| Index: content/renderer/media/webrtc_local_audio_track.cc
|
| diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
|
| index 99ada98099fa3163d58fbf46ed0185fe838b8f39..8fc1e4f9ae0e4b04199034335d7d2a1503002f6b 100644
|
| --- a/content/renderer/media/webrtc_local_audio_track.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_track.cc
|
| @@ -20,11 +20,12 @@ WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
|
| WebRtcLocalAudioTrackAdapter* adapter,
|
| const scoped_refptr<WebRtcAudioCapturer>& capturer,
|
| WebAudioCapturerSource* webaudio_source)
|
| - : MediaStreamTrack(adapter, true),
|
| + : MediaStreamTrack(true),
|
| adapter_(adapter),
|
| capturer_(capturer),
|
| webaudio_source_(webaudio_source) {
|
| DCHECK(capturer.get() || webaudio_source);
|
| + signal_thread_checker_.DetachFromThread();
|
|
|
| adapter_->Initialize(this);
|
|
|
| @@ -122,7 +123,11 @@ void WebRtcLocalAudioTrack::SetAudioProcessor(
|
| }
|
|
|
| void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| + // This method is called from webrtc, on the signaling thread, when the local
|
| + // description is set and from the main thread from WebMediaPlayerMS::load
|
| + // (via WebRtcLocalAudioRenderer::Start).
|
| + DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
|
| + signal_thread_checker_.CalledOnValidThread());
|
| DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
|
| base::AutoLock auto_lock(lock_);
|
|
|
| @@ -140,13 +145,18 @@ void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
|
| }
|
|
|
| void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| + // See AddSink for additional context. When local audio is stopped from
|
| + // webrtc, we'll be called here on the signaling thread.
|
| + DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
|
| + signal_thread_checker_.CalledOnValidThread());
|
| DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
|
|
|
| - base::AutoLock auto_lock(lock_);
|
| -
|
| - scoped_refptr<MediaStreamAudioTrackSink> removed_item = sinks_.Remove(
|
| - MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
|
| + scoped_refptr<MediaStreamAudioTrackSink> removed_item;
|
| + {
|
| + base::AutoLock auto_lock(lock_);
|
| + removed_item = sinks_.Remove(
|
| + MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
|
| + }
|
|
|
| // Clear the delegate to ensure that no more capture callbacks will
|
| // be sent to this sink. Also avoids a possible crash which can happen
|
| @@ -212,6 +222,12 @@ void WebRtcLocalAudioTrack::Start() {
|
| }
|
| }
|
|
|
| +void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
|
| + DCHECK(thread_checker_.CalledOnValidThread());
|
| + if (adapter_.get())
|
| + adapter_->set_enabled(enabled);
|
| +}
|
| +
|
| void WebRtcLocalAudioTrack::Stop() {
|
| DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
|
| @@ -249,4 +265,9 @@ void WebRtcLocalAudioTrack::Stop() {
|
| }
|
| }
|
|
|
| +webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
|
| + DCHECK(thread_checker_.CalledOnValidThread());
|
| + return adapter_.get();
|
| +}
|
| +
|
| } // namespace content
|
|
|