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Médias (1)
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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
Autres articles (57)
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Configurer la prise en compte des langues
15 novembre 2010, parAccéder à la configuration et ajouter des langues prises en compte
Afin de configurer la prise en compte de nouvelles langues, il est nécessaire de se rendre dans la partie "Administrer" du site.
De là, dans le menu de navigation, vous pouvez accéder à une partie "Gestion des langues" permettant d’activer la prise en compte de nouvelles langues.
Chaque nouvelle langue ajoutée reste désactivable tant qu’aucun objet n’est créé dans cette langue. Dans ce cas, elle devient grisée dans la configuration et (...) -
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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Emballe médias : à quoi cela sert ?
4 février 2011, parCe plugin vise à gérer des sites de mise en ligne de documents de tous types.
Il crée des "médias", à savoir : un "média" est un article au sens SPIP créé automatiquement lors du téléversement d’un document qu’il soit audio, vidéo, image ou textuel ; un seul document ne peut être lié à un article dit "média" ;
Sur d’autres sites (7182)
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ffmpeg concat drops audio frames
5 octobre 2017, par ShaunI have an mp4 file and I want to take two sequential sections of the video out and render them as individual files, later recombining them back into the original video. For instance, with my video
video.mp4
, I can runffmpeg -i video.mp4 -ss 56 -t 4 out1.mp4
ffmpeg -i video.mp4 -ss 60 -t 4 out2.mp4creating
out1.mp4
which contains 00:00:56 to 00:01:00 ofvideo.mp4
, andout2.mp4
which contains 00:01:00 to 00:01:04. However, later I want to be able to recombine them again quickly (i.e., without reencoding), so I use the concat demuxer,ffmpeg -f concat -safe 0 -i files.txt -c copy concat.mp4
where
files.txt
containsfile out1.mp4
file out2.mp4which theoretically should give me back 00:00:56 to 00:01:04 of
video.mp4
, however there are always dropped audio frames where the concatenation occurs, creating a very unpleasant sound artifact, an audio blip, if you will.I have tried using
async
and-af apad
on initially creating the two sections of the video but I am still faced with the same problem, and have not found the solution elsewhere. I have experienced this issue in multiple different use cases, so hopefully this simple example will shed some light on the real problem. -
displaying a baseline h264 frames stream in browsers
6 août 2021, par Thabet SabhaSo, I have a server that receives a live rtsp stream then generates baseline h264 frames using ffmpeg, which then are sent via an rtcDataChannel to browser, and while the frames arrive as intended, I can't figure out a way to display them on my html5 videoElement,
here is a simplified version of my current approach :


const remoteStream = new MediaSource();
myVideoElement.src = window.URL.createObjectURL(remoteStream);

// called when remoteStream.readyState === "open"
let sourceBuffer = remoteStream.addSourceBuffer('video/mp4; codecs="avc1.4d002a"');

// this gets called when ever a new frame is received from the webrtc data channel.
function onFrame(frame) {
 sourceBuffer.appendBuffer(new Uint8Array(frame));

 /*
 console.log(frame) ==> <buffer 00="00" 01="01" 41="41" 9b="9b" a0="a0" 22="22" 80="80" a5="a5" d7="d7" 42="42" ea="ea" 34="34" 14="14" 85="85" ba="ba" bc="bc" 1b="1b" f2="f2" 71="71" 0d="0d" 8b="8b" e1="e1" 3c="3c" 52="52" d5="d5" 8c="8c" ef="ef" c1="c1" 89="89" 10="10" c5="c5" 05="05" 78="78" ee="ee" 1d="1d" 03="03" 8d="8d" 2896="2896" more="more" bytes="bytes">
 */
}
</buffer>


ffmpeg options :


[
 "-rtsp_transport", "tcp",
 "-i", `${rtspCamURL}`, 
 "-framerate", "15",
 "-c:v", "libx264",
 "-vprofile", "baseline",
 "-b:v", "600k",
 "-bufsize", "600k",
 "-pix_fmt", "yuv420p",
 '-tune', 'zerolatency',
 "-preset", "ultrafast",
 "-f", "rawvideo",
 '-'
]; 



ffmpeg stream is then split using NAL delimiter (to generate individual frames) then each frame is sent via the data channel like so :

Buffer.concat([nalDelimiter, frame])
.

I am not sure if i'm missing something as i'm not getting any helpful errors due to the remoteSource closing as soon as the first frame arrives for some reason.


or does the media source just not support raw h264 frames, and if so is there a workaround to solve this issue ? (even if it has to do with changing the ffmpeg params.


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ffmpeg Error : Pattern type 'glob' was selected but globbing is not support ed by this libavformat build
14 septembre 2017, par Aryan NaimI’m trying to convert group of ".jpg" files acting as individual frames into 1 single mpeg video ".mp4"
Example parameters i used :
frame duration = 2 secs
frame rate = 30 fps
encoder = libx264 (mpeg)
input pattern = "*.jpg"
output pattern = video.mp4Based on ffmpeg wiki instructions at (https://trac.ffmpeg.org/wiki/Create%20a%20video%20slideshow%20from%20images), I issued this command :
ffmpeg -framerate 1/2 -pattern_type glob -i "*.jpg" -c:v libx264 -r 30 -pix_fmt yuv420p video.mp4
But I’m getting this error :
[image2 @ 049ab120] Pattern type 'glob' was selected but globbing is not
supported by this libavformat build *.jpg: Function not implementedWhich probably means the API pattern matching commands for my build/version have changed. By the way this my windows 32bit ffmpeg download build (
ffmpeg-20150702-git-03b2b40-win32-static
).How can I choose a group of files using pattern matching using ffmpeg ?