Support for remote audio into tracks

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 356a3a39289910fa98aa7fc852e66e0e07535ed7..2f7eb148da0df63e72eb1661914ba5797c0ba65c 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/audio_receive_stream.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
@@ -23,6 +24,7 @@
 namespace webrtc {
 
 class AudioDecoder;
+class AudioSinkInterface;
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
@@ -100,6 +102,12 @@ class AudioReceiveStream : public ReceiveStream {
   };
 
   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