Go ws+webrtc实现视频多人同步

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main.go

//go:build !js
// +build !js

// sfu-ws is a many-to-many websocket based SFU
package main

import (
	"encoding/json"
	"flag"
	"github.com/pion/webrtc/v4"
	"log"
	"net/http"
	"os"
	"sync"
	"text/template"
	"time"

	"github.com/gorilla/websocket"
	"github.com/pion/rtcp"
)

// nolint
var (
	addr     = flag.String("addr", ":8080", "http service address")
	upgrader = websocket.Upgrader{
		CheckOrigin: func(r *http.Request) bool { return true },
	}
	indexTemplate = &template.Template{}

	// lock for peerConnections and trackLocals
	listLock        sync.RWMutex
	peerConnections []peerConnectionState
	trackLocals     map[string]*webrtc.TrackLocalStaticRTP
)

type websocketMessage struct {
	Event string `json:"event"`
	Data  string `json:"data"`
}

type peerConnectionState struct {
	peerConnection *webrtc.PeerConnection
	websocket      *threadSafeWriter
}

func main() {
	// Parse the flags passed to program
	flag.Parse()

	// Init other state
	log.SetFlags(0)
	trackLocals = map[string]*webrtc.TrackLocalStaticRTP{}

	// Read index.html from disk into memory, serve whenever anyone requests /
	indexHTML, err := os.ReadFile("index.html")
	if err != nil {
		panic(err)
	}
	indexTemplate = template.Must(template.New("").Parse(string(indexHTML)))

	// websocket handler
	http.HandleFunc("/websocket", websocketHandler)

	// index.html handler
	http.HandleFunc("/", func(w http.ResponseWriter, r *http.Request) {
		if err := indexTemplate.Execute(w, "ws://"+r.Host+"/websocket"); err != nil {
			log.Fatal(err)
		}
	})

	// request a keyframe every 3 seconds
	go func() {
		for range time.NewTicker(time.Second * 3).C {
			dispatchKeyFrame()
		}
	}()

	// start HTTP server
	log.Fatal(http.ListenAndServe(*addr, nil)) // nolint:gosec
}

// Add to list of tracks and fire renegotation for all PeerConnections
func addTrack(t *webrtc.TrackRemote) *webrtc.TrackLocalStaticRTP {
	listLock.Lock()
	defer func() {
		listLock.Unlock()
		signalPeerConnections()
	}()

	// Create a new TrackLocal with the same codec as our incoming
	trackLocal, err := webrtc.NewTrackLocalStaticRTP(t.Codec().RTPCodecCapability, t.ID(), t.StreamID())
	if err != nil {
		panic(err)
	}

	trackLocals[t.ID()] = trackLocal
	return trackLocal
}

// Remove from list of tracks and fire renegotation for all PeerConnections
func removeTrack(t *webrtc.TrackLocalStaticRTP) {
	listLock.Lock()
	defer func() {
		listLock.Unlock()
		signalPeerConnections()
	}()

	delete(trackLocals, t.ID())
}

// signalPeerConnections updates each PeerConnection so that it is getting all the expected media tracks
func signalPeerConnections() {
	listLock.Lock()
	defer func() {
		listLock.Unlock()
		dispatchKeyFrame()
	}()

	attemptSync := func() (tryAgain bool) {
		for i := range peerConnections {
			if peerConnections[i].peerConnection.ConnectionState() == webrtc.PeerConnectionStateClosed {
				peerConnections = append(peerConnections[:i], peerConnections[i+1:]...)
				return true // We modified the slice, start from the beginning
			}

			// map of sender we already are seanding, so we don't double send
			existingSenders := map[string]bool{}

			for _, sender := range peerConnections[i].peerConnection.GetSenders() {
				if sender.Track() == nil {
					continue
				}

				existingSenders[sender.Track().ID()] = true

				// If we have a RTPSender that doesn't map to a existing track remove and signal
				if _, ok := trackLocals[sender.Track().ID()]; !ok {
					if err := peerConnections[i].peerConnection.RemoveTrack(sender); err != nil {
						return true
					}
				}
			}

			// Don't receive videos we are sending, make sure we don't have loopback
			for _, receiver := range peerConnections[i].peerConnection.GetReceivers() {
				if receiver.Track() == nil {
					continue
				}

				existingSenders[receiver.Track().ID()] = true
			}

			// Add all track we aren't sending yet to the PeerConnection
			for trackID := range trackLocals {
				if _, ok := existingSenders[trackID]; !ok {
					if _, err := peerConnections[i].peerConnection.AddTrack(trackLocals[trackID]); err != nil {
						return true
					}
				}
			}

			offer, err := peerConnections[i].peerConnection.CreateOffer(nil)
			if err != nil {
				return true
			}

			if err = peerConnections[i].peerConnection.SetLocalDescription(offer); err != nil {
				return true
			}

			offerString, err := json.Marshal(offer)
			if err != nil {
				return true
			}

			if err = peerConnections[i].websocket.WriteJSON(&websocketMessage{
				Event: "offer",
				Data:  string(offerString),
			}); err != nil {
				return true
			}
		}

		return
	}

	for syncAttempt := 0; ; syncAttempt++ {
		if syncAttempt == 25 {
			// Release the lock and attempt a sync in 3 seconds. We might be blocking a RemoveTrack or AddTrack
			go func() {
				time.Sleep(time.Second * 3)
				signalPeerConnections()
			}()
			return
		}

		if !attemptSync() {
			break
		}
	}
}

// dispatchKeyFrame sends a keyframe to all PeerConnections, used everytime a new user joins the call
func dispatchKeyFrame() {
	listLock.Lock()
	defer listLock.Unlock()

	for i := range peerConnections {
		for _, receiver := range peerConnections[i].peerConnection.GetReceivers() {
			if receiver.Track() == nil {
				continue
			}

			_ = peerConnections[i].peerConnection.WriteRTCP([]rtcp.Packet{
				&rtcp.PictureLossIndication{
					MediaSSRC: uint32(receiver.Track().SSRC()),
				},
			})
		}
	}
}

// Handle incoming websockets
func websocketHandler(w http.ResponseWriter, r *http.Request) {
	// Upgrade HTTP request to Websocket
	unsafeConn, err := upgrader.Upgrade(w, r, nil)
	if err != nil {
		log.Print("upgrade:", err)
		return
	}

	c := &threadSafeWriter{unsafeConn, sync.Mutex{}}

	// When this frame returns close the Websocket
	defer c.Close() //nolint

	// Create new PeerConnection
	peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{})
	if err != nil {
		log.Print(err)
		return
	}

	// When this frame returns close the PeerConnection
	defer peerConnection.Close() //nolint

	// Accept one audio and one video track incoming
	for _, typ := range []webrtc.RTPCodecType{webrtc.RTPCodecTypeVideo, webrtc.RTPCodecTypeAudio} {
		if _, err := peerConnection.AddTransceiverFromKind(typ, webrtc.RTPTransceiverInit{
			Direction: webrtc.RTPTransceiverDirectionRecvonly,
		}); err != nil {
			log.Print(err)
			return
		}
	}

	// Add our new PeerConnection to global list
	listLock.Lock()
	peerConnections = append(peerConnections, peerConnectionState{peerConnection, c})
	listLock.Unlock()

	// Trickle ICE. Emit server candidate to client
	peerConnection.OnICECandidate(func(i *webrtc.ICECandidate) {
		if i == nil {
			return
		}

		candidateString, err := json.Marshal(i.ToJSON())
		if err != nil {
			log.Println(err)
			return
		}

		if writeErr := c.WriteJSON(&websocketMessage{
			Event: "candidate",
			Data:  string(candidateString),
		}); writeErr != nil {
			log.Println(writeErr)
		}
	})

	// If PeerConnection is closed remove it from global list
	peerConnection.OnConnectionStateChange(func(p webrtc.PeerConnectionState) {
		switch p {
		case webrtc.PeerConnectionStateFailed:
			if err := peerConnection.Close(); err != nil {
				log.Print(err)
			}
		case webrtc.PeerConnectionStateClosed:
			signalPeerConnections()
		default:
		}
	})

	peerConnection.OnTrack(func(t *webrtc.TrackRemote, _ *webrtc.RTPReceiver) {
		// Create a track to fan out our incoming video to all peers
		trackLocal := addTrack(t)
		defer removeTrack(trackLocal)

		buf := make([]byte, 1500)
		for {
			i, _, err := t.Read(buf)
			if err != nil {
				return
			}

			if _, err = trackLocal.Write(buf[:i]); err != nil {
				return
			}
		}
	})

	// Signal for the new PeerConnection
	signalPeerConnections()

	message := &websocketMessage{}
	for {
		_, raw, err := c.ReadMessage()
		if err != nil {
			log.Println(err)
			return
		} else if err := json.Unmarshal(raw, &message); err != nil {
			log.Println(err)
			return
		}

		switch message.Event {
		case "candidate":
			candidate := webrtc.ICECandidateInit{}
			if err := json.Unmarshal([]byte(message.Data), &candidate); err != nil {
				log.Println(err)
				return
			}

			if err := peerConnection.AddICECandidate(candidate); err != nil {
				log.Println(err)
				return
			}
		case "answer":
			answer := webrtc.SessionDescription{}
			if err := json.Unmarshal([]byte(message.Data), &answer); err != nil {
				log.Println(err)
				return
			}

			if err := peerConnection.SetRemoteDescription(answer); err != nil {
				log.Println(err)
				return
			}
		}
	}
}

// Helper to make Gorilla Websockets threadsafe
type threadSafeWriter struct {
	*websocket.Conn
	sync.Mutex
}

func (t *threadSafeWriter) WriteJSON(v interface{}) error {
	t.Lock()
	defer t.Unlock()

	return t.Conn.WriteJSON(v)
}

index.html

<!DOCTYPE html>
<html>
<!--
  SPDX-FileCopyrightText: 2023 The Pion community <https://pion.ly>
  SPDX-License-Identifier: MIT
-->

<head>
    <meta charset="utf-8">
</head>
<body>
<h3> Local Video </h3>
<video id="localVideo" width="160" height="120" autoplay muted></video>
<br/>

<h3> Remote Video </h3>
<div id="remoteVideos"></div>
<br/>

<h3> Logs </h3>
<div id="logs"></div>


</body>

<script>
    navigator.mediaDevices.getUserMedia({video: true, audio: true})
        .then(stream => {
            let pc = new RTCPeerConnection()
            pc.ontrack = function (event) {
                if (event.track.kind === 'audio') {
                    return
                }

                let el = document.createElement(event.track.kind)
                el.srcObject = event.streams[0]
                el.autoplay = true
                el.controls = true
                document.getElementById('remoteVideos').appendChild(el)

                event.track.onmute = function (event) {
                    el.play()
                }

                event.streams[0].onremovetrack = ({track}) => {
                    if (el.parentNode) {
                        el.parentNode.removeChild(el)
                    }
                }
            }

            document.getElementById('localVideo').srcObject = stream
            stream.getTracks().forEach(track => pc.addTrack(track, stream))

            let ws = new WebSocket("{{.}}")
            pc.onicecandidate = e => {
                if (!e.candidate) {
                    return
                }

                ws.send(JSON.stringify({event: 'candidate', data: JSON.stringify(e.candidate)}))
            }

            ws.onclose = function (evt) {
                window.alert("Websocket has closed")
            }

            ws.onmessage = function (evt) {
                let msg = JSON.parse(evt.data)
                if (!msg) {
                    return console.log('failed to parse msg')
                }

                switch (msg.event) {
                    case 'offer':
                        let offer = JSON.parse(msg.data)
                        if (!offer) {
                            return console.log('failed to parse answer')
                        }
                        pc.setRemoteDescription(offer)
                        pc.createAnswer().then(answer => {
                            pc.setLocalDescription(answer)
                            ws.send(JSON.stringify({event: 'answer', data: JSON.stringify(answer)}))
                        })
                        return

                    case 'candidate':
                        let candidate = JSON.parse(msg.data)
                        if (!candidate) {
                            return console.log('failed to parse candidate')
                        }

                        pc.addIceCandidate(candidate)
                }
            }

            ws.onerror = function (evt) {
                console.log("ERROR: " + evt.data)
            }
        }).catch(window.alert)
</script>
</html>