Support for remote audio into tracks

talk/app/webrtc/mediastreamprovider.h

Index: talk/app/webrtc/mediastreamprovider.h
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
index a78b55a68c0a73e95cae68b9ab923c0b5a10200b..9ffbcad176ce02b2af38184c4b63a069e9af301d 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -28,11 +28,12 @@
 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 
+#include "talk/media/base/audiorenderer.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace cricket {
 
-class AudioRenderer;
 class VideoCapturer;
 class VideoRenderer;
 struct AudioOptions;
@@ -42,6 +43,8 @@ struct VideoOptions;
 
 namespace webrtc {
 
+class AudioSinkInterface;
+
 // TODO(deadbeef): Change the key from an ssrc to a "sender_id" or
 // "receiver_id" string, which will be the MSID in the short term and MID in
 // the long term.
@@ -67,6 +70,13 @@ class AudioProviderInterface {
   // |volume| is in the range of [0, 10].
   virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0;
 
+  // Allows for setting a direct audio sink for an incoming audio source.
+  // Only one audio sink is supported per ssrc and ownership of the sink is
+  // passed to the provider.
+  virtual void SetRawAudioSink(
+      uint32_t ssrc,
+      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
+
  protected:
   virtual ~AudioProviderInterface() {}
 };