Hello, Attachment is my Skype SILK codec patch for PJSIP 1.12. Only tested at centos 5.x and Freeswitch SILK codec impl. no SILK FEC, DTX. has PLC. For anyone who has interest to have a try SILK. regards, Gang On Wed, Nov 23, 2011 at 10:44 AM, Gang Liu <gangban.lau at gmail.com> wrote: > hello, > ? ? ? I am playing Skype SILK codec and considering how to add FEC > feature of SILK. > So is it able to access pjmedia stream jitter buffer from > pjmedia_codec implementation? > And Is there a way to bypass pjsua conference bridge resampling and > let samples directly > go to SILK codec in below case? > > ? ? ? PJSUA CONF_BRIDGE (16000hz) --- > resample port --> SILK 8000hz > codec instance > > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? | ?bypass > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?V > > ? ? ? PJSUA CONF_BRIDGE (16000hz) --> SILK (API sample rate 16Khz, > internal 8Khz) codec instance > > regards, > Gang > -------------- next part -------------- A non-text attachment was scrubbed... Name: silk_pjsip_1.12.tar.gz Type: application/x-gzip Size: 9390 bytes Desc: not available URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20111130/414eb6f1/attachment.gz>