Storing raw audio sink for default audio track.

webrtc/audio_receive_stream.h

Index: webrtc/audio_receive_stream.h
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
index daf45985d33c2523f85ee26891f4134643b66243..7590329cfe29f8f6979b1c0757dc3ad5568a004d 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/audio_receive_stream.h
@@ -15,7 +15,7 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
@@ -106,12 +106,12 @@ class AudioReceiveStream : public ReceiveStream {
   // 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;
+  virtual void SetSink(rtc::scoped_refptr<AudioSinkInterface> sink) = 0;
 };
 }  // namespace webrtc