- 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