ghostream/stream/webrtc/ingest.go

143 lines
3.6 KiB
Go

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