asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin P. Fleming	887103e83b	and now just to keep the libresample party going... if the functions from libresample are going to be in the main Asterisk binary, it makes sense for the header that defines them to be available without any special CFLAGS and to out-of-tree modules building against /usr/include/asterisk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	04a10c145b	go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	78f4b28552	Instead of linking libresample into the main Asterisk binary, build it as res_resample, and mark codec_resample as dependent upon res_resample. This prevents the linker from optimizing away libresample, and also makes it so the libresample code isn't linked in to multiple places. (I have another module in a branch that needs it, too.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	e637f2c038	make codec_resample build on __CYGWIN__, and make it load on FreeBSD (and probably other systems as well). Both need libresample.a to be specified in the linking phase, and cygwin needs <float.h> as other BSD. The checks for OS-specific headers should really be moved to some common header though. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e0dfa91db8	Use float.h to fix the build on FreeBSD. Also, add some other platforms as they are likely the same. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	21cb767db7	Merge changes from team/russell/codec_resample This commit imports libresample for use in Asterisk. It also adds a new codec module, codec_resample. This module uses libresample to re-sample signed linear audio between 8 kHz and 16 kHz. It also provides an alternative for converting between 16 kHz G.722 and 8 kHz signed linear when using G.722, which will likely be useful as some people have complained about volume issues when the current codec_g722 converts to 8 kHz signed linear. But, to test this, you will have to disable the g722-to-slin and g722-to-slin16 translators in codec_g722.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

6 Commits (7b319d9d69a4d389bcf000a5ecac7a0c521c63db)