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Autres articles (45)
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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 -
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 (...) -
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 (...)
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Transcode HLS Segments individually using FFMPEG
27 mai 2013, par rayhI am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).
Here is an example ffmpeg command line :
ffmpeg -threads 1 -nostdin -loglevel verbose \
-nostdin -y -i input.ts -c:a libfdk_aac \
-ac 2 -b:a 64k -y -metadata -vn output.tsInspecting an example sound file shows that there is a gap at the end of the audio :
And the start of the file looks suspiciously attenuated (although this may not be an issue) :
My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.
Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?
** UPDATE 1 **
Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)
** UPDATED 2 **
So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :
I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).
** UPDATE 3 **
According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.
For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.
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Create conversion queue using ffmpeg and C #
22 octobre 2017, par Alexei Agüero AlbaOk, the idea is to create a file queue that can be modified and reorganized (this is done) and for each file execute a ffmpeg process to convert it to another format.
For conversion use Xabe.FFmpeg and .Net 4.5 all using async and await.
The question would be how to execute an x number of processes in parallel (example 4) of that variable queue and when one of them finishes executing the next one, keeping in execution always the same amount in parallel. I can start from scratch but I need ideas on how to do this in the simplest way possible. The program itself is simple (with gui) takes a folder and its subfolders all the video files and queues them and starts the conversion, you can add other folders with more files, and independent files reorder them, to convert whichever is the greater.
At one point I found a package I think nuget (or github) that did exactly what I needed but I have not been able to get back.
Thanks for your help in advance.
Excuse the English because I use the translator of Google for being faster because my domain of this is limited but sufficient to understand the answers.
Ok, I found what I was looking for called ProcessManager is a nupkg package. It has 2 years of development but seems stable. The only drawback is that it does not allow me to organize the conversion queue once you have added the files, although I have to try some ideas that maybe functions.
var manager = new Manager(4); // Max 4 processes will be started simultaneously
manager.Start();
manager.ProcessErrorDataReceived += (sender, e) => Console.WriteLine(e.Data);
manager.ProcessOutputDataReceived += (sender, e) => Console.WriteLine(e.Data);
foreach (var videoFileName in Directory.EnumerateFiles("videos"))
{
var info = new ProcessInfo(
"ffprobe.exe",
string.Format("-v quiet -print_format json -show_format -show_streams \"{0}\"", videoFileName));
manager.Queue(info);
} -
Accessibility support in Ogg and liboggplay
19 février 2010, par silviaAt the recent FOMS/LCA in Wellington, New Zealand, we talked a lot about how Ogg could support accessibility. Technically, this means support for multiple text tracks (subtitles/captions), multiple audio tracks (audio descriptions parallel to main audio track), and multiple video tracks (sign (...)