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

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 844 matching lines @@
   std::vector<VoiceSenderInfo> senders;
   std::vector<VoiceReceiverInfo> receivers;
   RtpCodecParametersMap send_codecs;
   RtpCodecParametersMap receive_codecs;
 };
 
 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;
 };
 
 struct DataMediaInfo {
   void Clear() {
     senders.clear();
     receivers.clear();
   }
   std::vector<DataSenderInfo> senders;
@@ skipping 193 matching lines @@
   // The |ssrc| must correspond to a registered send stream.
   virtual bool SetVideoSend(
       uint32_t ssrc,
       bool enable,
       const VideoOptions* options,
       rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0;
   // Sets the sink object to be used for the specified stream.
   // 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;
 };
 
 enum DataMessageType {
   // Chrome-Internal use only.  See SctpDataMediaChannel for the actual PPID
   // values.
   DMT_NONE = 0,
   DMT_CONTROL = 1,
   DMT_BINARY = 2,
@@ skipping 109 matching lines @@
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_