| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index fa4dad0f156b6b204f968c6f43c111182eb033ba..7d026d9b9df0cbfd0a950c402f7510b459fdac6a 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -213,11 +213,13 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
|
| webrtc::AudioDeviceModule* adm,
|
| const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
|
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
|
| - rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
|
| + rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
|
| + rtc::TaskQueue* low_priority_worker_queue)
|
| : WebRtcVoiceEngine(adm,
|
| encoder_factory,
|
| decoder_factory,
|
| audio_mixer,
|
| + low_priority_worker_queue,
|
| new VoEWrapper()) {
|
| audio_state_ =
|
| webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer));
|
| @@ -228,17 +230,18 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
|
| const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
|
| const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
|
| rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
|
| + rtc::TaskQueue* low_priority_worker_queue,
|
| VoEWrapper* voe_wrapper)
|
| - : low_priority_worker_queue_("low-prio-worker-queue",
|
| - rtc::TaskQueue::Priority::LOW),
|
| - adm_(adm),
|
| + : adm_(adm),
|
| encoder_factory_(encoder_factory),
|
| decoder_factory_(decoder_factory),
|
| - voe_wrapper_(voe_wrapper) {
|
| + voe_wrapper_(voe_wrapper),
|
| + low_priority_worker_queue_(low_priority_worker_queue) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
|
| RTC_DCHECK(voe_wrapper);
|
| RTC_DCHECK(decoder_factory);
|
| + RTC_DCHECK(low_priority_worker_queue);
|
|
|
| signal_thread_checker_.DetachFromThread();
|
|
|
| @@ -660,8 +663,9 @@ void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
|
| bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
|
| int64_t max_size_bytes) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| + RTC_DCHECK(low_priority_worker_queue_);
|
| auto aec_dump = webrtc::AecDumpFactory::Create(file, max_size_bytes,
|
| - &low_priority_worker_queue_);
|
| + low_priority_worker_queue_);
|
| if (aec_dump) {
|
| apm()->AttachAecDump(std::move(aec_dump));
|
| }
|
| @@ -670,9 +674,10 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
|
|
|
| void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
|
| RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
|
| + RTC_DCHECK(low_priority_worker_queue_);
|
|
|
| auto aec_dump =
|
| - webrtc::AecDumpFactory::Create(filename, -1, &low_priority_worker_queue_);
|
| + webrtc::AecDumpFactory::Create(filename, -1, low_priority_worker_queue_);
|
| if (aec_dump) {
|
| apm()->AttachAecDump(std::move(aec_dump));
|
| }
|
|
|