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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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
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
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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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- 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
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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.
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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
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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
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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)
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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 ...
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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)
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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)
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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.
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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.
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- 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.
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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.)
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(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.
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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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Shaun Ruffell
Kevin P. Fleming
Jason Parker
Tilghman Lesher
Russell Bryant
Sean Bright
Brett Bryant
Jeff Peeler
Donny Kavanagh
Michiel van Baak
Joshua Colp
Luigi Rizzo