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 55282c8d62f7fa5c605637a0664fb3dc4d08e694..c6facb90a6b6fc8fcd9035de2feb93c652ad0d60 100644 |
--- a/content/renderer/media/webrtc_local_audio_renderer.cc |
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc |
@@ -263,11 +263,12 @@ void WebRtcLocalAudioRenderer::ReconfigureSink( |
source_params_ = params; |
- sink_params_ = media::AudioParameters(source_params_.format(), |
- source_params_.channel_layout(), source_params_.sample_rate(), |
- source_params_.bits_per_sample(), |
+ sink_params_ = media::AudioParameters( |
+ source_params_.format(), source_params_.channel_layout(), |
+ source_params_.sample_rate(), source_params_.bits_per_sample(), |
WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(), |
frames_per_buffer_), |
+ std::vector<media::Point>(), |
source_params_.effects()); |
{ |