Index: content/renderer/media/webrtc_audio_renderer.cc |
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc |
index 006f12fe4f7b12d8fdc085ef813b780987933bc9..8976ca66579f3bee523b6545d168f279efb71681 100644 |
--- a/content/renderer/media/webrtc_audio_renderer.cc |
+++ b/content/renderer/media/webrtc_audio_renderer.cc |
@@ -218,7 +218,7 @@ |
audio_delay_milliseconds_(0), |
fifo_delay_milliseconds_(0), |
sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- media::CHANNEL_LAYOUT_STEREO, 0, sample_rate, 16, |
+ media::CHANNEL_LAYOUT_STEREO, sample_rate, 16, |
frames_per_buffer, |
GetCurrentDuckingFlag(source_render_frame_id)) { |
WebRtcLogMessage(base::StringPrintf( |
@@ -285,7 +285,7 @@ |
DVLOG(1) << "Using WebRTC output buffer size: " << frames_per_10ms; |
source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- sink_params_.channel_layout(), sink_params_.channels(), 0, |
+ sink_params_.channel_layout(), sink_params_.channels(), |
sample_rate, 16, frames_per_10ms); |
// Update audio parameters for the sink, i.e., the native audio output stream. |
@@ -312,7 +312,7 @@ |
DVLOG(1) << "Using sink output buffer size: " << frames_per_buffer; |
sink_params_.Reset(sink_params_.format(), sink_params_.channel_layout(), |
- sink_params_.channels(), 0, sample_rate, 16, |
+ sink_params_.channels(), sample_rate, 16, |
frames_per_buffer); |
// Create a FIFO if re-buffering is required to match the source input with |