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.
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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)
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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
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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.
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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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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().
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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
Richard Mudgett
Kinsey Moore
Corey Farrell
Scott Griepentrog
Matthew Jordan
Joshua Colp
Kevin P. Fleming
Matthew Nicholson
David Vossel