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Collections - Formulaire de création rapide
19 février 2013, par
Mis à jour : Février 2013
Langue : français
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Les Miserables
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Mis à jour : Février 2013
Langue : English
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Ne pas afficher certaines informations : page d’accueil
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Mis à jour : Novembre 2011
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The Great Big Beautiful Tomorrow
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Mis à jour : Octobre 2011
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Richard Stallman et la révolution du logiciel libre - Une biographie autorisée (version epub)
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Mis à jour : Octobre 2011
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Rennes Emotion Map 2010-11
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Mis à jour : Juillet 2013
Langue : français
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Autres articles (97)
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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 (...) -
Installation en mode ferme
4 février 2011, parLe mode ferme permet d’héberger plusieurs sites de type MediaSPIP en n’installant qu’une seule fois son noyau fonctionnel.
C’est la méthode que nous utilisons sur cette même plateforme.
L’utilisation en mode ferme nécessite de connaïtre un peu le mécanisme de SPIP contrairement à la version standalone qui ne nécessite pas réellement de connaissances spécifique puisque l’espace privé habituel de SPIP n’est plus utilisé.
Dans un premier temps, vous devez avoir installé les mêmes fichiers que l’installation (...) -
Installation en mode standalone
4 février 2011, parL’installation de la distribution MediaSPIP se fait en plusieurs étapes : la récupération des fichiers nécessaires. À ce moment là deux méthodes sont possibles : en installant l’archive ZIP contenant l’ensemble de la distribution ; via SVN en récupérant les sources de chaque modules séparément ; la préconfiguration ; l’installation définitive ;
[mediaspip_zip]Installation de l’archive ZIP de MediaSPIP
Ce mode d’installation est la méthode la plus simple afin d’installer l’ensemble de la distribution (...)
Sur d’autres sites (7583)
-
ffmpeg : How can a MOV with transparent background be created ?
25 mars 2017, par MatI’m trying - with no success at all - to convert the green pixels of a background into transparent ones and output the result as clip with ffmpeg. N.b. I do not want to lay the clip over anything ; I’m not having a problem with that. What I want is a clip with transparent background for the OpenShot video editor (the chromakey filter of which doesn’t work satisfyingly).
What I have tried (amongst 1 zillion other things over the last 15 hrs.) was
ffmpeg.exe -i in.mov -vf chromakey=0x008001:0.115:0.0 -c:v qtrle out.mov
but the pixels simply would not be transparent. Seemingly, nothing happens. I reckon the filter is ok, because it works fine in a complex chain (overlaying a background image).
The output of ffprompt -show_stream -show_format of out.mov is as follows :
[STREAM]
index=0
codec_name=qtrle
codec_long_name=QuickTime Animation (RLE) video
profile=unknown
codec_type=video
codec_time_base=1/30
codec_tag_string=rle
codec_tag=0x20656c72
width=1920
height=1080
coded_width=1920
coded_height=1080
has_b_frames=0
sample_aspect_ratio=1:1
display_aspect_ratio=16:9
pix_fmt=bgra
level=-99
color_range=N/A
color_space=unknown
color_transfer=unknown
color_primaries=unknown
chroma_location=unspecified
field_order=progressive
timecode=N/A
refs=1
id=N/A
r_frame_rate=30/1
avg_frame_rate=30/1
time_base=1/15360
start_pts=0
start_time=0.000000
duration_ts=54789
duration=3.566992
bit_rate=822383192
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=107
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:language=eng
TAG:handler_name=DataHandler
TAG:encoder=Lavc57.64.101 qtrle
[/STREAM]
[STREAM]
index=1
codec_name=aac
codec_long_name=AAC (Advanced Audio Coding)
profile=LC
codec_type=audio
codec_time_base=1/44100
codec_tag_string=mp4a
codec_tag=0x6134706d
sample_fmt=fltp
sample_rate=44100
channels=2
channel_layout=stereo
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/44100
start_pts=926
start_time=0.020998
duration_ts=157481
duration=3.570998
bit_rate=132103
max_bit_rate=132103
bits_per_raw_sample=N/A
nb_frames=153
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:language=eng
TAG:handler_name=DataHandler
[/STREAM]
[FORMAT]
filename=out.mov
nb_streams=2
nb_programs=0
format_name=mov,mp4,m4a,3gp,3g2,mj2
format_long_name=QuickTime / MOV
start_time=0.000000
duration=3.567000
size=366708874
bit_rate=822447712
probe_score=100
TAG:major_brand=qt
TAG:minor_version=512
TAG:compatible_brands=qt
TAG:encoder=Lavf57.56.101
[/FORMAT]I have a "sample" clip which shows the behaviour I want, with the following stream and information :
[STREAM]
index=0
codec_name=qtrle
codec_long_name=QuickTime Animation (RLE) video
profile=unknown
codec_type=video
codec_time_base=1/24
codec_tag_string=rle
codec_tag=0x20656c72
width=1920
height=1080
coded_width=1920
coded_height=1080
has_b_frames=0
sample_aspect_ratio=0:1
display_aspect_ratio=0:1
pix_fmt=bgra
level=-99
color_range=N/A
color_space=unknown
color_transfer=unknown
color_primaries=unknown
chroma_location=unspecified
field_order=progressive
timecode=N/A
refs=1
id=N/A
r_frame_rate=24/1
avg_frame_rate=24/1
time_base=1/12288
start_pts=0
start_time=0.000000
duration_ts=74760
duration=6.083984
bit_rate=49226848
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=146
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:language=eng
TAG:handler_name=DataHandler
TAG:encoder=Lavc57.24.102 qtrle
[/STREAM]
[STREAM]
index=1
codec_name=aac
codec_long_name=AAC (Advanced Audio Coding)
profile=LC
codec_type=audio
codec_time_base=1/48000
codec_tag_string=mp4a
codec_tag=0x6134706d
sample_fmt=fltp
sample_rate=48000
channels=2
channel_layout=stereo
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/48000
start_pts=0
start_time=0.000000
duration_ts=293856
duration=6.122000
bit_rate=53537
max_bit_rate=128000
bits_per_raw_sample=N/A
nb_frames=288
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
DISPOSITION:timed_thumbnails=0
TAG:language=eng
TAG:handler_name=DataHandler
[/STREAM]
[FORMAT]
filename=templateOK.mov
nb_streams=2
nb_programs=0
format_name=mov,mp4,m4a,3gp,3g2,mj2
format_long_name=QuickTime / MOV
start_time=0.000000
duration=6.144000
size=37478506
bit_rate=48800138
probe_score=100
TAG:major_brand=qt
TAG:minor_version=512
TAG:compatible_brands=qt
TAG:encoder=Lavf57.25.100
[/FORMAT]and I simply am not able to spot the relevant difference.
The input, output and the working template can be found here.
(The security issue you might see when clicking the link comes from the server certificate being self-signed. You can accept a temporal exception. Btw : The ridiculous file size of the output file will be the next nut to crack. Probably something about compression.)
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fluent ffmpeg size output option not working
19 janvier 2017, par AshburySummary : I’m trying to limit output to 3mb,
.outputOptions('-fs 3000000')
isn’t working for me, the file is coming back with a size of119260428
or 119mb.Here is the code to try for yourself, all you need is a
test.mp3
large enough that the resultingtestoutput.ogg
is > 3mb :var ffmpeg = require("fluent-ffmpeg");
var command = ffmpeg();
var convertToOGG = function(){
var fileName = 'test.mp3'
ffmpeg.ffprobe(fileName, function(err, metadata) {
command
.input(fileName)
.inputFormat("mp3")
.audioChannels(1)
.outputOptions('-fs', 3000000)
.output('testoutput.ogg')
.on("progress", function(progress) {
console.log("Processing: " + progress.timemark);
})
.on("error", function(err, stdout, stderr) {
console.log("Cannot process video: " + err.message);
})
.on("end", function(stdout, stderr) {
ffmpeg.ffprobe('testoutput.ogg', function(err,metadata){
if(metadata.format.size >= 3000000){
console.log("didn't work")
}
})
})
.run();
});
};
convertToOGG();Per the fluent-ffmpeg documentation you should be able to use a ffmpeg command in an output option :
outputOption()
This method allows passing any output-related option to ffmpeg. You can call it with a single argument to pass a single option, optionnaly
with a space-separated parameter :/* Single option */
ffmpeg('/path/to/file.avi').outputOptions('-someOption');and in FFMPEG’s documentation :
-fs limit_size (output) Set the file size limit, expressed in bytes. No further chunk of bytes is written after the limit is exceeded. The
size of the output file is slightly more than the requested file size.It’s giving me no errors, just seemingly ignoring the file size limit of 99mb and outputting a 119.3mb file.
Edit - Looks like
-fs 3000000
is working for mp3 to wav, but still wont do mp3 to ogg. This is the output from running the command in terminal :✗ ffmpeg -i test.mp3 -fs 3000000 testoutput.ogg
ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
libavutil 55. 34.100 / 55. 34.100
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.100 / 57. 56.100
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
[mp3 @ 0x7fc6a4000000] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'test.mp3':
Metadata:
lyrics-eng : xxx
title : xxx
artist : xxx
album_artist : xxx
album : xxx
genre : xxx
Duration: 03:27:28.74, start: 0.000000, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, mono, s16p, 128 kb/s
Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 540x360, 90k tbr, 90k tbn, 90k tbc
Metadata:
title : Array
comment : Cover (front)
[swscaler @ 0x7fc6a4808800] deprecated pixel format used, make sure you did set range correctly
[ogg @ 0x7fc6a3815800] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
Output #0, ogg, to 'testoutput.ogg':
Metadata:
lyrics-eng : xxx
title : xxx
artist : xxx
album_artist : xxx
album : xxx
genre : xxx
encoder : Lavf57.56.100
Stream #0:0: Video: theora (libtheora), yuv444p, 540x360, q=2-31, 200 kb/s, 90k fps, 90k tbn, 90k tbc
Metadata:
title : Array
DESCRIPTION : Cover (front)
encoder : Lavc57.64.101 libtheora
lyrics-eng : xxx
artist : xxx
ALBUMARTIST : xxx
album : xxx
genre : xxx
Stream #0:1: Audio: vorbis (libvorbis), 44100 Hz, mono, fltp
Metadata:
encoder : Lavc57.64.101 libvorbis
lyrics-eng : xxx
title : xxx
artist : xxx
ALBUMARTIST : xxx
album : xxx
genre : xxx
Stream mapping:
Stream #0:1 -> #0:0 (mjpeg (native) -> theora (libtheora))
Stream #0:0 -> #0:1 (mp3 (native) -> vorbis (libvorbis))
Press [q] to stop, [?] for help
frame= 1 fps=0.0 q=-0.0 Lsize= 116465kB time=03:27:28.71 bitrate= 76.6kbits/s speed=61.2x
video:9kB audio:114907kB subtitle:0kB other streams:0kB global headers:6kB muxing overhead: 1.347787% -
Audio recorded with MediaRecorder on Chrome missing duration
27 octobre 2016, par suppp111I am recording audio (oga/vorbis) files with MediaRecorder. When I record these file through Chrome I get problems : I cannot edit the files on ffmpeg and when I try to play them on Firefox it says they are corrupt (they do play fine on Chrome though).
Looking at their metadata on ffmpeg I get this :
Input #0, matroska,webm, from '91.oga':
Metadata:
encoder : Chrome
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
[STREAM]
index=0
codec_name=opus
codec_long_name=Opus (Opus Interactive Audio Codec)
profile=unknown
codec_type=audio
codec_time_base=1/48000
codec_tag_string=[0][0][0][0]
codec_tag=0x0000
sample_fmt=fltp
sample_rate=48000
channels=1
channel_layout=mono
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/1000
start_pts=0
start_time=0.000000
duration_ts=N/A
duration=N/A
bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=1
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
TAG:language=eng
[/STREAM]
[FORMAT]
filename=91.oga
nb_streams=1
nb_programs=0
format_name=matroska,webm
format_long_name=Matroska / WebM
start_time=0.000000
duration=N/A
size=7195
bit_rate=N/A
probe_score=100
TAG:encoder=ChromeAs you can see there are problems with the duration. I have looked at posts like this :
How can I add predefined length to audio recorded from MediaRecorder in Chrome ?But even trying that, I got errors when trying to chop and merge files.For example when running :
ffmpeg -f concat -i 89_inputs.txt -c copy final.oga
I get a lot of this :
[oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57612, current: 1980; changing to 57613. This may result in incorrect timestamps in the output file.
[oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57613, current: 2041; changing to 57614. This may result in incorrect timestamps in the output file.
DTS -442721849179034176, next:42521 st:0 invalid dropping
PTS -442721849179034176, next:42521 invalid dropping st:0
[oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57614, current: 2041; changing to 57615. This may result in incorrect timestamps in the output file.
[oga @ 00000000006789c0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
DTS -442721849179031296, next:42521 st:0 invalid dropping
PTS -442721849179031296, next:42521 invalid dropping st:0Does anyone know what we need to do to audio files recorded from Chrome for them to be useful ? Or is there a problem with my setup ?
Recorder js :
if (navigator.getUserMedia) {
console.log('getUserMedia supported.');
var constraints = { audio: true };
var chunks = [];
var onSuccess = function(stream) {
var mediaRecorder = new MediaRecorder(stream);
record.onclick = function() {
mediaRecorder.start();
console.log(mediaRecorder.state);
console.log("recorder started");
record.style.background = "red";
stop.disabled = false;
record.disabled = true;
var aud = document.getElementById("audioClip");
start = aud.currentTime;
}
stop.onclick = function() {
console.log(mediaRecorder.state);
console.log("Recording request sent.");
mediaRecorder.stop();
}
mediaRecorder.onstop = function(e) {
console.log("data available after MediaRecorder.stop() called.");
var audio = document.createElement('audio');
audio.setAttribute('controls', '');
audio.setAttribute('id', 'audioClip');
audio.controls = true;
var blob = new Blob(chunks, { 'type' : 'audio/ogg; codecs="vorbis"' });
chunks = [];
var audioURL = window.URL.createObjectURL(blob);
audio.src = audioURL;
sendRecToPost(blob); // this just send the audio blob to the server by post
console.log("recorder stopped");
}