(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
iconv dependency for func_iconv.
This fixes some build issues on CYGWIN and FreeBSD and probably
other platforms where libiconv is not there by default
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95624 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines
Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then
calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is
dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension
into ast_merge_contexts_and_delete (sans the extra lock).
(this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the
problematic area experienced by the reporters of that issue)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95578 f38db490-d61c-443f-a65b-d21fe96a405b
to an 8 kHz endpoint, then codec_resample will automatically be used to properly
resample the audio before sending it to/from chan_console.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95527 f38db490-d61c-443f-a65b-d21fe96a405b
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
Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface. It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95412 f38db490-d61c-443f-a65b-d21fe96a405b
to add more entries. This required moving struct grab_desc to the common
header, and adding an entry in the Makefile.
On passing, cleanup some comments and file headers (some are still missing).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95313 f38db490-d61c-443f-a65b-d21fe96a405b
This makes it very convenient to enlarge images using the right-click
on the video window.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95264 f38db490-d61c-443f-a65b-d21fe96a405b
of the queue_exec function by reversing the logic of an if statement. This change makes the function
comply better with the coding guidelines. Since this change is purely a cosmetic change to the code, I am
only committing the change to trunk.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95233 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | 6 lines
Remove duplicate increment of the header count in the add_header() function.
(closes issue #11648)
Reported by: makoto
Patch provided by sergee, committed patch by me, inspired by comments from putnopvut
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95192 f38db490-d61c-443f-a65b-d21fe96a405b
The channel name is printed in verbose messages
maximumWordLength option added.
Duration of words that do not meet the minimum word duration will be logged
The duration of pre-greeting silence will be logged
Only consider us in the greeting if we actually detected a valid word duration.
(closes issue #11650, reported and patched by davevg)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95167 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec 2007) | 8 lines
I found a bug while browsing the queue code and managed to reproduce it in a small setup.
If a queue uses the ringall strategy, it was possible through unfortunate coincidence for a single member at a given penalty level to
make app_queue think that all members at that penalty level were unavailable and cause the members at the
next penalty level to be rung. With this patch, we will only move to the next penalty level if ALL the members
at a given penalty level are unreachable.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95096 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | 9 lines
Don't report a syntax error when an empty string is passed to ast_get_group.
Just return 0.
(closes issue #11540)
Reported by: tzafrir
Patches:
group_empty.diff uploaded by tzafrir (license 46)
-- slightly changed by me
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95025 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 lines
Include types.h in chan_h323 as without it it can not be compiled on some operating systems like FreeBSD to name one.
(closes issue #11585)
Reported by: sobomax
Patches:
chan_h323.c.diff uploaded by sobomax (license 359)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94934 f38db490-d61c-443f-a65b-d21fe96a405b
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r94831 | russell | 2007-12-27 09:16:56 -0600 (Thu, 27 Dec 2007) | 5 lines
Now that the contexts lock is a read/write lock, it should not be locked here
in ast_hint_state_changed(). This makes it get locked recursively which now
causes a deadlock.
(closes issue #11080, thanks to callguy for the access to a deadlocked machine)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94832 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94830 f38db490-d61c-443f-a65b-d21fe96a405b
- Doxygen fixes
Note:
- This application needs better documentation and a RESULT code in the
dialplan.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94826 f38db490-d61c-443f-a65b-d21fe96a405b
SDL is also detected at runtime).
Now we should be able to stream video even without a rendering device
(useful for remote monitoring).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94822 f38db490-d61c-443f-a65b-d21fe96a405b
rizzo