asterisk/funcs/func_talkdetect.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2014, Digium, Inc.
*
* Matt Jordan <mjordan@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 Function that raises events when talking is detected on a channel
*
* \author Matt Jordan <mjordan@digium.com>
*
* \ingroup functions
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
git migration: Refactor the ASTERISK_FILE_VERSION macro Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-12 02:38:22 +00:00
ASTERISK_REGISTER_FILE()
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/audiohook.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
/*** DOCUMENTATION
<function name="TALK_DETECT" language="en_US">
<synopsis>
Raises notifications when Asterisk detects silence or talking on a channel.
</synopsis>
<syntax>
<parameter name="action" required="true">
<optionlist>
<option name="remove">
<para>W/O. Remove talk detection from the channel.</para>
</option>
<option name="set">
<para>W/O. Enable TALK_DETECT and/or configure talk detection
parameters. Can be called multiple times to change parameters
on a channel with talk detection already enabled.</para>
<argument name="dsp_silence_threshold" required="false">
<para>The time in milliseconds before which a user is considered silent.</para>
</argument>
<argument name="dsp_talking_threshold" required="false">
<para>The time in milliseconds after which a user is considered talking.</para>
</argument>
</option>
</optionlist>
</parameter>
</syntax>
<description>
<para>The TALK_DETECT function enables events on the channel
it is applied to. These events can be emited over AMI, ARI, and
potentially other Asterisk modules that listen for the internal
notification.</para>
<para>The function has two parameters that can optionally be passed
when <literal>set</literal> on a channel: <replaceable>dsp_talking_threshold</replaceable>
and <replaceable>dsp_silence_threshold</replaceable>.</para>
<para><replaceable>dsp_talking_threshold</replaceable> is the time in milliseconds of sound
above what the dsp has established as base line silence for a user
before a user is considered to be talking. By default, the value of
<replaceable>silencethreshold</replaceable> from <filename>dsp.conf</filename>
is used. If this value is set too tight events may be
falsely triggered by variants in room noise.</para>
<para>Valid values are 1 through 2^31.</para>
<para><replaceable>dsp_silence_threshold</replaceable> is the time in milliseconds of sound
falling within what the dsp has established as baseline silence before
a user is considered be silent. If this value is set too low events
indicating the user has stopped talking may get falsely sent out when
the user briefly pauses during mid sentence.</para>
<para>The best way to approach this option is to set it slightly above
the maximum amount of ms of silence a user may generate during
natural speech.</para>
<para>By default this value is 2500ms. Valid values are 1
through 2^31.</para>
<para>Example:</para>
<para>same => n,Set(TALK_DETECT(set)=) ; Enable talk detection</para>
<para>same => n,Set(TALK_DETECT(set)=1200) ; Update existing talk detection's silence threshold to 1200 ms</para>
<para>same => n,Set(TALK_DETECT(remove)=) ; Remove talk detection</para>
<para>same => n,Set(TALK_DETECT(set)=,128) ; Enable and set talk threshold to 128</para>
<para>This function will set the following variables:</para>
<note>
<para>The TALK_DETECT function uses an audiohook to inspect the
voice media frames on a channel. Other functions, such as JITTERBUFFER,
DENOISE, and AGC use a similar mechanism. Audiohooks are processed
in the order in which they are placed on the channel. As such,
it typically makes sense to place functions that modify the voice
media data prior to placing the TALK_DETECT function, as this will
yield better results.</para>
<para>Example:</para>
<para>same => n,Set(DENOISE(rx)=on) ; Denoise received audio</para>
<para>same => n,Set(TALK_DETECT(set)=) ; Perform talk detection on the denoised received audio</para>
</note>
</description>
</function>
***/
#define DEFAULT_SILENCE_THRESHOLD 2500
/*! \brief Private data structure used with the function's datastore */
struct talk_detect_params {
/*! The audiohook for the function */
struct ast_audiohook audiohook;
/*! Our threshold above which we consider someone talking */
int dsp_talking_threshold;
/*! How long we'll wait before we decide someone is silent */
int dsp_silence_threshold;
/*! Whether or not the user is currently talking */
int talking;
/*! The time the current burst of talking started */
struct timeval talking_start;
/*! The DSP used to do the heavy lifting */
struct ast_dsp *dsp;
};
/*! \internal \brief Destroy the datastore */
static void datastore_destroy_cb(void *data) {
struct talk_detect_params *td_params = data;
ast_audiohook_destroy(&td_params->audiohook);
if (td_params->dsp) {
ast_dsp_free(td_params->dsp);
}
ast_free(data);
}
/*! \brief The channel datastore the function uses to store state */
static const struct ast_datastore_info talk_detect_datastore = {
.type = "talk_detect",
.destroy = datastore_destroy_cb
};
/*! \internal \brief An audiohook modification callback
*
* This processes the read side of a channel's voice data to see if
* they are talking
*
* \note We don't actually modify the audio, so this function always
* returns a 'failure' indicating that it didn't modify the data
*/
static int talk_detect_audiohook_cb(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
{
int total_silence;
int update_talking = 0;
struct ast_datastore *datastore;
struct talk_detect_params *td_params;
struct stasis_message *message;
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
return 1;
}
if (direction != AST_AUDIOHOOK_DIRECTION_READ) {
return 1;
}
if (frame->frametype != AST_FRAME_VOICE) {
return 1;
}
if (!(datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL))) {
return 1;
}
td_params = datastore->data;
ast_dsp_silence(td_params->dsp, frame, &total_silence);
if (total_silence < td_params->dsp_silence_threshold) {
if (!td_params->talking) {
update_talking = 1;
td_params->talking_start = ast_tvnow();
}
td_params->talking = 1;
} else {
if (td_params->talking) {
update_talking = 1;
}
td_params->talking = 0;
}
if (update_talking) {
struct ast_json *blob = NULL;
if (!td_params->talking) {
int64_t diff_ms = ast_tvdiff_ms(ast_tvnow(), td_params->talking_start);
diff_ms -= td_params->dsp_silence_threshold;
blob = ast_json_pack("{s: i}", "duration", diff_ms);
if (!blob) {
return 1;
}
}
ast_verb(4, "%s is now %s\n", ast_channel_name(chan),
td_params->talking ? "talking" : "silent");
message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan),
td_params->talking ? ast_channel_talking_start() : ast_channel_talking_stop(),
blob);
if (message) {
stasis_publish(ast_channel_topic(chan), message);
ao2_ref(message, -1);
}
ast_json_unref(blob);
}
return 1;
}
/*! \internal \brief Disable talk detection on the channel */
static int remove_talk_detect(struct ast_channel *chan)
{
struct ast_datastore *datastore = NULL;
struct talk_detect_params *td_params;
SCOPED_CHANNELLOCK(chan_lock, chan);
datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
if (!datastore) {
ast_log(AST_LOG_WARNING, "Cannot remove TALK_DETECT from %s: TALK_DETECT not currently enabled\n",
ast_channel_name(chan));
return -1;
}
td_params = datastore->data;
if (ast_audiohook_remove(chan, &td_params->audiohook)) {
ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT audiohook from channel %s\n",
ast_channel_name(chan));
return -1;
}
if (ast_channel_datastore_remove(chan, datastore)) {
ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT datastore from channel %s\n",
ast_channel_name(chan));
return -1;
}
ast_datastore_free(datastore);
return 0;
}
/*! \internal \brief Enable talk detection on the channel */
static int set_talk_detect(struct ast_channel *chan, int dsp_silence_threshold, int dsp_talking_threshold)
{
struct ast_datastore *datastore = NULL;
struct talk_detect_params *td_params;
SCOPED_CHANNELLOCK(chan_lock, chan);
datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
if (!datastore) {
datastore = ast_datastore_alloc(&talk_detect_datastore, NULL);
if (!datastore) {
return -1;
}
td_params = ast_calloc(1, sizeof(*td_params));
if (!td_params) {
ast_datastore_free(datastore);
return -1;
}
ast_audiohook_init(&td_params->audiohook,
AST_AUDIOHOOK_TYPE_MANIPULATE,
"TALK_DETECT",
AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
td_params->audiohook.manipulate_callback = talk_detect_audiohook_cb;
ast_set_flag(&td_params->audiohook, AST_AUDIOHOOK_TRIGGER_READ);
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
td_params->dsp = ast_dsp_new_with_rate(ast_format_get_sample_rate(ast_channel_rawreadformat(chan)));
if (!td_params->dsp) {
ast_datastore_free(datastore);
ast_free(td_params);
return -1;
}
datastore->data = td_params;
ast_channel_datastore_add(chan, datastore);
ast_audiohook_attach(chan, &td_params->audiohook);
} else {
/* Talk detection already enabled; update existing settings */
td_params = datastore->data;
}
td_params->dsp_talking_threshold = dsp_talking_threshold;
td_params->dsp_silence_threshold = dsp_silence_threshold;
ast_dsp_set_threshold(td_params->dsp, td_params->dsp_talking_threshold);
return 0;
}
/*! \internal \brief TALK_DETECT write function callback */
static int talk_detect_fn_write(struct ast_channel *chan, const char *function, char *data, const char *value)
{
int res;
if (!chan) {
return -1;
}
if (ast_strlen_zero(data)) {
ast_log(AST_LOG_WARNING, "TALK_DETECT requires an argument\n");
return -1;
}
if (!strcasecmp(data, "set")) {
int dsp_silence_threshold = DEFAULT_SILENCE_THRESHOLD;
int dsp_talking_threshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
if (!ast_strlen_zero(value)) {
char *parse = ast_strdupa(value);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(silence_threshold);
AST_APP_ARG(talking_threshold);
);
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.silence_threshold)) {
if (sscanf(args.silence_threshold, "%30d", &dsp_silence_threshold) != 1) {
ast_log(AST_LOG_WARNING, "Failed to parse %s for dsp_silence_threshold\n",
args.silence_threshold);
return -1;
}
if (dsp_silence_threshold < 1) {
ast_log(AST_LOG_WARNING, "Invalid value %d for dsp_silence_threshold\n",
dsp_silence_threshold);
return -1;
}
}
if (!ast_strlen_zero(args.talking_threshold)) {
if (sscanf(args.talking_threshold, "%30d", &dsp_talking_threshold) != 1) {
ast_log(AST_LOG_WARNING, "Failed to parse %s for dsp_talking_threshold\n",
args.talking_threshold);
return -1;
}
if (dsp_talking_threshold < 1) {
ast_log(AST_LOG_WARNING, "Invalid value %d for dsp_talking_threshold\n",
dsp_silence_threshold);
return -1;
}
}
}
res = set_talk_detect(chan, dsp_silence_threshold, dsp_talking_threshold);
} else if (!strcasecmp(data, "remove")) {
res = remove_talk_detect(chan);
} else {
ast_log(AST_LOG_WARNING, "TALK_DETECT: unknown option %s\n", data);
res = -1;
}
return res;
}
/*! \brief Definition of the TALK_DETECT function */
static struct ast_custom_function talk_detect_function = {
.name = "TALK_DETECT",
.write = talk_detect_fn_write,
};
/*! \internal \brief Unload the module */
static int unload_module(void)
{
int res = 0;
res |= ast_custom_function_unregister(&talk_detect_function);
return res;
}
/*! \internal \brief Load the module */
static int load_module(void)
{
int res = 0;
res |= ast_custom_function_register(&talk_detect_function);
return res ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Talk detection dialplan function");