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Merge changes from team/russell/codec_resample

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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
This commit is contained in:
Russell Bryant
2007-12-31 21:22:31 +00:00
parent f3e2f0bb0a
commit 21cb767db7
26 changed files with 10477 additions and 1 deletions
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11
main/libresample/Makefile.asterisk Normal file
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include $(ASTTOPDIR)/Makefile.moddir_rules
ASTCFLAGS+= -Isrc -Iinclude
libresample.a: src/resample.o src/resamplesubs.o src/filterkit.o
$(ECHO_PREFIX) echo " [AR] $^ -> $@"
$(CMD_PREFIX) $(AR) cr $@ $^
$(CMD_PREFIX) $(RANLIB) $@
clean::
rm -f src/*.o libresample.a