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: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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

(closes issue #16471)
Reported by: kjotte
Patches:
      M16471.diff uploaded by junky (license 177)
Tested by: kjotte, junky



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247770 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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: https://origsvn.digium.com/svn/asterisk/trunk@184677 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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: https://origsvn.digium.com/svn/asterisk/trunk@184628 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3