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srs/trunk/research/players/whip.html
Winlin 26f4ab9923
WebRTC: Add support for A/V only WHEP/WHEP player. v6.0.116 (#3964)
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Co-authored-by: john <hondaxiao@tencent.com>
2024-03-19 21:08:03 +08:00

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<!DOCTYPE html>
<html>
<head>
<title>SRS</title>
<meta charset="utf-8">
<style>
body{
padding-top: 30px;
}
</style>
<link rel="stylesheet" type="text/css" href="css/bootstrap.min.css"/>
<script type="text/javascript" src="js/jquery-1.12.2.min.js"></script>
<script type="text/javascript" src="js/adapter-7.4.0.min.js"></script>
<script type="text/javascript" src="js/srs.sdk.js"></script>
<script type="text/javascript" src="js/winlin.utility.js"></script>
<script type="text/javascript" src="js/srs.page.js"></script>
</head>
<body>
<img src='//ossrs.net/gif/v1/sls.gif?site=ossrs.net&path=/player/rtcpublisher'/>
<div class="navbar navbar-fixed-top">
<div class="navbar-inner">
<div class="container">
<a class="brand" href="https://github.com/ossrs/srs" target="_blank">SRS</a>
<div class="nav-collapse collapse">
<ul class="nav">
<li><a id="nav_srs_player" href="srs_player.html">LivePlayer</a></li>
<!--<li><a id="nav_rtc_player" href="rtc_player.html">RTC播放器</a></li>-->
<!--<li><a id="nav_rtc_publisher" href="rtc_publisher.html">RTC推流</a></li>-->
<li class="active"><a id="nav_whip" href="whip.html">WHIP</a></li>
<li><a id="nav_whep" href="whip.html">WHEP</a></li>
<li><a href="http://ossrs.net/srs.release/releases/app.html">iOS/Andriod</a></li>
<!--<li><a id="nav_srs_publisher" href="srs_publisher.html">SRS编码器</a></li>-->
<!--<li><a id="nav_srs_chat" href="srs_chat.html">SRS会议</a></li>-->
<!--<li><a id="nav_srs_bwt" href="srs_bwt.html">SRS测网速</a></li>-->
<!--<li><a id="nav_vlc" href="vlc.html">VLC播放器</a></li>-->
<!--<li><a id="nav_gb28181" href="srs_gb28181.html">GB28181</a></li>-->
<!--<li>
<a href="https://github.com/ossrs/srs">
<img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/ossrs/srs?style=social">
</a>
</li>-->
</ul>
</div>
</div>
</div>
</div>
<div class="container">
<div id="main_info" class="alert alert-info fade in">
<button type="button" class="close" data-dismiss="alert">×</button>
<strong><span>Usage:</span></strong> <span>Enter the WebRTC WHIP URL and click "Publish" to start publishing.</span>
</div>
<div class="form-inline">
URL:
<input type="text" id="txt_url" class="input-xxlarge" value="">
<button class="btn btn-primary" id="btn_publish">Publish</button>
</div>
<p></p>
<video id="rtc_media_player" width="320" autoplay muted></video>
<p></p>
<div class="form-inline">
Controls:
<label>
<input type="checkbox" id="ch_videoonly" style="margin-bottom: 8px"> Video Only
</label>
<label>
<input type="checkbox" id="ch_audioonly" style="margin-bottom: 8px"> Audio Only
</label>
</div>
<p></p>
SessionID: <span id='sessionid'></span>
<p></p>
Audio: <span id='acodecs'></span><br/>
Video: <span id='vcodecs'></span>
<p></p>
Simulator: <a href='#' id='simulator-drop'>Drop</a>
<footer>
<p></p>
<p><a href="https://github.com/ossrs/srs">SRS Team &copy; 2020</a></p>
</footer>
</div>
<script type="text/javascript">
$(function(){
var sdk = null; // Global handler to do cleanup when republishing.
var startPublish = 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); };
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
sdk.pc.onicegatheringstatechange = function (event) {
if (sdk.pc.iceGatheringState === "complete") {
$('#acodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "audio"));
$('#vcodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "video"));
}
};
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.publish(url, {
videoOnly: $('#ch_videoonly').prop('checked'),
audioOnly: $('#ch_audioonly').prop('checked'),
}).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
}).catch(function (reason) {
// Throw by sdk.
if (reason instanceof SrsError) {
if (reason.name === 'HttpsRequiredError') {
alert(`WebRTC推流必须是HTTPS或者localhost${reason.name} ${reason.message}`);
} else {
alert(`${reason.name} ${reason.message}`);
}
}
// See https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia#exceptions
if (reason instanceof DOMException) {
if (reason.name === 'NotFoundError') {
alert(`找不到麦克风和摄像头设备getUserMedia ${reason.name} ${reason.message}`);
} else if (reason.name === 'NotAllowedError') {
alert(`你禁止了网页访问摄像头和麦克风getUserMedia ${reason.name} ${reason.message}`);
} else if (['AbortError', 'NotAllowedError', 'NotFoundError', 'NotReadableError', 'OverconstrainedError', 'SecurityError', 'TypeError'].includes(reason.name)) {
alert(`getUserMedia ${reason.name} ${reason.message}`);
}
}
sdk.close();
$('#rtc_media_player').hide();
console.error(reason);
});
};
$('#rtc_media_player').hide();
var query = parse_query_string();
srs_init_whip("#txt_url", query);
$("#btn_publish").click(startPublish);
// Never play util windows loaded @see https://github.com/ossrs/srs/issues/2732
if (query.autostart === 'true') {
window.addEventListener("load", function(){ startPublish(); });
}
});
</script>
</body>
</html>