| 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);
|
|
|
| {
|
|
|