Move talk/app/webrtc to webrtc/api

webrtc/api/remoteaudiosource.cc

 /*
  * libjingle
  * Copyright 2014 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/remoteaudiosource.h"
+#include "webrtc/api/remoteaudiosource.h"
 
 #include <algorithm>
 #include <functional>
 #include <utility>
 
-#include "talk/app/webrtc/mediastreamprovider.h"
+#include "webrtc/api/mediastreamprovider.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
 
 namespace webrtc {
 
 class RemoteAudioSource::MessageHandler : public rtc::MessageHandler {
  public:
   explicit MessageHandler(RemoteAudioSource* source) : source_(source) {}
 
@@ skipping 122 matching lines @@
 }
 
 void RemoteAudioSource::OnMessage(rtc::Message* msg) {
   RTC_DCHECK(main_thread_->IsCurrent());
   sinks_.clear();
   state_ = MediaSourceInterface::kEnded;
   FireOnChanged();
 }
 
 }  // namespace webrtc