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Médias (29)
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#7 Ambience
16 octobre 2011, par
Mis à jour : Juin 2015
Langue : English
Type : Audio
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#6 Teaser Music
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#5 End Title
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#3 The Safest Place
16 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#4 Emo Creates
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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#2 Typewriter Dance
15 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Audio
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Creating farms of unique websites
13 avril 2011, parMediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...) -
Des sites réalisés avec MediaSPIP
2 mai 2011, parCette 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 (7190)
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FFMPEG Slow Video and Audio by Half
12 septembre 2021, par nomaamI am using the latest version of FFMPEG and am trying to slow video and audio by half


This is the best I could come up with


./ffmpeg.exe -i "C:/ffmpeg/input.mp4" -filter_complex "[0:v]setpts=1.5*PTS[v];[0:a]atempo=0.5[a]" -map "[v]" -map "[a]" "C://ffmpeg/output.mp4"



But the audio and video are not being slowed down at the same rate.


Any suggestions ?


Thanks.


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Cut every .mp4 in folder by half with ffmpeg
14 novembre 2022, par c00kieRaptorI have a folder with 200 mp4-files and I need to cut the last half of every one of them. So if a video is 140 seconds I need to cut out the last 70 seconds so that the resulting output is the preceding 70 seconds.


I would assume the loop would look a bit like this :


for i in *.mp4; do
 let A=$(ffprobe -v error -show_entries format=duration -of default=noprint_wrappers=1:nokey=1 $i)
 let A=${A%.*}
 let A=$A/2
 ffmpeg -sseof $A -i $i ${i%.mp4}
done



But I get "syntax error : invalid arithmetic operator (error token is ".246848")" from the first 'let'. This corresponds to the float of the video duration.
For all I know there could be many other mistakes with my loop


Also some files have spaces and special characters if that matters.


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ffmpef cannot open a simple microsoft wav file exported with Audacity
23 juillet 2013, par sebpiqI have exported a sound file to microsoft wav using Audacity.
I am trying to open this file with ffmpeg :ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg
and here's the ouput I get :
fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
libavutil 52. 18.100 / 52. 18.100
libavcodec 54. 92.100 / 54. 92.100
libavformat 54. 63.104 / 54. 63.104
libavdevice 54. 3.103 / 54. 3.103
libavfilter 3. 42.103 / 3. 42.103
libswscale 2. 2.100 / 2. 2.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 2.100 / 52. 2.100
[dca @ 0x7fd30c013600] Not a valid DCA frame
... SNIP ...
[dca @ 0x7fd5bc013600] Invalid bit allocation index
[dca @ 0x7fd5bc013600] error decoding block
Last message repeated 3 times
[dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
[dca @ 0x7fd5bc013600] error decoding block
[wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
[wav @ 0x7fd5bc013000] decoding for stream 0 failed
[wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
Consider increasing the value for the 'analyzeduration' and 'probesize' options
steps-stereo-16b-44khz.wav: could not find codec parametersIf I export the same file to .ogg or .aiff, no problem, the following works fine :
ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg
Any idea what could be wrong ?
A link to my wav file so you can try to reproduce.
NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.