When started with a verbose level of 3, asterisk can emit over 1500
verbose message that serve no real purpose other than to fill up
logs. When asterisk shuts down, it emits another 1100 that are of
even less use. Since the testsuite runs asterisk with a verbose
level of 3, and asterisk starts and stops for every one of the 700+
tests, the number of log messages is staggering. Besides taking up
resources, it also makes it hard to debug failing tests.
This commit changes the log level for those verbose messages to 5
instead of 3 which reduces the number of log messages to only a
handful. Of course, NOTICE, WARNING and ERROR message are
unaffected.
There's also one other minor change...
ast_context_remove_extension_callerid2() logs a DEBUG message
instead of an ERROR if the extension you're deleting doesn't exist.
The pjsip_config_wizard calls that function to clean up the config
and has been triggering that annoying error message for years.
Resolves: #582
(cherry picked from commit a433ed0d5a)
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are also removed. Only ao2_container_alloc remains due to
it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
This uses AO2_STRING_FIELD_HASH_FN and AO2_STRING_FIELD_CMP_FN where
possible in the Asterisk core.
This removes CMP_STOP from the result of CMP_FN callbacks for the
following structure types:
* ast_bucket_metadata
* ast_bucket_scheme
* generic_monitor_instance_list (ccss.c)
* ast_bucket_file (media_cache.c)
* named_acl
Change-Id: Ide4c1449a894bce70dea1fef664dade9b57578f1
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
Adds an identifier (with a getter and setter) to detect channels with
interleaved audio.
This is needed by the binaural bridge_softmix patch (ASTERISK-26292) and
was already discussed here:
http://lists.digium.com/pipermail/asterisk-dev/2016-October/075900.html
The identifier can be set during fmtp parsing (to be seen in the
res_format_attr_opus.c change).
ASTERISK-26292
Change-Id: I359801cc5f98c35671c48dabc81a7f4ee1183d63
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
In Asterisk 13, cached formats are created before their corresponding format-
attribute module is registered. Cached formats are involved when a local
extension is called. Therefore, ast_format_generate_sdp_fmtp did not work
on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264,
and format-attribute modules provided externally.
ASTERISK-25160 #close
Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354
Some codecs that may be a third party library to Asterisk need to have
knowledge of the format attributes that were negotiated. Unfortunately,
when the great format migration of Asterisk 13 occurred, that ability
was lost.
This patch adds an API call, ast_format_attribute_get, to the core
format API, along with updates to the unit test to check the new API
call. A new callback is also now available for format attribute modules,
such that they can provide the format attribute values they manage.
Note that the API returns a void *. This is done as the format attribute
modules themselves may store format attributes in any particular manner
they like. Care should be taken by consumers of the API to check the
return value before casting and dereferencing. Consumers will obviously
need to have a priori knowledge of the type of the format attribute as
well.
Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3
Format interfaces cannot be unregistered, so the modules that provide them
need to be held open except by shutdown.
ASTERISK-25054 #close
Reported by: Corey Farrell
Change-Id: Iadbd9675bf0d30b8fded5a739b163db3ea2db8f3
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
Versions of Asterisk prior to 12 defaulted to 8000 as a sample rate
if one was not provided by a format. In Asterisk 13, this was removed.
The result was that some calculations which involve dividing by the
sample rate resulted in dividing by 0. The fix being put in place
here is to have the same default fallback that was present in previous
versions of Asterisk.
Asterisk-24914 #close
Reported by Marcello Ceschia
........
Merged revisions 434046 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups. Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe. ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.
Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.
ASTERISK-24142 #close
Reported by: David Brillert
ASTERISK-24683 #close
Reported by: Peter Katzmann
ASTERISK-24805 #close
Reported by: Badalian Vyacheslav
ASTERISK-24881 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
........
Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When shutting down Asterisk the codecs are cleaned up. As a result anything
attempting to get a codec based on ID or details will find that no codec
exists. This currently occurs when determining the sample count of a frame.
This code did not take this situation into account.
This change fixes this by getting the codec directly from the format and
eliminates the lookup. This is both faster and also provides a guarantee
that the codec will exist and will be valid.
ASTERISK-24604 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4260/
........
Merged revisions 429497 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The supplied hash function to a container must be idempotent given the
object's key value to figure out which container bucket the object belongs
in. Returning a random number or the current container count is not
idempotent. The "computed hash" value doesn't help find the object later
in those cases.
* Fixed the format_list container to actually be a list since that is how
the container is used. Conceptually, if more than 283 formats were added
to the format_list then odd things may have happened before the fix.
........
Merged revisions 415728 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 415729 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ast_format_sdp_parse and ast_format_sdp_generate
the check checks for a valid interface and function
were potentially confusing, and hid an error in the
test of the presence of the function that is called
later. This patch clears up and corrects the test.
Review: https://reviewboard.asterisk.org/r/3208/
(closes issue ASTERISK-23098)
Reported by: marcelloceschia
Patches:
main_format.patch uploaded by marcelloceschia (license 6036)
ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 408137 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 408138 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes three memory leaks
* When we load a module with the LOAD_PRIORITY flag, we remove its entry from
the load order list. Unfortunately, we don't free the memory associated with
entry in the list. This patch corrects that and properly frees the memory
for the module in the list.
* When adding a custom format (such as SILK or CELT), the routine for adding
the format was leaking a reference. RAII_VAR cleans this up properly.
* We now de-ref the channel_snapshot appropriately when an endpoint is
disposed of
........
Merged revisions 391489 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 391507 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391521 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
Add the ability to specify what kind of locking an ao2 object has when it
is allocated. The locking could be one of: MUTEX, RWLOCK, or none.
New API:
ao2_t_alloc_options()
ao2_alloc_options()
ao2_t_container_alloc_options()
ao2_container_alloc_options()
ao2_rdlock()
ao2_wrlock()
ao2_tryrdlock()
ao2_trywrlock()
The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning
change. They no longer mean that the object is protected by an external
mechanism. They mean the lock associated with the object has already been
manually obtained by one of the ao2_lock calls. This change is necessary
for RWLOCK support since they are not reentrant. Also an operation on an
ao2 container may require promoting a read lock to a write lock by
releasing the already held read lock to re-acquire as a write lock.
Replaced API calls:
ao2_t_link_nolock()
ao2_link_nolock()
ao2_t_unlink_nolock()
ao2_unlink_nolock()
with the respective
ao2_t_link_flags()
ao2_link_flags()
ao2_t_unlink_flags()
ao2_unlink_flags()
API calls to be more flexible and to allow an anticipated enhancement to
control linking duplicate objects into a container.
The changes to format.c and format_cap.c are taking advantange of the new
ao2 locking options to simplify the use of the format capabilities
containers.
Review: https://reviewboard.asterisk.org/r/1554/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed typo in format_cap.c:joint_copy_helper() using the wrong variable.
* Fix potential race between checking if an interface exists and adding it
to the container in format.c:ast_format_attr_reg_interface().
* Fixed double rwlock destroy in format.c:ast_format_attr_init() error
exit path.
* Simplified format.c:find_interface() and format.c:has_interface().
........
Merged revisions 342824 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds pass-through support for CELT. CELT
formats are defined in codecs.conf and can be configured
to any sample rate a CELT endpoint supports. This patch also
addresses a crash in channel.c resulting from a frame list being
freed incorrectly. This crash was discovered while testing a CELT
translator which had to split encoded audio into multiple frames.
The codec translator is not a part of this patch, but may be
contributed in the future.
Review: https://reviewboard.asterisk.org/r/1294/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-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
George Joseph
Corey Farrell
Sean Bright
frahaase
Alexander Traud
Matt Jordan
Mark Michelson
Joshua Colp
Jonathan Rose
Richard Mudgett
Kinsey Moore
Scott Griepentrog
Kevin P. Fleming
Matthew Nicholson
David Vossel