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    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 (...)

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    5 septembre 2013, par

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    21 juin 2013, par

    Présentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
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Sur d’autres sites (8870)

  • How the ffmpeg astats crest factor of an audio track is calculated

    29 août 2017, par FranGar

    I’m scripting a ffmpeg chain process for my work. The aim is normalizing/compressing lot of audio files (mp3’s).
    It’s done in Python and the critical part is the line :

    ffmpeg -y -i "Input.mp3" -codec:a libmp3lame -b:a 96k -af acompressor=threshold=-15dB:ratio=5:attack=0.01:release=1000:knee=2,dynaudnorm=g=3:m=2:p=0.95 "Output.mp3"

    The python script it’s complete and working BUT the nature of the audios (voice recordings) are very different so I can’t use the same params for all of them.

    I make some experimenting with the values of the ffmpeg filter astats and i discovered that the crest factor (Standard ratio of peak to RMS level ) gave a good reference to programatically get the better params.

    In fact I saw that a recording with a nice dynamic range sound and smooth in shape, get crest values around 9-15 (the compress/normlz params will be somehow conservative). But audios with crest around 22-30 need more aggressive processing.
    (All empirically)

    Somebody can clarify how the crest values are really calculated ? Which are the peaks taken to account ? (Why the flat factor is always 0 ?)
    Or if somebody knows how to get a value representing the sound ’smoothness’ will be nice also.

    Thanks for the ideas.

  • FFmpeg - concatenate variable length intro / outro fixed segments and separate audio track

    25 septembre 2017, par bobmarksie

    I’m trying to create an app which generates a video based on 3 main parts : -

    1. Intro - Variable length video ( 20 to 30 seconds long)
    2. Segments - 3 clips from a "segments video" - this is a fixed length video (always 400 seconds) which contains 100 separate videos all precisely 4 seconds in length e.g. "segment 1" is from 0 to 4 seconds and "segment 3" is from 8 to 12 seconds.
    3. Outro - Variable length video ( 10 to 20 seconds long)

    The user picks the intro, segments and outro from a list of videos and the app concatenates these together (and extracts 3 segments at random). The user also picks an audio file which plays for the entire video. The generated file should look something like the following : -

    GENERATED VIDEO

    VIDEO   | Intro (20 - 30) |  Seg 1 (4) |  Seg 2 (4) |  Seg 3 (4) |  Outro (10 - 20)  |
    
--------+-----------------+------------+------------+------------+-------------------+
    
AUDIO   |        Audio track which spans full video (e.g. 4 minutes long) ...        |

    I have been able to hack the following together using ffmpeg (which works) : -

    ffmpeg -y \
    
  -i audio/audio-19.m4a \
    
  -i videos/intro/intro-23.mkv \
    
  -i videos/segments/segments-88.mkv \
    
  -i videos/outro/outro-12.mkv \
    
  -filter_complex \
    
    "[1:v]trim=0:30,setpts=PTS-STARTPTS[v0]; \
    
    [0:a]atrim=0:30,asetpts=PTS-STARTPTS[a0]; \
    
    [2:v]trim=20:24,setpts=PTS-STARTPTS[v1]; \
    
    [0:a]atrim=30:34,asetpts=PTS-STARTPTS[a1]; \
    
    [2:v]trim=60:64,setpts=PTS-STARTPTS[v2]; \
    
    [0:a]atrim=34:38,asetpts=PTS-STARTPTS[a2]; \
    
    [2:v]trim=132:136,setpts=PTS-STARTPTS[v3]; \
    
    [0:a]atrim=38:42,asetpts=PTS-STARTPTS[a3]; \
    
    [3:v]trim=0:20,setpts=PTS-STARTPTS[v4]; \
    
    [0:a]atrim=42:62,asetpts=PTS-STARTPTS[a4]; \
    
    [v0][a0][v1][a1][v2][a2][v3][a3][v4][a4]concat=n=5:v=1:a=1[out]" \
    
  -map "[out]" generated.mkv

    However, there are 2 problems with this solution : -

    1. I’ve had to define the length of the intro video ([1:v]trim=0:30 ...) and the outro video ([3:v]trim=0:20 ...) - these are variable and would be preferable if I could simply concat the entire video.
    2. Each of the audio tracks are trimmed (using a running total from each video length) e.g. [0:a]atrim=0:30 ... => [0:a]atrim=30:34 ... => [0:a]atrim=34:38 ... => etc. Would be much easier if it simply said - "here’s the audio track - trim depending on the length of the generated video"

    Any advice is much appreciated !

  • FFmpeg flac to wav 24h audio track concatenation resulting in wrong duration

    2 novembre 2017, par MadManMoon

    I’m using this cmd line to create a 24h audio track from 24 separate chunks :

    ffmpeg -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/359.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/369.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/371.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/381.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/383.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/393.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/395.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/405.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/407.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/417.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/419.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/429.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/431.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/441.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/443.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/783.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/453.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/475.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/591.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/485.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/491.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/497.flac -ss 0.0 -t 3600.0 -i /data/work/cutjoin/1125/503.flac -ss 0.0 -t 3599.0 -i /data/work/cutjoin/1125/505.flac -filter_complex [0][1][2][3][4][5][6][7][8][9][10][11][12][13][14][15][16][17][18][19][20][21][22][23]concat=n=24:v=0:a=1 -f wav -rf64 always /data/work/cutjoin/1125/cutjoin_test.wav

    But the duration of the output is time=23:34:45.05 when is supposed to be 23:59:59.0

    I systematically get the following error during the decode process

    [flac @ 0x2ecc4e0] sample/frame number mismatch in adjacent frames
    
[flac @ 0x2e734c0] CRC error at PTS 100130816.0kbits/s
    
[flac @ 0x2e734c0] overread: 1
    
[flac @ 0x2f55ae0] sample/frame number mismatch in adjacent frames
    
[flac @ 0x2f55ae0] sample/frame number mismatch in adjacent frames
    
[flac @ 0x2f5b000] sample/frame number mismatch in adjacent frames
    
[flac @ 0x2f94460] sample/frame number mismatch in adjacent frames

    is there a parameter which I have to add in order to accomplish such duration ?

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