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  • Les autorisations surchargées par les plugins

    27 avril 2010, par

    Mediaspip core
    autoriser_auteur_modifier() afin que les visiteurs soient capables de modifier leurs informations sur la page d’auteurs

  • List of compatible distributions

    26 avril 2011, par

    The table below is the list of Linux distributions compatible with the automated installation script of MediaSPIP. Distribution nameVersion nameVersion number Debian Squeeze 6.x.x Debian Weezy 7.x.x Debian Jessie 8.x.x Ubuntu The Precise Pangolin 12.04 LTS Ubuntu The Trusty Tahr 14.04
    If you want to help us improve this list, you can provide us access to a machine whose distribution is not mentioned above or send the necessary fixes to add (...)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

Sur d’autres sites (5316)

  • FFmpeg library : modified muxing sample for FLTP input and FLTP audio, loses audio

    21 janvier 2014, par taansari

    Based on muxing sample that comes with FFmpeg docs, I have modified it, from input format as S16 to FLTP (planar stereo), and outputting to webm format (stereo).

    Since input is now FLTP, I am filling two arrays, then encoding again to FLTP. There are no obvious errors given on screen, but the resulting webm video does not play any audio (just the video content). This is just proof of concept in understanding things ; here is an added (crude) function to fill up input FLTP stereo buffer :

    static void get_audio_frame_for_planar_stereo(int16_t **samples, int frame_size, int nb_channels)
    {
       int j, i, v[2];
       int16_t *q1 = (int16_t *) samples[0];
       int16_t *q2 = (int16_t *) samples[1];

       for (j = 0; j < frame_size; j++)
       {
           v[0] = (int)(sin(t) * 10000);
           v[1] = (int)(tan(t) * 10000);
           *q1++ = v[0];
           *q2++ = v[1];
           t     += tincr;
           tincr += tincr2;
       }
    }

    Which I am calling from inside write_audio_frame() function.

    Note also, wherever code reffered AV_SAMPLE_FMT_S16 as input, I have changed to AV_SAMPLE_FMT_FLTP.

    Whole workable source is here :

    https://gist.github.com/anonymous/05d1d7662e9feafc45a6

    When run with ffprobe.exe, with these instructions :

    ffprobe -show_packets output.webm >output.txt

    I see nothing out of ordinary, all pts/dts values appear to be in place :

    https://gist.github.com/anonymous/3ed0d6308700ab991704

    Could someone highlight cause of this mis-interpretation ?

    Thanks for your time...

    p.s. I am using Zeranoe FFmpeg Windows builds (32 bit), built on Jan 9 2014 22:04:35 with gcc 4.8.2.(GCC)

    Edit : Based on your guidance elsewhere, I tried the following :

       /* set options */
       //av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
       //av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
       //av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_FLTP, 0);
       //av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
       //av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
       //av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);

       av_opt_set_int(swr_ctx, "in_channel_layout",    AV_CH_LAYOUT_STEREO, 0);
       av_opt_set_int(swr_ctx, "in_sample_rate",       c->sample_rate, 0);
       av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);

       av_opt_set_int(swr_ctx, "out_channel_layout",    AV_CH_LAYOUT_STEREO, 0);
       av_opt_set_int(swr_ctx, "out_sample_rate",       c->sample_rate, 0);
       av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);

    And the revised function :

    static void get_audio_frame_for_planar_stereo(uint8_t **samples, int frame_size, int nb_channels)
    {
       int j, i;
       float v[2];
       float *q1 = (float *) samples[0];
       float *q2 = (float *) samples[1];

       for (j = 0; j < frame_size; j++)
       {
           v[0] = (tan(t) * 1);
           v[1] = (sin(t) * 1);
           *q1++ = v[0];
           *q2++ = v[1];
           t     += tincr;
           tincr += tincr2;
       }
    }

    Now it appears to be working properly. I tried changing function parameters from uint8_t** to float**, as well as src_samples_data from uint8_t** to float**, but did not make any difference, in a view.

    Updated code : https://gist.github.com/anonymous/35371b2c106961029c3d

    Thanks for highlighting the place(s) that result in this behavior !

  • Audio equalizer using FFmpeg

    28 septembre 2015, par Evgeniy Kharchenko

    I’m developing audio player using FFmpeg. I want to add audio equaliqer to my app, but I don’t know how to do it. I know that FFmpeg has a FFT functions, but I have count of samples that not mutch with any power of 2. I also tried to use other FFT libraries, but I have some noise on my audio after equalization.
    Can anybody help me to understand how to use FFmpeg’s FFT functions or any oter ways to implement audio-eq using FFmpeg ?

  • Overlay audio over video FFMEPG

    15 mars 2018, par Benjamin Sweney

    I’m wanting to overlay an audio recording over a video using FFMEPG PHP or more specially Laravel (if possible). The audio clip will need to replace the videos audio, or better if possible set the videos audio volume to a predefined value and overlay the audio.

    The audio clip will always be longer than or the same length as the video and can not be clipped. Side note, how will this be handled ? Black screen or frozen last frame, or ?

    Would appreciate someone pointing me in the right direction.

    Much appreciated !