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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • Les vidéos

    21 avril 2011, par

    Comme les documents de type "audio", Mediaspip affiche dans la mesure du possible les vidéos grâce à la balise html5 .
    Un des inconvénients de cette balise est qu’elle n’est pas reconnue correctement par certains navigateurs (Internet Explorer pour ne pas le nommer) et que chaque navigateur ne gère en natif que certains formats de vidéos.
    Son avantage principal quant à lui est de bénéficier de la prise en charge native de vidéos dans les navigateur et donc de se passer de l’utilisation de Flash et (...)

  • Personnaliser en ajoutant son logo, sa bannière ou son image de fond

    5 septembre 2013, par

    Certains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;

Sur d’autres sites (5874)

  • encoding XDCAM MXF with FFMPEG

    9 octobre 2013, par magingax

    Trying to encode XDCAM HD422 MXF with FFMPEG
    But can't know specific setting for encoder/format/mux
    Anyone can show me sample code or give some advice for that ?

  • 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 ?

  • MP4 Videos on website embed with html5 does not play on iOS

    31 août 2012, par Sidner

    So I have a couple of videos on my website that I shot using the iPhone 4 and then converted to mp4, webm and ogg, so that I can use them with html5. Thing is, the video does not play at all on the 4 iOS devices that I tested and neither on Chrome for Android.

    The Chrome issue could be because some of the mp4 are actually m4v files, but still after encoding with handbrake a video to the iphone 4 presset and in mp4 format, it still does not play.

    What happens, you ask ? Well, it shows the play button crossed out with a diagonal bar, the debug console on Safari does not show any message untill I try to access the video directly. Then it says : QuickTime Movie could not be played.

    What can I do ? I have been trying to encode with ffmpeg, have tried a handful of different solutions, some even found here on stackoverlow, but to no avail. The videos do get shorter, both in display size and MBs, but nothing works to fix the issue at hand.

    I've been trying to get this corrected for a couple of weeks now. Any help and/or suggestions are welcome.

    Thank you.

    By the way, all the videos are on a registred users section of the website, but I have one for debugin on the main page, so feel free to test.

    https://sidnerwebsite.sytes.net