[Freeswitch-dev] Speaker Identification Implementation

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.