| Index: webrtc/media/engine/fakewebrtccall.h
|
| diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
|
| index 89a644a2960a121e07fb8707ebcd8e9fd8226f3e..41a92dfac069fa576ec4e012b2dbca237094d27f 100644
|
| --- a/webrtc/media/engine/fakewebrtccall.h
|
| +++ b/webrtc/media/engine/fakewebrtccall.h
|
| @@ -35,8 +35,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
|
| public:
|
| struct TelephoneEvent {
|
| int payload_type = -1;
|
| - uint8_t event_code = 0;
|
| - uint32_t duration_ms = 0;
|
| + int event_code = 0;
|
| + int duration_ms = 0;
|
| };
|
|
|
| explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
|
| @@ -56,8 +56,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
|
| }
|
|
|
| // webrtc::AudioSendStream implementation.
|
| - bool SendTelephoneEvent(int payload_type, uint8_t event,
|
| - uint32_t duration_ms) override;
|
| + bool SendTelephoneEvent(int payload_type, int event,
|
| + int duration_ms) override;
|
| webrtc::AudioSendStream::Stats GetStats() const override;
|
|
|
| TelephoneEvent latest_telephone_event_;
|
|
|