livekit/pkg/sfu/buffer/buffer.go

// Copyright 2023 LiveKit, Inc.
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
//     http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.

package buffer

import (
	"encoding/binary"
	"errors"
	"fmt"
	"io"
	"strings"
	"sync"
	"time"

	"github.com/gammazero/deque"
	"github.com/pion/rtcp"
	"github.com/pion/rtp"
	"github.com/pion/rtp/codecs"
	"github.com/pion/sdp/v3"
	"github.com/pion/webrtc/v3"
	"go.uber.org/atomic"

	"github.com/livekit/livekit-server/pkg/sfu/audio"
	act "github.com/livekit/livekit-server/pkg/sfu/rtpextension/abscapturetime"
	dd "github.com/livekit/livekit-server/pkg/sfu/rtpextension/dependencydescriptor"
	"github.com/livekit/livekit-server/pkg/sfu/utils"
	sutils "github.com/livekit/livekit-server/pkg/utils"
	"github.com/livekit/mediatransportutil"
	"github.com/livekit/mediatransportutil/pkg/bucket"
	"github.com/livekit/mediatransportutil/pkg/nack"
	"github.com/livekit/mediatransportutil/pkg/twcc"
	"github.com/livekit/protocol/livekit"
	"github.com/livekit/protocol/logger"
)

const (
	ReportDelta = 1e9

	InitPacketBufferSizeVideo = 300
	InitPacketBufferSizeAudio = 70
)

type pendingPacket struct {
	arrivalTime int64
	packet      []byte
}

type ExtPacket struct {
	VideoLayer
	Arrival              int64
	ExtSequenceNumber    uint64
	ExtTimestamp         uint64
	Packet               *rtp.Packet
	Payload              interface{}
	KeyFrame             bool
	RawPacket            []byte
	DependencyDescriptor *ExtDependencyDescriptor
	AbsCaptureTimeExt    *act.AbsCaptureTime
}

// Buffer contains all packets
type Buffer struct {
	sync.RWMutex
	readCond        *sync.Cond
	bucket          *bucket.Bucket
	nacker          *nack.NackQueue
	maxVideoPkts    int
	maxAudioPkts    int
	codecType       webrtc.RTPCodecType
	payloadType     uint8
	extPackets      deque.Deque[*ExtPacket]
	pPackets        []pendingPacket
	closeOnce       sync.Once
	mediaSSRC       uint32
	clockRate       uint32
	lastReport      int64
	twccExtID       uint8
	audioLevelExtID uint8
	bound           bool
	closed          atomic.Bool
	mime            string

	snRangeMap *utils.RangeMap[uint64, uint64]

	latestTSForAudioLevelInitialized bool
	latestTSForAudioLevel            uint32

	twcc                    *twcc.Responder
	audioLevelParams        audio.AudioLevelParams
	audioLevel              *audio.AudioLevel
	enableAudioLossProxying bool

	lastPacketRead int

	pliThrottle int64

	rtpStats             *RTPStatsReceiver
	rrSnapshotId         uint32
	deltaStatsSnapshotId uint32
	ppsSnapshotId        uint32

	lastFractionLostToReport uint8 // Last fraction lost from subscribers, should report to publisher; Audio only

	// callbacks
	onClose            func()
	onRtcpFeedback     func([]rtcp.Packet)
	onRtcpSenderReport func()
	onFpsChanged       func()
	onFinalRtpStats    func(*livekit.RTPStats)

	// logger
	logger logger.Logger

	// dependency descriptor
	ddExtID  uint8
	ddParser *DependencyDescriptorParser

	paused              bool
	frameRateCalculator [DefaultMaxLayerSpatial + 1]FrameRateCalculator
	frameRateCalculated bool

	packetNotFoundCount   atomic.Uint32
	packetTooOldCount     atomic.Uint32
	extPacketTooMuchCount atomic.Uint32
	invalidPacketCount    atomic.Uint32

	primaryBufferForRTX *Buffer
	rtxPktBuf           []byte

	absCaptureTimeExtID uint8
}

// NewBuffer constructs a new Buffer
func NewBuffer(ssrc uint32, maxVideoPkts, maxAudioPkts int) *Buffer {
	l := logger.GetLogger() // will be reset with correct context via SetLogger
	b := &Buffer{
		mediaSSRC:    ssrc,
		maxVideoPkts: maxVideoPkts,
		maxAudioPkts: maxAudioPkts,
		snRangeMap:   utils.NewRangeMap[uint64, uint64](100),
		pliThrottle:  int64(500 * time.Millisecond),
		logger:       l.WithComponent(sutils.ComponentPub).WithComponent(sutils.ComponentSFU),
	}
	b.readCond = sync.NewCond(&b.RWMutex)
	b.extPackets.SetMinCapacity(7)
	return b
}

func (b *Buffer) SetLogger(logger logger.Logger) {
	b.Lock()
	defer b.Unlock()

	b.logger = logger.WithComponent(sutils.ComponentSFU).WithValues("ssrc", b.mediaSSRC)
	if b.rtpStats != nil {
		b.rtpStats.SetLogger(b.logger)
	}
}

func (b *Buffer) SetPaused(paused bool) {
	b.Lock()
	defer b.Unlock()

	b.paused = paused
}

func (b *Buffer) SetTWCCAndExtID(twcc *twcc.Responder, extID uint8) {
	b.Lock()
	defer b.Unlock()

	b.twcc = twcc
	b.twccExtID = extID
}

func (b *Buffer) SetAudioLevelParams(audioLevelParams audio.AudioLevelParams) {
	b.Lock()
	defer b.Unlock()

	b.audioLevelParams = audioLevelParams
}

func (b *Buffer) SetAudioLossProxying(enable bool) {
	b.Lock()
	defer b.Unlock()

	b.enableAudioLossProxying = enable
}

func (b *Buffer) Bind(params webrtc.RTPParameters, codec webrtc.RTPCodecCapability, bitrates int) {
	b.Lock()
	defer b.Unlock()
	if b.bound {
		return
	}

	b.rtpStats = NewRTPStatsReceiver(RTPStatsParams{
		ClockRate: codec.ClockRate,
		Logger:    b.logger,
	})
	b.rrSnapshotId = b.rtpStats.NewSnapshotId()
	b.deltaStatsSnapshotId = b.rtpStats.NewSnapshotId()
	b.ppsSnapshotId = b.rtpStats.NewSnapshotId()

	b.clockRate = codec.ClockRate
	b.lastReport = time.Now().UnixNano()
	b.mime = strings.ToLower(codec.MimeType)
	for _, codecParameter := range params.Codecs {
		if strings.EqualFold(codecParameter.MimeType, codec.MimeType) {
			b.payloadType = uint8(codecParameter.PayloadType)
			break
		}
	}

	if b.payloadType == 0 {
		b.logger.Warnw("could not find payload type for codec", nil, "codec", codec.MimeType, "parameters", params)
		b.payloadType = uint8(params.Codecs[0].PayloadType)
	}

	for _, ext := range params.HeaderExtensions {
		switch ext.URI {
		case dd.ExtensionURI:
			if IsSvcCodec(codec.MimeType) {
				b.ddExtID = uint8(ext.ID)
				frc := NewFrameRateCalculatorDD(b.clockRate, b.logger)
				for i := range b.frameRateCalculator {
					b.frameRateCalculator[i] = frc.GetFrameRateCalculatorForSpatial(int32(i))
				}
				b.ddParser = NewDependencyDescriptorParser(b.ddExtID, b.logger, func(spatial, temporal int32) {
					frc.SetMaxLayer(spatial, temporal)
				})
			}

		case sdp.AudioLevelURI:
			b.audioLevelExtID = uint8(ext.ID)
			b.audioLevel = audio.NewAudioLevel(b.audioLevelParams)

		case act.AbsCaptureTimeURI:
			b.absCaptureTimeExtID = uint8(ext.ID)
		}
	}

	switch {
	case strings.HasPrefix(b.mime, "audio/"):
		b.codecType = webrtc.RTPCodecTypeAudio
		b.bucket = bucket.NewBucket(InitPacketBufferSizeAudio)
	case strings.HasPrefix(b.mime, "video/"):
		b.codecType = webrtc.RTPCodecTypeVideo
		b.bucket = bucket.NewBucket(InitPacketBufferSizeVideo)
		if b.frameRateCalculator[0] == nil {
			if strings.EqualFold(codec.MimeType, webrtc.MimeTypeVP8) {
				b.frameRateCalculator[0] = NewFrameRateCalculatorVP8(b.clockRate, b.logger)
			}

			if strings.EqualFold(codec.MimeType, webrtc.MimeTypeVP9) {
				frc := NewFrameRateCalculatorVP9(b.clockRate, b.logger)
				for i := range b.frameRateCalculator {
					b.frameRateCalculator[i] = frc.GetFrameRateCalculatorForSpatial(int32(i))
				}
			}
		}
		if bitrates > 0 {
			pps := bitrates / 8 / 1200
			for pps > b.bucket.Capacity() {
				if b.bucket.Grow() >= b.maxVideoPkts {
					break
				}
			}
		}

	default:
		b.codecType = webrtc.RTPCodecType(0)
	}

	for _, fb := range codec.RTCPFeedback {
		switch fb.Type {
		case webrtc.TypeRTCPFBGoogREMB:
			b.logger.Debugw("Setting feedback", "type", webrtc.TypeRTCPFBGoogREMB)
			b.logger.Debugw("REMB not supported, RTCP feedback will not be generated")
		case webrtc.TypeRTCPFBNACK:
			// pion use a single mediaengine to manage negotiated codecs of peerconnection, that means we can't have different
			// codec settings at track level for same codec type, so enable nack for all audio receivers but don't create nack queue
			// for red codec.
			if strings.EqualFold(b.mime, "audio/red") {
				break
			}
			b.logger.Debugw("Setting feedback", "type", webrtc.TypeRTCPFBNACK)
			b.nacker = nack.NewNACKQueue(nack.NackQueueParamsDefault)
		}
	}

	for _, pp := range b.pPackets {
		b.calc(pp.packet, nil, pp.arrivalTime, false)
	}
	b.pPackets = nil
	b.bound = true
}

// Write adds an RTP Packet, ordering is not guaranteed, newer packets may arrive later
func (b *Buffer) Write(pkt []byte) (n int, err error) {
	var rtpPacket rtp.Packet
	err = rtpPacket.Unmarshal(pkt)
	if err != nil {
		return
	}

	b.Lock()
	if b.closed.Load() {
		b.Unlock()
		err = io.EOF
		return
	}

	if err = utils.ValidateRTPPacket(&rtpPacket, b.payloadType, b.mediaSSRC); err != nil {
		invalidPacketCount := b.invalidPacketCount.Inc()
		if (invalidPacketCount-1)%100 == 0 {
			b.logger.Warnw(
				"validating RTP packet failed", err,
				"version", rtpPacket.Version,
				"padding", rtpPacket.Padding,
				"marker", rtpPacket.Marker,
				"expectedPayloadType", b.payloadType,
				"payloadType", rtpPacket.PayloadType,
				"sequenceNumber", rtpPacket.SequenceNumber,
				"timestamp", rtpPacket.Timestamp,
				"expectedSSRC", b.mediaSSRC,
				"ssrc", rtpPacket.SSRC,
				"numExtensions", len(rtpPacket.Extensions),
				"payloadSize", len(rtpPacket.Payload),
				"rtpStats", b.rtpStats,
				"snRangeMap", b.snRangeMap,
			)
		}
	}

	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, rtpPacket.Marker)
		}
	}

	// libwebrtc will use 0 ssrc for probing, don't push the packet to pending queue to avoid memory increasing since
	// the Bind will not be called to consume the pending packets. More details in https://github.com/pion/webrtc/pull/2816
	if rtpPacket.SSRC == 0 {
		b.Unlock()
		return
	}

	// handle RTX packet
	if pb := b.primaryBufferForRTX; pb != nil {
		b.Unlock()

		// skip padding only packets
		if rtpPacket.Padding && len(rtpPacket.Payload) == 0 {
			return
		}

		pb.writeRTX(&rtpPacket, now)
		return
	}

	if !b.bound {
		packet := make([]byte, len(pkt))
		copy(packet, pkt)
		b.pPackets = append(b.pPackets, pendingPacket{
			packet:      packet,
			arrivalTime: now,
		})
		b.Unlock()
		b.readCond.Broadcast()
		return
	}

	b.payloadType = rtpPacket.PayloadType
	b.calc(pkt, &rtpPacket, now, false)
	b.Unlock()
	b.readCond.Broadcast()
	return
}

func (b *Buffer) SetPrimaryBufferForRTX(primaryBuffer *Buffer) {
	b.Lock()
	b.primaryBufferForRTX = primaryBuffer
	pkts := b.pPackets
	b.pPackets = nil
	b.Unlock()
	for _, pp := range pkts {
		var rtpPacket rtp.Packet
		err := rtpPacket.Unmarshal(pp.packet)
		if err != nil {
			continue
		}
		if rtpPacket.Padding && len(rtpPacket.Payload) == 0 {
			continue
		}
		primaryBuffer.writeRTX(&rtpPacket, pp.arrivalTime)
	}
}

func (b *Buffer) writeRTX(rtxPkt *rtp.Packet, arrivalTime int64) (n int, err error) {
	b.Lock()
	defer b.Unlock()
	if !b.bound {
		return
	}

	if b.rtxPktBuf == nil {
		b.rtxPktBuf = make([]byte, bucket.MaxPktSize)
	}

	repairedPkt := *rtxPkt
	repairedPkt.PayloadType = b.payloadType
	repairedPkt.SequenceNumber = binary.BigEndian.Uint16(rtxPkt.Payload[:2])
	repairedPkt.SSRC = b.mediaSSRC
	repairedPkt.Payload = rtxPkt.Payload[2:]
	n, err = repairedPkt.MarshalTo(b.rtxPktBuf)
	if err != nil {
		b.logger.Errorw("could not marshal repaired packet", err, "ssrc", b.mediaSSRC, "sn", repairedPkt.SequenceNumber)
		return
	}

	b.calc(b.rtxPktBuf[:n], &repairedPkt, arrivalTime, true)
	return
}

func (b *Buffer) Read(buff []byte) (n int, err error) {
	b.Lock()
	for {
		if b.closed.Load() {
			b.Unlock()
			return 0, io.EOF
		}
		if b.pPackets != nil && len(b.pPackets) > b.lastPacketRead {
			if len(buff) < len(b.pPackets[b.lastPacketRead].packet) {
				b.Unlock()
				return 0, bucket.ErrBufferTooSmall
			}

			n = copy(buff, b.pPackets[b.lastPacketRead].packet)
			b.lastPacketRead++
			b.Unlock()
			return
		}
		b.readCond.Wait()
	}
}

func (b *Buffer) ReadExtended(buf []byte) (*ExtPacket, error) {
	b.Lock()
	for {
		if b.closed.Load() {
			b.Unlock()
			return nil, io.EOF
		}
		if b.extPackets.Len() > 0 {
			ep := b.extPackets.PopFront()
			ep = b.patchExtPacket(ep, buf)
			if ep == nil {
				continue
			}

			b.Unlock()
			return ep, nil
		}
		b.readCond.Wait()
	}
}

func (b *Buffer) Close() error {
	b.closeOnce.Do(func() {
		b.closed.Store(true)

		if b.rtpStats != nil {
			b.rtpStats.Stop()
			b.logger.Debugw("rtp stats",
				"direction", "upstream",
				"stats", b.rtpStats,
			)
			if cb := b.getOnFinalRtpStats(); cb != nil {
				cb(b.rtpStats.ToProto())
			}
		}

		b.readCond.Broadcast()
		if cb := b.getOnClose(); cb != nil {
			cb()
		}
	})
	return nil
}

func (b *Buffer) OnClose(fn func()) {
	b.Lock()
	b.onClose = fn
	b.Unlock()
}

func (b *Buffer) getOnClose() func() {
	b.RLock()
	defer b.RUnlock()

	return b.onClose
}

func (b *Buffer) SetPLIThrottle(duration int64) {
	b.Lock()
	defer b.Unlock()

	b.pliThrottle = duration
}

func (b *Buffer) SendPLI(force bool) {
	b.RLock()
	rtpStats := b.rtpStats
	pliThrottle := b.pliThrottle
	b.RUnlock()

	if (rtpStats == nil && !force) || !rtpStats.CheckAndUpdatePli(pliThrottle, force) {
		return
	}

	b.logger.Debugw("send pli", "ssrc", b.mediaSSRC, "force", force)
	pli := []rtcp.Packet{
		&rtcp.PictureLossIndication{SenderSSRC: b.mediaSSRC, MediaSSRC: b.mediaSSRC},
	}

	if cb := b.getOnRtcpFeedback(); cb != nil {
		cb(pli)
	}
}

func (b *Buffer) SetRTT(rtt uint32) {
	b.Lock()
	defer b.Unlock()

	if rtt == 0 {
		return
	}

	if b.nacker != nil {
		b.nacker.SetRTT(rtt)
	}

	if b.rtpStats != nil {
		b.rtpStats.UpdateRtt(rtt)
	}
}

func (b *Buffer) calc(rawPkt []byte, rtpPacket *rtp.Packet, arrivalTime int64, isRTX bool) {
	defer func() {
		b.doNACKs()

		b.doReports(arrivalTime)
	}()

	if rtpPacket == nil {
		rtpPacket = &rtp.Packet{}
		if err := rtpPacket.Unmarshal(rawPkt); err != nil {
			b.logger.Errorw("could not unmarshal RTP packet", err)
			return
		}
	}

	// process header extensions always as padding packets could be used for probing
	b.processHeaderExtensions(rtpPacket, arrivalTime, isRTX)

	flowState := b.updateStreamState(rtpPacket, arrivalTime)
	if flowState.IsNotHandled {
		return
	}

	if len(rtpPacket.Payload) == 0 && (!flowState.IsOutOfOrder || flowState.IsDuplicate) {
		// drop padding only in-order or duplicate packet
		if !flowState.IsOutOfOrder {
			// in-order packet - increment sequence number offset for subsequent packets
			// Example:
			//   40 - regular packet - pass through as sequence number 40
			//   41 - missing packet - don't know what it is, could be padding or not
			//   42 - padding only packet - in-order - drop - increment sequence number offset to 1 -
			//        range[0, 42] = 0 offset
			//   41 - arrives out of order - get offset 0 from cache - passed through as sequence number 41
			//   43 - regular packet - offset = 1 (running offset) - passes through as sequence number 42
			//   44 - padding only - in order - drop - increment sequence number offset to 2
			//        range[0, 42] = 0 offset, range[43, 44] = 1 offset
			//   43 - regular packet - out of order + duplicate - offset = 1 from cache -
			//        adjusted sequence number is 42, will be dropped by RTX buffer AddPacket method as duplicate
			//   45 - regular packet - offset = 2 (running offset) - passed through with adjusted sequence number as 43
			//   44 - padding only - out-of-order + duplicate - dropped as duplicate
			//
			if err := b.snRangeMap.ExcludeRange(flowState.ExtSequenceNumber, flowState.ExtSequenceNumber+1); err != nil {
				b.logger.Errorw(
					"could not exclude range", err,
					"sn", rtpPacket.SequenceNumber,
					"esn", flowState.ExtSequenceNumber,
					"rtpStats", b.rtpStats,
					"snRangeMap", b.snRangeMap,
				)
			}
		}
		return
	}

	// add to RTX buffer using sequence number after accounting for dropped padding only packets
	snAdjustment, err := b.snRangeMap.GetValue(flowState.ExtSequenceNumber)
	if err != nil {
		b.logger.Errorw(
			"could not get sequence number adjustment", err,
			"sn", rtpPacket.SequenceNumber,
			"esn", flowState.ExtSequenceNumber,
			"payloadSize", len(rtpPacket.Payload),
			"rtpStats", b.rtpStats,
			"snRangeMap", b.snRangeMap,
		)
		return
	}
	flowState.ExtSequenceNumber -= snAdjustment
	rtpPacket.Header.SequenceNumber = uint16(flowState.ExtSequenceNumber)
	_, err = b.bucket.AddPacketWithSequenceNumber(rawPkt, rtpPacket.Header.SequenceNumber)
	if err != nil {
		if !flowState.IsDuplicate {
			if errors.Is(err, bucket.ErrPacketTooOld) {
				packetTooOldCount := b.packetTooOldCount.Inc()
				if (packetTooOldCount-1)%100 == 0 {
					b.logger.Warnw(
						"could not add packet to bucket", err,
						"count", packetTooOldCount,
						"flowState", &flowState,
						"snAdjustment", snAdjustment,
						"incomingSequenceNumber", flowState.ExtSequenceNumber+snAdjustment,
						"rtpStats", b.rtpStats,
						"snRangeMap", b.snRangeMap,
					)
				}
			} else if err != bucket.ErrRTXPacket {
				b.logger.Warnw(
					"could not add packet to bucket", err,
					"flowState", &flowState,
					"snAdjustment", snAdjustment,
					"incomingSequenceNumber", flowState.ExtSequenceNumber+snAdjustment,
					"rtpStats", b.rtpStats,
					"snRangeMap", b.snRangeMap,
				)
			}
		}
		return
	}

	ep := b.getExtPacket(rtpPacket, arrivalTime, flowState)
	if ep == nil {
		return
	}
	b.extPackets.PushBack(ep)

	if b.extPackets.Len() > b.bucket.Capacity() {
		if (b.extPacketTooMuchCount.Inc()-1)%100 == 0 {
			b.logger.Warnw("too much ext packets", nil, "count", b.extPackets.Len())
		}
	}

	b.doFpsCalc(ep)
}

func (b *Buffer) patchExtPacket(ep *ExtPacket, buf []byte) *ExtPacket {
	n, err := b.getPacket(buf, ep.Packet.SequenceNumber)
	if err != nil {
		packetNotFoundCount := b.packetNotFoundCount.Inc()
		if (packetNotFoundCount-1)%20 == 0 {
			b.logger.Warnw(
				"could not get packet from bucket", err,
				"sn", ep.Packet.SequenceNumber,
				"headSN", b.bucket.HeadSequenceNumber(),
				"count", packetNotFoundCount,
				"rtpStats", b.rtpStats,
				"snRangeMap", b.snRangeMap,
			)
		}
		return nil
	}
	ep.RawPacket = buf[:n]

	// patch RTP packet to point payload to new buffer
	pkt := *ep.Packet
	payloadStart := ep.Packet.Header.MarshalSize()
	payloadEnd := payloadStart + len(ep.Packet.Payload)
	if payloadEnd > n {
		b.logger.Warnw("unexpected marshal size", nil, "max", n, "need", payloadEnd)
		return nil
	}
	// TODO-REMOVE-AFTER-DEBUG START
	if payloadEnd != n {
		paddingEnd := payloadStart + int(ep.Packet.PaddingSize)
		if paddingEnd != n {
			b.logger.Warnw("unexpected marshal size", nil, "max", n, "payloadEnd", payloadEnd, "paddingEnd", paddingEnd)
		}
	}
	// check a few fields for validity
	checkVersion := (buf[0] & 0xc0) >> 6
	checkPayloadType := buf[1] & 0x7f
	checkSequenceNumber := binary.BigEndian.Uint16(buf[2:])
	checkSSRC := binary.BigEndian.Uint32(buf[8:])
	if checkVersion != pkt.Version || checkPayloadType != pkt.PayloadType || checkSequenceNumber != pkt.SequenceNumber || checkSSRC != pkt.SSRC {
		b.logger.Warnw(
			"rtp packet mismatch", nil,
			"version", fmt.Sprintf("%d != %d", checkVersion, pkt.Version),
			"payloadType", fmt.Sprintf("%d != %d", checkPayloadType, pkt.PayloadType),
			"sequenceNumber", fmt.Sprintf("%d != %d", checkSequenceNumber, pkt.SequenceNumber),
			"SSRC", fmt.Sprintf("%d != %d", checkSSRC, pkt.SSRC),
			"bytes", buf[0:16],
			"len", n,
			"headerSize", payloadStart,
			"payloadSize", payloadEnd-payloadStart,
		)
	}
	// TODO-REMOVE-AFTER-DEBUG END
	pkt.Payload = buf[payloadStart:payloadEnd]
	ep.Packet = &pkt

	return ep
}

func (b *Buffer) doFpsCalc(ep *ExtPacket) {
	if b.paused || b.frameRateCalculated || len(ep.Packet.Payload) == 0 {
		return
	}
	spatial := ep.Spatial
	if spatial < 0 || int(spatial) >= len(b.frameRateCalculator) {
		spatial = 0
	}
	if fr := b.frameRateCalculator[spatial]; fr != nil {
		if fr.RecvPacket(ep) {
			complete := true
			for _, fr2 := range b.frameRateCalculator {
				if fr2 != nil && !fr2.Completed() {
					complete = false
					break
				}
			}
			if complete {
				b.frameRateCalculated = true
				if f := b.onFpsChanged; f != nil {
					go f()
				}
			}
		}
	}
}

func (b *Buffer) updateStreamState(p *rtp.Packet, arrivalTime int64) RTPFlowState {
	flowState := b.rtpStats.Update(
		arrivalTime,
		p.Header.SequenceNumber,
		p.Header.Timestamp,
		p.Header.Marker,
		p.Header.MarshalSize(),
		len(p.Payload),
		int(p.PaddingSize),
	)

	if b.nacker != nil {
		b.nacker.Remove(p.SequenceNumber)

		if flowState.HasLoss {
			for lost := flowState.LossStartInclusive; lost != flowState.LossEndExclusive; lost++ {
				b.nacker.Push(uint16(lost))
			}
		}
	}

	return flowState
}

func (b *Buffer) processHeaderExtensions(p *rtp.Packet, arrivalTime int64, isRTX bool) {
	if b.audioLevelExtID != 0 && !isRTX {
		if !b.latestTSForAudioLevelInitialized {
			b.latestTSForAudioLevelInitialized = true
			b.latestTSForAudioLevel = p.Timestamp
		}
		if e := p.GetExtension(b.audioLevelExtID); e != nil {
			ext := rtp.AudioLevelExtension{}
			if err := ext.Unmarshal(e); err == nil {
				if (p.Timestamp - b.latestTSForAudioLevel) < (1 << 31) {
					duration := (int64(p.Timestamp) - int64(b.latestTSForAudioLevel)) * 1e3 / int64(b.clockRate)
					if duration > 0 {
						b.audioLevel.Observe(ext.Level, uint32(duration), arrivalTime)
					}

					b.latestTSForAudioLevel = p.Timestamp
				}
			}
		}
	}
}

func (b *Buffer) getExtPacket(rtpPacket *rtp.Packet, arrivalTime int64, flowState RTPFlowState) *ExtPacket {
	ep := &ExtPacket{
		Arrival:           arrivalTime,
		ExtSequenceNumber: flowState.ExtSequenceNumber,
		ExtTimestamp:      flowState.ExtTimestamp,
		Packet:            rtpPacket,
		VideoLayer: VideoLayer{
			Spatial:  InvalidLayerSpatial,
			Temporal: InvalidLayerTemporal,
		},
	}

	if len(rtpPacket.Payload) == 0 {
		// padding only packet, nothing else to do
		return ep
	}

	ep.Temporal = 0
	if b.ddParser != nil {
		ddVal, videoLayer, err := b.ddParser.Parse(ep.Packet)
		if err != nil {
			return nil
		} else if ddVal != nil {
			ep.DependencyDescriptor = ddVal
			ep.VideoLayer = videoLayer
			// DD-TODO : notify active decode target change if changed.
		}
	}
	switch b.mime {
	case "video/vp8":
		vp8Packet := VP8{}
		if err := vp8Packet.Unmarshal(rtpPacket.Payload); err != nil {
			b.logger.Warnw("could not unmarshal VP8 packet", err)
			return nil
		}
		ep.KeyFrame = vp8Packet.IsKeyFrame
		if ep.DependencyDescriptor == nil {
			ep.Temporal = int32(vp8Packet.TID)
		} else {
			// vp8 with DependencyDescriptor enabled, use the TID from the descriptor
			vp8Packet.TID = uint8(ep.Temporal)
		}
		ep.Payload = vp8Packet
		ep.Spatial = InvalidLayerSpatial // vp8 don't have spatial scalability, reset to invalid

	case "video/vp9":
		if ep.DependencyDescriptor == nil {
			var vp9Packet codecs.VP9Packet
			_, err := vp9Packet.Unmarshal(rtpPacket.Payload)
			if err != nil {
				b.logger.Warnw("could not unmarshal VP9 packet", err)
				return nil
			}
			ep.VideoLayer = VideoLayer{
				Spatial:  int32(vp9Packet.SID),
				Temporal: int32(vp9Packet.TID),
			}
			ep.Payload = vp9Packet
		}
		ep.KeyFrame = IsVP9KeyFrame(rtpPacket.Payload)

	case "video/h264":
		ep.KeyFrame = IsH264KeyFrame(rtpPacket.Payload)
		ep.Spatial = InvalidLayerSpatial // h.264 don't have spatial scalability, reset to invalid

	case "video/av1":
		ep.KeyFrame = IsAV1KeyFrame(rtpPacket.Payload)
	}

	if ep.KeyFrame {
		if b.rtpStats != nil {
			b.rtpStats.UpdateKeyFrame(1)
		}
	}

	if b.absCaptureTimeExtID != 0 {
		extData := rtpPacket.GetExtension(b.absCaptureTimeExtID)

		var actExt act.AbsCaptureTime
		if err := actExt.Unmarshal(extData); err == nil {
			ep.AbsCaptureTimeExt = &actExt
		}
	}

	return ep
}

func (b *Buffer) doNACKs() {
	if b.nacker == nil {
		return
	}

	if r, numSeqNumsNacked := b.buildNACKPacket(); r != nil {
		if cb := b.onRtcpFeedback; cb != nil {
			cb(r)
		}
		if b.rtpStats != nil {
			b.rtpStats.UpdateNack(uint32(numSeqNumsNacked))
		}
	}
}

func (b *Buffer) doReports(arrivalTime int64) {
	if arrivalTime-b.lastReport < ReportDelta {
		return
	}

	b.lastReport = arrivalTime

	// RTCP reports
	pkts := b.getRTCP()
	if pkts != nil {
		if cb := b.onRtcpFeedback; cb != nil {
			cb(pkts)
		}
	}

	b.mayGrowBucket()
}

func (b *Buffer) mayGrowBucket() {
	cap := b.bucket.Capacity()
	maxPkts := b.maxVideoPkts
	if b.codecType == webrtc.RTPCodecTypeAudio {
		maxPkts = b.maxAudioPkts
	}
	if cap >= maxPkts {
		return
	}
	oldCap := cap
	if deltaInfo := b.rtpStats.DeltaInfo(b.ppsSnapshotId); deltaInfo != nil {
		duration := deltaInfo.EndTime.Sub(deltaInfo.StartTime)
		if duration > 500*time.Millisecond {
			pps := int(time.Duration(deltaInfo.Packets) * time.Second / duration)
			for pps > cap && cap < maxPkts {
				cap = b.bucket.Grow()
			}
			if cap > oldCap {
				b.logger.Debugw("grow bucket", "from", oldCap, "to", cap, "pps", pps)
			}
		}
	}
}

func (b *Buffer) buildNACKPacket() ([]rtcp.Packet, int) {
	if nacks, numSeqNumsNacked := b.nacker.Pairs(); len(nacks) > 0 {
		pkts := []rtcp.Packet{&rtcp.TransportLayerNack{
			SenderSSRC: b.mediaSSRC,
			MediaSSRC:  b.mediaSSRC,
			Nacks:      nacks,
		}}
		return pkts, numSeqNumsNacked
	}
	return nil, 0
}

func (b *Buffer) buildReceptionReport() *rtcp.ReceptionReport {
	if b.rtpStats == nil {
		return nil
	}

	proxyLoss := b.lastFractionLostToReport
	if b.codecType == webrtc.RTPCodecTypeAudio && !b.enableAudioLossProxying {
		proxyLoss = 0
	}

	return b.rtpStats.GetRtcpReceptionReport(b.mediaSSRC, proxyLoss, b.rrSnapshotId)
}

func (b *Buffer) SetSenderReportData(rtpTime uint32, ntpTime uint64, packets uint32, octets uint32) {
	b.RLock()
	srData := &RTCPSenderReportData{
		RTPTimestamp: rtpTime,
		NTPTimestamp: mediatransportutil.NtpTime(ntpTime),
		At:           time.Now(),
		Packets:      packets,
		Octets:       octets,
	}

	didSet := false
	if b.rtpStats != nil {
		didSet = b.rtpStats.SetRtcpSenderReportData(srData)
	}
	b.RUnlock()

	if didSet {
		if cb := b.getOnRtcpSenderReport(); cb != nil {
			cb()
		}
	}
}

func (b *Buffer) GetSenderReportData() *RTCPSenderReportData {
	b.RLock()
	defer b.RUnlock()

	if b.rtpStats != nil {
		return b.rtpStats.GetRtcpSenderReportData()
	}

	return nil
}

func (b *Buffer) SetLastFractionLostReport(lost uint8) {
	b.Lock()
	defer b.Unlock()

	b.lastFractionLostToReport = lost
}

func (b *Buffer) getRTCP() []rtcp.Packet {
	var pkts []rtcp.Packet

	rr := b.buildReceptionReport()
	if rr != nil {
		pkts = append(pkts, &rtcp.ReceiverReport{
			SSRC:    b.mediaSSRC,
			Reports: []rtcp.ReceptionReport{*rr},
		})
	}

	return pkts
}

func (b *Buffer) GetPacket(buff []byte, sn uint16) (int, error) {
	b.Lock()
	defer b.Unlock()

	return b.getPacket(buff, sn)
}

func (b *Buffer) getPacket(buff []byte, sn uint16) (int, error) {
	if b.closed.Load() {
		return 0, io.EOF
	}
	return b.bucket.GetPacket(buff, sn)
}

func (b *Buffer) OnRtcpFeedback(fn func(fb []rtcp.Packet)) {
	b.Lock()
	b.onRtcpFeedback = fn
	b.Unlock()
}

func (b *Buffer) getOnRtcpFeedback() func(fb []rtcp.Packet) {
	b.RLock()
	defer b.RUnlock()

	return b.onRtcpFeedback
}

func (b *Buffer) OnRtcpSenderReport(fn func()) {
	b.Lock()
	b.onRtcpSenderReport = fn
	b.Unlock()
}

func (b *Buffer) getOnRtcpSenderReport() func() {
	b.RLock()
	defer b.RUnlock()

	return b.onRtcpSenderReport
}

func (b *Buffer) OnFinalRtpStats(fn func(*livekit.RTPStats)) {
	b.Lock()
	b.onFinalRtpStats = fn
	b.Unlock()
}

func (b *Buffer) getOnFinalRtpStats() func(*livekit.RTPStats) {
	b.RLock()
	defer b.RUnlock()

	return b.onFinalRtpStats
}

// GetMediaSSRC returns the associated SSRC of the RTP stream
func (b *Buffer) GetMediaSSRC() uint32 {
	return b.mediaSSRC
}

// GetClockRate returns the RTP clock rate
func (b *Buffer) GetClockRate() uint32 {
	return b.clockRate
}

func (b *Buffer) GetStats() *livekit.RTPStats {
	b.RLock()
	defer b.RUnlock()

	if b.rtpStats == nil {
		return nil
	}

	return b.rtpStats.ToProto()
}

func (b *Buffer) GetDeltaStats() *StreamStatsWithLayers {
	b.RLock()
	defer b.RUnlock()

	if b.rtpStats == nil {
		return nil
	}

	deltaStats := b.rtpStats.DeltaInfo(b.deltaStatsSnapshotId)
	if deltaStats == nil {
		return nil
	}

	return &StreamStatsWithLayers{
		RTPStats: deltaStats,
		Layers: map[int32]*RTPDeltaInfo{
			0: deltaStats,
		},
	}
}

func (b *Buffer) GetLastSenderReportTime() time.Time {
	b.RLock()
	defer b.RUnlock()

	if b.rtpStats == nil {
		return time.Time{}
	}

	return b.rtpStats.LastSenderReportTime()
}

func (b *Buffer) GetAudioLevel() (float64, bool) {
	b.RLock()
	defer b.RUnlock()

	if b.audioLevel == nil {
		return 0, false
	}

	return b.audioLevel.GetLevel(time.Now().UnixNano())
}

func (b *Buffer) OnFpsChanged(f func()) {
	b.Lock()
	b.onFpsChanged = f
	b.Unlock()
}

func (b *Buffer) GetTemporalLayerFpsForSpatial(layer int32) []float32 {
	if int(layer) >= len(b.frameRateCalculator) {
		return nil
	}

	if fc := b.frameRateCalculator[layer]; fc != nil {
		return fc.GetFrameRate()
	}
	return nil
}

// SVC-TODO: Have to use more conditions to differentiate between
// SVC-TODO: SVC and non-SVC (could be single layer or simulcast).
// SVC-TODO: May only need to differentiate between simulcast and non-simulcast
// SVC-TODO: i. e. may be possible to treat single layer as SVC to get proper/intended functionality.
func IsSvcCodec(mime string) bool {
	switch strings.ToLower(mime) {
	case "video/av1":
		fallthrough
	case "video/vp9":
		return true
	}
	return false
}

func IsRedCodec(mime string) bool {
	return strings.HasSuffix(strings.ToLower(mime), "red")
}