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path: root/apps/app_speech_utils.c
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2012-04-20Document Speech* apps hangup on failure and suggest TryExecTerry Wilson
The Speech API apps return -1 on failure, which will hang up the channel. This may not be desirable behavior for some, but it isn't something that can be changed without breaking people's dialplans or writing an option to all of the Speech apps that does what TryExec already does. This patch documents the hangup behavior of the apps, and suggests TryExec as the solution. (closes issue AST-813) ........ Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362816 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Finalize ast_channel opaquificationTerry Wilson
Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24Opaquification for ast_format structs in struct ast_channelTerry Wilson
Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20ast_channel opaquification of pointers and integral typesTerry Wilson
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24Opaquify channel stringfieldsTerry Wilson
Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14Merged revisions 328247 via svnmerge from Leif Madsen
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
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02Merged revisions 284610 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19Merged revisions 264334 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf tone during playback in speechbackground. (closes issue #16966) Reported by: asackheim ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01Swap reversed timevals.Mark Michelson
This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19Merged revisions 177383 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-06Move Speech* applications and functions documentation to XML.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-08Merged revisions 147517 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06Merged revisions 146799 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines Dialplan functions should not actually return 0, unless they have modified the workspace. To signal an error (and no change to the workspace), -1 should be returned instead. (closes issue #13340) Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146802 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
2008-05-22- 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
2008-05-01Modify TIMEOUT() to be accurate down to the millisecond.Tilghman Lesher
(closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30Lock around variables retrieved, and copy the values, if they stay persistent,Tilghman Lesher
since another thread could remove them. (closes issue #12541) Reported by: snuffy Patches: bug_12156_apps.diff uploaded by snuffy (license 35) Several additional changes by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-25Whitespace changes onlyTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07Merged revisions 106552 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines Safely use the strncat() function. (closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-30Add the 'n' option to SpeechBackground, which has the application not answer theRussell Bryant
channel if it has not already been answered. (closes SPD-51) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyLuigi Rizzo
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 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
2007-11-06"show application <foo>" changes for clarity.Mark Michelson
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-13Merged revisions 79334 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79334 | file | 2007-08-13 18:57:20 -0300 (Mon, 13 Aug 2007) | 2 lines Instead of accepting a single DTMF character accept a full string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-13Merged revisions 79207 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79207 | file | 2007-08-13 11:51:09 -0300 (Mon, 13 Aug 2007) | 2 lines Add an API call to allow the engine to know that DTMF was received. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵Tilghman Lesher
few other formatting cleanups, too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-25Merged revisions 77176 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines (closes issue #10303) Reported by: jtodd Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16Applications no longer need to call ast_module_user_add and ↵Joshua Colp
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16It is no longer required for each module that deals with a channel to call ↵Joshua Colp
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-11Use the linkedlists.h AST_LIST_NEXT macro for modifying the list of results.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-11Allow the native formats of a channel to influence the audio that is going ↵Joshua Colp
to the engine. The best format will try to be chosen with an ultimate fallback to signed linear if possible. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-22Merged revisions 71068 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri, 22 Jun 2007) | 12 lines Merged revisions 71065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 lines Fix a few silly usages of ast_playstream() - it only ever returns 0... Issue 10035 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-15Merged revisions 69558 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69558 | file | 2007-06-15 15:23:45 -0400 (Fri, 15 Jun 2007) | 2 lines Add support for setting the maximum length of acceptable DTMF in SpeechBackground. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Convert uses of strdup() to ast_strdup()Russell Bryant
(issue #9983, eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵Tilghman Lesher
guidelines changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13Merged revisions 61651 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr 2007) | 2 lines Do not bother looking for a result if none are present. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Merged revisions 60361 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2 lines Add support for returning different types of results (ie: NBest). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-03Merged revisions 59963 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59963 | file | 2007-04-03 15:40:59 -0400 (Tue, 03 Apr 2007) | 2 lines Don't clash when a person both speaks and uses DTMF. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26Merged revisions 59223 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59223 | file | 2007-03-26 16:34:14 -0300 (Mon, 26 Mar 2007) | 2 lines Add ability to specify no timeout. This means as soon as the prompt is done playing it moves on to the next priority. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26Merged revisions 59213 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59213 | file | 2007-03-26 14:13:06 -0400 (Mon, 26 Mar 2007) | 2 lines Make SpeechBackground obey the digit timeout value. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28Merged revisions 57053 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb 2007) | 2 lines Better handle timeouts when the individual speaks after everything has been played but before the timeout ends. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21Merged revisions 55947 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55947 | file | 2007-02-21 15:03:38 -0500 (Wed, 21 Feb 2007) | 2 lines Only start playing the next file if we have not been quieted. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16Merged revisions 54714 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54714 | file | 2007-02-15 19:48:48 -0500 (Thu, 15 Feb 2007) | 2 lines Don't let dtmf leak over into the engine and let it skew the results... also give DTMF results priority. (issue #9014 reported by surftek) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-08Merged revisions 53601 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb 2007) | 2 lines Fix timeout issue when utterance is longer then timeout itself. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-18Merged revisions 51251 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51251 | file | 2007-01-18 14:17:34 -0500 (Thu, 18 Jan 2007) | 2 lines Only start timeout once we reach the end of the files to play back. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-10Merged revisions 50433 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50433 | file | 2007-01-10 15:25:44 -0500 (Wed, 10 Jan 2007) | 2 lines Merge speech-multi branch which adds support for joining multiple sound files together to be played one after another in SpeechBackground. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-05const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way hereKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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