WebRTCStats added for surfacing RTCStats from WebRTC to Blink.

content/renderer/BUILD.gn

Index: content/renderer/BUILD.gn
diff --git a/content/renderer/BUILD.gn b/content/renderer/BUILD.gn
index c192a878dd8a5b92b08e0d412a5c71d34ad8efc1..ce31538840e583e35e4c2f401395d37c387dfafa 100644
--- a/content/renderer/BUILD.gn
+++ b/content/renderer/BUILD.gn
@@ -589,6 +589,8 @@ target(link_target_type, "renderer") {
       "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",
@@ -691,6 +693,7 @@ target(link_target_type, "renderer") {
       "//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) {