mirror of
https://github.com/jech/galene.git
synced 2024-11-09 18:25:58 +01:00
1063 lines
23 KiB
Go
1063 lines
23 KiB
Go
package rtpconn
|
|
|
|
import (
|
|
"errors"
|
|
"io"
|
|
"log"
|
|
"math/bits"
|
|
"sync"
|
|
"sync/atomic"
|
|
"time"
|
|
|
|
"github.com/pion/rtcp"
|
|
"github.com/pion/rtp"
|
|
"github.com/pion/webrtc/v3"
|
|
|
|
"sfu/conn"
|
|
"sfu/estimator"
|
|
"sfu/group"
|
|
"sfu/jitter"
|
|
"sfu/packetcache"
|
|
"sfu/rtptime"
|
|
)
|
|
|
|
type bitrate struct {
|
|
bitrate uint64
|
|
jiffies uint64
|
|
}
|
|
|
|
func (br *bitrate) Set(bitrate uint64, now uint64) {
|
|
atomic.StoreUint64(&br.bitrate, bitrate)
|
|
atomic.StoreUint64(&br.jiffies, now)
|
|
}
|
|
|
|
func (br *bitrate) Get(now uint64) uint64 {
|
|
ts := atomic.LoadUint64(&br.jiffies)
|
|
if now < ts || now-ts > receiverReportTimeout {
|
|
return ^uint64(0)
|
|
}
|
|
return atomic.LoadUint64(&br.bitrate)
|
|
}
|
|
|
|
type receiverStats struct {
|
|
loss uint32
|
|
jitter uint32
|
|
jiffies 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.jiffies, now)
|
|
}
|
|
|
|
func (s *receiverStats) Get(now uint64) (uint8, uint32) {
|
|
ts := atomic.LoadUint64(&s.jiffies)
|
|
if now < ts || now > ts+receiverReportTimeout {
|
|
return 0, 0
|
|
}
|
|
return uint8(atomic.LoadUint32(&s.loss)), atomic.LoadUint32(&s.jitter)
|
|
}
|
|
|
|
const receiverReportTimeout = 8 * rtptime.JiffiesPerSec
|
|
|
|
type iceConnection interface {
|
|
addICECandidate(candidate *webrtc.ICECandidateInit) error
|
|
flushICECandidates() error
|
|
}
|
|
|
|
type rtpDownTrack struct {
|
|
track *webrtc.Track
|
|
remote conn.UpTrack
|
|
maxBitrate *bitrate
|
|
rate *estimator.Estimator
|
|
stats *receiverStats
|
|
srTime uint64
|
|
srNTPTime uint64
|
|
remoteNTPTime uint64
|
|
remoteRTPTime uint32
|
|
cname atomic.Value
|
|
rtt uint64
|
|
}
|
|
|
|
func (down *rtpDownTrack) WriteRTP(packet *rtp.Packet) error {
|
|
return down.track.WriteRTP(packet)
|
|
}
|
|
|
|
func (down *rtpDownTrack) Accumulate(bytes uint32) {
|
|
down.rate.Accumulate(bytes)
|
|
}
|
|
|
|
func (down *rtpDownTrack) SetTimeOffset(ntp uint64, rtp uint32) {
|
|
atomic.StoreUint64(&down.remoteNTPTime, ntp)
|
|
atomic.StoreUint32(&down.remoteRTPTime, rtp)
|
|
}
|
|
|
|
func (down *rtpDownTrack) SetCname(cname string) {
|
|
down.cname.Store(cname)
|
|
}
|
|
|
|
type rtpDownConnection struct {
|
|
id string
|
|
pc *webrtc.PeerConnection
|
|
remote conn.Up
|
|
tracks []*rtpDownTrack
|
|
maxREMBBitrate *bitrate
|
|
iceCandidates []*webrtc.ICECandidateInit
|
|
}
|
|
|
|
func newDownConn(c group.Client, id string, remote conn.Up) (*rtpDownConnection, error) {
|
|
pc, err := c.Group().API().NewPeerConnection(group.IceConfiguration())
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
pc.OnTrack(func(remote *webrtc.Track, receiver *webrtc.RTPReceiver) {
|
|
log.Printf("Got track on downstream connection")
|
|
})
|
|
|
|
conn := &rtpDownConnection{
|
|
id: id,
|
|
pc: pc,
|
|
remote: remote,
|
|
maxREMBBitrate: new(bitrate),
|
|
}
|
|
|
|
return conn, nil
|
|
}
|
|
|
|
func (down *rtpDownConnection) GetMaxBitrate(now uint64) uint64 {
|
|
rate := down.maxREMBBitrate.Get(now)
|
|
var trackRate uint64
|
|
for _, t := range down.tracks {
|
|
r := t.maxBitrate.Get(now)
|
|
if r == ^uint64(0) {
|
|
if t.track.Kind() == webrtc.RTPCodecTypeAudio {
|
|
r = 128 * 1024
|
|
} else {
|
|
r = 512 * 1024
|
|
}
|
|
}
|
|
trackRate += r
|
|
}
|
|
if trackRate < rate {
|
|
return trackRate
|
|
}
|
|
return rate
|
|
}
|
|
|
|
func (down *rtpDownConnection) addICECandidate(candidate *webrtc.ICECandidateInit) error {
|
|
if down.pc.RemoteDescription() != nil {
|
|
return down.pc.AddICECandidate(*candidate)
|
|
}
|
|
down.iceCandidates = append(down.iceCandidates, candidate)
|
|
return nil
|
|
}
|
|
|
|
func flushICECandidates(pc *webrtc.PeerConnection, candidates []*webrtc.ICECandidateInit) error {
|
|
if pc.RemoteDescription() == nil {
|
|
return errors.New("flushICECandidates called in bad state")
|
|
}
|
|
|
|
var err error
|
|
for _, candidate := range candidates {
|
|
err2 := pc.AddICECandidate(*candidate)
|
|
if err == nil {
|
|
err = err2
|
|
}
|
|
}
|
|
return err
|
|
}
|
|
|
|
func (down *rtpDownConnection) flushICECandidates() error {
|
|
err := flushICECandidates(down.pc, down.iceCandidates)
|
|
down.iceCandidates = nil
|
|
return err
|
|
}
|
|
|
|
type rtpUpTrack struct {
|
|
track *webrtc.Track
|
|
label string
|
|
rate *estimator.Estimator
|
|
cache *packetcache.Cache
|
|
jitter *jitter.Estimator
|
|
lastPLI uint64
|
|
lastFIR uint64
|
|
firSeqno uint32
|
|
|
|
localCh chan localTrackAction
|
|
readerDone chan struct{}
|
|
|
|
mu sync.Mutex
|
|
cname string
|
|
local []conn.DownTrack
|
|
srTime uint64
|
|
srNTPTime uint64
|
|
srRTPTime uint32
|
|
}
|
|
|
|
type localTrackAction struct {
|
|
add bool
|
|
track conn.DownTrack
|
|
}
|
|
|
|
func (up *rtpUpTrack) notifyLocal(add bool, track conn.DownTrack) {
|
|
select {
|
|
case up.localCh <- localTrackAction{add, track}:
|
|
case <-up.readerDone:
|
|
}
|
|
}
|
|
|
|
func (up *rtpUpTrack) AddLocal(local conn.DownTrack) error {
|
|
up.mu.Lock()
|
|
for _, t := range up.local {
|
|
if t == local {
|
|
up.mu.Unlock()
|
|
return nil
|
|
}
|
|
}
|
|
up.local = append(up.local, local)
|
|
up.mu.Unlock()
|
|
|
|
up.notifyLocal(true, local)
|
|
return nil
|
|
}
|
|
|
|
func (up *rtpUpTrack) DelLocal(local conn.DownTrack) bool {
|
|
up.mu.Lock()
|
|
for i, l := range up.local {
|
|
if l == local {
|
|
up.local = append(up.local[:i], up.local[i+1:]...)
|
|
up.mu.Unlock()
|
|
up.notifyLocal(false, l)
|
|
return true
|
|
}
|
|
}
|
|
up.mu.Unlock()
|
|
return false
|
|
}
|
|
|
|
func (up *rtpUpTrack) getLocal() []conn.DownTrack {
|
|
up.mu.Lock()
|
|
defer up.mu.Unlock()
|
|
local := make([]conn.DownTrack, len(up.local))
|
|
copy(local, up.local)
|
|
return local
|
|
}
|
|
|
|
func (up *rtpUpTrack) GetRTP(seqno uint16, result []byte) uint16 {
|
|
return up.cache.Get(seqno, result)
|
|
}
|
|
|
|
func (up *rtpUpTrack) Label() string {
|
|
return up.label
|
|
}
|
|
|
|
func (up *rtpUpTrack) Codec() *webrtc.RTPCodec {
|
|
return up.track.Codec()
|
|
}
|
|
|
|
func (up *rtpUpTrack) hasRtcpFb(tpe, parameter string) bool {
|
|
for _, fb := range up.track.Codec().RTCPFeedback {
|
|
if fb.Type == tpe && fb.Parameter == parameter {
|
|
return true
|
|
}
|
|
}
|
|
return false
|
|
}
|
|
|
|
type rtpUpConnection struct {
|
|
id string
|
|
label string
|
|
pc *webrtc.PeerConnection
|
|
labels map[string]string
|
|
iceCandidates []*webrtc.ICECandidateInit
|
|
|
|
mu sync.Mutex
|
|
tracks []*rtpUpTrack
|
|
local []conn.Down
|
|
}
|
|
|
|
func (up *rtpUpConnection) getTracks() []*rtpUpTrack {
|
|
up.mu.Lock()
|
|
defer up.mu.Unlock()
|
|
tracks := make([]*rtpUpTrack, len(up.tracks))
|
|
copy(tracks, up.tracks)
|
|
return tracks
|
|
}
|
|
|
|
func (up *rtpUpConnection) Id() string {
|
|
return up.id
|
|
}
|
|
|
|
func (up *rtpUpConnection) Label() string {
|
|
return up.label
|
|
}
|
|
|
|
func (up *rtpUpConnection) AddLocal(local conn.Down) error {
|
|
up.mu.Lock()
|
|
defer up.mu.Unlock()
|
|
for _, t := range up.local {
|
|
if t == local {
|
|
return nil
|
|
}
|
|
}
|
|
up.local = append(up.local, local)
|
|
return nil
|
|
}
|
|
|
|
func (up *rtpUpConnection) DelLocal(local conn.Down) bool {
|
|
up.mu.Lock()
|
|
defer up.mu.Unlock()
|
|
for i, l := range up.local {
|
|
if l == local {
|
|
up.local = append(up.local[:i], up.local[i+1:]...)
|
|
return true
|
|
}
|
|
}
|
|
return false
|
|
}
|
|
|
|
func (up *rtpUpConnection) getLocal() []conn.Down {
|
|
up.mu.Lock()
|
|
defer up.mu.Unlock()
|
|
local := make([]conn.Down, len(up.local))
|
|
copy(local, up.local)
|
|
return local
|
|
}
|
|
|
|
func (up *rtpUpConnection) addICECandidate(candidate *webrtc.ICECandidateInit) error {
|
|
if up.pc.RemoteDescription() != nil {
|
|
return up.pc.AddICECandidate(*candidate)
|
|
}
|
|
up.iceCandidates = append(up.iceCandidates, candidate)
|
|
return nil
|
|
}
|
|
|
|
func (up *rtpUpConnection) flushICECandidates() error {
|
|
err := flushICECandidates(up.pc, up.iceCandidates)
|
|
up.iceCandidates = nil
|
|
return err
|
|
}
|
|
|
|
func getTrackMid(pc *webrtc.PeerConnection, track *webrtc.Track) string {
|
|
for _, t := range pc.GetTransceivers() {
|
|
if t.Receiver() != nil && t.Receiver().Track() == track {
|
|
return t.Mid()
|
|
}
|
|
}
|
|
return ""
|
|
}
|
|
|
|
// called locked
|
|
func (up *rtpUpConnection) complete() bool {
|
|
for mid := range up.labels {
|
|
found := false
|
|
for _, t := range up.tracks {
|
|
m := getTrackMid(up.pc, t.track)
|
|
if m == mid {
|
|
found = true
|
|
break
|
|
}
|
|
}
|
|
if !found {
|
|
return false
|
|
}
|
|
}
|
|
return true
|
|
}
|
|
|
|
func newUpConn(c group.Client, id string) (*rtpUpConnection, error) {
|
|
pc, err := c.Group().API().NewPeerConnection(group.IceConfiguration())
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio,
|
|
webrtc.RtpTransceiverInit{
|
|
Direction: webrtc.RTPTransceiverDirectionRecvonly,
|
|
},
|
|
)
|
|
if err != nil {
|
|
pc.Close()
|
|
return nil, err
|
|
}
|
|
|
|
_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo,
|
|
webrtc.RtpTransceiverInit{
|
|
Direction: webrtc.RTPTransceiverDirectionRecvonly,
|
|
},
|
|
)
|
|
if err != nil {
|
|
pc.Close()
|
|
return nil, err
|
|
}
|
|
|
|
up := &rtpUpConnection{id: id, pc: pc}
|
|
|
|
pc.OnTrack(func(remote *webrtc.Track, receiver *webrtc.RTPReceiver) {
|
|
up.mu.Lock()
|
|
|
|
mid := getTrackMid(pc, remote)
|
|
if mid == "" {
|
|
log.Printf("Couldn't get track's mid")
|
|
return
|
|
}
|
|
|
|
label, ok := up.labels[mid]
|
|
if !ok {
|
|
log.Printf("Couldn't get track's label")
|
|
isvideo := remote.Kind() == webrtc.RTPCodecTypeVideo
|
|
if isvideo {
|
|
label = "video"
|
|
} else {
|
|
label = "audio"
|
|
}
|
|
}
|
|
|
|
track := &rtpUpTrack{
|
|
track: remote,
|
|
label: label,
|
|
cache: packetcache.New(32),
|
|
rate: estimator.New(time.Second),
|
|
jitter: jitter.New(remote.Codec().ClockRate),
|
|
localCh: make(chan localTrackAction, 2),
|
|
readerDone: make(chan struct{}),
|
|
}
|
|
|
|
up.tracks = append(up.tracks, track)
|
|
|
|
go readLoop(up, track)
|
|
|
|
go rtcpUpListener(up, track, receiver)
|
|
|
|
complete := up.complete()
|
|
var tracks []conn.UpTrack
|
|
if complete {
|
|
tracks = make([]conn.UpTrack, len(up.tracks))
|
|
for i, t := range up.tracks {
|
|
tracks[i] = t
|
|
}
|
|
}
|
|
|
|
// pushConn might need to take the lock
|
|
up.mu.Unlock()
|
|
|
|
if complete {
|
|
clients := c.Group().GetClients(c)
|
|
for _, cc := range clients {
|
|
cc.PushConn(up.id, up, tracks, up.label)
|
|
}
|
|
go rtcpUpSender(up)
|
|
}
|
|
})
|
|
|
|
return up, nil
|
|
}
|
|
|
|
func readLoop(conn *rtpUpConnection, track *rtpUpTrack) {
|
|
writers := rtpWriterPool{conn: conn, track: track}
|
|
defer func() {
|
|
writers.close()
|
|
close(track.readerDone)
|
|
}()
|
|
|
|
isvideo := track.track.Kind() == webrtc.RTPCodecTypeVideo
|
|
buf := make([]byte, packetcache.BufSize)
|
|
var packet rtp.Packet
|
|
for {
|
|
bytes, err := track.track.Read(buf)
|
|
if err != nil {
|
|
if err != io.EOF {
|
|
log.Printf("%v", err)
|
|
}
|
|
break
|
|
}
|
|
track.rate.Accumulate(uint32(bytes))
|
|
|
|
err = packet.Unmarshal(buf[:bytes])
|
|
if err != nil {
|
|
log.Printf("%v", err)
|
|
continue
|
|
}
|
|
|
|
track.jitter.Accumulate(packet.Timestamp)
|
|
|
|
first, index :=
|
|
track.cache.Store(packet.SequenceNumber, buf[:bytes])
|
|
if packet.SequenceNumber-first > 24 {
|
|
found, first, bitmap := track.cache.BitmapGet()
|
|
if found {
|
|
err := conn.sendNACK(track, first, bitmap)
|
|
if err != nil {
|
|
log.Printf("%v", err)
|
|
}
|
|
}
|
|
}
|
|
|
|
_, rate := track.rate.Estimate()
|
|
delay := uint32(rtptime.JiffiesPerSec / 1024)
|
|
if rate > 512 {
|
|
delay = rtptime.JiffiesPerSec / rate / 2
|
|
}
|
|
|
|
writers.write(packet.SequenceNumber, index, delay,
|
|
isvideo, packet.Marker)
|
|
|
|
select {
|
|
case action := <-track.localCh:
|
|
err := writers.add(action.track, action.add)
|
|
if err != nil {
|
|
log.Printf("add/remove track: %v", err)
|
|
}
|
|
default:
|
|
}
|
|
}
|
|
}
|
|
|
|
var ErrUnsupportedFeedback = errors.New("unsupported feedback type")
|
|
var ErrRateLimited = errors.New("rate limited")
|
|
|
|
func (up *rtpUpConnection) sendPLI(track *rtpUpTrack) error {
|
|
if !track.hasRtcpFb("nack", "pli") {
|
|
return ErrUnsupportedFeedback
|
|
}
|
|
last := atomic.LoadUint64(&track.lastPLI)
|
|
now := rtptime.Jiffies()
|
|
if now >= last && now-last < rtptime.JiffiesPerSec/5 {
|
|
return ErrRateLimited
|
|
}
|
|
atomic.StoreUint64(&track.lastPLI, now)
|
|
return sendPLI(up.pc, track.track.SSRC())
|
|
}
|
|
|
|
func sendPLI(pc *webrtc.PeerConnection, ssrc uint32) error {
|
|
return pc.WriteRTCP([]rtcp.Packet{
|
|
&rtcp.PictureLossIndication{MediaSSRC: ssrc},
|
|
})
|
|
}
|
|
|
|
func (up *rtpUpConnection) sendFIR(track *rtpUpTrack, increment bool) error {
|
|
// we need to reliably increment the seqno, even if we are going
|
|
// to drop the packet due to rate limiting.
|
|
var seqno uint8
|
|
if increment {
|
|
seqno = uint8(atomic.AddUint32(&track.firSeqno, 1) & 0xFF)
|
|
} else {
|
|
seqno = uint8(atomic.LoadUint32(&track.firSeqno) & 0xFF)
|
|
}
|
|
|
|
if !track.hasRtcpFb("ccm", "fir") {
|
|
return ErrUnsupportedFeedback
|
|
}
|
|
last := atomic.LoadUint64(&track.lastFIR)
|
|
now := rtptime.Jiffies()
|
|
if now >= last && now-last < rtptime.JiffiesPerSec/5 {
|
|
return ErrRateLimited
|
|
}
|
|
atomic.StoreUint64(&track.lastFIR, now)
|
|
return sendFIR(up.pc, track.track.SSRC(), seqno)
|
|
}
|
|
|
|
func sendFIR(pc *webrtc.PeerConnection, ssrc uint32, seqno uint8) error {
|
|
return pc.WriteRTCP([]rtcp.Packet{
|
|
&rtcp.FullIntraRequest{
|
|
FIR: []rtcp.FIREntry{
|
|
{
|
|
SSRC: ssrc,
|
|
SequenceNumber: seqno,
|
|
},
|
|
},
|
|
},
|
|
})
|
|
}
|
|
|
|
func (up *rtpUpConnection) sendNACK(track *rtpUpTrack, first uint16, bitmap uint16) error {
|
|
if !track.hasRtcpFb("nack", "") {
|
|
return nil
|
|
}
|
|
err := sendNACK(up.pc, track.track.SSRC(), first, bitmap)
|
|
if err == nil {
|
|
track.cache.Expect(1 + bits.OnesCount16(bitmap))
|
|
}
|
|
return err
|
|
}
|
|
|
|
func sendNACK(pc *webrtc.PeerConnection, ssrc uint32, first uint16, bitmap uint16) error {
|
|
packet := rtcp.Packet(
|
|
&rtcp.TransportLayerNack{
|
|
MediaSSRC: ssrc,
|
|
Nacks: []rtcp.NackPair{
|
|
{
|
|
first,
|
|
rtcp.PacketBitmap(bitmap),
|
|
},
|
|
},
|
|
},
|
|
)
|
|
return pc.WriteRTCP([]rtcp.Packet{packet})
|
|
}
|
|
|
|
func sendRecovery(p *rtcp.TransportLayerNack, track *rtpDownTrack) {
|
|
var packet rtp.Packet
|
|
buf := make([]byte, packetcache.BufSize)
|
|
for _, nack := range p.Nacks {
|
|
for _, seqno := range nack.PacketList() {
|
|
l := track.remote.GetRTP(seqno, buf)
|
|
if l == 0 {
|
|
continue
|
|
}
|
|
err := packet.Unmarshal(buf[:l])
|
|
if err != nil {
|
|
continue
|
|
}
|
|
err = track.track.WriteRTP(&packet)
|
|
if err != nil {
|
|
log.Printf("WriteRTP: %v", err)
|
|
continue
|
|
}
|
|
track.rate.Accumulate(uint32(l))
|
|
}
|
|
}
|
|
}
|
|
|
|
func rtcpUpListener(conn *rtpUpConnection, track *rtpUpTrack, r *webrtc.RTPReceiver) {
|
|
buf := make([]byte, 1500)
|
|
|
|
for {
|
|
firstSR := false
|
|
n, err := r.Read(buf)
|
|
if err != nil {
|
|
if err != io.EOF {
|
|
log.Printf("Read RTCP: %v", err)
|
|
}
|
|
return
|
|
}
|
|
ps, err := rtcp.Unmarshal(buf[:n])
|
|
if err != nil {
|
|
log.Printf("Unmarshal RTCP: %v", err)
|
|
continue
|
|
}
|
|
|
|
jiffies := rtptime.Jiffies()
|
|
|
|
for _, p := range ps {
|
|
local := track.getLocal()
|
|
switch p := p.(type) {
|
|
case *rtcp.SenderReport:
|
|
track.mu.Lock()
|
|
if track.srTime == 0 {
|
|
firstSR = true
|
|
}
|
|
track.srTime = jiffies
|
|
track.srNTPTime = p.NTPTime
|
|
track.srRTPTime = p.RTPTime
|
|
track.mu.Unlock()
|
|
for _, l := range local {
|
|
l.SetTimeOffset(p.NTPTime, p.RTPTime)
|
|
}
|
|
case *rtcp.SourceDescription:
|
|
for _, c := range p.Chunks {
|
|
if c.Source != track.track.SSRC() {
|
|
continue
|
|
}
|
|
for _, i := range c.Items {
|
|
if i.Type != rtcp.SDESCNAME {
|
|
continue
|
|
}
|
|
track.mu.Lock()
|
|
track.cname = i.Text
|
|
track.mu.Unlock()
|
|
for _, l := range local {
|
|
l.SetCname(i.Text)
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if firstSR {
|
|
// this is the first SR we got for at least one track,
|
|
// quickly propagate the time offsets downstream
|
|
local := conn.getLocal()
|
|
for _, l := range local {
|
|
l, ok := l.(*rtpDownConnection)
|
|
if ok {
|
|
err := sendSR(l)
|
|
if err != nil {
|
|
log.Printf("sendSR: %v", err)
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
func sendUpRTCP(conn *rtpUpConnection) error {
|
|
conn.mu.Lock()
|
|
defer conn.mu.Unlock()
|
|
|
|
if len(conn.tracks) == 0 {
|
|
state := conn.pc.ConnectionState()
|
|
if state == webrtc.PeerConnectionStateClosed {
|
|
return io.ErrClosedPipe
|
|
}
|
|
return nil
|
|
}
|
|
|
|
now := rtptime.Jiffies()
|
|
|
|
reports := make([]rtcp.ReceptionReport, 0, len(conn.tracks))
|
|
for _, t := range conn.tracks {
|
|
updateUpTrack(t)
|
|
expected, lost, totalLost, eseqno := t.cache.GetStats(true)
|
|
if expected == 0 {
|
|
expected = 1
|
|
}
|
|
if lost >= expected {
|
|
lost = expected - 1
|
|
}
|
|
|
|
t.mu.Lock()
|
|
srTime := t.srTime
|
|
srNTPTime := t.srNTPTime
|
|
t.mu.Unlock()
|
|
|
|
var delay uint64
|
|
if srTime != 0 {
|
|
delay = (now - srTime) /
|
|
(rtptime.JiffiesPerSec / 0x10000)
|
|
}
|
|
|
|
reports = append(reports, rtcp.ReceptionReport{
|
|
SSRC: t.track.SSRC(),
|
|
FractionLost: uint8((lost * 256) / expected),
|
|
TotalLost: totalLost,
|
|
LastSequenceNumber: eseqno,
|
|
Jitter: t.jitter.Jitter(),
|
|
LastSenderReport: uint32(srNTPTime >> 16),
|
|
Delay: uint32(delay),
|
|
})
|
|
}
|
|
|
|
packets := []rtcp.Packet{
|
|
&rtcp.ReceiverReport{
|
|
Reports: reports,
|
|
},
|
|
}
|
|
|
|
rate := ^uint64(0)
|
|
for _, l := range conn.local {
|
|
r := l.GetMaxBitrate(now)
|
|
if r < rate {
|
|
rate = r
|
|
}
|
|
}
|
|
if rate < group.MinBitrate {
|
|
rate = group.MinBitrate
|
|
}
|
|
|
|
var ssrcs []uint32
|
|
for _, t := range conn.tracks {
|
|
if t.hasRtcpFb("goog-remb", "") {
|
|
continue
|
|
}
|
|
ssrcs = append(ssrcs, t.track.SSRC())
|
|
}
|
|
|
|
if len(ssrcs) > 0 {
|
|
packets = append(packets,
|
|
&rtcp.ReceiverEstimatedMaximumBitrate{
|
|
Bitrate: rate,
|
|
SSRCs: ssrcs,
|
|
},
|
|
)
|
|
}
|
|
return conn.pc.WriteRTCP(packets)
|
|
}
|
|
|
|
func rtcpUpSender(conn *rtpUpConnection) {
|
|
for {
|
|
time.Sleep(time.Second)
|
|
err := sendUpRTCP(conn)
|
|
if err != nil {
|
|
if err == io.EOF || err == io.ErrClosedPipe {
|
|
return
|
|
}
|
|
log.Printf("sendRR: %v", err)
|
|
}
|
|
}
|
|
}
|
|
|
|
func sendSR(conn *rtpDownConnection) error {
|
|
// since this is only called after all tracks have been created,
|
|
// there is no need for locking.
|
|
packets := make([]rtcp.Packet, 0, len(conn.tracks))
|
|
|
|
now := time.Now()
|
|
nowNTP := rtptime.TimeToNTP(now)
|
|
jiffies := rtptime.TimeToJiffies(now)
|
|
|
|
for _, t := range conn.tracks {
|
|
clockrate := t.track.Codec().ClockRate
|
|
|
|
var nowRTP uint32
|
|
|
|
remoteNTP := atomic.LoadUint64(&t.remoteNTPTime)
|
|
remoteRTP := atomic.LoadUint32(&t.remoteRTPTime)
|
|
if remoteNTP != 0 {
|
|
srTime := rtptime.NTPToTime(remoteNTP)
|
|
d := now.Sub(srTime)
|
|
if d > 0 && d < time.Hour {
|
|
delay := rtptime.FromDuration(
|
|
d, clockrate,
|
|
)
|
|
nowRTP = remoteRTP + uint32(delay)
|
|
}
|
|
|
|
p, b := t.rate.Totals()
|
|
packets = append(packets,
|
|
&rtcp.SenderReport{
|
|
SSRC: t.track.SSRC(),
|
|
NTPTime: nowNTP,
|
|
RTPTime: nowRTP,
|
|
PacketCount: p,
|
|
OctetCount: b,
|
|
})
|
|
atomic.StoreUint64(&t.srTime, jiffies)
|
|
atomic.StoreUint64(&t.srNTPTime, nowNTP)
|
|
}
|
|
|
|
cname, ok := t.cname.Load().(string)
|
|
if ok {
|
|
item := rtcp.SourceDescriptionItem{
|
|
Type: rtcp.SDESCNAME,
|
|
Text: cname,
|
|
}
|
|
packets = append(packets,
|
|
&rtcp.SourceDescription{
|
|
Chunks: []rtcp.SourceDescriptionChunk{
|
|
{
|
|
Source: t.track.SSRC(),
|
|
Items: []rtcp.SourceDescriptionItem{item},
|
|
},
|
|
},
|
|
},
|
|
)
|
|
}
|
|
}
|
|
|
|
if len(packets) == 0 {
|
|
state := conn.pc.ConnectionState()
|
|
if state == webrtc.PeerConnectionStateClosed {
|
|
return io.ErrClosedPipe
|
|
}
|
|
return nil
|
|
}
|
|
|
|
return conn.pc.WriteRTCP(packets)
|
|
}
|
|
|
|
func rtcpDownSender(conn *rtpDownConnection) {
|
|
for {
|
|
time.Sleep(time.Second)
|
|
err := sendSR(conn)
|
|
if err != nil {
|
|
if err == io.EOF || err == io.ErrClosedPipe {
|
|
return
|
|
}
|
|
log.Printf("sendSR: %v", err)
|
|
}
|
|
}
|
|
}
|
|
|
|
const (
|
|
minLossRate = 9600
|
|
initLossRate = 512 * 1000
|
|
maxLossRate = 1 << 30
|
|
)
|
|
|
|
func (track *rtpDownTrack) updateRate(loss uint8, now uint64) {
|
|
rate := track.maxBitrate.Get(now)
|
|
if rate < minLossRate || rate > maxLossRate {
|
|
// no recent feedback, reset
|
|
rate = initLossRate
|
|
}
|
|
if loss < 5 {
|
|
// if our actual rate is low, then we're not probing the
|
|
// bottleneck
|
|
r, _ := track.rate.Estimate()
|
|
actual := 8 * uint64(r)
|
|
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.maxBitrate.Set(rate, now)
|
|
}
|
|
|
|
func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RTPSender) {
|
|
var gotFir bool
|
|
lastFirSeqno := uint8(0)
|
|
|
|
buf := make([]byte, 1500)
|
|
|
|
for {
|
|
n, err := s.Read(buf)
|
|
if err != nil {
|
|
if err != io.EOF {
|
|
log.Printf("Read RTCP: %v", err)
|
|
}
|
|
return
|
|
}
|
|
ps, err := rtcp.Unmarshal(buf[:n])
|
|
if err != nil {
|
|
log.Printf("Unmarshal RTCP: %v", err)
|
|
continue
|
|
}
|
|
|
|
jiffies := rtptime.Jiffies()
|
|
|
|
for _, p := range ps {
|
|
switch p := p.(type) {
|
|
case *rtcp.PictureLossIndication:
|
|
remote, ok := conn.remote.(*rtpUpConnection)
|
|
if !ok {
|
|
continue
|
|
}
|
|
rt, ok := track.remote.(*rtpUpTrack)
|
|
if !ok {
|
|
continue
|
|
}
|
|
err := remote.sendPLI(rt)
|
|
if err != nil && err != ErrRateLimited {
|
|
log.Printf("sendPLI: %v", err)
|
|
}
|
|
case *rtcp.FullIntraRequest:
|
|
found := false
|
|
var seqno uint8
|
|
for _, entry := range p.FIR {
|
|
if entry.SSRC == track.track.SSRC() {
|
|
found = true
|
|
seqno = entry.SequenceNumber
|
|
break
|
|
}
|
|
}
|
|
if !found {
|
|
log.Printf("Misdirected FIR")
|
|
continue
|
|
}
|
|
|
|
increment := true
|
|
if gotFir {
|
|
increment = seqno != lastFirSeqno
|
|
}
|
|
gotFir = true
|
|
lastFirSeqno = seqno
|
|
|
|
remote, ok := conn.remote.(*rtpUpConnection)
|
|
if !ok {
|
|
continue
|
|
}
|
|
rt, ok := track.remote.(*rtpUpTrack)
|
|
if !ok {
|
|
continue
|
|
}
|
|
err := remote.sendFIR(rt, increment)
|
|
if err == ErrUnsupportedFeedback {
|
|
err := remote.sendPLI(rt)
|
|
if err != nil && err != ErrRateLimited {
|
|
log.Printf("sendPLI: %v", err)
|
|
}
|
|
} else if err != nil {
|
|
log.Printf("sendFIR: %v", err)
|
|
}
|
|
case *rtcp.ReceiverEstimatedMaximumBitrate:
|
|
conn.maxREMBBitrate.Set(p.Bitrate, jiffies)
|
|
case *rtcp.ReceiverReport:
|
|
for _, r := range p.Reports {
|
|
if r.SSRC == track.track.SSRC() {
|
|
handleReport(track, r, jiffies)
|
|
}
|
|
}
|
|
case *rtcp.SenderReport:
|
|
for _, r := range p.Reports {
|
|
if r.SSRC == track.track.SSRC() {
|
|
handleReport(track, r, jiffies)
|
|
}
|
|
}
|
|
case *rtcp.TransportLayerNack:
|
|
sendRecovery(p, track)
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
func handleReport(track *rtpDownTrack, report rtcp.ReceptionReport, jiffies uint64) {
|
|
track.stats.Set(report.FractionLost, report.Jitter, jiffies)
|
|
track.updateRate(report.FractionLost, jiffies)
|
|
|
|
if report.LastSenderReport != 0 {
|
|
jiffies := rtptime.Jiffies()
|
|
srTime := atomic.LoadUint64(&track.srTime)
|
|
if jiffies < srTime || jiffies-srTime > 8*rtptime.JiffiesPerSec {
|
|
return
|
|
}
|
|
srNTPTime := atomic.LoadUint64(&track.srNTPTime)
|
|
if report.LastSenderReport == uint32(srNTPTime>>16) {
|
|
delay := uint64(report.Delay) *
|
|
(rtptime.JiffiesPerSec / 0x10000)
|
|
if delay > jiffies-srTime {
|
|
return
|
|
}
|
|
rtt := (jiffies - srTime) - delay
|
|
oldrtt := atomic.LoadUint64(&track.rtt)
|
|
newrtt := rtt
|
|
if oldrtt > 0 {
|
|
newrtt = (3*oldrtt + rtt) / 4
|
|
}
|
|
atomic.StoreUint64(&track.rtt, newrtt)
|
|
}
|
|
}
|
|
}
|
|
|
|
func updateUpTrack(track *rtpUpTrack) {
|
|
now := rtptime.Jiffies()
|
|
|
|
clockrate := track.track.Codec().ClockRate
|
|
local := track.getLocal()
|
|
var maxrto uint64
|
|
for _, l := range local {
|
|
ll, ok := l.(*rtpDownTrack)
|
|
if ok {
|
|
_, j := ll.stats.Get(now)
|
|
jitter := uint64(j) *
|
|
(rtptime.JiffiesPerSec / uint64(clockrate))
|
|
rtt := atomic.LoadUint64(&ll.rtt)
|
|
rto := rtt + 4*jitter
|
|
if rto > maxrto {
|
|
maxrto = rto
|
|
}
|
|
}
|
|
}
|
|
_, r := track.rate.Estimate()
|
|
packets := int((uint64(r) * maxrto * 4) / rtptime.JiffiesPerSec)
|
|
if packets < 32 {
|
|
packets = 32
|
|
}
|
|
if packets > 256 {
|
|
packets = 256
|
|
}
|
|
track.cache.ResizeCond(packets)
|
|
}
|