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Autres articles (37)

  • Les formats acceptés

    28 janvier 2010, par

    Les commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
    ffmpeg -codecs ffmpeg -formats
    Les format videos acceptés en entrée
    Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
    Les formats vidéos de sortie possibles
    Dans un premier temps on (...)

  • Contribute to documentation

    13 avril 2011

    Documentation is vital to the development of improved technical capabilities.
    MediaSPIP welcomes documentation by users as well as developers - including : critique of existing features and functions articles contributed by developers, administrators, content producers and editors screenshots to illustrate the above translations of existing documentation into other languages
    To contribute, register to the project users’ mailing (...)

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

Sur d’autres sites (5244)

  • How to compile ffmpeg via Alchemy gcc ?

    3 juillet 2014, par Rella

    So I created ffmpeg configuration file that makes it pure C (platform independent, but only theoretically)

    So my config is simple (0.6.1,0.6.3 tested) :

    ./configure  --disable-doc  --disable-ffplay --disable-ffprobe  --disable-ffserver    --disable-avdevice   --disable-avfilter   --disable-pthreads  --disable-everything --enable-muxer=flv --enable-encoder=flv --enable-encoder=h263  --disable-mmx  --disable-shared   --prefix=bin/  --disable-protocols --disable-network --disable-optimizations --disable-debug  --disable-asm --disable-stripping

    Compiling this on Linux will resolve in 4 libs with totall size of 1 mb.

    But I need to compile ffmpeg with custom compiler (opensource gcc analog called Adobe Alchemy, lets us compile C/c++ into Flash)

    It gives me errors on nearly each and every file during standart Make :

    Array @ARGV missing the @ in argument 1 of shift() at /home/rupert/Downloads/alchemy-ubuntu-v0.5a/achacks/gcc line 218.
    cc1: error: unrecognized command line option "-Wtype-limits"
    cc1: error: unrecognized command line option "-fno-signed-zeros"

    So what shall I do - how to compile ffmpeg (at least smallest part of it) via alchemy ?

    Update
    If we would fix that errors manually (2 in configure.mak and one in alchemy gcc) we would get a really messy and long output like :

    > $ make -w install

    make: Entering directory `/home/rupert/Downloads/ffmpeg-0.6.1'
    AR  libavformat/libavformat.a
    llvm-ld: error opening 'avformat/libavformat.l.bc' for writing!
    llvm-ranlib: Archive file does not exist
    INSTALL libavformat/libavformat.a
    install: cannot stat `libavformat/libavformat.a': No such file or directory
    llvm-ranlib: Archive file does not exist
    AR  libavcodec/libavcodec.a
    llvm-ld: error opening 'avcodec/libavcodec.l.bc' for writing!
    llvm-ranlib: Archive file does not exist
    INSTALL libavcodec/libavcodec.a
    install: cannot stat `libavcodec/libavcodec.a': No such file or directory
    llvm-ranlib: Archive file does not exist
    AR  libswscale/libswscale.a
    llvm-ld: error opening 'swscale/libswscale.l.bc' for writing!
    llvm-ranlib: Archive file does not exist
    INSTALL libswscale/libswscale.a
    install: cannot stat `libswscale/libswscale.a': No such file or directory
    llvm-ranlib: Archive file does not exist
    AR  libavutil/libavutil.a
    llvm-ld: error opening 'avutil/libavutil.l.bc' for writing!
    llvm-ranlib: Archive file does not exist
    INSTALL libavutil/libavutil.a
    install: cannot stat `libavutil/libavutil.a': No such file or directory
    llvm-ranlib: Archive file does not exist
    INSTALL libavformat/avformat.h
    INSTALL libavformat/avio.h
    INSTALL libavformat/libavformat.pc
    INSTALL libavcodec/avcodec.h
    INSTALL libavcodec/avfft.h
    INSTALL libavcodec/dxva2.h
    INSTALL libavcodec/opt.h
    INSTALL libavcodec/vaapi.h
    INSTALL libavcodec/vdpau.h
    INSTALL libavcodec/xvmc.h
    INSTALL libavcodec/libavcodec.pc
    INSTALL libswscale/swscale.h
    INSTALL libswscale/libswscale.pc
    INSTALL libavutil/adler32.h
    INSTALL libavutil/attributes.h
    INSTALL libavutil/avstring.h
    INSTALL libavutil/avutil.h
    INSTALL libavutil/base64.h
    INSTALL libavutil/common.h
    INSTALL libavutil/crc.h
    INSTALL libavutil/error.h
    INSTALL libavutil/fifo.h
    INSTALL libavutil/intfloat_readwrite.h
    INSTALL libavutil/log.h
    INSTALL libavutil/lzo.h
    INSTALL libavutil/mathematics.h
    INSTALL libavutil/md5.h
    INSTALL libavutil/mem.h
    INSTALL libavutil/pixdesc.h
    INSTALL libavutil/pixfmt.h
    INSTALL libavutil/rational.h
    INSTALL libavutil/sha1.h
    INSTALL libavutil/avconfig.h
    INSTALL libavutil/libavutil.pc
    LD  ffmpeg_g
    WARNING: While resolving call to function 'main' arguments were dropped!
    Cannot yet select: 0x8e707e8: i32 = ConstantPool < i64 6881500230622117888> 0
    llc[0x86c7dec]
    make: *** [ffmpeg_g] Error 6
    make: Leaving directory `/home/rupert/Downloads/ffmpeg-0.6.1'
  • C# library for audio resampling that has the same abilities as FFmpeg

    21 avril 2013, par Designation

    I have to use a pure C# solution for resampling audio, which can produce me the exact same results as FFmpeg's audio sampling can.

    FFmpeg first builds some kind of polyphase filter bank, and then uses that for the sampling process (sorry for the vague phrasing, but I'm not too familiar with this topic). According to this brief documentation, the initialization can be customized this way :

    AVResampleContext* av_resample_init(
       int     out_rate,
       int     in_rate,
       int     filter_length,
       int     log2_phase_count,
       int     linear,
       double  cutoff  
       )

    The parameters are :

    • out_rate : output sample rate
    • in_rate : input sample rate
    • filter_length : length of each FIR filter in the filterbank relative to the cutoff freq
    • log2_phase_count : log2 of the number of entries in the polyphase filterbank
    • linear : if 1 then the used FIR filter will be linearly interpolated between the 2 closest, if 0 the closest will be used
    • cutoff : cutoff frequency, 1.0 corresponds to half the output sampling rate

    I'd need to use a C# library that is configurable in the same depth. I've been trying to use NAudio (more specifically, its WaveFormatConversionStream class), but there, I could only set the input and output sample rates, so I didn't get the expected results.

    So, is there a C# lib that could resample with the same settings as FFmpeg can ? Or one that has almost all of these settings or similar ones ? Note : I need a C# solution, not a wrapper !

  • configure : Maintain alphabetical order of components

    24 janvier 2016, par Timothy Gu
    configure : Maintain alphabetical order of components
    

    This is not "pure" alphabetical order since some components are
    intentionally kept together.

    • [DH] configure