Clean up PlatformThread.

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 195 matching lines @@
   // PLAYOUT
   if (!_outputFilename.empty() && _outputFile.OpenFile(
         _outputFilename.c_str(), false, false, false) == -1) {
     printf("Failed to open playout file %s!\n", _outputFilename.c_str());
     _playing = false;
     delete [] _playoutBuffer;
     _playoutBuffer = NULL;
     return -1;
   }
 
-  const char* threadName = "webrtc_audio_module_play_thread";
-  _ptrThreadPlay =
-      PlatformThread::CreateThread(PlayThreadFunc, this, threadName);
-  if (!_ptrThreadPlay->Start()) {
-      _ptrThreadPlay.reset();
-      _playing = false;
-      delete [] _playoutBuffer;
-      _playoutBuffer = NULL;
-      return -1;
-  }
-  _ptrThreadPlay->SetPriority(kRealtimePriority);
+  _ptrThreadPlay.reset(new rtc::PlatformThread(
+      PlayThreadFunc, this, "webrtc_audio_module_play_thread"));
+  _ptrThreadPlay->Start();
+  _ptrThreadPlay->SetPriority(rtc::kRealtimePriority);
   return 0;
 }
 
 int32_t FileAudioDevice::StopPlayout() {
   {
       CriticalSectionScoped lock(&_critSect);
       _playing = false;
   }
 
   // stop playout thread first
@@ skipping 32 matching lines @@
   if (!_inputFilename.empty() && _inputFile.OpenFile(
         _inputFilename.c_str(), true, true, false) == -1) {
     printf("Failed to open audio input file %s!\n",
            _inputFilename.c_str());
     _recording = false;
     delete[] _recordingBuffer;
     _recordingBuffer = NULL;
     return -1;
   }
 
-  const char* threadName = "webrtc_audio_module_capture_thread";
-  _ptrThreadRec = PlatformThread::CreateThread(RecThreadFunc, this, threadName);
+  _ptrThreadRec.reset(new rtc::PlatformThread(
+      RecThreadFunc, this, "webrtc_audio_module_capture_thread"));
 
-  if (!_ptrThreadRec->Start()) {
-      _ptrThreadRec.reset();
-      _recording = false;
-      delete [] _recordingBuffer;
-      _recordingBuffer = NULL;
-      return -1;
-  }
-  _ptrThreadRec->SetPriority(kRealtimePriority);
+  _ptrThreadRec->Start();
+  _ptrThreadRec->SetPriority(rtc::kRealtimePriority);
 
   return 0;
 }
 
 
 int32_t FileAudioDevice::StopRecording() {
   {
     CriticalSectionScoped lock(&_critSect);
     _recording = false;
   }
@@ skipping 241 matching lines @@
         _critSect.Enter();
       }
     }
 
     _critSect.Leave();
     SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
     return true;
 }
 
 }  // namespace webrtc