Commit Graph

3 Commits

Author SHA1 Message Date
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
2008-01-01 22:21:39 +00:00
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
2007-12-31 22:41:39 +00:00
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
2007-12-31 21:22:31 +00:00