Recherche avancée

Médias (0)

Mot : - Tags -/interaction

Aucun média correspondant à vos critères n’est disponible sur le site.

Autres articles (45)

  • Contribute to documentation

    13 avril 2011

    Documentation is vital to the development of improved technical capabilities.
    MediaSPIP welcomes documentation by users as well as developers - including : critique of existing features and functions articles contributed by developers, administrators, content producers and editors screenshots to illustrate the above translations of existing documentation into other languages
    To contribute, register to the project users’ mailing (...)

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

  • Use, discuss, criticize

    13 avril 2011, par

    Talk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
    The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
    A discussion list is available for all exchanges between users.

Sur d’autres sites (8563)

  • FFMPEG- Duration of audio file is inaccurate

    17 septembre 2015, par Tony Than

    I have video file (mp4). I want to detach audio stream (AAC format) from that file and save in PC.
    With below code, Generated aac file canplay now on KM player, but can not play on VLC player. Information of duration displays on player is wrong.
    Please help me with this problem.

    err = avformat_open_input(input_format_context, filename, NULL, NULL);
    if (err < 0) {
       return err;
    }

    /* If not enough info to get the stream parameters, we decode the
      first frames to get it. (used in mpeg case for example) */
    ret = avformat_find_stream_info(*input_format_context, 0);
    if (ret < 0) {
       av_log(NULL, AV_LOG_FATAL, "%s: could not find codec parameters\n", filename);
       return ret;
    }

    /* dump the file content */
    av_dump_format(*input_format_context, 0, filename, 0);

    for (size_t i = 0; i < (*input_format_context)->nb_streams; i++) {
       AVStream *st = (*input_format_context)->streams[i];
       if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
           *input_codec_context = st->codec;
           *input_audio_stream = st;

           FILE *file = NULL;
           file = fopen("C:\\Users\\MyPC\\Downloads\\Test.aac", "wb");
           AVPacket reading_packet;
           av_init_packet(&reading_packet);
           while (av_read_frame(*input_format_context, &reading_packet) == 0) {
               if (reading_packet.stream_index == (int) i) {

                   uint8_t adts_header[7];
                   unsigned int obj_type = 0;
                   unsigned int num_data_block = (reading_packet.size)/1024;
                   int rate_idx = st->codec->sample_rate, channels = st->codec->channels;

                    uint16_t frame_length;

                   // include the header length also
                    frame_length = reading_packet.size + 7;

                   /* We want the same metadata */
                   /* Generate ADTS header */
                   if(adts_header == NULL) return -1;
                   /* Sync point over a full byte */
                   adts_header[0] = 0xFF;
                   /* Sync point continued over first 4 bits + static 4 bits
                   * (ID, layer, protection)*/
                   adts_header[1] = 0xF1;
                   /* Object type over first 2 bits */
                   adts_header[2] = obj_type << 6;
                   /* rate index over next 4 bits */
                   adts_header[2] |= (rate_idx << 2);
                   /* channels over last 2 bits */
                   adts_header[2] |= (channels & 0x4) >> 2;
                   /* channels continued over next 2 bits + 4 bits at zero */
                   adts_header[3] = (channels & 0x3) << 6;
                   /* frame size over last 2 bits */
                   adts_header[3] |= (frame_length & 0x1800) >> 11;
                   /* frame size continued over full byte */
                   adts_header[4] = (frame_length & 0x1FF8) >> 3;
                   /* frame size continued first 3 bits */
                   adts_header[5] = (frame_length & 0x7) << 5;
                   /* buffer fullness (0x7FF for VBR) over 5 last bits*/
                   adts_header[5] |= 0x1F;
                   /* buffer fullness (0x7FF for VBR) continued over 6 first bits + 2 zeros
                   * number of raw data blocks */
                   adts_header[6] = 0xFA;
                   adts_header[6] |= num_data_block & 0x03; // Set raw Data blocks.

                   fwrite(adts_header, 1, 7, file);
                   fwrite(reading_packet.data, 1, reading_packet.size, file);
               }
               av_free_packet(&reading_packet);  
           }
           fclose(file);

           return 0;
       }
    }
  • FFMPEG merge mp4 file and mp3 file into mp4

    10 avril 2017, par Cường Trần

    I have video file in mp4 format (video.mp4), its length is 20 seconds. From 0 seconds to 10 seconds, the video has sound, and from 10 seconds to 20 seconds, there is no sound.

    I also have mp3 file (audio.mp3) and has length 10 seconds.

    I want to merge video.mp4 and audio.mp3 into result.mp4. The result.mp4 file should have video stream and its audio stream from 01 second to 10 seconds as original and audio stream from 10 seconds to 20 seconds of audio.mp3 as merged.

    I use the command to merge :

    ffmpeg -i video.mp4 -i audio.mp3 -filter_complex "aevalsrc=0:d=10[s1];[s1][1:a]concat=n=2:v=1:a=1[aout]" -c:v copy -map 0:v -map [aout] result.mp4

    But i get the result.mp4 with video : there is no sound from 01-10 seconds, only new sound from 10-20 seconds.

    It is the seem that my command don’t keep the sound from original mp4 file, it has removed it and just keep the new sound.

    Could you please help ?

  • Making a thumbnail using fluent-ffmpeg returns ffprobe

    29 avril 2019, par user10204157

    I am getting the following error :

    Error: ffprobe exited with code 1
    ffprobe version 4.1.1 Copyright (c) 2007-2019 the FFmpeg developers
     built with gcc 8.2.1 (GCC) 20190212
     configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass
    --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
     libavutil      56. 22.100 / 56. 22.100
     libavcodec     58. 35.100 / 58. 35.100
     libavformat    58. 20.100 / 58. 20.100
     libavdevice    58.  5.100 / 58.  5.100
     libavfilter     7. 40.101 /  7. 40.101
     libswscale      5.  3.100 /  5.  3.100
     libswresample   3.  3.100 /  3.  3.100
     libpostproc    55.  3.100 / 55.  3.100
    [tcp @ 0000020813d01740] Connection to tcp://stemuli.blob.core.windows.net:443 failed: Error number -138 occurred  <------
    https://stemuli.blob.core.windows.net/stemuli/mentor-lesson-video-76cbf390-4033-4a92-b127-7df22d5885f8.MOV: Unknown error

    Does this mean that azure’s connection is just being cut off ?