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2014-07-15codec_adpcm: Change description of codec "ADPCM" to "Dialogic ADPCM"Matthew Jordan
Technically, ADPCM is a method that can be applied to several codecs. Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec. See http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information about said codec. Review: https://reviewboard.asterisk.org/r/3744 patches: rb3744.patch uploaded by dennis.guse (License 6513) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-11format.c: Fix misuse of hash container function.Richard Mudgett
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
2014-05-09Allow Asterisk to compile under GCC 4.10Kinsey Moore
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
2014-03-27main/formats: Fix crash in ast_format_cmp during non-clean shutdown.Corey Farrell
* Update asterisk.h to reflect availability of ast_register_cleanup in 11.9. * Use ast_register_cleanup for format_attr_shutdown. (closes issue ASTERISK-23103) Reported by: JoshE ........ Merged revisions 411310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-14format.c: correct possible null pointer dereferenceScott Griepentrog
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
2013-08-23Add pass through support for Opus and VP8; Opus format attribute negotiationMatthew Jordan
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
2013-08-16Doxygen comment tweaks.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-12Fix memory leak while loading modules, adding formats, and destroying endpointsMatthew Jordan
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
2012-12-05Remove init_framer(). It no longer does anything.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-05Fix registering core show codecs/codec CLI commands twice.Richard Mudgett
........ Merged revisions 377241 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377244 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-02Fix a variety of ref counting issuesMatthew Jordan
This patch resolves a number of ref leaks that occur primarily on Asterisk shutdown. It adds a variety of shutdown routines to core portions of Asterisk such that they can reclaim resources allocate duringd initialization. Review: https://reviewboard.asterisk.org/r/2137 ........ Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374196 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13Add support for parsing SDP attributes, generating SDP attributes, and ↵Joshua Colp
passing it through. 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
2012-06-15Multiple revisions 369001-369002Kevin P. Fleming
........ 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
2012-03-13Use struct copy instead of memcpy().Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28Astobj2 locking enhancement.Richard Mudgett
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
2012-01-25Eliminate unnecessary rebuilds of main/format*.c.Kevin P. Fleming
These files have no need to include "asterisk/version.h", and doing so forces them to be rebuilt each time a Subversion checkout moves between 'modified' and 'unmodified' states. ........ Merged revisions 352516 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-07list all of the codecs associated with a particular format id for CLI ↵Matthew Nicholson
command "core show codec" AST-699 ........ Merged revisions 343533 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-31Misc format capability fixes.Richard Mudgett
* 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
2011-07-07Adds pass-through support for codec CELT.David Vossel
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
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵David Vossel
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
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
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
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