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Autres articles (54)

  • Submit bugs and patches

    13 avril 2011

    Unfortunately a software is never perfect.
    If you think you have found a bug, report it using our ticket system. Please to help us to fix it by providing the following information : the browser you are using, including the exact version as precise an explanation as possible of the problem if possible, the steps taken resulting in the problem a link to the site / page in question
    If you think you have solved the bug, fill in a ticket and attach to it a corrective patch.
    You may also (...)

  • 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 (...)

  • Contribute to translation

    13 avril 2011

    You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
    To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
    MediaSPIP is currently available in French and English (...)

Sur d’autres sites (11422)

  • avcodec/h264_cavlc : Fix undefined behavior on qscale overflow

    22 avril 2017, par Michael Niedermayer
    avcodec/h264_cavlc : Fix undefined behavior on qscale overflow
    

    Fixes : 1214/clusterfuzz-testcase-minimized-6130606599569408

    Found-by : continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
    Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>

    • [DH] libavcodec/h264_cavlc.c
  • avfilter/interlace : change lowpass_line function prototype

    20 avril 2017, par Thomas Mundt
    avfilter/interlace : change lowpass_line function prototype
    

    Signed-off-by : Thomas Mundt <tmundt75@gmail.com>
    Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>

    • [DH] libavfilter/interlace.h
    • [DH] libavfilter/tinterlace.h
    • [DH] libavfilter/vf_interlace.c
    • [DH] libavfilter/vf_tinterlace.c
    • [DH] libavfilter/x86/vf_interlace.asm
    • [DH] libavfilter/x86/vf_interlace_init.c
    • [DH] libavfilter/x86/vf_tinterlace_init.c
  • MediaPlayer miscalculates audio duration. getCurrentPosition() goes back when audio reaches the end

    26 mars 2017, par Kacy

    I’m streaming .ogg files from my server. The MediaPlayer plays the song to completion, but when it reaches the end of the song, getCurrentPosition() returns a time less than the actual duration of the song. For 1 song it’s only off by a second, but for another song it’s off by 13 seconds.

    I have a SeekBar that’s updated every second by calling getCurrentPosition(), and once the bar reaches the end, it actually jumps back a few seconds. And I can no longer use seekTo() into the segment of the song that’s past the miscalculated duration because I receive the error : Attempt to seek to past end of file.

    Chrome’s html video element simply displays the songs with the incorrect durations.

    Other tools like ffmpeg or this app on Windows 10 called PowerMediaPlayer both calculate the songs’ durations perfectly.

    I believe this problem is related to this answer, but the solution assumes ffmpeg also miscalculates the duration, which is not the case here. Using the -vn flag when converting audio files to ogg didn’t change anything.

    What are my options ? Is there a way to get the MediaPlayer to calculate the correct duration ?

    Update :

    Converting to mp3 has no issues (although I used audioBitrate('96k')instead of audioQuality(2), but I require a solution to get ogg files working. Below is how I’m using ffmpeg to perform the conversion. Another thing worth noting is that when I used 'vorbis' instead of 'libvorbis' for the codec the durations were off by less than a second, if at all. However, I must use 'libvorbis' because 'vorbis' is experimental and results in significanlty inconsistent bitrates across different files given the same audio quality parameter. (And it ignores audioBitrate() entirely.)

    ffmpeg( filepath )
       .toFormat( 'ogg' )
       .audioCodec( 'libvorbis' )
       .audioQuality( 2 )
       .output( destination )
       .run();