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Des sites réalisés avec MediaSPIP
2 mai 2011, parCette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page. -
(Dés)Activation de fonctionnalités (plugins)
18 février 2011, parPour gérer l’ajout et la suppression de fonctionnalités supplémentaires (ou plugins), MediaSPIP utilise à partir de la version 0.2 SVP.
SVP permet l’activation facile de plugins depuis l’espace de configuration de MediaSPIP.
Pour y accéder, il suffit de se rendre dans l’espace de configuration puis de se rendre sur la page "Gestion des plugins".
MediaSPIP est fourni par défaut avec l’ensemble des plugins dits "compatibles", ils ont été testés et intégrés afin de fonctionner parfaitement avec chaque (...) -
Les autorisations surchargées par les plugins
27 avril 2010, parMediaspip core
autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs
Sur d’autres sites (7761)
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Cutting sections with -ss and -t is not frame accurate
16 avril 2024, par Théo ChampionI'm using
ffmpeg 6.0
to extract small sections from longer videos. According to the doc i can use the-ss
option to specify the start time and the-t
option to specify the duration and this should result in frame accurate cuts (since FFmpeg 2.1).

However, in my testing i found that the cuts are not frame accurate. I'm using the following command :


ffmpeg -ss 10 -i ffmpeg -ss 10 -i https://storage.googleapis.com/klap-assets/Frame%20Counter%20%5B4K%2C%2060%20FPS%5D%20%E2%80%93%200100.mp4 -t 10 -y -vcodec libx264 -acodec aac -movflags +faststart out2.mp4 -t 10 -y -vcodec libx264 -acodec aac -movflags +faststart out.mp4



This result in a video that start at
9:57
(-2 frames) and ends at19:57
(-2 frames) instead of10:00
and20:00
.

Also weird is that ffprobe on the resulting video show its duration being exactly 10 seconds.


FFprobe of the resulting video :


ffprobe version 6.0 Copyright (c) 2007-2023 the FFmpeg developers
 built with Apple clang version 14.0.0 (clang-1400.0.29.202)
 configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/6.0 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --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-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-neon
 libavutil 58. 2.100 / 58. 2.100
 libavcodec 60. 3.100 / 60. 3.100
 libavformat 60. 3.100 / 60. 3.100
 libavdevice 60. 1.100 / 60. 1.100
 libavfilter 9. 3.100 / 9. 3.100
 libswscale 7. 1.100 / 7. 1.100
 libswresample 4. 10.100 / 4. 10.100
 libpostproc 57. 1.100 / 57. 1.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'out2.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf60.3.100
 Duration: 00:00:10.00, start: 0.000000, bitrate: 303 kb/s
 Stream #0:0[0x1](und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, smpte170m/bt470bg/bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 287 kb/s, 60 fps, 60 tbr, 15360 tbn (default)
 Metadata:
 handler_name : ISO Media file produced by Google Inc.
 vendor_id : [0][0][0][0]
 encoder : Lavc60.3.100 libx264
 Stream #0:1[0x2](und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 2 kb/s (default)
 Metadata:
 handler_name : ISO Media file produced by Google Inc.
 vendor_id : [0][0][0][0]



I've done some testing with other videos and results are never totally frame accurate.


What am i missing here ? How can i get frame accurate cuts with FFmpeg ?


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FFMPEG update from 5.0 to 6.0 out_0_0 buffer queued [closed]
16 mai 2023, par KevittoI've been using ffmpeg 5.0 for some time, encoding an audio stream to an rtp server, but since I updated to ffmpeg 6.0 I get this :


[out_0_0 @ 0x55ac187b60] 100 buffers queued in out_0_0, something may be wrong.



Below is the ffmpeg call :


ffmpeg -re -f alsa -i default:CARD:card1 -ac 2 -af aresample=async=1 -acodec libopus -b:a 48000 -f rtp "rtp://127.0.0.1:5002"



And here is the full startup log :


ffmpeg version 549430e Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 10 (Debian 10.2.1-6)
 configuration: --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --extra-libs='-lpthread -lm -latomic' --arch=arm64 --enable-gmp --enable-gpl --enable-libopus --enable-nonfree --enable-version3 --target-os=linux --enable-pthreads --enable-openssl --enable-hardcoded-tables
 libavutil 58. 2.100 / 58. 2.100
 libavcodec 60. 3.100 / 60. 3.100
 libavformat 60. 3.100 / 60. 3.100
 libavdevice 60. 1.100 / 60. 1.100
 libavfilter 9. 3.100 / 9. 3.100
 libswscale 7. 1.100 / 7. 1.100
 libswresample 4. 10.100 / 4. 10.100
 libpostproc 57. 1.100 / 57. 1.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'default:CARD=pisound':
 Duration: N/A, start: 1684250059.973334, bitrate: 1536 kb/s
 Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))
Press [q] to stop, [?] for help
Output #0, rtp, to 'rtp://127.0.0.1:5002':
 Metadata:
 encoder : Lavf60.3.100
 Stream #0:0: Audio: opus, 48000 Hz, stereo, s16, 48 kb/s
 Metadata:
 encoder : Lavc60.3.100 libopus
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 60.3.100
m=audio 5002 RTP/AVP 97
b=AS:48
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1

size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
[out_0_0 @ 0x559f530c60] 100 buffers queued in out_0_0, something may be wrong.
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.93x 
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.797x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.703x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.624x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.562x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.511x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.939x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.866x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.805x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.751x 
size= 10kB time=00:00:05.69 bitrate= 14.8kbits/s speed=0.707x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 13kB time=00:00:05.95 bitrate= 17.8kbits/s speed=0.696x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.939x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.89x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.847x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.806x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.77x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 21kB time=00:00:11.37 bitrate= 14.8kbits/s speed=0.981x 



I tried changing the output to
-f null /dev/null
to see if the rtp was the issue, but I get the same thing. I made sure the user running it was a member to the "audio" group andarecord -l
andaplay -l
both show the card with the right name and information. I even tried to use its hw code instead of the default name, and same issue.