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Autres articles (112)
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Support de tous types de médias
10 avril 2011Contrairement à beaucoup de logiciels et autres plate-formes modernes de partage de documents, MediaSPIP a l’ambition de gérer un maximum de formats de documents différents qu’ils soient de type : images (png, gif, jpg, bmp et autres...) ; audio (MP3, Ogg, Wav et autres...) ; vidéo (Avi, MP4, Ogv, mpg, mov, wmv et autres...) ; contenu textuel, code ou autres (open office, microsoft office (tableur, présentation), web (html, css), LaTeX, Google Earth) (...)
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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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avformat/dashdec : Check whitelist
15 janvier, par Michael Niedermayer -
WebRTC books – a brief review
1er janvier 2014, par silviaI just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.
Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.
Rob’s focus is very much on the features required in a typical Web application :
- video calls
- audio calls
- text chats
- file sharing
In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.
Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.
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Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.
Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.
Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.
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Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.
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rtmp audio out of sync, http works fine
21 janvier 2014, par marcaWe have encoded and distributed videos for some years now, using FFMPEG to produce h.264/mp4 files that have been working great for us. We have been using HTML mode and fall-backed to flash for browsers that does not support it natively using flowplayer.
We use cloudfront to serve our files from a s3 bucket and have been using http progressive streaming.
Recently we started distribute the files in flashmode over rtmp instead, using a cloudfront streaming distribution pointing to the same amazon s3 bucket.
All good for some weeks, until yesterday when we notice a couple of files with audio sync issues in rtmp mode.
The same file have no sync problems in flash with direct url to file.What can be the case ?
Not working when streamed via RTMP, but file work with http streaming/progressive.
You see the sync issue 15 sec's into the video.
rtmp ://s2xe2avk54qztf.cloudfront.net:1935/cfx/st/mp4:95fvOY255bdPspO3z6tEvGi3Em7/default.mp4
http://media.shootitlive.com/95fvOY255bdPspO3z6tEvGi3Em7/default.mp4Another file that have no sync issue at all.
rtmp ://s2xe2avk54qztf.cloudfront.net:1935/cfx/st/mp4:P4EuH2TZxfV6BvpupP6dxrrs7gD/default.mp4
http://media.shootitlive.com/P4EuH2TZxfV6BvpupP6dxrrs7gD/default.mp4Both files have the same format for video and audio and have been encoded the exact same way with ffmpeg. It's not player related as we see the audio sync issue on several players and when playing stream in VLC.