| 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_;
|
| webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
|
| webrtc::Call* call_ = nullptr;
|
| webrtc::AudioSendStream::Config config_;
|
|
|