| 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 86b657ffdbf6b6c144ff0904dbb3a9076295255c..e552fbff3b657b05a8e58a50507a735a5cc049c4 100644
|
| --- a/content/renderer/media/webrtc_local_audio_renderer.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_renderer.cc
|
| @@ -121,7 +121,8 @@ void WebRtcLocalAudioRenderer::Start() {
|
| MediaStreamAudioSink::AddToAudioTrack(this, audio_track_);
|
| // ...and |sink_| will get audio data from us.
|
| DCHECK(!sink_.get());
|
| - sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
|
| + sink_ = AudioDeviceFactory::NewOutputDevice(
|
| + source_render_frame_id_, session_id_, std::string(), url::Origin());
|
|
|
| base::AutoLock auto_lock(thread_lock_);
|
| last_render_time_ = base::TimeTicks::Now();
|
| @@ -242,7 +243,7 @@ void WebRtcLocalAudioRenderer::MaybeStartSink() {
|
| return;
|
|
|
| DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_.";
|
| - sink_->InitializeWithSessionId(sink_params_, this, session_id_);
|
| + sink_->Initialize(sink_params_, this);
|
| sink_->Start();
|
| sink_started_ = true;
|
| UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
|
| @@ -286,12 +287,11 @@ void WebRtcLocalAudioRenderer::ReconfigureSink(
|
|
|
| // Stop |sink_| and re-create a new one to be initialized with different audio
|
| // parameters. Then, invoke MaybeStartSink() to restart everything again.
|
| - if (sink_started_) {
|
| - sink_->Stop();
|
| - sink_started_ = false;
|
| - }
|
| + sink_->Stop();
|
| + sink_started_ = false;
|
|
|
| - sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
|
| + sink_ = AudioDeviceFactory::NewOutputDevice(
|
| + source_render_frame_id_, session_id_, std::string(), url::Origin());
|
| MaybeStartSink();
|
| }
|
|
|
|
|