Separate the WebRTC stream subroutine in a dedicated subroutine
This commit is contained in:
Yohann D'ANELLO
parent
defba52569
commit
32f877508d
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@ -15,156 +15,37 @@ import (
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"gitlab.crans.org/nounous/ghostream/stream/srt"
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)
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var (
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ffmpeg = make(map[string]*exec.Cmd)
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ffmpegInput = make(map[string]io.WriteCloser)
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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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srtPacket := <-inputChannel
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log.Println(len(inputChannel))
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switch srtPacket.PacketType {
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case "register":
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log.Printf("WebRTC RegisterStream %s", srtPacket.StreamName)
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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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log.Printf("Faited to open UDP listener %s", err)
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return
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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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log.Printf("Faited to open UDP listener %s", err)
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return
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}
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defer func() {
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if err = videoListener.Close(); err != nil {
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log.Printf("Faited to close UDP listener %s", err)
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}
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if err = audioListener.Close(); err != nil {
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log.Printf("Faited to close UDP listener %s", err)
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}
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}()
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ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-re", "-i", "pipe:0",
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"-an", "-vcodec", "libvpx", "-crf", "10", "-cpu-used", "5", "-b:v", "6000k", "-maxrate", "8000k", "-bufsize", "12000k", // TODO Change bitrate when changing quality
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"-qmin", "10", "-qmax", "42", "-threads", "4", "-deadline", "1", "-error-resilient", "1",
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"-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", "-qmin", "10", "-qmax", "42", "-error-resilient", "1", "-auto-alt-ref", "1",
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"-f", "rtp", "rtp://127.0.0.1:5005"}
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// Export stream to ascii art
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if telnet.Cfg.Enabled {
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ffmpegArgs = append(ffmpegArgs,
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"-an", "-f", "rawvideo", "-vf", fmt.Sprintf("scale=%dx%d", telnet.Cfg.Width, telnet.Cfg.Height), "-pix_fmt", "gray", "pipe:1")
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}
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ffmpeg = exec.Command("ffmpeg", ffmpegArgs...)
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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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// Receive raw video output and convert it to ASCII art, then forward it TCP
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if telnet.Cfg.Enabled {
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output, err := ffmpeg.StdoutPipe()
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if err != nil {
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panic(err)
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}
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go telnet.StartASCIIArtStream(srtPacket.StreamName, output)
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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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log.Printf("Failed to read from UDP: %s", err)
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continue
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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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log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
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continue
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}
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if videoTracks[srtPacket.StreamName] == nil {
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videoTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
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}
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// Write RTP srtPacket to all video tracks
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// Adapt payload and SSRC to match destination
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for _, videoTrack := range videoTracks[srtPacket.StreamName] {
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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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log.Printf("Failed to write to video track: %s", err)
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continue
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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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log.Printf("Failed to read from UDP: %s", err)
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continue
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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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log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
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continue
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}
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if audioTracks[srtPacket.StreamName] == nil {
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audioTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
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}
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// Write RTP srtPacket to all audio tracks
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// Adapt payload and SSRC to match destination
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for _, audioTrack := range audioTracks[srtPacket.StreamName] {
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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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log.Printf("Failed to write to audio track: %s", err)
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continue
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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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go registerStream(&srtPacket)
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break
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case "sendData":
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if _, ok := ffmpegInput[srtPacket.StreamName]; !ok {
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break
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}
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// FIXME send to stream srtPacket.StreamName
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if _, err := ffmpegInput.Write(srtPacket.Data); err != nil {
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if _, err := ffmpegInput[srtPacket.StreamName].Write(srtPacket.Data); err != nil {
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log.Printf("Failed to write data to ffmpeg input: %s", 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", srtPacket.StreamName)
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_ = ffmpeg[srtPacket.StreamName].Process.Kill()
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_ = ffmpegInput[srtPacket.StreamName].Close()
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delete(ffmpeg, srtPacket.StreamName)
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delete(ffmpegInput, srtPacket.StreamName)
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break
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default:
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log.Println("Unknown SRT srtPacket type:", srtPacket.PacketType)
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@ -175,3 +56,140 @@ func ingestFrom(inputChannel chan srt.Packet) {
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}
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}
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}
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func registerStream(srtPacket *srt.Packet) {
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log.Printf("WebRTC RegisterStream %s", srtPacket.StreamName)
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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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log.Printf("Faited to open UDP listener %s", err)
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return
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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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log.Printf("Faited to open UDP listener %s", err)
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return
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}
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// FIXME Close UDP listeners at the end of the stream, not the end of the routine
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/* defer func() {
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if err = videoListener.Close(); err != nil {
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log.Printf("Faited to close UDP listener %s", err)
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}
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if err = audioListener.Close(); err != nil {
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log.Printf("Faited to close UDP listener %s", err)
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}
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}() */
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ffmpegArgs := []string{"-re", "-i", "pipe:0",
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"-an", "-vcodec", "libvpx", "-crf", "10", "-cpu-used", "5", "-b:v", "6000k", "-maxrate", "8000k", "-bufsize", "12000k", // TODO Change bitrate when changing quality
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"-qmin", "10", "-qmax", "42", "-threads", "4", "-deadline", "1", "-error-resilient", "1",
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"-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", "-qmin", "10", "-qmax", "42", "-error-resilient", "1", "-auto-alt-ref", "1",
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"-f", "rtp", "rtp://127.0.0.1:5005"}
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// Export stream to ascii art
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if telnet.Cfg.Enabled {
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bitrate := fmt.Sprintf("%dk", telnet.Cfg.Width*telnet.Cfg.Height/telnet.Cfg.Delay)
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ffmpegArgs = append(ffmpegArgs,
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"-an", "-vf", fmt.Sprintf("scale=%dx%d", telnet.Cfg.Width, telnet.Cfg.Height),
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"-b:v", bitrate, "-minrate", bitrate, "-maxrate", bitrate, "-bufsize", bitrate, "-q", "42", "-pix_fmt", "gray", "-f", "rawvideo", "pipe:1")
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}
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ffmpeg[srtPacket.StreamName] = exec.Command("ffmpeg", ffmpegArgs...)
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input, err := ffmpeg[srtPacket.StreamName].StdinPipe()
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if err != nil {
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panic(err)
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}
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ffmpegInput[srtPacket.StreamName] = input
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errOutput, err := ffmpeg[srtPacket.StreamName].StderrPipe()
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if err != nil {
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panic(err)
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}
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// Receive raw video output and convert it to ASCII art, then forward it TCP
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if telnet.Cfg.Enabled {
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output, err := ffmpeg[srtPacket.StreamName].StdoutPipe()
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if err != nil {
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panic(err)
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}
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go telnet.StartASCIIArtStream(srtPacket.StreamName, output)
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}
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if err := ffmpeg[srtPacket.StreamName].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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log.Printf("Failed to read from UDP: %s", err)
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continue
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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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log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
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continue
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}
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if videoTracks[srtPacket.StreamName] == nil {
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videoTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
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}
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// Write RTP srtPacket to all video tracks
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// Adapt payload and SSRC to match destination
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for _, videoTrack := range videoTracks[srtPacket.StreamName] {
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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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log.Printf("Failed to write to video track: %s", err)
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continue
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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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log.Printf("Failed to read from UDP: %s", err)
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continue
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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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log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
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continue
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}
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if audioTracks[srtPacket.StreamName] == nil {
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audioTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
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}
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// Write RTP srtPacket to all audio tracks
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// Adapt payload and SSRC to match destination
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for _, audioTrack := range audioTracks[srtPacket.StreamName] {
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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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log.Printf("Failed to write to audio track: %s", err)
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continue
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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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}
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