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	Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			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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| 
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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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| 
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| /*** MODULEINFO
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| 	<support_level>extended</support_level>
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|  ***/
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| 
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| #include "asterisk.h"
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| 
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| ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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| 
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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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| 
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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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| 
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| static char *app = "BackgroundDetect";
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| 
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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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| 	struct ast_format origrformat;
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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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| 
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| 	ast_format_clear(&origrformat);
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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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| 
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| 	tmp = ast_strdupa(data);
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| 	AST_STANDARD_APP_ARGS(args, tmp);
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| 
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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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| 
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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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| 
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| 		ast_format_copy(&origrformat, ast_channel_readformat(chan));
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| 		if ((ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR))) {
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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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| 
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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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| 				
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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) && (fr->subclass.format.id == AST_FORMAT_SLINEAR) && 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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| 
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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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| 
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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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| 
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| 	if (res > -1) {
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| 		if (origrformat.id && 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_getformatname(&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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| 
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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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| 
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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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| 
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| AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Playback with Talk Detection");
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