143 lines
3.6 KiB
Go
143 lines
3.6 KiB
Go
package webrtc
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import (
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"bufio"
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"fmt"
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"io"
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"log"
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"net"
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"os/exec"
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"github.com/pion/rtp"
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"gitlab.crans.org/nounous/ghostream/stream/srt"
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)
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func ingestFrom(inputChannel chan srt.Packet) {
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// FIXME Clean code
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var ffmpeg *exec.Cmd
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var ffmpegInput io.WriteCloser
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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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// From https://github.com/pion/webrtc/blob/master/examples/rtp-to-webrtc/main.go
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// Open a UDP Listener for RTP Packets on port 5004
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videoListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
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if err != nil {
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panic(err)
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}
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audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5005})
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if err != nil {
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panic(err)
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}
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defer func() {
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if err = videoListener.Close(); err != nil {
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panic(err)
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}
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if err = audioListener.Close(); err != nil {
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panic(err)
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}
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}()
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ffmpeg = exec.Command("ffmpeg", "-re", "-i", "pipe:0",
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"-an", "-vcodec", "libvpx", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
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"-f", "rtp", "rtp://127.0.0.1:5004",
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"-vn", "-acodec", "libopus", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
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"-f", "rtp", "rtp://127.0.0.1:5005")
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fmt.Println("Waiting for RTP Packets, please run GStreamer or ffmpeg now")
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input, err := ffmpeg.StdinPipe()
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if err != nil {
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panic(err)
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}
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ffmpegInput = input
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errOutput, err := ffmpeg.StderrPipe()
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if err != nil {
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panic(err)
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}
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if err := ffmpeg.Start(); err != nil {
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panic(err)
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}
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// Receive video
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go func() {
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for {
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inboundRTPPacket := make([]byte, 1500) // UDP MTU
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n, _, err := videoListener.ReadFromUDP(inboundRTPPacket)
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if err != nil {
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panic(err)
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}
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packet := &rtp.Packet{}
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if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
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panic(err)
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}
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log.Printf("[Video] %s", packet)
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// Write RTP packet to all video tracks
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// Adapt payload and SSRC to match destination
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for _, videoTrack := range videoTracks {
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packet.Header.PayloadType = videoTrack.PayloadType()
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packet.Header.SSRC = videoTrack.SSRC()
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if writeErr := videoTrack.WriteRTP(packet); writeErr != nil {
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panic(err)
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}
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}
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}
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}()
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// Receive audio
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go func() {
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for {
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inboundRTPPacket := make([]byte, 1500) // UDP MTU
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n, _, err := audioListener.ReadFromUDP(inboundRTPPacket)
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if err != nil {
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panic(err)
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}
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packet := &rtp.Packet{}
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if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
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panic(err)
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}
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log.Printf("[Audio] %s", packet)
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for _, audioTrack := range audioTracks {
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packet.Header.PayloadType = audioTrack.PayloadType()
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packet.Header.SSRC = audioTrack.SSRC()
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if writeErr := audioTrack.WriteRTP(packet); writeErr != nil {
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panic(err)
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}
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}
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}
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}()
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go func() {
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scanner := bufio.NewScanner(errOutput)
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for scanner.Scan() {
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log.Printf("[WEBRTC FFMPEG %s] %s", "demo", scanner.Text())
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}
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}()
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break
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case "sendData":
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// FIXME send to stream packet.StreamName
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_, err := ffmpegInput.Write(packet.Data)
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if err != nil {
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panic(err)
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}
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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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