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Somos millones 1
21 juillet 2014, par
Mis à jour : Juin 2015
Langue : français
Type : Video
Autres articles (102)
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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 (...) -
MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)
Sur d’autres sites (9698)
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Change the default audio and video codec loaded by avformat_alloc_output_context2
8 juin 2021, par mythili3093I'm using ffmpeg library for live streaming via RTMP. I want to know how to give my choice of audio and video codec for the particular format in avformat_alloc_output_context2.


In Detail :


The following command works perfectly for me.


ffmpeg -re -stream_loop -1 -i ~/Downloads/Microsoft_Surface.mp4 -vcodec copy -c:a aac -b:a 160k -ar 44100 -strict -2 -f flv -flvflags no_duration_filesize rtmp://192.168.1.7/live/surface



In the output, I have set my audio codec to be aac and copied the video codec from input, which is H264.
I want to emulate this in the library, but don't know how to.


avformat_alloc_output_context2(&_ctx, NULL, "flv", NULL);



Above code sets oformat audio codec to ADPCM_SWF and video codec to FLV1. How to change that to AAC and H264 ?


So far, used av_guess_format to construct AVOutputFormat. It accepts only format as input. And I don't know where to mention audio and video codec.


AVOutputFormat* output_format = av_guess_format("flv", NULL, NULL);



Also tried giving filename to avformat_alloc_output_context2 with the rest of the parameters NULL.


AVOutputFormat* output_format = av_guess_format(NULL, "flv_acc_sample.flv", NULL);



This file has AAC audio and H264 video. But still ffmpeg loads oformat with ADPCM_SWF audio and FLV1 video codecs.


Searched stackoverflow for similar questions, but could not find the solution I was looking for.
Any hint/guidance is hugely appreciated. Thank you.


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How to incorporate screencasting into a rails project
13 mars 2015, par ruby_newbieI am working on a personal project and I want to incorporate screencasting with it. I want the screencast to be able to record the entire screen not just the page in a web browser so I think JS is out as a solution. I have searched and found a few clues but nothing concrete as to how i can make it happen short of learning Java. Here is what I have tried :
Rmov :https://rubygems.org/gems/rmov I cant get bundler to run bundle with this gem but this has the exact functionality I want. I think it has to do with quicktime being 32 bit but I am not sure. I opened an issue on Github in hopes to resolve it.
I saw this post : http://devblog.avdi.org/2013/01/21/my-screencasting-process/ which links to this script : https://github.com/avdi/Xcast/blob/master/Xcast
but I am failing on the require ’highline/system_extensions’ line (require cannot load such file).I found this : http://rvideo.rubyforge.org/ but I am still confused as to how to implement it. FFMPEG seems promising though and like it may be the tool I am looking for.
Does anyone know of a tool that would allow me to do this or a good tutorial that explains this process in detail ? Any help is appreciated and let me know if you need any more info that I can provide.
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Getting error "WebAssembly.Memory() : could not allocate memory" when running ffmpeg.wasm on android chrome browser
27 juin 2022, par Ravi KunduError in detail :


WebAssembly.Memory(): could not allocate memory
at https://*******/ffmpeg-core.js:22:82
at https://*******/ffmpeg.min.js:1:6506
at f (https://*******/ffmpeg.min.js:1:11322)
at Generator._invoke (https://*******/ffmpeg.min.js:1:11110)
at Generator.next (https://*******/ffmpeg.min.js:1:11747)
at i (https://*******/ffmpeg.min.js:1:4295)
at c (https://*******/ffmpeg.min.js:1:4498)



Code for ffmpeg :


const downloadWithFFMPEG = async () =>{
 const sourceBuffer = await fetch(recordingURL).then(r => r.arrayBuffer());
 await ffmpeg.load();
 await ffmpeg.FS(
 "writeFile",
 "input.webm",
 new Uint8Array(sourceBuffer, 0, sourceBuffer.byteLength)
 );
 await ffmpeg.run("-i", "input.webm", "-c", "copy", "output.mp4")
 const output = ffmpeg.FS("readFile", "output.mp4");
 var link = document.createElement('a')
 link.href = URL.createObjectURL(new Blob([output.buffer], { type: 'video/mp4;codecs=H264' }));
 link.download = this.data;
 link.click();
 recording = false;
}



Brief about problem :
The error only comes for android chrome browser. The same code works fine on pc/laptop chrome.
Have also enabled Webassembly-thread on chrome ://flags for android browser as someone suggested me to do it but still same error. Can someone help me ?