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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • 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 (...)

Sur d’autres sites (7726)

  • Python pydub issue : Can't repoen a wav file after it as been reencoded by ffpeg [Silved]

    28 avril 2024, par Trent Tompkins

    Solved by Michael Butscher. Updated code is now in the repo in the fixed branch. Thanks !

    


    I have a python script that reads in all the *.wav files in a directory and saves them in a folder called output after it adjusts their volume.

    


    lower.py :

    


    import os
from pydub import AudioSegment as ass

# get the current working directory
current_working_directory = os.getcwd()
all_files = os.listdir();

 


while len(all_files) > 0:
    file = all_files.pop()
    song = ass.from_wav(file)
    song = song - 16
    song.export('out\\'+file)



    


    https://github.com/tibberous/ChangeWavVolume/tree/fixed

    


    The problem is (and this took me a LONG time to figure out), after the files are output once, if you try to read them again with AudioSegment ::from_wav(file), ffmpeg throws an error saying it can't decode the file. The output wav files play in Windows Media Player, and even have their audio adjusted, but I am guessing there somehow an error either in the file data itself or the file headers.

    


    I put the code on github. Checkout the branch 'fixed'.

    


    Btw, I just installed everything, so I should have the current versions of everything. I am using the Chocolatey package manager for Windows 10. Read README.txt (in progress) to see how I setup my dev environment.

    


    

    

    Python 3.12.2
Chocolatey v2.2.2
C:\Users\moren>C:\ProgramData\chocolatey\bin\ffmpeg
ffmpeg version 7.0-essentials_build-www.gyan.dev Copyright (c) 2000-2024 the FFmpeg developers
  built with gcc 13.2.0 (Rev5, 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-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      59.  8.100 / 59.  8.100
  libavcodec     61.  3.100 / 61.  3.100
  libavformat    61.  1.100 / 61.  1.100
  libavdevice    61.  1.100 / 61.  1.100
  libavfilter    10.  1.100 / 10.  1.100
  libswscale      8.  1.100 /  8.  1.100
  libswresample   5.  1.100 /  5.  1.100
  libpostproc    58.  1.100 / 58.  1.100
Universal media converter

    


    


    



    I opened the wav files the program generated with Media Player, VLC Media Player and Audacity, thinking they might throw an error, but the only thing that can't seem to read the exported wav files is the script that created them !

    


  • lavu/version : fix minor version

    17 mai 2024, par Haihao Xiang
    lavu/version : fix minor version
    

    The latest version should be 59.18.100 since commit 01c5f4ad

    $ git diff 01c5f4ad 1..HEAD doc/APIchanges
    ...
    +200-¸05-10 - xxxxxxxxx - lavu 59.18.100 - cpu.h
    + Add AV_CPU_FLAG_RV_ZVBB.
    +

    Signed-off-by : Haihao Xiang <haihao.xiang@intel.com>

    • [DH] libavutil/version.h
  • Generation thumbnail from video using ffmpeg [closed]

    23 avril 2024, par godvlpr

    Try to create thumbnail from video using ffmpeg&#xA;This command work perfectly :

    &#xA;

    ffmpeg -i test.mp4 -ss 00:00:00 -vframes 1 thumbnail.jpg

    &#xA;

    But I need to push video to ffmpeg from stdin, and find solution with pipe :

    &#xA;

    cat test.mp4 | ffmpeg -f mp4 -i pipe:0  -ss 00:00:00 -vframes 1 thumbnail.jpg

    &#xA;

    But it's not work for me. Error :

    &#xA;

    ffmpeg version 7.0 Copyright (c) 2000-2024 the FFmpeg developers&#xA;  built with Apple clang version 15.0.0 (clang-1500.3.9.4)&#xA;  configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/7.0 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags=&#x27;-Wl,-ld_classic&#x27; --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libharfbuzz --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopenvino --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox --enable-neon&#xA;  libavutil      59.  8.100 / 59.  8.100&#xA;  libavcodec     61.  3.100 / 61.  3.100&#xA;  libavformat    61.  1.100 / 61.  1.100&#xA;  libavdevice    61.  1.100 / 61.  1.100&#xA;  libavfilter    10.  1.100 / 10.  1.100&#xA;  libswscale      8.  1.100 /  8.  1.100&#xA;  libswresample   5.  1.100 /  5.  1.100&#xA;  libpostproc    58.  1.100 / 58.  1.100&#xA;[mov,mp4,m4a,3gp,3g2,mj2 @ 0x14f706460] stream 0, offset 0x30: partial file&#xA;[mov,mp4,m4a,3gp,3g2,mj2 @ 0x14f706460] Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), none, 1920x1080, 3496 kb/s): unspecified pixel format&#xA;Consider increasing the value for the &#x27;analyzeduration&#x27; (0) and &#x27;probesize&#x27; (5000000) options&#xA;Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#x27;pipe:0&#x27;:&#xA;  Metadata:&#xA;    major_brand     : isom&#xA;    minor_version   : 512&#xA;    compatible_brands: isomiso2avc1mp41&#xA;    encoder         : Lavf58.26.100&#xA;  Duration: 00:00:02.01, start: 0.000000, bitrate: N/A&#xA;  Stream #0:0[0x1](und): Video: h264 (avc1 / 0x31637661), none, 1920x1080, 3496 kb/s, 25 fps, 25 tbr, 12800 tbn (default)&#xA;      Metadata:&#xA;        handler_name    : VideoHandler&#xA;        vendor_id       : [0][0][0][0]&#xA;  Stream #0:1[0x2](und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 96 kb/s (default)&#xA;      Metadata:&#xA;        handler_name    : SoundHandler&#xA;        vendor_id       : [0][0][0][0]&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (h264 (native) -> mjpeg (native))&#xA;[mov,mp4,m4a,3gp,3g2,mj2 @ 0x14f706460] stream 0, offset 0x30: partial file&#xA;[in#0/mov,mp4,m4a,3gp,3g2,mj2 @ 0x600002e2c400] Error during demuxing: Invalid data found when processing input&#xA;Cannot determine format of input 0:0 after EOF&#xA;[vf#0:0 @ 0x60000202c240] Task finished with error code: -1094995529 (Invalid data found when processing input)&#xA;[vf#0:0 @ 0x60000202c240] Terminating thread with return code -1094995529 (Invalid data found when processing input)&#xA;[vost#0:0/mjpeg @ 0x14f7080e0] Could not open encoder before EOF&#xA;[vost#0:0/mjpeg @ 0x14f7080e0] Task finished with error code: -22 (Invalid argument)&#xA;[vost#0:0/mjpeg @ 0x14f7080e0] Terminating thread with return code -22 (Invalid argument)&#xA;[out#0/image2 @ 0x60000272c180] Nothing was written into output file, because at least one of its streams received no packets.&#xA;frame=    0 fps=0.0 q=0.0 Lsize=       0KiB time=N/A bitrate=N/A speed=N/A    &#xA;Conversion failed!&#xA;

    &#xA;

    Tried on macos and ubuntu. What can be wrong here and how to fix this issue ? Maybe exists other solutions without pipes, or something else..

    &#xA;