FileWrapper[Impl] modifications and actually remove the "Impl" class.

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.
  */
 #include "webrtc/base/platform_thread.h"
@@ skipping 28 matching lines @@
     _lastCallPlayoutMillis(0),
     _lastCallRecordMillis(0),
     _outputFile(*FileWrapper::Create()),
     _inputFile(*FileWrapper::Create()),
     _outputFilename(outputFilename),
     _inputFilename(inputFilename),
     _clock(Clock::GetRealTimeClock()) {
 }
 
 FileAudioDevice::~FileAudioDevice() {
-  if (_outputFile.Open()) {
-      _outputFile.Flush();
-      _outputFile.CloseFile();
-  }
   delete &_outputFile;
-  if (_inputFile.Open()) {
-      _inputFile.Flush();
-      _inputFile.CloseFile();
-  }
   delete &_inputFile;
 }
 
 int32_t FileAudioDevice::ActiveAudioLayer(
     AudioDeviceModule::AudioLayer& audioLayer) const {
   return -1;
 }
 
 int32_t FileAudioDevice::Init() { return 0; }
 
@@ skipping 127 matching lines @@
 
   if (!_playoutBuffer) {
       _playoutBuffer = new int8_t[kPlayoutBufferSize];
   }
   if (!_playoutBuffer) {
     _playing = false;
     return -1;
   }
 
   // PLAYOUT
-  if (!_outputFilename.empty() && _outputFile.OpenFile(
-        _outputFilename.c_str(), false, false, false) == -1) {
+  if (!_outputFilename.empty() &&
+      _outputFile.OpenFile(_outputFilename.c_str(), false) == -1) {
     printf("Failed to open playout file %s!\n", _outputFilename.c_str());
     _playing = false;
     delete [] _playoutBuffer;
     _playoutBuffer = NULL;
     return -1;
   }
 
   _ptrThreadPlay.reset(new rtc::PlatformThread(
       PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
   _ptrThreadPlay->Start();
@@ skipping 11 matching lines @@
   if (_ptrThreadPlay) {
       _ptrThreadPlay->Stop();
       _ptrThreadPlay.reset();
   }
 
   CriticalSectionScoped lock(&_critSect);
 
   _playoutFramesLeft = 0;
   delete [] _playoutBuffer;
   _playoutBuffer = NULL;
-  if (_outputFile.Open()) {
-      _outputFile.Flush();
-      _outputFile.CloseFile();
-  }
-   return 0;
+  _outputFile.CloseFile();
+  return 0;
 }
 
 bool FileAudioDevice::Playing() const {
   return true;
 }
 
 int32_t FileAudioDevice::StartRecording() {
   _recording = true;
 
   // Make sure we only create the buffer once.
   _recordingBufferSizeIn10MS = _recordingFramesIn10MS *
                                kRecordingNumChannels *
                                2;
   if (!_recordingBuffer) {
       _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
   }
 
-  if (!_inputFilename.empty() && _inputFile.OpenFile(
-        _inputFilename.c_str(), true, true, false) == -1) {
+  if (!_inputFilename.empty() &&
+      _inputFile.OpenFile(_inputFilename.c_str(), true) == -1) {
     printf("Failed to open audio input file %s!\n",
            _inputFilename.c_str());
     _recording = false;
     delete[] _recordingBuffer;
     _recordingBuffer = NULL;
     return -1;
   }
 
   _ptrThreadRec.reset(new rtc::PlatformThread(
       RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
@@ skipping 211 matching lines @@
     _critSect.Enter();
 
     if (_lastCallPlayoutMillis == 0 ||
         currentTime - _lastCallPlayoutMillis >= 10) {
         _critSect.Leave();
         _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
         _critSect.Enter();
 
         _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
         assert(_playoutFramesLeft == _playoutFramesIn10MS);
-        if (_outputFile.Open()) {
+        if (_outputFile.is_open()) {
           _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
-          _outputFile.Flush();
         }
         _lastCallPlayoutMillis = currentTime;
     }
     _playoutFramesLeft = 0;
     _critSect.Leave();
 
     uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
     if(deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 bool FileAudioDevice::RecThreadProcess()
 {
     if (!_recording) {
         return false;
     }
 
     uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
     _critSect.Enter();
 
     if (_lastCallRecordMillis == 0 ||
         currentTime - _lastCallRecordMillis >= 10) {
-      if (_inputFile.Open()) {
+      if (_inputFile.is_open()) {
         if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
           _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
                                              _recordingFramesIn10MS);
         } else {
           _inputFile.Rewind();
         }
         _lastCallRecordMillis = currentTime;
         _critSect.Leave();
         _ptrAudioBuffer->DeliverRecordedData();
         _critSect.Enter();
       }
     }
 
     _critSect.Leave();
 
     uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
     if(deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 }  // namespace webrtc