| 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);
|
|
|