asterisk/main/libresample/Makefile.asterisk

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
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			`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`