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Autres articles (82)

  • Mise à jour de la version 0.1 vers 0.2

    24 juin 2013, par

    Explications des différents changements notables lors du passage de la version 0.1 de MediaSPIP à la version 0.3. Quelles sont les nouveautés
    Au niveau des dépendances logicielles Utilisation des dernières versions de FFMpeg (>= v1.2.1) ; Installation des dépendances pour Smush ; Installation de MediaInfo et FFprobe pour la récupération des métadonnées ; On n’utilise plus ffmpeg2theora ; On n’installe plus flvtool2 au profit de flvtool++ ; On n’installe plus ffmpeg-php qui n’est plus maintenu au (...)

  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

Sur d’autres sites (12132)

  • Screen Transfer [on hold]

    9 décembre 2013, par sanalism

    I need to develop a program that transfers a users screen and sound to another user over internet. The scenario can be visualized like this :

    There is a teacher in front of his computer and writes some code on his computer (ex:visual studio). He has a microphone. There is two students watching teacher's screen and listening his voice from their own computers.

    There are a few alternatives to get screen capture of a user and transfer it like ffmpeg, aforge or self coded screen capture program by c# (over udp). Tried all alternatives.

    However, the problem is all of them are so slow, or creates bad resolution of video. There should be an alternative way, because some programs reaches a high quality. For example teamviewer, windows remote desktop, logmein can send the screen with a high quality. More over, teamviewer can send sound also.

    Where should I begin to overcome this mission ? Which platform, language, protocol is useful ?

  • Extracting audio from video per video frame (C++) [on hold]

    5 septembre 2015, par PaulZyCZ

    Several days ago I had written an interface (in C) to simple DLL library which processes video in OpenCV using Qt5 framework. There is a function which takes frame ID and returns the raw picture data of the frame (so it can be played in Unity using C#). There is however no function to extract corresponding audio. So I am looking for a way to write such a function.

    I am thinking of a way to extract audio from video file (say MP4 with MPEG-4), with length of the corresponding video frame. But so far I have found only tutorials about extracting whole track (via FFMPEG) either to save or play the sound.

    Is there a way to extract only a "frame" (1/fps seconds of audio) ? Or would it be better to create indexes and extract whole track ?

    Thank you for answers.

    PS : Usual questions deal with whole track, not one frame-length sample specified by video frame ID. I have little knowledge of the FFMPEG and it has been several months since I used it last in uni project.

    I have frame, it’s ID, but no sound to it.

  • No Audio when streaming to Youtube from ffmpeg

    11 février 2019, par Sindre Svendby

    So I’m not able to get sound to youtube when streaming with ffmpeg.

    The command I try to run ;

    ffmpeg \
    -f v4l2 \
    -vcodec h264 \
    -video_size 864x480 \
    -r 24 \
    -i /dev/video1 \
    -f alsa \
    -thread_queue_size 1024 \
    -ac 2 \
    -i plughw:CARD=C920,DEV=0 \
    -c:a aac \
    -filter:a "volume=1.5" \
    -b:a 128k \
    -ar 44100 \
    -vcodec copy \
    -b:v 2000k  \
    -r 24 \
    -g 48 \
    -x264opts no-scenecut \
    -bufsize 4096k \
    -maxrate 2048k \
    -f flv \
    rtmp://a.rtmp.youtube.com/live2/STREAM_KODE

    if I switch out the rmtp stream, with a file, like test.flv.
    And then I try to watch this with VLC I do get sound.

    If I check the audio codec in VLC I do see that the codec is aac.
    And as far as I can see from their help pages it is aac that is the correct audio codec to send in.

    I’m not sure how to continue debug this issue, and ideas on this would be great.