Chromium Code Reviews| Index: webrtc/media/engine/webrtcvoiceengine.cc |
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
| index e066f309a74b09225c4a4f90d4786662e25a06a4..358c142a1b01bea7ba3a977f55279a78cc45221f 100644 |
| --- a/webrtc/media/engine/webrtcvoiceengine.cc |
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc |
| @@ -26,6 +26,7 @@ |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/base/helpers.h" |
| #include "webrtc/base/logging.h" |
| +#include "webrtc/base/race_checker.h" |
| #include "webrtc/base/stringencode.h" |
| #include "webrtc/base/stringutils.h" |
| #include "webrtc/base/trace_event.h" |
| @@ -1155,7 +1156,6 @@ 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 +1270,7 @@ 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_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_); |
| RTC_DCHECK(voe_audio_transport_); |
| voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data, |
| bits_per_sample, sample_rate, |
| @@ -1317,7 +1317,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
| } |
| rtc::ThreadChecker worker_thread_checker_; |
| - rtc::ThreadChecker audio_capture_thread_checker_; |
| + rtc::RaceChecker audio_capture_race_checker_; |
|
kwiberg-webrtc
2016/09/20 19:29:07
Again, no annotations on the member variables prov
|
| webrtc::AudioTransport* const voe_audio_transport_ = nullptr; |
|
pbos-webrtc
2016/09/20 19:38:03
GUARDED_BY(audio_capture_racer_checker_) if this i
kwiberg-webrtc
2016/09/20 20:05:25
Yes. You really need to annotate all the members t
|
| webrtc::Call* call_ = nullptr; |
| webrtc::AudioSendStream::Config config_; |