Implement loss-based congestion control on the down side.

client.go

@@ -591,7 +591,9 @@ func addDownTrack(c *client, id string, remoteTrack *upTrack, remoteConn *upConn
 	track := &downTrack{
 		track:          local,
 		remote:         remoteTrack,
-		maxBitrate: new(timeStampedBitrate),
+		maxLossBitrate: new(bitrate),
+		maxREMBBitrate: new(bitrate),
+		stats:          new(receiverStats),
 		rate:           estimator.New(time.Second),
 	}
 	conn.tracks = append(conn.tracks, track)
@@ -602,6 +604,41 @@ func addDownTrack(c *client, id string, remoteTrack *upTrack, remoteConn *upConn
 	return conn, s, nil
 }
 
+const (
+	minLossRate  = 9600
+	initLossRate = 512 * 1000
+	maxLossRate  = 1 << 30
+)
+
+func (track *downTrack) updateRate(loss uint8, now uint64) {
+	rate := track.maxLossBitrate.Get(now)
+	if rate > maxLossRate {
+		// no recent feedback, reset
+		rate = initLossRate
+	}
+	if loss < 5 {
+		// if our actual rate is low, then we're not probing the
+		// bottleneck
+		actual := 8 * uint64(track.rate.Estimate())
+		if actual >= (rate*7)/8 {
+			// loss < 0.02, multiply by 1.05
+			rate = rate * 269 / 256
+			if rate > maxLossRate {
+				rate = maxLossRate
+			}
+		}
+	} else if loss > 25 {
+		// loss > 0.1, multiply by (1 - loss/2)
+		rate = rate * (512 - uint64(loss)) / 512
+		if rate < minLossRate {
+			rate = minLossRate
+		}
+	}
+
+	// update unconditionally, to set the timestamp
+	track.maxLossBitrate.Set(rate, now)
+}
+
 func rtcpDownListener(g *group, conn *downConnection, track *downTrack, s *webrtc.RTPSender) {
 	for {
 		ps, err := s.ReadRTCP()
@@ -620,23 +657,26 @@ func rtcpDownListener(g *group, conn *downConnection, track *downTrack, s *webrt
 					log.Printf("sendPLI: %v", err)
 				}
 			case *rtcp.ReceiverEstimatedMaximumBitrate:
-				track.maxBitrate.Set(p.Bitrate,
-					mono.Microseconds(),
+				track.maxREMBBitrate.Set(
+					p.Bitrate, mono.Microseconds(),
 				)
 			case *rtcp.ReceiverReport:
 				for _, r := range p.Reports {
 					if r.SSRC == track.track.SSRC() {
-						atomic.StoreUint32(
-							&track.loss,
-							uint32(r.FractionLost),
+						now := mono.Microseconds()
+						track.stats.Set(
+							r.FractionLost,
+							r.Jitter,
+							now,
+						)
+						track.updateRate(
+							r.FractionLost,
+							now,
 						)
-						atomic.StoreUint32(
-							&track.jitter,
-							r.Jitter)
 					}
 				}
 			case *rtcp.TransportLayerNack:
-				maxBitrate := track.maxBitrate.Get(
+				maxBitrate := track.GetMaxBitrate(
 					mono.Microseconds(),
 				)
 				bitrate := track.rate.Estimate()
@@ -675,7 +715,7 @@ func updateUpBitrate(up *upConnection) {
 		track.maxBitrate = ^uint64(0)
 		local := track.getLocal()
 		for _, l := range local {
-			bitrate := l.maxBitrate.Get(now)
+			bitrate := l.GetMaxBitrate(now)
 			if bitrate == ^uint64(0) {
 				continue
 			}

group.go

@@ -72,34 +72,63 @@ type upConnection struct {
 	tracks     []*upTrack
 }
 
-type timeStampedBitrate struct {
+type bitrate struct {
 	bitrate      uint64
 	microseconds uint64
 }
 
-func (tb *timeStampedBitrate) Set(bitrate, us uint64) {
+func (br *bitrate) Set(bitrate uint64, now uint64) {
 	// this is racy -- a reader might read the
 	// data between the two writes.  This shouldn't
 	// matter, we'll recover at the next sample.
-	atomic.StoreUint64(&tb.bitrate, bitrate)
-	atomic.StoreUint64(&tb.microseconds, us)
+	atomic.StoreUint64(&br.bitrate, bitrate)
+	atomic.StoreUint64(&br.microseconds, now)
 }
 
-func (tb *timeStampedBitrate) Get(now uint64) uint64 {
-	ts := atomic.LoadUint64(&tb.microseconds)
+func (br *bitrate) Get(now uint64) uint64 {
+	ts := atomic.LoadUint64(&br.microseconds)
 	if now < ts || now > ts+4000000 {
 		return ^uint64(0)
 	}
-	return atomic.LoadUint64(&tb.bitrate)
+	return atomic.LoadUint64(&br.bitrate)
+}
+
+type receiverStats struct {
+	loss         uint32
+	jitter       uint32
+	microseconds uint64
+}
+
+func (s *receiverStats) Set(loss uint8, jitter uint32, now uint64) {
+	atomic.StoreUint32(&s.loss, uint32(loss))
+	atomic.StoreUint32(&s.jitter, jitter)
+	atomic.StoreUint64(&s.microseconds, now)
+}
+
+func (s *receiverStats) Get(now uint64) (uint8, uint32) {
+	ts := atomic.LoadUint64(&s.microseconds)
+	if now < ts || now > ts+4000000 {
+		return 0, 0
+	}
+	return uint8(atomic.LoadUint32(&s.loss)), atomic.LoadUint32(&s.jitter)
 }
 
 type downTrack struct {
 	track          *webrtc.Track
 	remote         *upTrack
-	maxBitrate *timeStampedBitrate
+	maxLossBitrate *bitrate
+	maxREMBBitrate *bitrate
 	rate           *estimator.Estimator
-	loss       uint32
-	jitter     uint32
+	stats          *receiverStats
+}
+
+func (down *downTrack) GetMaxBitrate(now uint64) uint64 {
+	br1 := down.maxLossBitrate.Get(now)
+	br2 := down.maxREMBBitrate.Get(now)
+	if br1 < br2 {
+		return br1
+	}
+	return br2
 }
 
 type downConnection struct {
@@ -725,15 +754,14 @@ func getClientStats(c *client) clientStats {
 	for _, down := range c.down {
 		conns := connStats{id: down.id}
 		for _, t := range down.tracks {
-			loss := atomic.LoadUint32(&t.loss)
-			jitter := time.Duration(atomic.LoadUint32(&t.jitter)) *
-				time.Second /
+			loss, jitter := t.stats.Get(mono.Microseconds())
+			j := time.Duration(jitter) * time.Second /
 				time.Duration(t.track.Codec().ClockRate)
 			conns.tracks = append(conns.tracks, trackStats{
 				bitrate:    uint64(t.rate.Estimate()) * 8,
-				maxBitrate: t.maxBitrate.Get(mono.Microseconds()),
-				loss:       uint8((loss * 100) / 256),
-				jitter:     jitter,
+				maxBitrate: t.GetMaxBitrate(mono.Microseconds()),
+				loss:       uint8(uint32(loss) * 100 / 256),
+				jitter:     j,
 			})
 		}
 		cs.down = append(cs.down, conns)