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 6bbdd08889cdf2078e292ae5a195d6f4a23b4d2e..1d3ffc85f0ca646e0b603e7fede3485792ba0892 100644 |
--- a/content/renderer/media/webrtc_local_audio_renderer.cc |
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc |
@@ -275,11 +275,12 @@ 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_), |
+ "", |
// If DUCKING is enabled on the source, it needs to be enabled on the |
// sink as well. |
source_params_.effects() | implicit_ducking_effect); |