Chromium Code Reviews| Index: webrtc/media/engine/webrtcvoiceengine.cc |
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
| index 235b9b3aafeb2641ac0b93a7fd0e8d660d1f728d..8a23c16e58cf743118d6472d7b3b3c7544d1b72a 100644 |
| --- a/webrtc/media/engine/webrtcvoiceengine.cc |
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc |
| @@ -1155,7 +1155,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
| // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore: |
| // RTC_DCHECK(voe_audio_transport); |
| RTC_DCHECK(call); |
| - audio_capture_thread_checker_.DetachFromThread(); |
| config_.rtp.ssrc = ssrc; |
| config_.rtp.c_name = c_name; |
| config_.voe_channel_id = ch; |
| @@ -1270,7 +1269,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
| int sample_rate, |
| size_t number_of_channels, |
| size_t number_of_frames) override { |
| - RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread()); |
| + RTC_DCHECK_NON_REENTRANT(audio_capture_non_reentrant_checker_); |
| RTC_DCHECK(voe_audio_transport_); |
| voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data, |
| bits_per_sample, sample_rate, |
| @@ -1317,7 +1316,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
| } |
| rtc::ThreadChecker worker_thread_checker_; |
| - rtc::ThreadChecker audio_capture_thread_checker_; |
| + rtc::NonReentrantChecker audio_capture_non_reentrant_checker_; |
|
kwiberg-webrtc
2016/09/20 10:55:22
That you can change the name of the checker withou
|
| webrtc::AudioTransport* const voe_audio_transport_ = nullptr; |
| webrtc::Call* call_ = nullptr; |
| webrtc::AudioSendStream::Config config_; |