This reverts commit 3c8116c8. It has
been a merge that has been acidentally squashed into one commit with the
consequence that all history has been lost. The next commit will be a
proper merge commit rectifying this.

Grzegorz Sluja authored 1 year ago and Yalu Zhang committed 1 year ago

Our commits to the previous version have been rebased.

ASTERISK-29980 #close

Change-Id: I7b347665822ea2774dd322276c09be67914d2065

Correct typos of the following word families:

voiced
denumerator
codeword
upsampling
constructed
residual
subroutine
conditional
quantizing
courtesy
number

ASTERISK-29714

Change-Id: I471fb8086a5277d8f05047fedee22cfa97a4252d

Correct typos of the following word families:

voiced
denumerator
codeword
upsampling
constructed
residual
subroutine
conditional
quantizing
courtesy
number

ASTERISK-29714

Change-Id: I471fb8086a5277d8f05047fedee22cfa97a4252d

This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.

ASTERISK-29335

Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90

This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.

ASTERISK-29335

Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90

This was dead code, test code introduced with Asterisk 13. This was
found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
partly fixes, not closes those two issues.

Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940

This patch fixes several issues reported by the lgtm code analysis tool:

https://lgtm.com/projects/g/asterisk/asterisk

Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:

* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards

Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25

ASTERISK-28511

Change-Id: If0d58598ce14aad3c786a1c0127b5f7b200b737d

ASTERISK-28511 #close

Change-Id: Idd07bf341e89ac999c7f5701d9b72b8a9cb11e82

If codec_speex fails to register a translator it would cause Asterisk to
exit instead of continue as a DECLINED module.

* Make unload_module() always return 0.  It is silly to fail unloading if
any translators we try to unregister were not even registered.

Change-Id: Ia262591f68333dad17673ba7104d11c88096f51a

This new option can be passed for ./configure or
./tests/CI/buildAsterisk.sh to prevent download/install of binary
modules.

Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES
will result in binary modules being enabled even if the build target is
incompatible with those modules.  This includes CI scripts which enable
categories before disabling specific modules.

If more binary modules are offered in the future this will help avoid
accidentally downloading them if unwanted or incompatible.  Adding a
binary module will only require creating a new menuselect entry similar
to the existing ones, it will not be necessary to modify the CI scripts.

Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166

The symbol FS is the sampling frequency. That symbol is not used in Asterisk at
all and was a copy-and-paste of the iLBC reference code from the IETF RFC.
However, in Solaris, that symbol is defined by another header already. To
compile in Solaris, that symbol has to go.

Change-Id: I91ddbe5be7c00069c3a25abd5f58d7b2f04c51b1

This change extends the existing AST_FRAME_RTCP frame type to be
able to contain additional RTCP message types, such as feedback
messages. The payload type is contained in the subclass which allows
knowing what is in the frame itself.

The RTCP feedback message type is now handled and REMB[1] messages
are raised with their containing information.

This also fixes a bug where all feedback messages were triggering
video updates instead of just FIR and FUR.

Finally RTCP frames are now passed up through the Asterisk core to
what is handling the channel, mapped appropriately in the case of
bridging, and written to an outgoing stream. Since RTCP frames are
on a per-stream basis this is only done on multistream capable
channels.

[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03

ASTERISK-27758
ASTERISK-26366

Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e

When the latest version of that library was installed, Asterisk did not build.

ASTERISK-27669
Reported by: Николай Михо

Change-Id: I27e09bb875fdd56423bd9fae1be85fddb428eb96

This reverts commit 028f4320.

Change-Id: Ieb91f825cb55202a937f5361c01d356e7662b70c

clang 5.0 warned about this.

ASTERISK-27558

Change-Id: Icc452ecb0d86bbeba78dae768cc472ec540699df

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

ASTERISK-27475 #close

Change-Id: If7384bc6ed002ef140dec69798d14c52b7cfd800

There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.

Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7

ASTERISK-23556
Reported by: Marcello Ceschia

Change-Id: Ic27e88e0336a0d83877dc857938659dc5560b93c

Change-Id: Iac40ecb20e10513d67bf0eaf61807f306067b258

In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE.  This prevents asterisk from exiting
if a module can't be loaded.  If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.

A new API was added to logger: ast_is_logger_initialized().  This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout.  If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.

Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25

This has not worked for some time and is no longer actively maintained.

Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99

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

POSIX defines poll.h. sys/poll.h should not be used as it is c-library
internal header which may or may not exist. Notably in musl including
sys/poll.h generates warning of being incorrect.

Change-Id: Ib318c1c7142a737bcf3caa4d8d72560bebe39252

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

Updated codecs/codecs.xml to add codec_opus to the external
download list.

ASTERISK-26409

Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4
(cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c)
(cherry picked from commit e9684f3acd0e8def0df582c1505dd39dd3fd1610)

The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
http://downloads.digium.com/pub/telephony/ are now listed in the
"External" sections of the "Resource Modules" and "Codec Translators"
pages in menuselect.  Any that are selected will automatically be
downloaded and installed when "make install" is run.  Their LICENSE and
README (if avaialble) files will be installed to
ASTVARLIBDIR/documentation/thirdparty/<product_name>.

Example use with codecs:

The codecs/codecs.xml file is a menuselect style xml file that lists
the codecs to be included.  Their support levels are 'external', which
triggers the download and install, and defaultenabled is no.  Also
because codec_g729a is actually in a directory named codec_g729 on the
download server, the newly added 'member_data' element is used to
override the default of the directory name being the package name.  You
can use the 'directory_name' attribute to keep default base URL
(http://downloads.digium.com/pub/telephony/) but use the new directory,
or you use the 'remote_url' attribute to specify a full URL to the
download directory.  In this case, you must still follow the same
subdirectory naming conventions as that used for the packages located
at 'http://downloads.digium.com/pub/telephony'.

A new configure option '--with-externals-cache' was added and like
'--with-sounds-cache' it allows the installer to cache tarballs so
they're not downloaded every time.

To assist with the download and install process, each external package
now has a manifest.xml file that, among other things, contains a package
version and checksums for each file in the tarball.  The manifest is
saved to both the cache directory and ASTMODDIR and together with the
manifest.xml on the downloads site, tells the install scripts whether
a download and/or update is needed.

bash and xmlstarlet are required for downloader operation.  If they're
not installed, the external items in menuselect will be unavailable.

Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a

ASTERISK-26217 #close

Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6

Modules must define AST_MODULE_SELF_SYM to be used as the name of a
generated function.  This produces a friendly error when it's not
defined.

ASTERISK-26278 #close

Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b

Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk
defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests
this.

ASTERISK-26218 #close
ASTERISK-26221 #close
Reported by: Aaron Meriwether

Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa

ASTERISK-25599 #close

Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e

ASTERISK-25599 #close

Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10

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

 * Add some type casting so tv_usec can really be a long, instead of
   some strange platform specific type.

 * Add some .dylib style files to .gitignore.

 * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer
   versions of GCC, when compiling the Homebrew formula for Asterisk,
   are not properly passing the -Xlinker options to the linker. Given
   that -Wl, does exactly the [same thing][], and does it properly, this
   patch changes the -Xlinker options to use -Wl, instead.

 [reasons unknown]: http://bit.ly/1SUbEYx
 [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html

Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd

There are 3 ways that calls directly to standard allocator functions can
be dealt with:
1. Block their use, cause them to generate an error.  This is the default.
2. Replace them with the Asterisk equivalent function calls.
3. Leave them alone.

This change allows one of these 3 options to be selected by any source.
The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT,
or ASTMM_IGNORE to use option 1, 2 or 3 respectively.  Normally ASTMM_BLOCK
is the correct option, so it is default when ASTMM_LIBC is not defined.
In some cases when building 3rd party code it is desirable to have it use
Asterisk functions, without changing the whole source - ASTMM_REDIRECT
accomplishes this.  When using 3rd party libraries sometimes a static
inline function will make use of malloc or free.  In these cases it may
be unsafe to replace the allocator in the header, as it's possible the
memory could be freed by the library using standard allocators.  For
those cases ASTMM_IGNORE is needed.

Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7

Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723

This switches codecs/lpc10/lpcini.c back to including "asterisk.h"
instead of <stdlib.h>.  lpcini.c allocates memory that is freed by
codec_lpc10.c, so it is important to use MALLOC_DEBUG allocator.
Added #define WRAP_LIBC_MALLOC to the start of the source to prevent
runtime symbol link error's.

Change-Id: I74f63fd09fdeb673ee7753122c3bb4722ab6e1ac