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2011-07-14Merged revisions 328247 via svnmerge from lmadsen4-0/+16
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: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-11Updates follow_talker video_mode in confbridge application.dvossel1-5/+28
follow_talker mode originally echoed the same video stream to all participants. As the primary talker switched around, the video stream would result in the talker seeing themselves. Now the primary talker sees the last person who was talking rather than themselves. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327640 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Video support for ConfBridge.dvossel1-7/+69
Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325931 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Fixes reliability issues with func_jitterbuffer's usage in the new ↵dvossel1-3/+21
ConfBridge application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317197 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-21New HD ConfBridge conferencing application.dvossel2-155/+563
Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314598 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵dvossel2-34/+158
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: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Asterisk media architecture conversion - no more format bitfieldsdvossel3-14/+35
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: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18fixes confbridge crash when no timing module is loaded.dvossel1-0/+3
(closes issue #16471) Reported by: kjotte Patches: M16471.diff uploaded by junky (license 177) Tested by: kjotte, junky git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247770 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman1-4/+4
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.russell1-1/+1
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Improve timing interface to remember which provider provided a timerkpfleming1-9/+8
The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184762 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Fix a potential timer leak in bridge_softmix.file1-2/+11
It is possible for a bridge to be created without actually being used. In that scenario a timing file descriptor would be opened and not closed. To fix this the timing file descriptor is now closed in the destroy callback, not the thread function. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184677 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Remove a cast that is not needed.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184639 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Fix a potential race condition when creating a software based mixing bridge.file1-7/+2
It was possible for no timer to become available between creating the bridge and starting it. We now open a timer when creating it and keep it open until the bridge is destroyed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184628 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merge phase 1 support for the new bridging architecture.file5-0/+1087
This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180369 f38db490-d61c-443f-a65b-d21fe96a405b
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