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Médias (1)
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Somos millones 1
21 juillet 2014, par
Mis à jour : Juin 2015
Langue : français
Type : Video
Autres articles (56)
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Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
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Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
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Changer son thème graphique
22 février 2011, parLe thème graphique ne touche pas à la disposition à proprement dite des éléments dans la page. Il ne fait que modifier l’apparence des éléments.
Le placement peut être modifié effectivement, mais cette modification n’est que visuelle et non pas au niveau de la représentation sémantique de la page.
Modifier le thème graphique utilisé
Pour modifier le thème graphique utilisé, il est nécessaire que le plugin zen-garden soit activé sur le site.
Il suffit ensuite de se rendre dans l’espace de configuration du (...)
Sur d’autres sites (10977)
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How to encode bmp images to H264, H265 losslessly with and without Nvenc
26 août 2020, par Hamza GhizaouiSo, let s assume that i have 5000 BMP images (1920x1080) that i want to encode into a video.


I managed to do that losslessly using FFMPEG using the following 4 encoders :


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- libx264
- libx265
- h264_nvenc
- hevc_nvenc










The problem is that when i try to test my results using PSNR, i get extremly high values, in the 30-70 range. What is happening ?
Is the problem with color space conversion ? (None of the encoders above seem to support RGB, a conversion to YUV occur and I m assuming that a lot of averaging is happening. stuff that PSNR test is very sensetive toward).


If my reasoning is correct, is it possible to encode BMP 100% losslessly. the resuling video will be streamed locally. the network is 10 GBS, so the size isn t that important.


thank you


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Error opening filters : Running ffmpeg through Groovy and ProcessBuilder
30 octobre 2016, par smeebThis question involves the popular Linux media processing utility "
ffmpeg
", but I think its really a Java 8ProcessBuilder
question at its core. Or possibly an issue w/ GroovyGStrings
.The
ffmpeg
command to split an MP4 video into frames/images based on a sampling rate is :ffmpeg -i /some/path/to/video.mp4 -vf "select=not(mod(n\,${samplingRate}))" -vsync vfr -q:v 2 img_%d.jpg
Where
${samplingRate}
is a positive integer and represents the number of frames to skip in between creating each frame. So for instance, if you had a video with 430 frames in it, and wanted to sample every 50th frame, your command would be :ffmpeg -i /some/path/to/video.mp4 -vf "select=not(mod(n\,50))" -vsync vfr -q:v 2 img_%d.jpg
Which would then create 8 (or maybe 9, not gonna do the math for a pretend example) JPG files called
img_1.jpg
,img_2.jpg
, ...etc. The first image would be the 50th frame in the video. The second image would be the 100th frame, etc. I have ran the above command directly from a terminal and confirmed that it is correct and succeeds without errors.I have a Groovy app, and am trying to run this
ffmpeg
command via a Java 8ProcessBuilder
, and am having issues with theselect=
argument. Currently I have :int numSamples = 50
Process frameSamplerProc = new ProcessBuilder(
'ffmpeg',
'-i',
"/home/myuser/some/path/to/video.mp4",
'-vf',
"\"select=not(mod(n\\,${numSamples}))\"",
'-vsync',
'vfr',
'-q:v',
'2',
'sample%d.jpg'
).redirectErrorStream(true).start()
frameSamplerProc.inputStream.eachLine { fsamplerLine ->
println(fsamplerLine)
}When I run thise code, I don’t get any errors, but I do see errors in
ffmpeg
’s output :ffmpeg version N-81995-gd790e48 Copyright (c) 2000-2016 the FFmpeg developers
// Omitting the next ~50 lines of output as it all looks normal
Metadata:
creation_time : 2016-10-27T20:20:01.000000Z
handler_name : SoundHandle
[AVFilterGraph @ 0x3eddc80] No such filter: '"select'
Error opening filters!I’m wondering if I’m not escaping the
select
argument properly or if I’m passing in a bad array to theProcessBuilder
. Can anybody spot where I’m going awry ? -
How to call ffmpeg code to keep same video quality ?
7 mai 2016, par seaguestI am using ffmpeg3 to convert mv.webm to mp4, it works fine when I use the command line.
$ ffprobe mv.webm
ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
configuration: --enable-gpl --enable-shared --enable-libx264
libavutil 55. 24.100 / 55. 24.100
libavcodec 57. 39.100 / 57. 39.100
libavformat 57. 36.100 / 57. 36.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 45.100 / 6. 45.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Input #0, matroska,webm, from 'mv.webm':
Metadata:
encoder : google
Duration: 00:04:56.36, start: 0.000000, bitrate: 836 kb/s
Stream #0:0: Video: vp8, yuv420p, 640x480, SAR 1:1 DAR 4:3, 25 fps, 25 tbr, 1k tbn (default)
Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)
$ ffmpeg -i mv.webm -acodec aac -vcodec libx264 mv1.mp4
$ ffprobe mv1.mp4
ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
configuration: --enable-gpl --enable-shared --enable-libx264
libavutil 55. 24.100 / 55. 24.100
libavcodec 57. 39.100 / 57. 39.100
libavformat 57. 36.100 / 57. 36.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 45.100 / 6. 45.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'n1.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.36.100
Duration: 00:04:56.36, start: 0.000000, bitrate: 628 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 500 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 120 kb/s (default)
Metadata:
handler_name : SoundHandlerI play it with VLC, there is no noise at all.
Here is my first question, in case we don’t specify the bitrate in the ffmpeg command, how are the bitrate of video/audio determined ?
Since I am developping in Golang, I use the framework gmf which calls ffmpeg, I use go to convert the video format without specifying bitrate, then I got the following info :
$ ffprobe mv2.mp4
ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
configuration: --enable-gpl --enable-shared --enable-libx264
libavutil 55. 24.100 / 55. 24.100
libavcodec 57. 39.100 / 57. 39.100
libavformat 57. 36.100 / 57. 36.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 45.100 / 6. 45.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'mv2.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.36.100
Duration: 00:04:56.36, start: 0.000000, bitrate: 540 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480, 385 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 147 kb/s (default)
Metadata:
handler_name : SoundHandlerThe bitrate is event better than mv1.mp4, however when I play it with VLC, I heard a lot of noises.
I tried to increase the audio bitrate (the way we set bitrate is like AVCodecContext->bit_rate = 441000...), but this doesn’t help.
Maybe there is some option that I need to add when calling ffmpeg code to keep the same quality as input source, could anyone help ?