Index: content/renderer/media/webrtc_audio_renderer.cc |
=================================================================== |
--- content/renderer/media/webrtc_audio_renderer.cc (revision 189875) |
+++ content/renderer/media/webrtc_audio_renderer.cc (working copy) |
@@ -158,10 +158,8 @@ |
return false; |
} |
- int channels = ChannelLayoutToChannelCount(channel_layout); |
source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- channel_layout, channels, 0, |
- 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 |
@@ -173,7 +171,7 @@ |
buffer_size = hardware_config->GetOutputBufferSize(); |
sink_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- channel_layout, channels, 0, 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 |