Commit Graph

378 Commits

Author SHA1 Message Date
David Brooks 48363c16e1 Fixes numerous spelling errors. Patch submitted by alecdavis.
(closes issue #15595)
Reported by: alecdavis



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-30 16:07:05 +00:00
Kevin P. Fleming 96e4e31eeb Merged revisions 207647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
  
  Ensure that user-provided CFLAGS and LDFLAGS are honored.
  
  This commit changes the build system so that user-provided flags (in ASTCFLAGS
  and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
  by the build system itself, so that the user can effectively override the
  build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
  be provided *either* in the environment before running 'make', or as variable
  assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
  is no longer necessary, so they are no longer documented, but are still supported
  so as not to break existing build systems that supply them when building Asterisk.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 13:28:04 +00:00
Sean Bright 6b5dbba90c Merged revisions 206635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line
  
  Only print debug info in codec_dahdi if we are asking for it.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-15 16:00:24 +00:00
David Vossel dcfe69ec64 fixes some memory leaks and redundant conditions
(closes issue #15269)
Reported by: contactmayankjain
Patches:
      patch.txt uploaded by contactmayankjain (license 740)
      memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
Tested by: contactmayankjain, dvossel




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-18 16:37:42 +00:00
Russell Bryant 19853c437e Shuttle some bits around to address some gain issues with G.722.
(closes AST-209)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15 17:59:08 +00:00
Russell Bryant 66c76daba1 Further simplify codec_g722 build.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15 17:37:12 +00:00
Russell Bryant 79b5f06f34 Actually force running make for g722.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15 17:24:39 +00:00
Shaun Ruffell c65029b74e Several changes to codec_dahdi to play nice with G723.
This commit brings in the changes that were living out on the
svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch.  codec_dahdi.c now
always uses signed linear as the simple codec so that a soft g729 codec will
not end up being preferred to the hardware codec.  There are also changes to
allow codec_dahdi.c to feed packets to the hardware in the native sample size of
the codec.  This solves problems with choppy audio when using G723. 



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 22:28:41 +00:00
Kevin P. Fleming 8d5deb312b Merged revisions 157859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
  
  the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
  
  with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
  
  while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-20 00:08:12 +00:00
Kevin P. Fleming 10d36d9f34 fix a few small things found by using sparse
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30 16:49:02 +00:00
Jason Parker ae0a736353 Merge codec_consistency branch. This should make sample usage much happier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17 21:35:23 +00:00
Tilghman Lesher a8775172aa When using MALLOC_DEBUG, codec_lpc10 leaks memory, because it matches a library
malloc() with an ast_free (which, of course, doesn't match up with known
allocated memory, so the free fails).
(closes issue #13702)
 Reported by: eliel
 Patches: 
       codec_lpc10_lpcini.c uploaded by eliel (license 64)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-15 16:41:54 +00:00
Tilghman Lesher 08af5bb312 Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating
when a file is invalid from when a file is missing.  This is most important when
we have two configuration files.  Consider the following example:

Old system:
sip.conf     users.conf     Old result               New result
========     ==========     ==========               ==========
Missing      Missing        SIP doesn't load         SIP doesn't load
Missing      OK             SIP doesn't load         SIP doesn't load
Missing      Invalid        SIP doesn't load         SIP doesn't load
OK           Missing        SIP loads                SIP loads
OK           OK             SIP loads                SIP loads
OK           Invalid        SIP loads incompletely   SIP doesn't load
Invalid      Missing        SIP doesn't load         SIP doesn't load
Invalid      OK             SIP doesn't load         SIP doesn't load
Invalid      Invalid        SIP doesn't load         SIP doesn't load

So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed.  Worse yet, the old
system would do this with no indication that anything was even wrong.

(closes issue #10690)
 Reported by: dtyoo
 Patches: 
       20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 23:30:03 +00:00
Russell Bryant c0c743a5fa Update instructions for getting libresample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-02 15:11:53 +00:00
Shaun Ruffell b39e0f3c66 Remove extraneous debugging messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-20 20:03:28 +00:00
Shaun Ruffell f75d794c69 Fix bug where the samples were not accurate when in G723 mode, which would
cause the timestamp field of the RTP header to be invalid.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-20 19:57:22 +00:00
Sean Bright db1ed285c4 More RSW merges. This should do it for the channels/ dir.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-09 14:12:34 +00:00
Shaun Ruffell f9ba4bf070 Updating codec_dahdi to the new transcoder interface.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-07 20:54:52 +00:00
Sean Bright 6305009cfc More merges from resolve-shadow warnings:
utils/
  codecs/
  and a change I missed from formats/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-07 15:16:48 +00:00
Russell Bryant c87f901cfd 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/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-21 14:47:41 +00:00
Russell Bryant 5de127e103 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/trunk@132388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-21 13:51:05 +00:00
Brett Bryant 5b7933fe5e 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/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11 18:09:35 +00:00
Tilghman Lesher 7b84cf6fa6 Convert casts to unions, to fix alignment issues on Solaris
(closes issue #12932)
 Reported by: snuffy
 Patches: 
       bug_12932_20080627.diff uploaded by snuffy (license 35)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-26 17:06:17 +00:00
Kevin P. Fleming fd4a60c459 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/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-25 23:05:28 +00:00
Jeff Peeler ef3b214728 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/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12 17:27:55 +00:00
Donny Kavanagh 2433d6ecb9 Revision 117802 changed frame.data to frame.data.ptr however codec_ilbc.c was not updated. This resolves that oversight.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10 19:03:11 +00:00
Jason Parker f7eb823a7a Fix a few places where frame data was used directly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22 17:10:53 +00:00
Michiel van Baak f1e9371da8 - revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- make data member of the ast_frame struct a named union instead of a void

Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.

The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.

This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data

Thanks russellb and kpfleming for the feedback.

(closes issue #12674)
Reported by: mvanbaak


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22 16:29:54 +00:00
Joshua Colp fc120bf827 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/trunk@115328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-05 22:13:57 +00:00
Jason Parker 7af650c18b 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/trunk@111857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28 21:46:02 +00:00
Kevin P. Fleming 789831ef9a 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/trunk@110881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 17:10:28 +00:00
Jason Parker e3ea6829bb 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/trunk@110475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21 14:36:17 +00:00
Russell Bryant f31cc97197 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/trunk@110339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 22:02:20 +00:00
Jason Parker 56bddc2a0d 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/trunk@109651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 19:24:15 +00:00
Kevin P. Fleming 4925e7b835 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/trunk@107466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 15:13:38 +00:00
Russell Bryant 5ca5d97673 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/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 00:24:58 +00:00
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
2008-01-16 01:17:25 +00:00
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
2008-01-15 23:31:53 +00:00
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
2008-01-11 19:05:24 +00:00
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
2008-01-11 18:48:07 +00:00
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
2008-01-11 17:17:54 +00:00
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
2008-01-11 03:37:19 +00:00
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
2008-01-10 23:16:09 +00:00
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
2008-01-10 23:10:00 +00:00
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
2008-01-10 00:17:02 +00:00
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
2008-01-10 00:11:02 +00:00
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
2008-01-09 17:30:13 +00:00
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
2008-01-02 18:21:04 +00:00
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
2008-01-02 14:05:30 +00:00
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
2008-01-02 01:00:44 +00:00