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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | 3 lines
Revert a change that introduces an unacceptable performance hit and is causing
memory leaks ... (from rev 97973)
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This fix was made in favor of the proposed patch since it doesn't involve changing
a core codec define.
(closes issue #11722, reported and initially patched by caio1982, final patch by me)
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Instead of using is16kHz(), implement a format_rate() function.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines
1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running. Don't just ignore it.
(Closes issue #11698)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines
Fix various timing calculations that made assumptions that the audio being
processed was at a sample rate of 8 kHz.
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not work for everyone, but it did for some. This set of changes makes trunk
start again for those having problems. Instead of building libresample as a
static library, it just links the object files in directly with the asterisk
binary.
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines
Change ast_translator_best_choice() to only pay attention to audio formats.
This fixes a problem where Asterisk claims that a translation path can not be
found for channels involving video.
(closes issue #11638)
Reported by: cwhuang
Tested by: cwhuang
Patch suggested by cwhuang, with some additional changes by me.
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r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines
Use the constant that I really meant to use here ...
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4 lines
What was I thinking when I wrote this masterpiece?
-1 + 1 = 0.. who woulda thunk it?.
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show translation return the usage information.
(closes issue #11316)
Reported by: eliel
Patches:
translate.c.patch uploaded by eliel (license 64)
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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
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- 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
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- 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.)
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
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(closes issue #10724)
Reported by: eliel
Patches:
chan_skinny.c.patch uploaded by eliel (license 64)
chan_oss.c.patch uploaded by eliel (license 64)
chan_mgcp.c.patch2 uploaded by eliel (license 64)
pbx_config.c.patch uploaded by seanbright (license 71)
iax2-provision.c.patch uploaded by eliel (license 64)
chan_gtalk.c.patch uploaded by eliel (license 64)
pbx_ael.c.patch uploaded by seanbright (license 71)
file.c.patch uploaded by seanbright (license 71)
image.c.patch uploaded by seanbright (license 71)
cli.c.patch uploaded by moy (license 222)
astobj2.c.patch uploaded by moy (license 222)
asterisk.c.patch uploaded by moy (license 222)
res_limit.c.patch uploaded by seanbright (license 71)
res_convert.c.patch uploaded by seanbright (license 71)
res_crypto.c.patch uploaded by seanbright (license 71)
app_osplookup.c.patch uploaded by seanbright (license 71)
app_rpt.c.patch uploaded by seanbright (license 71)
app_mixmonitor.c.patch uploaded by seanbright (license 71)
channel.c.patch uploaded by seanbright (license 71)
translate.c.patch uploaded by seanbright (license 71)
udptl.c.patch uploaded by seanbright (license 71)
threadstorage.c.patch uploaded by seanbright (license 71)
db.c.patch uploaded by seanbright (license 71)
cdr.c.patch uploaded by moy (license 222)
pbd_dundi.c.patch uploaded by moy (license 222)
app_osplookup-rev83558.patch uploaded by moy (license 222)
res_clioriginate.c.patch uploaded by moy (license 222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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options, activated via the menuselect stuff, which defaults to the old way. non-zero sample data added. Translate tables expressed in microseconds instead of milliseconds, with 5-digit data now instead of 3, giving 2 more digits of precision.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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(3-deep nested loop)
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(issue #9984, patches from eliel and dimas)
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guidelines changes
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67631 | file | 2007-06-06 09:18:39 -0400 (Wed, 06 Jun 2007) | 2 lines
Fix plc_samples warning when registering a translator. (issue #9897 reported by xylome)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65877 | qwell | 2007-05-24 11:14:02 -0400 (Thu, 24 May 2007) | 4 lines
Fix handling of zero-length frames when a codec is capable of native PLC.
Issue 9183, patch by Mihai.
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with the PLC code
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spends most of it's time only reading.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46714 | kpfleming | 2006-10-31 15:47:48 -0600 (Tue, 31 Oct 2006) | 2 lines
add an API so that translators can activate/deactivate themselves when needed
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) | 5 lines
Add a small tweak to the code that checks to see whether destination formats
are translatable based on the source format. If we have already determined
that there is no translation path in one direction, don't bother checking the
other direction.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46526 | kpfleming | 2006-10-30 16:19:55 -0600 (Mon, 30 Oct 2006) | 3 lines
when unregistering a translator, don't rebuild the translation matrix unless needed
when filtering formats out of an offer, ensure we check for translation ability in both directions
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) | 11 lines
- If the source has no audio or no video portion, do not call powerof() to
get the format index.
- Don't run through the audio and video loops if there is no audio or video
portion of the source
If 0 is passed to powerof, it will return -1. This value of -1 was then being
used as an array index in these loops, which caused a crash on some systems.
Other than this issue, this code works as we expected it to. If a format is
not in the source, and we have to translation path to it, it is not offered in
the list of acceptable destination formats.
(fixes issue #8231)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) | 2 lines
add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using
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r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) | 2 lines
ensure that the translation matrix is properly lock-protected every place it is used
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r46152 | kpfleming | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines
if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list
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r46153 | kpfleming | 2006-10-24 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines
code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines
add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen)
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reported by Mithraen)
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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