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_; |