More code for SRT handling

This commit is contained in:
Alexandre Iooss 2020-09-28 16:32:35 +02:00
parent a552b48730
commit 92f44c981d
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GPG Key ID: 6C79278F3FCDCC02
2 changed files with 42 additions and 16 deletions

1
go.mod
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@ -5,6 +5,7 @@ go 1.13
require ( require (
github.com/go-ldap/ldap/v3 v3.2.3 github.com/go-ldap/ldap/v3 v3.2.3
github.com/haivision/srtgo v0.0.0-20200731151239-e00427ae473a github.com/haivision/srtgo v0.0.0-20200731151239-e00427ae473a
github.com/pion/rtp v1.6.0
github.com/pion/webrtc/v3 v3.0.0-beta.5 github.com/pion/webrtc/v3 v3.0.0-beta.5
github.com/prometheus/client_golang v1.7.1 github.com/prometheus/client_golang v1.7.1
github.com/spf13/viper v1.7.1 github.com/spf13/viper v1.7.1

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@ -4,6 +4,7 @@ import (
"log" "log"
"github.com/haivision/srtgo" "github.com/haivision/srtgo"
"github.com/pion/rtp"
) )
// Options holds web package configuration // Options holds web package configuration
@ -16,7 +17,7 @@ func Serve(cfg *Options) {
log.Printf("SRT server listening on %s", cfg.ListenAddress) log.Printf("SRT server listening on %s", cfg.ListenAddress)
options := make(map[string]string) options := make(map[string]string)
options["transtype"] = "file" options["transtype"] = "live"
// FIXME: cfg.ListenAddress -> host and port // FIXME: cfg.ListenAddress -> host and port
sck := srtgo.NewSrtSocket("0.0.0.0", 9710, options) sck := srtgo.NewSrtSocket("0.0.0.0", 9710, options)
@ -25,24 +26,48 @@ func Serve(cfg *Options) {
for { for {
s, err := sck.Accept() s, err := sck.Accept()
if err != nil { if err != nil {
log.Println("Error occured while accepting request:", err) log.Println("Error occurred while accepting request:", err)
continue continue
} }
go func(s *srtgo.SrtSocket) {
buff := make([]byte, 2048) buff := make([]byte, 2048)
for {
n, err := s.Read(buff, 10000) n, err := s.Read(buff, 10000)
if err != nil { if err != nil {
log.Println("Error occured while reading SRT socket:", err) log.Println("Error occurred while reading SRT socket:", err)
break break
} }
if n == 0 { if n == 0 {
// End of stream // End of stream
break break
} }
// Unmarshal the incoming packet
packet := &rtp.Packet{}
if err = packet.Unmarshal(buff[:n]); err != nil {
log.Println("Error occured while unmarshaling SRT:", err)
break
}
// videoTrack, err := peerConnection.NewTrack(payloadType, packet.SSRC, "video", "pion")
// Read RTP packets forever and send them to the WebRTC Client
for {
n, err := s.Read(buff, 10000)
if err != nil {
log.Println("Error occured while reading SRT socket:", err)
break
}
log.Printf("Received %d bytes", n) log.Printf("Received %d bytes", n)
packet := &rtp.Packet{}
if err := packet.Unmarshal(buff[:n]); err != nil {
panic(err)
}
payloadType := uint8(22) // FIXME put vp8 payload
packet.Header.PayloadType = payloadType
//err := videoTrack.WriteRTP(packet)
} }
}(s)
} }
} }