Replace NULL with nullptr in all C++ files.

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/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 
 namespace webrtc {
 
 const int kRecordingFixedSampleRate = 48000;
 const size_t kRecordingNumChannels = 2;
 const int kPlayoutFixedSampleRate = 48000;
 const size_t kPlayoutNumChannels = 2;
 const size_t kPlayoutBufferSize =
     kPlayoutFixedSampleRate / 100 * kPlayoutNumChannels * 2;
 const size_t kRecordingBufferSize =
     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),
-    _inputFilename(inputFilename) {
-}
+                                 const char* outputFilename)
+    : _ptrAudioBuffer(nullptr),
+      _recordingBuffer(nullptr),
+      _playoutBuffer(nullptr),
+      _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),
+      _inputFilename(inputFilename) {}
 
 FileAudioDevice::~FileAudioDevice() {
   delete &_outputFile;
   delete &_inputFile;
 }
 
 int32_t FileAudioDevice::ActiveAudioLayer(
     AudioDeviceModule::AudioLayer& audioLayer) const {
   return -1;
 }
@@ skipping 137 matching lines @@
     _playing = false;
     return -1;
   }
 
   // PLAYOUT
   if (!_outputFilename.empty() &&
       !_outputFile.OpenFile(_outputFilename.c_str(), false)) {
     LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
     _playing = false;
     delete [] _playoutBuffer;
-    _playoutBuffer = NULL;
+    _playoutBuffer = nullptr;
     return -1;
   }
 
   _ptrThreadPlay.reset(new rtc::PlatformThread(
       PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
   _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);
       _playing = false;
   }
 
   // stop playout thread first
   if (_ptrThreadPlay) {
       _ptrThreadPlay->Stop();
       _ptrThreadPlay.reset();
   }
 
   CriticalSectionScoped lock(&_critSect);
 
   _playoutFramesLeft = 0;
   delete [] _playoutBuffer;
-  _playoutBuffer = NULL;
+  _playoutBuffer = nullptr;
   _outputFile.CloseFile();
 
   LOG(LS_INFO) << "Stopped playout capture to output file: "
                << _outputFilename;
   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)) {
     LOG(LS_ERROR) << "Failed to open audio input file: " << _inputFilename;
     _recording = false;
     delete[] _recordingBuffer;
-    _recordingBuffer = NULL;
+    _recordingBuffer = nullptr;
     return -1;
   }
 
   _ptrThreadRec.reset(new rtc::PlatformThread(
       RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
 
   _ptrThreadRec->Start();
   _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
   LOG(LS_INFO) << "Started recording from input file: "
@@ skipping 11 matching lines @@
 
   if (_ptrThreadRec) {
       _ptrThreadRec->Stop();
       _ptrThreadRec.reset();
   }
 
   CriticalSectionScoped lock(&_critSect);
   _recordingFramesLeft = 0;
   if (_recordingBuffer) {
       delete [] _recordingBuffer;
-      _recordingBuffer = NULL;
+      _recordingBuffer = nullptr;
   }
   _inputFile.CloseFile();
 
   LOG(LS_INFO) << "Stopped recording from input file: "
                << _inputFilename;
   return 0;
 }
 
 bool FileAudioDevice::Recording() const {
   return _recording;
@@ skipping 233 matching lines @@
 
     int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
     if (deltaTimeMillis < 10) {
       SleepMs(10 - deltaTimeMillis);
     }
 
     return true;
 }
 
 }  // namespace webrtc