Commit Graph

11 Commits

Author SHA1 Message Date
Tilghman Lesher 642bec4d6f AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10 19:20:57 +00:00
Tilghman Lesher a866a75900 Merge str_substitution branch.
This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result.  No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29 18:53:01 +00:00
Russell Bryant 6f314f4d42 Fix various spelling and grammatical issues in documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02 02:50:33 +00:00
Russell Bryant 5b168ee34b 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: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01 21:10:07 +00:00
Kevin P. Fleming 7df8b8b848 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: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-05 16:56:11 +00:00
Jason Parker f7eb823a7a Fix a few places where frame data was used directly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22 17:10:53 +00:00
Russell Bryant ea3fb96b29 Re-introduce proper error handling that was removed in recent commits.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-13 17:42:17 +00:00
Claude Patry df1912cd4f since we unregister, that has not been properly registered, i standardized this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-10 03:04:25 +00:00
Joshua Colp fc120bf827 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: https://origsvn.digium.com/svn/asterisk/trunk@115328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-05 22:13:57 +00:00
Brett Bryant e8c3130292 Add "read" capability to new libspeex functions in func_speex.c.
func_speex.c is based on contributions from Switchvox.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-01 18:28:38 +00:00
Brett Bryant 5634048c98 Add two new dialplan functions from libspeex for applying audio gain control
and denoising to a channel, AGC() and DENOISE(). Also included, is a change 
to the audiohook API to add a new function (ast_audiohook_remove) that can 
remove an audiohook from a channel before it is detached.

This code is based on a contribution from Switchvox.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-01 16:57:19 +00:00