
Recherche avancée
Autres articles (35)
-
Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
-
Creating farms of unique websites
13 avril 2011, parMediaSPIP 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 2011MediaSPIP 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 PeterMi'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 dstaleyI'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 JulianAccording 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 the00:00:00
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 extract14:15
from the end time to get the correct time of the result video. When it comes to multiple video-editing it's so annoying.