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Author SHA1 Message Date
sruffell
63f40a9bd9 Remove extraneous debugging messages.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139154 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-20 20:03:28 +00:00
sruffell
b51a73f888 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: http://svn.digium.com/svn/asterisk/trunk@139153 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-20 19:57:22 +00:00
seanbright
3d55cb9df3 More RSW merges. This should do it for the channels/ dir.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136917 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-09 14:12:34 +00:00
sruffell
cc06499d99 Updating codec_dahdi to the new transcoder interface.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136676 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07 20:54:52 +00:00
seanbright
f21f6ae82a More merges from resolve-shadow warnings:
utils/
  codecs/
  and a change I missed from formats/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136408 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07 15:16:48 +00:00
russell
4af9e5c085 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/trunk@132390 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21 14:47:41 +00:00
russell
d704c1aaf6 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/trunk@132388 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21 13:51:05 +00:00
bbryant
8e222897e6 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/trunk@130129 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11 18:09:35 +00:00
tilghman
a1fa45760e 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: http://svn.digium.com/svn/asterisk/trunk@125386 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-26 17:06:17 +00:00
kpfleming
ae1eb91abe 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/trunk@125138 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-25 23:05:28 +00:00
jpeeler
490730a6b3 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: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12 17:27:55 +00:00
juggie
fa257271e7 Revision 117802 changed frame.data to frame.data.ptr however codec_ilbc.c was not updated. This resolves that oversight.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121599 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-10 19:03:11 +00:00
qwell
dac7a3528e Fix a few places where frame data was used directly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117828 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22 17:10:53 +00:00
mvanbaak
c1210321e7 - 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: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22 16:29:54 +00:00
file
82f9045435 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/trunk@115328 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-05 22:13:57 +00:00
qwell
e53c6f4673 Merged revisions 111856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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/trunk@111857 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-28 21:46:02 +00:00
kpfleming
adfd7f5f13 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/trunk@110881 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26 17:10:28 +00:00
qwell
7271799a8e Merged revisions 110474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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/trunk@110475 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-21 14:36:17 +00:00
russell
9d0264511a 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/trunk@110339 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-20 22:02:20 +00:00
qwell
427d420574 Merged revisions 109648 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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/trunk@109651 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18 19:24:15 +00:00
kpfleming
a333628652 Merged revisions 107464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines

fix various other problems found by gcc 4.3

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107466 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11 15:13:38 +00:00
russell
5ffedddec1 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/trunk@106501 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 00:24:58 +00:00
file
35d5a377ed Merged revisions 98951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98952 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16 01:17:25 +00:00
russell
b61a98675c Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
2008-01-15 23:31:53 +00:00
russell
9c1c46c009 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: http://svn.digium.com/svn/asterisk/trunk@98308 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 19:05:24 +00:00
russell
ccce263b5c 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: http://svn.digium.com/svn/asterisk/trunk@98270 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 18:48:07 +00:00
russell
2f83bcc869 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: http://svn.digium.com/svn/asterisk/trunk@98218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 17:17:54 +00:00
russell
560327b0ec - 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: http://svn.digium.com/svn/asterisk/trunk@98081 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 03:37:19 +00:00
russell
300fa53d79 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: http://svn.digium.com/svn/asterisk/trunk@97975 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 23:16:09 +00:00
russell
57ccc02998 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: http://svn.digium.com/svn/asterisk/trunk@97974 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 23:10:00 +00:00
russell
a2a1eb045d Fix this so it doesn't force codec_g722 to get relinked every time
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97652 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 00:17:02 +00:00
russell
d4f2402f2e Ensure that libg722.a gets rebuilt if one of the files changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97650 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 00:11:02 +00:00
kpfleming
bf8028a2c8 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: http://svn.digium.com/svn/asterisk/trunk@97495 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09 17:30:13 +00:00
kpfleming
3f0b6d8d86 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: http://svn.digium.com/svn/asterisk/trunk@95894 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02 18:21:04 +00:00
kpfleming
933ddf410a 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: http://svn.digium.com/svn/asterisk/trunk@95816 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02 14:05:30 +00:00
russell
21969815a0 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: http://svn.digium.com/svn/asterisk/trunk@95697 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02 01:00:44 +00:00
file
59bc75d1f8 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: http://svn.digium.com/svn/asterisk/trunk@95648 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01 23:09:32 +00:00
rizzo
5503f69d4c 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: http://svn.digium.com/svn/asterisk/trunk@95625 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01 22:21:39 +00:00
russell
250f46db38 Use float.h to fix the build on FreeBSD. Also, add some other platforms as
they are likely the same.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95550 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31 22:41:39 +00:00
russell
04838b9d59 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: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31 21:22:31 +00:00
russell
806b3cfdd3 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: http://svn.digium.com/svn/asterisk/trunk@94877 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27 16:11:41 +00:00
qwell
7c0eb401e6 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: http://svn.digium.com/svn/asterisk/trunk@94541 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21 20:12:50 +00:00
kpfleming
d4e966efcc 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: http://svn.digium.com/svn/asterisk/trunk@93184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17 07:25:35 +00:00
tilghman
a4425cc28d Solaris compat fixes
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
2007-12-14 21:09:17 +00:00
rizzo
aa85540763 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: http://svn.digium.com/svn/asterisk/trunk@92082 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10 03:50:38 +00:00
rizzo
b50ce18fe8 normalize subdirs' Makefile by using ASTTOPDIR and not .. to reference
the top level directory.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92022 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-09 21:29:37 +00:00
rizzo
8cd33321ef remove a number of #include <fcntl.h> which are either
useless or done elsewhere



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89516 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 01:03:02 +00:00
rizzo
e8c3c0d206 remove some useless includes from codecs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89428 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 19:51:55 +00:00
rizzo
9cf442d7f7 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: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 18:52:04 +00:00
rizzo
883346d64a 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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16 20:04:58 +00:00