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ghostream
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Code Issues Releases Wiki Activity

Try to use the new H264 reader from pion, not working yet

This commit is contained in:
Yohann D'ANELLO 2020-11-07 16:58:59 +01:00
parent 4e0dea4eab
commit 9f8c3a907a
2 changed files with 41 additions and 38 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

65
stream/webrtc/ingest.go
View File

@ -4,10 +4,15 @@ package webrtc
import (
"bufio"
"fmt"
"github.com/pion/webrtc/v3/pkg/media"
"github.com/pion/webrtc/v3/pkg/media/h264reader"
"io"
"log"
"math/rand"
"net"
"os"
"os/exec"
"time"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
@ -22,19 +27,17 @@ func ingest(name string, q *messaging.Quality) {
// FIXME Mux into RTP without having multiple UDP listeners
firstPort := int(rand.Int31n(63535)) + 2000
// Open UDP listeners for RTP Packets
// Open UDP listener 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
f, _ := os.Open("CoffeeRun.h264")
h264, err := h264reader.NewReader(f)
// Start ffmpag to convert videoInput to audio UDP
ffmpeg, err := startFFmpeg(videoInput, firstPort)
if err != nil {
log.Printf("Error while starting ffmpeg: %s", err)
@ -43,31 +46,37 @@ func ingest(name string, q *messaging.Quality) {
// 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)
nal, h264Err := h264.NextNAL()
if h264Err == io.EOF {
fmt.Printf("All video frames parsed and sent")
return
}
if h264Err != nil {
log.Printf("Failed to read from H264: %s", h264Err)
break
}
packet := &rtp.Packet{}
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
continue
}
time.Sleep(time.Millisecond * 33)
nal.Data = append([]byte{0x00, 0x00, 0x00, 0x01}, nal.Data...)
if videoTracks[name] == nil {
videoTracks[name] = make([]*webrtc.Track, 0)
}
// Write RTP srtPacket to all video tracks
// Adapt payload and SSRC to match destination
var spsAndPpsCache []byte
if nal.UnitType == h264reader.NalUnitTypeSPS || nal.UnitType == h264reader.NalUnitTypePPS {
spsAndPpsCache = append(spsAndPpsCache, nal.Data...)
continue
} else if nal.UnitType == h264reader.NalUnitTypeCodedSliceIdr {
nal.Data = append(spsAndPpsCache, nal.Data...)
spsAndPpsCache = []byte{}
}
log.Println(nal.PictureOrderCount)
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
if h264Err = videoTrack.WriteSample(media.Sample{Data: nal.Data, Samples: 90000}); h264Err != nil {
panic(h264Err)
}
}
}
@ -110,10 +119,7 @@ func ingest(name string, q *messaging.Quality) {
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)
}
// Close UDP listener
if err = audioListener.Close(); err != nil {
log.Printf("Faited to close UDP listener: %s", err)
}
@ -124,10 +130,7 @@ func startFFmpeg(in <-chan []byte, listeningPort int) (ffmpeg *exec.Cmd, err err
ffmpegArgs := []string{"-hide_banner", "-loglevel", "error", "-i", "pipe:0",
// Audio
"-vn", "-c:a", "libopus", "-b:a", "96k",
"-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", listeningPort),
// Source
"-an", "-c:v", "copy",
"-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", listeningPort+1)}
"-f", "rtp", fmt.Sprintf("rtp://127.0.0.1:%d", listeningPort)}
ffmpeg = exec.Command("ffmpeg", ffmpegArgs...)
// Handle errors output

14
stream/webrtc/webrtc.go
View File

@ -75,8 +75,8 @@ func newPeerHandler(name string, localSdpChan chan webrtc.SessionDescription, re
}
// Create video track
codec, payloadType := getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "H264")
videoTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "video", "pion", codec)
payloadType := getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8")
videoTrack, err := peerConnection.NewTrack(payloadType, rand.Uint32(), "video", "pion")
if err != nil {
log.Println("Failed to create new video track", err)
localSdpChan <- webrtc.SessionDescription{}
@ -89,8 +89,8 @@ func newPeerHandler(name string, localSdpChan chan webrtc.SessionDescription, re
}
// Create audio track
codec, payloadType = getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus")
audioTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "audio", "pion", codec)
payloadType = getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus")
audioTrack, err := peerConnection.NewTrack(payloadType, rand.Uint32(), "audio", "pion")
if err != nil {
log.Println("Failed to create new audio track", err)
localSdpChan <- webrtc.SessionDescription{}
@ -168,14 +168,14 @@ func newPeerHandler(name string, localSdpChan chan webrtc.SessionDescription, re
// Search for Codec PayloadType
//
// Since we are answering we need to match the remote PayloadType
func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) (*webrtc.RTPCodec, uint8) {
func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) uint8 {
for _, codec := range m.GetCodecsByKind(codecType) {
if codec.Name == codecName {
return codec, codec.PayloadType
return codec.PayloadType
}
}
log.Printf("Remote peer does not support %s", codecName)
return nil, 0
return 0
}
// Serve WebRTC media streaming server