Commit Graph

21 Commits

Author SHA1 Message Date
Naveen Albert 51d262af12 xmldocs: Improve examples.
Use example tags instead of regular para tags
where possible.

ASTERISK-30090

Change-Id: Iada8bbfda08f30b118cedf2d040bbb21e4966ec5
2022-06-09 03:47:41 -05:00
Naveen Albert 19b5097d87 func_volume: Add read capability to function.
Up until now, the VOLUME function has been write
only, so that TX/RX values can be set but not
read afterwards. Now, previously set TX/RX values
can be read later.

ASTERISK-29439

Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f
2021-05-26 11:19:00 -05:00
Jean Aunis de66713fd5 func_volume: Accept decimal number as argument
Allow voice volume to be multiplied or divided by a floating point number.

ASTERISK-28813

Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c
2020-04-14 09:28:05 -05:00
Sean Bright fd0ca1c3f9 Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
2017-12-22 09:23:22 -05:00
Corey Farrell a6e5bae3ef Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
2016-10-27 09:53:55 -04:00
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00
Corey Farrell fbe0dfaf44 Fix dialplan function NULL channel safety issues
(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/
........

Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-27 19:21:44 +00:00
Sean Bright 0877ba5b37 Minor spelling fix to the VOLUME documentation.
........

Merged revisions 376919 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 376920 from http://svn.asterisk.org/svn/asterisk/branches/10
........

Merged revisions 376921 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-30 17:08:41 +00:00
Mark Michelson 21997aa7bb Fix a deadlock that occurs when func_volume is used on a local channel.
This was discovered by trying to perform a call forward to an extension
that makes use of func_volume. When the local channel is optimized away,
the datastore on the local;2 channel would have its audiohook destroyed
rather than detaching the audiohook from the channel and then destroying
it.

With this patch, func_volume's datastore destructor takes the proper
route of detaching the audiohook and then destroying it.

(closes issue ASTERISK-19611)
reported by Volker Sauer
Patches:
	ASTERISK-19611.patch uploaded by Mark Michelson (license #5049)
........

Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 368899 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-13 21:17:13 +00:00
Mark Michelson b5f0647fc8 Fix Coverity-reported ARRAY_VS_SINGLETON error.
As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none
took arguments. The proper thing to do for this case is to
pass NULL for the "args" parameter here. We were instead passing
a seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume.

(closes issue ASTERISK-19656)
........

Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 364900 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01 23:11:22 +00:00
Leif Madsen a525edea59 Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  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
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14 20:28:54 +00:00
Jonathan Rose 1c1c9c2bd4 Merged revisions 310587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  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
  
  ........
    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/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-14 15:40:43 +00:00
David Vossel d760e81f37 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: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
Tilghman Lesher d8e0c58437 Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Olle Johansson 9b12df5731 By copying this code I got bad comments in reviewboard... Better fix the original.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-31 18:17:38 +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
Luigi Rizzo 9c2aaeb701 remove some unnecessary includes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 20:42:06 +00:00
Luigi Rizzo fdb7f7ba3d 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: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 20:04:58 +00:00
Russell Bryant 040a5f20f9 * 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: https://origsvn.digium.com/svn/asterisk/trunk@81260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-28 18:32:56 +00:00
Joshua Colp 602198c402 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: https://origsvn.digium.com/svn/asterisk/trunk@78649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 19:30:52 +00:00