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Les Miserables
9 décembre 2019, par
Mis à jour : Décembre 2019
Langue : français
Type : Textuel
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VideoHandle
8 novembre 2019, par
Mis à jour : Novembre 2019
Langue : français
Type : Video
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Somos millones 1
21 juillet 2014, par
Mis à jour : Juin 2015
Langue : français
Type : Video
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Un test - mauritanie
3 avril 2014, par
Mis à jour : Avril 2014
Langue : français
Type : Textuel
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Pourquoi Obama lit il mes mails ?
4 février 2014, par
Mis à jour : Février 2014
Langue : français
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IMG 0222
6 octobre 2013, par
Mis à jour : Octobre 2013
Langue : français
Type : Image
Autres articles (43)
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Participer à sa traduction
10 avril 2011Vous 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, parPour 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, parWe 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 VaFancyAt 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=8/8The 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.
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cut fragment of mp3 from, to and apply fade in, fade out
23 juin 2021, par MeviaI am in need of a tool that will complete following tasks


- 

- provide
source
mp3 - provide
data
of fragment to cut (start
time,end
time) - start cutting from
source
wherestart
is - end cutting from
source
whereend
is - apply
X
secondsfade in
beginning ofstart
time - apply
Y
secondsfade out
beginningY
seconds subtracted fromend














By
fade
i meanincrease
/decrease
volume between0 - 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.


- provide
-
Merge commit ’6d93307f8df81808f0dcdbc064b848054a6e83b3’
26 janvier 2014, par Michael NiedermayerMerge commit ’6d93307f8df81808f0dcdbc064b848054a6e83b3’
* commit ’6d93307f8df81808f0dcdbc064b848054a6e83b3’ :
mpeg12 : check scantable indices in all decode_block functionsBenchmarks
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 skipsAll benchmarks are the best scores of several runs
Merged-by : Michael Niedermayer <michaelni@gmx.at>