content/renderer/media/webrtc_audio_renderer.h - Issue 539453003: Used 10ms native buffer size for webrtc audio renderer on Linux and Mac

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

Issue 539453003: Used 10ms native buffer size for webrtc audio renderer on Linux and Mac (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master

Patch Set: addressed Tommi's comments. Created 6 years, 3 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 61b0b24d141ba3c2ff2b6414df226ae8578614e7..580659933c9a86b36738e2aa7a77e625f20b23bd 100644

--- a/content/renderer/media/webrtc_audio_renderer.h

+++ b/content/renderer/media/webrtc_audio_renderer.h

@@ -99,6 +99,7 @@ class CONTENT_EXPORT WebRtcAudioRenderer

// Accessors to the sink audio parameters.

int channels() const { return sink_params_.channels(); }

int sample_rate() const { return sink_params_.sample_rate(); }

+ int frames_per_buffer() const { return sink_params_.frames_per_buffer(); }

private:

// MediaStreamAudioRenderer implementation. This is private since we want

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