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  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • Personnaliser les catégories

    21 juin 2013, par

    Formulaire de création d’une catégorie
    Pour ceux qui connaissent bien SPIP, une catégorie peut être assimilée à une rubrique.
    Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
    On peut modifier ce formulaire dans la partie :
    Administration > Configuration des masques de formulaire.
    Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
    Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

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  • Audacity vocal removal failed when ffmpeg-conversion was involved

    10 mars 2018, par fyang

    I downloaded some songs coded with FLAC, and Audacity could remove the vocals quite well.

    When I downloaded songs coded with ALAC, I must use ffmpeg to convert them to some other forms because Audacity didn’t recognise .m4a files.

    I used the command ffmpeg -i "song 01.m4a" -f flac "song 01.flac". Now Audacity could load the song, but its vocal removal failed to remove the vocals.

    I tried again with this command in order to be precise, ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac", and vocal removal did not work either.

    I tried to do it manually by splitting, inverting and changing both channels to mono, but the vocals were still there.

    I think the problem lies with the ffmpeg conversion step. Is there any fix ? Thanks !

    Below is the result of the conversion :

    ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac"
    
ffmpeg version N-90143-gb6652f5100 Copyright (c) 2000-2018 the FFmpeg developers
    
  built with gcc 7.3.0 (GCC)
    
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libmfx --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
    
  libavutil      56.  7.101 / 56.  7.101
    
  libavcodec     58. 12.102 / 58. 12.102
    
  libavformat    58.  9.100 / 58.  9.100
    
  libavdevice    58.  2.100 / 58.  2.100
    
  libavfilter     7. 12.100 /  7. 12.100
    
  libswscale      5.  0.101 /  5.  0.101
    
  libswresample   3.  0.101 /  3.  0.101
    
  libpostproc    55.  0.100 / 55.  0.100
    
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000019f2b258000] stream 0, timescale not set
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'song 01.m4a':
    
  Metadata:
    
    major_brand     : M4A
    
    minor_version   : 0
    
    compatible_brands: M4A mp42isom
    
    creation_time   : 2009-12-27T00:15:23.000000Z
    
    track           : 1/10
    
    genre           : 
    
    album           : 
    
    artist          : 
    
    comment         : ExactAudioCopy v0.95b4
    
    DISCID          : 
    
    iTunNORM        :  00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
    
    title           : song 01
    
    encoder         : iTunes 9.0.2.25
    
    date            : 2005
    
    album_artist    : 
    
    lyrics          : 
    
  Duration: 00:08:10.84, start: 0.000000, bitrate: 921 kb/s
    
    Stream #0:0(und): Audio: alac (alac / 0x63616C61), 44100 Hz, stereo, s16p, 920 kb/s (default)
    
    Metadata:
    
      creation_time   : 2009-12-27T00:15:23.000000Z
    
    Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 300x300 [SAR 100:100 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (alac (native) -> flac (native))
    
Press [q] to stop, [?] for help
    
[Parsed_pan_0 @ 0000019f2b2a6fc0] Pure channel mapping detected: 0 1
    
Output #0, flac, to 'song 01.flac':
    
  Metadata:
    
    major_brand     : M4A
    
    minor_version   : 0
    
    compatible_brands: M4A mp42isom
    
    lyrics          : 
    
    TRACKNUMBER     : 1/10
    
    genre           : 
    
    album           : 
    
    artist          : 
    
    DESCRIPTION     : ExactAudioCopy v0.95b4
    
    DISCID          : 
    
    iTunNORM        :  00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
    
    title           : song 01
    
    ALBUMARTIST     : 
    
    date            : 2005
    
    encoder         : Lavf58.9.100
    
    Stream #0:0(und): Audio: flac, 44100 Hz, stereo, s16, 128 kb/s (default)
    
    Metadata:
    
      creation_time   : 2009-12-27T00:15:23.000000Z
    
      encoder         : Lavc58.12.102 flac
    
size=   54518kB time=00:08:10.84 bitrate= 909.9kbits/s speed=  35x
    
video:0kB audio:54508kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.018294%

  • Convert audio files to mp3 using ffmpeg [closed]

    5 août, par Hrishikesh -Rishi- Choudhari

    I need to convert audio files to mp3 using ffmpeg.

    



    When I write the command as ffmpeg -i audio.ogg -acodec mp3 newfile.mp3, I get the error :

    



    FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  configuration: 
  libavutil     49.15. 0 / 49.15. 0
  libavcodec    52.20. 1 / 52.20. 1
  libavformat   52.31. 0 / 52.31. 0
  libavdevice   52. 1. 0 / 52. 1. 0
  built on Jun 24 2010 14:56:20, gcc: 4.4.1
Input #0, mp3, from 'ZHRE.mp3':
  Duration: 00:04:12.52, start: 0.000000, bitrate: 208 kb/s
    Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 256 kb/s
Output #0, mp3, to 'audio.mp3':
    Stream #0.0: Audio: 0x0000, 44100 Hz, stereo, s16, 64 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Unsupported codec for output stream #0.0


    



    I also ran this command :

    



     ffmpeg -formats | grep mp3


    



    and got this in response :

    



    FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  configuration: 
  libavutil     49.15. 0 / 49.15. 0
  libavcodec    52.20. 1 / 52.20. 1
  libavformat   52.31. 0 / 52.31. 0
  libavdevice   52. 1. 0 / 52. 1. 0
  built on Jun 24 2010 14:56:20, gcc: 4.4.1
 DE mp3             MPEG audio layer 3
 D A    mp3             MP3 (MPEG audio layer 3)
 D A    mp3adu          ADU (Application Data Unit) MP3 (MPEG audio layer 3)
 D A    mp3on4          MP3onMP4
 text2movsub remove_extra noise mov2textsub mp3decomp mp3comp mjpegadump imxdump h264_mp4toannexb dump_extra


    



    I guess that the mp3 codec isn't installed. Am I on the right track here ?

    


  • finding speed and tone of speech in an audio using python

    1er février 2018, par kRazzy R

    Given an audio , I want to calculate the pace of the speech. i.e how fast or slow is it.

    Currently I am doing the following :

    - convert speech to text and obtaining a transcript (using a free tool).

    - count number of words in transcript.

    - calculate length or duration of file.

    - finally, pace = (number of words in transcript / duration of file).

    However the accuracy of the pace obtained is dependent purely on transcription , which I think is an unnecessary step.

    Is there any python-library/sox/ffmpeg way that will enable me to

    • to calculate, in a straightforward way,the speed/pace of talk in an audio
    • dominant Pitches/tones of that audio ?

    I referred : I referred : http://sox.sourceforge.net/sox.html and https://digitalcardboard.com/blog/2009/08/25/the-sox-of-silence/

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