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Richard Stallman et le logiciel libre
19 octobre 2011, par
Mis à jour : Mai 2013
Langue : français
Type : Texte
Autres articles (44)
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Submit bugs and patches
13 avril 2011Unfortunately a software is never perfect.
If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
You may also (...) -
Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)
Sur d’autres sites (6349)
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mxfdec : set audio timebase to 1/samplerate
28 septembre 2013, par Anton Khirnovmxfdec : set audio timebase to 1/samplerate
Fixes sync in some samples (e.g. bugs 7581 and 8374 in VLC).
Based on a commit by Matthieu Bouron <matthieu.bouron@gmail.com>Reported-by : Jean-Baptiste Kempf <jb@videolan.org>
CC : libav-stable@libav.org -
ffmpeg hundreds of videos- to-image as shell script in keyboard maestro
28 juillet 2023, par LeopardiFirst I would like to say that I use MacOS, I am new using ffmpeg and keyboard maestro and have no little to none experience in coding. But I think I did learned a bit is the past few weeks trying to solve the problem I will ask now. Believe me I tried searching for answers online and did a lot of trial and error before coming here to ask my question.
so here is what I am trying to do :


I have a folder (/Users/Documents/clips) with 450 short AVI clips (foto1.avi, foto2.avi...., foto450.avi) from which I would like to extract all frames from. I know how to extract all frames from 1 clip and copy them to an existing directory :


*ffmpeg -i /Users/Documents/clips/foto1.avi /Users/Documents/Frames/Foto1/frame%06d.png *


With their command all frames from the clip Foto1.avi (frame000001.png, frame000002.png...., frame000132.png) are copied to the folder /Foto1.


With keyboard Maestro I created 450 folders named afters the 450 .avi clips.


Is there a way to write a ffmpeg command that will extract the frames of all 450 .avi clips to the folder with the same name as the file ?


I was hoping there would be a way of seeing variables in this kind of way :


ffmpeg -i /Users/Documents/clips/foto%.avi /Users/Documents/Frames/Foto%/frame%06d.png


I really appreciate any help. Thank you !


I tried searching for answers in forums, ffmpeg database. And since I have almost no knowledge on coding sometimes is hard to decipher and understand the meaning of the code.


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Transcode from a live m3u8 using -ss
20 août 2015, par pgmI’m trying to create a VOD hls clip from a live hls stream on adobe media server using ffmpeg and nodejs.
An example of the command I’m using looks like this :
ffmpeg -report -analyzeduration 999999999 -probesize 999999999 -ss 50 -i http://live.m3u8 -y -r 29.97 -threads 0 -hls_list_size 0 -c:v copy -a:v copy streamoutput.m3u8
The problem is the -ss param (start time) is calculating the start time from the live point on the stream, rather than from the first ’ts’ fragment. I’d like to be able to encode inside of a "DVR window," meaning seeking from the beginning of the stream, not from the live point of the stream.
Example : I use the param
-ss 50
and it won’t encode for 50 seconds until the live stream catches up, outputting this in the ffmpeg log :[h264 @ 0000000002beae00] non-existing PPS 0 referenced
[h264 @ 0000000002beae00] non-existing PPS 0 referenced
[h264 @ 0000000002beae00] decode_slice_header errorOnce the live stream catches up to the 50 second delay it begins encoding. It works this way when I use
-ss
as either an input parameter or output parameter.Is there a way to accomplish this ? I’ve noticed that if I leave
-ss
completely out of the command, it will start at the beginning of the stream, but as soon as it’s there, even as a 0, it will start at the "live point."Any help is much appreciated !