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  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Possibilité de déploiement en ferme

    12 avril 2011, par

    MediaSPIP peut être installé comme une ferme, avec un seul "noyau" hébergé sur un serveur dédié et utilisé par une multitude de sites différents.
    Cela permet, par exemple : de pouvoir partager les frais de mise en œuvre entre plusieurs projets / individus ; de pouvoir déployer rapidement une multitude de sites uniques ; d’éviter d’avoir à mettre l’ensemble des créations dans un fourre-tout numérique comme c’est le cas pour les grandes plate-formes tout public disséminées sur le (...)

Sur d’autres sites (5876)

  • convert lower bitrate mp3 to higher bit rate [closed]

    6 mars 2016, par janeman

    i had 2000 songs in 128 kbps but in my server space was not there so i have compressed those mp3 to 16kbps and found ok quality on my pc but when i was listening those songs on my iphone i found that sound quality is not good so now i wan t to convert those 16kbps songs to 48kbps which i think will give good sound quality so that users cant differentiate that songs are compressed and it will load fast as well .here is the ffmpeg code i have used to convert 16kbps file to 48kbps . i can see that the size of converted file has increased approx 3 fold but not sure whether it will give the better quality mp3 when i increase the bit rate through conversion .please suggest any way to get my file back with good mp3 quality

      <?php  
    
 if ( exec('ffmpeg  -i /pathtomp3folder/test.mp3  -acodec libmp3lame  -ab 48k -ac 1 -ar 44100  /pathtomp3folder/oputput.mp3'))
    

    
   {
    
  echo 'done!!';
    
  }
    

    
  ?>

    i can do below code as well wthout specifying the frequency which is best way to do

       ffmpeg  -i /pathtomp3folder/test.mp3  -acodec libmp3lame  -ab 48k -ac 1   /pathtomp3folder/oputput.mp3
    

    
  {
    
  echo 'done!!';
    
   }

    i dont see much difference in 16kbps and converted file quality however size got increased 3 times in converted file

    what is the minimum kbps we should prefer so that users cant detect the difference and i get high streaming rate as well without buffering .i am targetting mobile device so please suggest about mobile device

    size of 16 kbps file was 745kb and size of 48kbps file is 1840kb

    where i am doing mistake any idea or if anyone can share his experinene it will be a great help .i have stuggled to get my collection of 2000 songs back as i dont have backup and in my pc the quality of mp3 seems to be ok but when i play in iphone it clearly shows that its of low quality

    in my server ffmpeg is installed and working properly so it is easily converting the bit rate .one more thing i want to know when the bit rate is getting changed from 16kbps to 48 kbps what is the changes in quality ???/ as i can clearly see that size is increasing but the quality remains same

  • lavc/vaapi_encode_mpeg2 : fix frame rate calc error when use time_base.

    6 septembre 2017, par Jun Zhao
    lavc/vaapi_encode_mpeg2 : fix frame rate calc error when use time_base.

    fix frame rate calc error when use time_base.
    

    Signed-off-by : Yun Zhou <yunx.z.zhou@intel.com>
    
Signed-off-by : Jun Zhao <jun.zhao@intel.com>
    
Signed-off-by : Mark Thompson <sw@jkqxz.net>
    • [DH] libavcodec/vaapi_encode_mpeg2.c

  • aac : Validate the sbr sample rate before using the value

    11 février 2017, par Luca Barbato
    aac : Validate the sbr sample rate before using the value

    Avoid a floating point exception.
    

    Bug-Id : 1027
    
CC : libav-stable@libav.org
    • [DBH] libavcodec/aacsbr.c

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