Move talk/media to webrtc/media

talk/app/webrtc/remoteaudiosource.h

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 15 matching lines @@
  */
 
 #ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
 #define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
 
 #include <list>
 #include <string>
 
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/notifier.h"
-#include "talk/media/base/audiorenderer.h"
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/media/base/audiorenderer.h"
 
 namespace rtc {
 struct Message;
 class Thread;
 }  // namespace rtc
 
 namespace webrtc {
 
 class AudioProviderInterface;
 
@@ skipping 38 matching lines @@
   AudioObserverList audio_observers_;
   rtc::CriticalSection sink_lock_;
   std::list<AudioTrackSinkInterface*> sinks_;
   rtc::Thread* const main_thread_;
   SourceState state_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_