SizeLength、IndexLength、IndexDeltaLength

Bump golang.org/x/net from 0.2.0 to 0.7.0

.gitignore

@@ -1,2 +0,0 @@
-.vscode
-node_modules

README.md

@@ -1,63 +1,89 @@
 # RTSP插件
-
+rtsp插件提供rtsp协议的推拉流能力，以及向远程服务器推拉rtsp协议的能力。
 ## 插件地址
 
-github.com/Monibuca/plugin-rtsp
+https://github.com/Monibuca/plugin-rtsp
 
 ## 插件引入
 ```go
 import (
-    _ "github.com/Monibuca/plugin-rtsp"
+    _ "m7s.live/plugin/rtsp/v4"
 )
 ```
 
-## 默认插件配置
+## 推拉地址形式
-
-```toml
-[RTSP]
-# 端口接收推流
-ListenAddr = ":554"
-Reconnect = true
-[RTSP.AutoPullList]
-"live/rtsp1" = "rtsp://admin:admin@192.168.1.212:554/cam/realmonitor?channel=1&subtype=1"
-"live/rtsp2" = "rtsp://admin:admin@192.168.1.212:554/cam/realmonitor?channel=2&subtype=1"
 ```
+rtsp://localhost/live/test
+```
+- `localhost`是m7s的服务器域名或者IP地址，默认端口`554`可以不写，否则需要写
+- `live`代表`appName`
+- `test`代表`streamName`
+- m7s中`live/test`将作为`streamPath`为流的唯一标识
 
-- `ListenAddr`是监听的地址
-- `Reconnect` 是否自动重连
-- `RTSP.AutoPullList` 可以配置多项，用于自动拉流，key是streamPath，value是远程rtsp地址
-
-### 特殊功能
-
-当自动拉流列表中当的streamPath为sub/xxx 这种形式的话，在gb28181的分屏显示时会优先采用rtsp流，已实现分屏观看子码流效果
-## 插件功能
-
-### 接收RTSP协议的推流
 
 例如通过ffmpeg向m7s进行推流
 
 ```bash
-ffmpeg -i **** rtsp://localhost/live/test
+ffmpeg -i [视频源] -c:v h264 -c:a aac -f rtsp rtsp://localhost/live/test
 ```
 
 会在m7s内部形成一个名为live/test的流
 
-### 从远程拉取rtsp到m7s中
 
-可调用接口
+如果m7s中已经存在live/test流的话就可以用rtsp协议进行播放
-`/api/rtsp/pull?target=[RTSP地址]&streamPath=[流标识]`
+```bash
-
+ffplay rtsp://localhost/live/test
-## 使用编程方式拉流
-```go
-new(RTSPClient).PullStream("live/user1","rtsp://xxx.xxx.xxx.xxx/live/user1") 
 ```
 
-### 罗列所有的rtsp协议的流
 
-可调用接口
+## 配置
-`/api/rtsp/list`
 
-### 从m7s中拉取rtsp协议流
+```yaml
+rtsp:
+    publish:
+        pubaudio: true # 是否发布音频流
+        pubvideo: true # 是否发布视频流
+        kickexist: false # 剔出已经存在的发布者，用于顶替原有发布者
+        publishtimeout: 10s # 发布流默认过期时间，超过该时间发布者没有恢复流将被删除
+        delayclosetimeout: 0 # 自动关闭触发后延迟的时间(期间内如果有新的订阅则取消触发关闭)，0为关闭该功能，保持连接。
+        waitclosetimeout: 0 # 发布者断开后等待时间，超过该时间发布者没有恢复流将被删除，0为关闭该功能，由订阅者决定是否删除
+        buffertime: 0 # 缓存时间，用于时光回溯，0为关闭缓存
+    subscribe:
+        subaudio: true # 是否订阅音频流
+        subvideo: true # 是否订阅视频流
+        subaudioargname: ats # 订阅音频轨道参数名
+        subvideoargname: vts # 订阅视频轨道参数名
+        subdataargname: dts # 订阅数据轨道参数名
+        subaudiotracks: [] # 订阅音频轨道名称列表
+        subvideotracks: [] # 订阅视频轨道名称列表
+        submode: 0 # 订阅模式，0为跳帧追赶模式，1为不追赶（多用于录制），2为时光回溯模式
+        iframeonly: false # 只订阅关键帧
+        waittimeout: 10s # 等待发布者的超时时间，用于订阅尚未发布的流
+    pull:
+        repull: 0
+        pullonstart: {}
+        pullonsub: {}
+    push:
+        repush: 0
+        pushlist: {}
+    listenaddr: :554
+    udpaddr: :8000
+    rtcpaddr: :8001
+    readbuffersize: 2048
+    pullprotocol: 'auto'
+```
+:::tip 配置覆盖
+publish
+subscribe
+两项中未配置部分将使用全局配置
+:::
+## API
 
-直接通过协议rtsp://xxx.xxx.xxx.xxx/live/user1 即可播放
+### `rtsp/api/list`
-> h265 编码拉流尚未实现，敬请期待
+获取所有rtsp流
+
+### `rtsp/api/pull?target=[RTSP地址]&streamPath=[流标识]`
+从远程拉取rtsp到m7s中
+
+### `rtsp/api/push?target=[RTSP地址]&streamPath=[流标识]`
+将本地的流推送到远端

client.go

@@ -1,208 +1,152 @@
 package rtsp
 
 import (
-	"errors"
+	"github.com/aler9/gortsplib/v2"
-	"time"
+	"github.com/aler9/gortsplib/v2/pkg/url"
-	"unsafe"
+	"go.uber.org/zap"
-
+	"m7s.live/engine/v4"
-	. "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 {
+type RTSPPuller struct {
-	RTSPublisher
+	RTSPPublisher
-	Transport         gortsplib.Transport
+	engine.Puller
 	*gortsplib.Client `json:"-"`
+	gortsplib.Transport
 }
 
-// PullStream 从外部拉流
+func (p *RTSPPuller) Connect() error {
-func (rtsp *RTSPClient) PullStream(streamPath string, rtspUrl string) (err error) {
+	switch rtspConfig.PullProtocol {
-	rtsp.Stream = &Stream{
+	case "tcp", "TCP":
-		StreamPath: streamPath,
+		p.Transport = gortsplib.TransportTCP
-		Type:       "RTSP Pull",
+	case "udp", "UDP":
-		ExtraProp:  rtsp,
+		p.Transport = gortsplib.TransportUDP
-	}
+	default:
-	if result := rtsp.Publish(); result {
+		if p.Transport == gortsplib.TransportTCP {
-		rtsp.URL = rtspUrl
+			p.Transport = gortsplib.TransportUDP
-		if config.Reconnect {
-			go func() {
-				for rtsp.pullStream(); rtsp.Err() == nil; rtsp.pullStream() {
-					Printf("reconnecting:%s in 5 seconds", rtspUrl)
-					if rtsp.Transport == gortsplib.TransportTCP {
-						rtsp.Transport = gortsplib.TransportUDP
 		} else {
-						rtsp.Transport = gortsplib.TransportTCP
+			p.Transport = gortsplib.TransportTCP
 		}
-					time.Sleep(time.Second * 5)
 	}
-				if rtsp.IsTimeout {
+	p.Client = &gortsplib.Client{
-					go rtsp.PullStream(streamPath, rtspUrl)
+		// OnPacketRTP: func(ctx *gortsplib.ClientOnPacketRTPCtx) {
+		// 	if p.RTSPPublisher.Tracks[ctx.TrackID] != nil {
+		// 		p.RTSPPublisher.Tracks[ctx.TrackID].WriteRTPPack(ctx.Packet)
+		// 	}
+		// },
+		ReadBufferCount: rtspConfig.ReadBufferSize,
+		Transport:       &p.Transport,
 	}
-			}()
+
-		} else {
+	// parse URL
-			go rtsp.pullStream()
+	u, err := url.Parse(p.RemoteURL)
-		}
+	if err != nil {
-		return
-	}
-	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:
+	// connect to the server
-				vtrack = NewH265Track(96, sub.vt.ExtraData.NALUs)
+	if err = p.Client.Start(u.Scheme, u.Host); err != nil {
-				vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &H265Payloader{}, rtp.NewFixedSequencer(1), 90000)
+		return err
 	}
-			var st uint32
+	p.SetIO(p.Client)
-			onVideo := func(ts uint32, pack *VideoPack) {
+	return nil
-				for _, nalu := range pack.NALUs {
-					for _, pack := range vpacketer.Packetize(nalu, (ts-st)*90) {
-						rtp, _ := pack.Marshal()
-						rtsp.WritePacketRTP(trackId, rtp)
 }
-				}
+
-				st = ts
+func (p *RTSPPuller) Pull() (err error) {
-			}
+	u, _ := url.Parse(p.RemoteURL)
-			sub.OnVideo = func(ts uint32, pack *VideoPack) {
+	defer p.Stop()
-				if st = ts; st != 0 {
+	if _, err = p.Options(u); err != nil {
-					sub.OnVideo = onVideo
+		p.Error("Options", zap.Error(err))
-				}
-				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.at.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) pullStream() {
-	if client.Err() != nil {
 		return
 	}
-	client.Client = &gortsplib.Client{
+	// find published tracks
-		OnPacketRTP: func(trackID int, payload []byte) {
+	tracks, baseURL, _, err := p.Describe(u)
-			if f := client.processFunc[trackID]; f != nil {
+	if err != nil {
-				var clone []byte
+		p.Error("Describe", zap.Error(err))
-				f(append(clone, payload...))
+		return
 	}
-		},
+	p.tracks = tracks
-		Transport: &client.Transport,
+	err = p.SetTracks()
+	if err != nil {
+		p.Error("SetTracks", zap.Error(err))
+		return
+	}
+	if err = p.SetupAll(tracks, baseURL); err != nil {
+		p.Error("SetupAndPlay", zap.Error(err))
+		return
+	}
+	p.OnPacketRTPAny(p.OnPacket)
+	_, err = p.Play(nil)
+	if err != nil {
+		p.Error("Play", zap.Error(err))
+		return
+	}
+	return p.Wait()
+}
+
+type RTSPPusher struct {
+	RTSPSubscriber
+	engine.Pusher
+	*gortsplib.Client
+	gortsplib.Transport
+}
+
+func (p *RTSPPusher) OnEvent(event any) {
+	switch v := event.(type) {
+	case engine.VideoRTP:
+		p.Client.WritePacketRTP(p.videoTrack, v.Packet)
+	case engine.AudioRTP:
+		p.Client.WritePacketRTP(p.audioTrack, v.Packet)
+	default:
+		p.RTSPSubscriber.OnEvent(event)
+	}
+}
+func (p *RTSPPusher) Connect() error {
+	if p.Transport == gortsplib.TransportTCP {
+		p.Transport = gortsplib.TransportUDP
+	} else {
+		p.Transport = gortsplib.TransportTCP
+	}
+	p.Client = &gortsplib.Client{
+		ReadBufferCount: rtspConfig.ReadBufferSize,
+		Transport:       &p.Transport,
 	}
 	// parse URL
-	u, err := base.ParseURL(client.URL)
+	u, err := url.Parse(p.RemoteURL)
 	if err != nil {
-		Printf("ParseURL:%s error:%v", client.URL, err)
+		p.Error("url.Parse", zap.Error(err))
-		return
+		return err
 	}
 	// connect to the server
-	if err = client.Start(u.Scheme, u.Host); err != nil {
+	if err = p.Client.Start(u.Scheme, u.Host); err != nil {
-		Printf("connect:%s error:%v", client.URL, err)
+		p.Error("Client.Start", zap.Error(err))
+		return err
+	}
+	p.SetIO(p.Client)
+	_, err = p.Client.Options(u)
+	return err
+}
+func (p *RTSPPusher) Push() (err error) {
+	var u *url.URL
+	u, err = url.Parse(p.RemoteURL)
+	// startTime := time.Now()
+	// for len(p.tracks) < 2 {
+	// 	if time.Sleep(time.Second); time.Since(startTime) > time.Second*10 {
+	// 		return fmt.Errorf("timeout")
+	// 	}
+	// }
+	if _, err = p.Announce(u, p.tracks); err != nil {
+		p.Error("Announce", zap.Error(err))
 		return
 	}
-	//client.close should be after connected!
+	err = p.SetupAll(p.tracks, u)
-	defer client.Client.Close()
-	var res *base.Response
-	if res, err = client.Options(u); err != nil {
-		Printf("option:%s error:%v", client.URL, err)
-		return
-	}
-	Println(res)
-	// find published tracks
-	tracks, baseURL, res, err := client.Describe(u)
 	if err != nil {
-		Printf("Describe:%s error:%v", client.URL, err)
+		p.Error("Setup", zap.Error(err))
 		return
 	}
-	Println(res)
+
-	if client.processFunc == nil {
+	if _, err = p.Record(); err != nil {
-		client.setTracks(tracks)
+		p.Error("Record", zap.Error(err))
-	}
-	for _, track := range tracks {
-		if res, err = client.Setup(true, track, baseURL, 0, 0); err != nil {
-			Printf("Setup:%s error:%v", baseURL.String(), err)
 		return
 	}
-		Println(res)
+	p.PlayRTP()
-	}
-	// start reading tracks
-	if res, err = client.Play(nil); err != nil {
-		Printf("Play:%s error:%v", baseURL.String(), err)
 	return
 }
-	Println(res)
-	// wait until a fatal error
-	var fatalChan = make(chan error)
-	go func() {
-		fatalChan <- client.Wait()
-	}()
-	select {
-	case err := <-fatalChan:
-		Printf("Wait:%s error:%v", baseURL.String(), err)
-	case <-client.Done():
-		Printf("client:%s done", client.URL)
-	}
-}

go.mod

@@ -1,11 +1,47 @@
-module github.com/Monibuca/plugin-rtsp/v3
+module m7s.live/plugin/rtsp/v4
 
-go 1.16
+go 1.18
 
 require (
-	github.com/Monibuca/engine/v3 v3.4.1
+	github.com/aler9/gortsplib/v2 v2.1.0
-	github.com/Monibuca/utils/v3 v3.0.5
+	github.com/pion/rtp v1.7.13
-	github.com/aler9/gortsplib v0.0.0-20211212220644-6f374e396529
+	go.uber.org/zap v1.23.0
-	github.com/pion/rtp v1.7.4
+	m7s.live/engine/v4 v4.11.0
-	github.com/pion/sdp/v3 v3.0.4
+)
+
+require (
+	github.com/cnotch/ipchub v1.1.0 // indirect
+	github.com/go-ole/go-ole v1.2.6 // indirect
+	github.com/go-task/slim-sprig v0.0.0-20210107165309-348f09dbbbc0 // indirect
+	github.com/golang/mock v1.6.0 // indirect
+	github.com/google/pprof v0.0.0-20210407192527-94a9f03dee38 // indirect
+	github.com/google/uuid v1.3.0 // indirect
+	github.com/kr/text v0.2.0 // indirect
+	github.com/logrusorgru/aurora v2.0.3+incompatible // indirect
+	github.com/lucas-clemente/quic-go v0.31.0 // indirect
+	github.com/lufia/plan9stats v0.0.0-20220913051719-115f729f3c8c // indirect
+	github.com/marten-seemann/qtls-go1-18 v0.1.3 // indirect
+	github.com/marten-seemann/qtls-go1-19 v0.1.1 // indirect
+	github.com/mcuadros/go-defaults v1.2.0 // indirect
+	github.com/onsi/ginkgo/v2 v2.2.0 // indirect
+	github.com/pion/randutil v0.1.0 // indirect
+	github.com/pion/rtcp v1.2.10 // indirect
+	github.com/pion/sdp/v3 v3.0.6 // indirect
+	github.com/pion/webrtc/v3 v3.1.49 // indirect
+	github.com/power-devops/perfstat v0.0.0-20220216144756-c35f1ee13d7c // indirect
+	github.com/q191201771/naza v0.30.8 // indirect
+	github.com/shirou/gopsutil/v3 v3.22.10 // indirect
+	github.com/tklauser/go-sysconf v0.3.11 // indirect
+	github.com/tklauser/numcpus v0.6.0 // indirect
+	github.com/yusufpapurcu/wmi v1.2.2 // indirect
+	go.uber.org/atomic v1.10.0 // indirect
+	go.uber.org/multierr v1.8.0 // indirect
+	golang.org/x/crypto v0.3.0 // indirect
+	golang.org/x/exp v0.0.0-20221126150942-6ab00d035af9 // indirect
+	golang.org/x/mod v0.7.0 // indirect
+	golang.org/x/net v0.7.0 // indirect
+	golang.org/x/sync v0.1.0 // indirect
+	golang.org/x/sys v0.5.0 // indirect
+	golang.org/x/tools v0.3.0 // indirect
+	gopkg.in/yaml.v3 v3.0.1 // indirect
 )

go.sum

@@ -1,14 +1,10 @@
 github.com/BurntSushi/toml v0.3.1/go.mod h1:xHWCNGjB5oqiDr8zfno3MHue2Ht5sIBksp03qcyfWMU=
-github.com/BurntSushi/toml v0.4.1 h1:GaI7EiDXDRfa8VshkTj7Fym7ha+y8/XxIgD2okUIjLw=
+github.com/aler9/gortsplib/v2 v2.1.0 h1:HKQjFugTGH32FScSXPL2p4SGLpxGBRnyBSzR3GKYG9U=
-github.com/BurntSushi/toml v0.4.1/go.mod h1:CxXYINrC8qIiEnFrOxCa7Jy5BFHlXnUU2pbicEuybxQ=
+github.com/aler9/gortsplib/v2 v2.1.0/go.mod h1:jZy239Qg+C9OSvJrPs2vIPalm9/pam1ljFir5DdydnA=
-github.com/Monibuca/engine/v3 v3.4.1 h1:Ap2VbwTkMUkv80NPeUX2sNdV5Vz5nPVoU/6RU51PSAc=
+github.com/benbjohnson/clock v1.1.0 h1:Q92kusRqC1XV2MjkWETPvjJVqKetz1OzxZB7mHJLju8=
-github.com/Monibuca/engine/v3 v3.4.1/go.mod h1:rgAUey5ziRhlh6WugWyA5fYKyGOvcwhtTMDk4sukE7E=
+github.com/chzyer/logex v1.1.10/go.mod h1:+Ywpsq7O8HXn0nuIou7OrIPyXbp3wmkHB+jjWRnGsAI=
-github.com/Monibuca/utils/v3 v3.0.5 h1:w14x0HkWTbF4MmHbINLlOwe4VJNoSOeaQChMk5E/4es=
+github.com/chzyer/readline v0.0.0-20180603132655-2972be24d48e/go.mod h1:nSuG5e5PlCu98SY8svDHJxuZscDgtXS6KTTbou5AhLI=
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 golang.org/x/term v0.0.0-20201126162022-7de9c90e9dd1/go.mod h1:bj7SfCRtBDWHUb9snDiAeCFNEtKQo2Wmx5Cou7ajbmo=
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 golang.org/x/text v0.3.6/go.mod h1:5Zoc/QRtKVWzQhOtBMvqHzDpF6irO9z98xDceosuGiQ=
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 golang.org/x/tools v0.0.0-20180917221912-90fa682c2a6e/go.mod h1:n7NCudcB/nEzxVGmLbDWY5pfWTLqBcC2KZ6jyYvM4mQ=
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 gopkg.in/fsnotify.v1 v1.4.7/go.mod h1:Tz8NjZHkW78fSQdbUxIjBTcgA1z1m8ZHf0WmKUhAMys=
 gopkg.in/natefinch/lumberjack.v2 v2.0.0/go.mod h1:l0ndWWf7gzL7RNwBG7wST/UCcT4T24xpD6X8LsfU/+k=
 gopkg.in/tomb.v1 v1.0.0-20141024135613-dd632973f1e7/go.mod h1:dt/ZhP58zS4L8KSrWDmTeBkI65Dw0HsyUHuEVlX15mw=
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+gopkg.in/yaml.v3 v3.0.1/go.mod h1:K4uyk7z7BCEPqu6E+C64Yfv1cQ7kz7rIZviUmN+EgEM=
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+m7s.live/engine/v4 v4.11.0/go.mod h1:dx52RjysN9uocotAP2caLyb6j2uWwjcnstOy8C/U8sU=

main.go

@@ -1,112 +1,111 @@
 package rtsp
 
 import (
-	"encoding/json"
-	"fmt"
-	"log"
 	"net/http"
+	"strconv"
+	"sync"
 	"time"
 
-	. "github.com/Monibuca/engine/v3"
+	"github.com/aler9/gortsplib/v2"
-	. "github.com/Monibuca/utils/v3"
+	"go.uber.org/zap"
-	"github.com/aler9/gortsplib"
+	. "m7s.live/engine/v4"
+	"m7s.live/engine/v4/config"
+	"m7s.live/engine/v4/util"
 )
 
-var config = struct {
+type RTSPConfig struct {
+	config.Publish
+	config.Subscribe
+	config.Pull
+	config.Push
 	ListenAddr     string
 	UDPAddr        string
 	RTCPAddr       string
-	Timeout      int
+	ReadBufferSize int
-	Reconnect    bool
+	PullProtocol   string //tcp、udp、 auto（default）
-	AutoPullList map[string]string
+	sync.Map
-	AutoPushList map[string]string
-}{":554", ":8000", ":8001", 0, false, nil, nil}
-
-var pconfig = PluginConfig{
-	Name:   "RTSP",
-	Config: &config,
 }
 
-func init() {
+func (conf *RTSPConfig) OnEvent(event any) {
-	pconfig.Install(runPlugin)
+	switch v := event.(type) {
-}
+	case FirstConfig:
-
-func getRtspList() (info []*Stream) {
-	for _, s := range Streams.ToList() {
-		switch s.ExtraProp.(type) {
-		case *RTSPublisher:
-			info = append(info, s)
-		case *RTSPClient:
-			info = append(info, s)
-		}
-	}
-	return
-}
-func runPlugin() {
-	http.HandleFunc("/api/rtsp/list", func(w http.ResponseWriter, r *http.Request) {
-		CORS(w, r)
-		if r.URL.Query().Get("json") != "" {
-			if jsonData, err := json.Marshal(getRtspList()); err == nil {
-				w.Write(jsonData)
-			} else {
-				w.WriteHeader(500)
-			}
-			return
-		}
-		sse := NewSSE(w, r.Context())
-		var err error
-		for tick := time.NewTicker(time.Second); err == nil; <-tick.C {
-			err = sse.WriteJSON(getRtspList())
-		}
-	})
-	http.HandleFunc("/api/rtsp/pull", func(w http.ResponseWriter, r *http.Request) {
-		CORS(w, r)
-		targetURL := r.URL.Query().Get("target")
-		streamPath := r.URL.Query().Get("streamPath")
-		save := r.URL.Query().Get("save")
-		if err := (&RTSPClient{Transport: gortsplib.TransportTCP}).PullStream(streamPath, targetURL); err == nil {
-			if save == "1" {
-				if config.AutoPullList == nil {
-					config.AutoPullList = make(map[string]string)
-				}
-				config.AutoPullList[streamPath] = targetURL
-				if err := pconfig.Save(); err != nil {
-					Println(err)
-				}
-			}
-			w.Write([]byte(`{"code":0}`))
-		} else {
-			w.Write([]byte(fmt.Sprintf(`{"code":1,"msg":"%s"}`, err.Error())))
-		}
-	})
-	for streamPath, url := range config.AutoPullList {
-		if err := (&RTSPClient{Transport: gortsplib.TransportTCP}).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))
-	}
-}
-
-func ListenRtsp(addr string) error {
-	defer log.Println("rtsp server start!")
 		s := &gortsplib.Server{
-		Handler:           &RTSPServer{},
+			Handler:           conf,
-		RTSPAddress:       addr,
+			RTSPAddress:       conf.ListenAddr,
-		UDPRTPAddress:     config.UDPAddr,
+			UDPRTPAddress:     conf.UDPAddr,
-		UDPRTCPAddress:    config.RTCPAddr,
+			UDPRTCPAddress:    conf.RTCPAddr,
 			MulticastIPRange:  "224.1.0.0/16",
 			MulticastRTPPort:  8002,
 			MulticastRTCPPort: 8003,
 		}
-	return s.StartAndWait()
+		if err := s.Start(); err != nil {
+			RTSPPlugin.Error("server start", zap.Error(err))
+			v["enable"] = false
+		}
+		for streamPath, url := range conf.PullOnStart {
+			if err := RTSPPlugin.Pull(streamPath, url, new(RTSPPuller), 0); err != nil {
+				RTSPPlugin.Error("pull", zap.String("streamPath", streamPath), zap.String("url", url), zap.Error(err))
+			}
+		}
+	case SEpublish:
+		for streamPath, url := range conf.PushList {
+			if streamPath == v.Stream.Path {
+				if err := RTSPPlugin.Push(streamPath, url, new(RTSPPusher), false); err != nil {
+					RTSPPlugin.Error("push", zap.String("streamPath", streamPath), zap.String("url", url), zap.Error(err))
+				}
+			}
+		}
+	case *Stream: //按需拉流
+		for streamPath, url := range conf.PullOnSub {
+			if streamPath == v.Path {
+				if err := RTSPPlugin.Pull(streamPath, url, new(RTSPPuller), 0); err != nil {
+					RTSPPlugin.Error("pull", zap.String("streamPath", streamPath), zap.String("url", url), zap.Error(err))
+				}
+				break
+			}
+		}
+	}
+}
+
+var rtspConfig = &RTSPConfig{
+	ListenAddr:     ":554",
+	UDPAddr:        ":8000",
+	RTCPAddr:       ":8001",
+	ReadBufferSize: 2048,
+}
+var RTSPPlugin = InstallPlugin(rtspConfig)
+
+func filterStreams() (ss []*Stream) {
+	Streams.RLock()
+	defer Streams.RUnlock()
+	for _, s := range Streams.Map {
+		switch s.Publisher.(type) {
+		case *RTSPPublisher, *RTSPPuller:
+			ss = append(ss, s)
+		}
+	}
+	return
+}
+
+func (*RTSPConfig) API_list(w http.ResponseWriter, r *http.Request) {
+	util.ReturnJson(filterStreams, time.Second, w, r)
+}
+
+func (*RTSPConfig) API_Pull(rw http.ResponseWriter, r *http.Request) {
+	save, _ := strconv.Atoi(r.URL.Query().Get("save"))
+	err := RTSPPlugin.Pull(r.URL.Query().Get("streamPath"), r.URL.Query().Get("target"), new(RTSPPuller), save)
+	if err != nil {
+		http.Error(rw, err.Error(), http.StatusBadRequest)
+	} else {
+		rw.Write([]byte("ok"))
+	}
+}
+
+func (*RTSPConfig) API_Push(rw http.ResponseWriter, r *http.Request) {
+	err := RTSPPlugin.Push(r.URL.Query().Get("streamPath"), r.URL.Query().Get("target"), new(RTSPPusher), r.URL.Query().Has("save"))
+	if err != nil {
+		http.Error(rw, err.Error(), http.StatusBadRequest)
+	} else {
+		rw.Write([]byte("ok"))
+	}
 }

payloader.go

@@ -1,24 +0,0 @@
-package rtsp
-
-// AACayloader payloads AAC packets
-type AACPayloader struct{}
-
-// Payload fragments an AAC packet across one or more byte arrays
-func (p *AACPayloader) Payload(mtu uint16, payload []byte) [][]byte {
-	var out [][]byte
-	o := make([]byte, len(payload)+4)
-	//AU_HEADER_LENGTH,因为单位是bit, 除以8就是auHeader的字节长度；又因为单个auheader字节长度2字节，所以再除以2就是auheader的个数。
-	o[0] = 0x00 //高位
-	o[1] = 0x10 //低位
-	//AU_HEADER
-	o[2] = (byte)((len(payload) & 0x1fe0) >> 5) //高位
-	o[3] = (byte)((len(payload) & 0x1f) << 3)   //低位
-	copy(o[4:], payload)
-	return append(out, o)
-}
-
-type H265Payloader struct{}
-
-func (p *H265Payloader) Payload(mtu uint16, payload []byte) [][]byte {
-	return nil
-}

publisher.go

@@ -1,121 +1,78 @@
 package rtsp
 
 import (
-	"encoding/base64"
+	"github.com/aler9/gortsplib/v2/pkg/codecs/mpeg4audio"
-	"encoding/hex"
+	"github.com/aler9/gortsplib/v2/pkg/format"
-	"strconv"
+	"github.com/aler9/gortsplib/v2/pkg/media"
-	"strings"
+	"github.com/pion/rtp"
-
+	"go.uber.org/zap"
-	. "github.com/Monibuca/engine/v3"
+	. "m7s.live/engine/v4"
-	. "github.com/Monibuca/utils/v3"
+	"m7s.live/engine/v4/common"
-	"github.com/aler9/gortsplib"
+	. "m7s.live/engine/v4/track"
 )
 
-type RTSPublisher struct {
+type RTSPPublisher struct {
-	*Stream     `json:"-"`
+	Publisher
-	stream      *gortsplib.ServerStream
+	Tracks map[*media.Media]common.AVTrack `json:"-"`
-	processFunc []func([]byte)
+	RTSPIO
 }
 
-func (p *RTSPublisher) setTracks(tracks gortsplib.Tracks) {
+func (p *RTSPPublisher) SetTracks() error {
-	if p.processFunc != nil {
+	p.Tracks = make(map[*media.Media]common.AVTrack, len(p.tracks))
-		p.processFunc = p.processFunc[:len(tracks)]
+	defer func() {
-		return
+		for _, track := range p.Tracks {
-	} else {
+			p.Info("set track", zap.String("name", track.GetBase().Name))
-		p.processFunc = make([]func([]byte), len(tracks))
 		}
-	for i, track := range tracks {
+	}()
-		v, ok := track.Media.Attribute("rtpmap")
+	for _, track := range p.tracks {
-		if !ok {
+		for _, forma := range track.Formats {
-			continue
+			switch f := forma.(type) {
+			case *format.H264:
+				vt := NewH264(p.Stream, f.PayloadType())
+				p.Tracks[track] = vt
+				if len(f.SPS) > 0 {
+					vt.WriteSliceBytes(f.SPS)
+				}
+				if len(f.PPS) > 0 {
+					vt.WriteSliceBytes(f.PPS)
+				}
+			case *format.H265:
+				vt := NewH265(p.Stream, f.PayloadType())
+				p.Tracks[track] = vt
+				if len(f.VPS) > 0 {
+					vt.WriteSliceBytes(f.VPS)
+				}
+				if len(f.SPS) > 0 {
+					vt.WriteSliceBytes(f.SPS)
+				}
+				if len(f.PPS) > 0 {
+					vt.WriteSliceBytes(f.PPS)
+				}
+			case *format.MPEG4Audio:
+				at := NewAAC(p.Stream, f.PayloadType(), uint32(f.Config.SampleRate))
+				p.Tracks[track] = at
+				at.IndexDeltaLength = f.IndexDeltaLength
+				at.IndexLength = f.IndexLength
+				at.SizeLength = f.SizeLength
+				if f.Config.Type == mpeg4audio.ObjectTypeAACLC {
+					at.Mode = 1
+				}
+				at.Channels = uint8(f.Config.ChannelCount)
+				asc, _ := f.Config.Marshal()
+				// 复用AVCC写入逻辑，解析出AAC的配置信息
+				at.WriteSequenceHead(append([]byte{0xAF, 0x00}, asc...))
+			case *format.G711:
+				at := NewG711(p.Stream, !f.MULaw, f.PayloadType(), uint32(f.ClockRate()))
+				p.Tracks[track] = at
+				at.AVCCHead = []byte{(byte(at.CodecID) << 4) | (1 << 1)}
+			}
+		}
+	}
+	return nil
 }
 
-		fmtp := make(map[string]string)
+func (p *RTSPPublisher) OnPacket(m *media.Media, f format.Format, pack *rtp.Packet) {
-		if v, ok := track.Media.Attribute("fmtp"); ok {
+	if t, ok := p.Tracks[m]; ok {
-			if tmp := strings.SplitN(v, " ", 2); len(tmp) == 2 {
+		t.WriteRTPPack(pack)
-				for _, kv := range strings.Split(tmp[1], ";") {
-					kv = strings.Trim(kv, " ")
-
-					if len(kv) == 0 {
-						continue
-					}
-					tmp := strings.SplitN(kv, "=", 2)
-					if len(tmp) == 2 {
-						fmtp[strings.TrimSpace(tmp[0])] = tmp[1]
-					}
-				}
-			}
-		}
-
-		v = strings.TrimSpace(v)
-		vals := strings.Split(v, " ")
-		if len(vals) != 2 {
-			continue
-		}
-		timeScale := 0
-		keyval := strings.Split(vals[1], "/")
-		if i, err := strconv.Atoi(keyval[1]); err == nil {
-			timeScale = i
-		}
-		if len(keyval) >= 2 {
-			Printf("track %d is %s", i, keyval[0])
-			switch strings.ToLower(keyval[0]) {
-			case "h264":
-				vt := p.NewRTPVideo(7)
-				if conf, err := track.ExtractConfigH264(); err == nil {
-					vt.PushNalu(0, 0, conf.SPS, conf.PPS)
-				}
-				p.processFunc[i] = vt.Push
-			case "h265", "hevc":
-				vt := p.NewRTPVideo(12)
-				if v, ok := fmtp["sprop-vps"]; ok {
-					vps, _ := base64.StdEncoding.DecodeString(v)
-					vt.PushNalu(0, 0, vps)
-				}
-				if v, ok := fmtp["sprop-sps"]; ok {
-					sps, _ := base64.StdEncoding.DecodeString(v)
-					vt.PushNalu(0, 0, sps)
-				}
-				if v, ok := fmtp["sprop-pps"]; ok {
-					pps, _ := base64.StdEncoding.DecodeString(v)
-					vt.PushNalu(0, 0, pps)
-				}
-				p.processFunc[i] = vt.Push
-			case "pcma":
-				at := p.NewRTPAudio(7)
-				at.SoundRate = timeScale
-				at.SoundSize = 16
-				if len(keyval) >= 3 {
-					x, _ := strconv.Atoi(keyval[2])
-					at.Channels = byte(x)
-				} else {
-					at.Channels = 1
-				}
-				at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
-				p.processFunc[i] = at.Push
-			case "pcmu":
-				at := p.NewRTPAudio(8)
-				at.SoundRate = timeScale
-				at.SoundSize = 16
-				if len(keyval) >= 3 {
-					x, _ := strconv.Atoi(keyval[2])
-					at.Channels = byte(x)
-				} else {
-					at.Channels = 1
-				}
-				at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
-				p.processFunc[i] = at.Push
-			case "mpeg4-generic":
-				at := p.NewRTPAudio(0)
-				if config, ok := fmtp["config"]; ok {
-					asc, _ := hex.DecodeString(config)
-					at.SetASC(asc)
-				} else {
-					Println("aac no config")
-				}
-				at.SoundSize = 16
-				p.processFunc[i] = at.Push
-			}
-		}
 	}
 }

server.go

@@ -1,256 +1,118 @@
 package rtsp
 
 import (
-	"fmt"
+	"github.com/aler9/gortsplib/v2"
-	"sync"
+	"github.com/aler9/gortsplib/v2/pkg/base"
-	"unsafe"
+	"github.com/aler9/gortsplib/v2/pkg/media"
-
+	. "m7s.live/engine/v4"
-	"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"
 )
 
-// 接收RTSP推流：OnConnOpen->OnAnnounce->OnSetup->OnSessionOpen
+type RTSPIO struct {
-// 接收RTSP拉流：OnConnOpen->OnDescribe->OnSetup->OnSessionOpen
+	tracks     media.Medias
-type RTSPServer struct {
-	sync.Map
-}
-type RTSPSubscriber struct {
 	stream     *gortsplib.ServerStream
-	engine.Subscriber
+	audioTrack *media.Media
-	vt *engine.VideoTrack
+	videoTrack *media.Media
-	at *engine.AudioTrack
 }
 
-// called after a connection is opened.
+func (conf *RTSPConfig) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
-func (sh *RTSPServer) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
+	RTSPPlugin.Debug("conn opened")
-	Printf("rtsp conn opened")
 }
 
-// called after a connection is closed.
+func (conf *RTSPConfig) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
-func (sh *RTSPServer) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
+	RTSPPlugin.Debug("conn closed")
-	Printf("rtsp conn closed (%v)", ctx.Error)
+	if p, ok := conf.LoadAndDelete(ctx.Conn); ok {
-	if p, ok := sh.Load(ctx.Conn); ok {
+		p.(IIO).Stop()
-		switch v := p.(type) {
-		case *RTSPublisher:
-			v.Close()
-		case *RTSPSubscriber:
-			v.Close()
-		}
-		sh.Delete(ctx.Conn)
 	}
 }
 
-// called after a session is opened.
+func (conf *RTSPConfig) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
-func (sh *RTSPServer) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
+	RTSPPlugin.Debug("session opened")
-	Printf("rtsp session opened")
 }
 
-// called after a session is closed.
+func (conf *RTSPConfig) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
-func (sh *RTSPServer) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
+	RTSPPlugin.Debug("session closed")
-	Printf("rtsp session closed")
+	if p, ok := conf.LoadAndDelete(ctx.Session); ok {
-	if v, ok := sh.LoadAndDelete(ctx.Session); ok {
+		p.(IIO).Stop()
-		switch v := v.(type) {
-		case *RTSPublisher:
-			v.Close()
-		case *RTSPSubscriber:
-			v.Close()
-		}
 	}
 }
 
 // called after receiving a DESCRIBE request.
-func (sh *RTSPServer) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
+func (conf *RTSPConfig) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
-	Printf("describe request")
+	RTSPPlugin.Debug("describe request")
-	var err error
+	var suber RTSPSubscriber
-	if s := engine.FindStream(ctx.Path); s != nil {
+	suber.SetIO(ctx.Conn.NetConn())
-		var tracks gortsplib.Tracks
+	streamPath := ctx.Path
-		var stream *gortsplib.ServerStream
+	if ctx.Query != "" {
-		var sub RTSPSubscriber
+		streamPath = streamPath + "?" + ctx.Query
-		sub.Type = "RTSP pull"
-		sub.vt = s.WaitVideoTrack("h264", "h265")
-		sub.at = s.WaitAudioTrack("aac", "pcma", "pcmu")
-		ssrc := uintptr(unsafe.Pointer(stream))
-		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 nil, nil, err
 	}
-			case codec.CodecID_H265:
+	if err := RTSPPlugin.Subscribe(streamPath, &suber); err == nil {
-				vtrack = NewH265Track(96, sub.vt.ExtraData.NALUs)
+		RTSPPlugin.Debug("describe replay ok")
-				vpacketer = rtp.NewPacketizer(1200, 96, uint32(ssrc), &H265Payloader{}, rtp.NewFixedSequencer(1), 90000)
+		conf.Store(ctx.Conn, &suber)
-			}
-			var st uint32
-			onVideo := func(ts uint32, pack *engine.VideoPack) {
-				for _, nalu := range pack.NALUs {
-					for _, pack := range vpacketer.Packetize(nalu, (ts-st)*90) {
-						rtp, _ := pack.Marshal()
-						stream.WritePacketRTP(trackId, rtp)
-					}
-				}
-				st = ts
-			}
-			sub.OnVideo = func(ts uint32, pack *engine.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.at.CodecID])
-				apacketizer := rtp.NewPacketizer(1200, 97, uint32(ssrc), &codecs.G711Payloader{}, rtp.NewFixedSequencer(1), 8000)
-				sub.OnAudio = func(ts uint32, pack *engine.AudioPack) {
-					for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*8) {
-						buf, _ := pack.Marshal()
-						stream.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 *engine.AudioPack) {
-						for _, pack := range apacketizer.Packetize(pack.Raw, (ts-st)*uint32(mpegConf.SampleRate)/1000) {
-							buf, _ := pack.Marshal()
-							stream.WritePacketRTP(trackId, buf)
-						}
-						st = ts
-					}
-					tracks = append(tracks, atrack)
-				}
-			}
-		}
-		stream = gortsplib.NewServerStream(tracks)
-		sub.stream = stream
-		sh.Store(ctx.Conn, &sub)
 		return &base.Response{
 			StatusCode: base.StatusOK,
-		}, stream, nil
+		}, suber.stream, nil
-		// if stream, ok := s.ExtraProp.(*gortsplib.ServerStream); ok {
+	} else {
-		// 	return &base.Response{
-		// 		StatusCode: base.StatusOK,
-		// 	}, stream, nil
-		// }
-	}
 		return &base.Response{
 			StatusCode: base.StatusNotFound,
-	}, nil, nil
+		}, suber.stream, nil
+	}
 }
 
-// called after receiving an ANNOUNCE request.
+func (conf *RTSPConfig) OnSetup(ctx *gortsplib.ServerHandlerOnSetupCtx) (*base.Response, *gortsplib.ServerStream, error) {
-func (sh *RTSPServer) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
+	var resp base.Response
-	Printf("announce request")
+	resp.StatusCode = base.StatusOK
-	p := &RTSPublisher{
+	if p, ok := conf.Load(ctx.Conn); ok {
-		Stream: &engine.Stream{
+		switch v := p.(type) {
-			StreamPath: ctx.Path,
+		case *RTSPSubscriber:
-			Type:       "RTSP push",
+			return &resp, v.stream, nil
-		},
+		case *RTSPPublisher:
+			return &resp, v.stream, nil
 		}
-	p.ExtraProp = p
+	}
-	p.URL = ctx.Req.URL.String()
+	resp.StatusCode = base.StatusNotFound
-	if p.Publish() {
+	return &resp, nil, nil
-		p.setTracks(ctx.Tracks)
+}
-		p.stream = gortsplib.NewServerStream(ctx.Tracks)
+
-		sh.Store(ctx.Conn, p)
+func (conf *RTSPConfig) OnPlay(ctx *gortsplib.ServerHandlerOnPlayCtx) (*base.Response, error) {
-		sh.Store(ctx.Session, p)
+	var resp base.Response
+	resp.StatusCode = base.StatusNotFound
+	if p, ok := conf.Load(ctx.Conn); ok {
+		switch v := p.(type) {
+		case *RTSPSubscriber:
+			resp.StatusCode = base.StatusOK
+			go func() {
+				v.PlayRTP()
+				ctx.Session.Close()
+			}()
+		}
+	}
+	return &resp, nil
+}
+func (conf *RTSPConfig) OnRecord(ctx *gortsplib.ServerHandlerOnRecordCtx) (*base.Response, error) {
+	if p, ok := conf.Load(ctx.Session); ok {
+		ctx.Session.OnPacketRTPAny(p.(*RTSPPublisher).OnPacket)
+	}
+	return &base.Response{
+		StatusCode: base.StatusOK,
+	}, nil
+}
+func (conf *RTSPConfig) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
+	p := &RTSPPublisher{}
+	p.SetIO(ctx.Conn.NetConn())
+	if err := RTSPPlugin.Publish(ctx.Path, p); err == nil {
+		p.tracks = ctx.Medias
+		p.stream = gortsplib.NewServerStream(ctx.Medias)
+		if err = p.SetTracks(); err != nil {
+			return nil, err
+		}
+		conf.Store(ctx.Conn, p)
+		conf.Store(ctx.Session, p)
 	} else {
 		return &base.Response{
 			StatusCode: base.StatusBadRequest,
-		}, fmt.Errorf("streamPath is already exist")
+		}, nil
 	}
 	return &base.Response{
 		StatusCode: base.StatusOK,
 	}, nil
 }
-
-// called after receiving a SETUP request.
-func (sh *RTSPServer) OnSetup(ctx *gortsplib.ServerHandlerOnSetupCtx) (*base.Response, *gortsplib.ServerStream, error) {
-	Printf("setup request")
-	if p, ok := sh.Load(ctx.Conn); ok {
-		switch v := p.(type) {
-		case *RTSPublisher:
-			return &base.Response{
-				StatusCode: base.StatusOK,
-			}, v.stream, nil
-		case *RTSPSubscriber:
-			return &base.Response{
-				StatusCode: base.StatusOK,
-			}, v.stream, nil
-		}
-	}
-	return &base.Response{
-		StatusCode: base.StatusNotFound,
-	}, nil, nil
-}
-
-// called after receiving a PLAY request.
-func (sh *RTSPServer) OnPlay(ctx *gortsplib.ServerHandlerOnPlayCtx) (*base.Response, error) {
-	Printf("play request")
-	if p, ok := sh.Load(ctx.Conn); ok {
-		if sub := p.(*RTSPSubscriber); sub.Subscribe(ctx.Path) == nil {
-			go func() {
-				sub.Play(sub.at, sub.vt)
-				ctx.Conn.Close()
-			}()
-			return &base.Response{
-				StatusCode: base.StatusOK,
-			}, nil
-		}
-	}
-	return &base.Response{
-		StatusCode: base.StatusNotFound,
-	}, nil
-}
-
-// called after receiving a RECORD request.
-func (sh *RTSPServer) OnRecord(ctx *gortsplib.ServerHandlerOnRecordCtx) (*base.Response, error) {
-	Printf("record request")
-	return &base.Response{
-		StatusCode: base.StatusOK,
-	}, nil
-}
-
-// called after receiving a frame.
-func (sh *RTSPServer) OnPacketRTP(ctx *gortsplib.ServerHandlerOnPacketRTPCtx) {
-	if p, ok := sh.Load(ctx.Session); ok {
-		rtsp := p.(*RTSPublisher)
-		if rtsp.Err() != nil {
-			ctx.Session.Close()
-			return
-		}
-		if f := rtsp.processFunc[ctx.TrackID]; f != nil {
-			f(ctx.Payload)
-		}
-	}
-}

subscriber.go

@@ -0,0 +1,100 @@
+package rtsp
+
+import (
+	"github.com/aler9/gortsplib/v2"
+	"github.com/aler9/gortsplib/v2/pkg/codecs/mpeg4audio"
+	"github.com/aler9/gortsplib/v2/pkg/format"
+	"github.com/aler9/gortsplib/v2/pkg/media"
+	. "m7s.live/engine/v4"
+	"m7s.live/engine/v4/codec"
+	"m7s.live/engine/v4/track"
+)
+
+type RTSPSubscriber struct {
+	Subscriber
+	RTSPIO
+}
+
+func (s *RTSPSubscriber) OnEvent(event any) {
+	switch v := event.(type) {
+	case *track.Video:
+		if s.Video != nil {
+			return
+		}
+		switch v.CodecID {
+		case codec.CodecID_H264:
+			video := &media.Media{
+				Type: media.TypeVideo,
+				Formats: []format.Format{&format.H264{
+					PacketizationMode: 1,
+					PayloadTyp:        v.PayloadType,
+					SPS:               v.ParamaterSets[0],
+					PPS:               v.ParamaterSets[1],
+				}},
+			}
+			s.videoTrack = video
+			s.tracks = append(s.tracks, video)
+		case codec.CodecID_H265:
+			video := &media.Media{
+				Type: media.TypeVideo,
+				Formats: []format.Format{&format.H265{
+					PayloadTyp: v.PayloadType,
+					VPS:        v.ParamaterSets[0],
+					SPS:        v.ParamaterSets[1],
+					PPS:        v.ParamaterSets[2],
+				}},
+			}
+			s.videoTrack = video
+			s.tracks = append(s.tracks, video)
+		}
+		s.AddTrack(v)
+	case *track.Audio:
+		if s.Audio != nil {
+			return
+		}
+		switch v.CodecID {
+		case codec.CodecID_AAC:
+			audio := &media.Media{
+				Type: media.TypeAudio,
+				Formats: []format.Format{&format.MPEG4Audio{
+					PayloadTyp: v.PayloadType,
+					Config: &mpeg4audio.Config{
+						Type:         mpeg4audio.ObjectTypeAACLC,
+						SampleRate:   int(v.SampleRate),
+						ChannelCount: int(v.Channels),
+					},
+					SizeLength:       v.SizeLength,
+					IndexLength:      v.IndexLength,
+					IndexDeltaLength: v.IndexDeltaLength,
+				}},
+			}
+			s.audioTrack = audio
+			s.tracks = append(s.tracks, audio)
+		case codec.CodecID_PCMA:
+			audio := &media.Media{
+				Type:    media.TypeAudio,
+				Formats: []format.Format{&format.G711{}},
+			}
+			s.audioTrack = audio
+			s.tracks = append(s.tracks, audio)
+		case codec.CodecID_PCMU:
+			audio := &media.Media{
+				Type: media.TypeAudio,
+				Formats: []format.Format{&format.G711{
+					MULaw: true,
+				}},
+			}
+			s.audioTrack = audio
+			s.tracks = append(s.tracks, audio)
+		}
+		s.AddTrack(v)
+	case ISubscriber:
+		s.stream = gortsplib.NewServerStream(s.tracks)
+	case VideoRTP:
+		s.stream.WritePacketRTP(s.videoTrack, v.Packet)
+	case AudioRTP:
+		s.stream.WritePacketRTP(s.audioTrack, v.Packet)
+	default:
+		s.Subscriber.OnEvent(event)
+	}
+}

track.go

@@ -1,87 +0,0 @@
-package rtsp
-
-import (
-	"encoding/base64"
-	"fmt"
-	"strconv"
-
-	"github.com/aler9/gortsplib"
-	psdp "github.com/pion/sdp/v3"
-)
-
-// func NewTrackAAC(payloadType uint8, conf *gortsplib.TrackConfigAAC) (*gortsplib.Track, error) {
-// 	mpegConf, err := aac.MPEG4AudioConfig{
-// 		Type:              aac.MPEG4AudioType(conf.Type),
-// 		SampleRate:        conf.SampleRate,
-// 		ChannelCount:      conf.ChannelCount,
-// 		AOTSpecificConfig: conf.AOTSpecificConfig,
-// 	}.Encode()
-// 	if err != nil {
-// 		return nil, err
-// 	}
-
-// 	typ := strconv.FormatInt(int64(payloadType), 10)
-
-// 	return &gortsplib.Track{
-// 		Media: &psdp.MediaDescription{
-// 			MediaName: psdp.MediaName{
-// 				Media:   "audio",
-// 				Protos:  []string{"RTP", "AVP"},
-// 				Formats: []string{typ},
-// 			},
-// 			Attributes: []psdp.Attribute{
-// 				{
-// 					Key: "rtpmap",
-// 					Value: typ + " mpeg4-generic/" + strconv.FormatInt(int64(conf.SampleRate), 10) +
-// 						"/" + strconv.FormatInt(int64(conf.ChannelCount), 10),
-// 				},
-// 				{
-// 					Key: "fmtp",
-// 					Value: typ + " profile-level-id=1; " +
-// 						"mode=AAC-hbr; " +
-// 						"sizelength=6; " +
-// 						"indexlength=2; " +
-// 						"indexdeltalength=2; " +
-// 						"config=" + hex.EncodeToString(mpegConf),
-// 				},
-// 			},
-// 		},
-// 	}, nil
-// }
-func NewG711Track(payloadType uint8, law string) *gortsplib.Track {
-	return &gortsplib.Track{
-		Media: &psdp.MediaDescription{
-			MediaName: psdp.MediaName{
-				Media:   "audio",
-				Protos:  []string{"RTP", "AVP"},
-				Formats: []string{strconv.FormatInt(int64(payloadType), 10)}},
-			Attributes: []psdp.Attribute{
-				{
-					Key:   "rtpmap",
-					Value: fmt.Sprintf("%d %s/8000/1", payloadType, law),
-				},
-			},
-		},
-	}
-}
-func NewH265Track(payloadType uint8, sprop [][]byte) *gortsplib.Track {
-	return &gortsplib.Track{
-		Media: &psdp.MediaDescription{
-			MediaName: psdp.MediaName{
-				Media:   "video",
-				Protos:  []string{"RTP", "AVP"},
-				Formats: []string{fmt.Sprintf("%d", payloadType)},
-			},
-			Attributes: []psdp.Attribute{
-				{
-					Key:   "rtpmap",
-					Value: fmt.Sprintf("%d H265/90000", payloadType),
-				},
-				{
-					Key:   "fmtp",
-					Value: 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])),
-				},
-			},
-		},
-	}
-}