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mirror of https://gitlab.crans.org/nounous/ghostream.git synced 2024-12-22 09:12:19 +00:00

Separate the WebRTC stream subroutine in a dedicated subroutine

This commit is contained in:
Yohann D'ANELLO 2020-10-13 09:50:46 +02:00
parent defba52569
commit 32f877508d

View File

@ -15,156 +15,37 @@ import (
"gitlab.crans.org/nounous/ghostream/stream/srt"
)
var (
ffmpeg = make(map[string]*exec.Cmd)
ffmpegInput = make(map[string]io.WriteCloser)
)
func ingestFrom(inputChannel chan srt.Packet) {
// FIXME Clean code
var ffmpeg *exec.Cmd
var ffmpegInput io.WriteCloser
for {
var err error = nil
srtPacket := <-inputChannel
log.Println(len(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)
}
}()
ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-re", "-i", "pipe:0",
"-an", "-vcodec", "libvpx", "-crf", "10", "-cpu-used", "5", "-b:v", "6000k", "-maxrate", "8000k", "-bufsize", "12000k", // TODO Change bitrate when changing quality
"-qmin", "10", "-qmax", "42", "-threads", "4", "-deadline", "1", "-error-resilient", "1",
"-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5004",
"-vn", "-acodec", "libopus", "-cpu-used", "5", "-deadline", "1", "-qmin", "10", "-qmax", "42", "-error-resilient", "1", "-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5005"}
// Export stream to ascii art
if telnet.Cfg.Enabled {
ffmpegArgs = append(ffmpegArgs,
"-an", "-f", "rawvideo", "-vf", fmt.Sprintf("scale=%dx%d", telnet.Cfg.Width, telnet.Cfg.Height), "-pix_fmt", "gray", "pipe:1")
}
ffmpeg = exec.Command("ffmpeg", ffmpegArgs...)
input, err := ffmpeg.StdinPipe()
if err != nil {
panic(err)
}
ffmpegInput = input
errOutput, err := ffmpeg.StderrPipe()
if err != nil {
panic(err)
}
// Receive raw video output and convert it to ASCII art, then forward it TCP
if telnet.Cfg.Enabled {
output, err := ffmpeg.StdoutPipe()
if err != nil {
panic(err)
}
go telnet.StartASCIIArtStream(srtPacket.StreamName, output)
}
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())
}
}()
go registerStream(&srtPacket)
break
case "sendData":
if _, ok := ffmpegInput[srtPacket.StreamName]; !ok {
break
}
// FIXME send to stream srtPacket.StreamName
if _, err := ffmpegInput.Write(srtPacket.Data); err != nil {
if _, err := ffmpegInput[srtPacket.StreamName].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)
_ = ffmpeg[srtPacket.StreamName].Process.Kill()
_ = ffmpegInput[srtPacket.StreamName].Close()
delete(ffmpeg, srtPacket.StreamName)
delete(ffmpegInput, srtPacket.StreamName)
break
default:
log.Println("Unknown SRT srtPacket type:", srtPacket.PacketType)
@ -175,3 +56,140 @@ func ingestFrom(inputChannel chan srt.Packet) {
}
}
}
func registerStream(srtPacket *srt.Packet) {
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
}
// FIXME Close UDP listeners at the end of the stream, not the end of the routine
/* 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)
}
}() */
ffmpegArgs := []string{"-re", "-i", "pipe:0",
"-an", "-vcodec", "libvpx", "-crf", "10", "-cpu-used", "5", "-b:v", "6000k", "-maxrate", "8000k", "-bufsize", "12000k", // TODO Change bitrate when changing quality
"-qmin", "10", "-qmax", "42", "-threads", "4", "-deadline", "1", "-error-resilient", "1",
"-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5004",
"-vn", "-acodec", "libopus", "-cpu-used", "5", "-deadline", "1", "-qmin", "10", "-qmax", "42", "-error-resilient", "1", "-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5005"}
// Export stream to ascii art
if telnet.Cfg.Enabled {
bitrate := fmt.Sprintf("%dk", telnet.Cfg.Width*telnet.Cfg.Height/telnet.Cfg.Delay)
ffmpegArgs = append(ffmpegArgs,
"-an", "-vf", fmt.Sprintf("scale=%dx%d", telnet.Cfg.Width, telnet.Cfg.Height),
"-b:v", bitrate, "-minrate", bitrate, "-maxrate", bitrate, "-bufsize", bitrate, "-q", "42", "-pix_fmt", "gray", "-f", "rawvideo", "pipe:1")
}
ffmpeg[srtPacket.StreamName] = exec.Command("ffmpeg", ffmpegArgs...)
input, err := ffmpeg[srtPacket.StreamName].StdinPipe()
if err != nil {
panic(err)
}
ffmpegInput[srtPacket.StreamName] = input
errOutput, err := ffmpeg[srtPacket.StreamName].StderrPipe()
if err != nil {
panic(err)
}
// Receive raw video output and convert it to ASCII art, then forward it TCP
if telnet.Cfg.Enabled {
output, err := ffmpeg[srtPacket.StreamName].StdoutPipe()
if err != nil {
panic(err)
}
go telnet.StartASCIIArtStream(srtPacket.StreamName, output)
}
if err := ffmpeg[srtPacket.StreamName].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())
}
}()
}