Hi guys, our company wants to use speaker identification/verification (SID) in FreeSWITCH. However, only ASR and TTS are currently implemented so we decided to contribute by coding the SID interface as well as speaker identification/verification part in mod_mrcp. Is it possible to get our code commited and made available to public within a month ?? And what's the procedure in getting our code commited ?? I think there's a huge gap on the FS web in information for newbies wanting to help with coding... PS: I'm not sure whether this is the right place to ask such question but I've been to #asterisk-dev on IRC and nobody answered my question... Thanks for help.
Join #freeswitch on freenode... People are there right now that can help you On 8/10/11 3:29 AM, "Viktor Jamrich" <viktor_jamrich@centrum.sk> wrote:
Hi guys, our company wants to use speaker identification/verification (SID) in FreeSWITCH. However, only ASR and TTS are currently implemented so we decided to contribute by coding the SID interface as well as speaker identification/verification part in mod_mrcp. Is it possible to get our code commited and made available to public within a month ?? And what's the procedure in getting our code commited ?? I think there's a huge gap on the FS web in information for newbies wanting to help with coding...
PS: I'm not sure whether this is the right place to ask such question but I've been to #asterisk-dev on IRC and nobody answered my question...
Thanks for help.
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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#freeswitch-dev is much friendlier :) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 2011-08-15, at 5:35 PM, Ken Rice wrote:
Join #freeswitch on freenode... People are there right now that can help you
On 8/10/11 3:29 AM, "Viktor Jamrich" <viktor_jamrich@centrum.sk> wrote:
Hi guys, our company wants to use speaker identification/verification (SID) in FreeSWITCH. However, only ASR and TTS are currently implemented so we decided to contribute by coding the SID interface as well as speaker identification/verification part in mod_mrcp. Is it possible to get our code commited and made available to public within a month ?? And what's the procedure in getting our code commited ?? I think there's a huge gap on the FS web in information for newbies wanting to help with coding...
PS: I'm not sure whether this is the right place to ask such question but I've been to #asterisk-dev on IRC and nobody answered my question...
Thanks for help.
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
If you mean .. Biometric Voice (which is equivalent to fingerprint) Nuance just purchased a company and then there are a few European/Russian companies that OeM the product. -E
participants (4)
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EdPimentl
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Ken Rice
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Mathieu Rene
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Viktor Jamrich