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@ -66,167 +66,207 @@
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$(function(){
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// Async-await-promise based SRS RTC Player.
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function SrsRtcPlayerAsync() {
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var self = {
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play: async function(apiUrl, streamUrl) {
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self.pc.addTransceiver("audio", {direction: "recvonly"});
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self.pc.addTransceiver("video", {direction: "recvonly"});
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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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var session = await new Promise(function(resolve, reject) {
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = {
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api: apiUrl, streamurl: streamUrl, clientip: null, sdp: offer.sdp
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};
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console.log("Generated offer: ", data);
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$.ajax({
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type: "POST", url: apiUrl, data: JSON.stringify(data),
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contentType:'application/json', dataType: 'json'
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}).done(function(data) {
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console.log("Got answer: ", data);
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if (data.code) {
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reject(data); return;
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}
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resolve(data);
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}).fail(function(reason){
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reject(reason);
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});
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var self = {};
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// @url The WebRTC url to play with, for example:
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// webrtc://r.ossrs.net/live/livestream
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// or specifies the API port:
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// webrtc://r.ossrs.net:11985/live/livestream
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// or autostart the play:
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// webrtc://r.ossrs.net/live/livestream?autostart=true
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// or change the app from live to myapp:
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// webrtc://r.ossrs.net:11985/myapp/livestream
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// or change the stream from livestream to mystream:
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// webrtc://r.ossrs.net:11985/live/mystream
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// or set the api server to myapi.domain.com:
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// webrtc://myapi.domain.com/live/livestream
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// or set the candidate(ip) of answer:
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// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
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// or force to access https API:
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// webrtc://r.ossrs.net/live/livestream?schema=https
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// or use plaintext, without SRTP:
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// webrtc://r.ossrs.net/live/livestream?encrypt=false
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// or any other information, will pass-by in the query:
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// webrtc://r.ossrs.net/live/livestream?vhost=xxx
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// webrtc://r.ossrs.net/live/livestream?token=xxx
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self.play = async function(url) {
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var conf = self.__internal.prepareUrl(url);
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self.pc.addTransceiver("audio", {direction: "recvonly"});
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self.pc.addTransceiver("video", {direction: "recvonly"});
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var offer = await self.pc.createOffer();
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await self.pc.setLocalDescription(offer);
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var session = await new Promise(function(resolve, reject) {
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = {
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api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
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};
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console.log("Generated offer: ", data);
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$.ajax({
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type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
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contentType:'application/json', dataType: 'json'
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}).done(function(data) {
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console.log("Got answer: ", data);
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if (data.code) {
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reject(data); return;
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}
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resolve(data);
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}).fail(function(reason){
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reject(reason);
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});
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await self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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);
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return session;
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},
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close: function() {
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self.pc.close();
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},
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// callbacks.
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onaddstream: function (event) {}
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};
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self.pc = new RTCPeerConnection(null);
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self.pc.onaddstream = function (event) {
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if (self.onaddstream) {
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self.onaddstream(event);
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}
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});
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await self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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);
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return session;
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};
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return self;
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}
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// Promise based SRS RTC Player.
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function SrsRtcPlayerPromise() {
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var self = {
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play: function(apiUrl, streamUrl) {
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self.pc.addTransceiver("audio", {direction: "recvonly"});
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self.pc.addTransceiver("video", {direction: "recvonly"});
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return self.pc.createOffer().then(function(offer) {
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return self.pc.setLocalDescription(offer).then(function(){ return offer; });
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}).then(function(offer) {
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return new Promise(function(resolve, reject) {
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = {
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api: apiUrl, streamurl: streamUrl, clientip: null, sdp: offer.sdp
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};
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console.log("Generated offer: ", data);
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$.ajax({
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type: "POST", url: apiUrl, data: JSON.stringify(data),
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contentType:'application/json', dataType: 'json'
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}).done(function(data) {
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console.log("Got answer: ", data);
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if (data.code) {
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reject(data); return;
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}
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resolve(data);
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}).fail(function(reason){
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reject(reason);
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});
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});
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}).then(function(session) {
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return self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: session.sdp})
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).then(function(){ return session; });
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});
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},
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close: function() {
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self.pc.close();
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},
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// callbacks.
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onaddstream: function (event) {}
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// Close the publisher.
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self.close = function() {
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self.pc.close();
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};
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self.pc = new RTCPeerConnection(null);
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self.pc.onaddstream = function (event) {
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if (self.onaddstream) {
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self.onaddstream(event);
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}
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};
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return self;
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}
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// The callback when got remote stream.
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self.onaddstream = function (event) {};
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// Internal APIs.
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self.__internal = {
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defaultPath: '/rtc/v1/play/',
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parse: function (url) {
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// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
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var a = document.createElement("a");
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a.href = url.replace("rtmp://", "http://")
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.replace("webrtc://", "http://")
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.replace("rtc://", "http://");
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var vhost = a.hostname;
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var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
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var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);
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// parse the vhost in the params of app, that srs supports.
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app = app.replace("...vhost...", "?vhost=");
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if (app.indexOf("?") >= 0) {
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var params = app.substr(app.indexOf("?"));
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app = app.substr(0, app.indexOf("?"));
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if (params.indexOf("vhost=") > 0) {
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vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
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if (vhost.indexOf("&") > 0) {
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vhost = vhost.substr(0, vhost.indexOf("&"));
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}
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}
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}
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// Callback based SRS RTC Player.
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function SrsRtcPlayerCallbacks() {
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var self = {
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play: function(apiUrl, streamUrl, success, fail) {
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self.pc.addTransceiver("audio", {direction: "recvonly"});
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self.pc.addTransceiver("video", {direction: "recvonly"});
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self.pc.createOffer(function(offer){
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onOffer(offer);
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}, function(reason){
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fail(reason);
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});
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// when vhost equals to server, and server is ip,
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// the vhost is __defaultVhost__
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if (a.hostname === vhost) {
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var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
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if (re.test(a.hostname)) {
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vhost = "__defaultVhost__";
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}
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}
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var onOffer = function(offer) {
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self.pc.setLocalDescription(offer, function(){
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onOfferDone(offer);
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}, function(reason) {
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fail(reason);
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});
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};
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// parse the schema
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var schema = "rtmp";
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if (url.indexOf("://") > 0) {
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schema = url.substr(0, url.indexOf("://"));
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}
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var onOfferDone = function (offer) {
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// @see https://github.com/rtcdn/rtcdn-draft
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var data = {
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api: apiUrl, streamurl: streamUrl, clientip: null, sdp: offer.sdp
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};
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console.log("Generated offer: ", data);
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$.ajax({
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type: "POST", url: apiUrl, data: JSON.stringify(data),
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contentType:'application/json', dataType: 'json'
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}).done(function(data) {
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console.log("Got answer: ", data);
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if (data.code) {
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fail(data); return;
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}
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var port = a.port;
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if (!port) {
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if (schema === 'http') {
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port = 80;
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} else if (schema === 'https') {
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port = 443;
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} else if (schema === 'rtmp') {
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port = 1935;
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}
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}
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onAnswer(data);
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}).fail(function(reason){
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fail(reason);
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});
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var ret = {
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url: url,
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schema: schema,
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server: a.hostname, port: port,
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vhost: vhost, app: app, stream: stream
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};
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self.__internal.fill_query(a.search, ret);
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// For webrtc API, we use 443 if page is https, or schema specified it.
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if (!ret.port) {
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if (schema === 'webrtc' || schema === 'rtc') {
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if (ret.user_query.schema === 'https') {
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ret.port = 443;
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} else if (window.location.href.indexOf('https://') === 0) {
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ret.port = 443;
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} else {
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// For WebRTC, SRS use 1985 as default API port.
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ret.port = 1985;
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}
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}
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}
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var onAnswer = function(session) {
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var answer = session.sdp;
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self.pc.setRemoteDescription(
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new RTCSessionDescription({type: 'answer', sdp: answer})
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).then(function(){
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success(session);
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}).catch(function(reason) {
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fail(reason);
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});
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};
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return ret;
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},
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close: function() {
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self.pc.close();
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prepareUrl: function (webrtcUrl) {
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var urlObject = self.__internal.parse(webrtcUrl);
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// If user specifies the schema, use it as API schema.
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var schema = urlObject.user_query.schema;
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schema = schema ? schema + ':' : window.location.protocol;
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var port = urlObject.port || 1985;
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if (schema === 'https:') {
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port = urlObject.port || 443;
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}
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// @see https://github.com/rtcdn/rtcdn-draft
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var api = urlObject.user_query.play || self.__internal.defaultPath;
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if (api.lastIndexOf('/') !== api.length - 1) {
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api += '/';
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}
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apiUrl = schema + '//' + urlObject.server + ':' + port + api;
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for (var key in urlObject.user_query) {
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if (key !== 'api' && key !== 'play') {
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apiUrl += '&' + key + '=' + urlObject.user_query[key];
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}
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}
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// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
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var apiUrl = apiUrl.replace(api + '&', api + '?');
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var streamUrl = urlObject.url;
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return {apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port};
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},
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// callbacks.
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onaddstream: function (event) {}
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fill_query: function (query_string, obj) {
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// pure user query object.
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obj.user_query = {};
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if (query_string.length === 0) {
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return;
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}
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// split again for angularjs.
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if (query_string.indexOf("?") >= 0) {
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query_string = query_string.split("?")[1];
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}
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var queries = query_string.split("&");
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for (var i = 0; i < queries.length; i++) {
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var elem = queries[i];
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var query = elem.split("=");
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obj[query[0]] = query[1];
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obj.user_query[query[0]] = query[1];
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}
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// alias domain for vhost.
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if (obj.domain) {
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obj.vhost = obj.domain;
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}
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}
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};
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self.pc = new RTCPeerConnection(null);
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@ -235,98 +275,36 @@
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self.onaddstream(event);
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}
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};
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return self;
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}
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// Build RTC api url.
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var prepareUrl = function () {
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var apiUrl, streamUrl;
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if (true) {
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var urlObject = parse_rtmp_url($("#txt_url").val());
|
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// If user specifies the schema, use it as API schema.
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|
var schema = urlObject.user_query.schema;
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|
schema = schema? schema + ':' : window.location.protocol;
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var port = urlObject.port || 1985;
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if (schema === 'https:') {
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port = urlObject.port || 443;
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}
|
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|
// @see https://github.com/rtcdn/rtcdn-draft
|
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|
|
var api = urlObject.user_query.play || '/rtc/v1/play/';
|
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|
|
if (api.lastIndexOf('/') !== api.length - 1) {
|
|
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|
|
api += '/';
|
|
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|
|
}
|
|
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|
|
|
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|
|
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
|
|
|
|
|
apiUrl = apiUrl.replace(api + '&', api + '?');
|
|
|
|
|
|
|
|
|
|
streamUrl = urlObject.url;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
return {apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port};
|
|
|
|
|
};
|
|
|
|
|
var sdk = null; // Global handler to do cleanup when replaying.
|
|
|
|
|
var startPlay = function() {
|
|
|
|
|
$('#rtc_media_player').show();
|
|
|
|
|
|
|
|
|
|
// Start play with conf.
|
|
|
|
|
var playStream = function (conf) {
|
|
|
|
|
// Close PC when user replay.
|
|
|
|
|
if (sdk) {
|
|
|
|
|
sdk.close();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
// Use Callback style.
|
|
|
|
|
if (true) {
|
|
|
|
|
if (true) {
|
|
|
|
|
sdk = new SrsRtcPlayerAsync();
|
|
|
|
|
} else {
|
|
|
|
|
sdk = new SrsRtcPlayerPromise();
|
|
|
|
|
}
|
|
|
|
|
sdk.onaddstream = function (event) {
|
|
|
|
|
console.log('Start play, event: ', event);
|
|
|
|
|
$('#rtc_media_player').prop('srcObject', event.stream);
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
sdk.play(conf.apiUrl, conf.streamUrl).then(function(session){
|
|
|
|
|
var simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
|
|
|
|
|
$('#sessionid').html(session.sessionid);
|
|
|
|
|
$('#simulator-drop').attr('href', simulator + '?drop=1&username=' + session.sessionid);
|
|
|
|
|
}).catch(function (reason) {
|
|
|
|
|
sdk.close();
|
|
|
|
|
$('#rtc_media_player').hide();
|
|
|
|
|
console.error(reason);
|
|
|
|
|
});
|
|
|
|
|
} else if (false) {
|
|
|
|
|
sdk = new SrsRtcPlayerCallbacks();
|
|
|
|
|
sdk.onaddstream = function (event) {
|
|
|
|
|
console.log('Start play, event: ', event);
|
|
|
|
|
$('#rtc_media_player').prop('srcObject', event.stream);
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
sdk.play(conf.apiUrl, conf.streamUrl, function (session) {
|
|
|
|
|
var simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
|
|
|
|
|
$('#sessionid').html(session.sessionid);
|
|
|
|
|
$('#simulator-drop').attr('href', simulator + '?drop=1&username=' + session.sessionid);
|
|
|
|
|
}, function (reason) {
|
|
|
|
|
sdk.close();
|
|
|
|
|
$('#rtc_media_player').hide();
|
|
|
|
|
throw reason;
|
|
|
|
|
});
|
|
|
|
|
}
|
|
|
|
|
};
|
|
|
|
|
sdk = new SrsRtcPlayerAsync();
|
|
|
|
|
sdk.onaddstream = function (event) {
|
|
|
|
|
console.log('Start play, event: ', event);
|
|
|
|
|
$('#rtc_media_player').prop('srcObject', event.stream);
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
var sdk = null; // Global handler to do cleanup when replaying.
|
|
|
|
|
var startPlay = function() {
|
|
|
|
|
$('#rtc_media_player').show();
|
|
|
|
|
var conf = prepareUrl();
|
|
|
|
|
playStream(conf);
|
|
|
|
|
// 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();
|
|
|
|
|
|
winlin
4 years ago