Move talk/app/webrtc to webrtc/api

webrtc/api/rtpsender.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/rtpsender.h"
+#include "webrtc/api/rtpsender.h"
 
-#include "talk/app/webrtc/localaudiosource.h"
-#include "talk/app/webrtc/videosourceinterface.h"
+#include "webrtc/api/localaudiosource.h"
+#include "webrtc/api/videosourceinterface.h"
 #include "webrtc/base/helpers.h"
 
 namespace webrtc {
 
 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
 
 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
   rtc::CritScope lock(&lock_);
   if (sink_)
     sink_->OnClose();
@@ skipping 297 matching lines @@
   RTC_DCHECK(!stopped_ && can_send_track());
   const cricket::VideoOptions* options = nullptr;
   VideoSourceInterface* source = track_->GetSource();
   if (track_->enabled() && source) {
     options = source->options();
   }
   provider_->SetVideoSend(ssrc_, track_->enabled(), options);
 }
 
 }  // namespace webrtc