WEBRTC 视频直播记录

Posted 2021-09-14 你在看桥下风景

• 需求：解决RTMP/HLS/FLV 视频直播流延迟
• 背景：因为视频直播是VR视频直播，直播流是属于8K 4K,而VR直播流比平常的平面直播要大特别多，所以在网络分发（CDN）中会有存在延迟，及H5播放中因为网络问题存在的流缓存导致延迟增大，特别是hls及flv播放模式，网络问题越大导致的流延迟越高
• 优化目标：延迟在1s-2s（除去相机本身延迟）
• 优化方案：WEBRTC + SRS 服务
• 链接：
  SRS:https://github.com/ossrs/srs (身为前端的我并看不懂)
  WEBRTC:https://juejin.cn/post/684490... 简单介绍
  SRSWebRTCDemo:http://ossrs.net/srs.release/... SRSwebrtc演示

---

JS 资源
https://ossrs.net/players/js/...
https://ossrs.net/players/js/...
https://ossrs.net/players/js/...

---

• 实时：主要说WEBRTC方法，

•   <template>
          <div>
              <video id="rtc_media_player" autoplay></video>
              <!-- <video id="rtc_media_player" x-webkit-airplay=\'allow\' webkit-playsinline playsinline controls
              x5-video-player-type=\'h5\' x5-video-player-fullscreen x5-video-orientation=\'portrait\' crossOrigin=\'Anonymous\'
              allowsInlineMediaPlayback=\'true\' autoplay></video> -->
          </div>
      </template>
  

提供一个Video的标签，注释里面有他的一些属性，srs播放需要一个video的ID rtc_media_player

    $(function () {
        // Async-await-promise based SRS RTC Player.
        function SrsRtcPlayerAsync() {
            var self = {};
            self.play = async function (url) {
                var conf = self.__internal.prepareUrl(url);
                self.pc.addTransceiver("audio", { direction: "recvonly" });
                self.pc.addTransceiver("video", { direction: "recvonly" });

                var offer = await self.pc.createOffer();
                await self.pc.setLocalDescription(offer);
                var session = await new Promise(function (resolve, reject) {
                    // @see https://github.com/rtcdn/rtcdn-draft
                    var data = {
                        api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
                    };
                    console.log("Generated offer: ", data);

                    $.ajax({
                        type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
                        contentType: \'application/json\', dataType: \'json\'
                    }).done(function (data) {
                        console.log("Got answer: ", data);
                        if (data.code) {
                            reject(data); return;
                        }

                        resolve(data);
                    }).fail(function (reason) {
                        reject(reason);
                    });
                });
                console.log(session)
                await self.pc.setRemoteDescription(
                    new RTCSessionDescription({ type: \'answer\', sdp: session.sdp })
                );
                return session;
            };

            // Close the publisher.
            self.close = function () {
                self.pc.close();
            };

            // The callback when got remote stream.
            self.onaddstream = function (event) { };

            // Internal APIs.
            self.__internal = {
                defaultPath: \'/rtc/v1/play/\',
                prepareUrl: function (webrtcUrl) {
                    var urlObject = self.__internal.parse(webrtcUrl);

                    // If user specifies the schema, use it as API schema.
                    var schema = urlObject.user_query.schema;
                    schema = schema ? schema + \':\' : window.location.protocol;

                    var port = urlObject.port || 1985;
                    if (schema === \'https:\') {
                        port = urlObject.port || 443;
                    }

                    // @see https://github.com/rtcdn/rtcdn-draft
                    var api = urlObject.user_query.play || self.__internal.defaultPath;
                    if (api.lastIndexOf(\'/\') !== api.length - 1) {
                        api += \'/\';
                    }

                    apiUrl = schema + \'//\' + urlObject.server + \':\' + port + api;
                    for (var key in urlObject.user_query) {
                        if (key !== \'api\' && key !== \'play\') {
                            apiUrl += \'&\' + key + \'=\' + urlObject.user_query[key];
                        }
                    }
                    // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
                    var apiUrl = apiUrl.replace(api + \'&\', api + \'?\');

                    var streamUrl = urlObject.url;

                    return { apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port };
                },
                parse: function (url) {
                    // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
                    var a = document.createElement("a");
                    a.href = url.replace("rtmp://", "http://")
                        .replace("webrtc://", "http://")
                        .replace("rtc://", "http://");

                    var vhost = a.hostname;
                    var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
                    var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);

                    // parse the vhost in the params of app, that srs supports.
                    app = app.replace("...vhost...", "?vhost=");
                    if (app.indexOf("?") >= 0) {
                        var params = app.substr(app.indexOf("?"));
                        app = app.substr(0, app.indexOf("?"));

                        if (params.indexOf("vhost=") > 0) {
                            vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
                            if (vhost.indexOf("&") > 0) {
                                vhost = vhost.substr(0, vhost.indexOf("&"));
                            }
                        }
                    }

                    // when vhost equals to server, and server is ip,
                    // the vhost is __defaultVhost__
                    if (a.hostname === vhost) {
                        var re = /^(\\d+)\\.(\\d+)\\.(\\d+)\\.(\\d+)$/;
                        if (re.test(a.hostname)) {
                            vhost = "__defaultVhost__";
                        }
                    }

                    // parse the schema
                    var schema = "rtmp";
                    if (url.indexOf("://") > 0) {
                        schema = url.substr(0, url.indexOf("://"));
                    }

                    var port = a.port;
                    if (!port) {
                        if (schema === \'http\') {
                            port = 80;
                        } else if (schema === \'https\') {
                            port = 443;
                        } else if (schema === \'rtmp\') {
                            port = 1935;
                        }
                    }

                    var ret = {
                        url: url,
                        schema: schema,
                        server: a.hostname, port: port,
                        vhost: vhost, app: app, stream: stream
                    };
                    self.__internal.fill_query(a.search, ret);

                    // For webrtc API, we use 443 if page is https, or schema specified it.
                    if (!ret.port) {
                        if (schema === \'webrtc\' || schema === \'rtc\') {
                            if (ret.user_query.schema === \'https\') {
                                ret.port = 443;
                            } else if (window.location.href.indexOf(\'https://\') === 0) {
                                ret.port = 443;
                            } else {
                                // For WebRTC, SRS use 1985 as default API port.
                                ret.port = 1985;
                            }
                        }
                    }

                    return ret;
                },
                fill_query: function (query_string, obj) {
                    // pure user query object.
                    obj.user_query = {};

                    if (query_string.length === 0) {
                        return;
                    }

                    // split again for angularjs.
                    if (query_string.indexOf("?") >= 0) {
                        query_string = query_string.split("?")[1];
                    }

                    var queries = query_string.split("&");
                    for (var i = 0; i < queries.length; i++) {
                        var elem = queries[i];

                        var query = elem.split("=");
                        obj[query[0]] = query[1];
                        obj.user_query[query[0]] = query[1];
                    }

                    // alias domain for vhost.
                    if (obj.domain) {
                        obj.vhost = obj.domain;
                    }
                }
            };

            self.pc = new RTCPeerConnection(null);
            self.pc.onaddstream = function (event) {
                if (self.onaddstream) {
                    self.onaddstream(event);
                }
            };

            return self;
        }
        var sdk = null; // Global handler to do cleanup when replaying.
        var startPlay = function () {
            $(\'#rtc_media_player\').show();

            // Close PC when user replay.
            if (sdk) {
                sdk.close();
            }

            sdk = new SrsRtcPlayerAsync();
            sdk.onaddstream = function (event) {
                console.log(\'Start play, event: \', event);
                console.log(event.stream)
                $(\'#rtc_media_player\').prop(\'srcObject\', event.stream);
            };
            // For example:
            //      webrtc://r.ossrs.net/live/livestream
            var url = $("#txt_url").val();
            sdk.play(url).then(function (session) {
                $(\'#sessionid\').html(session.sessionid);
                $(\'#simulator-drop\').attr(\'href\', session.simulator + \'?drop=1&username=\' + session.sessionid);
            }).catch(function (reason) {
                sdk.close();
                $(\'#rtc_media_player\').hide();
                console.error(reason);
            });
        };

        $(\'#rtc_media_player\').hide();
        var query = parse_query_string();

        srs_init_rtc("#txt_url", query);
        $("#txt_url").val(\'webrtc://47.115.33.66/live/livestream\')
        $("#btn_play").click(function () {
            $(\'#rtc_media_player\').prop(\'muted\', false);
            startPlay();
            setTimeout(() => {
                // vrVideoinit()
            }, 1000);
        });

        if (query.autostart === \'true\') {
            $(\'#rtc_media_player\').prop(\'muted\', true);
            console.warn(\'For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a \' +
                \'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements\');

            startPlay();
        }
    });
    function vrVideoinit() {
        var scene, renderer;
        var container;
        //renderer = new THREE.WebGLRenderer();
        AVR.debug = true;
        if (!AVR.Broswer.isIE() && AVR.Broswer.webglAvailable()) {
            renderer = new THREE.WebGLRenderer();
        } else {
            renderer = new THREE.CanvasRenderer();
        }
        renderer.setPixelRatio(window.devicePixelRatio);
        container = document.getElementById(\'example\');
        container.appendChild(renderer.domElement);
        scene = new THREE.Scene();
        // fov 选项可调整初始视频远近
        var vr = new VR(scene, renderer, container, { "fov": 90 });
        //vr.playText="<img src=\'img/play90.png\' width=\'40\' height=\'40\'/>";
        vr.vrbox.radius = 600;
        if (AVR.isCrossScreen()) {
            // 调整vr视窗偏移量
            vr.effect.separation = 1.2;
        }
        vr.loadProgressManager.onLoad = function () {
            // 视频静音
            vr.video.muted = true;
        }
        //AVR.useGyroscope=false;
        vr.init(function () {

        });
        var videoDom = document.getElementById(\'rtc_media_player\')
        vr.play(videoDom, vr.resType.webrtcVideo);
        vr.video.addEventListener(\'canplay\', function () {
            vr.video.play()
        })
        vr.video.crossOrigin = "Anonymous";
        vr.video.onended = function () {
            console.log(\'结束？\')
            $("#example").hide()
            $(".shade").show()
            $("#iframeDoms").show()
        }
    }
    

SrsRtcPlayerAsync 里面是SRS 提供内置RTC方法，及RTC解析，划重点

            self.play = async function (url) {
                var conf = self.__internal.prepareUrl(url);
                self.pc.addTransceiver("audio", { direction: "recvonly" });
                self.pc.addTransceiver("video", { direction: "recvonly" });

                var offer = await self.pc.createOffer();
                await self.pc.setLocalDescription(offer);
                var session = await new Promise(function (resolve, reject) {
                    // @see https://github.com/rtcdn/rtcdn-draft
                    var data = {
                        api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
                    };
                    console.log("Generated offer: ", data);
                    $.ajax({
                        type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
                        contentType: \'application/json\', dataType: \'json\'
                    }).done(function (data) {
                        console.log("Got answer: ", data);
                        if (data.code) {
                            reject(data); return;
                        }

                        resolve(data);
                    }).fail(function (reason) {
                        reject(reason);
                    });
                });
                console.log(session)
                await self.pc.setRemoteDescription(
                    new RTCSessionDescription({ type: \'answer\', sdp: session.sdp })
                );
                return session;
            };

注意play方法中的 apiUrl，这个地址是需要后台提供支持的地址，也就是SRS服务中提供的WEBRTC 地址，会返回一些内置方法的参数

                $(\'#rtc_media_player\').prop(\'srcObject\', event.stream);

这个简单 给video 注入media 的Object，vrVideoinit方法是VR全景播放的方法，可忽略，部分代码提供