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| author | cevans@chromium.org <cevans@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-01-11 03:24:49 +0000 |
|---|---|---|
| committer | cevans@chromium.org <cevans@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-01-11 03:24:49 +0000 |
| commit | d6cdc8727c5e005a914cb13bcf959250a2843fd9 (patch) | |
| tree | 6c741ff1916bb146752e6cd29109de2a8f01ac35 | |
| parent | 12ff4e1e24b84c72dd487226c6c1dad649278d1b (diff) | |
| download | chromium_src-d6cdc8727c5e005a914cb13bcf959250a2843fd9.zip chromium_src-d6cdc8727c5e005a914cb13bcf959250a2843fd9.tar.gz chromium_src-d6cdc8727c5e005a914cb13bcf959250a2843fd9.tar.bz2 | |
Merge 175323
> Avoids crash in WebRTC audio clients for 96kHz render rate on Mac OSX.
>
> TBR=xians
> BUG=166523
> TEST=Misc set of WebRTC audio clients on Mac.
>
> Review URL: https://codereview.chromium.org/11773017
TBR=henrika@chromium.org
Review URL: https://codereview.chromium.org/11860003
git-svn-id: svn://svn.chromium.org/chrome/branches/1364/src@176249 0039d316-1c4b-4281-b951-d872f2087c98
| -rw-r--r-- | content/renderer/media/webrtc_audio_renderer.cc | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc index 182fb5b..104c89b 100644 --- a/content/renderer/media/webrtc_audio_renderer.cc +++ b/content/renderer/media/webrtc_audio_renderer.cc @@ -156,11 +156,11 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) { // Render side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback- // driven Core Audio implementation. Tests have shown that 10ms is a suitable - // frame size to use, both for 48kHz and 44.1kHz. + // frame size to use for 96kHz, 48kHz and 44.1kHz. // Use different buffer sizes depending on the current hardware sample rate. - if (sample_rate == 48000) { - buffer_size = 480; + if (sample_rate == 96000 || sample_rate == 48000) { + buffer_size = (sample_rate / 100); } else { // We do run at 44.1kHz at the actual audio layer, but ask for frames // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. |