asterisk

Commit Graph

Author	SHA1	Message	Date
Shaun Ruffell	732659ef67	Removing code that was commented out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Shaun Ruffell	7fbba1d580	Updated codec_dahdi to use the transcoder interface in the DAHDI. (Issue: DAHDI-42) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b266380e2	Merged revisions 132390 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	bc50069f75	Merged revisions 132388 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	559bbfc457	Merged revisions 130129 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	0b30f14b33	Merged revisions 125138 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125138 \| kpfleming \| 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) \| 18 lines Merged revisions 125132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 \| kpfleming \| 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) \| 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	f089bbf663	Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	0ed86b307e	Merged revisions 115328 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115328 \| file \| 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) \| 10 lines Merged revisions 115327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115327 \| file \| 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) \| 2 lines Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@115331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	557219e0dc	Merged revisions 111857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111857 \| qwell \| 2008-03-28 16:46:02 -0500 (Fri, 28 Mar 2008) \| 20 lines Merged revisions 111856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111856 \| qwell \| 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) \| 12 lines Allow gsm to compile correctly on x86 with gcc4 optimizations. (closes issue #11243) Reported by: whiskerp Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) Tested by: Seggy (IRC) Note: While I did write this patch, I would not have found this if fossil had not reported and fixed issue #12253. A huge thanks to him for helping to (indirectly) find the problem here. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@111858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	085ef90375	Merged revisions 110881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110881 \| kpfleming \| 2008-03-26 10:10:28 -0700 (Wed, 26 Mar 2008) \| 18 lines Merged revisions 110880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110880 \| kpfleming \| 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) \| 10 lines Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 \| kpfleming \| 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) \| 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@110882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	57d3a27f1b	Merged revisions 110475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110475 \| qwell \| 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) \| 15 lines Merged revisions 110474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110474 \| qwell \| 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) \| 7 lines Don't attempt to do optimizations of gsm on mips platforms either. (closes issue #12270) Reported by: zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@110476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	2adbae37a8	Merged revisions 110339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 \| russell \| 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) \| 3 lines Use the correct buffer for g722tolin16_sample. This shouldn't have caused any problems, but Qwell noticed the typo here. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@110340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	4f2ec37cbd	Merged revisions 109651 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109651 \| qwell \| 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) \| 15 lines Merged revisions 109648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109648 \| qwell \| 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) \| 7 lines Allow codecs that use log2comp (g726) to compile correctly on x86 with gcc4 optimizations. (closes issue #12253) Reported by: fossil Patches: log2comp.patch uploaded by fossil (license 140) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@109654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	19ca2892c0	Merged revisions 107466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107466 \| kpfleming \| 2008-03-11 10:13:38 -0500 (Tue, 11 Mar 2008) \| 10 lines Merged revisions 107464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107464 \| kpfleming \| 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) \| 2 lines fix various other problems found by gcc 4.3 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	ccf90e8186	Merged revisions 106501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 \| russell \| 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) \| 28 lines Merge changes from team/russell/g722-sillyness ... Fix a number of other places where the number of samples in a G722 frame was not properly handled because of various reasons. main/rtp.c: - When a G722 frame is read from the smoother, the number of samples in the frame must be divided by 2 before being sent out over the network. Even though G722 is 16 kHz, an error in some previous spec has made it so that we have to list the number of samples such as if it was 8 kHz. main/file.c: - When scheduling the next time to expect a frame, take into account that the format of the file we're reading from may not be 8 kHz. codecs/codec_g722.c: - When converting from G722 to slinear, g722_decode() expects its samples parameter to be in the silly (real samples / 2) format. Make it so. - When converting from slinear to G722, properly set the number of samples in the frame to be the number of bytes of output * 2. formats/format_pcm.c: - This format module handles G722, among a number of other formats. However, the read() and seek() functions did not account for the fact that G722 has 2 samples per byte. (closes issue #12130, reported by rickross, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@106502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	c692e1cacb	Merged revisions 98951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98951 \| file \| 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) \| 4 lines Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex. (closes issue #11693) Reported by: yzg ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	4fb04cb58a	Merged revisions 98943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98943 \| russell \| 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) \| 25 lines Commit a fix for some memory access errors pointed out by the valgrind2.txt output on issue #11698. The issue here is that it is possible for an instance of a translator to get destroyed while the frame allocated as a part of the translator is still being processed. Specifically, this is possible anywhere between a call to ast_read() and ast_frame_free(), which is _a lot_ of places in the code. The reason this happens is that the channel might get masqueraded during this time. During a masquerade, existing translation paths get destroyed. So, this patch fixes the issue in an API and ABI compatible way. (This one is for you, paravoid!) It changes an int in ast_frame to be used as flag bits. The 1 bit is still used to indicate that the frame contains timing information. Also, a second flag has been added to indicate that the frame came from a translator. When a frame with this flag gets released and has this flag, a function is called in translate.c to let it know that this frame is doing being processed. At this point, the flag gets cleared. Also, if the translator was requested to be destroyed while its internal frame still had this flag set, its destruction has been deffered until it finds out that the frame is no longer being processed. Admittedly, this feels like a hack. But, it does fix the issue, and I was not able to think of a better solution ... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	7bc92eda05	Kevin noted that the thing that I _actually_ changed here was that I converted 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: https://origsvn.digium.com/svn/asterisk/trunk@98308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	d1256bb8a6	Fix a bus error that happened when asterisk was built with optimizations on 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: https://origsvn.digium.com/svn/asterisk/trunk@98270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c67509e429	At one point during working on this module, I had the lin/lin16 versions of the framein callbacks different. However, they are now the same again, so remove the duplicate code and use the same functions for the lin/lin16 versions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	9387f036d8	- Fix the last set of places where incorrect assumptions were made about the sample length with g722. It is _2_ samples per byte, not 1. This was all over the place, and I believed it, and it is what caused me to take so long to figure out what was broken. - Update copyright information on codec_g722. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	a2c08fe0f0	Fix various issues in codec_g722. - The most common fix being made here is to fix all of the places where the number of output samples and output bytes gets updated in the translator state structure. - Fix a number of other places where the number of samples provided as an initialization value to a struct was incorrect. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	0a5bfb9eb8	Fix the buffer_samples value. For signed linear, the number of samples needed to fill the buffer is half the buffer size. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	0da0033675	Fix this so it doesn't force codec_g722 to get relinked every time git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	cc11e9569d	Ensure that libg722.a gets rebuilt if one of the files changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	c7ba18820e	Merged revisions 97491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97491 \| kpfleming \| 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) \| 2 lines report the same message whether Zaptel does not have transcoder support loaded or no transcoders were found ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	887103e83b	and now just to keep the libresample party going... if the functions from 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: https://origsvn.digium.com/svn/asterisk/trunk@95894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	04a10c145b	go back to including libresample in the main Asterisk binary, but this time 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: https://origsvn.digium.com/svn/asterisk/trunk@95816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	78f4b28552	Instead of linking libresample into the main Asterisk binary, build it as 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: https://origsvn.digium.com/svn/asterisk/trunk@95697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	d0944cde47	Fix building of codec_resample on platforms other then Cygwin. On everything else it actually gets built after codec_resample, so you can't exactly link it in since it doesn't exist. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
Russell Bryant	f24c226e09	I went looking for where we downloaded the g722 implementation and came across these two links. So, I'm adding them so they are available for reference later. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	4710e2b6da	codecs.conf really shouldn't be mandatory.. it never had been before, so let's go back to being optional. A big "thank you" to pnlarsson on IRC for allowing me access to his system to debug this. Closes issue #11584. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	100ef27af9	Merged revisions 93180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93180 \| kpfleming \| 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) \| 23 lines In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, rizzo brought up some issues related to the way that the metadata required for menuselect and the rest of the build system is extracted from the source files. Since I had a few hours to kill on an airplane today, I decided to improve this situation... so now the system caches the extracted metadata and uses it to build the menuselect 'tree' as much as it can. The result of this is that when a single source file is changed, only the metadata for that file needs to be extracted again, and the rest is used from the cache files. I also reduced the number of forked processes required to do the metadata extraction; it was actually possible to do most of what we needed in the Makefiles themselves without using any shell scripts at all! On my laptop, these changes resulted in an 80% decrease in the time required for the 'menuselect.makeopts' automatic check to occur after editing a single source file. While doing this work I also cleaned up a few minor things in the Makefiles, adding a check for 'awk' to the configure script and changed all remaining places we use 'grep' or 'awk' to use the ones found by the configure script, and changed the 'prep_tarball' script to build the menuselect metadata so that tarballs of Asterisk will include it and won't require the user to wait while it is extracted after unpacking. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7029e8fdd1	Solaris compat fixes Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11315) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	5490889153	Put into Makefile.moddir_rules the common instructions used to generate loadable and embedded module lists. Individual Makefiles now are a lot simpler, possibly as simple as this: -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps MODULE_PREFIX=cdr_ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because in a single directory we can combine various types of modules (app_, cdr_, func_, ... ) by simply listing them in the MODULE_PREFIX variable. The individual Makefiles can also create list of modules to be excluded by listing them in the variablel MODULE_EXCLUDE (see an example in channels/Makefile). With this change it becomes trivial to integrate a directory with locally created/modified sources into the main build. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	d652be0930	normalize subdirs' Makefile by using ASTTOPDIR and not .. to reference the top level directory. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	3fc2646808	remove a number of #include <fcntl.h> which are either useless or done elsewhere git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	7bcf0529c5	remove some useless includes from codecs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	0595b5e2aa	include "logger.h" and errno.h from asterisk.h - usage shows that they were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	edc78d6023	improve linked-list macros in two ways: - the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7c56918262	Commit some cleanups to the format type code. - Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	23a02e04c5	Merged revisions 89046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89046 \| qwell \| 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) \| 4 lines Correctly set the total number of channels from a zaptel transcoder board. SPD-49, patch by Matthew Nicholson. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	59c9ff7ef2	More changes to change return values from load_module functions. (issue #11096) Patches: codec_adpcm.c.patch uploaded by moy (license 222) codec_alaw.c.patch uploaded by moy (license 222) codec_a_mu.c.patch uploaded by moy (license 222) codec_g722.c.patch uploaded by moy (license 222) codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy (license 222) codec_lpc10.c.patch uploaded by moy (license 222) codec_speex.c.patch uploaded by moy (license 222) codec_ulaw.c.patch uploaded by moy (license 222) codec_zap.c.patch uploaded by moy (license 222) format_g723.c.patch uploaded by moy (license 222) format_g726.c.patch uploaded by moy (license 222) format_g729.c.patch uploaded by moy (license 222) format_gsm.c.patch uploaded by moy (license 222) format_h263.c.patch uploaded by moy (license 222) format_h264.c.patch uploaded by moy (license 222) format_ilbc.c.patch uploaded by moy (license 222) format_jpeg.c.patch uploaded by moy (license 222) format_ogg_vorbis.c.patch uploaded by moy (license 222) format_pcm.c.patch uploaded by moy (license 222) format_sln.c.patch uploaded by moy (license 222) format_vox.c.patch uploaded by moy (license 222) format_wav.c.patch uploaded by moy (license 222) format_wav_gsm.c.patch uploaded by moy (license 222) res_adsi.c.patch uploaded by eliel (license 64) res_ael_share.c.patch uploaded by eliel (license 64) res_clioriginate.c.patch uploaded by eliel (license 64) res_convert.c.patch uploaded by eliel (license 64) res_indications.c.patch uploaded by eliel (license 64) res_musiconhold.c.patch uploaded by eliel (license 64) res_smdi.c.patch uploaded by eliel (license 64) res_speech.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	a4b8ca220d	clean up assembler and preprocessor files if they are here too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	ebe4050128	Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b0f3e6097e	Convert NEW_CLI to AST_CLI. Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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357 Commits (568dab95918eb4f718bf05e91c95396db9e57322)