++it) {
-        (*it)->OnChannelReady(true);
+      // Take a snapshot of the connected channels and check to see whether
+      // each value is still in connected_channels_ before calling
+      // OnChannelReady().  This avoids problems where the set gets modified
+      // in response to OnChannelReady().
+      for (webrtc::DataChannel *ch : std::set<webrtc::DataChannel*>(
+               connected_channels_.begin(), connected_channels_.end())) {
+        if (connected_channels_.count(ch)) {
+          ch->OnChannelReady(true);
+        }
       }
     }
   }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/test/mockpeerconnectionobservers.h b/chromium/third_party/libjingle/source/talk/app/webrtc/test/mockpeerconnectionobservers.h
index f31b16c7443..d2697b4364e 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/test/mockpeerconnectionobservers.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/test/mockpeerconnectionobservers.h
@@ -98,8 +98,10 @@ class MockDataChannelObserver : public webrtc::DataChannelObserver {
     channel_->UnregisterObserver();
   }
 
-  virtual void OnStateChange() { state_ = channel_->state(); }
-  virtual void OnMessage(const DataBuffer& buffer) {
+  void OnBufferedAmountChange(uint64 previous_amount) override {}
+
+  void OnStateChange() override { state_ = channel_->state(); }
+  void OnMessage(const DataBuffer& buffer) override {
     last_message_.assign(buffer.data.data<char>(), buffer.data.size());
     ++received_message_count_;
   }
@@ -127,6 +129,7 @@ class MockStatsObserver : public webrtc::StatsObserver {
     stats_.number_of_reports = reports.size();
     for (const auto* r : reports) {
       if (r->type() == StatsReport::kStatsReportTypeSsrc) {
+        stats_.timestamp = r->timestamp();
         GetIntValue(r, StatsReport::kStatsValueNameAudioOutputLevel,
             &stats_.audio_output_level);
         GetIntValue(r, StatsReport::kStatsValueNameAudioInputLevel,
@@ -136,9 +139,11 @@ class MockStatsObserver : public webrtc::StatsObserver {
         GetIntValue(r, StatsReport::kStatsValueNameBytesSent,
             &stats_.bytes_sent);
       } else if (r->type() == StatsReport::kStatsReportTypeBwe) {
+        stats_.timestamp = r->timestamp();
         GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth,
             &stats_.available_receive_bandwidth);
       } else if (r->type() == StatsReport::kStatsReportTypeComponent) {
+        stats_.timestamp = r->timestamp();
         GetStringValue(r, StatsReport::kStatsValueNameDtlsCipher,
             &stats_.dtls_cipher);
         GetStringValue(r, StatsReport::kStatsValueNameSrtpCipher,
@@ -149,6 +154,7 @@ class MockStatsObserver : public webrtc::StatsObserver {
 
   bool called() const { return called_; }
   size_t number_of_reports() const { return stats_.number_of_reports; }
+  double timestamp() const { return stats_.timestamp; }
 
   int AudioOutputLevel() const {
     ASSERT(called_);
@@ -210,6 +216,7 @@ class MockStatsObserver : public webrtc::StatsObserver {
   struct {
     void Clear() {
       number_of_reports = 0;
+      timestamp = 0;
       audio_output_level = 0;
       audio_input_level = 0;
       bytes_received = 0;
@@ -220,6 +227,7 @@ class MockStatsObserver : public webrtc::StatsObserver {
     }
 
     size_t number_of_reports;
+    double timestamp;
     int audio_output_level;
     int audio_input_level;
     int bytes_received;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/umametrics.h b/chromium/third_party/libjingle/source/talk/app/webrtc/umametrics.h
index d029a698bdb..bb9e05230d3 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/umametrics.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/umametrics.h
@@ -65,6 +65,12 @@ enum PeerConnectionMetricsName {
   kTimeToConnect,           // In milliseconds.
   kLocalCandidates_IPv4,    // Number of IPv4 local candidates.
   kLocalCandidates_IPv6,    // Number of IPv6 local candidates.
+  kAudioSrtpCipher,         // Name of SRTP cipher used in audio channel.
+  kAudioSslCipher,          // Name of SSL cipher used in audio channel.
+  kVideoSrtpCipher,         // Name of SRTP cipher used in video channel.
+  kVideoSslCipher,          // Name of SSL cipher used in video channel.
+  kDataSrtpCipher,          // Name of SRTP cipher used in data channel.
+  kDataSslCipher,           // Name of SSL cipher used in data channel.
   kPeerConnectionMetricsName_Max
 };
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc
index 30aedcad194..7c78aea91fa 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc
@@ -29,8 +29,6 @@
 
 #include <string>
 
-#include "talk/media/webrtc/webrtcvideocapturer.h"
-
 namespace webrtc {
 
 static const char kVideoTrackKind[] = "video";
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc
index a93977d6d98..b6f23ca9d00 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc
@@ -539,27 +539,11 @@ static bool AddSsrcLine(uint32 ssrc_id, const std::string& attribute,
   return AddLine(os.str(), message);
 }
 
-// Split the message into two parts by the first delimiter.
-static bool SplitByDelimiter(const std::string& message,
-                             const char delimiter,
-                             std::string* field1,
-                             std::string* field2) {
-  // Find the first delimiter
-  size_t pos = message.find(delimiter);
-  if (pos == std::string::npos) {
-    return false;
-  }
-  *field1 = message.substr(0, pos);
-  // The rest is the value.
-  *field2 = message.substr(pos + 1);
-  return true;
-}
-
 // Get value only from <attribute>:<value>.
 static bool GetValue(const std::string& message, const std::string& attribute,
                      std::string* value, SdpParseError* error) {
   std::string leftpart;
-  if (!SplitByDelimiter(message, kSdpDelimiterColon, &leftpart, value)) {
+  if (!rtc::tokenize_first(message, kSdpDelimiterColon, &leftpart, value)) {
     return ParseFailedGetValue(message, attribute, error);
   }
   // The left part should end with the expected attribute.
@@ -972,7 +956,8 @@ bool ParseCandidate(const std::string& message, Candidate* candidate,
   // Makes sure |message| contains only one line.
   if (message.size() > first_line.size()) {
     std::string left, right;
-    if (SplitByDelimiter(message, kNewLine, &left, &right) && !right.empty()) {
+    if (rtc::tokenize_first(message, kNewLine, &left, &right) &&
+        !right.empty()) {
       return ParseFailed(message, 0, "Expect one line only", error);
     }
   }
@@ -989,8 +974,8 @@ bool ParseCandidate(const std::string& message, Candidate* candidate,
   std::string candidate_value;
 
   // |first_line| must be in the form of "candidate:<value>".
-  if (!SplitByDelimiter(first_line, kSdpDelimiterColon,
-                        &attribute_candidate, &candidate_value) ||
+  if (!rtc::tokenize_first(first_line, kSdpDelimiterColon, &attribute_candidate,
+                           &candidate_value) ||
       attribute_candidate != kAttributeCandidate) {
     if (is_raw) {
       std::ostringstream description;
@@ -1289,16 +1274,7 @@ void BuildMediaDescription(const ContentInfo* content_info,
 
   // RFC 4566
   // b=AS:<bandwidth>
-  // We should always use the default bandwidth for RTP-based data
-  // channels.  Don't allow SDP to set the bandwidth, because that
-  // would give JS the opportunity to "break the Internet".
-  // TODO(pthatcher): But we need to temporarily allow the SDP to control
-  // this for backwards-compatibility.  Once we don't need that any
-  // more, remove this.
-  bool support_dc_sdp_bandwidth_temporarily = true;
-  if (media_desc->bandwidth() >= 1000 &&
-      (media_type != cricket::MEDIA_TYPE_DATA ||
-       support_dc_sdp_bandwidth_temporarily)) {
+  if (media_desc->bandwidth() >= 1000) {
     InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
     os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
     AddLine(os.str(), message);
@@ -2216,7 +2192,7 @@ bool ParseMediaDescription(const std::string& message,
       for (size_t j = 3 ; j < fields.size(); ++j) {
         // TODO(wu): Remove when below bug is fixed.
         // https://bugzilla.mozilla.org/show_bug.cgi?id=996329
-        if (fields[j] == "" && j == fields.size() - 1) {
+        if (fields[j].empty() && j == fields.size() - 1) {
           continue;
         }
 
@@ -2264,17 +2240,6 @@ bool ParseMediaDescription(const std::string& message,
         if (!AddSctpDataCodec(data_desc, p))
           return false;
       }
-
-      // We should always use the default bandwidth for RTP-based data
-      // channels.  Don't allow SDP to set the bandwidth, because that
-      // would give JS the opportunity to "break the Internet".
-      // TODO(pthatcher): But we need to temporarily allow the SDP to control
-      // this for backwards-compatibility.  Once we don't need that any
-      // more, remove this.
-      bool support_dc_sdp_bandwidth_temporarily = true;
-      if (content.get() && !support_dc_sdp_bandwidth_temporarily) {
-        content->set_bandwidth(cricket::kAutoBandwidth);
-      }
     } else {
       LOG(LS_WARNING) << "Unsupported media type: " << line;
       continue;
@@ -2532,6 +2497,17 @@ bool ParseContent(const std::string& message,
           if (!GetValueFromString(line, bandwidth, &b, error)) {
             return false;
           }
+          // We should never use more than the default bandwidth for RTP-based
+          // data channels. Don't allow SDP to set the bandwidth, because
+          // that would give JS the opportunity to "break the Internet".
+          // See: https://code.google.com/p/chromium/issues/detail?id=280726
+          if (media_type == cricket::MEDIA_TYPE_DATA && IsRtp(protocol) &&
+              b > cricket::kDataMaxBandwidth / 1000) {
+            std::ostringstream description;
+            description << "RTP-based data channels may not send more than "
+                        << cricket::kDataMaxBandwidth / 1000 << "kbps.";
+            return ParseFailed(line, description.str(), error);
+          }
           media_desc->set_bandwidth(b * 1000);
         }
       }
@@ -2749,10 +2725,8 @@ bool ParseSsrcAttribute(const std::string& line, SsrcInfoVec* ssrc_infos,
   // a=ssrc:<ssrc-id> <attribute>
   // a=ssrc:<ssrc-id> <attribute>:<value>
   std::string field1, field2;
-  if (!SplitByDelimiter(line.substr(kLinePrefixLength),
-                        kSdpDelimiterSpace,
-                        &field1,
-                        &field2)) {
+  if (!rtc::tokenize_first(line.substr(kLinePrefixLength), kSdpDelimiterSpace,
+                           &field1, &field2)) {
     const size_t expected_fields = 2;
     return ParseFailedExpectFieldNum(line, expected_fields, error);
   }
@@ -2769,8 +2743,7 @@ bool ParseSsrcAttribute(const std::string& line, SsrcInfoVec* ssrc_infos,
 
   std::string attribute;
   std::string value;
-  if (!SplitByDelimiter(field2, kSdpDelimiterColon,
-                        &attribute, &value)) {
+  if (!rtc::tokenize_first(field2, kSdpDelimiterColon, &attribute, &value)) {
     std::ostringstream description;
     description << "Failed to get the ssrc attribute value from " << field2
                 << ". Expected format <attribute>:<value>.";
@@ -3016,22 +2989,13 @@ bool ParseRtpmapAttribute(const std::string& line,
   return true;
 }
 
-void PruneRight(const char delimiter, std::string* message) {
-  size_t trailing = message->find(delimiter);
-  if (trailing != std::string::npos) {
-    *message = message->substr(0, trailing);
-  }
-}
-
 bool ParseFmtpParam(const std::string& line, std::string* parameter,
                     std::string* value, SdpParseError* error) {
-  if (!SplitByDelimiter(line, kSdpDelimiterEqual, parameter, value)) {
+  if (!rtc::tokenize_first(line, kSdpDelimiterEqual, parameter, value)) {
     ParseFailed(line, "Unable to parse fmtp parameter. \'=\' missing.", error);
     return false;
   }
   // a=fmtp:<payload_type> <param1>=<value1>; <param2>=<value2>; ...
-  // When parsing the values the trailing ";" gets picked up. Remove them.
-  PruneRight(kSdpDelimiterSemicolon, value);
   return true;
 }
 
@@ -3042,44 +3006,52 @@ bool ParseFmtpAttributes(const std::string& line, const MediaType media_type,
       media_type != cricket::MEDIA_TYPE_VIDEO) {
     return true;
   }
-  std::vector<std::string> fields;
-  rtc::split(line.substr(kLinePrefixLength),
-                   kSdpDelimiterSpace, &fields);
+
+  std::string line_payload;
+  std::string line_params;
 
   // RFC 5576
   // a=fmtp:<format> <format specific parameters>
   // At least two fields, whereas the second one is any of the optional
   // parameters.
-  if (fields.size() < 2) {
+  if (!rtc::tokenize_first(line.substr(kLinePrefixLength), kSdpDelimiterSpace,
+                           &line_payload, &line_params)) {
     ParseFailedExpectMinFieldNum(line, 2, error);
     return false;
   }
 
+  // Parse out the payload information.
   std::string payload_type_str;
-  if (!GetValue(fields[0], kAttributeFmtp, &payload_type_str, error)) {
+  if (!GetValue(line_payload, kAttributeFmtp, &payload_type_str, error)) {
     return false;
   }
 
+  int payload_type = 0;
+  if (!GetPayloadTypeFromString(line_payload, payload_type_str, &payload_type,
+                                error)) {
+    return false;
+  }
+
+  // Parse out format specific parameters.
+  std::vector<std::string> fields;
+  rtc::split(line_params, kSdpDelimiterSemicolon, &fields);
+
   cricket::CodecParameterMap codec_params;
-  for (std::vector<std::string>::const_iterator iter = fields.begin() + 1;
-       iter != fields.end(); ++iter) {
-    std::string name;
-    std::string value;
-    if (iter->find(kSdpDelimiterEqual) == std::string::npos) {
+  for (auto& iter : fields) {
+    if (iter.find(kSdpDelimiterEqual) == std::string::npos) {
       // Only fmtps with equals are currently supported. Other fmtp types
       // should be ignored. Unknown fmtps do not constitute an error.
       continue;
     }
-    if (!ParseFmtpParam(*iter, &name, &value, error)) {
+
+    std::string name;
+    std::string value;
+    if (!ParseFmtpParam(rtc::string_trim(iter), &name, &value, error)) {
       return false;
     }
     codec_params[name] = value;
   }
 
-  int payload_type = 0;
-  if (!GetPayloadTypeFromString(line, payload_type_str, &payload_type, error)) {
-    return false;
-  }
   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
     UpdateCodec<AudioContentDescription, cricket::AudioCodec>(
         media_desc, payload_type, codec_params);
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc
index 862e93c539b..b6577682041 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc
@@ -1207,11 +1207,10 @@ class WebRtcSdpTest : public testing::Test {
         "a=fmtp:111 0-15,66,70\r\n"
         "a=fmtp:111 ";
     std::ostringstream os;
-    os << "minptime=" << params.min_ptime
-       << "; stereo=" << params.stereo
+    os << "minptime=" << params.min_ptime << "; stereo=" << params.stereo
        << "; sprop-stereo=" << params.sprop_stereo
        << "; useinbandfec=" << params.useinband
-       << " maxaveragebitrate=" << params.maxaveragebitrate << "\r\n"
+       << "; maxaveragebitrate=" << params.maxaveragebitrate << "\r\n"
        << "a=ptime:" << params.ptime << "\r\n"
        << "a=maxptime:" << params.max_ptime << "\r\n";
     sdp += os.str();
@@ -1222,7 +1221,7 @@ class WebRtcSdpTest : public testing::Test {
     os << "m=video 9 RTP/SAVPF 99 95\r\n"
        << "a=rtpmap:99 VP8/90000\r\n"
        << "a=rtpmap:95 RTX/90000\r\n"
-       << "a=fmtp:95 apt=99;rtx-time=1000\r\n";
+       << "a=fmtp:95 apt=99;\r\n";
     sdp += os.str();
 
     // Deserialize
@@ -1699,12 +1698,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
 
   std::string expected_sdp = kSdpString;
   expected_sdp.append(kSdpRtpDataChannelString);
-  // We want to test that serializing data content ignores bandwidth
-  // settings (it should always be the default).  Thus, we don't do
-  // the following:
-  // TODO(pthatcher): We need to temporarily allow the SDP to control
-  // this for backwards-compatibility.  Once we don't need that any
-  // more, remove this.
+  // Serializing data content shouldn't ignore bandwidth settings.
   InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
               "b=AS:100\r\n",
               &expected_sdp);
@@ -2260,28 +2254,39 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelAndNewPort) {
 }
 
 TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannelsAndBandwidth) {
-  AddRtpDataChannel();
+  // We want to test that deserializing data content limits bandwidth
+  // settings (it should never be greater than the default).
+  // This should prevent someone from using unlimited data bandwidth through
+  // JS and "breaking the Internet".
+  // See: https://code.google.com/p/chromium/issues/detail?id=280726
+  std::string sdp_with_bandwidth = kSdpString;
+  sdp_with_bandwidth.append(kSdpRtpDataChannelString);
+  InjectAfter("a=mid:data_content_name\r\n",
+              "b=AS:100\r\n",
+              &sdp_with_bandwidth);
+  JsepSessionDescription jdesc_with_bandwidth(kDummyString);
+
+  EXPECT_FALSE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
+}
+
+TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) {
+  AddSctpDataChannel();
   JsepSessionDescription jdesc(kDummyString);
-  // We want to test that deserializing data content ignores bandwidth
-  // settings (it should always be the default).  Thus, we don't do
-  // the following:
-  // TODO(pthatcher): We need to temporarily allow the SDP to control
-  // this for backwards-compatibility.  Once we don't need that any
-  // more, remove this.
   DataContentDescription* dcd = static_cast<DataContentDescription*>(
      GetFirstDataContent(&desc_)->description);
   dcd->set_bandwidth(100 * 1000);
   ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
 
   std::string sdp_with_bandwidth = kSdpString;
-  sdp_with_bandwidth.append(kSdpRtpDataChannelString);
+  sdp_with_bandwidth.append(kSdpSctpDataChannelString);
   InjectAfter("a=mid:data_content_name\r\n",
               "b=AS:100\r\n",
               &sdp_with_bandwidth);
   JsepSessionDescription jdesc_with_bandwidth(kDummyString);
 
-  EXPECT_TRUE(
-      SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
+  // SCTP has congestion control, so we shouldn't limit the bandwidth
+  // as we do for RTP.
+  EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_with_bandwidth));
 }
 
@@ -2504,7 +2509,41 @@ TEST_F(WebRtcSdpTest, DeserializeVideoFmtp) {
       "t=0 0\r\n"
       "m=video 3457 RTP/SAVPF 120\r\n"
       "a=rtpmap:120 VP8/90000\r\n"
-      "a=fmtp:120 x-google-min-bitrate=10; x-google-max-quantization=40\r\n";
+      "a=fmtp:120 x-google-min-bitrate=10;x-google-max-quantization=40\r\n";
+
+  // Deserialize
+  SdpParseError error;
+  EXPECT_TRUE(
+      webrtc::SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error));
+
+  const ContentInfo* vc = GetFirstVideoContent(jdesc_output.description());
+  ASSERT_TRUE(vc != NULL);
+  const VideoContentDescription* vcd =
+      static_cast<const VideoContentDescription*>(vc->description);
+  ASSERT_FALSE(vcd->codecs().empty());
+  cricket::VideoCodec vp8 = vcd->codecs()[0];
+  EXPECT_EQ("VP8", vp8.name);
+  EXPECT_EQ(120, vp8.id);
+  cricket::CodecParameterMap::iterator found =
+      vp8.params.find("x-google-min-bitrate");
+  ASSERT_TRUE(found != vp8.params.end());
+  EXPECT_EQ(found->second, "10");
+  found = vp8.params.find("x-google-max-quantization");
+  ASSERT_TRUE(found != vp8.params.end());
+  EXPECT_EQ(found->second, "40");
+}
+
+TEST_F(WebRtcSdpTest, DeserializeVideoFmtpWithSpace) {
+  JsepSessionDescription jdesc_output(kDummyString);
+
+  const char kSdpWithFmtpString[] =
+      "v=0\r\n"
+      "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
+      "s=-\r\n"
+      "t=0 0\r\n"
+      "m=video 3457 RTP/SAVPF 120\r\n"
+      "a=rtpmap:120 VP8/90000\r\n"
+      "a=fmtp:120   x-google-min-bitrate=10;  x-google-max-quantization=40\r\n";
 
   // Deserialize
   SdpParseError error;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc
index e2a9d60f2c8..25333289689 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc
@@ -44,6 +44,7 @@
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
@@ -489,6 +490,7 @@ WebRtcSession::WebRtcSession(
       mediastream_signaling_(mediastream_signaling),
       ice_observer_(NULL),
       ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
+      ice_connection_receiving_(true),
       older_version_remote_peer_(false),
       dtls_enabled_(false),
       data_channel_type_(cricket::DCT_NONE),
@@ -497,6 +499,7 @@ WebRtcSession::WebRtcSession(
 }
 
 WebRtcSession::~WebRtcSession() {
+  ASSERT(signaling_thread()->IsCurrent());
   // Destroy video_channel_ first since it may have a pointer to the
   // voice_channel_.
   if (video_channel_) {
@@ -523,6 +526,8 @@ bool WebRtcSession::Initialize(
     DTLSIdentityServiceInterface* dtls_identity_service,
     const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
   bundle_policy_ = rtc_configuration.bundle_policy;
+  rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
+  SetSslMaxProtocolVersion(options.ssl_max_version);
 
   // TODO(perkj): Take |constraints| into consideration. Return false if not all
   // mandatory constraints can be fulfilled. Note that |constraints|
@@ -642,6 +647,9 @@ bool WebRtcSession::Initialize(
   audio_options_.audio_jitter_buffer_max_packets.Set(
       rtc_configuration.audio_jitter_buffer_max_packets);
 
+  audio_options_.audio_jitter_buffer_fast_accelerate.Set(
+      rtc_configuration.audio_jitter_buffer_fast_accelerate);
+
   const cricket::VideoCodec default_codec(
       JsepSessionDescription::kDefaultVideoCodecId,
       JsepSessionDescription::kDefaultVideoCodecName,
@@ -1387,7 +1395,11 @@ void WebRtcSession::OnTransportCompleted(cricket::Transport* transport) {
   SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
   // Only report once when Ice connection is completed.
   if (old_state != PeerConnectionInterface::kIceConnectionCompleted) {
-    ReportBestConnectionState(transport);
+    cricket::TransportStats stats;
+    if (metrics_observer_ && transport->GetStats(&stats)) {
+      ReportBestConnectionState(stats);
+      ReportNegotiatedCiphers(stats);
+    }
   }
 }
 
@@ -1396,6 +1408,31 @@ void WebRtcSession::OnTransportFailed(cricket::Transport* transport) {
   SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
 }
 
+void WebRtcSession::OnTransportReceiving(cricket::Transport* transport) {
+  ASSERT(signaling_thread()->IsCurrent());
+  // The ice connection is considered receiving if at least one transport is
+  // receiving on any channels.
+  bool receiving = false;
+  for (const auto& kv : transport_proxies()) {
+    cricket::Transport* transport = kv.second->impl();
+    if (transport && transport->any_channel_receiving()) {
+      receiving = true;
+      break;
+    }
+  }
+  SetIceConnectionReceiving(receiving);
+}
+
+void WebRtcSession::SetIceConnectionReceiving(bool receiving) {
+  if (ice_connection_receiving_ == receiving) {
+    return;
+  }
+  ice_connection_receiving_ = receiving;
+  if (ice_observer_) {
+    ice_observer_->OnIceConnectionReceivingChange(receiving);
+  }
+}
+
 void WebRtcSession::OnTransportProxyCandidatesReady(
     cricket::TransportProxy* proxy, const cricket::Candidates& candidates) {
   ASSERT(signaling_thread()->IsCurrent());
@@ -1600,6 +1637,18 @@ bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
     }
   }
 
+  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 before when kMaxBundle policy is in effect.
   if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {
     const cricket::ContentGroup* bundle_group = desc->GetGroupByName(
@@ -1619,12 +1668,11 @@ bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
 
 bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content) {
   voice_channel_.reset(channel_manager_->CreateVoiceChannel(
-      this, content->name, true));
+      this, content->name, true, audio_options_));
   if (!voice_channel_) {
     return false;
   }
 
-  voice_channel_->SetChannelOptions(audio_options_);
   voice_channel_->SignalDtlsSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSetupFailure);
   return true;
@@ -1856,16 +1904,9 @@ bool WebRtcSession::ReadyToUseRemoteCandidate(
 
 // Walk through the ConnectionInfos to gather best connection usage
 // for IPv4 and IPv6.
-void WebRtcSession::ReportBestConnectionState(cricket::Transport* transport) {
-  if (!metrics_observer_) {
-    return;
-  }
-
-  cricket::TransportStats stats;
-  if (!transport->GetStats(&stats)) {
-    return;
-  }
-
+void WebRtcSession::ReportBestConnectionState(
+    const cricket::TransportStats& stats) {
+  DCHECK(metrics_observer_ != NULL);
   for (cricket::TransportChannelStatsList::const_iterator it =
          stats.channel_stats.begin();
        it != stats.channel_stats.end(); ++it) {
@@ -1881,11 +1922,48 @@ void WebRtcSession::ReportBestConnectionState(cricket::Transport* transport) {
                  AF_INET6) {
         metrics_observer_->IncrementCounter(kBestConnections_IPv6);
       } else {
-        ASSERT(false);
+        RTC_NOTREACHED();
       }
       return;
     }
   }
 }
 
+void WebRtcSession::ReportNegotiatedCiphers(
+    const cricket::TransportStats& stats) {
+  DCHECK(metrics_observer_ != NULL);
+  if (!dtls_enabled_ || stats.channel_stats.empty()) {
+    return;
+  }
+
+  const std::string& srtp_cipher = stats.channel_stats[0].srtp_cipher;
+  const std::string& ssl_cipher = stats.channel_stats[0].ssl_cipher;
+  if (srtp_cipher.empty() && ssl_cipher.empty()) {
+    return;
+  }
+
+  PeerConnectionMetricsName srtp_name;
+  PeerConnectionMetricsName ssl_name;
+  if (stats.content_name == cricket::CN_AUDIO) {
+    srtp_name = kAudioSrtpCipher;
+    ssl_name = kAudioSslCipher;
+  } else if (stats.content_name == cricket::CN_VIDEO) {
+    srtp_name = kVideoSrtpCipher;
+    ssl_name = kVideoSslCipher;
+  } else if (stats.content_name == cricket::CN_DATA) {
+    srtp_name = kDataSrtpCipher;
+    ssl_name = kDataSslCipher;
+  } else {
+    RTC_NOTREACHED();
+    return;
+  }
+
+  if (!srtp_cipher.empty()) {
+    metrics_observer_->AddHistogramSample(srtp_name, srtp_cipher);
+  }
+  if (!ssl_cipher.empty()) {
+    metrics_observer_->AddHistogramSample(ssl_name, ssl_cipher);
+  }
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h
index aa1deb523c2..30ebc1e49ee 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h
@@ -84,6 +84,8 @@ class IceObserver {
  public:
   IceObserver() {}
   // Called any time the IceConnectionState changes
+  // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
+  // conform to the w3c standard.
   virtual void OnIceConnectionChange(
       PeerConnectionInterface::IceConnectionState new_state) {}
   // Called any time the IceGatheringState changes
@@ -96,6 +98,9 @@ class IceObserver {
   // (via PeerConnectionObserver)
   virtual void OnIceComplete() {}
 
+  // Called whenever the state changes between receiving and not receiving.
+  virtual void OnIceConnectionReceivingChange(bool receiving) {}
+
  protected:
   ~IceObserver() {}
 
@@ -253,19 +258,6 @@ class WebRtcSession : public cricket::BaseSession,
     metrics_observer_ = metrics_observer;
   }
 
- protected:
-  // Don't fire a new description.  The only thing it's used for is to
-  // push new media descriptions to the BaseChannels.  But in
-  // WebRtcSession, we just push to the BaseChannels directly, so we
-  // don't need this (and it would cause the descriptions to be pushed
-  // down twice).
-  // TODO(pthatcher): Remove this method and signal completely from
-  // BaseSession once all the subclasses of BaseSession push to
-  // BaseChannels directly rather than relying on the signal, or once
-  // BaseChannel no longer listens to the event and requires
-  // descriptions to be pushed down.
-  virtual void SignalNewDescription() override {}
-
  private:
   // Indicates the type of SessionDescription in a call to SetLocalDescription
   // and SetRemoteDescription.
@@ -298,9 +290,8 @@ class WebRtcSession : public cricket::BaseSession,
       cricket::TransportProxy* proxy,
       const cricket::Candidates& candidates);
   virtual void OnCandidatesAllocationDone();
+  void OnTransportReceiving(cricket::Transport* transport) override;
 
-  // Creates local session description with audio and video contents.
-  bool CreateDefaultLocalDescription();
   // Enables media channels to allow sending of media.
   void EnableChannels();
   // Creates a JsepIceCandidate and adds it to the local session description
@@ -344,6 +335,7 @@ class WebRtcSession : public cricket::BaseSession,
 
   std::string BadStateErrMsg(State state);
   void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
+  void SetIceConnectionReceiving(bool receiving);
 
   bool ValidateBundleSettings(const cricket::SessionDescription* desc);
   bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
@@ -372,7 +364,9 @@ class WebRtcSession : public cricket::BaseSession,
 
   // Invoked when OnTransportCompleted is signaled to gather the usage
   // of IPv4/IPv6 as best connection.
-  void ReportBestConnectionState(cricket::Transport* transport);
+  void ReportBestConnectionState(const cricket::TransportStats& stats);
+
+  void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
 
   rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
   rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
@@ -381,6 +375,7 @@ class WebRtcSession : public cricket::BaseSession,
   MediaStreamSignaling* mediastream_signaling_;
   IceObserver* ice_observer_;
   PeerConnectionInterface::IceConnectionState ice_connection_state_;
+  bool ice_connection_receiving_;
   rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
   rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
   // Candidates that arrived before the remote description was set.
@@ -413,6 +408,9 @@ class WebRtcSession : public cricket::BaseSession,
   // Declares the bundle policy for the WebRTCSession.
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
+  // Declares the RTCP mux policy for the WebRTCSession.
+  PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc
index e4f39f822b0..0111f53cee3 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc
@@ -26,6 +26,7 @@
  */
 
 #include "talk/app/webrtc/audiotrack.h"
+#include "talk/app/webrtc/fakemetricsobserver.h"
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
 #include "talk/app/webrtc/mediastreamsignaling.h"
@@ -81,6 +82,7 @@ using webrtc::CreateSessionDescriptionRequest;
 using webrtc::DTLSIdentityRequestObserver;
 using webrtc::DTLSIdentityServiceInterface;
 using webrtc::FakeConstraints;
+using webrtc::FakeMetricsObserver;
 using webrtc::IceCandidateCollection;
 using webrtc::JsepIceCandidate;
 using webrtc::JsepSessionDescription;
@@ -156,8 +158,6 @@ static const char kSdpWithRtx[] =
     "a=rtpmap:96 rtx/90000\r\n"
     "a=fmtp:96 apt=0\r\n";
 
-static const int kAudioJitterBufferMaxPackets = 50;
-
 // Add some extra |newlines| to the |message| after |line|.
 static void InjectAfter(const std::string& line,
                         const std::string& newlines,
@@ -167,33 +167,6 @@ static void InjectAfter(const std::string& line,
                              tmp.c_str(), tmp.length(), message);
 }
 
-class FakeMetricsObserver : public webrtc::MetricsObserverInterface {
- public:
-  FakeMetricsObserver() { Reset(); }
-  void Reset() {
-    memset(peer_connection_metrics_counters_, 0,
-           sizeof(peer_connection_metrics_counters_));
-    memset(peer_connection_metrics_name_, 0,
-           sizeof(peer_connection_metrics_name_));
-  }
-
-  void IncrementCounter(webrtc::PeerConnectionMetricsCounter type) override {
-    peer_connection_metrics_counters_[type]++;
-  }
-  void AddHistogramSample(webrtc::PeerConnectionMetricsName type,
-                          int value) override {
-    ASSERT(peer_connection_metrics_name_[type] == 0);
-    peer_connection_metrics_name_[type] = value;
-  }
-
-  int peer_connection_metrics_counters_
-      [webrtc::kPeerConnectionMetricsCounter_Max];
-  int peer_connection_metrics_name_[webrtc::kPeerConnectionMetricsCounter_Max];
-
-  int AddRef() override { return 1; }
-  int Release() override { return 1; }
-};
-
 class MockIceObserver : public webrtc::IceObserver {
  public:
   MockIceObserver()
@@ -362,7 +335,8 @@ class WebRtcSessionTest : public testing::Test {
       stun_server_(cricket::TestStunServer::Create(Thread::Current(),
                                                    stun_socket_addr_)),
       turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
-      mediastream_signaling_(channel_manager_.get()) {
+      mediastream_signaling_(channel_manager_.get()),
+      metrics_observer_(new rtc::RefCountedObject<FakeMetricsObserver>()) {
     tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID);
 
     cricket::ServerAddresses stun_servers;
@@ -400,16 +374,11 @@ class WebRtcSessionTest : public testing::Test {
 
     EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
                                      identity_service, rtc_configuration));
-    session_->set_metrics_observer(&metrics_observer_);
+    session_->set_metrics_observer(metrics_observer_);
   }
 
   void Init() {
     PeerConnectionInterface::RTCConfiguration configuration;
-    configuration.type = PeerConnectionInterface::kAll;
-    configuration.bundle_policy =
-        PeerConnectionInterface::kBundlePolicyBalanced;
-    configuration.audio_jitter_buffer_max_packets =
-        kAudioJitterBufferMaxPackets;
     Init(NULL, configuration);
   }
 
@@ -417,20 +386,20 @@ class WebRtcSessionTest : public testing::Test {
       PeerConnectionInterface::IceTransportsType ice_transport_type) {
     PeerConnectionInterface::RTCConfiguration configuration;
     configuration.type = ice_transport_type;
-    configuration.bundle_policy =
-        PeerConnectionInterface::kBundlePolicyBalanced;
-    configuration.audio_jitter_buffer_max_packets =
-        kAudioJitterBufferMaxPackets;
     Init(NULL, configuration);
   }
 
   void InitWithBundlePolicy(
       PeerConnectionInterface::BundlePolicy bundle_policy) {
     PeerConnectionInterface::RTCConfiguration configuration;
-    configuration.type = PeerConnectionInterface::kAll;
     configuration.bundle_policy = bundle_policy;
-    configuration.audio_jitter_buffer_max_packets =
-        kAudioJitterBufferMaxPackets;
+    Init(NULL, configuration);
+  }
+
+  void InitWithRtcpMuxPolicy(
+      PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy) {
+    PeerConnectionInterface::RTCConfiguration configuration;
+    configuration.rtcp_mux_policy = rtcp_mux_policy;
     Init(NULL, configuration);
   }
 
@@ -438,11 +407,6 @@ class WebRtcSessionTest : public testing::Test {
     FakeIdentityService* identity_service = new FakeIdentityService();
     identity_service->set_should_fail(identity_request_should_fail);
     PeerConnectionInterface::RTCConfiguration configuration;
-    configuration.type = PeerConnectionInterface::kAll;
-    configuration.bundle_policy =
-        PeerConnectionInterface::kBundlePolicyBalanced;
-    configuration.audio_jitter_buffer_max_packets =
-        kAudioJitterBufferMaxPackets;
     Init(identity_service, configuration);
   }
 
@@ -1034,20 +998,18 @@ class WebRtcSessionTest : public testing::Test {
     ExpectedBestConnection best_connection_after_initial_ice_converged_;
 
     void VerifyBestConnectionAfterIceConverge(
-        const FakeMetricsObserver& metrics_observer) const {
+        const rtc::scoped_refptr<FakeMetricsObserver> metrics_observer) const {
       Verify(metrics_observer, best_connection_after_initial_ice_converged_);
     }
 
    private:
-    void Verify(const FakeMetricsObserver& metrics_observer,
+    void Verify(const rtc::scoped_refptr<FakeMetricsObserver> metrics_observer,
                 const ExpectedBestConnection& expected) const {
       EXPECT_EQ(
-          metrics_observer
-              .peer_connection_metrics_counters_[webrtc::kBestConnections_IPv4],
+          metrics_observer->GetCounter(webrtc::kBestConnections_IPv4),
           expected.ipv4_count_);
       EXPECT_EQ(
-          metrics_observer
-              .peer_connection_metrics_counters_[webrtc::kBestConnections_IPv6],
+          metrics_observer->GetCounter(webrtc::kBestConnections_IPv6),
           expected.ipv6_count_);
     }
   };
@@ -1147,7 +1109,7 @@ class WebRtcSessionTest : public testing::Test {
                    observer_.ice_connection_state_,
                    kIceCandidatesTimeout);
 
-    metrics_observer_.Reset();
+    metrics_observer_->Reset();
 
     // Clearing the rules, session should move back to completed state.
     loopback_network_manager.ClearRules(fss_.get());
@@ -1292,7 +1254,7 @@ class WebRtcSessionTest : public testing::Test {
   MockIceObserver observer_;
   cricket::FakeVideoMediaChannel* video_channel_;
   cricket::FakeVoiceMediaChannel* voice_channel_;
-  FakeMetricsObserver metrics_observer_;
+  rtc::scoped_refptr<FakeMetricsObserver> metrics_observer_;
 };
 
 TEST_F(WebRtcSessionTest, TestInitializeWithDtls) {
@@ -2614,6 +2576,89 @@ TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionInvalidIceCredentials) {
   EXPECT_FALSE(session_->SetRemoteDescription(modified_offer, &error));
 }
 
+// Test that candidates sent to the "video" transport do not get pushed down to
+// the "audio" transport channel when bundling using TransportProxy.
+TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
+  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
+
+  InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  options.use_rtp_mux = true;
+
+  SessionDescriptionInterface* offer = CreateRemoteOffer();
+  SetRemoteDescriptionWithoutError(offer);
+
+  SessionDescriptionInterface* answer = CreateAnswer(NULL);
+  SetLocalDescriptionWithoutError(answer);
+
+  EXPECT_EQ(session_->GetTransportProxy("audio")->impl(),
+            session_->GetTransportProxy("video")->impl());
+
+  cricket::Transport* t = session_->GetTransport("audio");
+
+  // Checks if one of the transport channels contains a connection using a given
+  // port.
+  auto connection_with_remote_port = [t](int port) {
+    cricket::TransportStats stats;
+    t->GetStats(&stats);
+    for (auto& chan_stat : stats.channel_stats) {
+      for (auto& conn_info : chan_stat.connection_infos) {
+        if (conn_info.remote_candidate.address().port() == port) {
+          return true;
+        }
+      }
+    }
+    return false;
+  };
+
+  EXPECT_FALSE(connection_with_remote_port(5000));
+  EXPECT_FALSE(connection_with_remote_port(5001));
+  EXPECT_FALSE(connection_with_remote_port(6000));
+
+  // The way the *_WAIT checks work is they only wait if the condition fails,
+  // which does not help in the case where state is not changing. This is
+  // problematic in this test since we want to verify that adding a video
+  // candidate does _not_ change state. So we interleave candidates and assume
+  // that messages are executed in the order they were posted.
+
+  // First audio candidate.
+  cricket::Candidate candidate0;
+  candidate0.set_address(rtc::SocketAddress("1.1.1.1", 5000));
+  candidate0.set_component(1);
+  candidate0.set_protocol("udp");
+  JsepIceCandidate ice_candidate0(kMediaContentName0, kMediaContentIndex0,
+                                  candidate0);
+  EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate0));
+
+  // Video candidate.
+  cricket::Candidate candidate1;
+  candidate1.set_address(rtc::SocketAddress("1.1.1.1", 6000));
+  candidate1.set_component(1);
+  candidate1.set_protocol("udp");
+  JsepIceCandidate ice_candidate1(kMediaContentName1, kMediaContentIndex1,
+                                  candidate1);
+  EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate1));
+
+  // Second audio candidate.
+  cricket::Candidate candidate2;
+  candidate2.set_address(rtc::SocketAddress("1.1.1.1", 5001));
+  candidate2.set_component(1);
+  candidate2.set_protocol("udp");
+  JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0,
+                                  candidate2);
+  EXPECT_TRUE(session_->ProcessIceMessage(&ice_candidate2));
+
+  EXPECT_TRUE_WAIT(connection_with_remote_port(5000), 1000);
+  EXPECT_TRUE_WAIT(connection_with_remote_port(5001), 1000);
+
+  // No need here for a _WAIT check since we are checking that state hasn't
+  // changed: if this is false we would be doing waits for nothing and if this
+  // is true then there will be no messages processed anyways.
+  EXPECT_FALSE(connection_with_remote_port(6000));
+}
+
 // kBundlePolicyBalanced bundle policy and answer contains BUNDLE.
 TEST_F(WebRtcSessionTest, TestBalancedBundleInAnswer) {
   InitWithBundlePolicy(PeerConnectionInterface::kBundlePolicyBalanced);
@@ -2789,6 +2834,46 @@ TEST_F(WebRtcSessionTest, TestMaxBundleWithSetRemoteDescriptionFirst) {
             session_->GetTransportProxy("video")->impl());
 }
 
+TEST_F(WebRtcSessionTest, TestRequireRtcpMux) {
+  InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyRequire);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  SessionDescriptionInterface* offer = CreateOffer(options);
+  SetLocalDescriptionWithoutError(offer);
+
+  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
+  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+
+  mediastream_signaling_.SendAudioVideoStream2();
+  SessionDescriptionInterface* answer =
+      CreateRemoteAnswer(session_->local_description());
+  SetRemoteDescriptionWithoutError(answer);
+
+  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
+  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+}
+
+TEST_F(WebRtcSessionTest, TestNegotiateRtcpMux) {
+  InitWithRtcpMuxPolicy(PeerConnectionInterface::kRtcpMuxPolicyNegotiate);
+  mediastream_signaling_.SendAudioVideoStream1();
+
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  SessionDescriptionInterface* offer = CreateOffer(options);
+  SetLocalDescriptionWithoutError(offer);
+
+  EXPECT_TRUE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
+  EXPECT_TRUE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+
+  mediastream_signaling_.SendAudioVideoStream2();
+  SessionDescriptionInterface* answer =
+      CreateRemoteAnswer(session_->local_description());
+  SetRemoteDescriptionWithoutError(answer);
+
+  EXPECT_FALSE(session_->GetTransportProxy("audio")->impl()->HasChannel(2));
+  EXPECT_FALSE(session_->GetTransportProxy("video")->impl()->HasChannel(2));
+}
+
 // This test verifies that SetLocalDescription and SetRemoteDescription fails
 // if BUNDLE is enabled but rtcp-mux is disabled in m-lines.
 TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
@@ -3431,7 +3516,7 @@ TEST_F(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
   ASSERT_TRUE(ch != NULL);
   ASSERT_EQ(1UL, ch->send_codecs().size());
   EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->send_codecs()[0].id);
-  EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
+  EXPECT_EQ(0, strcmp(cricket::kGoogleSctpDataCodecName,
                       ch->send_codecs()[0].name.c_str()));
   EXPECT_TRUE(ch->send_codecs()[0].GetParam(cricket::kCodecParamPort,
                                             &portnum));
@@ -3439,7 +3524,7 @@ TEST_F(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
 
   ASSERT_EQ(1UL, ch->recv_codecs().size());
   EXPECT_EQ(cricket::kGoogleSctpDataCodecId, ch->recv_codecs()[0].id);
-  EXPECT_TRUE(!strcmp(cricket::kGoogleSctpDataCodecName,
+  EXPECT_EQ(0, strcmp(cricket::kGoogleSctpDataCodecName,
                       ch->recv_codecs()[0].name.c_str()));
   EXPECT_TRUE(ch->recv_codecs()[0].GetParam(cricket::kCodecParamPort,
                                             &portnum));
@@ -3767,6 +3852,38 @@ TEST_F(WebRtcSessionTest, TestSetSocketOptionBeforeBundle) {
   EXPECT_EQ(8000, option_val);
 }
 
+// Test creating a session, request multiple offers, destroy the session
+// and make sure we got success/failure callbacks for all of the requests.
+// Background: crbug.com/507307
+TEST_F(WebRtcSessionTest, CreateOffersAndShutdown) {
+  Init();
+
+  rtc::scoped_refptr<WebRtcSessionCreateSDPObserverForTest> observers[100];
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  options.offer_to_receive_audio =
+      RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
+
+  for (auto& o : observers) {
+    o = new WebRtcSessionCreateSDPObserverForTest();
+    session_->CreateOffer(o, options);
+  }
+
+  session_.reset();
+
+  // Make sure we process pending messages on the current (signaling) thread
+  // before checking we we got our callbacks.  Quit() will do this and then
+  // immediately exit.  We won't need the queue after this point anyway.
+  rtc::Thread::Current()->Quit();
+
+  for (auto& o : observers) {
+    // We expect to have received a notification now even if the session was
+    // terminated.  The offer creation may or may not have succeeded, but we
+    // must have received a notification which, so the only invalid state
+    // is kInit.
+    EXPECT_NE(WebRtcSessionCreateSDPObserverForTest::kInit, o->state());
+  }
+}
+
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
index aab24cf0654..1909b0ed78d 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
@@ -40,6 +40,8 @@ namespace webrtc {
 namespace {
 static const char kFailedDueToIdentityFailed[] =
     " failed because DTLS identity request failed";
+static const char kFailedDueToSessionShutdown[] =
+    " failed because the session was shut down";
 
 static const uint64 kInitSessionVersion = 2;
 
@@ -183,6 +185,18 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
 }
 
 WebRtcSessionDescriptionFactory::~WebRtcSessionDescriptionFactory() {
+  ASSERT(signaling_thread_->IsCurrent());
+
+  // Fail any requests that were asked for before identity generation completed.
+  FailPendingRequests(kFailedDueToSessionShutdown);
+
+  // Process all pending notifications in the message queue.  If we don't do
+  // this, requests will linger and not know they succeeded or failed.
+  rtc::MessageList list;
+  signaling_thread_->Clear(this, rtc::MQID_ANY, &list);
+  for (auto& msg : list)
+    OnMessage(&msg);
+
   transport_desc_factory_.set_identity(NULL);
 }
 
@@ -400,6 +414,19 @@ void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
   PostCreateSessionDescriptionSucceeded(request.observer, answer);
 }
 
+void WebRtcSessionDescriptionFactory::FailPendingRequests(
+    const std::string& reason) {
+  ASSERT(signaling_thread_->IsCurrent());
+  while (!create_session_description_requests_.empty()) {
+    const CreateSessionDescriptionRequest& request =
+        create_session_description_requests_.front();
+    PostCreateSessionDescriptionFailed(request.observer,
+        ((request.type == CreateSessionDescriptionRequest::kOffer) ?
+            "CreateOffer" : "CreateAnswer") + reason);
+    create_session_description_requests_.pop();
+  }
+}
+
 void WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionFailed(
     CreateSessionDescriptionObserver* observer, const std::string& error) {
   CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
@@ -422,16 +449,7 @@ void WebRtcSessionDescriptionFactory::OnIdentityRequestFailed(int error) {
   LOG(LS_ERROR) << "Async identity request failed: error = " << error;
   identity_request_state_ = IDENTITY_FAILED;
 
-  std::string msg = kFailedDueToIdentityFailed;
-  while (!create_session_description_requests_.empty()) {
-    const CreateSessionDescriptionRequest& request =
-        create_session_description_requests_.front();
-    PostCreateSessionDescriptionFailed(
-        request.observer,
-        ((request.type == CreateSessionDescriptionRequest::kOffer) ?
-            "CreateOffer" : "CreateAnswer") + msg);
-    create_session_description_requests_.pop();
-  }
+  FailPendingRequests(kFailedDueToIdentityFailed);
 }
 
 void WebRtcSessionDescriptionFactory::SetIdentity(
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h
index 41798a485fb..860532dec93 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h
@@ -133,6 +133,8 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
 
   void InternalCreateOffer(CreateSessionDescriptionRequest request);
   void InternalCreateAnswer(CreateSessionDescriptionRequest request);
+  // Posts failure notifications for all pending session description requests.
+  void FailPendingRequests(const std::string& reason);
   void PostCreateSessionDescriptionFailed(
       CreateSessionDescriptionObserver* observer,
       const std::string& error);
@@ -145,17 +147,16 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
 
   std::queue<CreateSessionDescriptionRequest>
       create_session_description_requests_;
-  rtc::Thread* signaling_thread_;
-  MediaStreamSignaling* mediastream_signaling_;
+  rtc::Thread* const signaling_thread_;
+  MediaStreamSignaling* const mediastream_signaling_;
   cricket::TransportDescriptionFactory transport_desc_factory_;
   cricket::MediaSessionDescriptionFactory session_desc_factory_;
   uint64 session_version_;
   rtc::scoped_ptr<DTLSIdentityServiceInterface> identity_service_;
-  rtc::scoped_refptr<WebRtcIdentityRequestObserver>
-      identity_request_observer_;
-  WebRtcSession* session_;
-  std::string session_id_;
-  cricket::DataChannelType data_channel_type_;
+  rtc::scoped_refptr<WebRtcIdentityRequestObserver> identity_request_observer_;
+  WebRtcSession* const session_;
+  const std::string session_id_;
+  const cricket::DataChannelType data_channel_type_;
   IdentityRequestState identity_request_state_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcSessionDescriptionFactory);
diff --git a/chromium/third_party/libjingle/source/talk/build/common.gypi b/chromium/third_party/libjingle/source/talk/build/common.gypi
index 4735bc83e2c..36a96c5c558 100644
--- a/chromium/third_party/libjingle/source/talk/build/common.gypi
+++ b/chromium/third_party/libjingle/source/talk/build/common.gypi
@@ -34,7 +34,6 @@
     # TODO(ronghuawu): For now, disable the Chrome plugins, which causes a
     # flood of chromium-style warnings.
     'clang_use_chrome_plugins%': 0,
-    'libpeer_target_type%': 'static_library',
     'conditions': [
       ['OS=="android" or OS=="linux"', {
         'java_home%': '<!(python -c "import os; dir=os.getenv(\'JAVA_HOME\', \'/usr/lib/jvm/java-7-openjdk-amd64\'); assert os.path.exists(os.path.join(dir, \'include/jni.h\')), \'Point \\$JAVA_HOME or the java_home gyp variable to a directory containing include/jni.h!\'; print dir")',
@@ -79,20 +78,22 @@
       'HAVE_WEBRTC_VOICE',
     ],
     'conditions': [
-      # TODO(ronghuawu): Support dynamic library build.
-      ['"<(libpeer_target_type)"=="static_library"', {
-        'defines': [ 'LIBPEERCONNECTION_LIB=1' ],
-      }],
       ['OS=="linux"', {
         'defines': [
           'LINUX',
           'WEBRTC_LINUX',
         ],
+        # Remove Chromium's disabling of the -Wformat warning.
+        'cflags!': [
+          '-Wno-format',
+        ],
         'conditions': [
           ['clang==1', {
             'cflags': [
               '-Wall',
               '-Wextra',
+              '-Wformat',
+              '-Wformat-security',
               '-Wimplicit-fallthrough',
               '-Wmissing-braces',
               '-Wreorder',
diff --git a/chromium/third_party/libjingle/source/talk/build/isolate.gypi b/chromium/third_party/libjingle/source/talk/build/isolate.gypi
index 00c38324a06..13f3d50aff0 100644
--- a/chromium/third_party/libjingle/source/talk/build/isolate.gypi
+++ b/chromium/third_party/libjingle/source/talk/build/isolate.gypi
@@ -123,9 +123,6 @@
             '--extra-variable', 'mac_product_name', '<(mac_product_name)',
           ],
         }],
-        ["test_isolation_outdir!=''", {
-          'action': [ '--isolate-server', '<(test_isolation_outdir)' ],
-        }],
       ],
     },
   ],
diff --git a/chromium/third_party/libjingle/source/talk/codereview.settings b/chromium/third_party/libjingle/source/talk/codereview.settings
index db2535b99da..c138ebb7a79 100644
--- a/chromium/third_party/libjingle/source/talk/codereview.settings
+++ b/chromium/third_party/libjingle/source/talk/codereview.settings
@@ -1,9 +1,10 @@
 # This file is used by gcl to get repository specific information.
-CODE_REVIEW_SERVER: webrtc-codereview.appspot.com
+CODE_REVIEW_SERVER: codereview.webrtc.org
 CC_LIST: webrtc-reviews@webrtc.org
-VIEW_VC: http://code.google.com/p/webrtc/source/detail?r=
+VIEW_VC: https://chromium.googlesource.com/external/webrtc/+/
 TRY_ON_UPLOAD: False
 TRYSERVER_SVN_URL: svn://svn.chromium.org/chrome-try/try-webrtc
+TRYSERVER_PROJECT: webrtc
 TRYSERVER_ROOT: src/talk
 PROJECT: webrtc
-FORCE_HTTPS_COMMIT_URL: True
+PENDING_REF_PREFIX: refs/pending/
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/jni/Android.mk b/chromium/third_party/libjingle/source/talk/examples/android/jni/Android.mk
deleted file mode 100644
index 8e80160039b..00000000000
--- a/chromium/third_party/libjingle/source/talk/examples/android/jni/Android.mk
+++ /dev/null
@@ -1,2 +0,0 @@
-# This space intentionally left blank (required for Android build system).
-
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/res/values/strings.xml b/chromium/third_party/libjingle/source/talk/examples/android/res/values/strings.xml
index 428b0eb8be3..4f2f3777e22 100644
--- a/chromium/third_party/libjingle/source/talk/examples/android/res/values/strings.xml
+++ b/chromium/third_party/libjingle/source/talk/examples/android/res/values/strings.xml
@@ -87,6 +87,11 @@
     <string name="pref_audiocodec_dlg">Select default audio codec.</string>
     <string name="pref_audiocodec_default">OPUS</string>
 
+    <string name="pref_noaudioprocessing_key">audioprocessing_preference</string>
+    <string name="pref_noaudioprocessing_title">Disable audio processing.</string>
+    <string name="pref_noaudioprocessing_dlg">Disable audio processing pipeline.</string>
+    <string name="pref_noaudioprocessing_default">false</string>
+
     <string name="pref_miscsettings_key">misc_settings_key</string>
     <string name="pref_miscsettings_title">Miscellaneous settings.</string>
 
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/res/xml/preferences.xml b/chromium/third_party/libjingle/source/talk/examples/android/res/xml/preferences.xml
index ad0affb478e..73d8d5e254d 100644
--- a/chromium/third_party/libjingle/source/talk/examples/android/res/xml/preferences.xml
+++ b/chromium/third_party/libjingle/source/talk/examples/android/res/xml/preferences.xml
@@ -82,6 +82,12 @@
             android:dialogTitle="@string/pref_audiocodec_dlg"
             android:entries="@array/audioCodecs"
             android:entryValues="@array/audioCodecs" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_noaudioprocessing_key"
+            android:title="@string/pref_noaudioprocessing_title"
+            android:dialogTitle="@string/pref_noaudioprocessing_dlg"
+            android:defaultValue="@string/pref_noaudioprocessing_default" />
     </PreferenceCategory>
 
     <PreferenceCategory
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDAppClient.m b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDAppClient.m
index e4c2f81b481..ac99ca29971 100644
--- a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDAppClient.m
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDAppClient.m
@@ -34,6 +34,7 @@
 #import "RTCMediaConstraints.h"
 #import "RTCMediaStream.h"
 #import "RTCPair.h"
+#import "RTCPeerConnectionInterface.h"
 #import "RTCVideoCapturer.h"
 #import "RTCAVFoundationVideoSource.h"
 
@@ -41,6 +42,7 @@
 #import "ARDCEODTURNClient.h"
 #import "ARDJoinResponse.h"
 #import "ARDMessageResponse.h"
+#import "ARDSDPUtils.h"
 #import "ARDSignalingMessage.h"
 #import "ARDUtilities.h"
 #import "ARDWebSocketChannel.h"
@@ -343,10 +345,15 @@ static NSInteger const kARDAppClientErrorInvalidRoom = -6;
       [_delegate appClient:self didError:sdpError];
       return;
     }
+    // Prefer H264 if available.
+    RTCSessionDescription *sdpPreferringH264 =
+        [ARDSDPUtils descriptionForDescription:sdp
+                           preferredVideoCodec:@"H264"];
     [_peerConnection setLocalDescriptionWithDelegate:self
-                                  sessionDescription:sdp];
+                                  sessionDescription:sdpPreferringH264];
     ARDSessionDescriptionMessage *message =
-        [[ARDSessionDescriptionMessage alloc] initWithDescription:sdp];
+        [[ARDSessionDescriptionMessage alloc]
+            initWithDescription:sdpPreferringH264];
     [self sendSignalingMessage:message];
   });
 }
@@ -397,9 +404,11 @@ static NSInteger const kARDAppClientErrorInvalidRoom = -6;
 
   // Create peer connection.
   RTCMediaConstraints *constraints = [self defaultPeerConnectionConstraints];
-  _peerConnection = [_factory peerConnectionWithICEServers:_iceServers
-                                               constraints:constraints
-                                                  delegate:self];
+  RTCConfiguration *config = [[RTCConfiguration alloc] init];
+  config.iceServers = _iceServers;
+  _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];
@@ -438,8 +447,12 @@ static NSInteger const kARDAppClientErrorInvalidRoom = -6;
       ARDSessionDescriptionMessage *sdpMessage =
           (ARDSessionDescriptionMessage *)message;
       RTCSessionDescription *description = sdpMessage.sessionDescription;
+      // Prefer H264 if available.
+      RTCSessionDescription *sdpPreferringH264 =
+          [ARDSDPUtils descriptionForDescription:description
+                             preferredVideoCodec:@"H264"];
       [_peerConnection setRemoteDescriptionWithDelegate:self
-                                     sessionDescription:description];
+                                     sessionDescription:sdpPreferringH264];
       break;
     }
     case kARDSignalingMessageTypeCandidate: {
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDSDPUtils.h b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDSDPUtils.h
new file mode 100644
index 00000000000..2f14e6dec0b
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDSDPUtils.h
@@ -0,0 +1,41 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+#import <Foundation/Foundation.h>
+
+@class RTCSessionDescription;
+
+@interface ARDSDPUtils : NSObject
+
+// Updates the original SDP description to instead prefer the specified video
+// codec. We do this by placing the specified codec at the beginning of the
+// codec list if it exists in the sdp.
++ (RTCSessionDescription *)
+    descriptionForDescription:(RTCSessionDescription *)description
+          preferredVideoCodec:(NSString *)codec;
+
+@end
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDSDPUtils.m b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDSDPUtils.m
new file mode 100644
index 00000000000..157d6fc1f61
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ARDSDPUtils.m
@@ -0,0 +1,108 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+#import "ARDSDPUtils.h"
+
+#import "RTCSessionDescription.h"
+
+@implementation ARDSDPUtils
+
++ (RTCSessionDescription *)
+    descriptionForDescription:(RTCSessionDescription *)description
+          preferredVideoCodec:(NSString *)codec {
+  NSString *sdpString = description.description;
+  NSString *lineSeparator = @"\n";
+  NSString *mLineSeparator = @" ";
+  // Copied from PeerConnectionClient.java.
+  // TODO(tkchin): Move this to a shared C++ file.
+  NSMutableArray *lines =
+      [NSMutableArray arrayWithArray:
+          [sdpString componentsSeparatedByString:lineSeparator]];
+  int mLineIndex = -1;
+  NSString *codecRtpMap = nil;
+  // a=rtpmap:<payload type> <encoding name>/<clock rate>
+  // [/<encoding parameters>]
+  NSString *pattern =
+      [NSString stringWithFormat:@"^a=rtpmap:(\\d+) %@(/\\d+)+[\r]?$", codec];
+  NSRegularExpression *regex =
+      [NSRegularExpression regularExpressionWithPattern:pattern
+                                                options:0
+                                                  error:nil];
+  for (NSInteger i = 0; (i < lines.count) && (mLineIndex == -1 || !codecRtpMap);
+       ++i) {
+    NSString *line = lines[i];
+    if ([line hasPrefix:@"m=video"]) {
+      mLineIndex = i;
+      continue;
+    }
+    NSTextCheckingResult *codecMatches =
+        [regex firstMatchInString:line
+                          options:0
+                            range:NSMakeRange(0, line.length)];
+    if (codecMatches) {
+      codecRtpMap =
+          [line substringWithRange:[codecMatches rangeAtIndex:1]];
+      continue;
+    }
+  }
+  if (mLineIndex == -1) {
+    NSLog(@"No m=video line, so can't prefer %@", codec);
+    return description;
+  }
+  if (!codecRtpMap) {
+    NSLog(@"No rtpmap for %@", codec);
+    return description;
+  }
+  NSArray *origMLineParts =
+      [lines[mLineIndex] componentsSeparatedByString:mLineSeparator];
+  if (origMLineParts.count > 3) {
+    NSMutableArray *newMLineParts =
+        [NSMutableArray arrayWithCapacity:origMLineParts.count];
+    NSInteger origPartIndex = 0;
+    // Format is: m=<media> <port> <proto> <fmt> ...
+    [newMLineParts addObject:origMLineParts[origPartIndex++]];
+    [newMLineParts addObject:origMLineParts[origPartIndex++]];
+    [newMLineParts addObject:origMLineParts[origPartIndex++]];
+    [newMLineParts addObject:codecRtpMap];
+    for (; origPartIndex < origMLineParts.count; ++origPartIndex) {
+      if (![codecRtpMap isEqualToString:origMLineParts[origPartIndex]]) {
+        [newMLineParts addObject:origMLineParts[origPartIndex]];
+      }
+    }
+    NSString *newMLine =
+        [newMLineParts componentsJoinedByString:mLineSeparator];
+    [lines replaceObjectAtIndex:mLineIndex
+                     withObject:newMLine];
+  } else {
+    NSLog(@"Wrong SDP media description format: %@", lines[mLineIndex]);
+  }
+  NSString *mangledSdpString = [lines componentsJoinedByString:lineSeparator];
+  return [[RTCSessionDescription alloc] initWithType:description.type
+                                                 sdp:mangledSdpString];
+}
+
+@end
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
index b12a61a6182..2f07c7a4cc4 100644
--- a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
@@ -151,8 +151,10 @@
   self.remoteVideoTrack = nil;
   self.localVideoTrack = nil;
   [_client disconnect];
-  [self.presentingViewController dismissViewControllerAnimated:YES
-                                                    completion:nil];
+  if (![self isBeingDismissed]) {
+    [self.presentingViewController dismissViewControllerAnimated:YES
+                                                      completion:nil];
+  }
 }
 
 - (void)switchCamera {
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/Info.plist b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/Info.plist
index 0b66b1c68d4..fd1e26f8a54 100644
--- a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/Info.plist
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/Info.plist
@@ -2,69 +2,96 @@
 <!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
 <plist version="1.0">
 <dict>
-        <key>BuildMachineOSBuild</key>
-        <string>12E55</string>
-        <key>CFBundleDevelopmentRegion</key>
-        <string>en</string>
-        <key>CFBundleDisplayName</key>
-        <string>AppRTCDemo</string>
-        <key>CFBundleExecutable</key>
-        <string>AppRTCDemo</string>
-        <key>CFBundleIcons</key>
-        <dict>
-                <key>CFBundlePrimaryIcon</key>
-                <dict>
-                        <key>CFBundleIconFiles</key>
-                        <array>
-                                <string>Icon.png</string>
-                        </array>
-                </dict>
-        </dict>
-        <key>CFBundleIdentifier</key>
-        <string>com.google.AppRTCDemo</string>
-        <key>CFBundleInfoDictionaryVersion</key>
-        <string>6.0</string>
-        <key>CFBundleName</key>
-        <string>AppRTCDemo</string>
-        <key>CFBundlePackageType</key>
-        <string>APPL</string>
-        <key>CFBundleResourceSpecification</key>
-        <string>ResourceRules.plist</string>
-        <key>CFBundleShortVersionString</key>
-        <string>1.0</string>
-        <key>CFBundleSignature</key>
-        <string>????</string>
-        <key>CFBundleSupportedPlatforms</key>
-        <array>
-                <string>iPhoneOS</string>
-        </array>
-        <key>CFBundleVersion</key>
-        <string>1.0</string>
-        <key>UIRequiredDeviceCapabilities</key>
-        <array>
-                <string>armv7</string>
-        </array>
-        <key>UIStatusBarTintParameters</key>
-        <dict>
-                <key>UINavigationBar</key>
-                <dict>
-                        <key>Style</key>
-                        <string>UIBarStyleDefault</string>
-                        <key>Translucent</key>
-                        <false/>
-                </dict>
-        </dict>
-        <key>UISupportedInterfaceOrientations</key>
-        <array>
-                <string>UIInterfaceOrientationPortrait</string>
-        </array>
-        <key>UIAppFonts</key>
-        <array>
-                <string>Roboto-Regular.ttf</string>
-        </array>
-        <key>UIBackgroundModes</key>
-        <array>
-                <string>voip</string>
-        </array>
+  <key>BuildMachineOSBuild</key>
+  <string>12E55</string>
+  <key>CFBundleDevelopmentRegion</key>
+  <string>en</string>
+  <key>CFBundleDisplayName</key>
+  <string>AppRTCDemo</string>
+  <key>CFBundleExecutable</key>
+  <string>AppRTCDemo</string>
+  <key>CFBundleIcons</key>
+  <dict>
+    <key>CFBundlePrimaryIcon</key>
+    <dict>
+      <key>CFBundleIconFiles</key>
+      <array>
+        <string>Icon.png</string>
+      </array>
+    </dict>
+  </dict>
+  <key>CFBundleIdentifier</key>
+  <string>com.google.AppRTCDemo</string>
+  <key>CFBundleInfoDictionaryVersion</key>
+  <string>6.0</string>
+  <key>CFBundleName</key>
+  <string>AppRTCDemo</string>
+  <key>CFBundlePackageType</key>
+  <string>APPL</string>
+  <key>CFBundleShortVersionString</key>
+  <string>1.0</string>
+  <key>CFBundleSignature</key>
+  <string>????</string>
+  <key>CFBundleSupportedPlatforms</key>
+  <array>
+    <string>iPhoneOS</string>
+  </array>
+  <key>CFBundleVersion</key>
+  <string>1.0</string>
+  <key>UIStatusBarTintParameters</key>
+  <dict>
+    <key>UINavigationBar</key>
+    <dict>
+      <key>Style</key>
+      <string>UIBarStyleDefault</string>
+      <key>Translucent</key>
+      <false/>
+    </dict>
+  </dict>
+  <key>UISupportedInterfaceOrientations</key>
+  <array>
+    <string>UIInterfaceOrientationPortrait</string>
+  </array>
+  <key>UIAppFonts</key>
+  <array>
+    <string>Roboto-Regular.ttf</string>
+  </array>
+  <key>UIBackgroundModes</key>
+  <array>
+    <string>voip</string>
+  </array>
+  <key>UILaunchImages</key>
+  <array>
+    <dict>
+      <key>UILaunchImageMinimumOSVersion</key>
+      <string>7.0</string>
+      <key>UILaunchImageName</key>
+      <string>iPhone5</string>
+      <key>UILaunchImageOrientation</key>
+      <string>Portrait</string>
+      <key>UILaunchImageSize</key>
+      <string>{320, 568}</string>
+    </dict>
+    <dict>
+      <key>UILaunchImageMinimumOSVersion</key>
+      <string>8.0</string>
+      <key>UILaunchImageName</key>
+      <string>iPhone6</string>
+      <key>UILaunchImageOrientation</key>
+      <string>Portrait</string>
+      <key>UILaunchImageSize</key>
+      <string>{375, 667}</string>
+    </dict>
+    <dict>
+      <key>UILaunchImageMinimumOSVersion</key>
+      <string>8.0</string>
+      <key>UILaunchImageName</key>
+      <string>iPhone6p</string>
+      <key>UILaunchImageOrientation</key>
+      <string>Portrait</string>
+      <key>UILaunchImageSize</key>
+      <string>{414, 736}</string>
+    </dict>
+  </array>
 </dict>
 </plist>
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/Default-568h.png b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/Default-568h.png
deleted file mode 100644
index 2735148c93a..00000000000
--- a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/Default-568h.png
+++ /dev/null
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone5@2x.png b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone5@2x.png
new file mode 100644
index 00000000000..9d005fde061
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone5@2x.png
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone6@2x.png b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone6@2x.png
new file mode 100644
index 00000000000..fce3eb95b3c
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone6@2x.png
diff --git a/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone6p@3x.png b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone6p@3x.png
new file mode 100644
index 00000000000..aee20c22093
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/examples/objc/AppRTCDemo/ios/resources/iPhone6p@3x.png
diff --git a/chromium/third_party/libjingle/source/talk/libjingle.gyp b/chromium/third_party/libjingle/source/talk/libjingle.gyp
index 1c97f0f0a49..9de1c9bf6df 100755
--- a/chromium/third_party/libjingle/source/talk/libjingle.gyp
+++ b/chromium/third_party/libjingle/source/talk/libjingle.gyp
@@ -140,14 +140,18 @@
                 # included here, or better yet, build a proper .jar in webrtc
                 # and include it here.
                 'android_java_files': [
+                  'app/webrtc/java/android/org/webrtc/EglBase.java',
+                  'app/webrtc/java/android/org/webrtc/GlRectDrawer.java',
+                  'app/webrtc/java/android/org/webrtc/GlShader.java',
+                  'app/webrtc/java/android/org/webrtc/GlUtil.java',
                   'app/webrtc/java/android/org/webrtc/VideoRendererGui.java',
                   'app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java',
                   'app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java',
                   'app/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java',
-                  '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/AudioManagerAndroid.java',
                   '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java',
                   '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViERenderer.java',
                   '<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java',
+                  '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java',
                   '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java',
                   '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java',
                   '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java',
@@ -257,6 +261,8 @@
             'app/webrtc/objc/RTCPeerConnection+Internal.h',
             'app/webrtc/objc/RTCPeerConnection.mm',
             'app/webrtc/objc/RTCPeerConnectionFactory.mm',
+            'app/webrtc/objc/RTCPeerConnectionInterface+Internal.h',
+            'app/webrtc/objc/RTCPeerConnectionInterface.mm',
             'app/webrtc/objc/RTCPeerConnectionObserver.h',
             'app/webrtc/objc/RTCPeerConnectionObserver.mm',
             'app/webrtc/objc/RTCSessionDescription+Internal.h',
@@ -286,6 +292,7 @@
             'app/webrtc/objc/public/RTCPeerConnection.h',
             'app/webrtc/objc/public/RTCPeerConnectionDelegate.h',
             'app/webrtc/objc/public/RTCPeerConnectionFactory.h',
+            'app/webrtc/objc/public/RTCPeerConnectionInterface.h',
             'app/webrtc/objc/public/RTCSessionDescription.h',
             'app/webrtc/objc/public/RTCSessionDescriptionDelegate.h',
             'app/webrtc/objc/public/RTCStatsDelegate.h',
@@ -351,6 +358,9 @@
                 # Need to build against 10.7 framework for full ARC support
                 # on OSX.
                 'MACOSX_DEPLOYMENT_TARGET' : '10.7',
+                # RTCVideoTrack.mm uses code with partial availability.
+                # https://code.google.com/p/webrtc/issues/detail?id=4695
+                'WARNING_CFLAGS!': ['-Wpartial-availability'],
               },
               'link_settings': {
                 'xcode_settings': {
@@ -429,8 +439,6 @@
         'media/base/cryptoparams.h',
         'media/base/device.h',
         'media/base/fakescreencapturerfactory.h',
-        'media/base/filemediaengine.cc',
-        'media/base/filemediaengine.h',
         'media/base/hybriddataengine.h',
         'media/base/mediachannel.h',
         'media/base/mediacommon.h',
@@ -470,22 +478,20 @@
         'media/devices/videorendererfactory.h',
         'media/devices/yuvframescapturer.cc',
         'media/devices/yuvframescapturer.h',
-        'media/other/linphonemediaengine.h',
         'media/sctp/sctpdataengine.cc',
         'media/sctp/sctpdataengine.h',
         'media/webrtc/simulcast.cc',
         'media/webrtc/simulcast.h',
         'media/webrtc/webrtccommon.h',
-        'media/webrtc/webrtcexport.h',
         'media/webrtc/webrtcmediaengine.cc',
         'media/webrtc/webrtcmediaengine.h',
         'media/webrtc/webrtcmediaengine.cc',
         'media/webrtc/webrtcpassthroughrender.cc',
         'media/webrtc/webrtcpassthroughrender.h',
         'media/webrtc/webrtcvideocapturer.cc',
+        'media/webrtc/webrtcvideocapturer.h',
         'media/webrtc/webrtcvideocapturerfactory.h',
         'media/webrtc/webrtcvideocapturerfactory.cc',
-        'media/webrtc/webrtcvideocapturer.h',
         'media/webrtc/webrtcvideodecoderfactory.h',
         'media/webrtc/webrtcvideoencoderfactory.h',
         'media/webrtc/webrtcvideoengine2.cc',
@@ -600,6 +606,10 @@
               # deprecated functions and remove this flag.
               '-Wno-deprecated-declarations',
             ],
+            # Disable partial availability warning to prevent errors
+            # in macdevicemanagermm.mm using AVFoundation.
+            # https://code.google.com/p/webrtc/issues/detail?id=4695
+            'WARNING_CFLAGS!': ['-Wpartial-availability'],
           },
           'link_settings': {
             'xcode_settings': {
@@ -666,15 +676,11 @@
         'session/media/currentspeakermonitor.h',
         'session/media/mediamonitor.cc',
         'session/media/mediamonitor.h',
-        'session/media/mediarecorder.cc',
-        'session/media/mediarecorder.h',
         'session/media/mediasession.cc',
         'session/media/mediasession.h',
         'session/media/mediasink.h',
         'session/media/rtcpmuxfilter.cc',
         'session/media/rtcpmuxfilter.h',
-        'session/media/soundclip.cc',
-        'session/media/soundclip.h',
         'session/media/srtpfilter.cc',
         'session/media/srtpfilter.h',
         'session/media/typingmonitor.cc',
diff --git a/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp b/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp
index af2456fcd95..5cd31b6d108 100755
--- a/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp
+++ b/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp
@@ -173,6 +173,8 @@
             'examples/objc/AppRTCDemo/ARDMessageResponse.m',
             'examples/objc/AppRTCDemo/ARDMessageResponse+Internal.h',
             'examples/objc/AppRTCDemo/ARDRoomServerClient.h',
+            'examples/objc/AppRTCDemo/ARDSDPUtils.h',
+            'examples/objc/AppRTCDemo/ARDSDPUtils.m',
             'examples/objc/AppRTCDemo/ARDSignalingChannel.h',
             'examples/objc/AppRTCDemo/ARDSignalingMessage.h',
             'examples/objc/AppRTCDemo/ARDSignalingMessage.m',
@@ -220,7 +222,9 @@
           'conditions': [
             ['OS=="ios"', {
               'mac_bundle_resources': [
-                'examples/objc/AppRTCDemo/ios/resources/Default-568h.png',
+                'examples/objc/AppRTCDemo/ios/resources/iPhone5@2x.png',
+                'examples/objc/AppRTCDemo/ios/resources/iPhone6@2x.png',
+                'examples/objc/AppRTCDemo/ios/resources/iPhone6p@3x.png',
                 'examples/objc/AppRTCDemo/ios/resources/Roboto-Regular.ttf',
                 'examples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp.png',
                 'examples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp@2x.png',
@@ -290,6 +294,9 @@
                 # warning so we can compile successfully.
                 'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
                 'MACOSX_DEPLOYMENT_TARGET' : '10.8',
+                # SRWebSocket.m uses code with partial availability.
+                # https://code.google.com/p/webrtc/issues/detail?id=4695
+                'WARNING_CFLAGS!': ['-Wpartial-availability'],
               },
             }],
           ],
diff --git a/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp b/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp
index 3ca2f1ff367..d08a23682c3 100755
--- a/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp
+++ b/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp
@@ -84,7 +84,6 @@
       'sources': [
         'media/base/capturemanager_unittest.cc',
         'media/base/codec_unittest.cc',
-        'media/base/filemediaengine_unittest.cc',
         'media/base/rtpdataengine_unittest.cc',
         'media/base/rtpdump_unittest.cc',
         'media/base/rtputils_unittest.cc',
@@ -103,6 +102,7 @@
         'media/webrtc/webrtcvideocapturer_unittest.cc',
         'media/base/videoframe_unittest.h',
         'media/webrtc/webrtcvideoframe_unittest.cc',
+        'media/webrtc/webrtcvideoframefactory_unittest.cc',
 
         # Disabled because some tests fail.
         # TODO(ronghuawu): Reenable these tests.
@@ -156,7 +156,6 @@
         'session/media/channel_unittest.cc',
         'session/media/channelmanager_unittest.cc',
         'session/media/currentspeakermonitor_unittest.cc',
-        'session/media/mediarecorder_unittest.cc',
         'session/media/mediasession_unittest.cc',
         'session/media/rtcpmuxfilter_unittest.cc',
         'session/media/srtpfilter_unittest.cc',
@@ -199,6 +198,8 @@
         'app/webrtc/datachannel_unittest.cc',
         'app/webrtc/dtlsidentitystore_unittest.cc',
         'app/webrtc/dtmfsender_unittest.cc',
+        'app/webrtc/fakemetricsobserver.cc',
+        'app/webrtc/fakemetricsobserver.h',
         'app/webrtc/jsepsessiondescription_unittest.cc',
         'app/webrtc/localaudiosource_unittest.cc',
         'app/webrtc/mediastream_unittest.cc',
diff --git a/chromium/third_party/libjingle/source/talk/media/base/capturemanager.cc b/chromium/third_party/libjingle/source/talk/media/base/capturemanager.cc
index 95612187fa2..d3b3f82b556 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/capturemanager.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/capturemanager.cc
@@ -375,8 +375,10 @@ void CaptureManager::UnregisterVideoCapturer(
   // for the CaptureManager it doesn't matter as it will no longer receive any
   // frames from the VideoCapturer.
   SignalCapturerStateChange.stop(video_capturer->SignalStateChange);
-  video_capturer->Stop();
-  SignalCapturerStateChange(video_capturer, CS_STOPPED);
+  if (video_capturer->IsRunning()) {
+    video_capturer->Stop();
+    SignalCapturerStateChange(video_capturer, CS_STOPPED);
+  }
 }
 
 bool CaptureManager::StartWithBestCaptureFormat(
diff --git a/chromium/third_party/libjingle/source/talk/media/base/constants.cc b/chromium/third_party/libjingle/source/talk/media/base/constants.cc
index 562dad4a3ea..0d0a33c18aa 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/constants.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/constants.cc
@@ -128,9 +128,11 @@ const int kNumDefaultUnsignalledVideoRecvStreams = 0;
 
 const char kVp8CodecName[] = "VP8";
 const char kVp9CodecName[] = "VP9";
+const char kH264CodecName[] = "H264";
 
 const int kDefaultVp8PlType = 100;
 const int kDefaultVp9PlType = 101;
+const int kDefaultH264PlType = 107;
 const int kDefaultRedPlType = 116;
 const int kDefaultUlpfecType = 117;
 const int kDefaultRtxVp8PlType = 96;
diff --git a/chromium/third_party/libjingle/source/talk/media/base/constants.h b/chromium/third_party/libjingle/source/talk/media/base/constants.h
index 84216fb534e..d92cb22514c 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/constants.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/constants.h
@@ -158,9 +158,11 @@ extern const int kNumDefaultUnsignalledVideoRecvStreams;
 
 extern const char kVp8CodecName[];
 extern const char kVp9CodecName[];
+extern const char kH264CodecName[];
 
 extern const int kDefaultVp8PlType;
 extern const int kDefaultVp9PlType;
+extern const int kDefaultH264PlType;
 extern const int kDefaultRedPlType;
 extern const int kDefaultUlpfecType;
 extern const int kDefaultRtxVp8PlType;
diff --git a/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h b/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h
index ead9f79e33f..15fcb202f1c 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h
@@ -630,11 +630,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
   int max_bps_;
 };
 
-class FakeSoundclipMedia : public SoundclipMedia {
- public:
-  virtual bool PlaySound(const char* buf, int len, int flags) { return true; }
-};
-
 class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
  public:
   explicit FakeDataMediaChannel(void* unused)
@@ -716,9 +711,6 @@ class FakeBaseEngine {
       : loglevel_(-1),
         options_changed_(false),
         fail_create_channel_(false) {}
-  bool Init(rtc::Thread* worker_thread) { return true; }
-  void Terminate() {}
-
   void SetLogging(int level, const char* filter) {
     loglevel_ = level;
     logfilter_ = filter;
@@ -756,6 +748,8 @@ class FakeVoiceEngine : public FakeBaseEngine {
     // sanity checks against that.
     codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
   }
+  bool Init(rtc::Thread* worker_thread) { return true; }
+  void Terminate() {}
   int GetCapabilities() { return AUDIO_SEND | AUDIO_RECV; }
   AudioOptions GetAudioOptions() const {
     return options_;
@@ -769,12 +763,13 @@ class FakeVoiceEngine : public FakeBaseEngine {
     return true;
   }
 
-  VoiceMediaChannel* CreateChannel() {
+  VoiceMediaChannel* CreateChannel(const AudioOptions& options) {
     if (fail_create_channel_) {
-      return NULL;
+      return nullptr;
     }
 
     FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this);
+    ch->SetOptions(options);
     channels_.push_back(ch);
     return ch;
   }
@@ -784,7 +779,6 @@ class FakeVoiceEngine : public FakeBaseEngine {
   void UnregisterChannel(VoiceMediaChannel* channel) {
     channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
   }
-  SoundclipMedia* CreateSoundclip() { return new FakeSoundclipMedia(); }
 
   const std::vector<AudioCodec>& codecs() { return codecs_; }
   void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
@@ -867,6 +861,7 @@ class FakeVideoEngine : public FakeBaseEngine {
     // sanity checks against that.
     codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
   }
+  void Init() {}
   bool GetOptions(VideoOptions* options) const {
     *options = options_;
     return true;
diff --git a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine.cc b/chromium/third_party/libjingle/source/talk/media/base/filemediaengine.cc
deleted file mode 100644
index 0fc8d5610b1..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine.cc
+++ /dev/null
@@ -1,376 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 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.
- */
-
-#include "talk/media/base/filemediaengine.h"
-
-#include <algorithm>
-#include <limits.h>
-
-#include "talk/media/base/rtpdump.h"
-#include "talk/media/base/rtputils.h"
-#include "talk/media/base/streamparams.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/event.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/stream.h"
-
-namespace cricket {
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of FileMediaEngine.
-///////////////////////////////////////////////////////////////////////////
-int FileMediaEngine::GetCapabilities() {
-  int capabilities = 0;
-  if (!voice_input_filename_.empty()) {
-    capabilities |= AUDIO_SEND;
-  }
-  if (!voice_output_filename_.empty()) {
-    capabilities |= AUDIO_RECV;
-  }
-  if (!video_input_filename_.empty()) {
-    capabilities |= VIDEO_SEND;
-  }
-  if (!video_output_filename_.empty()) {
-    capabilities |= VIDEO_RECV;
-  }
-  return capabilities;
-}
-
-VoiceMediaChannel* FileMediaEngine::CreateChannel() {
-  rtc::FileStream* input_file_stream = NULL;
-  rtc::FileStream* output_file_stream = NULL;
-
-  if (voice_input_filename_.empty() && voice_output_filename_.empty())
-    return NULL;
-  if (!voice_input_filename_.empty()) {
-    input_file_stream = rtc::Filesystem::OpenFile(
-        rtc::Pathname(voice_input_filename_), "rb");
-    if (!input_file_stream) {
-      LOG(LS_ERROR) << "Not able to open the input audio stream file.";
-      return NULL;
-    }
-  }
-
-  if (!voice_output_filename_.empty()) {
-    output_file_stream = rtc::Filesystem::OpenFile(
-        rtc::Pathname(voice_output_filename_), "wb");
-    if (!output_file_stream) {
-      delete input_file_stream;
-      LOG(LS_ERROR) << "Not able to open the output audio stream file.";
-      return NULL;
-    }
-  }
-
-  return new FileVoiceChannel(input_file_stream, output_file_stream,
-                              rtp_sender_thread_);
-}
-
-VideoMediaChannel* FileMediaEngine::CreateVideoChannel(
-    const VideoOptions& options,
-    VoiceMediaChannel* voice_ch) {
-  rtc::FileStream* input_file_stream = NULL;
-  rtc::FileStream* output_file_stream = NULL;
-
-  if (video_input_filename_.empty() && video_output_filename_.empty())
-      return NULL;
-
-  if (!video_input_filename_.empty()) {
-    input_file_stream = rtc::Filesystem::OpenFile(
-        rtc::Pathname(video_input_filename_), "rb");
-    if (!input_file_stream) {
-      LOG(LS_ERROR) << "Not able to open the input video stream file.";
-      return NULL;
-    }
-  }
-
-  if (!video_output_filename_.empty()) {
-    output_file_stream = rtc::Filesystem::OpenFile(
-        rtc::Pathname(video_output_filename_), "wb");
-    if (!output_file_stream) {
-      delete input_file_stream;
-      LOG(LS_ERROR) << "Not able to open the output video stream file.";
-      return NULL;
-    }
-  }
-
-  FileVideoChannel* channel = new FileVideoChannel(
-      input_file_stream, output_file_stream, rtp_sender_thread_);
-  channel->SetOptions(options);
-  return channel;
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Definition of RtpSenderReceiver.
-///////////////////////////////////////////////////////////////////////////
-class RtpSenderReceiver : public rtc::MessageHandler {
- public:
-  RtpSenderReceiver(MediaChannel* channel,
-                    rtc::StreamInterface* input_file_stream,
-                    rtc::StreamInterface* output_file_stream,
-                    rtc::Thread* sender_thread);
-  virtual ~RtpSenderReceiver();
-
-  // Called by media channel. Context: media channel thread.
-  bool SetSend(bool send);
-  void SetSendSsrc(uint32 ssrc);
-  void OnPacketReceived(rtc::Buffer* packet);
-
-  // Override virtual method of parent MessageHandler. Context: Worker Thread.
-  virtual void OnMessage(rtc::Message* pmsg);
-
- private:
-  // Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_.
-  // Return true if successful.
-  bool ReadNextPacket(RtpDumpPacket* packet);
-  // Send a RTP packet to the network. The input parameter data points to the
-  // start of the RTP packet and len is the packet size. Return true if the sent
-  // size is equal to len.
-  bool SendRtpPacket(const void* data, size_t len);
-
-  MediaChannel* media_channel_;
-  rtc::scoped_ptr<rtc::StreamInterface> input_stream_;
-  rtc::scoped_ptr<rtc::StreamInterface> output_stream_;
-  rtc::scoped_ptr<RtpDumpLoopReader> rtp_dump_reader_;
-  rtc::scoped_ptr<RtpDumpWriter> rtp_dump_writer_;
-  rtc::Thread* sender_thread_;
-  bool own_sender_thread_;
-  // RTP dump packet read from the input stream.
-  RtpDumpPacket rtp_dump_packet_;
-  uint32 start_send_time_;
-  bool sending_;
-  bool first_packet_;
-  uint32 first_ssrc_;
-
-  DISALLOW_COPY_AND_ASSIGN(RtpSenderReceiver);
-};
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of RtpSenderReceiver.
-///////////////////////////////////////////////////////////////////////////
-RtpSenderReceiver::RtpSenderReceiver(
-    MediaChannel* channel,
-    rtc::StreamInterface* input_file_stream,
-    rtc::StreamInterface* output_file_stream,
-    rtc::Thread* sender_thread)
-    : media_channel_(channel),
-      input_stream_(input_file_stream),
-      output_stream_(output_file_stream),
-      sending_(false),
-      first_packet_(true) {
-  if (sender_thread == NULL) {
-    sender_thread_ = new rtc::Thread();
-    own_sender_thread_ = true;
-  } else {
-    sender_thread_ = sender_thread;
-    own_sender_thread_ = false;
-  }
-
-  if (input_stream_) {
-    rtp_dump_reader_.reset(new RtpDumpLoopReader(input_stream_.get()));
-    // Start the sender thread, which reads rtp dump records, waits based on
-    // the record timestamps, and sends the RTP packets to the network.
-    if (own_sender_thread_) {
-      sender_thread_->Start();
-    }
-  }
-
-  // Create a rtp dump writer for the output RTP dump stream.
-  if (output_stream_) {
-    rtp_dump_writer_.reset(new RtpDumpWriter(output_stream_.get()));
-  }
-}
-
-RtpSenderReceiver::~RtpSenderReceiver() {
-  if (own_sender_thread_) {
-    sender_thread_->Stop();
-    delete sender_thread_;
-  }
-}
-
-bool RtpSenderReceiver::SetSend(bool send) {
-  bool was_sending = sending_;
-  sending_ = send;
-  if (!was_sending && sending_) {
-    sender_thread_->PostDelayed(0, this);  // Wake up the send thread.
-    start_send_time_ = rtc::Time();
-  }
-  return true;
-}
-
-void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) {
-  if (rtp_dump_reader_) {
-    rtp_dump_reader_->SetSsrc(ssrc);
-  }
-}
-
-void RtpSenderReceiver::OnPacketReceived(rtc::Buffer* packet) {
-  if (rtp_dump_writer_) {
-    rtp_dump_writer_->WriteRtpPacket(packet->data(), packet->size());
-  }
-}
-
-void RtpSenderReceiver::OnMessage(rtc::Message* pmsg) {
-  if (!sending_) {
-    // If the sender thread is not sending, ignore this message. The thread goes
-    // to sleep until SetSend(true) wakes it up.
-    return;
-  }
-  if (!first_packet_) {
-    // Send the previously read packet.
-    SendRtpPacket(&rtp_dump_packet_.data[0], rtp_dump_packet_.data.size());
-  }
-
-  if (ReadNextPacket(&rtp_dump_packet_)) {
-    int wait = rtc::TimeUntil(
-        start_send_time_ + rtp_dump_packet_.elapsed_time);
-    wait = std::max(0, wait);
-    sender_thread_->PostDelayed(wait, this);
-  } else {
-    sender_thread_->Quit();
-  }
-}
-
-bool RtpSenderReceiver::ReadNextPacket(RtpDumpPacket* packet) {
-  while (rtc::SR_SUCCESS == rtp_dump_reader_->ReadPacket(packet)) {
-    uint32 ssrc;
-    if (!packet->GetRtpSsrc(&ssrc)) {
-      return false;
-    }
-    if (first_packet_) {
-      first_packet_ = false;
-      first_ssrc_ = ssrc;
-    }
-    if (ssrc == first_ssrc_) {
-      return true;
-    }
-  }
-  return false;
-}
-
-bool RtpSenderReceiver::SendRtpPacket(const void* data, size_t len) {
-  if (!media_channel_)
-    return false;
-
-  rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
-                     kMaxRtpPacketLen);
-  return media_channel_->SendPacket(&packet);
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of FileVoiceChannel.
-///////////////////////////////////////////////////////////////////////////
-FileVoiceChannel::FileVoiceChannel(
-    rtc::StreamInterface* input_file_stream,
-    rtc::StreamInterface* output_file_stream,
-    rtc::Thread* rtp_sender_thread)
-    : send_ssrc_(0),
-      rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
-                                                 output_file_stream,
-                                                 rtp_sender_thread)) {}
-
-FileVoiceChannel::~FileVoiceChannel() {}
-
-bool FileVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
-  // TODO(whyuan): Check the format of RTP dump input.
-  return true;
-}
-
-bool FileVoiceChannel::SetSend(SendFlags flag) {
-  return rtp_sender_receiver_->SetSend(flag != SEND_NOTHING);
-}
-
-bool FileVoiceChannel::AddSendStream(const StreamParams& sp) {
-  if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
-    LOG(LS_ERROR) << "FileVoiceChannel only supports one send stream.";
-    return false;
-  }
-  send_ssrc_ = sp.ssrcs[0];
-  rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
-  return true;
-}
-
-bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) {
-  if (ssrc != send_ssrc_)
-    return false;
-  send_ssrc_ = 0;
-  rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
-  return true;
-}
-
-void FileVoiceChannel::OnPacketReceived(
-    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
-  rtp_sender_receiver_->OnPacketReceived(packet);
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of FileVideoChannel.
-///////////////////////////////////////////////////////////////////////////
-FileVideoChannel::FileVideoChannel(
-    rtc::StreamInterface* input_file_stream,
-    rtc::StreamInterface* output_file_stream,
-    rtc::Thread* rtp_sender_thread)
-    : send_ssrc_(0),
-      rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
-                                                 output_file_stream,
-                                                 rtp_sender_thread)) {}
-
-FileVideoChannel::~FileVideoChannel() {}
-
-bool FileVideoChannel::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
-  // TODO(whyuan): Check the format of RTP dump input.
-  return true;
-}
-
-bool FileVideoChannel::SetSend(bool send) {
-  return rtp_sender_receiver_->SetSend(send);
-}
-
-bool FileVideoChannel::AddSendStream(const StreamParams& sp) {
-  if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
-    LOG(LS_ERROR) << "FileVideoChannel only support one send stream.";
-    return false;
-  }
-  send_ssrc_ = sp.ssrcs[0];
-  rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
-  return true;
-}
-
-bool FileVideoChannel::RemoveSendStream(uint32 ssrc) {
-  if (ssrc != send_ssrc_)
-    return false;
-  send_ssrc_ = 0;
-  rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
-  return true;
-}
-
-void FileVideoChannel::OnPacketReceived(
-    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
-  rtp_sender_receiver_->OnPacketReceived(packet);
-}
-
-}  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine.h b/chromium/third_party/libjingle/source/talk/media/base/filemediaengine.h
deleted file mode 100644
index c2e80c383e6..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine.h
+++ /dev/null
@@ -1,330 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 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.
- */
-
-#ifndef TALK_MEDIA_BASE_FILEMEDIAENGINE_H_
-#define TALK_MEDIA_BASE_FILEMEDIAENGINE_H_
-
-#include <string>
-#include <vector>
-
-#include "talk/media/base/codec.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/mediaengine.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/stream.h"
-
-namespace rtc {
-class StreamInterface;
-}
-
-namespace cricket {
-
-// A media engine contains a capturer, an encoder, and a sender in the sender
-// side and a receiver, a decoder, and a renderer in the receiver side.
-// FileMediaEngine simulates the capturer and the encoder via an input RTP dump
-// stream and simulates the decoder and the renderer via an output RTP dump
-// stream. Depending on the parameters of the constructor, FileMediaEngine can
-// act as file voice engine, file video engine, or both. Currently, we use
-// only the RTP dump packets. TODO(whyuan): Enable RTCP packets.
-class FileMediaEngine : public MediaEngineInterface {
- public:
-  FileMediaEngine() : rtp_sender_thread_(NULL) {}
-  virtual ~FileMediaEngine() {}
-
-  // Set the file name of the input or output RTP dump for voice or video.
-  // Should be called before the channel is created.
-  void set_voice_input_filename(const std::string& filename) {
-    voice_input_filename_ = filename;
-  }
-  void set_voice_output_filename(const std::string& filename) {
-    voice_output_filename_ = filename;
-  }
-  void set_video_input_filename(const std::string& filename) {
-    video_input_filename_ = filename;
-  }
-  void set_video_output_filename(const std::string& filename) {
-    video_output_filename_ = filename;
-  }
-
-  // Should be called before codecs() and video_codecs() are called. We need to
-  // set the voice and video codecs; otherwise, Jingle initiation will fail.
-  void set_voice_codecs(const std::vector<AudioCodec>& codecs) {
-    voice_codecs_ = codecs;
-  }
-  void set_video_codecs(const std::vector<VideoCodec>& codecs) {
-    video_codecs_ = codecs;
-  }
-
-  // Implement pure virtual methods of MediaEngine.
-  virtual bool Init(rtc::Thread* worker_thread) {
-    return true;
-  }
-  virtual void Terminate() {}
-  virtual int GetCapabilities();
-  virtual VoiceMediaChannel* CreateChannel();
-  virtual VideoMediaChannel* CreateVideoChannel(const VideoOptions& options,
-                                                VoiceMediaChannel* voice_ch);
-  virtual SoundclipMedia* CreateSoundclip() { return NULL; }
-  virtual AudioOptions GetAudioOptions() const { return AudioOptions(); }
-  virtual bool SetAudioOptions(const AudioOptions& options) { return true; }
-  virtual bool SetAudioDelayOffset(int offset) { return true; }
-  virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
-    return true;
-  }
-  virtual bool SetSoundDevices(const Device* in_dev, const Device* out_dev) {
-    return true;
-  }
-  virtual bool SetVideoCaptureDevice(const Device* cam_device) { return true; }
-  virtual bool SetVideoCapturer(VideoCapturer* /*capturer*/) {
-    return true;
-  }
-  virtual VideoCapturer* GetVideoCapturer() const {
-    return NULL;
-  }
-  virtual bool GetOutputVolume(int* level) {
-    *level = 0;
-    return true;
-  }
-  virtual bool SetOutputVolume(int level) { return true; }
-  virtual int GetInputLevel() { return 0; }
-  virtual bool SetLocalMonitor(bool enable) { return true; }
-  // TODO(whyuan): control channel send?
-  virtual bool SetVideoCapture(bool capture) { return true; }
-  virtual const std::vector<AudioCodec>& audio_codecs() {
-    return voice_codecs_;
-  }
-  virtual const std::vector<VideoCodec>& video_codecs() {
-    return video_codecs_;
-  }
-  virtual const std::vector<RtpHeaderExtension>& audio_rtp_header_extensions() {
-    return audio_rtp_header_extensions_;
-  }
-  virtual const std::vector<RtpHeaderExtension>& video_rtp_header_extensions() {
-    return video_rtp_header_extensions_;
-  }
-
-  virtual bool FindAudioCodec(const AudioCodec& codec) { return true; }
-  virtual bool FindVideoCodec(const VideoCodec& codec) { return true; }
-  virtual void SetVoiceLogging(int min_sev, const char* filter) {}
-  virtual void SetVideoLogging(int min_sev, const char* filter) {}
-  virtual bool StartAecDump(rtc::PlatformFile) { return false; }
-
-  virtual bool RegisterVideoProcessor(VideoProcessor* processor) {
-    return true;
-  }
-  virtual bool UnregisterVideoProcessor(VideoProcessor* processor) {
-    return true;
-  }
-  virtual bool RegisterVoiceProcessor(uint32 ssrc,
-                                      VoiceProcessor* processor,
-                                      MediaProcessorDirection direction) {
-    return true;
-  }
-  virtual bool UnregisterVoiceProcessor(uint32 ssrc,
-                                        VoiceProcessor* processor,
-                                        MediaProcessorDirection direction) {
-    return true;
-  }
-
-  virtual sigslot::repeater2<VideoCapturer*, CaptureState>&
-      SignalVideoCaptureStateChange() {
-    return signal_state_change_;
-  }
-
-  void set_rtp_sender_thread(rtc::Thread* thread) {
-    rtp_sender_thread_ = thread;
-  }
-
- private:
-  std::string voice_input_filename_;
-  std::string voice_output_filename_;
-  std::string video_input_filename_;
-  std::string video_output_filename_;
-  std::vector<AudioCodec> voice_codecs_;
-  std::vector<VideoCodec> video_codecs_;
-  std::vector<RtpHeaderExtension> audio_rtp_header_extensions_;
-  std::vector<RtpHeaderExtension> video_rtp_header_extensions_;
-  sigslot::repeater2<VideoCapturer*, CaptureState>
-     signal_state_change_;
-  rtc::Thread* rtp_sender_thread_;
-
-  DISALLOW_COPY_AND_ASSIGN(FileMediaEngine);
-};
-
-class RtpSenderReceiver;  // Forward declaration. Defined in the .cc file.
-
-class FileVoiceChannel : public VoiceMediaChannel {
- public:
-  FileVoiceChannel(rtc::StreamInterface* input_file_stream,
-      rtc::StreamInterface* output_file_stream,
-      rtc::Thread* rtp_sender_thread);
-  virtual ~FileVoiceChannel();
-
-  // Implement pure virtual methods of VoiceMediaChannel.
-  virtual bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
-    return true;
-  }
-  virtual bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
-  virtual bool SetRecvRtpHeaderExtensions(
-      const std::vector<RtpHeaderExtension>& extensions) {
-    return true;
-  }
-  virtual bool SetSendRtpHeaderExtensions(
-      const std::vector<RtpHeaderExtension>& extensions) {
-    return true;
-  }
-  virtual bool SetPlayout(bool playout) { return true; }
-  virtual bool SetSend(SendFlags flag);
-  virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
-    return false;
-  }
-  virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
-    return false;
-  }
-  virtual bool GetActiveStreams(AudioInfo::StreamList* actives) { return true; }
-  virtual int GetOutputLevel() { return 0; }
-  virtual int GetTimeSinceLastTyping() { return -1; }
-  virtual void SetTypingDetectionParameters(int time_window,
-    int cost_per_typing, int reporting_threshold, int penalty_decay,
-    int type_event_delay) {}
-
-  virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
-    return false;
-  }
-  virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
-    return false;
-  }
-  virtual bool SetRingbackTone(const char* buf, int len) { return true; }
-  virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
-    return true;
-  }
-  virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) {
-    return false;
-  }
-  virtual bool GetStats(VoiceMediaInfo* info) { return true; }
-
-  // Implement pure virtual methods of MediaChannel.
-  virtual void OnPacketReceived(rtc::Buffer* packet,
-                                const rtc::PacketTime& packet_time);
-  virtual void OnRtcpReceived(rtc::Buffer* packet,
-                              const rtc::PacketTime& packet_time) {}
-  virtual void OnReadyToSend(bool ready) {}
-  virtual bool AddSendStream(const StreamParams& sp);
-  virtual bool RemoveSendStream(uint32 ssrc);
-  virtual bool AddRecvStream(const StreamParams& sp) { return true; }
-  virtual bool RemoveRecvStream(uint32 ssrc) { return true; }
-  virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
-  virtual bool SetMaxSendBandwidth(int bps) { return true; }
-  virtual bool SetOptions(const AudioOptions& options) {
-    options_ = options;
-    return true;
-  }
-  virtual bool GetOptions(AudioOptions* options) const {
-    *options = options_;
-    return true;
-  }
-
- private:
-  uint32 send_ssrc_;
-  rtc::scoped_ptr<RtpSenderReceiver> rtp_sender_receiver_;
-  AudioOptions options_;
-
-  DISALLOW_COPY_AND_ASSIGN(FileVoiceChannel);
-};
-
-class FileVideoChannel : public VideoMediaChannel {
- public:
-  FileVideoChannel(rtc::StreamInterface* input_file_stream,
-      rtc::StreamInterface* output_file_stream,
-      rtc::Thread* rtp_sender_thread);
-  virtual ~FileVideoChannel();
-  // Implement pure virtual methods of VideoMediaChannel.
-  void DetachVoiceChannel() override {}
-  virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
-    return true;
-  }
-  virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
-  virtual bool GetSendCodec(VideoCodec* send_codec) {
-    *send_codec = VideoCodec();
-    return true;
-  }
-  virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) {
-    return true;
-  }
-  virtual bool SetRecvRtpHeaderExtensions(
-      const std::vector<RtpHeaderExtension>& extensions) {
-    return true;
-  }
-  virtual bool SetSendRtpHeaderExtensions(
-      const std::vector<RtpHeaderExtension>& extensions) {
-    return true;
-  }
-  virtual bool SetRender(bool render) { return true; }
-  virtual bool SetSend(bool send);
-  virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
-    return true;
-  }
-  virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
-    return false;
-  }
-  virtual bool GetStats(VideoMediaInfo* info) { return true; }
-  virtual bool SendIntraFrame() { return false; }
-  virtual bool RequestIntraFrame() { return false; }
-
-  // Implement pure virtual methods of MediaChannel.
-  virtual void OnPacketReceived(rtc::Buffer* packet,
-                                const rtc::PacketTime& packet_time);
-  virtual void OnRtcpReceived(rtc::Buffer* packet,
-                              const rtc::PacketTime& packet_time) {}
-  virtual void OnReadyToSend(bool ready) {}
-  virtual bool AddSendStream(const StreamParams& sp);
-  virtual bool RemoveSendStream(uint32 ssrc);
-  virtual bool AddRecvStream(const StreamParams& sp) { return true; }
-  virtual bool RemoveRecvStream(uint32 ssrc) { return true; }
-  virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
-  virtual bool SetMaxSendBandwidth(int bps) { return true; }
-  virtual bool SetOptions(const VideoOptions& options) {
-    options_ = options;
-    return true;
-  }
-  virtual bool GetOptions(VideoOptions* options) const {
-    *options = options_;
-    return true;
-  }
-  virtual void UpdateAspectRatio(int ratio_w, int ratio_h) {}
-
- private:
-  uint32 send_ssrc_;
-  rtc::scoped_ptr<RtpSenderReceiver> rtp_sender_receiver_;
-  VideoOptions options_;
-
-  DISALLOW_COPY_AND_ASSIGN(FileVideoChannel);
-};
-
-}  // namespace cricket
-
-#endif  // TALK_MEDIA_BASE_FILEMEDIAENGINE_H_
diff --git a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine_unittest.cc b/chromium/third_party/libjingle/source/talk/media/base/filemediaengine_unittest.cc
deleted file mode 100644
index 43c2c841076..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine_unittest.cc
+++ /dev/null
@@ -1,459 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 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.
- */
-
-#include <set>
-
-#include "talk/media/base/filemediaengine.h"
-#include "talk/media/base/rtpdump.h"
-#include "talk/media/base/streamparams.h"
-#include "talk/media/base/testutils.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/stream.h"
-
-namespace cricket {
-
-static const int kWaitTimeMs = 100;
-static const std::string kFakeFileName = "foobar";
-
-//////////////////////////////////////////////////////////////////////////////
-// Media channel sends RTP packets via NetworkInterface. Rather than sending
-// packets to the network, FileNetworkInterface writes packets to a stream and
-// feeds packets back to the channel via OnPacketReceived.
-//////////////////////////////////////////////////////////////////////////////
-class FileNetworkInterface : public MediaChannel::NetworkInterface {
- public:
-  FileNetworkInterface(rtc::StreamInterface* output, MediaChannel* ch)
-      : media_channel_(ch),
-        num_sent_packets_(0) {
-    if (output) {
-      dump_writer_.reset(new RtpDumpWriter(output));
-    }
-  }
-
-  // Implement pure virtual methods of NetworkInterface.
-  virtual bool SendPacket(rtc::Buffer* packet,
-                          rtc::DiffServCodePoint dscp) {
-    if (!packet) return false;
-
-    if (media_channel_) {
-      media_channel_->OnPacketReceived(packet, rtc::PacketTime());
-    }
-    if (dump_writer_.get() &&
-        rtc::SR_SUCCESS !=
-            dump_writer_->WriteRtpPacket(packet->data(), packet->size())) {
-      return false;
-    }
-
-    ++num_sent_packets_;
-    return true;
-  }
-
-  virtual bool SendRtcp(rtc::Buffer* packet,
-                        rtc::DiffServCodePoint dscp) { return false; }
-  virtual int SetOption(MediaChannel::NetworkInterface::SocketType type,
-      rtc::Socket::Option opt, int option) {
-    return 0;
-  }
-  virtual void SetDefaultDSCPCode(rtc::DiffServCodePoint dscp) {}
-
-  size_t num_sent_packets() const { return num_sent_packets_; }
-
- private:
-  MediaChannel* media_channel_;
-  rtc::scoped_ptr<RtpDumpWriter> dump_writer_;
-  size_t num_sent_packets_;
-
-  DISALLOW_COPY_AND_ASSIGN(FileNetworkInterface);
-};
-
-class FileMediaEngineTest : public testing::Test {
- public:
-  virtual void SetUp() {
-    ASSERT_TRUE(GetTempFilename(&voice_input_filename_));
-    ASSERT_TRUE(GetTempFilename(&voice_output_filename_));
-    ASSERT_TRUE(GetTempFilename(&video_input_filename_));
-    ASSERT_TRUE(GetTempFilename(&video_output_filename_));
-  }
-  virtual void TearDown() {
-    // Force to close the dump files, if opened.
-    voice_channel_.reset();
-    video_channel_.reset();
-
-    DeleteTempFile(voice_input_filename_);
-    DeleteTempFile(voice_output_filename_);
-    DeleteTempFile(video_input_filename_);
-    DeleteTempFile(video_output_filename_);
-  }
-
- protected:
-  void CreateEngineAndChannels(const std::string& voice_in,
-                               const std::string& voice_out,
-                               const std::string& video_in,
-                               const std::string& video_out,
-                               size_t ssrc_count) {
-    // Force to close the dump files, if opened.
-    voice_channel_.reset();
-    video_channel_.reset();
-
-    if (!voice_in.empty()) {
-      EXPECT_TRUE(WriteTestPacketsToFile(voice_in, ssrc_count));
-    }
-    if (!video_in.empty()) {
-      EXPECT_TRUE(WriteTestPacketsToFile(video_in, ssrc_count));
-    }
-
-    engine_.reset(new FileMediaEngine);
-    engine_->set_voice_input_filename(voice_in);
-    engine_->set_voice_output_filename(voice_out);
-    engine_->set_video_input_filename(video_in);
-    engine_->set_video_output_filename(video_out);
-    engine_->set_rtp_sender_thread(rtc::Thread::Current());
-
-    voice_channel_.reset(engine_->CreateChannel());
-    video_channel_.reset(engine_->CreateVideoChannel(VideoOptions(), NULL));
-  }
-
-  bool GetTempFilename(std::string* filename) {
-    rtc::Pathname temp_path;
-    if (!rtc::Filesystem::GetTemporaryFolder(temp_path, true, NULL)) {
-      return false;
-    }
-    temp_path.SetPathname(
-        rtc::Filesystem::TempFilename(temp_path, "fme-test-"));
-
-    if (filename) {
-      *filename = temp_path.pathname();
-    }
-    return true;
-  }
-
-  bool WriteTestPacketsToFile(const std::string& filename, size_t ssrc_count) {
-    rtc::scoped_ptr<rtc::StreamInterface> stream(
-        rtc::Filesystem::OpenFile(rtc::Pathname(filename), "wb"));
-    bool ret = (NULL != stream.get());
-    RtpDumpWriter writer(stream.get());
-
-    for (size_t i = 0; i < ssrc_count; ++i) {
-      ret &= RtpTestUtility::WriteTestPackets(
-          RtpTestUtility::GetTestPacketCount(), false,
-          static_cast<uint32>(RtpTestUtility::kDefaultSsrc + i),
-          &writer);
-    }
-    return ret;
-  }
-
-  void DeleteTempFile(std::string filename) {
-    rtc::Pathname pathname(filename);
-    if (rtc::Filesystem::IsFile(rtc::Pathname(pathname))) {
-      rtc::Filesystem::DeleteFile(pathname);
-    }
-  }
-
-  bool GetSsrcAndPacketCounts(rtc::StreamInterface* stream,
-                              size_t* ssrc_count, size_t* packet_count) {
-    rtc::scoped_ptr<RtpDumpReader> reader(new RtpDumpReader(stream));
-    size_t count = 0;
-    RtpDumpPacket packet;
-    std::set<uint32> ssrcs;
-    while (rtc::SR_SUCCESS == reader->ReadPacket(&packet)) {
-      count++;
-      uint32 ssrc;
-      if (!packet.GetRtpSsrc(&ssrc)) {
-        return false;
-      }
-      ssrcs.insert(ssrc);
-    }
-    if (ssrc_count) {
-      *ssrc_count = ssrcs.size();
-    }
-    if (packet_count) {
-      *packet_count = count;
-    }
-    return true;
-  }
-
-  static const uint32 kWaitTimeout = 3000;
-  std::string voice_input_filename_;
-  std::string voice_output_filename_;
-  std::string video_input_filename_;
-  std::string video_output_filename_;
-  rtc::scoped_ptr<FileMediaEngine> engine_;
-  rtc::scoped_ptr<VoiceMediaChannel> voice_channel_;
-  rtc::scoped_ptr<VideoMediaChannel> video_channel_;
-};
-
-TEST_F(FileMediaEngineTest, TestDefaultImplementation) {
-  CreateEngineAndChannels("", "", "", "", 1);
-  EXPECT_TRUE(engine_->Init(rtc::Thread::Current()));
-  EXPECT_EQ(0, engine_->GetCapabilities());
-  EXPECT_TRUE(NULL == voice_channel_.get());
-  EXPECT_TRUE(NULL == video_channel_.get());
-  EXPECT_TRUE(NULL == engine_->CreateSoundclip());
-  cricket::AudioOptions audio_options;
-  EXPECT_TRUE(engine_->SetAudioOptions(audio_options));
-  VideoEncoderConfig video_encoder_config;
-  EXPECT_TRUE(engine_->SetDefaultVideoEncoderConfig(video_encoder_config));
-  EXPECT_TRUE(engine_->SetSoundDevices(NULL, NULL));
-  EXPECT_TRUE(engine_->SetVideoCaptureDevice(NULL));
-  EXPECT_TRUE(engine_->SetOutputVolume(0));
-  EXPECT_EQ(0, engine_->GetInputLevel());
-  EXPECT_TRUE(engine_->SetLocalMonitor(true));
-  EXPECT_TRUE(engine_->SetVideoCapture(true));
-  EXPECT_EQ(0U, engine_->audio_codecs().size());
-  EXPECT_EQ(0U, engine_->video_codecs().size());
-  AudioCodec voice_codec;
-  EXPECT_TRUE(engine_->FindAudioCodec(voice_codec));
-  VideoCodec video_codec;
-  EXPECT_TRUE(engine_->FindVideoCodec(video_codec));
-  engine_->Terminate();
-}
-
-// Test that when file path is not pointing to a valid stream file, the channel
-// creation function should fail and return NULL.
-TEST_F(FileMediaEngineTest, TestBadFilePath) {
-  engine_.reset(new FileMediaEngine);
-  engine_->set_voice_input_filename(kFakeFileName);
-  engine_->set_video_input_filename(kFakeFileName);
-  EXPECT_TRUE(engine_->CreateChannel() == NULL);
-  EXPECT_TRUE(engine_->CreateVideoChannel(VideoOptions(), NULL) == NULL);
-}
-
-TEST_F(FileMediaEngineTest, TestCodecs) {
-  CreateEngineAndChannels("", "", "", "", 1);
-  std::vector<AudioCodec> voice_codecs = engine_->audio_codecs();
-  std::vector<VideoCodec> video_codecs = engine_->video_codecs();
-  EXPECT_EQ(0U, voice_codecs.size());
-  EXPECT_EQ(0U, video_codecs.size());
-
-  AudioCodec voice_codec(103, "ISAC", 16000, 0, 1, 0);
-  voice_codecs.push_back(voice_codec);
-  engine_->set_voice_codecs(voice_codecs);
-  voice_codecs = engine_->audio_codecs();
-  ASSERT_EQ(1U, voice_codecs.size());
-  EXPECT_EQ(voice_codec, voice_codecs[0]);
-
-  VideoCodec video_codec(96, "H264-SVC", 320, 240, 30, 0);
-  video_codecs.push_back(video_codec);
-  engine_->set_video_codecs(video_codecs);
-  video_codecs = engine_->video_codecs();
-  ASSERT_EQ(1U, video_codecs.size());
-  EXPECT_EQ(video_codec, video_codecs[0]);
-}
-
-// Test that the capabilities and channel creation of the Filemedia engine
-// depend on the stream parameters passed to its constructor.
-TEST_F(FileMediaEngineTest, TestGetCapabilities) {
-  CreateEngineAndChannels(voice_input_filename_, "", "", "", 1);
-  EXPECT_EQ(AUDIO_SEND, engine_->GetCapabilities());
-  EXPECT_TRUE(NULL != voice_channel_.get());
-  EXPECT_TRUE(NULL == video_channel_.get());
-
-  CreateEngineAndChannels(voice_input_filename_, voice_output_filename_, "", "",
-                          1);
-  EXPECT_EQ(AUDIO_SEND | AUDIO_RECV, engine_->GetCapabilities());
-  EXPECT_TRUE(NULL != voice_channel_.get());
-  EXPECT_TRUE(NULL == video_channel_.get());
-
-  CreateEngineAndChannels("", "", video_input_filename_, "", 1);
-  EXPECT_EQ(VIDEO_SEND, engine_->GetCapabilities());
-  EXPECT_TRUE(NULL == voice_channel_.get());
-  EXPECT_TRUE(NULL != video_channel_.get());
-
-  CreateEngineAndChannels(voice_input_filename_, voice_output_filename_,
-                          video_input_filename_, video_output_filename_, 1);
-  EXPECT_EQ(AUDIO_SEND | AUDIO_RECV | VIDEO_SEND | VIDEO_RECV,
-            engine_->GetCapabilities());
-  EXPECT_TRUE(NULL != voice_channel_.get());
-  EXPECT_TRUE(NULL != video_channel_.get());
-}
-
-// FileVideoChannel is the same as FileVoiceChannel in terms of receiving and
-// sending the RTP packets. We therefore test only FileVoiceChannel.
-
-// Test that SetSend() controls whether a voice channel sends RTP packets.
-TEST_F(FileMediaEngineTest, TestVoiceChannelSetSend) {
-  CreateEngineAndChannels(voice_input_filename_, voice_output_filename_, "", "",
-                          1);
-  EXPECT_TRUE(NULL != voice_channel_.get());
-  rtc::MemoryStream net_dump;
-  FileNetworkInterface net_interface(&net_dump, voice_channel_.get());
-  voice_channel_->SetInterface(&net_interface);
-
-  // The channel is not sending yet.
-  rtc::Thread::Current()->ProcessMessages(kWaitTimeMs);
-  EXPECT_EQ(0U, net_interface.num_sent_packets());
-
-  // The channel starts sending.
-  voice_channel_->SetSend(SEND_MICROPHONE);
-  EXPECT_TRUE_WAIT(net_interface.num_sent_packets() >= 1U, kWaitTimeout);
-
-  // The channel stops sending.
-  voice_channel_->SetSend(SEND_NOTHING);
-  // Wait until packets are all delivered.
-  rtc::Thread::Current()->ProcessMessages(kWaitTimeMs);
-  size_t old_number = net_interface.num_sent_packets();
-  rtc::Thread::Current()->ProcessMessages(kWaitTimeMs);
-  EXPECT_EQ(old_number, net_interface.num_sent_packets());
-
-  // The channel starts sending again.
-  voice_channel_->SetSend(SEND_MICROPHONE);
-  EXPECT_TRUE_WAIT(net_interface.num_sent_packets() > old_number, kWaitTimeout);
-
-  // When the function exits, the net_interface object is released. The sender
-  // thread may call net_interface to send packets, which results in a segment
-  // fault. We hence stop sending and wait until all packets are delivered
-  // before we exit this function.
-  voice_channel_->SetSend(SEND_NOTHING);
-  rtc::Thread::Current()->ProcessMessages(kWaitTimeMs);
-}
-
-// Test the sender thread of the channel. The sender sends RTP packets
-// continuously with proper sequence number, timestamp, and payload.
-TEST_F(FileMediaEngineTest, TestVoiceChannelSenderThread) {
-  CreateEngineAndChannels(voice_input_filename_, voice_output_filename_, "", "",
-                          1);
-  EXPECT_TRUE(NULL != voice_channel_.get());
-  rtc::MemoryStream net_dump;
-  FileNetworkInterface net_interface(&net_dump, voice_channel_.get());
-  voice_channel_->SetInterface(&net_interface);
-
-  voice_channel_->SetSend(SEND_MICROPHONE);
-  // Wait until the number of sent packets is no less than 2 * kPacketNumber.
-  EXPECT_TRUE_WAIT(
-      net_interface.num_sent_packets() >=
-          2 * RtpTestUtility::GetTestPacketCount(),
-      kWaitTimeout);
-  voice_channel_->SetSend(SEND_NOTHING);
-  // Wait until packets are all delivered.
-  rtc::Thread::Current()->ProcessMessages(kWaitTimeMs);
-  EXPECT_TRUE(RtpTestUtility::VerifyTestPacketsFromStream(
-      2 * RtpTestUtility::GetTestPacketCount(), &net_dump,
-      RtpTestUtility::kDefaultSsrc));
-
-  // Each sent packet is dumped to net_dump and is also feed to the channel
-  // via OnPacketReceived, which in turn writes the packets into voice_output_.
-  // We next verify the packets in voice_output_.
-  voice_channel_.reset();  // Force to close the files.
-  rtc::scoped_ptr<rtc::StreamInterface> voice_output_;
-  voice_output_.reset(rtc::Filesystem::OpenFile(
-      rtc::Pathname(voice_output_filename_), "rb"));
-  EXPECT_TRUE(voice_output_.get() != NULL);
-  EXPECT_TRUE(RtpTestUtility::VerifyTestPacketsFromStream(
-      2 * RtpTestUtility::GetTestPacketCount(), voice_output_.get(),
-      RtpTestUtility::kDefaultSsrc));
-}
-
-// Test that we can specify the ssrc for outgoing RTP packets.
-TEST_F(FileMediaEngineTest, TestVoiceChannelSendSsrc) {
-  CreateEngineAndChannels(voice_input_filename_, voice_output_filename_, "", "",
-                          1);
-  EXPECT_TRUE(NULL != voice_channel_.get());
-  const uint32 send_ssrc = RtpTestUtility::kDefaultSsrc + 1;
-  voice_channel_->AddSendStream(StreamParams::CreateLegacy(send_ssrc));
-
-  rtc::MemoryStream net_dump;
-  FileNetworkInterface net_interface(&net_dump, voice_channel_.get());
-  voice_channel_->SetInterface(&net_interface);
-
-  voice_channel_->SetSend(SEND_MICROPHONE);
-  // Wait until the number of sent packets is no less than 2 * kPacketNumber.
-  EXPECT_TRUE_WAIT(
-      net_interface.num_sent_packets() >=
-          2 * RtpTestUtility::GetTestPacketCount(),
-      kWaitTimeout);
-  voice_channel_->SetSend(SEND_NOTHING);
-  // Wait until packets are all delivered.
-  rtc::Thread::Current()->ProcessMessages(kWaitTimeMs);
-  EXPECT_TRUE(RtpTestUtility::VerifyTestPacketsFromStream(
-      2 * RtpTestUtility::GetTestPacketCount(), &net_dump, send_ssrc));
-
-  // Each sent packet is dumped to net_dump and is also feed to the channel
-  // via OnPacketReceived, which in turn writes the packets into voice_output_.
-  // We next verify the packets in voice_output_.
-  voice_channel_.reset();  // Force to close the files.
-  rtc::scoped_ptr<rtc::StreamInterface> voice_output_;
-  voice_output_.reset(rtc::Filesystem::OpenFile(
-      rtc::Pathname(voice_output_filename_), "rb"));
-  EXPECT_TRUE(voice_output_.get() != NULL);
-  EXPECT_TRUE(RtpTestUtility::VerifyTestPacketsFromStream(
-      2 * RtpTestUtility::GetTestPacketCount(), voice_output_.get(),
-      send_ssrc));
-}
-
-// Test the sender thread of the channel, where the input rtpdump has two SSRCs.
-TEST_F(FileMediaEngineTest, TestVoiceChannelSenderThreadTwoSsrcs) {
-  CreateEngineAndChannels(voice_input_filename_, voice_output_filename_, "", "",
-                          2);
-  // Verify that voice_input_filename_ contains 2 *
-  // RtpTestUtility::GetTestPacketCount() packets
-  // with different SSRCs.
-  rtc::scoped_ptr<rtc::StreamInterface> input_stream(
-      rtc::Filesystem::OpenFile(
-          rtc::Pathname(voice_input_filename_), "rb"));
-  ASSERT_TRUE(NULL != input_stream.get());
-  size_t ssrc_count;
-  size_t packet_count;
-  EXPECT_TRUE(GetSsrcAndPacketCounts(input_stream.get(), &ssrc_count,
-                                     &packet_count));
-  EXPECT_EQ(2U, ssrc_count);
-  EXPECT_EQ(2 * RtpTestUtility::GetTestPacketCount(), packet_count);
-  input_stream.reset();
-
-  // Send 2 * RtpTestUtility::GetTestPacketCount() packets and verify that all
-  // these packets have the same SSRCs (that is, the packets with different
-  // SSRCs are skipped by the filemediaengine).
-  EXPECT_TRUE(NULL != voice_channel_.get());
-  rtc::MemoryStream net_dump;
-  FileNetworkInterface net_interface(&net_dump, voice_channel_.get());
-  voice_channel_->SetInterface(&net_interface);
-  voice_channel_->SetSend(SEND_MICROPHONE);
-  EXPECT_TRUE_WAIT(
-      net_interface.num_sent_packets() >=
-          2 * RtpTestUtility::GetTestPacketCount(),
-      kWaitTimeout);
-  voice_channel_->SetSend(SEND_NOTHING);
-  // Wait until packets are all delivered.
-  rtc::Thread::Current()->ProcessMessages(kWaitTimeMs);
-  net_dump.Rewind();
-  EXPECT_TRUE(GetSsrcAndPacketCounts(&net_dump, &ssrc_count, &packet_count));
-  EXPECT_EQ(1U, ssrc_count);
-  EXPECT_GE(packet_count, 2 * RtpTestUtility::GetTestPacketCount());
-}
-
-// Test SendIntraFrame() and RequestIntraFrame() of video channel.
-TEST_F(FileMediaEngineTest, TestVideoChannelIntraFrame) {
-  CreateEngineAndChannels("", "", video_input_filename_, video_output_filename_,
-                          1);
-  EXPECT_TRUE(NULL != video_channel_.get());
-  EXPECT_FALSE(video_channel_->SendIntraFrame());
-  EXPECT_FALSE(video_channel_->RequestIntraFrame());
-}
-
-}  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h b/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h
index d77ddbb2483..e7af7a76aa6 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h
@@ -152,12 +152,14 @@ struct AudioOptions {
     stereo_swapping.SetFrom(change.stereo_swapping);
     audio_jitter_buffer_max_packets.SetFrom(
         change.audio_jitter_buffer_max_packets);
+    audio_jitter_buffer_fast_accelerate.SetFrom(
+        change.audio_jitter_buffer_fast_accelerate);
     typing_detection.SetFrom(change.typing_detection);
     aecm_generate_comfort_noise.SetFrom(change.aecm_generate_comfort_noise);
     conference_mode.SetFrom(change.conference_mode);
     adjust_agc_delta.SetFrom(change.adjust_agc_delta);
     experimental_agc.SetFrom(change.experimental_agc);
-    experimental_aec.SetFrom(change.experimental_aec);
+    extended_filter_aec.SetFrom(change.extended_filter_aec);
     delay_agnostic_aec.SetFrom(change.delay_agnostic_aec);
     experimental_ns.SetFrom(change.experimental_ns);
     aec_dump.SetFrom(change.aec_dump);
@@ -183,11 +185,13 @@ struct AudioOptions {
         highpass_filter == o.highpass_filter &&
         stereo_swapping == o.stereo_swapping &&
         audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
+        audio_jitter_buffer_fast_accelerate ==
+            o.audio_jitter_buffer_fast_accelerate &&
         typing_detection == o.typing_detection &&
         aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
         conference_mode == o.conference_mode &&
         experimental_agc == o.experimental_agc &&
-        experimental_aec == o.experimental_aec &&
+        extended_filter_aec == o.extended_filter_aec &&
         delay_agnostic_aec == o.delay_agnostic_aec &&
         experimental_ns == o.experimental_ns &&
         adjust_agc_delta == o.adjust_agc_delta &&
@@ -215,12 +219,14 @@ struct AudioOptions {
     ost << ToStringIfSet("swap", stereo_swapping);
     ost << ToStringIfSet("audio_jitter_buffer_max_packets",
                          audio_jitter_buffer_max_packets);
+    ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
+                         audio_jitter_buffer_fast_accelerate);
     ost << ToStringIfSet("typing", typing_detection);
     ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
     ost << ToStringIfSet("conference", conference_mode);
     ost << ToStringIfSet("agc_delta", adjust_agc_delta);
     ost << ToStringIfSet("experimental_agc", experimental_agc);
-    ost << ToStringIfSet("experimental_aec", experimental_aec);
+    ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
     ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
     ost << ToStringIfSet("experimental_ns", experimental_ns);
     ost << ToStringIfSet("aec_dump", aec_dump);
@@ -255,13 +261,15 @@ struct AudioOptions {
   Settable<bool> stereo_swapping;
   // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
   Settable<int> audio_jitter_buffer_max_packets;
+  // Audio receiver jitter buffer (NetEq) fast accelerate mode.
+  Settable<bool> audio_jitter_buffer_fast_accelerate;
   // Audio processing to detect typing.
   Settable<bool> typing_detection;
   Settable<bool> aecm_generate_comfort_noise;
   Settable<bool> conference_mode;
   Settable<int> adjust_agc_delta;
   Settable<bool> experimental_agc;
-  Settable<bool> experimental_aec;
+  Settable<bool> extended_filter_aec;
   Settable<bool> delay_agnostic_aec;
   Settable<bool> experimental_ns;
   Settable<bool> aec_dump;
@@ -446,22 +454,6 @@ struct VideoOptions {
   Settable<int> screencast_min_bitrate;
 };
 
-// A class for playing out soundclips.
-class SoundclipMedia {
- public:
-  enum SoundclipFlags {
-    SF_LOOP = 1,
-  };
-
-  virtual ~SoundclipMedia() {}
-
-  // Plays a sound out to the speakers with the given audio stream. The stream
-  // must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
-  // on this SoundclipMedia, it is stopped. If clip is NULL, nothing is played.
-  // Returns whether it was successful.
-  virtual bool PlaySound(const char *clip, int len, int flags) = 0;
-};
-
 struct RtpHeaderExtension {
   RtpHeaderExtension() : id(0) {}
   RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}
@@ -792,14 +784,15 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
         expand_rate(0),
         speech_expand_rate(0),
         secondary_decoded_rate(0),
+        accelerate_rate(0),
+        preemptive_expand_rate(0),
         decoding_calls_to_silence_generator(0),
         decoding_calls_to_neteq(0),
         decoding_normal(0),
         decoding_plc(0),
         decoding_cng(0),
         decoding_plc_cng(0),
-        capture_start_ntp_time_ms(-1) {
-  }
+        capture_start_ntp_time_ms(-1) {}
 
   int ext_seqnum;
   int jitter_ms;
@@ -813,6 +806,10 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
   float speech_expand_rate;
   // fraction of data out of secondary decoding, including FEC and RED.
   float secondary_decoded_rate;
+  // Fraction of data removed through time compression.
+  float accelerate_rate;
+  // Fraction of data inserted through time stretching.
+  float preemptive_expand_rate;
   int decoding_calls_to_silence_generator;
   int decoding_calls_to_neteq;
   int decoding_normal;
diff --git a/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h b/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h
index 25adcee7019..501333f94d6 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h
@@ -77,16 +77,13 @@ class MediaEngineInterface {
 
   // MediaChannel creation
   // Creates a voice media channel. Returns NULL on failure.
-  virtual VoiceMediaChannel *CreateChannel() = 0;
+  virtual VoiceMediaChannel* CreateChannel(const AudioOptions& options) = 0;
   // Creates a video media channel, paired with the specified voice channel.
   // Returns NULL on failure.
   virtual VideoMediaChannel* CreateVideoChannel(
       const VideoOptions& options,
       VoiceMediaChannel* voice_media_channel) = 0;
 
-  // Creates a soundclip object for playing sounds on. Returns NULL on failure.
-  virtual SoundclipMedia *CreateSoundclip() = 0;
-
   // Configuration
   // Gets global audio options.
   virtual AudioOptions GetAudioOptions() const = 0;
@@ -101,7 +98,6 @@ class MediaEngineInterface {
       = 0;
 
   // Device selection
-  // TODO(tschmelcher): Add method for selecting the soundclip device.
   virtual bool SetSoundDevices(const Device* in_device,
                                const Device* out_device) = 0;
 
@@ -172,30 +168,23 @@ class CompositeMediaEngine : public MediaEngineInterface {
   virtual bool Init(rtc::Thread* worker_thread) {
     if (!voice_.Init(worker_thread))
       return false;
-    if (!video_.Init(worker_thread)) {
-      voice_.Terminate();
-      return false;
-    }
+    video_.Init();
     return true;
   }
   virtual void Terminate() {
-    video_.Terminate();
     voice_.Terminate();
   }
 
   virtual int GetCapabilities() {
     return (voice_.GetCapabilities() | video_.GetCapabilities());
   }
-  virtual VoiceMediaChannel *CreateChannel() {
-    return voice_.CreateChannel();
+  virtual VoiceMediaChannel* CreateChannel(const AudioOptions& options) {
+    return voice_.CreateChannel(options);
   }
   virtual VideoMediaChannel* CreateVideoChannel(const VideoOptions& options,
                                                 VoiceMediaChannel* channel) {
     return video_.CreateChannel(options, channel);
   }
-  virtual SoundclipMedia *CreateSoundclip() {
-    return voice_.CreateSoundclip();
-  }
 
   virtual AudioOptions GetAudioOptions() const {
     return voice_.GetOptions();
@@ -276,11 +265,8 @@ class NullVoiceEngine {
   void Terminate() {}
   int GetCapabilities() { return 0; }
   // If you need this to return an actual channel, use FakeMediaEngine instead.
-  VoiceMediaChannel* CreateChannel() {
-    return NULL;
-  }
-  SoundclipMedia* CreateSoundclip() {
-    return NULL;
+  VoiceMediaChannel* CreateChannel(const AudioOptions& options) {
+    return nullptr;
   }
   bool SetDelayOffset(int offset) { return true; }
   AudioOptions GetOptions() const { return AudioOptions(); }
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h b/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h
index 90db020abec..e4d2829e88a 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h
@@ -496,7 +496,7 @@ class VideoMediaChannelTest : public testing::Test,
 
   virtual void SetUp() {
     cricket::Device device("test", "device");
-    EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
+    engine_.Init();
     channel_.reset(engine_.CreateChannel(cricket::VideoOptions(), NULL));
     EXPECT_TRUE(channel_.get() != NULL);
     ConnectVideoChannelError();
@@ -551,7 +551,6 @@ class VideoMediaChannelTest : public testing::Test,
   }
   virtual void TearDown() {
     channel_.reset();
-    engine_.Terminate();
   }
   void ConnectVideoChannelError() {
     channel_->SignalMediaError.connect(this,
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videoframe.cc b/chromium/third_party/libjingle/source/talk/media/base/videoframe.cc
index 79f52fcb4e1..7e79bf698c6 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videoframe.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/videoframe.cc
@@ -190,28 +190,6 @@ void VideoFrame::StretchToPlanes(
                 static_cast<int>(width), static_cast<int>(height), interpolate);
 }
 
-size_t VideoFrame::StretchToBuffer(size_t dst_width, size_t dst_height,
-                                   uint8* dst_buffer, size_t size,
-                                   bool interpolate, bool vert_crop) const {
-  if (!dst_buffer) {
-    LOG(LS_ERROR) << "NULL dst_buffer pointer.";
-    return 0;
-  }
-
-  size_t needed = SizeOf(dst_width, dst_height);
-  if (needed <= size) {
-    uint8* dst_y = dst_buffer;
-    uint8* dst_u = dst_y + dst_width * dst_height;
-    uint8* dst_v = dst_u + ((dst_width + 1) >> 1) * ((dst_height + 1) >> 1);
-    StretchToPlanes(dst_y, dst_u, dst_v,
-                    static_cast<int32>(dst_width),
-                    static_cast<int32>((dst_width + 1) >> 1),
-                    static_cast<int32>((dst_width + 1) >> 1),
-                    dst_width, dst_height, interpolate, vert_crop);
-  }
-  return needed;
-}
-
 void VideoFrame::StretchToFrame(VideoFrame* dst,
                                 bool interpolate, bool vert_crop) const {
   if (!dst) {
@@ -225,6 +203,8 @@ void VideoFrame::StretchToFrame(VideoFrame* dst,
                   interpolate, vert_crop);
   dst->SetElapsedTime(GetElapsedTime());
   dst->SetTimeStamp(GetTimeStamp());
+  // Stretched frame should have the same rotation as the source.
+  dst->SetRotation(GetVideoRotation());
 }
 
 VideoFrame* VideoFrame::Stretch(size_t dst_width, size_t dst_height,
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videoframe.h b/chromium/third_party/libjingle/source/talk/media/base/videoframe.h
index 0aac645128c..ac53ca201ba 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videoframe.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/videoframe.h
@@ -188,16 +188,6 @@ class VideoFrame {
       uint8 *y, uint8 *u, uint8 *v, int32 pitchY, int32 pitchU, int32 pitchV,
       size_t width, size_t height, bool interpolate, bool crop) const;
 
-  // Writes the frame into the given frame buffer, stretched to the given width
-  // and height, provided that it is of sufficient size. Returns the frame's
-  // actual size, regardless of whether it was written or not (like snprintf).
-  // If there is insufficient space, nothing is written. The parameter
-  // "interpolate" controls whether to interpolate or just take the
-  // nearest-point. The parameter "crop" controls whether to crop this frame to
-  // the aspect ratio of the given dimensions before stretching.
-  virtual size_t StretchToBuffer(size_t w, size_t h, uint8 *buffer, size_t size,
-                                 bool interpolate, bool crop) const;
-
   // Writes the frame into the target VideoFrame, stretched to the size of that
   // frame. The parameter "interpolate" controls whether to interpolate or just
   // take the nearest-point. The parameter "crop" controls whether to crop this
@@ -230,6 +220,7 @@ class VideoFrame {
                                        size_t pixel_height,
                                        int64_t elapsed_time,
                                        int64_t time_stamp) const = 0;
+  virtual void SetRotation(webrtc::VideoRotation rotation) = 0;
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/media/other/linphonemediaengine.cc b/chromium/third_party/libjingle/source/talk/media/other/linphonemediaengine.cc
deleted file mode 100644
index 06b6d76f12a..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/other/linphonemediaengine.cc
+++ /dev/null
@@ -1,276 +0,0 @@
-/*
- * libjingle
- * Copyright 2010 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.
- */
-
-#ifndef MSILBC_LIBRARY
-#define MSILBC_LIBRARY "/usr/lib/mediastreamer/plugins/libmsilbc.so"
-#endif
-
-// LinphoneMediaEngine is a Linphone implementation of MediaEngine
-extern "C" {
-#include <mediastreamer2/mediastream.h>
-#include <mediastreamer2/msfilter.h>
-#include <mediastreamer2/mssndcard.h>
-}
-
-#include "talk/media/other/linphonemediaengine.h"
-
-#include "talk/media/base/rtpdump.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/event.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/stream.h"
-
-#ifndef WIN32
-#include <libgen.h>
-#endif
-
-namespace cricket {
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of LinphoneMediaEngine.
-///////////////////////////////////////////////////////////////////////////
-LinphoneMediaEngine::LinphoneMediaEngine(const std::string& ringWav,  const std::string& callWav) : ring_wav_(ringWav), call_wav_(callWav) { }
-
-bool LinphoneMediaEngine::Init() {
-  ortp_init();
-  ms_init();
-
-#ifdef HAVE_ILBC
-#ifndef WIN32
-  char * path = strdup(MSILBC_LIBRARY);
-  char * dirc = dirname(path);
-  ms_load_plugins(dirc);
-#endif
-  if (ms_filter_codec_supported("iLBC"))
-    have_ilbc = 1;
-  else
-    have_ilbc = 0;
-#else
-  have_ilbc = 0;
-#endif
-
-#ifdef HAVE_SPEEX
-  voice_codecs_.push_back(AudioCodec(110, payload_type_speex_wb.mime_type, payload_type_speex_wb.clock_rate, 0, 1, 8));
-  voice_codecs_.push_back(AudioCodec(111, payload_type_speex_nb.mime_type, payload_type_speex_nb.clock_rate, 0, 1, 7));
-#endif
-
-#ifdef HAVE_ILBC
-  if (have_ilbc)
-    voice_codecs_.push_back(AudioCodec(102, payload_type_ilbc.mime_type, payload_type_ilbc.clock_rate, 0, 1, 4));
-#endif
-
-  voice_codecs_.push_back(AudioCodec(0, payload_type_pcmu8000.mime_type, payload_type_pcmu8000.clock_rate, 0, 1, 2));
-  voice_codecs_.push_back(AudioCodec(101, payload_type_telephone_event.mime_type, payload_type_telephone_event.clock_rate, 0, 1, 1));
-  return true;
-}
-
-void LinphoneMediaEngine::Terminate() {
-  fflush(stdout);
-}
-
-
-int LinphoneMediaEngine::GetCapabilities() {
-  int capabilities = 0;
-  capabilities |= AUDIO_SEND;
-  capabilities |= AUDIO_RECV;
-  return capabilities;
-}
-
-VoiceMediaChannel* LinphoneMediaEngine::CreateChannel() {
-  return new LinphoneVoiceChannel(this);
-}
-
-VideoMediaChannel* LinphoneMediaEngine::CreateVideoChannel(VoiceMediaChannel* voice_ch) {
-  return NULL;
-}
-
-bool LinphoneMediaEngine::FindAudioCodec(const AudioCodec &c) {
-  if (c.id == 0)
-    return true;
-  if (c.name == payload_type_telephone_event.mime_type)
-    return true;
-#ifdef HAVE_SPEEX
-  if (c.name == payload_type_speex_wb.mime_type && c.clockrate == payload_type_speex_wb.clock_rate)
-    return true;
-  if (c.name == payload_type_speex_nb.mime_type && c.clockrate == payload_type_speex_nb.clock_rate)
-    return true;
-#endif
-#ifdef HAVE_ILBC
-  if (have_ilbc && c.name == payload_type_ilbc.mime_type)
-    return true;
-#endif
-  return false;
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of LinphoneVoiceChannel.
-///////////////////////////////////////////////////////////////////////////
-LinphoneVoiceChannel::LinphoneVoiceChannel(LinphoneMediaEngine*eng)
-    : pt_(-1),
-      audio_stream_(0),
-      engine_(eng),
-      ring_stream_(0)
-{
-
-  rtc::Thread *thread = rtc::ThreadManager::CurrentThread();
-  rtc::SocketServer *ss = thread->socketserver();
-  socket_.reset(ss->CreateAsyncSocket(SOCK_DGRAM));
-
-  socket_->Bind(rtc::SocketAddress("localhost",3000));
-  socket_->SignalReadEvent.connect(this, &LinphoneVoiceChannel::OnIncomingData);
-
-}
-
-LinphoneVoiceChannel::~LinphoneVoiceChannel()
-{
-  fflush(stdout);
-  StopRing();
-
-  if (audio_stream_)
-    audio_stream_stop(audio_stream_);
-}
-
-bool LinphoneVoiceChannel::SetPlayout(bool playout) {
-  play_ = playout;
-  return true;
-}
-
-bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
-
-  bool first = true;
-  std::vector<AudioCodec>::const_iterator i;
-
-  ortp_set_log_level_mask(ORTP_MESSAGE|ORTP_WARNING|ORTP_ERROR|ORTP_FATAL);
-
-  for (i = codecs.begin(); i < codecs.end(); i++) {
-
-    if (!engine_->FindAudioCodec(*i))
-      continue;
-#ifdef HAVE_ILBC
-    if (engine_->have_ilbc && i->name == payload_type_ilbc.mime_type) {
-      rtp_profile_set_payload(&av_profile, i->id, &payload_type_ilbc);
-    }
-#endif
-#ifdef HAVE_SPEEX
-    if (i->name == payload_type_speex_wb.mime_type && i->clockrate == payload_type_speex_wb.clock_rate) {
-      rtp_profile_set_payload(&av_profile, i->id, &payload_type_speex_wb);
-    } else if (i->name == payload_type_speex_nb.mime_type && i->clockrate == payload_type_speex_nb.clock_rate) {
-      rtp_profile_set_payload(&av_profile, i->id, &payload_type_speex_nb);
-    }
-#endif
-
-    if (i->id == 0)
-      rtp_profile_set_payload(&av_profile, 0, &payload_type_pcmu8000);
-
-    if (i->name == payload_type_telephone_event.mime_type) {
-      rtp_profile_set_payload(&av_profile, i->id, &payload_type_telephone_event);
-    }
-
-    if (first) {
-      StopRing();
-      LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
-      pt_ = i->id;
-      audio_stream_ = audio_stream_start(&av_profile, 2000, "127.0.0.1", 3000, i->id, 250, 0);
-      first = false;
-    }
-  }
-
-  if (first) {
-    StopRing();
-    // We're being asked to set an empty list of codecs. This will only happen when
-    // working with a buggy client; let's try PCMU.
-    LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
-    audio_stream_ = audio_stream_start(&av_profile, 2000, "127.0.0.1", 3000, 0, 250, 0);
-  }
-
-  return true;
-}
-
-bool LinphoneVoiceChannel::SetSend(SendFlags flag) {
-  mute_ = !flag;
-  return true;
-}
-
-void LinphoneVoiceChannel::OnPacketReceived(rtc::Buffer* packet) {
-  const void* data = packet->data();
-  int len = packet->length();
-  uint8 buf[2048];
-  memcpy(buf, data, len);
-
-  /* We may receive packets with payload type 13: comfort noise. Linphone can't
-   * handle them, so let's ignore those packets.
-   */
-  int payloadtype = buf[1] & 0x7f;
-  if (play_ && payloadtype != 13)
-    socket_->SendTo(buf, len, rtc::SocketAddress("localhost",2000));
-}
-
-void LinphoneVoiceChannel::StartRing(bool bIncomingCall)
-{
-  MSSndCard *sndcard = NULL;
-  sndcard=ms_snd_card_manager_get_default_card(ms_snd_card_manager_get());
-  if (sndcard)
-  {
-    if (bIncomingCall)
-    {
-      if (engine_->GetRingWav().size() > 0)
-      {
-        LOG(LS_VERBOSE) << "incoming ring. sound file: " << engine_->GetRingWav().c_str() << "\n";
-        ring_stream_ = ring_start (engine_->GetRingWav().c_str(), 1, sndcard);
-      }
-    }
-    else
-    {
-      if (engine_->GetCallWav().size() > 0)
-      {
-        LOG(LS_VERBOSE) << "outgoing ring. sound file: " << engine_->GetCallWav().c_str() << "\n";
-        ring_stream_ = ring_start (engine_->GetCallWav().c_str(), 1, sndcard);
-      }
-    }
-  }
-}
-
-void LinphoneVoiceChannel::StopRing()
-{
-  if (ring_stream_) {
-    ring_stop(ring_stream_);
-    ring_stream_ = 0;
-  }
-}
-
-void LinphoneVoiceChannel::OnIncomingData(rtc::AsyncSocket *s)
-{
-  char *buf[2048];
-  int len;
-  len = s->Recv(buf, sizeof(buf));
-  rtc::Buffer packet(buf, len);
-  if (network_interface_ && !mute_)
-    network_interface_->SendPacket(&packet);
-}
-
-}
diff --git a/chromium/third_party/libjingle/source/talk/media/other/linphonemediaengine.h b/chromium/third_party/libjingle/source/talk/media/other/linphonemediaengine.h
deleted file mode 100644
index b17f4062dfa..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/other/linphonemediaengine.h
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * libjingle
- * Copyright 2010 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.
- */
-
-// LinphoneMediaEngine is a Linphone implementation of MediaEngine
-
-#ifndef TALK_SESSION_PHONE_LINPHONEMEDIAENGINE_H_
-#define TALK_SESSION_PHONE_LINPHONEMEDIAENGINE_H_
-
-#include <string>
-#include <vector>
-
-extern "C" {
-#include <mediastreamer2/mediastream.h>
-}
-
-#include "talk/media/base/codec.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/mediaengine.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace rtc {
-class StreamInterface;
-}
-
-namespace cricket {
-
-class LinphoneMediaEngine : public MediaEngineInterface {
- public:
-  LinphoneMediaEngine(const std::string& ringWav,  const std::string& callWav);
-  virtual ~LinphoneMediaEngine() {}
-
-  // Should be called before codecs() and video_codecs() are called. We need to
-  // set the voice and video codecs; otherwise, Jingle initiation will fail.
-  void set_voice_codecs(const std::vector<AudioCodec>& codecs) {
-    voice_codecs_ = codecs;
-  }
-  void set_video_codecs(const std::vector<VideoCodec>& codecs) {
-    video_codecs_ = codecs;
-  }
-
-  // Implement pure virtual methods of MediaEngine.
-  virtual bool Init();
-  virtual void Terminate();
-  virtual int GetCapabilities();
-  virtual VoiceMediaChannel* CreateChannel();
-  virtual VideoMediaChannel* CreateVideoChannel(VoiceMediaChannel* voice_ch);
-  virtual SoundclipMedia* CreateSoundclip() { return NULL; }
-  virtual bool SetAudioOptions(int options) { return true; }
-  virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
-    return true;
-  }
-  virtual bool SetSoundDevices(const Device* in_dev, const Device* out_dev) {
-    return true;
-  }
-  virtual bool SetVideoCaptureDevice(const Device* cam_device) { return true; }
-  virtual bool SetOutputVolume(int level) { return true; }
-  virtual int GetInputLevel() { return 0; }
-  virtual bool SetLocalMonitor(bool enable) { return true; }
-  // TODO: control channel send?
-  virtual bool SetVideoCapture(bool capture) { return true; }
-  virtual const std::vector<AudioCodec>& audio_codecs() {
-    return voice_codecs_;
-  }
-  virtual const std::vector<VideoCodec>& video_codecs() {
-    return video_codecs_;
-  }
-  virtual bool FindAudioCodec(const AudioCodec& codec);
-  virtual bool FindVideoCodec(const VideoCodec& codec) { return true; }
-  virtual void SetVoiceLogging(int min_sev, const char* filter) {}
-  virtual void SetVideoLogging(int min_sev, const char* filter) {}
-
-  std::string GetRingWav(){return ring_wav_;}
-  std::string GetCallWav(){return call_wav_;}
-
-  int have_ilbc;
-
- private:
-  std::string voice_input_filename_;
-  std::string voice_output_filename_;
-  std::string video_input_filename_;
-  std::string video_output_filename_;
-  std::vector<AudioCodec> voice_codecs_;
-  std::vector<VideoCodec> video_codecs_;
-
-  std::string ring_wav_;
-  std::string call_wav_;
-
-  DISALLOW_COPY_AND_ASSIGN(LinphoneMediaEngine);
-};
-
-class LinphoneVoiceChannel : public VoiceMediaChannel {
- public:
-  LinphoneVoiceChannel(LinphoneMediaEngine *eng);
-  virtual ~LinphoneVoiceChannel();
-
-  // Implement pure virtual methods of VoiceMediaChannel.
-  virtual bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) { return true; }
-  virtual bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
-  virtual bool SetPlayout(bool playout);
-  virtual bool SetSend(SendFlags flag);
-  virtual bool AddStream(uint32 ssrc) { return true; }
-  virtual bool RemoveStream(uint32 ssrc) { return true; }
-  virtual bool GetActiveStreams(AudioInfo::StreamList* actives) { return true; }
-  virtual int GetOutputLevel() { return 0; }
-  virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
-    return false;
-  }
-  virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
-    return false;
-  }
-  virtual void SetRingbackTone(const char* buf, int len) {}
-  virtual bool PlayRingbackTone(bool play, bool loop) { return true; }
-  virtual bool PressDTMF(int event, bool playout) { return true; }
-  virtual bool GetStats(VoiceMediaInfo* info) { return true; }
-
-  // Implement pure virtual methods of MediaChannel.
-  virtual void OnPacketReceived(rtc::Buffer* packet);
-  virtual void OnRtcpReceived(rtc::Buffer* packet) {}
-  virtual void SetSendSsrc(uint32 id) {}  // TODO: change RTP packet?
-  virtual bool SetRtcpCName(const std::string& cname) { return true; }
-  virtual bool Mute(bool on) { return mute_; }
-  virtual bool SetMaxSendBandwidth(int bps) { return true; }
-  virtual bool SetOptions(int options) { return true; }
-  virtual bool SetRecvRtpHeaderExtensions(
-      const std::vector<RtpHeaderExtension>& extensions) { return true; }
-  virtual bool SetSendRtpHeaderExtensions(
-      const std::vector<RtpHeaderExtension>& extensions) { return true; }
-
-  virtual void StartRing(bool bIncomingCall);
-  virtual void StopRing();
-
- private:
-  int pt_;
-  bool mute_;
-  bool play_;
-  AudioStream *audio_stream_;
-  LinphoneMediaEngine *engine_;
-  RingStream* ring_stream_;
-  rtc::scoped_ptr<rtc::AsyncSocket> socket_;
-  void OnIncomingData(rtc::AsyncSocket *s);
-
-  DISALLOW_COPY_AND_ASSIGN(LinphoneVoiceChannel);
-};
-
-}  // namespace cricket
-
-#endif  // TALK_SESSION_PHONE_LINPHONEMEDIAENGINE_H_
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.cc
index f78e8146dd8..9f6a0ff7409 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.cc
@@ -39,6 +39,10 @@ FakeAudioReceiveStream::FakeAudioReceiveStream(
     : config_(config), received_packets_(0) {
 }
 
+webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
+  return webrtc::AudioReceiveStream::Stats();
+}
+
 const webrtc::AudioReceiveStream::Config&
     FakeAudioReceiveStream::GetConfig() const {
   return config_;
@@ -108,7 +112,7 @@ int FakeVideoSendStream::GetLastHeight() const {
 }
 
 void FakeVideoSendStream::IncomingCapturedFrame(
-    const webrtc::I420VideoFrame& frame) {
+    const webrtc::VideoFrame& frame) {
   ++num_swapped_frames_;
   last_frame_.ShallowCopy(frame);
 }
@@ -141,7 +145,7 @@ bool FakeVideoSendStream::ReconfigureVideoEncoder(
   return true;
 }
 
-webrtc::VideoSendStreamInput* FakeVideoSendStream::Input() {
+webrtc::VideoCaptureInput* FakeVideoSendStream::Input() {
   return this;
 }
 
@@ -166,7 +170,7 @@ bool FakeVideoReceiveStream::IsReceiving() const {
   return receiving_;
 }
 
-void FakeVideoReceiveStream::InjectFrame(const webrtc::I420VideoFrame& frame,
+void FakeVideoReceiveStream::InjectFrame(const webrtc::VideoFrame& frame,
                                          int time_to_render_ms) {
   config_.renderer->RenderFrame(frame, time_to_render_ms);
 }
@@ -230,6 +234,14 @@ webrtc::Call::NetworkState FakeCall::GetNetworkState() const {
   return network_state_;
 }
 
+webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
+    const webrtc::AudioSendStream::Config& config) {
+  return nullptr;
+}
+
+void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+}
+
 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   audio_receive_streams_.push_back(new FakeAudioReceiveStream(config));
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.h b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.h
index e4b00d2517a..c500416d347 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtccall.h
@@ -42,6 +42,8 @@ class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
   explicit FakeAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config);
 
+  webrtc::AudioReceiveStream::Stats GetStats() const override;
+
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
 
   int received_packets() const { return received_packets_; }
@@ -53,7 +55,7 @@ class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
 };
 
 class FakeVideoSendStream : public webrtc::VideoSendStream,
-                            public webrtc::VideoSendStreamInput {
+                            public webrtc::VideoCaptureInput {
  public:
   FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
                       const webrtc::VideoEncoderConfig& encoder_config);
@@ -71,13 +73,13 @@ class FakeVideoSendStream : public webrtc::VideoSendStream,
   void SetStats(const webrtc::VideoSendStream::Stats& stats);
 
  private:
-  void IncomingCapturedFrame(const webrtc::I420VideoFrame& frame) override;
+  void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override;
   webrtc::VideoSendStream::Stats GetStats() override;
 
   bool ReconfigureVideoEncoder(
       const webrtc::VideoEncoderConfig& config) override;
 
-  webrtc::VideoSendStreamInput* Input() override;
+  webrtc::VideoCaptureInput* Input() override;
 
   void Start() override;
   void Stop() override;
@@ -91,7 +93,7 @@ class FakeVideoSendStream : public webrtc::VideoSendStream,
     webrtc::VideoCodecVP9 vp9;
   } vpx_settings_;
   int num_swapped_frames_;
-  webrtc::I420VideoFrame last_frame_;
+  webrtc::VideoFrame last_frame_;
   webrtc::VideoSendStream::Stats stats_;
 };
 
@@ -104,7 +106,7 @@ class FakeVideoReceiveStream : public webrtc::VideoReceiveStream {
 
   bool IsReceiving() const;
 
-  void InjectFrame(const webrtc::I420VideoFrame& frame, int time_to_render_ms);
+  void InjectFrame(const webrtc::VideoFrame& frame, int time_to_render_ms);
 
   void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
 
@@ -137,6 +139,10 @@ class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
   void SetStats(const webrtc::Call::Stats& stats);
 
  private:
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config) override;
   void DestroyAudioReceiveStream(
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideocapturemodule.h b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideocapturemodule.h
index 93406cd55ba..3c515555e7d 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideocapturemodule.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideocapturemodule.h
@@ -109,7 +109,7 @@ class FakeWebRtcVideoCaptureModule : public webrtc::VideoCaptureModule {
 
   bool SendFrame(int w, int h) {
     if (!running_) return false;
-    webrtc::I420VideoFrame sample;
+    webrtc::VideoFrame sample;
     // Setting stride based on width.
     if (sample.CreateEmptyFrame(w, h, w, (w + 1) / 2, (w + 1) / 2) < 0) {
       return false;
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h
index fed19c97076..755a3def14c 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -56,8 +56,6 @@ 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 kViECaptureIdBase = 10000;  // Make sure there is a gap.
-static const int kViECaptureIdMax = 11000;
 
 // Fake class for mocking out webrtc::VideoDecoder
 class FakeWebRtcVideoDecoder : public webrtc::VideoDecoder {
@@ -159,10 +157,9 @@ class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
     return codec_settings_;
   }
 
-  virtual int32 Encode(
-      const webrtc::I420VideoFrame& inputImage,
-            const webrtc::CodecSpecificInfo* codecSpecificInfo,
-            const std::vector<webrtc::VideoFrameType>* frame_types) {
+  virtual int32 Encode(const webrtc::VideoFrame& inputImage,
+                       const webrtc::CodecSpecificInfo* codecSpecificInfo,
+                       const std::vector<webrtc::VideoFrameType>* frame_types) {
     rtc::CritScope lock(&crit_);
     ++num_frames_encoded_;
     return WEBRTC_VIDEO_CODEC_OK;
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h
index c6d39bbb223..419170b24dc 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -152,6 +152,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
   WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
   WEBRTC_STUB(StopDebugRecording, ());
+  WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
   webrtc::EchoControlMobile* echo_control_mobile() const override {
     return NULL;
@@ -216,7 +217,8 @@ class FakeWebRtcVoiceEngine
           send_absolute_sender_time_ext_(-1),
           receive_absolute_sender_time_ext_(-1),
           associate_send_channel(-1),
-          neteq_capacity(-1) {
+          neteq_capacity(-1),
+          neteq_fast_accelerate(false) {
       memset(&send_codec, 0, sizeof(send_codec));
       memset(&rx_agc_config, 0, sizeof(rx_agc_config));
     }
@@ -254,6 +256,7 @@ class FakeWebRtcVoiceEngine
     webrtc::PacketTime last_rtp_packet_time;
     std::list<std::string> packets;
     int neteq_capacity;
+    bool neteq_fast_accelerate;
   };
 
   FakeWebRtcVoiceEngine(const cricket::AudioCodec* const* codecs,
@@ -409,6 +412,8 @@ class FakeWebRtcVoiceEngine
     if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
       ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
     }
+    ch->neteq_fast_accelerate =
+        config.Get<webrtc::NetEqFastAccelerate>().enabled;
     channels_[++last_channel_] = ch;
     return last_channel_;
   }
@@ -512,8 +517,6 @@ class FakeWebRtcVoiceEngine
   }
   WEBRTC_STUB(GetVersion, (char version[1024]));
   WEBRTC_STUB(LastError, ());
-  WEBRTC_STUB(SetOnHoldStatus, (int, bool, webrtc::OnHoldModes));
-  WEBRTC_STUB(GetOnHoldStatus, (int, bool&, webrtc::OnHoldModes&));
   WEBRTC_FUNC(AssociateSendChannel, (int channel,
                                      int accociate_send_channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
@@ -577,14 +580,6 @@ class FakeWebRtcVoiceEngine
     }
     return -1;
   }
-  WEBRTC_STUB(SetAMREncFormat, (int channel, webrtc::AmrMode mode));
-  WEBRTC_STUB(SetAMRDecFormat, (int channel, webrtc::AmrMode mode));
-  WEBRTC_STUB(SetAMRWbEncFormat, (int channel, webrtc::AmrMode mode));
-  WEBRTC_STUB(SetAMRWbDecFormat, (int channel, webrtc::AmrMode mode));
-  WEBRTC_STUB(SetISACInitTargetRate, (int channel, int rateBps,
-                                      bool useFixedFrameSize));
-  WEBRTC_STUB(SetISACMaxRate, (int channel, int rateBps));
-  WEBRTC_STUB(SetISACMaxPayloadSize, (int channel, int sizeBytes));
   WEBRTC_FUNC(SetRecPayloadType, (int channel,
                                   const webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
@@ -722,9 +717,6 @@ class FakeWebRtcVoiceEngine
     dtmf_info_.dtmf_length_ms = length_ms;
     return 0;
   }
-  WEBRTC_STUB(StartPlayingDtmfTone,
-      (int eventCode, int attenuationDb = 10));
-  WEBRTC_STUB(StopPlayingDtmfTone, ());
 
   // webrtc::VoEFile
   WEBRTC_FUNC(StartPlayingFileLocally, (int channel, const char* fileNameUTF8,
@@ -752,7 +744,6 @@ class FakeWebRtcVoiceEngine
     WEBRTC_CHECK_CHANNEL(channel);
     return (channels_[channel]->file) ? 1 : 0;
   }
-  WEBRTC_STUB(ScaleLocalFilePlayout, (int channel, float scale));
   WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
                                              const char* fileNameUTF8,
                                              bool loop,
@@ -766,7 +757,6 @@ class FakeWebRtcVoiceEngine
                                              float volumeScaling));
   WEBRTC_STUB(StopPlayingFileAsMicrophone, (int channel));
   WEBRTC_STUB(IsPlayingFileAsMicrophone, (int channel));
-  WEBRTC_STUB(ScaleFileAsMicrophonePlayout, (int channel, float scale));
   WEBRTC_STUB(StartRecordingPlayout, (int channel, const char* fileNameUTF8,
                                       webrtc::CodecInst* compression,
                                       int maxSizeBytes));
@@ -797,30 +787,8 @@ class FakeWebRtcVoiceEngine
     recording_microphone_ = false;
     return 0;
   }
-  WEBRTC_STUB(ConvertPCMToWAV, (const char* fileNameInUTF8,
-                                const char* fileNameOutUTF8));
-  WEBRTC_STUB(ConvertPCMToWAV, (webrtc::InStream* streamIn,
-                                webrtc::OutStream* streamOut));
-  WEBRTC_STUB(ConvertWAVToPCM, (const char* fileNameInUTF8,
-                                const char* fileNameOutUTF8));
-  WEBRTC_STUB(ConvertWAVToPCM, (webrtc::InStream* streamIn,
-                                webrtc::OutStream* streamOut));
-  WEBRTC_STUB(ConvertPCMToCompressed, (const char* fileNameInUTF8,
-                                       const char* fileNameOutUTF8,
-                                       webrtc::CodecInst* compression));
-  WEBRTC_STUB(ConvertPCMToCompressed, (webrtc::InStream* streamIn,
-                                       webrtc::OutStream* streamOut,
-                                       webrtc::CodecInst* compression));
-  WEBRTC_STUB(ConvertCompressedToPCM, (const char* fileNameInUTF8,
-                                     const char* fileNameOutUTF8));
-  WEBRTC_STUB(ConvertCompressedToPCM, (webrtc::InStream* streamIn,
-                                       webrtc::OutStream* streamOut));
-  WEBRTC_STUB(GetFileDuration, (const char* fileNameUTF8, int& durationMs,
-                                webrtc::FileFormats format));
-  WEBRTC_STUB(GetPlaybackPosition, (int channel, int& positionMs));
 
   // webrtc::VoEHardware
-  WEBRTC_STUB(GetCPULoad, (int&));
   WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
     return GetNumDevices(num);
   }
@@ -837,12 +805,6 @@ class FakeWebRtcVoiceEngine
   WEBRTC_STUB(SetPlayoutDevice, (int));
   WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
   WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
-  WEBRTC_STUB(GetPlayoutDeviceStatus, (bool&));
-  WEBRTC_STUB(GetRecordingDeviceStatus, (bool&));
-  WEBRTC_STUB(ResetAudioDevice, ());
-  WEBRTC_STUB(AudioDeviceControl, (unsigned int, unsigned int, unsigned int));
-  WEBRTC_STUB(SetLoudspeakerStatus, (bool enable));
-  WEBRTC_STUB(GetLoudspeakerStatus, (bool& enabled));
   WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
     recording_sample_rate_ = samples_per_sec;
     return 0;
@@ -860,7 +822,6 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
-  virtual bool BuiltInAECIsEnabled() const { return true; }
   virtual bool BuiltInAECIsAvailable() const { return false; }
 
   // webrtc::VoENetEqStats
@@ -912,9 +873,6 @@ class FakeWebRtcVoiceEngine
                                    size_t length));
 
   // webrtc::VoERTP_RTCP
-  WEBRTC_STUB(RegisterRTPObserver, (int channel,
-                                    webrtc::VoERTPObserver& observer));
-  WEBRTC_STUB(DeRegisterRTPObserver, (int channel));
   WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->send_ssrc = ssrc;
@@ -955,7 +913,6 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
 
-  WEBRTC_STUB(GetRemoteCSRCs, (int channel, unsigned int arrCSRC[15]));
   WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
   WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
   WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256]));
@@ -967,8 +924,6 @@ class FakeWebRtcVoiceEngine
                                   unsigned int& playoutTimestamp,
                                   unsigned int* jitter,
                                   unsigned short* fractionLost));
-  WEBRTC_STUB(GetRemoteRTCPSenderInfo, (int channel,
-                                        webrtc::SenderInfo* sender_info));
   WEBRTC_FUNC(GetRemoteRTCPReportBlocks,
               (int channel, std::vector<webrtc::ReportBlock>* receive_blocks)) {
     WEBRTC_CHECK_CHANNEL(channel);
@@ -986,11 +941,6 @@ class FakeWebRtcVoiceEngine
     }
     return 0;
   }
-  WEBRTC_STUB(SendApplicationDefinedRTCPPacket, (int channel,
-                                                 unsigned char subType,
-                                                 unsigned int name,
-                                                 const char* data,
-                                                 unsigned short dataLength));
   WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
                                  unsigned int& maxJitterMs,
                                  unsigned int& discardedPackets));
@@ -1035,15 +985,6 @@ class FakeWebRtcVoiceEngine
     channels_[channel]->nack_max_packets = maxNoPackets;
     return 0;
   }
-  WEBRTC_STUB(StartRTPDump, (int channel, const char* fileNameUTF8,
-                             webrtc::RTPDirections direction));
-  WEBRTC_STUB(StopRTPDump, (int channel, webrtc::RTPDirections direction));
-  WEBRTC_STUB(RTPDumpIsActive, (int channel, webrtc::RTPDirections direction));
-  WEBRTC_STUB(InsertExtraRTPPacket, (int channel, unsigned char payloadType,
-                                     bool markerBit, const char* payloadData,
-                                     unsigned short payloadSize));
-  WEBRTC_STUB(GetLastRemoteTimeStamp, (int channel,
-                                       uint32_t* lastRemoteTimeStamp));
 
   // webrtc::VoEVideoSync
   WEBRTC_STUB(GetPlayoutBufferSize, (int& bufferMs));
@@ -1060,14 +1001,10 @@ class FakeWebRtcVoiceEngine
   // webrtc::VoEVolumeControl
   WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
   WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
-  WEBRTC_STUB(SetSystemOutputMute, (bool));
-  WEBRTC_STUB(GetSystemOutputMute, (bool&));
   WEBRTC_STUB(SetMicVolume, (unsigned int));
   WEBRTC_STUB(GetMicVolume, (unsigned int&));
   WEBRTC_STUB(SetInputMute, (int, bool));
   WEBRTC_STUB(GetInputMute, (int, bool&));
-  WEBRTC_STUB(SetSystemInputMute, (bool));
-  WEBRTC_STUB(GetSystemInputMute, (bool&));
   WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&));
   WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&));
   WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&));
@@ -1253,14 +1190,6 @@ class FakeWebRtcVoiceEngine
     media_processor_ = NULL;
     return 0;
   }
-  WEBRTC_STUB(SetExternalRecordingStatus, (bool enable));
-  WEBRTC_STUB(SetExternalPlayoutStatus, (bool enable));
-  WEBRTC_STUB(ExternalRecordingInsertData,
-              (const int16_t speechData10ms[], int lengthSamples,
-               int samplingFreqHz, int current_delay_ms));
-  WEBRTC_STUB(ExternalPlayoutGetData,
-              (int16_t speechData10ms[], int samplingFreqHz,
-               int current_delay_ms, int& lengthSamples));
   WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
                               webrtc::AudioFrame* frame));
   WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
@@ -1269,6 +1198,11 @@ class FakeWebRtcVoiceEngine
     ASSERT(ch != channels_.end());
     return ch->second->neteq_capacity;
   }
+  bool GetNetEqFastAccelerate() const {
+    auto ch = channels_.find(last_channel_);
+    ASSERT(ch != channels_.end());
+    return ch->second->neteq_fast_accelerate;
+  }
 
  private:
   int GetNumDevices(int& num) {
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/simulcast.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/simulcast.cc
index 6a822b8ad29..759cfef93b2 100755
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/simulcast.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/simulcast.cc
@@ -411,7 +411,7 @@ void LogSimulcastSubstreams(const webrtc::VideoCodec& codec) {
 
 static const int kScreenshareMinBitrateKbps = 50;
 static const int kScreenshareMaxBitrateKbps = 6000;
-static const int kScreenshareDefaultTl0BitrateKbps = 100;
+static const int kScreenshareDefaultTl0BitrateKbps = 200;
 static const int kScreenshareDefaultTl1BitrateKbps = 1000;
 
 static const char* kScreencastLayerFieldTrialName =
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcexport.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcexport.h
deleted file mode 100644
index a2997dcf077..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcexport.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--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.
- */
-
-#ifndef TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
-#define TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
-
-// When building for Chrome a part of the code can be built into
-// a shared library, which is controlled by these macros.
-// For all other builds, we always build a static library.
-#if !defined(GOOGLE_CHROME_BUILD) && !defined(CHROMIUM_BUILD)
-#define LIBPEERCONNECTION_LIB 1
-#endif
-
-#ifndef NON_EXPORTED_BASE
-#ifdef WIN32
-
-// MSVC_SUPPRESS_WARNING disables warning |n| for the remainder of the line and
-// for the next line of the source file.
-#define MSVC_SUPPRESS_WARNING(n) __pragma(warning(suppress:n))
-
-// Allows exporting a class that inherits from a non-exported base class.
-// This uses suppress instead of push/pop because the delimiter after the
-// declaration (either "," or "{") has to be placed before the pop macro.
-//
-// Example usage:
-// class EXPORT_API Foo : NON_EXPORTED_BASE(public Bar) {
-//
-// MSVC Compiler warning C4275:
-// non dll-interface class 'Bar' used as base for dll-interface class 'Foo'.
-// Note that this is intended to be used only when no access to the base class'
-// static data is done through derived classes or inline methods. For more info,
-// see http://msdn.microsoft.com/en-us/library/3tdb471s(VS.80).aspx
-#define NON_EXPORTED_BASE(code) MSVC_SUPPRESS_WARNING(4275) \
-                                code
-
-#else  // Not WIN32
-#define NON_EXPORTED_BASE(code) code
-#endif  // WIN32
-#endif  // NON_EXPORTED_BASE
-
-#if defined (LIBPEERCONNECTION_LIB)
-  #define WRME_EXPORT
-#else
-  #if defined(WIN32)
-    #if defined(LIBPEERCONNECTION_IMPLEMENTATION)
-      #define WRME_EXPORT __declspec(dllexport)
-    #else
-      #define WRME_EXPORT __declspec(dllimport)
-    #endif
-  #else // defined(WIN32)
-    #if defined(LIBPEERCONNECTION_IMPLEMENTATION)
-      #define WRME_EXPORT __attribute__((visibility("default")))
-    #else
-      #define WRME_EXPORT
-    #endif
-  #endif
-#endif  // LIBPEERCONNECTION_LIB
-
-#endif  // TALK_MEDIA_WEBRTC_WEBRTCEXPORT_H_
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.cc
index ca36f7331d8..cf6a5cf2e59 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.cc
@@ -25,8 +25,6 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#if defined(LIBPEERCONNECTION_LIB) || defined(LIBPEERCONNECTION_IMPLEMENTATION)
-
 #include "talk/media/webrtc/webrtcmediaengine.h"
 #include "talk/media/webrtc/webrtcvideoengine2.h"
 #include "talk/media/webrtc/webrtcvoiceengine.h"
@@ -37,10 +35,9 @@ class WebRtcMediaEngine2
     : public CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine2> {
  public:
   WebRtcMediaEngine2(webrtc::AudioDeviceModule* adm,
-                     webrtc::AudioDeviceModule* adm_sc,
                      WebRtcVideoEncoderFactory* encoder_factory,
                      WebRtcVideoDecoderFactory* decoder_factory) {
-    voice_.SetAudioDeviceModule(adm, adm_sc);
+    voice_.SetAudioDeviceModule(adm);
     video_.SetExternalDecoderFactory(decoder_factory);
     video_.SetExternalEncoderFactory(encoder_factory);
   }
@@ -48,17 +45,14 @@ class WebRtcMediaEngine2
 
 }  // namespace cricket
 
-WRME_EXPORT
 cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
     webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
-  return new cricket::WebRtcMediaEngine2(adm, adm_sc, encoder_factory,
+  return new cricket::WebRtcMediaEngine2(adm, encoder_factory,
                                          decoder_factory);
 }
 
-WRME_EXPORT
 void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
   delete media_engine;
 }
@@ -69,13 +63,9 @@ namespace cricket {
 // ChannelManager.
 MediaEngineInterface* WebRtcMediaEngineFactory::Create(
     webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
     WebRtcVideoEncoderFactory* encoder_factory,
     WebRtcVideoDecoderFactory* decoder_factory) {
-  return CreateWebRtcMediaEngine(adm, adm_sc, encoder_factory, decoder_factory);
+  return CreateWebRtcMediaEngine(adm, encoder_factory, decoder_factory);
 }
 
 }  // namespace cricket
-
-#endif  // defined(LIBPEERCONNECTION_LIB) ||
-        // defined(LIBPEERCONNECTION_IMPLEMENTATION)
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.h
index 48da1763b59..0fd8a74eb66 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcmediaengine.h
@@ -29,7 +29,6 @@
 #define TALK_MEDIA_WEBRTCMEDIAENGINE_H_
 
 #include "talk/media/base/mediaengine.h"
-#include "talk/media/webrtc/webrtcexport.h"
 
 namespace webrtc {
 class AudioDeviceModule;
@@ -40,162 +39,17 @@ class WebRtcVideoDecoderFactory;
 class WebRtcVideoEncoderFactory;
 }
 
-#if !defined(LIBPEERCONNECTION_LIB) && \
-    !defined(LIBPEERCONNECTION_IMPLEMENTATION)
-
-WRME_EXPORT
-cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
-    webrtc::AudioDeviceModule* adm, webrtc::AudioDeviceModule* adm_sc,
-    cricket::WebRtcVideoEncoderFactory* encoder_factory,
-    cricket::WebRtcVideoDecoderFactory* decoder_factory);
-
-WRME_EXPORT
-void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine);
-
-#endif  // !defined(LIBPEERCONNECTION_LIB) &&
-        // !defined(LIBPEERCONNECTION_IMPLEMENTATION)
-
 namespace cricket {
 
 class WebRtcMediaEngineFactory {
  public:
-#if !defined(LIBPEERCONNECTION_LIB) && \
-    !defined(LIBPEERCONNECTION_IMPLEMENTATION)
-// A bare Create() isn't supported when using the delegating media
-// engine.
-#else
   static MediaEngineInterface* Create();
-#endif  // !defined(LIBPEERCONNECTION_LIB) &&
-        // !defined(LIBPEERCONNECTION_IMPLEMENTATION)
   static MediaEngineInterface* Create(
       webrtc::AudioDeviceModule* adm,
-      webrtc::AudioDeviceModule* adm_sc,
       WebRtcVideoEncoderFactory* encoder_factory,
       WebRtcVideoDecoderFactory* decoder_factory);
 };
 
 }  // namespace cricket
 
-
-#if !defined(LIBPEERCONNECTION_LIB) && \
-    !defined(LIBPEERCONNECTION_IMPLEMENTATION)
-
-namespace cricket {
-
-// TODO(pthacther): Move this code into webrtcmediaengine.cc once
-// Chrome compiles it.  Right now it relies on only the .h file.
-class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
- public:
-  DelegatingWebRtcMediaEngine(
-      webrtc::AudioDeviceModule* adm,
-      webrtc::AudioDeviceModule* adm_sc,
-      WebRtcVideoEncoderFactory* encoder_factory,
-      WebRtcVideoDecoderFactory* decoder_factory)
-      : delegate_(CreateWebRtcMediaEngine(
-          adm, adm_sc, encoder_factory, decoder_factory)) {
-  }
-  virtual ~DelegatingWebRtcMediaEngine() {
-    DestroyWebRtcMediaEngine(delegate_);
-  }
-  bool Init(rtc::Thread* worker_thread) override {
-    return delegate_->Init(worker_thread);
-  }
-  void Terminate() override { delegate_->Terminate(); }
-  int GetCapabilities() override { return delegate_->GetCapabilities(); }
-  VoiceMediaChannel* CreateChannel() override {
-    return delegate_->CreateChannel();
-  }
-  VideoMediaChannel* CreateVideoChannel(
-      const VideoOptions& options,
-      VoiceMediaChannel* voice_media_channel) override {
-    return delegate_->CreateVideoChannel(options, voice_media_channel);
-  }
-  SoundclipMedia* CreateSoundclip() override {
-    return delegate_->CreateSoundclip();
-  }
-  AudioOptions GetAudioOptions() const override {
-    return delegate_->GetAudioOptions();
-  }
-  bool SetAudioOptions(const AudioOptions& options) override {
-    return delegate_->SetAudioOptions(options);
-  }
-  bool SetAudioDelayOffset(int offset) override {
-    return delegate_->SetAudioDelayOffset(offset);
-  }
-  bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) override {
-    return delegate_->SetDefaultVideoEncoderConfig(config);
-  }
-  bool SetSoundDevices(const Device* in_device,
-                       const Device* out_device) override {
-    return delegate_->SetSoundDevices(in_device, out_device);
-  }
-  bool GetOutputVolume(int* level) override {
-    return delegate_->GetOutputVolume(level);
-  }
-  bool SetOutputVolume(int level) override {
-    return delegate_->SetOutputVolume(level);
-  }
-  int GetInputLevel() override { return delegate_->GetInputLevel(); }
-  bool SetLocalMonitor(bool enable) override {
-    return delegate_->SetLocalMonitor(enable);
-  }
-  const std::vector<AudioCodec>& audio_codecs() override {
-    return delegate_->audio_codecs();
-  }
-  const std::vector<RtpHeaderExtension>& audio_rtp_header_extensions()
-      override {
-    return delegate_->audio_rtp_header_extensions();
-  }
-  const std::vector<VideoCodec>& video_codecs() override {
-    return delegate_->video_codecs();
-  }
-  const std::vector<RtpHeaderExtension>& video_rtp_header_extensions()
-      override {
-    return delegate_->video_rtp_header_extensions();
-  }
-  void SetVoiceLogging(int min_sev, const char* filter) override {
-    delegate_->SetVoiceLogging(min_sev, filter);
-  }
-  void SetVideoLogging(int min_sev, const char* filter) override {
-    delegate_->SetVideoLogging(min_sev, filter);
-  }
-  bool StartAecDump(rtc::PlatformFile file) override {
-    return delegate_->StartAecDump(file);
-  }
-  bool RegisterVoiceProcessor(uint32 ssrc,
-                              VoiceProcessor* video_processor,
-                              MediaProcessorDirection direction) override {
-    return delegate_->RegisterVoiceProcessor(ssrc, video_processor, direction);
-  }
-  bool UnregisterVoiceProcessor(uint32 ssrc,
-                                VoiceProcessor* video_processor,
-                                MediaProcessorDirection direction) override {
-    return delegate_->UnregisterVoiceProcessor(ssrc, video_processor,
-        direction);
-  }
-  virtual sigslot::repeater2<VideoCapturer*, CaptureState>&
-      SignalVideoCaptureStateChange() {
-    return delegate_->SignalVideoCaptureStateChange();
-  }
-
- private:
-  cricket::MediaEngineInterface* delegate_;
-};
-
-// Used by PeerConnectionFactory to create a media engine passed into
-// ChannelManager.
-MediaEngineInterface* WebRtcMediaEngineFactory::Create(
-    webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
-    WebRtcVideoEncoderFactory* encoder_factory,
-    WebRtcVideoDecoderFactory* decoder_factory) {
-  return new cricket::DelegatingWebRtcMediaEngine(
-      adm, adm_sc, encoder_factory, decoder_factory);
-}
-
-}  // namespace cricket
-
-#endif  // !defined(LIBPEERCONNECTION_LIB) &&
-        // !defined(LIBPEERCONNECTION_IMPLEMENTATION)
-
 #endif  // TALK_MEDIA_WEBRTCMEDIAENGINE_H_
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.cc
index 89772884c95..e3178607629 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.cc
@@ -41,7 +41,7 @@ class PassthroughStream: public webrtc::VideoRenderCallback {
   virtual ~PassthroughStream() {
   }
   virtual int32_t RenderFrame(const uint32_t stream_id,
-                              const webrtc::I420VideoFrame& videoFrame) {
+                              const webrtc::VideoFrame& videoFrame) {
     rtc::CritScope cs(&stream_critical_);
     // Send frame for rendering directly
     if (running_ && renderer_) {
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.h
index b1e4fd3380a..685cfe881c2 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender.h
@@ -161,12 +161,12 @@ class WebRtcPassthroughRender : public webrtc::VideoRender {
   }
 
   int32_t SetStartImage(const uint32_t stream_id,
-                        const webrtc::I420VideoFrame& videoFrame) override {
+                        const webrtc::VideoFrame& videoFrame) override {
     return -1;
   }
 
   int32_t SetTimeoutImage(const uint32_t stream_id,
-                          const webrtc::I420VideoFrame& videoFrame,
+                          const webrtc::VideoFrame& videoFrame,
                           const uint32_t timeout) override {
     return -1;
   }
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender_unittest.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender_unittest.cc
index 9bbba0e340c..65eed05f8ed 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcpassthroughrender_unittest.cc
@@ -44,7 +44,7 @@ class WebRtcPassthroughRenderTest : public testing::Test {
     }
 
     virtual int32_t RenderFrame(const uint32_t stream_id,
-                                const webrtc::I420VideoFrame& videoFrame) {
+                                const webrtc::VideoFrame& videoFrame) {
       ++frame_num_;
       LOG(INFO) << "RenderFrame stream_id: " << stream_id
                 << " frame_num: " << frame_num_;
@@ -121,7 +121,7 @@ TEST_F(WebRtcPassthroughRenderTest, Streams) {
 }
 
 TEST_F(WebRtcPassthroughRenderTest, Renderer) {
-  webrtc::I420VideoFrame frame;
+  webrtc::VideoFrame frame;
   const int stream_id1 = 1234;
   const int stream_id2 = 5678;
   const int stream_id3 = 9012;  // A stream that doesn't exist.
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.cc
index 285ed01966b..f8c373db36b 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.cc
@@ -129,17 +129,19 @@ static bool FormatToCapability(const VideoFormat& format,
 
 WebRtcVideoCapturer::WebRtcVideoCapturer()
     : factory_(new WebRtcVcmFactory),
-      module_(NULL),
+      module_(nullptr),
       captured_frames_(0),
-      start_thread_(nullptr) {
+      start_thread_(nullptr),
+      async_invoker_(nullptr) {
   set_frame_factory(new WebRtcVideoFrameFactory());
 }
 
 WebRtcVideoCapturer::WebRtcVideoCapturer(WebRtcVcmFactoryInterface* factory)
     : factory_(factory),
-      module_(NULL),
+      module_(nullptr),
       captured_frames_(0),
-      start_thread_(nullptr) {
+      start_thread_(nullptr),
+      async_invoker_(nullptr) {
   set_frame_factory(new WebRtcVideoFrameFactory());
 }
 
@@ -200,15 +202,12 @@ bool WebRtcVideoCapturer::Init(const Device& device) {
     }
   }
   factory_->DestroyDeviceInfo(info);
-// TODO(fischman): Remove the following check
-// when capabilities for iOS are implemented
-// https://code.google.com/p/webrtc/issues/detail?id=2968
-#if !defined(IOS)
+
   if (supported.empty()) {
     LOG(LS_ERROR) << "Failed to find usable formats for id: " << device.id;
     return false;
   }
-#endif
+
   module_ = factory_->Create(0, vcm_id);
   if (!module_) {
     LOG(LS_ERROR) << "Failed to create capturer for id: " << device.id;
@@ -284,16 +283,17 @@ CaptureState WebRtcVideoCapturer::Start(const VideoFormat& capture_format) {
     LOG(LS_ERROR) << "The capturer has not been initialized";
     return CS_NO_DEVICE;
   }
-
-  rtc::CritScope cs(&critical_section_stopping_);
-  if (IsRunning()) {
+  if (start_thread_) {
     LOG(LS_ERROR) << "The capturer is already running";
+    DCHECK(start_thread_->IsCurrent())
+        << "Trying to start capturer on different threads";
     return CS_FAILED;
   }
 
-  DCHECK(!start_thread_);
-
   start_thread_ = rtc::Thread::Current();
+  DCHECK(!async_invoker_);
+  async_invoker_.reset(new rtc::AsyncInvoker());
+  captured_frames_ = 0;
 
   SetCaptureFormat(&capture_format);
 
@@ -303,44 +303,50 @@ CaptureState WebRtcVideoCapturer::Start(const VideoFormat& capture_format) {
     return CS_FAILED;
   }
 
-  std::string camera_id(GetId());
   uint32 start = rtc::Time();
   module_->RegisterCaptureDataCallback(*this);
   if (module_->StartCapture(cap) != 0) {
-    LOG(LS_ERROR) << "Camera '" << camera_id << "' failed to start";
+    LOG(LS_ERROR) << "Camera '" << GetId() << "' failed to start";
+    module_->DeRegisterCaptureDataCallback();
+    async_invoker_.reset();
+    SetCaptureFormat(nullptr);
     start_thread_ = nullptr;
     return CS_FAILED;
   }
 
-  LOG(LS_INFO) << "Camera '" << camera_id << "' started with format "
+  LOG(LS_INFO) << "Camera '" << GetId() << "' started with format "
                << capture_format.ToString() << ", elapsed time "
                << rtc::TimeSince(start) << " ms";
 
-  captured_frames_ = 0;
   SetCaptureState(CS_RUNNING);
   return CS_STARTING;
 }
 
-// Critical section blocks Stop from shutting down during callbacks from capture
-// thread to OnIncomingCapturedFrame. Note that the crit is try-locked in
-// OnFrameCaptured, as the lock ordering between this and the system component
-// controlling the camera is reversed: system frame -> OnIncomingCapturedFrame;
-// Stop -> system stop camera).
 void WebRtcVideoCapturer::Stop() {
-  rtc::CritScope cs(&critical_section_stopping_);
+  if (!start_thread_) {
+    LOG(LS_ERROR) << "The capturer is already stopped";
+    return;
+  }
+  DCHECK(start_thread_);
+  DCHECK(start_thread_->IsCurrent());
+  DCHECK(async_invoker_);
   if (IsRunning()) {
-    DCHECK(start_thread_);
-    rtc::Thread::Current()->Clear(this);
+    // The module is responsible for OnIncomingCapturedFrame being called, if
+    // we stop it we will get no further callbacks.
     module_->StopCapture();
-    module_->DeRegisterCaptureDataCallback();
-
-    // TODO(juberti): Determine if the VCM exposes any drop stats we can use.
-    double drop_ratio = 0.0;
-    std::string camera_id(GetId());
-    LOG(LS_INFO) << "Camera '" << camera_id << "' stopped after capturing "
-                 << captured_frames_ << " frames and dropping "
-                 << drop_ratio << "%";
   }
+  module_->DeRegisterCaptureDataCallback();
+
+  // TODO(juberti): Determine if the VCM exposes any drop stats we can use.
+  double drop_ratio = 0.0;
+  LOG(LS_INFO) << "Camera '" << GetId() << "' stopped after capturing "
+               << captured_frames_ << " frames and dropping "
+               << drop_ratio << "%";
+
+  // Clear any pending async invokes (that OnIncomingCapturedFrame may have
+  // caused).
+  async_invoker_.reset();
+
   SetCaptureFormat(NULL);
   start_thread_ = nullptr;
 }
@@ -364,38 +370,23 @@ bool WebRtcVideoCapturer::GetPreferredFourccs(
 
 void WebRtcVideoCapturer::OnIncomingCapturedFrame(
     const int32_t id,
-    const webrtc::I420VideoFrame& sample) {
-  // This would be a normal CritScope, except that it's possible that:
-  // (1) whatever system component producing this frame has taken a lock, and
-  // (2) Stop() probably calls back into that system component, which may take
-  // the same lock. Due to the reversed order, we have to try-lock in order to
-  // avoid a potential deadlock. Besides, if we can't enter because we're
-  // stopping, we may as well drop the frame.
-  rtc::TryCritScope cs(&critical_section_stopping_);
-  if (!cs.locked() || !IsRunning()) {
-    // Capturer has been stopped or is in the process of stopping.
-    return;
-  }
-
-  ++captured_frames_;
-  // Log the size and pixel aspect ratio of the first captured frame.
-  if (1 == captured_frames_) {
-    LOG(LS_INFO) << "Captured frame size "
-                 << sample.width() << "x" << sample.height()
-                 << ". Expected format " << GetCaptureFormat()->ToString();
-  }
-
+    const webrtc::VideoFrame& sample) {
+  // This can only happen between Start() and Stop().
+  DCHECK(start_thread_);
+  DCHECK(async_invoker_);
   if (start_thread_->IsCurrent()) {
-    SignalFrameCapturedOnStartThread(&sample);
+    SignalFrameCapturedOnStartThread(sample);
   } else {
     // This currently happens on with at least VideoCaptureModuleV4L2 and
     // possibly other implementations of WebRTC's VideoCaptureModule.
     // In order to maintain the threading contract with the upper layers and
     // consistency with other capturers such as in Chrome, we need to do a
     // thread hop.
-    start_thread_->Invoke<void>(
+    // Note that Stop() can cause the async invoke call to be cancelled.
+    async_invoker_->AsyncInvoke<void>(start_thread_,
+        // Note that this results in a shallow copying of the frame.
         rtc::Bind(&WebRtcVideoCapturer::SignalFrameCapturedOnStartThread,
-                  this, &sample));
+                  this, sample));
   }
 }
 
@@ -405,23 +396,35 @@ void WebRtcVideoCapturer::OnCaptureDelayChanged(const int32_t id,
 }
 
 void WebRtcVideoCapturer::SignalFrameCapturedOnStartThread(
-    const webrtc::I420VideoFrame* frame) {
+    const webrtc::VideoFrame frame) {
+  // This can only happen between Start() and Stop().
+  DCHECK(start_thread_);
   DCHECK(start_thread_->IsCurrent());
+  DCHECK(async_invoker_);
+
+  ++captured_frames_;
+  // Log the size and pixel aspect ratio of the first captured frame.
+  if (1 == captured_frames_) {
+    LOG(LS_INFO) << "Captured frame size "
+                 << frame.width() << "x" << frame.height()
+                 << ". Expected format " << GetCaptureFormat()->ToString();
+  }
+
   // Signal down stream components on captured frame.
   // The CapturedFrame class doesn't support planes. We have to ExtractBuffer
   // to one block for it.
   size_t length =
-      webrtc::CalcBufferSize(webrtc::kI420, frame->width(), frame->height());
+      webrtc::CalcBufferSize(webrtc::kI420, frame.width(), frame.height());
   capture_buffer_.resize(length);
   // TODO(magjed): Refactor the WebRtcCapturedFrame to avoid memory copy or
   // take over ownership of the buffer held by |frame| if that's possible.
-  webrtc::ExtractBuffer(*frame, length, &capture_buffer_[0]);
-  WebRtcCapturedFrame webrtc_frame(*frame, &capture_buffer_[0], length);
+  webrtc::ExtractBuffer(frame, length, &capture_buffer_[0]);
+  WebRtcCapturedFrame webrtc_frame(frame, &capture_buffer_[0], length);
   SignalFrameCaptured(this, &webrtc_frame);
 }
 
 // WebRtcCapturedFrame
-WebRtcCapturedFrame::WebRtcCapturedFrame(const webrtc::I420VideoFrame& sample,
+WebRtcCapturedFrame::WebRtcCapturedFrame(const webrtc::VideoFrame& sample,
                                          void* buffer,
                                          size_t length) {
   width = sample.width();
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.h
index 56896f9cd69..fe545ad7474 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideocapturer.h
@@ -35,8 +35,9 @@
 
 #include "talk/media/base/videocapturer.h"
 #include "talk/media/webrtc/webrtcvideoframe.h"
-#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/messagehandler.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_capture/include/video_capture.h"
 
@@ -81,7 +82,7 @@ class WebRtcVideoCapturer : public VideoCapturer,
  private:
   // Callback when a frame is captured by camera.
   virtual void OnIncomingCapturedFrame(const int32_t id,
-                                       const webrtc::I420VideoFrame& frame);
+                                       const webrtc::VideoFrame& frame);
   virtual void OnCaptureDelayChanged(const int32_t id,
                                      const int32_t delay);
 
@@ -91,7 +92,7 @@ class WebRtcVideoCapturer : public VideoCapturer,
   // directly from OnIncomingCapturedFrame.
   // TODO(tommi): Remove this workaround when we've updated the WebRTC capturers
   // to follow the same contract.
-  void SignalFrameCapturedOnStartThread(const webrtc::I420VideoFrame* frame);
+  void SignalFrameCapturedOnStartThread(const webrtc::VideoFrame frame);
 
   rtc::scoped_ptr<WebRtcVcmFactoryInterface> factory_;
   webrtc::VideoCaptureModule* module_;
@@ -99,14 +100,14 @@ class WebRtcVideoCapturer : public VideoCapturer,
   std::vector<uint8_t> capture_buffer_;
   rtc::Thread* start_thread_;  // Set in Start(), unset in Stop();
 
-  // Critical section to avoid Stop during an OnIncomingCapturedFrame callback.
-  rtc::CriticalSection critical_section_stopping_;
+  rtc::scoped_ptr<rtc::AsyncInvoker> async_invoker_;
 };
 
 struct WebRtcCapturedFrame : public CapturedFrame {
  public:
-  WebRtcCapturedFrame(const webrtc::I420VideoFrame& frame,
-                      void* buffer, size_t length);
+  WebRtcCapturedFrame(const webrtc::VideoFrame& frame,
+                      void* buffer,
+                      size_t length);
 };
 
 }  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc
deleted file mode 100644
index 6ff09a3c8a1..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc
+++ /dev/null
@@ -1 +0,0 @@
-// TODO(pbos): Remove this file when it's no longer built in Chromium.
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.cc
index b296b194556..0a2152e5e4c 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.cc
@@ -36,7 +36,6 @@
 #include "talk/media/base/videorenderer.h"
 #include "talk/media/webrtc/constants.h"
 #include "talk/media/webrtc/simulcast.h"
-#include "talk/media/webrtc/webrtcvideocapturer.h"
 #include "talk/media/webrtc/webrtcvideoencoderfactory.h"
 #include "talk/media/webrtc/webrtcvideoframe.h"
 #include "talk/media/webrtc/webrtcvoiceengine.h"
@@ -44,6 +43,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/call.h"
+#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
 #include "webrtc/system_wrappers/interface/field_trial.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
@@ -158,6 +158,10 @@ bool CodecIsInternallySupported(const std::string& codec_name) {
         webrtc::field_trial::FindFullName("WebRTC-SupportVP9");
     return group_name == "Enabled" || group_name == "EnabledByFlag";
   }
+  if (CodecNamesEq(codec_name, kH264CodecName)) {
+    return webrtc::H264Encoder::IsSupported() &&
+        webrtc::H264Decoder::IsSupported();
+  }
   return false;
 }
 
@@ -317,8 +321,6 @@ static const int kDefaultQpMax = 56;
 
 static const int kDefaultRtcpReceiverReportSsrc = 1;
 
-const char kH264CodecName[] = "H264";
-
 const int kMinBandwidthBps = 30000;
 const int kStartBandwidthBps = 300000;
 const int kMaxBandwidthBps = 2000000;
@@ -332,6 +334,10 @@ std::vector<VideoCodec> DefaultVideoCodecList() {
   }
   codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType,
                                                            kVp8CodecName));
+  if (CodecIsInternallySupported(kH264CodecName)) {
+    codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultH264PlType,
+                                                             kH264CodecName));
+  }
   codecs.push_back(
       VideoCodec::CreateRtxCodec(kDefaultRtxVp8PlType, kDefaultVp8PlType));
   codecs.push_back(VideoCodec(kDefaultRedPlType, kRedCodecName));
@@ -530,8 +536,7 @@ void DefaultUnsignalledSsrcHandler::SetDefaultRenderer(
 }
 
 WebRtcVideoEngine2::WebRtcVideoEngine2(WebRtcVoiceEngine* voice_engine)
-    : worker_thread_(NULL),
-      voice_engine_(voice_engine),
+    : voice_engine_(voice_engine),
       initialized_(false),
       call_factory_(&default_call_factory_),
       external_decoder_factory_(NULL),
@@ -551,10 +556,6 @@ WebRtcVideoEngine2::WebRtcVideoEngine2(WebRtcVoiceEngine* voice_engine)
 
 WebRtcVideoEngine2::~WebRtcVideoEngine2() {
   LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
-
-  if (initialized_) {
-    Terminate();
-  }
 }
 
 void WebRtcVideoEngine2::SetCallFactory(WebRtcCallFactory* call_factory) {
@@ -562,19 +563,9 @@ void WebRtcVideoEngine2::SetCallFactory(WebRtcCallFactory* call_factory) {
   call_factory_ = call_factory;
 }
 
-bool WebRtcVideoEngine2::Init(rtc::Thread* worker_thread) {
+void WebRtcVideoEngine2::Init() {
   LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
-  worker_thread_ = worker_thread;
-  DCHECK(worker_thread_ != NULL);
-
   initialized_ = true;
-  return true;
-}
-
-void WebRtcVideoEngine2::Terminate() {
-  LOG(LS_INFO) << "WebRtcVideoEngine2::Terminate";
-
-  initialized_ = false;
 }
 
 int WebRtcVideoEngine2::GetCapabilities() { return VIDEO_RECV | VIDEO_SEND; }
@@ -1181,7 +1172,7 @@ bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp,
   }
 
   receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
-      call_.get(), sp.ssrcs, external_decoder_factory_, default_stream, config,
+      call_.get(), sp, external_decoder_factory_, default_stream, config,
       recv_codecs_);
 
   return true;
@@ -1660,6 +1651,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
     const StreamParams& sp,
     const std::vector<webrtc::RtpExtension>& rtp_extensions)
     : ssrcs_(sp.ssrcs),
+      ssrc_groups_(sp.ssrc_groups),
       call_(call),
       external_encoder_factory_(external_encoder_factory),
       stream_(NULL),
@@ -1694,7 +1686,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
   DestroyVideoEncoder(&allocated_encoder_);
 }
 
-static void CreateBlackFrame(webrtc::I420VideoFrame* video_frame,
+static void CreateBlackFrame(webrtc::VideoFrame* video_frame,
                              int width,
                              int height) {
   video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2,
@@ -1711,8 +1703,8 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame(
     VideoCapturer* capturer,
     const VideoFrame* frame) {
   TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::InputFrame");
-  webrtc::I420VideoFrame video_frame(frame->GetVideoFrameBuffer(), 0, 0,
-                                     frame->GetVideoRotation());
+  webrtc::VideoFrame video_frame(frame->GetVideoFrameBuffer(), 0, 0,
+                                 frame->GetVideoRotation());
   rtc::CritScope cs(&lock_);
   if (stream_ == NULL) {
     // Frame input before send codecs are configured, dropping frame.
@@ -1759,7 +1751,7 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
     if (capturer == NULL) {
       if (stream_ != NULL) {
         LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
-        webrtc::I420VideoFrame black_frame;
+        webrtc::VideoFrame black_frame;
 
         CreateBlackFrame(&black_frame, last_dimensions_.width,
                          last_dimensions_.height);
@@ -1891,6 +1883,9 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
   } else if (type == webrtc::kVideoCodecVP9) {
     return AllocatedEncoder(
         webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp9), type, false);
+  } else if (type == webrtc::kVideoCodecH264) {
+    return AllocatedEncoder(
+        webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kH264), type, false);
   }
 
   // This shouldn't happen, we should not be trying to create something we don't
@@ -2116,6 +2111,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
       }
     }
   }
+  info.ssrc_groups = ssrc_groups_;
   info.framerate_input = stats.input_frame_rate;
   info.framerate_sent = stats.encode_frame_rate;
   info.avg_encode_ms = stats.avg_encode_time_ms;
@@ -2218,13 +2214,14 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
 
 WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
     webrtc::Call* call,
-    const std::vector<uint32>& ssrcs,
+    const StreamParams& sp,
     WebRtcVideoDecoderFactory* external_decoder_factory,
     bool default_stream,
     const webrtc::VideoReceiveStream::Config& config,
     const std::vector<VideoCodecSettings>& recv_codecs)
     : call_(call),
-      ssrcs_(ssrcs),
+      ssrcs_(sp.ssrcs),
+      ssrc_groups_(sp.ssrc_groups),
       stream_(NULL),
       default_stream_(default_stream),
       config_(config),
@@ -2297,6 +2294,11 @@ WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
         webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp9), type, false);
   }
 
+  if (type == webrtc::kVideoCodecH264) {
+    return AllocatedDecoder(
+        webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kH264), type, false);
+  }
+
   // This shouldn't happen, we should not be trying to create something we don't
   // support.
   DCHECK(false);
@@ -2381,7 +2383,7 @@ void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders(
 }
 
 void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RenderFrame(
-    const webrtc::I420VideoFrame& frame,
+    const webrtc::VideoFrame& frame,
     int time_to_render_ms) {
   rtc::CritScope crit(&renderer_lock_);
 
@@ -2448,6 +2450,7 @@ void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSize(int width,
 VideoReceiverInfo
 WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo() {
   VideoReceiverInfo info;
+  info.ssrc_groups = ssrc_groups_;
   info.add_ssrc(config_.rtp.remote_ssrc);
   webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
   info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.h
index 5ee2c1e185d..e3059758400 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.h
@@ -123,8 +123,7 @@ class WebRtcVideoEngine2 : public sigslot::has_slots<> {
   void SetCallFactory(WebRtcCallFactory* call_factory);
 
   // Basic video engine implementation.
-  bool Init(rtc::Thread* worker_thread);
-  void Terminate();
+  void Init();
 
   int GetCapabilities();
   bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
@@ -158,7 +157,6 @@ class WebRtcVideoEngine2 : public sigslot::has_slots<> {
  private:
   std::vector<VideoCodec> GetSupportedCodecs() const;
 
-  rtc::Thread* worker_thread_;
   WebRtcVoiceEngine* voice_engine_;
   std::vector<VideoCodec> video_codecs_;
   std::vector<RtpHeaderExtension> rtp_header_extensions_;
@@ -378,6 +376,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
         EXCLUSIVE_LOCKS_REQUIRED(lock_);
 
     const std::vector<uint32> ssrcs_;
+    const std::vector<SsrcGroup> ssrc_groups_;
     webrtc::Call* const call_;
     WebRtcVideoEncoderFactory* const external_encoder_factory_
         GUARDED_BY(lock_);
@@ -402,8 +401,8 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
   class WebRtcVideoReceiveStream : public webrtc::VideoRenderer {
    public:
     WebRtcVideoReceiveStream(
-        webrtc::Call*,
-        const std::vector<uint32>& ssrcs,
+        webrtc::Call* call,
+        const StreamParams& sp,
         WebRtcVideoDecoderFactory* external_decoder_factory,
         bool default_stream,
         const webrtc::VideoReceiveStream::Config& config,
@@ -417,7 +416,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
     void SetRecvCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
     void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
 
-    void RenderFrame(const webrtc::I420VideoFrame& frame,
+    void RenderFrame(const webrtc::VideoFrame& frame,
                      int time_to_render_ms) override;
     bool IsTextureSupported() const override;
     bool IsDefaultStream() const;
@@ -449,6 +448,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
 
     webrtc::Call* const call_;
     const std::vector<uint32> ssrcs_;
+    const std::vector<SsrcGroup> ssrc_groups_;
 
     webrtc::VideoReceiveStream* stream_;
     const bool default_stream_;
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index 21693eccc81..230ca3b2b06 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -74,7 +74,7 @@ void VerifyCodecHasDefaultFeedbackParams(const cricket::VideoCodec& codec) {
       cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir)));
 }
 
-static void CreateBlackFrame(webrtc::I420VideoFrame* video_frame,
+static void CreateBlackFrame(webrtc::VideoFrame* video_frame,
                              int width,
                              int height) {
   video_frame->CreateEmptyFrame(
@@ -173,13 +173,12 @@ class WebRtcVideoEngine2VoiceTest : public WebRtcVideoEngine2Test {
 TEST_F(WebRtcVideoEngine2VoiceTest, ConfiguresAvSyncForFirstReceiveChannel) {
   FakeCallFactory call_factory;
   engine_.SetCallFactory(&call_factory);
-
   voice_engine_.Init(rtc::Thread::Current());
-  engine_.Init(rtc::Thread::Current());
+  engine_.Init();
 
   rtc::scoped_ptr<VoiceMediaChannel> voice_channel(
-      voice_engine_.CreateChannel());
-  ASSERT_TRUE(voice_channel.get() != NULL);
+      voice_engine_.CreateChannel(cricket::AudioOptions()));
+  ASSERT_TRUE(voice_channel.get() != nullptr);
   WebRtcVoiceMediaChannel* webrtc_voice_channel =
       static_cast<WebRtcVoiceMediaChannel*>(voice_channel.get());
   ASSERT_NE(webrtc_voice_channel->voe_channel(), -1);
@@ -187,11 +186,11 @@ TEST_F(WebRtcVideoEngine2VoiceTest, ConfiguresAvSyncForFirstReceiveChannel) {
       engine_.CreateChannel(cricket::VideoOptions(), voice_channel.get()));
 
   FakeCall* fake_call = call_factory.GetCall();
-  ASSERT_TRUE(fake_call != NULL);
+  ASSERT_TRUE(fake_call != nullptr);
 
   webrtc::Call::Config call_config = fake_call->GetConfig();
 
-  ASSERT_TRUE(voice_engine_.voe()->engine() != NULL);
+  ASSERT_TRUE(voice_engine_.voe()->engine() != nullptr);
   ASSERT_EQ(voice_engine_.voe()->engine(), call_config.voice_engine);
 
   EXPECT_TRUE(channel->AddRecvStream(StreamParams::CreateLegacy(kSsrc)));
@@ -386,7 +385,7 @@ TEST_F(WebRtcVideoEngine2Test, CVOSetHeaderExtensionAfterCapturer) {
 }
 
 TEST_F(WebRtcVideoEngine2Test, SetSendFailsBeforeSettingCodecs) {
-  engine_.Init(rtc::Thread::Current());
+  engine_.Init();
   rtc::scoped_ptr<VideoMediaChannel> channel(
       engine_.CreateChannel(cricket::VideoOptions(), NULL));
 
@@ -399,7 +398,7 @@ TEST_F(WebRtcVideoEngine2Test, SetSendFailsBeforeSettingCodecs) {
 }
 
 TEST_F(WebRtcVideoEngine2Test, GetStatsWithoutSendCodecsSetDoesNotCrash) {
-  engine_.Init(rtc::Thread::Current());
+  engine_.Init();
   rtc::scoped_ptr<VideoMediaChannel> channel(
       engine_.CreateChannel(cricket::VideoOptions(), NULL));
   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
@@ -461,7 +460,7 @@ VideoMediaChannel* WebRtcVideoEngine2Test::SetUpForExternalEncoderFactory(
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     const std::vector<VideoCodec>& codecs) {
   engine_.SetExternalEncoderFactory(encoder_factory);
-  engine_.Init(rtc::Thread::Current());
+  engine_.Init();
 
   VideoMediaChannel* channel =
       engine_.CreateChannel(cricket::VideoOptions(), NULL);
@@ -474,7 +473,7 @@ VideoMediaChannel* WebRtcVideoEngine2Test::SetUpForExternalDecoderFactory(
     cricket::WebRtcVideoDecoderFactory* decoder_factory,
     const std::vector<VideoCodec>& codecs) {
   engine_.SetExternalDecoderFactory(decoder_factory);
-  engine_.Init(rtc::Thread::Current());
+  engine_.Init();
 
   VideoMediaChannel* channel =
       engine_.CreateChannel(cricket::VideoOptions(), NULL);
@@ -623,8 +622,7 @@ TEST_F(WebRtcVideoEngine2Test, ReportSupportedExternalCodecs) {
   cricket::FakeWebRtcVideoEncoderFactory encoder_factory;
   encoder_factory.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264");
   engine_.SetExternalEncoderFactory(&encoder_factory);
-
-  engine_.Init(rtc::Thread::Current());
+  engine_.Init();
 
   std::vector<cricket::VideoCodec> codecs(engine_.codecs());
   ASSERT_GE(codecs.size(), 2u);
@@ -822,7 +820,7 @@ class WebRtcVideoChannel2Test : public WebRtcVideoEngine2Test,
   WebRtcVideoChannel2Test() : fake_call_(NULL), last_ssrc_(0) {}
   void SetUp() override {
     engine_.SetCallFactory(this);
-    engine_.Init(rtc::Thread::Current());
+    engine_.Init();
     channel_.reset(engine_.CreateChannel(cricket::VideoOptions(), NULL));
     ASSERT_TRUE(fake_call_ != NULL) << "Call not created through factory.";
     last_ssrc_ = 123;
@@ -1134,11 +1132,11 @@ TEST_F(WebRtcVideoChannel2Test, IdenticalRecvExtensionsDoesntRecreateStream) {
       kRtpVideoRotationHeaderExtension, kVideoRotationId));
 
   EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
-  FakeVideoReceiveStream* send_stream =
+  FakeVideoReceiveStream* recv_stream =
       AddRecvStream(cricket::StreamParams::CreateLegacy(123));
 
   EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
-  ASSERT_EQ(3u, send_stream->GetConfig().rtp.extensions.size());
+  ASSERT_EQ(3u, recv_stream->GetConfig().rtp.extensions.size());
 
   // Setting the same extensions (even if in different order) shouldn't
   // reallocate the stream.
@@ -1440,7 +1438,8 @@ TEST_F(WebRtcVideoChannel2Test, UsesCorrectSettingsForScreencast) {
 
 TEST_F(WebRtcVideoChannel2Test,
        ConferenceModeScreencastConfiguresTemporalLayer) {
-  static const int kConferenceScreencastTemporalBitrateBps = 100000;
+  static const int kConferenceScreencastTemporalBitrateBps =
+      ScreenshareLayerConfig::GetDefault().tl0_bitrate_kbps * 1000;
   VideoOptions options;
   options.conference_mode.Set(true);
   channel_->SetOptions(options);
@@ -1592,8 +1591,6 @@ class Vp9SettingsTest : public WebRtcVideoChannel2Test {
   void TearDown() override {
     // Remove references to encoder_factory_ since this will be destroyed
     // before channel_ and engine_.
-    engine_.Terminate();
-    engine_.SetExternalEncoderFactory(nullptr);
     ASSERT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
   }
 
@@ -1736,7 +1733,7 @@ TEST_F(WebRtcVideoChannel2Test, EstimatesNtpStartTimeAndElapsedTimeCorrectly) {
   EXPECT_TRUE(channel_->SetRenderer(last_ssrc_, &renderer));
   EXPECT_TRUE(channel_->SetRender(true));
 
-  webrtc::I420VideoFrame video_frame;
+  webrtc::VideoFrame video_frame;
   CreateBlackFrame(&video_frame, 4, 4);
   video_frame.set_timestamp(kInitialTimestamp);
   // Initial NTP time is not available on the first frame, but should still be
@@ -2556,6 +2553,35 @@ TEST_F(WebRtcVideoChannel2Test,
   EXPECT_TRUE(channel_->AddRecvStream(sp));
 }
 
+TEST_F(WebRtcVideoChannel2Test, ReportsSsrcGroupsInStats) {
+  EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
+
+  static const uint32_t kSenderSsrcs[] = {4, 7, 10};
+  static const uint32_t kSenderRtxSsrcs[] = {5, 8, 11};
+
+  StreamParams sender_sp = cricket::CreateSimWithRtxStreamParams(
+      "cname", MAKE_VECTOR(kSenderSsrcs), MAKE_VECTOR(kSenderRtxSsrcs));
+
+  EXPECT_TRUE(channel_->AddSendStream(sender_sp));
+
+  static const uint32_t kReceiverSsrcs[] = {3};
+  static const uint32_t kReceiverRtxSsrcs[] = {2};
+
+  StreamParams receiver_sp = cricket::CreateSimWithRtxStreamParams(
+      "cname", MAKE_VECTOR(kReceiverSsrcs), MAKE_VECTOR(kReceiverRtxSsrcs));
+  EXPECT_TRUE(channel_->AddRecvStream(receiver_sp));
+
+  cricket::VideoMediaInfo info;
+  ASSERT_TRUE(channel_->GetStats(&info));
+
+  ASSERT_EQ(1u, info.senders.size());
+  ASSERT_EQ(1u, info.receivers.size());
+
+  EXPECT_NE(sender_sp.ssrc_groups, receiver_sp.ssrc_groups);
+  EXPECT_EQ(sender_sp.ssrc_groups, info.senders[0].ssrc_groups);
+  EXPECT_EQ(receiver_sp.ssrc_groups, info.receivers[0].ssrc_groups);
+}
+
 void WebRtcVideoChannel2Test::TestReceiverLocalSsrcConfiguration(
     bool receiver_first) {
   EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
@@ -2603,7 +2629,7 @@ class WebRtcVideoChannel2SimulcastTest : public WebRtcVideoEngine2SimulcastTest,
 
   void SetUp() override {
     engine_.SetCallFactory(this);
-    engine_.Init(rtc::Thread::Current());
+    engine_.Init();
     channel_.reset(engine_.CreateChannel(VideoOptions(), NULL));
     ASSERT_TRUE(fake_call_ != NULL) << "Call not created through factory.";
     last_ssrc_ = 123;
@@ -2899,16 +2925,6 @@ TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TestAdaptToOutputFormat) {
   FAIL() << "Not implemented.";
 }
 
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TestAdaptToCpuLoad) {
-  // TODO(pbos): Implement.
-  FAIL() << "Not implemented.";
-}
-
-TEST_F(WebRtcVideoEngine2SimulcastTest, DISABLED_TestAdaptToCpuLoadDisabled) {
-  // TODO(pbos): Implement.
-  FAIL() << "Not implemented.";
-}
-
 TEST_F(WebRtcVideoEngine2SimulcastTest,
        DISABLED_TestAdaptWithCpuOveruseObserver) {
   // TODO(pbos): Implement.
@@ -2929,7 +2945,7 @@ TEST_F(WebRtcVideoEngine2SimulcastTest,
 }
 
 TEST_F(WebRtcVideoEngine2SimulcastTest,
-       DISABLED_DontUseSimulcastAdapterOnNoneVp8Factory) {
+       DISABLED_DontUseSimulcastAdapterOnNonVp8Factory) {
   // TODO(pbos): Implement.
   FAIL() << "Not implemented.";
 }
@@ -2969,8 +2985,7 @@ TEST_F(WebRtcVideoChannel2SimulcastTest, DISABLED_SimulcastSend_480x300) {
   FAIL() << "Not implemented.";
 }
 
-TEST_F(WebRtcVideoChannel2SimulcastTest,
-       DISABLED_DISABLED_SimulcastSend_480x270) {
+TEST_F(WebRtcVideoChannel2SimulcastTest, DISABLED_SimulcastSend_480x270) {
   // TODO(pbos): Implement.
   FAIL() << "Not implemented.";
 }
@@ -2985,14 +3000,6 @@ TEST_F(WebRtcVideoChannel2SimulcastTest, DISABLED_SimulcastSend_320x180) {
   FAIL() << "Not implemented.";
 }
 
-// Test reset send codec with simulcast.
-// Disabled per b/6773425
-TEST_F(WebRtcVideoChannel2SimulcastTest,
-       DISABLED_DISABLED_SimulcastResetSendCodec) {
-  // TODO(pbos): Implement.
-  FAIL() << "Not implemented.";
-}
-
 // Test simulcast streams are decodeable with expected sizes.
 TEST_F(WebRtcVideoChannel2SimulcastTest, DISABLED_SimulcastStreams) {
   // TODO(pbos): Implement.
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe.h
index 06ac52bfb2b..790047277dd 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe.h
@@ -131,7 +131,9 @@ class WebRtcVideoFrame : public VideoFrame {
   const VideoFrame* GetCopyWithRotationApplied() const override;
 
  protected:
-  void SetRotation(webrtc::VideoRotation rotation) { rotation_ = rotation; }
+  void SetRotation(webrtc::VideoRotation rotation) override {
+    rotation_ = rotation;
+  }
 
  private:
   virtual VideoFrame* CreateEmptyFrame(int w, int h, size_t pixel_width,
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe_unittest.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe_unittest.cc
index 7386f51e7b9..daa8ffa46ef 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframe_unittest.cc
@@ -29,6 +29,7 @@
 
 #include "talk/media/base/videoframe_unittest.h"
 #include "talk/media/webrtc/webrtcvideoframe.h"
+#include "webrtc/test/fake_texture_frame.h"
 
 namespace {
 
@@ -297,10 +298,11 @@ TEST_F(WebRtcVideoFrameTest, InitRotated90DontApplyRotation) {
 }
 
 TEST_F(WebRtcVideoFrameTest, TextureInitialValues) {
-  void* dummy_handle = reinterpret_cast<void*>(0x1);
-  webrtc::TextureBuffer* buffer =
-      new rtc::RefCountedObject<webrtc::TextureBuffer>(dummy_handle, 640, 480,
-                                                       rtc::Callback0<void>());
+  webrtc::test::FakeNativeHandle* dummy_handle =
+      new webrtc::test::FakeNativeHandle();
+  webrtc::NativeHandleBuffer* buffer =
+      new rtc::RefCountedObject<webrtc::test::FakeNativeHandleBuffer>(
+          dummy_handle, 640, 480);
   cricket::WebRtcVideoFrame frame(buffer, 100, 200, webrtc::kVideoRotation_0);
   EXPECT_EQ(dummy_handle, frame.GetNativeHandle());
   EXPECT_EQ(640u, frame.GetWidth());
@@ -314,10 +316,11 @@ TEST_F(WebRtcVideoFrameTest, TextureInitialValues) {
 }
 
 TEST_F(WebRtcVideoFrameTest, CopyTextureFrame) {
-  void* dummy_handle = reinterpret_cast<void*>(0x1);
-  webrtc::TextureBuffer* buffer =
-      new rtc::RefCountedObject<webrtc::TextureBuffer>(dummy_handle, 640, 480,
-                                                       rtc::Callback0<void>());
+  webrtc::test::FakeNativeHandle* dummy_handle =
+      new webrtc::test::FakeNativeHandle();
+  webrtc::NativeHandleBuffer* buffer =
+      new rtc::RefCountedObject<webrtc::test::FakeNativeHandleBuffer>(
+          dummy_handle, 640, 480);
   cricket::WebRtcVideoFrame frame1(buffer, 100, 200, webrtc::kVideoRotation_0);
   cricket::VideoFrame* frame2 = frame1.Copy();
   EXPECT_EQ(frame1.GetNativeHandle(), frame2->GetNativeHandle());
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframefactory_unittest.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframefactory_unittest.cc
new file mode 100644
index 00000000000..4dbad0556f2
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoframefactory_unittest.cc
@@ -0,0 +1,125 @@
+/*
+ * libjingle
+ * Copyright 2015 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.
+ */
+
+#include <string.h>
+
+#include "talk/media/base/videoframe_unittest.h"
+#include "talk/media/webrtc/webrtcvideoframe.h"
+#include "talk/media/webrtc/webrtcvideoframefactory.h"
+
+class WebRtcVideoFrameFactoryTest
+    : public VideoFrameTest<cricket::WebRtcVideoFrameFactory> {
+ public:
+  WebRtcVideoFrameFactoryTest() {}
+
+  void InitFrame(webrtc::VideoRotation frame_rotation) {
+    const int frame_width = 1920;
+    const int frame_height = 1080;
+
+    // Build the CapturedFrame.
+    captured_frame_.fourcc = cricket::FOURCC_I420;
+    captured_frame_.pixel_width = 1;
+    captured_frame_.pixel_height = 1;
+    captured_frame_.elapsed_time = 1234;
+    captured_frame_.time_stamp = 5678;
+    captured_frame_.rotation = frame_rotation;
+    captured_frame_.width = frame_width;
+    captured_frame_.height = frame_height;
+    captured_frame_.data_size =
+        (frame_width * frame_height) +
+        ((frame_width + 1) / 2) * ((frame_height + 1) / 2) * 2;
+    captured_frame_buffer_.reset(new uint8[captured_frame_.data_size]);
+    // Initialize memory to satisfy DrMemory tests.
+    memset(captured_frame_buffer_.get(), 0, captured_frame_.data_size);
+    captured_frame_.data = captured_frame_buffer_.get();
+  }
+
+  void VerifyFrame(cricket::VideoFrame* dest_frame,
+                   webrtc::VideoRotation src_rotation,
+                   int src_width,
+                   int src_height,
+                   bool apply_rotation) {
+    if (!apply_rotation) {
+      EXPECT_EQ(dest_frame->GetRotation(), src_rotation);
+      EXPECT_EQ(dest_frame->GetWidth(), src_width);
+      EXPECT_EQ(dest_frame->GetHeight(), src_height);
+    } else {
+      EXPECT_EQ(dest_frame->GetRotation(), webrtc::kVideoRotation_0);
+      if (src_rotation == webrtc::kVideoRotation_90 ||
+          src_rotation == webrtc::kVideoRotation_270) {
+        EXPECT_EQ(dest_frame->GetWidth(), src_height);
+        EXPECT_EQ(dest_frame->GetHeight(), src_width);
+      } else {
+        EXPECT_EQ(dest_frame->GetWidth(), src_width);
+        EXPECT_EQ(dest_frame->GetHeight(), src_height);
+      }
+    }
+  }
+
+  void TestCreateAliasedFrame(bool apply_rotation) {
+    cricket::VideoFrameFactory& factory = factory_;
+    factory.SetApplyRotation(apply_rotation);
+    InitFrame(webrtc::kVideoRotation_270);
+    const cricket::CapturedFrame& captured_frame = get_captured_frame();
+    // Create the new frame from the CapturedFrame.
+    rtc::scoped_ptr<cricket::VideoFrame> frame;
+    int new_width = captured_frame.width / 2;
+    int new_height = captured_frame.height / 2;
+    frame.reset(factory.CreateAliasedFrame(&captured_frame, new_width,
+                                           new_height, new_width, new_height));
+    VerifyFrame(frame.get(), webrtc::kVideoRotation_270, new_width, new_height,
+                apply_rotation);
+
+    frame.reset(factory.CreateAliasedFrame(
+        &captured_frame, new_width, new_height, new_width / 2, new_height / 2));
+    VerifyFrame(frame.get(), webrtc::kVideoRotation_270, new_width / 2,
+                new_height / 2, apply_rotation);
+
+    // Reset the frame first so it's exclusive hence we could go through the
+    // StretchToFrame code path in CreateAliasedFrame.
+    frame.reset();
+    frame.reset(factory.CreateAliasedFrame(
+        &captured_frame, new_width, new_height, new_width / 2, new_height / 2));
+    VerifyFrame(frame.get(), webrtc::kVideoRotation_270, new_width / 2,
+                new_height / 2, apply_rotation);
+  }
+
+  const cricket::CapturedFrame& get_captured_frame() { return captured_frame_; }
+
+ private:
+  cricket::CapturedFrame captured_frame_;
+  rtc::scoped_ptr<uint8[]> captured_frame_buffer_;
+  cricket::WebRtcVideoFrameFactory factory_;
+};
+
+TEST_F(WebRtcVideoFrameFactoryTest, NoApplyRotation) {
+  TestCreateAliasedFrame(false);
+}
+
+TEST_F(WebRtcVideoFrameFactoryTest, ApplyRotation) {
+  TestCreateAliasedFrame(true);
+}
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc
index 62404db9f21..284afed4d11 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc
@@ -103,7 +103,6 @@ static const CodecPref kCodecPrefs[] = {
 
 #ifdef WIN32
 static const int kDefaultAudioDeviceId = -1;
-static const int kDefaultSoundclipDeviceId = -2;
 #else
 static const int kDefaultAudioDeviceId = 0;
 #endif
@@ -353,11 +352,12 @@ static AudioOptions GetDefaultEngineOptions() {
   options.highpass_filter.Set(true);
   options.stereo_swapping.Set(false);
   options.audio_jitter_buffer_max_packets.Set(50);
+  options.audio_jitter_buffer_fast_accelerate.Set(false);
   options.typing_detection.Set(true);
   options.conference_mode.Set(false);
   options.adjust_agc_delta.Set(0);
   options.experimental_agc.Set(false);
-  options.experimental_aec.Set(false);
+  options.extended_filter_aec.Set(false);
   options.delay_agnostic_aec.Set(false);
   options.experimental_ns.Set(false);
   options.aec_dump.Set(false);
@@ -368,103 +368,10 @@ static std::string GetEnableString(bool enable) {
   return enable ? "enable" : "disable";
 }
 
-class WebRtcSoundclipMedia : public SoundclipMedia {
- public:
-  explicit WebRtcSoundclipMedia(WebRtcVoiceEngine *engine)
-      : engine_(engine), webrtc_channel_(-1) {
-    engine_->RegisterSoundclip(this);
-  }
-
-  ~WebRtcSoundclipMedia() override {
-    engine_->UnregisterSoundclip(this);
-    if (webrtc_channel_ != -1) {
-      // We shouldn't have to call Disable() here. DeleteChannel() should call
-      // StopPlayout() while deleting the channel.  We should fix the bug
-      // inside WebRTC and remove the Disable() call bellow.  This work is
-      // tracked by bug http://b/issue?id=5382855.
-      PlaySound(NULL, 0, 0);
-      Disable();
-      if (engine_->voe_sc()->base()->DeleteChannel(webrtc_channel_)
-          == -1) {
-        LOG_RTCERR1(DeleteChannel, webrtc_channel_);
-      }
-    }
-  }
-
-  bool Init() {
-    if (!engine_->voe_sc()) {
-      return false;
-    }
-    webrtc_channel_ = engine_->CreateSoundclipVoiceChannel();
-    if (webrtc_channel_ == -1) {
-      LOG_RTCERR0(CreateChannel);
-      return false;
-    }
-    return true;
-  }
-
-  bool Enable() {
-    if (engine_->voe_sc()->base()->StartPlayout(webrtc_channel_) == -1) {
-      LOG_RTCERR1(StartPlayout, webrtc_channel_);
-      return false;
-    }
-    return true;
-  }
-
-  bool Disable() {
-    if (engine_->voe_sc()->base()->StopPlayout(webrtc_channel_) == -1) {
-      LOG_RTCERR1(StopPlayout, webrtc_channel_);
-      return false;
-    }
-    return true;
-  }
-
-  bool PlaySound(const char* buf, int len, int flags) override {
-    // The voe file api is not available in chrome.
-    if (!engine_->voe_sc()->file()) {
-      return false;
-    }
-    // Must stop playing the current sound (if any), because we are about to
-    // modify the stream.
-    if (engine_->voe_sc()->file()->StopPlayingFileLocally(webrtc_channel_)
-        == -1) {
-      LOG_RTCERR1(StopPlayingFileLocally, webrtc_channel_);
-      return false;
-    }
-
-    if (buf) {
-      stream_.reset(new WebRtcSoundclipStream(buf, len));
-      stream_->set_loop((flags & SF_LOOP) != 0);
-      stream_->Rewind();
-
-      // Play it.
-      if (engine_->voe_sc()->file()->StartPlayingFileLocally(
-          webrtc_channel_, stream_.get()) == -1) {
-        LOG_RTCERR2(StartPlayingFileLocally, webrtc_channel_, stream_.get());
-        LOG(LS_ERROR) << "Unable to start soundclip";
-        return false;
-      }
-    } else {
-      stream_.reset();
-    }
-    return true;
-  }
-
-  int GetLastEngineError() const { return engine_->voe_sc()->error(); }
-
- private:
-  WebRtcVoiceEngine *engine_;
-  int webrtc_channel_;
-  rtc::scoped_ptr<WebRtcSoundclipStream> stream_;
-};
-
 WebRtcVoiceEngine::WebRtcVoiceEngine()
     : voe_wrapper_(new VoEWrapper()),
-      voe_wrapper_sc_(new VoEWrapper()),
-      voe_wrapper_sc_initialized_(false),
       tracing_(new VoETraceWrapper()),
       adm_(NULL),
-      adm_sc_(NULL),
       log_filter_(SeverityToFilter(kDefaultLogSeverity)),
       is_dumping_aec_(false),
       desired_local_monitor_enable_(false),
@@ -474,14 +381,10 @@ WebRtcVoiceEngine::WebRtcVoiceEngine()
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
-                                     VoEWrapper* voe_wrapper_sc,
                                      VoETraceWrapper* tracing)
     : voe_wrapper_(voe_wrapper),
-      voe_wrapper_sc_(voe_wrapper_sc),
-      voe_wrapper_sc_initialized_(false),
       tracing_(tracing),
       adm_(NULL),
-      adm_sc_(NULL),
       log_filter_(SeverityToFilter(kDefaultLogSeverity)),
       is_dumping_aec_(false),
       desired_local_monitor_enable_(false),
@@ -593,11 +496,6 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() {
     adm_->Release();
     adm_ = NULL;
   }
-  if (adm_sc_) {
-    voe_wrapper_sc_.reset();
-    adm_sc_->Release();
-    adm_sc_ = NULL;
-  }
 
   // Test to see if the media processor was deregistered properly
   DCHECK(SignalRxMediaFrame.is_empty());
@@ -673,61 +571,12 @@ bool WebRtcVoiceEngine::InitInternal() {
   return true;
 }
 
-bool WebRtcVoiceEngine::EnsureSoundclipEngineInit() {
-  if (voe_wrapper_sc_initialized_) {
-    return true;
-  }
-  // Note that, if initialization fails, voe_wrapper_sc_initialized_ will still
-  // be false, so subsequent calls to EnsureSoundclipEngineInit will
-  // probably just fail again. That's acceptable behavior.
-#if defined(LINUX) && !defined(HAVE_LIBPULSE)
-  voe_wrapper_sc_->hw()->SetAudioDeviceLayer(webrtc::kAudioLinuxAlsa);
-#endif
-
-  // Initialize the VoiceEngine instance that we'll use to play out sound clips.
-  if (voe_wrapper_sc_->base()->Init(adm_sc_) == -1) {
-    LOG_RTCERR0_EX(Init, voe_wrapper_sc_->error());
-    return false;
-  }
-
-  // On Windows, tell it to use the default sound (not communication) devices.
-  // First check whether there is a valid sound device for playback.
-  // TODO(juberti): Clean this up when we support setting the soundclip device.
-#ifdef WIN32
-  // The SetPlayoutDevice may not be implemented in the case of external ADM.
-  // TODO(ronghuawu): We should only check the adm_sc_ here, but current
-  // PeerConnection interface never set the adm_sc_, so need to check both
-  // in order to determine if the external adm is used.
-  if (!adm_ && !adm_sc_) {
-    int num_of_devices = 0;
-    if (voe_wrapper_sc_->hw()->GetNumOfPlayoutDevices(num_of_devices) != -1 &&
-        num_of_devices > 0) {
-      if (voe_wrapper_sc_->hw()->SetPlayoutDevice(kDefaultSoundclipDeviceId)
-          == -1) {
-        LOG_RTCERR1_EX(SetPlayoutDevice, kDefaultSoundclipDeviceId,
-                       voe_wrapper_sc_->error());
-        return false;
-      }
-    } else {
-      LOG(LS_WARNING) << "No valid sound playout device found.";
-    }
-  }
-#endif
-  voe_wrapper_sc_initialized_ = true;
-  LOG(LS_INFO) << "Initialized WebRtc soundclip engine.";
-  return true;
-}
-
 void WebRtcVoiceEngine::Terminate() {
   LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
   initialized_ = false;
 
   StopAecDump();
 
-  if (voe_wrapper_sc_) {
-    voe_wrapper_sc_initialized_ = false;
-    voe_wrapper_sc_->base()->Terminate();
-  }
   voe_wrapper_->base()->Terminate();
   desired_local_monitor_enable_ = false;
 }
@@ -736,27 +585,17 @@ int WebRtcVoiceEngine::GetCapabilities() {
   return AUDIO_SEND | AUDIO_RECV;
 }
 
-VoiceMediaChannel *WebRtcVoiceEngine::CreateChannel() {
+VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
+    const AudioOptions& options) {
   WebRtcVoiceMediaChannel* ch = new WebRtcVoiceMediaChannel(this);
   if (!ch->valid()) {
     delete ch;
-    ch = NULL;
-  }
-  return ch;
-}
-
-SoundclipMedia *WebRtcVoiceEngine::CreateSoundclip() {
-  if (!EnsureSoundclipEngineInit()) {
-    LOG(LS_ERROR) << "Unable to create soundclip: soundclip engine failed to "
-                  << "initialize.";
-    return NULL;
+    return nullptr;
   }
-  WebRtcSoundclipMedia *soundclip = new WebRtcSoundclipMedia(this);
-  if (!soundclip->Init() || !soundclip->Enable()) {
-    delete soundclip;
-    return NULL;
+  if (!ch->SetOptions(options)) {
+    LOG(LS_WARNING) << "Failed to set options while creating channel.";
   }
-  return soundclip;
+  return ch;
 }
 
 bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
@@ -822,7 +661,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
   agc_mode = webrtc::kAgcFixedDigital;
   options.typing_detection.Set(false);
   options.experimental_agc.Set(false);
-  options.experimental_aec.Set(false);
+  options.extended_filter_aec.Set(false);
   options.experimental_ns.Set(false);
 #endif
 
@@ -833,7 +672,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
   if (options.delay_agnostic_aec.Get(&use_delay_agnostic_aec)) {
     if (use_delay_agnostic_aec) {
       options.echo_cancellation.Set(true);
-      options.experimental_aec.Set(true);
+      options.extended_filter_aec.Set(true);
       ec_mode = webrtc::kEcConference;
     }
   }
@@ -850,12 +689,14 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     // 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();
-    if (built_in_aec && !use_delay_agnostic_aec) {
+    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.
-      if (voe_wrapper_->hw()->EnableBuiltInAEC(echo_cancellation) == 0 &&
-          echo_cancellation) {
+      const bool enable_built_in_aec =
+          echo_cancellation && !use_delay_agnostic_aec;
+      if (voe_wrapper_->hw()->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.
         options.echo_cancellation.Set(false);
@@ -963,6 +804,14 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
         new webrtc::NetEqCapacityConfig(audio_jitter_buffer_max_packets));
   }
 
+  bool audio_jitter_buffer_fast_accelerate;
+  if (options.audio_jitter_buffer_fast_accelerate.Get(
+          &audio_jitter_buffer_fast_accelerate)) {
+    LOG(LS_INFO) << "NetEq fast mode? " << audio_jitter_buffer_fast_accelerate;
+    voe_config_.Set<webrtc::NetEqFastAccelerate>(
+        new webrtc::NetEqFastAccelerate(audio_jitter_buffer_fast_accelerate));
+  }
+
   bool typing_detection;
   if (options.typing_detection.Get(&typing_detection)) {
     LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection;
@@ -995,16 +844,16 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
   bool delay_agnostic_aec;
   if (delay_agnostic_aec_.Get(&delay_agnostic_aec)) {
     LOG(LS_INFO) << "Delay agnostic aec is enabled? " << delay_agnostic_aec;
-    config.Set<webrtc::ReportedDelay>(
-        new webrtc::ReportedDelay(!delay_agnostic_aec));
+    config.Set<webrtc::DelayAgnostic>(
+        new webrtc::DelayAgnostic(delay_agnostic_aec));
   }
 
-  experimental_aec_.SetFrom(options.experimental_aec);
-  bool experimental_aec;
-  if (experimental_aec_.Get(&experimental_aec)) {
-    LOG(LS_INFO) << "Experimental aec is enabled? " << experimental_aec;
-    config.Set<webrtc::DelayCorrection>(
-        new webrtc::DelayCorrection(experimental_aec));
+  extended_filter_aec_.SetFrom(options.extended_filter_aec);
+  bool extended_filter;
+  if (extended_filter_aec_.Get(&extended_filter)) {
+    LOG(LS_INFO) << "Extended filter aec is enabled? " << extended_filter;
+    config.Set<webrtc::ExtendedFilter>(
+        new webrtc::ExtendedFilter(extended_filter));
   }
 
   experimental_ns_.SetFrom(options.experimental_ns);
@@ -1532,19 +1381,6 @@ void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel *channel) {
   }
 }
 
-void WebRtcVoiceEngine::RegisterSoundclip(WebRtcSoundclipMedia *soundclip) {
-  soundclips_.push_back(soundclip);
-}
-
-void WebRtcVoiceEngine::UnregisterSoundclip(WebRtcSoundclipMedia *soundclip) {
-  SoundclipList::iterator i = std::find(soundclips_.begin(),
-                                        soundclips_.end(),
-                                        soundclip);
-  if (i != soundclips_.end()) {
-    soundclips_.erase(i);
-  }
-}
-
 // Adjusts the default AGC target level by the specified delta.
 // NB: If we start messing with other config fields, we'll want
 // to save the current webrtc::AgcConfig as well.
@@ -1563,8 +1399,7 @@ bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) {
   return true;
 }
 
-bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc) {
+bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
   if (initialized_) {
     LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
     return false;
@@ -1577,15 +1412,6 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
     adm_ = adm;
     adm_->AddRef();
   }
-
-  if (adm_sc_) {
-    adm_sc_->Release();
-    adm_sc_ = NULL;
-  }
-  if (adm_sc) {
-    adm_sc_ = adm_sc;
-    adm_sc_->AddRef();
-  }
   return true;
 }
 
@@ -1791,10 +1617,6 @@ int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
   return CreateVoiceChannel(voe_wrapper_.get());
 }
 
-int WebRtcVoiceEngine::CreateSoundclipVoiceChannel() {
-  return CreateVoiceChannel(voe_wrapper_sc_.get());
-}
-
 class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
     : public AudioRenderer::Sink {
  public:
@@ -3525,6 +3347,10 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
             static_cast<float>(ns.currentSpeechExpandRate) / (1 << 14);
         rinfo.secondary_decoded_rate =
             static_cast<float>(ns.currentSecondaryDecodedRate) / (1 << 14);
+        rinfo.accelerate_rate =
+            static_cast<float>(ns.currentAccelerateRate) / (1 << 14);
+        rinfo.preemptive_expand_rate =
+            static_cast<float>(ns.currentPreemptiveRate) / (1 << 14);
       }
 
       webrtc::AudioDecodingCallStats ds;
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h
index 242467dd990..35f2dbc93ef 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h
@@ -35,7 +35,6 @@
 
 #include "talk/media/base/rtputils.h"
 #include "talk/media/webrtc/webrtccommon.h"
-#include "talk/media/webrtc/webrtcexport.h"
 #include "talk/media/webrtc/webrtcvoe.h"
 #include "talk/session/media/channel.h"
 #include "webrtc/base/buffer.h"
@@ -48,15 +47,6 @@
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 
-#if !defined(LIBPEERCONNECTION_LIB) && \
-    !defined(LIBPEERCONNECTION_IMPLEMENTATION)
-// If you hit this, then you've tried to include this header from outside
-// the shared library.  An instance of this class must only be created from
-// within the library that actually implements it.  Otherwise use the
-// WebRtcMediaEngine to construct an instance.
-#error "Bogus include."
-#endif
-
 namespace webrtc {
 class VideoEngine;
 }
@@ -91,7 +81,6 @@ class AudioRenderer;
 class VoETraceWrapper;
 class VoEWrapper;
 class VoiceProcessor;
-class WebRtcSoundclipMedia;
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
@@ -100,36 +89,21 @@ class WebRtcVoiceEngine
     : public webrtc::VoiceEngineObserver,
       public webrtc::TraceCallback,
       public webrtc::VoEMediaProcess  {
+  friend class WebRtcVoiceMediaChannel;
+
  public:
   WebRtcVoiceEngine();
   // Dependency injection for testing.
-  WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
-                    VoEWrapper* voe_wrapper_sc,
-                    VoETraceWrapper* tracing);
+  WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
   ~WebRtcVoiceEngine();
   bool Init(rtc::Thread* worker_thread);
   void Terminate();
 
   int GetCapabilities();
-  VoiceMediaChannel* CreateChannel();
-
-  SoundclipMedia* CreateSoundclip();
+  VoiceMediaChannel* CreateChannel(const AudioOptions& options);
 
   AudioOptions GetOptions() const { return options_; }
   bool SetOptions(const AudioOptions& options);
-  // Overrides, when set, take precedence over the options on a
-  // per-option basis.  For example, if AGC is set in options and AEC
-  // is set in overrides, AGC and AEC will be both be set.  Overrides
-  // can also turn off options.  For example, if AGC is set to "on" in
-  // options and AGC is set to "off" in overrides, the result is that
-  // AGC will be off until different overrides are applied or until
-  // the overrides are cleared.  Only one set of overrides is present
-  // at a time (they do not "stack").  And when the overrides are
-  // cleared, the media engine's state reverts back to the options set
-  // via SetOptions.  This allows us to have both "persistent options"
-  // (the normal options) and "temporary options" (overrides).
-  bool SetOptionOverrides(const AudioOptions& options);
-  bool ClearOptionOverrides();
   bool SetDelayOffset(int offset);
   bool SetDevices(const Device* in_device, const Device* out_device);
   bool GetOutputVolume(int* level);
@@ -166,21 +140,15 @@ class WebRtcVoiceEngine
   void RegisterChannel(WebRtcVoiceMediaChannel *channel);
   void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
 
-  // May only be called by WebRtcSoundclipMedia.
-  void RegisterSoundclip(WebRtcSoundclipMedia *channel);
-  void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
-
   // Called by WebRtcVoiceMediaChannel to set a gain offset from
   // the default AGC target level.
   bool AdjustAgcLevel(int delta);
 
   VoEWrapper* voe() { return voe_wrapper_.get(); }
-  VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
   int GetLastEngineError();
 
-  // Set the external ADMs. This can only be called before Init.
-  bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
-                            webrtc::AudioDeviceModule* adm_sc);
+  // Set the external ADM. This can only be called before Init.
+  bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
 
   // Starts AEC dump using existing file.
   bool StartAecDump(rtc::PlatformFile file);
@@ -190,10 +158,8 @@ class WebRtcVoiceEngine
 
   // Create a VoiceEngine Channel.
   int CreateMediaVoiceChannel();
-  int CreateSoundclipVoiceChannel();
 
  private:
-  typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
   typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
   typedef sigslot::
       signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
@@ -202,7 +168,6 @@ class WebRtcVoiceEngine
   void ConstructCodecs();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
-  bool EnsureSoundclipEngineInit();
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
   // Applies either options or overrides.  Every option that is "set"
@@ -210,6 +175,19 @@ class WebRtcVoiceEngine
   // allows us to selectively turn on and off different options easily
   // at any time.
   bool ApplyOptions(const AudioOptions& options);
+  // Overrides, when set, take precedence over the options on a
+  // per-option basis.  For example, if AGC is set in options and AEC
+  // is set in overrides, AGC and AEC will be both be set.  Overrides
+  // can also turn off options.  For example, if AGC is set to "on" in
+  // options and AGC is set to "off" in overrides, the result is that
+  // AGC will be off until different overrides are applied or until
+  // the overrides are cleared.  Only one set of overrides is present
+  // at a time (they do not "stack").  And when the overrides are
+  // cleared, the media engine's state reverts back to the options set
+  // via SetOptions.  This allows us to have both "persistent options"
+  // (the normal options) and "temporary options" (overrides).
+  bool SetOptionOverrides(const AudioOptions& options);
+  bool ClearOptionOverrides();
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
@@ -250,13 +228,9 @@ class WebRtcVoiceEngine
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
-  // A secondary instance, for playing out soundclips (on the 'ring' device).
-  rtc::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
-  bool voe_wrapper_sc_initialized_;
   rtc::scoped_ptr<VoETraceWrapper> tracing_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_;
-  webrtc::AudioDeviceModule* adm_sc_;
   int log_filter_;
   std::string log_options_;
   bool is_dumping_aec_;
@@ -264,7 +238,6 @@ class WebRtcVoiceEngine
   std::vector<RtpHeaderExtension> rtp_header_extensions_;
   bool desired_local_monitor_enable_;
   rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
-  SoundclipList soundclips_;
   ChannelList channels_;
   // channels_ can be read from WebRtc callback thread. We need a lock on that
   // callback as well as the RegisterChannel/UnregisterChannel.
@@ -293,11 +266,11 @@ class WebRtcVoiceEngine
 
   rtc::CriticalSection signal_media_critical_;
 
-  // Cache received experimental_aec, delay_agnostic_aec and experimental_ns
+  // 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.
-  Settable<bool> experimental_aec_;
+  Settable<bool> extended_filter_aec_;
   Settable<bool> delay_agnostic_aec_;
   Settable<bool> experimental_ns_;
 };
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index bc14770ea40..0dc7b510eea 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -128,12 +128,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 
   WebRtcVoiceEngineTestFake()
       : voe_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
-        voe_sc_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
         trace_wrapper_(new FakeVoETraceWrapper()),
-        engine_(new FakeVoEWrapper(&voe_),
-                new FakeVoEWrapper(&voe_sc_),
-                trace_wrapper_),
-        channel_(NULL), soundclip_(NULL) {
+        engine_(new FakeVoEWrapper(&voe_), trace_wrapper_),
+        channel_(nullptr) {
     options_conference_.conference_mode.Set(true);
     options_adjust_agc_.adjust_agc_delta.Set(-10);
   }
@@ -141,8 +138,8 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     if (!engine_.Init(rtc::Thread::Current())) {
       return false;
     }
-    channel_ = engine_.CreateChannel();
-    return (channel_ != NULL);
+    channel_ = engine_.CreateChannel(cricket::AudioOptions());
+    return (channel_ != nullptr);
   }
   bool SetupEngine() {
     if (!SetupEngineWithoutStream()) {
@@ -168,15 +165,14 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     channel_->OnPacketReceived(&packet, rtc::PacketTime());
   }
   void TearDown() override {
-    delete soundclip_;
     delete channel_;
     engine_.Terminate();
   }
 
   void TestInsertDtmf(uint32 ssrc, bool caller) {
     EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-    channel_ = engine_.CreateChannel();
-    EXPECT_TRUE(channel_ != NULL);
+    channel_ = engine_.CreateChannel(cricket::AudioOptions());
+    EXPECT_TRUE(channel_ != nullptr);
     if (caller) {
       // if this is a caller, local description will be applied and add the
       // send stream.
@@ -335,11 +331,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 
  protected:
   cricket::FakeWebRtcVoiceEngine voe_;
-  cricket::FakeWebRtcVoiceEngine voe_sc_;
   FakeVoETraceWrapper* trace_wrapper_;
   cricket::WebRtcVoiceEngine engine_;
   cricket::VoiceMediaChannel* channel_;
-  cricket::SoundclipMedia* soundclip_;
 
   cricket::AudioOptions options_conference_;
   cricket::AudioOptions options_adjust_agc_;
@@ -348,29 +342,25 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 // Tests that our stub library "works".
 TEST_F(WebRtcVoiceEngineTestFake, StartupShutdown) {
   EXPECT_FALSE(voe_.IsInited());
-  EXPECT_FALSE(voe_sc_.IsInited());
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
   EXPECT_TRUE(voe_.IsInited());
-  // The soundclip engine is lazily initialized.
-  EXPECT_FALSE(voe_sc_.IsInited());
   engine_.Terminate();
   EXPECT_FALSE(voe_.IsInited());
-  EXPECT_FALSE(voe_sc_.IsInited());
 }
 
 // Tests that we can create and destroy a channel.
 TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) {
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  channel_ = engine_.CreateChannel();
-  EXPECT_TRUE(channel_ != NULL);
+  channel_ = engine_.CreateChannel(cricket::AudioOptions());
+  EXPECT_TRUE(channel_ != nullptr);
 }
 
 // Tests that we properly handle failures in CreateChannel.
 TEST_F(WebRtcVoiceEngineTestFake, CreateChannelFail) {
   voe_.set_fail_create_channel(true);
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  channel_ = engine_.CreateChannel();
-  EXPECT_TRUE(channel_ == NULL);
+  channel_ = engine_.CreateChannel(cricket::AudioOptions());
+  EXPECT_TRUE(channel_ == nullptr);
 }
 
 // Tests that the list of supported codecs is created properly and ordered
@@ -678,8 +668,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
 
 TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  channel_ = engine_.CreateChannel();
-  EXPECT_TRUE(channel_ != NULL);
+  channel_ = engine_.CreateChannel(cricket::AudioOptions());
+  EXPECT_TRUE(channel_ != nullptr);
   EXPECT_TRUE(channel_->SetSendCodecs(engine_.codecs()));
 
   int desired_bitrate = 128000;
@@ -1052,8 +1042,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCaller) {
 // Test that we can enable NACK with opus as callee.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  channel_ = engine_.CreateChannel();
-  EXPECT_TRUE(channel_ != NULL);
+  channel_ = engine_.CreateChannel(cricket::AudioOptions());
+  EXPECT_TRUE(channel_ != nullptr);
 
   int channel_num = voe_.GetLastChannel();
   std::vector<cricket::AudioCodec> codecs;
@@ -1632,8 +1622,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
 // Test that we set VAD and DTMF types correctly as callee.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  channel_ = engine_.CreateChannel();
-  EXPECT_TRUE(channel_ != NULL);
+  channel_ = engine_.CreateChannel(cricket::AudioOptions());
+  EXPECT_TRUE(channel_ != nullptr);
 
   int channel_num = voe_.GetLastChannel();
   std::vector<cricket::AudioCodec> codecs;
@@ -1749,8 +1739,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCaller) {
 // Test that we set up RED correctly as callee.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCallee) {
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  channel_ = engine_.CreateChannel();
-  EXPECT_TRUE(channel_ != NULL);
+  channel_ = engine_.CreateChannel(cricket::AudioOptions());
+  EXPECT_TRUE(channel_ != nullptr);
 
   int channel_num = voe_.GetLastChannel();
   std::vector<cricket::AudioCodec> codecs;
@@ -2076,6 +2066,12 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
       (1 << 14), info.receivers[0].speech_expand_rate);
   EXPECT_EQ(static_cast<float>(cricket::kNetStats.currentSecondaryDecodedRate) /
       (1 << 14), info.receivers[0].secondary_decoded_rate);
+  EXPECT_EQ(
+      static_cast<float>(cricket::kNetStats.currentAccelerateRate) / (1 << 14),
+      info.receivers[0].accelerate_rate);
+  EXPECT_EQ(
+      static_cast<float>(cricket::kNetStats.currentPreemptiveRate) / (1 << 14),
+      info.receivers[0].preemptive_expand_rate);
 }
 
 // Test that we can add and remove receive streams, and do proper send/playout.
@@ -2432,7 +2428,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) {
 // receive channel is created before the send channel.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  channel_ = engine_.CreateChannel();
+  channel_ = engine_.CreateChannel(cricket::AudioOptions());
   EXPECT_TRUE(channel_->SetOptions(options_conference_));
 
   EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
@@ -2652,35 +2648,6 @@ TEST_F(WebRtcVoiceEngineTestFake, PlayRingbackWithMultipleStreams) {
   EXPECT_EQ(0, voe_.IsPlayingFileLocally(channel_num));
 }
 
-// Tests creating soundclips, and make sure they come from the right engine.
-TEST_F(WebRtcVoiceEngineTestFake, CreateSoundclip) {
-  EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  EXPECT_FALSE(voe_sc_.IsInited());
-  soundclip_ = engine_.CreateSoundclip();
-  EXPECT_TRUE(voe_sc_.IsInited());
-  ASSERT_TRUE(soundclip_ != NULL);
-  EXPECT_EQ(0, voe_.GetNumChannels());
-  EXPECT_EQ(1, voe_sc_.GetNumChannels());
-  int channel_num = voe_sc_.GetLastChannel();
-  EXPECT_TRUE(voe_sc_.GetPlayout(channel_num));
-  delete soundclip_;
-  soundclip_ = NULL;
-  EXPECT_EQ(0, voe_sc_.GetNumChannels());
-  // Make sure the soundclip engine is uninitialized on shutdown, now that
-  // we've initialized it by creating a soundclip.
-  engine_.Terminate();
-  EXPECT_FALSE(voe_sc_.IsInited());
-}
-
-// Tests playing out a fake sound.
-TEST_F(WebRtcVoiceEngineTestFake, PlaySoundclip) {
-  static const char kZeroes[16000] = {};
-  EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  soundclip_ = engine_.CreateSoundclip();
-  ASSERT_TRUE(soundclip_ != NULL);
-  EXPECT_TRUE(soundclip_->PlaySound(kZeroes, sizeof(kZeroes), 0));
-}
-
 TEST_F(WebRtcVoiceEngineTestFake, MediaEngineCallbackOnError) {
   rtc::scoped_ptr<ChannelErrorListener> listener;
   cricket::WebRtcVoiceMediaChannel* media_channel;
@@ -2882,6 +2849,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
   EXPECT_EQ(ec_mode, webrtc::kEcConference);
   EXPECT_EQ(ns_mode, webrtc::kNsHighSuppression);
   EXPECT_EQ(50, voe_.GetNetEqCapacity());  // From GetDefaultEngineOptions().
+  EXPECT_FALSE(
+      voe_.GetNetEqFastAccelerate());  // From GetDefaultEngineOptions().
 
   // Turn echo cancellation off
   options.echo_cancellation.Set(false);
@@ -2926,7 +2895,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
 
   // Turn off echo cancellation and delay agnostic aec.
   options.delay_agnostic_aec.Set(false);
-  options.experimental_aec.Set(false);
+  options.extended_filter_aec.Set(false);
   options.echo_cancellation.Set(false);
   ASSERT_TRUE(engine_.SetOptions(options));
   voe_.GetEcStatus(ec_enabled, ec_mode);
@@ -3029,9 +2998,9 @@ TEST_F(WebRtcVoiceEngineTestFake, InitDoesNotOverwriteDefaultAgcConfig) {
 TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
   EXPECT_TRUE(SetupEngine());
   rtc::scoped_ptr<cricket::VoiceMediaChannel> channel1(
-      engine_.CreateChannel());
+      engine_.CreateChannel(cricket::AudioOptions()));
   rtc::scoped_ptr<cricket::VoiceMediaChannel> channel2(
-      engine_.CreateChannel());
+      engine_.CreateChannel(cricket::AudioOptions()));
 
   // Have to add a stream to make SetSend work.
   cricket::StreamParams stream1;
@@ -3149,7 +3118,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
 TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
   EXPECT_TRUE(SetupEngine());
   rtc::scoped_ptr<cricket::VoiceMediaChannel> channel(
-      engine_.CreateChannel());
+      engine_.CreateChannel(cricket::AudioOptions()));
   rtc::scoped_ptr<cricket::FakeNetworkInterface> network_interface(
       new cricket::FakeNetworkInterface);
   channel->SetInterface(network_interface.get());
@@ -3231,8 +3200,9 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOutputScaling) {
 TEST(WebRtcVoiceEngineTest, StartupShutdown) {
   cricket::WebRtcVoiceEngine engine;
   EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
-  cricket::VoiceMediaChannel* channel = engine.CreateChannel();
-  EXPECT_TRUE(channel != NULL);
+  cricket::VoiceMediaChannel* channel =
+      engine.CreateChannel(cricket::AudioOptions());
+  EXPECT_TRUE(channel != nullptr);
   delete channel;
   engine.Terminate();
 
@@ -3241,30 +3211,6 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
   engine.Terminate();
 }
 
-// Tests that the logging from the library is cleartext.
-TEST(WebRtcVoiceEngineTest, DISABLED_HasUnencryptedLogging) {
-  cricket::WebRtcVoiceEngine engine;
-  rtc::scoped_ptr<rtc::MemoryStream> stream(
-      new rtc::MemoryStream);
-  size_t size = 0;
-  bool cleartext = true;
-  rtc::LogMessage::AddLogToStream(stream.get(), rtc::LS_VERBOSE);
-  engine.SetLogging(rtc::LS_VERBOSE, "");
-  EXPECT_TRUE(engine.Init(rtc::Thread::Current()));
-  EXPECT_TRUE(stream->GetSize(&size));
-  EXPECT_GT(size, 0U);
-  engine.Terminate();
-  rtc::LogMessage::RemoveLogToStream(stream.get());
-  const char* buf = stream->GetBuffer();
-  for (size_t i = 0; i < size && cleartext; ++i) {
-    int ch = static_cast<int>(buf[i]);
-    ASSERT_GE(ch, 0) << "Out of bounds character in WebRtc VoE log: "
-                     << std::hex << ch;
-    cleartext = (isprint(ch) || isspace(ch));
-  }
-  EXPECT_TRUE(cleartext);
-}
-
 // Tests that the library is configured with the codecs we want.
 TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
   cricket::WebRtcVoiceEngine engine;
@@ -3354,7 +3300,8 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
   int num_channels = 0;
 
   while (num_channels < ARRAY_SIZE(channels)) {
-    cricket::VoiceMediaChannel* channel = engine.CreateChannel();
+    cricket::VoiceMediaChannel* channel =
+        engine.CreateChannel(cricket::AudioOptions());
     if (!channel)
       break;
 
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channel.cc b/chromium/third_party/libjingle/source/talk/session/media/channel.cc
index 0034f15d9b1..d30972db06f 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channel.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channel.cc
@@ -204,11 +204,6 @@ bool BaseChannel::Init() {
     return false;
   }
 
-  session_->SignalNewLocalDescription.connect(
-      this, &BaseChannel::OnNewLocalDescription);
-  session_->SignalNewRemoteDescription.connect(
-      this, &BaseChannel::OnNewRemoteDescription);
-
   // Both RTP and RTCP channels are set, we can call SetInterface on
   // media channel and it can set network options.
   media_channel_->SetInterface(this);
@@ -502,12 +497,6 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
     return false;
   }
 
-  // Signal to the media sink before protecting the packet.
-  {
-    rtc::CritScope cs(&signal_send_packet_cs_);
-    SignalSendPacketPreCrypto(packet->data(), packet->size(), rtcp);
-  }
-
   rtc::PacketOptions options(dscp);
   // Protect if needed.
   if (srtp_filter_.IsActive()) {
@@ -576,12 +565,6 @@ bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
     return false;
   }
 
-  // Signal to the media sink after protecting the packet.
-  {
-    rtc::CritScope cs(&signal_send_packet_cs_);
-    SignalSendPacketPostCrypto(packet->data(), packet->size(), rtcp);
-  }
-
   // Bon voyage.
   int ret =
       channel->SendPacket(packet->data<char>(), packet->size(), options,
@@ -622,12 +605,6 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
     signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
   }
 
-  // Signal to the media sink before unprotecting the packet.
-  {
-    rtc::CritScope cs(&signal_recv_packet_cs_);
-    SignalRecvPacketPostCrypto(packet->data(), packet->size(), rtcp);
-  }
-
   // Unprotect the packet, if needed.
   if (srtp_filter_.IsActive()) {
     char* data = packet->data<char>();
@@ -673,12 +650,6 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
     return;
   }
 
-  // Signal to the media sink after unprotecting the packet.
-  {
-    rtc::CritScope cs(&signal_recv_packet_cs_);
-    SignalRecvPacketPreCrypto(packet->data(), packet->size(), rtcp);
-  }
-
   // Push it down to the media channel.
   if (!rtcp) {
     media_channel_->OnPacketReceived(packet, packet_time);
@@ -687,24 +658,6 @@ void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
   }
 }
 
-void BaseChannel::OnNewLocalDescription(
-    BaseSession* session, ContentAction action) {
-  std::string error_desc;
-  if (!PushdownLocalDescription(
-          session->local_description(), action, &error_desc))  {
-    SetSessionError(session_, BaseSession::ERROR_CONTENT, error_desc);
-  }
-}
-
-void BaseChannel::OnNewRemoteDescription(
-    BaseSession* session, ContentAction action) {
-  std::string error_desc;
-  if (!PushdownRemoteDescription(
-          session->remote_description(), action, &error_desc))  {
-    SetSessionError(session_, BaseSession::ERROR_CONTENT, error_desc);
-  }
-}
-
 bool BaseChannel::PushdownLocalDescription(
     const SessionDescription* local_desc, ContentAction action,
     std::string* error_desc) {
@@ -1042,6 +995,18 @@ bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
   return true;
 }
 
+void BaseChannel::ActivateRtcpMux() {
+  worker_thread_->Invoke<void>(Bind(
+      &BaseChannel::ActivateRtcpMux_w, this));
+}
+
+void BaseChannel::ActivateRtcpMux_w() {
+  if (!rtcp_mux_filter_.IsActive()) {
+    rtcp_mux_filter_.SetActive();
+    set_rtcp_transport_channel(NULL);
+  }
+}
+
 bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
                                ContentSource src,
                                std::string* error_desc) {
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channel.h b/chromium/third_party/libjingle/source/talk/session/media/channel.h
index 441fe64b684..bb2dffdc905 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channel.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/channel.h
@@ -108,6 +108,11 @@ class BaseChannel
   bool writable() const { return writable_; }
   bool IsStreamMuted(uint32 ssrc);
 
+  // Activate RTCP mux, regardless of the state so far.  Once
+  // activated, it can not be deactivated, and if the remote
+  // description doesn't support RTCP mux, setting the remote
+  // description will fail.
+  void ActivateRtcpMux();
   bool PushdownLocalDescription(const SessionDescription* local_desc,
                                 ContentAction action,
                                 std::string* error_desc);
@@ -143,72 +148,6 @@ class BaseChannel
     srtp_filter_.set_signal_silent_time(silent_time);
   }
 
-  template <class T>
-  void RegisterSendSink(T* sink,
-                        void (T::*OnPacket)(const void*, size_t, bool),
-                        SinkType type) {
-    rtc::CritScope cs(&signal_send_packet_cs_);
-    if (SINK_POST_CRYPTO == type) {
-      SignalSendPacketPostCrypto.disconnect(sink);
-      SignalSendPacketPostCrypto.connect(sink, OnPacket);
-    } else {
-      SignalSendPacketPreCrypto.disconnect(sink);
-      SignalSendPacketPreCrypto.connect(sink, OnPacket);
-    }
-  }
-
-  void UnregisterSendSink(sigslot::has_slots<>* sink,
-                          SinkType type) {
-    rtc::CritScope cs(&signal_send_packet_cs_);
-    if (SINK_POST_CRYPTO == type) {
-      SignalSendPacketPostCrypto.disconnect(sink);
-    } else {
-      SignalSendPacketPreCrypto.disconnect(sink);
-    }
-  }
-
-  bool HasSendSinks(SinkType type) {
-    rtc::CritScope cs(&signal_send_packet_cs_);
-    if (SINK_POST_CRYPTO == type) {
-      return !SignalSendPacketPostCrypto.is_empty();
-    } else {
-      return !SignalSendPacketPreCrypto.is_empty();
-    }
-  }
-
-  template <class T>
-  void RegisterRecvSink(T* sink,
-                        void (T::*OnPacket)(const void*, size_t, bool),
-                        SinkType type) {
-    rtc::CritScope cs(&signal_recv_packet_cs_);
-    if (SINK_POST_CRYPTO == type) {
-      SignalRecvPacketPostCrypto.disconnect(sink);
-      SignalRecvPacketPostCrypto.connect(sink, OnPacket);
-    } else {
-      SignalRecvPacketPreCrypto.disconnect(sink);
-      SignalRecvPacketPreCrypto.connect(sink, OnPacket);
-    }
-  }
-
-  void UnregisterRecvSink(sigslot::has_slots<>* sink,
-                          SinkType type) {
-    rtc::CritScope cs(&signal_recv_packet_cs_);
-    if (SINK_POST_CRYPTO == type) {
-      SignalRecvPacketPostCrypto.disconnect(sink);
-    } else {
-      SignalRecvPacketPreCrypto.disconnect(sink);
-    }
-  }
-
-  bool HasRecvSinks(SinkType type) {
-    rtc::CritScope cs(&signal_recv_packet_cs_);
-    if (SINK_POST_CRYPTO == type) {
-      return !SignalRecvPacketPostCrypto.is_empty();
-    } else {
-      return !SignalRecvPacketPreCrypto.is_empty();
-    }
-  }
-
   BundleFilter* bundle_filter() { return &bundle_filter_; }
 
   const std::vector<StreamParams>& local_streams() const {
@@ -350,6 +289,7 @@ class BaseChannel
                  ContentAction action,
                  ContentSource src,
                  std::string* error_desc);
+  void ActivateRtcpMux_w();
   bool SetRtcpMux_w(bool enable,
                     ContentAction action,
                     ContentSource src,
@@ -371,13 +311,6 @@ class BaseChannel
   }
 
  private:
-  sigslot::signal3<const void*, size_t, bool> SignalSendPacketPreCrypto;
-  sigslot::signal3<const void*, size_t, bool> SignalSendPacketPostCrypto;
-  sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPreCrypto;
-  sigslot::signal3<const void*, size_t, bool> SignalRecvPacketPostCrypto;
-  rtc::CriticalSection signal_send_packet_cs_;
-  rtc::CriticalSection signal_recv_packet_cs_;
-
   rtc::Thread* worker_thread_;
   MediaEngineInterface* media_engine_;
   BaseSession* session_;
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc
index a55f6e8cb03..2573454b22f 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc
@@ -35,7 +35,6 @@
 #include "talk/media/base/testutils.h"
 #include "webrtc/p2p/base/fakesession.h"
 #include "talk/session/media/channel.h"
-#include "talk/session/media/mediarecorder.h"
 #include "talk/session/media/typingmonitor.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/gunit.h"
@@ -1140,6 +1139,89 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     EXPECT_TRUE(CheckNoRtcp2());
   }
 
+  // Check that RTP and RTCP are transmitted ok when both sides
+  // support mux and one the offerer requires mux.
+  void SendRequireRtcpMuxToRtcpMux() {
+    CreateChannels(RTCP | RTCP_MUX, RTCP | RTCP_MUX);
+    channel1_->ActivateRtcpMux();
+    EXPECT_TRUE(SendInitiate());
+    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());
+    EXPECT_TRUE(CheckRtp1());
+    EXPECT_TRUE(CheckRtp2());
+    EXPECT_TRUE(CheckNoRtp1());
+    EXPECT_TRUE(CheckNoRtp2());
+    EXPECT_TRUE(CheckRtcp1());
+    EXPECT_TRUE(CheckRtcp2());
+    EXPECT_TRUE(CheckNoRtcp1());
+    EXPECT_TRUE(CheckNoRtcp2());
+  }
+
+  // Check that RTP and RTCP are transmitted ok when both sides
+  // support mux and one the answerer requires rtcp mux.
+  void SendRtcpMuxToRequireRtcpMux() {
+    CreateChannels(RTCP | RTCP_MUX, RTCP | RTCP_MUX);
+    channel2_->ActivateRtcpMux();
+    EXPECT_TRUE(SendInitiate());
+    EXPECT_EQ(2U, GetTransport1()->channels().size());
+    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());
+    EXPECT_TRUE(CheckRtp1());
+    EXPECT_TRUE(CheckRtp2());
+    EXPECT_TRUE(CheckNoRtp1());
+    EXPECT_TRUE(CheckNoRtp2());
+    EXPECT_TRUE(CheckRtcp1());
+    EXPECT_TRUE(CheckRtcp2());
+    EXPECT_TRUE(CheckNoRtcp1());
+    EXPECT_TRUE(CheckNoRtcp2());
+  }
+
+  // Check that RTP and RTCP are transmitted ok when both sides
+  // require mux.
+  void SendRequireRtcpMuxToRequireRtcpMux() {
+    CreateChannels(RTCP | RTCP_MUX, RTCP | RTCP_MUX);
+    channel1_->ActivateRtcpMux();
+    channel2_->ActivateRtcpMux();
+    EXPECT_TRUE(SendInitiate());
+    EXPECT_EQ(1U, GetTransport1()->channels().size());
+    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());
+    EXPECT_TRUE(CheckRtp1());
+    EXPECT_TRUE(CheckRtp2());
+    EXPECT_TRUE(CheckNoRtp1());
+    EXPECT_TRUE(CheckNoRtp2());
+    EXPECT_TRUE(CheckRtcp1());
+    EXPECT_TRUE(CheckRtcp2());
+    EXPECT_TRUE(CheckNoRtcp1());
+    EXPECT_TRUE(CheckNoRtcp2());
+  }
+
+  // Check that SendAccept fails if the answerer doesn't support mux
+  // and the offerer requires it.
+  void SendRequireRtcpMuxToNoRtcpMux() {
+    CreateChannels(RTCP | RTCP_MUX, RTCP);
+    channel1_->ActivateRtcpMux();
+    EXPECT_TRUE(SendInitiate());
+    EXPECT_EQ(1U, GetTransport1()->channels().size());
+    EXPECT_EQ(2U, GetTransport2()->channels().size());
+    EXPECT_FALSE(SendAccept());
+  }
+
   // Check that RTCP data sent by the initiator before the accept is not muxed.
   void SendEarlyRtcpMuxToRtcp() {
     CreateChannels(RTCP | RTCP_MUX, RTCP);
@@ -1475,125 +1557,44 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     channel1_->StopMediaMonitor();
   }
 
-  void TestMediaSinks() {
-    CreateChannels(0, 0);
-    EXPECT_TRUE(SendInitiate());
-    EXPECT_TRUE(SendAccept());
-    EXPECT_FALSE(channel1_->HasSendSinks(cricket::SINK_POST_CRYPTO));
-    EXPECT_FALSE(channel1_->HasRecvSinks(cricket::SINK_POST_CRYPTO));
-    EXPECT_FALSE(channel1_->HasSendSinks(cricket::SINK_PRE_CRYPTO));
-    EXPECT_FALSE(channel1_->HasRecvSinks(cricket::SINK_PRE_CRYPTO));
-
-    rtc::Pathname path;
-    EXPECT_TRUE(rtc::Filesystem::GetTemporaryFolder(path, true, NULL));
-    path.SetFilename("sink-test.rtpdump");
-    rtc::scoped_ptr<cricket::RtpDumpSink> sink(
-        new cricket::RtpDumpSink(Open(path.pathname())));
-    sink->set_packet_filter(cricket::PF_ALL);
-    EXPECT_TRUE(sink->Enable(true));
-    channel1_->RegisterSendSink(
-        sink.get(), &cricket::RtpDumpSink::OnPacket, cricket::SINK_POST_CRYPTO);
-    EXPECT_TRUE(channel1_->HasSendSinks(cricket::SINK_POST_CRYPTO));
-    EXPECT_FALSE(channel1_->HasRecvSinks(cricket::SINK_POST_CRYPTO));
-    EXPECT_FALSE(channel1_->HasSendSinks(cricket::SINK_PRE_CRYPTO));
-    EXPECT_FALSE(channel1_->HasRecvSinks(cricket::SINK_PRE_CRYPTO));
-
-    // The first packet is recorded with header + data.
-    EXPECT_TRUE(SendRtp1());
-    // The second packet is recorded with header only.
-    sink->set_packet_filter(cricket::PF_RTPHEADER);
-    EXPECT_TRUE(SendRtp1());
-    // The third packet is not recorded since sink is disabled.
-    EXPECT_TRUE(sink->Enable(false));
-    EXPECT_TRUE(SendRtp1());
-     // The fourth packet is not recorded since sink is unregistered.
-    EXPECT_TRUE(sink->Enable(true));
-    channel1_->UnregisterSendSink(sink.get(), cricket::SINK_POST_CRYPTO);
-    EXPECT_TRUE(SendRtp1());
-    sink.reset();  // This will close the file.
-
-    // Read the recorded file and verify two packets.
-    rtc::scoped_ptr<rtc::StreamInterface> stream(
-        rtc::Filesystem::OpenFile(path, "rb"));
-
-    cricket::RtpDumpReader reader(stream.get());
-    cricket::RtpDumpPacket packet;
-    EXPECT_EQ(rtc::SR_SUCCESS, reader.ReadPacket(&packet));
-    std::string read_packet(reinterpret_cast<const char*>(&packet.data[0]),
-        packet.data.size());
-    EXPECT_EQ(rtp_packet_, read_packet);
-
-    EXPECT_EQ(rtc::SR_SUCCESS, reader.ReadPacket(&packet));
-    size_t len = 0;
-    packet.GetRtpHeaderLen(&len);
-    EXPECT_EQ(len, packet.data.size());
-    EXPECT_EQ(0, memcmp(&packet.data[0], rtp_packet_.c_str(), len));
-
-    EXPECT_EQ(rtc::SR_EOS, reader.ReadPacket(&packet));
-
-    // Delete the file for media recording.
-    stream.reset();
-    EXPECT_TRUE(rtc::Filesystem::DeleteFile(path));
-  }
-
   void TestSetContentFailure() {
     CreateChannels(0, 0);
-    typename T::Content content;
-    cricket::SessionDescription* sdesc_loc = new cricket::SessionDescription();
-    cricket::SessionDescription* sdesc_rem = new cricket::SessionDescription();
 
-    // Set up the session description.
-    CreateContent(0, kPcmuCodec, kH264Codec, &content);
-    sdesc_loc->AddContent(cricket::CN_AUDIO, cricket::NS_JINGLE_RTP,
-                          new cricket::AudioContentDescription());
-    sdesc_loc->AddContent(cricket::CN_VIDEO, cricket::NS_JINGLE_RTP,
-                          new cricket::VideoContentDescription());
-    session1_.set_local_description(sdesc_loc);
-    sdesc_rem->AddContent(cricket::CN_AUDIO, cricket::NS_JINGLE_RTP,
-                          new cricket::AudioContentDescription());
-    sdesc_rem->AddContent(cricket::CN_VIDEO, cricket::NS_JINGLE_RTP,
-                          new cricket::VideoContentDescription());
-    session1_.set_remote_description(sdesc_rem);
-
-    // Test failures in SetLocalContent.
-    media_channel1_->set_fail_set_recv_codecs(true);
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_SENTINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_CONTENT, session1_.error());
+    auto sdesc = cricket::SessionDescription();
+    sdesc.AddContent(cricket::CN_AUDIO, cricket::NS_JINGLE_RTP,
+                     new cricket::AudioContentDescription());
+    sdesc.AddContent(cricket::CN_VIDEO, cricket::NS_JINGLE_RTP,
+                     new cricket::VideoContentDescription());
+
+    std::string err;
     media_channel1_->set_fail_set_recv_codecs(true);
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_SENTACCEPT);
-    EXPECT_EQ(cricket::BaseSession::ERROR_CONTENT, session1_.error());
+    EXPECT_FALSE(channel1_->PushdownLocalDescription(
+        &sdesc, cricket::CA_OFFER, &err));
+    EXPECT_FALSE(channel1_->PushdownLocalDescription(
+        &sdesc, cricket::CA_ANSWER, &err));
 
-    // Test failures in SetRemoteContent.
     media_channel1_->set_fail_set_send_codecs(true);
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_RECEIVEDINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_CONTENT, session1_.error());
+    EXPECT_FALSE(channel1_->PushdownRemoteDescription(
+        &sdesc, cricket::CA_OFFER, &err));
     media_channel1_->set_fail_set_send_codecs(true);
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_RECEIVEDACCEPT);
-    EXPECT_EQ(cricket::BaseSession::ERROR_CONTENT, session1_.error());
+    EXPECT_FALSE(channel1_->PushdownRemoteDescription(
+        &sdesc, cricket::CA_ANSWER, &err));
   }
 
   void TestSendTwoOffers() {
     CreateChannels(0, 0);
 
-    // Set up the initial session description.
-    cricket::SessionDescription* sdesc = CreateSessionDescriptionWithStream(1);
-    session1_.set_local_description(sdesc);
-
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_SENTINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    std::string err;
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc1(
+        CreateSessionDescriptionWithStream(1));
+    EXPECT_TRUE(channel1_->PushdownLocalDescription(
+        sdesc1.get(), cricket::CA_OFFER, &err));
     EXPECT_TRUE(media_channel1_->HasSendStream(1));
 
-    // Update the local description and set the state again.
-    sdesc = CreateSessionDescriptionWithStream(2);
-    session1_.set_local_description(sdesc);
-
-    session1_.SetState(cricket::BaseSession::STATE_SENTINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc2(
+        CreateSessionDescriptionWithStream(2));
+    EXPECT_TRUE(channel1_->PushdownLocalDescription(
+        sdesc2.get(), cricket::CA_OFFER, &err));
     EXPECT_FALSE(media_channel1_->HasSendStream(1));
     EXPECT_TRUE(media_channel1_->HasSendStream(2));
   }
@@ -1601,19 +1602,17 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
   void TestReceiveTwoOffers() {
     CreateChannels(0, 0);
 
-    // Set up the initial session description.
-    cricket::SessionDescription* sdesc = CreateSessionDescriptionWithStream(1);
-    session1_.set_remote_description(sdesc);
-
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_RECEIVEDINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    std::string err;
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc1(
+        CreateSessionDescriptionWithStream(1));
+    EXPECT_TRUE(channel1_->PushdownRemoteDescription(
+        sdesc1.get(), cricket::CA_OFFER, &err));
     EXPECT_TRUE(media_channel1_->HasRecvStream(1));
 
-    sdesc = CreateSessionDescriptionWithStream(2);
-    session1_.set_remote_description(sdesc);
-    session1_.SetState(cricket::BaseSession::STATE_RECEIVEDINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc2(
+        CreateSessionDescriptionWithStream(2));
+    EXPECT_TRUE(channel1_->PushdownRemoteDescription(
+        sdesc2.get(), cricket::CA_OFFER, &err));
     EXPECT_FALSE(media_channel1_->HasRecvStream(1));
     EXPECT_TRUE(media_channel1_->HasRecvStream(2));
   }
@@ -1621,30 +1620,27 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
   void TestSendPrAnswer() {
     CreateChannels(0, 0);
 
-    // Set up the initial session description.
-    cricket::SessionDescription* sdesc = CreateSessionDescriptionWithStream(1);
-    session1_.set_remote_description(sdesc);
-
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_RECEIVEDINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    std::string err;
+    // Receive offer
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc1(
+        CreateSessionDescriptionWithStream(1));
+    EXPECT_TRUE(channel1_->PushdownRemoteDescription(
+        sdesc1.get(), cricket::CA_OFFER, &err));
     EXPECT_TRUE(media_channel1_->HasRecvStream(1));
 
-    // Send PRANSWER
-    sdesc = CreateSessionDescriptionWithStream(2);
-    session1_.set_local_description(sdesc);
-
-    session1_.SetState(cricket::BaseSession::STATE_SENTPRACCEPT);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    // Send PR answer
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc2(
+        CreateSessionDescriptionWithStream(2));
+    EXPECT_TRUE(channel1_->PushdownLocalDescription(
+        sdesc2.get(), cricket::CA_PRANSWER, &err));
     EXPECT_TRUE(media_channel1_->HasRecvStream(1));
     EXPECT_TRUE(media_channel1_->HasSendStream(2));
 
-    // Send ACCEPT
-    sdesc = CreateSessionDescriptionWithStream(3);
-    session1_.set_local_description(sdesc);
-
-    session1_.SetState(cricket::BaseSession::STATE_SENTACCEPT);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    // Send answer
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc3(
+        CreateSessionDescriptionWithStream(3));
+    EXPECT_TRUE(channel1_->PushdownLocalDescription(
+        sdesc3.get(), cricket::CA_ANSWER, &err));
     EXPECT_TRUE(media_channel1_->HasRecvStream(1));
     EXPECT_FALSE(media_channel1_->HasSendStream(2));
     EXPECT_TRUE(media_channel1_->HasSendStream(3));
@@ -1653,30 +1649,27 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
   void TestReceivePrAnswer() {
     CreateChannels(0, 0);
 
-    // Set up the initial session description.
-    cricket::SessionDescription* sdesc = CreateSessionDescriptionWithStream(1);
-    session1_.set_local_description(sdesc);
-
-    session1_.SetError(cricket::BaseSession::ERROR_NONE, "");
-    session1_.SetState(cricket::BaseSession::STATE_SENTINITIATE);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    std::string err;
+    // Send offer
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc1(
+        CreateSessionDescriptionWithStream(1));
+    EXPECT_TRUE(channel1_->PushdownLocalDescription(
+        sdesc1.get(), cricket::CA_OFFER, &err));
     EXPECT_TRUE(media_channel1_->HasSendStream(1));
 
-    // Receive PRANSWER
-    sdesc = CreateSessionDescriptionWithStream(2);
-    session1_.set_remote_description(sdesc);
-
-    session1_.SetState(cricket::BaseSession::STATE_RECEIVEDPRACCEPT);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    // Receive PR answer
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc2(
+        CreateSessionDescriptionWithStream(2));
+    EXPECT_TRUE(channel1_->PushdownRemoteDescription(
+        sdesc2.get(), cricket::CA_PRANSWER, &err));
     EXPECT_TRUE(media_channel1_->HasSendStream(1));
     EXPECT_TRUE(media_channel1_->HasRecvStream(2));
 
-    // Receive ACCEPT
-    sdesc = CreateSessionDescriptionWithStream(3);
-    session1_.set_remote_description(sdesc);
-
-    session1_.SetState(cricket::BaseSession::STATE_RECEIVEDACCEPT);
-    EXPECT_EQ(cricket::BaseSession::ERROR_NONE, session1_.error());
+    // Receive answer
+    rtc::scoped_ptr<cricket::SessionDescription> sdesc3(
+        CreateSessionDescriptionWithStream(3));
+    EXPECT_TRUE(channel1_->PushdownRemoteDescription(
+        sdesc3.get(), cricket::CA_ANSWER, &err));
     EXPECT_TRUE(media_channel1_->HasSendStream(1));
     EXPECT_FALSE(media_channel1_->HasRecvStream(2));
     EXPECT_TRUE(media_channel1_->HasRecvStream(3));
@@ -2085,6 +2078,22 @@ TEST_F(VoiceChannelTest, SendRtcpMuxToRtcpMux) {
   Base::SendRtcpMuxToRtcpMux();
 }
 
+TEST_F(VoiceChannelTest, SendRequireRtcpMuxToRtcpMux) {
+  Base::SendRequireRtcpMuxToRtcpMux();
+}
+
+TEST_F(VoiceChannelTest, SendRtcpMuxToRequireRtcpMux) {
+  Base::SendRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VoiceChannelTest, SendRequireRtcpMuxToRequireRtcpMux) {
+  Base::SendRequireRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VoiceChannelTest, SendRequireRtcpMuxToNoRtcpMux) {
+  Base::SendRequireRtcpMuxToNoRtcpMux();
+}
+
 TEST_F(VoiceChannelTest, SendEarlyRtcpMuxToRtcp) {
   Base::SendEarlyRtcpMuxToRtcp();
 }
@@ -2217,10 +2226,6 @@ TEST_F(VoiceChannelTest, TestInsertDtmf) {
                               3, 7, 120, cricket::DF_PLAY | cricket::DF_SEND));
 }
 
-TEST_F(VoiceChannelTest, TestMediaSinks) {
-  Base::TestMediaSinks();
-}
-
 TEST_F(VoiceChannelTest, TestSetContentFailure) {
   Base::TestSetContentFailure();
 }
@@ -2507,6 +2512,22 @@ TEST_F(VideoChannelTest, SendRtcpMuxToRtcpMux) {
   Base::SendRtcpMuxToRtcpMux();
 }
 
+TEST_F(VideoChannelTest, SendRequireRtcpMuxToRtcpMux) {
+  Base::SendRequireRtcpMuxToRtcpMux();
+}
+
+TEST_F(VideoChannelTest, SendRtcpMuxToRequireRtcpMux) {
+  Base::SendRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VideoChannelTest, SendRequireRtcpMuxToRequireRtcpMux) {
+  Base::SendRequireRtcpMuxToRequireRtcpMux();
+}
+
+TEST_F(VideoChannelTest, SendRequireRtcpMuxToNoRtcpMux) {
+  Base::SendRequireRtcpMuxToNoRtcpMux();
+}
+
 TEST_F(VideoChannelTest, SendEarlyRtcpMuxToRtcp) {
   Base::SendEarlyRtcpMuxToRtcp();
 }
@@ -2562,10 +2583,6 @@ TEST_F(VideoChannelTest, TestMediaMonitor) {
   Base::TestMediaMonitor();
 }
 
-TEST_F(VideoChannelTest, TestMediaSinks) {
-  Base::TestMediaSinks();
-}
-
 TEST_F(VideoChannelTest, TestSetContentFailure) {
   Base::TestSetContentFailure();
 }
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc
index 5013fc48b3b..52116ddb39c 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc
@@ -41,7 +41,6 @@
 #ifdef HAVE_SCTP
 #include "talk/media/sctp/sctpdataengine.h"
 #endif
-#include "talk/session/media/soundclip.h"
 #include "talk/session/media/srtpfilter.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/common.h"
@@ -311,9 +310,6 @@ void ChannelManager::Terminate_w() {
   while (!voice_channels_.empty()) {
     DestroyVoiceChannel_w(voice_channels_.back(), nullptr);
   }
-  while (!soundclips_.empty()) {
-    DestroySoundclip_w(soundclips_.back());
-  }
   if (!SetCaptureDevice_w(NULL)) {
     LOG(LS_WARNING) << "failed to delete video capturer";
   }
@@ -321,26 +317,32 @@ void ChannelManager::Terminate_w() {
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel(
-    BaseSession* session, const std::string& content_name, bool rtcp) {
+    BaseSession* session,
+    const std::string& content_name,
+    bool rtcp,
+    const AudioOptions& options) {
   return worker_thread_->Invoke<VoiceChannel*>(
-      Bind(&ChannelManager::CreateVoiceChannel_w, this,
-           session, content_name, rtcp));
+      Bind(&ChannelManager::CreateVoiceChannel_w, this, session, content_name,
+           rtcp, options));
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel_w(
-    BaseSession* session, const std::string& content_name, bool rtcp) {
+    BaseSession* session,
+    const std::string& content_name,
+    bool rtcp,
+    const AudioOptions& options) {
   ASSERT(initialized_);
   ASSERT(worker_thread_ == rtc::Thread::Current());
-  VoiceMediaChannel* media_channel = media_engine_->CreateChannel();
-  if (media_channel == NULL)
-    return NULL;
+  VoiceMediaChannel* media_channel = media_engine_->CreateChannel(options);
+  if (!media_channel)
+    return nullptr;
 
   VoiceChannel* voice_channel = new VoiceChannel(
       worker_thread_, media_engine_.get(), media_channel,
       session, content_name, rtcp);
   if (!voice_channel->Init()) {
     delete voice_channel;
-    return NULL;
+    return nullptr;
   }
   voice_channels_.push_back(voice_channel);
   return voice_channel;
@@ -504,45 +506,6 @@ void ChannelManager::DestroyDataChannel_w(DataChannel* data_channel) {
   delete data_channel;
 }
 
-Soundclip* ChannelManager::CreateSoundclip() {
-  return worker_thread_->Invoke<Soundclip*>(
-      Bind(&ChannelManager::CreateSoundclip_w, this));
-}
-
-Soundclip* ChannelManager::CreateSoundclip_w() {
-  ASSERT(initialized_);
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-
-  SoundclipMedia* soundclip_media = media_engine_->CreateSoundclip();
-  if (!soundclip_media) {
-    return NULL;
-  }
-
-  Soundclip* soundclip = new Soundclip(worker_thread_, soundclip_media);
-  soundclips_.push_back(soundclip);
-  return soundclip;
-}
-
-void ChannelManager::DestroySoundclip(Soundclip* soundclip) {
-  if (soundclip) {
-    worker_thread_->Invoke<void>(
-        Bind(&ChannelManager::DestroySoundclip_w, this, soundclip));
-  }
-}
-
-void ChannelManager::DestroySoundclip_w(Soundclip* soundclip) {
-  // Destroy soundclip.
-  ASSERT(initialized_);
-  Soundclips::iterator it = std::find(soundclips_.begin(),
-      soundclips_.end(), soundclip);
-  ASSERT(it != soundclips_.end());
-  if (it == soundclips_.end())
-    return;
-
-  soundclips_.erase(it);
-  delete soundclip;
-}
-
 bool ChannelManager::GetAudioOptions(std::string* in_name,
                                      std::string* out_name,
                                      AudioOptions* options) {
@@ -615,29 +578,6 @@ bool ChannelManager::SetAudioOptions_w(
   return ret;
 }
 
-// Sets Engine-specific audio options according to enabled experiments.
-bool ChannelManager::SetEngineAudioOptions(const AudioOptions& options) {
-  // If we're initialized, pass the settings to the media engine.
-  bool ret = false;
-  if (initialized_) {
-    ret = worker_thread_->Invoke<bool>(
-        Bind(&ChannelManager::SetEngineAudioOptions_w, this, options));
-  }
-
-  // If all worked well, save the audio options.
-  if (ret) {
-    audio_options_ = options;
-  }
-  return ret;
-}
-
-bool ChannelManager::SetEngineAudioOptions_w(const AudioOptions& options) {
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-  ASSERT(initialized_);
-
-  return media_engine_->SetAudioOptions(options);
-}
-
 bool ChannelManager::GetOutputVolume(int* level) {
   if (!initialized_) {
     return false;
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h
index 27f874b59d6..898ae1cb2db 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h
@@ -44,7 +44,6 @@ namespace cricket {
 
 const int kDefaultAudioDelayOffset = 0;
 
-class Soundclip;
 class VideoProcessor;
 class VoiceChannel;
 class VoiceProcessor;
@@ -104,8 +103,10 @@ class ChannelManager : public rtc::MessageHandler,
   // The operations below all occur on the worker thread.
 
   // Creates a voice channel, to be associated with the specified session.
-  VoiceChannel* CreateVoiceChannel(
-      BaseSession* session, const std::string& content_name, bool rtcp);
+  VoiceChannel* CreateVoiceChannel(BaseSession* session,
+                                   const std::string& content_name,
+                                   bool rtcp,
+                                   const AudioOptions& options);
   // Destroys a voice channel created with the Create API.
   void DestroyVoiceChannel(VoiceChannel* voice_channel,
                            VideoChannel* video_channel);
@@ -129,15 +130,9 @@ class ChannelManager : public rtc::MessageHandler,
   // Destroys a data channel created with the Create API.
   void DestroyDataChannel(DataChannel* data_channel);
 
-  // Creates a soundclip.
-  Soundclip* CreateSoundclip();
-  // Destroys a soundclip created with the Create API.
-  void DestroySoundclip(Soundclip* soundclip);
-
   // Indicates whether any channels exist.
   bool has_channels() const {
-    return (!voice_channels_.empty() || !video_channels_.empty() ||
-            !soundclips_.empty());
+    return (!voice_channels_.empty() || !video_channels_.empty());
   }
 
   // Configures the audio and video devices. A null pointer can be passed to
@@ -148,8 +143,6 @@ class ChannelManager : public rtc::MessageHandler,
   bool SetAudioOptions(const std::string& wave_in_device,
                        const std::string& wave_out_device,
                        const AudioOptions& options);
-  // Sets Engine-specific audio options according to enabled experiments.
-  bool SetEngineAudioOptions(const AudioOptions& options);
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   bool IsSameCapturer(const std::string& capturer_name,
@@ -253,7 +246,6 @@ class ChannelManager : public rtc::MessageHandler,
   typedef std::vector<VoiceChannel*> VoiceChannels;
   typedef std::vector<VideoChannel*> VideoChannels;
   typedef std::vector<DataChannel*> DataChannels;
-  typedef std::vector<Soundclip*> Soundclips;
 
   void Construct(MediaEngineInterface* me,
                  DataEngineInterface* dme,
@@ -263,8 +255,10 @@ class ChannelManager : public rtc::MessageHandler,
   bool InitMediaEngine_w();
   void DestructorDeletes_w();
   void Terminate_w();
-  VoiceChannel* CreateVoiceChannel_w(
-      BaseSession* session, const std::string& content_name, bool rtcp);
+  VoiceChannel* CreateVoiceChannel_w(BaseSession* session,
+                                     const std::string& content_name,
+                                     bool rtcp,
+                                     const AudioOptions& options);
   void DestroyVoiceChannel_w(VoiceChannel* voice_channel,
                              VideoChannel* video_channel);
   VideoChannel* CreateVideoChannel_w(BaseSession* session,
@@ -277,11 +271,8 @@ class ChannelManager : public rtc::MessageHandler,
       BaseSession* session, const std::string& content_name,
       bool rtcp, DataChannelType data_channel_type);
   void DestroyDataChannel_w(DataChannel* data_channel);
-  Soundclip* CreateSoundclip_w();
-  void DestroySoundclip_w(Soundclip* soundclip);
   bool SetAudioOptions_w(const AudioOptions& options, int delay_offset,
                          const Device* in_dev, const Device* out_dev);
-  bool SetEngineAudioOptions_w(const AudioOptions& options);
   bool SetCaptureDevice_w(const Device* cam_device);
   void OnVideoCaptureStateChange(VideoCapturer* capturer,
                                  CaptureState result);
@@ -306,7 +297,6 @@ class ChannelManager : public rtc::MessageHandler,
   VoiceChannels voice_channels_;
   VideoChannels video_channels_;
   DataChannels data_channels_;
-  Soundclips soundclips_;
 
   std::string audio_in_device_;
   std::string audio_out_device_;
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc
index b0abf0487ee..1ffdaf28369 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc
@@ -125,15 +125,15 @@ TEST_F(ChannelManagerTest, StartupShutdownOnThread) {
 TEST_F(ChannelManagerTest, CreateDestroyChannels) {
   EXPECT_TRUE(cm_->Init());
   cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
-      session_, cricket::CN_AUDIO, false);
-  EXPECT_TRUE(voice_channel != NULL);
+      session_, cricket::CN_AUDIO, false, AudioOptions());
+  EXPECT_TRUE(voice_channel != nullptr);
   cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
       session_, cricket::CN_VIDEO, false, VideoOptions(), voice_channel);
-  EXPECT_TRUE(video_channel != NULL);
+  EXPECT_TRUE(video_channel != nullptr);
   cricket::DataChannel* data_channel =
       cm_->CreateDataChannel(session_, cricket::CN_DATA,
                              false, cricket::DCT_RTP);
-  EXPECT_TRUE(data_channel != NULL);
+  EXPECT_TRUE(data_channel != nullptr);
   cm_->DestroyVideoChannel(video_channel);
   cm_->DestroyVoiceChannel(voice_channel, nullptr);
   cm_->DestroyDataChannel(data_channel);
@@ -148,15 +148,15 @@ TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
   delete session_;
   session_ = new cricket::FakeSession(&worker_, true);
   cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
-      session_, cricket::CN_AUDIO, false);
-  EXPECT_TRUE(voice_channel != NULL);
+      session_, cricket::CN_AUDIO, false, AudioOptions());
+  EXPECT_TRUE(voice_channel != nullptr);
   cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
       session_, cricket::CN_VIDEO, false, VideoOptions(), voice_channel);
-  EXPECT_TRUE(video_channel != NULL);
+  EXPECT_TRUE(video_channel != nullptr);
   cricket::DataChannel* data_channel =
       cm_->CreateDataChannel(session_, cricket::CN_DATA,
                              false, cricket::DCT_RTP);
-  EXPECT_TRUE(data_channel != NULL);
+  EXPECT_TRUE(data_channel != nullptr);
   cm_->DestroyVideoChannel(video_channel);
   cm_->DestroyVoiceChannel(voice_channel, nullptr);
   cm_->DestroyDataChannel(data_channel);
@@ -171,18 +171,18 @@ TEST_F(ChannelManagerTest, NoTransportChannelTest) {
   // The test is useless unless the session does not fail creating
   // cricket::TransportChannel.
   ASSERT_TRUE(session_->CreateChannel(
-      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP) == NULL);
+                  "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP) == nullptr);
 
   cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
-      session_, cricket::CN_AUDIO, false);
-  EXPECT_TRUE(voice_channel == NULL);
+      session_, cricket::CN_AUDIO, false, AudioOptions());
+  EXPECT_TRUE(voice_channel == nullptr);
   cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
       session_, cricket::CN_VIDEO, false, VideoOptions(), voice_channel);
-  EXPECT_TRUE(video_channel == NULL);
+  EXPECT_TRUE(video_channel == nullptr);
   cricket::DataChannel* data_channel =
       cm_->CreateDataChannel(session_, cricket::CN_DATA,
                              false, cricket::DCT_RTP);
-  EXPECT_TRUE(data_channel == NULL);
+  EXPECT_TRUE(data_channel == nullptr);
   cm_->Terminate();
 }
 
@@ -309,25 +309,6 @@ TEST_F(ChannelManagerTest, SetAudioOptions) {
   EXPECT_FALSE(cm_->SetAudioOptions("audio-in9", "audio-out2", options));
 }
 
-TEST_F(ChannelManagerTest, SetEngineAudioOptions) {
-  EXPECT_TRUE(cm_->Init());
-  // Test setting specific values.
-  AudioOptions options;
-  options.experimental_ns.Set(true);
-  EXPECT_TRUE(cm_->SetEngineAudioOptions(options));
-  bool experimental_ns = false;
-  EXPECT_TRUE(fme_->audio_options().experimental_ns.Get(&experimental_ns));
-  EXPECT_TRUE(experimental_ns);
-}
-
-TEST_F(ChannelManagerTest, SetEngineAudioOptionsBeforeInitFails) {
-  // Test that values that we set before Init are not applied.
-  AudioOptions options;
-  options.experimental_ns.Set(true);
-  EXPECT_FALSE(cm_->SetEngineAudioOptions(options));
-  EXPECT_FALSE(fme_->audio_options().experimental_ns.IsSet());
-}
-
 TEST_F(ChannelManagerTest, SetCaptureDeviceBeforeInit) {
   // Test that values that we set before Init are applied.
   EXPECT_TRUE(cm_->SetCaptureDevice("video-in2"));
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediarecorder.cc b/chromium/third_party/libjingle/source/talk/session/media/mediarecorder.cc
deleted file mode 100644
index 9ce84f38389..00000000000
--- a/chromium/third_party/libjingle/source/talk/session/media/mediarecorder.cc
+++ /dev/null
@@ -1,224 +0,0 @@
-/*
- * libjingle
- * Copyright 2010 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.
- */
-
-#include "talk/session/media/mediarecorder.h"
-
-#include <limits.h>
-
-#include <string>
-
-#include "talk/media/base/rtpdump.h"
-#include "webrtc/base/fileutils.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/pathutils.h"
-
-
-namespace cricket {
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of RtpDumpSink.
-///////////////////////////////////////////////////////////////////////////
-RtpDumpSink::RtpDumpSink(rtc::StreamInterface* stream)
-    : max_size_(INT_MAX),
-      recording_(false),
-      packet_filter_(PF_NONE) {
-  stream_.reset(stream);
-}
-
-RtpDumpSink::~RtpDumpSink() {}
-
-void RtpDumpSink::SetMaxSize(size_t size) {
-  rtc::CritScope cs(&critical_section_);
-  max_size_ = size;
-}
-
-bool RtpDumpSink::Enable(bool enable) {
-  rtc::CritScope cs(&critical_section_);
-
-  recording_ = enable;
-
-  // Create a file and the RTP writer if we have not done yet.
-  if (recording_ && !writer_) {
-    if (!stream_) {
-      return false;
-    }
-    writer_.reset(new RtpDumpWriter(stream_.get()));
-    writer_->set_packet_filter(packet_filter_);
-  } else if (!recording_ && stream_) {
-    stream_->Flush();
-  }
-  return true;
-}
-
-void RtpDumpSink::OnPacket(const void* data, size_t size, bool rtcp) {
-  rtc::CritScope cs(&critical_section_);
-
-  if (recording_ && writer_) {
-    size_t current_size;
-    if (writer_->GetDumpSize(&current_size) &&
-        current_size + RtpDumpPacket::kHeaderLength + size <= max_size_) {
-      if (!rtcp) {
-        writer_->WriteRtpPacket(data, size);
-      } else {
-        // TODO(whyuan): Enable recording RTCP.
-      }
-    }
-  }
-}
-
-void RtpDumpSink::set_packet_filter(int filter) {
-  rtc::CritScope cs(&critical_section_);
-  packet_filter_ = filter;
-  if (writer_) {
-    writer_->set_packet_filter(packet_filter_);
-  }
-}
-
-void RtpDumpSink::Flush() {
-  rtc::CritScope cs(&critical_section_);
-  if (stream_) {
-    stream_->Flush();
-  }
-}
-
-///////////////////////////////////////////////////////////////////////////
-// Implementation of MediaRecorder.
-///////////////////////////////////////////////////////////////////////////
-MediaRecorder::MediaRecorder() {}
-
-MediaRecorder::~MediaRecorder() {
-  rtc::CritScope cs(&critical_section_);
-  std::map<BaseChannel*, SinkPair*>::iterator itr;
-  for (itr = sinks_.begin(); itr != sinks_.end(); ++itr) {
-    delete itr->second;
-  }
-}
-
-bool MediaRecorder::AddChannel(VoiceChannel* channel,
-                               rtc::StreamInterface* send_stream,
-                               rtc::StreamInterface* recv_stream,
-                               int filter) {
-  return InternalAddChannel(channel, false, send_stream, recv_stream,
-                            filter);
-}
-bool MediaRecorder::AddChannel(VideoChannel* channel,
-                               rtc::StreamInterface* send_stream,
-                               rtc::StreamInterface* recv_stream,
-                               int filter) {
-  return InternalAddChannel(channel, true, send_stream, recv_stream,
-                            filter);
-}
-
-bool MediaRecorder::InternalAddChannel(BaseChannel* channel,
-                                       bool video_channel,
-                                       rtc::StreamInterface* send_stream,
-                                       rtc::StreamInterface* recv_stream,
-                                       int filter) {
-  if (!channel) {
-    return false;
-  }
-
-  rtc::CritScope cs(&critical_section_);
-  if (sinks_.end() != sinks_.find(channel)) {
-    return false;  // The channel was added already.
-  }
-
-  SinkPair* sink_pair = new SinkPair;
-  sink_pair->video_channel = video_channel;
-  sink_pair->filter = filter;
-  sink_pair->send_sink.reset(new RtpDumpSink(send_stream));
-  sink_pair->send_sink->set_packet_filter(filter);
-  sink_pair->recv_sink.reset(new RtpDumpSink(recv_stream));
-  sink_pair->recv_sink->set_packet_filter(filter);
-  sinks_[channel] = sink_pair;
-
-  return true;
-}
-
-void MediaRecorder::RemoveChannel(BaseChannel* channel,
-                                  SinkType type) {
-  rtc::CritScope cs(&critical_section_);
-  std::map<BaseChannel*, SinkPair*>::iterator itr = sinks_.find(channel);
-  if (sinks_.end() != itr) {
-    channel->UnregisterSendSink(itr->second->send_sink.get(), type);
-    channel->UnregisterRecvSink(itr->second->recv_sink.get(), type);
-    delete itr->second;
-    sinks_.erase(itr);
-  }
-}
-
-bool MediaRecorder::EnableChannel(
-    BaseChannel* channel, bool enable_send, bool enable_recv,
-    SinkType type) {
-  rtc::CritScope cs(&critical_section_);
-  std::map<BaseChannel*, SinkPair*>::iterator itr = sinks_.find(channel);
-  if (sinks_.end() == itr) {
-    return false;
-  }
-
-  SinkPair* sink_pair = itr->second;
-  RtpDumpSink* sink = sink_pair->send_sink.get();
-  sink->Enable(enable_send);
-  if (enable_send) {
-    channel->RegisterSendSink(sink, &RtpDumpSink::OnPacket, type);
-  } else {
-    channel->UnregisterSendSink(sink, type);
-  }
-
-  sink = sink_pair->recv_sink.get();
-  sink->Enable(enable_recv);
-  if (enable_recv) {
-    channel->RegisterRecvSink(sink, &RtpDumpSink::OnPacket, type);
-  } else {
-    channel->UnregisterRecvSink(sink, type);
-  }
-
-  if (sink_pair->video_channel &&
-      (sink_pair->filter & PF_RTPPACKET) == PF_RTPPACKET) {
-    // Request a full intra frame.
-    VideoChannel* video_channel = static_cast<VideoChannel*>(channel);
-    if (enable_send) {
-      video_channel->SendIntraFrame();
-    }
-    if (enable_recv) {
-      video_channel->RequestIntraFrame();
-    }
-  }
-
-  return true;
-}
-
-void MediaRecorder::FlushSinks() {
-  rtc::CritScope cs(&critical_section_);
-  std::map<BaseChannel*, SinkPair*>::iterator itr;
-  for (itr = sinks_.begin(); itr != sinks_.end(); ++itr) {
-    itr->second->send_sink->Flush();
-    itr->second->recv_sink->Flush();
-  }
-}
-
-}  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediarecorder.h b/chromium/third_party/libjingle/source/talk/session/media/mediarecorder.h
deleted file mode 100644
index 095f355c389..00000000000
--- a/chromium/third_party/libjingle/source/talk/session/media/mediarecorder.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * libjingle
- * Copyright 2010 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.
- */
-
-#ifndef TALK_SESSION_MEDIA_MEDIARECORDER_H_
-#define TALK_SESSION_MEDIA_MEDIARECORDER_H_
-
-#include <map>
-#include <string>
-
-#include "talk/session/media/channel.h"
-#include "talk/session/media/mediasink.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/sigslot.h"
-
-namespace rtc {
-class Pathname;
-class FileStream;
-}
-
-namespace cricket {
-
-class BaseChannel;
-class VideoChannel;
-class VoiceChannel;
-class RtpDumpWriter;
-
-// RtpDumpSink implements MediaSinkInterface by dumping the RTP/RTCP packets to
-// a file.
-class RtpDumpSink : public MediaSinkInterface, public sigslot::has_slots<> {
- public:
-  // Takes ownership of stream.
-  explicit RtpDumpSink(rtc::StreamInterface* stream);
-  virtual ~RtpDumpSink();
-
-  virtual void SetMaxSize(size_t size);
-  virtual bool Enable(bool enable);
-  virtual bool IsEnabled() const { return recording_; }
-  virtual void OnPacket(const void* data, size_t size, bool rtcp);
-  virtual void set_packet_filter(int filter);
-  int packet_filter() const { return packet_filter_; }
-  void Flush();
-
- private:
-  size_t max_size_;
-  bool recording_;
-  int packet_filter_;
-  rtc::scoped_ptr<rtc::StreamInterface> stream_;
-  rtc::scoped_ptr<RtpDumpWriter> writer_;
-  rtc::CriticalSection critical_section_;
-
-  DISALLOW_COPY_AND_ASSIGN(RtpDumpSink);
-};
-
-class MediaRecorder {
- public:
-  MediaRecorder();
-  virtual ~MediaRecorder();
-
-  bool AddChannel(VoiceChannel* channel,
-                  rtc::StreamInterface* send_stream,
-                  rtc::StreamInterface* recv_stream,
-                  int filter);
-  bool AddChannel(VideoChannel* channel,
-                  rtc::StreamInterface* send_stream,
-                  rtc::StreamInterface* recv_stream,
-                  int filter);
-  void RemoveChannel(BaseChannel* channel, SinkType type);
-  bool EnableChannel(BaseChannel* channel, bool enable_send, bool enable_recv,
-                     SinkType type);
-  void FlushSinks();
-
- private:
-  struct SinkPair {
-    bool video_channel;
-    int filter;
-    rtc::scoped_ptr<RtpDumpSink> send_sink;
-    rtc::scoped_ptr<RtpDumpSink> recv_sink;
-  };
-
-  bool InternalAddChannel(BaseChannel* channel,
-                          bool video_channel,
-                          rtc::StreamInterface* send_stream,
-                          rtc::StreamInterface* recv_stream,
-                          int filter);
-
-  std::map<BaseChannel*, SinkPair*> sinks_;
-  rtc::CriticalSection critical_section_;
-
-  DISALLOW_COPY_AND_ASSIGN(MediaRecorder);
-};
-
-}  // namespace cricket
-
-#endif  // TALK_SESSION_MEDIA_MEDIARECORDER_H_
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediarecorder_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/mediarecorder_unittest.cc
deleted file mode 100644
index 52a92457fa2..00000000000
--- a/chromium/third_party/libjingle/source/talk/session/media/mediarecorder_unittest.cc
+++ /dev/null
@@ -1,356 +0,0 @@
-/*
- * libjingle
- * Copyright 2010 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.
- */
-
-#include <string>
-
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/base/rtpdump.h"
-#include "talk/media/base/testutils.h"
-#include "webrtc/p2p/base/fakesession.h"
-#include "talk/session/media/channel.h"
-#include "talk/session/media/mediarecorder.h"
-#include "webrtc/base/bytebuffer.h"
-#include "webrtc/base/fileutils.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/thread.h"
-
-namespace cricket {
-
-rtc::StreamInterface* Open(const std::string& path) {
-  return rtc::Filesystem::OpenFile(
-      rtc::Pathname(path), "wb");
-}
-
-/////////////////////////////////////////////////////////////////////////
-// Test RtpDumpSink
-/////////////////////////////////////////////////////////////////////////
-class RtpDumpSinkTest : public testing::Test {
- public:
-  virtual void SetUp() {
-    EXPECT_TRUE(rtc::Filesystem::GetTemporaryFolder(path_, true, NULL));
-    path_.SetPathname(rtc::Filesystem::TempFilename(path_, "sink-test"));
-    sink_.reset(new RtpDumpSink(Open(path_.pathname())));
-
-    for (int i = 0; i < ARRAY_SIZE(rtp_buf_); ++i) {
-      RtpTestUtility::kTestRawRtpPackets[i].WriteToByteBuffer(
-          RtpTestUtility::kDefaultSsrc, &rtp_buf_[i]);
-    }
-  }
-
-  virtual void TearDown() {
-    stream_.reset();
-    EXPECT_TRUE(rtc::Filesystem::DeleteFile(path_));
-  }
-
- protected:
-  void OnRtpPacket(const RawRtpPacket& raw) {
-    rtc::ByteBuffer buf;
-    raw.WriteToByteBuffer(RtpTestUtility::kDefaultSsrc, &buf);
-    sink_->OnPacket(buf.Data(), buf.Length(), false);
-  }
-
-  rtc::StreamResult ReadPacket(RtpDumpPacket* packet) {
-    if (!stream_.get()) {
-      sink_.reset();  // This will close the file. So we can read it.
-      stream_.reset(rtc::Filesystem::OpenFile(path_, "rb"));
-      reader_.reset(new RtpDumpReader(stream_.get()));
-    }
-    return reader_->ReadPacket(packet);
-  }
-
-  rtc::Pathname path_;
-  rtc::scoped_ptr<RtpDumpSink> sink_;
-  rtc::ByteBuffer rtp_buf_[3];
-  rtc::scoped_ptr<rtc::StreamInterface> stream_;
-  rtc::scoped_ptr<RtpDumpReader> reader_;
-};
-
-TEST_F(RtpDumpSinkTest, TestRtpDumpSink) {
-  // By default, the sink is disabled. The 1st packet is not written.
-  EXPECT_FALSE(sink_->IsEnabled());
-  sink_->set_packet_filter(PF_ALL);
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[0]);
-
-  // Enable the sink. The 2nd packet is written.
-  EXPECT_TRUE(sink_->Enable(true));
-  EXPECT_TRUE(sink_->IsEnabled());
-  EXPECT_TRUE(rtc::Filesystem::IsFile(path_.pathname()));
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[1]);
-
-  // Disable the sink. The 3rd packet is not written.
-  EXPECT_TRUE(sink_->Enable(false));
-  EXPECT_FALSE(sink_->IsEnabled());
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[2]);
-
-  // Read the recorded file and verify it contains only the 2nd packet.
-  RtpDumpPacket packet;
-  EXPECT_EQ(rtc::SR_SUCCESS, ReadPacket(&packet));
-  EXPECT_TRUE(RtpTestUtility::VerifyPacket(
-      &packet, &RtpTestUtility::kTestRawRtpPackets[1], false));
-  EXPECT_EQ(rtc::SR_EOS, ReadPacket(&packet));
-}
-
-TEST_F(RtpDumpSinkTest, TestRtpDumpSinkMaxSize) {
-  EXPECT_TRUE(sink_->Enable(true));
-  sink_->set_packet_filter(PF_ALL);
-  sink_->SetMaxSize(strlen(RtpDumpFileHeader::kFirstLine) +
-                    RtpDumpFileHeader::kHeaderLength +
-                    RtpDumpPacket::kHeaderLength +
-                    RtpTestUtility::kTestRawRtpPackets[0].size());
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[0]);
-
-  // Exceed the limit size: the 2nd and 3rd packets are not written.
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[1]);
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[2]);
-
-  // Read the recorded file and verify that it contains only the first packet.
-  RtpDumpPacket packet;
-  EXPECT_EQ(rtc::SR_SUCCESS, ReadPacket(&packet));
-  EXPECT_TRUE(RtpTestUtility::VerifyPacket(
-      &packet, &RtpTestUtility::kTestRawRtpPackets[0], false));
-  EXPECT_EQ(rtc::SR_EOS, ReadPacket(&packet));
-}
-
-TEST_F(RtpDumpSinkTest, TestRtpDumpSinkFilter) {
-  // The default filter is PF_NONE.
-  EXPECT_EQ(PF_NONE, sink_->packet_filter());
-
-  // Set to PF_RTPHEADER before enable.
-  sink_->set_packet_filter(PF_RTPHEADER);
-  EXPECT_EQ(PF_RTPHEADER, sink_->packet_filter());
-  EXPECT_TRUE(sink_->Enable(true));
-  // We dump only the header of the first packet.
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[0]);
-
-  // Set the filter to PF_RTPPACKET. We dump all the second packet.
-  sink_->set_packet_filter(PF_RTPPACKET);
-  EXPECT_EQ(PF_RTPPACKET, sink_->packet_filter());
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[1]);
-
-  // Set the filter to PF_NONE. We do not dump the third packet.
-  sink_->set_packet_filter(PF_NONE);
-  EXPECT_EQ(PF_NONE, sink_->packet_filter());
-  OnRtpPacket(RtpTestUtility::kTestRawRtpPackets[2]);
-
-  // Read the recorded file and verify the header of the first packet and
-  // the whole packet for the second packet.
-  RtpDumpPacket packet;
-  EXPECT_EQ(rtc::SR_SUCCESS, ReadPacket(&packet));
-  EXPECT_TRUE(RtpTestUtility::VerifyPacket(
-      &packet, &RtpTestUtility::kTestRawRtpPackets[0], true));
-  EXPECT_EQ(rtc::SR_SUCCESS, ReadPacket(&packet));
-  EXPECT_TRUE(RtpTestUtility::VerifyPacket(
-      &packet, &RtpTestUtility::kTestRawRtpPackets[1], false));
-  EXPECT_EQ(rtc::SR_EOS, ReadPacket(&packet));
-}
-
-/////////////////////////////////////////////////////////////////////////
-// Test MediaRecorder
-/////////////////////////////////////////////////////////////////////////
-void TestMediaRecorder(BaseChannel* channel,
-                       FakeVideoMediaChannel* video_media_channel,
-                       int filter) {
-  // Create media recorder.
-  rtc::scoped_ptr<MediaRecorder> recorder(new MediaRecorder);
-  // Fail to EnableChannel before AddChannel.
-  EXPECT_FALSE(recorder->EnableChannel(channel, true, true, SINK_PRE_CRYPTO));
-  EXPECT_FALSE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_FALSE(channel->HasRecvSinks(SINK_PRE_CRYPTO));
-  EXPECT_FALSE(channel->HasSendSinks(SINK_POST_CRYPTO));
-  EXPECT_FALSE(channel->HasRecvSinks(SINK_POST_CRYPTO));
-
-  // Add the channel to the recorder.
-  rtc::Pathname path;
-  EXPECT_TRUE(rtc::Filesystem::GetTemporaryFolder(path, true, NULL));
-  std::string send_file =
-      rtc::Filesystem::TempFilename(path, "send");
-  std::string recv_file =
-      rtc::Filesystem::TempFilename(path, "recv");
-  if (video_media_channel) {
-    EXPECT_TRUE(recorder->AddChannel(static_cast<VideoChannel*>(channel),
-                                     Open(send_file), Open(recv_file), filter));
-  } else {
-    EXPECT_TRUE(recorder->AddChannel(static_cast<VoiceChannel*>(channel),
-                                     Open(send_file), Open(recv_file), filter));
-  }
-
-  // Enable recording only the sent media.
-  EXPECT_TRUE(recorder->EnableChannel(channel, true, false, SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_FALSE(channel->HasRecvSinks(SINK_POST_CRYPTO));
-  EXPECT_FALSE(channel->HasSendSinks(SINK_POST_CRYPTO));
-  EXPECT_FALSE(channel->HasRecvSinks(SINK_POST_CRYPTO));
-  if (video_media_channel) {
-    EXPECT_TRUE_WAIT(video_media_channel->sent_intra_frame(), 100);
-  }
-
-  // Enable recording only the received meida.
-  EXPECT_TRUE(recorder->EnableChannel(channel, false, true, SINK_PRE_CRYPTO));
-  EXPECT_FALSE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasRecvSinks(SINK_PRE_CRYPTO));
-  if (video_media_channel) {
-    EXPECT_TRUE(video_media_channel->requested_intra_frame());
-  }
-
-  // Enable recording both the sent and the received video.
-  EXPECT_TRUE(recorder->EnableChannel(channel, true, true, SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasRecvSinks(SINK_PRE_CRYPTO));
-
-  // Enable recording only headers.
-  if (video_media_channel) {
-    video_media_channel->set_sent_intra_frame(false);
-    video_media_channel->set_requested_intra_frame(false);
-  }
-  EXPECT_TRUE(recorder->EnableChannel(channel, true, true, SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasRecvSinks(SINK_PRE_CRYPTO));
-  if (video_media_channel) {
-    if ((filter & PF_RTPPACKET) == PF_RTPPACKET) {
-      // If record the whole RTP packet, trigers FIR.
-      EXPECT_TRUE(video_media_channel->requested_intra_frame());
-      EXPECT_TRUE(video_media_channel->sent_intra_frame());
-    } else {
-      // If record only the RTP header, does not triger FIR.
-      EXPECT_FALSE(video_media_channel->requested_intra_frame());
-      EXPECT_FALSE(video_media_channel->sent_intra_frame());
-    }
-  }
-
-  // Remove the voice channel from the recorder.
-  recorder->RemoveChannel(channel, SINK_PRE_CRYPTO);
-  EXPECT_FALSE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_FALSE(channel->HasRecvSinks(SINK_PRE_CRYPTO));
-
-  // Delete all files.
-  recorder.reset();
-  EXPECT_TRUE(rtc::Filesystem::DeleteFile(send_file));
-  EXPECT_TRUE(rtc::Filesystem::DeleteFile(recv_file));
-}
-
-// Fisrt start recording header and then start recording media. Verify that
-// differnt files are created for header and media.
-void TestRecordHeaderAndMedia(BaseChannel* channel,
-                              FakeVideoMediaChannel* video_media_channel) {
-  // Create RTP header recorder.
-  rtc::scoped_ptr<MediaRecorder> header_recorder(new MediaRecorder);
-
-  rtc::Pathname path;
-  EXPECT_TRUE(rtc::Filesystem::GetTemporaryFolder(path, true, NULL));
-  std::string send_header_file =
-      rtc::Filesystem::TempFilename(path, "send-header");
-  std::string recv_header_file =
-      rtc::Filesystem::TempFilename(path, "recv-header");
-  if (video_media_channel) {
-    EXPECT_TRUE(header_recorder->AddChannel(
-        static_cast<VideoChannel*>(channel),
-        Open(send_header_file), Open(recv_header_file), PF_RTPHEADER));
-  } else {
-    EXPECT_TRUE(header_recorder->AddChannel(
-        static_cast<VoiceChannel*>(channel),
-        Open(send_header_file), Open(recv_header_file), PF_RTPHEADER));
-  }
-
-  // Enable recording both sent and received.
-  EXPECT_TRUE(
-      header_recorder->EnableChannel(channel, true, true, SINK_POST_CRYPTO));
-  EXPECT_TRUE(channel->HasSendSinks(SINK_POST_CRYPTO));
-  EXPECT_TRUE(channel->HasRecvSinks(SINK_POST_CRYPTO));
-  EXPECT_FALSE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_FALSE(channel->HasRecvSinks(SINK_PRE_CRYPTO));
-  if (video_media_channel) {
-    EXPECT_FALSE(video_media_channel->sent_intra_frame());
-    EXPECT_FALSE(video_media_channel->requested_intra_frame());
-  }
-
-  // Verify that header files are created.
-  EXPECT_TRUE(rtc::Filesystem::IsFile(send_header_file));
-  EXPECT_TRUE(rtc::Filesystem::IsFile(recv_header_file));
-
-  // Create RTP header recorder.
-  rtc::scoped_ptr<MediaRecorder> recorder(new MediaRecorder);
-  std::string send_file =
-      rtc::Filesystem::TempFilename(path, "send");
-  std::string recv_file =
-      rtc::Filesystem::TempFilename(path, "recv");
-  if (video_media_channel) {
-    EXPECT_TRUE(recorder->AddChannel(
-        static_cast<VideoChannel*>(channel),
-        Open(send_file), Open(recv_file), PF_RTPPACKET));
-  } else {
-    EXPECT_TRUE(recorder->AddChannel(
-        static_cast<VoiceChannel*>(channel),
-        Open(send_file), Open(recv_file), PF_RTPPACKET));
-  }
-
-  // Enable recording both sent and received.
-  EXPECT_TRUE(recorder->EnableChannel(channel, true, true, SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasSendSinks(SINK_POST_CRYPTO));
-  EXPECT_TRUE(channel->HasRecvSinks(SINK_POST_CRYPTO));
-  EXPECT_TRUE(channel->HasSendSinks(SINK_PRE_CRYPTO));
-  EXPECT_TRUE(channel->HasRecvSinks(SINK_PRE_CRYPTO));
-  if (video_media_channel) {
-    EXPECT_TRUE_WAIT(video_media_channel->sent_intra_frame(), 100);
-    EXPECT_TRUE(video_media_channel->requested_intra_frame());
-  }
-
-  // Verify that media files are created.
-  EXPECT_TRUE(rtc::Filesystem::IsFile(send_file));
-  EXPECT_TRUE(rtc::Filesystem::IsFile(recv_file));
-
-  // Delete all files.
-  header_recorder.reset();
-  recorder.reset();
-  EXPECT_TRUE(rtc::Filesystem::DeleteFile(send_header_file));
-  EXPECT_TRUE(rtc::Filesystem::DeleteFile(recv_header_file));
-  EXPECT_TRUE(rtc::Filesystem::DeleteFile(send_file));
-  EXPECT_TRUE(rtc::Filesystem::DeleteFile(recv_file));
-}
-
-TEST(MediaRecorderTest, TestMediaRecorderVoiceChannel) {
-  // Create the voice channel.
-  FakeMediaEngine media_engine;
-  VoiceChannel channel(rtc::Thread::Current(), &media_engine,
-                       new FakeVoiceMediaChannel(NULL), NULL, "", false);
-  TestMediaRecorder(&channel, NULL, PF_RTPPACKET);
-  TestMediaRecorder(&channel, NULL, PF_RTPHEADER);
-  TestRecordHeaderAndMedia(&channel, NULL);
-}
-
-TEST(MediaRecorderTest, TestMediaRecorderVideoChannel) {
-  // Create the video channel.
-  FakeMediaEngine media_engine;
-  FakeVideoMediaChannel* media_channel = new FakeVideoMediaChannel(NULL);
-  VideoChannel channel(rtc::Thread::Current(), &media_engine,
-                       media_channel, NULL, "", false);
-  TestMediaRecorder(&channel, media_channel, PF_RTPPACKET);
-  TestMediaRecorder(&channel, media_channel, PF_RTPHEADER);
-  TestRecordHeaderAndMedia(&channel, media_channel);
-}
-
-}  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc b/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc
index b7285179613..b2999445193 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc
@@ -436,8 +436,8 @@ static bool AddStreamParams(
     StreamParamsVec* current_streams,
     MediaContentDescriptionImpl<C>* content_description,
     const bool add_legacy_stream) {
-  const bool include_rtx_stream =
-    ContainsRtxCodec(content_description->codecs());
+  const bool include_rtx_streams =
+      ContainsRtxCodec(content_description->codecs());
 
   if (streams.empty() && add_legacy_stream) {
     // TODO(perkj): Remove this legacy stream when all apps use StreamParams.
@@ -445,10 +445,10 @@ static bool AddStreamParams(
     if (IsSctp(content_description)) {
       GenerateSctpSids(*current_streams, &ssrcs);
     } else {
-      int num_ssrcs = include_rtx_stream ? 2 : 1;
+      int num_ssrcs = include_rtx_streams ? 2 : 1;
       GenerateSsrcs(*current_streams, num_ssrcs, &ssrcs);
     }
-    if (include_rtx_stream) {
+    if (include_rtx_streams) {
       content_description->AddLegacyStream(ssrcs[0], ssrcs[1]);
       content_description->set_multistream(true);
     } else {
@@ -492,11 +492,15 @@ static bool AddStreamParams(
         SsrcGroup group(kSimSsrcGroupSemantics, stream_param.ssrcs);
         stream_param.ssrc_groups.push_back(group);
       }
-      // Generate an extra ssrc for include_rtx_stream case.
-      if (include_rtx_stream) {
-        std::vector<uint32> rtx_ssrc;
-        GenerateSsrcs(*current_streams, 1, &rtx_ssrc);
-        stream_param.AddFidSsrc(ssrcs[0], rtx_ssrc[0]);
+      // Generate extra ssrcs for include_rtx_streams case.
+      if (include_rtx_streams) {
+        // Generate an RTX ssrc for every ssrc in the group.
+        std::vector<uint32> rtx_ssrcs;
+        GenerateSsrcs(*current_streams, static_cast<int>(ssrcs.size()),
+                      &rtx_ssrcs);
+        for (size_t i = 0; i < ssrcs.size(); ++i) {
+          stream_param.AddFidSsrc(ssrcs[i], rtx_ssrcs[i]);
+        }
         content_description->set_multistream(true);
       }
       stream_param.cname = cname;
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc
index f487baa9c56..7fd1d638239 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc
@@ -1767,6 +1767,47 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   EXPECT_EQ(expected_codecs, vcd->codecs());
 }
 
+// Test that when RTX is used in conjunction with simulcast, an RTX ssrc is
+// generated for each simulcast ssrc and correctly grouped.
+TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
+  MediaSessionOptions opts;
+  opts.recv_video = true;
+  opts.recv_audio = false;
+
+  // Add simulcast streams.
+  opts.AddSendVideoStream("stream1", "stream1label", 3);
+
+  // 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));
+  AddRtxCodec(VideoCodec::CreateRtxCodec(125, 97), &f1_codecs);
+  f1_.set_video_codecs(f1_codecs);
+
+  // Ensure that the offer has an RTX ssrc for each regular ssrc, and that there
+  // is a FID ssrc + grouping for each.
+  rtc::scoped_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
+  ASSERT_TRUE(offer.get() != NULL);
+  VideoContentDescription* desc = static_cast<VideoContentDescription*>(
+      offer->GetContentDescriptionByName(cricket::CN_VIDEO));
+  ASSERT_TRUE(desc != NULL);
+  EXPECT_TRUE(desc->multistream());
+  const StreamParamsVec& streams = desc->streams();
+  // Single stream.
+  ASSERT_EQ(1u, streams.size());
+  // Stream should have 6 ssrcs: 3 for video, 3 for RTX.
+  EXPECT_EQ(6u, streams[0].ssrcs.size());
+  // And should have a SIM group for the simulcast.
+  EXPECT_TRUE(streams[0].has_ssrc_group("SIM"));
+  // And a FID group for RTX.
+  EXPECT_TRUE(streams[0].has_ssrc_group("FID"));
+  std::vector<uint32> primary_ssrcs;
+  streams[0].GetPrimarySsrcs(&primary_ssrcs);
+  EXPECT_EQ(3u, primary_ssrcs.size());
+  std::vector<uint32> fid_ssrcs;
+  streams[0].GetFidSsrcs(primary_ssrcs, &fid_ssrcs);
+  EXPECT_EQ(3u, fid_ssrcs.size());
+}
+
 // Create an updated offer after creating an answer to the original offer and
 // verify that the RTP header extensions that were part of the original answer
 // are not changed in the updated offer.
diff --git a/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.cc b/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.cc
index f95199241e5..f21e0eeb5b7 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.cc
@@ -40,7 +40,16 @@ bool RtcpMuxFilter::IsActive() const {
          state_ == ST_ACTIVE;
 }
 
+void RtcpMuxFilter::SetActive() {
+  state_ = ST_ACTIVE;
+}
+
 bool RtcpMuxFilter::SetOffer(bool offer_enable, ContentSource src) {
+  if (state_ == ST_ACTIVE) {
+    // Fail if we try to deactivate and no-op if we try and activate.
+    return offer_enable;
+  }
+
   if (!ExpectOffer(offer_enable, src)) {
     LOG(LS_ERROR) << "Invalid state for change of RTCP mux offer";
     return false;
@@ -53,6 +62,11 @@ bool RtcpMuxFilter::SetOffer(bool offer_enable, ContentSource src) {
 
 bool RtcpMuxFilter::SetProvisionalAnswer(bool answer_enable,
                                          ContentSource src) {
+  if (state_ == ST_ACTIVE) {
+    // Fail if we try to deactivate and no-op if we try and activate.
+    return answer_enable;
+  }
+
   if (!ExpectAnswer(src)) {
     LOG(LS_ERROR) << "Invalid state for RTCP mux provisional answer";
     return false;
@@ -83,6 +97,11 @@ bool RtcpMuxFilter::SetProvisionalAnswer(bool answer_enable,
 }
 
 bool RtcpMuxFilter::SetAnswer(bool answer_enable, ContentSource src) {
+  if (state_ == ST_ACTIVE) {
+    // Fail if we try to deactivate and no-op if we try and activate.
+    return answer_enable;
+  }
+
   if (!ExpectAnswer(src)) {
     LOG(LS_ERROR) << "Invalid state for RTCP mux answer";
     return false;
@@ -100,19 +119,24 @@ bool RtcpMuxFilter::SetAnswer(bool answer_enable, ContentSource src) {
   return true;
 }
 
-bool RtcpMuxFilter::DemuxRtcp(const char* data, int len) {
-  // If we're muxing RTP/RTCP, we must inspect each packet delivered and
-  // determine whether it is RTP or RTCP. We do so by checking the packet type,
-  // and assuming RTP if type is 0-63 or 96-127. For additional details, see
-  // http://tools.ietf.org/html/rfc5761.
-  // Note that if we offer RTCP mux, we may receive muxed RTCP before we
-  // receive the answer, so we operate in that state too.
-  if (!offer_enable_ || state_ < ST_SENTOFFER) {
+// Check the RTP payload type.  If 63 < payload type < 96, it's RTCP.
+// For additional details, see http://tools.ietf.org/html/rfc5761.
+bool IsRtcp(const char* data, int len) {
+  if (len < 2) {
     return false;
   }
+  char pt = data[1] & 0x7F;
+  return (63 < pt) && (pt < 96);
+}
 
-  int type = (len >= 2) ? (static_cast<uint8>(data[1]) & 0x7F) : 0;
-  return (type >= 64 && type < 96);
+bool RtcpMuxFilter::DemuxRtcp(const char* data, int len) {
+  // If we're muxing RTP/RTCP, we must inspect each packet delivered
+  // and determine whether it is RTP or RTCP. We do so by looking at
+  // the RTP payload type (see IsRtcp).  Note that if we offer RTCP
+  // mux, we may receive muxed RTCP before we receive the answer, so
+  // we operate in that state too.
+  bool offered_mux = ((state_ == ST_SENTOFFER) && offer_enable_);
+  return (IsActive() || offered_mux) && IsRtcp(data, len);
 }
 
 bool RtcpMuxFilter::ExpectOffer(bool offer_enable, ContentSource source) {
diff --git a/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.h b/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.h
index 948b3c33f08..8888fd4ffb5 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter.h
@@ -41,6 +41,9 @@ class RtcpMuxFilter {
   // Whether the filter is active, i.e. has RTCP mux been properly negotiated.
   bool IsActive() const;
 
+  // Make the filter active, regardless of the current state.
+  void SetActive();
+
   // Specifies whether the offer indicates the use of RTCP mux.
   bool SetOffer(bool offer_enable, ContentSource src);
 
diff --git a/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter_unittest.cc
index 1305c891a26..d4e6376e6d7 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/rtcpmuxfilter_unittest.cc
@@ -212,3 +212,44 @@ TEST(RtcpMuxFilterTest, KeepFilterDisabledDuringUpdate) {
   EXPECT_TRUE(filter.SetAnswer(false, cricket::CS_LOCAL));
   EXPECT_FALSE(filter.IsActive());
 }
+
+// Test that we can SetActive and then can't deactivate.
+TEST(RtcpMuxFilterTest, SetActiveCantDeactivate) {
+  cricket::RtcpMuxFilter filter;
+  const char data[] = { 0, 73, 0, 0 };
+  const int len = 4;
+
+  filter.SetActive();
+  EXPECT_TRUE(filter.IsActive());
+  EXPECT_TRUE(filter.DemuxRtcp(data, len));
+
+  EXPECT_FALSE(filter.SetOffer(false, cricket::CS_LOCAL));
+  EXPECT_TRUE(filter.IsActive());
+  EXPECT_TRUE(filter.SetOffer(true, cricket::CS_LOCAL));
+  EXPECT_TRUE(filter.IsActive());
+
+  EXPECT_FALSE(filter.SetProvisionalAnswer(false, cricket::CS_REMOTE));
+  EXPECT_TRUE(filter.IsActive());
+  EXPECT_TRUE(filter.SetProvisionalAnswer(true, cricket::CS_REMOTE));
+  EXPECT_TRUE(filter.IsActive());
+
+  EXPECT_FALSE(filter.SetAnswer(false, cricket::CS_REMOTE));
+  EXPECT_TRUE(filter.IsActive());
+  EXPECT_TRUE(filter.SetAnswer(true, cricket::CS_REMOTE));
+  EXPECT_TRUE(filter.IsActive());
+
+  EXPECT_FALSE(filter.SetOffer(false, cricket::CS_REMOTE));
+  EXPECT_TRUE(filter.IsActive());
+  EXPECT_TRUE(filter.SetOffer(true, cricket::CS_REMOTE));
+  EXPECT_TRUE(filter.IsActive());
+
+  EXPECT_FALSE(filter.SetProvisionalAnswer(false, cricket::CS_LOCAL));
+  EXPECT_TRUE(filter.IsActive());
+  EXPECT_TRUE(filter.SetProvisionalAnswer(true, cricket::CS_LOCAL));
+  EXPECT_TRUE(filter.IsActive());
+
+  EXPECT_FALSE(filter.SetAnswer(false, cricket::CS_LOCAL));
+  EXPECT_TRUE(filter.IsActive());
+  EXPECT_TRUE(filter.SetAnswer(true, cricket::CS_LOCAL));
+  EXPECT_TRUE(filter.IsActive());
+}
diff --git a/chromium/third_party/libjingle/source/talk/session/media/soundclip.cc b/chromium/third_party/libjingle/source/talk/session/media/soundclip.cc
index 70a3b18e261..be8d0f99fce 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/soundclip.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/soundclip.cc
@@ -1,82 +1 @@
-/*
- * libjingle
- * Copyright 2004 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.
- */
-
-#include "talk/session/media/soundclip.h"
-
-namespace cricket {
-
-enum {
-  MSG_PLAYSOUND = 1,
-};
-
-struct PlaySoundMessageData : rtc::MessageData {
-  PlaySoundMessageData(const void *c,
-                       int l,
-                       SoundclipMedia::SoundclipFlags f)
-      : clip(c),
-        len(l),
-        flags(f),
-        result(false) {
-  }
-
-  const void *clip;
-  int len;
-  SoundclipMedia::SoundclipFlags flags;
-  bool result;
-};
-
-Soundclip::Soundclip(rtc::Thread *thread, SoundclipMedia *soundclip_media)
-    : worker_thread_(thread),
-      soundclip_media_(soundclip_media) {
-}
-
-bool Soundclip::PlaySound(const void *clip,
-                          int len,
-                          SoundclipMedia::SoundclipFlags flags) {
-  PlaySoundMessageData data(clip, len, flags);
-  worker_thread_->Send(this, MSG_PLAYSOUND, &data);
-  return data.result;
-}
-
-bool Soundclip::PlaySound_w(const void *clip,
-                            int len,
-                            SoundclipMedia::SoundclipFlags flags) {
-  return soundclip_media_->PlaySound(static_cast<const char *>(clip),
-                                     len,
-                                     flags);
-}
-
-void Soundclip::OnMessage(rtc::Message *message) {
-  ASSERT(message->message_id == MSG_PLAYSOUND);
-  PlaySoundMessageData *data =
-      static_cast<PlaySoundMessageData *>(message->pdata);
-  data->result = PlaySound_w(data->clip,
-                             data->len,
-                             data->flags);
-}
-
-}  // namespace cricket
+// TODO(solenberg): Remove this file when it's no longer built in Chromium.
diff --git a/chromium/third_party/libjingle/source/talk/session/media/soundclip.h b/chromium/third_party/libjingle/source/talk/session/media/soundclip.h
index aaf85797d4e..818ea9d4ebe 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/soundclip.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/soundclip.h
@@ -1,70 +1,2 @@
-/*
- * libjingle
- * Copyright 2004 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.
- */
+// TODO(solenberg): Remove this file when it's no longer built in Chromium.
 
-#ifndef TALK_SESSION_MEDIA_SOUNDCLIP_H_
-#define TALK_SESSION_MEDIA_SOUNDCLIP_H_
-
-#include "talk/media/base/mediaengine.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace rtc {
-
-class Thread;
-
-}
-
-namespace cricket {
-
-// Soundclip wraps SoundclipMedia to support marshalling calls to the proper
-// thread.
-class Soundclip : private rtc::MessageHandler {
- public:
-  Soundclip(rtc::Thread* thread, SoundclipMedia* soundclip_media);
-
-  // Plays a sound out to the speakers with the given audio stream. The stream
-  // must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
-  // on this Soundclip, it is stopped. If clip is NULL, nothing is played.
-  // Returns whether it was successful.
-  bool PlaySound(const void* clip,
-                 int len,
-                 SoundclipMedia::SoundclipFlags flags);
-
- private:
-  bool PlaySound_w(const void* clip,
-                   int len,
-                   SoundclipMedia::SoundclipFlags flags);
-
-  // From MessageHandler
-  virtual void OnMessage(rtc::Message* message);
-
-  rtc::Thread* worker_thread_;
-  rtc::scoped_ptr<SoundclipMedia> soundclip_media_;
-};
-
-}  // namespace cricket
-
-#endif  // TALK_SESSION_MEDIA_SOUNDCLIP_H_
diff --git a/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.cc b/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.cc
index dc93dd463d8..33e42c09d4a 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.cc
@@ -475,18 +475,27 @@ bool SrtpFilter::ParseKeyParams(const std::string& key_params,
 
 bool SrtpSession::inited_ = false;
 
+// This lock protects SrtpSession::inited_ and SrtpSession::sessions_.
+rtc::GlobalLockPod SrtpSession::lock_;
+
 SrtpSession::SrtpSession()
     : session_(NULL),
       rtp_auth_tag_len_(0),
       rtcp_auth_tag_len_(0),
       srtp_stat_(new SrtpStat()),
       last_send_seq_num_(-1) {
-  sessions()->push_back(this);
+  {
+    rtc::GlobalLockScope ls(&lock_);
+    sessions()->push_back(this);
+  }
   SignalSrtpError.repeat(srtp_stat_->SignalSrtpError);
 }
 
 SrtpSession::~SrtpSession() {
-  sessions()->erase(std::find(sessions()->begin(), sessions()->end(), this));
+  {
+    rtc::GlobalLockScope ls(&lock_);
+    sessions()->erase(std::find(sessions()->begin(), sessions()->end(), this));
+  }
   if (session_) {
     srtp_dealloc(session_);
   }
@@ -691,6 +700,7 @@ bool SrtpSession::SetKey(int type, const std::string& cs,
 
   int err = srtp_create(&session_, &policy);
   if (err != err_status_ok) {
+    session_ = NULL;
     LOG(LS_ERROR) << "Failed to create SRTP session, err=" << err;
     return false;
   }
@@ -702,6 +712,8 @@ bool SrtpSession::SetKey(int type, const std::string& cs,
 }
 
 bool SrtpSession::Init() {
+  rtc::GlobalLockScope ls(&lock_);
+
   if (!inited_) {
     int err;
     err = srtp_init();
@@ -729,6 +741,8 @@ bool SrtpSession::Init() {
 }
 
 void SrtpSession::Terminate() {
+  rtc::GlobalLockScope ls(&lock_);
+
   if (inited_) {
     int err = srtp_shutdown();
     if (err) {
@@ -760,6 +774,8 @@ void SrtpSession::HandleEvent(const srtp_event_data_t* ev) {
 }
 
 void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) {
+  rtc::GlobalLockScope ls(&lock_);
+
   for (std::list<SrtpSession*>::iterator it = sessions()->begin();
        it != sessions()->end(); ++it) {
     if ((*it)->session_ == ev->session) {
@@ -770,7 +786,7 @@ void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) {
 }
 
 std::list<SrtpSession*>* SrtpSession::sessions() {
-  LIBJINGLE_DEFINE_STATIC_LOCAL(std::list<SrtpSession*>, sessions, ());
+  RTC_DEFINE_STATIC_LOCAL(std::list<SrtpSession*>, sessions, ());
   return &sessions;
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h b/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h
index 43cb241b756..f171f5fe660 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h
@@ -36,6 +36,7 @@
 #include "talk/media/base/cryptoparams.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslotrepeater.h"
 
@@ -242,6 +243,7 @@ class SrtpSession {
   int rtcp_auth_tag_len_;
   rtc::scoped_ptr<SrtpStat> srtp_stat_;
   static bool inited_;
+  static rtc::GlobalLockPod lock_;
   int last_send_seq_num_;
   DISALLOW_COPY_AND_ASSIGN(SrtpSession);
 };