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### Description When converting between AAC and Opus formats (aac2opus or opus2aac), the `av_frame_get_buffer` API is frequently called. ### Objective The goal is to optimize the code logic and reduce the frequent allocation and deallocation of memory. In the case of aac2opus, av_frame_get_buffer is still frequently called. In the case of opus2aac, the goal is to avoid calling av_frame_get_buffer and reduce memory allocations. ### Additional Note Before calling the `av_audio_fifo_read` API, use `av_frame_make_writable` to check if the frame is writable. If it is not writable, create a new frame. --------- Co-authored-by: john <hondaxiao@tencent.com>
118 lines
4.9 KiB
HTML
118 lines
4.9 KiB
HTML
<!DOCTYPE html>
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<html>
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<head>
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<title>SRS</title>
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<meta charset="utf-8">
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<style>
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body{
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padding-top: 30px;
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}
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</style>
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<link rel="stylesheet" type="text/css" href="css/bootstrap.min.css"/>
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<script type="text/javascript" src="js/jquery-1.12.2.min.js"></script>
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<script type="text/javascript" src="js/adapter-7.4.0.min.js"></script>
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<script type="text/javascript" src="js/srs.sdk.js"></script>
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<script type="text/javascript" src="js/winlin.utility.js"></script>
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<script type="text/javascript" src="js/srs.page.js"></script>
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</head>
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<body>
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<img src='//ossrs.net/gif/v1/sls.gif?site=ossrs.net&path=/player/rtcpublisher'/>
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<div class="navbar navbar-fixed-top">
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<div class="navbar-inner">
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<div class="container">
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<a class="brand" href="https://github.com/ossrs/srs" target="_blank">SRS</a>
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<div class="nav-collapse collapse">
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<ul class="nav">
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<li><a id="nav_srs_player" href="srs_player.html">LivePlayer</a></li>
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<!--<li><a id="nav_rtc_player" href="rtc_player.html">RTC播放器</a></li>-->
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<!--<li><a id="nav_rtc_publisher" href="rtc_publisher.html">RTC推流</a></li>-->
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<li><a id="nav_whip" href="whip.html">WHIP</a></li>
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<li class="active"><a id="nav_whep" href="whip.html">WHEP</a></li>
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<li><a href="http://ossrs.net/srs.release/releases/app.html">iOS/Andriod</a></li>
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<!--<li><a id="nav_srs_publisher" href="srs_publisher.html">SRS编码器</a></li>-->
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<!--<li><a id="nav_srs_chat" href="srs_chat.html">SRS会议</a></li>-->
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<!--<li><a id="nav_srs_bwt" href="srs_bwt.html">SRS测网速</a></li>-->
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<!--<li><a id="nav_vlc" href="vlc.html">VLC播放器</a></li>-->
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<!--<li><a id="nav_gb28181" href="srs_gb28181.html">GB28181</a></li>-->
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<!--<li>
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<a href="https://github.com/ossrs/srs">
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<img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/ossrs/srs?style=social">
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</a>
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</li>-->
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</ul>
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</div>
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</div>
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</div>
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</div>
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<div class="container">
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<div id="main_info" class="alert alert-info fade in">
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<button type="button" class="close" data-dismiss="alert">×</button>
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<strong><span>Usage:</span></strong> <span>Enter the WebRTC WHEP URL and click "Play" to start playing.</span>
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</div>
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<div class="form-inline">
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URL:
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<input type="text" id="txt_url" class="input-xxlarge" value="">
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<button class="btn btn-primary" id="btn_play">Play</button>
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</div>
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<label></label>
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<video id="rtc_media_player" controls autoplay></video>
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<label></label>
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SessionID: <span id='sessionid'></span>
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<label></label>
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Simulator: <a href='#' id='simulator-drop'>Drop</a>
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<footer>
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<p></p>
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<p><a href="https://github.com/ossrs/srs">SRS Team © 2020</a></p>
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</footer>
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</div>
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<script type="text/javascript">
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$(function(){
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var sdk = null; // Global handler to do cleanup when republishing.
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var startPlay = function() {
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$('#rtc_media_player').show();
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// Close PC when user replay.
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if (sdk) {
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sdk.close();
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}
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sdk = new SrsRtcWhipWhepAsync();
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// User should set the stream when publish is done, @see https://webrtc.org/getting-started/media-devices
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// However SRS SDK provides a consist API like https://webrtc.org/getting-started/remote-streams
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$('#rtc_media_player').prop('srcObject', sdk.stream);
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// Optional callback, SDK will add track to stream.
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// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
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// For example: webrtc://r.ossrs.net/live/livestream
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var url = $("#txt_url").val();
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sdk.play(url).then(function(session){
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$('#sessionid').html(session.sessionid);
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$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
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}).catch(function (reason) {
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sdk.close();
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$('#rtc_media_player').hide();
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console.error(reason);
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});
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};
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$('#rtc_media_player').hide();
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var query = parse_query_string();
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srs_init_whep("#txt_url", query);
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$("#btn_play").click(startPlay);
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// Never play util windows loaded @see https://github.com/ossrs/srs/issues/2732
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if (query.autostart === 'true') {
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$('#rtc_media_player').prop('muted', true);
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console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
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'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');
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window.addEventListener("load", function(){ startPlay(); });
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}
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});
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</script>
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</body>
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</html>
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