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galene/client.go

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2020-04-24 19:38:21 +02:00
// Copyright (c) 2020 by Juliusz Chroboczek.
// This is not open source software. Copy it, and I'll break into your
// house and tell your three year-old that Santa doesn't exist.
package main
import (
"encoding/json"
"errors"
"io"
"log"
"math"
"math/bits"
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"os"
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"strings"
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"sync"
"sync/atomic"
"time"
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"sfu/estimator"
"sfu/jitter"
"sfu/mono"
"sfu/packetcache"
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"github.com/gorilla/websocket"
"github.com/pion/rtcp"
"github.com/pion/rtp"
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"github.com/pion/sdp"
"github.com/pion/webrtc/v2"
)
var iceConf webrtc.Configuration
var iceOnce sync.Once
func iceConfiguration() webrtc.Configuration {
iceOnce.Do(func() {
var iceServers []webrtc.ICEServer
file, err := os.Open(iceFilename)
if err != nil {
log.Printf("Open %v: %v", iceFilename, err)
return
}
defer file.Close()
d := json.NewDecoder(file)
err = d.Decode(&iceServers)
if err != nil {
log.Printf("Get ICE configuration: %v", err)
return
}
iceConf = webrtc.Configuration{
ICEServers: iceServers,
}
})
return iceConf
}
type protocolError string
func (err protocolError) Error() string {
return string(err)
}
type userError string
func (err userError) Error() string {
return string(err)
}
func errorToWSCloseMessage(err error) (string, []byte) {
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var code int
var text string
switch e := err.(type) {
case *websocket.CloseError:
code = websocket.CloseNormalClosure
case protocolError:
code = websocket.CloseProtocolError
text = string(e)
case userError:
code = websocket.CloseNormalClosure
text = string(e)
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default:
code = websocket.CloseInternalServerErr
}
return text, websocket.FormatCloseMessage(code, text)
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}
func isWSNormalError(err error) bool {
return websocket.IsCloseError(err,
websocket.CloseNormalClosure,
websocket.CloseGoingAway)
}
type clientMessage struct {
Type string `json:"type"`
Id string `json:"id,omitempty"`
Username string `json:"username,omitempty"`
Password string `json:"password,omitempty"`
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Permissions userPermission `json:"permissions,omitempty"`
Group string `json:"group,omitempty"`
Value string `json:"value,omitempty"`
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Me bool `json:"me,omitempty"`
Offer *webrtc.SessionDescription `json:"offer,omitempty"`
Answer *webrtc.SessionDescription `json:"answer,omitempty"`
Candidate *webrtc.ICECandidateInit `json:"candidate,omitempty"`
Del bool `json:"del,omitempty"`
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Request []string `json:"request,omitempty"`
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}
type closeMessage struct {
data []byte
}
func startClient(conn *websocket.Conn) (err error) {
var m clientMessage
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err = conn.SetReadDeadline(time.Now().Add(15 * time.Second))
if err != nil {
conn.Close()
return
}
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err = conn.ReadJSON(&m)
if err != nil {
conn.Close()
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return
}
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err = conn.SetReadDeadline(time.Time{})
if err != nil {
conn.Close()
return
}
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if m.Type != "handshake" {
conn.Close()
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return
}
c := &client{
id: m.Id,
username: m.Username,
actionCh: make(chan interface{}, 10),
done: make(chan struct{}),
}
defer close(c.done)
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c.writeCh = make(chan interface{}, 25)
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defer func() {
if isWSNormalError(err) {
err = nil
} else {
m, e := errorToWSCloseMessage(err)
if m != "" {
c.write(clientMessage{
Type: "error",
Value: m,
})
}
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select {
case c.writeCh <- closeMessage{e}:
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case <-c.writerDone:
}
}
close(c.writeCh)
c.writeCh = nil
}()
c.writerDone = make(chan struct{})
go clientWriter(conn, c.writeCh, c.writerDone)
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if strings.ContainsRune(m.Username, ' ') {
err = userError("don't put spaces in your username")
return
}
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g, users, err := addClient(m.Group, c, m.Username, m.Password)
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if err != nil {
return
}
c.group = g
defer delClient(c)
for _, u := range users {
c.write(clientMessage{
Type: "user",
Id: u.id,
Username: u.username,
})
}
clients := g.getClients(nil)
u := clientMessage{
Type: "user",
Id: c.id,
Username: c.username,
}
for _, c := range clients {
c.write(u)
}
defer func() {
clients := g.getClients(c)
u := clientMessage{
Type: "user",
Id: c.id,
Username: c.username,
Del: true,
}
for _, c := range clients {
c.write(u)
}
}()
return clientLoop(c, conn)
}
func getUpConn(c *client, id string) *upConnection {
c.mu.Lock()
defer c.mu.Unlock()
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if c.up == nil {
return nil
}
conn := c.up[id]
if conn == nil {
return nil
}
return conn
}
func getUpConns(c *client) []string {
c.mu.Lock()
defer c.mu.Unlock()
up := make([]string, 0, len(c.up))
for id := range c.up {
up = append(up, id)
}
return up
}
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func addUpConn(c *client, id string) (*upConnection, error) {
pc, err := groups.api.NewPeerConnection(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
}
conn := &upConnection{id: id, pc: pc}
c.mu.Lock()
defer c.mu.Unlock()
if c.up == nil {
c.up = make(map[string]*upConnection)
}
if c.up[id] != nil {
conn.pc.Close()
return nil, errors.New("Adding duplicate connection")
}
c.up[id] = conn
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pc.OnICECandidate(func(candidate *webrtc.ICECandidate) {
sendICE(c, id, candidate)
})
pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
if state == webrtc.ICEConnectionStateFailed {
c.action(connectionFailedAction{id: id})
}
})
go rtcpUpSender(c, conn)
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pc.OnTrack(func(remote *webrtc.Track, receiver *webrtc.RTPReceiver) {
c.mu.Lock()
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u, ok := c.up[id]
if !ok {
log.Printf("Unknown connection")
c.mu.Unlock()
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return
}
track := &upTrack{
track: remote,
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cache: packetcache.New(96),
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rate: estimator.New(time.Second),
jitter: jitter.New(remote.Codec().ClockRate),
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maxBitrate: ^uint64(0),
}
u.tracks = append(u.tracks, track)
done := len(u.tracks) >= u.trackCount
if remote.Kind() == webrtc.RTPCodecTypeVideo {
atomic.AddUint32(&c.group.videoCount, 1)
}
c.mu.Unlock()
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clients := c.group.getClients(c)
for _, cc := range clients {
cc.action(addTrackAction{track, u, done})
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if done && u.label != "" {
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cc.action(addLabelAction{id, u.label})
}
}
go upLoop(conn, track)
go rtcpUpListener(conn, track, receiver)
})
return conn, nil
}
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func upLoop(conn *upConnection, track *upTrack) {
buf := make([]byte, packetcache.BufSize)
var packet rtp.Packet
var local []*downTrack
var localTime uint64
for {
now := mono.Microseconds()
if now < localTime || now > localTime+500000 {
local = track.getLocal()
localTime = now
}
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bytes, err := track.track.Read(buf)
if err != nil {
if err != io.EOF {
log.Printf("%v", err)
}
break
}
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track.rate.Add(uint32(bytes))
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err = packet.Unmarshal(buf[:bytes])
if err != nil {
log.Printf("%v", err)
continue
}
track.jitter.Accumulate(packet.Timestamp)
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first := track.cache.Store(packet.SequenceNumber, buf[:bytes])
if packet.SequenceNumber-first > 24 {
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found, first, bitmap := track.cache.BitmapGet()
if found {
err := conn.sendNACK(track, first, bitmap)
if err != nil {
log.Printf("%v", err)
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}
}
}
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for _, l := range local {
err := l.track.WriteRTP(&packet)
if err != nil && err != io.ErrClosedPipe {
log.Printf("%v", err)
}
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l.rate.Add(uint32(bytes))
}
}
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}
func rtcpUpListener(conn *upConnection, track *upTrack, r *webrtc.RTPReceiver) {
for {
ps, err := r.ReadRTCP()
if err != nil {
if err != io.EOF {
log.Printf("ReadRTCP: %v", err)
}
return
}
for _, p := range ps {
switch p := p.(type) {
case *rtcp.SenderReport:
atomic.StoreUint32(&track.lastSenderReport,
uint32(p.NTPTime>>16))
atomic.StoreUint32(&track.lastSenderReportTime,
uint32(mono.Now(0x10000)))
case *rtcp.SourceDescription:
}
}
}
}
func sendRR(c *client, conn *upConnection) error {
c.mu.Lock()
if len(conn.tracks) == 0 {
c.mu.Unlock()
return nil
}
now := uint32(mono.Now(0x10000))
reports := make([]rtcp.ReceptionReport, 0, len(conn.tracks))
for _, t := range conn.tracks {
expected, lost, totalLost, eseqno := t.cache.GetStats(true)
if expected == 0 {
expected = 1
}
if lost >= expected {
lost = expected - 1
}
lastSR := atomic.LoadUint32(&t.lastSenderReport)
delay := now - atomic.LoadUint32(&t.lastSenderReportTime)
reports = append(reports, rtcp.ReceptionReport{
SSRC: t.track.SSRC(),
FractionLost: uint8((lost * 256) / expected),
TotalLost: totalLost,
LastSequenceNumber: eseqno,
Jitter: t.jitter.Jitter(),
LastSenderReport: lastSR,
Delay: delay,
})
}
c.mu.Unlock()
return conn.pc.WriteRTCP([]rtcp.Packet{
&rtcp.ReceiverReport{
SSRC: 1,
Reports: reports,
},
})
}
func rtcpUpSender(c *client, conn *upConnection) {
for {
time.Sleep(time.Second)
err := sendRR(c, conn)
if err != nil {
if err == io.EOF || err == io.ErrClosedPipe {
return
}
log.Printf("WriteRTCP: %v", err)
}
}
}
func delUpConn(c *client, id string) bool {
c.mu.Lock()
defer c.mu.Unlock()
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if c.up == nil {
return false
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}
conn := c.up[id]
if conn == nil {
return false
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}
for _, track := range conn.tracks {
if track.track.Kind() == webrtc.RTPCodecTypeVideo {
count := atomic.AddUint32(&c.group.videoCount,
^uint32(0))
if count == ^uint32(0) {
log.Printf("Negative track count!")
atomic.StoreUint32(&c.group.videoCount, 0)
}
}
}
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type clientId struct {
client *client
id string
}
cids := make([]clientId, 0)
clients := c.group.getClients(c)
for _, cc := range clients {
cc.mu.Lock()
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for _, otherconn := range cc.down {
if otherconn.remote == conn {
cids = append(cids, clientId{cc, otherconn.id})
}
}
cc.mu.Unlock()
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}
go func(cids []clientId) {
for _, cid := range cids {
cid.client.action(delConnAction{cid.id})
}
}(cids)
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conn.pc.Close()
delete(c.up, id)
return true
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}
func getDownConn(c *client, id string) *downConnection {
if c.down == nil {
return nil
}
c.mu.Lock()
defer c.mu.Unlock()
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conn := c.down[id]
if conn == nil {
return nil
}
return conn
}
func addDownConn(c *client, id string, remote *upConnection) (*downConnection, error) {
pc, err := groups.api.NewPeerConnection(iceConfiguration())
if err != nil {
return nil, err
}
pc.OnICECandidate(func(candidate *webrtc.ICECandidate) {
sendICE(c, id, candidate)
})
pc.OnTrack(func(remote *webrtc.Track, receiver *webrtc.RTPReceiver) {
log.Printf("Got track on downstream connection")
})
pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
if state == webrtc.ICEConnectionStateFailed {
c.action(connectionFailedAction{id: id})
}
})
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if c.down == nil {
c.down = make(map[string]*downConnection)
}
conn := &downConnection{id: id, pc: pc, remote: remote}
c.mu.Lock()
defer c.mu.Unlock()
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if c.down[id] != nil {
conn.pc.Close()
return nil, errors.New("Adding duplicate connection")
}
c.down[id] = conn
return conn, nil
}
func delDownConn(c *client, id string) bool {
c.mu.Lock()
defer c.mu.Unlock()
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if c.down == nil {
return false
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}
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conn := c.down[id]
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if conn == nil {
return false
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}
for _, track := range conn.tracks {
found := track.remote.delLocal(track)
if !found {
log.Printf("Couldn't find remote track")
}
}
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conn.pc.Close()
delete(c.down, id)
return true
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}
func addDownTrack(c *client, id string, remoteTrack *upTrack, remoteConn *upConnection) (*downConnection, *webrtc.RTPSender, error) {
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conn := getDownConn(c, id)
if conn == nil {
var err error
conn, err = addDownConn(c, id, remoteConn)
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if err != nil {
return nil, nil, err
}
}
local, err := conn.pc.NewTrack(
remoteTrack.track.PayloadType(),
remoteTrack.track.SSRC(),
remoteTrack.track.ID(),
remoteTrack.track.Label(),
)
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if err != nil {
return nil, nil, err
}
s, err := conn.pc.AddTrack(local)
if err != nil {
return nil, nil, err
}
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track := &downTrack{
track: local,
remote: remoteTrack,
maxLossBitrate: new(bitrate),
maxREMBBitrate: new(bitrate),
stats: new(receiverStats),
rate: estimator.New(time.Second),
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}
conn.tracks = append(conn.tracks, track)
remoteTrack.addLocal(track)
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go rtcpDownListener(conn, track, s)
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return conn, s, nil
}
const (
minLossRate = 9600
initLossRate = 512 * 1000
maxLossRate = 1 << 30
)
func (track *downTrack) updateRate(loss uint8, now uint64) {
rate := track.maxLossBitrate.Get(now)
if rate > maxLossRate {
// no recent feedback, reset
rate = initLossRate
}
if loss < 5 {
// if our actual rate is low, then we're not probing the
// bottleneck
actual := 8 * uint64(track.rate.Estimate())
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.maxLossBitrate.Set(rate, now)
}
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func rtcpDownListener(conn *downConnection, track *downTrack, s *webrtc.RTPSender) {
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for {
ps, err := s.ReadRTCP()
if err != nil {
if err != io.EOF {
log.Printf("ReadRTCP: %v", err)
}
return
}
for _, p := range ps {
switch p := p.(type) {
case *rtcp.PictureLossIndication:
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err := conn.remote.sendPLI(track.remote)
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if err != nil {
log.Printf("sendPLI: %v", err)
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}
case *rtcp.ReceiverEstimatedMaximumBitrate:
track.maxREMBBitrate.Set(
p.Bitrate, mono.Microseconds(),
)
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case *rtcp.ReceiverReport:
for _, r := range p.Reports {
if r.SSRC == track.track.SSRC() {
now := mono.Microseconds()
track.stats.Set(
r.FractionLost,
r.Jitter,
now,
)
track.updateRate(
r.FractionLost,
now,
)
}
}
case *rtcp.TransportLayerNack:
maxBitrate := track.GetMaxBitrate(
mono.Microseconds(),
)
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bitrate := track.rate.Estimate()
if uint64(bitrate) < maxBitrate {
sendRecovery(p, track)
}
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}
}
}
}
func trackKinds(down *downConnection) (audio bool, video bool) {
if down.pc == nil {
return
}
for _, s := range down.pc.GetSenders() {
track := s.Track()
if track == nil {
continue
}
switch track.Kind() {
case webrtc.RTPCodecTypeAudio:
audio = true
case webrtc.RTPCodecTypeVideo:
video = true
}
}
return
}
func updateUpBitrate(up *upConnection, maxVideoRate uint64) {
now := mono.Microseconds()
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for _, track := range up.tracks {
isvideo := track.track.Kind() == webrtc.RTPCodecTypeVideo
minrate := uint64(minAudioRate)
rate := ^uint64(0)
if isvideo {
minrate = minVideoRate
rate = maxVideoRate
if rate < minrate {
rate = minrate
}
}
local := track.getLocal()
for _, l := range local {
bitrate := l.GetMaxBitrate(now)
if bitrate == ^uint64(0) {
continue
}
if bitrate <= minrate {
rate = minrate
break
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}
if rate > bitrate {
rate = bitrate
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}
}
track.maxBitrate = rate
}
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}
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func (up *upConnection) sendPLI(track *upTrack) error {
if !track.hasRtcpFb("nack", "pli") {
return nil
}
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last := atomic.LoadUint64(&track.lastPLI)
now := mono.Microseconds()
if now >= last && now-last < 200000 {
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return nil
}
atomic.StoreUint64(&track.lastPLI, now)
return sendPLI(up.pc, track.track.SSRC())
}
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func sendPLI(pc *webrtc.PeerConnection, ssrc uint32) error {
return pc.WriteRTCP([]rtcp.Packet{
&rtcp.PictureLossIndication{MediaSSRC: ssrc},
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})
}
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func sendREMB(pc *webrtc.PeerConnection, ssrc uint32, bitrate uint64) error {
return pc.WriteRTCP([]rtcp.Packet{
&rtcp.ReceiverEstimatedMaximumBitrate{
Bitrate: bitrate,
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SSRCs: []uint32{ssrc},
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},
})
}
func (up *upConnection) sendNACK(track *upTrack, 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{
rtcp.NackPair{
first,
rtcp.PacketBitmap(bitmap),
},
},
},
)
return pc.WriteRTCP([]rtcp.Packet{packet})
}
func sendRecovery(p *rtcp.TransportLayerNack, track *downTrack) {
var packet rtp.Packet
for _, nack := range p.Nacks {
for _, seqno := range nack.PacketList() {
raw := track.remote.cache.Get(seqno)
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if raw == nil {
continue
}
err := packet.Unmarshal(raw)
if err != nil {
continue
}
err = track.track.WriteRTP(&packet)
if err != nil {
log.Printf("%v", err)
continue
}
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track.rate.Add(uint32(len(raw)))
}
}
}
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func countMediaStreams(data string) (int, error) {
desc := sdp.NewJSEPSessionDescription(false)
err := desc.Unmarshal(data)
if err != nil {
return 0, err
}
return len(desc.MediaDescriptions), nil
}
func negotiate(c *client, id string, pc *webrtc.PeerConnection) error {
offer, err := pc.CreateOffer(nil)
if err != nil {
return err
}
err = pc.SetLocalDescription(offer)
if err != nil {
return err
}
return c.write(clientMessage{
Type: "offer",
Id: id,
Offer: &offer,
})
}
func sendICE(c *client, id string, candidate *webrtc.ICECandidate) error {
if candidate == nil {
return nil
}
cand := candidate.ToJSON()
return c.write(clientMessage{
Type: "ice",
Id: id,
Candidate: &cand,
})
}
func gotOffer(c *client, offer webrtc.SessionDescription, id string) error {
var err error
up, ok := c.up[id]
if !ok {
up, err = addUpConn(c, id)
if err != nil {
return err
}
}
if c.username != "" {
up.label = c.username
}
n, err := countMediaStreams(offer.SDP)
if err != nil {
log.Printf("Couldn't parse SDP: %v", err)
n = 2
}
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up.trackCount = n
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err = up.pc.SetRemoteDescription(offer)
if err != nil {
return err
}
answer, err := up.pc.CreateAnswer(nil)
if err != nil {
return err
}
err = up.pc.SetLocalDescription(answer)
if err != nil {
return err
}
return c.write(clientMessage{
Type: "answer",
Id: id,
Answer: &answer,
})
}
func gotAnswer(c *client, answer webrtc.SessionDescription, id string) error {
conn := getDownConn(c, id)
if conn == nil {
return protocolError("unknown id in answer")
}
err := conn.pc.SetRemoteDescription(answer)
if err != nil {
return err
}
return nil
}
func gotICE(c *client, candidate *webrtc.ICECandidateInit, id string) error {
var pc *webrtc.PeerConnection
down := getDownConn(c, id)
if down != nil {
pc = down.pc
} else {
up := getUpConn(c, id)
if up == nil {
return errors.New("unknown id in ICE")
}
pc = up.pc
}
return pc.AddICECandidate(*candidate)
}
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func (c *client) setRequested(audio, video bool) error {
if audio == c.requestedAudio && video == c.requestedVideo {
return nil
}
if c.down != nil {
for id := range c.down {
c.write(clientMessage{
Type: "close",
Id: id,
})
delDownConn(c, id)
}
}
c.requestedAudio = audio
c.requestedVideo = video
go func() {
clients := c.group.getClients(c)
for _, cc := range clients {
cc.action(pushTracksAction{c})
}
}()
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return nil
}
func (c *client) requested(kind webrtc.RTPCodecType) bool {
switch kind {
case webrtc.RTPCodecTypeAudio:
return c.requestedAudio
case webrtc.RTPCodecTypeVideo:
return c.requestedVideo
default:
return false
}
}
func pushTracks(c *client, conn *upConnection, tracks []*upTrack, done bool, label string) {
for i, t := range tracks {
c.action(addTrackAction{t, conn, done && i == len(tracks)-1})
}
if done && label != "" {
c.action(addLabelAction{conn.id, conn.label})
}
}
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func clientLoop(c *client, conn *websocket.Conn) error {
read := make(chan interface{}, 1)
go clientReader(conn, read, c.done)
defer func() {
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c.setRequested(false, false)
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if c.up != nil {
for id := range c.up {
delUpConn(c, id)
}
}
}()
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c.write(clientMessage{
Type: "permissions",
Permissions: c.permissions,
})
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h := c.group.getChatHistory()
for _, m := range h {
err := c.write(clientMessage{
Type: "chat",
Id: m.id,
Username: m.user,
Value: m.value,
Me: m.me,
})
if err != nil {
return err
}
}
readTime := time.Now()
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ticker := time.NewTicker(time.Second)
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defer ticker.Stop()
slowTicker := time.NewTicker(10 * time.Second)
defer slowTicker.Stop()
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for {
select {
case m, ok := <-read:
if !ok {
return errors.New("reader died")
}
switch m := m.(type) {
case clientMessage:
readTime = time.Now()
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err := handleClientMessage(c, m)
if err != nil {
return err
}
case error:
return m
}
case a := <-c.actionCh:
switch a := a.(type) {
case addTrackAction:
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var down *downConnection
var err error
if c.requested(a.track.track.Kind()) {
down, _, err = addDownTrack(
c, a.remote.id, a.track,
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a.remote)
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if err != nil {
return err
}
} else {
down = getDownConn(c, a.remote.id)
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}
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if a.done && down != nil {
err = negotiate(c, a.remote.id, down.pc)
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if err != nil {
return err
}
}
case delConnAction:
found := delDownConn(c, a.id)
if found {
c.write(clientMessage{
Type: "close",
Id: a.id,
})
}
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case addLabelAction:
c.write(clientMessage{
Type: "label",
Id: a.id,
Value: a.label,
})
case pushTracksAction:
for _, u := range c.up {
tracks := make([]*upTrack, len(u.tracks))
copy(tracks, u.tracks)
go pushTracks(
a.c, u, tracks,
len(tracks) >= u.trackCount-1,
u.label,
)
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}
case connectionFailedAction:
found := delUpConn(c, a.id)
if found {
c.write(clientMessage{
Type: "error",
Value: "connection failed",
})
c.write(clientMessage{
Type: "abort",
Id: a.id,
})
continue
}
// What should we do if a downstream
// connection fails? Renegotiate?
case permissionsChangedAction:
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c.write(clientMessage{
Type: "permissions",
Permissions: c.permissions,
})
if !c.permissions.Present {
ids := getUpConns(c)
for _, id := range ids {
c.write(clientMessage{
Type: "abort",
Id: id,
})
delUpConn(c, id)
}
}
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case kickAction:
return userError("you have been kicked")
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default:
log.Printf("unexpected action %T", a)
return errors.New("unexpected action")
}
case <-ticker.C:
sendRateUpdate(c)
case <-slowTicker.C:
if time.Since(readTime) > 90*time.Second {
return errors.New("client is dead")
}
if time.Since(readTime) > 60*time.Second {
err := c.write(clientMessage{
Type: "ping",
})
if err != nil {
return err
}
}
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}
}
}
func handleClientMessage(c *client, m clientMessage) error {
switch m.Type {
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case "request":
var audio, video bool
for _, s := range m.Request {
switch(s) {
case "audio": audio = true
case "video": video = true
default: log.Printf("Unknown request %v", s)
}
}
err := c.setRequested(audio, video)
if err != nil {
return err
}
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case "offer":
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if !c.permissions.Present {
c.write(clientMessage{
Type: "abort",
Id: m.Id,
})
return c.error(userError("not authorised"))
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}
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if m.Offer == nil {
return protocolError("null offer")
}
err := gotOffer(c, *m.Offer, m.Id)
if err != nil {
return err
}
case "answer":
if m.Answer == nil {
return protocolError("null answer")
}
err := gotAnswer(c, *m.Answer, m.Id)
if err != nil {
return err
}
case "close":
found := delUpConn(c, m.Id)
if !found {
log.Printf("Deleting unknown up connection %v", m.Id)
}
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case "ice":
if m.Candidate == nil {
return protocolError("null candidate")
}
err := gotICE(c, m.Candidate, m.Id)
if err != nil {
log.Printf("ICE: %v", err)
}
case "chat":
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c.group.addToChatHistory(m.Id, m.Username, m.Value, m.Me)
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clients := c.group.getClients(c)
for _, cc := range clients {
cc.write(m)
}
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case "clearchat":
c.group.clearChatHistory()
m := clientMessage{Type: "clearchat"}
clients := c.group.getClients(nil)
for _, cc := range clients {
cc.write(m)
}
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case "op", "unop", "present", "unpresent":
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if !c.permissions.Op {
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c.error(userError("not authorised"))
return nil
}
err := setPermission(c.group, m.Id, m.Type)
if err != nil {
return c.error(err)
}
case "kick":
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if !c.permissions.Op {
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c.error(userError("not authorised"))
return nil
}
err := kickClient(c.group, m.Id)
if err != nil {
return c.error(err)
}
case "pong":
// nothing
case "ping":
c.write(clientMessage{
Type: "pong",
})
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default:
log.Printf("unexpected message: %v", m.Type)
return protocolError("unexpected message")
}
return nil
}
func sendRateUpdate(c *client) {
type remb struct {
pc *webrtc.PeerConnection
ssrc uint32
bitrate uint64
}
rembs := make([]remb, 0)
maxVideoRate := ^uint64(0)
count := atomic.LoadUint32(&c.group.videoCount)
if count >= 3 {
maxVideoRate = uint64(2000000 / math.Sqrt(float64(count)))
if maxVideoRate < minVideoRate {
maxVideoRate = minVideoRate
}
}
c.mu.Lock()
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for _, u := range c.up {
updateUpBitrate(u, maxVideoRate)
for _, t := range u.tracks {
if !t.hasRtcpFb("goog-remb", "") {
continue
}
bitrate := t.maxBitrate
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if bitrate == ^uint64(0) {
continue
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}
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rembs = append(rembs,
remb{u.pc, t.track.SSRC(), bitrate})
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}
}
c.mu.Unlock()
for _, r := range rembs {
err := sendREMB(r.pc, r.ssrc, r.bitrate)
if err != nil {
log.Printf("sendREMB: %v", err)
}
}
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}
func clientReader(conn *websocket.Conn, read chan<- interface{}, done <-chan struct{}) {
defer close(read)
for {
var m clientMessage
err := conn.ReadJSON(&m)
if err != nil {
select {
case read <- err:
return
case <-done:
return
}
}
select {
case read <- m:
case <-done:
return
}
}
}
func clientWriter(conn *websocket.Conn, ch <-chan interface{}, done chan<- struct{}) {
defer func() {
close(done)
conn.Close()
}()
for {
m, ok := <-ch
if !ok {
break
}
err := conn.SetWriteDeadline(
time.Now().Add(2 * time.Second))
if err != nil {
return
}
switch m := m.(type) {
case clientMessage:
err := conn.WriteJSON(m)
if err != nil {
return
}
case closeMessage:
err := conn.WriteMessage(websocket.CloseMessage, m.data)
if err != nil {
return
}
default:
log.Printf("clientWiter: unexpected message %T", m)
return
}
}
}