Files
livekit/pkg/sfu/buffer/buffer.go
T
Raja SubramanianandClaude Opus 5 bbdfd50d62 Fix simulcast RTX pairing broken by pion/webrtc#3470 (#4800)
pion no longer starts the repair stream reader when a custom BufferFactory
is set, so the mid/rid/rsid extensions were never observed and simulcast RTX
streams were never paired with their primary streams.

Extract the extensions on the buffer write path instead. Migrated publishers
send no extensions at all, so pair those from SimTracks.

Adds an integration test covering both paths, and moves the vnet setup it
shares with the downtrack test into pkg/testutils/vnettest.

Co-authored-by: Claude Opus 5 <noreply@anthropic.com>
2026-08-26 13:01:22 +05:30

541 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) {
var rtpPacket rtp.Packet
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
}