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Bug de détection d’ogg
22 mars 2013, par
Mis à jour : Avril 2013
Langue : français
Type : Video
Autres articles (53)
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Encodage et transformation en formats lisibles sur Internet
10 avril 2011MediaSPIP transforme et ré-encode les documents mis en ligne afin de les rendre lisibles sur Internet et automatiquement utilisables sans intervention du créateur de contenu.
Les vidéos sont automatiquement encodées dans les formats supportés par HTML5 : MP4, Ogv et WebM. La version "MP4" est également utilisée pour le lecteur flash de secours nécessaire aux anciens navigateurs.
Les documents audios sont également ré-encodés dans les deux formats utilisables par HTML5 :MP3 et Ogg. La version "MP3" (...) -
Personnaliser les catégories
21 juin 2013, parFormulaire de création d’une catégorie
Pour ceux qui connaissent bien SPIP, une catégorie peut être assimilée à une rubrique.
Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire.
Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...) -
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
Sur d’autres sites (12204)
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On ALAC’s Open Sourcing
1er novembre 2011, par Multimedia Mike — Codec TechnologyApple open sourced their lossless audio codec last week. Pretty awesome ! I have a theory that, given enough time, absolutely every codec will be open source in one way or another.
I know I shouldn’t bother reading internet conversation around any news related to multimedia technology. And if I do read it, I shouldn’t waste any effort getting annoyed about them. But here are some general corrections :
- ALAC is not in the same league as — nor is it a suitable replacement for — MP3/AAC/Vorbis or any other commonly used perceptual audio codec. It’s not a matter of better or worse ; they’re just different families of codecs designed for different purposes.
- Apple open sourced ALAC, not AAC– easy mistake, though there’s nothing to ‘open source’ about AAC (though people can, and will, argue about its absolute ‘open-ness’).
- There’s not much technical room to argue between ALAC and FLAC, the leading open source lossless audio compressor. Both perform similarly in terms of codec speeds (screamingly fast) and compression efficiency (results vary slightly depending on source material).
- Perhaps the most frustrating facet is the blithe ignorance about ALAC’s current open source status. While this event simply added an official “open source” status to the codec, ALAC has effectively been open source for a very long time. According to my notes, the ALAC decoding algorithm was reverse engineered in 2005 and added into FFmpeg in March of the same year. Then in 2008, Google — through their Summer of Code program — sponsored an open source ALAC encoder.
From the multimedia-savvy who are versed in these concepts, the conversation revolves around which would win in a fight, ALAC or FLAC ? And who between Apple and FFmpeg/Libav has a faster ALAC decoder ? The faster and more efficient ALAC encoder ? I contend that these issues don’t really matter. If you have any experience working with lossless audio encoders, you know that they tend to be ridiculously fast to both encode and decode and that many different lossless codecs compress at roughly the same ratios.
As for which encoder is the fastest : use whatever encoder is handiest and most familiar, either iTunes or FFmpeg/Libav.
As for whether to use FLAC or ALAC — if you’ve already been using one or the other for years, keep on using it. Support isn’t going to vanish. If you’re deciding which to use for a new project, again, perhaps choose based on software you’re already familiar with. Also, consider hardware support– ALAC enjoys iPod support, FLAC is probably better supported in a variety of non-iPod devices, though that may change going forward due to this open sourcing event.
For my part, I’m just ecstatic that the question of moral superiority based on open source status has been removed from the equation.
Code-wise, I’m interested in studying the official ALAC code to see if it has any corner-case modes that the existing open source decoders don’t yet account for. The source makes mention of multichannel (i.e., greater than stereo) configurations, but I don’t know if that’s in FFmpeg/Libav.
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Encode loopback capture data in mp3 using ffmpeg in VC++
16 octobre 2013, par NetCoder89I'm trying to work on loopback capture(What you hear) and record this file in mp3/aac format using VC++.
>I can capture audio and can create a .wav file i.e. not compressed but I want a compressed file so I'm encoding this through ffmpeg to write an mp3 file not a .wav.
However I'm not getting any way to do it directly ?
I referred this for loopback capture.Please share your experience and opinions.
Thanks !
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FFmpeg not encoding with libx264 library
29 octobre 2011, par bOkeifusHey people of StackOverflow. I have been having a strange issue that I'm not exactly sure what's going on. I am using FFmpeg to convert any incoming video files to h264 mp4 files using the libx264. This is the log that I get from running this line of code :
ffmpeg -y -i vdoname.flv -acodec libfaac -vcodec libx264 -sameq vid.mp4
This is the log output after running the line :
FFmpeg version SVN-r13428, Copyright (c) 2000-2008 Fabrice Bellard, et al.
configuration: --prefix=/usr --disable-static --enable-shared --enable-gpl --enable-nonfree --enable-pthreads --enable-liba52 --enable-liba52bin --enable-libamr-nb --enable-libamr-wb --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --disable-network --disable-ipv6 --disable-ffserver --disable-ffplay
libavutil version: 49.6.0
libavcodec version: 51.57.0
libavformat version: 52.14.0
libavdevice version: 52.0.0
built on Feb 17 2009 09:01:13, gcc: 4.1.2 20071124 (Red Hat 4.1.2-42)
Seems stream 0 codec frame rate differs from container frame rate: 1000.00 (1000/1) -> 49.92 (599/12)
Input #0, flv, from '/html/video/937.flv':
Duration: 00:00:10.08, start: 0.000000, bitrate: 48 kb/s
Stream #0.0: Video: flv, yuv420p, 720x480, 49.92 tb(r)
Stream #0.1: Audio: mp3, 22050 Hz, stereo, 48 kb/s
Output #0, h264, to '/html/flvideo/new_937.mp4':
Stream #0.0: Video: libx264, yuv420p, 720x480, q=2-31, 200 kb/s, 49.92 tb(c)
Stream #0.1: Audio: libfaac, 22050 Hz, stereo, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Stream #0.1 -> #0.1
[libx264 @ 0x11d58b0]using cpu capabilities: MMX MMXEXT SSE SSE2 3DNow!
Press [q] to stop encodingCan someone please help me out and tell me what to do to get this to work ? I guessed that the libx264 external library is not installed but it doesn't exactly say that in the log file and it looks as if it does find it but then doesn't actually encode the video.
Any and all help is greatly appreciated.