[Freeswitch-users] FreeSWITCH, MRCP and Perl

[mszlazak at aol.com]

 Yup, or just get pocketsphinx "tuned" up for telephony and then no one will have to bother with ASR vendors.

I believe that some speech data from a good size sample for training is needed to make it more "speaker independent" and better suited for use with phone calls. I have a list of things from the Sphinx forums that would be good to have for a telephony ready PocketSphinx. There is a "wsj" database but I don't know if that's would help??

Best. Mark.


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 12 Jan 2009 9:29 am
Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl









"With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the outbound socket interface, unicast and a bit of Perl."



Was mod_openmrcp not enough :) ?We really need someone to fund the writing of mod_unimrcp.




/b




On Jan 12, 2009, at 5:49 AM, David Knell wrote:



Hi all -

In case anyone's interested, I've documented how we interfaced FS with 
Lumenvox via MRCP using FS' event socket and unicast interfaces and a 
bit of Perl here: 
http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl

Three surprises: that it worked at all, that it works quite well and 
that it was really quite easy to do.

One thing I'm looking for: has anyone written a module which attaches a 
bug to an audio stream and forwards the audio as RTP to a specified 
IP/port to just allow audio to be tapped off a call and sent somewhere 
else to be listened to?

Cheers --

Dave

-- 
David Knell, Director, 3C Limited
T: 020 8114 8901 ?F: 020 3002 7257 ?M: 001 415 630 3031
http://www.3c.co.uk 


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=


 





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