Index: content/renderer/media/webrtc_local_audio_renderer.cc |
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc |
index addf9637a5ff68964a0cdfc471433d028de5981e..e4adbc28d3f72df61cdd50b243488a56f72b7f0b 100644 |
--- a/content/renderer/media/webrtc_local_audio_renderer.cc |
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc |
@@ -282,8 +282,7 @@ void WebRtcLocalAudioRenderer::ReconfigureSink( |
source_params_ = params; |
sink_params_ = media::AudioParameters(source_params_.format(), |
- source_params_.channel_layout(), source_params_.channels(), |
- source_params_.input_channels(), source_params_.sample_rate(), |
+ source_params_.channel_layout(), source_params_.sample_rate(), |
source_params_.bits_per_sample(), |
#if defined(OS_ANDROID) |
// On Android, input and output use the same sample rate. In order to |