Index: webrtc/audio/audio_send_stream.h |
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h |
index 61dd7f24b45f2662e897df12936f72bd86252c29..264f9b3a69f53d9c71e732919e10619e51edb26c 100644 |
--- a/webrtc/audio/audio_send_stream.h |
+++ b/webrtc/audio/audio_send_stream.h |
@@ -39,6 +39,7 @@ class AudioSendStream final : public webrtc::AudioSendStream { |
void Stop() override; |
bool SendTelephoneEvent(int payload_type, int event, |
int duration_ms) override; |
+ void SetMuted(bool muted) override; |
webrtc::AudioSendStream::Stats GetStats() const override; |
void SignalNetworkState(NetworkState state); |