Chromium Code Reviews| Index: content/renderer/media/audio_renderer_mixer_manager.cc |
| diff --git a/content/renderer/media/audio_renderer_mixer_manager.cc b/content/renderer/media/audio_renderer_mixer_manager.cc |
| index 05782d231fb618d66e8e824388f045e8866185db..5337b186a086521084d701ea86c58cb1c3a1ab59 100644 |
| --- a/content/renderer/media/audio_renderer_mixer_manager.cc |
| +++ b/content/renderer/media/audio_renderer_mixer_manager.cc |
| @@ -54,10 +54,13 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer( |
| // Create output parameters based on the audio hardware configuration for |
| // passing on to the output sink. Force to 16-bit output for now since we |
| - // know that works well for WebAudio and WebRTC. |
| + // know that works well for WebAudio and WebRTC. Force a 2048 buffer size |
| + // until we have renderer-side device changes since WebRTC audio mirroring |
| + // will use the input device's buffer size to drive mirroring so we need |
| + // the output device's buffer size to be big enough. |
| media::AudioParameters output_params( |
| media::AudioParameters::AUDIO_PCM_LOW_LATENCY, params.channel_layout(), |
| - hardware_sample_rate_, 16, hardware_buffer_size_); |
| + hardware_sample_rate_, 16, 2048); |
|
DaleCurtis
2013/01/15 21:00:08
You'll need to test this on mac/win and make sure
justinlin
2013/01/15 22:36:21
OK, will test.
|
| // If we've created invalid output parameters, simply pass on the input params |
| // and let the browser side handle automatic fallback. |