package webrtc import ( "net" "github.com/pion/rtcp" . "github.com/pion/webrtc/v3" . "m7s.live/engine/v4" "m7s.live/engine/v4/codec" "m7s.live/engine/v4/track" "m7s.live/engine/v4/util" ) type WebRTCSubscriber struct { Subscriber WebRTCIO videoTrack *TrackLocalStaticRTP audioTrack *TrackLocalStaticRTP videoSender *RTPSender audioSender *RTPSender DC *DataChannel flvHeadCache []byte } func (suber *WebRTCSubscriber) OnEvent(event any) { switch v := event.(type) { case *track.Video: switch v.CodecID { case codec.CodecID_H264: pli := "420028" // pli = fmt.Sprintf("%x", v.GetDecoderConfiguration().Raw[0][1:4]) // if !strings.Contains(suber.SDP, pli) { // list := reg_level.FindAllStringSubmatch(suber.SDP, -1) // if len(list) > 0 { // pli = list[0][1] // } // } suber.videoTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: MimeTypeH264, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=" + pli}, v.Name, suber.Subscriber.Stream.Path) case codec.CodecID_H265: // suber.videoTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: MimeTypeH265, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=" + pli}, "video", suber.Subscriber.Stream.Path) default: return } if suber.videoTrack == nil { suber.DC, _ = suber.PeerConnection.CreateDataChannel(suber.Subscriber.Stream.Path, nil) } else { suber.videoSender, _ = suber.PeerConnection.AddTrack(suber.videoTrack) go func() { rtcpBuf := make([]byte, 1500) for { if n, _, rtcpErr := suber.videoSender.Read(rtcpBuf); rtcpErr != nil { return } else { if p, err := rtcp.Unmarshal(rtcpBuf[:n]); err == nil { for _, pp := range p { switch pp.(type) { case *rtcp.PictureLossIndication: // fmt.Println("PictureLossIndication") } } } } } }() } suber.Subscriber.AddTrack(v) //接受这个track case *track.Audio: audioMimeType := MimeTypePCMA if v.CodecID == codec.CodecID_PCMU { audioMimeType = MimeTypePCMU } if v.CodecID == codec.CodecID_PCMA || v.CodecID == codec.CodecID_PCMU { suber.audioTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: audioMimeType}, v.Name, suber.Subscriber.Stream.Path) suber.audioSender, _ = suber.PeerConnection.AddTrack(suber.audioTrack) suber.Subscriber.AddTrack(v) //接受这个track } case VideoDeConf: if suber.DC != nil { if suber.flvHeadCache == nil { suber.flvHeadCache = make([]byte, 15) suber.flvHeadCache[0] = 9 suber.DC.Send(codec.FLVHeader) } suber.DC.Send(util.ConcatBuffers(codec.VideoAVCC2FLV(0, v))) } case VideoRTP: if suber.videoTrack != nil { suber.videoTrack.WriteRTP(&v.Packet) } else if suber.DC != nil { frame := suber.VideoReader.Frame dataSize := uint32(frame.AVCC.ByteLength) result := net.Buffers{suber.flvHeadCache[:11]} result = append(result, frame.AVCC.ToBuffers()...) ts := suber.VideoReader.AbsTime util.PutBE(suber.flvHeadCache[1:4], dataSize) util.PutBE(suber.flvHeadCache[4:7], ts) suber.flvHeadCache[7] = byte(ts >> 24) result = append(result, util.PutBE(suber.flvHeadCache[11:15], dataSize+11)) for _, data := range util.SplitBuffers(result, 65535) { for _, d := range data { suber.DC.Send(d) } } } case AudioRTP: suber.audioTrack.WriteRTP(&v.Packet) case ISubscriber: suber.OnConnectionStateChange(func(pcs PeerConnectionState) { suber.Info("Connection State has changed:" + pcs.String()) switch pcs { case PeerConnectionStateConnected: go suber.PlayRTP() case PeerConnectionStateDisconnected, PeerConnectionStateFailed: suber.Stop() suber.PeerConnection.Close() } }) default: suber.Subscriber.OnEvent(event) } } type WebRTCBatchSubscriber struct { WebRTCSubscriber } func (suber *WebRTCBatchSubscriber) OnEvent(event any) { switch event.(type) { case ISubscriber: default: suber.WebRTCSubscriber.OnEvent(event) } }