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  • 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 (...)

  • Le plugin : Podcasts.

    14 juillet 2010, par

    Le problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
    Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
    Types de fichiers supportés dans les flux
    Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...)

  • Selection of projects using MediaSPIP

    2 mai 2011, par

    The examples below are representative elements of MediaSPIP specific uses for specific projects.
    MediaSPIP farm @ Infini
    The non profit organizationInfini develops hospitality activities, internet access point, training, realizing innovative projects in the field of information and communication technologies and Communication, and hosting of websites. It plays a unique and prominent role in the Brest (France) area, at the national level, among the half-dozen such association. Its members (...)

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  • 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 ?

  • Transition effects between images using ImageMagick and FFMPEG [closed]

    7 mai 2013, par user1599914

    I'm trying to generate a video using still images. I'm using Image Magick's convert command to duplicate the images.

    convert screen6.png -duplicate 10 -background black -gravity center -extent 1024x576 4%06d.screen.transition.png

    I then use ffmpeg to encode the images.

    Problem 1 : How can I achieve good transition effects between the images ; effects such as circle in, box out, wheel, as well as kenburns effect.

    Problem 2 : How can I overlay the images with text, and make that text to animate or appear with some effects such as fade in, zoom out.

    Your help will be greatly appreciated.

  • resolution changed mid video - FFMPEG

    27 août 2012, par user1626099

    I have a source video (mpeg2video) which I'm transcoding to x264. The source contains 2 different programs recorded from TV. One is in 4:3 AR and the other 16:9 AR. When I play the source file through VLC the player correctly changes size to show the video at the correct AR. So far so good.

    When I transcode the conversion process auto detects the AR from the first few frames and then transcodes the whole video using this AR. If the 16:9 section comes first then the whole conversion is done in 16:9 and the 4:3 section looks stretch horizontally. If the 4:3 section is at the start of the source file then the whole transcode is done in 4:3 and the 16:9 section looks squashed horizontally.

    No black bars are ever visible.

    Here's my command :

    nice -n 17 ffmpeg -i source.mpg -acodec libfaac -ar 48000 -ab 192k -async 1 -copyts -vcodec libx264 -b 1250k -threads 2 -level 31 -map 0:0 -map 0:1 -map 0:2 -scodec copy -deinterlace output.mkv

    I don't fully understand what's going on. How do I get the same 'change in AR' mid video in the output file that I have in the input video ?