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    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
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  • Support audio et vidéo HTML5

    10 avril 2011

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    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

  • 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
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Sur d’autres sites (9342)

  • FFmpeg audio stream extraction on non-interleaved AVI - slow compared to AviSynth

    8 mai 2019, par LLL

    I want to extract the audio stream of an avi file as a wav file, it works but it is really slow ( 4-5fps) although I just want to copy the stream.

    Here is the type of stream I want to extract (ffprobe info) :
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s

    Going through AviSynth does it about 100 times faster, but I would prefer a pure FFmpeg solution. Why such a speed difference ? It looks like FFmpeg is reading and processing through the whole file whereas AviSynth can just extract the data without reading it.

    Example :
    ffmpeg -i file.avi -vn -ac 2 -c:a copy audio.wav
    or
    ffmpeg -i file.avi -map 0:a -ac 2 -c:a copy audio.wav
    both work fine but take time.

    Using an AviSynth script as input :
    ffmpeg -i script.avs -map 0:a -ac 2 -c:a copy audio.wav
    with script.avs containing just :
    AviSource("file.avi")
    does the same but almost instantaneously !

    Any idea why AviSynth is so much faster and if there is a way to get the same speed in FFmpeg ?

    Edit : adding logs
    Using FFmpeg directly :

    E:\>ffmpeg -i "file.avi" -map 0:a -c:a copy -y -benchmark "output.wav"
    
ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 8.2.1 (GCC) 20181201
    
  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 --enable-libopenmpt
    
  libavutil      56. 25.100 / 56. 25.100
    
  libavcodec     58. 43.100 / 58. 43.100
    
  libavformat    58. 25.100 / 58. 25.100
    
  libavdevice    58.  6.101 / 58.  6.101
    
  libavfilter     7. 47.100 /  7. 47.100
    
  libswscale      5.  4.100 /  5.  4.100
    
  libswresample   3.  4.100 /  3.  4.100
    
  libpostproc    55.  4.100 / 55.  4.100
    
[avi @ 0000018d3c38a680] non-interleaved AVI
    
Guessed Channel Layout for Input Stream #0.1 : stereo
    
Input #0, avi, from 'file.avi':
    
  Duration: 00:18:37.49, start: 0.000000, bitrate: 534682 kb/s
    
    Stream #0:0: Video: rawvideo, bgr24, 1280x720, 533183 kb/s, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
    
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    
Output #0, wav, to 'output.wav':
    
  Metadata:
    
    ISFT            : Lavf58.25.100
    
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    
Stream mapping:
    
  Stream #0:1 -> #0:0 (copy)
    
Press [q] to stop, [?] for help
    
size=  192445kB time=00:18:37.12 bitrate=1411.2kbits/s speed=4.77x
    
video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
    
bench: utime=1.188s stime=50.766s rtime=234.254s
    
bench: maxrss=17468kB

    Using AviSynth :

    E:\>ffmpeg -i "soundout.avs" -map 0:a -c:a copy -y -benchmark "output.wav"
    
ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 8.2.1 (GCC) 20181201
    
  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 --enable-libopenmpt
    
  libavutil      56. 25.100 / 56. 25.100
    
  libavcodec     58. 43.100 / 58. 43.100
    
  libavformat    58. 25.100 / 58. 25.100
    
  libavdevice    58.  6.101 / 58.  6.101
    
  libavfilter     7. 47.100 /  7. 47.100
    
  libswscale      5.  4.100 /  5.  4.100
    
  libswresample   3.  4.100 /  3.  4.100
    
  libpostproc    55.  4.100 / 55.  4.100
    
Guessed Channel Layout for Input Stream #0.1 : stereo
    
Input #0, avisynth, from 'soundout.avs':
    
  Duration: 00:18:37.49, start: 0.000000, bitrate: N/A
    
    Stream #0:0: Video: rawvideo (BGR[24] / 0x18524742), bgr24, 1280x720, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
    
    Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    
Output #0, wav, to 'output.wav':
    
  Metadata:
    
    ISFT            : Lavf58.25.100
    
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    
Stream mapping:
    
  Stream #0:1 -> #0:0 (copy)
    
Press [q] to stop, [?] for help
    
size=  192445kB time=00:18:37.11 bitrate=1411.2kbits/s speed= 155x
    
video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
    
bench: utime=0.234s stime=1.047s rtime=7.236s
    
bench: maxrss=23792kB

    Edit : tests after "reencoding" AVI file :
    Onto something...
    Say my original file is f.avi. Here is ffprobe’s results :

    [avi @ 0x55a9c4b1e740] non-interleaved AVI
    
Input #0, avi, from 'f.avi':
    
  Duration: 00:00:38.18, start: 0.000000, bitrate: 1104582 kb/s
    
    Stream #0:0: Video: rawvideo, bgr24, 1632x1200, 1104265 kb/s, 23.47 fps, 23.47 tbr, 23.47 tbn, 23.47 tbc
    
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s

    Extracting audio takes a long time.
    Now if I "reencode" the file in another AVI :

    ffmpeg -i f.avi -c copy f2.avi

    I can extract the audio from f2.avi in milliseconds !
    FFprobe on f2.avi :

    Input #0, avi, from 'f2.avi':
    
  Metadata:
    
    encoder         : Lavf57.56.101
    
  Duration: 00:00:38.18, start: 0.000000, bitrate: 1104456 kb/s
    
    Stream #0:0: Video: rawvideo, bgr24, 1632x1200, 1104265 kb/s, 23.47 fps, 23.47 tbr, 23.47 tbn, 23.47 tbc
    
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s

    It’s the same apart from the Metadata, which shouldn’t make a difference, but with this comparison I see the problem must have to do with the fact that the original is non-interleaved !
    I would assume it was easier to read and extract the audio from a non-interleaved file but maybe this is not conforming to AVI standards, hence the extra work needed ?

  • Why is audio stream extraction slow ? (compared to AviSynth)

    6 mai 2019, par LLL

    I want to extract the audio stream of an avi file as a wav file, it works but it is really slow ( 4-5fps) although I just want to copy the stream.

    Here is the type of stream I want to extract (ffprobe info) :
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s

    Going through AviSynth does it about 100 times faster, but I would prefer a pure FFmpeg solution. Why such a speed difference ? It looks like FFmpeg is reading and processing through the whole file whereas AviSynth can just extract the data without reading it.

    Example :
    ffmpeg -i file.avi -vn -ac 2 -c:a copy audio.wav
    or
    ffmpeg -i file.avi -map 0:a -ac 2 -c:a copy audio.wav
    both work fine but take time.

    Using an AviSynth script as input :
    ffmpeg -i script.avs -map 0:a -ac 2 -c:a copy audio.wav
    with script.avs containing just :
    AviSource("file.avi")
    does the same but almost instantaneously !

    Any idea why AviSynth is so much faster and if there is a way to get the same speed in FFmpeg ?

    Edit : adding logs
    Using FFmpeg directly :

    E:\>ffmpeg -i "file.avi" -map 0:a -c:a copy -y -benchmark "output.wav"
    
ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 8.2.1 (GCC) 20181201
    
  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 --enable-libopenmpt
    
  libavutil      56. 25.100 / 56. 25.100
    
  libavcodec     58. 43.100 / 58. 43.100
    
  libavformat    58. 25.100 / 58. 25.100
    
  libavdevice    58.  6.101 / 58.  6.101
    
  libavfilter     7. 47.100 /  7. 47.100
    
  libswscale      5.  4.100 /  5.  4.100
    
  libswresample   3.  4.100 /  3.  4.100
    
  libpostproc    55.  4.100 / 55.  4.100
    
[avi @ 0000018d3c38a680] non-interleaved AVI
    
Guessed Channel Layout for Input Stream #0.1 : stereo
    
Input #0, avi, from 'file.avi':
    
  Duration: 00:18:37.49, start: 0.000000, bitrate: 534682 kb/s
    
    Stream #0:0: Video: rawvideo, bgr24, 1280x720, 533183 kb/s, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
    
    Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    
Output #0, wav, to 'output.wav':
    
  Metadata:
    
    ISFT            : Lavf58.25.100
    
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    
Stream mapping:
    
  Stream #0:1 -> #0:0 (copy)
    
Press [q] to stop, [?] for help
    
size=  192445kB time=00:18:37.12 bitrate=1411.2kbits/s speed=4.77x
    
video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
    
bench: utime=1.188s stime=50.766s rtime=234.254s
    
bench: maxrss=17468kB

    Using AviSynth :

    E:\>ffmpeg -i "soundout.avs" -map 0:a -c:a copy -y -benchmark "output.wav"
    
ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
    
  built with gcc 8.2.1 (GCC) 20181201
    
  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 --enable-libopenmpt
    
  libavutil      56. 25.100 / 56. 25.100
    
  libavcodec     58. 43.100 / 58. 43.100
    
  libavformat    58. 25.100 / 58. 25.100
    
  libavdevice    58.  6.101 / 58.  6.101
    
  libavfilter     7. 47.100 /  7. 47.100
    
  libswscale      5.  4.100 /  5.  4.100
    
  libswresample   3.  4.100 /  3.  4.100
    
  libpostproc    55.  4.100 / 55.  4.100
    
Guessed Channel Layout for Input Stream #0.1 : stereo
    
Input #0, avisynth, from 'soundout.avs':
    
  Duration: 00:18:37.49, start: 0.000000, bitrate: N/A
    
    Stream #0:0: Video: rawvideo (BGR[24] / 0x18524742), bgr24, 1280x720, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
    
    Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    
Output #0, wav, to 'output.wav':
    
  Metadata:
    
    ISFT            : Lavf58.25.100
    
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
    
Stream mapping:
    
  Stream #0:1 -> #0:0 (copy)
    
Press [q] to stop, [?] for help
    
size=  192445kB time=00:18:37.11 bitrate=1411.2kbits/s speed= 155x
    
video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
    
bench: utime=0.234s stime=1.047s rtime=7.236s
    
bench: maxrss=23792kB

  • Replace Special Characters In Batch-File Variable Feeding ffmpeg program

    14 janvier 2019, par whereswaller

    I am attempting to write a batch-file that leverages ffmpeg.exe to convert all files in a folder structure to mp3 format (specifically 128 KBps).

    My batch-file is presently unable to process filenames (constructed by concatenating the %_SOURCE% and %%~F variables) containing certain special characters generating the following errors :

    No such file or directory

    • … ellipsis sign
    • – en dash
    • — em dash
    • − minus sign

    Invalid argument

    • ‘ and ’ curved single quotation marks
    • “ and ” curved double quotation marks

    Invalid argument (yet sometimes passes depending on where symbol is in the filename, for example, seems to work if placed between the n and t of Dont in C:\Users\Test\Documents\Input\Peter Bjorn And John - I Know You Dont Love Me.mp3)

    • - hyphen
    • ! exclamation mark
    • ~ tilde
    • ' non-curved single quotation mark
    • = equals sign
    • + plus sign
    • % percentage sign
    • ( open bracket

    How can I modify my batch-file script so that the %%~F variable escapes these characters correctly ?

    Example current filename input : C:\Users\Test\Documents\Input\Peter Bjorn And John - I Know You Don't Love Me.mp3

    Example desired filename input : C:\Users\Test\Documents\Input\Peter Bjorn And John - I Know You Don"^'"t Love Me.mp3

    Script (see line beginning C:\ffmpeg\bin\ffmpeg.exe) :

    @echo off
    
setlocal EnableExtensions DisableDelayedExpansion
    

    
rem // Define constants here:
    
set "_SOURCE=C:\Users\Test\Documents\Input" & rem // (absolute source path)
    
set "_TARGET=C:\Users\Test\Documents\Output"  & rem // (absolute target path)
    
set "_PATTERN=*.*" & rem // (pure file pattern for input files)
    
set "_FILEEXT=.mp3"   & rem // (pure file extension of output files)
    

    
pushd "%_TARGET%" || exit /B 1
    
for /F "delims=" %%F in ('
    
    cd /D "%_SOURCE%" ^&^& ^(rem/ list but do not copy: ^
    
        ^& xcopy /L /S /Y /I ".\%_PATTERN%" "%_TARGET%" ^
    
        ^| find ".\" ^& rem/ remove summary line;
    
    ^)
    
') do (
    
    2> nul mkdir "%%~dpF."
    

    
    rem // Set up the correct `ffmpeg` command line here:
    
    set "FFREPORT=file=C\:\\Users\\Test\\Documents\\Output\\ffreport-%%~F.log:level=32"
    
    "C:\ffmpeg\bin\ffmpeg.exe" -report -n -i "%_SOURCE%\%%~F" -vn -c:a libmp3lame -b:a 128k "%%~dpnF%_FILEEXT%"
    
    if not errorlevel 1  if exist "%%~dpnF%_FILEEXT%" del /f /q "%_SOURCE%\%%~F"
    

    
)
    
popd
    

    
endlocal
    
pause

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