Support for remote audio into tracks

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.
  */
 
 #ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioDecoder;
+class AudioSinkInterface;
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioReceiveStream : public ReceiveStream {
  public:
   struct Stats {
     uint32_t remote_ssrc = 0;
@@ skipping 57 matching lines @@
     // AudioDecoder instances once the Config is submitted to
     // Call::CreateReceiveStream().
     // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
     std::map<uint8_t, AudioDecoder*> decoder_map;
 
     // TODO(pbos): Remove config option once combined A/V BWE is always on.
     bool combined_audio_video_bwe = false;
   };
 
   virtual Stats GetStats() const = 0;
+
+  // Sets an audio sync that receives unmixed audio from the receive stream.

[the sun, 2015/12/11 16:32:05]

s/sync/sink

I think it is worth to mention that audio must still be pulled through the
AudioTransport interface and is only tapped here as a side effect.

[tommi (sloooow) - chröme, 2015/12/11 17:51:18]

Thanks.
Yes good point. Added documentation.

+  // Ownership of the sink is passed to the stream and can be used by the
+  // caller to do lifetime management (i.e. when the sink's dtor is called).
+  // Only one sink can be set and passing a null sink, clears an existing one.
+  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_