Try to use manually FFMPEG bindings to avoid having too much syscalls, not working yet

go.mod

@@ -3,6 +3,7 @@ module gitlab.crans.org/nounous/ghostream
 go 1.15
 
 require (
+	github.com/3d0c/gmf v0.0.0-20200614092945-e58d8d5a6035
 	github.com/go-ldap/ldap/v3 v3.2.3
 	github.com/gorilla/websocket v1.4.0
 	github.com/haivision/srtgo v0.0.0-20201025191851-67964e8f497a

go.sum

@@ -7,6 +7,8 @@ dmitri.shuralyov.com/html/belt v0.0.0-20180602232347-f7d459c86be0/go.mod h1:JLBr
 dmitri.shuralyov.com/service/change v0.0.0-20181023043359-a85b471d5412/go.mod h1:a1inKt/atXimZ4Mv927x+r7UpyzRUf4emIoiiSC2TN4=
 dmitri.shuralyov.com/state v0.0.0-20180228185332-28bcc343414c/go.mod h1:0PRwlb0D6DFvNNtx+9ybjezNCa8XF0xaYcETyp6rHWU=
 git.apache.org/thrift.git v0.0.0-20180902110319-2566ecd5d999/go.mod h1:fPE2ZNJGynbRyZ4dJvy6G277gSllfV2HJqblrnkyeyg=
+github.com/3d0c/gmf v0.0.0-20200614092945-e58d8d5a6035 h1:QZb1aMKxiYdGGieyIDmXuw9I9YcGWGViTrpQ6vcZX7Q=
+github.com/3d0c/gmf v0.0.0-20200614092945-e58d8d5a6035/go.mod h1:0QMRcUq2JsDECeAq7bj4h79k7XbhtTsrPUQf6G7qfPs=
 github.com/Azure/go-ntlmssp v0.0.0-20200615164410-66371956d46c h1:/IBSNwUN8+eKzUzbJPqhK839ygXJ82sde8x3ogr6R28=
 github.com/Azure/go-ntlmssp v0.0.0-20200615164410-66371956d46c/go.mod h1:chxPXzSsl7ZWRAuOIE23GDNzjWuZquvFlgA8xmpunjU=
 github.com/BurntSushi/toml v0.3.1/go.mod h1:xHWCNGjB5oqiDr8zfno3MHue2Ht5sIBksp03qcyfWMU=

stream/webrtc/ingest.go

@@ -3,17 +3,20 @@ package webrtc
 
 import (
 	"bufio"
+	"bytes"
 	"fmt"
-	"github.com/pion/rtp"
-	"github.com/pion/webrtc/v3"
-	"github.com/pion/webrtc/v3/pkg/media"
-	"github.com/pion/webrtc/v3/pkg/media/h264reader"
-	"gitlab.crans.org/nounous/ghostream/messaging"
 	"io"
 	"log"
 	"math/rand"
 	"net"
 	"os/exec"
+
+	"github.com/3d0c/gmf"
+	"github.com/pion/rtp"
+	"github.com/pion/webrtc/v3"
+	"github.com/pion/webrtc/v3/pkg/media"
+	"github.com/pion/webrtc/v3/pkg/media/h264reader"
+	"gitlab.crans.org/nounous/ghostream/messaging"
 )
 
 func ingest(name string, q *messaging.Quality) {
@@ -21,26 +24,98 @@ func ingest(name string, q *messaging.Quality) {
 	videoInput := make(chan []byte, 1024)
 	q.Register(videoInput)
 
-	// FIXME Mux into RTP without having multiple UDP listeners
+	inputCtx := gmf.NewCtx()
-	firstPort := int(rand.Int31n(63535)) + 2000
+	avioInputCtx, _ := gmf.NewAVIOContext(inputCtx, &gmf.AVIOHandlers{ReadPacket: func() ([]byte, int) {
+		data := <-videoInput
+		return data, len(data)
+	}})
+	log.Println("Open input")
+	inputCtx.SetPb(avioInputCtx).OpenInput("")
+	log.Println("Opened")
+	defer inputCtx.CloseInput()
+	defer avioInputCtx.Release()
 
-	// Open UDP listener for RTP Packets
+	if audioTracks[name] == nil {
-	audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: firstPort})
+		audioTracks[name] = make([]*webrtc.Track, 0)
-	if err != nil {
-		log.Printf("Faited to open UDP listener %s", err)
-		return
 	}
 
-	// Start ffmpag to convert videoInput to audio UDP
+	port := rand.Int()%64355 + 2000
-	ffmpeg, ffmpegOut, err := startFFmpeg(videoInput, firstPort)
+	audioListener, _ := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: port})
+
+	b := bytes.Buffer{}
+	videoOutputCtx, _ := gmf.NewOutputCtxWithFormatName("/dev/null", "h264")
+	avioOutputCtx, _ := gmf.NewAVIOContext(videoOutputCtx, &gmf.AVIOHandlers{WritePacket: func(data []byte) int {
+		n, _ := b.Write(data)
+		return n
+	}})
+
+	videoOutputCtx.SetPb(avioOutputCtx)
+	defer videoOutputCtx.CloseOutput()
+	defer avioOutputCtx.Release()
+
+	audioOutputCtx, _ := gmf.NewOutputCtxWithFormatName(fmt.Sprintf("rtp://127.0.0.1:%d", port), "rtp")
+	defer audioOutputCtx.CloseOutput()
+
+	log.Printf("%d streams", inputCtx.StreamsCnt())
+
+	c, err := gmf.FindEncoder("libopus")
 	if err != nil {
-		log.Printf("Error while starting ffmpeg: %s", err)
+		log.Printf("Error while searching opus codec: %s", err)
-		return
+	}
+
+	audioStream, _ := inputCtx.GetBestStream(gmf.AVMEDIA_TYPE_AUDIO)
+	ctx := gmf.NewCodecCtx(c, []*gmf.Option{
+		{Key: "time_base", Val: audioStream.CodecCtx().TimeBase().AVR()},
+		{Key: "ar", Val: audioStream.CodecCtx().SampleRate()},
+		{Key: "ac", Val: audioStream.CodecCtx().Channels()},
+	})
+	par := gmf.NewCodecParameters()
+	_ = par.FromContext(audioStream.CodecCtx())
+	defer par.Free()
+	_, _ = audioOutputCtx.AddStreamWithCodeCtx(ctx)
+	//c, err = gmf.FindEncoder("libx264")
+
+	videoStream, _ := inputCtx.GetBestStream(gmf.AVMEDIA_TYPE_VIDEO)
+	c, err = gmf.FindEncoder("libx264")
+	if err != nil {
+		log.Printf("Error while searching x264 codec: %s", err)
+	}
+
+	ctx = gmf.NewCodecCtx(c, []*gmf.Option{
+		{Key: "time_base", Val: gmf.AVR{Num: 1, Den: 25}},
+		{Key: "pixel_format", Val: gmf.AV_PIX_FMT_YUV420P},
+		// Save original
+		{Key: "video_size", Val: videoStream.CodecCtx().GetVideoSize()},
+		{Key: "b", Val: 500000},
+	})
+	par = gmf.NewCodecParameters()
+	_ = par.FromContext(videoStream.CodecCtx())
+	defer par.Free()
+	_, _ = videoOutputCtx.AddStreamWithCodeCtx(ctx)
+
+	for i := 0; i < inputCtx.StreamsCnt(); i++ {
+		srcStream, err := inputCtx.GetStream(i)
+		if err != nil {
+			log.Println("GetStream error")
+			continue
+		}
+
+		log.Println(srcStream.CodecCtx())
+	}
+	videoOutputCtx.Dump()
+	audioOutputCtx.Dump()
+
+	if err := videoOutputCtx.WriteHeader(); err != nil {
+		log.Printf("Unable to write video header: %s", err)
+	}
+
+	if err := audioOutputCtx.WriteHeader(); err != nil {
+		log.Printf("Unable to write audio header: %s", err)
 	}
 
 	// Receive video
 	go func() {
-		h264, _ := h264reader.NewReader(*ffmpegOut)
+		h264, _ := h264reader.NewReader(&b)
 		var spsAndPpsCache []byte
 
 		for {
@@ -67,6 +142,7 @@ func ingest(name string, q *messaging.Quality) {
 				nal.Data = append(spsAndPpsCache, nal.Data...)
 				spsAndPpsCache = []byte{}
 			}
+			log.Println(len(nal.Data))
 
 			for _, videoTrack := range videoTracks[name] {
 				if h264Err = videoTrack.WriteSample(media.Sample{Data: nal.Data, Samples: 90000}); h264Err != nil {
@@ -108,16 +184,31 @@ func ingest(name string, q *messaging.Quality) {
 		}
 	}()
 
-	// Wait for stopped ffmpeg
+	for packet := range inputCtx.GetNewPackets() {
-	if err = ffmpeg.Wait(); err != nil {
+		if packet.StreamIndex() == 0 {
-		log.Printf("Faited to wait for ffmpeg: %s", err)
+			if err := videoOutputCtx.WritePacketNoBuffer(packet); err != nil {
+				log.Printf("Error while writing packet: %s", err)
+			}
+		} else if packet.StreamIndex() == 1 {
+			packet = packet.Clone()
+			packet.SetStreamIndex(0)
+			if err := audioOutputCtx.WritePacketNoBuffer(packet); err != nil {
+				log.Printf("Error while writing packet: %s", err)
+			}
+		}
+		packet.Free()
 	}
 
-	// Close UDP listener
+	// Wait for stopped ffmpeg
-	if err = audioListener.Close(); err != nil {
+	/*	if err = ffmpeg.Wait(); err != nil {
-		log.Printf("Faited to close UDP listener: %s", err)
+			log.Printf("Faited to wait for ffmpeg: %s", err)
-	}
+		}
-	q.Unregister(videoInput)
+
+		// Close UDP listener
+		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, stdout *io.ReadCloser, err error) {