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• AppRTC : Google’s WebRTC test app and its parameters

  23 juillet 2014, par silvia

  If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

  When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

  We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

  Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

  Here are my favourite parameters :
  

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

  For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

  This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

  AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

  Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

  UPDATE 12 May 2014

  AppRTC now also supports the following bitrate controls :

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

  Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

• How to check if ffmpeg is done encoding a video with PHP ?

  7 juin 2012, par Kevin

  I need the specific code to check with PHP if it is done encoding a video with ffmpeg. I just need a simple true/false check. True if the encoding is done, and false if it is not done. Also the language is PHP, and I do have ffmpeg-php installed.

  More info : I am on Linux.

  Code is below

  convertToFlv( $input, $output );
  
  
  
  function convertToFlv( $input, $output ) {
  
  
  
  echo "Converting $input to $output";
  
  
  
  $command = "ffmpeg -y -i $input -acodec libfaac -ar 44100 -ab 96k -vcodec libx264 -level 41 -crf 20 -bufsize 20000k -maxrate 25000k -g 250 -r 20 -s 640x480 -coder 1 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -flags2 +brdo+dct8x8+bpyramid -me umh -subq 7 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -rc_eq 'blurCplx^(1-qComp)' -bf 16 -b_strategy 1 -bidir_refine 1 -refs 6 -deblockalpha 0 -deblockbeta 0 $output";
  
  
  
  shell_exec( $command );
  
  
  
  }

• NodeMediaServer MP4 change default video resolution 540p

  11 août 2021, par Farooq Zaman

  I have setup a nginx RTMP server and the purpose is to store videos streamed from mobile devices in MP4 format for later analysis. Although mobile devices are streaming videos in 720p resolution NodeMediaServer always store video in 540p resolution. How can I change this behaviour ? Following is NodeMediaServer configuration :

  


  const nodeMediaServerConfig = {
  rtmp: {
    port: 1936,
    chunk_size: 60000,
    gop_cache: true,
    ping: 60,
    ping_timeout: 10,
  },
  http: {
    port: 8000,
    mediaroot: './media',
    allow_origin: '*',
  },
  trans: {
    ffmpeg: '/usr/bin/ffmpeg',
    tasks: [
      {
        app: 'live',
        vcParam: [
          "-c:v",
          "libx264",
          "-vf",
          "scale=720:-1",
          "-b:v",
          "2800k",
          "-bufsize",
          "4200k",
          "-preset",
          "fast",
        ],
        ac: 'aac',
        acParam:["-b:a", "128k", "-ar", 48000],
        mp4: true,
        mp4Flags: '[movflags=faststart]',
      },
    ],
  },
};


  


  Any help in this matter is highly appreciated.