| 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());
|
|
|
| {
|
|
|