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  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

  • 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 (...)

  • Selection of projects using MediaSPIP

    2 mai 2011, par

    The examples below are representative elements of MediaSPIP specific uses for specific projects.
    MediaSPIP farm @ Infini
    The non profit organizationInfini develops hospitality activities, internet access point, training, realizing innovative projects in the field of information and communication technologies and Communication, and hosting of websites. It plays a unique and prominent role in the Brest (France) area, at the national level, among the half-dozen such association. Its members (...)

Sur d’autres sites (3600)

  • Why the fps of concatenate video file change with old video file ?

    23 juin 2016, par Zuiche

    I have two wmv video file which 25fps.
    I concatenate these video file using ffmpeg command line like this :

    ffmpeg -r 25 -f concat -i 1.txt -c copy output.wmv

    1.txt :

    file 'g:\1.wmv'
    
file 'g:\2.wmv'

    It works. But the fps of output.wmv is 30.
    How can I maintain the fps of concatenate video file ?

    UPDATE
    ======Console Output========

    ffmpeg version N-73266-g4aa0de6 Copyright (c) 2000-2015 the FFmpeg developers
    
built with gcc 4.9.2 (GCC)
    
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enable-zlib
    
libavutil      54. 27.100 / 54. 27.100
    
libavcodec     56. 45.101 / 56. 45.101
    
libavformat    56. 40.100 / 56. 40.100
    
libavdevice    56.  4.100 / 56.  4.100
    
libavfilter     5. 19.100 /  5. 19.100
    
libswscale      3.  1.101 /  3.  1.101
    
libswresample   1.  2.100 /  1.  2.100
    
libpostproc    53.  3.100 / 53.  3.100
    
[wmv3 @ 04d452a0] Extra data: 8 bits left, value: 20
    
[wmv3 @ 04d14d00] Extra data: 8 bits left, value: 20
    
Input #0, concat, from '1.txt':
    
Duration: N/A, start: 0.000000, bitrate: 4214 kb/s
    
Stream #0:0: Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 720x576, 4194 kb/s, SAR 16:15 DAR 4:3, 25 fps, 25 tbr, 1k tbn, 1k tbc
    
Stream #0:1: Audio: wmavoice ([10][0][0][0] / 0x000A), 22050 Hz, mono, flt, 20 kb/s
    
[asf @ 04d12120] Codec for stream 0 does not use global headers but container format requires global headers
    
[asf @ 04d12120] Codec for stream 1 does not use global headers but container format requires global headers
    
Output #0, asf, to 'output.wmv':
    
Metadata:
    
WM/EncodingSettings: Lavf56.40.100
    
Stream #0:0: Video: wmv3 (WMV3 / 0x33564D57), yuv420p, 720x576 [SAR 16:15 DAR 4:3], q=2-31, 4194 kb/s, 25 fps, 25 tbr, 1k tbn, 25 tbc
    
Stream #0:1: Audio: wmavoice ([10][0][0][0] / 0x000A), 22050 Hz, mono, 20 kb/s
    
Stream mapping:
    
Stream #0:0 -> #0:0 (copy)
    
Stream #0:1 -> #0:1 (copy)
    
Press [q] to stop, [?] for help
    
[wmv3 @ 04ce8880] Extra data: 8 bits left, value: 20
    
frame=  538 fps=0.0 q=-1.0 Lsize=    5723kB time=00:00:21.52 bitrate=2178.5kbits/s  
    
video:5593kB audio:62kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.206905%

    =========FFprobe Output=========

    ffprobe version N-60456-g4040b56-Sherpya Copyright (c) 2007-2014 the FFmpeg developers
    
built on Feb  9 2014 06:50:33 with gcc 4.8 (GCC)
    
[wmv3 @ 050684e0] Extra data: 8 bits left, value: 20
    
Input #0, asf, from 'output.wmv':
    
Metadata:
    
encoder         : Lavf56.40.100
    
Duration: 00:00:21.52, start: 0.000000, bitrate: 2178 kb/s
    
Stream #0:0: Video: wmv3 (Main) (WMV3 / 0x33564D57), yuv420p, 720x576, SAR 16:15 DAR 4:3, 25 tbr, 1k tbn, 1k tbc
    
Stream #0:1: Audio: wmavoice ([10][0][0][0] / 0x000A), 22050 Hz, mono, flt, 20 kb/s
    
[wmv3 @ 050684e0] Extra data: 8 bits left, value: 20

  • How to segment a video and then concatenate back into original one with ffmpeg

    23 décembre 2016, par steve

    I am surveying on distributed video transcoding with FFmpeg. I have found that there is a good script on https://github.com/nergdron/dve/blob/master/dve.

    The script mainly uses the segment and concatenate filters of FFmpeg. I want to do a simple test first. However, I can not split the video into segments and then concatenate back to original video (with the same codec). I have tried with the following command :

    a. Chunk the video

    ffmpeg -fflags +genpts -i Test.avi -map 0 -codec copy -f segment -segment_format avi -v error chunk-%03d.seg

    b. Building the chunking list :

    #!/bin/bash -e
    
set -e
    
echo "ffconcat version 1.0" > concat.txt
    
for f in `ls chunk-*.seg | sort`; do
    
echo "file $f" >> concat.txt
    
done

    c. Concatenate the chunks

    ffmpeg  -y -v error -i concat.txt -f concat -map 0 -c copy -f avi output.avi

    Then when I run ffprobe I get the following message which says it is a non-interleaved AVI :

        ffprobe version N-82301-g1bbb18f Copyright (c) 2007-2016 the FFmpeg developers
    
  built with gcc 5.4.0 (GCC)
    
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
    
  libavutil      55. 35.100 / 55. 35.100
    
  libavcodec     57. 66.101 / 57. 66.101
    
  libavformat    57. 57.100 / 57. 57.100
    
  libavdevice    57.  2.100 / 57.  2.100
    
  libavfilter     6. 66.100 /  6. 66.100
    
  libswscale      4.  3.100 /  4.  3.100
    
  libswresample   2.  4.100 /  2.  4.100
    
  libpostproc    54.  2.100 / 54.  2.100
    
[avi @ 00000000028e3700] non-interleaved AVI
    
Input #0, avi, from 'output.avi':
    
  Metadata:
    
    encoder         : Lavf57.57.100
    
  Duration: 74:43:47.82, start: 0.000000, bitrate: 17 kb/s
    
    Stream #0:0: Video: mpeg4 (Advanced Simple Profile) (XVID / 0x44495658), yuv420p, 640x368 [SAR 1:1 DAR 40:23], 23.98 fps, 23.98 tbr, 23.98 tbn, 23.98 tbc
    
    Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 112 kb/s

    I have tried a few other things without success. Any help would be greatly appreciated. Thanks in advance !!

  • How to call ffmpeg code without losing quality ?

    7 mai 2016, par seaguest

    I am using ffmpeg3 to convert mv.webm to mp4, it works fine when I use the command line.

    $ ffprobe mv.webm 
    
ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
    
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
    
  configuration: --enable-gpl --enable-shared --enable-libx264
    
  libavutil      55. 24.100 / 55. 24.100
    
  libavcodec     57. 39.100 / 57. 39.100
    
  libavformat    57. 36.100 / 57. 36.100
    
  libavdevice    57.  0.101 / 57.  0.101
    
  libavfilter     6. 45.100 /  6. 45.100
    
  libswscale      4.  1.100 /  4.  1.100
    
  libswresample   2.  0.101 /  2.  0.101
    
  libpostproc    54.  0.100 / 54.  0.100
    
Input #0, matroska,webm, from 'mv.webm':
    
  Metadata:
    
    encoder         : google
    
  Duration: 00:04:56.36, start: 0.000000, bitrate: 836 kb/s
    
    Stream #0:0: Video: vp8, yuv420p, 640x480, SAR 1:1 DAR 4:3, 25 fps, 25 tbr, 1k tbn (default)
    
    Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)
    

    
$ ffmpeg -i mv.webm  -acodec aac -vcodec libx264  mv1.mp4
    
$ ffprobe mv1.mp4 
    
ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
    
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
    
  configuration: --enable-gpl --enable-shared --enable-libx264
    
  libavutil      55. 24.100 / 55. 24.100
    
  libavcodec     57. 39.100 / 57. 39.100
    
  libavformat    57. 36.100 / 57. 36.100
    
  libavdevice    57.  0.101 / 57.  0.101
    
  libavfilter     6. 45.100 /  6. 45.100
    
  libswscale      4.  1.100 /  4.  1.100
    
  libswresample   2.  0.101 /  2.  0.101
    
  libpostproc    54.  0.100 / 54.  0.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'n1.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.36.100
    
  Duration: 00:04:56.36, start: 0.000000, bitrate: 628 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 500 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 120 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

    I play it with VLC, there is no noise at all.

    Here is my first question, in case we don’t specify the bitrate in the ffmpeg command, how are the bitrate of video/audio determined ?

    Since I am developping in Golang, I use the framework gmf which calls ffmpeg, I use go to convert the video format without specifying bitrate, then I got the following info :

    $ ffprobe mv2.mp4 
    
ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
    
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
    
  configuration: --enable-gpl --enable-shared --enable-libx264
    
  libavutil      55. 24.100 / 55. 24.100
    
  libavcodec     57. 39.100 / 57. 39.100
    
  libavformat    57. 36.100 / 57. 36.100
    
  libavdevice    57.  0.101 / 57.  0.101
    
  libavfilter     6. 45.100 /  6. 45.100
    
  libswscale      4.  1.100 /  4.  1.100
    
  libswresample   2.  0.101 /  2.  0.101
    
  libpostproc    54.  0.100 / 54.  0.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'mv2.mp4':
    
  Metadata:
    
    major_brand     : isom
    
    minor_version   : 512
    
    compatible_brands: isomiso2avc1mp41
    
    encoder         : Lavf57.36.100
    
  Duration: 00:04:56.36, start: 0.000000, bitrate: 540 kb/s
    
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480, 385 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
    
    Metadata:
    
      handler_name    : VideoHandler
    
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 147 kb/s (default)
    
    Metadata:
    
      handler_name    : SoundHandler

    The bitrate is event better than mv1.mp4, however when I play it with VLC, I heard a lot of noises in the audio, it seems some part of the audio becomes uncontinuous.

    Here is the gmf example code that I use for the video format conversion
    transcode.go

    Here are the two files :

    mv.webm - original file
    mv2.mp4 which has a lot of noises

    I tried to increase the audio bitrate (the way we set bitrate is like AVCodecContext->bit_rate = 441000...), but this doesn’t help.

    What should I do to call ffmpeg code to make it exactly the same as command line

    ffmpeg -i mv.webm  -acodec aac -vcodec libx264  mv1.mp4

     ?

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