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 845 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;
+  virtual bool FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0;

[Taylor Brandstetter, 2017/05/07 21:30:44]

nit: Why not "FillBandwidthEstimationInfo(BandwidthEstimationInfo*)"?

Also, what's the advantage of adding a new method over just adding
"bw_estimations" to VoiceMediaInfo? Is there a time FillBitrateInfo will be
called outside of the stats context?

[stefan-webrtc, 2017/05/08 07:12:56]

It only fills the "bitrate parts" (rtx, video bitrate) of the
BandwidthEstimationInfo, since that's the part that isn't possible to get
through Call::GetStats, since they are statistics of the send streams. I think
it would make sense to change this in the future, but I added this method now
since I don't know how we should change it, and it probably deserves a separate
CL.
I find it strange that the Call::Stats are collected via WebRtcVideoEngine2 (and
with your suggestion, WebRtcVoiceEngine). They are on a peerconnection/session
level and should only be collected once. That's why I decided to move it. I
don't think there's bw_estimations for both audio and video, there's only
bw_estimations for the currently active transport. Our current design is to have
one BWE per session.

[Taylor Brandstetter, 2017/05/08 17:47:41]

Can you describe this in a comment above FillBitrateInfo, maybe with a TODO?
I understand when you explain it that way. Can you describe this more in the CL
description?

   // 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_