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ANNEXE : Les plugins utilisés spécifiquement pour la ferme
5 mars 2010, parLe site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)
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Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang 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. -
Mediabox : ouvrir les images dans l’espace maximal pour l’utilisateur
8 février 2011, parLa visualisation des images est restreinte par la largeur accordée par le design du site (dépendant du thème utilisé). Elles sont donc visibles sous un format réduit. Afin de profiter de l’ensemble de la place disponible sur l’écran de l’utilisateur, il est possible d’ajouter une fonctionnalité d’affichage de l’image dans une boite multimedia apparaissant au dessus du reste du contenu.
Pour ce faire il est nécessaire d’installer le plugin "Mediabox".
Configuration de la boite multimédia
Dès (...)
Sur d’autres sites (9521)
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FFMPEG-Php installation on Amazon EC2 linux AMI
5 mai 2015, par HamzaI have installed FFMPEG on Amazon EC2 Linux AMI using this script now I am trying to install FFMPEG-PHP and after
./configure
it gives me this error onmake
command# make
/bin/sh /opt/ffmpeg-php-0.6.0/libtool --mode=compile cc -I. -I/opt/ffmpeg-php-0.6.0 -DPHP_ATOM_INC -I/opt/ffmpeg-php-0.6.0/include -I/opt/ffmpeg-php-0.6.0/main -I/opt/ffmpeg-php-0.6.0 -I/usr/include/php -I/usr/include/php/main -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext -I/usr/include/php/ext/date/lib -I/usr/local/include/ffmpeg -DHAVE_CONFIG_H -g -O2 -Wall -fno-strict-aliasing -c /opt/ffmpeg-php-0.6.0/ffmpeg-php.c -o ffmpeg-php.lo
libtool: compile: cc -I. -I/opt/ffmpeg-php-0.6.0 -DPHP_ATOM_INC -I/opt/ffmpeg-php-0.6.0/include -I/opt/ffmpeg-php-0.6.0/main -I/opt/ffmpeg-php-0.6.0 -I/usr/include/php -I/usr/include/php/main -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext -I/usr/include/php/ext/date/lib -I/usr/local/include/ffmpeg -DHAVE_CONFIG_H -g -O2 -Wall -fno-strict-aliasing -c /opt/ffmpeg-php-0.6.0/ffmpeg-php.c -fPIC -DPIC -o .libs/ffmpeg-php.o
In file included from /opt/ffmpeg-php-0.6.0/ffmpeg-php.c:42:0:
/usr/local/include/ffmpeg/avcodec.h:43:21: fatal error: version.h: No such file or directory
#include "version.h"I have searched this a lot but couldn’t manage to do it, Amazon support officer asked me to use CENTOS AMI instead of Linux AMI but I have everything hosted there, so I will appreciate if someone can help me with this.
Thank you,
Hamza
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Converting a rtsp stream to hls but got error "Non-monotonous DTS in output stream 0:0"
8 mai 2023, par DeviI'm converting a rtsp live camera stream to hls using ffmpeg, it is all good the first few hours, each segment file sizes around
1.1M
, which match the setting ofsegment_time
of 5 seconds. However, when I check back a few days later, the size of each segment file became20G
, and I found that some error are logged.

My current implementation is like this :


ffmpeg -fflags nobuffer \
 -rtsp_transport tcp \
 -i rtsp://abc.com/live.sdp \
 -fps_mode 0 \
 -copyts \
 -c copy \
 -movflags frag_keyframe+empty_moov \
 -hls_flags delete_segments+append_list \
 -f segment \
 -segment_list_flags live \
 -segment_time 5 \
 -segment_list_size 3 \
 -segment_format mpegts \
 -segment_list /app/mount/test/live.m3u8 \
 -segment_list_type m3u8 \
 -segment_wrap 10 \
 /app/mount/test/%04d.ts \
 2>&1 



Here's the logged error :


[segment @ 0x7fe5123101c0] Non-monotonous DTS in output stream 0:0; previous: 14179898636, current: 11332822107; changing to 14179898637. This may result in incorrect timestamps in the output file.
[segment @ 0x7fe5123101c0] stream:0 start_pts_time:157550 pts:14179898637 pts_time:157554 dts:14179898637 dts_time:157554 -> pts:14179898637 pts_time:157554 dts:14179898637 dts_time:157554
[NULL @ 0x7fe510744880] unknown SEI type 229
[segment @ 0x7fe5123101c0] Non-monotonous DTS in output stream 0:0; previous: 14179898637, current: 11332823907; changing to 14179898638. This may result in incorrect timestamps in the output file.
[segment @ 0x7fe5123101c0] stream:0 start_pts_time:157550 pts:14179898638 pts_time:157554 dts:14179898638 dts_time:157554 -> pts:14179898638 pts_time:157554 dts:14179898638 dts_time:157554
[segment @ 0x7fe5123101c0] Non-monotonous DTS in output stream 0:0; previous: 14179898638, current: 11332825707; changing to 14179898639. This may result in incorrect timestamps in the output file.
[segment @ 0x7fe5123101c0] stream:0 start_pts_time:157550 pts:14179898639 pts_time:157554 dts:14179898639 dts_time:157554 -> pts:14179898639 pts_time:157554 dts:14179898639 dts_time:157554



I've tried :


- 

- Changing the option
-fps_mode
withpassthrough
,vfr
,drop
. - Changing the option
-c
with-c copy
,-c:a copy -c:v libx264
.






None of the above works for me.


- Changing the option
-
ffmpeg rtsp stream to YouTube livestream not doing anything
29 mai 2018, par felixosthI’m using C# to initiate ffmpeg. I’ve a onvif bridge server for a CCTV VMS and I’m building a application to enable the user to livestream any CCTV camera to YouTube.
The rtsp stream to the camera looks like this :
rtsp ://onvif:bridge@localhost:554/live/xxxxx-xxxguidtocameraxxx-xxxxx
I’m new to ffmpeg and I’m using this arg :
-f lavfi -i anullsrc -rtsp_transport udp -i camerastreamurl -tune zerolatency -vcodec libx264 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp ://x.rtmp.youtube.com/live2/streamkey -loglevel debug
It seems like ffmpeg isn’t finding the camera stream, only the fake audio one. It just freezes.
This is the result of the debug log :
ffmpeg version N-91172-gebf85d3190 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 7.3.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
libavutil 56. 18.102 / 56. 18.102
libavcodec 58. 19.104 / 58. 19.104
libavformat 58. 17.100 / 58. 17.100
libavdevice 58. 4.100 / 58. 4.100
libavfilter 7. 24.100 / 7. 24.100
libswscale 5. 2.100 / 5. 2.100
libswresample 3. 2.100 / 3. 2.100
libpostproc 55. 2.100 / 55. 2.100
Splitting the commandline.
Reading option '-f' ... matched as option 'f' (force format) with argument 'lavfi'.
Reading option '-i' ... matched as input url with argument 'anullsrc'.
Reading option '-rtsp_transport' ... matched as AVOption 'rtsp_transport' with argument 'udp'.
Reading option '-i' ... matched as input url with argument 'rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b'.
Reading option '-tune' ... matched as AVOption 'tune' with argument 'zerolatency'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
Reading option '-pix_fmt' ... matched as option 'pix_fmt' (set pixel format) with argument '+'.
Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'aac'.
Reading option '-strict' ...Routing option strict to both codec and muxer layer
matched as AVOption 'strict' with argument 'experimental'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
Reading option 'rtmp://x.rtmp.youtube.com/live2/xxxxxxxx' ... matched as output url.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input url anullsrc.
Applying option f (force format) with argument lavfi.
Successfully parsed a group of options.
Opening an input file: anullsrc.
detected 8 logical cores
[AVFilterGraph @ 0000027a34bad7c0] query_formats: 2 queried, 3 merged, 0 already done, 0 delayed
[Parsed_anullsrc_0 @ 0000027a34badb80] sample_rate:44100 channel_layout:'stereo' nb_samples:1024
[lavfi @ 0000027a34babc80] All info found
Input #0, lavfi, from 'anullsrc':
Duration: N/A, start: 0.000000, bitrate: 705 kb/s
Stream #0:0, 1, 1/44100: Audio: pcm_u8, 44100 Hz, stereo, u8, 705 kb/s
Successfully opened the file.
Parsing a group of options: input url rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
Successfully parsed a group of options.
Opening an input file: rtsp://onvif:bridge@localhost:554/live/41cf4f34-e137-4559-8278-47d912c64c5b.
[tcp @ 0000027a34bb5980] No default whitelist set