// Package webrtc provides the backend to simulate a WebRTC client to send stream package webrtc import ( "bufio" "io" "log" "net" "os/exec" "github.com/pion/rtp" "github.com/pion/webrtc/v3" "gitlab.crans.org/nounous/ghostream/stream/srt" ) func ingestFrom(inputChannel chan srt.Packet) { // FIXME Clean code var ffmpeg *exec.Cmd var ffmpegInput io.WriteCloser for { var err error = nil srtPacket := <-inputChannel switch srtPacket.PacketType { case "register": log.Printf("WebRTC RegisterStream %s", srtPacket.StreamName) // Open a UDP Listener for RTP Packets on port 5004 videoListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004}) if err != nil { log.Printf("Faited to open UDP listener %s", err) return } audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5005}) if err != nil { log.Printf("Faited to open UDP listener %s", err) return } defer func() { if err = videoListener.Close(); err != nil { log.Printf("Faited to close UDP listener %s", err) } if err = audioListener.Close(); err != nil { log.Printf("Faited to close UDP listener %s", err) } }() ffmpeg = exec.Command("ffmpeg", "-hide_banner", "-loglevel", "error", "-re", "-i", "pipe:0", "-an", "-vcodec", "libvpx", "-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1", "-f", "rtp", "rtp://127.0.0.1:5004", "-vn", "-acodec", "libopus", "-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1", "-f", "rtp", "rtp://127.0.0.1:5005") input, err := ffmpeg.StdinPipe() if err != nil { panic(err) } ffmpegInput = input errOutput, err := ffmpeg.StderrPipe() if err != nil { panic(err) } if err := ffmpeg.Start(); err != nil { panic(err) } // Receive video go func() { for { inboundRTPPacket := make([]byte, 1500) // UDP MTU n, _, err := videoListener.ReadFromUDP(inboundRTPPacket) if err != nil { log.Printf("Failed to read from UDP: %s", err) continue } packet := &rtp.Packet{} if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil { log.Printf("Failed to unmarshal RTP srtPacket: %s", err) continue } if videoTracks[srtPacket.StreamName] == nil { videoTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0) } // Write RTP srtPacket to all video tracks // Adapt payload and SSRC to match destination for _, videoTrack := range videoTracks[srtPacket.StreamName] { packet.Header.PayloadType = videoTrack.PayloadType() packet.Header.SSRC = videoTrack.SSRC() if writeErr := videoTrack.WriteRTP(packet); writeErr != nil { log.Printf("Failed to write to video track: %s", err) continue } } } }() // Receive audio go func() { for { inboundRTPPacket := make([]byte, 1500) // UDP MTU n, _, err := audioListener.ReadFromUDP(inboundRTPPacket) if err != nil { log.Printf("Failed to read from UDP: %s", err) continue } packet := &rtp.Packet{} if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil { log.Printf("Failed to unmarshal RTP srtPacket: %s", err) continue } if audioTracks[srtPacket.StreamName] == nil { audioTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0) } // Write RTP srtPacket to all audio tracks // Adapt payload and SSRC to match destination for _, audioTrack := range audioTracks[srtPacket.StreamName] { packet.Header.PayloadType = audioTrack.PayloadType() packet.Header.SSRC = audioTrack.SSRC() if writeErr := audioTrack.WriteRTP(packet); writeErr != nil { log.Printf("Failed to write to audio track: %s", err) continue } } } }() go func() { scanner := bufio.NewScanner(errOutput) for scanner.Scan() { log.Printf("[WEBRTC FFMPEG %s] %s", "demo", scanner.Text()) } }() break case "sendData": // FIXME send to stream srtPacket.StreamName if _, err := ffmpegInput.Write(srtPacket.Data); err != nil { log.Printf("Failed to write data to ffmpeg input: %s", err) } break case "close": log.Printf("WebRTC CloseConnection %s", srtPacket.StreamName) break default: log.Println("Unknown SRT srtPacket type:", srtPacket.PacketType) break } if err != nil { log.Printf("Error occured while receiving SRT srtPacket of type %s: %s", srtPacket.PacketType, err) } } }