Move talk/app/webrtc to webrtc/api

webrtc/api/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 11 matching lines @@
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 // This file contains classes that implement RtpSenderInterface.
 // An RtpSender associates a MediaStreamTrackInterface with an underlying
 // transport (provided by AudioProviderInterface/VideoProviderInterface)
 
-#ifndef TALK_APP_WEBRTC_RTPSENDER_H_
-#define TALK_APP_WEBRTC_RTPSENDER_H_
+#ifndef WEBRTC_API_RTPSENDER_H_
+#define WEBRTC_API_RTPSENDER_H_
 
 #include <string>
 
-#include "talk/app/webrtc/mediastreamprovider.h"
-#include "talk/app/webrtc/rtpsenderinterface.h"
-#include "talk/app/webrtc/statscollector.h"
+#include "webrtc/api/mediastreamprovider.h"
+#include "webrtc/api/rtpsenderinterface.h"
+#include "webrtc/api/statscollector.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,
@@ skipping 135 matching lines @@
   std::string stream_id_;
   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_
+#endif  // WEBRTC_API_RTPSENDER_H_