加入转推功能

This commit is contained in:
dexter
2021-11-18 19:29:31 +08:00
parent ef106e42f8
commit 5094fd0db7
2 changed files with 104 additions and 7 deletions

View File

@@ -3,11 +3,16 @@ package rtsp
import (
"errors"
"time"
"unsafe"
. "github.com/Monibuca/engine/v3"
. "github.com/Monibuca/utils/v3"
"github.com/Monibuca/utils/v3/codec"
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/aac"
"github.com/aler9/gortsplib/pkg/base"
"github.com/pion/rtp"
"github.com/pion/rtp/codecs"
)
type RTSPClient struct {
@@ -48,7 +53,93 @@ func (rtsp *RTSPClient) PullStream(streamPath string, rtspUrl string) (err error
}
return errors.New("publish badname")
}
func (rtsp *RTSPClient) PushStream(streamPath string, rtspUrl string) (err error) {
if s := FindStream(streamPath); s != nil {
var tracks gortsplib.Tracks
var sub RTSPSubscriber
sub.Type = "RTSP push out"
sub.vt = s.WaitVideoTrack("h264", "h265")
sub.at = s.WaitAudioTrack("aac", "pcma", "pcmu")
ssrc := uintptr(unsafe.Pointer(&sub))
var trackIds = 0
if sub.vt != nil {
trackId := trackIds
var vtrack *gortsplib.Track
var vpacketer rtp.Packetizer
switch sub.vt.CodecID {
case codec.CodecID_H264:
if vtrack, err = gortsplib.NewTrackH264(96, &gortsplib.TrackConfigH264{
SPS: sub.vt.ExtraData.NALUs[0],
PPS: sub.vt.ExtraData.NALUs[1],
}); err == nil {
vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &codecs.H264Payloader{}, rtp.NewFixedSequencer(1), 90000)
} else {
return err
}
case codec.CodecID_H265:
vtrack = NewH265Track(96, sub.vt.ExtraData.NALUs)
vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &H265Payloader{}, rtp.NewFixedSequencer(1), 90000)
}
var st uint32
onVideo := func(ts uint32, pack *VideoPack) {
for _, nalu := range pack.NALUs {
for _, pack := range vpacketer.Packetize(nalu, (ts-st)*90) {
rtp, _ := pack.Marshal()
rtsp.WritePacketRTP(trackId, rtp)
}
}
st = ts
}
sub.OnVideo = func(ts uint32, pack *VideoPack) {
if st = ts; st != 0 {
sub.OnVideo = onVideo
}
onVideo(ts, pack)
}
tracks = append(tracks, vtrack)
trackIds++
}
if sub.at != nil {
var st uint32
trackId := trackIds
switch sub.at.CodecID {
case codec.CodecID_PCMA, codec.CodecID_PCMU:
atrack := NewG711Track(97, map[byte]string{7: "pcma", 8: "pcmu"}[sub.vt.CodecID])
apacketizer := rtp.NewPacketizer(1200, 97, uint32(ssrc), &codecs.G711Payloader{}, rtp.NewFixedSequencer(1), 8000)
sub.OnAudio = func(ts uint32, pack *AudioPack) {
for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*8) {
buf, _ := pack.Marshal()
rtsp.WritePacketRTP(trackId, buf)
}
st = ts
}
tracks = append(tracks, atrack)
case codec.CodecID_AAC:
var mpegConf aac.MPEG4AudioConfig
mpegConf.Decode(sub.at.ExtraData[2:])
conf := &gortsplib.TrackConfigAAC{
Type: int(mpegConf.Type),
SampleRate: mpegConf.SampleRate,
ChannelCount: mpegConf.ChannelCount,
AOTSpecificConfig: mpegConf.AOTSpecificConfig,
}
if atrack, err := gortsplib.NewTrackAAC(97, conf); err == nil {
apacketizer := rtp.NewPacketizer(1200, 97, uint32(ssrc), &AACPayloader{}, rtp.NewFixedSequencer(1), uint32(mpegConf.SampleRate))
sub.OnAudio = func(ts uint32, pack *AudioPack) {
for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*uint32(mpegConf.SampleRate)/1000) {
buf, _ := pack.Marshal()
rtsp.WritePacketRTP(trackId, buf)
}
st = ts
}
tracks = append(tracks, atrack)
}
}
}
return rtsp.StartPublishing(rtspUrl, tracks)
}
return errors.New("stream not exist")
}
func (client *RTSPClient) startStream() {
if client.Err() != nil {
return

10
main.go
View File

@@ -19,7 +19,8 @@ var config = struct {
Timeout int
Reconnect bool
AutoPullList map[string]string
}{":554", ":8000", ":8001", 0, false, nil}
AutoPushList map[string]string
}{":554", ":8000", ":8001", 0, false, nil, nil}
func init() {
InstallPlugin(&PluginConfig{
@@ -63,13 +64,18 @@ func runPlugin() {
w.Write([]byte(fmt.Sprintf(`{"code":1,"msg":"%s"}`, err.Error())))
}
})
if len(config.AutoPullList) > 0 {
for streamPath, url := range config.AutoPullList {
if err := (&RTSPClient{}).PullStream(streamPath, url); err != nil {
Println(err)
}
}
go AddHook(HOOK_PUBLISH, func(s *Stream) {
for streamPath, url := range config.AutoPushList {
if s.StreamPath == streamPath {
(&RTSPClient{}).PushStream(streamPath, url)
}
}
})
if config.ListenAddr != "" {
go log.Fatal(ListenRtsp(config.ListenAddr))
}