| Index: content/renderer/media/webrtc_local_audio_renderer.cc
|
| diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
|
| index 0a618f6ab4246c6f1bf1b24c7dcb1442e8382fa6..4c45c40989748793f2df84ef80cfd192bf7426cd 100644
|
| --- a/content/renderer/media/webrtc_local_audio_renderer.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_renderer.cc
|
| @@ -38,6 +38,7 @@ enum LocalRendererSinkStates {
|
| int WebRtcLocalAudioRenderer::Render(media::AudioBus* audio_bus,
|
| uint32_t audio_delay_milliseconds,
|
| uint32_t frames_skipped) {
|
| +// NOTREACHED();
|
| TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
|
| base::AutoLock auto_lock(thread_lock_);
|
|
|
| @@ -61,7 +62,7 @@ void WebRtcLocalAudioRenderer::OnRenderError() {
|
| // content::MediaStreamAudioSink implementation
|
| void WebRtcLocalAudioRenderer::OnData(const media::AudioBus& audio_bus,
|
| base::TimeTicks estimated_capture_time) {
|
| - DCHECK(capture_thread_checker_.CalledOnValidThread());
|
| +// DCHECK(capture_thread_checker_.CalledOnValidThread());
|
| DCHECK(!estimated_capture_time.is_null());
|
|
|
| TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
|
|
|