| Index: content/renderer/media/webrtc_audio_renderer.cc
|
| ===================================================================
|
| --- content/renderer/media/webrtc_audio_renderer.cc (revision 181072)
|
| +++ content/renderer/media/webrtc_audio_renderer.cc (working copy)
|
| @@ -157,7 +157,7 @@
|
| }
|
|
|
| source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
|
| - channel_layout, sample_rate, 16, buffer_size);
|
| + channel_layout, 0, sample_rate, 16, buffer_size);
|
|
|
| // Set up audio parameters for the sink, i.e., the native audio output stream.
|
| // We strive to open up using native parameters to achieve best possible
|
| @@ -169,7 +169,7 @@
|
|
|
| buffer_size = hardware_config->GetOutputBufferSize();
|
| sink_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
|
| - channel_layout, sample_rate, 16, buffer_size);
|
| + channel_layout, 0, sample_rate, 16, buffer_size);
|
|
|
| // Create a FIFO if re-buffering is required to match the source input with
|
| // the sink request. The source acts as provider here and the sink as
|
|
|