asterisk

Commit Graph

Author	SHA1	Message	Date
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	17 years ago
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	17 years ago
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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
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	17 years ago
Joshua Colp	5fc569f5f5	Merged revisions 108083 via svnmerge from 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	17 years ago
Joshua Colp	b0be65f2ef	mumble git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ddf7a8a2a0	file not found. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ef7cfaa2f8	Minor test... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
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	18 years ago
Joshua Colp	b8efdb304b	I am no longer Rockin' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	225f268e88	Testing something... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago
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	18 years ago

21 Commits (d32c2d6fd9d6970ad2cde1b29fefff445dc98fb1)