2014-05-30 12:42:57 +00:00
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2014, Digium, Inc.
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*
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* Matt Jordan <mjordan@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 Function that raises events when talking is detected on a channel
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*
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* \author Matt Jordan <mjordan@digium.com>
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*
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* \ingroup functions
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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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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/app.h"
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#include "asterisk/dsp.h"
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#include "asterisk/audiohook.h"
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#include "asterisk/stasis.h"
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#include "asterisk/stasis_channels.h"
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/*** DOCUMENTATION
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<function name="TALK_DETECT" language="en_US">
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<synopsis>
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Raises notifications when Asterisk detects silence or talking on a channel.
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</synopsis>
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<syntax>
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<parameter name="action" required="true">
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<optionlist>
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<option name="remove">
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<para>W/O. Remove talk detection from the channel.</para>
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</option>
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<option name="set">
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<para>W/O. Enable TALK_DETECT and/or configure talk detection
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parameters. Can be called multiple times to change parameters
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on a channel with talk detection already enabled.</para>
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<argument name="dsp_silence_threshold" required="false">
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<para>The time in milliseconds before which a user is considered silent.</para>
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</argument>
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<argument name="dsp_talking_threshold" required="false">
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<para>The time in milliseconds after which a user is considered talking.</para>
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</argument>
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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>The TALK_DETECT function enables events on the channel
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it is applied to. These events can be emited over AMI, ARI, and
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potentially other Asterisk modules that listen for the internal
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notification.</para>
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<para>The function has two parameters that can optionally be passed
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when <literal>set</literal> on a channel: <replaceable>dsp_talking_threshold</replaceable>
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and <replaceable>dsp_silence_threshold</replaceable>.</para>
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<para><replaceable>dsp_talking_threshold</replaceable> is the time in milliseconds of sound
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above what the dsp has established as base line silence for a user
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before a user is considered to be talking. By default, the value of
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<replaceable>silencethreshold</replaceable> from <filename>dsp.conf</filename>
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is used. If this value is set too tight events may be
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falsely triggered by variants in room noise.</para>
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<para>Valid values are 1 through 2^31.</para>
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<para><replaceable>dsp_silence_threshold</replaceable> is the time in milliseconds of sound
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falling within what the dsp has established as baseline silence before
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a user is considered be silent. If this value is set too low events
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indicating the user has stopped talking may get falsely sent out when
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the user briefly pauses during mid sentence.</para>
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<para>The best way to approach this option is to set it slightly above
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the maximum amount of ms of silence a user may generate during
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natural speech.</para>
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<para>By default this value is 2500ms. Valid values are 1
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through 2^31.</para>
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<para>Example:</para>
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<para>same => n,Set(TALK_DETECT(set)=) ; Enable talk detection</para>
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<para>same => n,Set(TALK_DETECT(set)=1200) ; Update existing talk detection's silence threshold to 1200 ms</para>
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<para>same => n,Set(TALK_DETECT(remove)=) ; Remove talk detection</para>
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<para>same => n,Set(TALK_DETECT(set)=,128) ; Enable and set talk threshold to 128</para>
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<para>This function will set the following variables:</para>
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<note>
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<para>The TALK_DETECT function uses an audiohook to inspect the
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voice media frames on a channel. Other functions, such as JITTERBUFFER,
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DENOISE, and AGC use a similar mechanism. Audiohooks are processed
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in the order in which they are placed on the channel. As such,
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it typically makes sense to place functions that modify the voice
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media data prior to placing the TALK_DETECT function, as this will
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yield better results.</para>
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<para>Example:</para>
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<para>same => n,Set(DENOISE(rx)=on) ; Denoise received audio</para>
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<para>same => n,Set(TALK_DETECT(set)=) ; Perform talk detection on the denoised received audio</para>
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</note>
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</description>
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</function>
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***/
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#define DEFAULT_SILENCE_THRESHOLD 2500
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/*! \brief Private data structure used with the function's datastore */
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struct talk_detect_params {
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/*! The audiohook for the function */
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struct ast_audiohook audiohook;
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/*! Our threshold above which we consider someone talking */
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int dsp_talking_threshold;
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/*! How long we'll wait before we decide someone is silent */
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int dsp_silence_threshold;
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/*! Whether or not the user is currently talking */
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int talking;
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/*! The time the current burst of talking started */
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struct timeval talking_start;
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/*! The DSP used to do the heavy lifting */
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struct ast_dsp *dsp;
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};
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/*! \internal \brief Destroy the datastore */
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static void datastore_destroy_cb(void *data) {
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struct talk_detect_params *td_params = data;
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ast_audiohook_destroy(&td_params->audiohook);
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if (td_params->dsp) {
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ast_dsp_free(td_params->dsp);
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}
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ast_free(data);
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}
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/*! \brief The channel datastore the function uses to store state */
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static const struct ast_datastore_info talk_detect_datastore = {
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.type = "talk_detect",
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.destroy = datastore_destroy_cb
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};
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/*! \internal \brief An audiohook modification callback
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*
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* This processes the read side of a channel's voice data to see if
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* they are talking
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*
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* \note We don't actually modify the audio, so this function always
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* returns a 'failure' indicating that it didn't modify the data
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*/
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static int talk_detect_audiohook_cb(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
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{
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int total_silence;
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int update_talking = 0;
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struct ast_datastore *datastore;
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struct talk_detect_params *td_params;
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struct stasis_message *message;
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if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
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return 1;
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}
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if (direction != AST_AUDIOHOOK_DIRECTION_READ) {
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return 1;
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}
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if (frame->frametype != AST_FRAME_VOICE) {
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return 1;
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}
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if (!(datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL))) {
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return 1;
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}
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td_params = datastore->data;
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ast_dsp_silence(td_params->dsp, frame, &total_silence);
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if (total_silence < td_params->dsp_silence_threshold) {
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if (!td_params->talking) {
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update_talking = 1;
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td_params->talking_start = ast_tvnow();
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}
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td_params->talking = 1;
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} else {
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if (td_params->talking) {
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update_talking = 1;
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}
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td_params->talking = 0;
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}
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if (update_talking) {
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struct ast_json *blob = NULL;
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if (!td_params->talking) {
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int64_t diff_ms = ast_tvdiff_ms(ast_tvnow(), td_params->talking_start);
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diff_ms -= td_params->dsp_silence_threshold;
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blob = ast_json_pack("{s: i}", "duration", diff_ms);
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if (!blob) {
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return 1;
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}
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}
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ast_verb(4, "%s is now %s\n", ast_channel_name(chan),
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td_params->talking ? "talking" : "silent");
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message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan),
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td_params->talking ? ast_channel_talking_start() : ast_channel_talking_stop(),
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blob);
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if (message) {
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stasis_publish(ast_channel_topic(chan), message);
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2014-11-04 19:46:33 +00:00
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ao2_ref(message, -1);
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2014-05-30 12:42:57 +00:00
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}
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ast_json_unref(blob);
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}
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return 1;
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}
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/*! \internal \brief Disable talk detection on the channel */
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static int remove_talk_detect(struct ast_channel *chan)
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{
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struct ast_datastore *datastore = NULL;
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struct talk_detect_params *td_params;
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SCOPED_CHANNELLOCK(chan_lock, chan);
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datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
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if (!datastore) {
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ast_log(AST_LOG_WARNING, "Cannot remove TALK_DETECT from %s: TALK_DETECT not currently enabled\n",
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ast_channel_name(chan));
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return -1;
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}
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td_params = datastore->data;
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if (ast_audiohook_remove(chan, &td_params->audiohook)) {
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ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT audiohook from channel %s\n",
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ast_channel_name(chan));
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return -1;
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}
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if (ast_channel_datastore_remove(chan, datastore)) {
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ast_log(AST_LOG_WARNING, "Failed to remove TALK_DETECT datastore from channel %s\n",
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ast_channel_name(chan));
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return -1;
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}
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ast_datastore_free(datastore);
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return 0;
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}
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/*! \internal \brief Enable talk detection on the channel */
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static int set_talk_detect(struct ast_channel *chan, int dsp_silence_threshold, int dsp_talking_threshold)
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{
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struct ast_datastore *datastore = NULL;
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struct talk_detect_params *td_params;
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SCOPED_CHANNELLOCK(chan_lock, chan);
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datastore = ast_channel_datastore_find(chan, &talk_detect_datastore, NULL);
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if (!datastore) {
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datastore = ast_datastore_alloc(&talk_detect_datastore, NULL);
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if (!datastore) {
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return -1;
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}
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td_params = ast_calloc(1, sizeof(*td_params));
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if (!td_params) {
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ast_datastore_free(datastore);
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return -1;
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}
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ast_audiohook_init(&td_params->audiohook,
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AST_AUDIOHOOK_TYPE_MANIPULATE,
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"TALK_DETECT",
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AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
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td_params->audiohook.manipulate_callback = talk_detect_audiohook_cb;
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ast_set_flag(&td_params->audiohook, AST_AUDIOHOOK_TRIGGER_READ);
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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
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td_params->dsp = ast_dsp_new_with_rate(ast_format_get_sample_rate(ast_channel_rawreadformat(chan)));
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2014-05-30 12:42:57 +00:00
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if (!td_params->dsp) {
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ast_datastore_free(datastore);
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ast_free(td_params);
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return -1;
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}
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datastore->data = td_params;
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ast_channel_datastore_add(chan, datastore);
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ast_audiohook_attach(chan, &td_params->audiohook);
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} else {
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/* Talk detection already enabled; update existing settings */
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td_params = datastore->data;
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}
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td_params->dsp_talking_threshold = dsp_talking_threshold;
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td_params->dsp_silence_threshold = dsp_silence_threshold;
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ast_dsp_set_threshold(td_params->dsp, td_params->dsp_talking_threshold);
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return 0;
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}
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/*! \internal \brief TALK_DETECT write function callback */
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static int talk_detect_fn_write(struct ast_channel *chan, const char *function, char *data, const char *value)
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{
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int res;
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if (!chan) {
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return -1;
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}
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if (ast_strlen_zero(data)) {
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ast_log(AST_LOG_WARNING, "TALK_DETECT requires an argument\n");
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return -1;
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}
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if (!strcasecmp(data, "set")) {
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int dsp_silence_threshold = DEFAULT_SILENCE_THRESHOLD;
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int dsp_talking_threshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
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if (!ast_strlen_zero(value)) {
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char *parse = ast_strdupa(value);
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(silence_threshold);
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AST_APP_ARG(talking_threshold);
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);
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AST_STANDARD_APP_ARGS(args, parse);
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if (!ast_strlen_zero(args.silence_threshold)) {
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if (sscanf(args.silence_threshold, "%30d", &dsp_silence_threshold) != 1) {
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ast_log(AST_LOG_WARNING, "Failed to parse %s for dsp_silence_threshold\n",
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args.silence_threshold);
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return -1;
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}
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if (dsp_silence_threshold < 1) {
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ast_log(AST_LOG_WARNING, "Invalid value %d for dsp_silence_threshold\n",
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dsp_silence_threshold);
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return -1;
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}
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}
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if (!ast_strlen_zero(args.talking_threshold)) {
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if (sscanf(args.talking_threshold, "%30d", &dsp_talking_threshold) != 1) {
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ast_log(AST_LOG_WARNING, "Failed to parse %s for dsp_talking_threshold\n",
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args.talking_threshold);
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return -1;
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}
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if (dsp_talking_threshold < 1) {
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ast_log(AST_LOG_WARNING, "Invalid value %d for dsp_talking_threshold\n",
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dsp_silence_threshold);
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return -1;
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}
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}
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}
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res = set_talk_detect(chan, dsp_silence_threshold, dsp_talking_threshold);
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} else if (!strcasecmp(data, "remove")) {
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res = remove_talk_detect(chan);
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} else {
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ast_log(AST_LOG_WARNING, "TALK_DETECT: unknown option %s\n", data);
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res = -1;
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}
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return res;
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}
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/*! \brief Definition of the TALK_DETECT function */
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static struct ast_custom_function talk_detect_function = {
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.name = "TALK_DETECT",
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.write = talk_detect_fn_write,
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};
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/*! \internal \brief Unload the module */
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static int unload_module(void)
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{
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int res = 0;
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res |= ast_custom_function_unregister(&talk_detect_function);
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return res;
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}
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/*! \internal \brief Load the module */
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static int load_module(void)
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{
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int res = 0;
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res |= ast_custom_function_register(&talk_detect_function);
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return res ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Talk detection dialplan function");
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