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 dadb55bfdbae513e4bea0e255ecb377e7faf73b2..8741429fca5719cca351042ac6438961dc3f10cd 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -862,13 +862,11 @@ 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;
 };
@@ -1082,6 +1080,15 @@ 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;
+  // This fills the "bitrate parts" (rtx, video bitrate) of the
+  // BandwidthEstimationInfo, since that part that isn't possible to get
+  // through webrtc::Call::GetStats, as they are statistics of the send
+  // streams.
+  // TODO(holmer): We should change this so that either BWE graphs doesn't
+  // need access to bitrates of the streams, or change the (RTC)StatsCollector
+  // so that it's getting the send stream stats separately by calling
+  // GetStats(), and merges with BandwidthEstimationInfo by itself.
+  virtual void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0;
   // Gets quality stats for the channel.
   virtual bool GetStats(VideoMediaInfo* info) = 0;
 };