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11 Commits

Author SHA1 Message Date
lmadsen e73cab2f3f Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

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  r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
  
  Merged revisions 328209 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
    
    Introduce <support_level> tags in MODULEINFO.
    This change introduces MODULEINFO into many modules in Asterisk in order to show
    the community support level for those modules. This is used by changes committed
    to menuselect by Russell Bryant recently (r917 in menuselect). More information about
    the support level types and what they mean is available on the wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-14 20:28:54 +00:00
jrose 6a23a123b5 Merged revisions 310587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r310587 | jrose | 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) | 15 lines
  
  Merged revisions 310585 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines
    
    Adds 'p' as an option to func_volume.  When it is on, the old behavior with DTMF controlling volume adjustment will be enforced.
    When it is off, DTMF will not be processed by the function.
    
    Programmed by Jonathan Rose
    Reviewed by David Vossel, Leif Madsen, and Russell Bryant
    
    http://reviewboard.digium.internal/r/93/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310588 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-14 15:40:43 +00:00
dvossel f27e928f05 Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22 23:04:49 +00:00
tilghman 3bacd4082e Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04 14:05:12 +00:00
oej d000fabce0 By copying this code I got bad comments in reviewboard... Better fix the original.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215023 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-31 18:17:38 +00:00
russell b1f91b97d2 Merge changes from team/group/appdocsxml
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01 21:10:07 +00:00
kpfleming 0891b8a53c make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05 16:56:11 +00:00
rizzo 5e75696a42 remove some unnecessary includes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89430 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 20:42:06 +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
russell ac52aa8cd6 * Constify the uid field of channel datastores
* Convert some spaces to tabs in func_volume
* Add a note in channel.h making it clear that none of the datastore API calls
  lock the channel they are given, so the channel should be locked before
  calling the functions that take a channel argument.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81260 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-28 18:32:56 +00:00
file ce30d7306b Merge audiohooks branch into trunk. This is a new API for developers to listen and manipulate the audio going through a channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78649 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08 19:30:52 +00:00