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  • 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 ;

  • Activation de l’inscription des visiteurs

    12 avril 2011, par

    Il est également possible d’activer l’inscription des visiteurs ce qui permettra à tout un chacun d’ouvrir soit même un compte sur le canal en question dans le cadre de projets ouverts par exemple.
    Pour ce faire, il suffit d’aller dans l’espace de configuration du site en choisissant le sous menus "Gestion des utilisateurs". Le premier formulaire visible correspond à cette fonctionnalité.
    Par défaut, MediaSPIP a créé lors de son initialisation un élément de menu dans le menu du haut de la page menant (...)

  • Supporting all media types

    13 avril 2011, par

    Unlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)

Sur d’autres sites (7069)

  • Trouble syncing libavformat/ffmpeg with x264 and RTP

    26 décembre 2012, par Jacob Peddicord

    I've been working on some streaming software that takes live feeds
    from various kinds of cameras and streams over the network using
    H.264. To accomplish this, I'm using the x264 encoder directly (with
    the "zerolatency" preset) and feeding NALs as they are available to
    libavformat to pack into RTP (ultimately RTSP). Ideally, this
    application should be as real-time as possible. For the most part,
    this has been working well.

    Unfortunately, however, there is some sort of synchronization issue :
    any video playback on clients seems to show a few smooth frames,
    followed by a short pause, then more frames ; repeat. Additionally,
    there appears to be approximately a 4-second delay. This happens with
    every video player I've tried : Totem, VLC, and basic gstreamer pipes.

    I've boiled it all down to a somewhat small test case :

    #include
    #include
    #include
    #include
    #include <libavformat></libavformat>avformat.h>
    #include <libswscale></libswscale>swscale.h>

    #define WIDTH       640
    #define HEIGHT      480
    #define FPS         30
    #define BITRATE     400000
    #define RTP_ADDRESS "127.0.0.1"
    #define RTP_PORT    49990

    struct AVFormatContext* avctx;
    struct x264_t* encoder;
    struct SwsContext* imgctx;

    uint8_t test = 0x80;


    void create_sample_picture(x264_picture_t* picture)
    {
       // create a frame to store in
       x264_picture_alloc(picture, X264_CSP_I420, WIDTH, HEIGHT);

       // fake image generation
       // disregard how wrong this is; just writing a quick test
       int strides = WIDTH / 8;
       uint8_t* data = malloc(WIDTH * HEIGHT * 3);
       memset(data, test, WIDTH * HEIGHT * 3);
       test = (test &lt;&lt; 1) | (test >> (8 - 1));

       // scale the image
       sws_scale(imgctx, (const uint8_t* const*) &amp;data, &amp;strides, 0, HEIGHT,
                 picture->img.plane, picture->img.i_stride);
    }

    int encode_frame(x264_picture_t* picture, x264_nal_t** nals)
    {
       // encode a frame
       x264_picture_t pic_out;
       int num_nals;
       int frame_size = x264_encoder_encode(encoder, nals, &amp;num_nals, picture, &amp;pic_out);

       // ignore bad frames
       if (frame_size &lt; 0)
       {
           return frame_size;
       }

       return num_nals;
    }

    void stream_frame(uint8_t* payload, int size)
    {
       // initalize a packet
       AVPacket p;
       av_init_packet(&amp;p);
       p.data = payload;
       p.size = size;
       p.stream_index = 0;
       p.flags = AV_PKT_FLAG_KEY;
       p.pts = AV_NOPTS_VALUE;
       p.dts = AV_NOPTS_VALUE;

       // send it out
       av_interleaved_write_frame(avctx, &amp;p);
    }

    int main(int argc, char* argv[])
    {
       // initalize ffmpeg
       av_register_all();

       // set up image scaler
       // (in-width, in-height, in-format, out-width, out-height, out-format, scaling-method, 0, 0, 0)
       imgctx = sws_getContext(WIDTH, HEIGHT, PIX_FMT_MONOWHITE,
                               WIDTH, HEIGHT, PIX_FMT_YUV420P,
                               SWS_FAST_BILINEAR, NULL, NULL, NULL);

       // set up encoder presets
       x264_param_t param;
       x264_param_default_preset(&amp;param, "ultrafast", "zerolatency");

       param.i_threads = 3;
       param.i_width = WIDTH;
       param.i_height = HEIGHT;
       param.i_fps_num = FPS;
       param.i_fps_den = 1;
       param.i_keyint_max = FPS;
       param.b_intra_refresh = 0;
       param.rc.i_bitrate = BITRATE;
       param.b_repeat_headers = 1; // whether to repeat headers or write just once
       param.b_annexb = 1;         // place start codes (1) or sizes (0)

       // initalize
       x264_param_apply_profile(&amp;param, "high");
       encoder = x264_encoder_open(&amp;param);

       // at this point, x264_encoder_headers can be used, but it has had no effect

       // set up streaming context. a lot of error handling has been ommitted
       // for brevity, but this should be pretty standard.
       avctx = avformat_alloc_context();
       struct AVOutputFormat* fmt = av_guess_format("rtp", NULL, NULL);
       avctx->oformat = fmt;

       snprintf(avctx->filename, sizeof(avctx->filename), "rtp://%s:%d", RTP_ADDRESS, RTP_PORT);
       if (url_fopen(&amp;avctx->pb, avctx->filename, URL_WRONLY) &lt; 0)
       {
           perror("url_fopen failed");
           return 1;
       }
       struct AVStream* stream = av_new_stream(avctx, 1);

       // initalize codec
       AVCodecContext* c = stream->codec;
       c->codec_id = CODEC_ID_H264;
       c->codec_type = AVMEDIA_TYPE_VIDEO;
       c->flags = CODEC_FLAG_GLOBAL_HEADER;
       c->width = WIDTH;
       c->height = HEIGHT;
       c->time_base.den = FPS;
       c->time_base.num = 1;
       c->gop_size = FPS;
       c->bit_rate = BITRATE;
       avctx->flags = AVFMT_FLAG_RTP_HINT;

       // write the header
       av_write_header(avctx);

       // make some frames
       for (int frame = 0; frame &lt; 10000; frame++)
       {
           // create a sample moving frame
           x264_picture_t* pic = (x264_picture_t*) malloc(sizeof(x264_picture_t));
           create_sample_picture(pic);

           // encode the frame
           x264_nal_t* nals;
           int num_nals = encode_frame(pic, &amp;nals);

           if (num_nals &lt; 0)
               printf("invalid frame size: %d\n", num_nals);

           // send out NALs
           for (int i = 0; i &lt; num_nals; i++)
           {
               stream_frame(nals[i].p_payload, nals[i].i_payload);
           }

           // free up resources
           x264_picture_clean(pic);
           free(pic);

           // stream at approx 30 fps
           printf("frame %d\n", frame);
           usleep(33333);
       }

       return 0;
    }

    This test shows black lines on a white background that
    should move smoothly to the left. It has been written for ffmpeg 0.6.5
    but the problem can be reproduced on 0.8 and 0.10 (from what I've tested so far). I've taken some shortcuts in error handling to make this example as short as
    possible while still showing the problem, so please excuse some of the
    nasty code. I should also note that while an SDP is not used here, I
    have tried using that already with similar results. The test can be
    compiled with :

    gcc -g -std=gnu99 streamtest.c -lswscale -lavformat -lx264 -lm -lpthread -o streamtest

    It can be played with gtreamer directly :

    gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink

    You should immediately notice the stuttering. One common "fix" I've
    seen all over the Internet is to add sync=false to the pipeline :

    gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink sync=false

    This causes playback to be smooth (and near-realtime), but is a
    non-solution and only works with gstreamer. I'd like to fix the
    problem at the source. I've been able to stream with near-identical
    parameters using raw ffmpeg and haven't had any issues :

    ffmpeg -re -i sample.mp4 -vcodec libx264 -vpre ultrafast -vpre baseline -b 400000 -an -f rtp rtp://127.0.0.1:49990 -an

    So clearly I'm doing something wrong. But what is it ?

  • Simple example of using ffmpeg to convert a movie into a format that can be viewed on Apple products

    16 janvier 2014, par ensnare

    I have a directory of AVI and MOV movies taken from my digital camera. I'd like to write a Python script to go through them, and for every AVI or MOV it finds, generate a file in place that can be viewed on Apple products (iPad, Apple TV, etc ...). My videos are 1080p, and I'm interested in preserving as much of the original quality as possible.

    Any examples of how to do this with ffmpeg ? Specifically, flags to pass to the executable ? Would I want to do a one or two-pass encoding (what's the difference ?)

    I'm hoping to do the scripting part in Python.

  • Revision 62702 : On affiche les boutons supplémentaires que si l’utilisateur a le droit de ...

    19 juin 2012, par kent1@… — Log

    On affiche les boutons supplémentaires que si l’utilisateur a le droit de modifier le document.
    On remet en place la génération de vignettes