| Index: webrtc/media/engine/fakewebrtccall.h
|
| diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
|
| index 59084fc4c8fabdaf5961acbfb6a3d44703b4ceba..3a34b34252108db92b4337f95c42dfc7d6593e91 100644
|
| --- a/webrtc/media/engine/fakewebrtccall.h
|
| +++ b/webrtc/media/engine/fakewebrtccall.h
|
| @@ -37,6 +37,7 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
|
| public:
|
| struct TelephoneEvent {
|
| int payload_type = -1;
|
| + int payload_frequency = -1;
|
| int event_code = 0;
|
| int duration_ms = 0;
|
| };
|
| @@ -54,7 +55,7 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
|
| void Start() override { sending_ = true; }
|
| void Stop() override { sending_ = false; }
|
|
|
| - bool SendTelephoneEvent(int payload_type, int event,
|
| + bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
|
| int duration_ms) override;
|
| void SetMuted(bool muted) override;
|
| webrtc::AudioSendStream::Stats GetStats() const override;
|
|
|