258 lines
7.5 KiB
C
258 lines
7.5 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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* Mark Spencer <markster@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 Playback a file with audio detect
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*
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* \author Mark Spencer <markster@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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#include "asterisk/lock.h"
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#include "asterisk/file.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/module.h"
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#include "asterisk/translate.h"
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#include "asterisk/utils.h"
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#include "asterisk/dsp.h"
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#include "asterisk/app.h"
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#include "asterisk/format.h"
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#include "asterisk/format_cache.h"
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/*** DOCUMENTATION
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<application name="BackgroundDetect" language="en_US">
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<synopsis>
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Background a file with talk detect.
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</synopsis>
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<syntax>
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<parameter name="filename" required="true" />
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<parameter name="sil">
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<para>If not specified, defaults to <literal>1000</literal>.</para>
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</parameter>
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<parameter name="min">
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<para>If not specified, defaults to <literal>100</literal>.</para>
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</parameter>
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<parameter name="max">
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<para>If not specified, defaults to <literal>infinity</literal>.</para>
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</parameter>
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<parameter name="analysistime">
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<para>If not specified, defaults to <literal>infinity</literal>.</para>
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</parameter>
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</syntax>
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<description>
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<para>Plays back <replaceable>filename</replaceable>, waiting for interruption from a given digit (the digit
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must start the beginning of a valid extension, or it will be ignored). During
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the playback of the file, audio is monitored in the receive direction, and if
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a period of non-silence which is greater than <replaceable>min</replaceable> ms yet less than
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<replaceable>max</replaceable> ms is followed by silence for at least <replaceable>sil</replaceable> ms,
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which occurs during the first <replaceable>analysistime</replaceable> ms, then the audio playback is
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aborted and processing jumps to the <replaceable>talk</replaceable> extension, if available.</para>
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</description>
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</application>
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***/
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static char *app = "BackgroundDetect";
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static int background_detect_exec(struct ast_channel *chan, const char *data)
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{
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int res = 0;
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char *tmp;
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struct ast_frame *fr;
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int notsilent = 0;
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struct timeval start = { 0, 0 };
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struct timeval detection_start = { 0, 0 };
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int sil = 1000;
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int min = 100;
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int max = -1;
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int analysistime = -1;
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int continue_analysis = 1;
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int x;
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RAII_VAR(struct ast_format *, origrformat, NULL, ao2_cleanup);
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struct ast_dsp *dsp = NULL;
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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(min);
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AST_APP_ARG(max);
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AST_APP_ARG(analysistime);
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);
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if (ast_strlen_zero(data)) {
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ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
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return -1;
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}
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tmp = ast_strdupa(data);
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AST_STANDARD_APP_ARGS(args, tmp);
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if (!ast_strlen_zero(args.silence) && (sscanf(args.silence, "%30d", &x) == 1) && (x > 0)) {
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sil = x;
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}
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if (!ast_strlen_zero(args.min) && (sscanf(args.min, "%30d", &x) == 1) && (x > 0)) {
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min = x;
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}
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if (!ast_strlen_zero(args.max) && (sscanf(args.max, "%30d", &x) == 1) && (x > 0)) {
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max = x;
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}
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if (!ast_strlen_zero(args.analysistime) && (sscanf(args.analysistime, "%30d", &x) == 1) && (x > 0)) {
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analysistime = x;
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}
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ast_debug(1, "Preparing detect of '%s', sil=%d, min=%d, max=%d, analysistime=%d\n", args.filename, sil, min, max, analysistime);
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do {
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if (ast_channel_state(chan) != AST_STATE_UP) {
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if ((res = ast_answer(chan))) {
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break;
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}
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}
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origrformat = ao2_bump(ast_channel_readformat(chan));
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if ((ast_set_read_format(chan, ast_format_slin))) {
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ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
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res = -1;
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break;
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}
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if (!(dsp = ast_dsp_new())) {
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ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
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res = -1;
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break;
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}
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ast_stopstream(chan);
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if (ast_streamfile(chan, tmp, ast_channel_language(chan))) {
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ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", ast_channel_name(chan), (char *)data);
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break;
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}
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detection_start = ast_tvnow();
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while (ast_channel_stream(chan)) {
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res = ast_sched_wait(ast_channel_sched(chan));
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if ((res < 0) && !ast_channel_timingfunc(chan)) {
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res = 0;
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break;
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}
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if (res < 0) {
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res = 1000;
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}
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res = ast_waitfor(chan, res);
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if (res < 0) {
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ast_log(LOG_WARNING, "Waitfor failed on %s\n", ast_channel_name(chan));
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break;
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} else if (res > 0) {
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fr = ast_read(chan);
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if (continue_analysis && analysistime >= 0) {
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/* If we have a limit for the time to analyze voice
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* frames and the time has not expired */
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if (ast_tvdiff_ms(ast_tvnow(), detection_start) >= analysistime) {
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continue_analysis = 0;
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ast_verb(3, "BackgroundDetect: Talk analysis time complete on %s.\n", ast_channel_name(chan));
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}
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}
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if (!fr) {
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res = -1;
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break;
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} else if (fr->frametype == AST_FRAME_DTMF) {
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char t[2];
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t[0] = fr->subclass.integer;
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t[1] = '\0';
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if (ast_canmatch_extension(chan, ast_channel_context(chan), t, 1,
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S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
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/* They entered a valid extension, or might be anyhow */
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res = fr->subclass.integer;
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ast_frfree(fr);
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break;
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}
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} else if ((fr->frametype == AST_FRAME_VOICE) &&
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(ast_format_cmp(fr->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL) && continue_analysis) {
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int totalsilence;
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int ms;
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res = ast_dsp_silence(dsp, fr, &totalsilence);
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if (res && (totalsilence > sil)) {
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/* We've been quiet a little while */
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if (notsilent) {
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/* We had heard some talking */
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ms = ast_tvdiff_ms(ast_tvnow(), start);
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ms -= sil;
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if (ms < 0)
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ms = 0;
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if ((ms > min) && ((max < 0) || (ms < max))) {
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char ms_str[12];
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ast_debug(1, "Found qualified token of %d ms\n", ms);
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/* Save detected talk time (in milliseconds) */
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snprintf(ms_str, sizeof(ms_str), "%d", ms);
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pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
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ast_goto_if_exists(chan, ast_channel_context(chan), "talk", 1);
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res = 0;
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ast_frfree(fr);
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break;
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} else {
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ast_debug(1, "Found unqualified token of %d ms\n", ms);
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}
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notsilent = 0;
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}
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} else {
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if (!notsilent) {
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/* Heard some audio, mark the begining of the token */
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start = ast_tvnow();
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ast_debug(1, "Start of voice token!\n");
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notsilent = 1;
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}
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}
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}
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ast_frfree(fr);
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}
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ast_sched_runq(ast_channel_sched(chan));
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}
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ast_stopstream(chan);
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} while (0);
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if (res > -1) {
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if (origrformat && ast_set_read_format(chan, origrformat)) {
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ast_log(LOG_WARNING, "Failed to restore read format for %s to %s\n",
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ast_channel_name(chan), ast_format_get_name(origrformat));
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}
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}
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if (dsp) {
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ast_dsp_free(dsp);
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}
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return res;
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}
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static int unload_module(void)
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{
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return ast_unregister_application(app);
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}
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static int load_module(void)
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{
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return ast_register_application_xml(app, background_detect_exec);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Playback with Talk Detection");
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