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21cb767db7
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
12 lines
274 B
Makefile
12 lines
274 B
Makefile
include $(ASTTOPDIR)/Makefile.moddir_rules
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ASTCFLAGS+= -Isrc -Iinclude
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libresample.a: src/resample.o src/resamplesubs.o src/filterkit.o
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$(ECHO_PREFIX) echo " [AR] $^ -> $@"
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$(CMD_PREFIX) $(AR) cr $@ $^
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$(CMD_PREFIX) $(RANLIB) $@
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clean::
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rm -f src/*.o libresample.a
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