Revert of Replace FileWrapper with File (in audio_device)

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_checker.h"
 #include "webrtc/modules/audio_device/include/audio_device.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
 
 class AudioDeviceObserver;
 
 class AudioDeviceBuffer {
  public:
   AudioDeviceBuffer();
   virtual ~AudioDeviceBuffer();
 
-  void SetId(uint32_t id) {}
+  void SetId(uint32_t id) {};
   int32_t RegisterAudioCallback(AudioTransport* audio_callback);
 
   int32_t InitPlayout();
   int32_t InitRecording();
 
   int32_t SetRecordingSampleRate(uint32_t fsHz);
   int32_t SetPlayoutSampleRate(uint32_t fsHz);
   int32_t RecordingSampleRate() const;
   int32_t PlayoutSampleRate() const;
 
@@ skipping 153 matching lines @@
   // all level measurements since the last log has been exactly zero.
   // In other words: this counter is incremented only if 20 measurements
   // (two per second) in a row equals zero. The member is only incremented on
   // the task queue and max once every 10th second.
   size_t num_rec_level_is_zero_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_