| 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..71488c3124e6be4d526ec3c6ed045538c1d35484 100644
|
| --- a/content/renderer/media/webrtc_local_audio_renderer.cc
|
| +++ b/content/renderer/media/webrtc_local_audio_renderer.cc
|
| @@ -263,12 +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_),
|
| - source_params_.effects());
|
| + std::vector<media::Point>(), source_params_.effects());
|
|
|
| {
|
| // Note: The max buffer is fairly large, but will rarely be used.
|
|
|