Wire-up BWE feedback for audio receive streams.

webrtc/audio_receive_stream.h

 /*
  *  Copyright (c) 2015 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 55 matching lines @@
     // Receive-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
       // Synchronization source (stream identifier) to be received.
       uint32_t remote_ssrc = 0;
 
       // Sender SSRC used for sending RTCP (such as receiver reports).
       uint32_t local_ssrc = 0;
 
+      // See draft-holmer-rmcat-transport-wide-cc-extensions for details.

[the sun, 2015/12/20 23:16:15]

// Enable send side bandwidth estimation.
// See .... for details.
?

[stefan-webrtc, 2015/12/21 08:01:29]

Done.

+      bool transport_cc = false;
+
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
     Transport* receive_transport = nullptr;
     Transport* rtcp_send_transport = nullptr;
 
     // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
     // level components.
     // TODO(solenberg): Remove when VoiceEngine channels are created outside
@@ skipping 23 matching lines @@
   // Only one sink can be set and passing a null sink, clears an existing one.
   // NOTE: Audio must still somehow be pulled through AudioTransport for audio
   // to stream through this sink. In practice, this happens if mixed audio
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
   virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_