<div dir="ltr">Hi Robert,<div><br></div><div>FreeSWITCH can be extended by writing custom modules. Modules can register a callback to get every audio frame passed by the core FS to them.</div><div>You can find a lot of examples in src/mod (best look into src/mod/applications, e.g. mod_avmd works with audio frames).</div><div><br></div><div>kind regards,</div><div><br clear="all"><div><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div><font style="background-color:rgb(255,255,255)" size="4" face="arial, helvetica, sans-serif"><img src="https://drive.google.com/a/dataandsignal.com/uc?id=1lUdzdLjEycxXcGhDz-bKqDX8-AIlZuCq&export=download" width="96" height="49"><br></font></div><div><font style="background-color:rgb(255,255,255)" size="4" face="arial, helvetica, sans-serif"><br></font></div><div><font style="background-color:rgb(255,255,255)" size="4" face="arial, helvetica, sans-serif">Piotr Gregor</font></div><div><span style="background-color:rgb(255,255,255)"><font size="2" face="arial, helvetica, sans-serif">Software Engineer</font></span></div><div><span style="font-family:arial,helvetica,sans-serif;font-size:x-small"><br></span></div><div><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">M: (+44) 07483 866 525 </span><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">L: (+44) 01256 597 470 </span><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">www: </span><a href="http://dataandsignal.com" style="font-family:arial,helvetica,sans-serif;font-size:x-small" target="_blank">dataandsignal.com</a></div><div><br></div><div><br><br><br></div></div></div></div></div></div></div></div></div></div></div></div></div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, Dec 14, 2020 at 4:56 PM R G <<a href="mailto:mitachundkrach@gmail.com">mitachundkrach@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hello everyone,<br>
<br>
i have a technical question about freeswitch and would like to know if <br>
my idea is technically possible.<br>
<br>
I've got a project at the university and my goal is to improve automatic <br>
speech recognition for bigbluebutton.<br>
<br>
In the summer semester a group of students of my university made a <br>
subtitling plugin[1] for bigbluebutton. This plugin adds a participant <br>
into the meeting and grabs his audiostream to send it to the kaldi ASR <br>
Software. When more than one person speaks at a time the detection <br>
decreases.<br>
<br>
My idea is to copy the incoming freeswitch audiostreams from every <br>
participant and send them directly as a copy to the ASR (kaldi can <br>
handle multiple audiostreams at once).<br>
<br>
In the Confluence-Wiki i searched for something in this direction but <br>
only found things like Dialplan to ring multiple telephones at once.<br>
<br>
I hope my idea is formulated understandably.<br>
<br>
<br>
Kind regards,<br>
<br>
Robert<br>
<br>
[1] <a href="https://github.com/3wille/bbb-kaldi-connector" rel="noreferrer" target="_blank">https://github.com/3wille/bbb-kaldi-connector</a><br>
<br>
<br>
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