apps/plugin-rtsp
1
0
Fork 0
mirror of https://github.com/Monibuca/plugin-rtsp.git synced 2025-09-26 19:51:14 +08:00
Code Issues Actions Packages Projects Releases Wiki Activity

增加强制指定拉流协议的配置

Browse Source
This commit is contained in:
dexter
2022-07-23 11:59:26 +08:00
parent ae37279dd1
commit 9b058153d2
7 changed files with 37 additions and 35 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
README.md
View File

@@ -24,7 +24,7 @@ rtsp://localhost/live/test
例如通过ffmpeg向m7s进行推流
```bash
ffmpeg -i [视频源] -f rtsp rtsp://localhost/live/test
ffmpeg -i [视频源] -c:v h264 -f rtsp rtsp://localhost/live/test
```
会在m7s内部形成一个名为live/test的流
@@ -63,6 +63,7 @@ rtsp:
udpaddr: :8000
rtcpaddr: :8001
readbuffersize: 2048
pullprotocol: 'auto'
```
:::tip 配置覆盖
publish

17
client.go
View File

@@ -14,14 +14,23 @@ type RTSPPuller struct {
}
func (p *RTSPPuller) Connect() error {
if p.Transport == gortsplib.TransportTCP {
p.Transport = gortsplib.TransportUDP
} else {
switch rtspConfig.PullProtocol {
case "tcp", "TCP":
p.Transport = gortsplib.TransportTCP
case "udp", "UDP":
p.Transport = gortsplib.TransportUDP
default:
if p.Transport == gortsplib.TransportTCP {
p.Transport = gortsplib.TransportUDP
} else {
p.Transport = gortsplib.TransportTCP
}
}
p.Client = &gortsplib.Client{
OnPacketRTP: func(ctx *gortsplib.ClientOnPacketRTPCtx) {
p.RTSPPublisher.Tracks[ctx.TrackID].WriteRTPPack(ctx.Packet)
if p.RTSPPublisher.Tracks[ctx.TrackID] != nil {
p.RTSPPublisher.Tracks[ctx.TrackID].WriteRTPPack(ctx.Packet)
}
},
ReadBufferCount: rtspConfig.ReadBufferSize,
Transport: &p.Transport,

1
main.go
View File

@@ -21,6 +21,7 @@ type RTSPConfig struct {
UDPAddr string
RTCPAddr string
ReadBufferSize int
PullProtocol string //tcp、udp、 auto（default）
sync.Map
}

4
publisher.go
View File

@@ -63,8 +63,8 @@ func (p *RTSPPublisher) SetTracks() error {
}
p.Tracks[trackId] = vt
t := track.(*gortsplib.TrackH264)
vt.WriteSlice(common.NALUSlice{t.SPS()})
vt.WriteSlice(common.NALUSlice{t.PPS()})
vt.WriteSlice(common.NALUSlice{t.SPS})
vt.WriteSlice(common.NALUSlice{t.PPS})
case "h265", "hevc":
vt := NewH265(p.Stream)
if payloadType, err := strconv.Atoi(vals[0]); err == nil {

4
server.go
View File

@@ -112,7 +112,9 @@ func (conf *RTSPConfig) OnPacketRTP(ctx *gortsplib.ServerHandlerOnPacketRTPCtx)
if p, ok := conf.Load(ctx.Session); ok {
switch v := p.(type) {
case *RTSPPublisher:
v.Tracks[ctx.TrackID].WriteRTPPack(ctx.Packet)
if v.Tracks[ctx.TrackID] != nil {
v.Tracks[ctx.TrackID].WriteRTPPack(ctx.Packet)
}
}
}
}

31
subscriber.go
View File

@@ -3,7 +3,6 @@ package rtsp
import (
"github.com/aler9/gortsplib"
"github.com/aler9/gortsplib/pkg/aac"
"go.uber.org/zap"
. "m7s.live/engine/v4"
"m7s.live/engine/v4/codec"
"m7s.live/engine/v4/track"
@@ -20,34 +19,36 @@ func (s *RTSPSubscriber) OnEvent(event any) {
switch v.CodecID {
case codec.CodecID_H264:
extra := v.DecoderConfiguration.Raw
if vtrack, err := gortsplib.NewTrackH264(v.DecoderConfiguration.PayloadType, extra[0], extra[1], nil); err == nil {
s.videoTrackId = len(s.tracks)
s.tracks = append(s.tracks, vtrack)
vtrack := &gortsplib.TrackH264{
PayloadType: v.DecoderConfiguration.PayloadType, SPS: extra[0], PPS: extra[1],
}
s.videoTrackId = len(s.tracks)
s.tracks = append(s.tracks, vtrack)
case codec.CodecID_H265:
if vtrack, err := NewH265Track(v.DecoderConfiguration.PayloadType, v.DecoderConfiguration.Raw); err == nil {
s.videoTrackId = len(s.tracks)
s.tracks = append(s.tracks, vtrack)
vtrack := &gortsplib.TrackH265{
PayloadType: v.DecoderConfiguration.PayloadType, VPS: v.DecoderConfiguration.Raw[0], SPS: v.DecoderConfiguration.Raw[1], PPS: v.DecoderConfiguration.Raw[2],
}
s.videoTrackId = len(s.tracks)
s.tracks = append(s.tracks, vtrack)
}
s.AddTrack(v)
case *track.Audio:
switch v.CodecID {
case codec.CodecID_AAC:
var mpegConf aac.MPEG4AudioConfig
mpegConf.Decode(v.DecoderConfiguration.Raw)
if atrack, err := gortsplib.NewTrackAAC(v.DecoderConfiguration.PayloadType, int(mpegConf.Type), mpegConf.SampleRate, mpegConf.ChannelCount, mpegConf.AOTSpecificConfig, 13, 3, 3); err == nil {
s.audioTrackId = len(s.tracks)
s.tracks = append(s.tracks, atrack)
} else {
v.Stream.Error("error creating AAC track", zap.Error(err))
mpegConf.Unmarshal(v.DecoderConfiguration.Raw)
atrack := &gortsplib.TrackAAC{
PayloadType: v.DecoderConfiguration.PayloadType, Config: &mpegConf, SizeLength: 13, IndexLength: 3, IndexDeltaLength: 3,
}
s.audioTrackId = len(s.tracks)
s.tracks = append(s.tracks, atrack)
case codec.CodecID_PCMA:
s.audioTrackId = len(s.tracks)
s.tracks = append(s.tracks, gortsplib.NewTrackPCMA())
s.tracks = append(s.tracks, &gortsplib.TrackPCMA{})
case codec.CodecID_PCMU:
s.audioTrackId = len(s.tracks)
s.tracks = append(s.tracks, gortsplib.NewTrackPCMU())
s.tracks = append(s.tracks, &gortsplib.TrackPCMU{})
}
s.AddTrack(v)
case ISubscriber:

12
track.go
View File

@@ -1,12 +0,0 @@
package rtsp
import (
"encoding/base64"
"fmt"
"github.com/aler9/gortsplib"
)
func NewH265Track(payloadType uint8, sprop [][]byte) (gortsplib.Track, error) {
return gortsplib.NewTrackGeneric("video", []string{fmt.Sprintf("%d", payloadType)}, fmt.Sprintf("%d H265/90000", payloadType), fmt.Sprintf("%d packetization-mode=1; sprop-vps=%s; sprop-sps=%s; sprop-pps=%s;", payloadType, base64.StdEncoding.EncodeToString(sprop[0]), base64.StdEncoding.EncodeToString(sprop[1]), base64.StdEncoding.EncodeToString(sprop[2])))
}