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Autres articles (72)
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List of compatible distributions
26 avril 2011, parThe table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...) -
Gestion des droits de création et d’édition des objets
8 février 2011, parPar défaut, beaucoup de fonctionnalités sont limitées aux administrateurs mais restent configurables indépendamment pour modifier leur statut minimal d’utilisation notamment : la rédaction de contenus sur le site modifiables dans la gestion des templates de formulaires ; l’ajout de notes aux articles ; l’ajout de légendes et d’annotations sur les images ;
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Dépôt de média et thèmes par FTP
31 mai 2013, parL’outil MédiaSPIP traite aussi les média transférés par la voie FTP. Si vous préférez déposer par cette voie, récupérez les identifiants d’accès vers votre site MédiaSPIP et utilisez votre client FTP favori.
Vous trouverez dès le départ les dossiers suivants dans votre espace FTP : config/ : dossier de configuration du site IMG/ : dossier des média déjà traités et en ligne sur le site local/ : répertoire cache du site web themes/ : les thèmes ou les feuilles de style personnalisées tmp/ : dossier de travail (...)
Sur d’autres sites (9366)
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What are good settings for transcoding videos uploaded to my app ?
14 mai 2020, par Dmitry MinkovskyI am working on an app that allows users to share videos. The problem is that many videos are very high bitrate. For example, A 4-minute H264 video from an old iPhone is encoded at 1080p and runs 17,000 kb/s ( 500 megabytes). Accepting and distributing such videos at this bitrate/resolution is not practical for a social application.



I have been playing with ffmpeg to transcode videos to smaller sizes and higher compression, but have not achieved acceptable results. For example :



ffmpeg \
 -i in.mov \
 -vf scale=w='if(gt(iw\,ih)\,780\,-2)':h='if(gt(iw\,ih)\,-2\,780)' \ 
 -c:v libx264 \
 -crf 28 \
 -preset medium \
 -pix_fmt yuv420p \
 -movflags +faststart \
 out.mp4




This command transcodes the above-mentioned 500MB file down to 70MB. It scales the larger dimension of the video to 780 pixels and compresses the video quite a bit. The results are okay, but the file is still large.



Taking the longer dimension down to 480 pixels, the file is reduced to 40MB. Still quite large, and now significantly degraded. Also, the transcoding still takes quite a long time : about 1-1.5x on my 4 year old i7 Macbook Pro with 16GB RAM.



I'm not sure how to improve on this. H265 is not supported in browsers. I am wondering :



- 

- How can I reduce size further ?
- How can I transcode faster than 1x without significantly reducing quality ? Even 2-3x doesn't seem great ?







Is this as good as it gets ?


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How can I improve the up-time of my coffee pot live stream ?
26 avril 2017, par tww0003Some Background on the Project :
Like most software developers I depend on coffee to keep me running, and so do my coworkers. I had an old iPhone sitting around, so I decided to pay homage to the first webcam and live stream my office coffee pot.
The stream has become popular within my company, so I want to make sure it will stay online with as little effort possible on my part. As of right now, it will occasionally go down, and I have to manually get it up and running again.
My Setup :
I have nginx set up on a digital ocean server (my nginx.conf is shown below), and downloaded an rtmp streaming app for my iPhone.
The phone is set to stream to
example.com/live/stream
and then I use an ffmpeg command to take that stream, strip the audio (the live stream is public and I don’t want coworkers to feel like they have to be careful about what they say), and then make it accessible atrtmp://example.com/live/coffee
andexample.com/hls/coffee.m3u8
.Since I’m not too familiar with ffmpeg, I had to google around and find the appropriate command to strip the coffee stream of the audio and I found this :
ffmpeg -i rtmp://localhost/live/stream -vcodec libx264 -vprofile baseline -acodec aac -strict -2 -f flv -an rtmp://localhost/live/coffee
Essentially all I know about this command is that the input stream comes from,
localhost/live/stream
, it strips the audio with-an
, and then it outputs tortmp://localhost/live/coffee
.I would assume that
ffmpeg -i rtmp://localhost/live/stream -an rtmp://localhost/live/coffee
would have the same effect, but the page I found the command on was dealing with ffmpeg, and nginx, so I figured the extra parameters were useful.What I’ve noticed with this command is that it will error out, taking the live stream down. I wrote a small bash script to rerun the command when it stops, but I don’t think this is the best solution.
Here is the bash script :
while true;
do
ffmpeg -i rtmp://localhost/live/stream -vcodec libx264 -vprofile baseline -acodec aac -strict -2 -f flv -an rtmp://localhost/live/coffee
echo 'Something went wrong. Retrying...'
sleep 1
doneI’m curious about 2 things :
- What is the best way to strip audio from an rtmp stream ?
- What is the proper configuration for nginx to ensure that my rtmp stream will stay up for as long as possible ?
Since I have close to 0 experience with nginx, ffmpeg, and rtmp streaming any help, or tips would be appreciated.
Here is my nginx.conf file :
worker_processes 1;
events {
worker_connections 1024;
}
http {
include mime.types;
default_type application/octet-stream;
sendfile on;
keepalive_timeout 65;
server {
listen 80;
server_name localhost;
location / {
root html;
index index.html index.htm;
}
error_page 500 502 503 504 /50x.html;
location = /50x.html {
root html;
}
location /stat {
rtmp_stat all;
rtmp_stat_stylesheet stat.xsl;
allow 127.0.0.1;
}
location /stat.xsl {
root html;
}
location /hls {
root /tmp;
add_header Cache-Control no-cache;
}
location /dash {
root /tmp;
add_header Cache-Control no-cache;
add_header Access-Control-Allow-Origin *;
}
}
}
rtmp {
server {
listen 1935;
chunk_size 4000;
application live {
live on;
hls on;
hls_path /tmp/hls;
dash on;
dash_path /tmp/dash;
}
}
}edit :
I’m also running into this same issue : https://trac.ffmpeg.org/ticket/4401 -
ffmpeg : Continiously encode and append base64 data chunks into output file
11 février 2021, par O.OI have a
.mov
file thats being written into by my iphone cam saved asinput.mov
and I have a script that's reading the currently updating file and I am trying to encode the video and audio codec into a.mkv
container.

I have little knowledge of this tool, but looking at similar Q/A's around
ffmpeg
usage I have found little on using base64 as input. But it is documented by ffmpeg for images, so I assume it is possible and I have also useddata:video/mp4
since these file types are very similar.

I have :


const ifRecordingStream = await fs.readStream('input.mov', 'base64', 4095);
ifRecordingStream.open();

ifRecordingStream.onData((chunk) => 
 execute(`ffmpeg -f concat -i "data:video/mp4;base64,${chunk} -c:v h264 -c:a aac output.mkv")
);



onData()
currently throwsLine {}: unknown keyword {}


Is my command wrong ?


ffmpeg -f concat -i "data:video/mp4;base64,${chunk}" -c:v h264 -c:a aac output.mkv


Any help at all is welcomed.