WebRTC offers multiple quality

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/webrtc/ingest.go

@@ -14,33 +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)
 
-	// Open a UDP Listener for RTP Packets on port 5004
+	// FIXME Bad code
-	audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
+	port := 5000
-	if err != nil {
+	var tracks map[string][]*webrtc.Track
-		log.Printf("Faited to open UDP listener %s", err)
+	qualityName := ""
-		return
+	switch q.Name {
+	case "audio":
+		port = 5004
+		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: 5005})
+
+	// 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)
+	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
@@ -51,49 +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
 				}
 			}
@@ -105,24 +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) (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", "rtp://127.0.0.1:5004",
+	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", "rtp://127.0.0.1:5005"}
+		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 {
@@ -205,7 +204,7 @@ 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)
 			if err != nil {
 				log.Printf("Failed to get quality '%s'", qualityName)
@@ -217,6 +216,7 @@ func Serve(streams *messaging.Streams, cfg *Options) {
 			go listenSdp(name, quality.WebRtcLocalSdp, quality.WebRtcRemoteSdp, cfg)
 		}
 	}
+}
 
 func listenSdp(name string, localSdp, remoteSdp chan webrtc.SessionDescription, cfg *Options) {
 	// Handle new connections

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/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>