| Index: content/renderer/media/webrtc_audio_renderer.cc
|
| diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
|
| index 0cd3bf2f346fc4882a5fb0b8390bf146370ffd4d..698af7b230653a35235cc80212202955984d4c3e 100644
|
| --- a/content/renderer/media/webrtc_audio_renderer.cc
|
| +++ b/content/renderer/media/webrtc_audio_renderer.cc
|
| @@ -273,10 +273,10 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
|
| source_ = source;
|
|
|
| // Configure the audio rendering client and start rendering.
|
| - sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
|
| -
|
| DCHECK_GE(session_id_, 0);
|
| - sink_->InitializeWithSessionId(sink_params_, this, session_id_);
|
| + sink_ =
|
| + AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_);
|
| + sink_->Initialize(sink_params_, this);
|
|
|
| sink_->Start();
|
|
|
| @@ -440,7 +440,7 @@ void WebRtcAudioRenderer::OnRenderError() {
|
| // Called by AudioPullFifo when more data is necessary.
|
| void WebRtcAudioRenderer::SourceCallback(
|
| int fifo_frame_delay, media::AudioBus* audio_bus) {
|
| - base::TimeTicks start_time = base::TimeTicks::Now() ;
|
| + base::TimeTicks start_time = base::TimeTicks::Now();
|
| DVLOG(2) << "WebRtcAudioRenderer::SourceCallback("
|
| << fifo_frame_delay << ", "
|
| << audio_bus->frames() << ")";
|
|
|