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https://gitlab.crans.org/nounous/ghostream.git
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Send SRT stream to WebRTC package
This commit is contained in:
parent
3bf27fb0b1
commit
ad934516a9
7
main.go
7
main.go
@ -104,15 +104,16 @@ func main() {
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remoteSdpChan := make(chan webrtc.SessionDescription)
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remoteSdpChan := make(chan webrtc.SessionDescription)
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localSdpChan := make(chan webrtc.SessionDescription)
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localSdpChan := make(chan webrtc.SessionDescription)
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// SRT channel for forwarding
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// SRT channel for forwarding and webrtc
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forwardingChannel := make(chan srt.Packet, 65536)
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forwardingChannel := make(chan srt.Packet, 65536)
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webrtcChannel := make(chan srt.Packet, 65536)
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// Start stream, web and monitoring server, and stream forwarding
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// Start stream, web and monitoring server, and stream forwarding
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go forwarding.Serve(cfg.Forwarding, forwardingChannel)
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go forwarding.Serve(cfg.Forwarding, forwardingChannel)
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go monitoring.Serve(&cfg.Monitoring)
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go monitoring.Serve(&cfg.Monitoring)
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go srt.Serve(&cfg.Srt, authBackend, forwardingChannel)
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go srt.Serve(&cfg.Srt, authBackend, forwardingChannel, webrtcChannel)
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go web.Serve(remoteSdpChan, localSdpChan, &cfg.Web)
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go web.Serve(remoteSdpChan, localSdpChan, &cfg.Web)
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go webrtc.Serve(remoteSdpChan, localSdpChan, &cfg.WebRTC)
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go webrtc.Serve(remoteSdpChan, localSdpChan, webrtcChannel, &cfg.WebRTC)
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// Wait for routines
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// Wait for routines
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select {}
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select {}
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@ -6,7 +6,7 @@ import (
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"github.com/haivision/srtgo"
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"github.com/haivision/srtgo"
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)
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)
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func handleStreamer(s *srtgo.SrtSocket, name string, clientDataChannels *[]chan Packet, forwardingChannel chan Packet) {
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func handleStreamer(s *srtgo.SrtSocket, name string, clientDataChannels *[]chan Packet, forwardingChannel, webrtcChannel chan Packet) {
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log.Printf("New SRT streamer for stream %s", name)
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log.Printf("New SRT streamer for stream %s", name)
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// Create a new buffer
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// Create a new buffer
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@ -15,6 +15,7 @@ func handleStreamer(s *srtgo.SrtSocket, name string, clientDataChannels *[]chan
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// Setup stream forwarding
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// Setup stream forwarding
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forwardingChannel <- Packet{StreamName: name, PacketType: "register", Data: nil}
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forwardingChannel <- Packet{StreamName: name, PacketType: "register", Data: nil}
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webrtcChannel <- Packet{StreamName: name, PacketType: "register", Data: nil}
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// Read RTP packets forever and send them to the WebRTC Client
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// Read RTP packets forever and send them to the WebRTC Client
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for {
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for {
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@ -36,12 +37,14 @@ func handleStreamer(s *srtgo.SrtSocket, name string, clientDataChannels *[]chan
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data := make([]byte, n)
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data := make([]byte, n)
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copy(data, buff[:n])
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copy(data, buff[:n])
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forwardingChannel <- Packet{StreamName: name, PacketType: "sendData", Data: data}
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forwardingChannel <- Packet{StreamName: name, PacketType: "sendData", Data: data}
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webrtcChannel <- Packet{StreamName: name, PacketType: "sendData", Data: data}
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for _, dataChannel := range *clientDataChannels {
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for _, dataChannel := range *clientDataChannels {
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dataChannel <- Packet{StreamName: name, PacketType: "sendData", Data: data}
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dataChannel <- Packet{StreamName: name, PacketType: "sendData", Data: data}
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}
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}
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}
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}
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forwardingChannel <- Packet{StreamName: name, PacketType: "close", Data: nil}
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forwardingChannel <- Packet{StreamName: name, PacketType: "close", Data: nil}
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webrtcChannel <- Packet{StreamName: name, PacketType: "close", Data: nil}
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}
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}
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func handleViewer(s *srtgo.SrtSocket, name string, dataChannel chan Packet, dataChannels *[]chan Packet) {
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func handleViewer(s *srtgo.SrtSocket, name string, dataChannel chan Packet, dataChannels *[]chan Packet) {
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@ -43,7 +43,7 @@ func splitHostPort(hostport string) (string, uint16) {
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}
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}
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// Serve SRT server
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// Serve SRT server
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func Serve(cfg *Options, authBackend auth.Backend, forwardingChannel chan Packet) {
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func Serve(cfg *Options, authBackend auth.Backend, forwardingChannel, webrtcChannel chan Packet) {
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// Start SRT in listening mode
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// Start SRT in listening mode
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log.Printf("SRT server listening on %s", cfg.ListenAddress)
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log.Printf("SRT server listening on %s", cfg.ListenAddress)
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host, port := splitHostPort(cfg.ListenAddress)
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host, port := splitHostPort(cfg.ListenAddress)
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@ -82,7 +82,7 @@ func Serve(cfg *Options, authBackend auth.Backend, forwardingChannel chan Packet
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}
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}
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}
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}
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go handleStreamer(s, name, &clientDataChannels, forwardingChannel)
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go handleStreamer(s, name, &clientDataChannels, forwardingChannel, webrtcChannel)
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} else {
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} else {
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// password was not provided so it is a viewer
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// password was not provided so it is a viewer
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name := split[0]
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name := split[0]
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@ -13,6 +13,7 @@ import (
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"github.com/pion/webrtc/v3/pkg/media/ivfreader"
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"github.com/pion/webrtc/v3/pkg/media/ivfreader"
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"github.com/pion/webrtc/v3/pkg/media/oggreader"
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"github.com/pion/webrtc/v3/pkg/media/oggreader"
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"gitlab.crans.org/nounous/ghostream/internal/monitoring"
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"gitlab.crans.org/nounous/ghostream/internal/monitoring"
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"gitlab.crans.org/nounous/ghostream/stream/srt"
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)
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)
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// Options holds web package configuration
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// Options holds web package configuration
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@ -242,10 +243,37 @@ func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecNa
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panic(fmt.Sprintf("Remote peer does not support %s", codecName))
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panic(fmt.Sprintf("Remote peer does not support %s", codecName))
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}
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}
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func waitForPackets(inputChannel chan srt.Packet) {
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for {
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var err error = nil
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packet := <-inputChannel
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switch packet.PacketType {
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case "register":
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log.Printf("WebRTC RegisterStream %s", packet.StreamName)
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break
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case "sendData":
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log.Printf("WebRTC SendPacket %s", packet.StreamName)
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// packet.Data
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break
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case "close":
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log.Printf("WebRTC CloseConnection %s", packet.StreamName)
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break
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default:
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log.Println("Unknown SRT packet type:", packet.PacketType)
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break
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}
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if err != nil {
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log.Printf("Error occured while receiving SRT packet of type %s: %s", packet.PacketType, err)
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}
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}
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}
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// Serve WebRTC media streaming server
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// Serve WebRTC media streaming server
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func Serve(remoteSdpChan, localSdpChan chan webrtc.SessionDescription, cfg *Options) {
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func Serve(remoteSdpChan, localSdpChan chan webrtc.SessionDescription, inputChannel chan srt.Packet, cfg *Options) {
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log.Printf("WebRTC server using UDP from %d to %d", cfg.MinPortUDP, cfg.MaxPortUDP)
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log.Printf("WebRTC server using UDP from %d to %d", cfg.MinPortUDP, cfg.MaxPortUDP)
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// FIXME: use data from inputChannel
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go waitForPackets(inputChannel)
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go playVideo()
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go playVideo()
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go playAudio()
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go playAudio()
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