Go语言怎么开发浏览器视频流rtsp转webrtc播放

本篇内容主要讲解"Go语言怎么开发浏览器视频流rtsp转webrtc播放"，感兴趣的朋友不妨来看看。本文介绍的方法操作简单快捷，实用性强。下面就让小编来带大家学习"Go语言怎么开发浏览器视频流rtsp转webrtc播放"吧!

1. 前言

前面我们测试了rtsp转hls方式，发现延迟比较大，不太适合我们的使用需求。接下来我们试一下webrtc的方式看下使用情况。

综合考虑下来，我们最好能找到一个go作为后端，前端兼容性较好的前后端方案来处理webrtc，这样我们就可以结合我们之前的cgo+onvif+gSoap实现方案来获取rtsp流，并且可以根据已经实现的ptz、预置点等功能接口做更多的扩展。

2. rtsp转webRTC

如下是找到的一个比较合适的开源方案，前端使用了jQuery、bootstrap等，后端使用go+gin来实现并将rtsp流解析转换为webRTC协议提供http相关接口给到前端，通过config.json配置rtsp地址和stun地址：

点击下载

此外，还带有stun，可以自行配置stun地址，便于进行内网穿透。

初步测试几乎看不出来延迟，符合预期，使用的jQuery+bootstrap+go+gin做的web，也符合我们的项目使用情况。

3. 初步测试结果

结果如下：

4. 结合我们之前的onvif+gSoap+cgo的方案做修改

我们在此项目的基础上，结合我们之前研究的onvif+cgo+gSoap的方案，将onvif获取到的相关数据提供接口到web端，增加ptz、调焦、缩放等功能。

我们在http.go中添加新的post接口：HTTPAPIServerStreamPtz来进行ptz和HTTPAPIServerStreamPreset进行预置点相关操作。

以下是部分代码，没有做太多的优化，也仅仅实现了ptz、调焦和缩放，算是打通了通路，具体项目需要可以再做优化。

4.1 go后端修改

增加了新的接口，并将之前cgo+onvif+gSoap的内容结合了进来，项目整体没有做更多的优化，只是为了演示，提供一个思路：

http.go（增加了两个post接口ptz和preset，采用cgo方式处理）：

package main/*#cgo CFLAGS: -I ./ -I /usr/local/#cgo LDFLAGS: -L ./build -lc_onvif_static -lpthread -ldl -lssl -lcrypto#include "client.h"#include "malloc.h"*/import "C"import (    "encoding/json"    "fmt"    "log"    "net/http"    "os"    "sort"    "strconv"    "time"    "unsafe"    "github.com/deepch/vdk/av"    webrtc "github.com/deepch/vdk/format/webrtcv3"    "github.com/gin-gonic/gin")type JCodec struct {    Type string}func serveHTTP() {    gin.SetMode(gin.ReleaseMode)    router := gin.Default()    router.Use(CORSMiddleware())    if _, err := os.Stat("./web"); !os.IsNotExist(err) {        router.LoadHTMLGlob("web/templates/*")        router.GET("/", HTTPAPIServerIndex)        router.GET("/stream/player/:uuid", HTTPAPIServerStreamPlayer)    }    router.POST("/stream/receiver/:uuid", HTTPAPIServerStreamWebRTC)    //增加新的post接口    router.POST("/stream/ptz/", HTTPAPIServerStreamPtz)    router.POST("/stream/preset/", HTTPAPIServerStreamPreset)    router.GET("/stream/codec/:uuid", HTTPAPIServerStreamCodec)    router.POST("/stream", HTTPAPIServerStreamWebRTC2)    router.StaticFS("/static", http.Dir("web/static"))    err := router.Run(Config.Server.HTTPPort)    if err != nil {        log.Fatalln("Start HTTP Server error", err)    }}//HTTPAPIServerIndex  indexfunc HTTPAPIServerIndex(c *gin.Context) {    _, all := Config.list()    if len(all) > 0 {        c.Header("Cache-Control", "no-cache, max-age=0, must-revalidate, no-store")        c.Header("Access-Control-Allow-Origin", "*")        c.Redirect(http.StatusMovedPermanently, "stream/player/"+all[0])    } else {        c.HTML(http.StatusOK, "index.tmpl", gin.H{            "port":    Config.Server.HTTPPort,            "version": time.Now().String(),        })    }}//HTTPAPIServerStreamPlayer stream playerfunc HTTPAPIServerStreamPlayer(c *gin.Context) {    _, all := Config.list()    sort.Strings(all)    c.HTML(http.StatusOK, "player.tmpl", gin.H{        "port":     Config.Server.HTTPPort,        "suuid":    c.Param("uuid"),        "suuidMap": all,        "version":  time.Now().String(),    })}//HTTPAPIServerStreamCodec stream codecfunc HTTPAPIServerStreamCodec(c *gin.Context) {    if Config.ext(c.Param("uuid")) {        Config.RunIFNotRun(c.Param("uuid"))        codecs := Config.coGe(c.Param("uuid"))        if codecs == nil {            return        }        var tmpCodec []JCodec        for _, codec := range codecs {            if codec.Type() != av.H264 && codec.Type() != av.PCM_ALAW && codec.Type() != av.PCM_MULAW && codec.Type() != av.OPUS {                log.Println("Codec Not Supported WebRTC ignore this track", codec.Type())                continue            }            if codec.Type().IsVideo() {                tmpCodec = append(tmpCodec, JCodec{Type: "video"})            } else {                tmpCodec = append(tmpCodec, JCodec{Type: "audio"})            }        }        b, err := json.Marshal(tmpCodec)        if err == nil {                        _, err = c.Writer.Write(b)                        if err != nil {                                log.Println("Write Codec Info error", err)                                return                        }                }        }}//HTTPAPIServerStreamWebRTC stream video over WebRTCfunc HTTPAPIServerStreamWebRTC(c *gin.Context) {        if !Config.ext(c.PostForm("suuid")) {                log.Println("Stream Not Found")                return        }        Config.RunIFNotRun(c.PostForm("suuid"))        codecs := Config.coGe(c.PostForm("suuid"))        if codecs == nil {                log.Println("Stream Codec Not Found")                return        }        var AudioOnly bool        if len(codecs) == 1 && codecs[0].Type().IsAudio() {                AudioOnly = true        }        muxerWebRTC := webrtc.NewMuxer(webrtc.Options{ICEServers: Config.GetICEServers(), ICEUsername: Config.GetICEUsername(), ICECredential: Config.GetICECredential(), PortMin: Config.GetWebRTCPortMin(), PortMax: Config.GetWebRTCPortMax()})        answer, err := muxerWebRTC.WriteHeader(codecs, c.PostForm("data"))        if err != nil {                log.Println("WriteHeader", err)                return        }        _, err = c.Writer.Write([]byte(answer))        if err != nil {                log.Println("Write", err)                return        }        go func() {                cid, ch := Config.clAd(c.PostForm("suuid"))                defer Config.clDe(c.PostForm("suuid"), cid)                defer muxerWebRTC.Close()                var videoStart bool                noVideo := time.NewTimer(10 * time.Second)                for {                        select {                        case <-noVideo.C:                                log.Println("noVideo")                                return                        case pck := <-ch:                                if pck.IsKeyFrame || AudioOnly {                                        noVideo.Reset(10 * time.Second)                                        videoStart = true                                }                                if !videoStart && !AudioOnly {                                        continue                                }                                err = muxerWebRTC.WritePacket(pck)                                if err != nil {                                        log.Println("WritePacket", err)                                        return                                }                        }                }        }()}func HTTPAPIServerStreamPtz(c *gin.Context) {        action := c.PostForm("action")        direction, err := strconv.Atoi(action)        if err != nil {                log.Println(err)                return        }        var soap C.P_Soap        soap = C.new_soap(soap)        username := C.CString("admin")        password := C.CString("admin")        serviceAddr := C.CString("http://40.40.40.101:80/onvif/device_service")        C.get_device_info(soap, username, password, serviceAddr)        mediaAddr := [200]C.char{}        C.get_capabilities(soap, username, password, serviceAddr, &mediaAddr[0])        profileToken := [200]C.char{}        C.get_profiles(soap, username, password, &profileToken[0], &mediaAddr[0])        videoSourceToken := [200]C.char{}        C.get_video_source(soap, username, password, &videoSourceToken[0], &mediaAddr[0])        switch direction {        case 0:                break        case 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11:                C.ptz(soap, username, password, C.int(direction), C.float(0.5), &profileToken[0], &mediaAddr[0])        case 12, 13, 14:                C.focus(soap, username, password, C.int(direction), C.float(0.5), &videoSourceToken[0], &mediaAddr[0])        default:                fmt.Println("Unknown direction.")        }        C.del_soap(soap)        C.free(unsafe.Pointer(username))        C.free(unsafe.Pointer(password))        C.free(unsafe.Pointer(serviceAddr))        c.JSON(http.StatusOK, gin.H{"message":"success"})}func HTTPAPIServerStreamPreset(c *gin.Context) {        var soap C.P_Soap        soap = C.new_soap(soap)        username := C.CString("admin")        password := C.CString("admin")        serviceAddr := C.CString("http://40.40.40.101:80/onvif/device_service")        C.get_device_info(soap, username, password, serviceAddr)        mediaAddr := [200]C.char{}        C.get_capabilities(soap, username, password, serviceAddr, &mediaAddr[0])        profileToken := [200]C.char{}        C.get_profiles(soap, username, password, &profileToken[0], &mediaAddr[0])        videoSourceToken := [200]C.char{}        C.get_video_source(soap, username, password, &videoSourceToken[0], &mediaAddr[0])        action := c.PostForm("action")        presetAction, err := strconv.Atoi(action)        if err != nil {                log.Println(err)                return        }        fmt.Println("请输入数字进行preset,1-4分别代表查询、设置、跳转、删除预置点；退出输入0:")        switch presetAction {        case 0:                break        case 1:                C.preset(soap, username, password, C.int(presetAction), nil, nil, &profileToken[0], &mediaAddr[0])        case 2:                fmt.Println("请输入要设置的预置点token信息：")                presentToken := ""                _, _ = fmt.Scanln(&presentToken)                fmt.Println("请输入要设置的预置点name信息长度不超过200：")                presentName := ""                _, _ = fmt.Scanln(&presentName)                C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), C.CString(presentName), &profileToken[0], &mediaAddr[0])        case 3:                fmt.Println("请输入要跳转的预置点token信息：")                presentToken := ""                _, _ = fmt.Scanln(&presentToken)                C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), nil, &profileToken[0], &mediaAddr[0])        case 4:                fmt.Println("请输入要删除的预置点token信息：")                presentToken := ""                _, _ = fmt.Scanln(&presentToken)                C.preset(soap, username, password, C.int(presetAction), C.CString(presentToken), nil, &profileToken[0], &mediaAddr[0])        default:                fmt.Println("unknown present action.")                break        }        C.del_soap(soap)        C.free(unsafe.Pointer(username))        C.free(unsafe.Pointer(password))        C.free(unsafe.Pointer(serviceAddr))                c.JSON(http.StatusOK, gin.H{"message":"success"})}func CORSMiddleware() gin.HandlerFunc {        return func(c *gin.Context) {                c.Header("Access-Control-Allow-Origin", "*")                c.Header("Access-Control-Allow-Credentials", "true")                c.Header("Access-Control-Allow-Headers", "Origin, X-Requested-With, Content-Type, Accept, Authorization, x-access-token")                c.Header("Access-Control-Expose-Headers", "Content-Length, Access-Control-Allow-Origin, Access-Control-Allow-Headers, Cache-Control, Content-Language, Content-Type")                c.Header("Access-Control-Allow-Methods", "POST, OPTIONS, GET, PUT")                if c.Request.Method == "OPTIONS" {                        c.AbortWithStatus(http.StatusNoContent)                        return                }                c.Next()        }}type Response struct {        Tracks []string `json:"tracks"`        Sdp64  string   `json:"sdp64"`}type ResponseError struct {        Error string `json:"error"`}func HTTPAPIServerStreamWebRTC2(c *gin.Context) {        url := c.PostForm("url")        if _, ok := Config.Streams[url]; !ok {                Config.Streams[url] = StreamST{                        URL:      url,                        OnDemand: true,                        Cl:       make(map[string]viewer),                }        }        Config.RunIFNotRun(url)        codecs := Config.coGe(url)        if codecs == nil {                log.Println("Stream Codec Not Found")                c.JSON(500, ResponseError{Error: Config.LastError.Error()})                return        }        muxerWebRTC := webrtc.NewMuxer(                webrtc.Options{                        ICEServers: Config.GetICEServers(),                        PortMin:    Config.GetWebRTCPortMin(),                        PortMax:    Config.GetWebRTCPortMax(),                },        )        sdp64 := c.PostForm("sdp64")        answer, err := muxerWebRTC.WriteHeader(codecs, sdp64)        if err != nil {                log.Println("Muxer WriteHeader", err)                c.JSON(500, ResponseError{Error: err.Error()})                return        }        response := Response{                Sdp64: answer,        }        for _, codec := range codecs {                if codec.Type() != av.H264 &&                        codec.Type() != av.PCM_ALAW &&                        codec.Type() != av.PCM_MULAW &&                        codec.Type() != av.OPUS {                        log.Println("Codec Not Supported WebRTC ignore this track", codec.Type())                        continue                }                if codec.Type().IsVideo() {                        response.Tracks = append(response.Tracks, "video")                } else {                        response.Tracks = append(response.Tracks, "audio")                }        }        c.JSON(200, response)        AudioOnly := len(codecs) == 1 && codecs[0].Type().IsAudio()        go func() {                cid, ch := Config.clAd(url)                defer Config.clDe(url, cid)                defer muxerWebRTC.Close()                var videoStart bool                noVideo := time.NewTimer(10 * time.Second)                for {                        select {                        case <-noVideo.C:                                log.Println("noVideo")                                return                        case pck := <-ch:                                if pck.IsKeyFrame || AudioOnly {                                        noVideo.Reset(10 * time.Second)                                        videoStart = true                                }                                if !videoStart && !AudioOnly {                                        continue                                }                                err = muxerWebRTC.WritePacket(pck)                                if err != nil {                                        log.Println("WritePacket", err)                                        return                                }                        }                }        }()}

4.2 前端修改

对于goland我们首先将.tmpl文件通过右键标记为html格式，然后再修改时就会有前端语法支持和补全支持，便于修改，否则默认是识别为文本的，之后我们修改player.tmpl和app.js，在player.tmpl中添加一些ptz的按钮并通过js与前后端进行数据交互：

player.tmpl:

    Play Stream {{ .suuid }}
    
        
            
                {{ range .suuidMap }}                {{ . }}                {{ end }}                
                
                    
                    
                    
                    
                    
                    
                    
                    
                    
                    
                
                
                    缩放+                    缩放-                
                
                    调焦+                    调焦-                    停止调焦                
            
        
        
                                                            
                            
            
        
    

app.js：

let stream = new MediaStream();let suuid = $('#suuid').val();let config = {  iceServers: [{    urls: ["stun:stun.l.google.com:19302"]  }]};const pc = new RTCPeerConnection(config);pc.onnegotiationneeded = handleNegotiationNeededEvent;let log = msg => {  document.getElementById('div')[xss_clean] += msg + '
'}pc.ontrack = function(event) {  stream.addTrack(event.track);  videoElem.srcObject = stream;  log(event.streams.length + ' track is delivered')}pc.oniceconnectionstatechange = e => log(pc.iceConnectionState)async function handleNegotiationNeededEvent() {  let offer = await pc.createOffer();  await pc.setLocalDescription(offer);  getRemoteSdp();}$(document).ready(function() {  $('#' + suuid).addClass('active');  getCodecInfo();});function getCodecInfo() {  $.get("../codec/" + suuid, function(data) {    try {      data = JSON.parse(data);    } catch (e) {      console.log(e);    } finally {      $.each(data,function(index,value){        pc.addTransceiver(value.Type, {          'direction': 'sendrecv'        })      })    }  });}let sendChannel = null;function getRemoteSdp() {  $.post("../receiver/"+ suuid, {    suuid: suuid,    data: btoa(pc.localDescription.sdp)  }, function(data) {    try {      pc.setRemoteDescription(new RTCSessionDescription({        type: 'answer',        sdp: atob(data)      }))    } catch (e) {      console.warn(e);    }  });}function ptz(direction) {  $.post("../ptz/", direction, function(data, status){    console.debug(_"Data: " + data + "nStatus: " + status);  });}function funTopClick() {  console.debug("top click");  ptz("action=1")}function funDownClick() {  console.debug("down click");  ptz("action=2")}function funLeftClick() {  console.debug("left click");  ptz("action=3")}function funRightClick() {  console.debug("right click");  ptz("action=4")}function funStopClick() {  console.debug("stop click");  ptz("action=9")}function funZoomClick(direction) {  console.debug("zoom click"+direction);  ptz("action="+direction)}function funFocusClick(direction) {  console.debug("focus click"+direction);  ptz("action="+direction)}

主要增加了一个扇形按钮和两组按钮组，然后将按钮的点击结合到app.js中进行处理，app.js中则发送post请求调用go后端接口。

4.3 项目结构和编译运行

项目结构如下，部分文件做了备份，实际可以不用：

$tree -a -I ".github|.idea|build".├── .gitignore├── CMakeLists.txt├── Dockerfile├── LICENSE├── README.md├── build.cmd├── client.c├── client.h├── config.go├── config.json├── config.json.bak├── doc│   ├── demo2.png│   ├── demo3.png│   └── demo4.png├── go.mod├── go.sum├── http.go├── main.go├── main.go.bak├── renovate.json├── soap│   ├── DeviceBinding.nsmap│   ├── ImagingBinding.nsmap│   ├── MediaBinding.nsmap│   ├── PTZBinding.nsmap│   ├── PullPointSubscriptionBinding.nsmap│   ├── RemoteDiscoveryBinding.nsmap│   ├── custom│   │   ├── README.txt│   │   ├── chrono_duration.cpp│   │   ├── chrono_duration.h│   │   ├── chrono_time_point.cpp│   │   ├── chrono_time_point.h│   │   ├── duration.c│   │   ├── duration.h│   │   ├── float128.c│   │   ├── float128.h│   │   ├── int128.c│   │   ├── int128.h│   │   ├── long_double.c│   │   ├── long_double.h│   │   ├── long_time.c│   │   ├── long_time.h│   │   ├── qbytearray_base64.cpp│   │   ├── qbytearray_base64.h│   │   ├── qbytearray_hex.cpp│   │   ├── qbytearray_hex.h│   │   ├── qdate.cpp│   │   ├── qdate.h│   │   ├── qdatetime.cpp│   │   ├── qdatetime.h│   │   ├── qstring.cpp│   │   ├── qstring.h│   │   ├── qtime.cpp│   │   ├── qtime.h│   │   ├── struct_timeval.c│   │   ├── struct_timeval.h│   │   ├── struct_tm.c│   │   ├── struct_tm.h│   │   ├── struct_tm_date.c│   │   └── struct_tm_date.h│   ├── dom.c│   ├── dom.h│   ├── duration.c│   ├── duration.h│   ├── mecevp.c│   ├── mecevp.h│   ├── onvif.h│   ├── smdevp.c│   ├── smdevp.h│   ├── soapC.c│   ├── soapClient.c│   ├── soapH.h│   ├── soapStub.h│   ├── stdsoap2.h│   ├── stdsoap2_ssl.c│   ├── struct_timeval.c│   ├── struct_timeval.h│   ├── threads.c│   ├── threads.h│   ├── typemap.dat│   ├── wsaapi.c│   ├── wsaapi.h│   ├── wsdd.nsmap│   ├── wsseapi.c│   └── wsseapi.h├── stream.go└── web    ├── static    │   ├── css    │   │   ├── bootstrap-grid.css    │   │   ├── bootstrap-grid.css.map    │   │   ├── bootstrap-grid.min.css    │   │   ├── bootstrap-grid.min.css.map    │   │   ├── bootstrap-reboot.css    │   │   ├── bootstrap-reboot.css.map    │   │   ├── bootstrap-reboot.min.css    │   │   ├── bootstrap-reboot.min.css.map    │   │   ├── bootstrap.css    │   │   ├── bootstrap.css.map    │   │   ├── bootstrap.min.css    │   │   ├── bootstrap.min.css.map    │   │   └── shanxing.css    │   └── js    │       ├── adapter-latest.js    │       ├── app.js    │       ├── bootstrap.bundle.js    │       ├── bootstrap.bundle.js.map    │       ├── bootstrap.bundle.min.js    │       ├── bootstrap.bundle.min.js.map    │       ├── bootstrap.js    │       ├── bootstrap.js.map    │       ├── bootstrap.min.js    │       ├── bootstrap.min.js.map    │       └── jquery-3.4.1.min.js    └── templates        ├── index.tmpl        └── player.tmpl8 directories, 111 files

关于cgo和onvif、gSoap部分这里就不多说了，不清楚的可以看前面的总结，gin、bootstramp、jQuery这些也需要一定的前后端概念学习和储备，在其它的分类总结中也零星分布了，不清楚的可以看一下，这里就不再多说了。

编译运行：

GOOS=linux GOARCH=amd64 CGO_ENABLE=1 GO111MODULE=on go run *.go

记得修改一下go.mod中对go版本的依赖，按照cgo的问题，目前至少需要1.18及以上，否则运行ptz可能出现分割违例问题，到我总结这里1.18已经发了正式版本了。

module github.com/deepch/RTSPtoWebRTCgo 1.18require (        github.com/deepch/vdk v0.0.0-20220309163430-c6529706436c        github.com/gin-gonic/gin v1.7.7)

4.4 结果展示

界面效果：

动态调试ptz：

动态调试缩放：

动态调试调焦：

到此，相信大家对"Go语言怎么开发浏览器视频流rtsp转webrtc播放"有了更深的了解，不妨来实际操作一番吧！这里是网站，更多相关内容可以进入相关频道进行查询，关注我们，继续学习！