Send rate updates over RTCP.

client.go

@@ -10,7 +10,6 @@ import (
 	"errors"
 	"io"
 	"log"
-	"math"
 	"os"
 	"strings"
 	"sync"
@@ -98,8 +97,6 @@ type clientMessage struct {
 	Answer      *webrtc.SessionDescription `json:"answer,omitempty"`
 	Candidate   *webrtc.ICECandidateInit   `json:"candidate,omitempty"`
 	Del         bool                       `json:"del,omitempty"`
-	AudioRate   int                        `json:"audiorate,omitempty"`
-	VideoRate   int                        `json:"videorate,omitempty"`
 }
 
 type closeMessage struct {
@@ -284,6 +281,7 @@ func addUpConn(c *client, id string) (*upConnection, error) {
 		u.pairs = append(u.pairs, trackPair{
 			remote:     remote,
 			local:      local,
+			maxBitrate: ^uint64(0),
 		})
 		done := len(u.pairs) >= u.trackCount
 		c.group.mu.Unlock()
@@ -442,12 +440,24 @@ func addDownTrack(c *client, id string, track *webrtc.Track, remote *upConnectio
 		return nil, nil, err
 	}
 
-	go rtcpListener(c.group, conn, s)
+	conn.tracks = append(conn.tracks,
+		downTrack{track.SSRC(), new(timeStampedBitrate)},
+	)
+
+	go rtcpListener(c.group, conn, s,
+		conn.tracks[len(conn.tracks)-1].maxBitrate)
 
 	return conn, s, nil
 }
 
-func rtcpListener(g *group, c *downConnection, s *webrtc.RTPSender) {
+var epoch = time.Now()
+
+func msSinceEpoch() uint64 {
+	return uint64(time.Since(epoch) / time.Millisecond)
+}
+
+func rtcpListener(g *group, c *downConnection, s *webrtc.RTPSender,
+	bitrate *timeStampedBitrate) {
 	for {
 		ps, err := s.ReadRTCP()
 		if err != nil {
@@ -460,16 +470,23 @@ func rtcpListener(g *group, c *downConnection, s *webrtc.RTPSender) {
 		for _, p := range ps {
 			switch p := p.(type) {
 			case *rtcp.PictureLossIndication:
-				err := sendPLI(c.remote, p.MediaSSRC)
+				err := sendPLI(c.remote.pc, p.MediaSSRC)
 				if err != nil {
 					log.Printf("sendPLI: %v", err)
 				}
 			case *rtcp.ReceiverEstimatedMaximumBitrate:
-				bitrate := uint32(math.MaxInt32)
+				ms := msSinceEpoch()
-				if p.Bitrate < math.MaxInt32 {
+				// this is racy -- a reader might read the
-					bitrate = uint32(p.Bitrate)
+				// data between the two writes.  This shouldn't
-				}
+				// matter, we'll recover at the next sample.
-				atomic.StoreUint32(&c.maxBitrate, bitrate)
+				atomic.StoreUint64(
+					&bitrate.bitrate,
+					p.Bitrate,
+				)
+				atomic.StoreUint64(
+					&bitrate.timestamp,
+					uint64(ms),
+				)
 			case *rtcp.ReceiverReport:
 			default:
 				log.Printf("RTCP: %T", p)
@@ -520,42 +537,61 @@ func splitBitrate(bitrate uint32, audio, video bool) (uint32, uint32) {
 	return audioRate, bitrate - audioRate
 }
 
-func updateBitrate(g *group, up *upConnection) (uint32, uint32) {
+func updateUpBitrate(g *group, up *upConnection) {
-	audio := uint32(math.MaxInt32)
+	for i := range up.pairs {
-	video := uint32(math.MaxInt32)
+		up.pairs[i].maxBitrate = ^uint64(0)
+	}
+
+	now := msSinceEpoch()
+
 	g.Range(func(c *client) bool {
 		for _, down := range c.down {
 			if down.remote == up {
-				bitrate := atomic.LoadUint32(&down.maxBitrate)
+				for _, dt := range down.tracks {
+					ms := atomic.LoadUint64(
+						&dt.maxBitrate.timestamp,
+					)
+					bitrate := atomic.LoadUint64(
+						&dt.maxBitrate.bitrate,
+					)
 					if bitrate == 0 {
-					bitrate = 256000
+						continue
-				} else if bitrate < 6000 {
+					}
-					bitrate = 6000
+
+					if now - ms > 5000 {
+						continue
+					}
+
+					for i, p := range up.pairs {
+						if p.local.SSRC() == dt.ssrc {
+							if p.maxBitrate > bitrate {
+								up.pairs[i].maxBitrate = bitrate
+								break
+							}
 						}
-				hasAudio, hasVideo := trackKinds(down)
-				a, v := splitBitrate(bitrate, hasAudio, hasVideo)
-				if a < audio {
-					audio = a
 					}
-				if v < video {
-					video = v
 				}
 			}
 		}
 		return true
 	})
-	up.maxAudioBitrate = audio
-	up.maxVideoBitrate = video
-	return audio, video
 }
 
-func sendPLI(up *upConnection, ssrc uint32) error {
+func sendPLI(pc *webrtc.PeerConnection, ssrc uint32) error {
-	// we use equal SSRC values on both sides
+	return pc.WriteRTCP([]rtcp.Packet{
-	return up.pc.WriteRTCP([]rtcp.Packet{
 		&rtcp.PictureLossIndication{MediaSSRC: ssrc},
 	})
 }
 
+func sendREMB(pc *webrtc.PeerConnection, ssrc uint32, bitrate uint64) error {
+	return pc.WriteRTCP([]rtcp.Packet{
+		&rtcp.ReceiverEstimatedMaximumBitrate{
+			Bitrate: bitrate,
+			SSRCs: []uint32{ssrc},
+		},
+	})
+}
+
 func countMediaStreams(data string) (int, error) {
 	desc := sdp.NewJSEPSessionDescription(false)
 	err := desc.Unmarshal(data)
@@ -709,7 +745,7 @@ func clientLoop(c *client, conn *websocket.Conn) error {
 
 	readTime := time.Now()
 
-	ticker := time.NewTicker(2 * time.Second)
+	ticker := time.NewTicker(time.Second)
 	defer ticker.Stop()
 	slowTicker := time.NewTicker(10 * time.Second)
 	defer slowTicker.Stop()
@@ -888,16 +924,19 @@ func handleClientMessage(c *client, m clientMessage) error {
 
 func sendRateUpdate(c *client) {
 	for _, u := range c.up {
-		oldaudio := u.maxAudioBitrate
+		updateUpBitrate(c.group, u)
-		oldvideo := u.maxVideoBitrate
+		for _, p := range u.pairs {
-		audio, video := updateBitrate(c.group, u)
+			bitrate := p.maxBitrate
-		if audio != oldaudio || video != oldvideo {
+			if bitrate != ^uint64(0) {
-			c.write(clientMessage{
+				if bitrate < 6000 {
-				Type:      "maxbitrate",
+					bitrate = 6000
-				Id:        u.id,
+				}
-				AudioRate: int(audio),
+				err := sendREMB(u.pc, p.remote.SSRC(),
-				VideoRate: int(video),
+					uint64(bitrate))
-			})
+				if err != nil {
+					log.Printf("sendREMB: %v", err)
+				}
+			}
 		}
 	}
 }

group.go

@@ -19,23 +19,31 @@ import (
 
 type trackPair struct {
 	remote, local *webrtc.Track
+	maxBitrate    uint64
 }
 
 type upConnection struct {
 	id         string
 	label      string
 	pc         *webrtc.PeerConnection
-	maxAudioBitrate uint32
-	maxVideoBitrate uint32
 	trackCount int
 	pairs      []trackPair
 }
 
+type timeStampedBitrate struct {
+	bitrate   uint64
+	timestamp uint64
+}
+type downTrack struct {
+	ssrc       uint32
+	maxBitrate *timeStampedBitrate
+}
+
 type downConnection struct {
 	id     string
 	pc     *webrtc.PeerConnection
 	remote *upConnection
-	maxBitrate uint32
+	tracks []downTrack
 }
 
 type client struct {

static/sfu.js

@@ -323,9 +323,6 @@ function serverConnect() {
             case 'ice':
                 gotICE(m.id, m.candidate);
                 break;
-            case 'maxbitrate':
-                setMaxBitrate(m.id, m.audiorate, m.videorate);
-                break;
             case 'label':
                 gotLabel(m.id, m.value);
                 break;
@@ -450,49 +447,6 @@ async function gotICE(id, candidate) {
         conn.iceCandidates.push(candidate)
 }
 
-let maxaudiorate, maxvideorate;
-
-async function setMaxBitrate(id, audio, video) {
-    let conn = up[id];
-    if(!conn)
-        throw new Error("Setting bitrate of unknown id");
-
-    let senders = conn.pc.getSenders();
-    for(let i = 0; i < senders.length; i++) {
-        let s = senders[i];
-        if(!s.track)
-            return;
-        let p = s.getParameters();
-        let bitrate;
-        if(s.track.kind == 'audio')
-            bitrate = audio;
-        else if(s.track.kind == 'video')
-            bitrate = video;
-        for(let j = 0; j < p.encodings.length; j++) {
-            let e = p.encodings[j];
-            if(bitrate)
-                e.maxBitrate = bitrate;
-            else
-                delete(e.maxBitrate);
-            await s.setParameters(p);
-        }
-    }
-
-    if((audio && audio < 128000) || (video && video < 256000)) {
-        let l = '';
-        if(audio)
-            l = `${Math.round(audio/1000)}kbps`
-        if(video) {
-            if(l)
-                l = l + ' + ';
-            l = l + `${Math.round(video/1000)}kbps`
-        }
-        setLabel(id, l)
-    } else {
-        setLabel(id);
-    }
-}
-
 async function addIceCandidates(conn) {
     let promises = []
     conn.iceCandidates.forEach(c => {