content/renderer/media/webrtc_audio_renderer.h - Issue 2067863003: Mixing audio with different latency requirements

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Unified Diff: content/renderer/media/webrtc_audio_renderer.h

Issue 2067863003: Mixing audio with different latency requirements (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master

Patch Set: android test fix Created 4 years, 6 months ago

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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,