Remove more CriticalSectionWrappers.

webrtc/modules/audio_device/dummy/file_audio_device.h

 /*
  *  Copyright (c) 2014 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_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
 #define WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
 
 #include <stdio.h>
 
 #include <memory>
 #include <string>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 namespace rtc {
 class PlatformThread;
 }  // namespace rtc
 
 namespace webrtc {
 class EventWrapper;
 
 // This is a fake audio device which plays audio from a file as its microphone
@@ skipping 137 matching lines @@
   bool PlayThreadProcess();
 
   int32_t _playout_index;
   int32_t _record_index;
   AudioDeviceModule::BufferType _playBufType;
   AudioDeviceBuffer* _ptrAudioBuffer;
   int8_t* _recordingBuffer;  // In bytes.
   int8_t* _playoutBuffer;  // In bytes.
   uint32_t _recordingFramesLeft;
   uint32_t _playoutFramesLeft;
-  CriticalSectionWrapper& _critSect;
+  rtc::CriticalSection _critSect;
 
   size_t _recordingBufferSizeIn10MS;
   size_t _recordingFramesIn10MS;
   size_t _playoutFramesIn10MS;
 
   // TODO(pbos): Make plain members instead of pointers and stop resetting them.
   std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
   std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
   bool _playing;
   bool _recording;
   int64_t _lastCallPlayoutMillis;
   int64_t _lastCallRecordMillis;
 
   FileWrapper& _outputFile;
   FileWrapper& _inputFile;
   std::string _outputFilename;
   std::string _inputFilename;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H