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..9a63564eceb8bff3ab24c8734878edf7dac8a1c4 100644 |
--- a/content/renderer/media/webrtc_audio_capturer.cc |
+++ b/content/renderer/media/webrtc_audio_capturer.cc |
@@ -343,11 +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, |
- device_info_.device.input.effects); |
+ channel_layout, sample_rate, 16, buffer_size); |
+ params.set_effects(device_info_.device.input.effects); |
{ |
base::AutoLock auto_lock(lock_); |