
Recherche avancée
Médias (91)
-
Spitfire Parade - Crisis
15 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
-
Wired NextMusic
14 mai 2011, par
Mis à jour : Février 2012
Langue : English
Type : Video
-
Video d’abeille en portrait
14 mai 2011, par
Mis à jour : Février 2012
Langue : français
Type : Video
-
Sintel MP4 Surround 5.1 Full
13 mai 2011, par
Mis à jour : Février 2012
Langue : English
Type : Video
-
Carte de Schillerkiez
13 mai 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Texte
-
Publier une image simplement
13 avril 2011, par ,
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (30)
-
Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...) -
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 (...) -
Les images
15 mai 2013
Sur d’autres sites (7388)
-
Merge commit 'e3e8eab359238486dc233f7aa89b7bb3cb19ec38'
11 novembre 2017, par James AlmerMerge commit 'e3e8eab359238486dc233f7aa89b7bb3cb19ec38'
* commit 'e3e8eab359238486dc233f7aa89b7bb3cb19ec38' :
vaapi_h265 : Add support for AUD NAL units
vaapi_h265 : Convert to use coded bitstream infrastructure
vaapi_h264 : Add support for SEI recovery points
vaapi_h264 : Add support for AUD NAL units
vaapi_h264 : Convert to use coded bitstream infrastructure
lavc : Add hevc_metadata bitstream filter
lavc : Add h264_redundant_pps bitstream filter
lavc : Add h264_metadata bitstream filter
lavc : Add trace_headers bitstream filter
lavc : Add coded bitstream read/write support for H.265
lavc : Add coded bitstream read/write support for H.264
lavc : Add coded bitstream read/write API
pixfmt : Support chroma-derived and ictcp color matrices
h264 : Add support for alternative transfer characterics SEI
vaapi_encode : Move quality option to common codeThis commit is a noop, see
9c878651dbc8c795894740af74670b591551f619
8c34a2024da77b50470e62789e4859b45959932e
f3571048669bf876681499f49e9df492f05f73c6
6734eef6b8b464139fdc140ec9bc9e8d74173869
b4c915f4b3e15c3e787e319b961e4389762f6309
9b0c7aa0e446eceec96ba8f4009e004fad29fba3
9c7d70b49b64aa5571772a7cdb9bc426174261e0
a308872b049e33f69f4b629a06f47e3681906b93
8b26306294ffe78cc73357e2ddd56dd463db50ab
03f982bbca4211108477e772db9a339517ecde37
2e29ca2a9f19ba9a5b189f322f38497d2e2e3db0
281b68b0265953ab2623a39484d927a0e921c405
0bc7575ced65bf4aa4678ac12d550aaf87890d0e
00179664bccd1dd6fa0d1c40db453528757bf6f7
038a51258c4c5d8b77f4f9efcce6f397e5755c24Merged-by : James Almer <jamrial@gmail.com>
-
Live audio using ffmpeg, javascript and nodejs
8 novembre 2017, par klausI am new to this thing. Please don’t hang me for the poor grammar. I am trying to create a proof of concept application which I will later extend. It does the following : We have a html page which asks for permission to use the microphone. We capture the microphone input and send it via websocket to a node js app.
JS (Client) :
var bufferSize = 4096;
var socket = new WebSocket(URL);
var myPCMProcessingNode = context.createScriptProcessor(bufferSize, 1, 1);
myPCMProcessingNode.onaudioprocess = function(e) {
var input = e.inputBuffer.getChannelData(0);
socket.send(convertFloat32ToInt16(input));
}
function convertFloat32ToInt16(buffer) {
l = buffer.length;
buf = new Int16Array(l);
while (l--) {
buf[l] = Math.min(1, buffer[l])*0x7FFF;
}
return buf.buffer;
}
navigator.mediaDevices.getUserMedia({audio:true, video:false})
.then(function(stream){
var microphone = context.createMediaStreamSource(stream);
microphone.connect(myPCMProcessingNode);
myPCMProcessingNode.connect(context.destination);
})
.catch(function(e){});In the server we take each incoming buffer, run it through ffmpeg, and send what comes out of the std out to another device using the node js ’http’ POST. The device has a speaker. We are basically trying to create a 1 way audio link from the browser to the device.
Node JS (Server) :
var WebSocketServer = require('websocket').server;
var http = require('http');
var children = require('child_process');
wsServer.on('request', function(request) {
var connection = request.accept(null, request.origin);
connection.on('message', function(message) {
if (message.type === 'utf8') { /*NOP*/ }
else if (message.type === 'binary') {
ffm.stdin.write(message.binaryData);
}
});
connection.on('close', function(reasonCode, description) {});
connection.on('error', function(error) {});
});
var ffm = children.spawn(
'./ffmpeg.exe'
,'-stdin -f s16le -ar 48k -ac 2 -i pipe:0 -acodec pcm_u8 -ar 48000 -f aiff pipe:1'.split(' ')
);
ffm.on('exit',function(code,signal){});
ffm.stdout.on('data', (data) => {
req.write(data);
});
var options = {
host: 'xxx.xxx.xxx.xxx',
port: xxxx,
path: '/path/to/service/on/device',
method: 'POST',
headers: {
'Content-Type': 'application/octet-stream',
'Content-Length': 0,
'Authorization' : 'xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx',
'Transfer-Encoding' : 'chunked',
'Connection': 'keep-alive'
}
};
var req = http.request(options, function(res) {});The device supports only continuous POST and only a couple of formats (ulaw, aiff, wav)
This solution doesn’t seem to work. In the device speaker we only hear something like white noise.
Also, I think I may have a problem with the buffer I am sending to the ffmpeg std in -> Tried to dump whatever comes out of the websocket to a .wav file then play it with VLC -> it plays everything in the record very fast -> 10 seconds of recording played in about 1 second.
I am new to audio processing and have searched for about 3 days now for solutions on how to improve this and found nothing.
I would ask from the community for 2 things :
-
Is something wrong with my approach ? What more can I do to make this work ? I will post more details if required.
-
If what I am doing is reinventing the wheel then I would like to know what other software / 3rd party service (like amazon or whatever) can accomplish the same thing.
Thank you.
-
-
stream the screen from one linux computer to another remote linux computer using ffpmeg
10 novembre 2017, par Revathi MI have tried to stream the screen using ffmpeg as follows :
ffmpeg -f x11grab -s 1600x900 -r 30 -i :0.0 -qscale 0.1 -vcodec huffyuv -f segment -segment_time 5 -reset_timestamps 1 "scrCap%03d.avi"
Here i need to transfer the video to another host using udp. So I have tried the following command,
ffmpeg -f x11grab -s 1600x900 -r 30 -i :0.0 -qscale 0.1 -vcodec huffyuv -f segment -segment_time 5 -reset_timestamps 1 "scrCap%03d.avi" -f mpegts udp://192.168.0.54:22
but it is not working properly.Also i don’t know how to identify udp port number in my linux system. Can you please help me to transfer the video to another host using udp protocol.