Remove more CriticalSectionWrappers.

webrtc/modules/audio_device/dummy/file_audio_device.cc

 /*
  *  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.
  */
 
@@ skipping 15 matching lines @@
     kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2;
 
 FileAudioDevice::FileAudioDevice(const int32_t id,
                                  const char* inputFilename,
                                  const char* outputFilename):
     _ptrAudioBuffer(NULL),
     _recordingBuffer(NULL),
     _playoutBuffer(NULL),
     _recordingFramesLeft(0),
     _playoutFramesLeft(0),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _recordingBufferSizeIn10MS(0),
     _recordingFramesIn10MS(0),
     _playoutFramesIn10MS(0),
     _playing(false),
     _recording(false),
     _lastCallPlayoutMillis(0),
     _lastCallRecordMillis(0),
     _outputFile(*FileWrapper::Create()),
     _inputFile(*FileWrapper::Create()),
     _outputFilename(outputFilename),
@@ skipping 107 matching lines @@
 int32_t FileAudioDevice::RecordingIsAvailable(bool& available) {
   if (_record_index == 0) {
     available = true;
     return _record_index;
   }
   available = false;
   return -1;
 }
 
 int32_t FileAudioDevice::InitRecording() {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   if (_recording) {
     return -1;
   }
 
   _recordingFramesIn10MS = static_cast<size_t>(kRecordingFixedSampleRate / 100);
 
   if (_ptrAudioBuffer) {
     _ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);
     _ptrAudioBuffer->SetRecordingChannels(kRecordingNumChannels);
@@ skipping 37 matching lines @@
   _ptrThreadPlay->Start();
   _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
 
   LOG(LS_INFO) << "Started playout capture to output file: "
                << _outputFilename;
   return 0;
 }
 
 int32_t FileAudioDevice::StopPlayout() {
   {
-      CriticalSectionScoped lock(&_critSect);
+      rtc::CritScope lock(&_critSect);
       _playing = false;
   }
 
   // stop playout thread first
   if (_ptrThreadPlay) {
       _ptrThreadPlay->Stop();
       _ptrThreadPlay.reset();
   }
 
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   _playoutFramesLeft = 0;
   delete [] _playoutBuffer;
   _playoutBuffer = NULL;
   _outputFile.CloseFile();
 
   LOG(LS_INFO) << "Stopped playout capture to output file: "
                << _outputFilename;
   return 0;
 }
@@ skipping 30 matching lines @@
 
   LOG(LS_INFO) << "Started recording from input file: "
                << _inputFilename;
 
   return 0;
 }
 
 
 int32_t FileAudioDevice::StopRecording() {
   {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _recording = false;
   }
 
   if (_ptrThreadRec) {
       _ptrThreadRec->Stop();
       _ptrThreadRec.reset();
   }
 
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   _recordingFramesLeft = 0;
   if (_recordingBuffer) {
       delete [] _recordingBuffer;
       _recordingBuffer = NULL;
   }
   _inputFile.CloseFile();
 
   LOG(LS_INFO) << "Stopped recording from input file: "
                << _inputFilename;
   return 0;
@@ skipping 146 matching lines @@
 
 void FileAudioDevice::ClearPlayoutWarning() {}
 
 void FileAudioDevice::ClearPlayoutError() {}
 
 void FileAudioDevice::ClearRecordingWarning() {}
 
 void FileAudioDevice::ClearRecordingError() {}
 
 void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   _ptrAudioBuffer = audioBuffer;
 
   // Inform the AudioBuffer about default settings for this implementation.
   // Set all values to zero here since the actual settings will be done by
   // InitPlayout and InitRecording later.
   _ptrAudioBuffer->SetRecordingSampleRate(0);
   _ptrAudioBuffer->SetPlayoutSampleRate(0);
   _ptrAudioBuffer->SetRecordingChannels(0);
   _ptrAudioBuffer->SetPlayoutChannels(0);
@@ skipping 70 matching lines @@
 
     int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
     if (deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 }  // namespace webrtc