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Autres articles (95)
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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...)
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Is there an efficient way to use ffmpeg to perform a large quantity of cuts from a single file ?
16 mars 2024, par Giuliano OliveriI'm trying to cut video files into smaller chunks. (each one being one word said in the video, so they're not all of equal size)


I've tried a lot of different approaches to try to be as efficient as possible, but I can't get the runtime to be under 2/3rd of the original video length. That's an issue because I'm trying to process 400+ hours of video.


Is there a more efficient way to do this ? Or am I doomed to run this for weeks ?


Here is the command for my best attempt so far


ffmpeg -hwaccel cuda -hwaccel_output_format cuda -ss start_timestamp -t to_timestamp -i file_name -vf "fps=30,scale_cuda=1280:720" -c:v h264_nvenc -y output_file



Note that the machine running the code has a 4090
This command is then executed via python, which gives it the right timestamps and file paths for each smaller clip in a for loop


I think it's wasting a lot of time calling a new process each time, however I haven't been able to get better results with a split filter ; but here's the ffmpeg-python code for that attempt :


Creation of the stream :


inp = (
 ffmpeg
 .input(file_name, hwaccel="cuda", hwaccel_output_format="cuda")
 .filter("fps",fps=30)
 .filter('scale_cuda', '1280','720')
 .filter_multi_output('split')
)



Which then gets called in a for loop


(
 ffmpeg
 .filter(inp, 'trim', start=row[1]['start'], end=row[1]['end'])
 .filter('setpts', 'PTS-STARTPTS')
 .output(output_file,vcodec='h264_nvenc')
 .run()
)



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Can ffmpeg write metadata encoder when transcoding alac/flac to aac audio file ?
11 juin 2022, par David II have a collection of alac and flac files from Bandcamp and an ffmpeg instance compiled with libfdk_aac https://trac.ffmpeg.org/wiki/CompilationGuide/Centos#libfdk_aac and am trying to convert these to lossy audio aac files for non-critical listening.


With
ffmpeg -i Liholesie\ -\ Shamanic\ Twilight\ -\ 09\ Gray\ Wings.m4a -c:a libfdk_aac -vbr 4 -c:v copy 09_Gray_wings_vbr4.m4a
an expected aac .m4a audio file is produced, album art included, works well. There's one slight detail missing :

During the ffmpeg conversion process ffmpeg says :


Output #0, ipod, to '09_Gray_wings_vbr4.m4a':
 Metadata:
 major_brand : M4A 
 minor_version : 512
 compatible_brands: M4A isomiso2
 title : Gray Wings
 artist : Liholesie
 album_artist : Liholesie
 album : Shamanic Twilight
 comment : Visit https://liholesie.bandcamp.com
 date : 2021
 track : 9
 encoder : Lavf59.24.100
 Stream #0:0: Video: mjpeg (Baseline), yuvj444p(pc, bt470bg/unknown/unknown), 700x700 [SAR 72:72 DAR 1:1], q=2-31, 90k tbr, 90k tbn (attached pic)
 Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16 (default)
 Metadata:
 handler_name : SoundHandler
 vendor_id : [0][0][0][0]
 encoder : Lavc59.33.100 libfdk_aac 
..





and the file produced looks like that when ffprobed except that the Metadata : encoder field is missing :


Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '09_Gray_wings_vbr4.m4a':
 Metadata:
 major_brand : M4A 
 minor_version : 512
 compatible_brands: M4A isomiso2
 title : Gray Wings
 artist : Liholesie
 album_artist : Liholesie
 album : Shamanic Twilight
 date : 2021
 encoder : Lavf59.24.100
 comment : Visit https://liholesie.bandcamp.com
 track : 9
 Duration: 00:06:57.78, start: 0.000000, bitrate: 155 kb/s
 Stream #0:0[0x2](und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 152 kb/s (default)
 Metadata:
 handler_name : SoundHandler
 vendor_id : [0][0][0][0]
 Stream #0:1[0x0]: Video: mjpeg (Baseline), yuvj444p(pc, bt470bg/unknown/unknown), 700x700 [SAR 72:72 DAR 1:1], 90k tbr, 90k tbn (attached pic)



Is there a way to write the encoder field in the Metadata section when transcoding (or is "encoder" not supported for aac m4a ? That would be weird since ffmpeg says what it says when specifying output during transcoding) .


Any hints on how to write a self-defined text to said tag during transcoding are also welcome.


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Using linux programs within app
15 octobre 2014, par user1503606Just going to ask this question feel free to say its a stupid question.
Is it possible to use a program like ffmpeg within a mobile app i currently use Titanium for app development so it is all written in javascript.
If i was going to go native and learn C objective C would this allow me to execute a program like ffmpeg or compile it into a app to run on the device ?
With a lot of my current apps i normally send/upload a file to my server then run all the file manipulation then download the newly encoded file.
It would be great if you could do this directly on the app in C.
How do people do this.