asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	573b7a2d18	Resolve FORWARD_NULL static analysis warnings This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@366168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Leif Madsen	7caa2349af	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/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	7d179abfd4	Merged revisions 326411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 \| tilghman \| 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) \| 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
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	14 years ago
Olle Johansson	8021cf48d8	Adding some clarifications to func_speex doxygen docs. The functions needed doesn't exist in Speex 1.05 which is what a lot of distros use. 1.2 seems to have been in beta status for years, and does include the sexy functions needed for func_speex to work. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	77031501a5	Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 \| tilghman \| 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) \| 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a866a75900	Merge str_substitution branch. This branch adds additional methods to dialplan functions, whereby the result buffers are now dynamic buffers, which can be expanded to the size of any result. No longer are variable substitutions limited to 4095 bytes of data. In addition, the common case of needing buffers much smaller than that will enable substitution to only take up the amount of memory actually needed. The existing variable substitution routines are still available, but users of those API calls should transition to using the dynamic-buffer APIs. Reviewboard: http://reviewboard.digium.com/r/174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	6f314f4d42	Fix various spelling and grammatical issues in documentation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
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	17 years ago
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	17 years ago
Jason Parker	f7eb823a7a	Fix a few places where frame data was used directly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ea3fb96b29	Re-introduce proper error handling that was removed in recent commits. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Claude Patry	df1912cd4f	since we unregister, that has not been properly registered, i standardized this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	fc120bf827	Merged revisions 115327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115327 \| file \| 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) \| 2 lines Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	e8c3130292	Add "read" capability to new libspeex functions in func_speex.c. func_speex.c is based on contributions from Switchvox. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114977 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

17 Commits (10)