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)); |
} |