WebRTC offers multiple quality

go.mod

@@ -5,7 +5,7 @@ go 1.13
 require (
 	github.com/go-ldap/ldap/v3 v3.2.3
 	github.com/gorilla/websocket v1.4.0
-	github.com/haivision/srtgo v0.0.0-20201025191851-67964e8f497a
+	github.com/haivision/srtgo v0.0.0-20200731151239-e00427ae473a
 	github.com/markbates/pkger v0.17.1
 	github.com/pion/rtp v1.6.0
 	github.com/pion/webrtc/v3 v3.0.0-beta.5

go.sum

@@ -122,8 +122,6 @@ github.com/grpc-ecosystem/grpc-gateway v1.5.0/go.mod h1:RSKVYQBd5MCa4OVpNdGskqpg
 github.com/grpc-ecosystem/grpc-gateway v1.9.0/go.mod h1:vNeuVxBJEsws4ogUvrchl83t/GYV9WGTSLVdBhOQFDY=
 github.com/haivision/srtgo v0.0.0-20200731151239-e00427ae473a h1:JliMkv/mAqM5+QzG6Hkw1XcVl1crU8yIQGnhppMv7s0=
 github.com/haivision/srtgo v0.0.0-20200731151239-e00427ae473a/go.mod h1:yVZ4oACfcnUAcxrh+0b6IuIWfkHLK3IAQ99tuuhRx54=
-github.com/haivision/srtgo v0.0.0-20201025191851-67964e8f497a h1:54abJQezjMoiP+xMQ3ZQbcDXFjqytAYm/n0EVqrYeXg=
-github.com/haivision/srtgo v0.0.0-20201025191851-67964e8f497a/go.mod h1:7izzTiCO3zc9ZIVTFMjxUiYL+kgryFP9rl3bsweqdmc=
 github.com/hashicorp/hcl v1.0.0 h1:0Anlzjpi4vEasTeNFn2mLJgTSwt0+6sfsiTG8qcWGx4=
 github.com/hashicorp/hcl v1.0.0/go.mod h1:E5yfLk+7swimpb2L/Alb/PJmXilQ/rhwaUYs4T20WEQ=
 github.com/hpcloud/tail v1.0.0/go.mod h1:ab1qPbhIpdTxEkNHXyeSf5vhxWSCs/tWer42PpOxQnU=
@@ -418,8 +416,6 @@ golang.org/x/sys v0.0.0-20200519105757-fe76b779f299/go.mod h1:h1NjWce9XRLGQEsW7w
 golang.org/x/sys v0.0.0-20200610111108-226ff32320da/go.mod h1:h1NjWce9XRLGQEsW7wpKNCjG9DtNlClVuFLEZdDNbEs=
 golang.org/x/sys v0.0.0-20200615200032-f1bc736245b1 h1:ogLJMz+qpzav7lGMh10LMvAkM/fAoGlaiiHYiFYdm80=
 golang.org/x/sys v0.0.0-20200615200032-f1bc736245b1/go.mod h1:h1NjWce9XRLGQEsW7wpKNCjG9DtNlClVuFLEZdDNbEs=
-golang.org/x/sys v0.0.0-20200926100807-9d91bd62050c h1:38q6VNPWR010vN82/SB121GujZNIfAUb4YttE2rhGuc=
-golang.org/x/sys v0.0.0-20200926100807-9d91bd62050c/go.mod h1:h1NjWce9XRLGQEsW7wpKNCjG9DtNlClVuFLEZdDNbEs=
 golang.org/x/text v0.0.0-20170915032832-14c0d48ead0c/go.mod h1:NqM8EUOU14njkJ3fqMW+pc6Ldnwhi/IjpwHt7yyuwOQ=
 golang.org/x/text v0.3.0/go.mod h1:NqM8EUOU14njkJ3fqMW+pc6Ldnwhi/IjpwHt7yyuwOQ=
 golang.org/x/text v0.3.1-0.20180807135948-17ff2d5776d2/go.mod h1:NqM8EUOU14njkJ3fqMW+pc6Ldnwhi/IjpwHt7yyuwOQ=

messaging/quality.go

@@ -10,6 +10,12 @@ import (
 // Quality holds a specific stream quality.
 // It makes packages able to subscribe to an incoming stream.
 type Quality struct {
+	// Type of the quality
+	Name string
+
+	// Source Stream
+	Stream *Stream
+
 	// Incoming data come from this channel
 	Broadcast chan<- []byte
 
@@ -27,8 +33,9 @@ type Quality struct {
 	WebRtcRemoteSdp chan webrtc.SessionDescription
 }
 
-func newQuality() (q *Quality) {
+func newQuality(name string, stream *Stream) (q *Quality) {
-	q = &Quality{}
+	q = &Quality{Name: name}
+	q.Stream = stream
 	broadcast := make(chan []byte, 1024)
 	q.Broadcast = broadcast
 	q.outputs = make(map[chan []byte]struct{})

messaging/stream.go

@@ -40,7 +40,7 @@ func (s *Stream) CreateQuality(name string) (quality *Quality, err error) {
 	}
 
 	s.lockQualities.Lock()
-	quality = newQuality()
+	quality = newQuality(name, s)
 	s.qualities[name] = quality
 	s.lockQualities.Unlock()
 	return quality, nil

stream/srt/handler.go

@@ -24,6 +24,17 @@ func handleStreamer(socket *srtgo.SrtSocket, streams *messaging.Streams, name st
 		socket.Close()
 		return
 	}
+
+	// Create sub-qualities
+	for _, qualityName := range []string{"audio", "480p", "360p", "240p"} {
+		_, err := stream.CreateQuality(qualityName)
+		if err != nil {
+			log.Printf("Error on quality creating: %s", err)
+			socket.Close()
+			return
+		}
+	}
+
 	log.Printf("New SRT streamer for stream '%s' quality 'source'", name)
 
 	// Read RTP packets forever and send them to the WebRTC Client

stream/srt/srt.go

@@ -1,6 +1,9 @@
 // Package srt serves a SRT server
 package srt
 
+// #include <srt/srt.h>
+import "C"
+
 import (
 	"log"
 	"net"
@@ -59,7 +62,7 @@ func Serve(streams *messaging.Streams, authBackend auth.Backend, cfg *Options) {
 
 	for {
 		// Wait for new connection
-		s, _, err := sck.Accept()
+		s, err := sck.Accept()
 		if err != nil {
 			// Something wrong happened
 			log.Println(err)
@@ -70,7 +73,7 @@ func Serve(streams *messaging.Streams, authBackend auth.Backend, cfg *Options) {
 		// Without this, the SRT buffer might get full before reading it
 
 		// streamid can be "name:password" for streamer or "name" for viewer
-		streamID, err := s.GetSockOptString(srtgo.SRTO_STREAMID)
+		streamID, err := s.GetSockOptString(C.SRTO_STREAMID)
 		if err != nil {
 			log.Print("Failed to get socket streamid")
 			continue

stream/webrtc/ingest.go

@@ -3,9 +3,7 @@ package webrtc
 
 import (
 	"bufio"
-	"fmt"
 	"log"
-	"math/rand"
 	"net"
 	"os/exec"
 
@@ -16,36 +14,61 @@ import (
 
 func ingest(name string, q *messaging.Quality) {
 	// Register to get stream
-	videoInput := make(chan []byte, 1024)
+	input := make(chan []byte, 1024)
-	q.Register(videoInput)
+	// FIXME Stream data should already be transcoded
+	source, _ := q.Stream.GetQuality("source")
+	source.Register(input)
 
-	// FIXME Mux into RTP without having multiple UDP listeners
+	// FIXME Bad code
-	firstPort := int(rand.Int31n(63535)) + 2000
+	port := 5000
-
+	var tracks map[string][]*webrtc.Track
-	// Open UDP listeners for RTP Packets
+	qualityName := ""
-	audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: firstPort})
+	switch q.Name {
-	if err != nil {
+	case "audio":
-		log.Printf("Faited to open UDP listener %s", err)
+		port = 5004
-		return
+		tracks = audioTracks
+		break
+	case "source":
+		port = 5005
+		tracks = videoTracks
+		qualityName = "@source"
+		break
+	case "480p":
+		port = 5006
+		tracks = videoTracks
+		qualityName = "@480p"
+		break
+	case "360p":
+		port = 5007
+		tracks = videoTracks
+		qualityName = "@360p"
+		break
+	case "240p":
+		port = 5008
+		tracks = videoTracks
+		qualityName = "@240p"
+		break
 	}
-	videoListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: firstPort + 1})
+
+	// Open a UDP Listener for RTP Packets
+	listener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: port})
 	if err != nil {
 		log.Printf("Faited to open UDP listener %s", err)
 		return
 	}
 
-	// Start ffmpag to convert videoInput to video and audio UDP
+	// Start ffmpag to convert input to video and audio UDP
-	ffmpeg, err := startFFmpeg(videoInput, firstPort)
+	ffmpeg, err := startFFmpeg(q, input)
 	if err != nil {
 		log.Printf("Error while starting ffmpeg: %s", err)
 		return
 	}
 
-	// Receive video
+	// Receive stream
 	go func() {
 		inboundRTPPacket := make([]byte, 1500) // UDP MTU
 		for {
-			n, _, err := videoListener.ReadFromUDP(inboundRTPPacket)
+			n, _, err := listener.ReadFromUDP(inboundRTPPacket)
 			if err != nil {
 				log.Printf("Failed to read from UDP: %s", err)
 				break
@@ -56,49 +79,13 @@ func ingest(name string, q *messaging.Quality) {
 				continue
 			}
 
-			if videoTracks[name] == nil {
+			// Write RTP srtPacket to all tracks
-				videoTracks[name] = make([]*webrtc.Track, 0)
-			}
-
-			// Write RTP srtPacket to all video tracks
 			// Adapt payload and SSRC to match destination
-			for _, videoTrack := range videoTracks[name] {
+			for _, track := range tracks[name+qualityName] {
-				packet.Header.PayloadType = videoTrack.PayloadType()
+				packet.Header.PayloadType = track.PayloadType()
-				packet.Header.SSRC = videoTrack.SSRC()
+				packet.Header.SSRC = track.SSRC()
-				if writeErr := videoTrack.WriteRTP(packet); writeErr != nil {
+				if writeErr := track.WriteRTP(packet); writeErr != nil {
-					log.Printf("Failed to write to video track: %s", err)
+					log.Printf("Failed to write to track: %s", writeErr)
-					continue
-				}
-			}
-		}
-	}()
-
-	// Receive audio
-	go func() {
-		inboundRTPPacket := make([]byte, 1500) // UDP MTU
-		for {
-			n, _, err := audioListener.ReadFromUDP(inboundRTPPacket)
-			if err != nil {
-				log.Printf("Failed to read from UDP: %s", err)
-				break
-			}
-			packet := &rtp.Packet{}
-			if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
-				log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
-				continue
-			}
-
-			if audioTracks[name] == nil {
-				audioTracks[name] = make([]*webrtc.Track, 0)
-			}
-
-			// Write RTP srtPacket to all audio tracks
-			// Adapt payload and SSRC to match destination
-			for _, audioTrack := range audioTracks[name] {
-				packet.Header.PayloadType = audioTrack.PayloadType()
-				packet.Header.SSRC = audioTrack.SSRC()
-				if writeErr := audioTrack.WriteRTP(packet); writeErr != nil {
-					log.Printf("Failed to write to audio track: %s", err)
 					continue
 				}
 			}
@@ -110,24 +97,47 @@ func ingest(name string, q *messaging.Quality) {
 		log.Printf("Faited to wait for ffmpeg: %s", err)
 	}
 
-	// Close UDP listeners
+	// Close UDP listener
-	if err = videoListener.Close(); err != nil {
+	if err = listener.Close(); err != nil {
 		log.Printf("Faited to close UDP listener: %s", err)
 	}
-	if err = audioListener.Close(); err != nil {
+	q.Unregister(input)
-		log.Printf("Faited to close UDP listener: %s", err)
-	}
-	q.Unregister(videoInput)
 }
 
-func startFFmpeg(in <-chan []byte, listeningPort int) (ffmpeg *exec.Cmd, err error) {
+func startFFmpeg(q *messaging.Quality, in <-chan []byte) (ffmpeg *exec.Cmd, err error) {
-	ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-i", "pipe:0",
+	// FIXME Use transcoders to downscale, then remux in RTP
-		// Audio
+	ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-i", "pipe:0"}
-		"-vn", "-c:a", "libopus", "-b:a", "160k",
+	switch q.Name {
-		"-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", listeningPort),
+	case "audio":
-		// Source
+		ffmpegArgs = append(ffmpegArgs, "-vn", "-c:a", "libopus", "-b:a", "160k",
-		"-an", "-c:v", "copy", "-b:v", "3000k", "-maxrate", "5000k", "-bufsize", "5000k",
+			"-f", "rtp", "rtp://127.0.0.1:5004")
-		"-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", listeningPort+1)}
+		break
+	case "source":
+		ffmpegArgs = append(ffmpegArgs, "-an", "-c:v", "copy",
+			"-f", "rtp", "rtp://127.0.0.1:5005")
+		break
+	case "480p":
+		ffmpegArgs = append(ffmpegArgs,
+			"-an", "-c:v", "libx264", "-b:v", "1200k", "-maxrate", "2000k", "-bufsize", "3000k",
+			"-preset", "ultrafast", "-profile", "main", "-tune", "zerolatency",
+			"-vf", "scale=854:480",
+			"-f", "rtp", "rtp://127.0.0.1:5006")
+		break
+	case "360p":
+		ffmpegArgs = append(ffmpegArgs,
+			"-an", "-c:v", "libx264", "-b:v", "800k", "-maxrate", "1200k", "-bufsize", "1500k",
+			"-preset", "ultrafast", "-profile", "main", "-tune", "zerolatency",
+			"-vf", "scale=480:360",
+			"-f", "rtp", "rtp://127.0.0.1:5007")
+		break
+	case "240p":
+		ffmpegArgs = append(ffmpegArgs,
+			"-an", "-c:v", "libx264", "-b:v", "500k", "-maxrate", "800k", "-bufsize", "1000k",
+			"-preset", "ultrafast", "-profile", "main", "-tune", "zerolatency",
+			"-vf", "scale=360:240",
+			"-f", "rtp", "rtp://127.0.0.1:5008")
+		break
+	}
 	ffmpeg = exec.Command("ffmpeg", ffmpegArgs...)
 
 	// Handle errors output

stream/webrtc/webrtc.go

@@ -40,7 +40,7 @@ func removeTrack(tracks []*webrtc.Track, track *webrtc.Track) []*webrtc.Track {
 
 // GetNumberConnectedSessions get the number of currently connected clients
 func GetNumberConnectedSessions(streamID string) int {
-	return len(videoTracks[streamID])
+	return len(audioTracks[streamID])
 }
 
 // newPeerHandler is called when server receive a new session description
@@ -117,21 +117,20 @@ func newPeerHandler(name string, localSdpChan chan webrtc.SessionDescription, re
 		quality = split[1]
 	}
 	log.Printf("New WebRTC session for stream %s, quality %s", streamID, quality)
-	// TODO Consider the quality
 
 	// Set the handler for ICE connection state
 	// This will notify you when the peer has connected/disconnected
 	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
 		log.Printf("Connection State has changed %s \n", connectionState.String())
-		if videoTracks[streamID] == nil {
+		if videoTracks[streamID+"@"+quality] == nil {
-			videoTracks[streamID] = make([]*webrtc.Track, 0, 1)
+			videoTracks[streamID+"@"+quality] = make([]*webrtc.Track, 0, 1)
 		}
 		if audioTracks[streamID] == nil {
 			audioTracks[streamID] = make([]*webrtc.Track, 0, 1)
 		}
 		if connectionState == webrtc.ICEConnectionStateConnected {
 			// Register tracks
-			videoTracks[streamID] = append(videoTracks[streamID], videoTrack)
+			videoTracks[streamID+"@"+quality] = append(videoTracks[streamID+"@"+quality], videoTrack)
 			audioTracks[streamID] = append(audioTracks[streamID], audioTrack)
 			monitoring.WebRTCConnectedSessions.Inc()
 		} else if connectionState == webrtc.ICEConnectionStateDisconnected {
@@ -205,16 +204,17 @@ func Serve(streams *messaging.Streams, cfg *Options) {
 
 		// Get specific quality
 		// FIXME: make it possible to forward other qualities
-		qualityName := "source"
+		for _, qualityName := range []string{"source", "audio", "480p", "360p", "240p"} {
-		quality, err := stream.GetQuality(qualityName)
+			quality, err := stream.GetQuality(qualityName)
-		if err != nil {
+			if err != nil {
-			log.Printf("Failed to get quality '%s'", qualityName)
+				log.Printf("Failed to get quality '%s'", qualityName)
-		}
+			}
 
-		// Start forwarding
+			// Start forwarding
-		log.Printf("Starting webrtc for '%s' quality '%s'", name, qualityName)
+			log.Printf("Starting webrtc for '%s' quality '%s'", name, qualityName)
-		go ingest(name, quality)
+			go ingest(name, quality)
-		go listenSdp(name, quality.WebRtcLocalSdp, quality.WebRtcRemoteSdp, cfg)
+			go listenSdp(name, quality.WebRtcLocalSdp, quality.WebRtcRemoteSdp, cfg)
+		}
 	}
 }
 

web/static/js/viewer.js

@@ -14,7 +14,7 @@ export function initViewerPage(stream, stunServers, viewersCounterRefreshPeriod)
     const viewer = document.getElementById("viewer");
 
     // Default quality
-    let quality = "source";
+    let quality = "240p";
 
     // Create WebSocket and WebRTC
     const websocket = new GsWebSocket();

web/template/index.html

@@ -21,7 +21,11 @@
   <ul>
     <li>
       <b>Serveur :</b>
-      <code>srt://{{.Cfg.Hostname}}:{{.Cfg.SRTServerPort}}?IDENTIFIANT:MOT_DE_PASS</code>,
+      <code>srt://{{.Cfg.Hostname}}:{{.Cfg.SRTServerPort}}</code>,
+    </li>
+    <li>
+      <b>Clé de stream :</b>
+      <code>IDENTIFIANT:MOT_DE_PASSE</code>
       avec <code>IDENTIFIANT</code> et <code>MOT_DE_PASSE</code>
       vos identifiants.
     </li>
@@ -48,81 +52,6 @@
     </code>
   </p>
 
-  <h2>Comment lire un flux depuis un lecteur externe ?</h2>
-  <p>
-    À l'heure actuelle, la plupart des lecteurs vidéos ne supportent
-    pas le protocole SRT, ou le supportent mal. Un travail est en
-    cours pour les rendre un maximum compatibles. Liste non exhaustive
-    des lecteurs vidéos testés :
-  </p>
-
-  <h3>FFPlay</h3>
-  <p>
-    Si FFMpeg est installé sur votre machine, il est accompagné d'un
-    lecteur vidéo nommé <code>ffplay</code>. Si FFMpeg est compilé
-    avec le support de SRT (c'est le cas sur la plupart des distributions,
-    sauf cas ci-dessous), il vous suffira d'exécuter :
-  </p>
-
-  <p>
-    <code>
-      ffplay -fflags nobuffer srt://{{.Cfg.Hostname}}:{{.Cfg.SRTServerPort}}?streamid=IDENTIFIANT
-    </code>
-  </p>
-
-  <h3>MPV</h3>
-
-  <p>
-    MPV supporte officiellement SRT depuis le 16 octobre 2020.
-    Néanmoins, la version stable de MPV est beaucoup plus vieille.
-    Vous devez alors utiliser une version de développement pour
-    pouvoir lire un flux SRT depuis MPV. L'installation se fait
-    depuis <a href="https://mpv.io/installation/"> cette page</a>.
-    Sous Arch Linux, il vous suffit de récupérer le paquet
-    <code>mpv-git</code> dans l'AUR. Pour lire le flux, il suffit
-    d'exécuter :
-  </p>
-
-  <p>
-    <code>
-      mpv srt://{{.Cfg.Hostname}}:{{.Cfg.SRTServerPort}}?streamid=IDENTIFIANT
-    </code>
-  </p>
-
-  <h3>VLC Media Player</h3>
-
-  <p>
-    Bien que VLC supporte officiellement le protocole SRT,
-    toutes les options ne sont pas encore implémentées,
-    notamment l'option pour choisir son stream.
-    <a href="https://patches.videolan.org/patch/30299/">Un patch</a>
-    a été soumis et est en attente d'acceptation.
-    Une fois le patch accepté, il sera appliqué dans les versions
-    de développement de VLC. Sous Arch Linux, il suffit de récupérer
-    le paquet <code>vlc-git</code> de l'AUR. Avec un VLC à jour,
-    il suffit d'exécuter :
-  </p>
-
-  <p>
-    <code>
-      vlc srt://{{.Cfg.Hostname}}:{{.Cfg.SRTServerPort}}?streamid=IDENTIFIANT
-    </code>
-  </p>
-
-  <p>
-    Ou bien d'aller dans Média -> Ouvrir un flux réseau et d'entrer l'URL
-    <code>srt://{{.Cfg.Hostname}}:{{.Cfg.SRTServerPort}}?streamid=IDENTIFIANT</code>.
-  </p>
-
-  <h3>Le protocole n'existe pas ou n'est pas supporté.</h3>
-  <p>
-    La technologie SRT est très récente et n'est pas supportée par
-    les dépôts stables. Ainsi, si vous avez Ubuntu &le; 20.04 ou
-    Debian &le; Buster, vous ne pourrez pas utiliser de lecteur vidéo
-    ni même diffuser avec votre machine. Vous devrez vous mettre à
-    jour vers Ubuntu 20.10 ou Debian Bullseye.
-  </p>
-
   <h2>Mentions légales</h2>
   <p>
     Le service de diffusion vidéo du Crans est un service d'hébergement

web/template/player.html

@@ -8,9 +8,9 @@
     <div class="controls">
       <span class="control-quality">
         <select id="quality">
-          <option value="source">Source</option>
+          <option value="240p">Source</option>
-          <option value="720p">720p</option>
           <option value="480p">480p</option>
+          <option value="360p">360p</option>
           <option value="240p">240p</option>
         </select>  
       </span>