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  • Script d’installation automatique de MediaSPIP

    25 avril 2011, par

    Afin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
    Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
    La documentation de l’utilisation du script d’installation (...)

  • Les notifications de la ferme

    1er décembre 2010, par

    Afin d’assurer une gestion correcte de la ferme, il est nécessaire de notifier plusieurs choses lors d’actions spécifiques à la fois à l’utilisateur mais également à l’ensemble des administrateurs de la ferme.
    Les notifications de changement de statut
    Lors d’un changement de statut d’une instance, l’ensemble des administrateurs de la ferme doivent être notifiés de cette modification ainsi que l’utilisateur administrateur de l’instance.
    À la demande d’un canal
    Passage au statut "publie"
    Passage au (...)

  • Initialisation de MediaSPIP (préconfiguration)

    20 février 2010, par

    Lors de l’installation de MediaSPIP, celui-ci est préconfiguré pour les usages les plus fréquents.
    Cette préconfiguration est réalisée par un plugin activé par défaut et non désactivable appelé MediaSPIP Init.
    Ce plugin sert à préconfigurer de manière correcte chaque instance de MediaSPIP. Il doit donc être placé dans le dossier plugins-dist/ du site ou de la ferme pour être installé par défaut avant de pouvoir utiliser le site.
    Dans un premier temps il active ou désactive des options de SPIP qui ne le (...)

Sur d’autres sites (7370)

  • ffmpeg error : Too large number of skipped frames xxxxx > 60000

    16 septembre 2019, par Fulkron

    I try to record on RasperryPI from usb Cam (/dev/video0) using segments to obtain sliced video.
    The command I’m using is :

    ffmpeg -input_format mjpeg -video_size 640x480 -i /dev/video0 -y -y -f v4l2 -an -pix_fmt yuv420p -codec copy -f segment -segment_time 60 -segment_wrap 10 -t 180 stream/looper%02d.avi

    I got an error after 2 minutes about skipped frames (see later for detail).
    Is I run the same command without segmentation is working.

    I tested on different HW, with different segment parameters, with different usb cam but the situation still the same.

    Here there is the output :

      ffmpeg version 4.1.4-1+rpt1~deb10u1 Copyright (c) 2000-2019 the FFmpeg developers
     built with gcc 8 (Raspbian 8.3.0-6+rpi1)
     configuration: --prefix=/usr --extra-version='1+rpt1~deb10u1' --toolchain=hardened --libdir=/usr/lib/arm-linux-gnueabihf --incdir=/usr/include/arm-linux-gnueabihf --arch=arm --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-omx-rpi --enable-mmal --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
     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
     libavresample   4.  0.  0 /  4.  0.  0
     libswscale      5.  3.100 /  5.  3.100
     libswresample   3.  3.100 /  3.  3.100
     libpostproc    55.  3.100 / 55.  3.100
    Input #0, video4linux2,v4l2, from '/dev/video0':
     Duration: N/A, start: 40185.241073, bitrate: N/A
       Stream #0:0: Video: mjpeg, yuvj422p(pc, bt470bg/unknown/unknown), 640x480, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
    [segment @ 0x9800b0] Opening 'stream/looper00.avi' for writing
    Output #0, segment, to 'stream/looper%02d.avi':
     Metadata:
       encoder         : Lavf58.20.100
       Stream #0:0: Video: mjpeg, yuvj422p(pc, bt470bg/unknown/unknown), 640x480, q=2-31, 25 fps, 25 tbr, 600 tbn, 1000k tbc
    Stream mapping:
     Stream #0:0 -> #0:0 (copy)
    Press [q] to stop, [?] for help
    [segment @ 0x9800b0] Opening 'stream/looper01.avi' for writing/A speed=1.01x    
    [segment @ 0x9800b0] Opening 'stream/looper02.avi' for writing/A speed=1.01x    
    [avi @ 0x9822a0] Too large number of skipped frames 72034 > 60000speed=1.01x    
    av_interleaved_write_frame(): Invalid argument
    [avi @ 0x9822a0] Too large number of skipped frames 72034 > 60000
    frame= 3011 fps= 25 q=-1.0 Lsize=N/A time=00:02:00.06 bitrate=N/A speed=1.01x

    I did not find any suggestion to solve it.
    Could you help me ?

    Thanks
    Dario

  • FFMPEG APC to MP3 Command

    7 octobre 2019, par Kuja

    Files compressed at a rate of 32 KBit / s using a special adpcm codec. How do I convert this file to mp3 ?

    ffmpeg -i "10_____________AO905326704670____10_________.apc"   audio.mp3

    ffmpeg version 3.4.6-0ubuntu0.18.04.1 Copyright (c) 2000-2019 the FFmpeg developers
     built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
     configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
     libavutil      55. 78.100 / 55. 78.100
     libavcodec     57.107.100 / 57.107.100
     libavformat    57. 83.100 / 57. 83.100
     libavdevice    57. 10.100 / 57. 10.100
     libavfilter     6.107.100 /  6.107.100
     libavresample   3.  7.  0 /  3.  7.  0
     libswscale      4.  8.100 /  4.  8.100
     libswresample   2.  9.100 /  2.  9.100
     libpostproc    54.  7.100 / 54.  7.100
    10_____________AO905326704670____10_________.apc: Invalid data found when processing input
  • Invalid data found when processing file with ffmpeg API

    18 septembre 2019, par Timmy K

    I have an m4a file containing two AAC-encoded audio streams. I want to decode it on Android with the ffmpeg library. However, avformat_open_input() fails with "Invalid data found when processing input".

    This file was created (i.e. the audio data encoded) with the same library.

    I’ve double-checked the accuracy of the file path I provide to the call.

    The file doesn’t appear to be corrupt. It plays with QuickTime player, and ffprobe reports the output below.

    I’ve looked at the source in ffmpeg’s utils.c but it’s not clear at which of several points (including nested calls) this invalidity of the data in the file is being decided.

    FFPROBE OUTPUT
    built with gcc 7.2.0 (crosstool-NG fa8859cb)
    configuration: --prefix=/home/ubuntu/miniconda3 --disable-doc --enable-
    shared --extra-cflags='-fPIC -I/home/ubuntu/miniconda3/include' --extra-
    cxxflags='=-fPIC' --extra-libs='-L/home/ubuntu/miniconda3/lib -lz' --enable-
    pic --disable-static --disable-gpl --disable-nonfree --disable-openssl --
    enable-libvpx --cc=/opt/conda/conda-
    bld/ffmpeg_1530807717919/_build_env/bin/x86_64-conda_cos6-linux-gnu-cc --cxx=/opt/conda/conda-bld/ffmpeg_1530807717919/_build_env/bin/x86_64-conda_cos6-linux-gnu-c++ --enable-libopus
       libavutil            55. 78.100 / 55. 78.100
       libavcodec         57.107.100 / 57.107.100
       libavformat        57. 83.100 / 57. 83.100
       libavdevice        57. 10.100 / 57. 10.100
       libavfilter         6.107.100 /    6.107.100
       libswscale            4.    8.100 /    4.    8.100
       libswresample     2.    9.100 /    2.    9.100
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'recording-2019-09-03-110626.m4a':
       Metadata:
           major_brand         : isom
           minor_version     : 512
           compatible_brands: isomiso2mp41
           encoder                 : Lavf58.27.103
       Duration: 00:00:05.04, start: 0.000000, bitrate: 637 kb/s
           Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 314 kb/s (default)
           Metadata:
               handler_name        : SoundHandler
           Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 315 kb/s (default)
           Metadata:
               handler_name        : SoundHandler

    The call that fails with "invalid data found" is :

    if ( (ret = avformat_open_input( &inputFormatContext, filePath, 0, 0)) < 0 )
     {
       __android_log_print(ANDROID_LOG_DEBUG, "MyTag", "Could not open input file '%s' error %s ", filePath, av_err2str(ret) );
         return false;
     }

    I would expect this to "just work" since I encoded and muxed the two audio tracks with the same build of the ffmpeg libraries.

    When I built the libraries, I configured the build with :

    —enable-decoder=aac,pcm_s16le —enable-encoder=aac,pcm_s16le —enable-demuxer=mp4,wav —enable-muxer=mp4,wav —enable-protocol=file,http

    When I iterate through the codecs using av_codec_iterate() and look for supported sample rates I see :

    Found codec aac
    supported sample rate 96000
    supported sample rate 88200
    supported sample rate 64000
    supported sample rate 48000
    supported sample rate 44100
    supported sample rate 32000
    supported sample rate 24000
    supported sample rate 22050
    supported sample rate 16000
    supported sample rate 12000
    supported sample rate 11025
    supported sample rate 8000
    supported sample rate 7350
    Found codec pcm_s16le
    Found codec aac
    Found codec pcm_s16le

    Each codec appears twice, which I thought was maybe the encoder and decoder. But I note that only one has any supported sample rates. My test files’ sample rate is 44100.

    Any thoughts on what other things I can check for ? I wondered whether I should create and pass an AVInputFormat object explicitly to the call, based upon what I know about the contents of the file. But I don’t know how to construct one.