Move talk/app/webrtc to webrtc/api

webrtc/api/rtpsender.h

 /*
- * libjingle
- * Copyright 2015 Google Inc.
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
  *
- * 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,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * 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.
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
  */
 
 // 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 "talk/media/base/audiorenderer.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"
 
 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 {
@@ skipping 134 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_