mirror of
https://github.com/lkmio/lkm.git
synced 2025-09-27 03:26:01 +08:00
730 lines
23 KiB
Go
730 lines
23 KiB
Go
package main
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import (
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"encoding/base64"
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"encoding/json"
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"fmt"
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"github.com/gorilla/mux"
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"github.com/gorilla/websocket"
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"github.com/lkmio/avformat/utils"
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"github.com/lkmio/lkm/flv"
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"github.com/lkmio/lkm/gb28181"
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"github.com/lkmio/lkm/hls"
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"github.com/lkmio/lkm/log"
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"github.com/lkmio/lkm/rtc"
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"github.com/lkmio/lkm/stream"
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"io"
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"net"
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"net/http"
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"os"
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"runtime"
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"strconv"
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"strings"
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"time"
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)
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type ApiServer struct {
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upgrader *websocket.Upgrader
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router *mux.Router
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}
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var apiServer *ApiServer
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func init() {
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apiServer = &ApiServer{
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upgrader: &websocket.Upgrader{
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CheckOrigin: func(r *http.Request) bool {
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return true
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},
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},
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router: mux.NewRouter(),
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}
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}
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func filterSourceID(f func(sourceId string, w http.ResponseWriter, req *http.Request), suffix string) func(http.ResponseWriter, *http.Request) {
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return func(w http.ResponseWriter, req *http.Request) {
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source, err := stream.Path2SourceID(req.URL.Path, suffix)
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if err != nil {
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log.Sugar.Errorf("拉流失败 解析流id发生err: %s path: %s", err.Error(), req.URL.Path)
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httpResponse(w, http.StatusBadRequest, err.Error())
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return
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}
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f(source, w, req)
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}
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}
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type IDS struct {
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// 内部SinkID可能是uint64或者string类型, 但外部传参均使用string类型，程序内部自行兼容ipv6.
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Sink string `json:"sink"`
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Source string `json:"source"`
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}
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func withJsonParams[T any](f func(params T, w http.ResponseWriter, req *http.Request), params interface{}) func(http.ResponseWriter, *http.Request) {
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return func(w http.ResponseWriter, req *http.Request) {
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newParams := new(T)
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if err := HttpDecodeJSONBody(w, req, newParams); err != nil {
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log.Sugar.Errorf("处理http请求失败 err: %s path: %s", err.Error(), req.URL.Path)
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httpResponseError(w, err.Error())
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return
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}
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f(*newParams, w, req)
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}
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}
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func startApiServer(addr string) {
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/**
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http://host:port/xxx.flv
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http://host:port/xxx.rtc
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http://host:port/xxx.m3u8
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http://host:port/xxx_0.ts
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ws://host:port/xxx.flv
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*/
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apiServer.router.Use(func(handler http.Handler) http.Handler {
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return http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
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// 添加 CORS 头以解决跨域问题
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w.Header().Set("Access-Control-Allow-Origin", "*")
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w.Header().Set("Access-Control-Allow-Methods", "GET, POST, OPTIONS, PUT, DELETE")
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w.Header().Set("Access-Control-Allow-Headers", "*")
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// 如果是OPTIONS请求，直接返回
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if r.Method == "OPTIONS" {
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w.WriteHeader(http.StatusOK)
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return
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}
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handler.ServeHTTP(w, r)
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})
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})
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// 点播, 映射录制资源
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// 放在最前面, 避免被后面的路由拦截
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apiServer.router.PathPrefix("/record/").Handler(http.StripPrefix("/record/", http.FileServer(http.Dir(stream.AppConfig.Record.Dir))))
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// {source}.flv和/{source}/{stream}.flv意味着, 推流id(路径)只能嵌套一层
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apiServer.router.HandleFunc("/{source}.flv", filterSourceID(apiServer.onFlv, ".flv"))
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apiServer.router.HandleFunc("/{source}/{stream}.flv", filterSourceID(apiServer.onFlv, ".flv"))
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if stream.AppConfig.Hls.Enable {
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apiServer.router.HandleFunc("/{source}.m3u8", filterSourceID(apiServer.onHLS, ".m3u8"))
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apiServer.router.HandleFunc("/{source}/{stream}.m3u8", filterSourceID(apiServer.onHLS, ".m3u8"))
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apiServer.router.HandleFunc("/{source}.ts", filterSourceID(apiServer.onTS, ".ts"))
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apiServer.router.HandleFunc("/{source}/{stream}.ts", filterSourceID(apiServer.onTS, ".ts"))
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}
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if stream.AppConfig.WebRtc.Enable {
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apiServer.router.HandleFunc("/{source}.rtc", filterSourceID(apiServer.onRtc, ".rtc"))
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apiServer.router.HandleFunc("/{source}/{stream}.rtc", filterSourceID(apiServer.onRtc, ".rtc"))
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}
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apiServer.router.HandleFunc("/api/v1/source/list", apiServer.OnSourceList) // 查询所有推流源
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apiServer.router.HandleFunc("/api/v1/source/close", withJsonParams(apiServer.OnSourceClose, &IDS{})) // 关闭推流源
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apiServer.router.HandleFunc("/api/v1/sink/list", withJsonParams(apiServer.OnSinkList, &IDS{})) // 查询某个推流源下，所有的拉流端列表
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apiServer.router.HandleFunc("/api/v1/sink/close", withJsonParams(apiServer.OnSinkClose, &IDS{})) // 关闭拉流端
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apiServer.router.HandleFunc("/api/v1/sink/add", withJsonParams(apiServer.OnSinkAdd, &GBOffer{})) // 级联/广播/JT转GB
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apiServer.router.HandleFunc("/api/v1/record/start", apiServer.OnRecordStart) // 开启录制
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apiServer.router.HandleFunc("/api/v1/record/stop", apiServer.OnRecordStop) // 关闭录制
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apiServer.router.HandleFunc("/api/v1/streams/statistics", nil) // 统计所有推拉流
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if stream.AppConfig.GB28181.Enable {
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apiServer.router.HandleFunc("/ws/v1/gb28181/talk", apiServer.OnGBTalk) // 对讲的主讲人WebSocket连接
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apiServer.router.HandleFunc("/api/v1/control/ws-talk/{device}/{channel}", apiServer.OnGBTalk) // 对讲的主讲人WebSocket连接
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apiServer.router.HandleFunc("/api/v1/gb28181/source/create", withJsonParams(apiServer.OnGBOfferCreate, &SourceSDP{}))
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apiServer.router.HandleFunc("/api/v1/gb28181/answer/set", withJsonParams(apiServer.OnGBSourceConnect, &SourceSDP{})) // 应答的sdp, 如果是active模式拉流, 设置对方的地址. 下载文件设置文件大小
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}
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apiServer.router.HandleFunc("/api/v1/gc/force", func(writer http.ResponseWriter, request *http.Request) {
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runtime.GC()
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})
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apiServer.router.HandleFunc("/api/v1/stream/info", apiServer.OnStreamInfo)
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apiServer.router.PathPrefix("/web/").Handler(http.StripPrefix("/web/", http.FileServer(http.Dir("./web"))))
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http.Handle("/", apiServer.router)
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srv := &http.Server{
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Handler: apiServer.router,
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Addr: addr,
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// Good practice: enforce timeouts for servers you create!
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WriteTimeout: 30 * time.Second,
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ReadTimeout: 30 * time.Second,
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}
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err := srv.ListenAndServe()
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if err != nil {
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panic(err)
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}
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}
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func (api *ApiServer) generateSinkID(_ string) stream.SinkID {
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return utils.RandStringBytes(18)
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}
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func (api *ApiServer) onFlv(sourceId string, w http.ResponseWriter, r *http.Request) {
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// 区分ws请求
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ws := true
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if !("upgrade" == strings.ToLower(r.Header.Get("Connection"))) {
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ws = false
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} else if !("websocket" == strings.ToLower(r.Header.Get("Upgrade"))) {
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ws = false
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} else if !("13" == r.Header.Get("Sec-Websocket-Version")) {
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ws = false
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}
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if ws {
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apiServer.onWSFlv(sourceId, w, r)
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} else {
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apiServer.onHttpFLV(sourceId, w, r)
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}
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}
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func (api *ApiServer) onWSFlv(sourceId string, w http.ResponseWriter, r *http.Request) {
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conn, err := api.upgrader.Upgrade(w, r, nil)
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if err != nil {
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log.Sugar.Errorf("ws拉流失败 source: %s err: %s", sourceId, err.Error())
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w.WriteHeader(http.StatusBadRequest)
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return
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}
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sink := flv.NewFLVSink(api.generateSinkID(r.RemoteAddr), sourceId, flv.NewWSConn(conn))
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ok := stream.SubscribeStream(sink, r.URL.Query())
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if utils.HookStateOK != ok {
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log.Sugar.Warnf("ws-flv 拉流失败 source: %s sink: %s", sourceId, sink.String())
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_ = conn.Close()
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} else {
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log.Sugar.Infof("ws-flv 拉流成功 source: %s sink: %s", sourceId, sink.String())
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}
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netConn := conn.NetConn()
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bytes := make([]byte, 64)
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for {
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if _, err := netConn.Read(bytes); err != nil {
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log.Sugar.Infof("ws-flv 断开连接 source: %s sink:%s", sourceId, sink.String())
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sink.Close()
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break
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}
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}
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}
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func (api *ApiServer) onHttpFLV(sourceId string, w http.ResponseWriter, r *http.Request) {
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w.Header().Set("Content-Type", "video/x-flv")
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w.Header().Set("Connection", "Keep-Alive")
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w.Header().Set("Transfer-Encoding", "chunked")
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var conn net.Conn
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if hj, ok := w.(http.Hijacker); !ok {
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log.Sugar.Errorf("http-flv 拉流失败 不支持hijacking. source: %s remote: %s", sourceId, r.RemoteAddr)
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http.Error(w, "webserver doesn't support hijacking", http.StatusInternalServerError)
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return
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} else {
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w.WriteHeader(http.StatusOK)
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var err error
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if conn, _, err = hj.Hijack(); err != nil {
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log.Sugar.Errorf("http-flv 拉流失败 source: %s remote: %s err: %s", sourceId, r.RemoteAddr, err.Error())
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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}
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sink := flv.NewFLVSink(api.generateSinkID(r.RemoteAddr), sourceId, conn)
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ok := stream.SubscribeStream(sink, r.URL.Query())
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if utils.HookStateOK != ok {
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log.Sugar.Warnf("http-flv 拉流失败 source: %s sink: %s", sourceId, sink.String())
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sink.Close()
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} else {
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log.Sugar.Infof("http-flv 拉流成功 source: %s sink: %s", sourceId, sink.String())
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}
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bytes := make([]byte, 64)
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for {
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if _, err := conn.Read(bytes); err != nil {
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log.Sugar.Infof("http-flv 断开连接 sink:%s", sink.String())
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sink.Close()
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break
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}
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}
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}
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func (api *ApiServer) onTS(source string, w http.ResponseWriter, r *http.Request) {
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sid := r.URL.Query().Get(hls.SessionIDKey)
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var sink stream.Sink
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if sid != "" {
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sink = hls.SinkManager.Find(stream.SinkID(sid))
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}
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if sink == nil {
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log.Sugar.Errorf("hls session with id '%s' has expired.", sid)
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w.WriteHeader(http.StatusForbidden)
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return
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}
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index := strings.LastIndex(source, "_")
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if index < 0 || index == len(source)-1 {
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w.WriteHeader(http.StatusBadRequest)
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return
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}
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seq := source[index+1:]
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tsPath := stream.AppConfig.Hls.TSPath(sink.GetSourceID(), seq)
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if _, err := os.Stat(tsPath); err != nil {
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w.WriteHeader(http.StatusNotFound)
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return
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}
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sink.(*hls.M3U8Sink).RefreshPlayingTime()
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w.Header().Set("Content-Type", "video/MP2T")
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http.ServeFile(w, r, tsPath)
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}
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func (api *ApiServer) onHLS(source string, w http.ResponseWriter, r *http.Request) {
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w.Header().Set("Content-Type", "application/vnd.apple.mpegurl")
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// 如果没有携带会话ID, 认为是首次拉流. Server将生成会话ID, 应答给拉流端, 后续拉流请求(.M3U8和.TS的HTTP请求)都将携带该会话ID.
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// 会话ID的Key为"hls_sid", 为避免冲突, 播放端和hook server不要再使用, 否则会一直拉流失败.
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sid := r.URL.Query().Get(hls.SessionIDKey)
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if sid == "" {
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sid = utils.RandStringBytes(10)
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query := r.URL.Query()
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query.Add(hls.SessionIDKey, sid)
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path := fmt.Sprintf("/%s.m3u8?%s", source, query.Encode())
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response := "#EXTM3U\r\n" +
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"#EXT-X-STREAM-INF:BANDWIDTH=1,AVERAGE-BANDWIDTH=1\r\n" +
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path + "\r\n"
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w.Write([]byte(response))
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return
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}
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sink := hls.SinkManager.Find(sid)
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if sink == nil {
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// 创建sink
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sink = hls.NewM3U8Sink(sid, source, sid)
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sink.(*hls.M3U8Sink).RefreshPlayingTime()
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if hls.SinkManager.Add(sink) {
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ok := stream.SubscribeStream(sink, r.URL.Query())
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if utils.HookStateOK != ok {
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log.Sugar.Warnf("m3u8拉流失败 source: %s sink: %s", source, sink.String())
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_ = hls.SinkManager.Remove(sink.GetID())
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w.WriteHeader(http.StatusForbidden)
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return
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}
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}
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}
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// 更新最近的M3U8文件
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playlist := sink.(*hls.M3U8Sink).GetPlaylist(nil)
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if playlist == "" {
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if playlist = sink.(*hls.M3U8Sink).GetPlaylist(r.Context()); playlist == "" {
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log.Sugar.Warnf("hls拉流失败 未能生成有效m3u8文件 sink: %s source: %s", sink.GetID(), sink.GetSourceID())
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w.WriteHeader(http.StatusInternalServerError)
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return
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}
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}
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w.Write([]byte(playlist))
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}
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func (api *ApiServer) onRtc(sourceId string, w http.ResponseWriter, r *http.Request) {
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v := struct {
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Type string `json:"type"`
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SDP string `json:"sdp"`
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}{}
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data, err := io.ReadAll(r.Body)
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var liveGBSWF bool
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if err != nil {
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log.Sugar.Errorf("rtc拉流失败 err: %s remote: %s", err.Error(), r.RemoteAddr)
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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} else if liveGBSWF = "livegbs" == r.URL.Query().Get("wf"); liveGBSWF {
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// 兼容livegbs前端播放webrtc
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offer, err := base64.StdEncoding.DecodeString(string(data))
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if err != nil {
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log.Sugar.Errorf("rtc拉流失败 err: %s remote: %s", err.Error(), r.RemoteAddr)
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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v.Type = "offer"
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v.SDP = string(offer)
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} else if err := json.Unmarshal(data, &v); err != nil {
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log.Sugar.Errorf("rtc拉流失败 err: %s remote: %s", err.Error(), r.RemoteAddr)
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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done := make(chan struct{})
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sink := rtc.NewSink(api.generateSinkID(r.RemoteAddr), sourceId, v.SDP, func(sdp string) {
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response := struct {
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Type string `json:"type"`
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SDP string `json:"sdp"`
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}{
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Type: "answer",
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SDP: sdp,
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}
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var body []byte
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body, err = json.Marshal(response)
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if err != nil {
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panic(err)
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}
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if liveGBSWF {
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body = []byte(base64.StdEncoding.EncodeToString([]byte(sdp)))
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} else {
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w.Header().Set("Content-Type", "application/json")
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}
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w.Write(body)
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close(done)
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})
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log.Sugar.Infof("rtc拉流请求 source: %s sink: %s sdp:%v", sourceId, sink.String(), v.SDP)
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ok := stream.SubscribeStream(sink, r.URL.Query())
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if utils.HookStateOK != ok {
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log.Sugar.Warnf("rtc拉流失败 source: %s sink: %s", sourceId, sink.String())
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w.WriteHeader(http.StatusForbidden)
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return
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}
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select {
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case <-r.Context().Done():
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log.Sugar.Infof("rtc拉流请求取消 source: %s sink: %s", sourceId, stream.SinkID2String(sink.GetID()))
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sink.Close()
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break
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case <-done:
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break
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}
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}
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func (api *ApiServer) OnSourceList(w http.ResponseWriter, r *http.Request) {
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sources := stream.SourceManager.All()
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type SourceDetails struct {
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ID string `json:"id"`
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Protocol string `json:"protocol"` // 推流协议
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Time time.Time `json:"time"` // 推流时间
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SinkCount int `json:"sink_count"` // 播放端计数
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Bitrate string `json:"bitrate"` // 码率统计
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Tracks []string `json:"tracks"` // 每路流编码器ID
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Urls []string `json:"urls"` // 拉流地址
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}
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var details []SourceDetails
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for _, source := range sources {
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var codecs []string
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tracks := source.OriginTracks()
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for _, track := range tracks {
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codecs = append(codecs, track.Stream.CodecID.String())
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}
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details = append(details, SourceDetails{
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ID: source.GetID(),
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Protocol: source.GetType().String(),
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Time: source.CreateTime(),
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SinkCount: source.GetTransStreamPublisher().SinkCount(),
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Bitrate: strconv.Itoa(source.GetBitrateStatistics().PreviousSecond()/1024) + "KBS", // 后续开发
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||
Tracks: codecs,
|
||
Urls: stream.GetStreamPlayUrls(source.GetID()),
|
||
})
|
||
}
|
||
|
||
httpResponseOK(w, details)
|
||
}
|
||
|
||
func (api *ApiServer) OnSinkList(v *IDS, w http.ResponseWriter, r *http.Request) {
|
||
source := stream.SourceManager.Find(v.Source)
|
||
if source == nil {
|
||
httpResponseOK(w, nil)
|
||
return
|
||
}
|
||
|
||
type SinkDetails struct {
|
||
ID string `json:"id"`
|
||
Protocol string `json:"protocol"` // 拉流协议
|
||
Time time.Time `json:"time"` // 拉流时间
|
||
Bitrate string `json:"bitrate"` // 码率统计
|
||
Tracks []string `json:"tracks"` // 每路流编码器ID
|
||
}
|
||
|
||
var details []SinkDetails
|
||
sinks := source.GetTransStreamPublisher().Sinks()
|
||
for _, sink := range sinks {
|
||
details = append(details,
|
||
SinkDetails{
|
||
ID: stream.SinkID2String(sink.GetID()),
|
||
Protocol: sink.GetProtocol().String(),
|
||
Time: sink.CreateTime(),
|
||
},
|
||
)
|
||
}
|
||
|
||
httpResponseOK(w, details)
|
||
}
|
||
|
||
func (api *ApiServer) OnSourceClose(v *IDS, w http.ResponseWriter, r *http.Request) {
|
||
log.Sugar.Infof("close source: %v", v.Source)
|
||
|
||
if source := stream.SourceManager.Find(v.Source); source != nil {
|
||
source.Close()
|
||
} else {
|
||
log.Sugar.Warnf("Source with ID %s does not exist.", v.Source)
|
||
}
|
||
|
||
httpResponseOK(w, nil)
|
||
}
|
||
|
||
func (api *ApiServer) OnSinkClose(v *IDS, w http.ResponseWriter, r *http.Request) {
|
||
log.Sugar.Infof("close sink: %v", v)
|
||
|
||
var sinkId stream.SinkID
|
||
i, err := strconv.ParseUint(v.Sink, 10, 64)
|
||
if err != nil {
|
||
sinkId = stream.SinkID(v.Sink)
|
||
} else {
|
||
sinkId = stream.SinkID(i)
|
||
}
|
||
|
||
if source := stream.SourceManager.Find(v.Source); source != nil {
|
||
if sink := source.GetTransStreamPublisher().FindSink(sinkId); sink != nil {
|
||
sink.Close()
|
||
|
||
if sink.GetProtocol() == stream.TransStreamHls {
|
||
_ = hls.SinkManager.Remove(sinkId)
|
||
}
|
||
}
|
||
} else {
|
||
log.Sugar.Warnf("Source with ID %s does not exist.", v.Source)
|
||
}
|
||
|
||
httpResponseOK(w, nil)
|
||
}
|
||
|
||
func (api *ApiServer) OnStreamInfo(w http.ResponseWriter, r *http.Request) {
|
||
id := r.URL.Query().Get("streamid")
|
||
source := stream.SourceManager.Find(id)
|
||
if source == nil || source.IsClosed() {
|
||
w.WriteHeader(http.StatusBadRequest)
|
||
httpResponseJson(w, "stream not found")
|
||
return
|
||
} else if !source.IsCompleted() {
|
||
// 在请求结束前, 每隔1秒检查track探测是否完成
|
||
ticker := time.NewTicker(time.Second)
|
||
defer ticker.Stop()
|
||
for !source.IsClosed() && !source.IsCompleted() && r.Context().Err() == nil {
|
||
select {
|
||
case <-ticker.C:
|
||
break
|
||
case <-r.Context().Done():
|
||
break
|
||
}
|
||
}
|
||
}
|
||
|
||
tracks := source.OriginTracks()
|
||
if len(tracks) < 1 {
|
||
return
|
||
}
|
||
|
||
var deviceId string
|
||
var channelId string
|
||
split := strings.Split(id, "/")
|
||
if len(split) < 2 {
|
||
return
|
||
}
|
||
|
||
deviceId = split[0]
|
||
channelId = split[1]
|
||
if len(split[1]) >= 20 {
|
||
channelId = split[1][:20]
|
||
}
|
||
|
||
var transport string
|
||
if stream.SourceType28181 == source.GetType() {
|
||
if gb28181.SetupUDP != source.(gb28181.GBSource).SetupType() {
|
||
transport = "TCP"
|
||
} else {
|
||
transport = "UDP"
|
||
}
|
||
}
|
||
|
||
var token string
|
||
cookie, err := r.Cookie("token")
|
||
if err == nil {
|
||
token = cookie.Value
|
||
}
|
||
|
||
urls := stream.GetStreamPlayUrlsMap(id)
|
||
liveGBSUrls := make(map[string]string)
|
||
for streamName, url := range urls {
|
||
url += "?stream_token=" + token
|
||
|
||
// 兼容livegbs前端播放webrtc
|
||
if streamName == "rtc" {
|
||
if strings.HasPrefix(url, "http") {
|
||
url = strings.Replace(url, "http", "webrtc", 1)
|
||
} else if strings.HasPrefix(url, "https") {
|
||
url = strings.Replace(url, "https", "webrtcs", 1)
|
||
}
|
||
|
||
url += "&wf=livegbs"
|
||
}
|
||
|
||
liveGBSUrls[streamName] = url
|
||
}
|
||
|
||
var recordStartTime string
|
||
if startTime := source.GetTransStreamPublisher().RecordStartTime(); !startTime.IsZero() {
|
||
recordStartTime = startTime.Format("2006-01-02 15:04:05")
|
||
}
|
||
|
||
gbSource := Source2GBSource(source)
|
||
var downloadInfo *DownloadInfo
|
||
if gbSource != nil && InviteTypeDownload == gbSource.GetSessionName() {
|
||
progress := gbSource.GetPlaybackProgress()
|
||
gbSource.GetTransStreamPublisher()
|
||
downloadInfo = &DownloadInfo{
|
||
PlaybackDuration: gbSource.GetDuration(),
|
||
PlaybackSpeed: gbSource.GetSpeed(),
|
||
PlaybackFileSize: gbSource.GetFileSize(),
|
||
PlaybackStartTime: gbSource.GetStartTime(),
|
||
PlaybackEndTime: gbSource.GetEndTime(),
|
||
PlaybackFileURL: gbSource.GetTransStreamPublisher().GetRecordStreamPlayUrl(),
|
||
PlaybackProgress: progress,
|
||
Progress: progress,
|
||
}
|
||
|
||
}
|
||
statistics := source.GetBitrateStatistics()
|
||
response := struct {
|
||
*DownloadInfo
|
||
AudioEnable bool `json:"AudioEnable"`
|
||
CDN string `json:"CDN"`
|
||
CascadeSize int `json:"CascadeSize"`
|
||
ChannelID string `json:"ChannelID"`
|
||
ChannelName string `json:"ChannelName"`
|
||
CloudRecord bool `json:"CloudRecord"`
|
||
DecodeSize int `json:"DecodeSize"`
|
||
DeviceID string `json:"DeviceID"`
|
||
Duration int `json:"Duration"`
|
||
FLV string `json:"FLV"`
|
||
HLS string `json:"HLS"`
|
||
InBitRate int `json:"InBitRate"`
|
||
InBytes int `json:"InBytes"`
|
||
NumOutputs int `json:"NumOutputs"`
|
||
Ondemand bool `json:"Ondemand"`
|
||
OutBytes int `json:"OutBytes"`
|
||
RTMP string `json:"RTMP"`
|
||
RTPCount int `json:"RTPCount"`
|
||
RTPLostCount int `json:"RTPLostCount"`
|
||
RTPLostRate int `json:"RTPLostRate"`
|
||
RTSP string `json:"RTSP"`
|
||
RecordStartAt string `json:"RecordStartAt"` // 录制时间
|
||
RelaySize int `json:"RelaySize"`
|
||
SMSID string `json:"SMSID"`
|
||
SnapURL string `json:"SnapURL"`
|
||
SourceAudioCodecName string `json:"SourceAudioCodecName"`
|
||
SourceAudioSampleRate int `json:"SourceAudioSampleRate"`
|
||
SourceVideoCodecName string `json:"SourceVideoCodecName"`
|
||
SourceVideoFrameRate int `json:"SourceVideoFrameRate"`
|
||
SourceVideoHeight int `json:"SourceVideoHeight"`
|
||
SourceVideoWidth int `json:"SourceVideoWidth"`
|
||
StartAt string `json:"StartAt"`
|
||
StreamID string `json:"StreamID"`
|
||
Transport string `json:"Transport"`
|
||
VideoFrameCount int `json:"VideoFrameCount"`
|
||
WEBRTC string `json:"WEBRTC"`
|
||
WS_FLV string `json:"WS_FLV"`
|
||
}{
|
||
DownloadInfo: downloadInfo,
|
||
AudioEnable: true,
|
||
CDN: "",
|
||
CascadeSize: 0,
|
||
ChannelID: channelId,
|
||
ChannelName: "",
|
||
CloudRecord: false,
|
||
DecodeSize: 0,
|
||
DeviceID: deviceId,
|
||
Duration: int(time.Since(source.CreateTime()).Seconds()),
|
||
FLV: liveGBSUrls["flv"],
|
||
HLS: liveGBSUrls["hls"],
|
||
InBitRate: statistics.PreviousSecond() * 8 / 1024,
|
||
InBytes: int(statistics.Total()),
|
||
NumOutputs: source.GetTransStreamPublisher().SinkCount(),
|
||
Ondemand: true,
|
||
OutBytes: 0,
|
||
RTMP: liveGBSUrls["rtmp"],
|
||
RTPCount: 0,
|
||
RTPLostCount: 0,
|
||
RTPLostRate: 0,
|
||
RTSP: liveGBSUrls["rtsp"],
|
||
RecordStartAt: recordStartTime,
|
||
RelaySize: 0,
|
||
SMSID: "",
|
||
SnapURL: "",
|
||
SourceVideoFrameRate: 0,
|
||
StartAt: source.CreateTime().Format("2006-01-02 15:04:05"),
|
||
StreamID: id,
|
||
Transport: transport,
|
||
VideoFrameCount: 0,
|
||
WEBRTC: liveGBSUrls["rtc"],
|
||
WS_FLV: liveGBSUrls["ws_flv"],
|
||
}
|
||
|
||
for _, track := range tracks {
|
||
if utils.AVMediaTypeAudio == track.Stream.MediaType {
|
||
response.SourceAudioCodecName = track.Stream.CodecID.String()
|
||
response.SourceAudioSampleRate = track.Stream.AudioConfig.SampleRate
|
||
} else if utils.AVMediaTypeVideo == track.Stream.MediaType {
|
||
response.SourceVideoCodecName = track.Stream.CodecID.String()
|
||
// response.SourceVideoFrameRate
|
||
if track.Stream.CodecParameters != nil {
|
||
response.SourceVideoWidth = track.Stream.CodecParameters.Width()
|
||
response.SourceVideoHeight = track.Stream.CodecParameters.Height()
|
||
}
|
||
}
|
||
}
|
||
|
||
httpResponseJson(w, &response)
|
||
}
|
||
|
||
func (api *ApiServer) OnRecordStart(w http.ResponseWriter, req *http.Request) {
|
||
streamId := req.FormValue("streamid")
|
||
source := stream.SourceManager.Find(streamId)
|
||
if source == nil {
|
||
log.Sugar.Errorf("OnRecordStart stream not found streamid %s", streamId)
|
||
w.WriteHeader(http.StatusNotFound)
|
||
} else if ok := source.GetTransStreamPublisher().StartRecord(); !ok {
|
||
w.WriteHeader(http.StatusBadRequest)
|
||
} else {
|
||
// 返回拉流地址
|
||
httpResponseJson(w, &struct {
|
||
DownloadURL string `json:"DownloadURL"`
|
||
}{
|
||
DownloadURL: source.GetTransStreamPublisher().GetRecordStreamPlayUrl(),
|
||
})
|
||
}
|
||
|
||
}
|
||
|
||
func (api *ApiServer) OnRecordStop(w http.ResponseWriter, req *http.Request) {
|
||
streamId := req.FormValue("streamid")
|
||
source := stream.SourceManager.Find(streamId)
|
||
if source == nil {
|
||
log.Sugar.Errorf("OnRecordStop stream not found streamid %s", streamId)
|
||
w.WriteHeader(http.StatusNotFound)
|
||
} else if err := source.GetTransStreamPublisher().StopRecord(); err != nil {
|
||
w.WriteHeader(http.StatusBadRequest)
|
||
httpResponseJson(w, err.Error())
|
||
}
|
||
}
|