Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
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package sourcertsp
import (
"net/url"
"sync"
"sync/atomic"
"time"
"github.com/aler9/gortsplib"
"github.com/aler9/rtsp-simple-server/stats"
)
const (
retryInterval = 5 * time.Second
)
type Parent interface {
Log(string, ...interface{})
OnSourceReady(gortsplib.Tracks)
OnSourceNotReady()
OnFrame(int, gortsplib.StreamType, []byte)
}
type Source struct {
ur string
proto gortsplib.StreamProtocol
readTimeout time.Duration
writeTimeout time.Duration
state bool
stats *stats.Stats
parent Parent
innerState bool
// in
innerTerminate chan struct{}
innerDone chan struct{}
stateChange chan bool
terminate chan struct{}
// out
done chan struct{}
}
func New(ur string,
proto gortsplib.StreamProtocol,
readTimeout time.Duration,
writeTimeout time.Duration,
state bool,
stats *stats.Stats,
parent Parent) *Source {
s := &Source{
ur: ur,
proto: proto,
readTimeout: readTimeout,
writeTimeout: writeTimeout,
state: state,
stats: stats,
parent: parent,
stateChange: make(chan bool),
terminate: make(chan struct{}),
done: make(chan struct{}),
}
atomic.AddInt64(s.stats.CountSourcesRtsp, +1)
go s.run()
s.SetRunning(s.state)
return s
}
func (s *Source) Close() {
close(s.terminate)
<-s.done
}
func (s *Source) IsSource() {}
func (s *Source) IsRunning() bool {
return s.state
}
func (s *Source) SetRunning(state bool) {
s.state = state
s.stateChange <- s.state
}
func (s *Source) run() {
defer close(s.done)
outer:
for {
select {
case state := <-s.stateChange:
if state {
if !s.innerState {
atomic.AddInt64(s.stats.CountSourcesRtspRunning, +1)
s.innerState = true
s.innerTerminate = make(chan struct{})
s.innerDone = make(chan struct{})
go s.runInner()
}
} else {
if s.innerState {
atomic.AddInt64(s.stats.CountSourcesRtspRunning, -1)
close(s.innerTerminate)
<-s.innerDone
s.innerState = false
}
}
case <-s.terminate:
break outer
}
}
if s.innerState {
atomic.AddInt64(s.stats.CountSourcesRtspRunning, -1)
close(s.innerTerminate)
<-s.innerDone
}
close(s.stateChange)
}
func (s *Source) runInner() {
defer close(s.innerDone)
for {
ok := func() bool {
ok := s.runInnerInner()
if !ok {
return false
}
t := time.NewTimer(retryInterval)
defer t.Stop()
select {
case <-t.C:
return true
case <-s.innerTerminate:
return false
}
}()
if !ok {
break
}
}
}
func (s *Source) runInnerInner() bool {
s.parent.Log("connecting to rtsp source")
u, _ := url.Parse(s.ur)
var conn *gortsplib.ConnClient
var err error
dialDone := make(chan struct{}, 1)
go func() {
defer close(dialDone)
conn, err = gortsplib.NewConnClient(gortsplib.ConnClientConf{
Host: u.Host,
ReadTimeout: s.readTimeout,
WriteTimeout: s.writeTimeout,
ReadBufferCount: 2,
})
}()
select {
case <-s.innerTerminate:
return false
case <-dialDone:
}
if err != nil {
s.parent.Log("rtsp source ERR: %s", err)
return true
}
_, err = conn.Options(u)
if err != nil {
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
return true
}
tracks, _, err := conn.Describe(u)
if err != nil {
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
return true
}
if s.proto == gortsplib.StreamProtocolUDP {
return s.runUDP(u, conn, tracks)
} else {
return s.runTCP(u, conn, tracks)
}
}
func (s *Source) runUDP(u *url.URL, conn *gortsplib.ConnClient, tracks gortsplib.Tracks) bool {
for _, track := range tracks {
_, err := conn.SetupUDP(u, gortsplib.TransportModePlay, track, 0, 0)
if err != nil {
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
return true
}
}
_, err := conn.Play(u)
if err != nil {
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
return true
}
s.parent.OnSourceReady(tracks)
s.parent.Log("rtsp source ready")
var wg sync.WaitGroup
// receive RTP packets
for trackId := range tracks {
wg.Add(1)
go func(trackId int) {
defer wg.Done()
for {
buf, err := conn.ReadFrameUDP(trackId, gortsplib.StreamTypeRtp)
if err != nil {
break
}
s.parent.OnFrame(trackId, gortsplib.StreamTypeRtp, buf)
}
}(trackId)
}
// receive RTCP packets
for trackId := range tracks {
wg.Add(1)
go func(trackId int) {
defer wg.Done()
for {
buf, err := conn.ReadFrameUDP(trackId, gortsplib.StreamTypeRtcp)
if err != nil {
break
}
s.parent.OnFrame(trackId, gortsplib.StreamTypeRtcp, buf)
}
}(trackId)
}
tcpConnDone := make(chan error)
go func() {
tcpConnDone <- conn.LoopUDP()
}()
var ret bool
outer:
for {
select {
case <-s.innerTerminate:
conn.Close()
<-tcpConnDone
ret = false
break outer
case err := <-tcpConnDone:
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
ret = true
break outer
}
}
wg.Wait()
s.parent.OnSourceNotReady()
return ret
}
func (s *Source) runTCP(u *url.URL, conn *gortsplib.ConnClient, tracks gortsplib.Tracks) bool {
for _, track := range tracks {
_, err := conn.SetupTCP(u, gortsplib.TransportModePlay, track)
if err != nil {
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
return true
}
}
_, err := conn.Play(u)
if err != nil {
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
return true
}
s.parent.OnSourceReady(tracks)
s.parent.Log("rtsp source ready")
tcpConnDone := make(chan error)
go func() {
for {
trackId, streamType, content, err := conn.ReadFrameTCP()
if err != nil {
tcpConnDone <- err
return
}
s.parent.OnFrame(trackId, streamType, content)
}
}()
var ret bool
outer:
for {
select {
case <-s.innerTerminate:
conn.Close()
<-tcpConnDone
ret = false
break outer
case err := <-tcpConnDone:
conn.Close()
s.parent.Log("rtsp source ERR: %s", err)
ret = true
break outer
}
}
s.parent.OnSourceNotReady()
return ret
}