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Author SHA1 Message Date
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
abd3e4040b Allow Asterisk to compile under GCC 4.10
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09 22:49:26 +00:00
Andrew Latham
6c20cf2d8a Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-18 14:17:40 +00:00
Alec L Davis
90f8c90b10 dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END
Instead of a recompile, allow values to be adjusted in dsp.conf

For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons.

Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3

(closes issue ASTERISK-17493)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2144/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-04 20:21:36 +00:00
Alec L Davis
36405ae659 dsp.c fix incorrect DTMF Digit_Duration.
it's always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2

(issue ASTERISK-16003)
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2145/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-04 20:08:22 +00:00
Alec L Davis
4af961a03a dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries.

Various countries have different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies.

Power level difference between frequencies for different Administrations/RPOAs
 NTT        = Max. 5 dB
 AT&T       = 4dB(reverse) to 8dB(normal)
 Danish     = Max. 6 dB
 Australian = Max. 10 dB
 Brazilian  = Max. 9 dB
 ETSI       = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03)

Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB.
Default is AT&T specifications

Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31 
;dtmf_reverse_twist=2.51 
;relax_dtmf_normal_twist=6.31 
;relax_dtmf_reverse_twist=3.98 


(closes issue ASTERISK-20442)
Reported by: tbsky
Tested by: tbsky,alecdavis

alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2141/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-04 04:50:16 +00:00
Alec L Davis
368b4c6166 dsp.c: remove more whitespace mentioned in review2107
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-21 09:11:39 +00:00
Alec L Davis
d3a23be26e dsp.c ast_dsp_call_progress use local short variable in loop, plus other cleanup
janitor cleanup. No functional change.

1). ast_dsp_call_progress: use 'short samp' instead of s[x] inside loop.
    apply same casting as other _init, dsp->energy = (int32_t) samp * (int32_t) samp

2). ast_dtmf_detect_init: move repeated setting of s->energy to outside of loop.
    do goertzel_init loop first before setting s->lasthit and s->current_hit, consistant with ast_dsp_digitreset()

3). ast_mf_detect_init:
    do goertzel_init loop first before setting s->hits[] and s->current_hit, consistant with ast_dsp_digitreset()

4). Don't chain init different variables, as the type may change

Review https://reviewboard.asterisk.org/r/2107/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-21 06:51:25 +00:00
Alec L Davis
ed442248e5 dsp.c: remove whitespace mentioned in review2107
Related https://reviewboard.asterisk.org/r/2107/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20 10:41:30 +00:00
Alec L Davis
d39b417074 dsp.c: in ast_mf_detect_init incorrectly sets goertzel samples to 160, should be MF_GSIZE
Remove unused goertzel_state_t member 'samples'.

Related https://reviewboard.asterisk.org/r/2097/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 18:56:39 +00:00
Alec L Davis
7af245a403 dsp.c: Fix multiple issues when no-interdigit delay is present, and fast DTMF 50ms/50ms
Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary;
  1. reseting of hits=0, when no signal, only need to set it once.
  2. incrementing of hits, when the hit is the same as the current hit.
  3. setting of lasthit, when it's the same as before.

Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3

& 3 spelling mistakes

(closes issue ASTERISK-19610)
alecdavis (license 585)
Reported by: Jean-Philippe Lord
Tested by: alecdavis
 
Review: https://reviewboard.asterisk.org/r/2085/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 07:43:32 +00:00
Alec L Davis
2f9f9ecffb dsp.c: optimize goerztzel sample loops, in dtmf_detect, mf_detect and tone_detect
use a temporary short int when repeatedly used to call goertzel_sample.

alecdavis (license 585)
Reported by: alecdavis
Tested by: alecdavis
 
Review: https://reviewboard.asterisk.org/r/2093/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 06:52:30 +00:00
Alec L Davis
1295d551f9 mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-26 23:10:30 +00:00
Kinsey Moore
9b16c8b0f6 Clean up and ensure proper usage of alloca()
This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().

(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31 20:21:43 +00:00
Kevin P. Fleming
166b4e2b30 Multiple revisions 369001-369002
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  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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 16:20:16 +00:00
Kinsey Moore
c5b3db1956 Kill off red blobs in most of main/*
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
2012-03-22 19:51:16 +00:00
Terry Wilson
0e5c761c28 Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-01 22:09:18 +00:00
Terry Wilson
04da92c379 Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
Jonathan Rose
fd04da5114 Fix an issue where dsp.c would interpret multiple dtmf events from a single key press.
When receiving calls from a mobile phone into a DISA system on a connection with
significant interference, the reporter's Asterisk system would interpret DTMF incorrectly
and replicate digits received. This patch resolves that by increasing the number of
frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and
adjusts dtmf_detect function to reset hits and misses only when an edge is detected.

(closes issue ASTERISK-17493)
Reported by: Alec Davis
Patches:
	bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546)
Review: https://reviewboard.asterisk.org/r/1130/
........

Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 349729 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-05 22:02:33 +00:00
Kevin P. Fleming
fdda494776 Improve T.38 gateway V.21 preamble detection.
This commit removes the V.21 preamble detection code previously added to the
generic DSP implementation in Asterisk, and instead enhances the res_fax module
to be able to utilize V.21 preamble detection functionality made available by
FAX technology modules. This commit also adds such support to res_fax_spandsp,
which uses the Spandsp modem tone detection code to do the V.21 preamble
detection.

There should be no functional change here, other than much more reliable V.21
preamble detection (and thus T.38 gateway initiation).
........

Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-28 18:59:16 +00:00
Matthew Nicholson
849992fde9 tweak the v21 detector to detect an additional pattern of hits and misses
........

Merged revisions 342605 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-27 20:11:11 +00:00
Paul Belanger
2e2381341e Clean up dsp.conf parsing
Review: https://reviewboard.asterisk.org/r/1434/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-13 18:11:33 +00:00
Gregory Nietsky
3b1cc6de8d dsp_process was enhanced to work with alaw and ulaw in addition to slin.
noticed that some functions could be refactored here it is.

Reported by: irroot
Tested by: irroot, mnicholson
Review: https://reviewboard.asterisk.org/r/1304/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-25 14:07:01 +00:00
Matthew Nicholson
e46aea196c Merged revisions 328162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

........
  r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines
  
  tune the v21 preamble detector to properly detect the desired sequence of hits
  and misses
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14 17:47:40 +00:00
Matthew Nicholson
3f44b08b7b do v21 detection instead of CED detection for the fax gateway
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-12 15:23:24 +00:00
David Vossel
1339a0a535 Video support for ConfBridge.
Review: https://reviewboard.asterisk.org/r/1288/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-30 20:33:15 +00:00
Jonathan Rose
1de75f0a4d Merged revisions 319261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines
  
  Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-16 21:08:50 +00:00
Russell Bryant
c28e2d380c Merged revisions 317429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) | 5 lines
  
  Only display inband DTMF warning if inband DTMF detection is enabled.
  
  (closes issue #18901)
  Reported by: irroot
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-05 22:12:10 +00:00
Russell Bryant
37aa52fd78 Merged revisions 316265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
  
  Fix a bunch of compiler warnings generated by gcc 4.6.0.
  
  Most of these are -Wunused-but-set-variable, but there were a few others
  mixed in here, as well.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03 20:45:32 +00:00
David Vossel
7f23115ad2 New HD ConfBridge conferencing application.
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
2011-04-21 18:11:40 +00:00
Jonathan Rose
846cfa0ef0 New Feature for chan_dahdi. 4 length pattern matching.
In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns.  The s
ntax remains the same and the method used to track the pattern history will only change when using the length
 4 patterns.

(closes issue SWP-3250)
Code:
        jrose
        rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-01 17:01:01 +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
Paul Belanger
1bc478656e Set threshold for silence detection defaults to 256
(closes issue #15685)
Reported by: david_s5
Patches:
      dsp-silence-threshold-init.diff uploaded by dant (license 670)
      issue15685.patch.v5 uploaded by pabelanger (license 224)
Tested by: danti

Review: https://reviewboard.asterisk.org/r/670/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07 17:34:45 +00:00
Tilghman Lesher
47ad8c27f5 Fix crash in DTMF detection.
What I did not originally see in my previous commit was that even though the
next digit could be detected before the previous was considered ended, the
detection of the next digit effectively ends the detection of the previous.
Therefore, the length moves in lockstep with the digit, and no separate counter
is needed for the length alone.

(closes issue #17371)
 Reported by: alecdavis

(closes issue #17474)
 Reported by: kenner


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-05 17:55:28 +00:00
Tilghman Lesher
07df131a7f Keep track of digit duration, when we're decoding inband to pass DTMF frames.
(closes issue #17235)
 Reported by: frawd
 Patches: 
       new_dtmf_dsp_len.patch uploaded by frawd (license 610)
       20100518__issue17235.diff.txt uploaded by tilghman (license 14)
 Tested by: frawd


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 16:42:20 +00:00
Tilghman Lesher
f55aff74ed Merged revisions 263949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) | 8 lines
  
  Because progress is called multiple times, across several frames, we must persist states when detecting multitone sequences.
  
  (closes issue #16749)
   Reported by: dant
   Patches: 
         dsp.c-bug16749-1.patch uploaded by dant (license 670)
   Tested by: dant
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 06:41:04 +00:00
Alec L Davis
ec0581fef4 restarts busydetector (if enabled) when DTMF is received after call is bridged.
(closes issue 0016389)
  Reported by: alecdavis
  Tested by: alecdavis
  Patch
    dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-20 08:22:35 +00:00
Alec L Davis
155931303b Whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15 03:26:49 +00:00
Alec L Davis
6c50fad99f restarts busydetector (if enabled) when DTMF is received.
(closes issue #16389)
  Reported by: alecdavis
  Tested by: alecdavis
  Patch
	dtmf_busydetector.diff.txt uploaded by alecdavis (license 585)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15 03:04:59 +00:00
Matthias Nick
71ca1b54cb Merged revisions 233014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | 11 lines
  
  Warning message gets displayed only once
  
  Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second.
  
  (closes issue #15769)
  Reported by: falves11
  Patches:
  	patch_15769_14.txt uploaded by mnick (license 874)
  Tested by: mnick, falves11
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04 15:38:33 +00:00
Tilghman Lesher
d8e0c58437 Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Russell Bryant
cd10bd931a Merged revisions 224931 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines
  
  Isolate frames returned from a DSP instance or codec translator.
  
  The reasoning for these changes are the same as what I wrote in the commit
  message for rev 222878.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21 03:09:04 +00:00
Tilghman Lesher
642bec4d6f AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10 19:20:57 +00:00
Kevin P. Fleming
e6b2e9a750 Const-ify the world (or at least a good part of it)
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:

- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments

In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.

Review: https://reviewboard.asterisk.org/r/251/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21 21:13:09 +00:00
Tilghman Lesher
bb80c835e0 Add an option, waitfordialtone, for UK analog lines which do not end a call
until the originating line hangs up.
(closes issue #12382)
 Reported by: one47
 Patches: 
       zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23)
       zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463)
 Tested by: fleed


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25 22:45:59 +00:00
Tilghman Lesher
455284ebc1 Add a bit of documentation (thanks, I-MOD) on what the silence threshold
constant actually does and what values are valid for it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-24 21:52:34 +00:00
Tilghman Lesher
08af5bb312 Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating
when a file is invalid from when a file is missing.  This is most important when
we have two configuration files.  Consider the following example:

Old system:
sip.conf     users.conf     Old result               New result
========     ==========     ==========               ==========
Missing      Missing        SIP doesn't load         SIP doesn't load
Missing      OK             SIP doesn't load         SIP doesn't load
Missing      Invalid        SIP doesn't load         SIP doesn't load
OK           Missing        SIP loads                SIP loads
OK           OK             SIP loads                SIP loads
OK           Invalid        SIP loads incompletely   SIP doesn't load
Invalid      Missing        SIP doesn't load         SIP doesn't load
Invalid      OK             SIP doesn't load         SIP doesn't load
Invalid      Invalid        SIP doesn't load         SIP doesn't load

So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed.  Worse yet, the old
system would do this with no indication that anything was even wrong.

(closes issue #10690)
 Reported by: dtyoo
 Patches: 
       20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 23:30:03 +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
Russell Bryant
6fd6286a11 arbitrary formatting change to test mantis change
(closes issue #12824)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-09 16:55:15 +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