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 1e0a335d742af0c647481af78d273f07d1e851c9..f4f677ee58f58e34b0918217ceae891ee3b8da92 100644 |
--- a/content/renderer/media/webrtc_audio_renderer.h |
+++ b/content/renderer/media/webrtc_audio_renderer.h |
@@ -73,10 +73,6 @@ 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<base::SingleThreadTaskRunner>& signaling_thread, |
const blink::WebMediaStream& media_stream, |