Try to mux into RTP with FFMPEG bindings
This commit is contained in:
Yohann D'ANELLO
parent
698b83fe6f
commit
fa281e6b31
1
go.mod
1
go.mod
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@ -3,6 +3,7 @@ module gitlab.crans.org/nounous/ghostream
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go 1.13
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require (
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github.com/3d0c/gmf v0.0.0-20200614092945-e58d8d5a6035
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github.com/go-ldap/ldap/v3 v3.2.3
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github.com/haivision/srtgo v0.0.0-20200731151239-e00427ae473a
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github.com/markbates/pkger v0.17.1
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2
go.sum
2
go.sum
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@ -7,6 +7,8 @@ dmitri.shuralyov.com/html/belt v0.0.0-20180602232347-f7d459c86be0/go.mod h1:JLBr
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dmitri.shuralyov.com/service/change v0.0.0-20181023043359-a85b471d5412/go.mod h1:a1inKt/atXimZ4Mv927x+r7UpyzRUf4emIoiiSC2TN4=
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dmitri.shuralyov.com/state v0.0.0-20180228185332-28bcc343414c/go.mod h1:0PRwlb0D6DFvNNtx+9ybjezNCa8XF0xaYcETyp6rHWU=
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git.apache.org/thrift.git v0.0.0-20180902110319-2566ecd5d999/go.mod h1:fPE2ZNJGynbRyZ4dJvy6G277gSllfV2HJqblrnkyeyg=
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github.com/3d0c/gmf v0.0.0-20200614092945-e58d8d5a6035 h1:QZb1aMKxiYdGGieyIDmXuw9I9YcGWGViTrpQ6vcZX7Q=
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github.com/3d0c/gmf v0.0.0-20200614092945-e58d8d5a6035/go.mod h1:0QMRcUq2JsDECeAq7bj4h79k7XbhtTsrPUQf6G7qfPs=
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github.com/Azure/go-ntlmssp v0.0.0-20200615164410-66371956d46c h1:/IBSNwUN8+eKzUzbJPqhK839ygXJ82sde8x3ogr6R28=
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github.com/Azure/go-ntlmssp v0.0.0-20200615164410-66371956d46c/go.mod h1:chxPXzSsl7ZWRAuOIE23GDNzjWuZquvFlgA8xmpunjU=
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github.com/BurntSushi/toml v0.3.1 h1:WXkYYl6Yr3qBf1K79EBnL4mak0OimBfB0XUf9Vl28OQ=
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@ -2,13 +2,14 @@
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package webrtc
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import (
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"github.com/3d0c/gmf"
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"github.com/pion/rtp"
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"github.com/pion/webrtc/v3"
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"gitlab.crans.org/nounous/ghostream/stream"
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"log"
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"net"
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"strings"
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"time"
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"gitlab.crans.org/nounous/ghostream/stream"
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)
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var (
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@ -52,11 +53,63 @@ func ingest(name string, input *stream.Stream, audio *stream.Stream) {
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audio.Register(audioInput)
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activeStream[name] = struct{}{}
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inputCtx := gmf.NewCtx()
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avioInputCtx, _ := gmf.NewAVIOContext(inputCtx, &gmf.AVIOHandlers{ReadPacket: func() ([]byte, int) {
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data := <-audioInput
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return data, len(data)
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}})
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log.Println("Open input")
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inputCtx.SetPb(avioInputCtx).OpenInput("")
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log.Println("Opened")
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defer inputCtx.CloseInput()
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defer avioInputCtx.Release()
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if audioTracks[name] == nil {
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audioTracks[name] = make([]*webrtc.Track, 0)
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}
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udpListener, _ := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 1234})
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outputCtx, _ := gmf.NewOutputCtxWithFormatName("rtp://127.0.0.1:1234", "rtp")
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avioOutputCtx, _ := gmf.NewAVIOContext(outputCtx, &gmf.AVIOHandlers{WritePacket: func(data []byte) int {
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n := len(data)
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log.Printf("Read %d bytes", n)
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return n
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}})
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// FIXME DON'T RAN AN UDP LISTENER, PLIZ GET DIRECTLY UDP PACKETS, WHY IS IT SO COMPLICATED????
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// outputCtx.SetPb(avioOutputCtx)
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defer outputCtx.CloseOutput()
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defer avioOutputCtx.Release()
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log.Printf("%d streams", inputCtx.StreamsCnt())
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for i := 0; i < inputCtx.StreamsCnt(); i++ {
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srcStream, err := inputCtx.GetStream(i)
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if err != nil {
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log.Println("GetStream error")
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}
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outputCtx.AddStreamWithCodeCtx(srcStream.CodecCtx())
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}
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outputCtx.Dump()
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if err := outputCtx.WriteHeader(); err != nil {
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log.Printf("Unable to write RTP header: %s", err)
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}
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// Receive audio data
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go func() {
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buff := make([]byte, 1500)
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for {
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n, _ := udpListener.Read(buff)
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if n == 0 {
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return
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}
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packet := &rtp.Packet{}
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if err := packet.Unmarshal(<-audioInput); err != nil {
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if err := packet.Unmarshal(buff[:n]); err != nil {
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log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
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continue
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}
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@ -78,6 +131,19 @@ func ingest(name string, input *stream.Stream, audio *stream.Stream) {
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}
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}()
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first := false
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for packet := range inputCtx.GetNewPackets() {
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if first { //if read from rtsp ,the first packets is wrong.
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if err := outputCtx.WritePacketNoBuffer(packet); err != nil {
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log.Printf("Error while writing packet: %s", err)
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}
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}
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first = true
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packet.Free()
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}
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select {}
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// TODO Register to all substreams and make RTP packets. Don't transcode in this package.
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@ -4,9 +4,9 @@ package audio
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import (
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"bufio"
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"fmt"
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"github.com/3d0c/gmf"
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"io"
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"log"
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"math/rand"
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"net"
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"os/exec"
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"strings"
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"time"
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@ -63,15 +63,15 @@ func transcode(input, output *stream.Stream, cfg *Options) {
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// Start ffmpeg to transcode video to audio
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videoInput := make(chan []byte, 1024)
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input.Register(videoInput)
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ffmpeg, rawvideo, err := startFFmpeg(videoInput, cfg)
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ffmpeg, audio, err := startFFmpeg(videoInput, cfg)
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if err != nil {
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log.Printf("Error while starting ffmpeg: %s", err)
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return
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}
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dataBuff := make([]byte, 1316) // UDP MTU
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dataBuff := make([]byte, gmf.IO_BUFFER_SIZE) // UDP MTU
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for {
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n, err := (*rawvideo).Read(dataBuff)
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n, err := (*audio).Read(dataBuff)
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if err != nil {
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log.Printf("An error occurred while reading input: %s", err)
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break
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@ -86,36 +86,36 @@ func transcode(input, output *stream.Stream, cfg *Options) {
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// Stop transcode
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_ = ffmpeg.Process.Kill()
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_ = rawvideo.Close()
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_ = (*audio).Close()
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}
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// Start a ffmpeg instance to convert stream into audio
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func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *net.UDPConn, error) {
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func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *io.ReadCloser, error) {
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// TODO in a future release: remove FFMPEG dependency and transcode directly using the libopus API
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// FIXME It seems impossible to get a RTP Packet from standard output.
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// We need to find a clean solution, without waiting on UDP listeners.
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// FIXME We should also not build RTP packets here.
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port := 0
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var udpListener *net.UDPConn
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var err error
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for {
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port = rand.Intn(65535)
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udpListener, err = net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: port})
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if err != nil {
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if strings.Contains(fmt.Sprintf("%s", err), "address already in use") {
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continue
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/* port := 0
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var udpListener *net.UDPConn
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var err error
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for {
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port = rand.Intn(65535)
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udpListener, err = net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: port})
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if err != nil {
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if strings.Contains(fmt.Sprintf("%s", err), "address already in use") {
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continue
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}
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return nil, nil, err
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}
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return nil, nil, err
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}
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break
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}
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break
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}*/
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bitrate := fmt.Sprintf("%dk", cfg.Bitrate)
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// Use copy audio codec, assume for now that libopus is used by the streamer
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ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-i", "pipe:0",
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"-vn", "-c:a", "copy", "-b:a", bitrate, "-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", port)}
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"-vn", "-c:a", "copy", "-b:a", bitrate, "-f", "mpegts", "pipe:1"}
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ffmpeg := exec.Command("ffmpeg", ffmpegArgs...)
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// Handle errors output
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@ -130,6 +130,12 @@ func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *net.UDPConn, error
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}
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}()
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// Handle text output
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output, err := ffmpeg.StdoutPipe()
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if err != nil {
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return nil, nil, err
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}
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// Handle stream input
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input, err := ffmpeg.StdinPipe()
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if err != nil {
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@ -143,5 +149,5 @@ func startFFmpeg(in <-chan []byte, cfg *Options) (*exec.Cmd, *net.UDPConn, error
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// Start process
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err = ffmpeg.Start()
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return ffmpeg, udpListener, err
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return ffmpeg, &output, err
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}
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