| 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 {
|
|
|