aboutsummaryrefslogtreecommitdiffstats
path: root/codecs
AgeCommit message (Collapse)AuthorFilesLines
2008-09-02Merged revisions 140566 via svnmerge from russell1-1/+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
2008-08-20Fix bug where the samples were not accurate when in G723 mode, which wouldsruffell1-5/+6
cause the timestamp field of the RTP header to be invalid. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@139155 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Removing code that was commented out.sruffell1-111/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@136674 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Updated codec_dahdi to use the transcoder interface in the DAHDI.sruffell1-38/+121
(Issue: DAHDI-42) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@136672 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Merged revisions 132390 via svnmerge from russell1-1/+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
2008-07-21Merged revisions 132388 via svnmerge from russell1-1/+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
2008-07-11Merged revisions 130129 via svnmerge from bbryant3-6/+6
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
2008-06-25Merged revisions 125138 via svnmerge from kpfleming1-1/+2
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@125146 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-17Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵jpeeler1-67/+66
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@123332 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-05Merged revisions 115328 via svnmerge from file1-0/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@115331 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-28Merged revisions 111857 via svnmerge from qwell1-3/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@111858 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26Merged revisions 110881 via svnmerge from kpfleming50-7359/+5
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@110882 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-21Merged revisions 110475 via svnmerge from qwell1-3/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@110476 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-20Merged revisions 110339 via svnmerge from russell1-3/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@110340 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18Merged revisions 109651 via svnmerge from qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@109654 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Merged revisions 107466 via svnmerge from kpfleming1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@107467 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Merged revisions 106501 via svnmerge from russell1-2/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@106502 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16Merged revisions 98951 via svnmerge from file1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@98952 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-15Merged revisions 98943 via svnmerge from russell1-17/+18
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: http://svn.digium.com/svn/asterisk/trunk@98944 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Kevin noted that the thing that I _actually_ changed here was that I convertedrussell1-3/+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
2008-01-11Fix a bus error that happened when asterisk was built with optimizations on russell1-2/+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
2008-01-11At one point during working on this module, I had the lin/lin16 versions of therussell1-34/+2
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: http://svn.digium.com/svn/asterisk/trunk@98218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 - Fix the last set of places where incorrect assumptions were made about therussell1-5/+9
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: http://svn.digium.com/svn/asterisk/trunk@98081 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Fix various issues in codec_g722.russell1-15/+50
- 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: http://svn.digium.com/svn/asterisk/trunk@97975 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Fix the buffer_samples value. For signed linear, the number of samples neededrussell1-1/+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
2008-01-10Fix this so it doesn't force codec_g722 to get relinked every timerussell1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97652 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Ensure that libg722.a gets rebuilt if one of the files changesrussell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97650 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97491 via svnmerge from kpfleming1-2/+4
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: http://svn.digium.com/svn/asterisk/trunk@97495 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02and now just to keep the libresample party going... if the functions from ↵kpfleming2-4/+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
2008-01-02go back to including libresample in the main Asterisk binary, but this time ↵kpfleming2-5/+1
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
2008-01-02Instead of linking libresample into the main Asterisk binary, build it asrussell2-4/+5
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
2008-01-01Fix building of codec_resample on platforms other then Cygwin. On everything ↵file1-1/+3
else it actually gets built after codec_resample, so you can't exactly link it in since it doesn't exist. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95648 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01make codec_resample build on __CYGWIN__, and make it load on FreeBSDrizzo2-1/+2
(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
2007-12-31Use float.h to fix the build on FreeBSD. Also, add some other platforms asrussell1-1/+1
they are likely the same. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95550 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Merge changes from team/russell/codec_resamplerussell3-0/+279
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
2007-12-27I went looking for where we downloaded the g722 implementation and came acrossrussell1-0/+3
these two links. So, I'm adding them so they are available for reference later. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94877 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21codecs.conf really shouldn't be mandatory.. it never had been before, so ↵qwell9-12/+12
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: http://svn.digium.com/svn/asterisk/trunk@94541 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17Merged revisions 93180 via svnmerge from kpfleming1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@93184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Solaris compat fixestilghman1-1/+0
Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11315) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93090 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10Put into Makefile.moddir_rules the common instructions used torizzo1-9/+1
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: http://svn.digium.com/svn/asterisk/trunk@92082 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-09normalize subdirs' Makefile by using ASTTOPDIR and not .. to referencerizzo1-1/+1
the top level directory. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92022 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22remove a number of #include <fcntl.h> which are eitherrizzo10-19/+0
useless or done elsewhere git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89516 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19remove some useless includes from codecsrizzo11-37/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89428 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo11-12/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo11-44/+0
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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08improve linked-list macros in two ways:kpfleming1-1/+1
- 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: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Commit some cleanups to the format type code.tilghman1-1/+75
- 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: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06Merged revisions 89046 via svnmerge from qwell1-2/+1
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: http://svn.digium.com/svn/asterisk/trunk@89047 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31More changes to change return values from load_module functions.qwell11-81/+132
(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: http://svn.digium.com/svn/asterisk/trunk@87889 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-29clean up assembler and preprocessor files if they are here tookpfleming3-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87467 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-07 23:40:00 +0000