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	patch provided in bugnote, with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			234 lines
		
	
	
		
			6.2 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			234 lines
		
	
	
		
			6.2 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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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include "asterisk/lock.h"
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#include "asterisk/file.h"
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#include "asterisk/logger.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/options.h"
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static char *app = "BackgroundDetect";
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static char *synopsis = "Background a file with talk detect";
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static char *descrip = 
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"  BackgroundDetect(filename[|sil[|min|[max]]]):  Plays  back  a  given\n"
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"filename, waiting for interruption from a given digit (the digit must\n"
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"start the beginning of a valid extension, or it will be ignored).\n"
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"During the playback of the file, audio is monitored in the receive\n"
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"direction, and if a period of non-silence which is greater than 'min' ms\n"
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"yet less than 'max' ms is followed by silence for at least 'sil' ms then\n"
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"the audio playback is aborted and processing jumps to the 'talk' extension\n"
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"if available.  If unspecified, sil, min, and max default to 1000, 100, and\n"
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"infinity respectively.\n";
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static int background_detect_exec(struct ast_channel *chan, void *data)
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{
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	int res = 0;
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	struct ast_module_user *u;
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	char *tmp;
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	char *options;
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	char *stringp;
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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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	int sil = 1000;
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	int min = 100;
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	int max = -1;
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	int x;
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	int origrformat=0;
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	struct ast_dsp *dsp;
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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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	u = ast_module_user_add(chan);
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	tmp = ast_strdupa(data);
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	stringp=tmp;
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	strsep(&stringp, "|");
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	options = strsep(&stringp, "|");
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	if (options) {
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		if ((sscanf(options, "%d", &x) == 1) && (x > 0))
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			sil = x;
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		options = strsep(&stringp, "|");
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		if (options) {
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			if ((sscanf(options, "%d", &x) == 1) && (x > 0))
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				min = x;
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			options = strsep(&stringp, "|");
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			if (options) {
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				if ((sscanf(options, "%d", &x) == 1) && (x > 0))
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					max = x;
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			}
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		}
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	}
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	if (option_debug)
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		ast_log(LOG_DEBUG, "Preparing detect of '%s', sil=%d,min=%d,max=%d\n", 
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						tmp, sil, min, max);
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	if (chan->_state != AST_STATE_UP) {
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		/* Otherwise answer unless we're supposed to send this while on-hook */
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		res = ast_answer(chan);
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	}
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	if (!res) {
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		origrformat = chan->readformat;
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		if ((res = ast_set_read_format(chan, AST_FORMAT_SLINEAR))) 
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			ast_log(LOG_WARNING, "Unable to set read format to linear!\n");
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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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	}
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	if (!res) {
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		ast_stopstream(chan);
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		res = ast_streamfile(chan, tmp, chan->language);
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		if (!res) {
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			while(chan->stream) {
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				res = ast_sched_wait(chan->sched);
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				if ((res < 0) && !chan->timingfunc) {
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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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				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", chan->name);
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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 (!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;
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						t[1] = '\0';
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						if (ast_canmatch_extension(chan, chan->context, t, 1, chan->cid.cid_num)) {
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							/* They entered a valid  extension, or might be anyhow */
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							res = fr->subclass;
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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 == AST_FORMAT_SLINEAR)) {
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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[10];
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									if (option_debug)
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										ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
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									/* Save detected talk time (in milliseconds) */ 
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									sprintf(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, chan->context, "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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									if (option_debug)
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										ast_log(LOG_DEBUG, "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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								if (option_debug)
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									ast_log(LOG_DEBUG, "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(chan->sched);
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			}
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			ast_stopstream(chan);
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		} else {
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			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
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			res = 0;
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		}
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	}
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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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				chan->name, 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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	ast_module_user_remove(u);
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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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	int res;
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	res = ast_unregister_application(app);
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	ast_module_user_hangup_all();
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	return res;	
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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(app, background_detect_exec, synopsis, descrip);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Playback with Talk Detection");
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