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, |