Commit Graph

80 Commits

Author SHA1 Message Date
Joshua Colp b2484d7db9 Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
(closes issue #14414)
Reported by: bluecrow76


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 15:59:17 +00:00
Mark Michelson e015e6f404 Get rid of an extra space.
I don't know how this crept back in when I had already
fixed it earlier



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 23:45:00 +00:00
Mark Michelson 9733b30ff0 Adding a new dialplan function AUDIOHOOK_INHERIT
This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel 
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor 
continue recording the call even after the transfer
has completed.

It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.

(closes issue #13538)
Reported by: mbit
Patches:
      13538.patch uploaded by putnopvut (license 60)
	  Tested by: putnopvut

Review: http://reviewboard.digium.com/r/102/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 22:26:16 +00:00
Mark Michelson 29a8fe20c8 Merged revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines

Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.

Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 23:04:44 +00:00
Sean Bright b69c8e6ab5 Another big chunk of changes from the RSW branch. Bunch of stuff from main/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-10 19:35:50 +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
Russell Bryant c4c3e2f875 Merged revisions 130634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines

Bump up the debug level for a message.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-14 10:39:23 +00:00
Mark Michelson 1a7806c836 Merged revisions 130236 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines

Remove redundant logic


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11 20:03:55 +00:00
Mark Michelson 48d39547ec Merged revisions 130173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines

Fix a typo in audiohook_read_frame_both.

While this change has not been proven to fix any
specific issue, it is incorrect and could cause
unforeseen problems.


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11 19:14:15 +00:00
Michiel van Baak f1e9371da8 - revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- make data member of the ast_frame struct a named union instead of a void

Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.

The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.

This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data

Thanks russellb and kpfleming for the feedback.

(closes issue #12674)
Reported by: mvanbaak


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22 16:29:54 +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
Joshua Colp dc8fe3910d Merged revisions 113296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines

If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute.
(closes issue #12296)
Reported by: jvandal

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-08 15:05:35 +00:00
Joshua Colp 30d85b3144 Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21 17:58:59 +00:00
Joshua Colp 5fc569f5f5 Merged revisions 108083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines

Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 18:29:33 +00:00
Joshua Colp b0be65f2ef *mumble*
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-20 03:52:57 +00:00
Joshua Colp ddf7a8a2a0 file not found.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-20 03:44:40 +00:00
Joshua Colp ef7cfaa2f8 Minor test...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-20 03:38:13 +00:00
Russell Bryant b7425090c8 Remove a duplicate lock of the audiohook lock when destroying manipulate
audiohooks.  This causes an error when we attempt to destroy the lock later
when freeing the audiohook.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13 00:10:00 +00:00
Joshua Colp b8efdb304b I am no longer Rockin'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-11 22:59:13 +00:00
Joshua Colp 225f268e88 Testing something...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-11 22:57:39 +00:00
Mark Michelson 6b08c442c7 Adding support for the "automixmonitor" dial and queue options.
This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.

This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.

Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to 
be committed.

(closes issue #10185, reported and patched by xmarksthespot)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-30 21:19:57 +00:00
Luigi Rizzo e0ff5fef5c remove a bunch of useless #include "options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:09:02 +00:00
Luigi Rizzo 9335ace850 another bunch of include removals (errno.h and asterisk/logger.h)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 19:09:03 +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
Luigi Rizzo 339d27ebe9 use %d and cast to int instead of %zd for size_t object,
this helps portability.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08 09:20:05 +00:00
Kevin P. Fleming edc78d6023 improve linked-list macros in two ways:
- 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08 05:28:47 +00:00
Joshua Colp a565584d05 Fix memory issue that crept up with Russell's testing. It is *not* proper to free the frame we get in ast_write.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-06 22:34:44 +00:00
Jason Parker d72ea80a00 Doxygen cleanups/fixes.
Closes issue #10654, patch by snuffy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-05 16:31:39 +00:00
Joshua Colp 937d83f7e4 Minor tweak. Don't manipulate volume of the audio in the buffer if no audio is actually there.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-21 15:51:49 +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