-
-
Notifications
You must be signed in to change notification settings - Fork 5.4k
/
srs.sdk.js
698 lines (596 loc) · 26.9 KB
/
srs.sdk.js
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
//
// Copyright (c) 2013-2021 Winlin
//
// SPDX-License-Identifier: MIT
//
'use strict';
function SrsError(name, message) {
this.name = name;
this.message = message;
this.stack = (new Error()).stack;
}
SrsError.prototype = Object.create(Error.prototype);
SrsError.prototype.constructor = SrsError;
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-awat-prmise based SRS RTC Publisher.
function SrsRtcPublisherAsync() {
var self = {};
// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
self.constraints = {
audio: true,
video: {
width: {ideal: 320, max: 576}
}
};
// @see https://github.com/rtcdn/rtcdn-draft
// @url The WebRTC url to play with, for example:
// webrtc://r.ossrs.net/live/livestream
// or specifies the API port:
// webrtc://r.ossrs.net:11985/live/livestream
// or autostart the publish:
// webrtc://r.ossrs.net/live/livestream?autostart=true
// or change the app from live to myapp:
// webrtc://r.ossrs.net:11985/myapp/livestream
// or change the stream from livestream to mystream:
// webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream
// or set the candidate(eip) of answer:
// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
// or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP:
// webrtc://r.ossrs.net/live/livestream?encrypt=false
// or any other information, will pass-by in the query:
// webrtc://r.ossrs.net/live/livestream?vhost=xxx
// webrtc://r.ossrs.net/live/livestream?token=xxx
self.publish = async function (url) {
var conf = self.__internal.prepareUrl(url);
self.pc.addTransceiver("audio", {direction: "sendonly"});
self.pc.addTransceiver("video", {direction: "sendonly"});
//self.pc.addTransceiver("video", {direction: "sendonly"});
//self.pc.addTransceiver("audio", {direction: "sendonly"});
if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
}
var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
stream.getTracks().forEach(function (track) {
self.pc.addTrack(track);
// Notify about local track when stream is ok.
self.ontrack && self.ontrack({track: track});
});
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, tid: conf.tid, streamurl: conf.streamUrl,
clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
const xhr = new XMLHttpRequest();
xhr.onload = function() {
if (xhr.readyState !== xhr.DONE) return;
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
const data = JSON.parse(xhr.responseText);
console.log("Got answer: ", data);
return data.code ? reject(xhr) : resolve(data);
}
xhr.open('POST', conf.apiUrl, true);
xhr.setRequestHeader('Content-type', 'application/json');
xhr.send(JSON.stringify(data));
});
await self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
);
session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
return session;
};
// Close the publisher.
self.close = function () {
self.pc && self.pc.close();
self.pc = null;
};
// The callback when got local stream.
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
self.ontrack = function (event) {
// Add track to stream of SDK.
self.stream.addTrack(event.track);
};
// Internal APIs.
self.__internal = {
defaultPath: '/rtc/v1/publish/',
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 += '/';
}
var 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 + '?');
var streamUrl = urlObject.url;
return {
apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
};
},
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.substring(1, a.pathname.lastIndexOf("/"));
var stream = a.pathname.slice(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.slice(app.indexOf("?"));
app = app.slice(0, app.indexOf("?"));
if (params.indexOf("vhost=") > 0) {
vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
if (vhost.indexOf("&") > 0) {
vhost = vhost.slice(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+)$/;
if (re.test(a.hostname)) {
vhost = "__defaultVhost__";
}
}
// parse the schema
var schema = "rtmp";
if (url.indexOf("://") > 0) {
schema = url.slice(0, url.indexOf("://"));
}
var port = a.port;
if (!port) {
// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
}
// Guess by schema.
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);
// To keep api consistent between player and publisher.
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
// @see https://webrtc.org/getting-started/media-devices
self.stream = new MediaStream();
return self;
}
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-await-promise based SRS RTC Player.
function SrsRtcPlayerAsync() {
var self = {};
// @see https://github.com/rtcdn/rtcdn-draft
// @url The WebRTC url to play with, for example:
// webrtc://r.ossrs.net/live/livestream
// or specifies the API port:
// webrtc://r.ossrs.net:11985/live/livestream
// webrtc://r.ossrs.net:80/live/livestream
// or autostart the play:
// webrtc://r.ossrs.net/live/livestream?autostart=true
// or change the app from live to myapp:
// webrtc://r.ossrs.net:11985/myapp/livestream
// or change the stream from livestream to mystream:
// webrtc://r.ossrs.net:11985/live/mystream
// or set the api server to myapi.domain.com:
// webrtc://myapi.domain.com/live/livestream
// or set the candidate(eip) of answer:
// webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
// or force to access https API:
// webrtc://r.ossrs.net/live/livestream?schema=https
// or use plaintext, without SRTP:
// webrtc://r.ossrs.net/live/livestream?encrypt=false
// or any other information, will pass-by in the query:
// webrtc://r.ossrs.net/live/livestream?vhost=xxx
// webrtc://r.ossrs.net/live/livestream?token=xxx
self.play = async function(url) {
var conf = self.__internal.prepareUrl(url);
self.pc.addTransceiver("audio", {direction: "recvonly"});
self.pc.addTransceiver("video", {direction: "recvonly"});
//self.pc.addTransceiver("video", {direction: "recvonly"});
//self.pc.addTransceiver("audio", {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, tid: conf.tid, streamurl: conf.streamUrl,
clientip: null, sdp: offer.sdp
};
console.log("Generated offer: ", data);
const xhr = new XMLHttpRequest();
xhr.onload = function() {
if (xhr.readyState !== xhr.DONE) return;
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
const data = JSON.parse(xhr.responseText);
console.log("Got answer: ", data);
return data.code ? reject(xhr) : resolve(data);
}
xhr.open('POST', conf.apiUrl, true);
xhr.setRequestHeader('Content-type', 'application/json');
xhr.send(JSON.stringify(data));
});
await self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
);
session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
return session;
};
// Close the player.
self.close = function() {
self.pc && self.pc.close();
self.pc = null;
};
// The callback when got remote track.
// Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
self.ontrack = function (event) {
// https://webrtc.org/getting-started/remote-streams
self.stream.addTrack(event.track);
};
// 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 += '/';
}
var 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 + '?');
var streamUrl = urlObject.url;
return {
apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).slice(0, 7)
};
},
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.substring(1, a.pathname.lastIndexOf("/"));
var stream = a.pathname.slice(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.slice(app.indexOf("?"));
app = app.slice(0, app.indexOf("?"));
if (params.indexOf("vhost=") > 0) {
vhost = params.slice(params.indexOf("vhost=") + "vhost=".length);
if (vhost.indexOf("&") > 0) {
vhost = vhost.slice(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.slice(0, url.indexOf("://"));
}
var port = a.port;
if (!port) {
// Finger out by webrtc url, if contains http or https port, to overwrite default 1985.
if (schema === 'webrtc' && url.indexOf(`webrtc://${a.host}:`) === 0) {
port = (url.indexOf(`webrtc://${a.host}:80`) === 0) ? 80 : 443;
}
// Guess by schema.
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);
// Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
self.stream = new MediaStream();
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
self.pc.ontrack = function(event) {
if (self.ontrack) {
self.ontrack(event);
}
};
return self;
}
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-awat-prmise based SRS RTC Publisher by WHIP.
function SrsRtcWhipWhepAsync() {
var self = {};
// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
self.constraints = {
audio: true,
video: {
width: {ideal: 320, max: 576}
}
};
// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
// @url The WebRTC url to publish with, for example:
// http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
// @options The options to control playing, supports:
// videoOnly: boolean, whether only play video, default to false.
// audioOnly: boolean, whether only play audio, default to false.
self.publish = async function (url, options) {
if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
if (!options?.videoOnly) {
self.pc.addTransceiver("audio", {direction: "sendonly"});
} else {
self.constraints.audio = false;
}
if (!options?.audioOnly) {
self.pc.addTransceiver("video", {direction: "sendonly"});
} else {
self.constraints.video = false;
}
if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
}
var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
stream.getTracks().forEach(function (track) {
self.pc.addTrack(track);
// Notify about local track when stream is ok.
self.ontrack && self.ontrack({track: track});
});
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
const answer = await new Promise(function (resolve, reject) {
console.log(`Generated offer: ${offer.sdp}`);
const xhr = new XMLHttpRequest();
xhr.onload = function() {
if (xhr.readyState !== xhr.DONE) return;
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
const data = xhr.responseText;
console.log("Got answer: ", data);
return data.code ? reject(xhr) : resolve(data);
}
xhr.open('POST', url, true);
xhr.setRequestHeader('Content-type', 'application/sdp');
xhr.send(offer.sdp);
});
await self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: answer})
);
return self.__internal.parseId(url, offer.sdp, answer);
};
// See https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
// @url The WebRTC url to play with, for example:
// http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream
// @options The options to control playing, supports:
// videoOnly: boolean, whether only play video, default to false.
// audioOnly: boolean, whether only play audio, default to false.
self.play = async function(url, options) {
if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
if (!options?.videoOnly) self.pc.addTransceiver("audio", {direction: "recvonly"});
if (!options?.audioOnly) self.pc.addTransceiver("video", {direction: "recvonly"});
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
const answer = await new Promise(function(resolve, reject) {
console.log(`Generated offer: ${offer.sdp}`);
const xhr = new XMLHttpRequest();
xhr.onload = function() {
if (xhr.readyState !== xhr.DONE) return;
if (xhr.status !== 200 && xhr.status !== 201) return reject(xhr);
const data = xhr.responseText;
console.log("Got answer: ", data);
return data.code ? reject(xhr) : resolve(data);
}
xhr.open('POST', url, true);
xhr.setRequestHeader('Content-type', 'application/sdp');
xhr.send(offer.sdp);
});
await self.pc.setRemoteDescription(
new RTCSessionDescription({type: 'answer', sdp: answer})
);
return self.__internal.parseId(url, offer.sdp, answer);
};
// Close the publisher.
self.close = function () {
self.pc && self.pc.close();
self.pc = null;
};
// The callback when got local stream.
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
self.ontrack = function (event) {
// Add track to stream of SDK.
self.stream.addTrack(event.track);
};
self.pc = new RTCPeerConnection(null);
// To keep api consistent between player and publisher.
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
// @see https://webrtc.org/getting-started/media-devices
self.stream = new MediaStream();
// Internal APIs.
self.__internal = {
parseId: (url, offer, answer) => {
let sessionid = offer.substr(offer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
sessionid = sessionid.substr(0, sessionid.indexOf('\n') - 1) + ':';
sessionid += answer.substr(answer.indexOf('a=ice-ufrag:') + 'a=ice-ufrag:'.length);
sessionid = sessionid.substr(0, sessionid.indexOf('\n'));
const a = document.createElement("a");
a.href = url;
return {
sessionid: sessionid, // Should be ice-ufrag of answer:offer.
simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
};
},
};
// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
self.pc.ontrack = function(event) {
if (self.ontrack) {
self.ontrack(event);
}
};
return self;
}
// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
function SrsRtcFormatSenders(senders, kind) {
var codecs = [];
senders.forEach(function (sender) {
var params = sender.getParameters();
params && params.codecs && params.codecs.forEach(function(c) {
if (kind && sender.track.kind !== kind) {
return;
}
if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
return;
}
var s = '';
s += c.mimeType.replace('audio/', '').replace('video/', '');
s += ', ' + c.clockRate + 'HZ';
if (sender.track.kind === "audio") {
s += ', channels: ' + c.channels;
}
s += ', pt: ' + c.payloadType;
codecs.push(s);
});
});
return codecs.join(", ");
}