关于前端:WEBRTC-视频直播记录

需要：解决RTMP/HLS/FLV 视频直播流提早
背景：因为视频直播是VR视频直播，直播流是属于8K 4K,而VR直播流比平时的立体直播要大特地多，所以在网络散发（CDN）中会有存在提早，及H5播放中因为网络问题存在的流缓存导致提早增大，特地是hls及flv播放模式，网络问题越大导致的流提早越高
优化指标：提早在1s-2s（除去相机自身提早）
优化计划：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... 简略介绍
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：次要说WEBRTC办法，

  <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，正文外面有他的一些属性，srs播放须要一个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('完结？')            $("#example").hide()            $(".shade").show()            $("#iframeDoms").show()        }    }

SrsRtcPlayerAsync 外面是SRS 提供内置RTC办法，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;            };

留神play办法中的 apiUrl，这个地址是须要后盾提供反对的地址，也就是SRS服务中提供的WEBRTC 地址，会返回一些内置办法的参数

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

这个简略给video 注入media 的Object，vrVideoinit办法是VR全景播放的办法，可疏忽，局部代码提供