Index: webrtc/modules/audio_device/dummy/file_audio_device.cc |
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc |
index d59eae2ae87bb84466bbf7e55073f9016f78dbaa..3fff40b0f9c03de391277151453f5832f29777fa 100644 |
--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc |
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc |
@@ -213,17 +213,10 @@ int32_t FileAudioDevice::StartPlayout() { |
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; |
} |
@@ -276,17 +269,11 @@ int32_t FileAudioDevice::StartRecording() { |
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; |
} |