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  • Amélioration de la version de base

    13 septembre 2013

    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
    Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)

  • 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.

  • Gestion des droits de création et d’édition des objets

    8 février 2011, par

    Par défaut, beaucoup de fonctionnalités sont limitées aux administrateurs mais restent configurables indépendamment pour modifier leur statut minimal d’utilisation notamment : la rédaction de contenus sur le site modifiables dans la gestion des templates de formulaires ; l’ajout de notes aux articles ; l’ajout de légendes et d’annotations sur les images ;

Sur d’autres sites (7330)

  • Ffmpeg converted video with image not playing on discord

    16 février 2023, par GiantBooley

    I am trying to convert music8.jpg and music8.mp3 into a video with the audio and image.

    


    I have already tried

    


    ffmpeg -i music8.mp3 -i music8.jpg -pix_fmt yuv420p music8.mp4


    


    It plays fine in windows media player but when I send it in discord, it sees it as corrupted and doesn't play.

    


  • FFmpeg AVI to MP4 transcode not playing fully in Quicktime

    11 août 2014, par Joel Kennedy

    I’m trying to transcode an AVI video to an MP4 stream on the fly using FFmpeg which can be watched on a webpage with a HTML5 video tag. I already know that MP4 isn’t designed for streaming, however it is possible if segmented so I’m still trying to use MP4 as it’s the method that I’ve found is best supported by multiple browsers.

    My stream plays fine in many browsers except for Safari on OS X, which is using Quicktime to play the HTML5 video.

    Quicktime only plays a second of the video, and then stops

    You can see in the image above that Quicktime only plays the video for a second, and then stops. I’ve tested the same video source in Google Chrome on OS X, and it continues playing fine even after a second has elapsed.

    I believe this is something to do with the set Duration in the tkhd and mdhd atoms ? User vbence described that "FFMpeg’s muxer will set the Duration in both tkhd and mdhd atoms to 0xffffffff for each track. This causes problems in some players (for example Quicktime will not play such files). You should find a tool and change it to zero (0x00000000)" in an answer here.

    Is there a way of getting FFmpeg to set a longer duration on the video to get Quicktime to continue playing it instead of stopping ? Because I’m transcoding on the fly, I wouldn’t want to use a separate tool apart from FFmpeg as in my situation I can’t transcode and save the video file, I have to transcode and pipe it to the client on request.

    The command that I’m using with FFmpeg to transcode the video is the following :

    ffmpeg -re -i
    http://www.example.com/video/stream?path=videopathhere -g 52
    -reset_timestamps 1 -vsync 1 -flags global_header -vcodec libx264 -strict -2 -f mp4 -b:v 512k -s 1280x720 -movflags frag_keyframe+empty_moov+faststart pipe :

    Any help or pointers in the right direction would be greatly appreciated.

  • CoreAudio : how to retrieve actual sampling rate of raw data ?

    13 août 2014, par jyavenard

    When attempting to play AAC-HE content in an mp4 container, the reported sampling rate found in the mp4 container appears to be half of the actual sampling rate.

    E.g it appears as 24kHz instead of 48kHz.

    Using the FFmpeg AAC decoder, retrieving the actual sampling rate can be done by simply decoding an audio packet using

    avcodec_decode_audio4

    And looking at AVCodecContext::sample_rate which will be updated appropriately. From that it’s easy to adapt the output.

    With CoreAudio decoder, I would use a AudioConverterRef set the input and output AudioStreamBasicDescription
    and call AudioConverterFillComplexBuffer

    As the converter performs all the required internal conversion including resampling it’s fine. But it plays the content after resampling it to 24kHz (as that’s what the input AudioStreamBasicDescription contains.

    Would there be a way to retrieve the actual sampling rate as found be the decoder (rather than the demuxer) in a similar fashion as one can with FFmpeg ?

    Would prefer to avoid losing audio quality if at all possible, and not downmix data

    Thanks