Index: content/renderer/media/webrtc_audio_capturer.cc |
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc |
index e1f71c3af038d7598fa44324afde62970b3b0073..15a10b6848d39e96ccc4fb868c20f5e156ade589 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.cc |
+++ b/content/renderer/media/webrtc_audio_capturer.cc |
@@ -244,7 +244,7 @@ WebRtcAudioCapturer::WebRtcAudioCapturer( |
: constraints_(constraints), |
audio_processor_(new rtc::RefCountedObject<MediaStreamAudioProcessor>( |
constraints, |
- device_info.device.input.effects, |
+ device_info.device.input, |
audio_device)), |
running_(false), |
render_frame_id_(render_frame_id), |
@@ -343,10 +343,8 @@ void WebRtcAudioCapturer::SetCapturerSourceInternal( |
// which would normally be used by default. |
// bits_per_sample is always 16 for now. |
media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- channel_layout, |
- sample_rate, |
- 16, |
- buffer_size, |
+ channel_layout, sample_rate, 16, buffer_size, |
+ std::vector<media::Point>(), |
device_info_.device.input.effects); |
{ |