ghostream/stream/webrtc/ingest.go

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// Package webrtc provides the backend to simulate a WebRTC client to send stream
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package webrtc
import (
"bufio"
"fmt"
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"log"
"math/rand"
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"net"
"os/exec"
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"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"gitlab.crans.org/nounous/ghostream/messaging"
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)
func ingest(name string, q *messaging.Quality) {
// Register to get stream
videoInput := make(chan []byte, 1024)
q.Register(videoInput)
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// FIXME Mux into RTP without having multiple UDP listeners
firstPort := int(rand.Int31n(63535)) + 2000
// Open UDP listeners for RTP Packets
audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: firstPort})
if err != nil {
log.Printf("Faited to open UDP listener %s", err)
return
}
videoListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: firstPort + 1})
if err != nil {
log.Printf("Faited to open UDP listener %s", err)
return
}
// Start ffmpag to convert videoInput to video and audio UDP
ffmpeg, err := startFFmpeg(videoInput, firstPort)
if err != nil {
log.Printf("Error while starting ffmpeg: %s", err)
return
}
// Receive video
go func() {
inboundRTPPacket := make([]byte, 1500) // UDP MTU
for {
n, _, err := videoListener.ReadFromUDP(inboundRTPPacket)
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if err != nil {
log.Printf("Failed to read from UDP: %s", err)
break
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}
packet := &rtp.Packet{}
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
continue
}
if videoTracks[name] == nil {
videoTracks[name] = make([]*webrtc.Track, 0)
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}
// Write RTP srtPacket to all video tracks
// Adapt payload and SSRC to match destination
for _, videoTrack := range videoTracks[name] {
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() {
inboundRTPPacket := make([]byte, 1500) // UDP MTU
for {
n, _, err := audioListener.ReadFromUDP(inboundRTPPacket)
if err != nil {
log.Printf("Failed to read from UDP: %s", err)
break
}
packet := &rtp.Packet{}
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
continue
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}
if audioTracks[name] == nil {
audioTracks[name] = make([]*webrtc.Track, 0)
}
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// Write RTP srtPacket to all audio tracks
// Adapt payload and SSRC to match destination
for _, audioTrack := range audioTracks[name] {
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
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}
}
}
}()
// Wait for stopped ffmpeg
if err = ffmpeg.Wait(); err != nil {
log.Printf("Faited to wait for ffmpeg: %s", err)
}
// Close UDP listeners
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)
}
q.Unregister(videoInput)
}
func startFFmpeg(in <-chan []byte, listeningPort int) (ffmpeg *exec.Cmd, err error) {
ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-i", "pipe:0",
// Audio
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"-vn", "-c:a", "libopus", "-b:a", "96k",
"-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", listeningPort),
// Source
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"-an", "-c:v", "copy",
"-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", listeningPort+1)}
ffmpeg = exec.Command("ffmpeg", ffmpegArgs...)
// Handle errors output
errOutput, err := ffmpeg.StderrPipe()
if err != nil {
return nil, err
}
go func() {
scanner := bufio.NewScanner(errOutput)
for scanner.Scan() {
log.Printf("[WEBRTC FFMPEG %s] %s", "demo", scanner.Text())
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}
}()
// Handle stream input
input, err := ffmpeg.StdinPipe()
if err != nil {
return nil, err
}
go func() {
for data := range in {
if _, err := input.Write(data); err != nil {
log.Printf("Failed to write data to ffmpeg input: %s", err)
}
}
// End of stream
ffmpeg.Process.Kill()
}()
// Start process
err = ffmpeg.Start()
return ffmpeg, err
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}