Merge pull request #20 from ziminghua/aler9

解决发布超时后，重连RTSP，无法播放的BUG

.gitignore

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

README.en.md

@@ -1,70 +0,0 @@
-_[简体中文](https://github.com/Monibuca/plugin-rtsp) | English_
-# RTSP Plugin
-
-The RTSP plugin provides the ability to push and pull the RTSP protocol and also to push and pull the RTSP protocol to remote servers.
-
-## Plugin address
-
-https://github.com/Monibuca/plugin-rtsp
-
-## Plugin introduction
-
-```go
-import (
-    _ "m7s.live/plugin/rtsp/v4"
-)
-```
-
-## Push and Pull address form
-
-```
-rtsp://localhost/live/test
-```
-- `localhost` is the m7s server domain name or IP address, and the default port `554` can be omitted, otherwise it is required to be written.
-- `live` represents `appName`
-- `test` represents `streamName`
-- `live/test` in m7s will serve as the stream identity.
-
-For example, push stream to m7s through ffmpeg
-
-```bash
-ffmpeg -i [video source] -c:v h264 -c:a aac -f rtsp rtsp://localhost/live/test
-```
-
-This will create a stream named `live/test` inside m7s.
-
-If the `live/test` stream already exists in m7s, then you can use the RTSP protocol to play it.
-
-```bash
-ffplay rtsp://localhost/live/test
-```
-
-## Configuration
-
-```yaml
-rtsp:
-    publish: # Refer to the global configuration format
-    subscribe: # Refer to the global configuration format
-    pull: # Format reference document https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
-    push: # Format reference document https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
-    listenaddr: :554
-    udpaddr: :8000
-    rtcpaddr: :8001
-    readbuffercount: 2048
-    writebuffercount: 2048
-```
-:::tip Configuration override
-publish and subscribe, any section not configured will use global configuration.
-:::
-
-## API
-
-### `rtsp/api/list`
-Get all RTSP streams
-
-### `rtsp/api/pull?target=[RTSP address]&streamPath=[Stream identity]&save=[0|1|2]`
-Pull the RTSP to m7s from a remote server
-- save meaning: 0, do not save; 1, save to pullonstart; 2, save to pullonsub
-- The RTSP address needs to be urlencoded to prevent special characters from affecting parsing
-### `rtsp/api/push?target=[RTSP address]&streamPath=[Stream identity]`
-Push local streams to remote servers

README.md

@@ -1,68 +1,63 @@
-_[English](https://github.com/Monibuca/plugin-rtsp/blob/v4/README.en.md) | 简体中文_
 # 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
 
-https://github.com/Monibuca/plugin-rtsp
+github.com/Monibuca/plugin-rtsp
 
 ## 插件引入
 ```go
 import (
-    _ "m7s.live/plugin/rtsp/v4"
+    _ "github.com/Monibuca/plugin-rtsp"
 )
 ```
 
-## 推拉地址形式
+## 默认插件配置
-```
-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
 
+```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"
+```
+
+- `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 [视频源] -c:v h264 -c:a aac -f rtsp rtsp://localhost/live/test
+ffmpeg -i **** 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
 
-如果m7s中已经存在live/test流的话就可以用rtsp协议进行播放
+可调用接口
-```bash
+`/api/rtsp/pull?target=[RTSP地址]&streamPath=[流标识]`
-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") 
 ```
 
-## 配置
+### 罗列所有的rtsp协议的流
 
-```yaml
+可调用接口
-rtsp:
+`/api/rtsp/list`
-    publish: # 参考全局配置格式
-    subscribe: # 参考全局配置格式
-    pull: # 格式参考文档 https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
-    push: # 格式参考文档 https://m7s.live/guide/config.html#%E6%8F%92%E4%BB%B6%E9%85%8D%E7%BD%AE
-    listenaddr: :554
-    udpaddr: :8000
-    rtcpaddr: :8001
-    readbuffercount: 2048 # 读取缓存队列大小
-    writebuffercount: 2048 # 写出缓存队列大小
-```
-:::tip 配置覆盖
-publish
-subscribe
-两项中未配置部分将使用全局配置
-:::
-## API
 
-### `rtsp/api/list`
+### 从m7s中拉取rtsp协议流
-获取所有rtsp流
 
-### `rtsp/api/pull?target=[RTSP地址]&streamPath=[流标识]&save=[0|1|2]`
+直接通过协议rtsp://xxx.xxx.xxx.xxx/live/user1 即可播放
-从远程拉取rtsp到m7s中
+> h265 编码拉流尚未实现，敬请期待
-- save含义：0、不保存；1、保存到pullonstart；2、保存到pullonsub
-- RTSP地址需要进行urlencode 防止其中的特殊字符影响解析
-### `rtsp/api/push?target=[RTSP地址]&streamPath=[流标识]`
-将本地的流推送到远端

client.go

@@ -1,150 +1,214 @@
 package rtsp
 
 import (
-	"context"
+	"errors"
-	"net"
+	"time"
+	"unsafe"
 
-	"github.com/bluenviron/gortsplib/v4"
+	. "github.com/Monibuca/engine/v3"
-	"github.com/bluenviron/gortsplib/v4/pkg/base"
+	. "github.com/Monibuca/utils/v3"
-	"github.com/bluenviron/gortsplib/v4/pkg/url"
+	"github.com/Monibuca/utils/v3/codec"
-	"go.uber.org/zap"
+	"github.com/aler9/gortsplib"
-	"m7s.live/engine/v4"
+	"github.com/aler9/gortsplib/pkg/aac"
+	"github.com/aler9/gortsplib/pkg/base"
+	"github.com/pion/rtp"
+	"github.com/pion/rtp/codecs"
 )
 
 type RTSPClient struct {
-	*gortsplib.Client `json:"-" yaml:"-"`
+	RTSPublisher
-	gortsplib.Transport
+	Transport         gortsplib.Transport
-	DialContext func(ctx context.Context, network, address string) (net.Conn, error) `json:"-" yaml:"-"`
+	*gortsplib.Client `json:"-"`
-}
-type RTSPPuller struct {
-	RTSPPublisher
-	engine.Puller
-	RTSPClient
 }
 
-func (p *RTSPClient) Disconnect() {
+// PullStream 从外部拉流
-	if p.Client != nil {
+func (rtsp *RTSPClient) PullStream(streamPath string, rtspUrl string) (err error) {
-		p.Client.Close()
+	rtsp.Stream = &Stream{
+		StreamPath: streamPath,
+		Type:       "RTSP Pull",
+		ExtraProp:  rtsp,
 	}
-}
+	if result := rtsp.Publish(); result {
-
+		rtsp.URL = rtspUrl
-func (p *RTSPPuller) Connect() error {
+		if config.Reconnect {
-	client := &gortsplib.Client{
+			go func() {
-		DialContext: p.DialContext,
+				for rtsp.pullStream(); rtsp.Err() == nil; rtsp.pullStream() {
-
+					Printf("reconnecting:%s in 5 seconds", rtspUrl)
-		AnyPortEnable: true,
+					if rtsp.Transport == gortsplib.TransportTCP {
+						rtsp.Transport = gortsplib.TransportUDP
+					} else {
+						rtsp.Transport = gortsplib.TransportTCP
+					}
+					time.Sleep(time.Second * 5)
+				}
+				if rtsp.IsTimeout {
+					rtsp.processFunc = nil
+					go rtsp.PullStream(streamPath, rtspUrl)
+				}
+			}()
+		} else {
+			go rtsp.pullStream()
+		}
+		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:
+				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.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{
+		OnPacketRTP: func(trackID int, payload []byte) {
+			// Println("OnPacketRTP", trackID, len(payload))
+			if f := client.processFunc[trackID]; f != nil {
+				var clone []byte
+				f(append(clone, payload...))
+			}
+		},
+		ReadBufferSize: config.ReadBufferSize,
+		Transport:      &client.Transport,
 	}
-	p.Transport = gortsplib.TransportTCP
-	client.Transport = &p.Transport
 	// parse URL
-	u, err := url.Parse(p.RemoteURL)
+	u, err := base.ParseURL(client.URL)
 	if err != nil {
-		return err
+		Printf("ParseURL:%s error:%v", client.URL, err)
+		return
 	}
 	// connect to the server
 	if err = client.Start(u.Scheme, u.Host); err != nil {
-		return err
+		Printf("connect:%s error:%v", client.URL, err)
+		return
 	}
-	p.Client = client
+	client.OnClose = func() {
-	p.SetIO(p.Client)
+		client.Client.Close()
-	return nil
+	}
-}
+	//client.close should be after connected!
-
+	defer client.Client.Close()
-func (p *RTSPPuller) Pull() (err error) {
-	u, _ := url.Parse(p.RemoteURL)
 	var res *base.Response
-	if res, err = p.Options(u); err != nil {
+	if res, err = client.Options(u); err != nil {
-		p.Error("Options", zap.Error(err))
+		Printf("option:%s error:%v", client.URL, err)
 		return
 	}
-	p.Debug("Options", zap.Any("res", res))
+	Println(res)
 	// find published tracks
-	session, res, err := p.Describe(u)
+	tracks, baseURL, res, err := client.Describe(u)
 	if err != nil {
-		p.Error("Describe", zap.Error(err))
+		Printf("Describe:%s error:%v", client.URL, err)
-		return err
-	}
-	p.Debug("Describe", zap.Any("res", res))
-	p.session = session
-	err = p.SetTracks()
-	if err != nil {
-		p.Error("SetTracks", zap.Error(err))
-		return err
-	}
-	if err = p.SetupAll(session.BaseURL, session.Medias); err != nil {
-		p.Error("SetupAndPlay", zap.Error(err))
-		return err
-	}
-	p.OnPacketRTPAny(p.OnPacket)
-	res, err = p.Play(nil)
-	p.Debug("Play", zap.Any("res", res))
-	if err != nil {
-		p.Error("Play", zap.Error(err))
-		return err
-	}
-	return p.Wait()
-}
-
-type RTSPPusher struct {
-	RTSPSubscriber
-	engine.Pusher
-	RTSPClient
-}
-
-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 {
-	p.Client = &gortsplib.Client{
-		DialContext:    p.DialContext,
-		WriteQueueSize: rtspConfig.WriteBufferCount,
-	}
-	// parse URL
-	u, err := url.Parse(p.RemoteURL)
-	if err != nil {
-		p.Error("url.Parse", zap.Error(err))
-		return err
-	}
-	// connect to the server
-	if err = p.Client.Start(u.Scheme, u.Host); err != nil {
-		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.session); err != nil {
-		p.Error("Announce", zap.Error(err))
 		return
 	}
-	err = p.SetupAll(p.session.BaseURL, p.session.Medias)
+	Println(res)
-	if err != nil {
+	if client.processFunc == nil {
-		p.Error("Setup", zap.Error(err))
+		client.setTracks(tracks)
+	}
+	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)
+	}
+	// start reading tracks
+	if res, err = client.Play(nil); err != nil {
+		Printf("Play:%s error:%v", baseURL.String(), err)
 		return
 	}
-
+	Println(res)
-	if _, err = p.Record(); err != nil {
+	// wait until a fatal error
-		p.Error("Record", zap.Error(err))
+	var fatalChan = make(chan error)
-		return
+	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)
 	}
-	p.PlayRTP()
-	return
 }

go.mod

@@ -1,49 +1,11 @@
-module m7s.live/plugin/rtsp/v4
+module github.com/Monibuca/plugin-rtsp/v3
 
-go 1.18
+go 1.16
 
 require (
-  github.com/bluenviron/gortsplib/v4 v4.2.1
+	github.com/Monibuca/engine/v3 v3.4.1
-	github.com/bluenviron/mediacommon v1.5.0
+	github.com/Monibuca/utils/v3 v3.0.5
-	github.com/pion/rtp v1.8.2
+	github.com/aler9/gortsplib v0.0.0-20211212220644-6f374e396529
-	go.uber.org/zap v1.24.0
+	github.com/pion/rtp v1.7.4
-	m7s.live/engine/v4 v4.13.12
+	github.com/pion/sdp/v3 v3.0.4
 )
-
-require (
-	github.com/cnotch/ipchub v1.1.0 // indirect
-	github.com/denisbrodbeck/machineid v1.0.1 // indirect
-	github.com/go-ole/go-ole v1.2.6 // indirect
-	github.com/go-task/slim-sprig v0.0.0-20230315185526-52ccab3ef572 // indirect
-	github.com/golang/mock v1.6.0 // indirect
-	github.com/google/pprof v0.0.0-20210407192527-94a9f03dee38 // indirect
-	github.com/google/uuid v1.3.1 // indirect
-	github.com/logrusorgru/aurora/v4 v4.0.0 // indirect
-	github.com/lufia/plan9stats v0.0.0-20230326075908-cb1d2100619a // indirect
-	github.com/mcuadros/go-defaults v1.2.0 // indirect
-	github.com/onsi/ginkgo/v2 v2.9.5 // 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 v3.2.14 // indirect
-	github.com/power-devops/perfstat v0.0.0-20221212215047-62379fc7944b // indirect
-	github.com/q191201771/naza v0.30.11 // indirect
-	github.com/quic-go/qtls-go1-20 v0.3.1 // indirect
-	github.com/quic-go/quic-go v0.37.4 // indirect
-	github.com/shirou/gopsutil/v3 v3.23.7 // indirect
-	github.com/shoenig/go-m1cpu v0.1.6 // indirect
-	github.com/tklauser/go-sysconf v0.3.11 // indirect
-	github.com/tklauser/numcpus v0.6.0 // indirect
-	github.com/yapingcat/gomedia v0.0.0-20230727105416-c491e66c9d2a // indirect
-	github.com/yusufpapurcu/wmi v1.2.3 // indirect
-	go.uber.org/atomic v1.10.0 // indirect
-	go.uber.org/multierr v1.11.0 // indirect
-	golang.org/x/crypto v0.14.0 // indirect
-	golang.org/x/exp v0.0.0-20221205204356-47842c84f3db // indirect
-	golang.org/x/mod v0.10.0 // indirect
-	golang.org/x/net v0.17.0 // indirect
-	golang.org/x/sync v0.2.0 // indirect
-	golang.org/x/sys v0.13.0 // indirect
-	golang.org/x/tools v0.9.1 // indirect
-	gopkg.in/yaml.v3 v3.0.1 // indirect
-)

go.sum

@@ -1,14 +1,14 @@
 github.com/BurntSushi/toml v0.3.1/go.mod h1:xHWCNGjB5oqiDr8zfno3MHue2Ht5sIBksp03qcyfWMU=
-github.com/benbjohnson/clock v1.1.0 h1:Q92kusRqC1XV2MjkWETPvjJVqKetz1OzxZB7mHJLju8=
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 golang.org/x/sys v0.0.0-20201119102817-f84b799fce68/go.mod h1:h1NjWce9XRLGQEsW7wpKNCjG9DtNlClVuFLEZdDNbEs=
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 golang.org/x/sys v0.0.0-20210423082822-04245dca01da/go.mod h1:h1NjWce9XRLGQEsW7wpKNCjG9DtNlClVuFLEZdDNbEs=
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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=
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main.go

@@ -1,101 +1,113 @@
 package rtsp
 
 import (
+	"encoding/json"
+	"fmt"
+	"log"
 	"net/http"
-	"strconv"
+	"time"
-	"sync"
 
-	"github.com/bluenviron/gortsplib/v4"
+	. "github.com/Monibuca/engine/v3"
-	"go.uber.org/zap"
+	. "github.com/Monibuca/utils/v3"
-	. "m7s.live/engine/v4"
+	"github.com/aler9/gortsplib"
-	"m7s.live/engine/v4/config"
-	"m7s.live/engine/v4/util"
 )
 
-type RTSPConfig struct {
+var config = struct {
-	config.Publish
+	ListenAddr     string
-	config.Subscribe
+	UDPAddr        string
-	config.Pull
+	RTCPAddr       string
-	config.Push
+	Timeout        int
-	ListenAddr       string `default:":554"`
+	Reconnect      bool
-	UDPAddr          string `default:":8000"`
+	AutoPullList   map[string]string
-	RTCPAddr         string `default:":8001"`
+	AutoPushList   map[string]string
-	WriteBufferCount int    `default:"2048"`
+	ReadBufferSize int
-	sync.Map
+}{":554", ":8000", ":8001", 0, false, nil, nil, 2048}
-	server *gortsplib.Server
+
+var pconfig = PluginConfig{
+	Name:   "RTSP",
+	Config: &config,
 }
 
-func (conf *RTSPConfig) OnEvent(event any) {
+func init() {
-	switch v := event.(type) {
+	pconfig.Install(runPlugin)
-	case FirstConfig:
+}
-		conf.server = &gortsplib.Server{
+
-			Handler:           conf,
+func getRtspList() (info []*Stream) {
-			RTSPAddress:       conf.ListenAddr,
+	for _, s := range Streams.ToList() {
-			UDPRTPAddress:     conf.UDPAddr,
+		switch s.ExtraProp.(type) {
-			UDPRTCPAddress:    conf.RTCPAddr,
+		case *RTSPublisher:
-			MulticastIPRange:  "224.1.0.0/16",
+			info = append(info, s)
-			MulticastRTPPort:  8002,
+		case *RTSPClient:
-			MulticastRTCPPort: 8003,
+			info = append(info, s)
-			WriteQueueSize:    conf.WriteBufferCount,
-		}
-		if err := conf.server.Start(); err != nil {
-			RTSPPlugin.Error("server start", zap.Error(err))
-			RTSPPlugin.Disabled = true
-		}
-		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:
-		if url, ok := conf.PushList[v.Target.Path]; ok {
-			if err := RTSPPlugin.Push(v.Target.Path, url, new(RTSPPusher), false); err != nil {
-				RTSPPlugin.Error("push", zap.String("streamPath", v.Target.Path), zap.String("url", url), zap.Error(err))
-			}
-		}
-	case InvitePublish: //按需拉流
-		if remoteURL := conf.CheckPullOnSub(v.Target); remoteURL != "" {
-			if err := RTSPPlugin.Pull(v.Target, remoteURL, new(RTSPPuller), 0); err != nil {
-				RTSPPlugin.Error("pull", zap.String("streamPath", v.Target), zap.String("url", remoteURL), zap.Error(err))
-			}
 		}
 	}
-}
-
-var rtspConfig = &RTSPConfig{}
-var RTSPPlugin = InstallPlugin(rtspConfig)
-
-func filterStreams() (ss []*Stream) {
-	Streams.Range(func(key string, s *Stream) {
-		switch s.Publisher.(type) {
-		case *RTSPPublisher, *RTSPPuller:
-			ss = append(ss, 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)
+		}
+	}
 
-func (*RTSPConfig) API_list(w http.ResponseWriter, r *http.Request) {
+	go AddHook(HOOK_PUBLISH, func(s *Stream) {
-	util.ReturnFetchValue(filterStreams, w, r)
+		for streamPath, url := range config.AutoPushList {
-}
+			if s.StreamPath == streamPath {
-
+				(&RTSPClient{}).PushStream(streamPath, url)
-func (*RTSPConfig) API_Pull(rw http.ResponseWriter, r *http.Request) {
+			}
-	query := r.URL.Query()
+		}
-	save, _ := strconv.Atoi(query.Get("save"))
+	})
-	err := RTSPPlugin.Pull(query.Get("streamPath"), query.Get("target"), new(RTSPPuller), save)
+	if config.ListenAddr != "" {
-	if err != nil {
+		go log.Fatal(ListenRtsp(config.ListenAddr))
-		util.ReturnError(util.APIErrorQueryParse, err.Error(), rw, r)
-	} else {
-		util.ReturnOK(rw, r)
 	}
 }
 
-func (*RTSPConfig) API_Push(rw http.ResponseWriter, r *http.Request) {
+func ListenRtsp(addr string) error {
-	query := r.URL.Query()
+	defer log.Println("rtsp server start!")
-	err := RTSPPlugin.Push(query.Get("streamPath"), query.Get("target"), new(RTSPPusher), query.Has("save"))
+	s := &gortsplib.Server{
-	if err != nil {
+		Handler:           &RTSPServer{},
-		util.ReturnError(util.APIErrorQueryParse, err.Error(), rw, r)
+		RTSPAddress:       addr,
-	} else {
+		UDPRTPAddress:     config.UDPAddr,
-		util.ReturnOK(rw, r)
+		UDPRTCPAddress:    config.RTCPAddr,
+		MulticastIPRange:  "224.1.0.0/16",
+		MulticastRTPPort:  8002,
+		MulticastRTCPPort: 8003,
 	}
+	return s.StartAndWait()
 }

payloader.go

@@ -0,0 +1,24 @@
+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

@@ -1,119 +1,121 @@
 package rtsp
 
 import (
+	"encoding/base64"
+	"encoding/hex"
+	"strconv"
 	"strings"
 
-	"github.com/bluenviron/gortsplib/v4/pkg/description"
+	. "github.com/Monibuca/engine/v3"
-	"github.com/bluenviron/gortsplib/v4/pkg/format"
+	. "github.com/Monibuca/utils/v3"
-	"github.com/bluenviron/mediacommon/pkg/codecs/mpeg4audio"
+	"github.com/aler9/gortsplib"
-	"github.com/pion/rtp"
-	"go.uber.org/zap"
-	. "m7s.live/engine/v4"
-	"m7s.live/engine/v4/common"
-	. "m7s.live/engine/v4/track"
 )
 
-type RTSPPublisher struct {
+type RTSPublisher struct {
-	Publisher
+	*Stream     `json:"-"`
-	Tracks map[*description.Media]common.AVTrack `json:"-" yaml:"-"`
+	stream      *gortsplib.ServerStream
-	RTSPIO
+	processFunc []func([]byte)
 }
 
-func (p *RTSPPublisher) SetTracks() error {
+func (p *RTSPublisher) setTracks(tracks gortsplib.Tracks) {
-	p.Tracks = make(map[*description.Media]common.AVTrack, len(p.session.Medias))
+	if p.processFunc != nil {
-	defer func() {
+		p.processFunc = p.processFunc[:len(tracks)]
-		for _, track := range p.Tracks {
+		return
-			p.Info("set track", zap.String("name", track.GetName()))
+	} else {
+		p.processFunc = make([]func([]byte), len(tracks))
+	}
+	for i, track := range tracks {
+		v, ok := track.Media.Attribute("rtpmap")
+		if !ok {
+			continue
 		}
-	}()
+
-	for _, track := range p.session.Medias {
+		fmtp := make(map[string]string)
-		for _, forma := range track.Formats {
+		if v, ok := track.Media.Attribute("fmtp"); ok {
-			switch f := forma.(type) {
+			if tmp := strings.SplitN(v, " ", 2); len(tmp) == 2 {
-			case *format.H264:
+				for _, kv := range strings.Split(tmp[1], ";") {
-				vt := p.VideoTrack
+					kv = strings.Trim(kv, " ")
-				if vt == nil {
+
-					vt = NewH264(p.Stream, f.PayloadType())
+					if len(kv) == 0 {
-					p.VideoTrack = vt
+						continue
-				}
-				p.Tracks[track] = p.VideoTrack
-				if len(f.SPS) > 0 {
-					vt.WriteSliceBytes(f.SPS)
-				}
-				if len(f.PPS) > 0 {
-					vt.WriteSliceBytes(f.PPS)
-				}
-			case *format.H265:
-				vt := p.VideoTrack
-				if vt == nil {
-					vt = NewH265(p.Stream, f.PayloadType())
-					p.VideoTrack = vt
-				}
-				p.Tracks[track] = p.VideoTrack
-				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 := p.AudioTrack
-				if at == nil {
-					at := NewAAC(p.Stream, f.PayloadType(), uint32(f.Config.SampleRate))
-					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)
+					tmp := strings.SplitN(kv, "=", 2)
-					asc, _ := f.Config.Marshal()
+					if len(tmp) == 2 {
-					// 复用AVCC写入逻辑，解析出AAC的配置信息
+						fmtp[strings.TrimSpace(tmp[0])] = tmp[1]
-					at.WriteSequenceHead(append([]byte{0xAF, 0x00}, asc...))
-					p.AudioTrack = at
-				}
-				p.Tracks[track] = p.AudioTrack
-			case *format.G711:
-				at := p.AudioTrack
-				if at == nil {
-					at := NewG711(p.Stream, !f.MULaw, f.PayloadType(), uint32(f.ClockRate()))
-					p.AudioTrack = at
-				}
-				p.Tracks[track] = p.AudioTrack
-			default:
-				rtpMap := strings.ToLower(forma.RTPMap())
-				if strings.Contains(rtpMap, "pcm") {
-					isMulaw := false
-					if strings.Contains(rtpMap, "pcmu") {
-						isMulaw = true
 					}
-					at := p.AudioTrack
-					if at == nil {
-						at := NewG711(p.Stream, !isMulaw, f.PayloadType(), uint32(f.ClockRate()))
-						p.AudioTrack = at
-					}
-					p.Tracks[track] = p.AudioTrack
-				} else {
-					p.Warn("unknown format", zap.Any("format", f.FMTP()))
 				}
 			}
 		}
-	}
-	if p.VideoTrack == nil {
-		p.Config.PubVideo = false
-		p.Info("no video track")
-	}
-	if p.AudioTrack == nil {
-		p.Config.PubAudio = false
-		p.Info("no audio track")
-	}
-	return nil
-}
 
-func (p *RTSPPublisher) OnPacket(m *description.Media, f format.Format, pack *rtp.Packet) {
+		v = strings.TrimSpace(v)
-	if t, ok := p.Tracks[m]; ok {
+		vals := strings.Split(v, " ")
-		t.WriteRTPPack(pack)
+		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

@@ -1,121 +1,264 @@
 package rtsp
 
 import (
-	"github.com/bluenviron/gortsplib/v4"
+	"fmt"
-	"github.com/bluenviron/gortsplib/v4/pkg/base"
+	"sync"
-	"github.com/bluenviron/gortsplib/v4/pkg/description"
+	"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 RTSPIO struct {
+// 接收RTSP推流：OnConnOpen->OnAnnounce->OnSetup->OnSessionOpen
-	server     *gortsplib.Server
+// 接收RTSP拉流：OnConnOpen->OnDescribe->OnSetup->OnSessionOpen
-	session    *description.Session
+type RTSPServer struct {
-	tracks     []*description.Media
+	sync.Map
-	stream     *gortsplib.ServerStream
+}
-	audioTrack *description.Media
+type RTSPSubscriber struct {
-	videoTrack *description.Media
+	stream *gortsplib.ServerStream
+	engine.Subscriber
+	vt *engine.VideoTrack
+	at *engine.AudioTrack
 }
 
-func (conf *RTSPConfig) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
+// called after a connection is opened.
-	RTSPPlugin.Debug("conn opened")
+func (sh *RTSPServer) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
+	Printf("rtsp conn opened")
 }
 
-func (conf *RTSPConfig) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
+// called after a connection is closed.
-	RTSPPlugin.Debug("conn closed")
+func (sh *RTSPServer) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
-	if p, ok := conf.LoadAndDelete(ctx.Conn); ok {
+	Printf("rtsp conn closed (%v)", ctx.Error)
-		p.(IIO).Stop(zap.String("conn", "closed"))
+	if p, ok := sh.Load(ctx.Conn); ok {
+		switch v := p.(type) {
+		case *RTSPublisher:
+			v.Close()
+		case *RTSPSubscriber:
+			v.Close()
+		}
+		sh.Delete(ctx.Conn)
 	}
 }
 
-func (conf *RTSPConfig) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
+// called after a session is opened.
-	RTSPPlugin.Debug("session opened")
+func (sh *RTSPServer) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
+	Printf("rtsp session opened")
 }
 
-func (conf *RTSPConfig) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
+// called after a session is closed.
-	RTSPPlugin.Debug("session closed")
+func (sh *RTSPServer) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
-	conf.Delete(ctx.Session)
+	Printf("rtsp session closed")
+	if v, ok := sh.LoadAndDelete(ctx.Session); ok {
+		switch v := v.(type) {
+		case *RTSPublisher:
+			v.Close()
+		case *RTSPSubscriber:
+			v.Close()
+		}
+	}
 }
 
 // called after receiving a DESCRIBE request.
-func (conf *RTSPConfig) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
+func (sh *RTSPServer) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
-	RTSPPlugin.Debug("describe request")
+	Printf("describe request")
-	var suber RTSPSubscriber
+	var err error
-	suber.server = conf.server
+	if s := engine.FindStream(ctx.Path); s != nil {
-	suber.RemoteAddr = ctx.Conn.NetConn().RemoteAddr().String()
+		var tracks gortsplib.Tracks
-	suber.SetIO(ctx.Conn.NetConn())
+		var stream *gortsplib.ServerStream
-	streamPath := ctx.Path
+		var sub RTSPSubscriber
-	if ctx.Query != "" {
+		sub.Type = "RTSP pull"
-		streamPath = streamPath + "?" + ctx.Query
+		sub.vt = s.WaitVideoTrack("h264", "h265")
-	}
+		sub.at = s.WaitAudioTrack("aac", "pcma", "pcmu")
-	if err := RTSPPlugin.Subscribe(streamPath, &suber); err == nil {
+		ssrc := uintptr(unsafe.Pointer(&stream))
-		RTSPPlugin.Debug("describe replay ok")
+		var trackIds = 0
-		conf.Store(ctx.Conn, &suber)
+		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:
+				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 *engine.VideoPack) {
+				for i, nalu := range pack.NALUs {
+					var samples uint32
+					if i == len(pack.NALUs)-1 {
+						samples = (ts - st) * 90
+					} else {
+						samples = 0
+					}
+					packs := vpacketer.Packetize(nalu, samples)
+					for j, rtpack := range packs {
+						rtpack.Marker = i == len(pack.NALUs)-1 && j == len(packs)-1
+						rtp, _ := rtpack.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,
-		}, suber.stream, nil
+		}, stream, nil
-	} else {
+		// if stream, ok := s.ExtraProp.(*gortsplib.ServerStream); ok {
-		return &base.Response{
+		// 	return &base.Response{
-			StatusCode: base.StatusNotFound,
+		// 		StatusCode: base.StatusOK,
-		}, suber.stream, nil
+		// 	}, stream, nil
-	}
+		// }
-}
-
-func (conf *RTSPConfig) OnSetup(ctx *gortsplib.ServerHandlerOnSetupCtx) (*base.Response, *gortsplib.ServerStream, error) {
-	var resp base.Response
-	resp.StatusCode = base.StatusOK
-	if p, ok := conf.Load(ctx.Conn); ok {
-		switch v := p.(type) {
-		case *RTSPSubscriber:
-			return &resp, v.stream, nil
-		case *RTSPPublisher:
-			return &resp, v.stream, nil
-		}
-	}
-	resp.StatusCode = base.StatusNotFound
-	return &resp, nil, nil
-}
-
-func (conf *RTSPConfig) OnPlay(ctx *gortsplib.ServerHandlerOnPlayCtx) (*base.Response, error) {
-	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,
+		StatusCode: base.StatusNotFound,
-	}, nil
+	}, nil, nil
 }
-func (conf *RTSPConfig) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
+
-	p := &RTSPPublisher{}
+// called after receiving an ANNOUNCE request.
-	p.SetIO(ctx.Conn.NetConn())
+func (sh *RTSPServer) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
-	if err := RTSPPlugin.Publish(ctx.Path, p); err == nil {
+	Printf("announce request")
-		p.session = ctx.Description
+	p := &RTSPublisher{
-		p.stream = gortsplib.NewServerStream(conf.server, ctx.Description)
+		Stream: &engine.Stream{
-		if err = p.SetTracks(); err != nil {
+			StreamPath: ctx.Path,
-			return nil, err
+			Type:       "RTSP push",
-		}
+		},
-		conf.Store(ctx.Conn, p)
+	}
-		conf.Store(ctx.Session, p)
+	p.ExtraProp = p
+	p.URL = ctx.Req.URL.String()
+	if p.Publish() {
+		p.setTracks(ctx.Tracks)
+		p.stream = gortsplib.NewServerStream(ctx.Tracks)
+		sh.Store(ctx.Conn, p)
+		sh.Store(ctx.Session, p)
 	} else {
 		return &base.Response{
 			StatusCode: base.StatusBadRequest,
-		}, nil
+		}, fmt.Errorf("streamPath is already exist")
 	}
 	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

@@ -1,105 +0,0 @@
-package rtsp
-
-import (
-	"github.com/bluenviron/gortsplib/v4"
-	"github.com/bluenviron/gortsplib/v4/pkg/description"
-	"github.com/bluenviron/gortsplib/v4/pkg/format"
-	"github.com/bluenviron/mediacommon/pkg/codecs/mpeg4audio"
-	. "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 := &description.Media{
-				Type: description.MediaTypeVideo,
-				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 := &description.Media{
-				Type: description.MediaTypeVideo,
-				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 := &description.Media{
-				Type: description.MediaTypeAudio,
-				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 := &description.Media{
-				Type:    description.MediaTypeAudio,
-				Formats: []format.Format{&format.G711{}},
-			}
-			s.audioTrack = audio
-			s.tracks = append(s.tracks, audio)
-		case codec.CodecID_PCMU:
-			audio := &description.Media{
-				Type: description.MediaTypeAudio,
-				Formats: []format.Format{&format.G711{
-					MULaw: true,
-				}},
-			}
-			s.audioTrack = audio
-			s.tracks = append(s.tracks, audio)
-		}
-		s.AddTrack(v)
-	case ISubscriber:
-		s.session = &description.Session{
-			Medias:  s.tracks,
-		}
-		if s.server != nil {
-			s.stream = gortsplib.NewServerStream(s.server, s.session)
-		}
-	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

@@ -0,0 +1,87 @@
+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])),
+				},
+			},
+		},
+	}
+}