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  • Pas question de marché, de cloud etc...

    10 avril 2011

    Le vocabulaire utilisé sur ce site essaie d’éviter toute référence à la mode qui fleurit allègrement
    sur le web 2.0 et dans les entreprises qui en vivent.
    Vous êtes donc invité à bannir l’utilisation des termes "Brand", "Cloud", "Marché" etc...
    Notre motivation est avant tout de créer un outil simple, accessible à pour tout le monde, favorisant
    le partage de créations sur Internet et permettant aux auteurs de garder une autonomie optimale.
    Aucun "contrat Gold ou Premium" n’est donc prévu, aucun (...)

  • 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 (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

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  • Ffmpeg won't copy excerpt of video correctly

    5 mai 2022, par JR Jr.

    I am using a DOS batch to automate and copy excerpts of various high definition videos (mkv's with more than 1GB each).
The script is very convenient and runs fast and fine, but Ffmpeg is not doing its job correctly (it seems Murphy's law is inexorably enthralled into technology, things never come easy, which is why I love and hate it).

    


    Anyway, to cut a long story short, each time the batch job runs, a code like below is executed (pardon me the indiscrete folder name, it's about the 90's TV show, it's not porn !).

    


    call "C:\ffmpeg\bin\ffmpeg.exe" -y -i "D:\100\Sexo\S01\SATC - S01E03 - Bay of Married Pigs.mkv" -ss 00:18:05 -to 00:19:15 -codec copy "002-SATC - S01E03 - Bay of Married Pigs-00_18_05-00_19_15.mp4"


    


    The problem is that the first 6 seconds or so of the resulting video has no video, only audio with a frozen image that only starts to move after about 6 seconds, which is a huge defect, not to mention very annoying (a big let down, after all my meticulous scripting work :(). And this happens for most of the files, except a few ones.

    


    Even though this is copying and changing the format from mkv to mp4, per another thread on this site (https://askubuntu.com/questions/396883/how-to-simply-convert-video-files-i-e-mkv-to-mp4), this is not re-encoding, so this is not the issue. Actually, the same problem occurs even if I don't change the format from mkv to mp4.

    


    Even though I foresee a "there's no way to fix this", let me ask : is there a way to fix this ? Hopefully there is a way.

    


  • runtime error when linking ffmpeg libraries in qt creator

    6 juillet 2012, par dxthegreat

    I'm quite new around here but i hear that if you want a question answered, stackoverflow is the place to ask it >.<. So i hope that my question isn't too trivial that everyone will get annoyed at my lack of research (I've tried googling for two days already D= no progress !)
    I've also asked this question in the Qt forums, but i figured i'd ask here too.

    so...

    For the last few days I’ve been fiddling around with opengl and the like, trying to write a video player.

    However, when i try to import the ffmpeg libraries (avcodec, avformat, avutils etc), an error occurs on runtime (the program compiles fine). When compiled and run in debug mode, the error message gives me only a memory address and error code 135 (DLL not found).

    This error occurs when i include a function from those libraries in my code (e.g. av_register_all()) and it occurs regardless of whether the function is actually called.

    So i’m thinking that I’m doing something wrong when linking these libraries.
    I’m currently using :
    Windows vista (32bit),
    Qt creator 2.4.1 based on Qt 4.7.4 (32bit),
    Zeranoe’s FFmpeg build git-3233ad4 (2012-06-30)

    My .pro file consists of :

    QT       += core gui opengl

    TARGET = test
    TEMPLATE = app


    SOURCES += main.cpp\
           mainwindow.cpp \
       glwidget.cpp

    HEADERS += mainwindow.h \
       glwidget.h \

    FORMS    += mainwindow.ui


    LIBS += -L"$$_PRO_FILE_PWD_/libraries/ffmpeg/libs/" -lavcodec -lavformat -lavutil
    INCLUDEPATH += libraries/ffmpeg/includes

    I’ve tried many variations to the LIBS += line and checked my filepath many times. However, the DLL not found error occurs in all of these variations =(.

    Is there something I’m forgetting when doing these includes ?

    Thanks in advance >.<,
    (young and naive) aspiring dev

  • Why does libmp3lame add zeros to the start of the MP3 ?

    29 mars 2016, par Phlox Midas

    I have a uncompressed .wav file that I turn into a 96k MP3 file :

    ffmpeg.exe -i song.wav -vn -b:a 96000 -ac 2 -ar 48000 -acodec libmp3lame -y song.mp3

    The input file has 637386 samples. The output has 639360 samples. The extra samples in the MP3 are all zeros at the beginning of the file. This happens in every file I’ve translated and with more codecs than just libmp3lame. Is this an FFMPEG bug or a codec bug ? Why are these added ? Is there a way to stop them from being added ?

    Edit : Simplified example and console output :

    ffmpeg.exe -i song.wav -y song.mp3

    ffmpeg version N-55796-gb74213d Copyright (c) 2000-2013 the FFmpeg developers
     built on Aug 26 2013 19:43:51 with gcc 4.7.3 (GCC)
     configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
     libavutil      52. 42.100 / 52. 42.100
     libavcodec     55. 29.100 / 55. 29.100
     libavformat    55. 14.102 / 55. 14.102
     libavdevice    55.  3.100 / 55.  3.100
     libavfilter     3. 82.102 /  3. 82.102
     libswscale      2.  5.100 /  2.  5.100
     libswresample   0. 17.103 /  0. 17.103
     libpostproc    52.  3.100 / 52.  3.100
    Guessed Channel Layout for  Input Stream #0.0 : stereo
    Input #0, wav, from 'song.wav':
     Duration: 00:00:13.28, bitrate: 1538 kb/s
       Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s16, 1536 kb/s
    Output #0, mp3, to 'song.mp3':
     Metadata:
       TSSE            : Lavf55.14.102
       Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s16p
    Stream mapping:
     Stream #0:0 -> #0:0 (pcm_s16le -> libmp3lame)
    Press [q] to stop, [?] for help
    size=     208kB time=00:00:13.29 bitrate= 128.4kbits/s
    video:0kB audio:208kB subtitle:0 global headers:0kB muxing overhead 0.111205%

    Number of samples in wav : 637386

    Number of samples in mp3 : 639984