| Index: content/renderer/media/webrtc_audio_renderer.h
|
| diff --git a/content/renderer/media/webrtc_audio_renderer.h b/content/renderer/media/webrtc_audio_renderer.h
|
| index 119853472b64f6e0b1174a07349b34319c48bfdb..9accf6a3059b5c4bc70dfb8381f1d35ddce11f35 100644
|
| --- a/content/renderer/media/webrtc_audio_renderer.h
|
| +++ b/content/renderer/media/webrtc_audio_renderer.h
|
| @@ -69,6 +69,10 @@ class CONTENT_EXPORT WebRtcAudioRenderer
|
| float volume_;
|
| };
|
|
|
| +
|
| + // Returns platform specific optimal buffer size for rendering audio.
|
| + static int GetOptimalBufferSize(int sample_rate, int hardware_buffer_size);
|
| +
|
| WebRtcAudioRenderer(
|
| const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
|
| int source_render_view_id,
|
|
|