📦 NEW: 增加批量订阅功能

batcher.go

@@ -0,0 +1,96 @@
+package webrtc
+
+import (
+	"encoding/json"
+
+	. "github.com/pion/webrtc/v3"
+	"go.uber.org/zap"
+)
+
+type Signal struct {
+	Type       string   `json:"type"`
+	StreamList []string `json:"streamList"`
+	Offer      string   `json:"offer"`
+	Answer 	 string   `json:"answer"`
+}
+
+type WebRTCBatcher struct {
+	WebRTCIO
+	PageSize      int
+	PageNum       int
+	subscribers   []*WebRTCBatchSubscriber
+	signalChannel *DataChannel
+}
+
+func (suber *WebRTCBatcher) Start() (err error) {
+	suber.OnICECandidate(func(ice *ICECandidate) {
+		if ice != nil {
+			WebRTCPlugin.Info(ice.ToJSON().Candidate)
+		}
+	})
+	suber.OnDataChannel(func(d *DataChannel) {
+		WebRTCPlugin.Info("OnDataChannel:" + d.Label())
+		suber.signalChannel = d
+		suber.signalChannel.OnMessage(suber.Signal)
+	})
+	if err = suber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: suber.SDP}); err != nil {
+		return
+	}
+	suber.OnConnectionStateChange(func(pcs PeerConnectionState) {
+		WebRTCPlugin.Info("Connection State has changed:" + pcs.String())
+		switch pcs {
+		case PeerConnectionStateConnected:
+
+		case PeerConnectionStateDisconnected, PeerConnectionStateFailed:
+			for _, sub := range suber.subscribers {
+				go sub.Stop()
+			}
+			suber.PeerConnection.Close()
+		}
+	})
+	return
+}
+
+func (suber *WebRTCBatcher) Signal(msg DataChannelMessage) {
+	var s Signal
+	// var offer SessionDescription
+	if err := json.Unmarshal(msg.Data, &s); err != nil {
+		WebRTCPlugin.Error("Signal", zap.Error(err))
+	} else {
+		switch s.Type {
+		case "subscribe":
+
+			// if err = suber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: s.Offer}); err != nil {
+			// 	WebRTCPlugin.Error("Signal SetRemoteDescription", zap.Error(err))
+			// 	return
+			// }
+			// var answer string
+			// if answer, err = suber.GetAnswer(); err == nil {
+			// 	b, _ := json.Marshal(map[string]string{"type": "answer", "answer": answer})
+			// 	err = suber.signalChannel.Send(b)
+			// }
+			// if err != nil {
+			// 	WebRTCPlugin.Error("Signal GetAnswer", zap.Error(err))
+			// 	return
+			// }
+			sub := &WebRTCBatchSubscriber{}
+			sub.WebRTCIO = suber.WebRTCIO
+			for _, streamPath := range s.StreamList {
+				if err = WebRTCPlugin.Subscribe(streamPath, sub); err == nil {
+					suber.subscribers = append(suber.subscribers, sub)
+					go sub.PlayRTP()
+				}
+			}
+			// if offer, err = suber.CreateOffer(nil); err == nil {
+			// 	b, _ := json.Marshal(offer)
+			// 	err = suber.signalChannel.SendText(string(b))
+			// 	suber.SetLocalDescription(offer)
+			// }
+		case "answer":
+			if err = suber.SetRemoteDescription(SessionDescription{Type: SDPTypeAnswer, SDP: s.Answer}); err != nil {
+				WebRTCPlugin.Error("Signal SetRemoteDescription", zap.Error(err))
+				return
+			}
+		}
+	}
+}

main.go

@@ -39,10 +39,10 @@ import (
 // 	"stun:stun01.sipphone.com",
 // }}
 
-// type udpConn struct {
+//	type udpConn struct {
-// 	conn *net.UDPConn
+//		conn *net.UDPConn
-// 	port int
+//		port int
-// }
+//	}
 var (
 	reg_level = regexp.MustCompile("profile-level-id=(4.+f)")
 )
@@ -121,7 +121,7 @@ func (conf *WebRTCConfig) Play_(w http.ResponseWriter, r *http.Request) {
 	if err = WebRTCPlugin.Subscribe(streamPath, &suber); err != nil {
 		http.Error(w, err.Error(), http.StatusBadRequest)
 		return
-	}     
+	}
 	if sdp, err := suber.GetAnswer(); err == nil {
 		w.Write([]byte(sdp))
 	} else {
@@ -176,32 +176,22 @@ var webrtcConfig = &WebRTCConfig{
 
 var WebRTCPlugin = engine.InstallPlugin(webrtcConfig)
 
-
 func (conf *WebRTCConfig) Batch(w http.ResponseWriter, r *http.Request) {
 	w.Header().Set("Content-Type", "application/sdp")
 	bytes, err := ioutil.ReadAll(r.Body)
-	var suber WebRTCSubscriber
+	var suber WebRTCBatcher
 	suber.SDP = string(bytes)
 	if suber.PeerConnection, err = conf.api.NewPeerConnection(Configuration{}); err != nil {
 		http.Error(w, err.Error(), http.StatusInternalServerError)
 		return
 	}
-	suber.OnICECandidate(func(ice *ICECandidate) {
+	if err = suber.Start(); err != nil {
-		if ice != nil {
-			suber.Info(ice.ToJSON().Candidate)
-		}
-	})
-	if err = suber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: suber.SDP}); err != nil {
 		http.Error(w, err.Error(), http.StatusInternalServerError)
 		return
 	}
-	// if err = WebRTCPlugin.Subscribe(streamPath, &suber); err != nil {
-	// 	http.Error(w, err.Error(), http.StatusBadRequest)
-	// 	return
-	// }
 	if sdp, err := suber.GetAnswer(); err == nil {
 		w.Write([]byte(sdp))
 	} else {
 		http.Error(w, err.Error(), http.StatusBadRequest)
 	}
-}
+}

subscriber.go

@@ -2,6 +2,7 @@ package webrtc
 
 import (
 	"fmt"
+	"net"
 	"strings"
 
 	"github.com/pion/rtcp"
@@ -9,25 +10,37 @@ import (
 	. "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
+	videoTrack   *TrackLocalStaticRTP
-	audioTrack *TrackLocalStaticRTP
+	audioTrack   *TrackLocalStaticRTP
+	DC           *DataChannel
+	flvHeadCache []byte
 }
 
 func (suber *WebRTCSubscriber) OnEvent(event any) {
 	switch v := event.(type) {
 	case *track.Video:
-		if v.CodecID == codec.CodecID_H264 {
+		switch v.CodecID {
+		case codec.CodecID_H264:
 			pli := "42001f"
 			pli = fmt.Sprintf("%x", v.GetDecoderConfiguration().Raw[0][1:4])
 			if !strings.Contains(suber.SDP, pli) {
 				pli = reg_level.FindAllStringSubmatch(suber.SDP, -1)[0][1]
 			}
-			suber.videoTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: MimeTypeH264, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=" + pli}, "video", "m7s")
+			suber.videoTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: MimeTypeH264, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=" + pli}, "video", 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 {
 			rtpSender, _ := suber.PeerConnection.AddTrack(suber.videoTrack)
 			go func() {
 				rtcpBuf := make([]byte, 1500)
@@ -47,20 +60,46 @@ func (suber *WebRTCSubscriber) OnEvent(event any) {
 					}
 				}
 			}()
-			suber.Subscriber.AddTrack(v) //接受这个track
 		}
+		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}, "audio", "m7s")
+			suber.audioTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: audioMimeType}, "audio", suber.Subscriber.Stream.Path)
 			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(v.FLV))
+		}
 	case VideoRTP:
-		suber.videoTrack.WriteRTP(&v.Packet)
+		if suber.videoTrack != nil {
+			suber.videoTrack.WriteRTP(&v.Packet)
+		} else if suber.DC != nil {
+			frame := suber.Video.Frame
+			dataSize := uint32(util.SizeOfBuffers(frame.AVCC))
+			result := net.Buffers{suber.flvHeadCache[:11]}
+			result = append(result, frame.AVCC...)
+			ts := frame.AbsTime - suber.SkipTS
+			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:
@@ -79,9 +118,15 @@ func (suber *WebRTCSubscriber) OnEvent(event any) {
 	}
 }
 
-// WebRTCBatcher 批量订阅者
+type WebRTCBatchSubscriber struct {
-type WebRTCBatcher struct {
+	WebRTCSubscriber
-	WebRTCIO
+}
-	PageSize int
+
-	PageNum  int
+func (suber *WebRTCBatchSubscriber) OnEvent(event any) {
+	switch event.(type) {
+	case ISubscriber:
+
+	default:
+		suber.WebRTCSubscriber.OnEvent(event)
+	}
 }