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| author | xians@chromium.org <xians@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2011-10-07 15:51:13 +0000 |
|---|---|---|
| committer | xians@chromium.org <xians@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2011-10-07 15:51:13 +0000 |
| commit | 257bc741f4b5a7b3c801dae7b26f900f6f09ff2a (patch) | |
| tree | 094537f89ba01b397e4bf8db1c382a6a43d40ae7 /content/renderer/media/webrtc_audio_device_impl.cc | |
| parent | 2526ddaed31aa448d34afcb7f03298fb076353e8 (diff) | |
| download | chromium_src-257bc741f4b5a7b3c801dae7b26f900f6f09ff2a.zip chromium_src-257bc741f4b5a7b3c801dae7b26f900f6f09ff2a.tar.gz chromium_src-257bc741f4b5a7b3c801dae7b26f900f6f09ff2a.tar.bz2 | |
ALSA should support 10ms playout buffer size now.
Review URL: http://codereview.chromium.org/8183018
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@104489 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content/renderer/media/webrtc_audio_device_impl.cc')
| -rw-r--r-- | content/renderer/media/webrtc_audio_device_impl.cc | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc index 748a890..eb2cef33 100644 --- a/content/renderer/media/webrtc_audio_device_impl.cc +++ b/content/renderer/media/webrtc_audio_device_impl.cc @@ -348,12 +348,12 @@ int32_t WebRtcAudioDeviceImpl::Init() { input_channels = 1; output_channels = 1; // Based on tests using the current ALSA implementation in Chrome, we have - // found that the best combination is 20ms on the input side and 30ms on the + // found that the best combination is 20ms on the input side and 10ms on the // output side. - // TODO(henrika): It might be possible to reduce the input and output buffer + // TODO(henrika): It might be possible to reduce the input buffer // size and reduce the delay even more. input_buffer_size = 2 * 480; - output_buffer_size = 3 * 480; + output_buffer_size = 480; #else DLOG(ERROR) << "Unsupported platform"; return -1; |