81 Commits (e3de4bc46e0332050f3fd296202abe94d13c20a5)

Author	SHA1	Message	Date
Richard Mudgett	7f916d621a	audiohook.c: Fix freeing a frame and still using it. ... Memory corruption happened to the media frame caches when an audio hook freed a frame when it shouldn't. I think the freed frame was because a jitter buffer interpolated a missing frame and the audio hook unconditionally freed it. * Made audiohook.c:audio_audiohook_write_list() not free an interpolated frame if it is the same frame as what was passed into the routine. * Made plc.c:normalise_history() use memmove() instead of memcpy() on a memory block that could overlap. Found by valgrind investigating this issue. ASTERISK-27238 ASTERISK-27412 Change-Id: I548d86894281fc4529aefeb9f161f2131ecc6fde	8 years ago
Joshua Colp	ae696132a2	frame: Better handle interpolated frames. ... Interpolated frames are frames which contain a number of samples but have no actual data. Audiohooks did not handle this case when translating an incoming frame into signed linear. It assumed that a frame would always contain media when it may not. If this occurs audiohooks will now immediately return and not act on the frame. As well for users of ast_trans_frameout the function has been changed to be a bit more sane and ensure that the data pointer on a frame is set to NULL if no data is actually on the frame. This allows the various spots in Asterisk that check for an interpolated frame based on the presence of a data pointer to work as expected. ASTERISK-26926 Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b	8 years ago
Aaron An	d5b480afca	audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor. ... Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22	8 years ago
Sean Bright	3aee199913	audiohooks: Muting a hook can mute underlying frames ... If an audiohook is placed on a channel that does not require transcoding, muting that hook will cause the underlying frames to be muted as well. The original patch is from David Woolley but I have modified slightly. ASTERISK-21094 #close Reported by: David Woolley Patches: ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded by David Woolley Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed	8 years ago
Jonathan Rose	072d94183c	Fix crash in audiohook translate to slin ... This patch fixes a crash which would occur when an audiohook was applied to a channel using an audio codec that could not be translated to signed linear (such as when using pass-through codecs like OPUS or when the codec translator module for the format in use is not loaded). ASTERISK-25498 #close Reported by: Ben Langfeld Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384	10 years ago
Richard Mudgett	cea5dc7b8a	audiohook.c: Simplify variable usage in audiohook_read_frame_both(). ... Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c	10 years ago
Richard Mudgett	b3a56bee83	audiohook.c: Fix MixMonitor crash when using the r() or t() options. ... The built frame format in audiohook_read_frame_both() is now set to a signed linear format before the rx and tx frames are duplicated instead of only for the mixed audio frame duplication. ASTERISK-25322 #close Reported by Sean Pimental Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538	10 years ago
Matt Jordan	4d8f47f4bf	Merge "audiohook: Use manipulated frame instead of dropping it." into 13	10 years ago
Joshua Colp	f509730cb9	audiohook: Use manipulated frame instead of dropping it. ... Previous changes to sample rate support in audiohooks accidentally removed code responsible for allowing the manipulate audiohooks to work. Without this code the manipulated frame would be dropped and not used. This change restores it. ASTERISK-25253 #close Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13	10 years ago
Joshua Colp	f1493f900e	audiohook: Read the correct number of samples based on audiohook format. ... Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d	10 years ago
Kevin Harwell	b1e8c0b9eb	audiohook.c: Difference in read/write rates caused continuous buffer resets ... Currently, everytime a sample rate change occurs (on read or write) the associated factory buffers are reset. If the requested sample rate on a read differed from that of a write then the buffers are continually reset on every read and write. This has the side effect of emptying the buffer, thus there being no data to read and then write to a file in the case of call recording. This patch fixes it so that an audiohook_list's rate always maintains the maximum sample rate among hooks and formats. Audiohook sample rates are only overwritten by this value when slin native compatibility is turned on. Also, the audiohook sample rate can only overwrite the list's sample rate when its rate is greater than that of the list or if compatibility is turned off. This keeps the rate from constantly switching/resetting. ASTERISK-24944 #close Reported by: Ronald Raikes Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f	10 years ago
Matthew Jordan	b9fd61f2c7	main/audiohook: Update internal sample rate on reads ... When an audiohook is created (which is used by the various Spy applications and Snoop channel in Asterisk 13+), it initially is given a sample rate of 8kHz. It is expected, however, that this rate may change based on the media that passes through the audiohook. However, the read/write operations on the audiohook behave very differently. When a frame is written to the audiohook, the format of the frame is checked against the internal sample rate. If the rate of the format does not match the internal sample rate, the internal sample rate is updated and a new SLIN format is chosen based on that sample rate. This works just fine. When a frame is read, however, we do something quite different. If the format rate matches the internal sample rate, all is fine. However, if the rates don't match, the audiohook attempts to "fix up" the number of samples that were requested. This can result in some seriously large number of samples being requested from the read/write factories. Consider the worst case - 192kHz SLIN. If we attempt to read 20ms worth of audio produced at that rate, we'd request 3840 samples (192000 / (1000 / 20)). However, if the audiohook is still expecting an internal sample rate of 8000, we'll attempt to "fix up" the requested samples to: samples_converted = samples * (ast_format_get_sample_rate(format) / (float) audiohook->hook_internal_samp_rate); which is: 92160 = 3840 * (192000 / 8000) This results in us attempting to read 92160 samples from our factories, as opposed to the 3840 that we actually wanted. On a 64-bit machine, this miraculously survives - despite allocating up to two buffers of length 92160 on the stack. The 32-bit machines aren't quite so lucky. Even in the case where this works, we will either (a) get way more samples than we wanted; or (b) get about 3840 samples, assuming the timing is pretty good on the machine. Either way, the calculation being performed is wrong, based on the API users expectations. My first inclination was to allocate the buffers on the heap. As it is, however, there's at least two drawbacks with doing this: (1) It's a bit complicated, as the size of the buffers may change during the lifetime of the audiohook (ew). (2) The stack is faster (yay); the heap is slower (boo). Since our calculation is flat out wrong in the first place, this patch fixes this issue by instead updating the internal sample rate based on the format passed into the read operation. This causes us to read the correct number of samples, and has the added benefit of setting the audihook with the right SLIN format. Note that this issue was caught by the Asterisk Test Suite as a result of r432195 in the 13 branch. Because this issue is also theoretically possible in Asterisk 11, the change is being made here as well. Review: https://reviewboard.asterisk.org/r/4475/ ........ Merged revisions 432810 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	d51169cd36	audiohooks: Clean references to formats ... Cleanup references to in_translate[x].format and out_translate[x].format in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	cff192429b	audiohooks: Reevaluate the bridge technology when an audiohook is added or removed. ... Adding a mixmonitor to a channel causes the bridge to change technologies from native to simple_bridge so the call can be recorded. However, when the mixmonitor is stopped the bridge does not switch back to the native technology. * Added unbridge requests to reevaluate the bridge when a channel audiohook is removed. * Moved the unbridge request into ast_audiohook_attach() ensure that the bridge reevaluates whenever an audiohook is attached. This simplified the mixmonitor and chan_spy start code as well. * Added defensive code to stop_mixmonitor_full() in case additional arguments are ever added to the StopMixMonitor application. * Made ast_framehook_detach() not do an unbridge request if the framehook does not exist. * Made ast_framehook_list_fixup() do an unbridge request if there are any framehooks. Also simplified the loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4046/ ........ Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
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	11 years ago
Jonathan Rose	af4cd65143	Channels: Masquerades to automatically move frame/audio hooks ... Whenever possible, audiohooks and framehooks will now be copied over to the channel that the masquerading channel gets cloned into. This should occur for all audiohooks and most framehooks. As a result, in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now deprecated and its behavior is essentially the new default for all audiohooks, plus some additional audiohooks/framehooks. Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	53968c00b3	TALK_DETECT: A channel function that raises events when talking is detected ... This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
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. ........ 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	11 years ago
Olle Johansson	f65dd23bf4	Formatting: Remove invisible characters ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Olle Johansson	2a4205df2c	Formatting only. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	eda7126862	ari: Add Snoop operation for spying/whispering on channels. ... The Snoop operation can be invoked on a channel to spy or whisper on it. It returns a channel that any channel operations can then be invoked on (such as record to do monitoring). (closes issue ASTERISK-22780) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3003/ ........ Merged revisions 403117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	643fb1ed14	Minor optimizations. ... * Made ast_audiohook_detach_list() and ast_audiohook_write_list_empty() NULL tolerant. * Made ast_audiohook_detach_list() return void since it is a destructor. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Andrew Latham	fd98835f1f	Doxygen Updates Janitor Work ... * Whitespace, doc-blocks, spelling, case, missing and incorrect tags. * Add cleanup to Makefile for the Doxygen configuration update * Start updating Doxygen configuration for cleaner output * Enable inclusion of configuration files into documentation * remove mantisworkflow... * update documentation README * Add markup to Tilghman's email and talk with him about updating his email, he knows... * no code changes on this commit other than the mentioned Makefile change (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8018b879a2	Clean up doxygen warnings ... This patch fixes numerous doxygen warnings across Asterisk. It also updates the makefile to regenerate the doxygen configuration on the local system before running doxygen to help prevent warnings/errors on the local system. Much thanks to Andrew for tackling one of the Asterisk janitor projects! (issue ASTERISK-20259) Reported by: Andrew Latham Patches: doxygen_partial.diff uploaded by Andrew Latham (license 5985) make_progdocs.diff uploaded by Andrew Latham (license 5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	166b4e2b30	Multiple revisions 369001-369002 ... ........ 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	13 years ago
Matthew Jordan	54143892af	Only log a failure to get read/write samples from factories if it didn't happen ... In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. ........ Merged revisions 364965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
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	13 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types ... Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	b80fcd77e5	Revert a change to audio_audiohook_write_list that had no affect. ... When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. ........ Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355623 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	9e48f6799d	Use ast_audiohook_write_list_empty to determine if our lists are empty instead ... of duplicating that logic. ........ Merged revisions 349289 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349290 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	8017be6fa9	Once an audiohook is attached to a channel, we continue to transcode all of the ... frames, even after all of the hooks are detached. This patch short-cicuits us out before we transcode unnecessarily. ........ Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349145 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
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	14 years ago
Jonathan Rose	16c5dda8ab	Fixes null reference bug introduced by audio hook changes that affects various OS distributions. Thanks David. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6e36042f64	Mix Monitor: Now with r and t options. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd 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	14 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() ... (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
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	15 years ago
David Vossel	395a35900a	Merged revisions 279949 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279949 | dvossel | 2010-07-27 15:57:00 -0500 (Tue, 27 Jul 2010) | 31 lines Merged revisions 279946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	c672763af8	Fix some doxygen warnings. ... (closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d4358a46a9	Merged revisions 260049 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines Fixes crash in audiohook_write_list The middle_frame in the audiohook_write_list function was being freed if a audiohook manipulator returned a failure. This is incorrect logic. This patch resolves this and adds detailed descriptions of how this function should work and why manipulator failures must be ignored. (closes issue #17052) Reported by: dvossel Tested by: dvossel (closes issue #16196) Reported by: atis Review: https://reviewboard.asterisk.org/r/623/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Julian Lyndon-Smith	d85650e4aa	Added MixMonitorMute manager command ... Added a new manager command to mute/unmute MixMonitor audio on a channel. Added a new feature to audiohooks so that you can mute either read / write (or both) types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself. (closes issue #16740) Reported by: jmls Review: https://reviewboard.asterisk.org/r/487/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	bf06747778	fixes AUDIOHOOK_INHERIT regression ... During the process of removing an audiohook from one channel and attaching it to another the audiohook's status is updated to DONE and then back to whatever it was previously. Typically updating the status after setting it to DONE is not a good idea because DONE can trigger unrecoverable audiohook destruction events... because of this a conditional check was added to audiohook_update_status to explicitly prevent the audiohook from ever changing after being set to DONE. It was this check that prevented audiohook inherit from work properly though. Now ast_audiohook_move_by_source is treated as a special exception, as the audiohook must be returned to its previous status after attaching it to the new channel. This is only a safe operation because the audiohook's lock is held the entire time, otherwise this could cause trouble. (closes issue #16522) Reported by: corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3595fbb70c	audiohook signal trigger on every status change ... (issue #14618) Review: https://reviewboard.asterisk.org/r/434/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
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	16 years ago
Tilghman Lesher	77031501a5	Merged revisions 224855 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a7fd763ecc	Merged revisions 197537 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	2d186315d2	Drop my IRC nickname. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1030a25ac9	Modify headers and macros, according to Russell's suggestions on the -dev list ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	547b5c7e90	audio_audiohook_write_list() did not correctly update sample size after ast_translate. ... audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	d40f86db1b	Merged revisions 185196 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago