Merge pull request #20 from ziminghua/aler9

解决发布超时后，重连RTSP，无法播放的BUG

LICENSE

@@ -0,0 +1,21 @@
+The MIT License (MIT)
+
+Copyright (c) 2019-present, dexter
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in
+all copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+THE SOFTWARE.

README.md

@@ -1,24 +1,63 @@
-# Monibuca 的RTSP 插件
+# RTSP插件
 
-主要功能是对RTSP地址进行拉流转换
+## 插件地址
 
-## 插件名称
+github.com/Monibuca/plugin-rtsp
 
-RTSP
+## 插件引入
+```go
+import (
+    _ "github.com/Monibuca/plugin-rtsp"
+)
+```
+
+## 默认插件配置
 
-## 配置
 ```toml
 [RTSP]
-BufferLength  = 2048
+# 端口接收推流
-AutoPull     = false
+ListenAddr = ":554"
-RemoteAddr   = "rtsp://localhost/${streamPath}"
+Reconnect = true
+[RTSP.AutoPullList]
+"live/rtsp1" = "rtsp://admin:admin@192.168.1.212:554/cam/realmonitor?channel=1&subtype=1"
+"live/rtsp2" = "rtsp://admin:admin@192.168.1.212:554/cam/realmonitor?channel=2&subtype=1"
 ```
-- BufferLength是指解析拉取的rtp包的缓冲大小
-- AutoPull是指当有用户订阅一个新房间的时候自动向远程拉流转发
-- RemoteAddr 指远程拉流地址，其中${streamPath}是占位符，实际使用流路径替换。
 
+- `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
 
-## 使用方法(拉流转发)
+### 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
+```
+
+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(RTSP).Publish("live/user1","rtsp://xxx.xxx.xxx.xxx/live/user1") 
+new(RTSPClient).PullStream("live/user1","rtsp://xxx.xxx.xxx.xxx/live/user1") 
-```
+```
+
+### 罗列所有的rtsp协议的流
+
+可调用接口
+`/api/rtsp/list`
+
+### 从m7s中拉取rtsp协议流
+
+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

client.go

@@ -1,546 +1,214 @@
-package rtspplugin
+package rtsp
 
 import (
-	"crypto/md5"
+	"errors"
-	b64 "encoding/base64"
-	"encoding/hex"
-	"fmt"
-	"io"
-	"log"
-	"net"
-	"net/url"
-	"regexp"
-	"strconv"
-	"strings"
 	"time"
+	"unsafe"
+
+	. "github.com/Monibuca/engine/v3"
+	. "github.com/Monibuca/utils/v3"
+	"github.com/Monibuca/utils/v3/codec"
+	"github.com/aler9/gortsplib"
+	"github.com/aler9/gortsplib/pkg/aac"
+	"github.com/aler9/gortsplib/pkg/base"
+	"github.com/pion/rtp"
+	"github.com/pion/rtp/codecs"
 )
 
-var (
+type RTSPClient struct {
-	VideoWidth  int
+	RTSPublisher
-	VideoHeight int
+	Transport         gortsplib.Transport
-	spropReg    *regexp.Regexp
+	*gortsplib.Client `json:"-"`
-	configReg   *regexp.Regexp
-)
-
-func init() {
-	spropReg, _ = regexp.Compile("sprop-parameter-sets=([^;]+)")
-	configReg, _ = regexp.Compile("config=([^;]+)")
 }
 
-type RtspClient struct {
+// PullStream 从外部拉流
-	socket              net.Conn
+func (rtsp *RTSPClient) PullStream(streamPath string, rtspUrl string) (err error) {
-	OutGoing            chan []byte //out chanel
+	rtsp.Stream = &Stream{
-	Signals             chan bool   //Signals quit
+		StreamPath: streamPath,
-	host                string      //host
+		Type:       "RTSP Pull",
-	port                string      //port
+		ExtraProp:  rtsp,
-	uri                 string      //url
-	auth                bool        //aut
-	login               string
-	password            string   //password
-	session             string   //rtsp session
-	responce            string   //responce string
-	bauth               string   //string b auth
-	track               []string //rtsp track
-	cseq                int      //qury number
-	videow              int
-	videoh              int
-	SPS                 []byte
-	PPS                 []byte
-	Header              string
-	AudioSpecificConfig []byte
-}
-
-//RtspClientNew 返回空的初始化对象
-func RtspClientNew(bufferLength int) *RtspClient {
-	Obj := &RtspClient{
-		cseq:     1,                               //查询起始号码
-		Signals:  make(chan bool, 1),              //一个消息缓冲通道
-		OutGoing: make(chan []byte, bufferLength), //输出通道
 	}
-	return Obj
+	if result := rtsp.Publish(); result {
-}
+		rtsp.URL = rtspUrl
-
+		if config.Reconnect {
-func (this *RtspClient) Client(rtsp_url string) (bool, string) {
+			go func() {
-	//Check back url
+				for rtsp.pullStream(); rtsp.Err() == nil; rtsp.pullStream() {
-	if !this.ParseUrl(rtsp_url) {
+					Printf("reconnecting:%s in 5 seconds", rtspUrl)
-		return false, "Incorrect url"
+					if rtsp.Transport == gortsplib.TransportTCP {
-	}
+						rtsp.Transport = gortsplib.TransportUDP
-	//Install connect to camera
-	if !this.Connect() {
-		return false, "cannot connect"
-	}
-	//Phase 1 options camera phase 1
-	//Send options request
-	if !this.Write("OPTIONS " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\n\r\n") {
-		return false, "Unable to send options message"
-	}
-	//Read the response to the options request
-	if status, message := this.Read(); !status {
-		return false, "Unable to read options response connection lost"
-	} else if status && strings.Contains(message, "Digest") {
-		if !this.AuthDigest("OPTIONS", message) {
-			return false, "Summary of authorization required"
-		}
-	} else if status && strings.Contains(message, "Basic") {
-		if !this.AuthBasic("OPTIONS", message) {
-			return false, "Need certification Basic"
-		}
-	} else if !strings.Contains(message, "200") {
-		return false, "error OPTIONS not status code 200 OK " + message
-	}
-
-	////////////PHASE 2 DESCRIBE
-	log.Println("DESCRIBE " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + this.bauth + "\r\n\r\n")
-	if !this.Write("DESCRIBE " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + this.bauth + "\r\n\r\n") {
-		return false, "Unable to send query DESCRIBE"
-	}
-	if status, message := this.Read(); !status {
-		return false, "Can't read response for decscribe connection loss?"
-	} else if status && strings.Contains(message, "Digest") {
-		if !this.AuthDigest("DESCRIBE", message) {
-			return false, "Summary of authorization required"
-		}
-	} else if status && strings.Contains(message, "Basic") {
-		if !this.AuthBasic("DESCRIBE", message) {
-			return false, "Basis of authorization required"
-		}
-	} else if !strings.Contains(message, "200") {
-		return false, "error DESCRIBE not status code 200 OK " + message
-	} else {
-		this.Header = message
-		this.track = this.ParseMedia(message)
-
-	}
-	if len(this.track) == 0 {
-		return false, "error track not found "
-	}
-	//PHASE 3 SETUP
-	log.Println("SETUP " + this.uri + "/" + this.track[0] + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nTransport: RTP/AVP/TCP;unicast;interleaved=0-1" + this.bauth + "\r\n\r\n")
-	if !this.Write("SETUP " + this.uri + "/" + this.track[0] + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nTransport: RTP/AVP/TCP;unicast;interleaved=0-1" + this.bauth + "\r\n\r\n") {
-		return false, ""
-	}
-	if status, message := this.Read(); !status {
-		return false, "Unable to read response for missing setup connection."
-
-	} else if !strings.Contains(message, "200") {
-		if strings.Contains(message, "401") {
-			str := this.AuthDigest_Only("SETUP", message)
-			if !this.Write("SETUP " + this.uri + "/" + this.track[0] + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nTransport: RTP/AVP/TCP;unicast;interleaved=0-1" + this.bauth + str + "\r\n\r\n") {
-				return false, ""
-			}
-			if status, message := this.Read(); !status {
-				return false, "Unable to read response for missing setup connection."
-
-			} else if !strings.Contains(message, "200") {
-
-				return false, "error SETUP not status code 200 OK " + message
-
-			} else {
-				this.session = ParseSession(message)
-			}
-		} else {
-			return false, "error SETUP not status code 200 OK " + message
-		}
-	} else {
-		log.Println(message)
-		this.session = ParseSession(message)
-		log.Println(this.session)
-	}
-	if len(this.track) > 1 {
-
-		if !this.Write("SETUP " + this.uri + "/" + this.track[1] + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nTransport: RTP/AVP/TCP;unicast;interleaved=2-3" + "\r\nSession: " + this.session + this.bauth + "\r\n\r\n") {
-			return false, ""
-		}
-		if status, message := this.Read(); !status {
-			return false, "Unable to read response for missing setup audio connection."
-
-		} else if !strings.Contains(message, "200") {
-			if strings.Contains(message, "401") {
-				str := this.AuthDigest_Only("SETUP", message)
-				if !this.Write("SETUP " + this.uri + "/" + this.track[1] + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nTransport: RTP/AVP/TCP;unicast;interleaved=2-3" + this.bauth + str + "\r\n\r\n") {
-					return false, ""
-				}
-				if status, message := this.Read(); !status {
-					return false, "Unable to read response for missing setup audio connection."
-
-				} else if !strings.Contains(message, "200") {
-
-					return false, "error SETUP not status code 200 OK " + message
-
-				} else {
-					log.Println(message)
-					this.session = ParseSession(message)
-				}
-			} else {
-				return false, "error SETUP not status code 200 OK " + message
-			}
-		} else {
-			log.Println(message)
-			this.session = ParseSession(message)
-		}
-	}
-
-	//PHASE 4 SETUP
-	log.Println("PLAY " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nSession: " + this.session + this.bauth + "\r\n\r\n")
-	if !this.Write("PLAY " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nSession: " + this.session + this.bauth + "\r\n\r\n") {
-		return false, ""
-	}
-	if status, message := this.Read(); !status {
-		return false, "Unable to read play response lost connection"
-
-	} else if !strings.Contains(message, "200") {
-		//return false, "Ошибка PLAY not status code 200 OK " + message
-		if strings.Contains(message, "401") {
-			str := this.AuthDigest_Only("PLAY", message)
-			if !this.Write("PLAY " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nSession: " + this.session + this.bauth + str + "\r\n\r\n") {
-				return false, ""
-			}
-			if status, message := this.Read(); !status {
-				return false, "Unable to read play response lost connection"
-
-			} else if !strings.Contains(message, "200") {
-
-				return false, "error PLAY not status code 200 OK " + message
-
-			} else {
-				//this.session = ParseSession(message)
-				log.Print(message)
-				go this.RtspRtpLoop()
-				return true, "ok"
-			}
-		} else {
-			return false, "error PLAY not status code 200 OK " + message
-		}
-	} else {
-		log.Print(message)
-		go this.RtspRtpLoop()
-		return true, "ok"
-	}
-	return false, "other error"
-}
-
-/*
-	The RTP header has the following format:
-    0                   1                   2                   3
-    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
-   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-   |                           timestamp                           |
-   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-   |           synchronization source (SSRC) identifier            |
-   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-   |            contributing source (CSRC) identifiers             |
-   |                             ....                              |
-   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-   version (V): 2 bits
-      This field identifies the version of RTP.  The version defined by
-      this specification is two (2).  (The value 1 is used by the first
-      draft version of RTP and the value 0 is used by the protocol
-      initially implemented in the "vat" audio tool.)
-   padding (P): 1 bit
-      If the padding bit is set, the packet contains one or more
-      additional padding octets at the end which are not part of the
-      payload.  The last octet of the padding contains a count of how
-      many padding octets should be ignored, including itself.  Padding
-      may be needed by some encryption algorithms with fixed block sizes
-      or for carrying several RTP packets in a lower-layer protocol data
-      unit.
-   extension (X): 1 bit
-      If the extension bit is set, the fixed header MUST be followed by
-      exactly one header extension, with a format defined in Section
-      5.3.1.
-*/
-func (this *RtspClient) RtspRtpLoop() {
-	defer func() {
-		this.Signals <- true
-	}()
-	header := make([]byte, 4)
-	payload := make([]byte, 4096)
-	//sync := make([]byte, 256)
-	sync_b := make([]byte, 1)
-	timer := time.Now()
-	for {
-		if int(time.Now().Sub(timer).Seconds()) > 50 {
-			if !this.Write("OPTIONS " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + "\r\nSession: " + this.session + this.bauth + "\r\n\r\n") {
-				return
-			}
-			timer = time.Now()
-		}
-		this.socket.SetDeadline(time.Now().Add(50 * time.Second))
-		//read rtp hdr 4
-		if n, err := io.ReadFull(this.socket, header); err != nil || n != 4 {
-			//rtp hdr read error
-			return
-		}
-		//log.Println(header)
-		if header[0] != 36 {
-			//log.Println("desync?", this.host)
-			for {
-				///////////////////////////skeep/////////////////////////////////////
-				if n, err := io.ReadFull(this.socket, sync_b); err != nil && n != 1 {
-					return
-				} else if sync_b[0] == 36 {
-					header[0] = 36
-					if n, err := io.ReadFull(this.socket, header[1:]); err != nil && n == 3 {
-						return
-					}
-					break
-				}
-			}
-			/*
-				//вычитываем 256 в попытке отсять мусор обрезать RTSP
-				if string(header) == "RTSP" {
-					if n, err := io.ReadFull(this.socket, sync); err != nil && n == 256 {
-						return
 					} else {
-						rtsp_rtp := []byte(strings.Split(string(sync), "\r\n\r\n")[1])
+						rtsp.Transport = gortsplib.TransportTCP
-						//отправим все что есть в буфере
-						this.SendBufer(rtsp_rtp)
-						continue
 					}
+					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 {
-					log.Println("full desync")
+					return err
-					return
 				}
-			*/
+			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 {
-		payloadLen := (int)(header[2])<<8 + (int)(header[3])
+			var st uint32
-		//log.Println("payloadLen", payloadLen)
+			trackId := trackIds
-		if payloadLen > 4096 || payloadLen < 12 {
+			switch sub.at.CodecID {
-			log.Println("desync", this.uri, payloadLen)
+			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,
+	}
+	// parse URL
+	u, err := base.ParseURL(client.URL)
+	if err != nil {
+		Printf("ParseURL:%s error:%v", client.URL, err)
+		return
+	}
+	// connect to the server
+	if err = client.Start(u.Scheme, u.Host); err != nil {
+		Printf("connect:%s error:%v", client.URL, err)
+		return
+	}
+	client.OnClose = func() {
+		client.Client.Close()
+	}
+	//client.close should be after connected!
+	defer client.Client.Close()
+	var res *base.Response
+	if res, err = client.Options(u); err != nil {
+		Printf("option:%s error:%v", client.URL, err)
+		return
+	}
+	Println(res)
+	// find published tracks
+	tracks, baseURL, res, err := client.Describe(u)
+	if err != nil {
+		Printf("Describe:%s error:%v", client.URL, err)
+		return
+	}
+	Println(res)
+	if client.processFunc == nil {
+		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
 		}
-		if n, err := io.ReadFull(this.socket, payload[:payloadLen]); err != nil || n != payloadLen {
+		Println(res)
-			return
-		} else {
-			this.OutGoing <- append(header, payload[:n]...)
-		}
 	}
-
+	// start reading tracks
-}
+	if res, err = client.Play(nil); err != nil {
-
+		Printf("Play:%s error:%v", baseURL.String(), err)
-//unsafe!
+		return
-func (this *RtspClient) SendBufer(bufer []byte) {
+	}
-	//Here you need to send all the packages from the send all buffer?
+	Println(res)
-	payload := make([]byte, 4096)
+	// wait until a fatal error
-	for {
+	var fatalChan = make(chan error)
-		if len(bufer) < 4 {
+	go func() {
-			log.Fatal("bufer small")
+		fatalChan <- client.Wait()
-		}
+	}()
-		dataLength := (int)(bufer[2])<<8 + (int)(bufer[3])
+	select {
-		if dataLength > len(bufer)+4 {
+	case err := <-fatalChan:
-			if n, err := io.ReadFull(this.socket, payload[:dataLength-len(bufer)+4]); err != nil {
+		Printf("Wait:%s error:%v", baseURL.String(), err)
-				return
+	case <-client.Done():
-			} else {
+		Printf("client:%s done", client.URL)
-				this.OutGoing <- append(bufer, payload[:n]...)
-				return
-			}
-
-		} else {
-			this.OutGoing <- bufer[:dataLength+4]
-			bufer = bufer[dataLength+4:]
-		}
 	}
 }
-func (this *RtspClient) Connect() bool {
-	d := &net.Dialer{Timeout: 3 * time.Second}
-	conn, err := d.Dial("tcp", this.host+":"+this.port)
-	if err != nil {
-		return false
-	}
-	this.socket = conn
-	return true
-}
-func (this *RtspClient) Write(message string) bool {
-	this.cseq += 1
-	if _, e := this.socket.Write([]byte(message)); e != nil {
-		return false
-	}
-	return true
-}
-func (this *RtspClient) Read() (bool, string) {
-	buffer := make([]byte, 4096)
-	if nb, err := this.socket.Read(buffer); err != nil || nb <= 0 {
-		log.Println("socket read failed", err)
-		return false, ""
-	} else {
-		return true, string(buffer[:nb])
-	}
-}
-func (this *RtspClient) AuthBasic(phase string, message string) bool {
-	this.bauth = "\r\nAuthorization: Basic " + b64.StdEncoding.EncodeToString([]byte(this.login+":"+this.password))
-	if !this.Write(phase + " " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + this.bauth + "\r\n\r\n") {
-		return false
-	}
-	if status, message := this.Read(); status && strings.Contains(message, "200") {
-		this.track = this.ParseMedia(message)
-		return true
-	}
-	return false
-}
-func (this *RtspClient) AuthDigest(phase string, message string) bool {
-	nonce := ParseDirective(message, "nonce")
-	realm := ParseDirective(message, "realm")
-	hs1 := GetMD5Hash(this.login + ":" + realm + ":" + this.password)
-	hs2 := GetMD5Hash(phase + ":" + this.uri)
-	responce := GetMD5Hash(hs1 + ":" + nonce + ":" + hs2)
-	dauth := "\r\n" + `Authorization: Digest username="` + this.login + `", realm="` + realm + `", nonce="` + nonce + `", uri="` + this.uri + `", response="` + responce + `"`
-	if !this.Write(phase + " " + this.uri + " RTSP/1.0\r\nCSeq: " + strconv.Itoa(this.cseq) + dauth + "\r\n\r\n") {
-		return false
-	}
-	if status, message := this.Read(); status && strings.Contains(message, "200") {
-		this.track = this.ParseMedia(message)
-		return true
-	}
-	return false
-}
-func (this *RtspClient) AuthDigest_Only(phase string, message string) string {
-	nonce := ParseDirective(message, "nonce")
-	realm := ParseDirective(message, "realm")
-	hs1 := GetMD5Hash(this.login + ":" + realm + ":" + this.password)
-	hs2 := GetMD5Hash(phase + ":" + this.uri)
-	responce := GetMD5Hash(hs1 + ":" + nonce + ":" + hs2)
-	dauth := "\r\n" + `Authorization: Digest username="` + this.login + `", realm="` + realm + `", nonce="` + nonce + `", uri="` + this.uri + `", response="` + responce + `"`
-	return dauth
-}
-func (this *RtspClient) ParseUrl(rtsp_url string) bool {
-
-	u, err := url.Parse(rtsp_url)
-	if err != nil {
-		return false
-	}
-	phost := strings.Split(u.Host, ":")
-	this.host = phost[0]
-	if len(phost) == 2 {
-		this.port = phost[1]
-	} else {
-		this.port = "554"
-	}
-	this.login = u.User.Username()
-	this.password, this.auth = u.User.Password()
-	if u.RawQuery != "" {
-		this.uri = "rtsp://" + this.host + ":" + this.port + u.Path + "?" + string(u.RawQuery)
-	} else {
-		this.uri = "rtsp://" + this.host + ":" + this.port + u.Path
-	}
-	return true
-}
-func (this *RtspClient) Close() {
-	if this.socket != nil {
-		this.socket.Close()
-	}
-}
-func ParseDirective(header, name string) string {
-	index := strings.Index(header, name)
-	if index == -1 {
-		return ""
-	}
-	start := 1 + index + strings.Index(header[index:], `"`)
-	end := start + strings.Index(header[start:], `"`)
-	return strings.TrimSpace(header[start:end])
-}
-func ParseSession(header string) string {
-	mparsed := strings.Split(header, "\r\n")
-	for _, element := range mparsed {
-		if strings.Contains(element, "Session:") {
-			if strings.Contains(element, ";") {
-				fist := strings.Split(element, ";")[0]
-				return fist[9:]
-			} else {
-				return element[9:]
-			}
-		}
-	}
-	return ""
-}
-
-// func ParseMedia(header string) []string {
-// 	letters := []string{}
-// 	mparsed := strings.Split(header, "\r\n")
-// 	paste := ""
-
-// 	// if true {
-// 	// 	log.Println("headers", header)
-// 	// }
-
-// 	for _, element := range mparsed {
-// 		if strings.Contains(element, "a=control:") && !strings.Contains(element, "*") {
-// 			paste = element[10:]
-// 			if strings.Contains(element, "/") {
-// 				striped := strings.Split(element, "/")
-// 				paste = striped[len(striped)-1]
-// 			}
-// 			letters = append(letters, paste)
-// 		}
-
-// 		dimensionsPrefix := "a=x-dimensions:"
-// 		if strings.HasPrefix(element, dimensionsPrefix) {
-// 			dims := []int{}
-// 			for _, s := range strings.Split(element[len(dimensionsPrefix):], ",") {
-// 				v := 0
-// 				fmt.Sscanf(s, "%d", &v)
-// 				if v <= 0 {
-// 					break
-// 				}
-// 				dims = append(dims, v)
-// 			}
-// 			if len(dims) == 2 {
-// 				VideoWidth = dims[0]
-// 				VideoHeight = dims[1]
-// 			}
-// 		}
-// 		if strings.Contains(element, "sprop-parameter-sets") {
-// 			group := spropReg.FindAllStringSubmatch(element, -1)
-// 			log.Println(group[1])
-// 		}
-// 	}
-// 	return letters
-// }
-func GetMD5Hash(text string) string {
-	hash := md5.Sum([]byte(text))
-	return hex.EncodeToString(hash[:])
-}
-func (this *RtspClient) ParseMedia(header string) []string {
-	letters := []string{}
-	log.Println(header)
-	mparsed := strings.Split(header, "\r\n")
-	paste := ""
-	for _, element := range mparsed {
-		if strings.Contains(element, "a=control:") && !strings.Contains(element, "*") {
-			paste = element[10:]
-			if strings.Contains(element, "/") {
-				striped := strings.Split(element, "/")
-				paste = striped[len(striped)-1]
-			}
-			letters = append(letters, paste)
-		}
-
-		dimensionsPrefix := "a=x-dimensions:"
-		if strings.HasPrefix(element, dimensionsPrefix) {
-			dims := []int{}
-			for _, s := range strings.Split(element[len(dimensionsPrefix):], ",") {
-				v := 0
-				fmt.Sscanf(s, "%d", &v)
-				if v <= 0 {
-					break
-				}
-				dims = append(dims, v)
-			}
-			if len(dims) == 2 {
-				this.videow = dims[0]
-				this.videoh = dims[1]
-			}
-		}
-		group := spropReg.FindAllStringSubmatch(element, -1)
-		if len(group) > 0 {
-			group := strings.Split(group[0][1], ",")
-			this.SPS, _ = b64.StdEncoding.DecodeString(group[0])
-			this.PPS, _ = b64.StdEncoding.DecodeString(group[1])
-		} else if group = configReg.FindAllStringSubmatch(element, -1); len(group) > 0 {
-			this.AudioSpecificConfig, _ = hex.DecodeString(group[0][1])
-		}
-	}
-	return letters
-}

go.mod

@@ -1,8 +1,11 @@
-module github.com/Monibuca/plugin-rtsp
+module github.com/Monibuca/plugin-rtsp/v3
 
-go 1.13
+go 1.16
 
 require (
-	github.com/Monibuca/engine v1.2.1
+	github.com/Monibuca/engine/v3 v3.4.1
-	github.com/Monibuca/engine/v2 v2.0.0 // indirect
+	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
 )

go.sum

@@ -1,47 +1,101 @@
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main.go

@@ -1,220 +1,113 @@
-package rtspplugin
+package rtsp
 
 import (
-	"bytes"
+	"encoding/json"
 	"fmt"
 	"log"
 	"net/http"
-	"strings"
-	"sync"
 	"time"
 
-	. "github.com/Monibuca/engine/v2"
+	. "github.com/Monibuca/engine/v3"
-	. "github.com/Monibuca/engine/v2/avformat"
+	. "github.com/Monibuca/utils/v3"
-	"github.com/Monibuca/engine/v2/util"
+	"github.com/aler9/gortsplib"
 )
 
-var collection = sync.Map{}
 var config = struct {
-	BufferLength int
+	ListenAddr     string
-	AutoPull     bool
+	UDPAddr        string
-	RemoteAddr   string
+	RTCPAddr       string
-}{2048, true, "rtsp://localhost/${streamPath}"}
+	Timeout        int
+	Reconnect      bool
+	AutoPullList   map[string]string
+	AutoPushList   map[string]string
+	ReadBufferSize int
+}{":554", ":8000", ":8001", 0, false, nil, nil, 2048}
+
+var pconfig = PluginConfig{
+	Name:   "RTSP",
+	Config: &config,
+}
 
 func init() {
-	InstallPlugin(&PluginConfig{
+	pconfig.Install(runPlugin)
-		Name:   "RTSP",
+}
-		Type:   PLUGIN_PUBLISHER | PLUGIN_HOOK,
+
-		Config: &config,
+func getRtspList() (info []*Stream) {
-		Run:    runPlugin,
+	for _, s := range Streams.ToList() {
-		HotConfig: map[string]func(interface{}){
+		switch s.ExtraProp.(type) {
-			"AutoPull": func(value interface{}) {
+		case *RTSPublisher:
-				config.AutoPull = value.(bool)
+			info = append(info, s)
-			},
+		case *RTSPClient:
-		},
+			info = append(info, s)
-	})
+		}
+	}
+	return
 }
 func runPlugin() {
-	OnSubscribeHooks.AddHook(func(s *Subscriber) {
+	http.HandleFunc("/api/rtsp/list", func(w http.ResponseWriter, r *http.Request) {
-		if config.AutoPull && s.Publisher == nil {
+		CORS(w, r)
-			new(RTSP).Publish(s.StreamPath, strings.Replace(config.RemoteAddr, "${streamPath}", s.StreamPath, -1))
+		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())
-	http.HandleFunc("/rtsp/list", func(w http.ResponseWriter, r *http.Request) {
-		sse := util.NewSSE(w, r.Context())
 		var err error
 		for tick := time.NewTicker(time.Second); err == nil; <-tick.C {
-			var info []*RTSPInfo
+			err = sse.WriteJSON(getRtspList())
-			collection.Range(func(key, value interface{}) bool {
-				rtsp := value.(*RTSP)
-				pinfo := &rtsp.RTSPInfo
-				pinfo.BufferRate = len(rtsp.OutGoing) * 100 / config.BufferLength
-				info = append(info, pinfo)
-				return true
-			})
-			err = sse.WriteJSON(info)
 		}
 	})
-	http.HandleFunc("/rtsp/pull", func(w http.ResponseWriter, r *http.Request) {
+	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")
-		var err error
+		save := r.URL.Query().Get("save")
-		if err == nil {
+		if err := (&RTSPClient{Transport: gortsplib.TransportTCP}).PullStream(streamPath, targetURL); err == nil {
-			new(RTSP).Publish(streamPath, targetURL)
+			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 {
-type RTSP struct {
+			Println(err)
-	Publisher
-	*RtspClient
-	RTSPInfo
-}
-type RTSPInfo struct {
-	SyncCount  int64
-	Header     *string
-	BufferRate int
-	StreamInfo *StreamInfo
-}
-
-func (rtsp *RTSP) run() {
-	fuBuffer := []byte{}
-	iframeHead := []byte{0x17, 0x01, 0, 0, 0}
-	pframeHead := []byte{0x27, 0x01, 0, 0, 0}
-	spsHead := []byte{0xE1, 0, 0}
-	ppsHead := []byte{0x01, 0, 0}
-	nalLength := []byte{0, 0, 0, 0}
-	avcsent := false
-	aacsent := false
-	handleNALU := func(nalType byte, payload []byte, ts int64) {
-		rtsp.SyncCount++
-		vl := len(payload)
-		switch nalType {
-		// case NALU_SPS:
-		// 	r := bytes.NewBuffer([]byte{})
-		// 	r.Write(RTMP_AVC_HEAD)
-		// 	util.BigEndian.PutUint16(spsHead[1:], uint16(vl))
-		// 	r.Write(spsHead)
-		// 	r.Write(payload)
-		// case NALU_PPS:
-		// 	r := bytes.NewBuffer([]byte{})
-		// 	util.BigEndian.PutUint16(ppsHead[1:], uint16(vl))
-		// 	r.Write(ppsHead)
-		// 	r.Write(payload)
-		// 	rtsp.PushVideo(0, r.Bytes())
-		// avcsent = true
-		case NALU_IDR_Picture:
-			if !avcsent {
-				r := bytes.NewBuffer([]byte{})
-				r.Write(RTMP_AVC_HEAD)
-				util.BigEndian.PutUint16(spsHead[1:], uint16(len(rtsp.SPS)))
-				r.Write(spsHead)
-				r.Write(rtsp.SPS)
-				util.BigEndian.PutUint16(ppsHead[1:], uint16(len(rtsp.PPS)))
-				r.Write(ppsHead)
-				r.Write(rtsp.PPS)
-				rtsp.PushVideo(0, r.Bytes())
-				avcsent = true
-			}
-			r := bytes.NewBuffer([]byte{})
-			util.BigEndian.PutUint24(iframeHead[2:], 0)
-			r.Write(iframeHead)
-			util.BigEndian.PutUint32(nalLength, uint32(vl))
-			r.Write(nalLength)
-			r.Write(payload)
-			rtsp.PushVideo(uint32(ts), r.Bytes())
-		case NALU_Non_IDR_Picture:
-			r := bytes.NewBuffer([]byte{})
-			util.BigEndian.PutUint24(pframeHead[2:], 0)
-			r.Write(pframeHead)
-			util.BigEndian.PutUint32(nalLength, uint32(vl))
-			r.Write(nalLength)
-			r.Write(payload)
-			rtsp.PushVideo(uint32(ts), r.Bytes())
 		}
 	}
-	for {
-		select {
-		case <-rtsp.Done():
-			return
-		case data, ok := <-rtsp.OutGoing:
-			if ok && data[0] == 36 {
-				if data[1] == 0 {
-					cc := data[4] & 0xF
-					//rtp header
-					rtphdr := 12 + cc*4
 
-					//packet time
+	go AddHook(HOOK_PUBLISH, func(s *Stream) {
-					ts := (int64(data[8]) << 24) + (int64(data[9]) << 16) + (int64(data[10]) << 8) + (int64(data[11]))
+		for streamPath, url := range config.AutoPushList {
-
+			if s.StreamPath == streamPath {
-					//packet number
+				(&RTSPClient{}).PushStream(streamPath, url)
-					//packno := (int64(data[6]) << 8) + int64(data[7])
-					data = data[4+rtphdr:]
-					nalType := data[0] & 0x1F
-
-					if nalType >= 1 && nalType <= 23 {
-						handleNALU(nalType, data, ts)
-					} else if nalType == 28 {
-						isStart := data[1]&0x80 != 0
-						isEnd := data[1]&0x40 != 0
-						nalType := data[1] & 0x1F
-						//nri := (data[1]&0x60)>>5
-						nal := data[0]&0xE0 | data[1]&0x1F
-						if isStart {
-							fuBuffer = []byte{0}
-						}
-						fuBuffer = append(fuBuffer, data[2:]...)
-						if isEnd {
-							fuBuffer[0] = nal
-							handleNALU(nalType, fuBuffer, ts)
-						}
-					}
-
-				} else if data[1] == 2 {
-					// audio
-					if !aacsent {
-						rtsp.PushAudio(0, append([]byte{0xAF, 0x00}, rtsp.AudioSpecificConfig...))
-						aacsent = true
-					}
-					cc := data[4] & 0xF
-					rtphdr := 12 + cc*4
-					payload := data[4+rtphdr:]
-					auHeaderLen := (int16(payload[0]) << 8) + int16(payload[1])
-					auHeaderLen = auHeaderLen >> 3
-					auHeaderCount := int(auHeaderLen / 2)
-					var auLenArray []int
-					for iIndex := 0; iIndex < int(auHeaderCount); iIndex++ {
-						auHeaderInfo := (int16(payload[2+2*iIndex]) << 8) + int16(payload[2+2*iIndex+1])
-						auLen := auHeaderInfo >> 3
-						auLenArray = append(auLenArray, int(auLen))
-					}
-					startOffset := 2 + 2*auHeaderCount
-					for _, auLen := range auLenArray {
-						endOffset := startOffset + auLen
-						addHead := []byte{0xAF, 0x01}
-						rtsp.PushAudio(0, append(addHead, payload[startOffset:endOffset]...))
-						startOffset = startOffset + auLen
-					}
-				}
 			}
 		}
+	})
+	if config.ListenAddr != "" {
+		go log.Fatal(ListenRtsp(config.ListenAddr))
 	}
 }
-func (rtsp *RTSP) Publish(streamPath string, rtspUrl string) (result bool) {
+
-	if result = rtsp.Publisher.Publish(streamPath); result {
+func ListenRtsp(addr string) error {
-		rtsp.Type = "RTSP"
+	defer log.Println("rtsp server start!")
-		rtsp.RTSPInfo.StreamInfo = &rtsp.Stream.StreamInfo
+	s := &gortsplib.Server{
-		rtsp.RtspClient = RtspClientNew(config.BufferLength)
+		Handler:           &RTSPServer{},
-		rtsp.RTSPInfo.Header = &rtsp.RtspClient.Header
+		RTSPAddress:       addr,
-		if status, message := rtsp.RtspClient.Client(rtspUrl); !status {
+		UDPRTPAddress:     config.UDPAddr,
-			log.Println(message)
+		UDPRTCPAddress:    config.RTCPAddr,
-			return false
+		MulticastIPRange:  "224.1.0.0/16",
-		}
+		MulticastRTPPort:  8002,
-		collection.Store(streamPath, rtsp)
+		MulticastRTCPPort: 8003,
-		go rtsp.run()
 	}
-	return
+	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

@@ -0,0 +1,121 @@
+package rtsp
+
+import (
+	"encoding/base64"
+	"encoding/hex"
+	"strconv"
+	"strings"
+
+	. "github.com/Monibuca/engine/v3"
+	. "github.com/Monibuca/utils/v3"
+	"github.com/aler9/gortsplib"
+)
+
+type RTSPublisher struct {
+	*Stream     `json:"-"`
+	stream      *gortsplib.ServerStream
+	processFunc []func([]byte)
+}
+
+func (p *RTSPublisher) setTracks(tracks gortsplib.Tracks) {
+	if p.processFunc != nil {
+		p.processFunc = p.processFunc[:len(tracks)]
+		return
+	} else {
+		p.processFunc = make([]func([]byte), len(tracks))
+	}
+	for i, track := range tracks {
+		v, ok := track.Media.Attribute("rtpmap")
+		if !ok {
+			continue
+		}
+
+		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
+			}
+		}
+	}
+}

server.go

@@ -0,0 +1,264 @@
+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"
+)
+
+// 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 {
+	stream *gortsplib.ServerStream
+	engine.Subscriber
+	vt *engine.VideoTrack
+	at *engine.AudioTrack
+}
+
+// called after a connection is opened.
+func (sh *RTSPServer) OnConnOpen(ctx *gortsplib.ServerHandlerOnConnOpenCtx) {
+	Printf("rtsp 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)
+	}
+}
+
+// called after a session is opened.
+func (sh *RTSPServer) OnSessionOpen(ctx *gortsplib.ServerHandlerOnSessionOpenCtx) {
+	Printf("rtsp 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()
+		}
+	}
+}
+
+// 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
+				}
+			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,
+		}, stream, nil
+		// if stream, ok := s.ExtraProp.(*gortsplib.ServerStream); ok {
+		// 	return &base.Response{
+		// 		StatusCode: base.StatusOK,
+		// 	}, stream, nil
+		// }
+	}
+	return &base.Response{
+		StatusCode: base.StatusNotFound,
+	}, nil, 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",
+		},
+	}
+	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,
+		}, 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)
+		}
+	}
+}

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])),
+				},
+			},
+		},
+	}
+}

ui/dist/demo.html

@@ -1,19 +0,0 @@
-<meta charset="utf-8">
-<title>plugin-rtsp demo</title>
-<script src="https://unpkg.com/vue"></script>
-<script src="./plugin-rtsp.umd.js"></script>
-
-<link rel="stylesheet" href="./plugin-rtsp.css">
-
-
-<div id="app">
-  <demo></demo>
-</div>
-
-<script>
-new Vue({
-  components: {
-    demo: plugin-rtsp
-  }
-}).$mount('#app')
-</script>

ui/dist/plugin-rtsp.common.js

@@ -1,408 +0,0 @@
-module.exports =
-/******/ (function(modules) { // webpackBootstrap
-/******/ 	// The module cache
-/******/ 	var installedModules = {};
-/******/
-/******/ 	// The require function
-/******/ 	function __webpack_require__(moduleId) {
-/******/
-/******/ 		// Check if module is in cache
-/******/ 		if(installedModules[moduleId]) {
-/******/ 			return installedModules[moduleId].exports;
-/******/ 		}
-/******/ 		// Create a new module (and put it into the cache)
-/******/ 		var module = installedModules[moduleId] = {
-/******/ 			i: moduleId,
-/******/ 			l: false,
-/******/ 			exports: {}
-/******/ 		};
-/******/
-/******/ 		// Execute the module function
-/******/ 		modules[moduleId].call(module.exports, module, module.exports, __webpack_require__);
-/******/
-/******/ 		// Flag the module as loaded
-/******/ 		module.l = true;
-/******/
-/******/ 		// Return the exports of the module
-/******/ 		return module.exports;
-/******/ 	}
-/******/
-/******/
-/******/ 	// expose the modules object (__webpack_modules__)
-/******/ 	__webpack_require__.m = modules;
-/******/
-/******/ 	// expose the module cache
-/******/ 	__webpack_require__.c = installedModules;
-/******/
-/******/ 	// define getter function for harmony exports
-/******/ 	__webpack_require__.d = function(exports, name, getter) {
-/******/ 		if(!__webpack_require__.o(exports, name)) {
-/******/ 			Object.defineProperty(exports, name, { enumerable: true, get: getter });
-/******/ 		}
-/******/ 	};
-/******/
-/******/ 	// define __esModule on exports
-/******/ 	__webpack_require__.r = function(exports) {
-/******/ 		if(typeof Symbol !== 'undefined' && Symbol.toStringTag) {
-/******/ 			Object.defineProperty(exports, Symbol.toStringTag, { value: 'Module' });
-/******/ 		}
-/******/ 		Object.defineProperty(exports, '__esModule', { value: true });
-/******/ 	};
-/******/
-/******/ 	// create a fake namespace object
-/******/ 	// mode & 1: value is a module id, require it
-/******/ 	// mode & 2: merge all properties of value into the ns
-/******/ 	// mode & 4: return value when already ns object
-/******/ 	// mode & 8|1: behave like require
-/******/ 	__webpack_require__.t = function(value, mode) {
-/******/ 		if(mode & 1) value = __webpack_require__(value);
-/******/ 		if(mode & 8) return value;
-/******/ 		if((mode & 4) && typeof value === 'object' && value && value.__esModule) return value;
-/******/ 		var ns = Object.create(null);
-/******/ 		__webpack_require__.r(ns);
-/******/ 		Object.defineProperty(ns, 'default', { enumerable: true, value: value });
-/******/ 		if(mode & 2 && typeof value != 'string') for(var key in value) __webpack_require__.d(ns, key, function(key) { return value[key]; }.bind(null, key));
-/******/ 		return ns;
-/******/ 	};
-/******/
-/******/ 	// getDefaultExport function for compatibility with non-harmony modules
-/******/ 	__webpack_require__.n = function(module) {
-/******/ 		var getter = module && module.__esModule ?
-/******/ 			function getDefault() { return module['default']; } :
-/******/ 			function getModuleExports() { return module; };
-/******/ 		__webpack_require__.d(getter, 'a', getter);
-/******/ 		return getter;
-/******/ 	};
-/******/
-/******/ 	// Object.prototype.hasOwnProperty.call
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-  var i
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-var render = function () {var _vm=this;var _h=_vm.$createElement;var _c=_vm._self._c||_h;return _c('div',[_c('mu-data-table',{attrs:{"data":_vm.Streams,"columns":_vm.columns},scopedSlots:_vm._u([{key:"default",fn:function(ref){
-var item = ref.row;
-return [_c('td',[_vm._v(_vm._s(item.StreamInfo.StreamPath))]),_c('td',[_c('StartTime',{attrs:{"value":item.StreamInfo.StartTime}})],1),_c('td',[_c('Progress',{attrs:{"stroke-width":20,"percent":Math.ceil(item.BufferRate),"text-inside":""}})],1),_c('td',[_vm._v(_vm._s(item.SyncCount))]),_c('td',[_c('mu-button',{attrs:{"flat":""},on:{"click":function($event){return _vm.showHeader(item)}}},[_vm._v("头信息")])],1)]}}])}),_c('mu-dialog',{attrs:{"title":"拉流转发","width":"360","open":_vm.openPull},on:{"update:open":function($event){_vm.openPull=$event}}},[_c('mu-text-field',{attrs:{"label":"rtsp url","label-float":"","help-text":"Please enter URL of rtsp..."},model:{value:(_vm.remoteAddr),callback:function ($$v) {_vm.remoteAddr=$$v},expression:"remoteAddr"}}),_c('mu-text-field',{attrs:{"label":"streamPath","label-float":"","help-text":"Please enter streamPath to publish."},model:{value:(_vm.streamPath),callback:function ($$v) {_vm.streamPath=$$v},expression:"streamPath"}}),_c('mu-button',{attrs:{"slot":"actions","flat":"","color":"primary"},on:{"click":_vm.addPull},slot:"actions"},[_vm._v("确定")])],1)],1)}
-var staticRenderFns = []
-
-
-// CONCATENATED MODULE: ./src/App.vue?vue&type=template&id=0225c7f1&
-
-// CONCATENATED MODULE: ./node_modules/cache-loader/dist/cjs.js??ref--0-0!./node_modules/vue-loader/lib??vue-loader-options!./src/App.vue?vue&type=script&lang=js&
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-//
-
-let listES = null;
-/* harmony default export */ var Appvue_type_script_lang_js_ = ({
-    components: {
-        StartTime
-    },
-    data() {
-        return {
-            currentStream: null,
-            Streams: null,
-            remoteAddr: "",
-            streamPath: "",
-            openPull: false,
-            columns:["StreamPath","开始时间","缓冲","同步数","操作" ].map(title=>({title}))
-        };
-    },
-
-    methods: {
-        fetchlist() {
-            listES = new EventSource(this.apiHost + "/rtsp/list");
-            listES.onmessage = evt => {
-                if (!evt.data) return;
-                this.Streams = JSON.parse(evt.data) || [];
-                this.Streams.sort((a, b) =>
-                    a.StreamInfo.StreamPath > b.StreamInfo.StreamPath ? 1 : -1
-                );
-            };
-        },
-        showHeader(item) {
-            this.$Modal.info({
-                title: "RTSP Header",
-                width: "1000px",
-                scrollable: true,
-                content: item.Header
-            });
-        },
-        addPull() {
-            this.openPull = false;
-            this.ajax
-                .getJSON(this.apiHost + "/rtsp/pull", {
-                    target: this.remoteAddr,
-                    streamPath: this.streamPath
-                })
-                .then(x => {
-                    if (x.code == 0) {
-                        this.$toast.success("已启动拉流");
-                    } else {
-                        this.$toast.error(x.msg);
-                    }
-                });
-        }
-    },
-    mounted() {
-        this.fetchlist();
-        let _this = this;
-        this.$parent.titleOps = [
-            {
-                template:'<m-button @click="onClick">拉流转发</m-button>',
-                methods:{
-                    onClick(){
-                        _this.openPull = true;
-                    }
-                }
-            }
-        ]
-    },
-    destroyed() {
-        listES.close();
-    }
-});
-
-// CONCATENATED MODULE: ./src/App.vue?vue&type=script&lang=js&
- /* harmony default export */ var src_Appvue_type_script_lang_js_ = (Appvue_type_script_lang_js_); 
-// EXTERNAL MODULE: ./src/App.vue?vue&type=style&index=0&lang=css&
-var Appvue_type_style_index_0_lang_css_ = __webpack_require__("034f");
-
-// CONCATENATED MODULE: ./node_modules/vue-loader/lib/runtime/componentNormalizer.js
-/* globals __VUE_SSR_CONTEXT__ */
-
-// IMPORTANT: Do NOT use ES2015 features in this file (except for modules).
-// This module is a runtime utility for cleaner component module output and will
-// be included in the final webpack user bundle.
-
-function normalizeComponent (
-  scriptExports,
-  render,
-  staticRenderFns,
-  functionalTemplate,
-  injectStyles,
-  scopeId,
-  moduleIdentifier, /* server only */
-  shadowMode /* vue-cli only */
-) {
-  // Vue.extend constructor export interop
-  var options = typeof scriptExports === 'function'
-    ? scriptExports.options
-    : scriptExports
-
-  // render functions
-  if (render) {
-    options.render = render
-    options.staticRenderFns = staticRenderFns
-    options._compiled = true
-  }
-
-  // functional template
-  if (functionalTemplate) {
-    options.functional = true
-  }
-
-  // scopedId
-  if (scopeId) {
-    options._scopeId = 'data-v-' + scopeId
-  }
-
-  var hook
-  if (moduleIdentifier) { // server build
-    hook = function (context) {
-      // 2.3 injection
-      context =
-        context || // cached call
-        (this.$vnode && this.$vnode.ssrContext) || // stateful
-        (this.parent && this.parent.$vnode && this.parent.$vnode.ssrContext) // functional
-      // 2.2 with runInNewContext: true
-      if (!context && typeof __VUE_SSR_CONTEXT__ !== 'undefined') {
-        context = __VUE_SSR_CONTEXT__
-      }
-      // inject component styles
-      if (injectStyles) {
-        injectStyles.call(this, context)
-      }
-      // register component module identifier for async chunk inferrence
-      if (context && context._registeredComponents) {
-        context._registeredComponents.add(moduleIdentifier)
-      }
-    }
-    // used by ssr in case component is cached and beforeCreate
-    // never gets called
-    options._ssrRegister = hook
-  } else if (injectStyles) {
-    hook = shadowMode
-      ? function () { injectStyles.call(this, this.$root.$options.shadowRoot) }
-      : injectStyles
-  }
-
-  if (hook) {
-    if (options.functional) {
-      // for template-only hot-reload because in that case the render fn doesn't
-      // go through the normalizer
-      options._injectStyles = hook
-      // register for functional component in vue file
-      var originalRender = options.render
-      options.render = function renderWithStyleInjection (h, context) {
-        hook.call(context)
-        return originalRender(h, context)
-      }
-    } else {
-      // inject component registration as beforeCreate hook
-      var existing = options.beforeCreate
-      options.beforeCreate = existing
-        ? [].concat(existing, hook)
-        : [hook]
-    }
-  }
-
-  return {
-    exports: scriptExports,
-    options: options
-  }
-}
-
-// CONCATENATED MODULE: ./src/App.vue
-
-
-
-
-
-
-/* normalize component */
-
-var component = normalizeComponent(
-  src_Appvue_type_script_lang_js_,
-  render,
-  staticRenderFns,
-  false,
-  null,
-  null,
-  null
-  
-)
-
-/* harmony default export */ var App = (component.exports);
-// CONCATENATED MODULE: ./node_modules/@vue/cli-service/lib/commands/build/entry-lib.js
-
-
-/* harmony default export */ var entry_lib = __webpack_exports__["default"] = (App);
-
-
-
-/***/ })
-
-/******/ })["default"];
-});
-//# sourceMappingURL=plugin-rtsp.umd.js.map

ui/dist/plugin-rtsp.umd.min.js

@@ -1,2 +0,0 @@
-(function(t,e){"object"===typeof exports&&"object"===typeof module?module.exports=e():"function"===typeof define&&define.amd?define([],e):"object"===typeof exports?exports["plugin-rtsp"]=e():t["plugin-rtsp"]=e()})("undefined"!==typeof self?self:this,(function(){return function(t){var e={};function r(n){if(e[n])return e[n].exports;var o=e[n]={i:n,l:!1,exports:{}};return t[n].call(o.exports,o,o.exports,r),o.l=!0,o.exports}return r.m=t,r.c=e,r.d=function(t,e,n){r.o(t,e)||Object.defineProperty(t,e,{enumerable:!0,get:n})},r.r=function(t){"undefined"!==typeof Symbol&&Symbol.toStringTag&&Object.defineProperty(t,Symbol.toStringTag,{value:"Module"}),Object.defineProperty(t,"__esModule",{value:!0})},r.t=function(t,e){if(1&e&&(t=r(t)),8&e)return t;if(4&e&&"object"===typeof t&&t&&t.__esModule)return t;var n=Object.create(null);if(r.r(n),Object.defineProperty(n,"default",{enumerable:!0,value:t}),2&e&&"string"!=typeof t)for(var o in t)r.d(n,o,function(e){return t[e]}.bind(null,o));return n},r.n=function(t){var e=t&&t.__esModule?function(){return t["default"]}:function(){return t};return r.d(e,"a",e),e},r.o=function(t,e){return Object.prototype.hasOwnProperty.call(t,e)},r.p="",r(r.s="fb15")}({"034f":function(t,e,r){"use strict";var n=r("85ec"),o=r.n(n);o.a},"85ec":function(t,e,r){},f6fd:function(t,e){(function(t){var e="currentScript",r=t.getElementsByTagName("script");e in t||Object.defineProperty(t,e,{get:function(){try{throw new Error}catch(n){var t,e=(/.*at [^\(]*\((.*):.+:.+\)$/gi.exec(n.stack)||[!1])[1];for(t in r)if(r[t].src==e||"interactive"==r[t].readyState)return r[t];return null}}})})(document)},fb15:function(t,e,r){"use strict";var n;(r.r(e),"undefined"!==typeof window)&&(r("f6fd"),(n=window.document.currentScript)&&(n=n.src.match(/(.+\/)[^/]+\.js(\?.*)?$/))&&(r.p=n[1]));var o=function(){var t=this,e=t.$createElement,r=t._self._c||e;return r("div",[r("mu-data-table",{attrs:{data:t.Streams,columns:t.columns},scopedSlots:t._u([{key:"default",fn:function(e){var n=e.row;return[r("td",[t._v(t._s(n.StreamInfo.StreamPath))]),r("td",[r("StartTime",{attrs:{value:n.StreamInfo.StartTime}})],1),r("td",[r("Progress",{attrs:{"stroke-width":20,percent:Math.ceil(n.BufferRate),"text-inside":""}})],1),r("td",[t._v(t._s(n.SyncCount))]),r("td",[r("mu-button",{attrs:{flat:""},on:{click:function(e){return t.showHeader(n)}}},[t._v("头信息")])],1)]}}])}),r("mu-dialog",{attrs:{title:"拉流转发",width:"360",open:t.openPull},on:{"update:open":function(e){t.openPull=e}}},[r("mu-text-field",{attrs:{label:"rtsp url","label-float":"","help-text":"Please enter URL of rtsp..."},model:{value:t.remoteAddr,callback:function(e){t.remoteAddr=e},expression:"remoteAddr"}}),r("mu-text-field",{attrs:{label:"streamPath","label-float":"","help-text":"Please enter streamPath to publish."},model:{value:t.streamPath,callback:function(e){t.streamPath=e},expression:"streamPath"}}),r("mu-button",{attrs:{slot:"actions",flat:"",color:"primary"},on:{click:t.addPull},slot:"actions"},[t._v("确定")])],1)],1)},a=[];let s=null;var i={components:{StartTime:StartTime},data(){return{currentStream:null,Streams:null,remoteAddr:"",streamPath:"",openPull:!1,columns:["StreamPath","开始时间","缓冲","同步数","操作"].map(t=>({title:t}))}},methods:{fetchlist(){s=new EventSource(this.apiHost+"/rtsp/list"),s.onmessage=t=>{t.data&&(this.Streams=JSON.parse(t.data)||[],this.Streams.sort((t,e)=>t.StreamInfo.StreamPath>e.StreamInfo.StreamPath?1:-1))}},showHeader(t){this.$Modal.info({title:"RTSP Header",width:"1000px",scrollable:!0,content:t.Header})},addPull(){this.openPull=!1,this.ajax.getJSON(this.apiHost+"/rtsp/pull",{target:this.remoteAddr,streamPath:this.streamPath}).then(t=>{0==t.code?this.$toast.success("已启动拉流"):this.$toast.error(t.msg)})}},mounted(){this.fetchlist();let t=this;this.$parent.titleOps=[{template:'<m-button @click="onClick">拉流转发</m-button>',methods:{onClick(){t.openPull=!0}}}]},destroyed(){s.close()}},l=i;r("034f");function u(t,e,r,n,o,a,s,i){var l,u="function"===typeof t?t.options:t;if(e&&(u.render=e,u.staticRenderFns=r,u._compiled=!0),n&&(u.functional=!0),a&&(u._scopeId="data-v-"+a),s?(l=function(t){t=t||this.$vnode&&this.$vnode.ssrContext||this.parent&&this.parent.$vnode&&this.parent.$vnode.ssrContext,t||"undefined"===typeof __VUE_SSR_CONTEXT__||(t=__VUE_SSR_CONTEXT__),o&&o.call(this,t),t&&t._registeredComponents&&t._registeredComponents.add(s)},u._ssrRegister=l):o&&(l=i?function(){o.call(this,this.$root.$options.shadowRoot)}:o),l)if(u.functional){u._injectStyles=l;var c=u.render;u.render=function(t,e){return l.call(e),c(t,e)}}else{var d=u.beforeCreate;u.beforeCreate=d?[].concat(d,l):[l]}return{exports:t,options:u}}var c=u(l,o,a,!1,null,null,null),d=c.exports;e["default"]=d}})["default"]}));
-//# sourceMappingURL=plugin-rtsp.umd.min.js.map

ui/package.json

@@ -1,15 +0,0 @@
-{
-  "name": "dashboard",
-  "version": "1.0.0",
-  "description": "dashboard of rtsp plugin for monibuca",
-  "main": "index.js",
-  "scripts": {
-    "build": "vue-cli-service build --target lib --name plugin-rtsp"
-  },
-  "author": "dexter",
-  "license": "ISC",
-  "devDependencies": {
-    "@vue/cli-service": "^4.2.3",
-    "vue-template-compiler": "^2.6.11"
-  }
-}

ui/src/App.vue

@@ -1,118 +0,0 @@
-<template>
-    <div>
-        <mu-data-table :data="Streams" :columns="columns">
-            <template #default="{row:item}">
-                <td>{{item.StreamInfo.StreamPath}}</td>
-                <td><StartTime :value="item.StreamInfo.StartTime"></StartTime></td>
-                <td><Progress :stroke-width="20" :percent="Math.ceil(item.BufferRate)" text-inside /></td>
-                <td>{{item.SyncCount}}</td>
-                <td><mu-button flat @click="showHeader(item)">头信息</mu-button></td>
-            </template>
-        </mu-data-table>
-        <mu-dialog title="拉流转发" width="360" :open.sync="openPull">
-            <mu-text-field v-model="remoteAddr" label="rtsp url" label-float help-text="Please enter URL of rtsp...">
-            </mu-text-field>
-            <mu-text-field v-model="streamPath" label="streamPath" label-float
-                help-text="Please enter streamPath to publish."></mu-text-field>
-            <mu-button slot="actions" flat color="primary" @click="addPull">确定</mu-button>
-        </mu-dialog>
-    </div>
-</template>
-
-<script>
-let listES = null;
-export default {
-    components: {
-        StartTime
-    },
-    data() {
-        return {
-            currentStream: null,
-            Streams: null,
-            remoteAddr: "",
-            streamPath: "",
-            openPull: false,
-            columns:["StreamPath","开始时间","缓冲","同步数","操作" ].map(title=>({title}))
-        };
-    },
-
-    methods: {
-        fetchlist() {
-            listES = new EventSource(this.apiHost + "/rtsp/list");
-            listES.onmessage = evt => {
-                if (!evt.data) return;
-                this.Streams = JSON.parse(evt.data) || [];
-                this.Streams.sort((a, b) =>
-                    a.StreamInfo.StreamPath > b.StreamInfo.StreamPath ? 1 : -1
-                );
-            };
-        },
-        showHeader(item) {
-            this.$Modal.info({
-                title: "RTSP Header",
-                width: "1000px",
-                scrollable: true,
-                content: item.Header
-            });
-        },
-        addPull() {
-            this.openPull = false;
-            this.ajax
-                .getJSON(this.apiHost + "/rtsp/pull", {
-                    target: this.remoteAddr,
-                    streamPath: this.streamPath
-                })
-                .then(x => {
-                    if (x.code == 0) {
-                        this.$toast.success("已启动拉流");
-                    } else {
-                        this.$toast.error(x.msg);
-                    }
-                });
-        }
-    },
-    mounted() {
-        this.fetchlist();
-        let _this = this;
-        this.$parent.titleOps = [
-            {
-                template:'<m-button @click="onClick">拉流转发</m-button>',
-                methods:{
-                    onClick(){
-                        _this.openPull = true;
-                    }
-                }
-            }
-        ]
-    },
-    destroyed() {
-        listES.close();
-    }
-};
-</script>
-
-<style>
-.empty {
-    color: #eb5e46;
-    width: 100%;
-    min-height: 500px;
-    display: flex;
-    justify-content: center;
-    align-items: center;
-}
-
-.layout {
-    padding-bottom: 30px;
-    display: flex;
-    flex-wrap: wrap;
-}
-.ts-info {
-    width: 300px;
-}
-
-.hls-info {
-    width: 350px;
-    display: flex;
-    flex-direction: column;
-}
-</style>