WebRTCStats added for surfacing RTCStats from WebRTC to Blink.

content/renderer/BUILD.gn

 # Copyright 2014 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 import("//build/config/features.gni")
 import("//build/config/ui.gni")
 import("//media/media_options.gni")
 import("//third_party/webrtc/build/webrtc.gni")
 import("//tools/ipc_fuzzer/ipc_fuzzer.gni")
 
@@ skipping 595 matching lines @@
       "media/webrtc/media_stream_track_metrics.cc",
       "media/webrtc/media_stream_track_metrics.h",
       "media/webrtc/media_stream_video_webrtc_sink.cc",
       "media/webrtc/media_stream_video_webrtc_sink.h",
       "media/webrtc/peer_connection_dependency_factory.cc",
       "media/webrtc/peer_connection_dependency_factory.h",
       "media/webrtc/peer_connection_remote_audio_source.cc",
       "media/webrtc/peer_connection_remote_audio_source.h",
       "media/webrtc/processed_local_audio_source.cc",
       "media/webrtc/processed_local_audio_source.h",
+      "media/webrtc/rtc_stats.cc",
+      "media/webrtc/rtc_stats.h",
       "media/webrtc/stun_field_trial.cc",
       "media/webrtc/stun_field_trial.h",
       "media/webrtc/track_observer.cc",
       "media/webrtc/track_observer.h",
       "media/webrtc/webrtc_audio_sink.cc",
       "media/webrtc/webrtc_audio_sink.h",
       "media/webrtc/webrtc_media_stream_adapter.cc",
       "media/webrtc/webrtc_media_stream_adapter.h",
       "media/webrtc/webrtc_video_capturer_adapter.cc",
       "media/webrtc/webrtc_video_capturer_adapter.h",
@@ skipping 58 matching lines @@
       "//third_party/webrtc/api:libjingle_peerconnection",
       "//third_party/webrtc/base:rtc_base",
       "//third_party/webrtc/common_video",
       "//third_party/webrtc/media:rtc_media",
       "//third_party/webrtc/modules/audio_device",
       "//third_party/webrtc/modules/audio_processing",
       "//third_party/webrtc/modules/video_coding:webrtc_h264",
       "//third_party/webrtc/p2p:libstunprober",
       "//third_party/webrtc/p2p:rtc_p2p",
       "//third_party/webrtc/pc:rtc_pc",
+      "//third_party/webrtc/stats",
       "//third_party/webrtc/system_wrappers",
     ]
     if (rtc_use_h264) {
       deps += [ "//third_party/openh264:encoder" ]
     }
   } else {
     sources += [
       "media/webrtc_logging.h",
       "media/webrtc_logging_noop.cc",
     ]
@@ skipping 203 matching lines @@
     # For the defines in mojo_media_config.
     public_configs = [ "//media/mojo/services:mojo_media_config" ]
   }
 
   if (!is_component_build) {
     public_deps = [
       ":renderer",
     ]
   }
 }