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Autres articles (111)
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Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
Supporting all media types
13 avril 2011, parUnlike most software and media-sharing platforms, MediaSPIP aims to manage as many different media types as possible. The following are just a few examples from an ever-expanding list of supported formats : images : png, gif, jpg, bmp and more audio : MP3, Ogg, Wav and more video : AVI, MP4, OGV, mpg, mov, wmv and more text, code and other data : OpenOffice, Microsoft Office (Word, PowerPoint, Excel), web (html, CSS), LaTeX, Google Earth and (...)
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Script d’installation automatique de MediaSPIP
25 avril 2011, parAfin de palier aux difficultés d’installation dues principalement aux dépendances logicielles coté serveur, un script d’installation "tout en un" en bash a été créé afin de faciliter cette étape sur un serveur doté d’une distribution Linux compatible.
Vous devez bénéficier d’un accès SSH à votre serveur et d’un compte "root" afin de l’utiliser, ce qui permettra d’installer les dépendances. Contactez votre hébergeur si vous ne disposez pas de cela.
La documentation de l’utilisation du script d’installation (...)
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Can you think of a reason why windows might not enable audio if noone is logged in ?
3 juillet 2017, par Caius JardI’m having a bizarre problem with some virtual servers created to record podcasts. They run on amazon AWS as windows server 2012 instances and a small c# app tells FFMPEG to do the heavy lifting of capturing from the virtual screen and reading from the virtual sound card (Virtual Audio Cable : https://en.wikipedia.org/wiki/Virtual_Audio_Cable) via DirectShow filters
The problem I have is if I leave the machine to do its stuff unattended, the recordings are sometimes silent. If I log in via VNC and watch it doing its stuff the audio is recorded just fine. All other aspects of the test op are the same, and the virtual machine is shut down between successive recordings so each one should theoretically be a clean slate. The app runs under a logged in session (hence the use of VNC rather than RDP)
I’m now wondering if there is some optimisation of the windows sound engine whereby it doesn’t bother playing audio if it thinks noone is listening. The confusing thing to me is that not every virtual machine suffers these problems ; some of them record fine (and they’re all created from the same seed virtual hard disk image) in unattended mode
I’m asking this question with the aim of getting together a list of things I can check/look into/debug.. I don’t have much knowledge of how MME/DirectSound/WASAPI work internally...
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What is the best way to merge .mkv and .mka files using ffmpeg ?
28 juin 2017, par RobertI’m using ffmpeg to merge .mkv and .mka files into .mp4 files. My current command looks like this :
ffmpeg -i video.mkv -i audio.mka output_path.mp4
The audio and video files are pre-signed urls from Amazon S3. Even on a server with sufficient resources, this process is going very slowly. I’ve researched situations where you can tell ffmpeg to skip re-encoding each frame, but I think that in my situation it actually does need to re-encode each frame.
I’ve downloaded 2 sample files to my macbook pro and have installed ffmpeg locally via homebrew. When I run the command
ffmpeg -i video.mkv -i audio.mka -c copy output.mp4
I get the following output :
ffmpeg version 3.3.2 Copyright (c) 2000-2017 the FFmpeg developers
built with Apple LLVM version 8.1.0 (clang-802.0.42)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.3.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
libavutil 55. 58.100 / 55. 58.100
libavcodec 57. 89.100 / 57. 89.100
libavformat 57. 71.100 / 57. 71.100
libavdevice 57. 6.100 / 57. 6.100
libavfilter 6. 82.100 / 6. 82.100
libavresample 3. 5. 0 / 3. 5. 0
libswscale 4. 6.100 / 4. 6.100
libswresample 2. 7.100 / 2. 7.100
libpostproc 54. 5.100 / 54. 5.100
Input #0, matroska,webm, from '319_audio_1498590673766.mka':
Metadata:
encoder : GStreamer matroskamux version 1.8.1.1
creation_time : 2017-06-27T19:10:58.000000Z
Duration: 00:00:03.53, start: 2.831000, bitrate: 50 kb/s
Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
Metadata:
title : Audio
Input #1, matroska,webm, from '319_video_1498590673766.mkv':
Metadata:
encoder : GStreamer matroskamux version 1.8.1.1
creation_time : 2017-06-27T19:10:58.000000Z
Duration: 00:00:03.97, start: 2.851000, bitrate: 224 kb/s
Stream #1:0(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 4:3, 30 tbr, 1k tbn, 1k tbc (default)
Metadata:
title : Video
[mp4 @ 0x7fa4f0806800] Could not find tag for codec vp8 in stream #0, codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Stream mapping:
Stream #1:0 -> #0:0 (copy)
Stream #0:0 -> #0:1 (copy)
Last message repeated 1 timesSo it appears that the specific encodings I’m working with are vp8 videos and opus audio files, which I believe are incompatible with the .mp4 output container. I would appreciate answers that cover ways of optimally merging vp8 and opus into .mp4 output or answers that point me in the direction of output media formats that are both compatible with vp8 & opus and are playable on web and mobile devices so that I can bypass the re-encoding step altogether.
EDIT :
Just wanted to provide a benchmark after following LordNeckbeard’s advice :
4 min 41 second video transcoded locally on my mac
LordNeckbeard’s approach : 15 mins 55 seconds (955 seconds)
Current approach : 18 mins 49 seconds (1129 seconds)
18% speed increase -
How to set the destination folder of a Node.js fluent-ffmpeg screenshot to your AWS S3 bucket using getSignedUrl() ?
10 juillet 2017, par Madhavi MohoniI’m writing a program to generate .png thumbnails (with the same name, in the same folder) for a set of .mp4 videos in my Amazon S3 bucket. For this example, I’m going to create a /folder/file.png for a /folder/file.mp4 in the bucket. I’ve managed to set the source URL using the s3 object and getSignedUrl as follows :
var srcurl = s3.getSignedUrl('getObject', {
Bucket: 'bucket-name',
Key: '/folder/file.mp4'
});and
new ffmpeg({ source: srcurl })
.screenshots({
count: 1,
filename: '%f'.substr(0, '%f'.indexOf('.')) + '.png',
/* To shorten the long string that's returned */
folder: desturl,
size: MAX_WIDTH + 'x' + MAX_HEIGHT
});The destination URL has to be the same folder as the source. So I set it as follows :
var desturl = s3.getSignedUrl('putObject', {
Bucket: 'bucket-name',
Key: '/folder/file' + '.png'
});This combination doesn’t work - is there a way to do this correctly ?