| 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 addf9637a5ff68964a0cdfc471433d028de5981e..ac44a210be3a655320f58b3219838cef2eb2c077 100644
|
| --- a/content/renderer/media/webrtc_local_audio_renderer.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_renderer.cc
|
| @@ -153,7 +153,7 @@ void WebRtcLocalAudioRenderer::Stop() {
|
| // Stop the output audio stream, i.e, stop asking for data to render.
|
| // It is safer to call Stop() on the |sink_| to clean up the resources even
|
| // when the |sink_| is never started.
|
| - if (sink_) {
|
| + if (sink_.get()) {
|
| sink_->Stop();
|
| sink_ = NULL;
|
| }
|
| @@ -316,7 +316,7 @@ void WebRtcLocalAudioRenderer::ReconfigureSink(
|
| loopback_fifo_.reset(new_fifo);
|
| }
|
|
|
| - if (!sink_)
|
| + if (!sink_.get())
|
| return; // WebRtcLocalAudioRenderer has not yet been started.
|
|
|
| // Stop |sink_| and re-create a new one to be initialized with different audio
|
|
|