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  • Dépôt de média et thèmes par FTP

    31 mai 2013, par

    L’outil MédiaSPIP traite aussi les média transférés par la voie FTP. Si vous préférez déposer par cette voie, récupérez les identifiants d’accès vers votre site MédiaSPIP et utilisez votre client FTP favori.
    Vous trouverez dès le départ les dossiers suivants dans votre espace FTP : config/ : dossier de configuration du site IMG/ : dossier des média déjà traités et en ligne sur le site local/ : répertoire cache du site web themes/ : les thèmes ou les feuilles de style personnalisées tmp/ : dossier de travail (...)

  • Qualité du média après traitement

    21 juin 2013, par

    Le bon réglage du logiciel qui traite les média est important pour un équilibre entre les partis ( bande passante de l’hébergeur, qualité du média pour le rédacteur et le visiteur, accessibilité pour le visiteur ). Comment régler la qualité de son média ?
    Plus la qualité du média est importante, plus la bande passante sera utilisée. Le visiteur avec une connexion internet à petit débit devra attendre plus longtemps. Inversement plus, la qualité du média est pauvre et donc le média devient dégradé voire (...)

  • 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 (...)

Sur d’autres sites (9868)

  • ffmpeg : Create a fake shadow below alpha channel webm/png sequence

    6 mai 2021, par Beneos Battlemaps

    Purpose : I'd like to render out animated 3D meshes as png sequence to use them as animated tokens for virtual tabletop games. To make the mesh looks more natural I'd like to create a fake show beneath the actual token.

    


    Problem : I have a png sequence 1 (as well as a webm file created with ffmpet out of this png sequence if it makes it easier) with alpha channel. To create the webm I use :
ffmpeg -framerate 24 -f image2 -i Idle_Top.%04d.png -c:v libvpx-vp9 -crf 25 -pix_fmt yuva420p Idle_Top.webm (If its relevant). I'd like to render out the png sequence to a webm file that have the current images as well as the transparent shadow beneath the token combined.

    


    Possible workflow : I think a good way to achieve the wanted shadow effect is to use the alpha channel image as a mask on a black picture with the same resolution as the source image 2. Then you have a complete black version of the image. Then you need to place this image beneath the colored image and make a offset of 10px left and 10px down to create the ilusion of perspective 3. At the end the black image below the colored image must have a transparency as well ( 30% visibility should be enough) 4.

    


    Workflow overview

    


    Assets : I've put the webm file and the png files on my gDrive https://drive.google.com/drive/folders/1wznGaPwhKc2UyPpSZBSISa1gs3oixsHR?usp=sharing

    


    Though I work with ffmpeg on a regular basis I have no clue where to start. Can you please help me out with this interesting problem ?

    


    Best regards
Ben

    


  • Generate waveforms for audio files with large amount of channels

    3 mai 2021, par motio

    I want to generate .png files displaying the waveforms of interleaved audio .wav files using the FFmpeg libraries. http://ffmpeg.org/documentation.html

    


    If the interleaved audio file contains maximum 8 channels, I manage to successfully achieve this using the following command line :

    


    ffmpeg -i 8_channels_input.wav -y -filter_complex "showwavespic=s=1920x1200:split_channels=1" -frames:v 1 8_channels_waveform_output.png


    


    However, if the interleaved audio file contains more than 8 channels, FFmpeg gives the following :

    


    


    Input #0, wav, from '30_channels_input.wav' : Duration : 00:00:02.08,
bitrate : 31752 kb/s Stream #0:0 : Audio : pcm_s24le ([1][0][0][0] /
0x0001), 44100 Hz, 30 channels, s32 (24 bit), 31752 kb/s Stream
mapping : Stream #0:0 (pcm_s24le) -> showwavespic showwavespic ->
Stream #0:0 (png) Press [q] to stop, [?] for help [auto_resampler_0 @
0x7faf5d60a3c0] Cannot select channel layout for the link between
filters auto_resampler_0 and Parsed_showwavespic_0. [auto_resampler_0
@ 0x7faf5d60a3c0] Unknown channel layouts not supported, try
specifying a channel layout using 'aformat=channel_layouts=something'.
Error reinitializing filters ! Failed to inject frame into filter
network : Invalid argument Error while processing the decoded data for
stream #0:0 Conversion failed !

    


    


    Here is the related documentation (c.f. bottom of the page) :
https://trac.ffmpeg.org/wiki/AudioChannelManipulation

    


    My problem is :
I need to generate the visual waveforms of audio files containing up to 30 channels.
All my attempts were unsuccessful so far (I am trying to define custom channel layouts and I am not sure if I am on the right track here).

    


    To simplify, I need to complete/modify the following command to make it work :

    


    ffmpeg -i 30_channels_input.wav -y -filter_complex "showwavespic=s=1920x1200:split_channels=1" -frames:v 1 30_channels_waveform_output.png 


    


    [EDIT] Remarks :

    


    I manage to generate the waveforms of a 10 channels input by combining existing layouts :

    


    ffmpeg -i 10_channels_input.wav -y -filter_complex "aformat=channel_layouts=7.1+downmix, showwavespic=s=1920x1200:split_channels=1" -frames:v 1 10_channels_waveform_output.png


    


    However, if you attempt to do it for a 30 channels input by combining 5x 6.0 layouts :

    


    ffmpeg -i 30_channels_input.wav -y -filter_complex "aformat=channel_layouts=6.0+6.0+6.0+6.0+6.0, showwavespic=s=1920x1200:split_channels=1" -frames:v 1 30_channels_waveform_output.png


    


    FFmpeg gives the following :

    


    


    [auto_resampler_0 @ 0x7ffd7002a480] [SWR @ 0x7ffd7013a000] Rematrix is
needed between 30 channels and 6.0 but there is not enough information
to do it [auto_resampler_0 @ 0x7ffd7002a480] Failed to configure
output pad on auto_resampler_0 Error reinitializing filters ! Failed to
inject frame into filter network : Invalid argument Error while
processing the decoded data for stream #0:0

    


    


    My assumption is that I need to create a custom layout using 30 unique channel IDs (c.f. https://trac.ffmpeg.org/wiki/AudioChannelManipulation bottom of the page) instead of combining existing layouts.

    


    It seems that only 25 channel IDs are available though. Creating a custom layout with 30 channels is maybe not possible at all...

    


    [EDIT 2] Remarks :

    


    I finally found the documentation I was looking for. But it still seems that generating the waveforms of 30 channels would be difficult.

    


    Here is how to create custom layouts :
https://ffmpeg.org/ffmpeg-utils.html

    


    


    A custom channel layout can be specified as a sequence of terms,
separated by ’+’ or ’|’. Each term can be :

    


    the name of a standard channel layout (e.g. ‘mono’, ‘stereo’, ‘4.0’,
‘quad’, ‘5.0’, etc.) the name of a single channel (e.g. ‘FL’, ‘FR’,
‘FC’, ‘LFE’, etc.) a number of channels, in decimal, followed by ’c’,
yielding the default channel layout for that number of channels (see
the function av_get_default_channel_layout). Note that not all channel
counts have a default layout. a number of channels, in decimal,
followed by ’C’, yielding an unknown channel layout with the specified
number of channels. Note that not all channel layout specification
strings support unknown channel layouts. a channel layout mask, in
hexadecimal starting with "0x" (see the AV_CH_* macros in
libavutil/channel_layout.h. Before libavutil version 53 the trailing
character "c" to specify a number of channels was optional, but now it
is required, while a channel layout mask can also be specified as a
decimal number (if and only if not followed by "c" or "C").

    


    See also the function av_get_channel_layout defined in
libavutil/channel_layout.h.

    


    


    e.g.

    


    Therefore, for 11 channels input :

    


    ffmpeg -i 11_channels_input.wav -y -filter_complex "aformat=channel_layouts=FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL, showwavespic=s=1920x1200:split_channels=1" -frames:v 1 11_waveform_output.png


    


    —> worked for me

    


    while :

    


    ffmpeg -i 11_channels_input.wav -y -filter_complex "aformat=channel_layouts=11c, showwavespic=s=1920x1200:split_channels=1" -frames:v 1 11_waveform_output.png


    


    —> does not work

    


    But :

    


    ffmpeg -i 24_channels_input.wav -y -filter_complex "aformat=channel_layouts=24c, showwavespic=s=1920x1200:split_channels=1" -frames:v 1 24_waveform_output.png


    


    —> does work

    


    and finally, what I am still trying to achieve :

    


    ffmpeg -i 30_channels_input.wav -y -filter_complex "aformat=channel_layouts=30c, showwavespic=s=1920x1200:split_channels=1" -frames:v 1 30_waveform_output.png


    


    —> does not work

    


    p.s.

    


      

    • I execute these commands in sub shells via Ruby scripts (puts %x...)
    • 


    • My system : macOS 10.15.6 | zsh | FFmpeg 4.4 | Ruby 2.6.3
    • 


    


  • FFmpeg - MJPEG decoding - getting different values

    27 décembre 2016, par ahmadh

    I have a set of JPEG frames which I am muxing into an avi, which gives me a mjpeg video. This is the command I run on the console :

    ffmpeg -y -start_number 0 -i %06d.JPEG -codec copy vid.avi

    When I try to demux the video using ffmpeg C api, I get frames which are slightly different in values. Demuxing code looks something like this :

    AVFormatContext* fmt_ctx = NULL;
    AVCodecContext* cdc_ctx = NULL;
    AVCodec* vid_cdc = NULL;
    int ret;
    unsigned int height, width;

    ....
    // read_nframes is the number of frames to read
    output_arr = new unsigned char [height * width * 3 *
                                   sizeof(unsigned char) * read_nframes];

    avcodec_open2(cdc_ctx, vid_cdc, NULL);

    int num_bytes;
    uint8_t* buffer = NULL;
    const AVPixelFormat out_format = AV_PIX_FMT_RGB24;

    num_bytes = av_image_get_buffer_size(out_format, width, height, 1);
    buffer = (uint8_t*)av_malloc(num_bytes * sizeof(uint8_t));

    AVFrame* vid_frame = NULL;
    vid_frame = av_frame_alloc();
    AVFrame* conv_frame = NULL;
    conv_frame = av_frame_alloc();

    av_image_fill_arrays(conv_frame->data, conv_frame->linesize, buffer,
                        out_format, width, height, 1);

    struct SwsContext *sws_ctx = NULL;
    sws_ctx = sws_getContext(width, height, cdc_ctx->pix_fmt,
                            width, height, out_format,
                            SWS_BILINEAR, NULL,NULL,NULL);

    int frame_num = 0;
    AVPacket vid_pckt;
    while (av_read_frame(fmt_ctx, &vid_pckt) >=0) {
       ret = avcodec_send_packet(cdc_ctx, &vid_pckt);
       if (ret < 0)
           break;

       ret = avcodec_receive_frame(cdc_ctx, vid_frame);
       if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF)
           break;
       if (ret >= 0) {
           // convert image from native format to planar GBR
           sws_scale(sws_ctx, vid_frame->data,
                     vid_frame->linesize, 0, vid_frame->height,
                     conv_frame->data, conv_frame->linesize);

           unsigned char* r_ptr = output_arr +
               (height * width * sizeof(unsigned char) * 3 * frame_num);
           unsigned char* g_ptr = r_ptr + (height * width * sizeof(unsigned char));
           unsigned char* b_ptr = g_ptr + (height * width * sizeof(unsigned char));
           unsigned int pxl_i = 0;

           for (unsigned int r = 0; r < height; ++r) {
               uint8_t* avframe_r = conv_frame->data[0] + r*conv_frame->linesize[0];
               for (unsigned int c = 0; c < width; ++c) {
                   r_ptr[pxl_i] = avframe_r[0];
                   g_ptr[pxl_i]   = avframe_r[1];
                   b_ptr[pxl_i]   = avframe_r[2];
                   avframe_r += 3;
                   ++pxl_i;
               }
           }

           ++frame_num;

           if (frame_num >= read_nframes)
               break;
       }
    }

    ...

    In my experience around two-thirds of the pixel values are different, each by +-1 (in a range of [0,255]). I am wondering is it due to some decoding scheme FFmpeg uses for reading JPEG frames ? I tried encoding and decoding png frames, and it works perfectly fine.

    In short my goal is to get the same pixel by pixel values for each JPEG frame as I would I have gotten if I was reading the JPEG images directly. Here is the stand-alone code I used. It includes cmake files to build code, and a couple of jpeg frames with the converted avi file to test this problem. (give —filetype png to test the png decoding).