Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.

webrtc/media/base/mediachannel.h

Index: webrtc/media/base/mediachannel.h
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 6f14b7aa2e5ec95c53d873feb0a6bbbf3ed56d06..db2fa375f8b802e4b160c2a6443c36246b278cdb 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -863,13 +863,11 @@ struct VideoMediaInfo {
   void Clear() {
     senders.clear();
     receivers.clear();
-    bw_estimations.clear();
     send_codecs.clear();
     receive_codecs.clear();
   }
   std::vector<VideoSenderInfo> senders;
   std::vector<VideoReceiverInfo> receivers;
-  std::vector<BandwidthEstimationInfo> bw_estimations;
   RtpCodecParametersMap send_codecs;
   RtpCodecParametersMap receive_codecs;
 };
@@ -1083,6 +1081,7 @@ class VideoMediaChannel : public MediaChannel {
   // If SSRC is 0, the sink is used for the 'default' stream.
   virtual bool SetSink(uint32_t ssrc,
                        rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0;
+  virtual void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0;
   // Gets quality stats for the channel.
   virtual bool GetStats(VideoMediaInfo* info) = 0;
 };