fix: use engine's aac format

feat: add LATM support
feat: support LATM
2025-10-31 02:46:47 +08:00 · 2024-09-27 10:05:32 +08:00 · 2024-09-26 20:34:00 +08:00 · 2024-09-26 14:08:06 +08:00 · 2024-07-09 19:59:32 +08:00 · 2023-12-13 15:27:10 +08:00
12 changed files with 964 additions and 833 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -1,2 +0,0 @@
-.vscode
-node_modules
--- a/README.en.md
+++ b/README.en.md
@@ -0,0 +1,70 @@
+_[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
--- a/README.md
+++ b/README.md
@@ -1,63 +1,68 @@
+_[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

-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
-
-```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"
+## 推拉地址形式
 ```
+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
-`/api/rtsp/pull?target=[RTSP地址]&streamPath=[流标识]`
-
-## 使用编程方式拉流
-```go
-new(RTSPClient).PullStream("live/user1","rtsp://xxx.xxx.xxx.xxx/live/user1") 
+如果m7s中已经存在live/test流的话就可以用rtsp协议进行播放
+```bash
+ffplay rtsp://localhost/live/test
 ```

-### 罗列所有的rtsp协议的流
+## 配置

-可调用接口
-`/api/rtsp/list`
+```yaml
+rtsp:
+    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

-### 从m7s中拉取rtsp协议流
+### `rtsp/api/list`
+获取所有rtsp流

-直接通过协议rtsp://xxx.xxx.xxx.xxx/live/user1 即可播放
-> h265 编码拉流尚未实现，敬请期待
+### `rtsp/api/pull?target=[RTSP地址]&streamPath=[流标识]&save=[0|1|2]`
+从远程拉取rtsp到m7s中
+- 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
--- a/client.go
+++ b/client.go
@@ -1,210 +1,169 @@
 package rtsp

 import (
-	"errors"
-	"time"
-	"unsafe"
+	"context"
+	"net"

-	. "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"
+	"github.com/bluenviron/gortsplib/v4"
+	"github.com/bluenviron/gortsplib/v4/pkg/base"
+	"github.com/bluenviron/gortsplib/v4/pkg/url"
+	"go.uber.org/zap"
+	"m7s.live/engine/v4"
 )

 type RTSPClient struct {
-	RTSPublisher
-	Transport         gortsplib.Transport
-	*gortsplib.Client `json:"-"`
+	*gortsplib.Client `json:"-" yaml:"-"`
+	gortsplib.Transport
+	DialContext func(ctx context.Context, network, address string) (net.Conn, error) `json:"-" yaml:"-"`
+}
+type RTSPPuller struct {
+	RTSPPublisher
+	engine.Puller
+	RTSPClient
 }

-// PullStream 从外部拉流
-func (rtsp *RTSPClient) PullStream(streamPath string, rtspUrl string) (err error) {
-	rtsp.Stream = &Stream{
-		StreamPath: streamPath,
-		Type:       "RTSP Pull",
-		ExtraProp:  rtsp,
+func (p *RTSPClient) Close() error {
+	if p.Client != nil {
+		p.Client.Close()
 	}
-	if result := rtsp.Publish(); result {
-		rtsp.URL = rtspUrl
-		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
-					}
-					time.Sleep(time.Second * 5)
-				}
-				if rtsp.IsTimeout {
-					go rtsp.PullStream(streamPath, rtspUrl)
-				}
-			}()
-		} else {
-			go rtsp.pullStream()
-		}
-		return
-	}
-	return errors.New("publish badname")
+	return nil
 }
-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
+
+func (p *RTSPClient) Disconnect() {
+	if p.Client != nil {
+		p.Client.Close()
 	}
-			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: 10000,
-		Transport:      &client.Transport,
+
+func (p *RTSPPuller) Connect() error {
+	client := &gortsplib.Client{
+		DialContext: p.DialContext,
+
+		AnyPortEnable: true,
 	}
+	p.Transport = gortsplib.TransportTCP
+	client.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)
-		return
+		return err
 	}
 	// connect to the server
 	if err = client.Start(u.Scheme, u.Host); err != nil {
-		Printf("connect:%s error:%v", client.URL, err)
-		return
+		return err
 	}
-	//client.close should be after connected!
-	defer client.Client.Close()
+	p.Client = client
+	p.SetIO(p)
+	return nil
+}
+
+func (p *RTSPPuller) Pull() (err error) {
+	u, _ := url.Parse(p.RemoteURL)
 	var res *base.Response
-	if res, err = client.Options(u); err != nil {
-		Printf("option:%s error:%v", client.URL, err)
+	if rtspConfig.SendOptions {
+		if res, err = p.Options(u); err != nil {
+			p.Error("Options", zap.Error(err))
 			return
 		}
-	Println(res)
+	}
+	p.Debug("Options", zap.Any("res", res))
 	// find published tracks
-	tracks, baseURL, res, err := client.Describe(u)
+	session, res, err := p.Describe(u)
 	if err != nil {
-		Printf("Describe:%s error:%v", client.URL, err)
-		return
+		p.Error("Describe", zap.Error(err))
+		return err
 	}
-	Println(res)
-	if client.processFunc == nil {
-		client.setTracks(tracks)
+	p.Debug("Describe", zap.Any("res", res))
+	p.session = session
+	err = p.SetTracks()
+	if err != nil {
+		p.Error("SetTracks", zap.Error(err))
+		return 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
+	if err = p.SetupAll(session.BaseURL, session.Medias); err != nil {
+		p.Error("SetupAndPlay", zap.Error(err))
+		return err
 	}
-		Println(res)
+	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
 	}
-	// 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)
+	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,
+	}
+	p.Transport = gortsplib.TransportTCP
+	p.Client.Transport = &p.Transport
+	// 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)
+	if rtspConfig.SendOptions {
+		_, 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")
+	// 	}
+	// }
+	var res *base.Response
+	if res, err = p.Announce(u, p.session); err != nil {
+		p.Error("Announce", zap.Error(err))
+		return
+	} else {
+		p.Debug("Announce", zap.Any("res", res))
+	}
+	err = p.SetupAll(u, p.session.Medias)
+	if err != nil {
+		p.Error("Setup", zap.Error(err))
+		return
+	}
+
+	if res, err = p.Record(); err != nil {
+		p.Error("Record", zap.Error(err))
+		return
+	} else {
+		p.Debug("Record", zap.Any("res", res))
+	}
+	p.PlayRTP()
+	return
+}
--- a/go.mod
+++ b/go.mod
@@ -1,11 +1,49 @@
-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/Monibuca/utils/v3 v3.0.5
-	github.com/aler9/gortsplib v0.0.0-20211212220644-6f374e396529
-	github.com/pion/rtp v1.7.4
-	github.com/pion/sdp/v3 v3.0.4
+	github.com/bluenviron/gortsplib/v4 v4.6.2
+	github.com/bluenviron/mediacommon v1.5.1
+	github.com/pion/rtp v1.8.3
+	go.uber.org/zap v1.26.0
+	m7s.live/engine/v4 v4.15.2
+)
+
+require (
+	github.com/deepch/vdk v0.0.27 // 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 v0.0.0-20230309165930-d61513b1440d // indirect
+	github.com/google/uuid v1.4.0 // 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 v1.2.12 // indirect
+	github.com/pion/sdp/v3 v3.0.6 // indirect
+	github.com/pion/webrtc/v3 v3.2.20 // indirect
+	github.com/pkg/errors v0.9.1 // indirect
+	github.com/power-devops/perfstat v0.0.0-20221212215047-62379fc7944b // indirect
+	github.com/q191201771/naza v0.30.48 // indirect
+	github.com/quic-go/qtls-go1-20 v0.3.3 // indirect
+	github.com/quic-go/quic-go v0.38.1 // indirect
+	github.com/shirou/gopsutil/v3 v3.23.8 // indirect
+	github.com/shoenig/go-m1cpu v0.1.6 // indirect
+	github.com/tklauser/go-sysconf v0.3.12 // indirect
+	github.com/tklauser/numcpus v0.6.1 // indirect
+	github.com/yapingcat/gomedia v0.0.0-20230905155010-55b9713fcec1 // indirect
+	github.com/yusufpapurcu/wmi v1.2.3 // indirect
+	go.uber.org/multierr v1.11.0 // indirect
+	golang.org/x/crypto v0.16.0 // indirect
+	golang.org/x/exp v0.0.0-20230905200255-921286631fa9 // indirect
+	golang.org/x/mod v0.12.0 // indirect
+	golang.org/x/net v0.19.0 // indirect
+	golang.org/x/sync v0.3.0 // indirect
+	golang.org/x/sys v0.15.0 // indirect
+	golang.org/x/tools v0.13.0 // indirect
+	gopkg.in/yaml.v3 v3.0.1 // indirect
 )
--- a/go.sum
+++ b/go.sum
@@ -1,101 +1,271 @@
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--- a/main.go
+++ b/main.go
@@ -1,112 +1,102 @@
 package rtsp

 import (
-	"encoding/json"
-	"fmt"
-	"log"
 	"net/http"
-	"time"
+	"strconv"
+	"sync"

-	. "github.com/Monibuca/engine/v3"
-	. "github.com/Monibuca/utils/v3"
-	"github.com/aler9/gortsplib"
+	"github.com/bluenviron/gortsplib/v4"
+	"go.uber.org/zap"
+	. "m7s.live/engine/v4"
+	"m7s.live/engine/v4/config"
+	"m7s.live/engine/v4/util"
 )

-var config = struct {
-	ListenAddr   string
-	UDPAddr      string
-	RTCPAddr     string
-	Timeout      int
-	Reconnect    bool
-	AutoPullList map[string]string
-	AutoPushList map[string]string
-}{":554", ":8000", ":8001", 0, false, nil, nil}
-
-var pconfig = PluginConfig{
-	Name:   "RTSP",
-	Config: &config,
+type RTSPConfig struct {
+	config.Publish
+	config.Subscribe
+	config.Pull
+	config.Push
+	ListenAddr       string `default:":554" desc:"rtsp监听地址"`
+	UDPAddr          string `default:":8000" desc:"udp rtp监听地址"`
+	RTCPAddr         string `default:":8001" desc:"udp rtcp监听地址"`
+	WriteBufferCount int    `default:"2048" desc:"rtsp写缓冲区大小"`
+	SendOptions      bool   `default:"true" desc:"是否发送options请求"`
+	sync.Map
+	server *gortsplib.Server
 }

-func init() {
-	pconfig.Install(runPlugin)
-}
-
-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{},
-		RTSPAddress:       addr,
-		UDPRTPAddress:     config.UDPAddr,
-		UDPRTCPAddress:    config.RTCPAddr,
+func (conf *RTSPConfig) OnEvent(event any) {
+	switch v := event.(type) {
+	case FirstConfig:
+		conf.server = &gortsplib.Server{
+			Handler:           conf,
+			RTSPAddress:       conf.ListenAddr,
+			UDPRTPAddress:     conf.UDPAddr,
+			UDPRTCPAddress:    conf.RTCPAddr,
 			MulticastIPRange:  "224.1.0.0/16",
 			MulticastRTPPort:  8002,
 			MulticastRTCPPort: 8003,
+			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 remoteURL := conf.CheckPush(v.Target.Path); remoteURL != "" {
+			if err := RTSPPlugin.Push(v.Target.Path, remoteURL, new(RTSPPusher), false); err != nil {
+				RTSPPlugin.Error("push", zap.String("streamPath", v.Target.Path), zap.String("url", remoteURL), 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 (*RTSPConfig) API_list(w http.ResponseWriter, r *http.Request) {
+	util.ReturnFetchValue(filterStreams, w, r)
+}
+
+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 err != nil {
+		util.ReturnError(util.APIErrorQueryParse, err.Error(), rw, r)
+	} else {
+		util.ReturnOK(rw, r)
+	}
+}
+
+func (*RTSPConfig) API_Push(rw http.ResponseWriter, r *http.Request) {
+	query := r.URL.Query()
+	err := RTSPPlugin.Push(query.Get("streamPath"), query.Get("target"), new(RTSPPusher), query.Has("save"))
+	if err != nil {
+		util.ReturnError(util.APIErrorQueryParse, err.Error(), rw, r)
+	} else {
+		util.ReturnOK(rw, r)
 	}
-	return s.StartAndWait()
 }
--- a/payloader.go
+++ b/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
-}
--- a/publisher.go
+++ b/publisher.go
@@ -1,121 +1,134 @@
 package rtsp

 import (
-	"encoding/base64"
-	"encoding/hex"
-	"strconv"
 	"strings"

-	. "github.com/Monibuca/engine/v3"
-	. "github.com/Monibuca/utils/v3"
-	"github.com/aler9/gortsplib"
+	"github.com/bluenviron/gortsplib/v4/pkg/description"
+	"github.com/bluenviron/gortsplib/v4/pkg/format"
+	"github.com/bluenviron/mediacommon/pkg/codecs/mpeg4audio"
+	"github.com/pion/rtp"
+	"go.uber.org/zap"
+	. "m7s.live/engine/v4"
+	"m7s.live/engine/v4/codec"
+	"m7s.live/engine/v4/common"
+	. "m7s.live/engine/v4/track"
+	"m7s.live/engine/v4/util"
 )

-type RTSPublisher struct {
-	*Stream     `json:"-"`
-	stream      *gortsplib.ServerStream
-	processFunc []func([]byte)
+type RTSPPublisher struct {
+	Publisher
+	Tracks map[*description.Media]common.AVTrack `json:"-" yaml:"-"`
+	RTSPIO
 }

-func (p *RTSPublisher) setTracks(tracks gortsplib.Tracks) {
-	if p.processFunc != nil {
-		p.processFunc = p.processFunc[:len(tracks)]
-		return
+func (p *RTSPPublisher) SetTracks() error {
+	p.Tracks = make(map[*description.Media]common.AVTrack, len(p.session.Medias))
+	defer func() {
+		for _, track := range p.Tracks {
+			p.Info("set track", zap.String("name", track.GetName()))
+		}
+	}()
+	for _, track := range p.session.Medias {
+		for _, forma := range track.Formats {
+			switch f := forma.(type) {
+			case *format.H264:
+				vt := p.VideoTrack
+				if vt == nil {
+					vt = p.CreateVideoTrack(codec.CodecID_H264, byte(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 := p.VideoTrack
+				if vt == nil {
+					vt = p.CreateVideoTrack(codec.CodecID_H265, byte(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.AV1:
+				vt := p.VideoTrack
+				if vt == nil {
+					vt = p.CreateVideoTrack(codec.CodecID_AV1, byte(f.PayloadType()))
+				}
+				p.Tracks[track] = vt
+			case *format.MPEG4Audio:
+				at := p.AudioTrack
+				if at == nil {
+					conf := f.Config
+					if f.LATM && f.StreamMuxConfig != nil && len(f.StreamMuxConfig.Programs) > 0 && len(f.StreamMuxConfig.Programs[0].Layers) > 0 {
+						conf = f.StreamMuxConfig.Programs[0].Layers[0].AudioSpecificConfig
+					}
+					at := p.CreateAudioTrack(codec.CodecID_AAC, byte(f.PayloadType()), uint32(conf.SampleRate)).(*AAC)
+					at.AACFormat = f
+					at.AACDecoder.LATM = f.LATM
+					at.AACDecoder.IndexDeltaLength = f.IndexDeltaLength
+					at.AACDecoder.IndexLength = f.IndexLength
+					at.AACDecoder.SizeLength = f.SizeLength
+					if conf.Type == mpeg4audio.ObjectTypeAACLC {
+						at.Mode = 1
+					}
+					at.Channels = uint8(conf.ChannelCount)
+					asc, _ := conf.Marshal()
+					// 复用AVCC写入逻辑，解析出AAC的配置信息
+					at.WriteSequenceHead(append([]byte{0xAF, 0x00}, asc...))
+				}
+				p.Tracks[track] = p.AudioTrack
+			case *format.G711:
+				at := p.AudioTrack
+				if at == nil {
+					at = p.CreateAudioTrack(util.Conditoinal(f.MULaw, codec.CodecID_PCMU, codec.CodecID_PCMA), byte(f.PayloadType()), uint32(f.ClockRate()))
+				}
+				p.Tracks[track] = at
+			case *format.Opus:
+				at := p.AudioTrack
+				if at == nil {
+					p.CreateAudioTrack(codec.CodecID_OPUS, byte(f.PayloadType()), uint32(f.ClockRate()))
+				}
+				p.Tracks[track] = at
+			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 = p.CreateAudioTrack(util.Conditoinal(isMulaw, codec.CodecID_PCMU, codec.CodecID_PCMA), byte(f.PayloadType()), uint32(f.ClockRate()))
+					}
+					p.Tracks[track] = at
 				} else {
-		p.processFunc = make([]func([]byte), len(tracks))
+					p.Warn("unknown format", zap.Any("format", f.FMTP()))
 				}
-	for i, track := range tracks {
-		v, ok := track.Media.Attribute("rtpmap")
-		if !ok {
-			continue
 			}
+		}
+	}
+	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
+}

-		fmtp := make(map[string]string)
-		if v, ok := track.Media.Attribute("fmtp"); ok {
-			if tmp := strings.SplitN(v, " ", 2); len(tmp) == 2 {
-				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
-			}
-		}
+func (p *RTSPPublisher) OnPacket(m *description.Media, f format.Format, pack *rtp.Packet) {
+	if t, ok := p.Tracks[m]; ok {
+		t.WriteRTPPack(pack)
 	}
 }
--- a/server.go
+++ b/server.go
@@ -1,257 +1,122 @@
 package rtsp

 import (
-	"fmt"
-	"sync"
-	"unsafe"
-
-	"github.com/Monibuca/engine/v3"
-	. "github.com/Monibuca/utils/v3"
-	"github.com/Monibuca/utils/v3/codec"
-	"github.com/aler9/gortsplib"
-	"github.com/aler9/gortsplib/pkg/aac"
-	"github.com/aler9/gortsplib/pkg/base"
-	"github.com/pion/rtp"
-	"github.com/pion/rtp/codecs"
+	"github.com/bluenviron/gortsplib/v4"
+	"github.com/bluenviron/gortsplib/v4/pkg/base"
+	"github.com/bluenviron/gortsplib/v4/pkg/description"
+	"go.uber.org/zap"
+	"m7s.live/engine/v4/common"
 )

-// 接收RTSP推流：OnConnOpen->OnAnnounce->OnSetup->OnSessionOpen
-// 接收RTSP拉流：OnConnOpen->OnDescribe->OnSetup->OnSessionOpen
-type RTSPServer struct {
-	sync.Map
-}
-type RTSPSubscriber struct {
+type RTSPIO struct {
+	server     *gortsplib.Server
+	session    *description.Session
+	tracks     []*description.Media
 	stream     *gortsplib.ServerStream
-	engine.Subscriber
-	vt *engine.VideoTrack
-	at *engine.AudioTrack
+	audioTrack *description.Media
+	videoTrack *description.Media
 }

-// called after a connection is opened.
-func (sh *RTSPServer) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
-	Printf("rtsp conn opened")
+func (conf *RTSPConfig) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
+	RTSPPlugin.Debug("conn opened")
 }

-// called after a connection is closed.
-func (sh *RTSPServer) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
-	Printf("rtsp conn closed (%v)", ctx.Error)
-	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) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
+	RTSPPlugin.Debug("conn closed")
+	if p, ok := conf.LoadAndDelete(ctx.Conn); ok {
+		p.(common.IIO).Stop(zap.String("conn", "closed"))
 	}
 }

-// called after a session is opened.
-func (sh *RTSPServer) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
-	Printf("rtsp session opened")
+func (conf *RTSPConfig) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
+	RTSPPlugin.Debug("session opened")
 }

-// called after a session is closed.
-func (sh *RTSPServer) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
-	Printf("rtsp session closed")
-	if v, ok := sh.LoadAndDelete(ctx.Session); ok {
-		switch v := v.(type) {
-		case *RTSPublisher:
-			v.Close()
-		case *RTSPSubscriber:
-			v.Close()
-		}
-	}
+func (conf *RTSPConfig) OnSessionClose(ctx *gortsplib.ServerHandlerOnSessionCloseCtx) {
+	RTSPPlugin.Debug("session closed")
+	conf.Delete(ctx.Session)
 }

 // called after receiving a DESCRIBE request.
-func (sh *RTSPServer) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
-	Printf("describe request")
-	var err error
-	if s := engine.FindStream(ctx.Path); s != nil {
-		var tracks gortsplib.Tracks
-		var stream *gortsplib.ServerStream
-		var sub RTSPSubscriber
-		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
+func (conf *RTSPConfig) OnDescribe(ctx *gortsplib.ServerHandlerOnDescribeCtx) (*base.Response, *gortsplib.ServerStream, error) {
+	RTSPPlugin.Debug("describe request", zap.String("sdp", string(ctx.Request.Body)))
+	var suber RTSPSubscriber
+	suber.server = conf.server
+	suber.RemoteAddr = ctx.Conn.NetConn().RemoteAddr().String()
+	suber.SetIO(ctx.Conn.NetConn())
+	streamPath := ctx.Path
+	if ctx.Query != "" {
+		streamPath = streamPath + "?" + ctx.Query
 	}
-			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 {
-					for _, rtpack := range vpacketer.Packetize(nalu, (ts-st)*90) {
-						rtpack.Marker = i == len(pack.NALUs)-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)
+	if err := RTSPPlugin.Subscribe(streamPath, &suber); err == nil {
+		RTSPPlugin.Debug("describe replay ok")
+		conf.Store(ctx.Conn, &suber)
 		return &base.Response{
 			StatusCode: base.StatusOK,
-		}, stream, nil
-		// if stream, ok := s.ExtraProp.(*gortsplib.ServerStream); ok {
-		// 	return &base.Response{
-		// 		StatusCode: base.StatusOK,
-		// 	}, stream, nil
-		// }
-	}
+		}, suber.stream, nil
+	} else {
 		return &base.Response{
 			StatusCode: base.StatusNotFound,
-	}, nil, nil
+		}, suber.stream, nil
+	}
 }

-// called after receiving an ANNOUNCE request.
-func (sh *RTSPServer) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
-	Printf("announce request")
-	p := &RTSPublisher{
-		Stream: &engine.Stream{
-			StreamPath: ctx.Path,
-			Type:       "RTSP push",
-		},
+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
 		}
-	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)
+	}
+	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,
+	}, nil
+}
+func (conf *RTSPConfig) OnAnnounce(ctx *gortsplib.ServerHandlerOnAnnounceCtx) (*base.Response, error) {
+	RTSPPlugin.Debug("annouce request", zap.String("sdp", string(ctx.Request.Body)))
+	p := &RTSPPublisher{}
+	p.SetIO(ctx.Conn.NetConn())
+	if err := RTSPPlugin.Publish(ctx.Path, p); err == nil {
+		p.session = ctx.Description
+		p.stream = gortsplib.NewServerStream(conf.server, ctx.Description)
+		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)
-		}
-	}
-}
--- a/subscriber.go
+++ b/subscriber.go
@@ -0,0 +1,134 @@
+package rtsp
+
+import (
+	"fmt"
+
+	"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:
+			s.videoTrack = &description.Media{
+				Type: description.MediaTypeVideo,
+				Formats: []format.Format{&format.H264{
+					PacketizationMode: 1,
+					PayloadTyp:        v.PayloadType,
+					SPS:               v.ParamaterSets[0],
+					PPS:               v.ParamaterSets[1],
+				}},
+			}
+		case codec.CodecID_H265:
+			s.videoTrack = &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],
+				}},
+			}
+		case codec.CodecID_AV1:
+			var idx, profile, tail int
+			idx = int(v.ParamaterSets[1][0])
+			profile = int(v.ParamaterSets[1][1])
+			tail = int(v.ParamaterSets[1][2])
+			s.videoTrack = &description.Media{
+				Type: description.MediaTypeVideo,
+				Formats: []format.Format{&format.AV1{
+					PayloadTyp: v.PayloadType,
+					LevelIdx:   &idx,
+					Profile:    &profile,
+					Tier:       &tail,
+				}},
+			}
+		}
+		if s.videoTrack != nil {
+			s.tracks = append(s.tracks, s.videoTrack)
+			s.AddTrack(v)
+		}
+	case *track.Audio:
+		if s.Audio != nil {
+			return
+		}
+		switch v.CodecID {
+		case codec.CodecID_AAC:
+			f := v.AACFormat
+			if f == nil {
+				f = &format.MPEG4Audio{
+					PayloadTyp: v.PayloadType,
+					Config: &mpeg4audio.Config{
+						Type:         mpeg4audio.ObjectTypeAACLC,
+						SampleRate:   int(v.SampleRate),
+						ChannelCount: int(v.Channels),
+					},
+					SizeLength:       v.AACDecoder.SizeLength,
+					IndexLength:      v.AACDecoder.IndexLength,
+					IndexDeltaLength: v.AACDecoder.IndexDeltaLength,
+				}
+			}
+			s.audioTrack = &description.Media{
+				Type:    description.MediaTypeAudio,
+				Formats: []format.Format{f},
+			}
+		case codec.CodecID_PCMA:
+			s.audioTrack = &description.Media{
+				Type: description.MediaTypeAudio,
+				Formats: []format.Format{&format.Generic{
+					PayloadTyp: v.PayloadType,
+					ClockRat:   int(v.SampleRate),
+					RTPMa:      fmt.Sprintf("PCMA/%d", v.SampleRate),
+				}},
+			}
+		case codec.CodecID_PCMU:
+			s.audioTrack = &description.Media{
+				Type: description.MediaTypeAudio,
+				Formats: []format.Format{&format.Generic{
+					PayloadTyp: v.PayloadType,
+					ClockRat:   int(v.SampleRate),
+					RTPMa:      fmt.Sprintf("PCMU/%d", v.SampleRate),
+				}},
+			}
+		case codec.CodecID_OPUS:
+			s.audioTrack = &description.Media{
+				Type: description.MediaTypeAudio,
+				Formats: []format.Format{&format.Opus{
+					PayloadTyp: v.PayloadType,
+				}},
+			}
+		}
+		if s.audioTrack != nil {
+			s.tracks = append(s.tracks, s.audioTrack)
+			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)
+	}
+}
--- a/track.go
+++ b/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])),
-				},
-			},
-		},
-	}
-}
Author	SHA1	Message	Date
langhuihui	4dc893e9fb	fix: use engine's aac format	2024-09-27 10:05:32 +08:00
langhuihui	c943c2a0ec	feat: add LATM support	2024-09-26 20:34:00 +08:00
langhuihui	eafbf28a02	feat: support LATM	2024-09-26 14:08:06 +08:00
langhuihui	97d72548c5	feat: use new engine 4.15.1	2024-07-09 19:59:32 +08:00
langhuihui	5e722bf849	feat: add desc to config	2023-12-13 15:27:10 +08:00
langhuihui	6c1207ae43	feat: add av1 and opus	2023-11-17 17:31:51 +08:00
langhuihui	c794c0514f	rollback	2023-11-13 17:28:29 +08:00
langhuihui	e65f970097	fix: replace client wait	2023-11-12 12:07:14 +08:00
langhuihui	855038c132	Update dependencies and add audio format support	2023-11-03 14:12:05 +08:00
langhuihui	2a62472b1d	feat: push use tcp default	2023-10-27 09:52:19 +08:00
langhuihui	3e4dcdf649	feat: add sendoptions config to avoid send options	2023-10-25 10:22:29 +08:00
langhuihui	9ac21a130e	fix: auto close client	2023-10-25 08:57:33 +08:00
langhuihui	c13d8c659d	feat: update gortsplib to v4	2023-10-23 14:20:15 +08:00
langhuihui	c4700713a8	feat: remove pull protocol config	2023-10-11 17:01:09 +08:00
langhuihui	62409f14b5	chore: update engine version	2023-09-20 19:24:26 +08:00
langhuihui	df9c5a69cc	fix: memory leak	2023-09-15 14:23:59 +08:00
dexter	4a5ba9b834	Merge pull request #44 from umaYnit/v4 fix: 修复错误的流地址导致的Disconnect时panic	2023-08-31 08:42:12 +08:00
Ynit	5df8f237de	fix: 修复错误的流地址导致的Disconnect时panic	2023-08-31 00:12:07 +08:00
langhuihui	9936b2ca20	fix: upgrade gortsplib	2023-08-25 15:00:57 +08:00
langhuihui	943ba46f70	chore: fit engine udpate	2023-08-14 10:04:37 +08:00
langhuihui	035c3647eb	docs: update readme	2023-08-06 18:05:06 +08:00
langhuihui	17874da782	docs: udpate readme	2023-08-06 15:26:50 +08:00
langhuihui	24b63ca07c	feat: add stop reason, pull protocol default is tcp now	2023-08-06 15:15:26 +08:00
dexter	eb91dd1cee	Merge pull request #42 from Monibuca/rtsp-pcma Update publisher.go	2023-07-24 13:45:37 +08:00
banshan	e5d0d7d5d6	Update publisher.go ffmpeg rtsp pcma 推流	2023-07-24 13:43:44 +08:00
langhuihui	311b8732a8	fit engine update	2023-06-16 22:42:43 +08:00
langhuihui	351ff9fa0c	refactor: remove some stupid code	2023-05-25 14:15:09 +08:00
langhuihui	9defe4eec3	fit engine update	2023-05-23 20:58:59 +08:00
langhuihui	8e84bb637a	适配引擎修改	2023-04-24 14:13:46 +08:00
langhuihui	dc62df1861	更新readme	2023-04-23 13:00:11 +08:00
langhuihui	1ad45bfbc1	gortsplib升级到v3.2.1 增加默认缓存大小，解决高码率推拉流问题增加英文文档	2023-04-23 11:26:02 +08:00
langhuihui	a4773b4044	使用内置的协议自动选择机制	2023-04-15 08:48:44 +08:00
dexter	225bb482d5	适配引擎修改	2023-03-28 19:48:59 +08:00
dexter	ea5ea6f48a	SizeLength、IndexLength、IndexDeltaLength	2023-03-03 22:06:38 +08:00
dexter	beda2dd9e0	适配引擎修改	2023-02-26 10:38:53 +08:00
dexter	344ab6f463	移除clockrate	2023-02-20 00:19:39 +08:00
dexter	d5e73d11db	Merge pull request #32 from Monibuca/dependabot/go_modules/golang.org/x/net-0.7.0 Bump golang.org/x/net from 0.2.0 to 0.7.0	2023-02-19 10:22:56 +08:00
dependabot[bot]	12350b8af0	Bump golang.org/x/net from 0.2.0 to 0.7.0 Bumps [golang.org/x/net](https://github.com/golang/net) from 0.2.0 to 0.7.0. - [Release notes](https://github.com/golang/net/releases) - [Commits](https://github.com/golang/net/compare/v0.2.0...v0.7.0) --- updated-dependencies: - dependency-name: golang.org/x/net dependency-type: indirect ... Signed-off-by: dependabot[bot] <support@github.com>	2023-02-18 03:22:09 +00:00
dexter	3b671606b6	添加时钟频率配置	2023-02-08 14:19:36 +08:00
dexter	70321ccd1f	更新依赖库aler9 到2.1.0	2023-02-03 21:36:33 +08:00
dexter	5ddbd7a852	优化代码	2023-02-02 12:08:16 +08:00
dexter	f32f44db40	适配引擎升级	2023-02-01 11:00:36 +08:00
dexter	b44efd9749	推拉在异常时对连接关闭	2023-01-18 21:20:06 +08:00
dexter	7f4bfe4c78	使用rtsp拉流时将参数传入订阅者	2023-01-02 21:12:45 +08:00
dexter	187335d311	适配engine v4.9.5	2022-12-31 22:18:50 +08:00
dexter	1f3f4bfb0e	适配引擎4.9.3	2022-12-19 09:06:17 +08:00
dexter	3cbf22576b	🐛 FIX: 关闭流后断开rtsp拉流或者推流连接	2022-11-23 09:22:23 +08:00
dexter	1206a4636f	👌 IMPROVE: 升级gortsplib版本	2022-11-17 23:48:13 +08:00
dexter	6063f42181	🐛 FIX: enable not enabled	2022-10-10 21:51:08 +08:00
dexter	2e6679b3ea	👌 IMPROVE: 添加初始化监听失败日志	2022-10-03 17:57:49 +08:00
dexter	022857c3d7	👌 IMPROVE: 增加一些错误信息输出	2022-09-06 18:56:19 +08:00
dexter	0501a84da6	👌 IMPROVE: 导出RTSP插件	2022-09-04 12:43:42 +08:00
dexter	7d08e06922	兼容sdp中没有包含sps和pps的情况	2022-08-21 14:49:54 +08:00
dexter	754a28e506	更新依赖版本	2022-07-29 20:43:39 +08:00
dexter	9b058153d2	增加强制指定拉流协议的配置	2022-07-23 11:59:26 +08:00
dexter	ae37279dd1	aler9未处理回调中带有err的情况	2022-07-03 01:14:23 +08:00
dexter	93d6eedff2	适配engine 4.4.0	2022-06-25 20:11:32 +08:00
dexter	2c1d908d7e	跟随引擎4.3.0版本	2022-06-19 23:44:28 +08:00
dexter	ced54a94a4	添加序列化屏蔽字段	2022-06-19 15:30:06 +08:00
dexter	f9bc450d01	跟随aler9升级	2022-06-05 21:09:37 +08:00
dexter	44fb77d121	去掉g711代码	2022-05-27 13:42:06 +08:00
dexter	1b6981cccb	使用官方创建的g711	2022-05-27 12:41:00 +08:00
dexter	2cd295d32c	payloadType推拉对应	2022-05-26 20:35:27 +08:00
dexter	57ec489fef	engine已实现等待track功能	2022-05-26 11:20:11 +08:00
dexter	a2100768b4	rtp降为v1	2022-05-15 16:32:45 +08:00
dexter	76956b16d1	对推流可能无法解析的情况进行报错	2022-05-10 21:58:35 +08:00
dexter	91e1726920	暂时修复向远程推流	2022-05-10 15:30:14 +08:00
dexter	4a0da71ee9	修改推流	2022-05-09 22:03:55 +08:00
langhuihui	701d55469d	update readme	2022-05-04 21:38:20 +08:00
langhuihui	e273d0010e	将pcma的payloadType设置为8	2022-05-03 22:28:21 +08:00
langhuihui	96834e26a1	加上空格	2022-05-03 22:24:07 +08:00
dexter	7d0451c204	优化代码	2022-04-24 09:55:37 +08:00
langhuihui	c7c8858d36	添加API接口	2022-04-05 21:19:08 +08:00
dexter	731521f771	升级playblock	2022-03-27 11:33:34 +08:00
dexter	17334dd106	发布功能完善	2022-03-20 00:16:16 +08:00
dexter	2d9838bdfa	加上client逻辑	2022-03-19 01:15:19 +08:00
dexter	bd5a146ea6	v4 update	2022-03-18 01:24:32 +08:00
dexter	e411d30e91	Merge pull request #17 from jianglieshan/aler9 fix:修改rtsp插件作为服务端出流时，ssrc为0的bug	2022-01-16 13:12:11 +08:00
jianglieshan	709a4cee7b	fix:修改rtsp插件作为服务端出流时，ssrc为0的bug	2022-01-15 16:50:40 +08:00
dexter	a90f52769d	Merge pull request #16 from ziminghua/aler9 增加RTSPClient关闭的事件订阅，同步关闭客户端连接	2022-01-11 17:00:40 +08:00
訾明华	3764a26bbd	增加RTSPClient关闭的事件订阅，同步关闭客户端连接增加RTSPClient关闭的事件订阅，同步关闭客户端连接	2022-01-11 16:52:14 +08:00
dexter	2533ab2604	Merge pull request #15 from ziminghua/aler9 多slice的情况下，同步同一帧的时间戳	2022-01-11 11:40:38 +08:00
訾明华	db07f0d588	多slice的情况下，同步同一帧的时间戳 `vpacketer.Packetize`再打包的过程中会把当前的timestamp+samples作为下一次打包的时间戳，如果多slice会连续传递samples导致同一帧的时间戳不一致	2022-01-11 11:34:29 +08:00
dexter	f110513d70	增加配置项ReadBufferSize	2021-12-29 22:59:45 +08:00
dexter	8901f4c117	修复bug	2021-12-29 22:16:11 +08:00