Index: content/renderer/media/webrtc_local_audio_renderer.h |
diff --git a/content/renderer/media/webrtc_local_audio_renderer.h b/content/renderer/media/webrtc_local_audio_renderer.h |
index d33c384975002ab70473339e02d6d543f27103b8..7d86ae6a436317df46dfaf5d786a67ee8e8605fd 100644 |
--- a/content/renderer/media/webrtc_local_audio_renderer.h |
+++ b/content/renderer/media/webrtc_local_audio_renderer.h |
@@ -21,13 +21,13 @@ |
#include "content/public/renderer/media_stream_audio_sink.h" |
#include "content/renderer/media/webrtc_audio_device_impl.h" |
#include "content/renderer/media/webrtc_local_audio_track.h" |
+#include "media/base/audio_renderer_sink.h" |
#include "media/base/output_device.h" |
#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
namespace media { |
class AudioBus; |
class AudioShifter; |
-class AudioOutputDevice; |
class AudioParameters; |
} |
@@ -133,7 +133,7 @@ class CONTENT_EXPORT WebRtcLocalAudioRenderer |
const scoped_refptr<base::SingleThreadTaskRunner> task_runner_; |
// The sink (destination) for rendered audio. |
- scoped_refptr<media::AudioOutputDevice> sink_; |
+ scoped_refptr<media::RestartableAudioRendererSink> sink_; |
// This does all the synchronization/resampling/smoothing. |
scoped_ptr<media::AudioShifter> audio_shifter_; |