Hi, And now the patch to integrate webRTC codecs :) : http://code.google.com/p/csipsimple/source/detail?r=997 It allows to use : * iLBC fix point implemantation (very usefull for old mobiles with CPU that does not support floating point). * iSAC : the codec of Gtalk : 16kHz and 32kHz (floating point only for 32kHz) It also duplicate implementations of g722 and L16. In android makefile I removed the build of these one. And also g711, this one is built with webrtc (webrtc does not allow to exclude it from build), but is excluded by the codec glue with pjsip. Regarding webRTC source code, they'll hopefully add other codecs soon. For example they already have placeholders for AMR and g729. For amr, I guess they'll use opencore so no need of IPP support. There is just a little annoying point : WebRTC has its own way to attribute dynamic payload types. It does that more or less dynamically (which I think a good idea, see my old post : http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2011-January/012504.html and their code : http://code.google.com/p/webrtc/source/browse/trunk/src/modules/audio_coding/main/source/acm_codec_database.cc). As consequence it will probably conflict with already defined codecs. For now to fix that, I'll modify the type.h of pjmedia-codecs to reserve correctly types assigned by webRTC. If anybody has a better idea about this point, it's welcome.
