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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 -
Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...)
Sur d’autres sites (11252)
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nginx RTMP to HLS : FFMPG error when trying multiple bitrate output [closed]
28 mai 2014, par user3685074I’m currently trying to convert my RTMP Livestream into a HLS with 3 quality-settings.
I followed this guide
I’ve compiled my own FFMPEG and it’s working if I just convert 1 file.
It seems libx264 isn’t able to do multiple encodings at the same time ?I’m using these command :
exec /usr/local/bin/ffmpeg -i rtmp://localhost/src/$name
-c:a libfdk_aac -b:a 32k -c:v libx264 -b:v 128K -f flv rtmp://localhost/hls/$name_low
-c:a libfdk_aac -b:a 64k -c:v libx264 -b:v 256K -f flv rtmp://localhost/hls/$name_mid
-c:a libfdk_aac -b:a 128k -c:v libx264 -b:v 512K -f flv rtmp://localhost/hls/$name_hi 2>>/tmp/ffmpeg.log;this is the output :
ffmpeg version N-63519-g61917a1 Copyright (c) 2000-2014 the FFmpeg developers
built on May 28 2014 18:06:42 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-x11grab --enable-libvpx --enable-libmp3lame --enable-librtmp --enable-libspeex --enable-libfdk_aac
libavutil 52. 87.100 / 52. 87.100
libavcodec 55. 65.100 / 55. 65.100
libavformat 55. 41.100 / 55. 41.100
libavdevice 55. 13.101 / 55. 13.101
libavfilter 4. 5.100 / 4. 5.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
Metadata:
Server NGINX RTMP (github.com/arut/nginx-rtmp-module)
width 1280.00
height 720.00
displayWidth 1280.00
displayHeight 720.00
duration 0.00
framerate 25.00
fps 25.00
videodatarate 390.00
videocodecid 0.00
audiodatarate 27.00
audiocodecid 11.00
Input #0, flv, from 'rtmp://localhost/src/test':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Duration: 00:00:00.00, start: 0.080000, bitrate: N/A
Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 399 kb/s, 25 fps, 25 tbr, 1k tbn, 50 tbc
Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 27 kb/s
[libx264 @ 0x5260380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x5260380] profile High, level 3.1
[libx264 @ 0x5260380] 264 - core 142 r2431 f23da7c - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=24 lookahead_threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=128 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
[libx264 @ 0x525a920] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
Output #0, flv, to 'rtmp://localhost/hls/test_low':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Stream #0:0: Video: h264 (libx264), yuv420p, 1280x720, q=-1--1, 128 kb/s, 25 fps, 90k tbn, 25 tbc
Metadata:
encoder : Lavc55.65.100 libx264
Stream #0:1: Audio: aac (libfdk_aac), 16000 Hz, mono, s16, 32 kb/s
Metadata:
encoder : Lavc55.65.100 libfdk_aac
Output #1, flv, to 'rtmp://localhost/hls/test_mid':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Stream #1:0: Video: h264, yuv420p, 1280x720, q=-1--1, 256 kb/s, 25 fps, 90k tbn, 25 tbc
Metadata:
encoder : Lavc55.65.100 libx264
Stream #1:1: Audio: aac, 16000 Hz, mono, s16
Metadata:
encoder : Lavc55.65.100 libfdk_aac
Output #2, flv, to 'rtmp://localhost/hls/test_hi':
Metadata:
Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 25
profile :
level :
Stream #2:0: Video: h264, yuv420p, 1280x720, q=-1--1, 25 fps, 90k tbn, 25 tbc
Metadata:
encoder : Lavc55.65.100 libx264
Stream #2:1: Audio: aac, 16000 Hz, mono, s16
Metadata:
encoder : Lavc55.65.100 libfdk_aac
Stream mapping:
Stream #0:0 -> #0:0 (h264 -> libx264)
Stream #0:1 -> #0:1 (libspeex -> libfdk_aac)
Stream #0:0 -> #1:0 (h264 -> libx264)
Stream #0:1 -> #1:1 (libspeex -> libfdk_aac)
Stream #0:0 -> #2:0 (h264 -> libx264)
Stream #0:1 -> #2:1 (libspeex -> libfdk_aac)
Error while opening encoder for output stream #1:0 - maybe incorrect parameters such as bit_rate, rate, width or heightI hope you can help me and sorry for my bad english.
Greetz
Kevin -
ffmpeg screencasting with webcam
6 septembre 2017, par curiouserI want to use the screen view of the computer in real-time for a fake webcam. I did the video action on the bottom. But the frames like the video (2.34) are forming on top of each other. How can I solve this problem.
https://www.youtube.com/watch?v=6AZRiW3hHrw
Load the module
sudo modprobe v4l2loopback exclusive_caps=1
Find the dummy device
v4l2-ctl --list-devices
Start the virtual-webcam (change "/dev/video1" to reflect your system)
ffmpeg -f x11grab -r 15 -s 1920x1080 -i :0.0+0,0 -vcodec rawvideo -pix_fmt yuv420p -threads 0 -f v4l2 /dev/video1 -
How i can slice audio file with using node.js
11 février 2018, par Maxim Cherevatovtell me please how i can slice audio file with using node.js ? Now i read the documentation for ffmpeg module, but don’t understand how to slice audio file with using this module.
I found this code, but it gives an error error : NaNffmpeg('music/ant.mp3')
.setStartTime('00:00:03')
.setDuration('10')
.output('music/ant.mp3')
.on('end', function(err) {
if(!err)
{
console.log('conversion Done');
}
})
.on('error', function(err){
console.log('error: ', +err);
}).run();
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