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 571 matching lines @@
       "media/rtc_certificate.cc",
       "media/rtc_certificate.h",
       "media/rtc_certificate_generator.cc",
       "media/rtc_certificate_generator.h",
       "media/rtc_data_channel_handler.cc",
       "media/rtc_data_channel_handler.h",
       "media/rtc_dtmf_sender_handler.cc",
       "media/rtc_dtmf_sender_handler.h",
       "media/rtc_peer_connection_handler.cc",
       "media/rtc_peer_connection_handler.h",
+      "media/rtc_stats.cc",

[perkj_chrome, 2016/09/07 14:34:36]

these files depend on webrtc third_party? Then please move to media/webrtc

[hbos_chromium, 2016/09/08 08:38:31]

Done.

+      "media/rtc_stats.h",
       "media/secure_display_link_tracker.h",
       "media/speech_recognition_audio_sink.cc",
       "media/speech_recognition_audio_sink.h",
       "media/track_audio_renderer.cc",
       "media/track_audio_renderer.h",
       "media/user_media_client_impl.cc",
       "media/video_track_adapter.cc",
       "media/video_track_adapter.h",
       "media/video_track_recorder.cc",
       "media/video_track_recorder.h",
@@ skipping 82 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",
     ]
   }
 }