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  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Configuration spécifique d’Apache

    4 février 2011, par

    Modules spécifiques
    Pour la configuration d’Apache, il est conseillé d’activer certains modules non spécifiques à MediaSPIP, mais permettant d’améliorer les performances : mod_deflate et mod_headers pour compresser automatiquement via Apache les pages. Cf ce tutoriel ; mode_expires pour gérer correctement l’expiration des hits. Cf ce tutoriel ;
    Il est également conseillé d’ajouter la prise en charge par apache du mime-type pour les fichiers WebM comme indiqué dans ce tutoriel.
    Création d’un (...)

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

Sur d’autres sites (6546)

  • How to use x264 for encoding with ffmpeg ?

    21 mai 2014, par mmmaaak

    I tryed to use ffmpeg for encoding video/ But it fails on initialization of AVCodecContext annd AVCodec.
    What I do :

    _codec = avcodec_find_encoder(CODEC_ID_H264);
    _codecContext = avcodec_alloc_context3(_codec);
    _codecContext->coder_type = 0;
    _codecContext->me_cmp|= 1;
    _codecContext->me_method=ME_HEX;
    _codecContext->me_subpel_quality = 0;
    _codecContext->me_range = 16;
    _codecContext->gop_size = 12;
    _codecContext->scenechange_threshold = 40;
    _codecContext->i_quant_factor = 0.71;
    _codecContext->b_frame_strategy = 1;
    _codecContext->qcompress = 0.5;
    _codecContext->qmin = 2;
    _codecContext->qmax = 31;
    _codecContext->max_qdiff = 4;
    _codecContext->max_b_frames = 3;
    _codecContext->refs = 3;
    _codecContext->trellis = 1;
    _codecContext->width = format.biWidth;
    _codecContext->height = format.biHeight;
    _codecContext->time_base.num = 1;
    _codecContext->time_base.den = 30;
    _codecContext->pix_fmt = PIX_FMT_YUV420P;
    _codecContext->chromaoffset = 0;
    _codecContext->thread_count =1;
    _codecContext->bit_rate = (int)(128000.f * 0.80f);
    _codecContext->bit_rate_tolerance = (int) (128000.f * 0.20f);
    int error = avcodec_open2(_codecContext, _codec, NULL);
    if(error<   )
    {
       std::cout<<"Open codec fail. Error "<code>

    In such way ii fails on avopen_codec2() with :

    Program received signal SIGSEGV, Segmentation fault.
    [Switching to Thread 0xae1fdb70 (LWP 30675)]
    0xb2eb2cbb in x264_param_default () from /usr/lib/libx264.so.120

    If i comment all AVCodecContext parameters settins - I have :

    [libx264 @ 0xac75edd0] invalid width x height (0x0)

    And avcodec_open retunrs negative value. Which steps, I’m doing, are wrong ?

    Thanks for any help (ffmpeg 0.10 && libx264 daily snapshot for yesterday)

  • 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.

  • Trying to execute a SPLICE effect with libsox

    4 décembre 2012, par cube

    When two input files and one output file is required for an effect such as the splice effect, how should the sox_open_read() method be called ?

    A typical libsox routine would look something like :

    in=sox_open_read(file_path_in, NULL, NULL, NULL);
       out=sox_open_write(file_path_out, &in->signal, NULL, NULL, NULL, NULL);
    chain = sox_create_effects_chain(&in->encoding, &out->encoding);

    e = sox_create_effect(sox_find_effect("input"));
    args[0] = (char *)in, assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);

    assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);

       e = sox_create_effect(sox_find_effect("speed"));
       args[0] = 5,
       assert(sox_effect_options(e, 1, args) == SOX_SUCCESS);
       assert(sox_add_effect(chain, e, &in->signal, &in->signal) == SOX_SUCCESS);

    However, I want to use the splice effect which requires 2 input files to combine into one output.

    Any help would be appreciated.