Files
livekit/pkg/sfu/buffer/buffer.go
T
Raja SubramanianandClaude Fable 5.1 9b20c1da57 sfu/buffer: stop allocating per ingested packet (#4951)
* sfu/buffer: stop allocating per ingested packet

Buffer.Write allocated six objects for every RTP packet. They were
the parsed rtp.Packet, a copy of it in patchExtPacket, and the flow
state, whose address two rare log calls took. The rest were an
AbsCaptureTime, present on about one packet per second, and the
extension slice that pion grows with append.

Unmarshal into a pooled scratch packet. Give ExtPacket its own backing
rtp.Packet and an [8]rtp.Extension array, and copy the header into
them. Patch the payload in place. If the abs-capture-time extension is
present, parse it. Log a copy of the flow state.

Per packet, Write plus ReadExtended. Opus: 6 allocs, 448 B -> 0.
VP8: 6 allocs, 416 B -> 1 alloc, 32 B. Opus 240 -> 142 ns, VP8 269 -> 204 ns.

Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>

* sfu/buffer: point extension payloads at the bucket copy on read

Queued ExtPackets kept extension payload slices into the write input,
which the transport reuses for the next packet. ReadExtended now parses
the header again from the bucket copy, so extensions and CSRCs point
into RawPacket like the payload does.

The test writes every packet through one input buffer and checks the
audio level of each packet read back.

Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>

* sfu/buffer: test that a dropped padding packet keeps the shifted sequence number on read

The bucket stores the shifted number, so re-parsing the header on read
returns it. This test guards that.

Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>

---------

Co-authored-by: Claude Fable 5.1 <noreply@anthropic.com>
2026-10-06 18:21:11 +05:30

542 lines
12 KiB
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"
"io"
"github.com/pion/rtcp"
"github.com/pion/rtp"
"github.com/pion/webrtc/v4"
sutils "github.com/livekit/livekit-server/pkg/utils"
"github.com/livekit/mediatransportutil/pkg/bucket"
"github.com/livekit/mediatransportutil/pkg/twcc"
"github.com/livekit/protocol/livekit"
"github.com/livekit/protocol/utils/mono"
)
const (
rtcpReceiverReportDelta = 1e9
InitPacketBufferSizeVideo = 300
InitPacketBufferSizeAudio = 70
)
var (
errInvalidCodec = errors.New("invalid codec")
)
var _ BufferProvider = (*Buffer)(nil)
type pendingPacket struct {
arrivalTime int64
packet []byte
}
// Buffer contains all packets
type Buffer struct {
*BufferBase
pPackets []pendingPacket
lastReportAt int64
isBound bool
twcc *twcc.Responder
twccExtID uint8
enableAudioLossProxying bool
lastFractionLostToReport uint8 // Last fraction lost from subscribers, should report to publisher; Audio only
lastPacketRead int
// callbacks
onClose func()
onRtcpFeedback func([]rtcp.Packet)
onFinalRtpStats func(*livekit.RTPStats)
onNotifyRTX func(uint32, uint32, string)
primaryBufferForRTX *Buffer
rtxPktBuf []byte
streamInfoProbe *StreamInfoProbe
warnedPendingOverflow bool
}
// StreamInfoProbe identifies a stream from the mid/rid/rsid header extensions of its
// packets. It runs on the write path, i. e. as SRTP pushes into this buffer, because
// nothing reads remote streams through pion's interceptor chain.
type StreamInfoProbe struct {
MidExtID uint8
RidExtID uint8
RsidExtID uint8
// Tries bounds how many packets are inspected before giving up.
Tries int
// OnFound is called at most once, in a goroutine, as it can re-enter this buffer.
OnFound func(ssrc uint32, mid, rid, rsid string)
}
func NewBuffer(ssrc uint32, maxVideoPkts, maxAudioPkts int) *Buffer {
b := &Buffer{}
b.BufferBase = NewBufferBase(BufferBaseParams{
SSRC: ssrc,
MaxVideoPkts: maxVideoPkts,
MaxAudioPkts: maxAudioPkts,
LoggerComponents: []string{sutils.ComponentPub, sutils.ComponentSFU},
SendPLI: b.sendPLI,
IsReportingEnabled: true,
})
return b
}
func (b *Buffer) SetTWCCAndExtID(twcc *twcc.Responder, extID uint8) {
b.Lock()
defer b.Unlock()
b.twcc = twcc
b.twccExtID = extID
}
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) error {
b.Lock()
if b.isBound {
b.Unlock()
return nil
}
if err := b.BufferBase.BindLocked(params, codec, bitrates); err != nil {
b.Unlock()
return err
}
b.lastReportAt = mono.UnixNano()
if len(b.pPackets) != 0 {
b.logger.Debugw("releasing queued packets on bind", "count", len(b.pPackets))
}
var rtcpPackets []rtcp.Packet
for _, pp := range b.pPackets {
rtcpPackets = append(rtcpPackets, b.calc(pp.packet, nil, pp.arrivalTime, true, false)...)
}
b.pPackets = nil
b.isBound = true
b.Unlock()
if len(rtcpPackets) != 0 {
if cb := b.getOnRtcpFeedback(); cb != nil {
cb(rtcpPackets)
}
}
return nil
}
// Write adds an RTP Packet, ordering is not guaranteed, newer packets may arrive later
func (b *Buffer) Write(pkt []byte) (n int, err error) {
rtpPacket := rtpPacketFactory.Get().(*rtp.Packet)
defer rtpPacketFactory.Put(rtpPacket)
err = rtpPacket.Unmarshal(pkt)
if err != nil {
return
}
b.Lock()
if b.BufferBase.IsClosed() {
b.Unlock()
err = io.EOF
return
}
now := mono.UnixNano()
if b.twcc != nil && b.twccExtID != 0 {
if ext := rtpPacket.GetExtension(b.twccExtID); len(ext) >= 2 {
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
}
if b.streamInfoProbe != nil {
b.probeStreamInfoLocked(rtpPacket)
}
// 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.isBound {
packet := make([]byte, len(pkt))
copy(packet, pkt)
if len(b.pPackets) == 0 {
b.logger.Debugw("received first packet")
}
startIdx := 0
overflow := len(b.pPackets) - max(b.BufferBase.MaxVideoPkts(), b.BufferBase.MaxAudioPkts())
if overflow > 0 {
startIdx = overflow
// a stream that keeps arriving but never binds drops every packet from here
// on; for an RTX stream it means the pairing was never established
if !b.warnedPendingOverflow {
b.warnedPendingOverflow = true
b.logger.Warnw(
"unbound buffer overflowing, dropping packets", nil,
"ssrc", b.BufferBase.SSRC(),
"pending", len(b.pPackets),
)
}
}
b.pPackets = append(b.pPackets[startIdx:], pendingPacket{
packet: packet,
arrivalTime: now,
})
b.BufferBase.NotifyRead()
b.Unlock()
return
}
rtcpPackets := b.calc(pkt, rtpPacket, now, false, false)
b.Unlock()
if len(rtcpPackets) != 0 {
if cb := b.getOnRtcpFeedback(); cb != nil {
cb(rtcpPackets)
}
}
return
}
// SetStreamInfoProbe installs probe and runs it over packets already queued.
func (b *Buffer) SetStreamInfoProbe(probe *StreamInfoProbe) {
b.Lock()
defer b.Unlock()
b.streamInfoProbe = probe
for _, pp := range b.pPackets {
if b.streamInfoProbe == nil {
return
}
var rtpPacket rtp.Packet
if err := rtpPacket.Unmarshal(pp.packet); err != nil {
continue
}
b.probeStreamInfoLocked(&rtpPacket)
}
}
// probeStreamInfoLocked inspects one packet, clearing the probe once the stream is
// identified or the try budget runs out.
func (b *Buffer) probeStreamInfoLocked(rtpPacket *rtp.Packet) {
probe := b.streamInfoProbe
var mid, rid, rsid string
if ext := rtpPacket.GetExtension(probe.MidExtID); ext != nil {
mid = string(ext)
}
if ext := rtpPacket.GetExtension(probe.RidExtID); ext != nil {
rid = string(ext)
}
if ext := rtpPacket.GetExtension(probe.RsidExtID); ext != nil {
rsid = string(ext)
}
if mid != "" && (rid != "" || rsid != "") {
b.streamInfoProbe = nil
b.logger.Debugw("stream found", "ssrc", rtpPacket.SSRC, "mid", mid, "rid", rid, "rsid", rsid)
go probe.OnFound(rtpPacket.SSRC, mid, rid, rsid)
return
}
// ignore padding only packets for probe count
if rtpPacket.Padding && len(rtpPacket.Payload) == 0 {
return
}
if probe.Tries--; probe.Tries <= 0 {
b.streamInfoProbe = nil
}
}
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) NotifyRTX(ssrc uint32, repairSSRC uint32, rsid string) {
if onNotifyRTX := b.getOnNotifyRTX(); onNotifyRTX != nil {
onNotifyRTX(ssrc, repairSSRC, rsid)
}
}
func (b *Buffer) writeRTX(rtxPkt *rtp.Packet, arrivalTime int64) {
b.Lock()
defer b.Unlock()
if !b.isBound {
return
}
if rtxPkt.PayloadType != b.rtxPayloadType {
b.logger.Debugw("unexpected rtx payload type", "expected", b.rtxPayloadType, "actual", rtxPkt.PayloadType)
return
}
if b.rtxPktBuf == nil {
b.rtxPktBuf = make([]byte, bucket.RTPMaxPktSize)
}
if len(rtxPkt.Payload) < 2 {
b.logger.Warnw("rtx payload too short", nil, "size", len(rtxPkt.Payload))
return
}
repairedPkt := *rtxPkt
repairedPkt.PayloadType = b.payloadType
repairedPkt.SequenceNumber = binary.BigEndian.Uint16(rtxPkt.Payload[:2])
repairedPkt.SSRC = b.BufferBase.SSRC()
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.BufferBase.SSRC(), "sn", repairedPkt.SequenceNumber)
return
}
b.calc(b.rtxPktBuf[:n], &repairedPkt, arrivalTime, false, true)
}
func (b *Buffer) Read(buff []byte) (n int, err error) {
b.Lock()
for {
if b.BufferBase.IsClosed() {
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.BufferBase.WaitRead()
}
}
func (b *Buffer) Close() error {
stats, err := b.BufferBase.CloseWithReason("close")
if err != nil {
return err
}
if stats != nil {
if cb := b.getOnFinalRtpStats(); cb != nil {
cb(stats)
}
}
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) sendPLI() {
ssrc := b.BufferBase.SSRC()
if ssrc == 0 {
return
}
b.logger.Debugw("send pli", "mediaSSRC", ssrc)
pli := []rtcp.Packet{
&rtcp.PictureLossIndication{
SenderSSRC: ssrc,
MediaSSRC: ssrc,
},
}
if cb := b.getOnRtcpFeedback(); cb != nil {
cb(pli)
}
}
func (b *Buffer) calc(rawPkt []byte, rtpPacket *rtp.Packet, arrivalTime int64, isBuffered bool, isRTX bool) []rtcp.Packet {
b.BufferBase.HandleIncomingPacketLocked(
rawPkt,
rtpPacket,
arrivalTime,
isBuffered,
isRTX,
nil,
0,
)
return b.getRTCPPackets(arrivalTime)
}
func (b *Buffer) getRTCPPackets(arrivalTime int64) []rtcp.Packet {
var pkts []rtcp.Packet
if nackPkt := b.getNACKPacket(); nackPkt != nil {
pkts = append(pkts, nackPkt)
}
if receiverReport := b.getRTCPReceiverReport(arrivalTime); receiverReport != nil {
pkts = append(pkts, receiverReport)
}
return pkts
}
func (b *Buffer) getNACKPacket() *rtcp.TransportLayerNack {
if nacks := b.BufferBase.GetNACKPairsLocked(); len(nacks) > 0 {
ssrc := b.BufferBase.SSRC()
return &rtcp.TransportLayerNack{
SenderSSRC: ssrc,
MediaSSRC: ssrc,
Nacks: nacks,
}
}
return nil
}
func (b *Buffer) getRTCPReceiverReport(arrivalTime int64) *rtcp.ReceiverReport {
if arrivalTime-b.lastReportAt < rtcpReceiverReportDelta {
return nil
}
b.lastReportAt = arrivalTime
proxyLoss := b.lastFractionLostToReport
if b.codecType == webrtc.RTPCodecTypeAudio && !b.enableAudioLossProxying {
proxyLoss = 0
}
if receptionReport := b.BufferBase.GetRtcpReceptionReportLocked(proxyLoss); receptionReport != nil {
return &rtcp.ReceiverReport{
SSRC: b.BufferBase.SSRC(),
Reports: []rtcp.ReceptionReport{*receptionReport},
}
}
return nil
}
func (b *Buffer) SetLastFractionLostReport(lost uint8) {
b.Lock()
defer b.Unlock()
b.lastFractionLostToReport = lost
}
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) 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
}
func (b *Buffer) OnNotifyRTX(fn func(ssrc uint32, repairSSRC uint32, rsid string)) {
b.Lock()
b.onNotifyRTX = fn
b.Unlock()
}
func (b *Buffer) getOnNotifyRTX() func(ssrc uint32, repairSSRC uint32, rsid string) {
b.RLock()
defer b.RUnlock()
return b.onNotifyRTX
}