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
485 lines
14 KiB
C
485 lines
14 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* Matthew Fredrickson <creslin@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Trivial application to record a sound file
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*
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* \author Matthew Fredrickson <creslin@digium.com>
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*
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* \ingroup applications
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/file.h"
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#include "asterisk/pbx.h"
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#include "asterisk/module.h"
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#include "asterisk/app.h"
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#include "asterisk/channel.h"
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#include "asterisk/dsp.h" /* use dsp routines for silence detection */
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#include "asterisk/format_cache.h"
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/*** DOCUMENTATION
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<application name="Record" language="en_US">
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<synopsis>
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Record to a file.
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</synopsis>
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<syntax>
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<parameter name="filename" required="true" argsep=".">
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<argument name="filename" required="true" />
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<argument name="format" required="true">
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<para>Is the format of the file type to be recorded (wav, gsm, etc).</para>
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</argument>
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</parameter>
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<parameter name="silence">
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<para>Is the number of seconds of silence to allow before returning.</para>
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</parameter>
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<parameter name="maxduration">
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<para>Is the maximum recording duration in seconds. If missing
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or 0 there is no maximum.</para>
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</parameter>
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<parameter name="options">
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<optionlist>
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<option name="a">
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<para>Append to existing recording rather than replacing.</para>
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</option>
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<option name="n">
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<para>Do not answer, but record anyway if line not yet answered.</para>
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</option>
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<option name="o">
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<para>Exit when 0 is pressed, setting the variable <variable>RECORD_STATUS</variable>
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to <literal>OPERATOR</literal> instead of <literal>DTMF</literal></para>
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</option>
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<option name="q">
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<para>quiet (do not play a beep tone).</para>
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</option>
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<option name="s">
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<para>skip recording if the line is not yet answered.</para>
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</option>
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<option name="t">
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<para>use alternate '*' terminator key (DTMF) instead of default '#'</para>
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</option>
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<option name="x">
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<para>Ignore all terminator keys (DTMF) and keep recording until hangup.</para>
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</option>
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<option name="k">
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<para>Keep recorded file upon hangup.</para>
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</option>
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<option name="y">
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<para>Terminate recording if *any* DTMF digit is received.</para>
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</option>
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</optionlist>
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</parameter>
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</syntax>
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<description>
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<para>If filename contains <literal>%d</literal>, these characters will be replaced with a number
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incremented by one each time the file is recorded.
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Use <astcli>core show file formats</astcli> to see the available formats on your system
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User can press <literal>#</literal> to terminate the recording and continue to the next priority.
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If the user hangs up during a recording, all data will be lost and the application will terminate.</para>
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<variablelist>
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<variable name="RECORDED_FILE">
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<para>Will be set to the final filename of the recording.</para>
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</variable>
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<variable name="RECORD_STATUS">
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<para>This is the final status of the command</para>
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<value name="DTMF">A terminating DTMF was received ('#' or '*', depending upon option 't')</value>
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<value name="SILENCE">The maximum silence occurred in the recording.</value>
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<value name="SKIP">The line was not yet answered and the 's' option was specified.</value>
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<value name="TIMEOUT">The maximum length was reached.</value>
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<value name="HANGUP">The channel was hung up.</value>
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<value name="ERROR">An unrecoverable error occurred, which resulted in a WARNING to the logs.</value>
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</variable>
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</variablelist>
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</description>
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</application>
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***/
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#define OPERATOR_KEY '0'
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static char *app = "Record";
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enum {
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OPTION_APPEND = (1 << 0),
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OPTION_NOANSWER = (1 << 1),
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OPTION_QUIET = (1 << 2),
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OPTION_SKIP = (1 << 3),
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OPTION_STAR_TERMINATE = (1 << 4),
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OPTION_IGNORE_TERMINATE = (1 << 5),
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OPTION_KEEP = (1 << 6),
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FLAG_HAS_PERCENT = (1 << 7),
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OPTION_ANY_TERMINATE = (1 << 8),
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OPTION_OPERATOR_EXIT = (1 << 9),
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};
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AST_APP_OPTIONS(app_opts,{
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AST_APP_OPTION('a', OPTION_APPEND),
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AST_APP_OPTION('k', OPTION_KEEP),
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AST_APP_OPTION('n', OPTION_NOANSWER),
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AST_APP_OPTION('o', OPTION_OPERATOR_EXIT),
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AST_APP_OPTION('q', OPTION_QUIET),
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AST_APP_OPTION('s', OPTION_SKIP),
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AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
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AST_APP_OPTION('y', OPTION_ANY_TERMINATE),
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AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
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});
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/*!
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* \internal
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* \brief Used to determine what action to take when DTMF is received while recording
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* \since 13.0.0
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*
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* \param chan channel being recorded
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* \param flags option flags in use by the record application
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* \param dtmf_integer the integer value of the DTMF key received
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* \param terminator key currently set to be pressed for normal termination
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*
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* \retval 0 do not exit
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* \retval -1 do exit
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*/
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static int record_dtmf_response(struct ast_channel *chan, struct ast_flags *flags, int dtmf_integer, int terminator)
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{
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if ((dtmf_integer == OPERATOR_KEY) &&
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(ast_test_flag(flags, OPTION_OPERATOR_EXIT))) {
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "OPERATOR");
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return -1;
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}
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if ((dtmf_integer == terminator) ||
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(ast_test_flag(flags, OPTION_ANY_TERMINATE))) {
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "DTMF");
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return -1;
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}
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return 0;
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}
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static int record_exec(struct ast_channel *chan, const char *data)
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{
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int res = 0;
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int count = 0;
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char *ext = NULL, *opts[0];
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char *parse, *dir, *file;
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int i = 0;
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char tmp[256];
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struct ast_filestream *s = NULL;
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struct ast_frame *f = NULL;
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struct ast_dsp *sildet = NULL; /* silence detector dsp */
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int totalsilence = 0;
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int dspsilence = 0;
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int silence = 0; /* amount of silence to allow */
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int gotsilence = 0; /* did we timeout for silence? */
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int maxduration = 0; /* max duration of recording in milliseconds */
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int gottimeout = 0; /* did we timeout for maxduration exceeded? */
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int terminator = '#';
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RAII_VAR(struct ast_format *, rfmt, NULL, ao2_cleanup);
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int ioflags;
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struct ast_silence_generator *silgen = NULL;
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struct ast_flags flags = { 0, };
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(filename);
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AST_APP_ARG(silence);
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AST_APP_ARG(maxduration);
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AST_APP_ARG(options);
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);
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int ms;
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struct timeval start;
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/* The next few lines of code parse out the filename and header from the input string */
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if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
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ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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return -1;
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}
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parse = ast_strdupa(data);
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AST_STANDARD_APP_ARGS(args, parse);
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if (args.argc == 4)
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ast_app_parse_options(app_opts, &flags, opts, args.options);
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if (!ast_strlen_zero(args.filename)) {
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if (strstr(args.filename, "%d"))
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ast_set_flag(&flags, FLAG_HAS_PERCENT);
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ext = strrchr(args.filename, '.'); /* to support filename with a . in the filename, not format */
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if (!ext)
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ext = strchr(args.filename, ':');
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if (ext) {
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*ext = '\0';
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ext++;
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}
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}
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if (!ext) {
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ast_log(LOG_WARNING, "No extension specified to filename!\n");
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pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
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return -1;
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}
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if (args.silence) {
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if ((sscanf(args.silence, "%30d", &i) == 1) && (i > -1)) {
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silence = i * 1000;
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} else if (!ast_strlen_zero(args.silence)) {
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ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
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}
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}
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if (args.maxduration) {
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if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
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/* Convert duration to milliseconds */
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maxduration = i * 1000;
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else if (!ast_strlen_zero(args.maxduration))
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ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", args.maxduration);
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}
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if (ast_test_flag(&flags, OPTION_STAR_TERMINATE))
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terminator = '*';
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if (ast_test_flag(&flags, OPTION_IGNORE_TERMINATE))
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terminator = '\0';
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/* done parsing */
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/* these are to allow the use of the %d in the config file for a wild card of sort to
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create a new file with the inputed name scheme */
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if (ast_test_flag(&flags, FLAG_HAS_PERCENT)) {
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AST_DECLARE_APP_ARGS(fname,
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AST_APP_ARG(piece)[100];
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);
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char *tmp2 = ast_strdupa(args.filename);
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char countstring[15];
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int idx;
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/* Separate each piece out by the format specifier */
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AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%');
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do {
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int tmplen;
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/* First piece has no leading percent, so it's copied verbatim */
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ast_copy_string(tmp, fname.piece[0], sizeof(tmp));
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tmplen = strlen(tmp);
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for (idx = 1; idx < fname.argc; idx++) {
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if (fname.piece[idx][0] == 'd') {
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/* Substitute the count */
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snprintf(countstring, sizeof(countstring), "%d", count);
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ast_copy_string(tmp + tmplen, countstring, sizeof(tmp) - tmplen);
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tmplen += strlen(countstring);
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} else if (tmplen + 2 < sizeof(tmp)) {
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/* Unknown format specifier - just copy it verbatim */
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tmp[tmplen++] = '%';
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tmp[tmplen++] = fname.piece[idx][0];
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}
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/* Copy the remaining portion of the piece */
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ast_copy_string(tmp + tmplen, &(fname.piece[idx][1]), sizeof(tmp) - tmplen);
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}
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count++;
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} while (ast_fileexists(tmp, ext, ast_channel_language(chan)) > 0);
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pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
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} else
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ast_copy_string(tmp, args.filename, sizeof(tmp));
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/* end of routine mentioned */
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|
if (ast_channel_state(chan) != AST_STATE_UP) {
|
|
if (ast_test_flag(&flags, OPTION_SKIP)) {
|
|
/* At the user's option, skip if the line is not up */
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SKIP");
|
|
return 0;
|
|
} else if (!ast_test_flag(&flags, OPTION_NOANSWER)) {
|
|
/* Otherwise answer unless we're supposed to record while on-hook */
|
|
res = ast_answer(chan);
|
|
}
|
|
}
|
|
|
|
if (res) {
|
|
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", ast_channel_name(chan));
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
goto out;
|
|
}
|
|
|
|
if (!ast_test_flag(&flags, OPTION_QUIET)) {
|
|
/* Some code to play a nice little beep to signify the start of the record operation */
|
|
res = ast_streamfile(chan, "beep", ast_channel_language(chan));
|
|
if (!res) {
|
|
res = ast_waitstream(chan, "");
|
|
} else {
|
|
ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", ast_channel_name(chan));
|
|
}
|
|
ast_stopstream(chan);
|
|
}
|
|
|
|
/* The end of beep code. Now the recording starts */
|
|
|
|
if (silence > 0) {
|
|
rfmt = ao2_bump(ast_channel_readformat(chan));
|
|
res = ast_set_read_format(chan, ast_format_slin);
|
|
if (res < 0) {
|
|
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
sildet = ast_dsp_new();
|
|
if (!sildet) {
|
|
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
|
|
}
|
|
|
|
/* Create the directory if it does not exist. */
|
|
dir = ast_strdupa(tmp);
|
|
if ((file = strrchr(dir, '/')))
|
|
*file++ = '\0';
|
|
ast_mkdir (dir, 0777);
|
|
|
|
ioflags = ast_test_flag(&flags, OPTION_APPEND) ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
|
|
s = ast_writefile(tmp, ext, NULL, ioflags, 0, AST_FILE_MODE);
|
|
|
|
if (!s) {
|
|
ast_log(LOG_WARNING, "Could not create file %s\n", args.filename);
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
goto out;
|
|
}
|
|
|
|
if (ast_opt_transmit_silence)
|
|
silgen = ast_channel_start_silence_generator(chan);
|
|
|
|
/* Request a video update */
|
|
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
|
|
|
|
if (maxduration <= 0)
|
|
maxduration = -1;
|
|
|
|
start = ast_tvnow();
|
|
while ((ms = ast_remaining_ms(start, maxduration))) {
|
|
ms = ast_waitfor(chan, ms);
|
|
if (ms < 0) {
|
|
break;
|
|
}
|
|
|
|
if (maxduration > 0 && ms == 0) {
|
|
break;
|
|
}
|
|
|
|
f = ast_read(chan);
|
|
if (!f) {
|
|
res = -1;
|
|
break;
|
|
}
|
|
if (f->frametype == AST_FRAME_VOICE) {
|
|
res = ast_writestream(s, f);
|
|
|
|
if (res) {
|
|
ast_log(LOG_WARNING, "Problem writing frame\n");
|
|
ast_frfree(f);
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
break;
|
|
}
|
|
|
|
if (silence > 0) {
|
|
dspsilence = 0;
|
|
ast_dsp_silence(sildet, f, &dspsilence);
|
|
if (dspsilence) {
|
|
totalsilence = dspsilence;
|
|
} else {
|
|
totalsilence = 0;
|
|
}
|
|
if (totalsilence > silence) {
|
|
/* Ended happily with silence */
|
|
ast_frfree(f);
|
|
gotsilence = 1;
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SILENCE");
|
|
break;
|
|
}
|
|
}
|
|
} else if (f->frametype == AST_FRAME_VIDEO) {
|
|
res = ast_writestream(s, f);
|
|
|
|
if (res) {
|
|
ast_log(LOG_WARNING, "Problem writing frame\n");
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
ast_frfree(f);
|
|
break;
|
|
}
|
|
} else if (f->frametype == AST_FRAME_DTMF) {
|
|
if (record_dtmf_response(chan, &flags, f->subclass.integer, terminator)) {
|
|
ast_frfree(f);
|
|
break;
|
|
}
|
|
}
|
|
ast_frfree(f);
|
|
}
|
|
|
|
if (maxduration > 0 && !ms) {
|
|
gottimeout = 1;
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "TIMEOUT");
|
|
}
|
|
|
|
if (!f) {
|
|
ast_debug(1, "Got hangup\n");
|
|
res = -1;
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "HANGUP");
|
|
if (!ast_test_flag(&flags, OPTION_KEEP)) {
|
|
ast_filedelete(args.filename, NULL);
|
|
}
|
|
}
|
|
|
|
if (gotsilence) {
|
|
ast_stream_rewind(s, silence - 1000);
|
|
ast_truncstream(s);
|
|
} else if (!gottimeout) {
|
|
/* Strip off the last 1/4 second of it */
|
|
ast_stream_rewind(s, 250);
|
|
ast_truncstream(s);
|
|
}
|
|
ast_closestream(s);
|
|
|
|
if (silgen)
|
|
ast_channel_stop_silence_generator(chan, silgen);
|
|
|
|
out:
|
|
if ((silence > 0) && rfmt) {
|
|
res = ast_set_read_format(chan, rfmt);
|
|
if (res) {
|
|
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(chan));
|
|
}
|
|
}
|
|
|
|
if (sildet) {
|
|
ast_dsp_free(sildet);
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
return ast_unregister_application(app);
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
return ast_register_application_xml(app, record_exec);
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial Record Application");
|