asterisk/main/app.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Convenient Application Routines
*
* \author Mark Spencer <markster@digium.com>
*/
/*!
* Application Skeleton is an example of creating an application for Asterisk.
* \example app_skel.c
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#ifdef HAVE_SYS_STAT_H
#include <sys/stat.h>
#endif
#include <regex.h> /* for regcomp(3) */
#include <sys/file.h> /* for flock(2) */
#include <signal.h> /* for pthread_sigmask(3) */
#include <stdlib.h> /* for closefrom(3) */
#include <sys/types.h>
#include <sys/wait.h> /* for waitpid(2) */
#ifndef HAVE_CLOSEFROM
#include <dirent.h> /* for opendir(3) */
#endif
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/file.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/indications.h"
#include "asterisk/linkedlists.h"
#include "asterisk/threadstorage.h"
#include "asterisk/test.h"
#include "asterisk/module.h"
#include "asterisk/astobj2.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/json.h"
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
2014-07-20 22:06:33 +00:00
#include "asterisk/format_cache.h"
AST_THREADSTORAGE_PUBLIC(ast_str_thread_global_buf);
static pthread_t shaun_of_the_dead_thread = AST_PTHREADT_NULL;
struct zombie {
pid_t pid;
AST_LIST_ENTRY(zombie) list;
};
static AST_LIST_HEAD_STATIC(zombies, zombie);
#ifdef HAVE_CAP
static cap_t child_cap;
#endif
/*!
* \brief Define \ref stasis topic objects
* @{
*/
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
static struct stasis_topic *queue_topic_all;
static struct stasis_topic_pool *queue_topic_pool;
/*! @} */
static void *shaun_of_the_dead(void *data)
{
struct zombie *cur;
int status;
for (;;) {
if (!AST_LIST_EMPTY(&zombies)) {
/* Don't allow cancellation while we have a lock. */
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, NULL);
AST_LIST_LOCK(&zombies);
AST_LIST_TRAVERSE_SAFE_BEGIN(&zombies, cur, list) {
if (waitpid(cur->pid, &status, WNOHANG) != 0) {
AST_LIST_REMOVE_CURRENT(list);
ast_free(cur);
}
}
AST_LIST_TRAVERSE_SAFE_END
AST_LIST_UNLOCK(&zombies);
pthread_setcancelstate(PTHREAD_CANCEL_ENABLE, NULL);
}
pthread_testcancel();
/* Wait for 60 seconds, without engaging in a busy loop. */
ast_poll(NULL, 0, AST_LIST_FIRST(&zombies) ? 5000 : 60000);
}
return NULL;
}
#define AST_MAX_FORMATS 10
static AST_RWLIST_HEAD_STATIC(groups, ast_group_info);
/*!
* \brief This function presents a dialtone and reads an extension into 'collect'
* which must be a pointer to a **pre-initialized** array of char having a
* size of 'size' suitable for writing to. It will collect no more than the smaller
* of 'maxlen' or 'size' minus the original strlen() of collect digits.
* \param chan struct.
* \param context
* \param collect
* \param size
* \param maxlen
* \param timeout timeout in milliseconds
*/
int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect, size_t size, int maxlen, int timeout)
{
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
struct ast_tone_zone_sound *ts;
int res = 0, x = 0;
if (maxlen > size) {
maxlen = size;
}
if (!timeout) {
if (ast_channel_pbx(chan) && ast_channel_pbx(chan)->dtimeoutms) {
timeout = ast_channel_pbx(chan)->dtimeoutms;
} else {
timeout = 5000;
}
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
}
if ((ts = ast_get_indication_tone(ast_channel_zone(chan), "dial"))) {
res = ast_playtones_start(chan, 0, ts->data, 0);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
ts = ast_tone_zone_sound_unref(ts);
} else {
ast_log(LOG_NOTICE, "Huh....? no dial for indications?\n");
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
}
for (x = strlen(collect); x < maxlen; ) {
res = ast_waitfordigit(chan, timeout);
if (!ast_ignore_pattern(context, collect)) {
ast_playtones_stop(chan);
}
if (res < 1) {
break;
}
if (res == '#') {
break;
}
collect[x++] = res;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (!ast_matchmore_extension(chan, context, collect, 1,
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
break;
}
}
if (res >= 0) {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
res = ast_exists_extension(chan, context, collect, 1,
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL)) ? 1 : 0;
}
return res;
}
enum ast_getdata_result ast_app_getdata(struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout)
{
return ast_app_getdata_terminator(c, prompt, s, maxlen, timeout, NULL);
}
enum ast_getdata_result ast_app_getdata_terminator(struct ast_channel *c, const char *prompt, char *s,
int maxlen, int timeout, char *terminator)
{
int res = 0, to, fto;
char *front, *filename;
/* XXX Merge with full version? XXX */
if (maxlen)
s[0] = '\0';
if (!prompt)
prompt = "";
filename = ast_strdupa(prompt);
while ((front = ast_strsep(&filename, '&', AST_STRSEP_STRIP | AST_STRSEP_TRIM))) {
if (!ast_strlen_zero(front)) {
res = ast_streamfile(c, front, ast_channel_language(c));
if (res)
continue;
}
if (ast_strlen_zero(filename)) {
/* set timeouts for the last prompt */
fto = ast_channel_pbx(c) ? ast_channel_pbx(c)->rtimeoutms : 6000;
to = ast_channel_pbx(c) ? ast_channel_pbx(c)->dtimeoutms : 2000;
if (timeout > 0) {
fto = to = timeout;
}
if (timeout < 0) {
fto = to = 1000000000;
}
} else {
/* there is more than one prompt, so
* get rid of the long timeout between
* prompts, and make it 50ms */
fto = 50;
to = ast_channel_pbx(c) ? ast_channel_pbx(c)->dtimeoutms : 2000;
}
res = ast_readstring(c, s, maxlen, to, fto, (terminator ? terminator : "#"));
if (res == AST_GETDATA_EMPTY_END_TERMINATED) {
return res;
}
if (!ast_strlen_zero(s)) {
return res;
}
}
return res;
}
/* The lock type used by ast_lock_path() / ast_unlock_path() */
static enum AST_LOCK_TYPE ast_lock_type = AST_LOCK_TYPE_LOCKFILE;
int ast_app_getdata_full(struct ast_channel *c, const char *prompt, char *s, int maxlen, int timeout, int audiofd, int ctrlfd)
{
int res, to = 2000, fto = 6000;
if (!ast_strlen_zero(prompt)) {
res = ast_streamfile(c, prompt, ast_channel_language(c));
if (res < 0) {
return res;
}
}
if (timeout > 0) {
fto = to = timeout;
}
if (timeout < 0) {
fto = to = 1000000000;
}
res = ast_readstring_full(c, s, maxlen, to, fto, "#", audiofd, ctrlfd);
return res;
}
int ast_app_exec_macro(struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const char *macro_args)
{
struct ast_app *macro_app;
int res;
macro_app = pbx_findapp("Macro");
if (!macro_app) {
ast_log(LOG_WARNING,
"Cannot run 'Macro(%s)'. The application is not available.\n", macro_args);
return -1;
}
if (autoservice_chan) {
ast_autoservice_start(autoservice_chan);
}
ast_debug(4, "%s Original location: %s,%s,%d\n", ast_channel_name(macro_chan),
ast_channel_context(macro_chan), ast_channel_exten(macro_chan),
ast_channel_priority(macro_chan));
res = pbx_exec(macro_chan, macro_app, macro_args);
ast_debug(4, "Macro exited with status %d\n", res);
/*
* Assume anything negative from Macro is an error.
* Anything else is success.
*/
if (res < 0) {
res = -1;
} else {
res = 0;
}
ast_debug(4, "%s Ending location: %s,%s,%d\n", ast_channel_name(macro_chan),
ast_channel_context(macro_chan), ast_channel_exten(macro_chan),
ast_channel_priority(macro_chan));
if (autoservice_chan) {
ast_autoservice_stop(autoservice_chan);
}
if (ast_check_hangup_locked(macro_chan)) {
ast_queue_hangup(macro_chan);
}
return res;
}
int ast_app_run_macro(struct ast_channel *autoservice_chan, struct ast_channel *macro_chan, const char *macro_name, const char *macro_args)
{
int res;
char *args_str;
size_t args_len;
if (ast_strlen_zero(macro_args)) {
return ast_app_exec_macro(autoservice_chan, macro_chan, macro_name);
}
/* Create the Macro application argument string. */
args_len = strlen(macro_name) + strlen(macro_args) + 2;
args_str = ast_malloc(args_len);
if (!args_str) {
return -1;
}
snprintf(args_str, args_len, "%s,%s", macro_name, macro_args);
res = ast_app_exec_macro(autoservice_chan, macro_chan, args_str);
ast_free(args_str);
return res;
}
/* BUGBUG this is not thread safe. */
static const struct ast_app_stack_funcs *app_stack_callbacks;
void ast_install_stack_functions(const struct ast_app_stack_funcs *funcs)
{
app_stack_callbacks = funcs;
}
const char *ast_app_expand_sub_args(struct ast_channel *chan, const char *args)
{
const struct ast_app_stack_funcs *funcs;
const char *new_args;
funcs = app_stack_callbacks;
if (!funcs || !funcs->expand_sub_args || !ast_module_running_ref(funcs->module)) {
ast_log(LOG_WARNING,
"Cannot expand 'Gosub(%s)' arguments. The app_stack module is not available.\n",
args);
return NULL;
}
new_args = funcs->expand_sub_args(chan, args);
ast_module_unref(funcs->module);
return new_args;
}
int ast_app_exec_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const char *sub_args, int ignore_hangup)
{
const struct ast_app_stack_funcs *funcs;
int res;
funcs = app_stack_callbacks;
if (!funcs || !funcs->run_sub || !ast_module_running_ref(funcs->module)) {
ast_log(LOG_WARNING,
"Cannot run 'Gosub(%s)'. The app_stack module is not available.\n",
sub_args);
return -1;
}
if (autoservice_chan) {
ast_autoservice_start(autoservice_chan);
}
res = funcs->run_sub(sub_chan, sub_args, ignore_hangup);
ast_module_unref(funcs->module);
if (autoservice_chan) {
ast_autoservice_stop(autoservice_chan);
}
if (!ignore_hangup && ast_check_hangup_locked(sub_chan)) {
ast_queue_hangup(sub_chan);
}
return res;
}
int ast_app_run_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const char *sub_location, const char *sub_args, int ignore_hangup)
{
int res;
char *args_str;
size_t args_len;
if (ast_strlen_zero(sub_args)) {
return ast_app_exec_sub(autoservice_chan, sub_chan, sub_location, ignore_hangup);
}
/* Create the Gosub application argument string. */
args_len = strlen(sub_location) + strlen(sub_args) + 3;
args_str = ast_malloc(args_len);
if (!args_str) {
return -1;
}
snprintf(args_str, args_len, "%s(%s)", sub_location, sub_args);
res = ast_app_exec_sub(autoservice_chan, sub_chan, args_str, ignore_hangup);
ast_free(args_str);
return res;
}
/*! \brief The container for the voicemail provider */
static AO2_GLOBAL_OBJ_STATIC(vm_provider);
/*! Voicemail not registered warning */
static int vm_warnings;
int ast_vm_is_registered(void)
{
struct ast_vm_functions *table;
int is_registered;
table = ao2_global_obj_ref(vm_provider);
is_registered = table ? 1 : 0;
ao2_cleanup(table);
return is_registered;
}
int __ast_vm_register(const struct ast_vm_functions *vm_table, struct ast_module *module)
{
RAII_VAR(struct ast_vm_functions *, table, NULL, ao2_cleanup);
if (!vm_table->module_name) {
ast_log(LOG_ERROR, "Voicemail provider missing required information.\n");
return -1;
}
if (vm_table->module_version != VM_MODULE_VERSION) {
ast_log(LOG_ERROR, "Voicemail provider '%s' has incorrect version\n",
vm_table->module_name);
return -1;
}
table = ao2_global_obj_ref(vm_provider);
if (table) {
ast_log(LOG_WARNING, "Voicemail provider already registered by %s.\n",
table->module_name);
return AST_MODULE_LOAD_DECLINE;
}
table = ao2_alloc_options(sizeof(*table), NULL, AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!table) {
return -1;
}
*table = *vm_table;
table->module = module;
ao2_global_obj_replace_unref(vm_provider, table);
return 0;
}
void ast_vm_unregister(const char *module_name)
{
struct ast_vm_functions *table;
table = ao2_global_obj_ref(vm_provider);
if (table && !strcmp(table->module_name, module_name)) {
ao2_global_obj_release(vm_provider);
}
ao2_cleanup(table);
}
#ifdef TEST_FRAMEWORK
/*! \brief Holding container for the voicemail provider used while testing */
static AO2_GLOBAL_OBJ_STATIC(vm_provider_holder);
static int provider_is_swapped = 0;
void ast_vm_test_swap_table_in(const struct ast_vm_functions *vm_table)
{
RAII_VAR(struct ast_vm_functions *, holding_table, NULL, ao2_cleanup);
RAII_VAR(struct ast_vm_functions *, new_table, NULL, ao2_cleanup);
if (provider_is_swapped) {
ast_log(LOG_ERROR, "Attempted to swap in test function table without swapping out old test table.\n");
return;
}
holding_table = ao2_global_obj_ref(vm_provider);
if (holding_table) {
ao2_global_obj_replace_unref(vm_provider_holder, holding_table);
}
new_table = ao2_alloc_options(sizeof(*new_table), NULL, AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!new_table) {
return;
}
*new_table = *vm_table;
ao2_global_obj_replace_unref(vm_provider, new_table);
provider_is_swapped = 1;
}
void ast_vm_test_swap_table_out(void)
{
RAII_VAR(struct ast_vm_functions *, held_table, NULL, ao2_cleanup);
if (!provider_is_swapped) {
ast_log(LOG_ERROR, "Attempted to swap out test function table, but none is currently installed.\n");
return;
}
held_table = ao2_global_obj_ref(vm_provider_holder);
if (!held_table) {
return;
}
ao2_global_obj_replace_unref(vm_provider, held_table);
ao2_global_obj_release(vm_provider_holder);
provider_is_swapped = 0;
}
#endif
/*! \brief The container for the voicemail greeter provider */
static AO2_GLOBAL_OBJ_STATIC(vm_greeter_provider);
/*! Voicemail greeter not registered warning */
static int vm_greeter_warnings;
int ast_vm_greeter_is_registered(void)
{
struct ast_vm_greeter_functions *table;
int is_registered;
table = ao2_global_obj_ref(vm_greeter_provider);
is_registered = table ? 1 : 0;
ao2_cleanup(table);
return is_registered;
}
int __ast_vm_greeter_register(const struct ast_vm_greeter_functions *vm_table, struct ast_module *module)
{
RAII_VAR(struct ast_vm_greeter_functions *, table, NULL, ao2_cleanup);
if (!vm_table->module_name) {
ast_log(LOG_ERROR, "Voicemail greeter provider missing required information.\n");
return -1;
}
if (vm_table->module_version != VM_GREETER_MODULE_VERSION) {
ast_log(LOG_ERROR, "Voicemail greeter provider '%s' has incorrect version\n",
vm_table->module_name);
return -1;
}
table = ao2_global_obj_ref(vm_greeter_provider);
if (table) {
ast_log(LOG_WARNING, "Voicemail greeter provider already registered by %s.\n",
table->module_name);
return AST_MODULE_LOAD_DECLINE;
}
table = ao2_alloc_options(sizeof(*table), NULL, AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!table) {
return -1;
}
*table = *vm_table;
table->module = module;
ao2_global_obj_replace_unref(vm_greeter_provider, table);
return 0;
}
void ast_vm_greeter_unregister(const char *module_name)
{
struct ast_vm_greeter_functions *table;
table = ao2_global_obj_ref(vm_greeter_provider);
if (table && !strcmp(table->module_name, module_name)) {
ao2_global_obj_release(vm_greeter_provider);
}
ao2_cleanup(table);
}
#ifdef TEST_FRAMEWORK
static ast_vm_test_create_user_fn *ast_vm_test_create_user_func = NULL;
static ast_vm_test_destroy_user_fn *ast_vm_test_destroy_user_func = NULL;
void ast_install_vm_test_functions(ast_vm_test_create_user_fn *vm_test_create_user_func,
ast_vm_test_destroy_user_fn *vm_test_destroy_user_func)
{
ast_vm_test_create_user_func = vm_test_create_user_func;
ast_vm_test_destroy_user_func = vm_test_destroy_user_func;
}
void ast_uninstall_vm_test_functions(void)
{
ast_vm_test_create_user_func = NULL;
ast_vm_test_destroy_user_func = NULL;
}
#endif
static void vm_warn_no_provider(void)
{
if (vm_warnings++ % 10 == 0) {
ast_verb(3, "No voicemail provider registered.\n");
}
}
#define VM_API_CALL(res, api_call, api_parms) \
do { \
struct ast_vm_functions *table; \
table = ao2_global_obj_ref(vm_provider); \
if (!table) { \
vm_warn_no_provider(); \
} else if (table->api_call) { \
ast_module_ref(table->module); \
(res) = table->api_call api_parms; \
ast_module_unref(table->module); \
} \
ao2_cleanup(table); \
} while (0)
static void vm_greeter_warn_no_provider(void)
{
if (vm_greeter_warnings++ % 10 == 0) {
ast_verb(3, "No voicemail greeter provider registered.\n");
}
}
#define VM_GREETER_API_CALL(res, api_call, api_parms) \
do { \
struct ast_vm_greeter_functions *table; \
table = ao2_global_obj_ref(vm_greeter_provider); \
if (!table) { \
vm_greeter_warn_no_provider(); \
} else if (table->api_call) { \
ast_module_ref(table->module); \
(res) = table->api_call api_parms; \
ast_module_unref(table->module); \
} \
ao2_cleanup(table); \
} while (0)
int ast_app_has_voicemail(const char *mailboxes, const char *folder)
{
int res = 0;
VM_API_CALL(res, has_voicemail, (mailboxes, folder));
return res;
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/*!
* \internal
* \brief Function used as a callback for ast_copy_recording_to_vm when a real one isn't installed.
* \param vm_rec_data Stores crucial information about the voicemail that will basically just be used
* to figure out what the name of the recipient was supposed to be
*/
int ast_app_copy_recording_to_vm(struct ast_vm_recording_data *vm_rec_data)
{
int res = -1;
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
VM_API_CALL(res, copy_recording_to_vm, (vm_rec_data));
return res;
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
}
int ast_app_inboxcount(const char *mailboxes, int *newmsgs, int *oldmsgs)
{
int res = 0;
if (newmsgs) {
*newmsgs = 0;
}
if (oldmsgs) {
*oldmsgs = 0;
}
VM_API_CALL(res, inboxcount, (mailboxes, newmsgs, oldmsgs));
return res;
}
int ast_app_inboxcount2(const char *mailboxes, int *urgentmsgs, int *newmsgs, int *oldmsgs)
{
int res = 0;
if (newmsgs) {
*newmsgs = 0;
}
if (oldmsgs) {
*oldmsgs = 0;
}
if (urgentmsgs) {
*urgentmsgs = 0;
}
VM_API_CALL(res, inboxcount2, (mailboxes, urgentmsgs, newmsgs, oldmsgs));
return res;
}
int ast_app_sayname(struct ast_channel *chan, const char *mailbox_id)
{
int res = -1;
VM_GREETER_API_CALL(res, sayname, (chan, mailbox_id));
return res;
}
int ast_app_messagecount(const char *mailbox_id, const char *folder)
{
int res = 0;
VM_API_CALL(res, messagecount, (mailbox_id, folder));
return res;
}
const char *ast_vm_index_to_foldername(int id)
{
const char *res = NULL;
VM_API_CALL(res, index_to_foldername, (id));
return res;
}
struct ast_vm_mailbox_snapshot *ast_vm_mailbox_snapshot_create(const char *mailbox,
const char *context,
const char *folder,
int descending,
enum ast_vm_snapshot_sort_val sort_val,
int combine_INBOX_and_OLD)
{
struct ast_vm_mailbox_snapshot *res = NULL;
VM_API_CALL(res, mailbox_snapshot_create, (mailbox, context, folder, descending,
sort_val, combine_INBOX_and_OLD));
return res;
}
struct ast_vm_mailbox_snapshot *ast_vm_mailbox_snapshot_destroy(struct ast_vm_mailbox_snapshot *mailbox_snapshot)
{
struct ast_vm_mailbox_snapshot *res = NULL;
VM_API_CALL(res, mailbox_snapshot_destroy, (mailbox_snapshot));
return res;
}
int ast_vm_msg_move(const char *mailbox,
const char *context,
size_t num_msgs,
const char *oldfolder,
const char *old_msg_ids[],
const char *newfolder)
{
int res = 0;
VM_API_CALL(res, msg_move, (mailbox, context, num_msgs, oldfolder, old_msg_ids,
newfolder));
return res;
}
int ast_vm_msg_remove(const char *mailbox,
const char *context,
size_t num_msgs,
const char *folder,
const char *msgs[])
{
int res = 0;
VM_API_CALL(res, msg_remove, (mailbox, context, num_msgs, folder, msgs));
return res;
}
int ast_vm_msg_forward(const char *from_mailbox,
const char *from_context,
const char *from_folder,
const char *to_mailbox,
const char *to_context,
const char *to_folder,
size_t num_msgs,
const char *msg_ids[],
int delete_old)
{
int res = 0;
VM_API_CALL(res, msg_forward, (from_mailbox, from_context, from_folder, to_mailbox,
to_context, to_folder, num_msgs, msg_ids, delete_old));
return res;
}
int ast_vm_msg_play(struct ast_channel *chan,
const char *mailbox,
const char *context,
const char *folder,
const char *msg_num,
ast_vm_msg_play_cb *cb)
{
int res = 0;
VM_API_CALL(res, msg_play, (chan, mailbox, context, folder, msg_num, cb));
return res;
}
#ifdef TEST_FRAMEWORK
int ast_vm_test_create_user(const char *context, const char *mailbox)
{
if (ast_vm_test_create_user_func) {
return ast_vm_test_create_user_func(context, mailbox);
}
return 0;
}
int ast_vm_test_destroy_user(const char *context, const char *mailbox)
{
if (ast_vm_test_destroy_user_func) {
return ast_vm_test_destroy_user_func(context, mailbox);
}
return 0;
}
#endif
static int external_sleep(struct ast_channel *chan, int ms)
{
usleep(ms * 1000);
return 0;
}
static int sf_stream(struct ast_channel *chan, struct ast_channel *chan2, const char *digits, int frequency, int is_external)
{
/* Bell System Technical Journal 39 (Nov. 1960) */
#define SF_ON 67
#define SF_OFF 33
#define SF_BETWEEN 600
const char *ptr;
int res;
struct ast_silence_generator *silgen = NULL, *silgen2 = NULL;
char *freq;
int (*my_sleep)(struct ast_channel *chan, int ms);
if (frequency >= 100000) {
ast_log(LOG_WARNING, "Frequency too large: %d\n", frequency);
return -1;
}
if (is_external) {
my_sleep = external_sleep;
} else {
my_sleep = ast_safe_sleep;
}
/* Need a quiet time before sending digits. */
if (ast_opt_transmit_silence) {
silgen = ast_channel_start_silence_generator(chan);
if (chan2) {
silgen2 = ast_channel_start_silence_generator(chan2);
}
}
if (chan2) {
ast_autoservice_start(chan2);
}
res = my_sleep(chan, 100);
if (chan2) {
ast_autoservice_stop(chan2);
}
if (res) {
goto sf_stream_cleanup;
}
/* len(SF_ON) + len(SF_OFF) + len(0) + maxlen(frequency) + /,/ + null terminator = 2 + 2 + 1 + 5 at most + 3 + 1 = 14 */
#define SF_BUF_LEN 20
freq = ast_alloca(SF_BUF_LEN); /* min 20 to avoid compiler warning about insufficient buffer */
/* pauses need to send audio, so send 0 Hz */
snprintf(freq, SF_BUF_LEN, "%d/%d,%d/%d", frequency, SF_ON, 0, SF_OFF);
for (ptr = digits; *ptr; ptr++) {
if (*ptr == 'w') {
/* 'w' -- wait half a second */
if (chan2) {
ast_autoservice_start(chan2);
}
res = my_sleep(chan, 500);
if (chan2) {
ast_autoservice_stop(chan2);
}
if (res) {
break;
}
} else if (*ptr == 'h' || *ptr == 'H') {
/* 'h' -- 2600 Hz for half a second, but
only to far end of trunk, not near end */
ast_playtones_start(chan, 0, "2600", 0);
if (chan2) {
ast_playtones_start(chan2, 0, "0", 0);
ast_autoservice_start(chan2);
}
res = my_sleep(chan, 250);
ast_senddigit_mf_end(chan);
if (chan2) {
ast_autoservice_stop(chan2);
ast_senddigit_mf_end(chan2);
}
if (res) {
break;
}
} else if (strchr("0123456789*#ABCDabcdwWfF", *ptr)) {
if (*ptr == 'f' || *ptr == 'F') {
/* ignore return values if not supported by channel */
ast_indicate(chan, AST_CONTROL_FLASH);
} else if (*ptr == 'W') {
/* ignore return values if not supported by channel */
ast_indicate(chan, AST_CONTROL_WINK);
} else {
/* Character represents valid SF */
int beeps;
if (*ptr == '*') {
beeps = 11;
} else if (*ptr == '#') {
beeps = 12;
} else if (*ptr == 'D') {
beeps = 13;
} else if (*ptr == 'C') {
beeps = 14;
} else if (*ptr == 'B') {
beeps = 15;
} else if (*ptr == 'A') {
beeps = 16;
} else {
beeps = (*ptr == '0') ? 10 : *ptr - '0';
}
while (beeps-- > 0) {
ast_playtones_start(chan, 0, freq, 0);
if (chan2) {
ast_playtones_start(chan2, 0, freq, 0);
ast_autoservice_start(chan2);
}
res = my_sleep(chan, SF_ON + SF_OFF);
ast_senddigit_mf_end(chan);
if (chan2) {
ast_autoservice_stop(chan2);
ast_senddigit_mf_end(chan2);
}
if (res) {
break;
}
}
}
/* pause between digits */
ast_playtones_start(chan, 0, "0", 0);
if (chan2) {
ast_playtones_start(chan2, 0, "0", 0);
ast_autoservice_start(chan2);
}
res = my_sleep(chan, SF_BETWEEN);
if (chan2) {
ast_autoservice_stop(chan2);
ast_senddigit_mf_end(chan2);
}
ast_senddigit_mf_end(chan);
if (res) {
break;
}
} else {
ast_log(LOG_WARNING, "Illegal SF character '%c' in string. (0-9A-DwWfFhH allowed)\n", *ptr);
}
}
sf_stream_cleanup:
if (silgen) {
ast_channel_stop_silence_generator(chan, silgen);
}
if (silgen2) {
ast_channel_stop_silence_generator(chan2, silgen2);
}
return res;
}
static int mf_stream(struct ast_channel *chan, struct ast_channel *chan2, const char *digits, int between, unsigned int duration,
unsigned int durationkp, unsigned int durationst, int is_external)
{
const char *ptr;
int res;
struct ast_silence_generator *silgen = NULL, *silgen2 = NULL;
int (*my_sleep)(struct ast_channel *chan, int ms);
if (is_external) {
my_sleep = external_sleep;
} else {
my_sleep = ast_safe_sleep;
}
if (!between) {
between = 100;
}
/* Need a quiet time before sending digits. */
if (ast_opt_transmit_silence) {
silgen = ast_channel_start_silence_generator(chan);
if (chan2) {
silgen2 = ast_channel_start_silence_generator(chan2);
}
}
if (chan2) {
ast_autoservice_start(chan2);
}
res = my_sleep(chan, 100);
if (chan2) {
ast_autoservice_stop(chan2);
}
if (res) {
goto mf_stream_cleanup;
}
for (ptr = digits; *ptr; ptr++) {
if (*ptr == 'w') {
/* 'w' -- wait half a second */
if (chan2) {
ast_autoservice_start(chan2);
}
res = my_sleep(chan, 500);
if (chan2) {
ast_autoservice_stop(chan2);
}
if (res) {
break;
}
} else if (*ptr == 'h' || *ptr == 'H') {
/* 'h' -- 2600 Hz for half a second, but
only to far end of trunk, not near end */
ast_playtones_start(chan, 0, "2600", 0);
if (chan2) {
ast_playtones_start(chan2, 0, "0", 0);
ast_autoservice_start(chan2);
}
res = my_sleep(chan, 250);
ast_senddigit_mf_end(chan);
if (chan2) {
ast_autoservice_stop(chan2);
ast_senddigit_mf_end(chan2);
}
if (res) {
break;
}
} else if (strchr("0123456789*#ABCwWfF", *ptr)) {
if (*ptr == 'f' || *ptr == 'F') {
/* ignore return values if not supported by channel */
ast_indicate(chan, AST_CONTROL_FLASH);
} else if (*ptr == 'W') {
/* ignore return values if not supported by channel */
ast_indicate(chan, AST_CONTROL_WINK);
} else {
/* Character represents valid MF */
ast_senddigit_mf(chan, *ptr, duration, durationkp, durationst, is_external);
if (chan2) {
ast_senddigit_mf(chan2, *ptr, duration, durationkp, durationst, is_external);
}
}
/* pause between digits */
/* The DSP code in Asterisk does not currently properly receive repeated tones
if no audio is sent in the middle. Simply sending audio (even 0 Hz)
works around this limitation and guarantees the correct behavior.
*/
ast_playtones_start(chan, 0, "0", 0);
if (chan2) {
ast_playtones_start(chan2, 0, "0", 0);
ast_autoservice_start(chan2);
}
res = my_sleep(chan, between);
ast_senddigit_mf_end(chan);
if (chan2) {
ast_autoservice_stop(chan2);
ast_senddigit_mf_end(chan2);
}
if (res) {
break;
}
} else {
ast_log(LOG_WARNING, "Illegal MF character '%c' in string. (0-9*#ABCwWfFhH allowed)\n", *ptr);
}
}
mf_stream_cleanup:
if (silgen) {
ast_channel_stop_silence_generator(chan, silgen);
}
if (silgen2) {
ast_channel_stop_silence_generator(chan2, silgen2);
}
return res;
}
static int dtmf_stream(struct ast_channel *chan, const char *digits, int between, unsigned int duration, int is_external)
{
const char *ptr;
int res;
struct ast_silence_generator *silgen = NULL;
int (*my_sleep)(struct ast_channel *chan, int ms);
int (*my_senddigit)(struct ast_channel *chan, char digit, unsigned int duration);
if (is_external) {
my_sleep = external_sleep;
my_senddigit = ast_senddigit_external;
} else {
my_sleep = ast_safe_sleep;
my_senddigit = ast_senddigit;
}
if (!between) {
between = 100;
}
/* Need a quiet time before sending digits. */
if (ast_opt_transmit_silence) {
silgen = ast_channel_start_silence_generator(chan);
}
res = my_sleep(chan, 100);
if (res) {
goto dtmf_stream_cleanup;
}
for (ptr = digits; *ptr; ptr++) {
if (*ptr == 'w') {
/* 'w' -- wait half a second */
res = my_sleep(chan, 500);
if (res) {
break;
}
} else if (*ptr == 'W') {
/* 'W' -- wait a second */
res = my_sleep(chan, 1000);
if (res) {
break;
}
} else if (strchr("0123456789*#abcdfABCDF", *ptr)) {
if (*ptr == 'f' || *ptr == 'F') {
/* ignore return values if not supported by channel */
ast_indicate(chan, AST_CONTROL_FLASH);
} else {
/* Character represents valid DTMF */
my_senddigit(chan, *ptr, duration);
}
/* pause between digits */
res = my_sleep(chan, between);
if (res) {
break;
}
} else {
ast_log(LOG_WARNING, "Illegal DTMF character '%c' in string. (0-9*#aAbBcCdD allowed)\n", *ptr);
}
}
dtmf_stream_cleanup:
if (silgen) {
ast_channel_stop_silence_generator(chan, silgen);
}
return res;
}
int ast_sf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits, int frequency, int is_external)
{
int res;
if (frequency <= 0) {
frequency = 2600;
}
if (!is_external && !chan2 && peer && ast_autoservice_start(peer)) {
return -1;
}
res = sf_stream(chan, chan2, digits, frequency, is_external);
if (!is_external && !chan2 && peer && ast_autoservice_stop(peer)) {
res = -1;
}
return res;
}
int ast_mf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits,
int between, unsigned int duration, unsigned int durationkp, unsigned int durationst, int is_external)
{
int res;
if (!is_external && !chan2 && peer && ast_autoservice_start(peer)) {
return -1;
}
res = mf_stream(chan, chan2, digits, between, duration, durationkp, durationst, is_external);
if (!is_external && !chan2 && peer && ast_autoservice_stop(peer)) {
res = -1;
}
return res;
}
int ast_dtmf_stream(struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between, unsigned int duration)
{
int res;
if (peer && ast_autoservice_start(peer)) {
return -1;
}
res = dtmf_stream(chan, digits, between, duration, 0);
if (peer && ast_autoservice_stop(peer)) {
res = -1;
}
return res;
}
void ast_dtmf_stream_external(struct ast_channel *chan, const char *digits, int between, unsigned int duration)
{
dtmf_stream(chan, digits, between, duration, 1);
}
struct linear_state {
int fd;
int autoclose;
int allowoverride;
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
2014-07-20 22:06:33 +00:00
struct ast_format *origwfmt;
};
static void linear_release(struct ast_channel *chan, void *params)
{
struct linear_state *ls = params;
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
2014-07-20 22:06:33 +00:00
if (ls->origwfmt && ast_set_write_format(chan, ls->origwfmt)) {
ast_log(LOG_WARNING, "Unable to restore channel '%s' to format '%s'\n",
ast_channel_name(chan), ast_format_get_name(ls->origwfmt));
}
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
2014-07-20 22:06:33 +00:00
ao2_cleanup(ls->origwfmt);
if (ls->autoclose) {
close(ls->fd);
}
ast_free(params);
}
static int linear_generator(struct ast_channel *chan, void *data, int len, int samples)
{
short buf[2048 + AST_FRIENDLY_OFFSET / 2];
struct linear_state *ls = data;
struct ast_frame f = {
.frametype = AST_FRAME_VOICE,
.data.ptr = buf + AST_FRIENDLY_OFFSET / 2,
.offset = AST_FRIENDLY_OFFSET,
};
int res;
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
2014-07-20 22:06:33 +00:00
f.subclass.format = ast_format_slin;
len = samples * 2;
if (len > sizeof(buf) - AST_FRIENDLY_OFFSET) {
ast_log(LOG_WARNING, "Can't generate %d bytes of data!\n" , len);
len = sizeof(buf) - AST_FRIENDLY_OFFSET;
}
res = read(ls->fd, buf + AST_FRIENDLY_OFFSET/2, len);
if (res > 0) {
f.datalen = res;
f.samples = res / 2;
ast_write(chan, &f);
if (res == len) {
return 0;
}
}
return -1;
}
static void *linear_alloc(struct ast_channel *chan, void *params)
{
struct linear_state *ls = params;
if (!params) {
return NULL;
}
/* In this case, params is already malloc'd */
if (ls->allowoverride) {
ast_set_flag(ast_channel_flags(chan), AST_FLAG_WRITE_INT);
} else {
ast_clear_flag(ast_channel_flags(chan), AST_FLAG_WRITE_INT);
}
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
2014-07-20 22:06:33 +00:00
ls->origwfmt = ao2_bump(ast_channel_writeformat(chan));
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
2014-07-20 22:06:33 +00:00
if (ast_set_write_format(chan, ast_format_slin)) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, "Unable to set '%s' to linear format (write)\n", ast_channel_name(chan));
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
2014-07-20 22:06:33 +00:00
ao2_cleanup(ls->origwfmt);
ast_free(ls);
ls = params = NULL;
}
return params;
}
static struct ast_generator linearstream =
{
.alloc = linear_alloc,
.release = linear_release,
.generate = linear_generator,
};
int ast_linear_stream(struct ast_channel *chan, const char *filename, int fd, int allowoverride)
{
struct linear_state *lin;
char tmpf[256];
int autoclose = 0;
if (fd < 0) {
if (ast_strlen_zero(filename)) {
return -1;
}
autoclose = 1;
if (filename[0] == '/') {
ast_copy_string(tmpf, filename, sizeof(tmpf));
} else {
snprintf(tmpf, sizeof(tmpf), "%s/%s/%s", ast_config_AST_DATA_DIR, "sounds", filename);
}
fd = open(tmpf, O_RDONLY);
if (fd < 0) {
ast_log(LOG_WARNING, "Unable to open file '%s': %s\n", tmpf, strerror(errno));
return -1;
}
}
lin = ast_calloc(1, sizeof(*lin));
if (!lin) {
if (autoclose) {
close(fd);
}
return -1;
}
lin->fd = fd;
lin->allowoverride = allowoverride;
lin->autoclose = autoclose;
return ast_activate_generator(chan, &linearstream, lin);
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
static int control_streamfile(struct ast_channel *chan,
const char *file,
const char *fwd,
const char *rev,
const char *stop,
const char *suspend,
const char *restart,
int skipms,
long *offsetms,
const char *lang,
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
ast_waitstream_fr_cb cb)
{
char *breaks = NULL;
char *end = NULL;
int blen = 2;
int res;
long pause_restart_point = 0;
long offset = 0;
struct ast_silence_generator *silgen = NULL;
if (!file) {
return -1;
}
if (offsetms) {
offset = *offsetms * 8; /* XXX Assumes 8kHz */
}
if (lang == NULL) {
lang = ast_channel_language(chan);
}
if (stop) {
blen += strlen(stop);
}
if (suspend) {
blen += strlen(suspend);
}
if (restart) {
blen += strlen(restart);
}
if (blen > 2) {
breaks = ast_alloca(blen + 1);
breaks[0] = '\0';
if (stop) {
strcat(breaks, stop);
}
if (suspend) {
strcat(breaks, suspend);
}
if (restart) {
strcat(breaks, restart);
}
}
if ((end = strchr(file, ':'))) {
if (!strcasecmp(end, ":end")) {
*end = '\0';
end++;
} else {
end = NULL;
}
}
for (;;) {
ast_stopstream(chan);
res = ast_streamfile(chan, file, lang);
if (!res) {
if (pause_restart_point) {
ast_seekstream(ast_channel_stream(chan), pause_restart_point, SEEK_SET);
pause_restart_point = 0;
}
else if (end || offset < 0) {
if (offset == -8) {
offset = 0;
}
ast_verb(3, "ControlPlayback seek to offset %ld from end\n", offset);
ast_seekstream(ast_channel_stream(chan), offset, SEEK_END);
end = NULL;
offset = 0;
} else if (offset) {
ast_verb(3, "ControlPlayback seek to offset %ld\n", offset);
ast_seekstream(ast_channel_stream(chan), offset, SEEK_SET);
offset = 0;
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
if (cb) {
res = ast_waitstream_fr_w_cb(chan, breaks, fwd, rev, skipms, cb);
} else {
res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
}
}
if (res < 1) {
break;
}
/* We go at next loop if we got the restart char */
if ((restart && strchr(restart, res)) || res == AST_CONTROL_STREAM_RESTART) {
ast_debug(1, "we'll restart the stream here at next loop\n");
pause_restart_point = 0;
ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
"Control: %s\r\n",
ast_channel_name(chan),
"Restart");
continue;
}
if ((suspend && strchr(suspend, res)) || res == AST_CONTROL_STREAM_SUSPEND) {
pause_restart_point = ast_tellstream(ast_channel_stream(chan));
if (ast_opt_transmit_silence) {
silgen = ast_channel_start_silence_generator(chan);
}
ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
"Control: %s\r\n",
ast_channel_name(chan),
"Pause");
for (;;) {
ast_stopstream(chan);
if (!(res = ast_waitfordigit(chan, 1000))) {
continue;
} else if (res == -1 || (suspend && strchr(suspend, res)) || (stop && strchr(stop, res))
|| res == AST_CONTROL_STREAM_SUSPEND || res == AST_CONTROL_STREAM_STOP) {
break;
}
}
if (silgen) {
ast_channel_stop_silence_generator(chan, silgen);
silgen = NULL;
}
if ((suspend && (res == *suspend)) || res == AST_CONTROL_STREAM_SUSPEND) {
res = 0;
ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
"Control: %s\r\n",
ast_channel_name(chan),
"Unpause");
continue;
}
}
if (res == -1) {
break;
}
/* if we get one of our stop chars, return it to the calling function */
if ((stop && strchr(stop, res)) || res == AST_CONTROL_STREAM_STOP) {
ast_test_suite_event_notify("PLAYBACK","Channel: %s\r\n"
"Control: %s\r\n",
ast_channel_name(chan),
"Stop");
break;
}
}
if (pause_restart_point) {
offset = pause_restart_point;
} else {
if (ast_channel_stream(chan)) {
offset = ast_tellstream(ast_channel_stream(chan));
} else {
offset = -8; /* indicate end of file */
}
}
if (offsetms) {
*offsetms = offset / 8; /* samples --> ms ... XXX Assumes 8 kHz */
}
ast_stopstream(chan);
return res;
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
int ast_control_streamfile_w_cb(struct ast_channel *chan,
const char *file,
const char *fwd,
const char *rev,
const char *stop,
const char *suspend,
const char *restart,
int skipms,
long *offsetms,
ast_waitstream_fr_cb cb)
{
return control_streamfile(chan, file, fwd, rev, stop, suspend, restart, skipms, offsetms, NULL, cb);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
}
int ast_control_streamfile(struct ast_channel *chan, const char *file,
const char *fwd, const char *rev,
const char *stop, const char *suspend,
const char *restart, int skipms, long *offsetms)
{
return control_streamfile(chan, file, fwd, rev, stop, suspend, restart, skipms, offsetms, NULL, NULL);
}
int ast_control_streamfile_lang(struct ast_channel *chan, const char *file,
const char *fwd, const char *rev, const char *stop, const char *suspend,
const char *restart, int skipms, const char *lang, long *offsetms)
{
return control_streamfile(chan, file, fwd, rev, stop, suspend, restart, skipms, offsetms, lang, NULL);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
}
enum control_tone_frame_response_result {
CONTROL_TONE_RESPONSE_FAILED = -1,
CONTROL_TONE_RESPONSE_NORMAL = 0,
CONTROL_TONE_RESPONSE_FINISHED = 1,
};
static enum control_tone_frame_response_result control_tone_frame_response(struct ast_channel *chan, struct ast_frame *fr, struct ast_tone_zone_sound *ts, const char *tone, int *paused)
{
switch (fr->subclass.integer) {
case AST_CONTROL_STREAM_STOP:
ast_playtones_stop(chan);
return CONTROL_TONE_RESPONSE_FINISHED;
case AST_CONTROL_STREAM_SUSPEND:
if (*paused) {
*paused = 0;
if (ast_playtones_start(chan, 0, ts ? ts->data : tone, 0)) {
return CONTROL_TONE_RESPONSE_FAILED;
}
} else {
*paused = 1;
ast_playtones_stop(chan);
}
return CONTROL_TONE_RESPONSE_NORMAL;
case AST_CONTROL_STREAM_RESTART:
ast_playtones_stop(chan);
if (ast_playtones_start(chan, 0, ts ? ts->data : tone, 0)) {
return CONTROL_TONE_RESPONSE_FAILED;
}
return CONTROL_TONE_RESPONSE_NORMAL;
case AST_CONTROL_STREAM_REVERSE:
ast_log(LOG_NOTICE, "Media control operation 'reverse' not supported for media type 'tone'\n");
return CONTROL_TONE_RESPONSE_NORMAL;
case AST_CONTROL_STREAM_FORWARD:
ast_log(LOG_NOTICE, "Media control operation 'forward' not supported for media type 'tone'\n");
return CONTROL_TONE_RESPONSE_NORMAL;
case AST_CONTROL_HANGUP:
case AST_CONTROL_BUSY:
case AST_CONTROL_CONGESTION:
return CONTROL_TONE_RESPONSE_FINISHED;
}
return CONTROL_TONE_RESPONSE_NORMAL;
}
static int parse_tone_uri(char *tone_parser,
const char **tone_indication,
const char **tone_zone)
{
*tone_indication = strsep(&tone_parser, ";");
if (ast_strlen_zero(tone_parser)) {
/* Only the indication is included */
return 0;
}
if (!(strncmp(tone_parser, "tonezone=", 9))) {
*tone_zone = tone_parser + 9;
} else {
ast_log(LOG_ERROR, "Unexpected Tone URI component: %s\n", tone_parser);
return -1;
}
return 0;
}
int ast_control_tone(struct ast_channel *chan, const char *tone)
{
struct ast_tone_zone *zone = NULL;
struct ast_tone_zone_sound *ts;
int paused = 0;
int res = 0;
const char *tone_indication = NULL;
const char *tone_zone = NULL;
char *tone_uri_parser;
if (ast_strlen_zero(tone)) {
return -1;
}
tone_uri_parser = ast_strdupa(tone);
if (parse_tone_uri(tone_uri_parser, &tone_indication, &tone_zone)) {
return -1;
}
if (tone_zone) {
zone = ast_get_indication_zone(tone_zone);
}
ts = ast_get_indication_tone(zone ? zone : ast_channel_zone(chan), tone_indication);
if (ast_playtones_start(chan, 0, ts ? ts->data : tone_indication, 0)) {
res = -1;
}
while (!res) {
struct ast_frame *fr;
if (ast_waitfor(chan, -1) < 0) {
res = -1;
break;
}
fr = ast_read_noaudio(chan);
if (!fr) {
res = -1;
break;
}
if (fr->frametype != AST_FRAME_CONTROL) {
continue;
}
res = control_tone_frame_response(chan, fr, ts, tone_indication, &paused);
if (res == CONTROL_TONE_RESPONSE_FINISHED) {
res = 0;
break;
} else if (res == CONTROL_TONE_RESPONSE_FAILED) {
res = -1;
break;
}
}
if (ts) {
ast_tone_zone_sound_unref(ts);
}
if (zone) {
ast_tone_zone_unref(zone);
}
return res;
}
int ast_play_and_wait(struct ast_channel *chan, const char *fn)
{
int d = 0;
if ((d = ast_streamfile(chan, fn, ast_channel_language(chan)))) {
return d;
}
d = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
return d;
}
/*!
* \brief Construct a silence frame of the same duration as \a orig.
*
* The \a orig frame must be \ref ast_format_slin.
*
* \param orig Frame as basis for silence to generate.
* \return New frame of silence; free with ast_frfree().
* \retval NULL on error.
*/
static struct ast_frame *make_silence(const struct ast_frame *orig)
{
struct ast_frame *silence;
size_t size;
size_t datalen;
size_t samples = 0;
if (!orig) {
return NULL;
}
do {
if (ast_format_cmp(orig->subclass.format, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
ast_log(LOG_WARNING, "Attempting to silence non-slin frame\n");
return NULL;
}
samples += orig->samples;
orig = AST_LIST_NEXT(orig, frame_list);
} while (orig);
ast_verb(4, "Silencing %zu samples\n", samples);
datalen = sizeof(short) * samples;
size = sizeof(*silence) + datalen;
silence = ast_calloc(1, size);
if (!silence) {
return NULL;
}
silence->mallocd = AST_MALLOCD_HDR;
silence->frametype = AST_FRAME_VOICE;
silence->data.ptr = (void *)(silence + 1);
silence->samples = samples;
silence->datalen = datalen;
silence->subclass.format = ao2_bump(ast_format_slin);
return silence;
}
/*!
* \brief Sets a channel's read format to \ref ast_format_slin, recording
* its original format.
*
* \param chan Channel to modify.
* \param[out] orig_format Output variable to store channel's original read
* format.
* \return 0 on success.
* \return -1 on error.
*/
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
2014-07-20 22:06:33 +00:00
static int set_read_to_slin(struct ast_channel *chan, struct ast_format **orig_format)
{
if (!chan || !orig_format) {
return -1;
}
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
2014-07-20 22:06:33 +00:00
*orig_format = ao2_bump(ast_channel_readformat(chan));
return ast_set_read_format(chan, ast_format_slin);
}
static int global_silence_threshold = 128;
static int global_maxsilence = 0;
/*! Optionally play a sound file or a beep, then record audio and video from the channel.
* \param chan Channel to playback to/record from.
* \param playfile Filename of sound to play before recording begins.
* \param recordfile Filename to record to.
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
* \param maxtime Maximum length of recording (in seconds).
* \param fmt Format(s) to record message in. Multiple formats may be specified by separating them with a '|'.
* \param duration Where to store actual length of the recorded message (in milliseconds).
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
* \param sound_duration Where to store the length of the recorded message (in milliseconds), minus any silence
* \param beep Whether to play a beep before starting to record.
* \param silencethreshold
* \param maxsilence Length of silence that will end a recording (in milliseconds).
* \param path Optional filesystem path to unlock.
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
* \param prepend If true, prepend the recorded audio to an existing file and follow prepend mode recording rules
* \param acceptdtmf DTMF digits that will end the recording.
* \param canceldtmf DTMF digits that will cancel the recording.
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
* \param skip_confirmation_sound If true, don't play auth-thankyou at end. Nice for custom recording prompts in apps.
* \param if_exists
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
*
* \retval -1 failure or hangup
* \retval 'S' Recording ended from silence timeout
* \retval 't' Recording ended from the message exceeding the maximum duration, or via DTMF in prepend mode
* \retval dtmfchar Recording ended via the return value's DTMF character for either cancel or accept.
*/
static int __ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence, const char *path, int prepend, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
{
int d = 0;
char *fmts;
char comment[256];
int x, fmtcnt = 1, res = -1, outmsg = 0;
struct ast_filestream *others[AST_MAX_FORMATS];
const char *sfmt[AST_MAX_FORMATS];
char *stringp = NULL;
time_t start, end;
struct ast_dsp *sildet = NULL; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int olddspsilence = 0;
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
2014-07-20 22:06:33 +00:00
struct ast_format *rfmt = NULL;
struct ast_silence_generator *silgen = NULL;
char prependfile[PATH_MAX];
int ioflags; /* IO flags for writing output file */
ioflags = O_CREAT|O_WRONLY;
switch (if_exists) {
case AST_RECORD_IF_EXISTS_FAIL:
ioflags |= O_EXCL;
break;
case AST_RECORD_IF_EXISTS_OVERWRITE:
ioflags |= O_TRUNC;
break;
case AST_RECORD_IF_EXISTS_APPEND:
ioflags |= O_APPEND;
break;
clang compiler warnings: Fix autological comparisons This fixes autological comparison warnings in the following: * chan_skinny: letohl may return a signed or unsigned value, depending on the macro chosen * func_curl: Provide a specific cast to CURLoption to prevent mismatch * cel: Fix enum comparisons where the enum can never be negative * enum: Fix comparison of return result of dn_expand, which returns a signed int value * event: Fix enum comparisons where the enum can never be negative * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be negative * presencestate: Use the actual enum value for INVALID state * security_events: Fix enum comparisons where the enum can never be negative * udptl: Don't bother to check if the return value from encode_length is less than 0, as it returns an unsigned int * translate: Since the parameters are unsigned int, don't bother checking to see if they are negative. The cast to unsigned int would already blow past the matrix bounds. * res_pjsip_exten_state: Use a temporary value to cache the return of ast_hint_presence_state * res_stasis_playback: Fix enum comparisons where the enum can never be negative * res_stasis_recording: Add an enum value for the case where the recording operation is in error; fix enum comparisons * resource_bridges: Use enum value as opposed to -1 * resource_channels: Use enum value as opposed to -1 Review: https://reviewboard.asterisk.org/r/4533 ASTERISK-24917 Reported by: dkdegroot patches: rb4533.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 434470 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-04-09 12:57:21 +00:00
case AST_RECORD_IF_EXISTS_ERROR:
ast_assert(0);
break;
}
if (silencethreshold < 0) {
silencethreshold = global_silence_threshold;
}
if (maxsilence < 0) {
maxsilence = global_maxsilence;
}
/* barf if no pointer passed to store duration in */
if (!duration) {
ast_log(LOG_WARNING, "Error play_and_record called without duration pointer\n");
return -1;
}
ast_debug(1, "play_and_record: %s, %s, '%s'\n", playfile ? playfile : "<None>", recordfile, fmt);
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
snprintf(comment, sizeof(comment), "Playing %s, Recording to: %s on %s\n", playfile ? playfile : "<None>", recordfile, ast_channel_name(chan));
if (playfile || beep) {
if (!beep) {
d = ast_play_and_wait(chan, playfile);
}
if (d > -1) {
d = ast_stream_and_wait(chan, "beep", "");
}
if (d < 0) {
return -1;
}
}
if (prepend) {
ast_copy_string(prependfile, recordfile, sizeof(prependfile));
strncat(prependfile, "-prepend", sizeof(prependfile) - strlen(prependfile) - 1);
}
fmts = ast_strdupa(fmt);
stringp = fmts;
strsep(&stringp, "|");
ast_debug(1, "Recording Formats: sfmts=%s\n", fmts);
sfmt[0] = ast_strdupa(fmts);
while ((fmt = strsep(&stringp, "|"))) {
if (fmtcnt > AST_MAX_FORMATS - 1) {
ast_log(LOG_WARNING, "Please increase AST_MAX_FORMATS in file.h\n");
break;
}
/*
* Storage for 'fmt' is on the stack and held by 'fmts', which is maintained for
* the rest of this function. So okay to not duplicate 'fmt' here, but only keep
* a pointer to it.
*/
sfmt[fmtcnt++] = fmt;
}
end = start = time(NULL); /* pre-initialize end to be same as start in case we never get into loop */
for (x = 0; x < fmtcnt; x++) {
others[x] = ast_writefile(prepend ? prependfile : recordfile, sfmt[x], comment, ioflags, 0, AST_FILE_MODE);
ast_verb(3, "x=%d, open writing: %s format: %s, %p\n", x, prepend ? prependfile : recordfile, sfmt[x], others[x]);
if (!others[x]) {
break;
}
}
if (path) {
ast_unlock_path(path);
}
if (maxsilence > 0) {
sildet = ast_dsp_new(); /* Create the silence detector */
if (!sildet) {
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
return -1;
}
ast_dsp_set_threshold(sildet, silencethreshold);
res = set_read_to_slin(chan, &rfmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
ast_dsp_free(sildet);
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
2014-07-20 22:06:33 +00:00
ao2_cleanup(rfmt);
return -1;
}
}
if (!prepend) {
/* Request a video update */
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
if (ast_opt_transmit_silence) {
silgen = ast_channel_start_silence_generator(chan);
}
}
if (x == fmtcnt) {
/* Loop, writing the packets we read to the writer(s), until
* we have reason to stop. */
struct ast_frame *f;
int paused = 0;
int muted = 0;
time_t pause_start = 0;
int paused_secs = 0;
int pausedsilence = 0;
for (;;) {
if (!(res = ast_waitfor(chan, 2000))) {
ast_debug(1, "One waitfor failed, trying another\n");
/* Try one more time in case of masq */
if (!(res = ast_waitfor(chan, 2000))) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, "No audio available on %s??\n", ast_channel_name(chan));
res = -1;
}
}
if (res < 0) {
f = NULL;
break;
}
if (!(f = ast_read(chan))) {
break;
}
if (f->frametype == AST_FRAME_VOICE) {
/* write each format */
if (paused) {
/* It's all good */
res = 0;
} else {
struct ast_frame *silence = NULL;
struct ast_frame *orig = f;
if (muted) {
silence = make_silence(orig);
if (!silence) {
ast_log(LOG_WARNING, "Error creating silence\n");
break;
}
f = silence;
}
for (x = 0; x < fmtcnt; x++) {
if (prepend && !others[x]) {
break;
}
res = ast_writestream(others[x], f);
}
ast_frame_dtor(silence);
f = orig;
}
/* Silence Detection */
if (maxsilence > 0) {
dspsilence = 0;
ast_dsp_silence(sildet, f, &dspsilence);
if (olddspsilence > dspsilence) {
totalsilence += olddspsilence;
}
olddspsilence = dspsilence;
if (paused) {
/* record how much silence there was while we are paused */
pausedsilence = dspsilence;
} else if (dspsilence > pausedsilence) {
/* ignore the paused silence */
dspsilence -= pausedsilence;
} else {
/* dspsilence has reset, reset pausedsilence */
pausedsilence = 0;
}
if (dspsilence > maxsilence) {
/* Ended happily with silence */
ast_verb(3, "Recording automatically stopped after a silence of %d seconds\n", dspsilence/1000);
res = 'S';
outmsg = 2;
break;
}
}
/* Exit on any error */
if (res) {
ast_log(LOG_WARNING, "Error writing frame\n");
break;
}
} else if (f->frametype == AST_FRAME_VIDEO) {
/* Write only once */
ast_writestream(others[0], f);
} else if (f->frametype == AST_FRAME_DTMF) {
if (prepend) {
/* stop recording with any digit */
ast_verb(3, "User ended message by pressing %c\n", f->subclass.integer);
res = 't';
outmsg = 2;
break;
}
if (strchr(acceptdtmf, f->subclass.integer)) {
ast_verb(3, "User ended message by pressing %c\n", f->subclass.integer);
res = f->subclass.integer;
outmsg = 2;
break;
}
if (strchr(canceldtmf, f->subclass.integer)) {
ast_verb(3, "User canceled message by pressing %c\n", f->subclass.integer);
res = f->subclass.integer;
outmsg = 0;
break;
}
} else if (f->frametype == AST_FRAME_CONTROL) {
if (f->subclass.integer == AST_CONTROL_RECORD_CANCEL) {
ast_verb(3, "Message canceled by control\n");
outmsg = 0; /* cancels the recording */
res = 0;
break;
} else if (f->subclass.integer == AST_CONTROL_RECORD_STOP) {
ast_verb(3, "Message ended by control\n");
res = 0;
break;
} else if (f->subclass.integer == AST_CONTROL_RECORD_SUSPEND) {
paused = !paused;
ast_verb(3, "Message %spaused by control\n",
paused ? "" : "un");
if (paused) {
pause_start = time(NULL);
} else {
paused_secs += time(NULL) - pause_start;
}
} else if (f->subclass.integer == AST_CONTROL_RECORD_MUTE) {
muted = !muted;
ast_verb(3, "Message %smuted by control\n",
muted ? "" : "un");
/* We can only silence slin frames, so
* set the mode, if we haven't already
* for sildet
*/
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
2014-07-20 22:06:33 +00:00
if (muted && !rfmt) {
ast_verb(3, "Setting read format to linear mode\n");
res = set_read_to_slin(chan, &rfmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
break;
}
}
}
}
if (maxtime && !paused) {
end = time(NULL);
if (maxtime < (end - start - paused_secs)) {
ast_verb(3, "Took too long, cutting it short...\n");
res = 't';
outmsg = 2;
break;
}
}
ast_frfree(f);
}
if (!f) {
ast_verb(3, "User hung up\n");
res = -1;
outmsg = 1;
} else {
ast_frfree(f);
}
} else {
ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", recordfile, sfmt[x]);
}
if (!prepend) {
if (silgen) {
ast_channel_stop_silence_generator(chan, silgen);
}
}
/*!\note
* Instead of asking how much time passed (end - start), calculate the number
* of seconds of audio which actually went into the file. This fixes a
* problem where audio is stopped up on the network and never gets to us.
*
* Note that we still want to use the number of seconds passed for the max
* message, otherwise we could get a situation where this stream is never
* closed (which would create a resource leak).
*/
*duration = others[0] ? ast_tellstream(others[0]) / 8000 : 0;
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
if (sound_duration) {
*sound_duration = *duration;
}
if (!prepend) {
/* Reduce duration by a total silence amount */
if (olddspsilence <= dspsilence) {
totalsilence += dspsilence;
}
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
if (sound_duration) {
if (totalsilence > 0) {
*sound_duration -= (totalsilence - 200) / 1000;
}
if (*sound_duration < 0) {
*sound_duration = 0;
}
}
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
if (dspsilence > 0) {
*duration -= (dspsilence - 200) / 1000;
}
if (*duration < 0) {
*duration = 0;
}
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
for (x = 0; x < fmtcnt; x++) {
if (!others[x]) {
break;
}
/*!\note
* If we ended with silence, trim all but the first 200ms of silence
* off the recording. However, if we ended with '#', we don't want
* to trim ANY part of the recording.
*/
if (res > 0 && dspsilence) {
/* rewind only the trailing silence */
ast_stream_rewind(others[x], dspsilence - 200);
}
ast_truncstream(others[x]);
ast_closestream(others[x]);
}
} else if (prepend && outmsg) {
struct ast_filestream *realfiles[AST_MAX_FORMATS];
struct ast_frame *fr;
for (x = 0; x < fmtcnt; x++) {
snprintf(comment, sizeof(comment), "Opening the real file %s.%s\n", recordfile, sfmt[x]);
realfiles[x] = ast_readfile(recordfile, sfmt[x], comment, O_RDONLY, 0, 0);
if (!others[x]) {
break;
}
if (!realfiles[x]) {
ast_closestream(others[x]);
continue;
}
/*!\note Same logic as above. */
if (dspsilence) {
ast_stream_rewind(others[x], dspsilence - 200);
}
ast_truncstream(others[x]);
/* add the original file too */
while ((fr = ast_readframe(realfiles[x]))) {
ast_writestream(others[x], fr);
ast_frfree(fr);
}
ast_closestream(others[x]);
ast_closestream(realfiles[x]);
ast_filerename(prependfile, recordfile, sfmt[x]);
ast_verb(4, "Recording Format: sfmts=%s, prependfile %s, recordfile %s\n", sfmt[x], prependfile, recordfile);
ast_filedelete(prependfile, sfmt[x]);
}
} else {
for (x = 0; x < fmtcnt; x++) {
if (!others[x]) {
break;
}
ast_closestream(others[x]);
}
}
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
2014-07-20 22:06:33 +00:00
if (rfmt && ast_set_read_format(chan, rfmt)) {
ast_log(LOG_WARNING, "Unable to restore format %s to channel '%s'\n", ast_format_get_name(rfmt), ast_channel_name(chan));
}
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
2014-07-20 22:06:33 +00:00
ao2_cleanup(rfmt);
Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26 14:17:13 +00:00
if ((outmsg == 2) && (!skip_confirmation_sound)) {
ast_stream_and_wait(chan, "auth-thankyou", "");
}
if (sildet) {
ast_dsp_free(sildet);
}
return res;
}
static const char default_acceptdtmf[] = "#";
static const char default_canceldtmf[] = "";
int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
{
Remove undocumented default behavior of ast_play_and_record_full acceptdtmf. ast_play_and_record_full() has a parameter called "acceptdtmf" that is a string of acceptable DTMF digits that may be pressed by a caller to end and accept the recording. ARI uses this function in order to perform recording, and it provides options for what is passed as acceptdtmf to ast_play_and_record_full(). By default, ARI passes an empty string, with the intention that no DTMF can be used to end the recording. The problem is that ast_play_and_record_full() attempts to be "helpful" by setting "#" as the acceptdtmf if an empty string or NULL pointer has been passed in. With ARI, this results in unexpected behavior occurring if you have attempted to intercept "#" yourself in order to perform some other manipulation of the live recording. This change removes the "helpful" behavior by no longer accepting "#" as a default acceptdtmf if none is specified by the caller of ast_play_and_record_full(). This makes the ARI scenario work as expected. The other callers of ast_play_and_record_full() are app_voicemail and app_minivm, and in both cases, they pass an explicit "#" to ast_play_and_record_full() as acceptdtmf, so they are unaffected by this change. ........ Merged revisions 422964 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422965 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-11 22:17:39 +00:00
return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, ""), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
}
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence, const char *path)
{
return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold, maxsilence, path, 0, default_acceptdtmf, default_canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
}
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence)
{
return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, NULL, 1, default_acceptdtmf, default_canceldtmf, 1, AST_RECORD_IF_EXISTS_OVERWRITE);
}
/* Channel group core functions */
int ast_app_group_split_group(const char *data, char *group, int group_max, char *category, int category_max)
{
int res = 0;
char tmp[256];
char *grp = NULL, *cat = NULL;
if (!ast_strlen_zero(data)) {
ast_copy_string(tmp, data, sizeof(tmp));
grp = tmp;
if ((cat = strchr(tmp, '@'))) {
*cat++ = '\0';
}
}
if (!ast_strlen_zero(grp)) {
ast_copy_string(group, grp, group_max);
} else {
*group = '\0';
}
if (!ast_strlen_zero(cat)) {
ast_copy_string(category, cat, category_max);
}
return res;
}
int ast_app_group_set_channel(struct ast_channel *chan, const char *data)
{
int res = 0;
char group[80] = "", category[80] = "";
struct ast_group_info *gi = NULL;
size_t len = 0;
if (ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category))) {
return -1;
}
/* Calculate memory we will need if this is new */
len = sizeof(*gi) + strlen(group) + 1;
if (!ast_strlen_zero(category)) {
len += strlen(category) + 1;
}
AST_RWLIST_WRLOCK(&groups);
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&groups, gi, group_list) {
if ((gi->chan == chan) && ((ast_strlen_zero(category) && ast_strlen_zero(gi->category)) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category)))) {
AST_RWLIST_REMOVE_CURRENT(group_list);
ast_free(gi);
break;
}
}
AST_RWLIST_TRAVERSE_SAFE_END;
if (ast_strlen_zero(group)) {
/* Enable unsetting the group */
} else if ((gi = ast_calloc(1, len))) {
gi->chan = chan;
gi->group = (char *) gi + sizeof(*gi);
strcpy(gi->group, group);
if (!ast_strlen_zero(category)) {
gi->category = (char *) gi + sizeof(*gi) + strlen(group) + 1;
strcpy(gi->category, category);
}
AST_RWLIST_INSERT_TAIL(&groups, gi, group_list);
} else {
res = -1;
}
AST_RWLIST_UNLOCK(&groups);
return res;
}
int ast_app_group_get_count(const char *group, const char *category)
{
struct ast_group_info *gi = NULL;
int count = 0;
if (ast_strlen_zero(group)) {
return 0;
}
AST_RWLIST_RDLOCK(&groups);
AST_RWLIST_TRAVERSE(&groups, gi, group_list) {
if (!strcasecmp(gi->group, group) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category)))) {
count++;
}
}
AST_RWLIST_UNLOCK(&groups);
return count;
}
int ast_app_group_match_get_count(const char *groupmatch, const char *category)
{
struct ast_group_info *gi = NULL;
regex_t regexbuf_group;
regex_t regexbuf_category;
int count = 0;
if (ast_strlen_zero(groupmatch)) {
ast_log(LOG_NOTICE, "groupmatch empty\n");
return 0;
}
/* if regex compilation fails, return zero matches */
if (regcomp(&regexbuf_group, groupmatch, REG_EXTENDED | REG_NOSUB)) {
ast_log(LOG_ERROR, "Regex compile failed on: %s\n", groupmatch);
return 0;
}
if (!ast_strlen_zero(category) && regcomp(&regexbuf_category, category, REG_EXTENDED | REG_NOSUB)) {
ast_log(LOG_ERROR, "Regex compile failed on: %s\n", category);
regfree(&regexbuf_group);
return 0;
}
AST_RWLIST_RDLOCK(&groups);
AST_RWLIST_TRAVERSE(&groups, gi, group_list) {
if (!regexec(&regexbuf_group, gi->group, 0, NULL, 0) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !regexec(&regexbuf_category, gi->category, 0, NULL, 0)))) {
count++;
}
}
AST_RWLIST_UNLOCK(&groups);
regfree(&regexbuf_group);
if (!ast_strlen_zero(category)) {
regfree(&regexbuf_category);
}
return count;
}
int ast_app_group_update(struct ast_channel *old, struct ast_channel *new)
{
struct ast_group_info *gi = NULL;
AST_RWLIST_WRLOCK(&groups);
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&groups, gi, group_list) {
if (gi->chan == old) {
gi->chan = new;
} else if (gi->chan == new) {
AST_RWLIST_REMOVE_CURRENT(group_list);
ast_free(gi);
}
}
AST_RWLIST_TRAVERSE_SAFE_END;
AST_RWLIST_UNLOCK(&groups);
return 0;
}
int ast_app_group_discard(struct ast_channel *chan)
{
struct ast_group_info *gi = NULL;
AST_RWLIST_WRLOCK(&groups);
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&groups, gi, group_list) {
if (gi->chan == chan) {
AST_RWLIST_REMOVE_CURRENT(group_list);
ast_free(gi);
}
}
AST_RWLIST_TRAVERSE_SAFE_END;
AST_RWLIST_UNLOCK(&groups);
return 0;
}
int ast_app_group_list_wrlock(void)
{
return AST_RWLIST_WRLOCK(&groups);
}
int ast_app_group_list_rdlock(void)
{
return AST_RWLIST_RDLOCK(&groups);
}
struct ast_group_info *ast_app_group_list_head(void)
{
return AST_RWLIST_FIRST(&groups);
}
int ast_app_group_list_unlock(void)
{
return AST_RWLIST_UNLOCK(&groups);
}
unsigned int __ast_app_separate_args(char *buf, char delim, int remove_chars, char **array, int arraylen)
{
int argc;
char *scan, *wasdelim = NULL;
int paren = 0, quote = 0, bracket = 0;
if (!array || !arraylen) {
return 0;
}
memset(array, 0, arraylen * sizeof(*array));
if (!buf) {
return 0;
}
scan = buf;
for (argc = 0; *scan && (argc < arraylen - 1); argc++) {
array[argc] = scan;
for (; *scan; scan++) {
if (*scan == '(') {
paren++;
} else if (*scan == ')') {
if (paren) {
paren--;
}
} else if (*scan == '[') {
bracket++;
} else if (*scan == ']') {
if (bracket) {
bracket--;
}
} else if (*scan == '"' && delim != '"') {
quote = quote ? 0 : 1;
if (remove_chars) {
/* Remove quote character from argument */
memmove(scan, scan + 1, strlen(scan));
scan--;
}
} else if (*scan == '\\') {
if (remove_chars) {
/* Literal character, don't parse */
memmove(scan, scan + 1, strlen(scan));
} else {
scan++;
}
} else if ((*scan == delim) && !paren && !quote && !bracket) {
wasdelim = scan;
*scan++ = '\0';
break;
}
}
}
/* If the last character in the original string was the delimiter, then
* there is one additional argument. */
if (*scan || (scan > buf && (scan - 1) == wasdelim)) {
array[argc++] = scan;
}
return argc;
}
static enum AST_LOCK_RESULT ast_lock_path_lockfile(const char *path)
{
char *s;
char *fs;
int res;
int fd;
int lp = strlen(path);
time_t start;
s = ast_alloca(lp + 10);
fs = ast_alloca(lp + 20);
snprintf(fs, strlen(path) + 19, "%s/.lock-%08lx", path, (unsigned long)ast_random());
fd = open(fs, O_WRONLY | O_CREAT | O_EXCL, AST_FILE_MODE);
if (fd < 0) {
ast_log(LOG_ERROR, "Unable to create lock file '%s': %s\n", path, strerror(errno));
return AST_LOCK_PATH_NOT_FOUND;
}
close(fd);
snprintf(s, strlen(path) + 9, "%s/.lock", path);
start = time(NULL);
while (((res = link(fs, s)) < 0) && (errno == EEXIST) && (time(NULL) - start < 5)) {
sched_yield();
}
unlink(fs);
if (res) {
ast_log(LOG_WARNING, "Failed to lock path '%s': %s\n", path, strerror(errno));
return AST_LOCK_TIMEOUT;
} else {
ast_debug(1, "Locked path '%s'\n", path);
return AST_LOCK_SUCCESS;
}
}
static int ast_unlock_path_lockfile(const char *path)
{
char *s;
int res;
s = ast_alloca(strlen(path) + 10);
snprintf(s, strlen(path) + 9, "%s/%s", path, ".lock");
if ((res = unlink(s))) {
ast_log(LOG_ERROR, "Could not unlock path '%s': %s\n", path, strerror(errno));
} else {
ast_debug(1, "Unlocked path '%s'\n", path);
}
return res;
}
struct path_lock {
AST_LIST_ENTRY(path_lock) le;
int fd;
char *path;
};
static AST_LIST_HEAD_STATIC(path_lock_list, path_lock);
static void path_lock_destroy(struct path_lock *obj)
{
if (obj->fd >= 0) {
close(obj->fd);
}
if (obj->path) {
ast_free(obj->path);
}
ast_free(obj);
}
static enum AST_LOCK_RESULT ast_lock_path_flock(const char *path)
{
char *fs;
int res;
int fd;
time_t start;
struct path_lock *pl;
struct stat st, ost;
fs = ast_alloca(strlen(path) + 20);
snprintf(fs, strlen(path) + 19, "%s/lock", path);
if (lstat(fs, &st) == 0) {
if ((st.st_mode & S_IFMT) == S_IFLNK) {
ast_log(LOG_WARNING, "Unable to create lock file "
"'%s': it's already a symbolic link\n",
fs);
return AST_LOCK_FAILURE;
}
if (st.st_nlink > 1) {
ast_log(LOG_WARNING, "Unable to create lock file "
"'%s': %u hard links exist\n",
fs, (unsigned int) st.st_nlink);
return AST_LOCK_FAILURE;
}
}
if ((fd = open(fs, O_WRONLY | O_CREAT, 0600)) < 0) {
ast_log(LOG_WARNING, "Unable to create lock file '%s': %s\n",
fs, strerror(errno));
return AST_LOCK_PATH_NOT_FOUND;
}
if (!(pl = ast_calloc(1, sizeof(*pl)))) {
/* We don't unlink the lock file here, on the possibility that
* someone else created it - better to leave a little mess
* than create a big one by destroying someone else's lock
* and causing something to be corrupted.
*/
close(fd);
return AST_LOCK_FAILURE;
}
pl->fd = fd;
pl->path = ast_strdup(path);
time(&start);
while (
#ifdef SOLARIS
((res = fcntl(pl->fd, F_SETLK, fcntl(pl->fd, F_GETFL) | O_NONBLOCK)) < 0) &&
#else
((res = flock(pl->fd, LOCK_EX | LOCK_NB)) < 0) &&
#endif
(errno == EWOULDBLOCK) &&
(time(NULL) - start < 5))
usleep(1000);
if (res) {
ast_log(LOG_WARNING, "Failed to lock path '%s': %s\n",
path, strerror(errno));
/* No unlinking of lock done, since we tried and failed to
* flock() it.
*/
path_lock_destroy(pl);
return AST_LOCK_TIMEOUT;
}
/* Check for the race where the file is recreated or deleted out from
* underneath us.
*/
if (lstat(fs, &st) != 0 && fstat(pl->fd, &ost) != 0 &&
st.st_dev != ost.st_dev &&
st.st_ino != ost.st_ino) {
ast_log(LOG_WARNING, "Unable to create lock file '%s': "
"file changed underneath us\n", fs);
path_lock_destroy(pl);
return AST_LOCK_FAILURE;
}
/* Success: file created, flocked, and is the one we started with */
AST_LIST_LOCK(&path_lock_list);
AST_LIST_INSERT_TAIL(&path_lock_list, pl, le);
AST_LIST_UNLOCK(&path_lock_list);
ast_debug(1, "Locked path '%s'\n", path);
return AST_LOCK_SUCCESS;
}
static int ast_unlock_path_flock(const char *path)
{
char *s;
struct path_lock *p;
s = ast_alloca(strlen(path) + 20);
AST_LIST_LOCK(&path_lock_list);
AST_LIST_TRAVERSE_SAFE_BEGIN(&path_lock_list, p, le) {
if (!strcmp(p->path, path)) {
AST_LIST_REMOVE_CURRENT(le);
break;
}
}
AST_LIST_TRAVERSE_SAFE_END;
AST_LIST_UNLOCK(&path_lock_list);
if (p) {
snprintf(s, strlen(path) + 19, "%s/lock", path);
unlink(s);
path_lock_destroy(p);
ast_debug(1, "Unlocked path '%s'\n", path);
} else {
ast_debug(1, "Failed to unlock path '%s': "
"lock not found\n", path);
}
return 0;
}
void ast_set_lock_type(enum AST_LOCK_TYPE type)
{
ast_lock_type = type;
}
enum AST_LOCK_RESULT ast_lock_path(const char *path)
{
enum AST_LOCK_RESULT r = AST_LOCK_FAILURE;
switch (ast_lock_type) {
case AST_LOCK_TYPE_LOCKFILE:
r = ast_lock_path_lockfile(path);
break;
case AST_LOCK_TYPE_FLOCK:
r = ast_lock_path_flock(path);
break;
}
return r;
}
int ast_unlock_path(const char *path)
{
int r = 0;
switch (ast_lock_type) {
case AST_LOCK_TYPE_LOCKFILE:
r = ast_unlock_path_lockfile(path);
break;
case AST_LOCK_TYPE_FLOCK:
r = ast_unlock_path_flock(path);
break;
}
return r;
}
int ast_record_review(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, const char *path)
{
int silencethreshold;
int maxsilence = 0;
int res = 0;
int cmd = 0;
int max_attempts = 3;
int attempts = 0;
int recorded = 0;
int message_exists = 0;
/* Note that urgent and private are for flagging messages as such in the future */
/* barf if no pointer passed to store duration in */
if (!duration) {
ast_log(LOG_WARNING, "Error ast_record_review called without duration pointer\n");
return -1;
}
cmd = '3'; /* Want to start by recording */
silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
while ((cmd >= 0) && (cmd != 't')) {
switch (cmd) {
case '1':
if (!message_exists) {
/* In this case, 1 is to record a message */
cmd = '3';
break;
} else {
ast_stream_and_wait(chan, "vm-msgsaved", "");
cmd = 't';
return res;
}
case '2':
/* Review */
ast_verb(3, "Reviewing the recording\n");
cmd = ast_stream_and_wait(chan, recordfile, AST_DIGIT_ANY);
break;
case '3':
message_exists = 0;
/* Record */
ast_verb(3, "R%secording\n", recorded == 1 ? "e-r" : "");
recorded = 1;
Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20 23:02:25 +00:00
if ((cmd = ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, NULL, silencethreshold, maxsilence, path)) == -1) {
/* User has hung up, no options to give */
return cmd;
}
if (cmd == '0') {
break;
} else if (cmd == '*') {
break;
} else {
/* If all is well, a message exists */
message_exists = 1;
cmd = 0;
}
break;
case '4':
case '5':
case '6':
case '7':
case '8':
case '9':
case '*':
case '#':
cmd = ast_play_and_wait(chan, "vm-sorry");
break;
default:
if (message_exists) {
cmd = ast_play_and_wait(chan, "vm-review");
} else {
if (!(cmd = ast_play_and_wait(chan, "vm-torerecord"))) {
cmd = ast_waitfordigit(chan, 600);
}
}
if (!cmd) {
cmd = ast_waitfordigit(chan, 6000);
}
if (!cmd) {
attempts++;
}
if (attempts > max_attempts) {
cmd = 't';
}
}
}
if (cmd == 't') {
cmd = 0;
}
return cmd;
}
#define RES_UPONE (1 << 16)
#define RES_EXIT (1 << 17)
#define RES_REPEAT (1 << 18)
#define RES_RESTART ((1 << 19) | RES_REPEAT)
static int ast_ivr_menu_run_internal(struct ast_channel *chan, struct ast_ivr_menu *menu, void *cbdata);
static int ivr_dispatch(struct ast_channel *chan, struct ast_ivr_option *option, char *exten, void *cbdata)
{
int res;
int (*ivr_func)(struct ast_channel *, void *);
char *c;
char *n;
switch (option->action) {
case AST_ACTION_UPONE:
return RES_UPONE;
case AST_ACTION_EXIT:
return RES_EXIT | (((unsigned long)(option->adata)) & 0xffff);
case AST_ACTION_REPEAT:
return RES_REPEAT | (((unsigned long)(option->adata)) & 0xffff);
case AST_ACTION_RESTART:
return RES_RESTART ;
case AST_ACTION_NOOP:
return 0;
case AST_ACTION_BACKGROUND:
res = ast_stream_and_wait(chan, (char *)option->adata, AST_DIGIT_ANY);
if (res < 0) {
ast_log(LOG_NOTICE, "Unable to find file '%s'!\n", (char *)option->adata);
res = 0;
}
return res;
case AST_ACTION_PLAYBACK:
res = ast_stream_and_wait(chan, (char *)option->adata, "");
if (res < 0) {
ast_log(LOG_NOTICE, "Unable to find file '%s'!\n", (char *)option->adata);
res = 0;
}
return res;
case AST_ACTION_MENU:
if ((res = ast_ivr_menu_run_internal(chan, (struct ast_ivr_menu *)option->adata, cbdata)) == -2) {
/* Do not pass entry errors back up, treat as though it was an "UPONE" */
res = 0;
}
return res;
case AST_ACTION_WAITOPTION:
if (!(res = ast_waitfordigit(chan, ast_channel_pbx(chan) ? ast_channel_pbx(chan)->rtimeoutms : 10000))) {
return 't';
}
return res;
case AST_ACTION_CALLBACK:
ivr_func = option->adata;
res = ivr_func(chan, cbdata);
return res;
case AST_ACTION_TRANSFER:
res = ast_parseable_goto(chan, option->adata);
return 0;
case AST_ACTION_PLAYLIST:
case AST_ACTION_BACKLIST:
res = 0;
c = ast_strdupa(option->adata);
while ((n = strsep(&c, ";"))) {
if ((res = ast_stream_and_wait(chan, n,
(option->action == AST_ACTION_BACKLIST) ? AST_DIGIT_ANY : ""))) {
break;
}
}
ast_stopstream(chan);
return res;
default:
ast_log(LOG_NOTICE, "Unknown dispatch function %u, ignoring!\n", option->action);
return 0;
}
return -1;
}
static int option_exists(struct ast_ivr_menu *menu, char *option)
{
int x;
for (x = 0; menu->options[x].option; x++) {
if (!strcasecmp(menu->options[x].option, option)) {
return x;
}
}
return -1;
}
static int option_matchmore(struct ast_ivr_menu *menu, char *option)
{
int x;
for (x = 0; menu->options[x].option; x++) {
if ((!strncasecmp(menu->options[x].option, option, strlen(option))) &&
(menu->options[x].option[strlen(option)])) {
return x;
}
}
return -1;
}
static int read_newoption(struct ast_channel *chan, struct ast_ivr_menu *menu, char *exten, int maxexten)
{
int res = 0;
int ms;
while (option_matchmore(menu, exten)) {
ms = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->dtimeoutms : 5000;
if (strlen(exten) >= maxexten - 1) {
break;
}
if ((res = ast_waitfordigit(chan, ms)) < 1) {
break;
}
exten[strlen(exten) + 1] = '\0';
exten[strlen(exten)] = res;
}
return res > 0 ? 0 : res;
}
static int ast_ivr_menu_run_internal(struct ast_channel *chan, struct ast_ivr_menu *menu, void *cbdata)
{
/* Execute an IVR menu structure */
int res = 0;
int pos = 0;
int retries = 0;
char exten[AST_MAX_EXTENSION] = "s";
if (option_exists(menu, "s") < 0) {
strcpy(exten, "g");
if (option_exists(menu, "g") < 0) {
ast_log(LOG_WARNING, "No 's' nor 'g' extension in menu '%s'!\n", menu->title);
return -1;
}
}
while (!res) {
while (menu->options[pos].option) {
if (!strcasecmp(menu->options[pos].option, exten)) {
res = ivr_dispatch(chan, menu->options + pos, exten, cbdata);
ast_debug(1, "IVR Dispatch of '%s' (pos %d) yields %d\n", exten, pos, res);
if (res < 0) {
break;
} else if (res & RES_UPONE) {
return 0;
} else if (res & RES_EXIT) {
return res;
} else if (res & RES_REPEAT) {
int maxretries = res & 0xffff;
if ((res & RES_RESTART) == RES_RESTART) {
retries = 0;
} else {
retries++;
}
if (!maxretries) {
maxretries = 3;
}
if ((maxretries > 0) && (retries >= maxretries)) {
ast_debug(1, "Max retries %d exceeded\n", maxretries);
return -2;
} else {
if (option_exists(menu, "g") > -1) {
strcpy(exten, "g");
} else if (option_exists(menu, "s") > -1) {
strcpy(exten, "s");
}
}
pos = 0;
continue;
} else if (res && strchr(AST_DIGIT_ANY, res)) {
ast_debug(1, "Got start of extension, %c\n", res);
exten[1] = '\0';
exten[0] = res;
if ((res = read_newoption(chan, menu, exten, sizeof(exten)))) {
break;
}
if (option_exists(menu, exten) < 0) {
if (option_exists(menu, "i")) {
ast_debug(1, "Invalid extension entered, going to 'i'!\n");
strcpy(exten, "i");
pos = 0;
continue;
} else {
ast_debug(1, "Aborting on invalid entry, with no 'i' option!\n");
res = -2;
break;
}
} else {
ast_debug(1, "New existing extension: %s\n", exten);
pos = 0;
continue;
}
}
}
pos++;
}
ast_debug(1, "Stopping option '%s', res is %d\n", exten, res);
pos = 0;
if (!strcasecmp(exten, "s")) {
strcpy(exten, "g");
} else {
break;
}
}
return res;
}
int ast_ivr_menu_run(struct ast_channel *chan, struct ast_ivr_menu *menu, void *cbdata)
{
int res = ast_ivr_menu_run_internal(chan, menu, cbdata);
/* Hide internal coding */
return res > 0 ? 0 : res;
}
char *ast_read_textfile(const char *filename)
{
int fd, count = 0, res;
char *output = NULL;
struct stat filesize;
if (stat(filename, &filesize) == -1) {
ast_log(LOG_WARNING, "Error can't stat %s\n", filename);
return NULL;
}
count = filesize.st_size + 1;
if ((fd = open(filename, O_RDONLY)) < 0) {
ast_log(LOG_WARNING, "Cannot open file '%s' for reading: %s\n", filename, strerror(errno));
return NULL;
}
if ((output = ast_malloc(count))) {
res = read(fd, output, count - 1);
if (res == count - 1) {
output[res] = '\0';
} else {
ast_log(LOG_WARNING, "Short read of %s (%d of %d): %s\n", filename, res, count - 1, strerror(errno));
ast_free(output);
output = NULL;
}
}
close(fd);
return output;
}
static int parse_options(const struct ast_app_option *options, void *_flags, char **args, char *optstr, int flaglen)
{
char *s, *arg;
int curarg, res = 0;
unsigned int argloc;
struct ast_flags *flags = _flags;
struct ast_flags64 *flags64 = _flags;
if (flaglen == 32) {
ast_clear_flag(flags, AST_FLAGS_ALL);
} else {
flags64->flags = 0;
}
if (!optstr) {
return 0;
}
s = optstr;
while (*s) {
curarg = *s++ & 0x7f; /* the array (in app.h) has 128 entries */
argloc = options[curarg].arg_index;
if (*s == '(') {
int paren = 1, quote = 0;
int parsequotes = (s[1] == '"') ? 1 : 0;
/* Has argument */
arg = ++s;
for (; *s; s++) {
if (*s == '(' && !quote) {
paren++;
} else if (*s == ')' && !quote) {
/* Count parentheses, unless they're within quotes (or backslashed, below) */
paren--;
} else if (*s == '"' && parsequotes) {
/* Leave embedded quotes alone, unless they are the first character */
quote = quote ? 0 : 1;
ast_copy_string(s, s + 1, INT_MAX);
s--;
} else if (*s == '\\') {
if (!quote) {
/* If a backslash is found outside of quotes, remove it */
ast_copy_string(s, s + 1, INT_MAX);
} else if (quote && s[1] == '"') {
/* Backslash for a quote character within quotes, remove the backslash */
ast_copy_string(s, s + 1, INT_MAX);
} else {
/* Backslash within quotes, keep both characters */
s++;
}
}
if (paren == 0) {
break;
}
}
/* This will find the closing paren we found above, or none, if the string ended before we found one. */
if ((s = strchr(s, ')'))) {
if (argloc) {
args[argloc - 1] = arg;
}
*s++ = '\0';
} else {
ast_log(LOG_WARNING, "Missing closing parenthesis for argument '%c' in string '%s'\n", curarg, arg);
res = -1;
break;
}
} else if (argloc) {
args[argloc - 1] = "";
}
if (!options[curarg].flag) {
ast_log(LOG_WARNING, "Unrecognized option: '%c'\n", curarg);
}
if (flaglen == 32) {
ast_set_flag(flags, options[curarg].flag);
} else {
ast_set_flag64(flags64, options[curarg].flag);
}
}
return res;
}
int ast_app_parse_options(const struct ast_app_option *options, struct ast_flags *flags, char **args, char *optstr)
{
return parse_options(options, flags, args, optstr, 32);
}
int ast_app_parse_options64(const struct ast_app_option *options, struct ast_flags64 *flags, char **args, char *optstr)
{
return parse_options(options, flags, args, optstr, 64);
}
void ast_app_options2str64(const struct ast_app_option *options, struct ast_flags64 *flags, char *buf, size_t len)
{
unsigned int i, found = 0;
for (i = 32; i < 128 && found < len; i++) {
if (ast_test_flag64(flags, options[i].flag)) {
buf[found++] = i;
}
}
buf[found] = '\0';
}
int ast_get_encoded_char(const char *stream, char *result, size_t *consumed)
{
int i;
*consumed = 1;
*result = 0;
if (ast_strlen_zero(stream)) {
*consumed = 0;
return -1;
}
if (*stream == '\\') {
*consumed = 2;
switch (*(stream + 1)) {
case 'n':
*result = '\n';
break;
case 'r':
*result = '\r';
break;
case 't':
*result = '\t';
break;
case 'x':
/* Hexadecimal */
if (strchr("0123456789ABCDEFabcdef", *(stream + 2)) && *(stream + 2) != '\0') {
*consumed = 3;
if (*(stream + 2) <= '9') {
*result = *(stream + 2) - '0';
} else if (*(stream + 2) <= 'F') {
*result = *(stream + 2) - 'A' + 10;
} else {
*result = *(stream + 2) - 'a' + 10;
}
} else {
ast_log(LOG_ERROR, "Illegal character '%c' in hexadecimal string\n", *(stream + 2));
return -1;
}
if (strchr("0123456789ABCDEFabcdef", *(stream + 3)) && *(stream + 3) != '\0') {
*consumed = 4;
*result <<= 4;
if (*(stream + 3) <= '9') {
*result += *(stream + 3) - '0';
} else if (*(stream + 3) <= 'F') {
*result += *(stream + 3) - 'A' + 10;
} else {
*result += *(stream + 3) - 'a' + 10;
}
}
break;
case '0':
/* Octal */
*consumed = 2;
for (i = 2; ; i++) {
if (strchr("01234567", *(stream + i)) && *(stream + i) != '\0') {
(*consumed)++;
ast_debug(5, "result was %d, ", *result);
*result <<= 3;
*result += *(stream + i) - '0';
ast_debug(5, "is now %d\n", *result);
} else {
break;
}
}
break;
default:
*result = *(stream + 1);
}
} else {
*result = *stream;
*consumed = 1;
}
return 0;
}
char *ast_get_encoded_str(const char *stream, char *result, size_t result_size)
{
char *cur = result;
size_t consumed;
while (cur < result + result_size - 1 && !ast_get_encoded_char(stream, cur, &consumed)) {
cur++;
stream += consumed;
}
*cur = '\0';
return result;
}
int ast_str_get_encoded_str(struct ast_str **str, int maxlen, const char *stream)
{
char next, *buf;
size_t offset = 0;
size_t consumed;
if (strchr(stream, '\\')) {
while (!ast_get_encoded_char(stream, &next, &consumed)) {
if (offset + 2 > ast_str_size(*str) && maxlen > -1) {
ast_str_make_space(str, maxlen > 0 ? maxlen : (ast_str_size(*str) + 48) * 2 - 48);
}
if (offset + 2 > ast_str_size(*str)) {
break;
}
buf = ast_str_buffer(*str);
buf[offset++] = next;
stream += consumed;
}
buf = ast_str_buffer(*str);
buf[offset++] = '\0';
ast_str_update(*str);
} else {
ast_str_set(str, maxlen, "%s", stream);
}
return 0;
}
void ast_close_fds_above_n(int n)
{
closefrom(n + 1);
}
int ast_safe_fork(int stop_reaper)
{
sigset_t signal_set, old_set;
int pid;
/* Don't let the default signal handler for children reap our status */
if (stop_reaper) {
ast_replace_sigchld();
}
/* GCC 4.9 gives a bogus "right-hand operand of comma expression has
* no effect" warning */
(void) sigfillset(&signal_set);
pthread_sigmask(SIG_BLOCK, &signal_set, &old_set);
pid = fork();
if (pid != 0) {
/* Fork failed or parent */
pthread_sigmask(SIG_SETMASK, &old_set, NULL);
if (!stop_reaper && pid > 0) {
struct zombie *cur = ast_calloc(1, sizeof(*cur));
if (cur) {
cur->pid = pid;
AST_LIST_LOCK(&zombies);
AST_LIST_INSERT_TAIL(&zombies, cur, list);
AST_LIST_UNLOCK(&zombies);
if (shaun_of_the_dead_thread == AST_PTHREADT_NULL) {
if (ast_pthread_create_background(&shaun_of_the_dead_thread, NULL, shaun_of_the_dead, NULL)) {
ast_log(LOG_ERROR, "Shaun of the Dead wants to kill zombies, but can't?!!\n");
shaun_of_the_dead_thread = AST_PTHREADT_NULL;
}
}
}
}
return pid;
} else {
/* Child */
#ifdef HAVE_CAP
cap_set_proc(child_cap);
#endif
/* Before we unblock our signals, return our trapped signals back to the defaults */
signal(SIGHUP, SIG_DFL);
signal(SIGCHLD, SIG_DFL);
signal(SIGINT, SIG_DFL);
signal(SIGURG, SIG_DFL);
signal(SIGTERM, SIG_DFL);
signal(SIGPIPE, SIG_DFL);
signal(SIGXFSZ, SIG_DFL);
/* unblock important signal handlers */
if (pthread_sigmask(SIG_UNBLOCK, &signal_set, NULL)) {
ast_log(LOG_WARNING, "unable to unblock signals: %s\n", strerror(errno));
_exit(1);
}
return pid;
}
}
void ast_safe_fork_cleanup(void)
{
ast_unreplace_sigchld();
}
int ast_app_parse_timelen(const char *timestr, int *result, enum ast_timelen unit)
{
int res;
char u[10];
#ifdef HAVE_LONG_DOUBLE_WIDER
long double amount;
#define FMT "%30Lf%9s"
#else
double amount;
#define FMT "%30lf%9s"
#endif
if (!timestr) {
return -1;
}
res = sscanf(timestr, FMT, &amount, u);
if (res == 0 || res == EOF) {
#undef FMT
return -1;
} else if (res == 2) {
switch (u[0]) {
case 'h':
case 'H':
unit = TIMELEN_HOURS;
if (u[1] != '\0') {
return -1;
}
break;
case 's':
case 'S':
unit = TIMELEN_SECONDS;
if (u[1] != '\0') {
return -1;
}
break;
case 'm':
case 'M':
if (toupper(u[1]) == 'S') {
unit = TIMELEN_MILLISECONDS;
if (u[2] != '\0') {
return -1;
}
} else if (u[1] == '\0') {
unit = TIMELEN_MINUTES;
} else {
return -1;
}
break;
default:
return -1;
}
}
switch (unit) {
case TIMELEN_HOURS:
amount *= 60;
/* fall-through */
case TIMELEN_MINUTES:
amount *= 60;
/* fall-through */
case TIMELEN_SECONDS:
amount *= 1000;
/* fall-through */
case TIMELEN_MILLISECONDS:
;
}
*result = amount > INT_MAX ? INT_MAX : (int) amount;
return 0;
}
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
struct stasis_topic *ast_queue_topic_all(void)
{
return queue_topic_all;
}
struct stasis_topic *ast_queue_topic(const char *queuename)
{
return stasis_topic_pool_get_topic(queue_topic_pool, queuename);
}
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
static void app_cleanup(void)
{
#ifdef HAS_CAP
cap_free(child_cap);
#endif
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
ao2_cleanup(queue_topic_pool);
queue_topic_pool = NULL;
ao2_cleanup(queue_topic_all);
queue_topic_all = NULL;
}
int app_init(void)
{
ast_register_cleanup(app_cleanup);
#ifdef HAVE_CAP
child_cap = cap_from_text("cap_net_admin-eip");
#endif
queue_topic_all = stasis_topic_create("queue:all");
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
if (!queue_topic_all) {
return -1;
}
queue_topic_pool = stasis_topic_pool_create(queue_topic_all);
if (!queue_topic_pool) {
return -1;
}
return 0;
}