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Autres articles (47)
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(Dés)Activation de fonctionnalités (plugins)
18 février 2011, parPour gérer l’ajout et la suppression de fonctionnalités supplémentaires (ou plugins), MediaSPIP utilise à partir de la version 0.2 SVP.
SVP permet l’activation facile de plugins depuis l’espace de configuration de MediaSPIP.
Pour y accéder, il suffit de se rendre dans l’espace de configuration puis de se rendre sur la page "Gestion des plugins".
MediaSPIP est fourni par défaut avec l’ensemble des plugins dits "compatibles", ils ont été testés et intégrés afin de fonctionner parfaitement avec chaque (...) -
Le profil des utilisateurs
12 avril 2011, parChaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...) -
Automated installation script of MediaSPIP
25 avril 2011, parTo overcome the difficulties mainly due to the installation of server side software dependencies, an "all-in-one" installation script written in bash was created to facilitate this step on a server with a compatible Linux distribution.
You must have access to your server via SSH and a root account to use it, which will install the dependencies. Contact your provider if you do not have that.
The documentation of the use of this installation script is available here.
The code of this (...)
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ffmpeg command line to manipulate audio track with multi language m2ts file
8 octobre 2016, par user1320370I like to manipulate an audio track inside an m2ts file with some ffmpeg command line.
In short I need to manipulate the audio track with ID 2 of an m2ts file that is AC3 format (dolby digital 5.1 or 2.0).
The modifications are :
- add equalizer filter : 60 Hz + 3dB, 80Hz +2 dB, 100 Hz +1 dB
- increase the volume : +2 dB
The m2ts file is multilingual for this reason I need to change a specific ID track.
The AC3 file is lossy, I hope that eventual audio modifications don’t add more compression like for example with Photoshop when re-saving a jpeg file.
On lossless file (wav or flac) I’m sure I can save and re-save without loss of quality but on AC3 I’m not sure.Someone can please post the command line to do it without loss of quality ?
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Slow audio-video sync drift when merging wav and mp4 with ffmpeg
27 juin 2018, par charlie80I have an
mp4
file with only a single video stream (no audio) and awav
audio file that I would like to add to the video usingffmpeg
. The audio and the video have been recorded simultaneously during a conference, the former from a mixer output on a PC and the latter from a digital videocamera.I am using this
ffmpeg
command :ffmpeg -i incontro3.mp4 -itsoffset 18.39 -i audio_mix.wav -c:v copy -c:a aac final-video.mp4
where I’m using the
-itsoffset 18.39
option since I know that 18.39s is the video-audio delay.The problem I’m experiencing is that in the output file, while the audio is perfectly in sync with the video at the beginning, it slowly drifts out of sync during the movie.
The output if
ffprobe
on the video file is :Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'incontro3.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.25.100
Duration: 00:47:22.56, start: 0.000000, bitrate: 888 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 886 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
Metadata:
handler_name : VideoHandlerand the
ffprobe
output for the audio file is :Input #0, wav, from 'audio_mix.wav':
Metadata:
track : 5
encoder : Lavf57.25.100
Duration: 00:46:32.20, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/sI’m using the latest
ffmpeg
Zeranoe windows build git-9591ca7 (2016-05-25).Thanks in anticipation for any help/ideas !
UPDATE 1 : It looks like the problem is upstream the video-audio merging, and could be in the concatenation and conversion of theMTS
files generated by the video camera into themp4
video. I will follow up as I make any progress in understanding...
UPDATE 2 : The problem is not in the initial merging of the
MTS
files generated by the camera. Or, at least, it occurs identically if I merge them withcat
or withffmpeg -f concat
UPDATE 3 : Following @Mulvya’s suggestion, I observed that the drift rate is constant (at least as far as I can tell judging by eye). I also tried to superimpose the A/V tracks with another software, and the drift is exactly the same, thereby ruling out
ffmpeg
as culprit. My (bad) feeling is that the issue could be related to the internal clocks of the digital video camera and the laptop used for audio recording running at slightly different rates (see here the report of an identical issue I just found). -
Slow audio-video sync drift when merging wav and mp4 with ffmpeg
30 mai 2016, par charlie80I have an
mp4
file with only a single video stream (no audio) and awav
audio file that I would like to add to the video usingffmpeg
. The audio and the video have been recorded simultaneously during a conference, the former from a mixer output on a PC and the latter from a digital videocamera.I am using this
ffmpeg
command :ffmpeg -i incontro3.mp4 -itsoffset 18.39 -i audio_mix.wav -c:v copy -c:a aac final-video.mp4
where I’m using the
-itsoffset 18.39
option since I know that 18.39s is the video-audio delay.The problem I’m experiencing is that in the output file, while the audio is perfectly in sync with the video at the beginning, it slowly drifts out of sync during the movie.
The output if
ffprobe
on the video file is :Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'incontro3.mp4':
Metadata:
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.25.100
Duration: 00:47:22.56, start: 0.000000, bitrate: 888 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 886 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
Metadata:
handler_name : VideoHandlerand the
ffprobe
output for the audio file is :Input #0, wav, from 'audio_mix.wav':
Metadata:
track : 5
encoder : Lavf57.25.100
Duration: 00:46:32.20, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/sI’m using the latest
ffmpeg
Zeranoe windows build git-9591ca7 (2016-05-25).Thanks in anticipation for any help/ideas !
UPDATE 1 : It looks like the problem is upstream the video-audio merging, and could be in the concatenation and conversion of theMTS
files generated by the video camera into themp4
video. I will follow up as I make any progress in understanding...
UPDATE 2 : The problem is not in the initial merging of the
MTS
files generated by the camera. Or, at least, it occurs identically if I merge them withcat
or withffmpeg -f concat
UPDATE 3 : Following @Mulvya’s suggestion, I observed that the drift rate is constant (at least as far as I can tell judging by eye). I also tried to superimpose the A/V tracks with another software, and the drift is exactly the same, thereby ruling out
ffmpeg
as culprit. My (bad) feeling is that the issue could be related to the internal clocks of the digital video camera and the laptop used for audio recording running at slightly different rates (see here the report of an identical issue I just found).