JS 資源
https://ossrs.net/players/js/...
https://ossrs.net/players/js/...
https://ossrs.net/players/js/...html
<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>提供一個Video的標籤，註釋裏面有他的一些屬性，srs播放須要一個video的ID rtc_media_player前端
$(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 選項可調整初始視頻遠近
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()) {
// 調整vr視窗偏移量
vr.effect.separation = 1.2;
}
vr.loadProgressManager.onLoad = function () {
// 視頻靜音
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('結束？')
$("#example").hide()
$(".shade").show()
$("#iframeDoms").show()
}
}
SrsRtcPlayerAsync 裏面是SRS 提供內置RTC方法，及RTC解析，劃重點java
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，這個地址是須要後臺提供支持的地址，也就是SRS服務中提供的WEBRTC 地址，會返回一些內置方法的參數git
$('#rtc_media_player').prop('srcObject', event.stream);
這個簡單 給video 注入media 的Object，vrVideoinit方法是VR全景播放的方法，可忽略，部分代碼提供angularjs