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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.
     */
    
    
     * \brief Playback a file with audio detect
    
     *
     * \author Mark Spencer <markster@digium.com>
    
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     * \ingroup applications
    
    	<support_level>core</support_level>
    
    #include "asterisk.h"
    
    
    #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/format.h"
    #include "asterisk/format_cache.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)
    
    	struct ast_frame *fr;
    
    	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;
    
    	RAII_VAR(struct ast_format *, origrformat, NULL, ao2_cleanup);
    
    	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);
    
    	if (ast_strlen_zero(data)) {
    
    		ast_log(LOG_WARNING, "BackgroundDetect requires an argument (filename)\n");
    		return -1;
    	}
    
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    	if (!ast_strlen_zero(args.silence) && (sscanf(args.silence, "%30d", &x) == 1) && (x > 0)) {
    
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    	if (!ast_strlen_zero(args.min) && (sscanf(args.min, "%30d", &x) == 1) && (x > 0)) {
    
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    	if (!ast_strlen_zero(args.max) && (sscanf(args.max, "%30d", &x) == 1) && (x > 0)) {
    
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    	if (!ast_strlen_zero(args.analysistime) && (sscanf(args.analysistime, "%30d", &x) == 1) && (x > 0)) {
    
    	ast_debug(1, "Preparing detect of '%s', sil=%d, min=%d, max=%d, analysistime=%d\n", args.filename, sil, min, max, analysistime);
    
    		if (ast_channel_state(chan) != AST_STATE_UP) {
    
    		origrformat = ao2_bump(ast_channel_readformat(chan));
    		if ((ast_set_read_format(chan, ast_format_slin))) {
    
    			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))) {
    
    			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", ast_channel_name(chan), (char *)data);
    
    		while (ast_channel_stream(chan)) {
    			res = ast_sched_wait(ast_channel_sched(chan));
    
    			if ((res < 0) && !ast_channel_timingfunc(chan)) {
    
    				ast_log(LOG_WARNING, "Waitfor failed on %s\n", ast_channel_name(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", ast_channel_name(chan));
    
    				} else if (fr->frametype == AST_FRAME_DTMF) {
    					char t[2];
    
    					t[0] = fr->subclass.integer;
    
    					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;
    
    				} else if ((fr->frametype == AST_FRAME_VOICE) &&
    				(ast_format_cmp(fr->subclass.format, ast_format_slin) == AST_FORMAT_CMP_EQUAL) && 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))) {
    
    								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_sched_runq(ast_channel_sched(chan));
    
    	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",
    
    				ast_channel_name(chan), ast_format_get_name(origrformat));
    
    		ast_dsp_free(dsp);
    
    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_EXTENDED(ASTERISK_GPL_KEY, "Playback with Talk Detection");