mirror of
https://github.com/jech/galene.git
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1324 lines
26 KiB
Go
1324 lines
26 KiB
Go
// Copyright (c) 2020 by Juliusz Chroboczek.
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// This is not open source software. Copy it, and I'll break into your
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// house and tell your three year-old that Santa doesn't exist.
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package main
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import (
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"encoding/json"
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"errors"
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"io"
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"log"
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"math"
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"math/bits"
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"os"
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"strings"
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"sync"
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"sync/atomic"
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"time"
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"sfu/estimator"
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"sfu/jitter"
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"sfu/mono"
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"sfu/packetcache"
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"github.com/gorilla/websocket"
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"github.com/pion/rtcp"
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"github.com/pion/rtp"
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"github.com/pion/sdp"
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"github.com/pion/webrtc/v2"
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)
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var iceConf webrtc.Configuration
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var iceOnce sync.Once
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func iceConfiguration() webrtc.Configuration {
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iceOnce.Do(func() {
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var iceServers []webrtc.ICEServer
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file, err := os.Open(iceFilename)
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if err != nil {
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log.Printf("Open %v: %v", iceFilename, err)
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return
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}
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defer file.Close()
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d := json.NewDecoder(file)
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err = d.Decode(&iceServers)
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if err != nil {
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log.Printf("Get ICE configuration: %v", err)
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return
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}
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iceConf = webrtc.Configuration{
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ICEServers: iceServers,
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}
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})
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return iceConf
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}
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type protocolError string
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func (err protocolError) Error() string {
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return string(err)
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}
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type userError string
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func (err userError) Error() string {
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return string(err)
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}
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func errorToWSCloseMessage(err error) (string, []byte) {
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var code int
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var text string
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switch e := err.(type) {
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case *websocket.CloseError:
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code = websocket.CloseNormalClosure
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case protocolError:
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code = websocket.CloseProtocolError
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text = string(e)
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case userError:
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code = websocket.CloseNormalClosure
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text = string(e)
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default:
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code = websocket.CloseInternalServerErr
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}
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return text, websocket.FormatCloseMessage(code, text)
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}
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func isWSNormalError(err error) bool {
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return websocket.IsCloseError(err,
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websocket.CloseNormalClosure,
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websocket.CloseGoingAway)
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}
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type clientMessage struct {
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Type string `json:"type"`
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Id string `json:"id,omitempty"`
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Username string `json:"username,omitempty"`
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Password string `json:"password,omitempty"`
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Permissions userPermission `json:"permissions,omitempty"`
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Group string `json:"group,omitempty"`
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Value string `json:"value,omitempty"`
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Me bool `json:"me,omitempty"`
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Offer *webrtc.SessionDescription `json:"offer,omitempty"`
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Answer *webrtc.SessionDescription `json:"answer,omitempty"`
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Candidate *webrtc.ICECandidateInit `json:"candidate,omitempty"`
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Del bool `json:"del,omitempty"`
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Request []string `json:"request,omitempty"`
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}
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type closeMessage struct {
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data []byte
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}
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func startClient(conn *websocket.Conn) (err error) {
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var m clientMessage
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err = conn.SetReadDeadline(time.Now().Add(15 * time.Second))
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if err != nil {
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conn.Close()
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return
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}
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err = conn.ReadJSON(&m)
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if err != nil {
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conn.Close()
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return
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}
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err = conn.SetReadDeadline(time.Time{})
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if err != nil {
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conn.Close()
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return
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}
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if m.Type != "handshake" {
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conn.Close()
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return
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}
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c := &client{
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id: m.Id,
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username: m.Username,
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actionCh: make(chan interface{}, 10),
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done: make(chan struct{}),
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}
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defer close(c.done)
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c.writeCh = make(chan interface{}, 25)
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defer func() {
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if isWSNormalError(err) {
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err = nil
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} else {
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m, e := errorToWSCloseMessage(err)
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if m != "" {
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c.write(clientMessage{
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Type: "error",
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Value: m,
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})
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}
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select {
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case c.writeCh <- closeMessage{e}:
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case <-c.writerDone:
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}
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}
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close(c.writeCh)
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c.writeCh = nil
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}()
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c.writerDone = make(chan struct{})
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go clientWriter(conn, c.writeCh, c.writerDone)
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if strings.ContainsRune(m.Username, ' ') {
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err = userError("don't put spaces in your username")
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return
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}
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g, users, err := addClient(m.Group, c, m.Username, m.Password)
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if err != nil {
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return
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}
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c.group = g
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defer delClient(c)
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for _, u := range users {
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c.write(clientMessage{
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Type: "user",
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Id: u.id,
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Username: u.username,
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})
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}
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clients := g.getClients(nil)
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u := clientMessage{
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Type: "user",
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Id: c.id,
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Username: c.username,
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}
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for _, c := range clients {
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c.write(u)
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}
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defer func() {
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clients := g.getClients(c)
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u := clientMessage{
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Type: "user",
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Id: c.id,
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Username: c.username,
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Del: true,
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}
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for _, c := range clients {
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c.write(u)
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}
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}()
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return clientLoop(c, conn)
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}
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func getUpConn(c *client, id string) *upConnection {
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c.mu.Lock()
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defer c.mu.Unlock()
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if c.up == nil {
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return nil
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}
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conn := c.up[id]
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if conn == nil {
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return nil
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}
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return conn
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}
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func getUpConns(c *client) []string {
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c.mu.Lock()
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defer c.mu.Unlock()
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up := make([]string, 0, len(c.up))
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for id := range c.up {
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up = append(up, id)
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}
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return up
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}
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func addUpConn(c *client, id string) (*upConnection, error) {
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pc, err := groups.api.NewPeerConnection(iceConfiguration())
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if err != nil {
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return nil, err
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}
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_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio,
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webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionRecvonly,
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},
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)
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if err != nil {
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pc.Close()
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return nil, err
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}
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_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo,
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webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionRecvonly,
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},
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)
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if err != nil {
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pc.Close()
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return nil, err
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}
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conn := &upConnection{id: id, pc: pc}
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c.mu.Lock()
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defer c.mu.Unlock()
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if c.up == nil {
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c.up = make(map[string]*upConnection)
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}
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if c.up[id] != nil {
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conn.pc.Close()
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return nil, errors.New("Adding duplicate connection")
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}
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c.up[id] = conn
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pc.OnICECandidate(func(candidate *webrtc.ICECandidate) {
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sendICE(c, id, candidate)
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})
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pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
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if state == webrtc.ICEConnectionStateFailed {
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c.action(connectionFailedAction{id: id})
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}
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})
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go rtcpUpSender(c, conn)
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pc.OnTrack(func(remote *webrtc.Track, receiver *webrtc.RTPReceiver) {
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c.mu.Lock()
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u, ok := c.up[id]
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if !ok {
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log.Printf("Unknown connection")
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c.mu.Unlock()
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return
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}
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track := &upTrack{
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track: remote,
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cache: packetcache.New(96),
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rate: estimator.New(time.Second),
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jitter: jitter.New(remote.Codec().ClockRate),
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maxBitrate: ^uint64(0),
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}
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u.tracks = append(u.tracks, track)
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done := len(u.tracks) >= u.trackCount
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if remote.Kind() == webrtc.RTPCodecTypeVideo {
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atomic.AddUint32(&c.group.videoCount, 1)
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}
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c.mu.Unlock()
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clients := c.group.getClients(c)
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for _, cc := range clients {
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cc.action(addTrackAction{track, u, done})
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if done && u.label != "" {
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cc.action(addLabelAction{id, u.label})
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}
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}
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go upLoop(conn, track)
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go rtcpUpListener(conn, track, receiver)
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})
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return conn, nil
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}
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func upLoop(conn *upConnection, track *upTrack) {
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buf := make([]byte, packetcache.BufSize)
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var packet rtp.Packet
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var local []*downTrack
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var localTime uint64
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for {
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now := mono.Microseconds()
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if now < localTime || now > localTime+500000 {
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local = track.getLocal()
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localTime = now
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}
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bytes, err := track.track.Read(buf)
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if err != nil {
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if err != io.EOF {
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log.Printf("%v", err)
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}
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break
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}
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track.rate.Add(uint32(bytes))
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err = packet.Unmarshal(buf[:bytes])
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if err != nil {
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log.Printf("%v", err)
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continue
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}
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track.jitter.Accumulate(packet.Timestamp)
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first := track.cache.Store(packet.SequenceNumber, buf[:bytes])
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if packet.SequenceNumber-first > 24 {
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found, first, bitmap := track.cache.BitmapGet()
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if found {
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err := conn.sendNACK(track, first, bitmap)
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if err != nil {
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log.Printf("%v", err)
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}
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}
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}
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for _, l := range local {
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err := l.track.WriteRTP(&packet)
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if err != nil && err != io.ErrClosedPipe {
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log.Printf("%v", err)
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}
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l.rate.Add(uint32(bytes))
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}
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}
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}
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func rtcpUpListener(conn *upConnection, track *upTrack, r *webrtc.RTPReceiver) {
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for {
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ps, err := r.ReadRTCP()
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if err != nil {
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if err != io.EOF {
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log.Printf("ReadRTCP: %v", err)
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}
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return
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}
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for _, p := range ps {
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switch p := p.(type) {
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case *rtcp.SenderReport:
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atomic.StoreUint32(&track.lastSenderReport,
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uint32(p.NTPTime>>16))
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atomic.StoreUint32(&track.lastSenderReportTime,
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uint32(mono.Now(0x10000)))
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case *rtcp.SourceDescription:
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}
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}
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}
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}
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func sendRR(c *client, conn *upConnection) error {
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c.mu.Lock()
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if len(conn.tracks) == 0 {
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c.mu.Unlock()
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return nil
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}
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now := uint32(mono.Now(0x10000))
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reports := make([]rtcp.ReceptionReport, 0, len(conn.tracks))
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for _, t := range conn.tracks {
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expected, lost, totalLost, eseqno := t.cache.GetStats(true)
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if expected == 0 {
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expected = 1
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}
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if lost >= expected {
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lost = expected - 1
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}
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lastSR := atomic.LoadUint32(&t.lastSenderReport)
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delay := now - atomic.LoadUint32(&t.lastSenderReportTime)
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reports = append(reports, rtcp.ReceptionReport{
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SSRC: t.track.SSRC(),
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FractionLost: uint8((lost * 256) / expected),
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TotalLost: totalLost,
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LastSequenceNumber: eseqno,
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Jitter: t.jitter.Jitter(),
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LastSenderReport: lastSR,
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Delay: delay,
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})
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}
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c.mu.Unlock()
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return conn.pc.WriteRTCP([]rtcp.Packet{
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&rtcp.ReceiverReport{
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SSRC: 1,
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Reports: reports,
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},
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})
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}
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func rtcpUpSender(c *client, conn *upConnection) {
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for {
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time.Sleep(time.Second)
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err := sendRR(c, conn)
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if err != nil {
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if err == io.EOF || err == io.ErrClosedPipe {
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return
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}
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log.Printf("WriteRTCP: %v", err)
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}
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}
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}
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func delUpConn(c *client, id string) bool {
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c.mu.Lock()
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defer c.mu.Unlock()
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if c.up == nil {
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return false
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}
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conn := c.up[id]
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if conn == nil {
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return false
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}
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for _, track := range conn.tracks {
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if track.track.Kind() == webrtc.RTPCodecTypeVideo {
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count := atomic.AddUint32(&c.group.videoCount,
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^uint32(0))
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if count == ^uint32(0) {
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log.Printf("Negative track count!")
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atomic.StoreUint32(&c.group.videoCount, 0)
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}
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}
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}
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type clientId struct {
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client *client
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id string
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}
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cids := make([]clientId, 0)
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clients := c.group.getClients(c)
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for _, cc := range clients {
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cc.mu.Lock()
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for _, otherconn := range cc.down {
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if otherconn.remote == conn {
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cids = append(cids, clientId{cc, otherconn.id})
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}
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}
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cc.mu.Unlock()
|
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}
|
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|
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for _, cid := range cids {
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cid.client.action(delConnAction{cid.id})
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}
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|
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conn.pc.Close()
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delete(c.up, id)
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return true
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}
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|
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func getDownConn(c *client, id string) *downConnection {
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if c.down == nil {
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return nil
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}
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|
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c.mu.Lock()
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defer c.mu.Unlock()
|
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conn := c.down[id]
|
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if conn == nil {
|
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return nil
|
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}
|
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return conn
|
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}
|
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|
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func addDownConn(c *client, id string, remote *upConnection) (*downConnection, error) {
|
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pc, err := groups.api.NewPeerConnection(iceConfiguration())
|
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if err != nil {
|
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return nil, err
|
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}
|
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|
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pc.OnICECandidate(func(candidate *webrtc.ICECandidate) {
|
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sendICE(c, id, candidate)
|
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})
|
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|
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pc.OnTrack(func(remote *webrtc.Track, receiver *webrtc.RTPReceiver) {
|
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log.Printf("Got track on downstream connection")
|
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})
|
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|
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pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
|
|
if state == webrtc.ICEConnectionStateFailed {
|
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c.action(connectionFailedAction{id: id})
|
|
}
|
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})
|
|
|
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if c.down == nil {
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c.down = make(map[string]*downConnection)
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}
|
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conn := &downConnection{id: id, pc: pc, remote: remote}
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|
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c.mu.Lock()
|
|
defer c.mu.Unlock()
|
|
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()
|
|
|
|
if c.down == nil {
|
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return false
|
|
}
|
|
conn := c.down[id]
|
|
if conn == nil {
|
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return false
|
|
}
|
|
|
|
for _, track := range conn.tracks {
|
|
found := track.remote.delLocal(track)
|
|
if !found {
|
|
log.Printf("Couldn't find remote track")
|
|
}
|
|
}
|
|
conn.pc.Close()
|
|
delete(c.down, id)
|
|
return true
|
|
}
|
|
|
|
func addDownTrack(c *client, id string, remoteTrack *upTrack, remoteConn *upConnection) (*downConnection, *webrtc.RTPSender, error) {
|
|
conn := getDownConn(c, id)
|
|
if conn == nil {
|
|
var err error
|
|
conn, err = addDownConn(c, id, remoteConn)
|
|
if err != nil {
|
|
return nil, nil, err
|
|
}
|
|
}
|
|
|
|
local, err := conn.pc.NewTrack(
|
|
remoteTrack.track.PayloadType(),
|
|
remoteTrack.track.SSRC(),
|
|
remoteTrack.track.ID(),
|
|
remoteTrack.track.Label(),
|
|
)
|
|
if err != nil {
|
|
return nil, nil, err
|
|
}
|
|
|
|
s, err := conn.pc.AddTrack(local)
|
|
if err != nil {
|
|
return nil, nil, err
|
|
}
|
|
|
|
track := &downTrack{
|
|
track: local,
|
|
remote: remoteTrack,
|
|
maxLossBitrate: new(bitrate),
|
|
maxREMBBitrate: new(bitrate),
|
|
stats: new(receiverStats),
|
|
rate: estimator.New(time.Second),
|
|
}
|
|
conn.tracks = append(conn.tracks, track)
|
|
remoteTrack.addLocal(track)
|
|
|
|
go rtcpDownListener(conn, track, s)
|
|
|
|
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)
|
|
}
|
|
|
|
func rtcpDownListener(conn *downConnection, track *downTrack, s *webrtc.RTPSender) {
|
|
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:
|
|
err := conn.remote.sendPLI(track.remote)
|
|
if err != nil {
|
|
log.Printf("sendPLI: %v", err)
|
|
}
|
|
case *rtcp.ReceiverEstimatedMaximumBitrate:
|
|
track.maxREMBBitrate.Set(
|
|
p.Bitrate, mono.Microseconds(),
|
|
)
|
|
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(),
|
|
)
|
|
bitrate := track.rate.Estimate()
|
|
if uint64(bitrate) < maxBitrate {
|
|
sendRecovery(p, track)
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
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()
|
|
|
|
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
|
|
}
|
|
if rate > bitrate {
|
|
rate = bitrate
|
|
}
|
|
}
|
|
track.maxBitrate = rate
|
|
}
|
|
}
|
|
|
|
func (up *upConnection) sendPLI(track *upTrack) error {
|
|
if !track.hasRtcpFb("nack", "pli") {
|
|
return nil
|
|
}
|
|
last := atomic.LoadUint64(&track.lastPLI)
|
|
now := mono.Microseconds()
|
|
if now >= last && now-last < 200000 {
|
|
return nil
|
|
}
|
|
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 sendREMB(pc *webrtc.PeerConnection, ssrc uint32, bitrate uint64) error {
|
|
return pc.WriteRTCP([]rtcp.Packet{
|
|
&rtcp.ReceiverEstimatedMaximumBitrate{
|
|
Bitrate: bitrate,
|
|
SSRCs: []uint32{ssrc},
|
|
},
|
|
})
|
|
}
|
|
|
|
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)
|
|
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
|
|
}
|
|
track.rate.Add(uint32(len(raw)))
|
|
}
|
|
}
|
|
}
|
|
|
|
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
|
|
}
|
|
up.trackCount = n
|
|
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)
|
|
}
|
|
|
|
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
|
|
|
|
for _, cc := range c.group.getClients(c) {
|
|
cc.action(pushTracksAction{c})
|
|
}
|
|
|
|
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 clientLoop(c *client, conn *websocket.Conn) error {
|
|
read := make(chan interface{}, 1)
|
|
go clientReader(conn, read, c.done)
|
|
|
|
defer func() {
|
|
c.setRequested(false, false)
|
|
if c.up != nil {
|
|
for id := range c.up {
|
|
delUpConn(c, id)
|
|
}
|
|
}
|
|
}()
|
|
|
|
c.write(clientMessage{
|
|
Type: "permissions",
|
|
Permissions: c.permissions,
|
|
})
|
|
|
|
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()
|
|
|
|
ticker := time.NewTicker(time.Second)
|
|
defer ticker.Stop()
|
|
slowTicker := time.NewTicker(10 * time.Second)
|
|
defer slowTicker.Stop()
|
|
|
|
for {
|
|
select {
|
|
case m, ok := <-read:
|
|
if !ok {
|
|
return errors.New("reader died")
|
|
}
|
|
switch m := m.(type) {
|
|
case clientMessage:
|
|
readTime = time.Now()
|
|
err := handleClientMessage(c, m)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
case error:
|
|
return m
|
|
}
|
|
case a := <-c.actionCh:
|
|
switch a := a.(type) {
|
|
case addTrackAction:
|
|
var down *downConnection
|
|
var err error
|
|
if c.requested(a.track.track.Kind()) {
|
|
down, _, err = addDownTrack(
|
|
c, a.remote.id, a.track,
|
|
a.remote)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
} else {
|
|
down = getDownConn(c, a.remote.id)
|
|
}
|
|
if a.done && down != nil {
|
|
err = negotiate(c, a.remote.id, down.pc)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
}
|
|
case delConnAction:
|
|
found := delDownConn(c, a.id)
|
|
if found {
|
|
c.write(clientMessage{
|
|
Type: "close",
|
|
Id: a.id,
|
|
})
|
|
}
|
|
case addLabelAction:
|
|
c.write(clientMessage{
|
|
Type: "label",
|
|
Id: a.id,
|
|
Value: a.label,
|
|
})
|
|
case pushTracksAction:
|
|
for _, u := range c.up {
|
|
var done bool
|
|
for i, t := range u.tracks {
|
|
done = i >= u.trackCount-1
|
|
a.c.action(addTrackAction{
|
|
t, u, done,
|
|
})
|
|
}
|
|
if done && u.label != "" {
|
|
a.c.action(addLabelAction{
|
|
u.id, u.label,
|
|
})
|
|
|
|
}
|
|
}
|
|
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:
|
|
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)
|
|
}
|
|
}
|
|
case kickAction:
|
|
return userError("you have been kicked")
|
|
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
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
func handleClientMessage(c *client, m clientMessage) error {
|
|
switch m.Type {
|
|
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
|
|
}
|
|
case "offer":
|
|
if !c.permissions.Present {
|
|
c.write(clientMessage{
|
|
Type: "abort",
|
|
Id: m.Id,
|
|
})
|
|
return c.error(userError("not authorised"))
|
|
}
|
|
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)
|
|
}
|
|
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":
|
|
c.group.addToChatHistory(m.Id, m.Username, m.Value, m.Me)
|
|
clients := c.group.getClients(c)
|
|
for _, cc := range clients {
|
|
cc.write(m)
|
|
}
|
|
case "clearchat":
|
|
c.group.clearChatHistory()
|
|
m := clientMessage{Type: "clearchat"}
|
|
clients := c.group.getClients(nil)
|
|
for _, cc := range clients {
|
|
cc.write(m)
|
|
}
|
|
case "op", "unop", "present", "unpresent":
|
|
if !c.permissions.Op {
|
|
c.error(userError("not authorised"))
|
|
return nil
|
|
}
|
|
err := setPermission(c.group, m.Id, m.Type)
|
|
if err != nil {
|
|
return c.error(err)
|
|
}
|
|
case "kick":
|
|
if !c.permissions.Op {
|
|
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",
|
|
})
|
|
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()
|
|
for _, u := range c.up {
|
|
updateUpBitrate(u, maxVideoRate)
|
|
for _, t := range u.tracks {
|
|
if !t.hasRtcpFb("goog-remb", "") {
|
|
continue
|
|
}
|
|
bitrate := t.maxBitrate
|
|
if bitrate == ^uint64(0) {
|
|
continue
|
|
}
|
|
rembs = append(rembs,
|
|
remb{u.pc, t.track.SSRC(), bitrate})
|
|
}
|
|
}
|
|
c.mu.Unlock()
|
|
|
|
for _, r := range rembs {
|
|
err := sendREMB(r.pc, r.ssrc, r.bitrate)
|
|
if err != nil {
|
|
log.Printf("sendREMB: %v", err)
|
|
}
|
|
}
|
|
}
|
|
|
|
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
|
|
}
|
|
}
|
|
}
|