Storing raw audio sink for default audio track.

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 94 matching lines @@
 
     // 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 sink that receives unmixed audio from the receive stream.
   // 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.
+  // 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_