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    /*
    
     * Asterisk -- An open source telephony toolkit.
    
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     *
    
     * Copyright (C) 1999 - 2005, Digium, Inc.
     *
     * Matthew Fredrickson <creslin@digium.com>
    
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     *
    
     * 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.
    
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     *
     * 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 Trivial application to record a sound file
    
     * \author Matthew Fredrickson <creslin@digium.com>
     *
    
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     * \ingroup applications
    
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     */
    
    
    /*** MODULEINFO
    	<support_level>core</support_level>
     ***/
    
    #include "asterisk.h"
    
    ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
    
    
    #include "asterisk/file.h"
    #include "asterisk/pbx.h"
    #include "asterisk/module.h"
    
    #include "asterisk/app.h"
    
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    #include "asterisk/dsp.h"	/* use dsp routines for silence detection */
    
    /*** DOCUMENTATION
    	<application name="Record" language="en_US">
    		<synopsis>
    			Record to a file.
    		</synopsis>
    		<syntax>
    			<parameter name="filename" required="true" argsep=".">
    				<argument name="filename" required="true" />
    				<argument name="format" required="true">
    					<para>Is the format of the file type to be recorded (wav, gsm, etc).</para>
    				</argument>
    			</parameter>
    			<parameter name="silence">
    				<para>Is the number of seconds of silence to allow before returning.</para>
    			</parameter>
    			<parameter name="maxduration">
    				<para>Is the maximum recording duration in seconds. If missing
    				or 0 there is no maximum.</para>
    			</parameter>
    			<parameter name="options">
    				<optionlist>
    					<option name="a">
    						<para>Append to existing recording rather than replacing.</para>
    					</option>
    					<option name="n">
    						<para>Do not answer, but record anyway if line not yet answered.</para>
    					</option>
    					<option name="q">
    						<para>quiet (do not play a beep tone).</para>
    					</option>
    					<option name="s">
    						<para>skip recording if the line is not yet answered.</para>
    					</option>
    					<option name="t">
    						<para>use alternate '*' terminator key (DTMF) instead of default '#'</para>
    					</option>
    					<option name="x">
    						<para>Ignore all terminator keys (DTMF) and keep recording until hangup.</para>
    					</option>
    
    					        <para>Keep recorded file upon hangup.</para>
    
    					</option>
    					<option name="y">
    					        <para>Terminate recording if *any* DTMF digit is received.</para>
    					</option>
    
    				</optionlist>
    			</parameter>
    		</syntax>
    		<description>
    			<para>If filename contains <literal>%d</literal>, these characters will be replaced with a number
    			incremented by one each time the file is recorded.
    			Use <astcli>core show file formats</astcli> to see the available formats on your system
    			User can press <literal>#</literal> to terminate the recording and continue to the next priority.
    
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    			If the user hangs up during a recording, all data will be lost and the application will terminate.</para>
    
    			<variablelist>
    				<variable name="RECORDED_FILE">
    					<para>Will be set to the final filename of the recording.</para>
    				</variable>
    
    				<variable name="RECORD_STATUS">
    					<para>This is the final status of the command</para>
    
    					<value name="DTMF">A terminating DTMF was received ('#' or '*', depending upon option 't')</value>
    					<value name="SILENCE">The maximum silence occurred in the recording.</value>
    					<value name="SKIP">The line was not yet answered and the 's' option was specified.</value>
    					<value name="TIMEOUT">The maximum length was reached.</value>
    					<value name="HANGUP">The channel was hung up.</value>
    					<value name="ERROR">An unrecoverable error occurred, which resulted in a WARNING to the logs.</value>
    
    			</variablelist>
    		</description>
    	</application>
    
     ***/
    
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    static char *app = "Record";
    
    
    enum {
    	OPTION_APPEND = (1 << 0),
    	OPTION_NOANSWER = (1 << 1),
    	OPTION_QUIET = (1 << 2),
    	OPTION_SKIP = (1 << 3),
    	OPTION_STAR_TERMINATE = (1 << 4),
    	OPTION_IGNORE_TERMINATE = (1 << 5),
    
    	OPTION_KEEP = (1 << 6),
    	FLAG_HAS_PERCENT = (1 << 7),
    
    };
    
    AST_APP_OPTIONS(app_opts,{
    	AST_APP_OPTION('a', OPTION_APPEND),
    
    	AST_APP_OPTION('k', OPTION_KEEP),	
    
    	AST_APP_OPTION('n', OPTION_NOANSWER),
    	AST_APP_OPTION('q', OPTION_QUIET),
    	AST_APP_OPTION('s', OPTION_SKIP),
    	AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
    
    	AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
    });
    
    static int record_exec(struct ast_channel *chan, const char *data)
    
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    {
    	int res = 0;
    	int count = 0;
    
    	char *ext = NULL, *opts[0];
    	char *parse, *dir, *file;
    
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    	int i = 0;
    
    	char tmp[256];
    
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    	struct ast_frame *f = NULL;
    
    	struct ast_dsp *sildet = NULL;   	/* silence detector dsp */
    
    	int totalsilence = 0;
    	int dspsilence = 0;
    	int silence = 0;		/* amount of silence to allow */
    	int gotsilence = 0;		/* did we timeout for silence? */
    
    	int maxduration = 0;		/* max duration of recording in milliseconds */
    
    	int gottimeout = 0;		/* did we timeout for maxduration exceeded? */
    
    	int terminator = '#';
    
    	int waitres;
    	struct ast_silence_generator *silgen = NULL;
    
    	AST_DECLARE_APP_ARGS(args,
    		AST_APP_ARG(filename);
    		AST_APP_ARG(silence);
    		AST_APP_ARG(maxduration);
    		AST_APP_ARG(options);
    	);
    
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    	/* The next few lines of code parse out the filename and header from the input string */
    
    	if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
    
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    		ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
    
    		pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
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    		return -1;
    	}
    
    	parse = ast_strdupa(data);
    	AST_STANDARD_APP_ARGS(args, parse);
    	if (args.argc == 4)
    		ast_app_parse_options(app_opts, &flags, opts, args.options);
    
    	if (!ast_strlen_zero(args.filename)) {
    		if (strstr(args.filename, "%d"))
    			ast_set_flag(&flags, FLAG_HAS_PERCENT);
    		ext = strrchr(args.filename, '.'); /* to support filename with a . in the filename, not format */
    
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    		if (ext) {
    			*ext = '\0';
    			ext++;
    		}
    
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    	}
    
    	if (!ext) {
    		ast_log(LOG_WARNING, "No extension specified to filename!\n");
    
    		pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
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    		if ((sscanf(args.silence, "%30d", &i) == 1) && (i > -1)) {
    
    			silence = i * 1000;
    
    		} else if (!ast_strlen_zero(args.silence)) {
    			ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
    
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    		if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
    
    			/* Convert duration to milliseconds */
    			maxduration = i * 1000;
    
    		else if (!ast_strlen_zero(args.maxduration))
    			ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", args.maxduration);
    
    
    	if (ast_test_flag(&flags, OPTION_STAR_TERMINATE))
    		terminator = '*';
    	if (ast_test_flag(&flags, OPTION_IGNORE_TERMINATE))
    		terminator = '\0';
    
    
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    	/* done parsing */
    
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    	/* these are to allow the use of the %d in the config file for a wild card of sort to
    	  create a new file with the inputed name scheme */
    
    	if (ast_test_flag(&flags, FLAG_HAS_PERCENT)) {
    
    		AST_DECLARE_APP_ARGS(fname,
    			AST_APP_ARG(piece)[100];
    		);
    
    		char countstring[15];
    
    
    		/* Separate each piece out by the format specifier */
    		AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%');
    
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    		do {
    
    			int tmplen;
    			/* First piece has no leading percent, so it's copied verbatim */
    			ast_copy_string(tmp, fname.piece[0], sizeof(tmp));
    			tmplen = strlen(tmp);
    
    			for (idx = 1; idx < fname.argc; idx++) {
    				if (fname.piece[idx][0] == 'd') {
    
    					/* Substitute the count */
    					snprintf(countstring, sizeof(countstring), "%d", count);
    					ast_copy_string(tmp + tmplen, countstring, sizeof(tmp) - tmplen);
    					tmplen += strlen(countstring);
    				} else if (tmplen + 2 < sizeof(tmp)) {
    					/* Unknown format specifier - just copy it verbatim */
    					tmp[tmplen++] = '%';
    
    					tmp[tmplen++] = fname.piece[idx][0];
    
    				}
    				/* Copy the remaining portion of the piece */
    
    				ast_copy_string(tmp + tmplen, &(fname.piece[idx][1]), sizeof(tmp) - tmplen);
    
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    			count++;
    
    		} while (ast_fileexists(tmp, ext, ast_channel_language(chan)) > 0);
    
    		pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
    
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    	} else
    
    		ast_copy_string(tmp, args.filename, sizeof(tmp));
    
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    	/* end of routine mentioned */
    
    	if (ast_channel_state(chan) != AST_STATE_UP) {
    
    			/* At the user's option, skip if the line is not up */
    
    			pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SKIP");
    
    			return 0;
    
    		} else if (!ast_test_flag(&flags, OPTION_NOANSWER)) {
    
    			/* Otherwise answer unless we're supposed to record while on-hook */
    			res = ast_answer(chan);
    		}
    
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    	}
    
    		ast_log(LOG_WARNING, "Could not answer channel '%s'\n", ast_channel_name(chan));
    
    		pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
    		/* Some code to play a nice little beep to signify the start of the record operation */
    
    		res = ast_streamfile(chan, "beep", ast_channel_language(chan));
    
    		if (!res) {
    			res = ast_waitstream(chan, "");
    		} else {
    
    			ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", ast_channel_name(chan));
    
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    		}
    
    	/* The end of beep code.  Now the recording starts */
    
    		ast_format_copy(&rfmt, ast_channel_readformat(chan));
    
    		res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
    
    		if (res < 0) {
    			ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
    
    			pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
    			return -1;
    		}
    		sildet = ast_dsp_new();
    		if (!sildet) {
    			ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
    
    			pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
    		ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
    
    
    	/* Create the directory if it does not exist. */
    	dir = ast_strdupa(tmp);
    	if ((file = strrchr(dir, '/')))
    		*file++ = '\0';
    	ast_mkdir (dir, 0777);
    
    
    	ioflags = ast_test_flag(&flags, OPTION_APPEND) ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
    	s = ast_writefile(tmp, ext, NULL, ioflags, 0, AST_FILE_MODE);
    
    
    		ast_log(LOG_WARNING, "Could not create file %s\n", args.filename);
    
    		pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
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    		silgen = ast_channel_start_silence_generator(chan);
    
    	/* Request a video update */
    	ast_indicate(chan, AST_CONTROL_VIDUPDATE);
    
    	if (maxduration <= 0)
    		maxduration = -1;
    
    	while ((waitres = ast_waitfor(chan, maxduration)) > -1) {
    		if (maxduration > 0) {
    			if (waitres == 0) {
    				gottimeout = 1;
    
    				pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "TIMEOUT");
    
    		f = ast_read(chan);
    		if (!f) {
    			res = -1;
    			break;
    		}
    		if (f->frametype == AST_FRAME_VOICE) {
    			res = ast_writestream(s, f);
    
    			if (res) {
    				ast_log(LOG_WARNING, "Problem writing frame\n");
    
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    				ast_frfree(f);
    
    				pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
    			if (silence > 0) {
    				dspsilence = 0;
    				ast_dsp_silence(sildet, f, &dspsilence);
    				if (dspsilence) {
    					totalsilence = dspsilence;
    				} else {
    					totalsilence = 0;
    
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    				}
    
    				if (totalsilence > silence) {
    					/* Ended happily with silence */
    
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    					ast_frfree(f);
    
    					pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SILENCE");
    
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    					break;
    				}
    
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    			}
    
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    		} else if (f->frametype == AST_FRAME_VIDEO) {
    
    			res = ast_writestream(s, f);
    
    			if (res) {
    				ast_log(LOG_WARNING, "Problem writing frame\n");
    
    				pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
    
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    				ast_frfree(f);
    
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    		} else if ((f->frametype == AST_FRAME_DTMF) &&
    
    			   ((f->subclass.integer == terminator) ||
    			    (ast_test_flag(&flags, OPTION_ANY_TERMINATE)))) {
    
    			pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "DTMF");
    
    		pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "HANGUP");
    
    		if (!ast_test_flag(&flags, OPTION_KEEP)) {
    			ast_filedelete(args.filename, NULL);
    		}
    
    		ast_truncstream(s);
    	} else if (!gottimeout) {
    		/* Strip off the last 1/4 second of it */
    		ast_stream_rewind(s, 250);
    		ast_truncstream(s);
    	}
    	ast_closestream(s);
    
    	if (silgen)
    		ast_channel_stop_silence_generator(chan, silgen);
    
    	if ((silence > 0) && rfmt.id) {
    		res = ast_set_read_format(chan, &rfmt);
    
    			ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(chan));
    
    		if (sildet)
    			ast_dsp_free(sildet);
    
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    	return res;
    }
    
    
    static int unload_module(void)
    
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    {
    
    	return ast_unregister_application(app);
    
    static int load_module(void)
    
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    {
    
    	return ast_register_application_xml(app, record_exec);
    
    AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial Record Application");