dect/asterisk - Asterisk with DECT support

Age	Commit message (Collapse)	Author	Files	Lines
2008-09-02	Merged revisions 140566 via svnmerge from	russell	1	-1/+2
	https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 \| russell \| 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) \| 2 lines Update instructions for getting libresample ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@140567 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21	Merged revisions 132390 via svnmerge from	russell	1	-1/+8
	https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 \| russell \| 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) \| 16 lines Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132391 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21	Merged revisions 132388 via svnmerge from	russell	1	-1/+1
	https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 \| russell \| 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) \| 3 lines Enable higher quality resampling, as it doesn't have a noticeable performance impact on my machine .. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@132389 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11	Merged revisions 130129 via svnmerge from	bbryant	1	-2/+2
	https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 \| bbryant \| 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) \| 8 lines Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@130130 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11	Kevin noted that the thing that I _actually_ changed here was that I converted	russell	1	-3/+3
	a value from a double, to a float, back to a double. Sure enough, when I changed my interim variable back to a double, it still blows up. Switching all of these to a float fixes the problem. This seems like a compiler bug where a double passed as an argument isn't getting properly aligned, so I'll have to see if I can replicate it with a small test program. (related to issue #11725) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98308 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11	Fix a bus error that happened when asterisk was built with optimizations on	russell	1	-2/+6
	with platforms that explode on unaligned access. I'm not exactly sure why this fixes it, but it fixed it on the machine I was testing on. If it makes sense to you, feel free to enlighten me. :) (closes issue #11725, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98270 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10	Fix the buffer_samples value. For signed linear, the number of samples needed	russell	1	-1/+1
	to fill the buffer is half the buffer size. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97974 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02	and now just to keep the libresample party going... if the functions from ↵	kpfleming	1	-2/+1
	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: http://svn.digium.com/svn/asterisk/trunk@95894 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02	go back to including libresample in the main Asterisk binary, but this time ↵	kpfleming	1	-4/+0
	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: http://svn.digium.com/svn/asterisk/trunk@95816 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02	Instead of linking libresample into the main Asterisk binary, build it as	russell	1	-0/+4
	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: http://svn.digium.com/svn/asterisk/trunk@95697 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01	make codec_resample build on __CYGWIN__, and make it load on FreeBSD	rizzo	1	-1/+1
	(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: http://svn.digium.com/svn/asterisk/trunk@95625 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31	Use float.h to fix the build on FreeBSD. Also, add some other platforms as	russell	1	-1/+1
	they are likely the same. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95550 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31	Merge changes from team/russell/codec_resample	russell	1	-0/+234
	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: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b