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 |