Index: content/renderer/media/webrtc_audio_capturer.cc |
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc |
index 994f77b150c6887b2ffb6718ec6efe7c9d134ee9..301fc212c74432bf0f37272e5fa3a79ff1fbe373 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.cc |
+++ b/content/renderer/media/webrtc_audio_capturer.cc |
@@ -264,7 +264,7 @@ void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) { |
// Add with a tag, so we remember to call OnSetFormat() on the new |
// track. |
scoped_refptr<TrackOwner> track_owner(new TrackOwner(track)); |
- tracks_.AddAndTag(track_owner); |
+ tracks_.AddAndTag(track_owner.get()); |
} |
} |
@@ -387,7 +387,7 @@ void WebRtcAudioCapturer::Start() { |
DCHECK(thread_checker_.CalledOnValidThread()); |
DVLOG(1) << "WebRtcAudioCapturer::Start()"; |
base::AutoLock auto_lock(lock_); |
- if (running_ || !source_) |
+ if (running_ || !source_.get()) |
return; |
// Start the data source, i.e., start capturing data from the current source. |
@@ -554,8 +554,8 @@ void WebRtcAudioCapturer::OnCaptureError() { |
media::AudioParameters WebRtcAudioCapturer::source_audio_parameters() const { |
base::AutoLock auto_lock(lock_); |
- return audio_processor_ ? |
- audio_processor_->InputFormat() : media::AudioParameters(); |
+ return audio_processor_.get() ? audio_processor_->InputFormat() |
+ : media::AudioParameters(); |
} |
bool WebRtcAudioCapturer::GetPairedOutputParameters( |