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Autres articles (79)

  • Gestion générale des documents

    13 mai 2011, par

    MédiaSPIP ne modifie jamais le document original mis en ligne.
    Pour chaque document mis en ligne il effectue deux opérations successives : la création d’une version supplémentaire qui peut être facilement consultée en ligne tout en laissant l’original téléchargeable dans le cas où le document original ne peut être lu dans un navigateur Internet ; la récupération des métadonnées du document original pour illustrer textuellement le fichier ;
    Les tableaux ci-dessous expliquent ce que peut faire MédiaSPIP (...)

  • Personnaliser les catégories

    21 juin 2013, par

    Formulaire 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 (...)

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

Sur d’autres sites (15224)

  • fate/all : add missing file protocol dependencies

    30 juin, par Nicolas Gaullier
    fate/all : add missing file protocol dependencies

    First, always require file protocol when FATE suite is used.
    
Then, add missing dependencies while removing duplicates.
    

    Signed-off-by : Nicolas Gaullier <nicolas.gaullier@cji.paris>
    
Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
    • [DH] tests/Makefile
    • [DH] tests/fate/cbs.mak
    • [DH] tests/fate/cover-art.mak
    • [DH] tests/fate/demux.mak
    • [DH] tests/fate/filter-audio.mak
    • [DH] tests/fate/filter-video.mak
    • [DH] tests/fate/image.mak
    • [DH] tests/fate/microsoft.mak
    • [DH] tests/fate/mpc.mak
    • [DH] tests/fate/pcm.mak
    • [DH] tests/fate/pixfmt.mak
    • [DH] tests/fate/seek.mak
    • [DH] tests/fate/subtitles.mak
    • [DH] tests/fate/vpx.mak
    • [DH] tests/fate/wavpack.mak

  • How to generate pcm audio file by setting 20 bit depth ?

    6 juillet 2015, par vip007

    No option using ffmpeg

    I am using ffmpeg to generate audio data. I am using following command

    ffmpeg.exe -i in.wav -ar 44100  -acodec pcm_s16le  -ac 1 out.wav

    but there is no option to convert to 20 bit depth pcm audio. (something like pcm_s20le).

    I want to perform some operations on apple codec (e.g. encoding pcm audio data to alac).

    On ALAC source : http://alac.macosforge.org/trac/browser/trunk/ReadMe.txt , it has been mentioned like this ;

    Apple Lossless supports the following features
    1. Bit depths 16, 20, 24 and 32 bits.

    Media Foundation on windows throwing exception for 20 bit depth

    I tried sample encoding and transcoding application mentioned on MSDN https://code.msdn.microsoft.com/windowsapps/media-transcode-sample-f7ba5269

    with following configuration but it failed

    spTypeOut->SetUINT32(MF_MT_AUDIO_BITS_PER_SAMPLE, 20);  

    NOTE : but i put 16 instead of 20 ; it created smoothly.

    Is there any way to generate such 20 bit PCM audio ? What is so special about 20 bit depth that it is not being considered on ffmpeg and media foundation but on apple codec ?

  • Transcode HLS Segments individually using FFMPEG

    27 mai 2013, par rayh

    I 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.ts

    Inspecting an example sound file shows that there is a gap at the end of the audio :

    End

    And the start of the file looks suspiciously attenuated (although this may not be an issue) :

    Start

    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 ?)

    Original Start
    Original End

    ** 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 :

    Side-by-side start
    Side-by-side end

    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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