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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Playback a file with audio detect
*
* \author Mark Spencer <markster@digium.com>
*
* \ingroup applications
*/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision: 211580 $")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/utils.h"
#include "asterisk/dsp.h"
#include "asterisk/app.h"
/*** DOCUMENTATION
<application name="BackgroundDetect" language="en_US">
<synopsis>
Background a file with talk detect.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="sil">
<para>If not specified, defaults to <literal>1000</literal>.</para>
</parameter>
<parameter name="min">
<para>If not specified, defaults to <literal>100</literal>.</para>
</parameter>
<parameter name="max">
<para>If not specified, defaults to <literal>infinity</literal>.</para>
</parameter>
<parameter name="analysistime">
<para>If not specified, defaults to <literal>infinity</literal>.</para>
</parameter>
</syntax>
<description>
<para>Plays back <replaceable>filename</replaceable>, waiting for interruption from a given digit (the digit
must start the beginning of a valid extension, or it will be ignored). During
the playback of the file, audio is monitored in the receive direction, and if
a period of non-silence which is greater than <replaceable>min</replaceable> ms yet less than
<replaceable>max</replaceable> ms is followed by silence for at least <replaceable>sil</replaceable> ms,
which occurs during the first <replaceable>analysistime</replaceable> ms, then the audio playback is
aborted and processing jumps to the <replaceable>talk</replaceable> extension, if available.</para>
</description>
</application>
***/
static char *app = "BackgroundDetect";
static int background_detect_exec(struct ast_channel *chan, void *data)
{
int res = 0;
char *tmp;
struct ast_frame *fr;
int notsilent = 0;
struct timeval start = { 0, 0 };
struct timeval detection_start = { 0, 0 };
int sil = 1000;
int min = 100;
int max = -1;
int analysistime = -1;
int continue_analysis = 1;
int x;
int origrformat = 0;
struct ast_dsp *dsp = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
AST_APP_ARG(silence);
AST_APP_ARG(min);
AST_APP_ARG(max);
AST_APP_ARG(analysistime);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
return -1;
}
tmp = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, tmp);
if (!ast_strlen_zero(args.silence) && (sscanf(args.silence, "%30d", &x) == 1) && (x > 0)) {
sil = x;
}
if (!ast_strlen_zero(args.min) && (sscanf(args.min, "%30d", &x) == 1) && (x > 0)) {
min = x;
}
if (!ast_strlen_zero(args.max) && (sscanf(args.max, "%30d", &x) == 1) && (x > 0)) {
max = x;
}
if (!ast_strlen_zero(args.analysistime) && (sscanf(args.analysistime, "%30d", &x) == 1) && (x > 0)) {
analysistime = x;
}
ast_debug(1, "Preparing detect of '%s', sil=%d, min=%d, max=%d, analysistime=%d\n", args.filename, sil, min, max, analysistime);
do {
if (chan->_state != AST_STATE_UP) {
if ((res = ast_answer(chan))) {
break;
}
}
origrformat = chan->readformat;
if ((ast_set_read_format(chan, AST_FORMAT_SLINEAR))) {
ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
res = -1;
break;
}
if (!(dsp = ast_dsp_new())) {
ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
res = -1;
break;
}
ast_stopstream(chan);
if (ast_streamfile(chan, tmp, chan->language)) {
ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
break;
}
detection_start = ast_tvnow();
while (chan->stream) {
res = ast_sched_wait(chan->sched);
if ((res < 0) && !chan->timingfunc) {
res = 0;
break;
}
if (res < 0) {
res = 1000;
}
res = ast_waitfor(chan, res);
if (res < 0) {
ast_log(LOG_WARNING, "Waitfor failed on %s\n", chan->name);
break;
} else if (res > 0) {
fr = ast_read(chan);
if (continue_analysis && analysistime >= 0) {
/* If we have a limit for the time to analyze voice
* frames and the time has not expired */
if (ast_tvdiff_ms(ast_tvnow(), detection_start) >= analysistime) {
continue_analysis = 0;
ast_verb(3, "BackgroundDetect: Talk analysis time complete on %s.\n", chan->name);
}
}
if (!fr) {
res = -1;
break;
} else if (fr->frametype == AST_FRAME_DTMF) {
char t[2];
t[0] = fr->subclass;
t[1] = '\0';
if (ast_canmatch_extension(chan, chan->context, t, 1, chan->cid.cid_num)) {
/* They entered a valid extension, or might be anyhow */
res = fr->subclass;
ast_frfree(fr);
break;
}
} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass == AST_FORMAT_SLINEAR) && continue_analysis) {
int totalsilence;
int ms;
res = ast_dsp_silence(dsp, fr, &totalsilence);
if (res && (totalsilence > sil)) {
/* We've been quiet a little while */
if (notsilent) {
/* We had heard some talking */
ms = ast_tvdiff_ms(ast_tvnow(), start);
ms -= sil;
if (ms < 0)
ms = 0;
if ((ms > min) && ((max < 0) || (ms < max))) {
char ms_str[12];
ast_debug(1, "Found qualified token of %d ms\n", ms);
/* Save detected talk time (in milliseconds) */
snprintf(ms_str, sizeof(ms_str), "%d", ms);
pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
ast_goto_if_exists(chan, chan->context, "talk", 1);
res = 0;
ast_frfree(fr);
break;
} else {
ast_debug(1, "Found unqualified token of %d ms\n", ms);
}
notsilent = 0;
}
} else {
if (!notsilent) {
/* Heard some audio, mark the begining of the token */
start = ast_tvnow();
ast_debug(1, "Start of voice token!\n");
notsilent = 1;
}
}
}
ast_frfree(fr);
}
ast_sched_runq(chan->sched);
}
ast_stopstream(chan);
} while (0);
if (res > -1) {
if (origrformat && ast_set_read_format(chan, origrformat)) {
ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n",
chan->name, ast_getformatname(origrformat));
}
}
if (dsp) {
ast_dsp_free(dsp);
}
return res;
}
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
return ast_register_application_xml(app, background_detect_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Playback with Talk Detection");
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