Move talk/app/webrtc to webrtc/api

talk/media/webrtc/fakewebrtccall.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,
@@ skipping 13 matching lines @@
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/media/webrtc/fakewebrtccall.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "talk/media/base/rtputils.h"
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/audio/audio_sink.h"
 
 namespace cricket {
 FakeAudioSendStream::FakeAudioSendStream(
     const webrtc::AudioSendStream::Config& config) : config_(config) {
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
 const webrtc::AudioSendStream::Config&
     FakeAudioSendStream::GetConfig() const {
   return config_;
@@ skipping 387 matching lines @@
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket