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Autres articles (35)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

  • Contribute to a better visual interface

    13 avril 2011

    MediaSPIP is based on a system of themes and templates. Templates define the placement of information on the page, and can be adapted to a wide range of uses. Themes define the overall graphic appearance of the site.
    Anyone can submit a new graphic theme or template and make it available to the MediaSPIP community.

Sur d’autres sites (5672)

  • ffmpeg video download shows errors in log

    15 juin 2022, par PeterM

    i'm trying to download media (video) files from sharepoint (i have view access), but during the process i get several "Connection to tcp ://xxx.xxxxx.ms:443 failed : Error number -138 occurred" or "HTTP error 503 Service Unavailable" errors (see following screenshots : tcp error HTTP error 503 )

    


    i get more errors with the following parameters :

    


    ffmpeg -i "https://theURLtoTheManifestYouCopiedHere" -codec copy downloadedVideo.mp4


    


    less errors with :

    


    ffmpeg -re -vsync 1 -i "https://theURLtoTheManifestYouCopiedHere" -codec copy downloadedVideo.mp4


    


    the options suggested in this article didn't help : https://medium.com/intrasonics/robust-continuous-audio-recording-c1948895bb49

    


    the output video is ok, but sound seems to be missing occasionally (a few seconds worth)

    


    ffmpeg version used is 2022-06-12-git-4d45f5acbd-essentials_build-www.gyan.dev (executable for windows)

    


    any advice ?

    


  • Use ffmpeg to convert FLAC to AAC using the aac_mf encoder

    19 juin 2022, par dstaley

    I'm using ffmpeg to convert audio from FLAC to AAC and I'd like to use the aac_mf encoder, which uses Microsoft Media Foundation. If I use the native AAC encoder built into ffmpeg it works with the following command :

    


    ffmpeg -i input.flac -acodec aac -vn output.m4a


    


    However, the following command gives an error :

    


    > ffmpeg -i input.flac -acodec aac_mf -vn output.m4a
ffmpeg version 5.0.1-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 11.2.0 (Rev7, 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-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --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. 17.100 / 57. 17.100
  libavcodec     59. 18.100 / 59. 18.100
  libavformat    59. 16.100 / 59. 16.100
  libavdevice    59.  4.100 / 59.  4.100
  libavfilter     8. 24.100 /  8. 24.100
  libswscale      6.  4.100 /  6.  4.100
  libswresample   4.  3.100 /  4.  3.100
  libpostproc    56.  3.100 / 56.  3.100
Input #0, flac, from '.\input.flac':
  Duration: 00:04:09.17, start: 0.000000, bitrate: 833 kb/s
  Stream #0:0: Audio: flac, 44100 Hz, stereo, s16
Stream mapping:
  Stream #0:0 -> #0:0 (flac (native) -> aac (aac_mf))
Press [q] to stop, [?] for help
[aac_mf @ 000001bad81c6400] MFT name: 'Microsoft AAC Audio Encoder MFT'
[ipod @ 000001bad81fb640] track 0: codec frame size is not set
Output #0, ipod, to 'output.m4a':
  Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 128 kb/s
    Metadata:
      encoder         : Lavc59.18.100 aac_mf
[aac_mf @ 000001bad81c6400] nb_samples (4096) != frame_size (0)
Audio encoding failed
Conversion failed!


    


  • ffmpeg, how to cut to exactly at the start and end time ? [closed]

    28 mars 2024, par Wayne Julian

    According to this article you can only cut the time, but not specify when to cut from the original video. So the end time will always be wrong. You need to calculate each time on the timeline when your video will end.

    


    e.g. -ss 01:30:40 -to 03:03:05 will cut your video in the wrong order, because it starting the calculation not from the 00:00:00 of the video, but from the -ss 01:30:40

    


    Is there a workaround for this ?

    


    example command :
ffmpeg -ss 00:14:15 -an -i  "2022-05-30.mp4" -to 01:30:40 -crf 23 -c:v libx264 PART1.mp4

    


    I will lose 14:15 minutes here, it is clearly a bug.
I need to extract 14:15 from the end time to get the correct time of the result video. When it comes to multiple video-editing it's so annoying.