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; |