Removes usage of system_wrappers/include/clock.h in audio_device/

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/logging.h"
@@ skipping 24 matching lines @@
     _recordingBufferSizeIn10MS(0),
     _recordingFramesIn10MS(0),
     _playoutFramesIn10MS(0),
     _playing(false),
     _recording(false),
     _lastCallPlayoutMillis(0),
     _lastCallRecordMillis(0),
     _outputFile(*FileWrapper::Create()),
     _inputFile(*FileWrapper::Create()),
     _outputFilename(outputFilename),
-    _inputFilename(inputFilename),
-    _clock(Clock::GetRealTimeClock()) {
+    _inputFilename(inputFilename) {
 }
 
 FileAudioDevice::~FileAudioDevice() {
   delete &_outputFile;
   delete &_inputFile;
 }
 
 int32_t FileAudioDevice::ActiveAudioLayer(
     AudioDeviceModule::AudioLayer& audioLayer) const {
   return -1;
@@ skipping 416 matching lines @@
     return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
 }
 
 bool FileAudioDevice::RecThreadFunc(void* pThis)
 {
     return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
 }
 
 bool FileAudioDevice::PlayThreadProcess()
 {
-    if(!_playing) {
+    if (!_playing) {
         return false;
     }
-    uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
+    int64_t currentTime = rtc::TimeMillis();
     _critSect.Enter();
 
     if (_lastCallPlayoutMillis == 0 ||
         currentTime - _lastCallPlayoutMillis >= 10) {
         _critSect.Leave();
         _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
         _critSect.Enter();
 
         _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
         assert(_playoutFramesLeft == _playoutFramesIn10MS);
         if (_outputFile.is_open()) {
           _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
         }
         _lastCallPlayoutMillis = currentTime;
     }
     _playoutFramesLeft = 0;
     _critSect.Leave();
 
-    uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
-    if(deltaTimeMillis < 10) {
+    int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+    if (deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 bool FileAudioDevice::RecThreadProcess()
 {
     if (!_recording) {
         return false;
     }
 
-    uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
+    int64_t currentTime = rtc::TimeMillis();
     _critSect.Enter();
 
     if (_lastCallRecordMillis == 0 ||
         currentTime - _lastCallRecordMillis >= 10) {
       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) {
+    int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+    if (deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 }  // namespace webrtc