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Autres articles (93)
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Utilisation et configuration du script
19 janvier 2011, parInformations spécifiques à la distribution Debian
Si vous utilisez cette distribution, vous devrez activer les dépôts "debian-multimedia" comme expliqué ici :
Depuis la version 0.3.1 du script, le dépôt peut être automatiquement activé à la suite d’une question.
Récupération du script
Le script d’installation peut être récupéré de deux manières différentes.
Via svn en utilisant la commande pour récupérer le code source à jour :
svn co (...) -
Configuration spécifique d’Apache
4 février 2011, parModules spécifiques
Pour la configuration d’Apache, il est conseillé d’activer certains modules non spécifiques à MediaSPIP, mais permettant d’améliorer les performances : mod_deflate et mod_headers pour compresser automatiquement via Apache les pages. Cf ce tutoriel ; mode_expires pour gérer correctement l’expiration des hits. Cf ce tutoriel ;
Il est également conseillé d’ajouter la prise en charge par apache du mime-type pour les fichiers WebM comme indiqué dans ce tutoriel.
Création d’un (...) -
MediaSPIP Player : problèmes potentiels
22 février 2011, parLe lecteur ne fonctionne pas sur Internet Explorer
Sur Internet Explorer (8 et 7 au moins), le plugin utilise le lecteur Flash flowplayer pour lire vidéos et son. Si le lecteur ne semble pas fonctionner, cela peut venir de la configuration du mod_deflate d’Apache.
Si dans la configuration de ce module Apache vous avez une ligne qui ressemble à la suivante, essayez de la supprimer ou de la commenter pour voir si le lecteur fonctionne correctement : /** * GeSHi (C) 2004 - 2007 Nigel McNie, (...)
Sur d’autres sites (3361)
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FFMPEG video encoding time increases over time
8 août 2018, par NStofI am creating a fairly basic DVR in C# using the FFMPEG wrapper in Accord.net. Essentially it works by getting the raw frames from the on-board frame grabber, displays the frame on screen and places it in a buffer. In a Separate thread in 20 second intervals, the frames are taken from the buffer and using Accord.Video.FFMPEG.VideoFileWriter.WriteVideoFrame(Bitmap bmp) the frame is encoded and saved to disk. Video is saved in two minute file chunks.
This works happily for about 24-30 hours, however after that it looks like the average time it takes to encode/save each frame increases to the point where it gets new frames faster than they can be saved. This causes buffer to grow and ends in tears.What I would like to know is why does the time it takes to complete the WriteVideoFrame(Bitmap bmp) function increase over time.
What I think I know so far :
I do not know if this problem is caused by something in FFMPEG or in the Accord.net wrapper.
I am reasonably sure it is not caused by hardware. Neither CPU usage nor HDD are working particularly hard. In fact, when I monitor the CPU usage, it does not work any harder or less hard when the encoding time increases.
When I stop recording (close and dispose all Accord.net objects) and start again, it does not reset the encoding speed. Only when I close the software and start it again does it fix the problem.
Any thoughts and help with this will be greatly appreciated. If more information is required, please let me know. -
How to achieve real time video editing on Android ?
3 octobre 2024, par ItamarI've been working recently on a video-editing-related project on Android, and am desperately looking for resources related to video editing on the platform.


The only video-editing related "method" or information I could find was using the FFmpeg library, which is pretty rich in features and capabilities but works really slow, operations such as reversing a 10-second video can take as long as 30 seconds, which delivers a really poor user experience.


That being said, it seems that there are tons of Android video-editing apps that are capable of doing everything FFmpeg can, only in almost immediate periods of time (apps such as InShot, PocketVideo or even the previous musical.ly TikTok).


I've tried researching and searching for information on the topic in almost any reasonable place (Google, GitHub, YouTube, the Android developer center, and even on "support" pages of the above-mentioned apps) to no avail (no explanatory documents, no open source libraries, not even demo apps), if anyone could shed some light on the subject that would be much, much appreciated !


Thanks.


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generating silence audio using ffmpeg with specific bitrate and duration time less than one second cannot get the audio with the desired bitrate
28 août 2018, par bambooomI’d like to generate a silence audio file using ffmpeg with specific bitrate and the duration is less than one second. I found that the generated audio file does not have the desired bitrate. But if I expand the duration to 2 seconds, the bitrate seems to be correct.
For example :
ffmpeg -f lavfi -i anullsrc=r=22050:cl=mono -t 0.3 -acodec mp3 -ab 48k -y silence.mp3
The generated silence.mp3’s bitrate seems not 48kb/s :
$ ffprobe silence.mp3
ffprobe version 3.4.2 Copyright (c) 2007-2018 the FFmpeg developers
built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --disable-jack --enable-gpl --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, mp3, from 'silence.mp3':
Metadata:
encoder : Lavf57.83.100
Duration: 00:00:00.37, start: 0.050113, bitrate: 52 kb/s
Stream #0:0: Audio: mp3, 22050 Hz, mono, s16p, 48 kb/sIf the duration is 2 seconds, bitrate seems OK :
$ ffmpeg -f lavfi -i anullsrc=r=22050:cl=mono -t 2 -acodec mp3 -ab 48k -y silence.mp3
$ ffprobe silence.mp3
ffprobe version 3.4.2 Copyright (c) 2007-2018 the FFmpeg developers
built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --disable-jack --enable-gpl --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, mp3, from 'silence.mp3':
Metadata:
encoder : Lavf57.83.100
Duration: 00:00:02.06, start: 0.050113, bitrate: 48 kb/s
Stream #0:0: Audio: mp3, 22050 Hz, mono, s16p, 48 kb/sIs there any way to keep the bitrate as desired ?