Handle cases of long mute/rollover of time stamp. (#2842)

* Handle cases of long mute/rollover of time stamp.

There are cases where the track is muted for long enough for timestamp
roll over to happen. There are no packets in that window (typically
there should be black frames (for video) or silence (for audio)). But,
maybe the pause based implementation of mute is causing this.

Anyhow, use time since last packet to gauge how much roll over should
have happened and use that to update time stamp. There will be really
edge cases where this could also fail (for e. g. packet time is affected
by propagation delay, so it could theoretically happen that mute/unmute
+ packet reception could happen exactly around that rollover point and
  miscalculate, but should be rare).

As this happen per packet on receive side, changing time to `UnixNano()`
to make it more efficient to check this.

* spelling

* tests

* test util

* tests

pkg/sfu/buffer/buffer.go

@@ -45,20 +45,20 @@ import (
 )
 
 const (
-	ReportDelta = time.Second
+	ReportDelta = 1e9
 
 	InitPacketBufferSizeVideo = 300
 	InitPacketBufferSizeAudio = 70
 )
 
 type pendingPacket struct {
-	arrivalTime time.Time
+	arrivalTime int64
 	packet      []byte
 }
 
 type ExtPacket struct {
 	VideoLayer
-	Arrival              time.Time
+	Arrival              int64
 	ExtSequenceNumber    uint64
 	ExtTimestamp         uint64
 	Packet               *rtp.Packet
@@ -84,7 +84,7 @@ type Buffer struct {
 	closeOnce       sync.Once
 	mediaSSRC       uint32
 	clockRate       uint32
-	lastReport      time.Time
+	lastReport      int64
 	twccExtID       uint8
 	audioLevelExtID uint8
 	bound           bool
@@ -212,7 +212,7 @@ func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapabili
 	b.ppsSnapshotId = b.rtpStats.NewSnapshotId()
 
 	b.clockRate = codec.ClockRate
-	b.lastReport = time.Now()
+	b.lastReport = time.Now().UnixNano()
 	b.mime = strings.ToLower(codec.MimeType)
 	for _, codecParameter := range params.Codecs {
 		if strings.EqualFold(codecParameter.MimeType, codec.MimeType) {
@@ -340,10 +340,10 @@ func (b *Buffer) Write(pkt []byte) (n int, err error) {
 		}
 	}
 
-	now := time.Now()
+	now := time.Now().UnixNano()
 	if b.twcc != nil && b.twccExtID != 0 && !b.closed.Load() {
 		if ext := rtpPacket.GetExtension(b.twccExtID); ext != nil {
-			b.twcc.Push(rtpPacket.SSRC, binary.BigEndian.Uint16(ext[0:2]), now.UnixNano(), rtpPacket.Marker)
+			b.twcc.Push(rtpPacket.SSRC, binary.BigEndian.Uint16(ext[0:2]), now, rtpPacket.Marker)
 		}
 	}
 
@@ -373,7 +373,7 @@ func (b *Buffer) Write(pkt []byte) (n int, err error) {
 	}
 
 	b.payloadType = rtpPacket.PayloadType
-	b.calc(pkt, &rtpPacket, time.Now(), false)
+	b.calc(pkt, &rtpPacket, now, false)
 	b.Unlock()
 	b.readCond.Broadcast()
 	return
@@ -398,7 +398,7 @@ func (b *Buffer) SetPrimaryBufferForRTX(primaryBuffer *Buffer) {
 	}
 }
 
-func (b *Buffer) writeRTX(rtxPkt *rtp.Packet, arrivalTime time.Time) (n int, err error) {
+func (b *Buffer) writeRTX(rtxPkt *rtp.Packet, arrivalTime int64) (n int, err error) {
 	b.Lock()
 	defer b.Unlock()
 	if !b.bound {
@@ -541,7 +541,7 @@ func (b *Buffer) SetRTT(rtt uint32) {
 	}
 }
 
-func (b *Buffer) calc(rawPkt []byte, rtpPacket *rtp.Packet, arrivalTime time.Time, isRTX bool) {
+func (b *Buffer) calc(rawPkt []byte, rtpPacket *rtp.Packet, arrivalTime int64, isRTX bool) {
 	defer func() {
 		b.doNACKs()
 
@@ -740,7 +740,7 @@ func (b *Buffer) doFpsCalc(ep *ExtPacket) {
 	}
 }
 
-func (b *Buffer) updateStreamState(p *rtp.Packet, arrivalTime time.Time) RTPFlowState {
+func (b *Buffer) updateStreamState(p *rtp.Packet, arrivalTime int64) RTPFlowState {
 	flowState := b.rtpStats.Update(
 		arrivalTime,
 		p.Header.SequenceNumber,
@@ -764,7 +764,7 @@ func (b *Buffer) updateStreamState(p *rtp.Packet, arrivalTime time.Time) RTPFlow
 	return flowState
 }
 
-func (b *Buffer) processHeaderExtensions(p *rtp.Packet, arrivalTime time.Time, isRTX bool) {
+func (b *Buffer) processHeaderExtensions(p *rtp.Packet, arrivalTime int64, isRTX bool) {
 	if b.audioLevelExtID != 0 && !isRTX {
 		if !b.latestTSForAudioLevelInitialized {
 			b.latestTSForAudioLevelInitialized = true
@@ -786,7 +786,7 @@ func (b *Buffer) processHeaderExtensions(p *rtp.Packet, arrivalTime time.Time, i
 	}
 }
 
-func (b *Buffer) getExtPacket(rtpPacket *rtp.Packet, arrivalTime time.Time, flowState RTPFlowState) *ExtPacket {
+func (b *Buffer) getExtPacket(rtpPacket *rtp.Packet, arrivalTime int64, flowState RTPFlowState) *ExtPacket {
 	ep := &ExtPacket{
 		Arrival:           arrivalTime,
 		ExtSequenceNumber: flowState.ExtSequenceNumber,
@@ -887,8 +887,8 @@ func (b *Buffer) doNACKs() {
 	}
 }
 
-func (b *Buffer) doReports(arrivalTime time.Time) {
-	if time.Since(b.lastReport) < ReportDelta {
+func (b *Buffer) doReports(arrivalTime int64) {
+	if arrivalTime-b.lastReport < ReportDelta {
 		return
 	}
 
@@ -1090,7 +1090,7 @@ func (b *Buffer) GetAudioLevel() (float64, bool) {
 		return 0, false
 	}
 
-	return b.audioLevel.GetLevel(time.Now())
+	return b.audioLevel.GetLevel(time.Now().UnixNano())
 }
 
 func (b *Buffer) OnFpsChanged(f func()) {