Bring enable_webrtc back

Returns the GN args and BUILDFLAG to disable WebRTC code shrinking the Chromium binary by a 15%. Reverting https://chromium-review.googlesource.com/1044220, https://chromium-review.googlesource.com/1059408 and fixes later bitrot. Change-Id: Ic46c2832dbfacaeffa6a00a12a8a144cf0adf7f0 Reviewed-by: Michal Klocek <michal.klocek@qt.io>
author: Allan Sandfeld Jensen <allan.jensen@qt.io> 2020-08-03 13:19:41 +0200
committer: Allan Sandfeld Jensen <allan.jensen@qt.io> 2020-12-03 17:29:43 +0100
commit: 7eb6adb6cac7c604d4717bd3ab0f0e8cca452fe5 (patch)
tree: 5218647eee4c52fad271362896e49c11ef1a1658
parent: 0199422d32f37be484483a523f7e88a3e1def4f2 (diff)
download: qtwebengine-chromium-7eb6adb6cac7c604d4717bd3ab0f0e8cca452fe5.tar.gz
80 files changed, 850 insertions, 205 deletions
diff --git a/chromium/chrome/browser/BUILD.gn b/chromium/chrome/browser/BUILD.gn
index 348d5b4d745..6d344b1e3a7 100644
--- a/chromium/chrome/browser/BUILD.gn
+++ b/chromium/chrome/browser/BUILD.gn
@@ -25,6 +25,7 @@ import("//components/signin/features.gni")
 import("//components/spellcheck/spellcheck_build_features.gni")
 import("//device/vr/buildflags/buildflags.gni")
 import("//extensions/buildflags/buildflags.gni")
+import("//media/media_options.gni")
 import("//net/features.gni")
 import("//pdf/features.gni")
 import("//ppapi/buildflags/buildflags.gni")
@@ -699,62 +700,6 @@ static_library("browser") {
     "media/platform_verification_impl.h",
     "media/router/media_router_feature.cc",
     "media/router/media_router_feature.h",
-    "media/webrtc/audio_debug_recordings_handler.cc",
-    "media/webrtc/audio_debug_recordings_handler.h",
-    "media/webrtc/camera_pan_tilt_zoom_permission_context.cc",
-    "media/webrtc/camera_pan_tilt_zoom_permission_context.h",
-    "media/webrtc/desktop_media_list.h",
-    "media/webrtc/desktop_media_list_base.cc",
-    "media/webrtc/desktop_media_list_base.h",
-    "media/webrtc/desktop_media_picker.cc",
-    "media/webrtc/desktop_media_picker.h",
-    "media/webrtc/desktop_media_picker_factory.cc",
-    "media/webrtc/desktop_media_picker_factory.h",
-    "media/webrtc/desktop_media_picker_manager.cc",
-    "media/webrtc/desktop_media_picker_manager.h",
-    "media/webrtc/media_capture_devices_dispatcher.cc",
-    "media/webrtc/media_capture_devices_dispatcher.h",
-    "media/webrtc/media_stream_capture_indicator.cc",
-    "media/webrtc/media_stream_capture_indicator.h",
-    "media/webrtc/media_stream_device_permission_context.cc",
-    "media/webrtc/media_stream_device_permission_context.h",
-    "media/webrtc/media_stream_device_permissions.cc",
-    "media/webrtc/media_stream_device_permissions.h",
-    "media/webrtc/native_desktop_media_list.cc",
-    "media/webrtc/native_desktop_media_list.h",
-    "media/webrtc/permission_bubble_media_access_handler.cc",
-    "media/webrtc/permission_bubble_media_access_handler.h",
-    "media/webrtc/webrtc_event_log_history.cc",
-    "media/webrtc/webrtc_event_log_history.h",
-    "media/webrtc/webrtc_event_log_manager.cc",
-    "media/webrtc/webrtc_event_log_manager.h",
-    "media/webrtc/webrtc_event_log_manager_common.cc",
-    "media/webrtc/webrtc_event_log_manager_common.h",
-    "media/webrtc/webrtc_event_log_manager_keyed_service.cc",
-    "media/webrtc/webrtc_event_log_manager_keyed_service.h",
-    "media/webrtc/webrtc_event_log_manager_keyed_service_factory.cc",
-    "media/webrtc/webrtc_event_log_manager_keyed_service_factory.h",
-    "media/webrtc/webrtc_event_log_manager_local.cc",
-    "media/webrtc/webrtc_event_log_manager_local.h",
-    "media/webrtc/webrtc_event_log_manager_remote.cc",
-    "media/webrtc/webrtc_event_log_manager_remote.h",
-    "media/webrtc/webrtc_event_log_uploader.cc",
-    "media/webrtc/webrtc_event_log_uploader.h",
-    "media/webrtc/webrtc_log_buffer.cc",
-    "media/webrtc/webrtc_log_buffer.h",
-    "media/webrtc/webrtc_log_uploader.cc",
-    "media/webrtc/webrtc_log_uploader.h",
-    "media/webrtc/webrtc_log_util.cc",
-    "media/webrtc/webrtc_log_util.h",
-    "media/webrtc/webrtc_logging_controller.cc",
-    "media/webrtc/webrtc_logging_controller.h",
-    "media/webrtc/webrtc_rtp_dump_handler.cc",
-    "media/webrtc/webrtc_rtp_dump_handler.h",
-    "media/webrtc/webrtc_rtp_dump_writer.cc",
-    "media/webrtc/webrtc_rtp_dump_writer.h",
-    "media/webrtc/webrtc_text_log_handler.cc",
-    "media/webrtc/webrtc_text_log_handler.h",
-    "media/webrtc/window_icon_util.h",
     "memory/chrome_browser_main_extra_parts_memory.cc",
     "memory/chrome_browser_main_extra_parts_memory.h",
     "memory/enterprise_memory_limit_evaluator.cc",
@@ -2174,9 +2119,6 @@ static_library("browser") {
     "//components/web_resource",
     "//components/webdata/common",
     "//components/webdata_services",
-    "//components/webrtc",
-    "//components/webrtc_logging/browser",
-    "//components/webrtc_logging/common",
     "//content/app/resources",
     "//content/public/browser",
     "//content/public/common",
@@ -2209,7 +2151,6 @@ static_library("browser") {
     "//printing",
     "//printing/buildflags",
     "//rlz/buildflags",
-    "//services/audio/public/cpp",
     "//services/cert_verifier:lib",
     "//services/data_decoder/public/cpp",
     "//services/device/public/cpp:device_features",
@@ -2249,7 +2190,6 @@ static_library("browser") {
     "//third_party/libyuv",
     "//third_party/metrics_proto",
     "//third_party/re2",
-    "//third_party/webrtc_overrides:webrtc_component",
     "//third_party/widevine/cdm:buildflags",
     "//third_party/widevine/cdm:headers",
     "//third_party/zlib",
@@ -6091,6 +6031,78 @@ static_library("browser") {
     ]
   }
 
+  if (enable_webrtc) {
+    sources += [
+      "media/webrtc/audio_debug_recordings_handler.cc",
+      "media/webrtc/audio_debug_recordings_handler.h",
+      "media/webrtc/desktop_media_list.h",
+      "media/webrtc/desktop_media_list_base.cc",
+      "media/webrtc/desktop_media_list_base.h",
+      "media/webrtc/desktop_media_picker.cc",
+      "media/webrtc/desktop_media_picker.h",
+      "media/webrtc/desktop_media_picker_factory.cc",
+      "media/webrtc/desktop_media_picker_factory.h",
+      "media/webrtc/desktop_media_picker_manager.cc",
+      "media/webrtc/desktop_media_picker_manager.h",
+      "media/webrtc/media_capture_devices_dispatcher.cc",
+      "media/webrtc/media_capture_devices_dispatcher.h",
+      "media/webrtc/media_stream_capture_indicator.cc",
+      "media/webrtc/media_stream_capture_indicator.h",
+      "media/webrtc/media_stream_device_permission_context.cc",
+      "media/webrtc/media_stream_device_permission_context.h",
+      "media/webrtc/media_stream_device_permissions.cc",
+      "media/webrtc/media_stream_device_permissions.h",
+      "media/webrtc/media_stream_devices_controller.cc",
+      "media/webrtc/media_stream_devices_controller.h",
+      "media/webrtc/native_desktop_media_list.cc",
+      "media/webrtc/native_desktop_media_list.h",
+      "media/webrtc/permission_bubble_media_access_handler.cc",
+      "media/webrtc/permission_bubble_media_access_handler.h",
+      "media/webrtc/webrtc_event_log_history.cc",
+      "media/webrtc/webrtc_event_log_history.h",
+      "media/webrtc/webrtc_event_log_manager.cc",
+      "media/webrtc/webrtc_event_log_manager.h",
+      "media/webrtc/webrtc_event_log_manager_common.cc",
+      "media/webrtc/webrtc_event_log_manager_common.h",
+      "media/webrtc/webrtc_event_log_manager_keyed_service.cc",
+      "media/webrtc/webrtc_event_log_manager_keyed_service.h",
+      "media/webrtc/webrtc_event_log_manager_keyed_service_factory.cc",
+      "media/webrtc/webrtc_event_log_manager_keyed_service_factory.h",
+      "media/webrtc/webrtc_event_log_manager_local.cc",
+      "media/webrtc/webrtc_event_log_manager_local.h",
+      "media/webrtc/webrtc_event_log_manager_remote.cc",
+      "media/webrtc/webrtc_event_log_manager_remote.h",
+      "media/webrtc/webrtc_event_log_uploader.cc",
+      "media/webrtc/webrtc_event_log_uploader.h",
+      "media/webrtc/webrtc_log_buffer.cc",
+      "media/webrtc/webrtc_log_buffer.h",
+      "media/webrtc/webrtc_log_uploader.cc",
+      "media/webrtc/webrtc_log_uploader.h",
+      "media/webrtc/webrtc_log_util.cc",
+      "media/webrtc/webrtc_log_util.h",
+      "media/webrtc/webrtc_logging_controller.cc",
+      "media/webrtc/webrtc_logging_controller.h",
+      "media/webrtc/webrtc_rtp_dump_handler.cc",
+      "media/webrtc/webrtc_rtp_dump_handler.h",
+      "media/webrtc/webrtc_rtp_dump_writer.cc",
+      "media/webrtc/webrtc_rtp_dump_writer.h",
+      "media/webrtc/webrtc_text_log_handler.cc",
+      "media/webrtc/webrtc_text_log_handler.h",
+      "media/webrtc/window_icon_util.h",
+      "media/webrtc/window_icon_util_chromeos.cc",
+      "media/webrtc/window_icon_util_mac.mm",
+      "media/webrtc/window_icon_util_win.cc",
+    ]
+    deps += [
+      "//components/webrtc",
+      "//components/webrtc_logging/browser",
+      "//components/webrtc_logging/common",
+      "//content/public/browser",
+      "//services/audio/public/cpp",
+      "//third_party/webrtc_overrides:webrtc_component",
+    ]
+  }
+
   if (use_cups) {
     configs += [ "//printing:cups" ]
     if (is_chromeos) {
diff --git a/chromium/chrome/browser/extensions/BUILD.gn b/chromium/chrome/browser/extensions/BUILD.gn
index 18315b380b6..e083ef7e712 100644
--- a/chromium/chrome/browser/extensions/BUILD.gn
+++ b/chromium/chrome/browser/extensions/BUILD.gn
@@ -9,6 +9,7 @@ import("//chrome/common/features.gni")
 import("//chromeos/services/cfm/public/buildflags/buildflags.gni")
 import("//components/nacl/features.gni")
 import("//extensions/buildflags/buildflags.gni")
+import("//media/media_options.gni")
 import("//mojo/public/tools/bindings/mojom.gni")
 import("//pdf/features.gni")
 import("//rlz/buildflags/buildflags.gni")
@@ -408,7 +409,6 @@ static_library("extensions") {
     "api/webrtc_audio_private/webrtc_audio_private_api.h",
     "api/webrtc_desktop_capture_private/webrtc_desktop_capture_private_api.cc",
     "api/webrtc_desktop_capture_private/webrtc_desktop_capture_private_api.h",
-    "api/webrtc_logging_private/webrtc_logging_private_api.cc",
     "api/webrtc_logging_private/webrtc_logging_private_api.h",
     "api/webstore_private/extension_install_status.cc",
     "api/webstore_private/extension_install_status.h",
@@ -1237,6 +1237,13 @@ static_library("extensions") {
     deps += [ "//rlz:rlz_lib" ]
   }
 
+  if (enable_webrtc) {
+    sources += [ "api/webrtc_logging_private/webrtc_logging_private_api.cc" ]
+  } else {
+    sources +=
+        [ "api/webrtc_logging_private/webrtc_logging_private_api_stub.cc" ]
+  }
+
   if (toolkit_views) {
     deps += [ "//ui/views" ]
   }
diff --git a/chromium/chrome/browser/extensions/api/preference/preference_api.cc b/chromium/chrome/browser/extensions/api/preference/preference_api.cc
index 6f608df9995..2f1f7576673 100644
--- a/chromium/chrome/browser/extensions/api/preference/preference_api.cc
+++ b/chromium/chrome/browser/extensions/api/preference/preference_api.cc
@@ -129,6 +129,7 @@ const PrefMappingEntry kPrefMapping[] = {
      APIPermission::kPrivacy, APIPermission::kPrivacy},
     {"translationServiceEnabled", prefs::kOfferTranslateEnabled,
      APIPermission::kPrivacy, APIPermission::kPrivacy},
+#if BUILDFLAG(ENABLE_WEBRTC)
     // webRTCMultipleRoutesEnabled and webRTCNonProxiedUdpEnabled have been
     // replaced by webRTCIPHandlingPolicy. Leaving it for backward
     // compatibility. TODO(guoweis): Remove this in M50.
@@ -140,6 +141,7 @@ const PrefMappingEntry kPrefMapping[] = {
      APIPermission::kPrivacy, APIPermission::kPrivacy},
     {"webRTCUDPPortRange", prefs::kWebRTCUDPPortRange, APIPermission::kPrivacy,
      APIPermission::kPrivacy},
+#endif
     // accessibilityFeatures.animationPolicy is available for
     // all platforms but the others from accessibilityFeatures
     // is only available for OS_CHROMEOS.
diff --git a/chromium/chrome/browser/extensions/api/preference/preference_apitest.cc b/chromium/chrome/browser/extensions/api/preference/preference_apitest.cc
index a48dbdf5957..31dfba251f4 100644
--- a/chromium/chrome/browser/extensions/api/preference/preference_apitest.cc
+++ b/chromium/chrome/browser/extensions/api/preference/preference_apitest.cc
@@ -165,10 +165,12 @@ IN_PROC_BROWSER_TEST_F(ExtensionPreferenceApiTest, Standard) {
   prefs->SetBoolean(password_manager::prefs::kCredentialsEnableService, false);
   prefs->SetBoolean(prefs::kSafeBrowsingEnabled, false);
   prefs->SetBoolean(prefs::kSearchSuggestEnabled, false);
+#if BUILDFLAG(ENABLE_WEBRTC)
   prefs->SetBoolean(prefs::kWebRTCMultipleRoutesEnabled, false);
   prefs->SetBoolean(prefs::kWebRTCNonProxiedUdpEnabled, false);
   prefs->SetString(prefs::kWebRTCIPHandlingPolicy,
                    blink::kWebRTCIPHandlingDefaultPublicInterfaceOnly);
+#endif
 
   const char kExtensionPath[] = "preference/standard";
 
diff --git a/chromium/chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_private_api.h b/chromium/chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_private_api.h
index e371b984dd6..620daa5e736 100644
--- a/chromium/chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_private_api.h
+++ b/chromium/chrome/browser/extensions/api/webrtc_logging_private/webrtc_logging_private_api.h
@@ -7,12 +7,15 @@
 
 #include <string>
 
-#include "chrome/browser/media/webrtc/audio_debug_recordings_handler.h"
-#include "chrome/browser/media/webrtc/webrtc_logging_controller.h"
 #include "chrome/common/extensions/api/webrtc_logging_private.h"
 #include "extensions/browser/extension_function.h"
 #include "media/media_buildflags.h"
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "chrome/browser/media/webrtc/audio_debug_recordings_handler.h"
+#include "chrome/browser/media/webrtc/webrtc_logging_controller.h"
+#endif
+
 namespace content {
 
 class RenderProcessHost;
@@ -25,6 +28,7 @@ class WebrtcLoggingPrivateFunction : public ExtensionFunction {
  protected:
   ~WebrtcLoggingPrivateFunction() override {}
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Returns the RenderProcessHost associated with the given |request|
   // authorized by the |security_origin|. Returns null and sets |*error| to an
   // appropriate error if unauthorized or the RPH does not exist.
@@ -37,6 +41,7 @@ class WebrtcLoggingPrivateFunction : public ExtensionFunction {
       const api::webrtc_logging_private::RequestInfo& request,
       const std::string& security_origin,
       std::string* error);
+#endif
 };
 
 class WebrtcLoggingPrivateFunctionWithGenericCallback
@@ -44,6 +49,7 @@ class WebrtcLoggingPrivateFunctionWithGenericCallback
  protected:
   ~WebrtcLoggingPrivateFunctionWithGenericCallback() override {}
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Finds the appropriate logging controller for performing the task and
   // prepares a generic callback object for when the task is completed.  If the
   // logging controller can't be found for the given request+origin, the
@@ -57,6 +63,7 @@ class WebrtcLoggingPrivateFunctionWithGenericCallback
 
   // Must be called on UI thread.
   void FireCallback(bool success, const std::string& error_message);
+#endif
 };
 
 class WebrtcLoggingPrivateFunctionWithUploadCallback
@@ -64,9 +71,11 @@ class WebrtcLoggingPrivateFunctionWithUploadCallback
  protected:
   ~WebrtcLoggingPrivateFunctionWithUploadCallback() override {}
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Must be called on UI thread.
   void FireCallback(bool success, const std::string& report_id,
                     const std::string& error_message);
+#endif
 };
 
 class WebrtcLoggingPrivateFunctionWithRecordingDoneCallback
@@ -74,11 +83,13 @@ class WebrtcLoggingPrivateFunctionWithRecordingDoneCallback
  protected:
   ~WebrtcLoggingPrivateFunctionWithRecordingDoneCallback() override {}
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Must be called on UI thread.
   void FireErrorCallback(const std::string& error_message);
   void FireCallback(const std::string& prefix_path,
                     bool did_stop,
                     bool did_manual_stop);
+#endif
 };
 
 class WebrtcLoggingPrivateSetMetaDataFunction
@@ -283,10 +294,12 @@ class WebrtcLoggingPrivateGetLogsDirectoryFunction
   // ExtensionFunction overrides.
   ResponseAction Run() override;
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Must be called on UI thread.
   void FireErrorCallback(const std::string& error_message);
   void FireCallback(const std::string& filesystem_id,
                     const std::string& base_name);
+#endif
 };
 
 }  // namespace extensions
diff --git a/chromium/chrome/browser/ui/BUILD.gn b/chromium/chrome/browser/ui/BUILD.gn
index 5fe4e7e4722..83b55f3be57 100644
--- a/chromium/chrome/browser/ui/BUILD.gn
+++ b/chromium/chrome/browser/ui/BUILD.gn
@@ -20,6 +20,7 @@ import("//components/offline_pages/buildflags/features.gni")
 import("//components/signin/features.gni")
 import("//device/vr/buildflags/buildflags.gni")
 import("//extensions/buildflags/buildflags.gni")
+import("//media/media_options.gni")
 import("//ppapi/buildflags/buildflags.gni")
 import("//printing/buildflags/buildflags.gni")
 import("//rlz/buildflags/buildflags.gni")
@@ -242,8 +243,6 @@ static_library("ui") {
     "webui/media/media_engagement_ui.h",
     "webui/media/media_history_ui.cc",
     "webui/media/media_history_ui.h",
-    "webui/media/webrtc_logs_ui.cc",
-    "webui/media/webrtc_logs_ui.h",
     "webui/memory_internals_ui.cc",
     "webui/memory_internals_ui.h",
     "webui/metrics_handler.cc",
@@ -532,7 +531,6 @@ static_library("ui") {
     "//components/viz/host",
     "//components/web_cache/browser",
     "//components/web_resource",
-    "//components/webrtc_logging/browser",
     "//content/app/resources",
     "//content/browser/webrtc/resources",
     "//content/public/common",
@@ -4470,6 +4468,14 @@ static_library("ui") {
     }
   }
 
+  if (enable_webrtc) {
+    sources += [
+      "webui/media/webrtc_logs_ui.cc",
+      "webui/media/webrtc_logs_ui.h",
+    ]
+    deps += [ "//components/webrtc_logging/browser" ]
+  }
+
   if (safe_browsing_mode == 1) {
     deps += [
       "//chrome/browser/ui/webui/reset_password:mojo_bindings",
diff --git a/chromium/chrome/browser/ui/webui/chrome_web_ui_controller_factory.cc b/chromium/chrome/browser/ui/webui/chrome_web_ui_controller_factory.cc
index c9abbd97037..022c7c0f553 100644
--- a/chromium/chrome/browser/ui/webui/chrome_web_ui_controller_factory.cc
+++ b/chromium/chrome/browser/ui/webui/chrome_web_ui_controller_factory.cc
@@ -56,7 +56,6 @@
 #include "chrome/browser/ui/webui/media/media_engagement_ui.h"
 #include "chrome/browser/ui/webui/media/media_feeds_ui.h"
 #include "chrome/browser/ui/webui/media/media_history_ui.h"
-#include "chrome/browser/ui/webui/media/webrtc_logs_ui.h"
 #include "chrome/browser/ui/webui/memory_internals_ui.h"
 #include "chrome/browser/ui/webui/net_export_ui.h"
 #include "chrome/browser/ui/webui/net_internals/net_internals_ui.h"
@@ -110,6 +109,10 @@
 #include "chrome/browser/ui/webui/nacl_ui.h"
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "chrome/browser/ui/webui/media/webrtc_logs_ui.h"
+#endif
+
 #if BUILDFLAG(ENABLE_PRINT_PREVIEW)
 #include "chrome/browser/ui/webui/print_preview/print_preview_ui.h"
 #endif
@@ -857,8 +860,10 @@ WebUIFactoryFunction GetWebUIFactoryFunction(WebUI* web_ui,
   }
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (url.host_piece() == chrome::kChromeUIWebRtcLogsHost)
     return &NewWebUI<WebRtcLogsUI>;
+#endif
 #if !defined(OS_ANDROID)
   if (url.host_piece() == chrome::kChromeUIMediaRouterInternalsHost &&
       media_router::MediaRouterEnabled(profile)) {
diff --git a/chromium/chrome/common/extensions/api/BUILD.gn b/chromium/chrome/common/extensions/api/BUILD.gn
index 08951222cc6..a68fd0faec5 100644
--- a/chromium/chrome/common/extensions/api/BUILD.gn
+++ b/chromium/chrome/common/extensions/api/BUILD.gn
@@ -7,6 +7,7 @@ import("//build/config/ui.gni")
 import("//chrome/common/extensions/api/api_sources.gni")
 import("//chrome/common/features.gni")
 import("//extensions/buildflags/buildflags.gni")
+import("//media/media_options.gni")
 import("//tools/json_schema_compiler/json_features.gni")
 import("//tools/json_schema_compiler/json_schema_api.gni")
 
diff --git a/chromium/chrome/common/features.gni b/chromium/chrome/common/features.gni
index 2ff23e50e1a..86d9ce1e7a6 100644
--- a/chromium/chrome/common/features.gni
+++ b/chromium/chrome/common/features.gni
@@ -83,6 +83,7 @@ chrome_grit_defines = [
   "enable_supervised_users=$enable_supervised_users",
   "enable_tab_search=$enable_tab_search",
   "enable_vr=$enable_vr",
+  "enable_webrtc=$enable_webrtc",
   "enable_webui_tab_strip=$enable_webui_tab_strip",
   "safe_browsing_mode=$safe_browsing_mode",
   "optimize_webui=$optimize_webui",
diff --git a/chromium/chrome/common/webui_url_constants.cc b/chromium/chrome/common/webui_url_constants.cc
index e6687146475..2584f68f654 100644
--- a/chromium/chrome/common/webui_url_constants.cc
+++ b/chromium/chrome/common/webui_url_constants.cc
@@ -394,7 +394,9 @@ const char kChromeUITabSearchHost[] = "tab-search";
 const char kChromeUITabSearchURL[] = "chrome://tab-search/";
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 const char kChromeUIWebRtcLogsHost[] = "webrtc-logs";
+#endif
 
 // Settings sub pages.
 
@@ -572,7 +574,9 @@ const char* const kChromeHostURLs[] = {
 #if BUILDFLAG(ENABLE_SERVICE_DISCOVERY)
     kChromeUIDevicesHost,
 #endif
+#if BUILDFLAG(ENABLE_WEBRTC)
     kChromeUIWebRtcLogsHost,
+#endif
 };
 const size_t kNumberOfChromeHostURLs = base::size(kChromeHostURLs);
 
diff --git a/chromium/chrome/common/webui_url_constants.h b/chromium/chrome/common/webui_url_constants.h
index 23decba2688..a72ca753355 100644
--- a/chromium/chrome/common/webui_url_constants.h
+++ b/chromium/chrome/common/webui_url_constants.h
@@ -343,7 +343,9 @@ extern const char kChromeUITabSearchHost[];
 extern const char kChromeUITabSearchURL[];
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 extern const char kChromeUIWebRtcLogsHost[];
+#endif
 
 // Settings sub-pages.
 extern const char kAccessibilitySubPage[];
diff --git a/chromium/chrome/renderer/BUILD.gn b/chromium/chrome/renderer/BUILD.gn
index ce49b592498..ac3494de703 100644
--- a/chromium/chrome/renderer/BUILD.gn
+++ b/chromium/chrome/renderer/BUILD.gn
@@ -84,8 +84,6 @@ static_library("renderer") {
     "media/flash_embed_rewrite.h",
     "media/media_feeds.cc",
     "media/media_feeds.h",
-    "media/webrtc_logging_agent_impl.cc",
-    "media/webrtc_logging_agent_impl.h",
     "net/net_error_helper.cc",
     "net/net_error_helper.h",
     "net/net_error_helper_core.cc",
@@ -326,12 +324,22 @@ static_library("renderer") {
       "//components/guest_view/renderer",
       "//extensions:extensions_resources",
       "//extensions/renderer",
-      "//media/cast:net",
-      "//media/cast:receiver",
-      "//media/cast:sender",
       "//services/network/public/cpp",
     ]
     public_deps += [ "//ipc" ]
+    if (enable_webrtc) {
+      deps += [
+        "//media/cast:net",
+        "//media/cast:receiver",
+        "//media/cast:sender",
+      ]
+    }
+  }
+  if (enable_webrtc) {
+    sources += [
+      "media/webrtc_logging_agent_impl.cc",
+      "media/webrtc_logging_agent_impl.h",
+    ]
   }
   if (enable_spellcheck) {
     deps += [ "//components/spellcheck/renderer:renderer" ]
@@ -441,4 +449,11 @@ static_library("test_support") {
     "//testing/gmock",
     "//testing/gtest",
   ]
+
+  if (enable_webrtc) {
+    sources += [
+      "media/mock_webrtc_logging_message_filter.cc",
+      "media/mock_webrtc_logging_message_filter.h",
+    ]
+  }
 }
diff --git a/chromium/chrome/test/BUILD.gn b/chromium/chrome/test/BUILD.gn
index 184924ac716..9179faef50c 100644
--- a/chromium/chrome/test/BUILD.gn
+++ b/chromium/chrome/test/BUILD.gn
@@ -28,6 +28,7 @@ import("//components/os_crypt/features.gni")
 import("//components/signin/features.gni")
 import("//components/spellcheck/spellcheck_build_features.gni")
 import("//extensions/buildflags/buildflags.gni")
+import("//media/media_options.gni")
 import("//mojo/public/tools/bindings/mojom.gni")
 import("//net/features.gni")
 import("//ppapi/buildflags/buildflags.gni")
@@ -2817,7 +2818,34 @@ if (!is_android) {
     if (enable_captive_portal_detection) {
       sources += [ "../browser/captive_portal/captive_portal_browsertest.cc" ]
     }
-
+    if (!enable_webrtc) {
+      sources -= [
+        "../browser/media/webrtc/webrtc_apprtc_browsertest.cc",
+        "../browser/media/webrtc/webrtc_browsertest.cc",
+        "../browser/media/webrtc/webrtc_browsertest_base.cc",
+        "../browser/media/webrtc/webrtc_browsertest_base.h",
+        "../browser/media/webrtc/webrtc_browsertest_common.cc",
+        "../browser/media/webrtc/webrtc_browsertest_common.h",
+        "../browser/media/webrtc/webrtc_browsertest_perf.cc",
+        "../browser/media/webrtc/webrtc_browsertest_perf.h",
+        "../browser/media/webrtc/webrtc_desktop_capture_browsertest.cc",
+        "../browser/media/webrtc/webrtc_disable_encryption_flag_browsertest.cc",
+        "../browser/media/webrtc/webrtc_getdisplaymedia_browsertest.cc",
+        "../browser/media/webrtc/webrtc_getmediadevices_browsertest.cc",
+        "../browser/media/webrtc/webrtc_internals_integration_browsertest.cc",
+        "../browser/media/webrtc/webrtc_internals_perf_browsertest.cc",
+        "../browser/media/webrtc/webrtc_stats_perf_browsertest.cc",
+        "../browser/media/webrtc/webrtc_video_display_perf_browsertest.cc",
+        "../browser/media/webrtc/webrtc_video_quality_browsertest.cc",
+        "../browser/media/webrtc/webrtc_webcam_browsertest.cc",
+      ]
+      if (enable_extensions) {
+        sources -= [
+          "../browser/extensions/api/webrtc_audio_private/webrtc_audio_private_browsertest.cc",
+          "../browser/extensions/api/webrtc_logging_private/webrtc_logging_private_apitest.cc",
+        ]
+      }
+    }
     if (is_mac) {
       deps += [ "//third_party/ocmock" ]
       data_deps += [
@@ -4623,9 +4651,6 @@ test("unit_tests") {
       "../browser/devtools/protocol/cast_handler_unittest.cc",
       "../browser/devtools/serialize_host_descriptions_unittest.cc",
       "../browser/download/download_dir_policy_handler_unittest.cc",
-      "../browser/media/webrtc/webrtc_log_uploader_unittest.cc",
-      "../browser/media/webrtc/webrtc_rtp_dump_handler_unittest.cc",
-      "../browser/media/webrtc/webrtc_rtp_dump_writer_unittest.cc",
       "../browser/policy/local_sync_policy_handler_unittest.cc",
       "../browser/renderer_context_menu/mock_render_view_context_menu.cc",
       "../browser/renderer_context_menu/mock_render_view_context_menu.h",
@@ -4666,7 +4691,6 @@ test("unit_tests") {
       "../browser/ui/views/passwords/password_save_unsynced_credentials_locally_view_unittest.cc",
       "../browser/ui/views/passwords/password_save_update_with_account_store_view_unittest.cc",
       "../browser/ui/views/passwords/post_save_compromised_bubble_view_unittest.cc",
-      "../renderer/media/webrtc_logging_agent_impl_unittest.cc",
     ]
 
     deps += [ "//services/network:test_support" ]
@@ -5582,6 +5606,14 @@ test("unit_tests") {
       "../browser/sessions/tab_loader_unittest.cc",
     ]
   }
+  if (enable_webrtc) {
+    sources += [
+      "../browser/media/webrtc/webrtc_log_uploader_unittest.cc",
+      "../browser/media/webrtc/webrtc_rtp_dump_handler_unittest.cc",
+      "../browser/media/webrtc/webrtc_rtp_dump_writer_unittest.cc",
+      "../renderer/media/chrome_webrtc_log_message_delegate_unittest.cc",
+    ]
+  }
   if (is_chromeos) {
     deps += [
       "//chrome/browser/chromeos:unit_tests",
diff --git a/chromium/chrome/tools/service_discovery_sniffer/BUILD.gn b/chromium/chrome/tools/service_discovery_sniffer/BUILD.gn
index 4c45f5456b7..e73f7cfaa36 100644
--- a/chromium/chrome/tools/service_discovery_sniffer/BUILD.gn
+++ b/chromium/chrome/tools/service_discovery_sniffer/BUILD.gn
@@ -3,6 +3,7 @@
 # found in the LICENSE file.
 
 import("//build/config/features.gni")
+import("//media/media_options.gni")
 
 executable("service_discovery_sniffer") {
   testonly = true
@@ -16,6 +17,9 @@ executable("service_discovery_sniffer") {
     "//base/test:test_support",
     "//chrome/browser",
     "//net",
-    "//third_party/webrtc_overrides:webrtc_component",
   ]
+
+  if (enable_webrtc) {
+    deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+  }
 }
diff --git a/chromium/components/BUILD.gn b/chromium/components/BUILD.gn
index 6582ce9985b..aafc1d4fa5c 100644
--- a/chromium/components/BUILD.gn
+++ b/chromium/components/BUILD.gn
@@ -303,9 +303,13 @@ test("components_unittests") {
       "//components/web_cache/browser:unit_tests",
       "//components/web_package:unit_tests",
       "//components/webcrypto:unit_tests",
-      "//components/webrtc_logging/browser:unit_tests",
-      "//components/webrtc_logging/common:unit_tests",
     ]
+    if (enable_webrtc && !is_ios) {
+      deps += [
+        "//components/webrtc_logging/browser:unit_tests",
+        "//components/webrtc_logging/common:unit_tests",
+      ]
+    }
 
     if (!is_fuchsia) {  # !iOS and !Fuchsia
       deps += [
diff --git a/chromium/components/webrtc_logging/browser/BUILD.gn b/chromium/components/webrtc_logging/browser/BUILD.gn
index 3fe701dc181..a82ed798cd6 100644
--- a/chromium/components/webrtc_logging/browser/BUILD.gn
+++ b/chromium/components/webrtc_logging/browser/BUILD.gn
@@ -2,6 +2,10 @@
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
+import("//media/media_options.gni")
+
+assert(enable_webrtc)
+
 source_set("browser") {
   sources = [
     "log_cleanup.cc",
diff --git a/chromium/components/webrtc_logging/common/BUILD.gn b/chromium/components/webrtc_logging/common/BUILD.gn
index 1a47354693f..7eacccc034f 100644
--- a/chromium/components/webrtc_logging/common/BUILD.gn
+++ b/chromium/components/webrtc_logging/common/BUILD.gn
@@ -2,6 +2,10 @@
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
+import("//media/media_options.gni")
+
+assert(enable_webrtc)
+
 source_set("common") {
   sources = [
     "partial_circular_buffer.cc",
diff --git a/chromium/content/app/content_main_runner_impl.cc b/chromium/content/app/content_main_runner_impl.cc
index a021f7b288c..a8c7d4457da 100644
--- a/chromium/content/app/content_main_runner_impl.cc
+++ b/chromium/content/app/content_main_runner_impl.cc
@@ -139,7 +139,6 @@
 #include "third_party/boringssl/src/include/openssl/crypto.h"
 #include "third_party/skia/include/core/SkFontMgr.h"
 #include "third_party/skia/include/ports/SkFontMgr_android.h"
-#include "third_party/webrtc_overrides/init_webrtc.h"  // nogncheck
 
 #if BUILDFLAG(ENABLE_PLUGINS)
 #include "content/common/pepper_plugin_list.h"
@@ -161,8 +160,13 @@
 #include "content/public/common/zygote/sandbox_support_linux.h"
 #include "content/public/common/zygote/zygote_handle.h"
 #include "media/base/media_switches.h"
+
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "third_party/webrtc_overrides/init_webrtc.h"  // nogncheck
 #endif
 
+#endif // BUILDFLAG(USE_ZYGOTE_HANDLE)
+
 #if defined(OS_ANDROID)
 #include "content/browser/android/browser_startup_controller.h"
 #include "content/common/android/cpu_affinity.h"
@@ -357,7 +361,9 @@ void PreSandboxInit() {
   // Ensure access to the library CDMs before the sandbox is turned on.
   PreloadLibraryCdms();
 #endif
+#if BUILDFLAG(ENABLE_WEBRTC)
   InitializeWebRtcModule();
+#endif
 
   // Set the android SkFontMgr for blink. We need to ensure this is done
   // before the sandbox is initialized to allow the font manager to access
diff --git a/chromium/content/browser/BUILD.gn b/chromium/content/browser/BUILD.gn
index 8bbc49e20e9..949eea2f10b 100644
--- a/chromium/content/browser/BUILD.gn
+++ b/chromium/content/browser/BUILD.gn
@@ -118,7 +118,6 @@ source_set("browser") {
     "//content/browser/process_internals:mojo_bindings",
     "//content/browser/resources/media:media_internals_resources",
     "//content/browser/service_worker:service_worker_proto",
-    "//content/browser/webrtc/resources",
     "//content/common",
     "//content/common:buildflags",
     "//content/common:mojo_bindings",
@@ -197,10 +196,6 @@ source_set("browser") {
     "//services/tracing/public/cpp:traced_process",
     "//services/tracing/public/cpp/background_tracing",
     "//services/tracing/public/mojom",
-    "//services/video_capture:lib",
-    "//services/video_capture/public/cpp",
-    "//services/video_capture/public/mojom:constants",
-    "//services/video_capture/public/uma",
     "//services/viz/privileged/mojom",
     "//services/viz/public/cpp/gpu",
     "//services/viz/public/mojom",
@@ -227,7 +222,6 @@ source_set("browser") {
     "//third_party/libyuv",
     "//third_party/re2",
     "//third_party/sqlite",
-    "//third_party/webrtc_overrides:webrtc_component",
     "//third_party/zlib",
     "//third_party/zlib/google:zip",
     "//ui/accessibility",
@@ -1105,8 +1099,6 @@ source_set("browser") {
     "loader/single_request_url_loader_factory.cc",
     "loader/single_request_url_loader_factory.h",
     "loader/url_loader_throttles.cc",
-    "loader/webrtc_connections_observer.cc",
-    "loader/webrtc_connections_observer.h",
     "locks/lock_manager.cc",
     "locks/lock_manager.h",
     "log_console_message.cc",
@@ -1504,8 +1496,6 @@ source_set("browser") {
     "renderer_host/keep_alive_handle_factory.h",
     "renderer_host/legacy_render_widget_host_win.cc",
     "renderer_host/legacy_render_widget_host_win.h",
-    "renderer_host/media/aec_dump_manager_impl.cc",
-    "renderer_host/media/aec_dump_manager_impl.h",
     "renderer_host/media/audio_input_device_manager.cc",
     "renderer_host/media/audio_input_device_manager.h",
     "renderer_host/media/audio_input_stream_handle.cc",
@@ -1537,20 +1527,10 @@ source_set("browser") {
     "renderer_host/media/media_stream_track_metrics_host.h",
     "renderer_host/media/media_stream_ui_proxy.cc",
     "renderer_host/media/media_stream_ui_proxy.h",
-    "renderer_host/media/peer_connection_tracker_host.cc",
-    "renderer_host/media/peer_connection_tracker_host.h",
-    "renderer_host/media/ref_counted_video_source_provider.cc",
-    "renderer_host/media/ref_counted_video_source_provider.h",
     "renderer_host/media/render_frame_audio_input_stream_factory.cc",
     "renderer_host/media/render_frame_audio_input_stream_factory.h",
     "renderer_host/media/render_frame_audio_output_stream_factory.cc",
     "renderer_host/media/render_frame_audio_output_stream_factory.h",
-    "renderer_host/media/service_launched_video_capture_device.cc",
-    "renderer_host/media/service_launched_video_capture_device.h",
-    "renderer_host/media/service_video_capture_device_launcher.cc",
-    "renderer_host/media/service_video_capture_device_launcher.h",
-    "renderer_host/media/service_video_capture_provider.cc",
-    "renderer_host/media/service_video_capture_provider.h",
     "renderer_host/media/video_capture_controller.cc",
     "renderer_host/media/video_capture_controller.h",
     "renderer_host/media/video_capture_controller_event_handler.h",
@@ -1560,10 +1540,6 @@ source_set("browser") {
     "renderer_host/media/video_capture_manager.cc",
     "renderer_host/media/video_capture_manager.h",
     "renderer_host/media/video_capture_provider.h",
-    "renderer_host/media/video_capture_provider_switcher.cc",
-    "renderer_host/media/video_capture_provider_switcher.h",
-    "renderer_host/media/virtual_video_capture_devices_changed_observer.cc",
-    "renderer_host/media/virtual_video_capture_devices_changed_observer.h",
     "renderer_host/mixed_content_navigation_throttle.cc",
     "renderer_host/mixed_content_navigation_throttle.h",
     "renderer_host/native_web_keyboard_event_aura.cc",
@@ -1590,8 +1566,6 @@ source_set("browser") {
     "renderer_host/overscroll_controller.cc",
     "renderer_host/overscroll_controller.h",
     "renderer_host/overscroll_controller_delegate.h",
-    "renderer_host/p2p/socket_dispatcher_host.cc",
-    "renderer_host/p2p/socket_dispatcher_host.h",
     "renderer_host/page_lifecycle_state_manager.cc",
     "renderer_host/page_lifecycle_state_manager.h",
     "renderer_host/popup_menu_helper_mac.h",
@@ -2007,14 +1981,6 @@ source_set("browser") {
     "webauth/is_uvpaa.cc",
     "webauth/webauth_request_security_checker.cc",
     "webauth/webauth_request_security_checker.h",
-    "webrtc/webrtc_internals.cc",
-    "webrtc/webrtc_internals.h",
-    "webrtc/webrtc_internals_connections_observer.h",
-    "webrtc/webrtc_internals_message_handler.cc",
-    "webrtc/webrtc_internals_message_handler.h",
-    "webrtc/webrtc_internals_ui.cc",
-    "webrtc/webrtc_internals_ui.h",
-    "webrtc/webrtc_internals_ui_observer.h",
     "websockets/websocket_connector_impl.cc",
     "websockets/websocket_connector_impl.h",
     "websockets/websocket_handshake_request_info_impl.cc",
@@ -2175,12 +2141,52 @@ source_set("browser") {
     ]
   }
 
+  if (enable_webrtc) {
+    sources += [
+      "loader/webrtc_connections_observer.cc",
+      "loader/webrtc_connections_observer.h",
+      "renderer_host/media/aec_dump_manager_impl.cc",
+      "renderer_host/media/aec_dump_manager_impl.h",
+      "renderer_host/media/ref_counted_video_source_provider.cc",
+      "renderer_host/media/ref_counted_video_source_provider.h",
+      "renderer_host/media/peer_connection_tracker_host.cc",
+      "renderer_host/media/peer_connection_tracker_host.h",
+      "renderer_host/media/service_launched_video_capture_device.cc",
+      "renderer_host/media/service_launched_video_capture_device.h",
+      "renderer_host/media/service_video_capture_device_launcher.cc",
+      "renderer_host/media/service_video_capture_device_launcher.h",
+      "renderer_host/media/service_video_capture_provider.cc",
+      "renderer_host/media/service_video_capture_provider.h",
+      "renderer_host/media/video_capture_provider_switcher.cc",
+      "renderer_host/media/video_capture_provider_switcher.h",
+      "renderer_host/media/virtual_video_capture_devices_changed_observer.cc",
+      "renderer_host/media/virtual_video_capture_devices_changed_observer.h",
+      "renderer_host/p2p/socket_dispatcher_host.cc",
+      "renderer_host/p2p/socket_dispatcher_host.h",
+      "webrtc/webrtc_internals.cc",
+      "webrtc/webrtc_internals.h",
+      "webrtc/webrtc_internals_message_handler.cc",
+      "webrtc/webrtc_internals_message_handler.h",
+      "webrtc/webrtc_internals_ui.cc",
+      "webrtc/webrtc_internals_ui.h",
+      "webrtc/webrtc_internals_ui_observer.h",
+    ]
+
+    deps += [
+      "//content/browser/webrtc/resources",
+      "//jingle:jingle_glue",
+      "//services/video_capture:lib",
+      "//services/video_capture/public/cpp",
+      "//services/video_capture/public/mojom:constants",
+      "//services/video_capture/public/uma",
+      "//third_party/webrtc_overrides:webrtc_component",
+    ]
+  }
+
   # Desktop/Window/WebContents screen capture implementations, conditionally
   # built depending on the available implementations for each platform.
   if (enable_screen_capture && !is_android) {
     sources += [
-      "media/capture/desktop_capture_device.cc",
-      "media/capture/desktop_capture_device.h",
       "media/capture/frame_sink_video_capture_device.cc",
       "media/capture/frame_sink_video_capture_device.h",
       "media/capture/mouse_cursor_overlay_controller.cc",
@@ -2188,7 +2194,6 @@ source_set("browser") {
       "media/capture/web_contents_video_capture_device.cc",
       "media/capture/web_contents_video_capture_device.h",
     ]
-    deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
     if (use_aura) {
       sources += [
         "media/capture/aura_window_video_capture_device.cc",
@@ -2222,6 +2227,13 @@ source_set("browser") {
         "//chromeos/lacros",
       ]
     }
+    if (enable_webrtc) {
+      sources += [
+        "media/capture/desktop_capture_device.cc",
+        "media/capture/desktop_capture_device.h",
+      ]
+      deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+    }
   }
 
   if (is_win) {
diff --git a/chromium/content/browser/browser_main_loop.cc b/chromium/content/browser/browser_main_loop.cc
index dce878e0f11..4826e0c49ec 100644
--- a/chromium/content/browser/browser_main_loop.cc
+++ b/chromium/content/browser/browser_main_loop.cc
@@ -92,7 +92,6 @@
 #include "content/browser/tracing/background_tracing_manager_impl.h"
 #include "content/browser/tracing/tracing_controller_impl.h"
 #include "content/browser/utility_process_host.h"
-#include "content/browser/webrtc/webrtc_internals.h"
 #include "content/browser/webui/content_web_ui_controller_factory.h"
 #include "content/browser/webui/url_data_manager.h"
 #include "content/common/content_switches_internal.h"
@@ -224,6 +223,10 @@
 #include "content/browser/media/cdm_registry_impl.h"
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "content/browser/webrtc/webrtc_internals.h"
+#endif
+
 #if defined(USE_X11)
 #include "gpu/config/gpu_driver_bug_workaround_type.h"
 #include "ui/base/ui_base_features.h"
@@ -1307,12 +1310,14 @@ int BrowserMainLoop::BrowserThreadsStarted() {
       new media::DeviceMonitorMac(base::ThreadTaskRunnerHandle::Get()));
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Instantiated once using CreateSingletonInstance(), and accessed only using
   // GetInstance(), which is not allowed to create the object. This allows us
   // to ensure that it cannot be used before objects it relies on have been
   // created; namely, WebRtcEventLogManager.
   // Allowed to leak when the browser exits.
   WebRTCInternals::CreateSingletonInstance();
+#endif
 
   // MediaStreamManager needs the IO thread to be created.
   {
diff --git a/chromium/content/browser/gpu/gpu_process_host.cc b/chromium/content/browser/gpu/gpu_process_host.cc
index 30e7a3e1c65..eec6c13f39d 100644
--- a/chromium/content/browser/gpu/gpu_process_host.cc
+++ b/chromium/content/browser/gpu/gpu_process_host.cc
@@ -240,7 +240,9 @@ static const char* const kSwitchNames[] = {
     switches::kDisableLogging,
     switches::kDisableShaderNameHashing,
     switches::kDisableSkiaRuntimeOpts,
+#if BUILDFLAG(ENABLE_WEBRTC)
     switches::kDisableWebRtcHWEncoding,
+#endif
     switches::kEnableGpuRasterization,
     switches::kEnableLogging,
     switches::kEnableDeJelly,
diff --git a/chromium/content/browser/network_service_client.cc b/chromium/content/browser/network_service_client.cc
index 172b799c866..8b2b8418272 100644
--- a/chromium/content/browser/network_service_client.cc
+++ b/chromium/content/browser/network_service_client.cc
@@ -86,10 +86,12 @@ NetworkServiceClient::NetworkServiceClient(
 #endif
   }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   webrtc_connections_observer_ =
       std::make_unique<content::WebRtcConnectionsObserver>(base::BindRepeating(
           &NetworkServiceClient::OnPeerToPeerConnectionsCountChange,
           base::Unretained(this)));
+#endif
 }
 
 NetworkServiceClient::~NetworkServiceClient() {
diff --git a/chromium/content/browser/network_service_client.h b/chromium/content/browser/network_service_client.h
index ca7cb5a7ee8..fcb21195586 100644
--- a/chromium/content/browser/network_service_client.h
+++ b/chromium/content/browser/network_service_client.h
@@ -14,6 +14,7 @@
 #include "base/unguessable_token.h"
 #include "build/build_config.h"
 #include "content/common/content_export.h"
+#include "media/media_buildflags.h"
 #include "mojo/public/cpp/bindings/pending_receiver.h"
 #include "mojo/public/cpp/bindings/receiver.h"
 #include "mojo/public/cpp/bindings/remote.h"
@@ -114,7 +115,9 @@ class CONTENT_EXPORT NetworkServiceClient
 
   std::unique_ptr<base::MemoryPressureListener> memory_pressure_listener_;
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   std::unique_ptr<WebRtcConnectionsObserver> webrtc_connections_observer_;
+#endif
 
 #if defined(OS_ANDROID)
   std::unique_ptr<base::android::ApplicationStatusListener>
diff --git a/chromium/content/browser/renderer_host/media/in_process_launched_video_capture_device.cc b/chromium/content/browser/renderer_host/media/in_process_launched_video_capture_device.cc
index 5631156ad6f..daa53471d4b 100644
--- a/chromium/content/browser/renderer_host/media/in_process_launched_video_capture_device.cc
+++ b/chromium/content/browser/renderer_host/media/in_process_launched_video_capture_device.cc
@@ -155,7 +155,8 @@ void InProcessLaunchedVideoCaptureDevice::
                                             gfx::NativeViewId window_id,
                                             base::OnceClosure done_cb) {
   DCHECK(device_task_runner_->BelongsToCurrentThread());
-#if defined(ENABLE_SCREEN_CAPTURE) && !defined(OS_ANDROID)
+#if defined(ENABLE_SCREEN_CAPTURE) && BUILDFLAG(ENABLE_WEBRTC) && \
+    !defined(OS_ANDROID)
   DesktopCaptureDevice* desktop_device =
       static_cast<DesktopCaptureDevice*>(device);
   desktop_device->SetNotificationWindowId(window_id);
diff --git a/chromium/content/browser/renderer_host/media/in_process_video_capture_device_launcher.cc b/chromium/content/browser/renderer_host/media/in_process_video_capture_device_launcher.cc
index 6a6009a8afb..d88ed5268c8 100644
--- a/chromium/content/browser/renderer_host/media/in_process_video_capture_device_launcher.cc
+++ b/chromium/content/browser/renderer_host/media/in_process_video_capture_device_launcher.cc
@@ -38,7 +38,9 @@
 #if defined(USE_AURA)
 #include "content/browser/media/capture/aura_window_video_capture_device.h"
 #endif
+#if BUILDFLAG(ENABLE_WEBRTC)
 #include "content/browser/media/capture/desktop_capture_device.h"
+#endif
 #endif  // defined(OS_ANDROID)
 #endif  // BUILDFLAG(ENABLE_SCREEN_CAPTURE)
 
@@ -318,8 +320,10 @@ void InProcessVideoCaptureDeviceLauncher::DoStartDeviceCaptureOnDeviceThread(
   DCHECK(device_task_runner_->BelongsToCurrentThread());
   DCHECK(video_capture_system_);
 
-  std::unique_ptr<media::VideoCaptureDevice> video_capture_device =
-      video_capture_system_->CreateDevice(device_id);
+  std::unique_ptr<media::VideoCaptureDevice> video_capture_device;
+#if BUILDFLAG(ENABLE_WEBRTC)
+  video_capture_device = video_capture_system_->CreateDevice(device_id);
+#endif
 
   if (video_capture_device)
     video_capture_device->AllocateAndStart(params, std::move(device_client));
@@ -394,7 +398,7 @@ void InProcessVideoCaptureDeviceLauncher::DoStartDesktopCaptureOnDeviceThread(
   std::unique_ptr<media::VideoCaptureDevice> video_capture_device;
 #if defined(OS_ANDROID)
   video_capture_device = std::make_unique<ScreenCaptureDeviceAndroid>();
-#else
+#elif BUILDFLAG(ENABLE_WEBRTC)
   if (!video_capture_device)
     video_capture_device = DesktopCaptureDevice::Create(desktop_id);
 #endif  // defined (OS_ANDROID)
diff --git a/chromium/content/browser/renderer_host/media/media_stream_manager.cc b/chromium/content/browser/renderer_host/media/media_stream_manager.cc
index 472b3f879b5..884eb393ba0 100644
--- a/chromium/content/browser/renderer_host/media/media_stream_manager.cc
+++ b/chromium/content/browser/renderer_host/media/media_stream_manager.cc
@@ -760,6 +760,7 @@ MediaStreamManager::MediaStreamManager(
     }
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
     if (base::FeatureList::IsEnabled(features::kMojoVideoCapture)) {
       video_capture_provider = std::make_unique<VideoCaptureProviderSwitcher>(
           std::make_unique<ServiceVideoCaptureProvider>(
@@ -767,7 +768,10 @@ MediaStreamManager::MediaStreamManager(
           InProcessVideoCaptureProvider::CreateInstanceForNonDeviceCapture(
               std::move(device_task_runner),
               base::BindRepeating(&SendVideoCaptureLogMessage)));
-    } else {
+    } else
+#endif
+    {
+
       video_capture::uma::LogVideoCaptureServiceEvent(
           video_capture::uma::BROWSER_USING_LEGACY_CAPTURE);
       video_capture_provider = InProcessVideoCaptureProvider::CreateInstance(
diff --git a/chromium/content/browser/renderer_host/render_process_host_browsertest.cc b/chromium/content/browser/renderer_host/render_process_host_browsertest.cc
index f4e62a4ce67..202a6ccfebe 100644
--- a/chromium/content/browser/renderer_host/render_process_host_browsertest.cc
+++ b/chromium/content/browser/renderer_host/render_process_host_browsertest.cc
@@ -946,6 +946,19 @@ class CaptureStreamRenderProcessHostTest : public RenderProcessHostTest {
   }
 };
 
+// These tests contain WebRTC calls and cannot be run when it isn't enabled.
+#if !BUILDFLAG(ENABLE_WEBRTC)
+#define GetUserMediaIncrementsVideoCaptureStreams \
+  DISABLED_GetUserMediaIncrementsVideoCaptureStreams
+#define StopResetsVideoCaptureStreams DISABLED_StopResetsVideoCaptureStreams
+#define KillProcessZerosVideoCaptureStreams \
+  DISABLED_KillProcessZerosVideoCaptureStreams
+#define GetUserMediaAudioOnlyIncrementsMediaStreams \
+  DISABLED_GetUserMediaAudioOnlyIncrementsMediaStreams
+#define KillProcessZerosAudioCaptureStreams \
+  DISABLED_KillProcessZerosAudioCaptureStreams
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
+
 // Tests that video capture stream count increments when getUserMedia() is
 // called.
 IN_PROC_BROWSER_TEST_F(CaptureStreamRenderProcessHostTest,
diff --git a/chromium/content/browser/renderer_host/render_process_host_impl.cc b/chromium/content/browser/renderer_host/render_process_host_impl.cc
index 3a2941e4e77..c72e548289e 100644
--- a/chromium/content/browser/renderer_host/render_process_host_impl.cc
+++ b/chromium/content/browser/renderer_host/render_process_host_impl.cc
@@ -118,10 +118,8 @@
 #include "content/browser/renderer_host/embedded_frame_sink_provider_impl.h"
 #include "content/browser/renderer_host/file_utilities_host_impl.h"
 #include "content/browser/renderer_host/media/media_stream_manager.h"
-#include "content/browser/renderer_host/media/media_stream_track_metrics_host.h"
 #include "content/browser/renderer_host/media/peer_connection_tracker_host.h"
 #include "content/browser/renderer_host/media/video_capture_host.h"
-#include "content/browser/renderer_host/p2p/socket_dispatcher_host.h"
 #include "content/browser/renderer_host/pepper/pepper_message_filter.h"
 #include "content/browser/renderer_host/pepper/pepper_renderer_connection.h"
 #include "content/browser/renderer_host/plugin_registry_impl.h"
@@ -166,7 +164,6 @@
 #include "content/public/browser/resource_context.h"
 #include "content/public/browser/resource_coordinator_service.h"
 #include "content/public/browser/site_isolation_policy.h"
-#include "content/public/browser/webrtc_log.h"
 #include "content/public/common/child_process_host.h"
 #include "content/public/common/content_client.h"
 #include "content/public/common/content_constants.h"
@@ -276,6 +273,13 @@
 #include "content/browser/net/reporting_service_proxy.h"
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "content/browser/renderer_host/media/media_stream_track_metrics_host.h"
+#include "content/browser/renderer_host/p2p/socket_dispatcher_host.h"
+#include "content/browser/webrtc/webrtc_internals.h"
+#include "content/public/browser/webrtc_log.h"
+#endif
+
 #if BUILDFLAG(USE_MINIKIN_HYPHENATION)
 #include "content/browser/hyphenation/hyphenation_impl.h"
 #endif
@@ -1994,8 +1998,10 @@ void RenderProcessHostImpl::CreateMessageFilters() {
   AddFilter(new PepperRendererConnection(GetID()));
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   p2p_socket_dispatcher_host_ =
       std::make_unique<P2PSocketDispatcherHost>(GetID());
+#endif
 
   scoped_refptr<ServiceWorkerContextWrapper> service_worker_context(
       static_cast<ServiceWorkerContextWrapper*>(
@@ -2255,11 +2261,15 @@ void RenderProcessHostImpl::DumpProfilingData(base::OnceClosure callback) {
 
 PeerConnectionTrackerHost*
 RenderProcessHostImpl::GetPeerConnectionTrackerHost() {
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (!peer_connection_tracker_host_) {
     peer_connection_tracker_host_ =
         std::make_unique<PeerConnectionTrackerHost>(this);
   }
   return peer_connection_tracker_host_.get();
+#else
+  return nullptr;
+#endif
 }
 
 // static
@@ -2429,14 +2439,17 @@ void RenderProcessHostImpl::RegisterMojoInterfaces() {
       base::ThreadPool::CreateSequencedTaskRunner(
           {base::MayBlock(), base::TaskPriority::USER_VISIBLE}));
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Note, the base::Unretained() is safe because the target object has an IO
   // thread deleter and the callback is also targeting the IO thread.  When
   // the RPHI is destroyed it also triggers the destruction of the registry
   // on the IO thread.
   media_stream_track_metrics_host_.reset(new MediaStreamTrackMetricsHost());
+
   registry->AddInterface(base::BindRepeating(
       &MediaStreamTrackMetricsHost::BindReceiver,
       base::Unretained(media_stream_track_metrics_host_.get())));
+#endif
 
   AddUIThreadInterface(
       registry.get(),
@@ -2447,10 +2460,12 @@ void RenderProcessHostImpl::RegisterMojoInterfaces() {
   registry->AddInterface(
       base::BindRepeating(&metrics::CreateSingleSampleMetricsProvider));
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   AddUIThreadInterface(
       registry.get(),
       base::BindRepeating(&RenderProcessHostImpl::BindPeerConnectionTrackerHost,
                           weak_factory_.GetWeakPtr()));
+#endif
 
   AddUIThreadInterface(
       registry.get(),
@@ -2463,10 +2478,12 @@ void RenderProcessHostImpl::RegisterMojoInterfaces() {
       base::BindRepeating(&CreateReportingServiceProxy, GetID()));
 #endif  // BUILDFLAG(ENABLE_REPORTING)
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   AddUIThreadInterface(
       registry.get(),
       base::BindRepeating(&RenderProcessHostImpl::BindP2PSocketManager,
                           weak_factory_.GetWeakPtr()));
+#endif
 
   AddUIThreadInterface(
       registry.get(),
@@ -2526,10 +2543,12 @@ void RenderProcessHostImpl::RegisterMojoInterfaces() {
       base::BindRepeating(&RenderProcessHostImpl::BindVideoDecoderService,
                           weak_factory_.GetWeakPtr()));
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   AddUIThreadInterface(
       registry.get(),
       base::BindRepeating(&RenderProcessHostImpl::BindAecDumpManager,
                           weak_factory_.GetWeakPtr()));
+#endif
 
   // ---- Please do not register interfaces below this line ------
   //
@@ -2651,10 +2670,13 @@ void RenderProcessHostImpl::BindWebDatabaseHostImpl(
       base::BindOnce(&WebDatabaseHostImpl::Create, GetID(),
                      base::WrapRefCounted(db_tracker), std::move(receiver)));
 }
+
+#if BUILDFLAG(ENABLE_WEBRTC)
 void RenderProcessHostImpl::BindAecDumpManager(
     mojo::PendingReceiver<blink::mojom::AecDumpManager> receiver) {
   aec_dump_manager_.AddReceiver(std::move(receiver));
 }
+#endif
 
 void RenderProcessHostImpl::CreateOneShotSyncService(
     mojo::PendingReceiver<blink::mojom::OneShotBackgroundSyncService>
@@ -2677,10 +2699,12 @@ void RenderProcessHostImpl::BindPushMessagingManager(
   push_messaging_manager_->AddPushMessagingReceiver(std::move(receiver));
 }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 void RenderProcessHostImpl::BindP2PSocketManager(
     mojo::PendingReceiver<network::mojom::P2PSocketManager> receiver) {
   p2p_socket_dispatcher_host_->BindReceiver(std::move(receiver));
 }
+#endif
 
 void RenderProcessHostImpl::CreateMediaLogRecordHost(
     mojo::PendingReceiver<content::mojom::MediaInternalLogRecords> receiver) {
@@ -3469,6 +3493,7 @@ void RenderProcessHostImpl::PropagateBrowserCommandLineToRenderer(
 #if BUILDFLAG(ENABLE_PLUGINS)
     switches::kEnablePepperTesting,
 #endif
+#if BUILDFLAG(ENABLE_WEBRTC)
     switches::kDisableWebRtcHWDecoding,
     switches::kDisableWebRtcHWEncoding,
     switches::kEnableWebRtcSrtpAesGcm,
@@ -3476,6 +3501,7 @@ void RenderProcessHostImpl::PropagateBrowserCommandLineToRenderer(
     switches::kEnableWebRtcStunOrigin,
     switches::kEnforceWebRtcIPPermissionCheck,
     switches::kWebRtcMaxCaptureFramerate,
+#endif
     switches::kEnableLowEndDeviceMode,
     switches::kDisableLowEndDeviceMode,
     switches::kDisallowNonExactResourceReuse,
@@ -3504,6 +3530,7 @@ void RenderProcessHostImpl::PropagateBrowserCommandLineToRenderer(
   BrowserChildProcessHostImpl::CopyFeatureAndFieldTrialFlags(renderer_cmd);
   BrowserChildProcessHostImpl::CopyTraceStartupFlags(renderer_cmd);
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Only run the Stun trials in the first renderer.
   if (!has_done_stun_trials &&
       browser_cmd.HasSwitch(switches::kWebRtcStunProbeTrialParameter)) {
@@ -3513,6 +3540,7 @@ void RenderProcessHostImpl::PropagateBrowserCommandLineToRenderer(
         browser_cmd.GetSwitchValueASCII(
             switches::kWebRtcStunProbeTrialParameter));
   }
+#endif
 
   // Disable databases in incognito mode.
   if (GetBrowserContext()->IsOffTheRecord() &&
@@ -3847,11 +3875,13 @@ void RenderProcessHostImpl::Cleanup() {
                                     "render_process_host", this,
                                     "browser_context", browser_context_);
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (is_initialized_) {
     GetIOThreadTaskRunner({})->PostTask(
         FROM_HERE,
         base::BindOnce(&WebRtcLog::ClearLogMessageCallback, GetID()));
   }
+#endif
 
   if (!keep_alive_start_time_.is_null()) {
     UMA_HISTOGRAM_LONG_TIMES("BrowserRenderProcessHost.KeepAliveDuration",
@@ -3982,6 +4012,7 @@ void RenderProcessHostImpl::FilterURL(bool empty_allowed, GURL* url) {
   FilterURL(this, empty_allowed, url);
 }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 void RenderProcessHostImpl::EnableAudioDebugRecordings(
     const base::FilePath& file_path) {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
@@ -4014,6 +4045,7 @@ void RenderProcessHostImpl::EnableWebRtcEventLogOutput(int lid,
 void RenderProcessHostImpl::DisableWebRtcEventLogOutput(int lid) {
   GetPeerConnectionTrackerHost()->StopEventLog(lid);
 }
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
 
 IPC::ChannelProxy* RenderProcessHostImpl::GetChannel() {
   return channel_.get();
@@ -5040,8 +5072,10 @@ void RenderProcessHostImpl::OnProcessLaunched() {
 #endif
   }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   aec_dump_manager_.set_pid(GetProcess().Pid());
   aec_dump_manager_.AutoStart();
+#endif
 
   tracing_registration_ = TracingServiceController::Get().RegisterClient(
       GetProcess().Pid(),
@@ -5136,10 +5170,12 @@ void RenderProcessHostImpl::CreateAgentMetricsCollectorHost(
   }
 }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 void RenderProcessHostImpl::BindPeerConnectionTrackerHost(
     mojo::PendingReceiver<blink::mojom::PeerConnectionTrackerHost> receiver) {
   GetPeerConnectionTrackerHost()->BindReceiver(std::move(receiver));
 }
+#endif
 
 #if BUILDFLAG(ENABLE_MDNS)
 void RenderProcessHostImpl::CreateMdnsResponder(
diff --git a/chromium/content/browser/renderer_host/render_process_host_impl.h b/chromium/content/browser/renderer_host/render_process_host_impl.h
index 008bb38131d..d28b50b5a67 100644
--- a/chromium/content/browser/renderer_host/render_process_host_impl.h
+++ b/chromium/content/browser/renderer_host/render_process_host_impl.h
@@ -114,8 +114,6 @@ class CodeCacheHostImpl;
 class FileSystemManagerImpl;
 class InProcessChildThreadParams;
 class IsolationContext;
-class MediaStreamTrackMetricsHost;
-class P2PSocketDispatcherHost;
 class PermissionServiceContext;
 class PeerConnectionTrackerHost;
 class PluginRegistryImpl;
@@ -134,6 +132,11 @@ class StoragePartition;
 class StoragePartitionImpl;
 struct ChildProcessTerminationInfo;
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+class MediaStreamTrackMetricsHost;
+class P2PSocketDispatcherHost;
+#endif
+
 typedef base::Thread* (*RendererMainThreadFactoryFunction)(
     const InProcessChildThreadParams& params,
     int32_t renderer_client_id);
@@ -234,6 +237,7 @@ class CONTENT_EXPORT RenderProcessHostImpl
   bool FastShutdownStarted() override;
   base::TimeDelta GetChildProcessIdleTime() override;
   void FilterURL(bool empty_allowed, GURL* url) override;
+#if BUILDFLAG(ENABLE_WEBRTC)
   void EnableAudioDebugRecordings(const base::FilePath& file) override;
   void DisableAudioDebugRecordings() override;
   WebRtcStopRtpDumpCallback StartRtpDump(
@@ -242,6 +246,7 @@ class CONTENT_EXPORT RenderProcessHostImpl
       WebRtcRtpPacketCallback packet_callback) override;
   void EnableWebRtcEventLogOutput(int lid, int output_period_ms) override;
   void DisableWebRtcEventLogOutput(int lid) override;
+#endif
   void BindReceiver(mojo::GenericPendingReceiver receiver) override;
   std::unique_ptr<base::PersistentMemoryAllocator> TakeMetricsAllocator()
       override;
@@ -799,12 +804,16 @@ class CONTENT_EXPORT RenderProcessHostImpl
       mojo::PendingReceiver<media::mojom::InterfaceFactory> receiver);
   void BindWebDatabaseHostImpl(
       mojo::PendingReceiver<blink::mojom::WebDatabaseHost> receiver);
+#if BUILDFLAG(ENABLE_WEBRTC)
   void BindAecDumpManager(
       mojo::PendingReceiver<blink::mojom::AecDumpManager> receiver);
+#endif
   void BindPushMessagingManager(
       mojo::PendingReceiver<blink::mojom::PushMessaging> receiver);
+#if BUILDFLAG(ENABLE_WEBRTC)
   void BindP2PSocketManager(
       mojo::PendingReceiver<network::mojom::P2PSocketManager> receiver);
+#endif
   void CreateMediaLogRecordHost(
       mojo::PendingReceiver<content::mojom::MediaInternalLogRecords> receiver);
 #if BUILDFLAG(ENABLE_PLUGINS)
@@ -900,8 +909,10 @@ class CONTENT_EXPORT RenderProcessHostImpl
   void CreateAgentMetricsCollectorHost(
       mojo::PendingReceiver<blink::mojom::AgentMetricsCollectorHost> receiver);
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   void BindPeerConnectionTrackerHost(
       mojo::PendingReceiver<blink::mojom::PeerConnectionTrackerHost> receiver);
+#endif
 
 #if BUILDFLAG(ENABLE_MDNS)
   void CreateMdnsResponder(
@@ -1139,13 +1150,16 @@ class CONTENT_EXPORT RenderProcessHostImpl
   // and calling through RenderProcessHostObserver::RenderProcessExited.
   bool within_process_died_observer_;
 
-  std::unique_ptr<P2PSocketDispatcherHost> p2p_socket_dispatcher_host_;
+#if BUILDFLAG(ENABLE_WEBRTC)
+  std::unique_ptr <P2PSocketDispatcherHost>
+      p2p_socket_dispatcher_host_;
 
   // Must be accessed on UI thread.
   AecDumpManagerImpl aec_dump_manager_;
 
   std::unique_ptr<MediaStreamTrackMetricsHost, BrowserThread::DeleteOnIOThread>
       media_stream_track_metrics_host_;
+#endif
 
   std::unique_ptr<AgentMetricsCollectorHost> agent_metrics_collector_host_;
 
diff --git a/chromium/content/browser/webrtc/webrtc_internals.cc b/chromium/content/browser/webrtc/webrtc_internals.cc
index b1465594228..f9ae80ad7bd 100644
--- a/chromium/content/browser/webrtc/webrtc_internals.cc
+++ b/chromium/content/browser/webrtc/webrtc_internals.cc
@@ -106,6 +106,10 @@ WebRTCInternals::WebRTCInternals(int aggregate_updates_ms,
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
   DCHECK(!g_webrtc_internals);
 
+// TODO(grunell): Shouldn't all the webrtc_internals* files be excluded from the
+// build if WebRTC is disabled?
+// https://crbug.com/817446
+#if BUILDFLAG(ENABLE_WEBRTC)
   audio_debug_recordings_file_path_ =
       GetContentClient()->browser()->GetDefaultDownloadDirectory();
   event_log_recordings_file_path_ = audio_debug_recordings_file_path_;
@@ -137,6 +141,7 @@ WebRTCInternals::WebRTCInternals(int aggregate_updates_ms,
     event_log_recordings_ = true;
     event_log_recordings_file_path_.clear();
   }
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
 
   g_webrtc_internals = this;
 }
@@ -362,6 +367,7 @@ void WebRTCInternals::UpdateObserver(WebRTCInternalsUIObserver* observer) {
 void WebRTCInternals::EnableAudioDebugRecordings(
     content::WebContents* web_contents) {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
+#if BUILDFLAG(ENABLE_WEBRTC)
 #if defined(OS_ANDROID)
   EnableAudioDebugRecordingsOnAllRenderProcessHosts();
 #else
@@ -376,10 +382,12 @@ void WebRTCInternals::EnableAudioDebugRecordings(
       base::FilePath::StringType(), web_contents->GetTopLevelNativeWindow(),
       nullptr);
 #endif
+#endif
 }
 
 void WebRTCInternals::DisableAudioDebugRecordings() {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (!audio_debug_recording_session_)
     return;
   audio_debug_recording_session_.reset();
@@ -393,6 +401,7 @@ void WebRTCInternals::DisableAudioDebugRecordings() {
        !i.IsAtEnd(); i.Advance()) {
     i.GetCurrentValue()->DisableAudioDebugRecordings();
   }
+#endif
 }
 
 bool WebRTCInternals::IsAudioDebugRecordingsEnabled() const {
@@ -409,6 +418,7 @@ void WebRTCInternals::EnableLocalEventLogRecordings(
     content::WebContents* web_contents) {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
   DCHECK(CanToggleEventLogRecordings());
+#if BUILDFLAG(ENABLE_WEBRTC)
 #if defined(OS_ANDROID)
   WebRtcEventLogger* const logger = WebRtcEventLogger::Get();
   if (logger) {
@@ -427,9 +437,11 @@ void WebRTCInternals::EnableLocalEventLogRecordings(
       event_log_recordings_file_path_, nullptr, 0, FILE_PATH_LITERAL(""),
       web_contents->GetTopLevelNativeWindow(), nullptr);
 #endif
+#endif
 }
 
 void WebRTCInternals::DisableLocalEventLogRecordings() {
+#if BUILDFLAG(ENABLE_WEBRTC)
   event_log_recordings_ = false;
   // Tear down the dialog since the user has unchecked the event log checkbox.
   select_file_dialog_ = nullptr;
@@ -438,6 +450,7 @@ void WebRTCInternals::DisableLocalEventLogRecordings() {
   if (logger) {
     logger->DisableLocalLogging(base::OnceCallback<void(bool)>());
   }
+#endif
 }
 
 bool WebRTCInternals::IsEventLogRecordingsEnabled() const {
@@ -478,6 +491,7 @@ void WebRTCInternals::RenderProcessExited(
 void WebRTCInternals::FileSelected(const base::FilePath& path,
                                    int /* unused_index */,
                                    void* /*unused_params */) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
   switch (selection_type_) {
     case SelectionType::kRtcEventLogs: {
@@ -496,9 +510,11 @@ void WebRTCInternals::FileSelected(const base::FilePath& path,
     }
     default: { NOTREACHED(); }
   }
+#endif
 }
 
 void WebRTCInternals::FileSelectionCanceled(void* params) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
   switch (selection_type_) {
     case SelectionType::kRtcEventLogs:
@@ -511,6 +527,7 @@ void WebRTCInternals::FileSelectionCanceled(void* params) {
       NOTREACHED();
   }
   select_file_dialog_ = nullptr;
+#endif
 }
 
 void WebRTCInternals::OnRendererExit(int render_process_id) {
@@ -566,6 +583,7 @@ void WebRTCInternals::OnRendererExit(int render_process_id) {
   }
 }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 void WebRTCInternals::EnableAudioDebugRecordingsOnAllRenderProcessHosts() {
   DCHECK_CURRENTLY_ON(BrowserThread::UI);
   DCHECK(!audio_debug_recording_session_);
@@ -582,6 +600,7 @@ void WebRTCInternals::EnableAudioDebugRecordingsOnAllRenderProcessHosts() {
         audio_debug_recordings_file_path_);
   }
 }
+#endif
 
 void WebRTCInternals::MaybeClosePeerConnection(base::DictionaryValue* record) {
   bool is_open;
diff --git a/chromium/content/browser/webrtc/webrtc_internals.h b/chromium/content/browser/webrtc/webrtc_internals.h
index a2761e98bf7..3801a6c03ab 100644
--- a/chromium/content/browser/webrtc/webrtc_internals.h
+++ b/chromium/content/browser/webrtc/webrtc_internals.h
@@ -172,9 +172,11 @@ class CONTENT_EXPORT WebRTCInternals : public RenderProcessHostObserver,
   // Called when a renderer exits (including crashes).
   void OnRendererExit(int render_process_id);
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Enables diagnostic audio recordings on all render process hosts using
   // |audio_debug_recordings_file_path_|.
   void EnableAudioDebugRecordingsOnAllRenderProcessHosts();
+#endif
 
   // Updates the number of open PeerConnections. Called when a PeerConnection
   // is stopped or removed.
diff --git a/chromium/content/browser/webui/content_web_ui_controller_factory.cc b/chromium/content/browser/webui/content_web_ui_controller_factory.cc
index 6012229fe08..774b1403b73 100644
--- a/chromium/content/browser/webui/content_web_ui_controller_factory.cc
+++ b/chromium/content/browser/webui/content_web_ui_controller_factory.cc
@@ -16,13 +16,16 @@
 #include "content/browser/service_worker/service_worker_internals_ui.h"
 #include "content/browser/tracing/tracing_ui.h"
 #include "content/browser/ukm_internals_ui.h"
-#include "content/browser/webrtc/webrtc_internals_ui.h"
 #include "content/public/browser/storage_partition.h"
 #include "content/public/browser/web_contents.h"
 #include "content/public/browser/web_ui.h"
 #include "content/public/common/url_constants.h"
 #include "media/media_buildflags.h"
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "content/browser/webrtc/webrtc_internals_ui.h"
+#endif
+
 namespace content {
 
 WebUI::TypeID ContentWebUIControllerFactory::GetWebUIType(
@@ -86,8 +89,10 @@ ContentWebUIControllerFactory::CreateWebUIControllerForURL(WebUI* web_ui,
   if (url.host_piece() == kChromeUITracingHost)
     return std::make_unique<TracingUI>(web_ui);
 #endif
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (url.host_piece() == kChromeUIWebRTCInternalsHost)
     return std::make_unique<WebRTCInternalsUI>(web_ui);
+#endif
   if (url.host_piece() == kChromeUIProcessInternalsHost)
     return std::make_unique<ProcessInternalsUI>(web_ui);
   if (url.host_piece() == kChromeUIConversionInternalsHost)
diff --git a/chromium/content/common/BUILD.gn b/chromium/content/common/BUILD.gn
index 82018ea14fc..d499253e23f 100644
--- a/chromium/content/common/BUILD.gn
+++ b/chromium/content/common/BUILD.gn
@@ -7,6 +7,7 @@ import("//build/config/features.gni")
 import("//build/config/ui.gni")
 import("//content/public/common/zygote/features.gni")
 import("//ipc/features.gni")
+import("//media/media_options.gni")
 import("//mojo/public/tools/bindings/mojom.gni")
 import("//ppapi/buildflags/buildflags.gni")
 import("//sandbox/features.gni")
@@ -335,6 +336,15 @@ source_set("common") {
     deps += [ "//ppapi/proxy:ipc_sources" ]
   }
 
+  if (enable_webrtc) {
+    sources += [
+      "p2p_messages.h",
+      "p2p_socket_type.h",
+    ]
+    deps += [
+      "//third_party/webrtc_overrides:webrtc_component",
+    ]
+  }
   if (!use_x11 && !use_ozone) {
     sources -= [ "cursors/webcursor_ozone.cc" ]
   }
diff --git a/chromium/content/public/browser/BUILD.gn b/chromium/content/public/browser/BUILD.gn
index bd83dfa0a32..1defedb6c48 100644
--- a/chromium/content/public/browser/BUILD.gn
+++ b/chromium/content/public/browser/BUILD.gn
@@ -4,6 +4,7 @@
 
 import("//build/config/ui.gni")
 import("//device/vr/buildflags/buildflags.gni")
+import("//media/media_options.gni")
 import("//ppapi/buildflags/buildflags.gni")
 import("//third_party/protobuf/proto_library.gni")
 import("//third_party/webrtc/webrtc.gni")
@@ -138,8 +139,6 @@ source_set("browser_sources") {
     "cors_origin_pattern_setter.cc",
     "cors_origin_pattern_setter.h",
     "dedicated_worker_service.h",
-    "desktop_capture.cc",
-    "desktop_capture.h",
     "desktop_media_id.cc",
     "desktop_media_id.h",
     "desktop_streams_registry.h",
@@ -411,8 +410,6 @@ source_set("browser_sources") {
     "web_ui_url_loader_factory.h",
     "webrtc_event_logger.cc",
     "webrtc_event_logger.h",
-    "webrtc_log.cc",
-    "webrtc_log.h",
     "websocket_handshake_request_info.h",
     "worker_type.h",
   ]
@@ -459,9 +456,7 @@ source_set("browser_sources") {
     "//services/service_manager/public/cpp",
     "//services/tracing/public/cpp",
     "//services/tracing/public/mojom",
-    "//services/video_capture/public/mojom",
     "//services/viz/public/mojom",
-    "//third_party/webrtc_overrides:webrtc_component",
 
     # We expose skia headers in the public API.
     "//skia",
@@ -554,6 +549,19 @@ source_set("browser_sources") {
       "//device/vr/public/mojom",
     ]
   }
+
+  if (enable_webrtc) {
+    sources += [
+      "desktop_capture.cc",
+      "desktop_capture.h",
+      "webrtc_log.cc",
+      "webrtc_log.h",
+    ]
+    public_deps += [
+      "//services/video_capture/public/mojom",
+      "//third_party/webrtc_overrides:webrtc_component",
+    ]
+  }
 }
 
 if (is_android) {
diff --git a/chromium/content/public/browser/render_process_host.h b/chromium/content/public/browser/render_process_host.h
index 941825ec995..1de0bb5a095 100644
--- a/chromium/content/public/browser/render_process_host.h
+++ b/chromium/content/public/browser/render_process_host.h
@@ -333,6 +333,7 @@ class CONTENT_EXPORT RenderProcessHost : public IPC::Sender,
   // |empty_allowed| must be set to false for navigations for security reasons.
   virtual void FilterURL(bool empty_allowed, GURL* url) = 0;
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   virtual void EnableAudioDebugRecordings(const base::FilePath& file) = 0;
   virtual void DisableAudioDebugRecordings() = 0;
 
@@ -356,6 +357,7 @@ class CONTENT_EXPORT RenderProcessHost : public IPC::Sender,
   // identified locally within the RPH using the ID |lid|.
   virtual void EnableWebRtcEventLogOutput(int lid, int output_period_ms) = 0;
   virtual void DisableWebRtcEventLogOutput(int lid) = 0;
+#endif
 
   // Asks the renderer process to bind |receiver|. |receiver| arrives in the
   // renderer process and is carried through the following flow, stopping if any
diff --git a/chromium/content/public/browser/webrtc_log.h b/chromium/content/public/browser/webrtc_log.h
index 6aacf6585b2..dc0d69c97d4 100644
--- a/chromium/content/public/browser/webrtc_log.h
+++ b/chromium/content/public/browser/webrtc_log.h
@@ -12,6 +12,10 @@
 #include "content/common/content_export.h"
 #include "media/media_buildflags.h"
 
+#if !BUILDFLAG(ENABLE_WEBRTC)
+#error "WebRTC not enabled."
+#endif
+
 namespace content {
 
 class CONTENT_EXPORT WebRtcLog {
diff --git a/chromium/content/public/common/content_switches.cc b/chromium/content/public/common/content_switches.cc
index 5696dc4c8e7..c439b65e1e0 100644
--- a/chromium/content/public/common/content_switches.cc
+++ b/chromium/content/public/common/content_switches.cc
@@ -830,6 +830,7 @@ const char kWebOtpBackendSmsVerification[] = "web-otp-backend-sms-verification";
 // Enables User Consent backend for Web OTP API.
 const char kWebOtpBackendUserConsent[] = "web-otp-backend-user-consent";
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 // Disables encryption of RTP Media for WebRTC. When Chrome embeds Content, it
 // ignores this switch on its stable and beta channels.
 const char kDisableWebRtcEncryption[]      = "disable-webrtc-encryption";
@@ -877,6 +878,7 @@ const char kWebRtcMaxCpuConsumptionPercentage[] =
 // interval. Please see SetupStunProbeTrial in
 // chrome_browser_field_trials_desktop.cc for more detail.
 const char kWebRtcStunProbeTrialParameter[] = "webrtc-stun-probe-trial";
+#endif // BUILDFLAG(ENABLE_WEBRTC)
 
 // Enable capture and local storage of WebRTC event logs without visiting
 // chrome://webrtc-internals. This is useful for automated testing. It accepts
diff --git a/chromium/content/public/common/content_switches.h b/chromium/content/public/common/content_switches.h
index bc16eb6cdde..5f5970964d9 100644
--- a/chromium/content/public/common/content_switches.h
+++ b/chromium/content/public/common/content_switches.h
@@ -227,6 +227,7 @@ CONTENT_EXPORT extern const char kZygoteProcess[];
 CONTENT_EXPORT extern const char kWebOtpBackend[];
 CONTENT_EXPORT extern const char kWebOtpBackendSmsVerification[];
 CONTENT_EXPORT extern const char kWebOtpBackendUserConsent[];
+#if BUILDFLAG(ENABLE_WEBRTC)
 CONTENT_EXPORT extern const char kDisableWebRtcEncryption[];
 CONTENT_EXPORT extern const char kDisableWebRtcHWDecoding[];
 CONTENT_EXPORT extern const char kDisableWebRtcHWEncoding[];
@@ -239,6 +240,11 @@ CONTENT_EXPORT extern const char kWaitForDebuggerWebUI[];
 extern const char kWebRtcMaxCaptureFramerate[];
 extern const char kWebRtcMaxCpuConsumptionPercentage[];
 CONTENT_EXPORT extern const char kWebRtcStunProbeTrialParameter[];
+#endif
+
+// TODO(eladalon): Move into "#if BUILDFLAG(ENABLE_WEBRTC)" section when
+// WebRTCInternals is excluded from the build for non-WebRTC builds.
+// https://crbug.com/817446
 CONTENT_EXPORT extern const char kWebRtcLocalEventLogging[];
 CONTENT_EXPORT extern const char kDisableScrollToTextFragment[];
 
diff --git a/chromium/content/public/renderer/BUILD.gn b/chromium/content/public/renderer/BUILD.gn
index dc6d10885ef..401ca36d693 100644
--- a/chromium/content/public/renderer/BUILD.gn
+++ b/chromium/content/public/renderer/BUILD.gn
@@ -89,7 +89,6 @@ target(link_target_type, "renderer_sources") {
     "//skia",
     "//third_party/blink/public:blink_headers",
     "//third_party/blink/public/common",
-    "//third_party/webrtc_overrides:webrtc_component",
     "//third_party/widevine/cdm:headers",
     "//ui/base",
     "//ui/base/ime",
@@ -108,6 +107,12 @@ target(link_target_type, "renderer_sources") {
     deps += [ "//sandbox" ]
   }
 
+  if (enable_webrtc) {
+    deps += [
+      "//third_party/webrtc_overrides:webrtc_component",
+    ]
+  }
+
   if (enable_library_cdms) {
     sources += [
       "key_system_support.cc",
diff --git a/chromium/content/renderer/BUILD.gn b/chromium/content/renderer/BUILD.gn
index dc2c25bb670..80ce235c3fc 100644
--- a/chromium/content/renderer/BUILD.gn
+++ b/chromium/content/renderer/BUILD.gn
@@ -307,7 +307,6 @@ target(link_target_type, "renderer") {
     "//content/gpu:gpu_sources",
     "//content/public/child:child_sources",
     "//content/public/common:service_names",
-    "//crypto",
     "//crypto:platform",
     "//device/base/synchronization",
     "//device/gamepad/public/cpp:shared_with_blink",
@@ -442,7 +441,7 @@ target(link_target_type, "renderer") {
     }
   }
 
-  if (enable_plugins) {
+  if (enable_plugins && enable_webrtc) {
     sources += [
       "pepper/pepper_media_stream_audio_track_host.cc",
       "pepper/pepper_media_stream_audio_track_host.h",
diff --git a/chromium/content/renderer/media/media_factory.cc b/chromium/content/renderer/media/media_factory.cc
index e328190798e..576b569ea57 100644
--- a/chromium/content/renderer/media/media_factory.cc
+++ b/chromium/content/renderer/media/media_factory.cc
@@ -692,6 +692,7 @@ blink::WebMediaPlayer* MediaFactory::CreateWebMediaPlayerForMediaStream(
     blink::WebLocalFrame* frame,
     viz::FrameSinkId parent_frame_sink_id,
     const cc::LayerTreeSettings& settings) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   RenderThreadImpl* const render_thread = RenderThreadImpl::current();
 
   scoped_refptr<base::SingleThreadTaskRunner>
@@ -726,6 +727,9 @@ blink::WebMediaPlayer* MediaFactory::CreateWebMediaPlayerForMediaStream(
                      parent_frame_sink_id,
                      blink::WebSurfaceLayerBridge::ContainsVideo::kYes),
       std::move(submitter), GetVideoSurfaceLayerMode());
+#else
+  return NULL;
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
 }
 
 media::RendererWebMediaPlayerDelegate*
diff --git a/chromium/content/renderer/pepper/content_renderer_pepper_host_factory.cc b/chromium/content/renderer/pepper/content_renderer_pepper_host_factory.cc
index 9648fe8ee4c..3af256c5395 100644
--- a/chromium/content/renderer/pepper/content_renderer_pepper_host_factory.cc
+++ b/chromium/content/renderer/pepper/content_renderer_pepper_host_factory.cc
@@ -148,9 +148,11 @@ ContentRendererPepperHostFactory::CreateResourceHost(
                                                       resource);
     case PpapiHostMsg_WebSocket_Create::ID:
       return std::make_unique<PepperWebSocketHost>(host_, instance, resource);
+#if BUILDFLAG(ENABLE_WEBRTC)
     case PpapiHostMsg_MediaStreamVideoTrack_Create::ID:
       return std::make_unique<PepperMediaStreamVideoTrackHost>(host_, instance,
                                                                resource);
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
   }
 
   // Dev interfaces.
diff --git a/chromium/content/renderer/pepper/pepper_media_device_manager.cc b/chromium/content/renderer/pepper/pepper_media_device_manager.cc
index c00f38126ad..f8d10ec9320 100644
--- a/chromium/content/renderer/pepper/pepper_media_device_manager.cc
+++ b/chromium/content/renderer/pepper/pepper_media_device_manager.cc
@@ -105,6 +105,7 @@ PepperMediaDeviceManager::~PepperMediaDeviceManager() {
 
 void PepperMediaDeviceManager::EnumerateDevices(PP_DeviceType_Dev type,
                                                 DevicesOnceCallback callback) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   bool request_audio_input = type == PP_DEVICETYPE_DEV_AUDIOCAPTURE;
   bool request_video_input = type == PP_DEVICETYPE_DEV_VIDEOCAPTURE;
   bool request_audio_output = type == PP_DEVICETYPE_DEV_AUDIOOUTPUT;
@@ -115,11 +116,20 @@ void PepperMediaDeviceManager::EnumerateDevices(PP_DeviceType_Dev type,
       false /* request_audio_input_capabilities */,
       base::BindOnce(&PepperMediaDeviceManager::DevicesEnumerated, AsWeakPtr(),
                      std::move(callback), ToMediaDeviceType(type)));
+#else
+  base::ThreadTaskRunnerHandle::Get()->PostTask(
+      FROM_HERE, base::BindOnce(&PepperMediaDeviceManager::DevicesEnumerated,
+                                AsWeakPtr(), std::move(callback), ToMediaDeviceType(type),
+                                std::vector<blink::WebMediaDeviceInfoArray>(),
+                                std::vector<blink::mojom::VideoInputDeviceCapabilitiesPtr>(),
+                                std::vector<blink::mojom::AudioInputDeviceCapabilitiesPtr>()));
+#endif
 }
 
 size_t PepperMediaDeviceManager::StartMonitoringDevices(
     PP_DeviceType_Dev type,
     const DevicesCallback& callback) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   bool subscribe_audio_input = type == PP_DEVICETYPE_DEV_AUDIOCAPTURE;
   bool subscribe_video_input = type == PP_DEVICETYPE_DEV_VIDEOCAPTURE;
   bool subscribe_audio_output = type == PP_DEVICETYPE_DEV_AUDIOOUTPUT;
@@ -136,10 +146,14 @@ size_t PepperMediaDeviceManager::StartMonitoringDevices(
   subscriptions.push_back(Subscription{subscription_id, callback});
 
   return subscription_id;
+#else
+  return 0u;
+#endif
 }
 
 void PepperMediaDeviceManager::StopMonitoringDevices(PP_DeviceType_Dev type,
                                                      size_t subscription_id) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   SubscriptionList& subscriptions =
       device_change_subscriptions_[ToMediaDeviceType(type)];
   base::EraseIf(subscriptions,
@@ -147,6 +161,7 @@ void PepperMediaDeviceManager::StopMonitoringDevices(PP_DeviceType_Dev type,
                   return subscription.first == subscription_id;
                 });
   receivers_.Remove(subscription_id);
+#endif
 }
 
 int PepperMediaDeviceManager::OpenDevice(PP_DeviceType_Dev type,
@@ -172,11 +187,18 @@ int PepperMediaDeviceManager::OpenDevice(PP_DeviceType_Dev type,
     return request_id;
   }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   GetMediaStreamDispatcherHost()->OpenDevice(
       request_id, device_id,
       PepperMediaDeviceManager::FromPepperDeviceType(type),
       base::BindOnce(&PepperMediaDeviceManager::OnDeviceOpened, AsWeakPtr(),
                      request_id));
+#else
+  base::ThreadTaskRunnerHandle::Get()->PostTask(
+      FROM_HERE,
+      base::BindOnce(&PepperMediaDeviceManager::OnDeviceOpened, AsWeakPtr(),
+                     request_id, false, std::string(), blink::MediaStreamDevice()));
+#endif
 
   return request_id;
 }
@@ -184,20 +206,25 @@ int PepperMediaDeviceManager::OpenDevice(PP_DeviceType_Dev type,
 void PepperMediaDeviceManager::CancelOpenDevice(int request_id) {
   open_callbacks_.erase(request_id);
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   GetMediaStreamDispatcherHost()->CancelRequest(request_id);
+#endif
 }
 
 void PepperMediaDeviceManager::CloseDevice(const std::string& label) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (!GetMediaStreamDeviceObserver()->RemoveStream(
           blink::WebString::FromUTF8(label)))
     return;
 
   GetMediaStreamDispatcherHost()->CloseDevice(label);
+#endif
 }
 
 base::UnguessableToken PepperMediaDeviceManager::GetSessionID(
     PP_DeviceType_Dev type,
     const std::string& label) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   switch (type) {
     case PP_DEVICETYPE_DEV_AUDIOCAPTURE:
       return GetMediaStreamDeviceObserver()->GetAudioSessionId(
@@ -209,6 +236,9 @@ base::UnguessableToken PepperMediaDeviceManager::GetSessionID(
       NOTREACHED();
       return base::UnguessableToken();
   }
+#else
+  return base::UnguessableToken();
+#endif
 }
 
 // static
@@ -248,9 +278,11 @@ void PepperMediaDeviceManager::OnDeviceOpened(
     return;
   }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (success)
     GetMediaStreamDeviceObserver()->AddStream(blink::WebString::FromUTF8(label),
                                               device);
+#endif
 
   OpenDeviceCallback callback = std::move(iter->second);
   open_callbacks_.erase(iter);
diff --git a/chromium/content/renderer/pepper/resource_converter.cc b/chromium/content/renderer/pepper/resource_converter.cc
index 9f40c432b5e..5c831aec5fb 100644
--- a/chromium/content/renderer/pepper/resource_converter.cc
+++ b/chromium/content/renderer/pepper/resource_converter.cc
@@ -162,6 +162,7 @@ bool DOMMediaStreamTrackToResource(
     std::unique_ptr<IPC::Message>* create_message) {
   DCHECK(!dom_media_stream_track.IsNull());
   *pending_renderer_id = 0;
+#if BUILDFLAG(ENABLE_WEBRTC)
   const blink::WebMediaStreamTrack track = dom_media_stream_track.Component();
   const std::string id = track.Source().Id().Utf8();
 
@@ -187,6 +188,7 @@ bool DOMMediaStreamTrackToResource(
         new PpapiPluginMsg_MediaStreamAudioTrack_CreateFromPendingHost(id));
     return true;
   }
+#endif
   return false;
 }
 
diff --git a/chromium/content/renderer/render_frame_impl.cc b/chromium/content/renderer/render_frame_impl.cc
index 23fc00b4f30..1ee3762639c 100644
--- a/chromium/content/renderer/render_frame_impl.cc
+++ b/chromium/content/renderer/render_frame_impl.cc
@@ -6250,9 +6250,11 @@ void RenderFrameImpl::InitializeMediaStreamDeviceObserver() {
   if (!render_thread)  // Will be NULL during unit tests.
     return;
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   DCHECK(!web_media_stream_device_observer_);
   web_media_stream_device_observer_ =
       std::make_unique<blink::WebMediaStreamDeviceObserver>(GetWebFrame());
+#endif
 }
 
 void RenderFrameImpl::PrepareRenderViewForNavigation(
diff --git a/chromium/content/renderer/render_view_impl.cc b/chromium/content/renderer/render_view_impl.cc
index fec5e2312b7..8fb9d4f38b1 100644
--- a/chromium/content/renderer/render_view_impl.cc
+++ b/chromium/content/renderer/render_view_impl.cc
@@ -1043,6 +1043,7 @@ void RenderViewImpl::SetPageFrozen(bool frozen) {
 
 #if defined(OS_ANDROID)
 void RenderViewImpl::SuspendVideoCaptureDevices(bool suspend) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (!main_render_frame_)
     return;
 
@@ -1055,6 +1056,7 @@ void RenderViewImpl::SuspendVideoCaptureDevices(bool suspend) {
       media_stream_device_observer->GetNonScreenCaptureDevices();
   RenderThreadImpl::current()->video_capture_impl_manager()->SuspendDevices(
       video_devices, suspend);
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
 }
 #endif  // defined(OS_ANDROID)
 
diff --git a/chromium/content/renderer/renderer_blink_platform_impl.cc b/chromium/content/renderer/renderer_blink_platform_impl.cc
index ca435bda981..119e70d2749 100644
--- a/chromium/content/renderer/renderer_blink_platform_impl.cc
+++ b/chromium/content/renderer/renderer_blink_platform_impl.cc
@@ -496,8 +496,12 @@ scoped_refptr<media::AudioCapturerSource>
 RendererBlinkPlatformImpl::NewAudioCapturerSource(
     blink::WebLocalFrame* web_frame,
     const media::AudioSourceParameters& params) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   return blink::WebAudioDeviceFactory::NewAudioCapturerSource(
       web_frame->GetLocalFrameToken(), params);
+#else
+  return nullptr;
+#endif
 }
 
 viz::RasterContextProvider*
@@ -505,6 +509,7 @@ RendererBlinkPlatformImpl::SharedMainThreadContextProvider() {
   return RenderThreadImpl::current()->SharedMainThreadContextProvider().get();
 }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
 bool RendererBlinkPlatformImpl::RTCSmoothnessAlgorithmEnabled() {
   return !base::CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kDisableRTCSmoothnessAlgorithm);
@@ -673,6 +678,7 @@ bool RendererBlinkPlatformImpl::AllowsLoopbackInPeerConnection() {
   return base::CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kAllowLoopbackInPeerConnection);
 }
+#endif // BUILDFLAG(ENABLE_WEBRTC)
 
 blink::WebVideoCaptureImplManager*
 RendererBlinkPlatformImpl::GetVideoCaptureImplManager() {
diff --git a/chromium/content/renderer/renderer_blink_platform_impl.h b/chromium/content/renderer/renderer_blink_platform_impl.h
index 30af7f4fed5..823ae98d0f0 100644
--- a/chromium/content/renderer/renderer_blink_platform_impl.h
+++ b/chromium/content/renderer/renderer_blink_platform_impl.h
@@ -21,6 +21,7 @@
 #include "content/child/blink_platform_impl.h"
 #include "content/common/content_export.h"
 #include "content/renderer/top_level_blame_context.h"
+#include "media/media_buildflags.h"
 #include "mojo/public/cpp/bindings/pending_remote.h"
 #include "mojo/public/cpp/bindings/shared_remote.h"
 #include "services/network/public/mojom/url_loader_factory.mojom.h"
@@ -138,6 +139,7 @@ class CONTENT_EXPORT RendererBlinkPlatformImpl : public BlinkPlatformImpl {
       blink::WebLocalFrame* web_frame,
       const media::AudioSourceParameters& params) override;
   viz::RasterContextProvider* SharedMainThreadContextProvider() override;
+#if BUILDFLAG(ENABLE_WEBRTC)
   bool RTCSmoothnessAlgorithmEnabled() override;
   base::Optional<double> GetWebRtcMaxCaptureFrameRate() override;
   scoped_refptr<media::AudioRendererSink> NewAudioRendererSink(
@@ -167,6 +169,7 @@ class CONTENT_EXPORT RendererBlinkPlatformImpl : public BlinkPlatformImpl {
   bool IsWebRtcSrtpAesGcmEnabled() override;
   bool IsWebRtcSrtpEncryptedHeadersEnabled() override;
   bool AllowsLoopbackInPeerConnection() override;
+#endif
 
   blink::WebVideoCaptureImplManager* GetVideoCaptureImplManager() override;
 
diff --git a/chromium/content/renderer/renderer_main.cc b/chromium/content/renderer/renderer_main.cc
index 9e0f9a37b43..e162c30cf4e 100644
--- a/chromium/content/renderer/renderer_main.cc
+++ b/chromium/content/renderer/renderer_main.cc
@@ -40,7 +40,6 @@
 #include "services/tracing/public/cpp/trace_startup.h"
 #include "third_party/blink/public/platform/platform.h"
 #include "third_party/blink/public/platform/scheduler/web_thread_scheduler.h"
-#include "third_party/webrtc_overrides/init_webrtc.h"  // nogncheck
 #include "ui/base/ui_base_switches.h"
 
 #if defined(OS_ANDROID)
@@ -66,6 +65,10 @@
 #include "content/renderer/pepper/pepper_plugin_registry.h"
 #endif
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "third_party/webrtc_overrides/init_webrtc.h"  // nogncheck
+#endif
+
 #if BUILDFLAG(MOJO_RANDOM_DELAYS_ENABLED)
 #include "mojo/public/cpp/bindings/lib/test_random_mojo_delays.h"
 #endif
@@ -185,11 +188,13 @@ int RendererMain(const MainFunctionParams& parameters) {
   // Load pepper plugins before engaging the sandbox.
   PepperPluginRegistry::GetInstance();
 #endif
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Initialize WebRTC before engaging the sandbox.
   // NOTE: On linux, this call could already have been made from
   // zygote_main_linux.cc.  However, calling multiple times from the same thread
   // is OK.
   InitializeWebRtcModule();
+#endif
 
   {
     bool should_run_loop = true;
diff --git a/chromium/content/test/BUILD.gn b/chromium/content/test/BUILD.gn
index 003f7fe2119..4fd83be5c90 100644
--- a/chromium/content/test/BUILD.gn
+++ b/chromium/content/test/BUILD.gn
@@ -445,7 +445,6 @@ static_library("test_support") {
     "//testing/gmock",
     "//testing/gtest",
     "//third_party/blink/public/strings:strings_grit",
-    "//third_party/webrtc_overrides:webrtc_component",
     "//ui/accessibility:ax_enums_mojo",
     "//ui/base",
     "//ui/base:test_support",
@@ -530,6 +529,27 @@ static_library("test_support") {
     ]
   }
 
+  if (enable_webrtc) {
+    sources += [
+      "../renderer/media/stream/mock_mojo_media_stream_dispatcher_host.cc",
+      "../renderer/media/stream/mock_mojo_media_stream_dispatcher_host.h",
+      "../renderer/media/webrtc/mock_data_channel_impl.cc",
+      "../renderer/media/webrtc/mock_data_channel_impl.h",
+      "../renderer/media/webrtc/mock_peer_connection_dependency_factory.cc",
+      "../renderer/media/webrtc/mock_peer_connection_dependency_factory.h",
+      "../renderer/media/webrtc/mock_peer_connection_impl.cc",
+      "../renderer/media/webrtc/mock_peer_connection_impl.h",
+      "../renderer/media/webrtc/mock_web_rtc_peer_connection_handler_client.cc",
+      "../renderer/media/webrtc/mock_web_rtc_peer_connection_handler_client.h",
+      "../renderer/media/webrtc/test/webrtc_stats_report_obtainer.cc",
+      "../renderer/media/webrtc/test/webrtc_stats_report_obtainer.h",
+    ]
+
+    deps += [
+      "//third_party/webrtc_overrides:webrtc_component",
+    ]
+  }
+
   if (use_glib) {
     configs += [ "//build/config/linux:glib" ]
   }
@@ -1122,31 +1142,6 @@ test("content_browsertests") {
     "../browser/web_package/signed_exchange_subresource_prefetch_browsertest.cc",
     "../browser/web_package/web_bundle_browsertest.cc",
     "../browser/webkit_browsertest.cc",
-    "../browser/webrtc/webrtc_audio_browsertest.cc",
-    "../browser/webrtc/webrtc_audio_debug_recordings_browsertest.cc",
-    "../browser/webrtc/webrtc_browsertest.cc",
-    "../browser/webrtc/webrtc_capture_from_element_browsertest.cc",
-    "../browser/webrtc/webrtc_constraints_browsertest.cc",
-    "../browser/webrtc/webrtc_content_browsertest_base.cc",
-    "../browser/webrtc/webrtc_content_browsertest_base.h",
-    "../browser/webrtc/webrtc_data_browsertest.cc",
-    "../browser/webrtc/webrtc_datachannel_browsertest.cc",
-    "../browser/webrtc/webrtc_depth_capture_browsertest.cc",
-    "../browser/webrtc/webrtc_getusermedia_browsertest.cc",
-    "../browser/webrtc/webrtc_image_capture_browsertest.cc",
-    "../browser/webrtc/webrtc_internals_browsertest.cc",
-    "../browser/webrtc/webrtc_ip_permissions_browsertest.cc",
-    "../browser/webrtc/webrtc_media_recorder_browsertest.cc",
-    "../browser/webrtc/webrtc_stress_image_capture_browsertest.cc",
-    "../browser/webrtc/webrtc_stress_pause_browsertest.cc",
-    "../browser/webrtc/webrtc_stress_resolution_switch_browsertest.cc",
-    "../browser/webrtc/webrtc_stress_source_switch_browsertest.cc",
-    "../browser/webrtc/webrtc_video_capture_browsertest.cc",
-    "../browser/webrtc/webrtc_video_capture_service_browsertest.cc",
-    "../browser/webrtc/webrtc_video_capture_service_enumeration_browsertest.cc",
-    "../browser/webrtc/webrtc_video_capture_shared_device_browsertest.cc",
-    "../browser/webrtc/webrtc_webcam_browsertest.cc",
-    "../browser/webrtc/webrtc_webcam_browsertest.h",
     "../browser/webui/web_ui_browsertest.cc",
     "../browser/webui/web_ui_mojo_browsertest.cc",
     "../browser/webui/web_ui_navigation_browsertest.cc",
@@ -1524,6 +1519,40 @@ test("content_browsertests") {
         [ "../browser/compositor/image_transport_factory_browsertest.cc" ]
   }
 
+  if (enable_webrtc) {
+    sources += [
+      "../browser/webrtc/webrtc_audio_browsertest.cc",
+      "../browser/webrtc/webrtc_audio_debug_recordings_browsertest.cc",
+      "../browser/webrtc/webrtc_browsertest.cc",
+      "../browser/webrtc/webrtc_capture_from_element_browsertest.cc",
+      "../browser/webrtc/webrtc_constraints_browsertest.cc",
+      "../browser/webrtc/webrtc_content_browsertest_base.cc",
+      "../browser/webrtc/webrtc_content_browsertest_base.h",
+      "../browser/webrtc/webrtc_data_browsertest.cc",
+      "../browser/webrtc/webrtc_datachannel_browsertest.cc",
+      "../browser/webrtc/webrtc_depth_capture_browsertest.cc",
+      "../browser/webrtc/webrtc_getusermedia_browsertest.cc",
+      "../browser/webrtc/webrtc_image_capture_browsertest.cc",
+      "../browser/webrtc/webrtc_internals_browsertest.cc",
+      "../browser/webrtc/webrtc_ip_permissions_browsertest.cc",
+      "../browser/webrtc/webrtc_media_recorder_browsertest.cc",
+      "../browser/webrtc/webrtc_stress_image_capture_browsertest.cc",
+      "../browser/webrtc/webrtc_stress_pause_browsertest.cc",
+      "../browser/webrtc/webrtc_stress_resolution_switch_browsertest.cc",
+      "../browser/webrtc/webrtc_stress_source_switch_browsertest.cc",
+      "../browser/webrtc/webrtc_video_capture_browsertest.cc",
+      "../browser/webrtc/webrtc_video_capture_service_browsertest.cc",
+      "../browser/webrtc/webrtc_video_capture_service_enumeration_browsertest.cc",
+      "../browser/webrtc/webrtc_video_capture_shared_device_browsertest.cc",
+      "../browser/webrtc/webrtc_webcam_browsertest.cc",
+      "../browser/webrtc/webrtc_webcam_browsertest.h",
+    ]
+    deps += [
+      "//content/common:buildflags",
+      "//testing/perf",
+    ]
+  }
+
   if (enable_plugins) {
     sources += [
       "../browser/plugin_service_impl_browsertest.cc",
@@ -1994,8 +2023,6 @@ test("content_unittests") {
     "../browser/web_package/web_bundle_reader_unittest.cc",
     "../browser/web_package/web_bundle_url_loader_factory_unittest.cc",
     "../browser/web_package/web_bundle_utils_unittest.cc",
-    "../browser/webrtc/webrtc_internals_message_handler_unittest.cc",
-    "../browser/webrtc/webrtc_internals_unittest.cc",
     "../browser/webui/web_ui_data_source_unittest.cc",
     "../browser/webui/web_ui_message_handler_unittest.cc",
     "../browser/webui/web_ui_unittest.cc",
@@ -2212,7 +2239,6 @@ test("content_unittests") {
     "//third_party/icu",
     "//third_party/inspector_protocol:crdtp_test",
     "//third_party/leveldatabase",
-    "//third_party/libyuv",
     "//third_party/metrics_proto",
     "//third_party/re2",
     "//third_party/sqlite",
@@ -2238,7 +2264,6 @@ test("content_unittests") {
     "//ui/gl",
     "//ui/gl:test_support",
     "//ui/latency:test_support",
-    "//ui/shell_dialogs:shell_dialogs",
   ]
 
   data_deps = [
@@ -2252,6 +2277,8 @@ test("content_unittests") {
   }
 
   if (enable_plugins) {
+    # Put WebRTC-related plugins sources in the "enable_webrtc &&
+    # enable_plugins" section below.
     sources += [
       "../browser/plugin_content_origin_allowlist_unittest.cc",
       "../browser/plugin_service_impl_unittest.cc",
@@ -2283,16 +2310,105 @@ test("content_unittests") {
     ]
   }
 
+  if (enable_webrtc) {
+    # Put WebRTC-related plugins sources in the "enable_webrtc &&
+    # enable_plugins" section below.
+    sources += [
+      "../renderer/media/stream/media_stream_audio_processor_unittest.cc",
+      "../renderer/media/stream/media_stream_audio_unittest.cc",
+      "../renderer/media/stream/media_stream_constraints_util_audio_unittest.cc",
+      "../renderer/media/stream/media_stream_constraints_util_sets_unittest.cc",
+      "../renderer/media/stream/media_stream_constraints_util_unittest.cc",
+      "../renderer/media/stream/media_stream_constraints_util_video_content_unittest.cc",
+      "../renderer/media/stream/media_stream_constraints_util_video_device_unittest.cc",
+      "../renderer/media/stream/media_stream_device_observer_unittest.cc",
+      "../renderer/media/stream/media_stream_video_renderer_sink_unittest.cc",
+      "../renderer/media/stream/media_stream_video_source_unittest.cc",
+      "../renderer/media/stream/media_stream_video_track_unittest.cc",
+      "../renderer/media/stream/mock_constraint_factory.cc",
+      "../renderer/media/stream/mock_constraint_factory.h",
+      "../renderer/media/stream/mock_media_stream_registry.cc",
+      "../renderer/media/stream/mock_media_stream_registry.h",
+      "../renderer/media/stream/mock_media_stream_video_sink.cc",
+      "../renderer/media/stream/mock_media_stream_video_sink.h",
+      "../renderer/media/stream/mock_media_stream_video_source.cc",
+      "../renderer/media/stream/mock_media_stream_video_source.h",
+      "../renderer/media/stream/processed_local_audio_source_unittest.cc",
+      "../renderer/media/stream/user_media_client_impl_unittest.cc",
+      "../renderer/media/stream/webmediaplayer_ms_unittest.cc",
+      "../renderer/media/webrtc/fake_rtc_rtp_transceiver.cc",
+      "../renderer/media/webrtc/fake_rtc_rtp_transceiver.h",
+      "../renderer/media/webrtc/media_stream_remote_video_source_unittest.cc",
+      "../renderer/media/webrtc/media_stream_track_metrics_unittest.cc",
+      "../renderer/media/webrtc/media_stream_video_webrtc_sink_unittest.cc",
+      "../renderer/media/webrtc/peer_connection_dependency_factory_unittest.cc",
+      "../renderer/media/webrtc/peer_connection_tracker_unittest.cc",
+      "../renderer/media/webrtc/rtc_peer_connection_handler_unittest.cc",
+      "../renderer/media/webrtc/rtc_rtp_receiver_unittest.cc",
+      "../renderer/media/webrtc/rtc_rtp_sender_unittest.cc",
+      "../renderer/media/webrtc/rtc_rtp_transceiver_unittest.cc",
+      "../renderer/media/webrtc/rtc_stats_unittest.cc",
+      "../renderer/media/webrtc/rtc_video_decoder_adapter_unittest.cc",
+      "../renderer/media/webrtc/rtc_video_decoder_unittest.cc",
+      "../renderer/media/webrtc/rtc_video_encoder_unittest.cc",
+      "../renderer/media/webrtc/stun_field_trial_unittest.cc",
+      "../renderer/media/webrtc/task_queue_factory_unittest.cc",
+      "../renderer/media/webrtc/transceiver_state_surfacer_unittest.cc",
+      "../renderer/media/webrtc/transmission_encoding_info_handler_unittest.cc",
+      "../renderer/media/webrtc/two_keys_adapter_map_unittest.cc",
+      "../renderer/media/webrtc/webrtc_audio_renderer_unittest.cc",
+      "../renderer/media/webrtc/webrtc_media_stream_track_adapter_map_unittest.cc",
+      "../renderer/media/webrtc/webrtc_media_stream_track_adapter_unittest.cc",
+      "../renderer/media/webrtc/webrtc_set_description_observer_unittest.cc",
+      "../renderer/media/webrtc/webrtc_video_track_source_unittest.cc",
+      "../renderer/media/webrtc_local_audio_source_provider_unittest.cc",
+      "../renderer/media_capture_from_element/canvas_capture_handler_unittest.cc",
+      "../renderer/media_capture_from_element/html_audio_element_capturer_source_unittest.cc",
+      "../renderer/media_capture_from_element/html_video_element_capturer_source_unittest.cc",
+      "../renderer/media_recorder/audio_track_recorder_unittest.cc",
+      "../renderer/media_recorder/media_recorder_handler_unittest.cc",
+      "../renderer/media_recorder/video_track_recorder_unittest.cc",
+      "../renderer/p2p/filtering_network_manager_unittest.cc",
+      "../renderer/p2p/ipc_network_manager_unittest.cc",
+    ]
+    deps += [
+      "//services/device/public/mojom",
+      "//third_party/libyuv",
+      "//third_party/opus",
+      "//third_party/webrtc_overrides:webrtc_component",
+      "//ui/shell_dialogs:shell_dialogs",
+    ]
+
+    if (is_linux || is_mac || is_win) {
+      sources +=
+          [ "../browser/media/capture/desktop_capture_device_unittest.cc" ]
+      deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+    }
+
+    if (is_chromecast) {
+      sources -= [
+        "../renderer/media_recorder/audio_track_recorder_unittest.cc",
+        "../renderer/media_recorder/media_recorder_handler_unittest.cc",
+        "../renderer/media_recorder/video_track_recorder_unittest.cc",
+      ]
+    }
+  }
+
+  if (enable_webrtc && enable_plugins) {
+    sources += [
+      "../renderer/media/pepper/pepper_to_video_track_adapter_unittest.cc",
+      "../renderer/media/pepper/video_track_to_pepper_adapter_unittest.cc",
+    ]
+  }
+
   # Screen capture unit tests.
   if (enable_screen_capture && !is_android) {
     deps += [
       "//content/common:buildflags",
       "//media/capture:test_support",
-      "//third_party/libyuv",
-      "//third_party/webrtc_overrides:webrtc_component",
+      "//third_party/libyuv"
     ]
     sources += [
-      "../browser/media/capture/desktop_capture_device_unittest.cc",
       "../browser/media/capture/frame_sink_video_capture_device_unittest.cc",
     ]
     if (is_chromeos) {
diff --git a/chromium/media/BUILD.gn b/chromium/media/BUILD.gn
index 228d631ec41..dcf1b138a80 100644
--- a/chromium/media/BUILD.gn
+++ b/chromium/media/BUILD.gn
@@ -40,6 +40,7 @@ buildflag_header("media_buildflags") {
     "ENABLE_OPENH264=$media_use_openh264",
     "ENABLE_PLATFORM_MPEG_H_AUDIO=$enable_platform_mpeg_h_audio",
     "ENABLE_MSE_MPEG2TS_STREAM_PARSER=$enable_mse_mpeg2ts_stream_parser",
+    "ENABLE_WEBRTC=$enable_webrtc",
     "ENABLE_WEB_SPEECH=$enable_web_speech",
     "USE_CHROMEOS_MEDIA_ACCELERATION=$use_vaapi||$use_v4l2_codec",
     "USE_PROPRIETARY_CODECS=$proprietary_codecs",
diff --git a/chromium/media/audio/BUILD.gn b/chromium/media/audio/BUILD.gn
index 5e4dc90597d..63dce701f15 100644
--- a/chromium/media/audio/BUILD.gn
+++ b/chromium/media/audio/BUILD.gn
@@ -77,8 +77,6 @@ source_set("audio") {
     "audio_input_device.h",
     "audio_input_ipc.cc",
     "audio_input_ipc.h",
-    "audio_input_stream_data_interceptor.cc",
-    "audio_input_stream_data_interceptor.h",
     "audio_io.h",
     "audio_manager.cc",
     "audio_manager.h",
@@ -314,6 +312,13 @@ source_set("audio") {
     ]
   }
 
+  if (enable_webrtc) {
+    sources += [
+      "audio_input_stream_data_interceptor.cc",
+      "audio_input_stream_data_interceptor.h",
+    ]
+  }
+
   configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
 }
 
@@ -385,7 +390,6 @@ source_set("unit_tests") {
     "audio_debug_recording_session_impl_unittest.cc",
     "audio_encoders_unittest.cc",
     "audio_input_device_unittest.cc",
-    "audio_input_stream_data_interceptor_unittest.cc",
     "audio_input_unittest.cc",
     "audio_manager_unittest.cc",
     "audio_output_device_unittest.cc",
@@ -413,6 +417,10 @@ source_set("unit_tests") {
     "//media:media_config",
   ]
 
+  if (enable_webrtc) {
+    sources += [ "audio_input_stream_data_interceptor_unittest.cc" ]
+  }
+
   if (is_android) {
     sources += [ "android/audio_android_unittest.cc" ]
     deps += [ "//ui/gl" ]
diff --git a/chromium/media/audio/audio_manager.cc b/chromium/media/audio/audio_manager.cc
index 710130e7b95..01ccb57ac88 100644
--- a/chromium/media/audio/audio_manager.cc
+++ b/chromium/media/audio/audio_manager.cc
@@ -115,7 +115,9 @@ std::unique_ptr<AudioManager> AudioManager::Create(
     AudioLogFactory* audio_log_factory) {
   std::unique_ptr<AudioManager> manager =
       CreateAudioManager(std::move(audio_thread), audio_log_factory);
+#if BUILDFLAG(ENABLE_WEBRTC)
   manager->InitializeDebugRecording();
+#endif
   return manager;
 }
 
diff --git a/chromium/media/audio/audio_manager_base.cc b/chromium/media/audio/audio_manager_base.cc
index ff38a76975b..323255c207d 100644
--- a/chromium/media/audio/audio_manager_base.cc
+++ b/chromium/media/audio/audio_manager_base.cc
@@ -27,7 +27,9 @@
 #include "media/base/media_switches.h"
 
 #include "base/logging.h"
+#if BUILDFLAG(ENABLE_WEBRTC)
 #include "media/audio/audio_input_stream_data_interceptor.h"
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
 
 namespace media {
 
@@ -307,6 +309,7 @@ AudioInputStream* AudioManagerBase::MakeAudioInputStream(
                      input_stream_count());
     }
 
+#if BUILDFLAG(ENABLE_WEBRTC)
     if (!params.IsBitstreamFormat() && debug_recording_manager_) {
       // Using unretained for |debug_recording_manager_| is safe since it
       // outlives the audio thread, on which streams are operated.
@@ -321,6 +324,7 @@ AudioInputStream* AudioManagerBase::MakeAudioInputStream(
               AudioDebugRecordingStreamType::kInput, params),
           stream);
     }
+#endif  // BUILDFLAG(ENABLE_WEBRTC)
   }
 
   return stream;
diff --git a/chromium/media/gpu/BUILD.gn b/chromium/media/gpu/BUILD.gn
index e3d961939c3..b1a84b1ad9a 100644
--- a/chromium/media/gpu/BUILD.gn
+++ b/chromium/media/gpu/BUILD.gn
@@ -96,8 +96,6 @@ component("gpu") {
 
   if (is_android) {
     sources += [
-      "android/android_video_encode_accelerator.cc",
-      "android/android_video_encode_accelerator.h",
       "android/android_video_surface_chooser.cc",
       "android/android_video_surface_chooser.h",
       "android/android_video_surface_chooser_impl.cc",
@@ -152,6 +150,14 @@ component("gpu") {
     if (enable_vulkan) {
       deps += [ "//gpu/vulkan:vulkan" ]
     }
+
+    if (enable_webrtc) {
+      deps += [ "//third_party/libyuv" ]
+      sources += [
+        "android/android_video_encode_accelerator.cc",
+        "android/android_video_encode_accelerator.h",
+      ]
+    }
   }
 
   if (use_v4l2_codec) {
diff --git a/chromium/media/gpu/gpu_video_encode_accelerator_factory.cc b/chromium/media/gpu/gpu_video_encode_accelerator_factory.cc
index 8d665bb0320..b76582675d2 100644
--- a/chromium/media/gpu/gpu_video_encode_accelerator_factory.cc
+++ b/chromium/media/gpu/gpu_video_encode_accelerator_factory.cc
@@ -16,7 +16,7 @@
 #if BUILDFLAG(USE_V4L2_CODEC)
 #include "media/gpu/v4l2/v4l2_video_encode_accelerator.h"
 #endif
-#if defined(OS_ANDROID)
+#if defined(OS_ANDROID) && BUILDFLAG(ENABLE_WEBRTC)
 #include "media/gpu/android/android_video_encode_accelerator.h"
 #endif
 #if defined(OS_MAC)
@@ -51,7 +51,7 @@ std::unique_ptr<VideoEncodeAccelerator> CreateVaapiVEA() {
 }
 #endif
 
-#if defined(OS_ANDROID)
+#if defined(OS_ANDROID) && BUILDFLAG(ENABLE_WEBRTC)
 std::unique_ptr<VideoEncodeAccelerator> CreateAndroidVEA() {
   return base::WrapUnique<VideoEncodeAccelerator>(
       new AndroidVideoEncodeAccelerator());
@@ -98,7 +98,7 @@ std::vector<VEAFactoryFunction> GetVEAFactoryFunctions(
 #if BUILDFLAG(USE_V4L2_CODEC)
   vea_factory_functions.push_back(base::BindRepeating(&CreateV4L2VEA));
 #endif
-#if defined(OS_ANDROID)
+#if defined(OS_ANDROID) && BUILDFLAG(ENABLE_WEBRTC)
   vea_factory_functions.push_back(base::BindRepeating(&CreateAndroidVEA));
 #endif
 #if defined(OS_MAC)
diff --git a/chromium/media/media_options.gni b/chromium/media/media_options.gni
index ea1e280d714..e0421429cbd 100644
--- a/chromium/media/media_options.gni
+++ b/chromium/media/media_options.gni
@@ -89,6 +89,8 @@ declare_args() {
 
   enable_web_speech = true
 
+  enable_webrtc = true
+
   # Enable HLS with SAMPLE-AES decryption.
   enable_hls_sample_aes = proprietary_codecs && (is_chromecast || is_fuchsia)
 
diff --git a/chromium/remoting/BUILD.gn b/chromium/remoting/BUILD.gn
index ceb99d97d38..2ec6a320367 100644
--- a/chromium/remoting/BUILD.gn
+++ b/chromium/remoting/BUILD.gn
@@ -2,6 +2,7 @@
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
+import("//media/media_options.gni")
 import("//remoting/build/config/remoting_build.gni")
 
 group("remoting_all") {
@@ -158,7 +159,9 @@ test("remoting_unittests") {
     ]
   }
 
-  deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+  if (enable_webrtc) {
+    deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+  }
 
   if (is_android) {
     deps += [ "//net/android:net_java" ]
@@ -201,9 +204,14 @@ if (enable_remoting_host) {
       "//net:test_support",
       "//remoting/base",
       "//testing/gtest",
-      "//third_party/webrtc_overrides:webrtc_component",
     ]
 
+    if (enable_webrtc) {
+      deps += [
+        "//third_party/webrtc_overrides:webrtc_component",
+      ]
+    }
+
     if (is_win) {
       defines += [ "_ALT_NO_EXCEPTIONS" ]
 
diff --git a/chromium/remoting/host/BUILD.gn b/chromium/remoting/host/BUILD.gn
index 04ae4e6ea86..40422297fb3 100644
--- a/chromium/remoting/host/BUILD.gn
+++ b/chromium/remoting/host/BUILD.gn
@@ -3,6 +3,7 @@
 # found in the LICENSE file.
 
 import("//build/util/process_version.gni")
+import("//media/media_options.gni")
 import("//remoting/build/config/remoting_build.gni")
 
 # This file depends on the legacy global sources assignment filter. It should
@@ -336,7 +337,6 @@ static_library("common") {
     "//remoting/protocol",
     "//remoting/resources",
     "//services/network:network_service",
-    "//third_party/webrtc_overrides:webrtc_component",
 
     # //remoting uses the power_save_blocker directly. See crbug.com/689423
     "//services/device/wake_lock/power_save_blocker",
@@ -453,6 +453,10 @@ static_library("common") {
 
     public_deps += [ "//remoting/host/win" ]
   }
+
+  if (enable_webrtc) {
+    deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+  }
 }
 
 static_library("test_support") {
@@ -482,10 +486,15 @@ static_library("test_support") {
     ":common",
     "//remoting/base:test_support",
     "//remoting/host/file_transfer:test_support",
-    "//third_party/libjingle_xmpp",
     "//third_party/protobuf:protobuf_lite",
-    "//third_party/webrtc_overrides:webrtc_component",
   ]
+
+  if (enable_webrtc) {
+    public_deps += [
+      "//third_party/libjingle_xmpp",
+      "//third_party/webrtc_overrides:webrtc_component",
+    ]
+  }
 }
 
 # The host portions of the remoting unit tests.
@@ -615,10 +624,7 @@ if (enable_remoting_host) {
   executable("remoting_start_host") {
     sources = [ "setup/start_host_entry_point.cc" ]
 
-    deps = [
-      "//third_party/libjingle_xmpp",
-      "//third_party/webrtc_overrides:webrtc_component",
-    ]
+    deps = []
 
     configs += [ "//build/config/compiler:wexit_time_destructors" ]
 
@@ -646,6 +652,13 @@ if (enable_remoting_host) {
         "//remoting/host/setup",
       ]
     }
+
+    if (enable_webrtc) {
+      deps += [
+        "//third_party/libjingle_xmpp",
+        "//third_party/webrtc_overrides:webrtc_component",
+      ]
+    }
   }
 
   action_foreach("remoting_native_messaging_manifests") {
@@ -744,8 +757,6 @@ if (enable_me2me_host) {
       "//remoting/base",
       "//remoting/host",
       "//remoting/proto",
-      "//third_party/libjingle_xmpp",
-      "//third_party/webrtc_overrides:webrtc_component",
     ]
 
     if (is_posix) {
@@ -759,6 +770,13 @@ if (enable_me2me_host) {
       deps += [ "//components/policy:generated" ]
     }
 
+    if (enable_webrtc) {
+      deps += [
+        "//third_party/libjingle_xmpp",
+        "//third_party/webrtc_overrides:webrtc_component",
+      ]
+    }
+
     if (use_gtk) {
       deps += [ "//build/config/linux/gtk" ]
     }
diff --git a/chromium/remoting/host/it2me/BUILD.gn b/chromium/remoting/host/it2me/BUILD.gn
index 4038303cdaa..7baddde20a6 100644
--- a/chromium/remoting/host/it2me/BUILD.gn
+++ b/chromium/remoting/host/it2me/BUILD.gn
@@ -4,6 +4,7 @@
 
 import("//build/config/features.gni")
 import("//build/config/ui.gni")
+import("//media/media_options.gni")
 import("//remoting/remoting_locales.gni")
 import("//remoting/remoting_options.gni")
 import("//remoting/remoting_version.gni")
@@ -213,7 +214,6 @@ if (!is_chromeos && enable_remoting_host) {
         "//remoting/host",
         "//remoting/host/native_messaging",
         "//remoting/proto",
-        "//third_party/webrtc_overrides:webrtc_component",
         "//ui/gfx",
       ]
       if (is_mac) {
@@ -234,6 +234,10 @@ if (!is_chromeos && enable_remoting_host) {
         }
       }
 
+      if (enable_webrtc) {
+        deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+      }
+
       if (use_gtk) {
         deps += [ "//build/config/linux/gtk" ]
       }
diff --git a/chromium/remoting/host/win/BUILD.gn b/chromium/remoting/host/win/BUILD.gn
index 8f527a22284..e0ec78df1a4 100644
--- a/chromium/remoting/host/win/BUILD.gn
+++ b/chromium/remoting/host/win/BUILD.gn
@@ -136,7 +136,6 @@ source_set("win") {
     "//remoting/protocol",
     "//remoting/resources",
     "//services/device/wake_lock/power_save_blocker",
-    "//third_party/webrtc_overrides:webrtc_component",
     "//third_party/wtl",
     "//ui/base",
     "//ui/events:dom_keycode_converter",
@@ -146,6 +145,10 @@ source_set("win") {
   if (!is_ios) {
     deps += [ "//components/policy:generated" ]
   }
+
+  if (enable_webrtc) {
+    deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
+  }
 }
 
 source_set("unit_tests") {
diff --git a/chromium/remoting/remoting_enable.gni b/chromium/remoting/remoting_enable.gni
index 879214d9a8d..8a24badbe0c 100644
--- a/chromium/remoting/remoting_enable.gni
+++ b/chromium/remoting/remoting_enable.gni
@@ -3,11 +3,12 @@
 # found in the LICENSE file.
 
 import("//build/config/ui.gni")
+import("//media/media_options.gni")
 
 if (is_ios) {
   import("//build/config/ios/ios_sdk.gni")
 }
 
 declare_args() {
-  enable_remoting = !is_chromecast && !is_fuchsia
+  enable_remoting = !is_chromecast && !is_fuchsia && enable_webrtc
 }
diff --git a/chromium/third_party/blink/common/BUILD.gn b/chromium/third_party/blink/common/BUILD.gn
index 330261bf50b..cd2031c669b 100644
--- a/chromium/third_party/blink/common/BUILD.gn
+++ b/chromium/third_party/blink/common/BUILD.gn
@@ -2,6 +2,7 @@
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
+import("//media/media_options.gni")
 import("//testing/libfuzzer/fuzzer_test.gni")
 import("//testing/test.gni")
 import("//third_party/blink/renderer/build/scripts/scripts.gni")
@@ -193,6 +194,12 @@ source_set("common") {
     deps += [ "//media" ]
   }
 
+  if (!enable_webrtc) {
+    sources -= [
+      "peerconnection/webrtc_ip_handling_policy.cc",
+    ]
+  }
+
   if (is_android || is_win) {
     sources += [
       "font_unique_name_lookup/font_table_matcher.cc",
diff --git a/chromium/third_party/blink/renderer/modules/media_capabilities/media_capabilities.cc b/chromium/third_party/blink/renderer/modules/media_capabilities/media_capabilities.cc
index 9b833e9c636..50bd77c314b 100644
--- a/chromium/third_party/blink/renderer/modules/media_capabilities/media_capabilities.cc
+++ b/chromium/third_party/blink/renderer/modules/media_capabilities/media_capabilities.cc
@@ -21,7 +21,7 @@
 #include "media/learning/common/media_learning_tasks.h"
 #include "media/learning/common/target_histogram.h"
 #include "media/learning/mojo/public/mojom/learning_task_controller.mojom-blink.h"
-#include "media/mojo/mojom/media_metrics_provider.mojom-blink.h"
+#include "media/media_buildflags.h"
 #include "media/mojo/mojom/media_types.mojom-blink.h"
 #include "media/video/gpu_video_accelerator_factories.h"
 #include "media/video/supported_video_decoder_config.h"
@@ -67,12 +67,15 @@
 #include "third_party/blink/renderer/platform/media_capabilities/web_media_capabilities_info.h"
 #include "third_party/blink/renderer/platform/media_capabilities/web_media_configuration.h"
 #include "third_party/blink/renderer/platform/network/parsed_content_type.h"
-#include "third_party/blink/renderer/platform/peerconnection/transmission_encoding_info_handler.h"
 #include "third_party/blink/renderer/platform/wtf/functional.h"
 #include "third_party/blink/renderer/platform/wtf/vector.h"
 #include "ui/gfx/geometry/rect.h"
 #include "ui/gfx/geometry/size.h"
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "third_party/blink/renderer/platform/peerconnection/transmission_encoding_info_handler.h"
+#endif
+
 namespace blink {
 
 namespace {
@@ -819,12 +822,14 @@ ScriptPromise MediaCapabilities::encodingInfo(
   }
 
   if (configuration->type() == "transmission") {
+#if BUILDFLAG(ENABLE_WEBRTC)
     if (auto* handler = TransmissionEncodingInfoHandler::Instance()) {
       handler->EncodingInfo(ToWebMediaConfiguration(configuration),
                             WTF::Bind(&OnMediaCapabilitiesEncodingInfo,
                                       WrapPersistent(resolver)));
       return promise;
     }
+#endif
     resolver->Reject(MakeGarbageCollected<DOMException>(
         DOMExceptionCode::kInvalidStateError,
         "Platform error: could not get EncodingInfoHandler."));
diff --git a/chromium/third_party/blink/renderer/modules/mediastream/local_media_stream_audio_source.cc b/chromium/third_party/blink/renderer/modules/mediastream/local_media_stream_audio_source.cc
index de235d0db4f..b07ed18df12 100644
--- a/chromium/third_party/blink/renderer/modules/mediastream/local_media_stream_audio_source.cc
+++ b/chromium/third_party/blink/renderer/modules/mediastream/local_media_stream_audio_source.cc
@@ -8,6 +8,7 @@
 
 #include "base/strings/stringprintf.h"
 #include "media/audio/audio_source_parameters.h"
+#include "media/media_buildflags.h"
 #include "third_party/blink/public/mojom/mediastream/media_stream.mojom-blink.h"
 #include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
 #include "third_party/blink/public/platform/platform.h"
@@ -80,6 +81,7 @@ bool LocalMediaStreamAudioSource::EnsureSourceIsStarted() {
   if (!consumer_frame_)
     return false;
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   VLOG(1) << "Starting local audio input device (session_id="
           << device().session_id() << ") with audio parameters={"
           << GetAudioParameters().AsHumanReadableString() << "}.";
@@ -91,6 +93,10 @@ bool LocalMediaStreamAudioSource::EnsureSourceIsStarted() {
   source_->Initialize(GetAudioParameters(), this);
   source_->Start();
   return true;
+#else
+  LOG(ERROR) << "Audio Capture not supported";
+  return false;
+#endif
 }
 
 void LocalMediaStreamAudioSource::EnsureSourceIsStopped() {
diff --git a/chromium/third_party/blink/renderer/modules/mediastream/media_stream_renderer_factory.cc b/chromium/third_party/blink/renderer/modules/mediastream/media_stream_renderer_factory.cc
index 3c622fdffd8..24faad2eeaa 100644
--- a/chromium/third_party/blink/renderer/modules/mediastream/media_stream_renderer_factory.cc
+++ b/chromium/third_party/blink/renderer/modules/mediastream/media_stream_renderer_factory.cc
@@ -6,6 +6,7 @@
 
 #include <utility>
 
+#include "media/media_buildflags.h"
 #include "third_party/blink/public/platform/modules/mediastream/web_media_stream.h"
 #include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
 #include "third_party/blink/public/platform/platform.h"
@@ -34,11 +35,15 @@ namespace {
 // Note that if there are more than one open capture devices the function
 // will not be able to pick an appropriate device and return 0.
 base::UnguessableToken GetSessionIdForWebRtcAudioRenderer() {
+#if BUILDFLAG(ENABLE_WEBRTC)
   WebRtcAudioDeviceImpl* audio_device =
       PeerConnectionDependencyFactory::GetInstance()->GetWebRtcAudioDevice();
   return audio_device
              ? audio_device->GetAuthorizedDeviceSessionIdForAudioRenderer()
              : base::UnguessableToken();
+#else
+  return base::UnguessableToken();
+#endif
 }
 
 void SendLogMessage(const WTF::String& message) {
@@ -57,6 +62,7 @@ MediaStreamRendererFactory::GetVideoRenderer(
     const WebMediaStreamVideoRenderer::RepaintCB& repaint_cb,
     scoped_refptr<base::SingleThreadTaskRunner> io_task_runner,
     scoped_refptr<base::SingleThreadTaskRunner> main_render_task_runner) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   DCHECK(!web_stream.IsNull());
 
   DVLOG(1) << "MediaStreamRendererFactory::GetVideoRenderer stream:"
@@ -73,6 +79,10 @@ MediaStreamRendererFactory::GetVideoRenderer(
   return new MediaStreamVideoRendererSink(video_components[0].Get(), repaint_cb,
                                           std::move(io_task_runner),
                                           std::move(main_render_task_runner));
+#else
+  WebRtcLogMessage("Error: No WebRTC support.");
+  return nullptr;
+#endif
 }
 
 scoped_refptr<WebMediaStreamAudioRenderer>
@@ -81,6 +91,7 @@ MediaStreamRendererFactory::GetAudioRenderer(
     WebLocalFrame* web_frame,
     const WebString& device_id,
     base::RepeatingCallback<void()> on_render_error_callback) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   DCHECK(!web_stream.IsNull());
   SendLogMessage(String::Format("%s({web_stream_id=%s}, {device_id=%s})",
                                 __func__, web_stream.Id().Utf8().c_str(),
@@ -174,6 +185,10 @@ MediaStreamRendererFactory::GetAudioRenderer(
         "%s => (ERROR: CreateSharedAudioRendererProxy failed)", __func__));
   }
   return ret;
+#else
+  WebRtcLogMessage("Error: No WebRTC support.");
+  return nullptr;
+#endif
 }
 
 }  // namespace blink
diff --git a/chromium/third_party/blink/renderer/modules/mediastream/processed_local_audio_source.cc b/chromium/third_party/blink/renderer/modules/mediastream/processed_local_audio_source.cc
index ef0df3146eb..8678c7b5d4c 100644
--- a/chromium/third_party/blink/renderer/modules/mediastream/processed_local_audio_source.cc
+++ b/chromium/third_party/blink/renderer/modules/mediastream/processed_local_audio_source.cc
@@ -16,6 +16,7 @@
 #include "media/audio/audio_source_parameters.h"
 #include "media/base/channel_layout.h"
 #include "media/base/sample_rates.h"
+#include "media/media_buildflags.h"
 #include "media/webrtc/audio_processor_controls.h"
 #include "media/webrtc/webrtc_switches.h"
 #include "third_party/blink/public/mojom/mediastream/media_stream.mojom-blink.h"
@@ -206,6 +207,7 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
   if (device_is_modified)
     SetDevice(modified_device);
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   // Create the MediaStreamAudioProcessor, bound to the WebRTC audio device
   // module.
   WebRtcAudioDeviceImpl* const rtc_audio_device =
@@ -300,9 +302,13 @@ bool ProcessedLocalAudioSource::EnsureSourceIsStarted() {
   rtc_audio_device->AddAudioCapturer(this);
 
   return true;
+#else
+  return false;
+#endif
 }
 
 void ProcessedLocalAudioSource::EnsureSourceIsStopped() {
+#if BUILDFLAG(ENABLE_WEBRTC)
   DCHECK(GetTaskRunner()->BelongsToCurrentThread());
 
   if (!source_)
@@ -323,6 +329,9 @@ void ProcessedLocalAudioSource::EnsureSourceIsStopped() {
     audio_processor_->Stop();
 
   DVLOG(1) << "Stopped WebRTC audio pipeline for consumption.";
+#else
+  return;
+#endif
 }
 
 scoped_refptr<webrtc::AudioProcessorInterface>
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/BUILD.gn b/chromium/third_party/blink/renderer/modules/peerconnection/BUILD.gn
index c4651cf9818..a49e0771fcf 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/BUILD.gn
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/BUILD.gn
@@ -2,6 +2,7 @@
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
+import("//media/media_options.gni")
 import("//third_party/blink/renderer/modules/modules.gni")
 
 blink_modules_sources("peerconnection") {
@@ -173,6 +174,13 @@ blink_modules_sources("peerconnection") {
 
   public_deps = [ "//third_party/webrtc_overrides:webrtc_component" ]
   deps = [ "//third_party/abseil-cpp:absl" ]
+
+  if (!enable_webrtc) {
+    sources -= [
+      "peer_connection_dependency_factory.cc",
+      "peer_connection_dependency_factory.h",
+    ]
+  }
 }
 
 source_set("test_support") {
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/media_stream_video_webrtc_sink.cc b/chromium/third_party/blink/renderer/modules/peerconnection/media_stream_video_webrtc_sink.cc
index ef1b12c7c1b..8de4055164d 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/media_stream_video_webrtc_sink.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/media_stream_video_webrtc_sink.cc
@@ -269,6 +269,7 @@ MediaStreamVideoWebRtcSink::MediaStreamVideoWebRtcSink(
       ContentHintTypeToWebRtcContentHint(component->ContentHint()));
   video_track_->set_enabled(component->Enabled());
 
+#if BUILDFLAG(ENABLE_WEBRTC)
   source_adapter_ = base::MakeRefCounted<WebRtcVideoSourceAdapter>(
       factory->GetWebRtcNetworkTaskRunner(), video_source_.get(),
       refresh_interval,
@@ -282,7 +283,7 @@ MediaStreamVideoWebRtcSink::MediaStreamVideoWebRtcSink(
       ConvertToBaseRepeatingCallback(CrossThreadBindRepeating(
           &WebRtcVideoSourceAdapter::OnVideoFrameOnIO, source_adapter_)),
       false);
-
+#endif
   DVLOG(3) << "MediaStreamVideoWebRtcSink ctor() : is_screencast "
            << is_screencast;
 }
@@ -291,8 +292,10 @@ MediaStreamVideoWebRtcSink::~MediaStreamVideoWebRtcSink() {
   DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
   DVLOG(3) << "MediaStreamVideoWebRtcSink dtor().";
   weak_factory_.InvalidateWeakPtrs();
-  MediaStreamVideoSink::DisconnectFromTrack();
-  source_adapter_->ReleaseSourceOnMainThread();
+  if (source_adapter_) {
+    MediaStreamVideoSink::DisconnectFromTrack();
+    source_adapter_->ReleaseSourceOnMainThread();
+  }
 }
 
 void MediaStreamVideoWebRtcSink::OnEnabledChanged(bool enabled) {
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_certificate_generator.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_certificate_generator.cc
index 77c2a8c053b..44548003044 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_certificate_generator.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_certificate_generator.cc
@@ -10,13 +10,14 @@
 #include "base/bind.h"
 #include "base/macros.h"
 #include "base/single_thread_task_runner.h"
+#include "media/media_buildflags.h"
 #include "third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.h"
 #include "third_party/blink/renderer/platform/wtf/thread_safe_ref_counted.h"
 #include "third_party/webrtc/api/scoped_refptr.h"
 #include "third_party/webrtc/rtc_base/rtc_certificate.h"
 #include "third_party/webrtc/rtc_base/rtc_certificate_generator.h"
 
-namespace blink {
+    namespace blink {
 namespace {
 
 // A certificate generation request spawned by
@@ -88,6 +89,7 @@ void GenerateCertificateWithOptionalExpiration(
     blink::RTCCertificateCallback completion_callback,
     scoped_refptr<base::SingleThreadTaskRunner> task_runner) {
   DCHECK(key_params.IsValid());
+#if BUILDFLAG(ENABLE_WEBRTC)
   auto* pc_dependency_factory =
       blink::PeerConnectionDependencyFactory::GetInstance();
   pc_dependency_factory->EnsureInitialized();
@@ -97,6 +99,9 @@ void GenerateCertificateWithOptionalExpiration(
           task_runner, pc_dependency_factory->GetWebRtcNetworkTaskRunner());
   request->GenerateCertificateAsync(key_params, expires_ms,
                                     std::move(completion_callback));
+#else
+  std::move(completion_callback).Run(nullptr);
+#endif
 }
 
 }  // namespace
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.cc
index 82a155cd0aa..09fabeb6778 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.cc
@@ -48,6 +48,7 @@ std::unique_ptr<DtlsTransportProxy> CreateProxy(
     ExecutionContext* context,
     webrtc::DtlsTransportInterface* native_transport,
     DtlsTransportProxy::Delegate* delegate) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   LocalFrame* frame = To<LocalDOMWindow>(context)->GetFrame();
   scoped_refptr<base::SingleThreadTaskRunner> proxy_thread =
       frame->GetTaskRunner(TaskType::kNetworking);
@@ -57,6 +58,9 @@ std::unique_ptr<DtlsTransportProxy> CreateProxy(
 
   return DtlsTransportProxy::Create(*frame, proxy_thread, host_thread,
                                     native_transport, delegate);
+#else
+  return nullptr;
+#endif
 }
 
 }  // namespace
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_ice_transport.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_ice_transport.cc
index 984e4d1e6b3..9872de029fc 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_ice_transport.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_ice_transport.cc
@@ -4,6 +4,7 @@
 
 #include "third_party/blink/renderer/modules/peerconnection/rtc_ice_transport.h"
 
+#include "media/media_buildflags.h"
 #include "third_party/blink/public/platform/platform.h"
 #include "third_party/blink/public/web/web_local_frame.h"
 #include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_ice_gather_options.h"
@@ -69,9 +70,13 @@ class DtlsIceTransportAdapterCrossThreadFactory
 
   std::unique_ptr<IceTransportAdapter> ConstructOnWorkerThread(
       IceTransportAdapter::Delegate* delegate) override {
+#if BUILDFLAG(ENABLE_WEBRTC)
     DCHECK(ice_transport_);
     return std::make_unique<IceTransportAdapterImpl>(delegate,
                                                      std::move(ice_transport_));
+#else
+    return nullptr;
+#endif
   }
 
  private:
@@ -82,6 +87,7 @@ class DefaultIceTransportAdapterCrossThreadFactory
     : public IceTransportAdapterCrossThreadFactory {
  public:
   void InitializeOnMainThread(LocalFrame& frame) override {
+#if BUILDFLAG(ENABLE_WEBRTC)
     DCHECK(!port_allocator_);
     DCHECK(!async_resolver_factory_);
 
@@ -91,15 +97,20 @@ class DefaultIceTransportAdapterCrossThreadFactory
         frame.Client()->GetWebFrame());
     async_resolver_factory_ =
         rtc_dependency_factory->CreateAsyncResolverFactory();
+#endif
   }
 
   std::unique_ptr<IceTransportAdapter> ConstructOnWorkerThread(
       IceTransportAdapter::Delegate* delegate) override {
+#if BUILDFLAG(ENABLE_WEBRTC)
     DCHECK(port_allocator_);
     DCHECK(async_resolver_factory_);
     return std::make_unique<IceTransportAdapterImpl>(
         delegate, std::move(port_allocator_),
         std::move(async_resolver_factory_));
+#else
+    return nullptr;
+#endif
   }
 
  private:
@@ -110,6 +121,7 @@ class DefaultIceTransportAdapterCrossThreadFactory
 }  // namespace
 
 RTCIceTransport* RTCIceTransport::Create(ExecutionContext* context) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   scoped_refptr<base::SingleThreadTaskRunner> proxy_thread =
       context->GetTaskRunner(TaskType::kNetworking);
 
@@ -120,12 +132,16 @@ RTCIceTransport* RTCIceTransport::Create(ExecutionContext* context) {
   return MakeGarbageCollected<RTCIceTransport>(
       context, std::move(proxy_thread), std::move(host_thread),
       std::make_unique<DefaultIceTransportAdapterCrossThreadFactory>());
+#else
+  return nullptr;
+#endif
 }
 
 RTCIceTransport* RTCIceTransport::Create(
     ExecutionContext* context,
     rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport,
     RTCPeerConnection* peer_connection) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   scoped_refptr<base::SingleThreadTaskRunner> proxy_thread =
       context->GetTaskRunner(TaskType::kNetworking);
 
@@ -138,6 +154,9 @@ RTCIceTransport* RTCIceTransport::Create(
       std::make_unique<DtlsIceTransportAdapterCrossThreadFactory>(
           std::move(ice_transport)),
       peer_connection);
+#else
+  return nullptr;
+#endif
 }
 
 RTCIceTransport* RTCIceTransport::Create(
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.cc
index 47a1fa272e0..fd5dff1a56b 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_peer_connection.cc
@@ -789,12 +789,14 @@ RTCPeerConnection::RTCPeerConnection(
     peer_handler_ =
         std::move(g_create_rpc_peer_connection_handler_callback_.Get()).Run();
   } else {
+#if BUILDFLAG(ENABLE_WEBRTC)
     peer_handler_ =
         PeerConnectionDependencyFactory::GetInstance()
             ->CreateRTCPeerConnectionHandler(
                 this, window->GetTaskRunner(TaskType::kInternalMedia),
                 force_encoded_audio_insertable_streams_,
                 force_encoded_video_insertable_streams_);
+#endif
   }
 
   if (!peer_handler_) {
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_quic_transport.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_quic_transport.cc
index 1d0953d5835..ea91a48696b 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_quic_transport.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_quic_transport.cc
@@ -4,6 +4,7 @@
 
 #include "third_party/blink/renderer/modules/peerconnection/rtc_quic_transport.h"
 
+#include "media/media_buildflags.h"
 #include "net/quic/platform/impl/quic_chromium_clock.h"
 #include "net/quic/quic_chromium_alarm_factory.h"
 #include "third_party/blink/public/platform/platform.h"
@@ -89,10 +90,14 @@ RTCQuicTransport* RTCQuicTransport::Create(
     RTCIceTransport* transport,
     const HeapVector<Member<RTCCertificate>>& certificates,
     ExceptionState& exception_state) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   return Create(context, transport, certificates, exception_state,
                 std::make_unique<DefaultP2PQuicTransportFactory>(
                     PeerConnectionDependencyFactory::GetInstance()
                         ->GetWebRtcNetworkTaskRunner()));
+#else
+  return nullptr;
+#endif
 }
 
 RTCQuicTransport* RTCQuicTransport::Create(
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc
index 2a6c88f6e4e..df57cb25748 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc
@@ -4,6 +4,7 @@
 
 #include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.h"
 
+#include "media/media_buildflags.h"
 #include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_insertable_streams.h"
 #include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_rtcp_parameters.h"
 #include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_rtp_capabilities.h"
@@ -15,7 +16,6 @@
 #include "third_party/blink/renderer/core/loader/document_loader.h"
 #include "third_party/blink/renderer/core/streams/readable_stream.h"
 #include "third_party/blink/renderer/core/streams/writable_stream.h"
-#include "third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_dtls_transport.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_underlying_sink.h"
 #include "third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_underlying_source.h"
@@ -34,6 +34,10 @@
 #include "third_party/blink/renderer/platform/wtf/text/wtf_string.h"
 #include "third_party/webrtc/api/rtp_parameters.h"
 
+#if BUILDFLAG(ENABLE_WEBRTC)
+#include "third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.h"
+#endif
+
 namespace blink {
 
 RTCRtpReceiver::RTCRtpReceiver(RTCPeerConnection* pc,
@@ -275,6 +279,7 @@ void RTCRtpReceiver::Trace(Visitor* visitor) const {
 }
 
 RTCRtpCapabilities* RTCRtpReceiver::getCapabilities(const String& kind) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (kind != "audio" && kind != "video")
     return nullptr;
 
@@ -321,6 +326,9 @@ RTCRtpCapabilities* RTCRtpReceiver::getCapabilities(const String& kind) {
   capabilities->setHeaderExtensions(header_extensions);
 
   return capabilities;
+#else
+  return nullptr;
+#endif
 }
 
 RTCRtpReceiveParameters* RTCRtpReceiver::getParameters() {
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc
index 83d2e0ca06a..a3cce3fc662 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc
@@ -9,6 +9,7 @@
 #include <tuple>
 #include <utility>
 
+#include "media/media_buildflags.h"
 #include "third_party/blink/renderer/bindings/core/v8/script_promise_resolver.h"
 #include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_insertable_streams.h"
 #include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_rtcp_parameters.h"
@@ -720,6 +721,7 @@ void RTCRtpSender::Trace(Visitor* visitor) const {
 }
 
 RTCRtpCapabilities* RTCRtpSender::getCapabilities(const String& kind) {
+#if BUILDFLAG(ENABLE_WEBRTC)
   if (kind != "audio" && kind != "video")
     return nullptr;
 
@@ -773,6 +775,9 @@ RTCRtpCapabilities* RTCRtpSender::getCapabilities(const String& kind) {
   capabilities->setHeaderExtensions(header_extensions);
 
   return capabilities;
+#else
+  return nullptr;
+#endif
 }
 
 void RTCRtpSender::RegisterEncodedAudioStreamCallback() {
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_transceiver.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_transceiver.cc
index fc46e57ab85..eb7103179e9 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_transceiver.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_rtp_transceiver.cc
@@ -35,6 +35,7 @@ String TransceiverDirectionToString(
       NOTREACHED();
       return String();
   }
+  NOTREACHED();
 }
 
 String OptionalTransceiverDirectionToString(
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_sctp_transport.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_sctp_transport.cc
index 2fedeb937be..f2fe1b794ad 100644
--- a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_sctp_transport.cc
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_sctp_transport.cc
@@ -7,6 +7,7 @@
 #include <limits>
 #include <memory>
 
+#include "media/media_buildflags.h"
 #include "third_party/blink/public/web/web_local_frame.h"
 #include "third_party/blink/renderer/core/dom/events/event.h"
 #include "third_party/blink/renderer/core/frame/local_dom_window.h"
@@ -64,8 +65,13 @@ RTCSctpTransport::RTCSctpTransport(
     : RTCSctpTransport(context,
                        native_transport,
                        context->GetTaskRunner(TaskType::kNetworking),
+#if BUILDFLAG(ENABLE_WEBRTC)
                        PeerConnectionDependencyFactory::GetInstance()
-                           ->GetWebRtcNetworkTaskRunner()) {}
+                           ->GetWebRtcNetworkTaskRunner()
+#else
+                       nullptr
+#endif
+                       ) {}
 
 RTCSctpTransport::RTCSctpTransport(
     ExecutionContext* context,
diff --git a/chromium/third_party/blink/renderer/platform/BUILD.gn b/chromium/third_party/blink/renderer/platform/BUILD.gn
index f51deafc4c8..a63f64a0130 100644
--- a/chromium/third_party/blink/renderer/platform/BUILD.gn
+++ b/chromium/third_party/blink/renderer/platform/BUILD.gn
@@ -1717,6 +1717,27 @@ component("platform") {
     deps += [ "//third_party/libavif" ]
   }
 
+  if (!enable_webrtc) {
+    sources -= [
+      "peerconnection/audio_codec_factory.cc",
+      "peerconnection/rtc_video_decoder_adapter.cc",
+      "peerconnection/rtc_video_decoder_adapter.h",
+      "peerconnection/rtc_video_decoder_factory.cc",
+      "peerconnection/rtc_video_decoder_factory.h",
+      "peerconnection/rtc_video_encoder.cc",
+      "peerconnection/rtc_video_encoder.h",
+      "peerconnection/rtc_video_encoder_factory.cc",
+      "peerconnection/rtc_video_encoder_factory.h",
+      "peerconnection/transmission_encoding_info_handler.cc",
+      "peerconnection/transmission_encoding_info_handler.h",
+      "peerconnection/video_codec_factory.cc",
+      "peerconnection/video_codec_factory.h",
+    ]
+    deps -= [
+      "//third_party/webrtc_overrides:webrtc_component"
+    ]
+  }
+
   if (current_cpu == "x86" || current_cpu == "x64") {
     deps += [ ":blink_x86_avx" ]
     sources += [ "audio/cpu/x86/audio_delay_dsp_kernel_sse2.cc" ]
diff --git a/chromium/tools/ipc_fuzzer/message_tools/message_list.cc b/chromium/tools/ipc_fuzzer/message_tools/message_list.cc
index 09d2bfbd4a2..acc238076f8 100644
--- a/chromium/tools/ipc_fuzzer/message_tools/message_list.cc
+++ b/chromium/tools/ipc_fuzzer/message_tools/message_list.cc
@@ -55,6 +55,9 @@ static bool check_msgtable() {
   exemptions.push_back(NaClMsgStart);
 #endif  // !BUILDFLAG(ENABLE_NACL)
 
+#if !BUILDFLAG(ENABLE_WEBRTC)
+  exemptions.push_back(WebRtcLoggingMsgStart);
+#endif
 
 #if !defined(OS_ANDROID)
   exemptions.push_back(EncryptedMediaMsgStart);
author	Allan Sandfeld Jensen <allan.jensen@qt.io>	2020-08-03 13:19:41 +0200
committer	Allan Sandfeld Jensen <allan.jensen@qt.io>	2020-12-03 17:29:43 +0100
commit	7eb6adb6cac7c604d4717bd3ab0f0e8cca452fe5 (patch)
tree	5218647eee4c52fad271362896e49c11ef1a1658
parent	0199422d32f37be484483a523f7e88a3e1def4f2 (diff)
download	qtwebengine-chromium-7eb6adb6cac7c604d4717bd3ab0f0e8cca452fe5.tar.gz