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livetalking
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add webrtc push

This commit is contained in:
lipku 2024-04-27 18:08:57 +08:00
parent 027e15201a
commit 4137e5bce6
5 changed files with 873 additions and 4 deletions
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18
README.md
View File

@ -107,18 +107,34 @@ python app.py --fullbody --fullbody_img data/fullbody/img --fullbody_offset_x 10
- ernerf训练第三步torso如果训练的不好，在拼接处会有接缝。可以在上面的命令加上--torso_imgs data/xxx/torso_imgs，torso不用模型推理，直接用训练数据集里的torso图片。这种方式可能头颈处会有些人工痕迹。
### 3.6 webrtc
#### 3.6.1 p2p模式
此种模式不需要srs
```
python app.py --transport webrtc
```
用浏览器打开http://serverip:8010/webrtc.html
#### 3.6.2 通过srs一对多
启动srs
```
export CANDIDATE='<服务器外网ip>'
docker run --rm --env CANDIDATE=$CANDIDATE \
-p 1935:1935 -p 8080:8080 -p 1985:1985 -p 8000:8000/udp \
registry.cn-hangzhou.aliyuncs.com/ossrs/srs:5 \
objs/srs -c conf/rtc.conf
```
然后运行
```
python app.py --transport rtcpush --push_url 'http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream'
```
用浏览器打开http://serverip:8010/rtcpush.html
## 4. Docker Run
不需要第1步的安装，直接运行。
```
docker run --gpus all -it --network=host --rm registry.cn-hangzhou.aliyuncs.com/lipku/nerfstream:v1.3
```
srs的运行同2.1
docker版本已经不是最新代码，可以作为一个空环境，把最新代码拷进去运行。
## 5. Data flow
![](/assets/dataflow.png)

34
app.py
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@ -14,6 +14,7 @@ from threading import Thread,Event
import multiprocessing
from aiohttp import web
import aiohttp
from aiortc import RTCPeerConnection, RTCSessionDescription
from webrtc import HumanPlayer
@ -244,6 +245,33 @@ async def on_shutdown(app):
coros = [pc.close() for pc in pcs]
await asyncio.gather(*coros)
pcs.clear()
async def post(url,data):
try:
async with aiohttp.ClientSession() as session:
async with session.post(url,data=data) as response:
return await response.text()
except aiohttp.ClientError as e:
print(f'Error: {e}')
async def run(push_url):
pc = RTCPeerConnection()
pcs.add(pc)
@pc.on("connectionstatechange")
async def on_connectionstatechange():
print("Connection state is %s" % pc.connectionState)
if pc.connectionState == "failed":
await pc.close()
pcs.discard(pc)
player = HumanPlayer(nerfreal)
audio_sender = pc.addTrack(player.audio)
video_sender = pc.addTrack(player.video)
await pc.setLocalDescription(await pc.createOffer())
answer = await post(push_url,pc.localDescription.sdp)
await pc.setRemoteDescription(RTCSessionDescription(sdp=answer,type='answer'))
##########################################
if __name__ == '__main__':
@ -344,8 +372,8 @@ if __name__ == '__main__':
# parser.add_argument('--asr_model', type=str, default='facebook/wav2vec2-large-960h-lv60-self')
# parser.add_argument('--asr_model', type=str, default='facebook/hubert-large-ls960-ft')
parser.add_argument('--transport', type=str, default='rtmp') #rtmp webrtc
parser.add_argument('--push_url', type=str, default='rtmp://localhost/live/livestream')
parser.add_argument('--transport', type=str, default='rtmp') #rtmp webrtc rtcpush
parser.add_argument('--push_url', type=str, default='rtmp://localhost/live/livestream') #http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
parser.add_argument('--asr_save_feats', action='store_true')
# audio FPS
@ -437,6 +465,8 @@ if __name__ == '__main__':
loop.run_until_complete(runner.setup())
site = web.TCPSite(runner, '0.0.0.0', 8010)
loop.run_until_complete(site.start())
if opt.transport=='rtcpush':
loop.run_until_complete(run(opt.push_url))
loop.run_forever()
Thread(target=run_server, args=(web.AppRunner(appasync),)).start()

2
asrreal.py
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@ -191,7 +191,7 @@ class ASR:
if not self.terminated:
self.frames = self.frames[-(self.stride_left_size + self.stride_right_size):]
print(f'[INFO] frame_to_text... ')
#print(f'[INFO] frame_to_text... ')
#t = time.time()
logits, labels, text = self.__frame_to_text(inputs)
#print(f'-------wav2vec time:{time.time()-t:.4f}s')

125
web/rtcpush.html Normal file
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@ -0,0 +1,125 @@
<!DOCTYPE html>
<html>
<head>
<meta charset="UTF-8"/>
<meta name="viewport" content="width=device-width, initial-scale=1.0" />
<title>WebRTC webcam</title>
<style>
button {
padding: 8px 16px;
}
video {
width: 100%;
}
.option {
margin-bottom: 8px;
}
#media {
max-width: 1280px;
}
</style>
</head>
<body>
<div class="option">
<input id="use-stun" type="checkbox"/>
<label for="use-stun">Use STUN server</label>
</div>
<button class="btn btn-primary" id="btn_play">Start</button>
<form class="form-inline" id="echo-form">
<div class="form-group">
<p>input text</p>
<textarea cols="2" rows="3" style="width:600px;height:50px;" class="form-control" id="message">test</textarea>
</div>
<button type="submit" class="btn btn-default">Send</button>
</form>
<div id="media">
<h2>Media</h2>
<video id="rtc_media_player" style="width:600px;" controls autoplay></video>
</div>
<script src="srs.sdk.js"></script>
<script type="text/javascript" src="http://cdn.sockjs.org/sockjs-0.3.4.js"></script>
<script type="text/javascript" src="https://ajax.aspnetcdn.com/ajax/jquery/jquery-2.1.1.min.js"></script>
</body>
<script type="text/javascript" charset="utf-8">
$(document).ready(function() {
var host = window.location.hostname
var ws = new WebSocket("ws://"+host+":8000/humanecho");
//document.getElementsByTagName("video")[0].setAttribute("src", aa["video"]);
ws.onopen = function() {
console.log('Connected');
};
ws.onmessage = function(e) {
console.log('Received: ' + e.data);
data = e
var vid = JSON.parse(data.data);
console.log(typeof(vid),vid)
//document.getElementsByTagName("video")[0].setAttribute("src", vid["video"]);
};
ws.onclose = function(e) {
console.log('Closed');
};
$('#echo-form').on('submit', function(e) {
e.preventDefault();
var message = $('#message').val();
console.log('Sending: ' + message);
ws.send(message);
$('#message').val('');
});
});
$(function(){
var sdk = null; // Global handler to do cleanup when republishing.
var startPlay = function() {
$('#rtc_media_player').show();
// Close PC when user replay.
if (sdk) {
sdk.close();
}
sdk = new SrsRtcWhipWhepAsync();
// User should set the stream when publish is done, @see https://webrtc.org/getting-started/media-devices
// However SRS SDK provides a consist API like https://webrtc.org/getting-started/remote-streams
$('#rtc_media_player').prop('srcObject', sdk.stream);
// Optional callback, SDK will add track to stream.
// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
var host = window.location.hostname
// For example: webrtc://r.ossrs.net/live/livestream
var url = "http://"+host+":1985/rtc/v1/whep/?app=live&stream=livestream"
sdk.play(url).then(function(session){
//$('#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_whep("#txt_url", query);
$("#btn_play").click(startPlay);
// Never play util windows loaded @see https://github.com/ossrs/srs/issues/2732
// 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');
// window.addEventListener("load", function(){ startPlay(); });
// }
});
</script>
</html>

698
web/srs.sdk.js Normal file
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@ -0,0 +1,698 @@
//
// 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(", ");
}