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  • Participer à sa traduction

    10 avril 2011

    Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
    Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
    Actuellement MediaSPIP n’est disponible qu’en français et (...)

  • Les statuts des instances de mutualisation

    13 mars 2010, par

    Pour des raisons de compatibilité générale du plugin de gestion de mutualisations avec les fonctions originales de SPIP, les statuts des instances sont les mêmes que pour tout autre objets (articles...), seuls leurs noms dans l’interface change quelque peu.
    Les différents statuts possibles sont : prepa (demandé) qui correspond à une instance demandée par un utilisateur. Si le site a déjà été créé par le passé, il est passé en mode désactivé. publie (validé) qui correspond à une instance validée par un (...)

  • Other interesting software

    13 avril 2011, par

    We don’t claim to be the only ones doing what we do ... and especially not to assert claims to be the best either ... What we do, we just try to do it well and getting better ...
    The following list represents softwares that tend to be more or less as MediaSPIP or that MediaSPIP tries more or less to do the same, whatever ...
    We don’t know them, we didn’t try them, but you can take a peek.
    Videopress
    Website : http://videopress.com/
    License : GNU/GPL v2
    Source code : (...)

Sur d’autres sites (4679)

  • Something wrong with recording video and merging audio and video by using FFmpeg

    16 mars 2014, par VaFancy

    At the beginning, I have to admit that I am a newbie in FFmpeg and also raspbian. Unfortunately, I need to solve this problem as soon as possible.

    I tried to record videos on my Raspberry PI by using PIcamera and USB webcam separately. When I used command :

    ffmpeg -t 10 -f video4linux2 -r 25 -i /dev/video0 out.h264

    However, something was very wrong because there was nothing saved in that video file. I copy the log.

    [video4linux2,v4l2 @ 0x267ad80] The driver changed the time per frame from 1/25 to 1/30
    Input #0, video4linux2,v4l2, from '/dev/video0':
     Duration: N/A, start: 37562.908020, bitrate: 147456 kb/s
       Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 147456 kb/s, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
    File 'out.h264' already exists. Overwrite ? [y/N] y
    No pixel format specified, yuv422p for H.264 encoding chosen.
    Use -pix_fmt yuv420p for compatibility with outdated media players.
    [libx264 @ 0x267ca60] using cpu capabilities: ARMv6
    [libx264 @ 0x267ca60] profile High 4:2:2, level 3.0, 4:2:2 8-bit
    Output #0, h264, to 'out.h264':
     Metadata:
       encoder         : Lavf55.34.101
       Stream #0:0: Video: h264 (libx264), yuv422p, 640x480, q=-1--1, 90k tbn, 25 tbc
    Stream mapping:
     Stream #0:0 -> #0:0 (rawvideo -> libx264)
    Press [q] to stop, [?] for help
    DTS 140730117085, next:18648459 st:0 invalid dropping00.00 bitrate=N/A    
    PTS 140730117085, next:18648459 invalid dropping st:0
    DTS 140740452429, next:6538221 st:0 invalid dropping:00.00 bitrate=N/A    
    PTS 140740452429, next:6538221 invalid dropping st:0
    frame=   44 fps=1.1 q=28.0 size=      18kB time=00:00:00.04 bitrate=3712.6kbits/
    frame=   45 fps=0.8 q=28.0 size=      22kB time=00:00:00.08 bitrate=2262.8kbits/
    frame=   46 fps=0.7 q=28.0 size=      27kB time=00:00:00.12 bitrate=1813.4kbits/
    frame=   47 fps=0.6 q=28.0 size=      30kB time=00:00:00.16 bitrate=1559.2kbits/
    frame=   48 fps=0.5 q=28.0 size=      34kB time=00:00:00.20 bitrate=1397.7kbits/
    frame=   49 fps=0.5 q=28.0 size=      39kB time=00:00:00.24 bitrate=1324.3kbits/
    frame=   50 fps=0.4 q=28.0 size=      42kB time=00:00:00.28 bitrate=1241.7kbits/

    As you can see, the size of each frame was only few kb and fps was so low. Thus each frame was taken by using a long time(4s in this case). Then I test my USB webcam by using command :

    ffplay -f video4linux2 /dev/video0

    The video stream which displayed on the screen was so unnatural. It looked like there was such a long delay between each frame. So the video was hardly moved. The log I have copied :

    pi@raspberrypi ~ $ ffplay -f video4linux2 /dev/video0
    avplay version 0.8.6-6:0.8.6-1+rpi1, Copyright (c) 2003-2013 the Libav developers
     built on Mar 31 2013 13:58:10 with gcc 4.6.3
    [video4linux2 @ 0x1ad83c0] Estimating duration from bitrate, this may be inaccurate
    Input #0, video4linux2, from '/dev/video0':
     Duration: N/A, start: 36107.154598, bitrate: 147456 kb/s
       Stream #0.0: Video: rawvideo, yuyv422, 640x480, 147456 kb/s, 30 tbr, 1000k tbn, 30 tbc
    [avsink @ 0x1ad9960] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out'
    [scale @ 0x1ad9d20] w:640 h:480 fmt:yuyv422 -> w:640 h:480 fmt:yuv420p flags:0x4
    36163.33 A-V:  0.000 s:0.0 aq=    0KB vq= 3000KB sq=    0B f=8/8

    The final problem is that no sound was added in video file when I tried to add some in. I used the command :

    avconv -i /home/pi/Desktop/test.mp4 -i /home/pi/Desktop/test.mp3 -map 0:0 -vcodec copy -map 1:0 -acodec copy output.h264

    I have checked separately that the video and audio were both able to play. However the merged file only had video but no audio. The log is shown below, it seems that the process was succeed.

    pi@raspberrypi ~ $ avconv -i /home/pi/Desktop/test.mp4 -i /home/pi/Desktop/test.mp3 -map 0:0 -vcodec copy -map 1:0 -acodec copy output.h264
    avconv version 0.8.6-6:0.8.6-1+rpi1, Copyright (c) 2000-2013 the Libav developers
     built on Mar 31 2013 13:58:10 with gcc 4.6.3
    [h264 @ 0x1aa6680] max_analyze_duration reached
    [h264 @ 0x1aa6680] Estimating duration from bitrate, this may be inaccurate
    Input #0, h264, from '/home/pi/Desktop/test.mp4':
     Duration: N/A, bitrate: N/A
       Stream #0.0: Video: h264 (High), yuv420p, 640x480, 25 fps, 25 tbr, 1200k tbn, 2400k tbc
    [mp3 @ 0x1aaa120] Estimating duration from bitrate, this may be inaccurate
    Input #1, mp3, from '/home/pi/Desktop/test.mp3':
     Metadata:
       title           : We Are Legend (Original Mix)
       artist          : Dimitri Vegas & Like Mike & Steve Aoki
       album           : SPINNIN RECORDS
       encoded_by      : iTunes 11.1.5
       TBP             : 128
       genre           : Progressive House
       date            : 2013
     Duration: 00:00:11.14, start: 0.000000, bitrate: 191 kb/s
       Stream #1.0: Audio: mp3, 48000 Hz, stereo, s16, 192 kb/s
    File 'output.h264' already exists. Overwrite ? [y/N] y
    Output #0, h264, to 'output.h264':
     Metadata:
       encoder         : Lavf53.21.1
       Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 90k tbn, 1200k tbc
       Stream #0.1: Audio: libmp3lame, 48000 Hz, stereo, 192 kb/s
    Stream mapping:
     Stream #0:0 -> #0:0 (copy)
     Stream #1:0 -> #0:1 (copy)
    Press ctrl-c to stop encoding
    frame=  144 fps=  0 q=-1.0 Lsize=    1684kB time=5.06 bitrate=2724.7kbits/s    
    video:1566kB audio:119kB global headers:0kB muxing overhead 0.000000%

    Could somebody helps me, please. And sorry for my bad english.

    BTW : When I finally captured 200 frames, it turns out all of them were green.

  • cut fragment of mp3 from, to and apply fade in, fade out

    23 juin 2021, par Mevia

    I am in need of a tool that will complete following tasks

    


      

    1. provide source mp3
    2. 


    3. provide data of fragment to cut (start time, end time)
    4. 


    5. start cutting from source where start is
    6. 


    7. end cutting from source where end is
    8. 


    9. apply X seconds fade in beginning of start time
    10. 


    11. apply Y seconds fade out beginning Y seconds subtracted from end
    12. 


    


    By fade i mean increase / decrease volume between 0 - 100% / 100% - 0

    


    I already have an interface that allows to prepare data.

    


    On the backend side, I am working with node.js.

    


    After some research i found a tool FFmpeg that was widely recommended to achieve my goal.

    


    After further research i came up with following code :

    


    const { stdout, stderr } = await exec('ffmpeg -y -i "' + sourcePath + '" -ss ' + start + ' -to ' + end + ' -af "afade=in:st=' + start + ':d=' + fadeLength + ',afade=out:st=' + (end - 3) + ':d=' + fadeLength + '" "' + outputPath + '"');

/*
Breakdown:
    ffmpeg
        -y
        -i "sourcePath"
        -ss start
        -to end
        -af "afade=in:st=(start):d=(fadeLength),afade=out:st=(end - fadeLength):d=3"
        "outputPath"

Example:
    ffmpeg
        -y
        -i "/source/test.mp3"
        -ss 89.382716
        -to 172.817231
        -af "afade=in:st=89.382716:d=5,afade=out:st=(172.817231 - 5):d=5"
        "/output/test_cut.mp3"
*/


    


    It's working, but processor throws warnings :

    


    [mp3 @ 0x7fc50a80c800] Estimating duration from bitrate, this may be inaccurate
[swscaler @ 0x114800000] deprecated pixel format used, make sure you did set range correctly
[mp3 @ 0x7fc50a819a00] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2


    


    Full log :

    


    ffmpeg version 4.4 Copyright (c) 2000-2021 the FFmpeg developers
    built with Apple clang version 12.0.0 (clang-1200.0.32.29)
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.4_2 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-avresample --enable-videotoolbox
    libavutil      56. 70.100 / 56. 70.100
    libavcodec     58.134.100 / 58.134.100
    libavformat    58. 76.100 / 58. 76.100
    libavdevice    58. 13.100 / 58. 13.100
    libavfilter     7.110.100 /  7.110.100
    libavresample   4.  0.  0 /  4.  0.  0
    libswscale      5.  9.100 /  5.  9.100
    libswresample   3.  9.100 /  3.  9.100
    libpostproc    55.  9.100 / 55.  9.100
[mp3 @ 0x7fc50a80c800] Estimating duration from bitrate, this may be inaccurate
Input #0, mp3, from 'test.mp3':
    Metadata:
        title           : Test
        artist          : Test
    Duration: 00:03:22.50, start: 0.000000, bitrate: 320 kb/s
    Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s
    Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 320x240, 90k tbr, 90k tbn, 90k tbc (attached pic)
        Metadata:
            comment         : Other
Stream mapping:
    Stream #0:1 -> #0:0 (mjpeg (native) -> png (native))
    Stream #0:0 -> #0:1 (mp3 (mp3float) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
[swscaler @ 0x114800000] deprecated pixel format used, make sure you did set range correctly
[mp3 @ 0x7fc50a819a00] Frame rate very high for a muxer not efficiently supporting it.
Please consider specifying a lower framerate, a different muxer or -vsync 2
Output #0, mp3, to 'test (CUT).mp3':
    Metadata:
        TIT2            : Test
        TPE1            : Test
        TSSE            : Lavf58.76.100
    Stream #0:0: Video: png, rgb24, 320x240, q=2-31, 200 kb/s, 90k fps, 90k tbn (attached pic)
        Metadata:
            comment         : Other
            encoder         : Lavc58.134.100 png
    Stream #0:1: Audio: mp3, 44100 Hz, stereo, fltp
        Metadata:
            encoder         : Lavc58.134.100 libmp3lame
frame=    0 fps=0.0 q=0.0 size=       0kB time=00:00:39.89 bitrate=   0.0kbits/s speed=31.4x    
[mp3 @ 0x7fc50a819a00] No packets were sent for some of the attached pictures.
frame=    0 fps=0.0 q=0.0 Lsize=     626kB time=00:00:40.02 bitrate= 128.2kbits/s speed=31.4x    
video:0kB audio:626kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.053810%


    


    So my question is, how to change execution command to have it executed properly (avoid warnings and errors) ?

    


    Important thing is that i am only interested in cutting it, if possible avoid re-encondig because i am already working with 320 mp3's that have been produced from master wav's.

    


  • Merge commit ’6d93307f8df81808f0dcdbc064b848054a6e83b3’

    26 janvier 2014, par Michael Niedermayer
    Merge commit ’6d93307f8df81808f0dcdbc064b848054a6e83b3’
    

    * commit ’6d93307f8df81808f0dcdbc064b848054a6e83b3’ :
    mpeg12 : check scantable indices in all decode_block functions

    Benchmarks

    Before :
    1878 decicycles in mpeg2_decode_block_non_intra, 8388487 runs, 121 skips
    1700 decicycles in mpeg2_decode_block_intra, 4194239 runs, 65 skips
    1808 decicycles in mpeg2_fast_decode_block_non_intra, 8388492 runs, 116 skips
    1669 decicycles in mpeg2_fast_decode_block_intra, 4194248 runs, 56 skips
    — 
    2056 decicycles in mpeg1_decode_block_inter, 65535 runs, 1 skips
    2346 decicycles in mpeg1_decode_block_intra, 32768 runs, 0 skips
    2011 decicycles in mpeg1_fast_decode_block_inter, 65533 runs, 3 skips


    After :
    1858 decicycles in mpeg2_decode_block_non_intra, 8388490 runs, 118 skips
    1691 decicycles in mpeg2_decode_block_intra, 4194233 runs, 71 skips
    1823 decicycles in mpeg2_fast_decode_block_non_intra, 8388493 runs, 115 skips
    1681 decicycles in mpeg2_fast_decode_block_intra, 4194238 runs, 66 skips
    — 
    2010 decicycles in mpeg1_decode_block_inter, 65535 runs, 1 skips
    2322 decicycles in mpeg1_decode_block_intra, 32766 runs, 2 skips
    1995 decicycles in mpeg1_fast_decode_block_inter, 65535 runs, 1 skips

    All benchmarks are the best scores of several runs

    Merged-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavcodec/mpeg12dec.c