When a format has no pre-recorded sound files, Asterisk has to transcode between
formats. For this, Asterisk has a fixed translation table. If the pre-recorded
sound files are not available in the same sample rate, Asterisk has not only to
transcode but also to resample.

Asterisk has pre-recorded files for SLN (8000 kHz) and SLN16 (16000 kHz).
However before this change, Asterisk did not take the sample rate into account,
because the translation paths to SLN and SLN16 got the same score/weight in the
table. Consequently, you might have got narrow-band audio with siren14, speex32,
silk24, and silk12 although those are (ultra) wide-band audio codecs.

With this change, the distance in sample-rates is taken into account. Now on the
Command-Line interface (CLI) 'core show channels', you should see:
(slin@16000)->(slin@32000)->(speex@32000).

ASTERISK-23735
Reported by: Richard Kenner

Change-Id: I9448295c1978be26f8633b6066395e7bbbe2e213

ASTERISK-24662

Change-Id: I3822956984292c99c48bca8e97807e498ccc0e88

Interpolated frames are frames which contain a number of
samples but have no actual data. Audiohooks did not
handle this case when translating an incoming frame into
signed linear. It assumed that a frame would always contain
media when it may not. If this occurs audiohooks will now
immediately return and not act on the frame.

As well for users of ast_trans_frameout the function has
been changed to be a bit more sane and ensure that the data
pointer on a frame is set to NULL if no data is actually
on the frame. This allows the various spots in Asterisk that
check for an interpolated frame based on the presence of a
data pointer to work as expected.

ASTERISK-26926

Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b

This change adds experimental support for providing RTCP
feedback information to codec modules so they can dynamically
change themselves based on conditions.

ASTERISK-26584

Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857

ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966

This change introduced some fax test failures
that have not yet been addressed. So this is
not forgotten I'm submitting a change which
reverts it.

This reverts:
d56fc3b3.

ASTERISK-25629

Change-Id: Ibc2f23c38643f5a2c89cf8915ae2d805b81bc3d5

ASTERISK-25629 #close

Change-Id: I66c0086e6c17764b8141ec60a3e2aaefe088eb78

If the destination format's name differed from the codec name then the
translator's explict_dst field would be improperly set. In some circumstances
it would end up setting it to a newly created format that has the same name
as the codec when it actually needed to be the given destination codec.

This could cause the translation path to use the wrong format. For instance,
if an endpoint had specified 'myulaw' as a format the translator could end up
using a 'ulaw' format (with whatever/default settings) instead. If the format
attribute settings differed between the two then there may unexpected results
during processing.

This patch removes the name check when building the translation path. This
should make it always set the translator's explicit_dst to the given destination
format as long as the sample rate and types match.

Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5

This reverts commit 5bfef2a8 as it
caused fax test failures.

ASTERISK-25629

Change-Id: I79de974dc4f63a1cafe0d2509169fd9a6b3cbaf4

ASTERISK-25629 #close

Change-Id: Ibfcf0670e094e9718d82fd9920f1fb2dae122006

ASTERISK-25616 #close

Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319

Previously, a trancoding module did not have access to the joint but cached
format. Therefore, the module did not have access to the attributes negotiated
via SDP (line fmtp). Now, a translation module receives the joint format.

ASTERISK-25545 #close

Change-Id: Id6878a989b50573298dab115d3371ea369e1a718

When Asterisk translates between codecs, each with a different frame size (for
example between iLBC 30 and Speex-WB), too large frames were created by
ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame
length, creating several frames when necessary. Affects all transcoding modules
which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex.

ASTERISK-25353 #close

Change-Id: I2e229569d73191d66a4e43fef35432db24000212

Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e

Given a source capability of h264 and ulaw, a destination capability of
h264 and g722 then ast_translator_best_choice() would pick h264 as the
best choice even though h264 is a video codec and Asterisk only supports
translation of audio codecs.  When the audio starts flowing, there are
warnings about a codec mismatch when the channel tries to write a frame to
the peer.

* Made ast_translator_best_choice() only select audio codecs.

* Restore a check in channel.c:set_format() lost after v1.8 to prevent
trying to set a non-audio codec.

This is an intermediate patch for a series of patches aimed at improving
translation path choices for ASTERISK-24841.

This patch is a complete enough fix for ASTERISK-21777 as the v11 version
of ast_translator_best_choice() does the same thing.  However, chan_sip.c
still somehow tries to call ast_codec_choose() which then calls
ast_best_codec() with a capability set that doesn't contain any audio
formats for the incoming call.  The remaining warning message seems to be
a benign transient.

ASTERISK-21777 #close
Reported by: Nick Ruggles

ASTERISK-24380 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/4605/
........

Merged revisions 434614 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 434615 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434616 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a 'core restart now' or 'core stop now' is executed and a channel is
currently in a media operation, the translator matrix can be destroyed while a
channel is currently blocked on getting the best translation choice
(see ast_translator_best_choice). When the channel gets the mutex, the
translation matrix now has invalid memory, and Asterisk crashes.

This patch does two things:
(1) We now only clean up the translation matrix on a graceful shutdown. In that
    case, there are no channels, and so there is no risk of this occurring.
(2) We also now set the __matrix and __indextable to NULL. In some initial
    backtraces when this occurred, it looked as if there was a memory corruption
    occurring, and it wasn't until we determined that something had restarted
    Asterisk that the issue became clear. By setting these to NULL on shutdown,
    it becomes a bit easier to determine why a crash is occurring.

Note that we could litter the code with NULL checks on the __matrix, but the
act of making the translation matrix cleaned up on shutdown should preclude
this issue from occurring in the first place, and this part of the code needs
to be as fast as possible.

Review: https://reviewboard.asterisk.org/r/4457/
........

Merged revisions 432453 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432455 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix the AMI Status action read and write translation path strings from
growing for each channel in the status event list by reseting the ast
string given to ast_translate_path_to_str() to fill in the given
translation path.
........

Merged revisions 426079 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426080 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Empty here means that there are no formats in the format_cap structure
or the only format in it is the "none" format.

I've added calls to check the emptiness of a format_cap in a few places
in order to short-circuit operations that would otherwise be pointless
as well as to prevent some assertions from being triggered in cases
where channels with no formats are used.
........

Merged revisions 423414 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423415 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
........

Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.

This change moves code around a bit so that the frame is now
freed after it has been completely used.

(closes issue ASTERISK-22788)
Reported by: Corey Farrell
Patches:
	translate-access-after-free-11up.patch uploaded by coreyfarrell (license 5909)
	translate-access-after-free-1.8.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 403014 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 403015 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 403016 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:

* channel.c: When copying variables from a parent channel to a child channel,
  specify the channels involved. Do not log anything for a variable that is not
  inherited; the fact that it doesn't have an _ or __ already signifies that it
  won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
  to use these debug messages, and for each format that is registered (on
  startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
  For short tests in the Asterisk Test Suite, this should make finding the
  actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
  Often, description elements - which are not required - are not provided.
  This debug message adds no additional value, as it is not indicative of an
  error or helpful in debugging which element did not contain a 'blah' element
  as a child. If an element is supposed to contain a child element, then that
  XML tree should have failed validation in the first place.

Review: https://reviewboard.asterisk.org/r/2966/
........

Merged revisions 402150 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 402151 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402154 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402155 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21170)
Reported by: NITESH BANSAL
Patches:
    dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
Review: https://reviewboard.asterisk.org/r/2938/
........

Merged revisions 401619 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401620 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 401621 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Consistently compare format2index() return value so matrix_get() cannot
get passed negative values.

* Optimize ast_translator_best_choice() to defer initializing things until
needed.  Also cached the matrix_get() return value rather than repeatedly
calling it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401039 65c4cc65-6c06-0410-ace0-fbb531ad65f3

main/config.c - cleanup cache fie includes
res/res_security_log.c - unregister logger level
channesl/chan_sip.c - cleanup io context and notify_types
main/translator.c - cleanup at shutdown
main/named_acl.c - cleanup cli commands
main/indications.c - ast_get_indication_tone() unref default_tone_zone if used

(closes issues ASTERISK-22378)
Reported by: Corey Farrell
Patches:
     config_shutdown.patch uploaded by coreyfarrell (license 5909)
     res_security_log.patch uploaded by coreyfarrell (license 5909)
     chan_sip-11.patch uploaded by coreyfarrell (license 5909)
     indications_refleak.patch uploaded by coreyfarrell (license 5909)
     named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license 5909)
     translate_shutdown.patch uploaded by coreyfarrell (license 5909)

........

Merged revisions 398102 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 398103 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 398116 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398124 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This support includes codecs such as H.263, H.264, SILK, and CELT. You are able to set up a call and have attribute information pass. This should help considerably with video calls.

Review: https://reviewboard.asterisk.org/r/2005/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370055 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
........
  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
........

Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Everything still compiled after making these changes, so I assume these
whitespace-only changes didn't break anything (and shouldn't have).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308624 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

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
  
  improved translation paths for wideband codecs
  
  The problem I'm addressing is that Asterisk's current
  method of building the least cost translation paths
  between codecs does not take into account sample rate.
  For instance, it was possible for siren14 (a 32khz codec),
  to contain the a translation path to siren7 (a 16khz
  audio codec) that goes through slin at 8khz.  In this
  case Asterisk takes a 32khz codec, down samples it to
  8khz and then up samples it to 16khz which is terrible
  regardless if it is computationally less expensive.  This
  patch now builds translation paths that give priority to
  maintaining the best possible sample rate before taking
  into consideration computational cost.  This patch also
  adds cli commands to expose what translation paths are
  actually being used.
  
  Changes:
  1. Translation paths will never contain a step that changes
  the sample rate unless absolutely necessary.
  2. When choosing the best codec to make two channels compatible.
  Shared codecs with the highest sample rate are given priority.
  3. A new cli command to show all translation paths available
  for a specific codec 'core show translation paths [codec name]'
  has been added.
  4. 'core show translation' which displays the translation
  matrix now includes the new higher bit audio codecs in the table.
  5. 'core show channel [channel name]'  now displays the
  translation paths if translation is used.
  
  (closes issue #16841)
  Reported by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/842/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17092)
Reported by: moy
Patches:
      translate.rev254273.patch uploaded by moy (license 222)
Tested by: moy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The logic for handling generic PLC is now handled in ast_write in
channel.c instead of in translation code.

Review: https://reviewboard.asterisk.org/r/683/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267492 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16395)
 Reported by: jtodd


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233239 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232017 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

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) | 2 lines
  
  Revert 225169, as this doesn't account for the possibility of a list of frames.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225172 65c4cc65-6c06-0410-ace0-fbb531ad65f3