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2009-04-10Modify headers and macros, according to Russell's suggestions on the -dev listTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03audio_audiohook_write_list() did not correctly update sample size after ↵David Vossel
ast_translate. audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-31Merged revisions 185196 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02Fix issue where changing the volume of both directions of audio did not work.Joshua Colp
(closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06Always detach and destroy the whisper and barge audiohooks. Additionally ↵Joshua Colp
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
2008-12-19Get rid of an extra space.Mark Michelson
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
2008-12-19Adding a new dialplan function AUDIOHOOK_INHERITMark Michelson
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
2008-10-14Merged revisions 149204 via svnmerge from Mark Michelson
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
2008-08-10Another big chunk of changes from the RSW branch. Bunch of stuff from main/Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-05make datastore creation and destruction a generic API since it is not really ↵Kevin P. Fleming
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
2008-07-14Merged revisions 130634 via svnmerge from Russell Bryant
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
2008-07-11Merged revisions 130236 via svnmerge from Mark Michelson
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
2008-07-11Merged revisions 130173 via svnmerge from Mark Michelson
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
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causeMichiel van Baak
- 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
2008-05-01Add two new dialplan functions from libspeex for applying audio gain control Brett Bryant
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
2008-04-08Merged revisions 113296 via svnmerge from Joshua Colp
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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21Merge over ast_audiohook_volume branch. This adds API calls for use by ↵Joshua Colp
developers to adjust the volume on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12Merged revisions 108083 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-20*mumble*Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-20file not found.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-20Minor test...Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13Remove a duplicate lock of the audiohook lock when destroying manipulateRussell Bryant
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
2008-01-11I am no longer Rockin'Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-11Testing something...Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-30Adding support for the "automixmonitor" dial and queue options.Mark Michelson
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
2007-11-21remove a bunch of useless #include "options.h"Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19another bunch of include removals (errno.h and asterisk/logger.h)Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16Start untangling header inclusion in a way that does not affectLuigi Rizzo
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
2007-11-08use %d and cast to int instead of %zd for size_t object,Luigi Rizzo
this helps portability. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08improve linked-list macros in two ways:Kevin P. Fleming
- 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
2007-09-06Fix memory issue that crept up with Russell's testing. It is *not* proper to ↵Joshua Colp
free the frame we get in ast_write. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-05Doxygen cleanups/fixes.Jason Parker
Closes issue #10654, patch by snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-21Minor tweak. Don't manipulate volume of the audio in the buffer if no audio ↵Joshua Colp
is actually there. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08Merge audiohooks branch into trunk. This is a new API for developers to ↵Joshua Colp
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
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