WebRTC is registering to the audio-transcoded stream

This commit is contained in:
Yohann D'ANELLO 2020-10-18 22:07:11 +02:00
parent 20776d897c
commit 698b83fe6f
2 changed files with 135 additions and 94 deletions

View File

@ -2,15 +2,12 @@
package webrtc
import (
"bufio"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"log"
"net"
"os/exec"
"strings"
"time"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
"gitlab.crans.org/nounous/ghostream/stream"
)
@ -35,7 +32,9 @@ func autoIngest(streams map[string]*stream.Stream) {
// Start ingestion
log.Printf("Starting webrtc for '%s'", name)
go ingest(name, st)
// FIXME Ensure that the audio stream exist, but that's poop code
time.Sleep(time.Second)
go ingest(name, st, streams[name+"@audio"])
}
// Regulary pull stream list,
@ -45,76 +44,19 @@ func autoIngest(streams map[string]*stream.Stream) {
}
}
func ingest(name string, input *stream.Stream) {
func ingest(name string, input *stream.Stream, audio *stream.Stream) {
// Register to get stream
videoInput := make(chan []byte, 1024)
input.Register(videoInput)
audioInput := make(chan []byte, 1024)
audio.Register(audioInput)
activeStream[name] = struct{}{}
// TODO Register to all substreams and make RTP packets. Don't transcode in this package.
// 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
}
// Start ffmpag to convert videoInput to video and audio UDP
ffmpeg, err := startFFmpeg(videoInput)
if err != nil {
log.Printf("Error while starting ffmpeg: %s", err)
return
}
// Receive video
// Receive audio data
go func() {
inboundRTPPacket := make([]byte, 1500) // UDP MTU
for {
n, _, err := videoListener.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
}
if videoTracks[name] == nil {
videoTracks[name] = make([]*webrtc.Track, 0)
}
// 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 {
if err := packet.Unmarshal(<-audioInput); err != nil {
log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
continue
}
@ -129,29 +71,116 @@ func ingest(name string, input *stream.Stream) {
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)
log.Printf("Failed to write to audio track: %s", writeErr)
continue
}
}
}
}()
// Wait for stopped ffmpeg
if err = ffmpeg.Wait(); err != nil {
log.Printf("Faited to wait for ffmpeg: %s", err)
}
select {}
// 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)
}
// TODO Register to all substreams and make RTP packets. Don't transcode in this package.
/* // 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
}
// Start ffmpag to convert videoInput to video and audio UDP
ffmpeg, err := startFFmpeg(videoInput)
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)
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
}
if videoTracks[name] == nil {
videoTracks[name] = make([]*webrtc.Track, 0)
}
// 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
}
if audioTracks[name] == nil {
audioTracks[name] = make([]*webrtc.Track, 0)
}
// 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
}
}
}
}()
// 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)
}*/
delete(activeStream, name)
}
func startFFmpeg(in <-chan []byte) (ffmpeg *exec.Cmd, err error) {
/* func startFFmpeg(in <-chan []byte) (ffmpeg *exec.Cmd, err error) {
ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-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",
@ -192,4 +221,4 @@ func startFFmpeg(in <-chan []byte) (ffmpeg *exec.Cmd, err error) {
// Start process
err = ffmpeg.Start()
return ffmpeg, err
}
} */

View File

@ -4,8 +4,9 @@ package audio
import (
"bufio"
"fmt"
"io"
"log"
"math/rand"
"net"
"os/exec"
"strings"
"time"
@ -84,20 +85,37 @@ func transcode(input, output *stream.Stream, cfg *Options) {
}
// Stop transcode
ffmpeg.Process.Kill()
_ = ffmpeg.Process.Kill()
_ = rawvideo.Close()
}
// Start a ffmpeg instance to convert stream into audio
func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *io.ReadCloser, error) {
func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *net.UDPConn, error) {
// TODO in a future release: remove FFMPEG dependency and transcode directly using the libopus API
// FIXME It seems impossible to get a RTP Packet from standard output.
// We need to find a clean solution, without waiting on UDP listeners.
// FIXME We should also not build RTP packets here.
port := 0
var udpListener *net.UDPConn
var err error
for {
port = rand.Intn(65535)
udpListener, err = net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: port})
if err != nil {
if strings.Contains(fmt.Sprintf("%s", err), "address already in use") {
continue
}
return nil, nil, err
}
break
}
bitrate := fmt.Sprintf("%dk", cfg.Bitrate)
// Use copy audio codec, assume for now that libopus is used by the streamer
ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-i", "pipe:0",
"-vn", "-c:a", "copy", "-b:a", bitrate, "-f", "rtp", "pipe:1"}
"-vn", "-c:a", "copy", "-b:a", bitrate, "-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", port)}
ffmpeg := exec.Command("ffmpeg", ffmpegArgs...)
// Handle errors output
@ -112,12 +130,6 @@ func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *io.ReadCloser, err
}
}()
// Handle audio output
output, err := ffmpeg.StdoutPipe()
if err != nil {
return nil, nil, err
}
// Handle stream input
input, err := ffmpeg.StdinPipe()
if err != nil {
@ -125,11 +137,11 @@ func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *io.ReadCloser, err
}
go func() {
for data := range in {
input.Write(data)
_, _ = input.Write(data)
}
}()
// Start process
err = ffmpeg.Start()
return ffmpeg, &output, err
return ffmpeg, udpListener, err
}