Move talk/app/webrtc to webrtc/api

webrtc/api/rtpsenderinterface.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 interfaces for RtpSenders
 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
-#ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
-#define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+#ifndef WEBRTC_API_RTPSENDERINTERFACE_H_
+#define WEBRTC_API_RTPSENDERINTERFACE_H_
 
 #include <string>
 
-#include "talk/app/webrtc/proxy.h"
-#include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/session/media/mediasession.h"
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/proxy.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
 class RtpSenderInterface : public rtc::RefCountInterface {
  public:
   // Returns true if successful in setting the track.
   // Fails if an audio track is set on a video RtpSender, or vice-versa.
   virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
@@ skipping 31 matching lines @@
 PROXY_CONSTMETHOD0(uint32_t, ssrc)
 PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
 PROXY_CONSTMETHOD0(std::string, id)
 PROXY_METHOD1(void, set_stream_id, const std::string&)
 PROXY_CONSTMETHOD0(std::string, stream_id)
 PROXY_METHOD0(void, Stop)
 END_PROXY()
 
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
+#endif  // WEBRTC_API_RTPSENDERINTERFACE_H_