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  • Support de tous types de médias

    10 avril 2011

    Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)

  • 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 (...)

  • Changer son thème graphique

    22 février 2011, par

    Le thème graphique ne touche pas à la disposition à proprement dite des éléments dans la page. Il ne fait que modifier l’apparence des éléments.
    Le placement peut être modifié effectivement, mais cette modification n’est que visuelle et non pas au niveau de la représentation sémantique de la page.
    Modifier le thème graphique utilisé
    Pour modifier le thème graphique utilisé, il est nécessaire que le plugin zen-garden soit activé sur le site.
    Il suffit ensuite de se rendre dans l’espace de configuration du (...)

Sur d’autres sites (9806)

  • FFmpeg stdin "output file is empty, nothing was encoded"

    2 février 2023, par brock

    Just trying to stdin and stdout a simple CAF to MP3 conversion. Output looks exactly the same except using stdin does not encode anything. Windows 10. I'm going bananas here. Please advise.

    


    Using - (stdin)...

    


    >type test.caf | ffmpeg -i - -f mp3 - > test.mp3
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.1.1 (GCC) 20190807
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --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-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, caf, from 'pipe:':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: adpcm_ima_qt (ima4 / 0x34616D69), 48000 Hz, stereo, s16p, 384 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_ima_qt (native) -> mp3 (libmp3lame))
Output #0, mp3, to 'pipe:':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
    TSSE            : Lavf58.29.100
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
    Metadata:
      encoder         : Lavc58.54.100 libmp3lame
size=       0kB time=00:00:00.00 bitrate=N/A speed=   0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)


    


    Using -i...

    


    >ffmpeg -i test.caf -f mp3 - > test.mp3
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.1.1 (GCC) 20190807
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --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-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, caf, from 'test.caf':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: adpcm_ima_qt (ima4 / 0x34616D69), 48000 Hz, stereo, s16p, 384 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (adpcm_ima_qt (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'pipe:':
  Metadata:
    approximate duration in seconds: 3.1
    source bit depth: I16
    TSSE            : Lavf58.29.100
    Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
    Metadata:
      encoder         : Lavc58.54.100 libmp3lame
size=      49kB time=00:00:03.12 bitrate= 129.3kbits/s speed=34.4x
video:0kB audio:49kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.246501%


    


    EDIT : Command syntax is ok. Works as expected with a WAV file. I upgraded FFmpeg now I get errors using stdin. So it is the file that is to blame. However, I do find it odd that using -i is fine, but stdin is not.

    


    >ffmpeg -i - -f mp3 - > test.mp3 < test.caf
ffmpeg version 5.1.2-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 12.1.0 (Rev2, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
[caf @ 0000023dc65b1140] skipping CAF chunk: 00000000 ([0][0][0][0]), size 0
    Last message repeated 324 times
[caf @ 0000023dc65b1140] skipping CAF chunk: 00000000 ([0][0][0][0]), size 431131746560
[caf @ 0000023dc65b1140] skipping CAF chunk: 00000000 ([0][0][0][0]), size 173911953188585728
[caf @ 0000023dc65b1140] skipping CAF chunk: 00FFFF02 ([0][255][255][2]), size -7250317618881344622
pipe:: Invalid data found when processing input


    


  • swr_convert is trying to write to an empty buffer. Is this a bug or am I doing something wrong ?

    14 janvier 2023, par MBucari

    I'm using FFmpe's swr_convert to convert AV_SAMPLE_FMT_FLTP audio. I've been successful converting to a different sample format (e.g. AV_SAMPLE_FMT_FLT and AV_SAMPLE_FMT_S16), but I'm running into trouble when I'm trying to keep the AV_SAMPLE_FMT_FLTP sample format but change the sample rate.

    


    When converting AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_FLTP, swr_convert attempts to write to an empty buffer.

    


    I'm using swr_convert to convert from 22050 Hz AV_SAMPLE_FMT_FLTP to 16000 Hz AV_SAMPLE_FMT_FLTP.

    


    I initialized SwrContext like so :

    


     if (swr_alloc_set_opts2(
            &resample_context,
            &pAVContext->ch_layout, AV_SAMPLE_FMT_FLTP, 16000,
            &pAVContext->ch_layout, AV_SAMPLE_FMT_FLTP, 22050, 0, NULL) < 0)
            return ERR_SWR_INIT_FAIL;

        if(swr_init(resample_context) < 0)
            return ERR_SWR_INIT_FAIL;


    


    and when I call it like this, the program tries to write to a null buffer and crashes.

    


            samples_decoded = swr_convert(ctx->pSwrContext,
            &pDecodedAudio, numOutSamples,
            (const uint8_t**)&pDecodedFrame->data, pDecodedFrame->nb_samples);



    


    So far I've traced the problem to swr_convert_internal

    


    if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
       && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
        //Sample format is planar and input format is same as output format
        if(preout==in){
            out_count= FFMIN(out_count, in_count); 
            av_assert0(s->in.planar);
            copy(out, in, out_count);
            return out_count;
        }
        else if(preout==postin) preout= midbuf= postin= out;
        else if(preout==midbuf) preout= midbuf= out;
        else                    preout= out;
    }


    


    That if bit of code assigns out to preout, but out's data is unitialized. Later on FFmpeg tries to write to the uninitialized block.

    


    I've tested this in 5.1 and in the snapshot build, and it crashes both of them.

    


    So, am I doing something wrong, or is this a bug ?

    


  • avfilter/palettegen : rename local variable box_weight to weight

    27 décembre 2022, par Clément Bœsch
    avfilter/palettegen : rename local variable box_weight to weight

    This variable is used only for the running weight (used to reach the
    
target median). The places where we actually need the box weight are
    
changed to use box->weight.
    • [DH] libavfilter/vf_palettegen.c

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