asterisk/apps/app_talkdetect.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Playback a file with audio detect
*
* \author Mark Spencer <markster@digium.com>
*
* \ingroup applications
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#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, const char *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;
struct ast_format origrformat;
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);
);
ast_format_clear(&origrformat);
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 (ast_channel_state(chan) != AST_STATE_UP) {
if ((res = ast_answer(chan))) {
break;
}
}
ast_format_copy(&origrformat, ast_channel_readformat(chan));
if ((ast_set_read_format_by_id(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, ast_channel_language(chan))) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", ast_channel_name(chan), (char *)data);
break;
}
detection_start = ast_tvnow();
while (ast_channel_stream(chan)) {
res = ast_sched_wait(ast_channel_sched(chan));
if ((res < 0) && !chan->timingfunc) {
res = 0;
break;
}
if (res < 0) {
res = 1000;
}
res = ast_waitfor(chan, res);
if (res < 0) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, "Waitfor failed on %s\n", ast_channel_name(chan));
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;
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ast_verb(3, "BackgroundDetect: Talk analysis time complete on %s.\n", ast_channel_name(chan));
}
}
if (!fr) {
res = -1;
break;
} else if (fr->frametype == AST_FRAME_DTMF) {
char t[2];
t[0] = fr->subclass.integer;
t[1] = '\0';
if (ast_canmatch_extension(chan, ast_channel_context(chan), t, 1,
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
/* They entered a valid extension, or might be anyhow */
res = fr->subclass.integer;
ast_frfree(fr);
break;
}
} else if ((fr->frametype == AST_FRAME_VOICE) && (fr->subclass.format.id == 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, ast_channel_context(chan), "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(ast_channel_sched(chan));
}
ast_stopstream(chan);
} while (0);
if (res > -1) {
if (origrformat.id && ast_set_read_format(chan, &origrformat)) {
ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n",
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_channel_name(chan), 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");