Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/test/fake_audio_device.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
 #define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/event.h"
-#include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
+#include "webrtc/rtc_base/array_view.h"
+#include "webrtc/rtc_base/buffer.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/event.h"
+#include "webrtc/rtc_base/platform_thread.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class EventTimerWrapper;
 
 namespace test {
 
 // FakeAudioDevice implements an AudioDevice module that can act both as a
 // capturer and a renderer. It will use 10ms audio frames.
@@ skipping 102 matching lines @@
   std::vector<int16_t> playout_buffer_ GUARDED_BY(lock_);
   rtc::BufferT<int16_t> recording_buffer_ GUARDED_BY(lock_);
 
   std::unique_ptr<EventTimerWrapper> tick_;
   rtc::PlatformThread thread_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_