| 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_);
|
|
|