| Index: webrtc/audio/audio_send_stream.cc
|
| diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
|
| index 9aa2fc004c2295dc2c2c6bdb97996f6d0d6848c0..d08bfeaa1ef70c43de1de0fd57f1e252966542ec 100644
|
| --- a/webrtc/audio/audio_send_stream.cc
|
| +++ b/webrtc/audio/audio_send_stream.cc
|
| @@ -127,6 +127,11 @@ bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
|
| channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
|
| }
|
|
|
| +void AudioSendStream::SetMuted(bool muted) {
|
| + RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
| + channel_proxy_->SetInputMute(muted);
|
| +}
|
| +
|
| webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
|
| RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
| webrtc::AudioSendStream::Stats stats;
|
|
|