Perform congestion control at the connection level.

REMB applies to the whole transport, not to individual tracks.

conn.go

@@ -34,11 +34,11 @@ type upTrack interface {
 }
 
 type downConnection interface {
+	GetMaxBitrate(now uint64) uint64
 }
 
 type downTrack interface {
 	WriteRTP(packat *rtp.Packet) error
 	Accumulate(bytes uint32)
-	GetMaxBitrate(now uint64) uint64
 	setTimeOffset(ntp uint64, rtp uint32)
 }

disk.go

@@ -382,10 +382,10 @@ func (conn *diskConn) initWriter(width, height uint32) error {
 	return nil
 }
 
+func (down *diskConn) GetMaxBitrate(now uint64) uint64 {
+	return ^uint64(0)
+}
+
 func (t *diskTrack) Accumulate(bytes uint32) {
 	return
 }
-
-func (down *diskTrack) GetMaxBitrate(now uint64) uint64 {
-	return ^uint64(0)
-}

group.go

@@ -37,8 +37,7 @@ type chatHistoryEntry struct {
 }
 
 const (
-	minVideoRate = 200000
+	minBitrate = 200000
-	minAudioRate = 9600
 )
 
 type group struct {
@@ -506,8 +505,9 @@ type clientStats struct {
 }
 
 type connStats struct {
-	id     string
+	id         string
-	tracks []trackStats
+	maxBitrate uint64
+	tracks     []trackStats
 }
 
 type trackStats struct {
@@ -560,7 +560,9 @@ func getClientStats(c *webClient) clientStats {
 	}
 
 	for _, up := range c.up {
-		conns := connStats{id: up.id}
+		conns := connStats{
+			id: up.id,
+		}
 		tracks := up.getTracks()
 		for _, t := range tracks {
 			expected, lost, _, _ := t.cache.GetStats(false)
@@ -572,10 +574,9 @@ func getClientStats(c *webClient) clientStats {
 				(time.Second / time.Duration(t.jitter.HZ()))
 			rate, _ := t.rate.Estimate()
 			conns.tracks = append(conns.tracks, trackStats{
-				bitrate:    uint64(rate) * 8,
+				bitrate: uint64(rate) * 8,
-				maxBitrate: atomic.LoadUint64(&t.maxBitrate),
+				loss:    loss,
-				loss:       loss,
+				jitter:  jitter,
-				jitter:     jitter,
 			})
 		}
 		cs.up = append(cs.up, conns)
@@ -584,10 +585,13 @@ func getClientStats(c *webClient) clientStats {
 		return cs.up[i].id < cs.up[j].id
 	})
 
+	jiffies := rtptime.Jiffies()
 	for _, down := range c.down {
-		conns := connStats{id: down.id}
+		conns := connStats{
+			id:         down.id,
+			maxBitrate: down.GetMaxBitrate(jiffies),
+		}
 		for _, t := range down.tracks {
-			jiffies := rtptime.Jiffies()
 			rate, _ := t.rate.Estimate()
 			rtt := rtptime.ToDuration(atomic.LoadUint64(&t.rtt),
 				rtptime.JiffiesPerSec)
@@ -596,7 +600,7 @@ func getClientStats(c *webClient) clientStats {
 				time.Duration(t.track.Codec().ClockRate)
 			conns.tracks = append(conns.tracks, trackStats{
 				bitrate:    uint64(rate) * 8,
-				maxBitrate: t.GetMaxBitrate(jiffies),
+				maxBitrate: t.maxBitrate.Get(jiffies),
 				loss:       uint8(uint32(loss) * 100 / 256),
 				rtt:        rtt,
 				jitter:     j,

rtpconn.go

@@ -70,17 +70,16 @@ type iceConnection interface {
 }
 
 type rtpDownTrack struct {
-	track          *webrtc.Track
+	track         *webrtc.Track
-	remote         upTrack
+	remote        upTrack
-	maxLossBitrate *bitrate
+	maxBitrate    *bitrate
-	maxREMBBitrate *bitrate
+	rate          *estimator.Estimator
-	rate           *estimator.Estimator
+	stats         *receiverStats
-	stats          *receiverStats
+	srTime        uint64
-	srTime         uint64
+	srNTPTime     uint64
-	srNTPTime      uint64
+	remoteNTPTime uint64
-	remoteNTPTime  uint64
+	remoteRTPTime uint32
-	remoteRTPTime  uint32
+	rtt           uint64
-	rtt            uint64
 }
 
 func (down *rtpDownTrack) WriteRTP(packet *rtp.Packet) error {
@@ -91,26 +90,18 @@ func (down *rtpDownTrack) Accumulate(bytes uint32) {
 	down.rate.Accumulate(bytes)
 }
 
-func (down *rtpDownTrack) GetMaxBitrate(now uint64) uint64 {
-	br1 := down.maxLossBitrate.Get(now)
-	br2 := down.maxREMBBitrate.Get(now)
-	if br1 < br2 {
-		return br1
-	}
-	return br2
-}
-
 func (down *rtpDownTrack) setTimeOffset(ntp uint64, rtp uint32) {
 	atomic.StoreUint64(&down.remoteNTPTime, ntp)
 	atomic.StoreUint32(&down.remoteRTPTime, rtp)
 }
 
 type rtpDownConnection struct {
-	id            string
+	id             string
-	pc            *webrtc.PeerConnection
+	pc             *webrtc.PeerConnection
-	remote        upConnection
+	remote         upConnection
-	tracks        []*rtpDownTrack
+	tracks         []*rtpDownTrack
-	iceCandidates []*webrtc.ICECandidateInit
+	maxREMBBitrate *bitrate
+	iceCandidates  []*webrtc.ICECandidateInit
 }
 
 func newDownConn(id string, remote upConnection) (*rtpDownConnection, error) {
@@ -124,14 +115,35 @@ func newDownConn(id string, remote upConnection) (*rtpDownConnection, error) {
 	})
 
 	conn := &rtpDownConnection{
-		id:     id,
+		id:             id,
-		pc:     pc,
+		pc:             pc,
-		remote: remote,
+		remote:         remote,
+		maxREMBBitrate: new(bitrate),
 	}
 
 	return conn, nil
 }
 
+func (down *rtpDownConnection) GetMaxBitrate(now uint64) uint64 {
+	rate := down.maxREMBBitrate.Get(now)
+	var trackRate uint64
+	for _, t := range down.tracks {
+		r := t.maxBitrate.Get(now)
+		if r == ^uint64(0) {
+			if t.track.Kind() == webrtc.RTPCodecTypeAudio {
+				r = 128 * 1024
+			} else {
+				r = 512 * 1024
+			}
+		}
+		trackRate += r
+	}
+	if trackRate < rate {
+		return trackRate
+	}
+	return rate
+}
+
 func (down *rtpDownConnection) addICECandidate(candidate *webrtc.ICECandidateInit) error {
 	if down.pc.RemoteDescription() != nil {
 		return down.pc.AddICECandidate(*candidate)
@@ -162,15 +174,14 @@ func (down *rtpDownConnection) flushICECandidates() error {
 }
 
 type rtpUpTrack struct {
-	track      *webrtc.Track
+	track    *webrtc.Track
-	label      string
+	label    string
-	rate       *estimator.Estimator
+	rate     *estimator.Estimator
-	cache      *packetcache.Cache
+	cache    *packetcache.Cache
-	jitter     *jitter.Estimator
+	jitter   *jitter.Estimator
-	maxBitrate uint64
+	lastPLI  uint64
-	lastPLI    uint64
+	lastFIR  uint64
-	lastFIR    uint64
+	firSeqno uint32
-	firSeqno   uint32
 
 	localCh    chan localTrackAction
 	writerDone chan struct{}
@@ -422,7 +433,6 @@ func newUpConn(c client, id string) (*rtpUpConnection, error) {
 			cache:      packetcache.New(32),
 			rate:       estimator.New(time.Second),
 			jitter:     jitter.New(remote.Codec().ClockRate),
-			maxBitrate: ^uint64(0),
 			localCh:    make(chan localTrackAction, 2),
 			writerDone: make(chan struct{}),
 		}
@@ -690,15 +700,6 @@ func sendFIR(pc *webrtc.PeerConnection, ssrc uint32, seqno uint8) error {
 	})
 }
 
-func sendREMB(pc *webrtc.PeerConnection, ssrc uint32, bitrate uint64) error {
-	return pc.WriteRTCP([]rtcp.Packet{
-		&rtcp.ReceiverEstimatedMaximumBitrate{
-			Bitrate: bitrate,
-			SSRCs:   []uint32{ssrc},
-		},
-	})
-}
-
 func (up *rtpUpConnection) sendNACK(track *rtpUpTrack, first uint16, bitmap uint16) error {
 	if !track.hasRtcpFb("nack", "") {
 		return nil
@@ -797,7 +798,7 @@ func rtcpUpListener(conn *rtpUpConnection, track *rtpUpTrack, r *webrtc.RTPRecei
 	}
 }
 
-func sendRR(conn *rtpUpConnection) error {
+func sendUpRTCP(conn *rtpUpConnection) error {
 	conn.mu.Lock()
 	defer conn.mu.Unlock()
 
@@ -813,6 +814,7 @@ func sendRR(conn *rtpUpConnection) error {
 
 	reports := make([]rtcp.ReceptionReport, 0, len(conn.tracks))
 	for _, t := range conn.tracks {
+		updateUpTrack(t)
 		expected, lost, totalLost, eseqno := t.cache.GetStats(true)
 		if expected == 0 {
 			expected = 1
@@ -843,17 +845,46 @@ func sendRR(conn *rtpUpConnection) error {
 		})
 	}
 
-	return conn.pc.WriteRTCP([]rtcp.Packet{
+	packets := []rtcp.Packet{
 		&rtcp.ReceiverReport{
 			Reports: reports,
 		},
-	})
+	}
+
+	rate := ^uint64(0)
+	for _, l := range conn.local {
+		r := l.GetMaxBitrate(now)
+		if r < rate {
+			rate = r
+		}
+	}
+	if rate < minBitrate {
+		rate = minBitrate
+	}
+
+	var ssrcs []uint32
+	for _, t := range conn.tracks {
+		if t.hasRtcpFb("goog-remb", "") {
+			continue
+		}
+		ssrcs = append(ssrcs, t.track.SSRC())
+	}
+
+	if len(ssrcs) > 0 {
+		packets = append(packets,
+			&rtcp.ReceiverEstimatedMaximumBitrate{
+				Bitrate: rate,
+				SSRCs:   ssrcs,
+			},
+		)
+	}
+	return conn.pc.WriteRTCP(packets)
 }
 
 func rtcpUpSender(conn *rtpUpConnection) {
 	for {
 		time.Sleep(time.Second)
-		err := sendRR(conn)
+		err := sendUpRTCP(conn)
 		if err != nil {
 			if err == io.EOF || err == io.ErrClosedPipe {
 				return
@@ -936,7 +967,7 @@ const (
 )
 
 func (track *rtpDownTrack) updateRate(loss uint8, now uint64) {
-	rate := track.maxLossBitrate.Get(now)
+	rate := track.maxBitrate.Get(now)
 	if rate < minLossRate || rate > maxLossRate {
 		// no recent feedback, reset
 		rate = initLossRate
@@ -962,7 +993,7 @@ func (track *rtpDownTrack) updateRate(loss uint8, now uint64) {
 	}
 
 	// update unconditionally, to set the timestamp
-	track.maxLossBitrate.Set(rate, now)
+	track.maxBitrate.Set(rate, now)
 }
 
 func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RTPSender) {
@@ -1034,7 +1065,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 					log.Printf("sendFIR: %v", err)
 				}
 			case *rtcp.ReceiverEstimatedMaximumBitrate:
-				track.maxREMBBitrate.Set(p.Bitrate, jiffies)
+				conn.maxREMBBitrate.Set(p.Bitrate, jiffies)
 			case *rtcp.ReceiverReport:
 				for _, r := range p.Reports {
 					if r.SSRC == track.track.SSRC() {
@@ -1048,11 +1079,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
 					}
 				}
 			case *rtcp.TransportLayerNack:
-				maxBitrate := track.GetMaxBitrate(jiffies)
+				sendRecovery(p, track)
-				bitrate, _ := track.rate.Estimate()
-				if uint64(bitrate)*7/8 < maxBitrate {
-					sendRecovery(p, track)
-				}
 			}
 		}
 	}
@@ -1086,37 +1113,18 @@ func handleReport(track *rtpDownTrack, report rtcp.ReceptionReport, jiffies uint
 	}
 }
 
-func updateUpTrack(track *rtpUpTrack) uint64 {
+func updateUpTrack(track *rtpUpTrack) {
 	now := rtptime.Jiffies()
 
-	isvideo := track.track.Kind() == webrtc.RTPCodecTypeVideo
 	clockrate := track.track.Codec().ClockRate
-	minrate := uint64(minAudioRate)
-	rate := ^uint64(0)
-	if isvideo {
-		minrate = minVideoRate
-		rate = ^uint64(0)
-	}
 	local := track.getLocal()
 	var maxrto uint64
 	for _, l := range local {
-		bitrate := l.GetMaxBitrate(now)
-		if bitrate == ^uint64(0) {
-			continue
-		}
-		if bitrate <= minrate {
-			rate = minrate
-			break
-		}
-		if rate > bitrate {
-			rate = bitrate
-		}
 		ll, ok := l.(*rtpDownTrack)
 		if ok {
 			_, j := ll.stats.Get(now)
 			jitter := uint64(j) *
-				(rtptime.JiffiesPerSec /
+				(rtptime.JiffiesPerSec / uint64(clockrate))
-					uint64(clockrate))
 			rtt := atomic.LoadUint64(&ll.rtt)
 			rto := rtt + 4*jitter
 			if rto > maxrto {
@@ -1124,7 +1132,6 @@ func updateUpTrack(track *rtpUpTrack) uint64 {
 			}
 		}
 	}
-	track.maxBitrate = rate
 	_, r := track.rate.Estimate()
 	packets := int((uint64(r) * maxrto * 4) / rtptime.JiffiesPerSec)
 	if packets < 32 {
@@ -1134,6 +1141,4 @@ func updateUpTrack(track *rtpUpTrack) uint64 {
 		packets = 256
 	}
 	track.cache.ResizeCond(packets)
-
-	return rate
 }

webclient.go

@@ -380,12 +380,11 @@ func addDownTrack(c *webClient, conn *rtpDownConnection, remoteTrack upTrack, re
 	}
 
 	track := &rtpDownTrack{
-		track:          local,
+		track:      local,
-		remote:         remoteTrack,
+		remote:     remoteTrack,
-		maxLossBitrate: new(bitrate),
+		maxBitrate: new(bitrate),
-		maxREMBBitrate: new(bitrate),
+		stats:      new(receiverStats),
-		stats:          new(receiverStats),
+		rate:       estimator.New(time.Second),
-		rate:           estimator.New(time.Second),
 	}
 	conn.tracks = append(conn.tracks, track)
 
@@ -692,10 +691,8 @@ func clientLoop(c *webClient, conn *websocket.Conn) error {
 
 	readTime := time.Now()
 
-	ticker := time.NewTicker(time.Second)
+	ticker := time.NewTicker(10 * time.Second)
 	defer ticker.Stop()
-	slowTicker := time.NewTicker(10 * time.Second)
-	defer slowTicker.Stop()
 
 	for {
 		select {
@@ -766,7 +763,7 @@ func clientLoop(c *webClient, conn *websocket.Conn) error {
 					go a.c.pushConn(u.id, u, ts, u.label)
 				}
 			case connectionFailedAction:
-				down := getDownConn(c, a.id);
+				down := getDownConn(c, a.id)
 				if down == nil {
 					log.Printf("Failed indication for " +
 						"unknown connection")
@@ -804,8 +801,6 @@ func clientLoop(c *webClient, conn *websocket.Conn) error {
 				return errors.New("unexpected action")
 			}
 		case <-ticker.C:
-			sendRateUpdate(c)
-		case <-slowTicker.C:
 			if time.Since(readTime) > 90*time.Second {
 				return errors.New("client is dead")
 			}
@@ -1022,27 +1017,6 @@ func handleClientMessage(c *webClient, m clientMessage) error {
 	return nil
 }
 
-func sendRateUpdate(c *webClient) {
-	up := getUpConns(c)
-
-	for _, u := range up {
-		tracks := u.getTracks()
-		for _, t := range tracks {
-			rate := updateUpTrack(t)
-			if !t.hasRtcpFb("goog-remb", "") {
-				continue
-			}
-			if rate == ^uint64(0) {
-				continue
-			}
-			err := sendREMB(u.pc, t.track.SSRC(), rate)
-			if err != nil {
-				log.Printf("sendREMB: %v", err)
-			}
-		}
-	}
-}
-
 func clientReader(conn *websocket.Conn, read chan<- interface{}, done <-chan struct{}) {
 	defer close(read)
 	for {

webserver.go

@@ -171,16 +171,26 @@ func statsHandler(w http.ResponseWriter, r *http.Request) {
 		for _, cs := range gs.clients {
 			fmt.Fprintf(w, "<tr><td>%v</td></tr>\n", cs.id)
 			for _, up := range cs.up {
-				fmt.Fprintf(w, "<tr><td></td><td>Up</td><td>%v</td></tr>\n",
+				fmt.Fprintf(w, "<tr><td></td><td>Up</td><td>%v</td>",
 					up.id)
+				if up.maxBitrate > 0 {
+					fmt.Fprintf(w, "<td>%v</td>",
+						up.maxBitrate)
+				}
+				fmt.Fprintf(w, "</tr>\n")
 				for _, t := range up.tracks {
 					printTrack(w, t)
 				}
 			}
-			for _, up := range cs.down {
+			for _, down := range cs.down {
-				fmt.Fprintf(w, "<tr><td></td><td>Down</td><td> %v</td></tr>\n",
+				fmt.Fprintf(w, "<tr><td></td><td>Down</td><td> %v</td>",
-					up.id)
+					down.id)
-				for _, t := range up.tracks {
+				if down.maxBitrate > 0 {
+					fmt.Fprintf(w, "<td>%v</td>",
+						down.maxBitrate)
+				}
+				fmt.Fprintf(w, "</tr>\n")
+				for _, t := range down.tracks {
 					printTrack(w, t)
 				}
 			}