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  • Multilang : améliorer l’interface pour les blocs multilingues

    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
    Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela.

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

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  • Something missing during transcoding audio-only pieces from FLV format to AAC format ?

    21 octobre 2012, par MaiTiano

    I have ten consecutive flv pieces cut from a song in flv format. Each flv piece has about 10 seconds long.

    After I transcode these flv format audio piece into wav format by ffmpeg, I drag the new 10 wav file into foobar player and play them automatically(one-by-one played automatically). I found there is no "broken/pause" feeling in during switching from the end part of previous wav file to the start part of the next wav file.

    However, after I transcode these flv format audio piece into AAC format by ffmpeg, and do the similar listening test in the foobar player, I can hear/feel obvious breakpoint when file changing.

    Is it possible for ffmpeg to miss some audio signal during flv->aac transcoding ? Because, the transcoded aac files are supposed to be played one-by-one contiguously just like there is one file playing, in other words, there should not be the breakpoint feeling happened between two consecutive file playing !

    Any advices ? Many thanks.

  • executable binary cannot run on android marshmallow

    30 décembre 2015, par alijandro

    I built ffmpeg executable binary with shared libraries on Android. But when I run it on Marshmallow, I got the following error, the executable can’t run.

    $ LD_LIBRARY_PATH=./lib ./bin/ffmpeg
    
CANNOT LINK EXECUTABLE: cannot find "libavformat.so" from verneed[0] in DT_NEEDED list for "./bin/ffmpeg"
    
page record for 0xb6eee00c was not found (block_size=16)

    I already added --extra-ldexeflags="-pie" when compiled the binary.

    The executable binary run properly on pre Marshmallow device.

    I didn’t encounter such problem before, did I miss something important ? How can I make this binary work on Marshmallow ?

    More information about my environment.

    I used android-ndk-r10e with

    SYSROOT=$ANDROID_NDK_ROOT/platforms/android-19/arch-arm and toolchains version is

    arm-linux-androideabi-gcc (GCC) 4.9 20140827 (prerelease)

    If I built ffmpeg into a single executable binary (build the static libraries and then build into binary), it run properly.

  • Why low qmax value improve video quality ?

    14 novembre 2013, par theateist

    Maybe my questions doesn't make sense due to not understanding but please explain me what I miss because I did read posts and wiki and still it's not clear to me.

    As I understand setting low value for qmax will improve the quality by increasing the bitrate.
    Maybe I didn't understood something but isn't lowing the Q(quantization) will decrease the quantization levels and thus the bitrate which means degradation in quality ? Or in ffmpeg lowing Q means increasing the quantization levels ? If the last is true so it make sense that lower qmax improves the quality.

    If the above is true, so increasing qmax will decrease the quantization levels which means less bits for coding a quantization level. So, if number of bits for a level is lower, so total bits per frame will be lower, so how the encoder manage to get to the desired bitrate ?

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