Index: webrtc/media/engine/fakewebrtccall.h |
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h |
index 2d1cb7415b1d7247c2ab18061a4fe1fcbc214857..4a0e41cb5959e7ab99ca78f9a8d882431cb533f4 100644 |
--- a/webrtc/media/engine/fakewebrtccall.h |
+++ b/webrtc/media/engine/fakewebrtccall.h |
@@ -46,6 +46,7 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream { |
void SetStats(const webrtc::AudioSendStream::Stats& stats); |
TelephoneEvent GetLatestTelephoneEvent() const; |
bool IsSending() const { return sending_; } |
+ bool muted() const { return muted_; } |
private: |
// webrtc::AudioSendStream implementation. |
@@ -54,12 +55,14 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream { |
bool SendTelephoneEvent(int payload_type, int event, |
int duration_ms) override; |
+ void SetMuted(bool muted) override; |
webrtc::AudioSendStream::Stats GetStats() const override; |
TelephoneEvent latest_telephone_event_; |
webrtc::AudioSendStream::Config config_; |
webrtc::AudioSendStream::Stats stats_; |
bool sending_ = false; |
+ bool muted_ = false; |
}; |
class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |