Update includes for webrtc/{base => rtc_base} rename (1/3)

webrtc/modules/audio_device/audio_device_buffer.h

 /*
  *  Copyright (c) 2012 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_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
 
-#include "webrtc/base/buffer.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/task_queue.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/rtc_base/buffer.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/task_queue.h"
+#include "webrtc/rtc_base/thread_annotations.h"
+#include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Delta times between two successive playout callbacks are limited to this
 // value before added to an internal array.
 const size_t kMaxDeltaTimeInMs = 500;
 // TODO(henrika): remove when no longer used by external client.
 const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
@@ skipping 226 matching lines @@
 // Should *never* be defined in production builds. Only used for testing.
 // When defined, the output signal will be replaced by a sinus tone at 440Hz.
 #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
   double phase_ ACCESS_ON(playout_thread_checker_);
 #endif
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_