Move talk/app/webrtc to webrtc/api

webrtc/api/rtpreceiver.cc

 /*
  * 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,
  *     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.
  */
 
-#include "talk/app/webrtc/rtpreceiver.h"
+#include "webrtc/api/rtpreceiver.h"
 
-#include "talk/app/webrtc/videosourceinterface.h"
+#include "webrtc/api/videosourceinterface.h"
 
 namespace webrtc {
 
 AudioRtpReceiver::AudioRtpReceiver(AudioTrackInterface* track,
                                    uint32_t ssrc,
                                    AudioProviderInterface* provider)
     : id_(track->id()),
       track_(track),
       ssrc_(ssrc),
       provider_(provider),
@@ skipping 57 matching lines @@
 void VideoRtpReceiver::Stop() {
   // TODO(deadbeef): Need to do more here to fully stop receiving packets.
   if (!provider_) {
     return;
   }
   provider_->SetVideoPlayout(ssrc_, false, nullptr);
   provider_ = nullptr;
 }
 
 }  // namespace webrtc