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  • Participer à sa traduction

    10 avril 2011

    Vous 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 (...)

  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • Ajouter notes et légendes aux images

    7 février 2011, par

    Pour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
    Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
    Modification lors de l’ajout d’un média
    Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)

Sur d’autres sites (10844)

  • Is there any way to edit just the metadata of an AVI file, like mkvpropedit does with MKV files ?

    20 juillet 2016, par hmj6jmh

    I am trying to swap the order of 2 audio tracks of an AVI file using ffmpeg.

    ffmpeg -i "infile.avi" -map 0:0 -map 0:2 -map 0:1 -c copy "outfile.avi"

    This works but the file is being re-encoded, not copied. It takes much longer than a straight copy to finish and the resulting file is larger than the original.

    Is there any way to edit just the metadata of an AVI file, like mkvpropedit does with MKV files ? Command line tool preferred.

    General
    
Complete name                            : infile.avi
    
Format                                   : AVI
    
Format/Info                              : Audio Video Interleave
    
File size                                : 451 MiB
    
Duration                                 : 43mn 11s
    
Overall bit rate                         : 1 460 Kbps
    
Writing application                      : VirtualDubMod 1.5.10.2 (build 2540/release)
    
Writing library                          : VirtualDubMod build 2540/release
    

    
Video
    
ID                                       : 0
    
Format                                   : MPEG-4 Visual
    
Format profile                           : Advanced Simple@L5
    
Format settings, BVOP                    : 3
    
Format settings, QPel                    : No
    
Format settings, GMC                     : No warppoints
    
Format settings, Matrix                  : Default (MPEG)
    
Muxing mode                              : Packed bitstream
    
Codec ID                                 : XVID
    
Codec ID/Hint                            : XviD
    
Duration                                 : 43mn 11s
    
Bit rate                                 : 1 190 Kbps
    
Width                                    : 656 pixels
    
Height                                   : 368 pixels
    
Display aspect ratio                     : 16:9
    
Frame rate                               : 23.976 fps
    
Color space                              : YUV
    
Chroma subsampling                       : 4:2:0
    
Bit depth                                : 8 bits
    
Scan type                                : Progressive
    
Compression mode                         : Lossy
    
Bits/(Pixel*Frame)                       : 0.206
    
Stream size                              : 367 MiB (82%)
    
Writing library                          : XviD 1.2.1 (UTC 2008-12-04)
    

    
Audio #1
    
ID                                       : 1
    
Format                                   : MPEG Audio
    
Format version                           : Version 1
    
Format profile                           : Layer 3
    
Codec ID                                 : 55
    
Codec ID/Hint                            : MP3
    
Duration                                 : 43mn 11s
    
Bit rate mode                            : Constant
    
Bit rate                                 : 128 Kbps
    
Channel count                            : 2 channels
    
Sampling rate                            : 48.0 KHz
    
Compression mode                         : Lossy
    
Stream size                              : 39.5 MiB (9%)
    
Alignment                                : Aligned on interleaves
    
Interleave, duration                     : 42 ms (1.00 video frame)
    
Interleave, preload duration             : 504 ms
    
Writing library                          : LAME3.98r
    
Encoding settings                        : -m s -V 4 -q 2 -lowpass 17 -b 128
    

    
Audio #2
    
ID                                       : 2
    
Format                                   : MPEG Audio
    
Format version                           : Version 1
    
Format profile                           : Layer 3
    
Codec ID                                 : 55
    
Codec ID/Hint                            : MP3
    
Duration                                 : 43mn 11s
    
Bit rate mode                            : Constant
    
Bit rate                                 : 128 Kbps
    
Channel count                            : 2 channels
    
Sampling rate                            : 48.0 KHz
    
Compression mode                         : Lossy
    
Stream size                              : 39.5 MiB (9%)
    
Alignment                                : Aligned on interleaves
    
Interleave, duration                     : 42 ms (1.00 video frame)
    
Interleave, preload duration             : 504 ms
    
Writing library                          : LAME3.98r
    
Encoding settings                        : -m s -V 4 -q 2 -lowpass 17 -b 128

  • avcodec : Remove libaacplus

    24 janvier 2016, par Timothy Gu
    avcodec : Remove libaacplus

    TODO : bump minor
    

    It’s inferior in quality to fdk-aac and has an arguably more problematic
    
license.
    

    As early as 2012, a HydrogenAudio user reported :
    

    > It has however one huge advantage : much better quality at low bitrates than
    
> faac and libaacplus.
    

    (https://hydrogenaud.io/index.php?PHPSESSID=ckiq394pdglka0kj2fin6ij8t7&topic=95989.msg804633#msg804633)
    

    I myself have made a few spectrograms for a comparison of the two
    
encoders as well. The FDK output is consistently better than the
    
libaacplus one, in all bitrates I tested.
    

    libaacplus license is 3GPP + LGPLv2. 3GPP copyright notice is completely
    
proprietory, as follows :
    

    > No part may be reproduced except as authorized by written permission.
    
>
    
> The copyright and the foregoing restriction extend to reproduction in
    
> all media.
    
>
    
> © 2008, 3GPP Organizational Partners (ARIB, ATIS, CCSA, ETSI, TTA, TTC).
    
>
    
> All rights reserved.
    

    (The latest 26410-d00 zip from 3GPP has the same notice, but the copyright
    
year is changed to 2015)
    

    The copyright part of the FDK AAC license (section 2) is a copyleft
    
license that permits redistribution under certain conditions (and
    
therefore the LGPL + libfdk-aac combination is not prohibited by
    
configure) :
    

    > Redistribution and use in source and binary forms, with or without
    
> modification, are permitted without payment of copyright license fees
    
> provided that you satisfy the following conditions :
    
>
    
> You must retain the complete text of this software license in
    
> redistributions of the FDK AAC Codec or your modifications thereto in
    
> source code form.
    
>
    
> You must retain the complete text of this software license in the
    
> documentation and/or other materials provided with redistributions of
    
> the FDK AAC Codec or your modifications thereto in binary form.
    
>
    
> You must make available free of charge copies of the complete source
    
> code of the FDK AAC Codec and your modifications thereto to recipients
    
> of copies in binary form.
    
>
    
> The name of Fraunhofer may not be used to endorse or promote products
    
> derived from this library without prior written permission.
    
>
    
> You may not charge copyright license fees for anyone to use, copy or
    
> distribute the FDK AAC Codec software or your modifications thereto.
    
>
    
> Your modified versions of the FDK AAC Codec must carry prominent
    
> notices stating that you changed the software and the date of any
    
> change. For modified versions of the FDK AAC Codec, the term
    
> "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the
    
> term "Third-Party Modified Version of the Fraunhofer FDK AAC Codec
    
> Library for Android."
    • [DH] Changelog
    • [DH] LICENSE.md
    • [DH] configure
    • [DH] doc/general.texi
    • [DH] doc/platform.texi
    • [DH] libavcodec/Makefile
    • [DH] libavcodec/allcodecs.c
    • [DH] libavcodec/libaacplus.c

  • ffmpeg and Red5 Issue : Increase in number of ffmpeg simultaneous streams to Red5 resulting in packet loss

    30 octobre 2014, par kajarigd

    I have a screen sharing app written in flex, using which one person can share his screen with another person via Red5 server (Version : 1.0.3). Platform is Windows Server 2008. Now, I want to load test this Red 5 server to find out maximum how many simultaneous screen sharing session it can allow, without any quality compromise. By quality I mean, speed of transmission and no data loss during transmission. I simulated the load using ffmpeg command.

    For this, instead of transmitting a live captured screen, I am transmitting (uploading) a FLV file stored in my local to the Red5 server using ffmpeg command. In the receiving client side, I am starting to download (transmitting) this same FLV file after 5 secs since the upload has started. This is working fine when I am running this test for less than 10 pairs of upstreaming-downstreaming sessions. But, when the number is increasing beyond 10, I am observing significant packet loss in transmission.

    Here are the commands I am running in a loop. The loop count is the number of streaming pairs.

    1. upstreaming : ffmpeg -re -i  -f flv -ar 22050 "rtmp://" -report
    2. downstreaming : ffmpeg -re -i "rtmp:// live=1"  -report

    The and are set in such a way, that in the downstream I will download the same uploaded file. "rtmp ://" are the same in both the cases. I am not doing the upstream in record mode, hence, no physical file is getting saved in the server side. When I am analyzing the file I received in the receiving client side, it is a poor quality video due to frame loss. Uploading and downloading machines are two different machines. I ran the test for many hours, repeating the same 10 simultaneous streaming sets. Each set is consistently giving the same results.

    What is puzzling me is, this is working fine without any packet loss for less that 10 simultaneous streaming. I searched about it in various forums, but none of the answers were applicable for this scenario. For a while I was thinking that Red5 has limited capacity, but I found many posts saying Red5 can easily scale up to take very big load. Does that mean, the problem is in my configuration ? I am not sure which are to focus on.

    An example log snippet :

    Lots of missing data at downstream side. For e.g. between frames 101 and 102 there is a difference of 25 sec. On replaying the video there is a stoppage for this much time.In this time gap all the frames are lost.

    frame=  101 fps=1.0 q=14.5 size=    2650kB time=00:01:41.00 bitrate= 214.9kbits/s
    
frame=  102 fps=1.0 q=13.2 size=    2763kB time=00:02:06.00 bitrate= 179.6kbits/s

    Any help is appreciated !

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