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  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

  • Pas question de marché, de cloud etc...

    10 avril 2011

    Le vocabulaire utilisé sur ce site essaie d’éviter toute référence à la mode qui fleurit allègrement
    sur le web 2.0 et dans les entreprises qui en vivent.
    Vous êtes donc invité à bannir l’utilisation des termes "Brand", "Cloud", "Marché" etc...
    Notre motivation est avant tout de créer un outil simple, accessible à pour tout le monde, favorisant
    le partage de créations sur Internet et permettant aux auteurs de garder une autonomie optimale.
    Aucun "contrat Gold ou Premium" n’est donc prévu, aucun (...)

  • Dépôt de média et thèmes par FTP

    31 mai 2013, par

    L’outil MédiaSPIP traite aussi les média transférés par la voie FTP. Si vous préférez déposer par cette voie, récupérez les identifiants d’accès vers votre site MédiaSPIP et utilisez votre client FTP favori.
    Vous trouverez dès le départ les dossiers suivants dans votre espace FTP : config/ : dossier de configuration du site IMG/ : dossier des média déjà traités et en ligne sur le site local/ : répertoire cache du site web themes/ : les thèmes ou les feuilles de style personnalisées tmp/ : dossier de travail (...)

Sur d’autres sites (10758)

  • How to decode mp3 to pcm by ffmpeg

    30 janvier 2017, par Meph-

    I need decode mp3 audio data to pcm. I have data which starts with mp3 header. Api-example.c doesn’t work, output is strange :

    enter image description here

    command ffmpeg -i input.mp3 output.wav
    is great, this is what i need. But I cant find way how to do that in code. Does anybody know, where some tutorial with ffmpeg library is ? Thanks

    Edit 2.7.13 :

    Hi again,
    I rebuilt the audio decode example method from ffmpeg and my problem is probably here :

    len = avcodec_decode_audio4(avCodecContext,avFrame, &got_frame,&avPacket);    
    int data_size = av_samples_get_buffer_size(NULL,avFrame->channels,avFrame->nb_samples,AV_SAMPLE_FMT_S16P,1);

    data_size is size of data frame from decoder, it depends on number of channels, number of data samples and data type(my data are 16bit PCM stereo encoded to mp3 to 1152 samples of mp3 frame)

    If I open an output file in audacity, correct parameters, which give correct output, are stereo (right), 8bit pcm (wrong) and half sample rate (also wrong), what’s it happened ?

    data before encoding :
    16bit PCM 44100Hz, stereo

    data after decoding :
    8bit PCM 22050Hz, stereo ---> ???!!!

    I’m tired of this....

  • bash variable changes in loop with ffmpeg

    17 septembre 2018, par Mike

    I wrote a skript to quickly create short preview clips from vides I recorded on timestamps that I found worth checking out later for cutting.
    My file with the timestamps is written like this

    FILE_NAME1#MM:SS MM:SS
    FILE_NAME2#MM:SS MM:SS MM:SS MM:SS

    example :

    MAH01728#02:47 03:34 03:44 05:00 06:08 06:55

    The script looks like this :

    #!/bin/bash
    while read f
    do

    file=$(echo $f | cut -d"#" -f1)
    filename=${file}".MP4"
    timestamps=$(echo $f | cut -d"#" -f2)

    for time in $timestamps
    do
     ffmpeg -ss 00:${time}.0 -i "orig/${filename}" -c copy -t 10 "preview/${file}_${time}.MP4"
    done
    done < $1

    The script gets half of the previews that I want and on the other the filename is messed up and ffmpeg complains that the file is not found :

    orig/714.MP4: No such file or directory
    orig/00:58 01:25.MP4: No such file or directory

    So I modified the script for trouble shooting and just put an echo in front of the ffmpeg command - now all file names are correct. What am I missing ?

    ffmpeg -ss 00:01:47.0 -i orig/MAH01714.MP4 -c copy -t 10 preview/MAH01714_01:47.MP4
    ffmpeg -ss 00:02:00.0 -i orig/MAH01713.MP4 -c copy -t 10 preview/MAH01713_02:00.MP4
    ffmpeg -ss 00:00:58.0 -i orig/MAH01712.MP4 -c copy -t 10 preview/MAH01712_00:58.MP4
    ffmpeg -ss 00:01:25.0 -i orig/MAH01712.MP4 -c copy -t 10 preview/MAH01712_01:25.MP4
  • Converting variable bitrate on the fly with ffmpeg

    4 septembre 2016, par Łukasz Bezwerchny

    i do cut some video files with either avidemux or virtualdub using directstream copy, some of the files comes with variable bitrate, the problem is that after such cutting my default video cataloging software shows those files lenght as 0:0, i managed to fix this problem with ffmpeg using "-vcodec copy -acodec copy". The output seem to be fine now but i have another problem, these new files have sound problem, on pot player everything is working fine but on mpc and my cataloging software which also plays media sound gets cut off at the half of the movie, for example 2h movie has only 30min of sound. I did look at the using mediainfo software and it shows me something like : video lenght 1:55, audio lenght 32min. I think the problem is the vbr, i did manage to solve this problem by extracting mp3 file from the troublesome video and convert it to 128kb constant bitrate and again merge the video and new audio file and it seems fine. Video and audio match at the whole video lenght. It’s just a bit of tedious to get things done for a big amount of files to fix, is there a parameter that i could use in one command line to do the job in one process, i mean like :
    ffmpeg -i test.avi -vcodec copy -"convert vbr to 128kb" "save to test2.avi" ?