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. |