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  • 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.
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    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 (...)

  • 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

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
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    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

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  • python subprocess ffmpeg return code = 69

    13 juin 2023, par Tim Chen

    I try to call ffmpeg through the subprocess.run(['ffmpeg', '-i', file_name, output_file_name], capture_output=True, text=True) command in python to convert the audio file incoming from the front end to wav format file. The backend code is as follows, using python+fastapi :

    


    @app.post("/api/upload/convert")
async def convert_upload_file(request: Request, file: UploadFile = File(...)):
    token = uuid.uuid4().hex
    tmpFileName = os.path.join(os.path.dirname(__file__), token)
    with open(tmpFileName, "wb") as buffer:
        buffer.write(await file.read())
    await file.seek(0)
    output_path = tmpFileName + '-output.wav'
    command = ['ffmpeg', '-i', tmpFileName, output_path]
    result = subprocess.run(command, capture_output=True, text=True)


    


    This code usually works, but there are some scenarios where it doesn't work. The audio file is recorded by js code (specifically navigator.mediaDevices.getUserMedia({audio: true})).
The code of the audio recorded in windows chrome can run normally and get the converted wav file, but the audio recorded from ios15 safari for more than 3 seconds cannot be converted, prompting returncode=69. The error message is as follows :

    


    CompletedProcess(args=['ffmpeg', '-i', '5cfb52c503a646bda0f422b517c8014a', '5cfb52c503a646bda0f422b517c8014a-output.wav'], returncode=69, stdout='', stderr="
ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil      56. 70.100 / 56. 70.100
libavcodec     58.134.100 / 58.134.100
libavformat    58. 76.100 / 58. 76.100
libavdevice    58. 13.100 / 58. 13.100
libavfilter     7.110.100 /  7.110.100
libswscale      5.  9.100 /  5.  9.100
libswresample   3.  9.100 /  3.  9.100
libpostproc    55.  9.100 / 55.  9.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '5cfb52c503a646bda0f422b517c8014a':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    creation_time   : 2023-06-11T16:36:53.000000Z
  Duration: 00:00:07.06, start: 0.000000, bitrate: 187 kb/s
  Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 184 kb/s (default)
    Metadata:
      creation_time   : 2023-06-11T16:36:53.000000Z
      handler_name    : Core Media Audio
      vendor_id       : [0][0][0][0]
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '5cfb52c503a646bda0f422b517c8014a-output.wav':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    ISFT            : Lavf58.76.100
  Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s (default)
    Metadata:
      creation_time   : 2023-06-11T16:36:53.000000Z
      handler_name    : Core Media Audio
      vendor_id       : [0][0][0][0]
      encoder         : Lavc58.134.100 pcm_s16le
size=       2kB time=00:00:00.00 bitrate=N/A speed=N/A    
[aac @ 0x55f1f8f19fc0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x55f1f8f19fc0] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x55f1f8f19fc0] If you want to help, upload a sample of this file to https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[aac @ 0x55f1f8f19fc0] Multiple frames in a packet.
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Number of bands (18) exceeds limit (13).
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Prediction is not allowed in AAC-LC.
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.


    


    For the abnormal code, I tried to execute ffmpeg -i input output.wav after fastapi handle request on the command line and subprocess.run(['ffmpeg', '-i', file_name, output_path], capture_output =True, text=True), all succeeded, which means that the final file must be normal, otherwise the subsequent verification work will get the same error.

    


    This confuses me, is there some information I'm missing ?

    


  • ffmpeg unable to convert vhs-captured .ts files [closed]

    4 avril, par fredko

    I'm digitizing vhs tapes with a Hauppauge Colossus capture card and Mediaportal on windows 7. They're captured as .ts files and the .ts files play well, with no sign of corruption.

    


    But I'd like to convert them to .mkv (losslessly from the .ts), and ffmpeg (version 2023-08-28-git-b5273c619d-essentials_build-www.gyan.dev) fails at this. I use

    


    ffmpeg.exe -i test.ts -c copy test.mkv


    


    and get these errors repeated many times

    


    [mpegts @ 00000000003d6d40] Packet corrupt (stream = 0, dts = 46105).
[mpegts @ 00000000003d6d40] Packet corrupt (stream = 0, dts = 49107).
[h264 @ 00000000003fc580] non-existing PPS 0 referenced
    Last message repeated 1 times
[h264 @ 00000000003fc580] decode_slice_header error
[h264 @ 00000000003fc580] no frame!
...etc...
[in#0/mpegts @ 00000000003d6b80] corrupt input packet in stream 0
[mpegts @ 00000000003d6d40] Packet corrupt (stream = 0, dts = 247305).
...etc...


    


    The resulting mkv file is much smaller than the .ts, and displays solid black in mpc-hc. Ffprobe gives the same errors and ends with

    


    Input #0, mpegts, from 'test.ts':
  Duration: 00:24:26.80, start: 0.099956, bitrate: 4486 kb/s
  Program 137 
  Stream #0:0[0x30]: Video: h264 (Main) (HDMV / 0x564D4448), yuv420p(top first),
 720x480 [SAR 10:11 DAR 15:11], 29.97 fps, 29.97 tbr, 90k tbn
  Stream #0:1[0x40]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo,
 fltp, 188 kb/s


    


    Ffmpeg seems to be complaining about corruption, though again the .ts files play fine. Is there some way to use ffmpeg to convert these files to mkv ? Or is the problem with the capture setup ?

    


  • PHP-FFMPEG Waveform is a little noisy, how to fix that ?

    18 avril 2023, par Kranchi

    I made a waveform through sample code :

    


    $waveform = $audio->waveform(2000, 500, array('#FFA500'));
$waveform->save('waveform.png');


    


    But result has some noise in top & bottom of edges :
https://gcdnb.pbrd.co/images/80v6Up6etCrD.png?o=1

    


    So I made a tiny change in line 140 of src\FFMpeg\Media\Waveform.php :

    


    //'showwavespic=colors='.$this->compileColors().':s='.$this->width.'x'.$this->height,
'showwavespic=draw=full:colors='.$this->compileColors().':s='.$this->width.'x'.$this->height,


    


    And problem fixed :
https://gcdnb.pbrd.co/images/KsCFsOujmJ9W.png?o=1

    


    I'm not sure is this a bug or problem in FFMpeg dll file and my question :

    


    is it possible to send the new command or override save method to avoid hacking the original file ? (the save method is using some protected methods)

    


    https://github.com/PHP-FFMpeg/PHP-FFMpeg/blob/master/src/FFMpeg/Media/Waveform.php

    


    Thanks.

    



    


    PHP-FFMpeg : 23 | FFMpeg : 2023-02-27-git-891ed24f77 (Win X64) |
PHP : 8.2