package webrtc import ( "fmt" "time" "github.com/pion/rtcp" . "github.com/pion/webrtc/v3" . "m7s.live/engine/v4" . "m7s.live/engine/v4/track" ) type WebRTCPublisher struct { Publisher WebRTCIO } func (puber *WebRTCPublisher) OnEvent(event any) { switch v := event.(type) { case IPublisher: puber.OnICEConnectionStateChange(func(connectionState ICEConnectionState) { puber.Info("Connection State has changed:" + connectionState.String()) switch connectionState { case ICEConnectionStateDisconnected, ICEConnectionStateFailed: puber.Stop() } }) puber.OnTrack(func(track *TrackRemote, receiver *RTPReceiver) { if codec := track.Codec(); track.Kind() == RTPCodecTypeAudio { if puber.Equal(v) || puber.AudioTrack == nil { switch codec.PayloadType { case 8: puber.AudioTrack = NewG711(puber.Stream, true) case 0: puber.AudioTrack = NewG711(puber.Stream, false) default: puber.AudioTrack = nil return } } for { b := make([]byte, 1460) if i, _, err := track.Read(b); err == nil { puber.AudioTrack.WriteRTP(b[:i]) } else { return } } } else { go func() { ticker := time.NewTicker(time.Millisecond * webrtcConfig.PLI) for { select { case <-ticker.C: if rtcpErr := puber.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: uint32(track.SSRC())}}); rtcpErr != nil { fmt.Println(rtcpErr) } case <-puber.Done(): return } } }() if puber.Equal(v) { puber.VideoTrack = NewH264(puber.Stream) } for { b := make([]byte, 1460) if i, _, err := track.Read(b); err == nil { puber.VideoTrack.WriteRTP(b[:i]) } else { return } } } }) } puber.Publisher.OnEvent(event) }