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 (
"sync"
"sync/atomic"
"time"
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/base"
"github.com/aler9/rtsp-simple-server/internal/logger"
"github.com/aler9/rtsp-simple-server/internal/stats"
)
const (
retryPause = 5 * time.Second
)
// Parent is implemented by path.Path.
type Parent interface {
Log(logger.Level, string, ...interface{})
OnSourceSetReady(gortsplib.Tracks)
OnSourceSetNotReady()
OnFrame(int, gortsplib.StreamType, []byte)
}
// Source is a RTSP source.
type Source struct {
ur string
proto *gortsplib.StreamProtocol
readTimeout time.Duration
writeTimeout time.Duration
readBufferCount uint64
wg *sync.WaitGroup
stats *stats.Stats
parent Parent
// in
terminate chan struct{}
}
// New allocates a Source.
func New(ur string,
proto *gortsplib.StreamProtocol,
readTimeout time.Duration,
writeTimeout time.Duration,
readBufferCount uint64,
wg *sync.WaitGroup,
stats *stats.Stats,
parent Parent) *Source {
s := &Source{
ur: ur,
proto: proto,
readTimeout: readTimeout,
writeTimeout: writeTimeout,
readBufferCount: readBufferCount,
wg: wg,
stats: stats,
parent: parent,
terminate: make(chan struct{}),
}
atomic.AddInt64(s.stats.CountSourcesRtsp, +1)
s.log(logger.Info, "started")
s.wg.Add(1)
go s.run()
return s
}
// Close closes a Source.
func (s *Source) Close() {
atomic.AddInt64(s.stats.CountSourcesRtsp, -1)
s.log(logger.Info, "stopped")
close(s.terminate)
}
// IsSource implements path.source.
func (s *Source) IsSource() {}
// IsSourceExternal implements path.sourceExternal.
func (s *Source) IsSourceExternal() {}
func (s *Source) log(level logger.Level, format string, args ...interface{}) {
s.parent.Log(level, "[rtsp source] "+format, args...)
}
func (s *Source) run() {
defer s.wg.Done()
for {
ok := func() bool {
ok := s.runInner()
if !ok {
return false
}
t := time.NewTimer(retryPause)
defer t.Stop()
select {
case <-t.C:
return true
case <-s.terminate:
return false
}
}()
if !ok {
break
}
}
}
func (s *Source) runInner() bool {
s.log(logger.Info, "connecting")
var conn *gortsplib.ClientConn
var err error
dialDone := make(chan struct{}, 1)
go func() {
defer close(dialDone)
conf := gortsplib.ClientConf{
StreamProtocol: s.proto,
ReadTimeout: s.readTimeout,
WriteTimeout: s.writeTimeout,
ReadBufferCount: s.readBufferCount,
OnRequest: func(req *base.Request) {
s.log(logger.Debug, "c->s %v", req)
},
OnResponse: func(res *base.Response) {
s.log(logger.Debug, "s->c %v", res)
},
}
conn, err = conf.DialRead(s.ur)
}()
select {
case <-s.terminate:
return false
case <-dialDone:
}
if err != nil {
s.log(logger.Info, "ERR: %s", err)
return true
}
tracks := conn.Tracks()
s.log(logger.Info, "ready")
s.parent.OnSourceSetReady(tracks)
defer s.parent.OnSourceSetNotReady()
done := conn.ReadFrames(func(trackID int, streamType gortsplib.StreamType, payload []byte) {
s.parent.OnFrame(trackID, streamType, payload)
})
for {
select {
case <-s.terminate:
conn.Close()
<-done
return false
case err := <-done:
conn.Close()
s.log(logger.Info, "ERR: %s", err)
return true
}
}
}