mirror of
https://gitlab.crans.org/nounous/ghostream.git
synced 2024-12-22 15:02:19 +00:00
Separate the WebRTC stream subroutine in a dedicated subroutine
This commit is contained in:
Yohann D'ANELLO
parent
defba52569
commit
32f877508d
@ -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())
|
||||
}
|
||||
}()
|
||||
}
|
||||
|
||||
Loading…
Reference in New Issue
Block a user