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Autres articles (86)
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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 (...) -
Librairies et logiciels spécifiques aux médias
10 décembre 2010, parPour un fonctionnement correct et optimal, plusieurs choses sont à prendre en considération.
Il est important, après avoir installé apache2, mysql et php5, d’installer d’autres logiciels nécessaires dont les installations sont décrites dans les liens afférants. Un ensemble de librairies multimedias (x264, libtheora, libvpx) utilisées pour l’encodage et le décodage des vidéos et sons afin de supporter le plus grand nombre de fichiers possibles. Cf. : ce tutoriel ; FFMpeg avec le maximum de décodeurs et (...)
Sur d’autres sites (7617)
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ffmpeg, download video stream from url
1er décembre 2014, par samiullahI am developing application to save any online webinar(video or audio) given the url using ffmpeg library. Url usually contain other contents like text, images as well. So i have problem, how to separately get video stream from url using ffmpeg(or some other better free library). Url may be for any site, not only for youtube, but as an example, link may be like
http://wildkingdom.magnify.net/video/Dangerous-Moments-in-the-Wild-K ;recent or
http://www.youtube.com/watch?v=wnrJJYm7qIw
etc -
Record linux screen (audio/video) and stream to an API
5 juin 2024, par JoelI'm running a browser on a Linux server, and I'm trying to figure out the best way to capture the audio/video of the screen, and stream that with as little latency as possible to an API on another server.


Requirements :


- 

- Stream the audio from server A to server B so that server B can pipe that forward to an online transcription service.
- Stream the audio and video from server A to server B so that server B can store the contents to some kind of blob storage. If the stream is killed for some reason before it's ended, the partial contents should be saved and still be playable.






I've successfully captured the audio/video to disk using
ffmpeg
(see below), but I'm really unsure how I should change that to stream the contents to an API instead.

The API is currently ASP.NET Core 8, but it could be node/python/whatever really.


ffmpeg -y -video_size 1920x1080 -framerate 30 -f x11grab -i :0 -f pulse -i default -c:v libx264 -pix_fmt yuv420p -c:a aac -strict experimental /app/recordings/output.mp4



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Merge audio (m4s) segments into one
20 avril 2022, par akinuriI recently started learning Laravel, and currently watching an online course. Online courses are fine, but I like to have local copies, so I'm trying to download/merge segmented audio from Laracasts : Laravel 8 From Scratch series.


I've written some scripts (in Python) that does the following :


- 

- Download the
master.json
- Read
master.json
and download audio segments - Merge the segments into a single file (the file is not playable yet)
- Process the audio file via
ffmpeg
(now it's playable, but has issues)










I think there's a problem with the step 3 and/or 4.


In step/script 3, I create a new file, and add the contents of the segments to the file in binary.


Then (step/script 4), run a
ffmpeg
command in python :ffmpeg -i merged-file.mp4 -c copy processed-file.mp4


However, the final file doesn't work/play as expected. There's a delay in the beginning, and some parts seem to be cut off/skipped.


There are three possibilities :


- 

- Segment files are problematic (not likely ?)
- I'm doing the merging wrong
- I'm doing the
ffmpeg
processing wrong








Can someone guide me here ?



The issues/colored parts in the
ffmpeg
output are :

...
[mov,mp4,m4a,3gp,3g2,mj2 @ 000001cfbc0de780] could not find corresponding track id 2
[mov,mp4,m4a,3gp,3g2,mj2 @ 000001cfbc0de780] could not find corresponding trex (id 2)
...
[aac @ 000001cfbc0f0380] Number of bands (31) exceeds limit (6).
...
[mp4 @ 000001cfbc20ecc0] track 0: codec frame size is not set
...
[mp4 @ 000001cfbc20ecc0] Non-monotonous DTS in output stream 0:0; previous: 318318, current: 286286; changing to 318319. This may result in incorrect timestamps in the output file.
...



Everything required for a test case is located in GitHub (akinuri/dump/m4s-segments/). Screenshot of the contents :





Note : there are two types/formats of audio in the
master.json
:mp42
anddash
.dash
works as expected, and seem to be used in limited videos/courses. On the other hand,mp42
appears more. So I need a way to makemp42
work.

- Download the