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  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

Sur d’autres sites (5825)

  • Writing a series of images into a video file using libavcodec (ffmpeg)

    19 novembre 2013, par user2978372

    Requirements :

    I have a bunch of images (to be more specific they are 1024x768, 24bpp RGB PNG files) that I want to encode into a video files.

    And I need to use 'libavcodec' library, not 'ffmpeg' tool. (well I know they are basically same in the origin, I am emphasizing because someone may answer to use 'ffmpeg' tool, but that's not a solution what I am looking for)

    I am using h264 encoder.

    Target :
    A high quality video with equal resolution (1024 x 768), YUV420P
    each image has a duration of 1 second.
    24 fps

    Problems :
    i've tried with many different (but same resolution and bits) png images, and all have failed to output a good video.

    For some series of images, only the frames of first second was shown in a good shape, but the remaining frames was distorted and color changed (lighter).

    For some series of images, it seemed the images were zoomed-in and distorted again.

    and etc.

    Question :
    I am a total AV newbie and I need someone to verify my steps for encoding. I am total AV newbie.

    1) av_register_all()

    2) avcodec_register_all()

    3) avcodec_find_encoder()

    4) avcodec_alloc_context3()

    5) sets codec configuraton to context.

    6) avcodec_open2()

    7) opens a output file using fopen_s()

    8)

    for(int second=1; second<=10; ++seconds)
    {

     Read a image from local using Gdiplus

     Create a gdiplus bitmap and draw the image onto this bitmap

     Get the raw byte data using LockBits

     Transfer this RGB raw byte into YUV420 frame using 'swscontext', 'sws_scale'

     for(int f=0; f<24; ++f)
     {
       av_init_packet(&pkt);
       pkt.data = NULL;
       pkt.size = 0;

       pFrame->pts = f;
       ret = avcodec_encode_video2(pContext, &pkt, pFrame, &got_output);

       if(got_output)
       {
           fwrite(pkt.data, 1, pkt.size, outputFile);
           av_free_packet(&pkt);
       }
     }
    }

    /* get the delayed frames */
    for (got_output = 1; got_output; i++) {
       fflush(stdout);

       ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
       if (ret < 0) {
           fprintf(stderr, "Error encoding frame\n");
           exit(1);
       }

       if (got_output) {
           printf("Write frame %3d (size=%5d)\n", i, pkt.size);
           fwrite(pkt.data, 1, pkt.size, f);
           av_free_packet(&pkt);
       }
    }

    close everything required

    I am sure that I misunderstood some steps using ffmpeg api. The above pseudo codes are based on the 'encoding' example of ffmpeg. which part am I doing wrong ? please can someone help me ?

    p.s sorry about broken english. english is not my natvie language. I tried my best =P

  • avformat/http: check the auth string contents not the pointer which cannot be NULL

    14 octobre 2013, par Michael Niedermayer
    avformat/http: check the auth string contents not the pointer which cannot be NULL
    

    It appears this bug originates from a "work in progress" patch from
    ffmpeg-devel that was heavily redesigned by and integrated in libav

    And that patch even had a reply and review on the mailing list pointing
    out that it had a bug.

    This fixes a deadlock with ffserver

    See : [FFmpeg-devel] [PATCH] Fix HTTP authentication problem for POST actions.
    [FFmpeg-devel] [PATCH 1/3] Introduce auth_phase flag, which will be true if authorization needs to be sent, but the type of authorization is not known yet Partial fix #3036
    [FFmpeg-devel] [PATCH 2/3] Only add Transfer-Encoding header when not in authorization phase, because server will wait (indefinitely) for data when receiving this header Partial fix #3036
    [FFmpeg-devel] [PATCH 3/3] Only allow posting data and/or forcing a 200 code, enabling posting isml chunks, -after- we did a possible first request to get a 403 from the server telling us which type of authentication to apply Final part fix #3036
    See : 71549a857b13edf4c4f95037de6ed5bb4c4bd4af
    Signed-off-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavformat/http.c
  • FFMpeg copy stream SEI type 162 error

    13 juillet 2015, par Vishal Shah

    I am using FFMpeg to copy an RTSP stream to an RTMP stream. Both of which are network streams. This is the command I am using.

    ffmpeg -re -i rtsp ://url.to/rtsp/stream -c copy -f flv rtmp ://target.rtmp/uri/stream

    I am receiving data on the youtube live control room. But After a point I get an error like this in my FFMpeg console and then the data transfer on youtube live stops.

    SEI type 162 size 1496 truncated at 1280

    Can someone throw light on what this error means and how to resolve it ?

    Note that the RTSP stream is over the network