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  • Mise à jour de la version 0.1 vers 0.2

    24 juin 2013, par

    Explications des différents changements notables lors du passage de la version 0.1 de MediaSPIP à la version 0.3. Quelles sont les nouveautés
    Au niveau des dépendances logicielles Utilisation des dernières versions de FFMpeg (>= v1.2.1) ; Installation des dépendances pour Smush ; Installation de MediaInfo et FFprobe pour la récupération des métadonnées ; On n’utilise plus ffmpeg2theora ; On n’installe plus flvtool2 au profit de flvtool++ ; On n’installe plus ffmpeg-php qui n’est plus maintenu au (...)

  • Les tâches Cron régulières de la ferme

    1er décembre 2010, par

    La gestion de la ferme passe par l’exécution à intervalle régulier de plusieurs tâches répétitives dites Cron.
    Le super Cron (gestion_mutu_super_cron)
    Cette tâche, planifiée chaque minute, a pour simple effet d’appeler le Cron de l’ensemble des instances de la mutualisation régulièrement. Couplée avec un Cron système sur le site central de la mutualisation, cela permet de simplement générer des visites régulières sur les différents sites et éviter que les tâches des sites peu visités soient trop (...)

  • Des sites réalisés avec MediaSPIP

    2 mai 2011, par

    Cette page présente quelques-uns des sites fonctionnant sous MediaSPIP.
    Vous pouvez bien entendu ajouter le votre grâce au formulaire en bas de page.

Sur d’autres sites (12124)

  • FFMPEG Duration convert To Frames On PowerShell

    1er juin 2020, par Ilhamzac

    I am new to FFMPEG

    



    I need to get some duration's of some audio and video files.
my out put comes in seconds and milliseconds i guess

    



    I need to convert this milliseconds into frames
FrameCount should be 30

    



    This is my code

    



    $audioId = "$id.m4a"
$dur = ffprobe $audioId -show_format 2>&1 | sed -n 's/duration=//p'
echo $dur


    



    Output be like, its a loop running

    



    3.478042
3.455979
3.522021



    



    my question is , i need to keep three seconds as it is , on the right side the milliseconds i need it to be divided by 30

    



    that means i need the answers be like

    



    3.15934
3.15199
3.17400


    



    thank you

    


  • ffmpeg filter_complex 100 buffers queued

    20 novembre 2019, par Artem Kuznetsov

    I need to mount complex video scenes from several files.
    Many trims, atrims, overlays, amix.
    Maybe it needs to be divided into several calls ?

    And I use something like this :

    ffmpeg -y -i 1.mp4 -i 2.mp4 -i 3.mp4 -i 4.mp4 -i 5.mp4 -i 6.mp4 -i 7.mp4 -i 8 -i 9.mp4 -i 10.mp4 -i 11.mp4 -i 12.mp4 -i 13.mp4 -f lavfi -i anullsrc \
    
-filter_complex "[0:v]trim=end=0.456, setpts=PTS-STARTPTS[out0]; \
    
[0:a]atrim=end=0.456, asetpts=PTS-STARTPTS[a_out0]; \
    
[0:v]trim=0.456:969.456, setpts=PTS-STARTPTS, scale=iw*180/ih:180[out1]; \
    
[1:v][out1]overlay=main_w-overlay_w-0:main_h-overlay_h:eof_action=pass[out2]; \
    
[0:a]atrim=0.456:969.456, asetpts=PTS-STARTPTS[a_out1]; \
    
[0:v]trim=969.456:974.138, setpts=PTS-STARTPTS[out3]; \
    
[0:a]atrim=969.456:974.138, asetpts=PTS-STARTPTS[a_out2]; \
    
[0:v]trim=974.138:1382.138, setpts=PTS-STARTPTS, scale=iw*180/ih:180[out4]; \
    
[2:v][out4]overlay=main_w-overlay_w-0:main_h-overlay_h:eof_action=pass[out5]; \
    
[0:a]atrim=974.138:1382.138, asetpts=PTS-STARTPTS[a_out3]; \
    
[2:a][a_out3]amix=inputs=2:duration=first[a_out4]; \
    
[0:v]trim=1382.138:1408.739, setpts=PTS-STARTPTS[out6]; \
    
[0:a]atrim=1382.138:1408.739, asetpts=PTS-STARTPTS[a_out5]; \
    
[0:v]trim=1408.739:2089.758, setpts=PTS-STARTPTS, scale=iw*180/ih:180[out7]; \
    
[3:v][out7]overlay=main_w-overlay_w-0:main_h-overlay_h:eof_action=pass[out8]; \
    
[0:a]atrim=1408.739:2089.758, asetpts=PTS-STARTPTS[a_out6]; \
    
[3:a][a_out6]amix=inputs=2:duration=first[a_out7]; \
    
[0:v]trim=2089.758:2112.511, setpts=PTS-STARTPTS[out9]; \
    
[0:a]atrim=2089.758:2112.511, asetpts=PTS-STARTPTS[a_out8]; \
    
[0:v]trim=2112.511:2358.528, setpts=PTS-STARTPTS, scale=iw*180/ih:180[out10]; \
    
[4:v][out10]overlay=main_w-overlay_w-0:main_h-overlay_h:eof_action=pass[out11]; \
    
[0:a]atrim=2112.511:2358.528, asetpts=PTS-STARTPTS[a_out9]; \
    
...
    
[9:v]trim=18.184:36.184, setpts=PTS-STARTPTS, scale=iw*180/ih:180[out26]; \
    
[10:v][out26]overlay=main_w-overlay_w-0:main_h-overlay_h:eof_action=pass[out27]; \
    
[9:a]atrim=18.184:36.184, asetpts=PTS-STARTPTS[a_out18]; \
    
[9:v]trim=36.184:90.237, setpts=PTS-STARTPTS[out28]; \
    
[9:a]atrim=36.184:90.237, asetpts=PTS-STARTPTS[a_out19]; \
    
[9:v]trim=90.237:345.237, setpts=PTS-STARTPTS, scale=iw*180/ih:180[out29]; \
    
[11:v][out29]overlay=main_w-overlay_w-0:main_h-overlay_h:eof_action=pass[out30]; \
    
[9:a]atrim=90.237:345.237, asetpts=PTS-STARTPTS[a_out20]; \
    
[9:v]trim=345.237:414.306, setpts=PTS-STARTPTS[out31]; \
    
[9:a]atrim=345.237:414.306, asetpts=PTS-STARTPTS[a_out21]; \
    
[9:v]trim=414.306:585.314, setpts=PTS-STARTPTS, scale=iw*180/ih:180[out32]; \
    
[12:v][out32]overlay=main_w-overlay_w-0:main_h-overlay_h:eof_action=pass[out33]; \
    
[9:a]atrim=414.306:585.314, asetpts=PTS-STARTPTS[a_out22]; \
    
[12:a][a_out22]amix=inputs=2:duration=first[a_out23]; \
    
[9:v]trim=start=585.314, setpts=PTS-STARTPTS[out34]; \
    
[9:a]atrim=start=585.314, asetpts=PTS-STARTPTS[a_out24]; \
    
[out0][a_out0][out2][a_out1][out3][a_out2][out5][a_out4][out6][a_out5][out8][a_out7][out9][a_out8][out11][a_out10][out12][a_out11][out14][a_out12][out15][a_out13][out18][a_out14][out19][silence0][out20][a_out15][out23][a_out16][out24][silence1][out25][a_out17][out27][a_out18][out28][a_out19][out30][a_out20][out31][a_out21][out33][a_out23][out34][a_out24]concat=n=23:v=1:a=1[out][a_out]" \
    
-map [out] -map [a_out] \
    
-c:v libx264 -crf 18 -pix_fmt yuv420p \
    
-c:a aac \
    
output.mp4

    Is this optimal ? Can this be written somehow differently ?
    Now I have this problem :

    [out_0_1 @ 0x7fb2b8ca96c0] 100 buffers queued in out_0_1, something may be wrong. speed=8.51x      
    
[out_0_1 @ 0x7fb2b8ca96c0] 1000 buffers queued in out_0_1, something may be wrong.

  • Concatenating two AAC decreasing number of frames packets

    29 décembre 2019, par Ahmed Hawary

    I am trying to concatenate two m4a (aac) files using the FFmpeg with the following command :

    ffmpeg -f concat -i input.txt -codec copy output.m4a

    the first file number of frames using afinfo on macOS :

    File type ID:   m4af
    
Data format:     1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
no channel layout.
    
estimated duration: 8.473832 sec
    
audio bytes: 68931
    
audio packets: 367
    
bit rate: 64710 bits per second
    
packet size upper bound: 391
    
maximum packet size: 391
    
audio data file offset: 44
    
not optimized
    
audio 373696 valid frames + 2112 priming + 0 remainder = 375808
    
format list:
    
[ 0] format:      1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
Channel layout: Mono

    The second file metadata :

    File type ID:   m4af
    
Num Tracks:     1
    
Data format:     1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
no channel layout.
    
estimated duration: 5.594558 sec
    
audio bytes: 46077
    
audio packets: 243
    
bit rate: 65329 bits per second
    
packet size upper bound: 340
    
maximum packet size: 340
    
audio data file offset: 44
    
not optimized
    
audio 246720 valid frames + 2112 priming + 0 remainder = 248832
    
format list:
    
[ 0] format:      1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
Channel layout: Mono

    the resulted audio files metadate :

    File type ID:   m4af
    
Num Tracks:     1
    
Data format:     1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
no channel layout.
    
estimated duration: 14.070998 sec
    
audio bytes: 122792
    
audio packets: 607
    
bit rate: 69696 bits per second
    
packet size upper bound: 293
    
maximum packet size: 293
    
audio data file offset: 40
    
not optimized
    
audio 620531 valid frames + 1024 priming + 13 remainder = 621568
    
format list:
    
[ 0] format:      1 ch,  44100 Hz, 'aac ' (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame
    
Channel layout: Mono

    The problem is that the total number of frames should be 367+293 = 610 and the resulted number of frames is 607. And the duration is 14.070998 sec instead of 14.06839 sec

    Any ideas if I am doing anything wrong here ? I need to precisely concatenate the two files without any loss or gain in the input frames.

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