Move talk/app/webrtc to webrtc/api

webrtc/api/test/fakeaudiocapturemodule.h

 /*
  * libjingle
  * Copyright 2012 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 16 matching lines @@
 
 // This class implements an AudioCaptureModule that can be used to detect if
 // audio is being received properly if it is fed by another AudioCaptureModule
 // in some arbitrary audio pipeline where they are connected. It does not play
 // out or record any audio so it does not need access to any hardware and can
 // therefore be used in the gtest testing framework.
 
 // Note P postfix of a function indicates that it should only be called by the
 // processing thread.
 
-#ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
-#define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
 namespace rtc {
@@ skipping 228 matching lines @@
   int frames_received_;
 
   // Protects variables that are accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_;
   // Protects |audio_callback_| that is accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_callback_;
 };
 
-#endif  // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_
+#endif  // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_