<commit_before><!DOCTYPE html>
<html>
<head>

  <meta name="keywords" content="WebRTC, HTML5, JavaScript" />
  <meta name="description" content="AppRTC URL parameters" />
  <meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
  <meta name="viewport" content="width=device-width, initial-scale=1.0, minimum-scale=1.0">

  <base target="_blank">

  <title>AppRTC parameters</title>

  <link rel="icon" sizes="192x192" href="images/webrtc-icon-192x192.png">
  <link href="//fonts.googleapis.com/css?family=Roboto:300,400,500,700" rel="stylesheet" type="text/css">
  <link rel="stylesheet" href="https://googlechrome.github.io/webrtc/samples/web/css/main.css" />

  <style>
  div#container {
    max-width: 65em;
  }
    h2 {
      font-size: 1.5em;
      font-weight: 500;
    }
    h3 {
      border-top: none;
    }
    section {
      border-bottom: 1px solid #eee;
      margin: 0 0 1.5em 0;
      padding: 0 0 1.5em 0;
    }
    section:last-child {
      border-bottom: none;
      margin: 0;
      padding: 0;
    }
    td {
      padding: 0.5em 1em;
      vertical-align: top;
    }
    td:first-of-type {
      text-align: right;
    }
  </style>
</head>
<body>
  <div id="container">

    <h1>AppRTC parameters</h1>

    <section>

      <p>A number of settings for the <a href="https://apprtc.appspot.com">AppRTC video chat application</a> can be changed by adding URL parameters.</p>

      <p>For example: <a href="https://apprtc.appspot.com/?hd=true&amp;stereo=true&amp;debug=loopback">https://apprtc.appspot.com/?hd=true&amp;stereo=true&amp;debug=loopback</a></p>

      <p>The file using the parameters is <a href="https://github.com/GoogleChrome/webrtc/blob/master/samples/web/content/apprtc/apprtc.py" title="apprtc.py file in AppRTC repo">apprtc.py</a>. More Google-specific parameters are available from the <a href="//code.google.com/p/webrtc/source/browse/trunk/talk/app/webrtc/mediaconstraintsinterface.h" title="mediaconstraintsinterface.h">MediaConstraints interface</a>.</p>

      <p>For more information see <a href="http://gingertech.net/2014/03/19/apprtc-googles-webrtc-test-app-and-its-parameters/">AppRTC : Google's WebRTC test app and its parameters</a>.</p>

    </section>

    <section>

      <table>
        <tr><td><a href="https://apprtc.appspot.com?hd=true">hd=true</a></td><td>Use HD camera resolution <a href="http://googlechrome.github.io/webrtc/samples/web/content/getusermedia/resolution/" title="Video constraints demo">constraints</a>, i.e. minWidth: 1280, minHeight: 720</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?stereo=true">stereo=true</a></td><td>Turn on stereo audio</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?debug=loopback">debug=loopback</a></td><td>Connect to yourself, e.g. to test firewalls</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?ts=https://turn.example.org">ts=[turnserver]</a></td><td>Set TURN server different from the default</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=true&amp;video=false">audio=true&amp;video=false</a></td><td>Audio only</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=false">audio=false</a></td><td>Video only</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=googEchoCancellation=false,googAutoGainControl=true">audio=googEchoCancellation=false,googAutoGainControl=true</a></td><td>Disable echo cancellation and enable gain control</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=googNoiseReduction=true">audio=googNoiseReduction=true</a></td><td>Enable noise reduction</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?asc=ISAC/16000">asc=ISAC/16000</a></td><td>Set preferred audio send codec to be ISAC at 16kHz (use on Android)</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?arc=opus/48000">arc=opus/48000</a></td><td>Set preferred audio receive codec Opus at 48kHz</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?dtls=false">dtls=false</a></td><td>Disable <a href="https://en.wikipedia.org/wiki/Datagram_Transport_Layer_Security" title="Wikipedia article: DTLS">Datagram Transport Layer Security</a></td></tr>
        <tr><td><a href="https://apprtc.appspot.com?dscp=true">dscp=true</a></td><td>Enable DSCP</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?ipv6=true">ipv6=true</a></td><td>Enable IPv6</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?arbr=48000">arbr=[bitrate]</a></td><td>Set audio receive bitrate, kbps</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?asbr=16000">asbr=[bitrate]</a></td><td>Set audio send bitrate</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?vsbr=1000">vsbr=[bitrate]</a></td><td>Set video receive bitrate</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?vrbr=8000">vrbr=[bitrate]</a></td><td>Set video send bitrate</td></tr>
      </table>

    </section>

    <a href="//github.com/GoogleChrome/webrtc" title="View the repository" id="viewSource">github.com/GoogleChrome/webrtc</a>

  </div>

  <script src="samples/web/js/lib/ga.js"></script>

</body>
</html><commit_msg>fix main.css link in params.html<commit_after><!DOCTYPE html>
<html>
<head>

  <meta name="keywords" content="WebRTC, HTML5, JavaScript" />
  <meta name="description" content="AppRTC URL parameters" />
  <meta http-equiv="Content-Type" content="text/html; charset=utf-8" />
  <meta name="viewport" content="width=device-width, initial-scale=1.0, minimum-scale=1.0">

  <base target="_blank">

  <title>AppRTC parameters</title>

  <link rel="icon" sizes="192x192" href="images/webrtc-icon-192x192.png">
  <link href="//fonts.googleapis.com/css?family=Roboto:300,400,500,700" rel="stylesheet" type="text/css">
  <link rel="stylesheet" href="http://webrtc.github.io/samples/src/css/main.css" />

  <style>
  div#container {
    max-width: 65em;
  }
    h2 {
      font-size: 1.5em;
      font-weight: 500;
    }
    h3 {
      border-top: none;
    }
    section {
      border-bottom: 1px solid #eee;
      margin: 0 0 1.5em 0;
      padding: 0 0 1.5em 0;
    }
    section:last-child {
      border-bottom: none;
      margin: 0;
      padding: 0;
    }
    td {
      padding: 0.5em 1em;
      vertical-align: top;
    }
    td:first-of-type {
      text-align: right;
    }
  </style>
</head>
<body>
  <div id="container">

    <h1>AppRTC parameters</h1>

    <section>

      <p>A number of settings for the <a href="https://apprtc.appspot.com">AppRTC video chat application</a> can be changed by adding URL parameters.</p>

      <p>For example: <a href="https://apprtc.appspot.com/?hd=true&amp;stereo=true&amp;debug=loopback">https://apprtc.appspot.com/?hd=true&amp;stereo=true&amp;debug=loopback</a></p>

      <p>The file using the parameters is <a href="https://github.com/GoogleChrome/webrtc/blob/master/samples/web/content/apprtc/apprtc.py" title="apprtc.py file in AppRTC repo">apprtc.py</a>. More Google-specific parameters are available from the <a href="//code.google.com/p/webrtc/source/browse/trunk/talk/app/webrtc/mediaconstraintsinterface.h" title="mediaconstraintsinterface.h">MediaConstraints interface</a>.</p>

      <p>For more information see <a href="http://gingertech.net/2014/03/19/apprtc-googles-webrtc-test-app-and-its-parameters/">AppRTC : Google's WebRTC test app and its parameters</a>.</p>

    </section>

    <section>

      <table>
        <tr><td><a href="https://apprtc.appspot.com?hd=true">hd=true</a></td><td>Use HD camera resolution <a href="http://googlechrome.github.io/webrtc/samples/web/content/getusermedia/resolution/" title="Video constraints demo">constraints</a>, i.e. minWidth: 1280, minHeight: 720</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?stereo=true">stereo=true</a></td><td>Turn on stereo audio</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?debug=loopback">debug=loopback</a></td><td>Connect to yourself, e.g. to test firewalls</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?ts=https://turn.example.org">ts=[turnserver]</a></td><td>Set TURN server different from the default</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=true&amp;video=false">audio=true&amp;video=false</a></td><td>Audio only</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=false">audio=false</a></td><td>Video only</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=googEchoCancellation=false,googAutoGainControl=true">audio=googEchoCancellation=false,googAutoGainControl=true</a></td><td>Disable echo cancellation and enable gain control</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?audio=googNoiseReduction=true">audio=googNoiseReduction=true</a></td><td>Enable noise reduction</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?asc=ISAC/16000">asc=ISAC/16000</a></td><td>Set preferred audio send codec to be ISAC at 16kHz (use on Android)</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?arc=opus/48000">arc=opus/48000</a></td><td>Set preferred audio receive codec Opus at 48kHz</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?dtls=false">dtls=false</a></td><td>Disable <a href="https://en.wikipedia.org/wiki/Datagram_Transport_Layer_Security" title="Wikipedia article: DTLS">Datagram Transport Layer Security</a></td></tr>
        <tr><td><a href="https://apprtc.appspot.com?dscp=true">dscp=true</a></td><td>Enable DSCP</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?ipv6=true">ipv6=true</a></td><td>Enable IPv6</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?arbr=48000">arbr=[bitrate]</a></td><td>Set audio receive bitrate, kbps</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?asbr=16000">asbr=[bitrate]</a></td><td>Set audio send bitrate</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?vsbr=1000">vsbr=[bitrate]</a></td><td>Set video receive bitrate</td></tr>
        <tr><td><a href="https://apprtc.appspot.com?vrbr=8000">vrbr=[bitrate]</a></td><td>Set video send bitrate</td></tr>
      </table>

    </section>

    <a href="//github.com/GoogleChrome/webrtc" title="View the repository" id="viewSource">github.com/GoogleChrome/webrtc</a>

  </div>
</body>
</html>
{'fraction_non_alphanumeric': 0.20046665370406377, 'fraction_numerical': 0.02274936807310908, 'mean_word_length': 4.264585465711361, 'pattern_counts': {'":': 0, '<': 383, '<?xml version=': 0, '>': 383, 'https://': 47, 'lorem ipsum': 0, 'www.': 0, 'xml': 0}, 'pii_count': 0, 'substrings_counts': 0, 'word_list_counts': {'cursed_substrings.json': 0, 'profanity_word_list.json': 0, 'sexual_word_list.json': 0, 'zh_pornsignals.json': 0}}
{'dir': 'git-commits-cleaned', 'id': '6585814', 'n_tokens_mistral': 3931, 'n_tokens_neox': 3673, 'n_words': 557}