Move ThreadWrapper to ProcessThread in base.

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 <string>
 
 #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"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 class EventWrapper;
-class ThreadWrapper;
+class PlatformThread;
 
 // This is a fake audio device which plays audio from a file as its microphone
 // and plays out into a file.
 class FileAudioDevice : public AudioDeviceGeneric {
  public:
   // Constructs a file audio device with |id|. It will read audio from
   // |inputFilename| and record output audio to |outputFilename|.
   //
   // The input file should be a readable 48k stereo raw file, and the output
   // file should point to a writable location. The output format will also be
@@ skipping 135 matching lines @@
   int8_t* _recordingBuffer;  // In bytes.
   int8_t* _playoutBuffer;  // In bytes.
   uint32_t _recordingFramesLeft;
   uint32_t _playoutFramesLeft;
   CriticalSectionWrapper& _critSect;
 
   size_t _recordingBufferSizeIn10MS;
   size_t _recordingFramesIn10MS;
   size_t _playoutFramesIn10MS;
 
-  rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
-  rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
+  rtc::scoped_ptr<PlatformThread> _ptrThreadRec;
+  rtc::scoped_ptr<PlatformThread> _ptrThreadPlay;
 
   bool _playing;
   bool _recording;
   uint64_t _lastCallPlayoutMillis;
   uint64_t _lastCallRecordMillis;
 
   FileWrapper& _outputFile;
   FileWrapper& _inputFile;
   std::string _outputFilename;
   std::string _inputFilename;
 
   Clock* _clock;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H