diff options
Diffstat (limited to 'main.go')
| -rw-r--r-- | main.go | 236 |
1 files changed, 236 insertions, 0 deletions
@@ -0,0 +1,236 @@ +package main + +import ( + "encoding/json" + "flag" + "fmt" + "log" + "net/http" + "strconv" + + "github.com/gorilla/websocket" + "github.com/pion/rtwatch/gst" + "github.com/pion/webrtc/v2" +) + +const homeHTML = `<!DOCTYPE html> +<html lang="en"> + <head> + <title>synced-playback</title> + </head> + <body id="body"> + <video id="video1" autoplay playsinline></video> + + <div> + <input type="number" id="seekTime" value="30"> + <button type="button" onClick="seekClick()">Seek</button> + <button type="button" onClick="playClick()">Play</button> + <button type="button" onClick="pauseClick()">Pause</button> + </div> + + <script> + let conn = new WebSocket('ws://' + window.location.host + '/ws') + let pc = new RTCPeerConnection() + + window.seekClick = () => { + conn.send(JSON.stringify({event: 'seek', data: document.getElementById('seekTime').value})) + } + window.playClick = () => { + conn.send(JSON.stringify({event: 'play', data: ''})) + } + window.pauseClick = () => { + conn.send(JSON.stringify({event: 'pause', data: ''})) + } + + pc.ontrack = function (event) { + if (event.track.kind === 'audio') { + return + } + var el = document.getElementById('video1') + el.srcObject = event.streams[0] + el.autoplay = true + el.controls = true + } + + conn.onopen = () => { + pc.createOffer({offerToReceiveVideo: true, offerToReceiveAudio: true}).then(offer => { + pc.setLocalDescription(offer) + conn.send(JSON.stringify({event: 'offer', data: JSON.stringify(offer)})) + }) + } + conn.onclose = evt => { + console.log('Connection closed') + } + conn.onmessage = evt => { + let msg = JSON.parse(evt.data) + if (!msg) { + return console.log('failed to parse msg') + } + + switch (msg.event) { + case 'answer': + answer = JSON.parse(msg.data) + if (!answer) { + return console.log('failed to parse answer') + } + pc.setRemoteDescription(answer) + } + } + window.conn = conn + </script> + </body> +</html> +` + +var ( + upgrader = websocket.Upgrader{ + ReadBufferSize: 1024, + WriteBufferSize: 1024, + } + + peerConnectionConfig = webrtc.Configuration{} + + audioTrack = &webrtc.Track{} + videoTrack = &webrtc.Track{} + pipeline = &gst.Pipeline{} +) + +type websocketMessage struct { + Event string `json:"event"` + Data string `json:"data"` +} + +func main() { + containerPath := "" + httpListenAddress := "" + flag.StringVar(&containerPath, "container-path", "", "path to the media file you want to playback") + flag.StringVar(&httpListenAddress, "http-listen-address", ":8080", "address for HTTP server to listen on") + flag.Parse() + + if containerPath == "" { + panic("-container-path must be specified") + } + + pc, err := webrtc.NewPeerConnection(webrtc.Configuration{ + ICEServers: []webrtc.ICEServer{ + { + URLs: []string{"stun:stun.l.google.com:19302"}, + }, + }, + }) + if err != nil { + log.Fatal(err) + } + + videoTrack, err = pc.NewTrack(webrtc.DefaultPayloadTypeH264, 5000, "synced-video", "synced-video") + if err != nil { + log.Fatal(err) + } + + audioTrack, err = pc.NewTrack(webrtc.DefaultPayloadTypeOpus, 5001, "synced-video", "synced-video") + if err != nil { + log.Fatal(err) + } + + pipeline = gst.CreatePipeline(containerPath, audioTrack, videoTrack) + pipeline.Start() + + http.HandleFunc("/", serveHome) + http.HandleFunc("/ws", serveWs) + + fmt.Println(fmt.Sprintf("Video file '%s' is now available on '%s', have fun!", containerPath, httpListenAddress)) + log.Fatal(http.ListenAndServe(httpListenAddress, nil)) +} + +func handleWebsocketMessage(pc *webrtc.PeerConnection, ws *websocket.Conn, message *websocketMessage) error { + switch message.Event { + case "play": + pipeline.Play() + case "pause": + pipeline.Pause() + case "seek": + i, err := strconv.ParseInt(message.Data, 0, 64) + if err != nil { + log.Print(err) + } + pipeline.SeekToTime(i) + case "offer": + offer := webrtc.SessionDescription{} + if err := json.Unmarshal([]byte(message.Data), &offer); err != nil { + return err + } + + if err := pc.SetRemoteDescription(offer); err != nil { + return err + } + + answer, err := pc.CreateAnswer(nil) + if err != nil { + return err + } + if err := pc.SetLocalDescription(answer); err != nil { + return err + } + + answerString, err := json.Marshal(answer) + if err != nil { + return err + } + + if err = ws.WriteJSON(&websocketMessage{ + Event: "answer", + Data: string(answerString), + }); err != nil { + return err + } + } + return nil +} + +func serveWs(w http.ResponseWriter, r *http.Request) { + ws, err := upgrader.Upgrade(w, r, nil) + if err != nil { + if _, ok := err.(websocket.HandshakeError); !ok { + log.Println(err) + } + return + } + + peerConnection, err := webrtc.NewPeerConnection(peerConnectionConfig) + if err != nil { + log.Print(err) + return + } else if _, err = peerConnection.AddTrack(audioTrack); err != nil { + log.Print(err) + return + } else if _, err = peerConnection.AddTrack(videoTrack); err != nil { + log.Print(err) + return + } + + defer func() { + if err := peerConnection.Close(); err != nil { + log.Println(err) + } + }() + + message := &websocketMessage{} + for { + _, msg, err := ws.ReadMessage() + if err != nil { + break + } else if err := json.Unmarshal(msg, &message); err != nil { + log.Print(err) + return + } + + if err := handleWebsocketMessage(peerConnection, ws, message); err != nil { + log.Print(err) + } + } +} + +func serveHome(w http.ResponseWriter, r *http.Request) { + w.Header().Set("Content-Type", "text/html; charset=utf-8") + fmt.Fprintf(w, homeHTML) +} |