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27 Commits

Author SHA1 Message Date
Alexander Traud
876600ce6e codec_resample: Increase buffer for Opus Codec with FEC. 
ASTERISK-25599 #close

Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e
 2015-12-08 08:43:05 -06:00
Alexander Traud
b76c196e13 codec_resample: Increase buffer for Opus Codec. 
ASTERISK-25599 #close

Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10
 2015-12-01 07:59:19 -06:00
Matthew Jordan
a2c912e997 media formats: re-architect handling of media for performance improvements 
In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
 1. Asterisk was limited in how many formats it could handle.
 2. Formats, being a bit field, could not include any attribute information.
    A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.

Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.

Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.

Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.

Generally, the new philosophy for handling formats is as follows:
 * The ast_format structure is reference counted. This removed a large amount
   of the memory allocations and copying that was done in prior versions.
 * In order to prevent race conditions while keeping things performant, the
   ast_format structure is immutable by convention and lock-free. Violate this
   tenet at your peril!
 * Because formats are reference counted, codecs are also reference counted.
   The Asterisk core generally provides built-in codecs and caches the
   ast_format structures created to represent them. Generally, to prevent
   inordinate amounts of module reference bumping, codecs and formats can be
   added at run-time but cannot be removed.
 * All compatibility with the bit field representation of codecs/formats has
   been moved to a compatibility API. The primary user of this representation
   is chan_iax2, which must continue to maintain its bit-field usage of formats
   for interoperability concerns.
 * When a format is negotiated with attributes, or when a format cannot be
   represented by one of the cached formats, a new format object is created or
   cloned from an existing format. That format may have the same codec
   underlying it, but is a different format than a version of the format with
   different attributes or without attributes.
 * While formats are reference counted objects, the reference count maintained
   on the format should be manipulated with care. Formats are generally cached
   and will persist for the lifetime of Asterisk and do not explicitly need
   to have their lifetime modified. An exception to this is when the user of a
   format does not know where the format came from *and* the user may outlive
   the provider of the format. This occurs, for example, when a format is read
   from a channel: the channel may have a format with attributes (hence,
   non-cached) and the user of the format may last longer than the channel (if
   the reference to the channel is released prior to the format's reference).

For more information on this work, see the API design notes:
  https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite

Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.

There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).

Reviews:
 https://reviewboard.asterisk.org/r/3814
 https://reviewboard.asterisk.org/r/3808
 https://reviewboard.asterisk.org/r/3805
 https://reviewboard.asterisk.org/r/3803
 https://reviewboard.asterisk.org/r/3801
 https://reviewboard.asterisk.org/r/3798
 https://reviewboard.asterisk.org/r/3800
 https://reviewboard.asterisk.org/r/3794
 https://reviewboard.asterisk.org/r/3793
 https://reviewboard.asterisk.org/r/3792
 https://reviewboard.asterisk.org/r/3791
 https://reviewboard.asterisk.org/r/3790
 https://reviewboard.asterisk.org/r/3789
 https://reviewboard.asterisk.org/r/3788
 https://reviewboard.asterisk.org/r/3787
 https://reviewboard.asterisk.org/r/3786
 https://reviewboard.asterisk.org/r/3784
 https://reviewboard.asterisk.org/r/3783
 https://reviewboard.asterisk.org/r/3778
 https://reviewboard.asterisk.org/r/3774
 https://reviewboard.asterisk.org/r/3775
 https://reviewboard.asterisk.org/r/3772
 https://reviewboard.asterisk.org/r/3761
 https://reviewboard.asterisk.org/r/3754
 https://reviewboard.asterisk.org/r/3753
 https://reviewboard.asterisk.org/r/3751
 https://reviewboard.asterisk.org/r/3750
 https://reviewboard.asterisk.org/r/3748
 https://reviewboard.asterisk.org/r/3747
 https://reviewboard.asterisk.org/r/3746
 https://reviewboard.asterisk.org/r/3742
 https://reviewboard.asterisk.org/r/3740
 https://reviewboard.asterisk.org/r/3739
 https://reviewboard.asterisk.org/r/3738
 https://reviewboard.asterisk.org/r/3737
 https://reviewboard.asterisk.org/r/3736
 https://reviewboard.asterisk.org/r/3734
 https://reviewboard.asterisk.org/r/3722
 https://reviewboard.asterisk.org/r/3713
 https://reviewboard.asterisk.org/r/3703
 https://reviewboard.asterisk.org/r/3689
 https://reviewboard.asterisk.org/r/3687
 https://reviewboard.asterisk.org/r/3674
 https://reviewboard.asterisk.org/r/3671
 https://reviewboard.asterisk.org/r/3667
 https://reviewboard.asterisk.org/r/3665
 https://reviewboard.asterisk.org/r/3625
 https://reviewboard.asterisk.org/r/3602
 https://reviewboard.asterisk.org/r/3519
 https://reviewboard.asterisk.org/r/3518
 https://reviewboard.asterisk.org/r/3516
 https://reviewboard.asterisk.org/r/3515
 https://reviewboard.asterisk.org/r/3512
 https://reviewboard.asterisk.org/r/3506
 https://reviewboard.asterisk.org/r/3413
 https://reviewboard.asterisk.org/r/3410
 https://reviewboard.asterisk.org/r/3387
 https://reviewboard.asterisk.org/r/3388
 https://reviewboard.asterisk.org/r/3389
 https://reviewboard.asterisk.org/r/3390
 https://reviewboard.asterisk.org/r/3321
 https://reviewboard.asterisk.org/r/3320
 https://reviewboard.asterisk.org/r/3319
 https://reviewboard.asterisk.org/r/3318
 https://reviewboard.asterisk.org/r/3266
 https://reviewboard.asterisk.org/r/3265
 https://reviewboard.asterisk.org/r/3234
 https://reviewboard.asterisk.org/r/3178

ASTERISK-23114 #close
Reported by: mjordan
  media_formats_translation_core.diff uploaded by kharwell (License 6464)
  rb3506.diff uploaded by mjordan (License 6283)
  media_format_app_file.diff uploaded by kharwell (License 6464) 
  misc-2.diff uploaded by file (License 5000)
  chan_mild-3.diff uploaded by file (License 5000) 
  chan_obscure.diff uploaded by file (License 5000) 
  jingle.diff uploaded by file (License 5000) 
  funcs.diff uploaded by file (License 5000) 
  formats.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  bridges.diff uploaded by file (License 5000) 
  mf-codecs-2.diff uploaded by file (License 5000) 
  mf-app_fax.diff uploaded by file (License 5000) 
  mf-apps-3.diff uploaded by file (License 5000) 
  media-formats-3.diff uploaded by file (License 5000) 

ASTERISK-23715
  rb3713.patch uploaded by coreyfarrell (License 5909)
  rb3689.patch uploaded by mjordan (License 6283)
  
ASTERISK-23957
  rb3722.patch uploaded by mjordan (License 6283) 
  mf-attributes-3.diff uploaded by file (License 5000) 

ASTERISK-23958
Tested by: jrose
  rb3822.patch uploaded by coreyfarrell (License 5909) 
  rb3800.patch uploaded by jrose (License 6182)
  chan_sip.diff uploaded by mjordan (License 6283) 
  rb3747.patch uploaded by jrose (License 6182)

ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
  sip_cleanup.diff uploaded by opticron (License 6273)
  chan_sip_caps.diff uploaded by mjordan (License 6283) 
  rb3751.patch uploaded by coreyfarrell (License 5909) 
  chan_sip-3.diff uploaded by file (License 5000) 

ASTERISK-23960 #close
Tested by: opticron
  direct_media.diff uploaded by opticron (License 6273) 
  pjsip-direct-media.diff uploaded by file (License 5000) 
  format_cap_remove.diff uploaded by opticron (License 6273) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  chan_pjsip-2.diff uploaded by file (License 5000) 

ASTERISK-23966 #close
Tested by: rmudgett
  rb3803.patch uploaded by rmudgetti (License 5621)
  chan_dahdi.diff uploaded by file (License 5000) 
  
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
  rb3814.patch uploaded by rmudgett (License 5621) 
  moh_cleanup.diff uploaded by opticron (License 6273) 
  bridge_leak.diff uploaded by opticron (License 6273) 
  translate.diff uploaded by file (License 5000) 
  rb3795.patch uploaded by rmudgett (License 5621) 
  tls_fix.diff uploaded by mjordan (License 6283) 
  fax-mf-fix-2.diff uploaded by file (License 5000) 
  rtp_transfer_stuff uploaded by mjordan (License 6283) 
  rb3787.patch uploaded by rmudgett (License 5621) 
  media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) 
  format_cache_case_fix.diff uploaded by opticron (License 6273) 
  rb3774.patch uploaded by rmudgett (License 5621) 
  rb3775.patch uploaded by rmudgett (License 5621) 
  rtp_engine_fix.diff uploaded by opticron (License 6273) 
  rtp_crash_fix.diff uploaded by opticron (License 6273) 
  rb3753.patch uploaded by mjordan (License 6283) 
  rb3750.patch uploaded by mjordan (License 6283) 
  rb3748.patch uploaded by rmudgett (License 5621) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  rb3740.patch uploaded by mjordan (License 6283) 
  rb3739.patch uploaded by mjordan (License 6283) 
  rb3734.patch uploaded by mjordan (License 6283) 
  rb3689.patch uploaded by mjordan (License 6283) 
  rb3674.patch uploaded by coreyfarrell (License 5909) 
  rb3671.patch uploaded by coreyfarrell (License 5909) 
  rb3667.patch uploaded by coreyfarrell (License 5909) 
  rb3665.patch uploaded by mjordan (License 6283) 
  rb3625.patch uploaded by coreyfarrell (License 5909) 
  rb3602.patch uploaded by coreyfarrell (License 5909) 
  format_compatibility-2.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-07-20 22:06:33 +00:00
Kinsey Moore
510b3b3594 Allow codec_resample to be unloaded 
Ensure that trans_size is correct to prevent uninitialized entries from
preventing reload.

(closes issue ASTERISK-21401)
Reported by: Corey Farrell
Tested by: Corey Farrell
Patches:
    codec_resample-unload.patch uploaded by Corey Farrell
........

Merged revisions 385582 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2013-04-12 22:26:17 +00:00
Walter Doekes
fc63e07135 Avoid cppcheck warnings; removing unused vars and a bit of cleanup. 
Patch by: junky
Review: https://reviewboard.asterisk.org/r/1743/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2012-04-17 18:57:40 +00:00
David Vossel
80a4611fd7 Merged revisions 330940 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011) | 2 lines
  
  The slin resampler is no longer dependent on an external library, but the dependency was not removed correctly.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2011-08-05 15:57:06 +00:00
Leif Madsen
a525edea59 Merged revisions 328247 via svnmerge from 
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
 2011-07-14 20:28:54 +00:00
David Vossel
e3222d8111 Fixes error with frame datalen being calculated from samples when this is not allwaya accurate. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2011-04-20 16:37:15 +00:00
David Vossel
b1f4589536 Remove libresample dependency from codec_resample.c 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2011-03-18 16:27:23 +00:00
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
 2011-02-22 23:04:49 +00:00
David Vossel
c26c190711 Asterisk media architecture conversion - no more format bitfields 
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
 2011-02-03 16:22:10 +00:00
Jason Parker
ae0a736353 Merge codec_consistency branch. This should make sample usage much happier. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-10-17 21:35:23 +00:00
Russell Bryant
c0c743a5fa Update instructions for getting libresample 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-09-02 15:11:53 +00:00
Russell Bryant
c87f901cfd Remove libresample from the Asterisk source tree. It is now available in its 
own repository, and must be installed like any other library for Asterisk to
use.  The two modules that require it are codec_resample and app_jack.

To install libresample:

$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install

This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-07-21 14:47:41 +00:00
Russell Bryant
5de127e103 Enable higher quality resampling, as it doesn't have a noticeable performance 
impact on my machine ..


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-07-21 13:51:05 +00:00
Brett Bryant
5b7933fe5e Janitor patch to change uses of sizeof to ARRAY_LEN 
(closes issue #13054)
Reported by: pabelanger
Patches:
      ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-07-11 18:09:35 +00:00
Tilghman Lesher
7b84cf6fa6 Convert casts to unions, to fix alignment issues on Solaris 
(closes issue #12932)
 Reported by: snuffy
 Patches: 
       bug_12932_20080627.diff uploaded by snuffy (license 35)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-06-26 17:06:17 +00:00
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
 2008-05-22 16:29:54 +00:00
Russell Bryant
7bc92eda05 Kevin noted that the thing that I _actually_ changed here was that I converted 
a value from a double, to a float, back to a double.  Sure enough, when I changed
my interim variable back to a double, it still blows up.  Switching all of these
to a float fixes the problem.  This seems like a compiler bug where a double passed
as an argument isn't getting properly aligned, so I'll have to see if I can replicate
it with a small test program.

(related to issue #11725)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-11 19:05:24 +00:00
Russell Bryant
d1256bb8a6 Fix a bus error that happened when asterisk was built with optimizations on 
with platforms that explode on unaligned access.  I'm not exactly sure why
this fixes it, but it fixed it on the machine I was testing on.  If it makes
sense to you, feel free to enlighten me.  :)

(closes issue #11725, patched by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-11 18:48:07 +00:00
Russell Bryant
0a5bfb9eb8 Fix the buffer_samples value. For signed linear, the number of samples needed 
to fill the buffer is half the buffer size.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-10 23:10:00 +00:00
Kevin P. Fleming
887103e83b and now just to keep the libresample party going... if the functions from libresample are going to be in the main Asterisk binary, it makes sense for the header that defines them to be available without any special CFLAGS and to out-of-tree modules building against /usr/include/asterisk 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-02 18:21:04 +00:00
Kevin P. Fleming
04a10c145b go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-02 14:05:30 +00:00
Russell Bryant
78f4b28552 Instead of linking libresample into the main Asterisk binary, build it as 
res_resample, and mark codec_resample as dependent upon res_resample.  This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places.  (I have another module
in a branch that needs it, too.)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-02 01:00:44 +00:00
Luigi Rizzo
e637f2c038 make codec_resample build on __CYGWIN__, and make it load on FreeBSD 
(and probably other systems as well).
Both need libresample.a to be specified in the linking phase,
and cygwin needs <float.h> as other BSD.

The checks for OS-specific headers should really be moved to some
common header though.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-01 22:21:39 +00:00
Russell Bryant
e0dfa91db8 Use float.h to fix the build on FreeBSD. Also, add some other platforms as 
they are likely the same.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-31 22:41:39 +00:00
Russell Bryant
21cb767db7 Merge changes from team/russell/codec_resample 
This commit imports libresample for use in Asterisk.  It also adds a new codec
module, codec_resample.  This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.

It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz 
signed linear.  But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-31 21:22:31 +00:00