Move talk/media to webrtc/media

talk/app/webrtc/rtpsender.h

 /*
  * libjingle
  * Copyright 2015 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 19 matching lines @@
 // transport (provided by AudioProviderInterface/VideoProviderInterface)
 
 #ifndef TALK_APP_WEBRTC_RTPSENDER_H_
 #define TALK_APP_WEBRTC_RTPSENDER_H_
 
 #include <string>
 
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/rtpsenderinterface.h"
 #include "talk/app/webrtc/statscollector.h"
-#include "talk/media/base/audiorenderer.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/audiorenderer.h"
 
 namespace webrtc {
 
 // LocalAudioSinkAdapter receives data callback as a sink to the local
 // AudioTrack, and passes the data to the sink of AudioRenderer.
 class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
                               public cricket::AudioRenderer {
  public:
   LocalAudioSinkAdapter();
   virtual ~LocalAudioSinkAdapter();
@@ skipping 132 matching lines @@
   VideoProviderInterface* provider_;
   rtc::scoped_refptr<VideoTrackInterface> track_;
   uint32_t ssrc_ = 0;
   bool cached_track_enabled_ = false;
   bool stopped_ = false;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_RTPSENDER_H_