Move talk/app/webrtc to webrtc/api

talk/media/webrtc/fakewebrtccall.h

 /*
  * 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 21 matching lines @@
 //   webrtc::AudioSendStream
 //   webrtc::AudioReceiveStream
 //   webrtc::VideoSendStream
 //   webrtc::VideoReceiveStream
 
 #ifndef TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
 #define TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_
 
 #include <vector>
 
-#include "webrtc/call.h"
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
+#include "webrtc/call.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace cricket {
 class FakeAudioSendStream final : public webrtc::AudioSendStream {
  public:
   struct TelephoneEvent {
     int payload_type = -1;
     uint8_t event_code = 0;
@@ skipping 205 matching lines @@
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_