git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
427 lines
13 KiB
C
427 lines
13 KiB
C
/*
|
|
* Asterisk -- An open source telephony toolkit.
|
|
*
|
|
* Copyright (C) 1999 - 2005, Digium, Inc.
|
|
*
|
|
* Matthew Fredrickson <creslin@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.
|
|
*/
|
|
|
|
/*! \file
|
|
*
|
|
* \brief Trivial application to record a sound file
|
|
*
|
|
* \author Matthew Fredrickson <creslin@digium.com>
|
|
*
|
|
* \ingroup applications
|
|
*/
|
|
|
|
#include "asterisk.h"
|
|
|
|
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
|
|
|
#include "asterisk/file.h"
|
|
#include "asterisk/pbx.h"
|
|
#include "asterisk/module.h"
|
|
#include "asterisk/app.h"
|
|
#include "asterisk/channel.h"
|
|
#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>
|
|
<option name="k">
|
|
<para>Keep recording if channel hangs up.</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.
|
|
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>
|
|
</variable>
|
|
</variablelist>
|
|
</description>
|
|
</application>
|
|
|
|
***/
|
|
|
|
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)
|
|
{
|
|
int res = 0;
|
|
int count = 0;
|
|
char *ext = NULL, *opts[0];
|
|
char *parse, *dir, *file;
|
|
int i = 0;
|
|
char tmp[256];
|
|
|
|
struct ast_filestream *s = NULL;
|
|
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 rfmt = 0;
|
|
int ioflags;
|
|
int waitres;
|
|
struct ast_silence_generator *silgen = NULL;
|
|
struct ast_flags flags = { 0, };
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(filename);
|
|
AST_APP_ARG(silence);
|
|
AST_APP_ARG(maxduration);
|
|
AST_APP_ARG(options);
|
|
);
|
|
|
|
/* 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 */
|
|
ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
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 */
|
|
if (!ext)
|
|
ext = strchr(args.filename, ':');
|
|
if (ext) {
|
|
*ext = '\0';
|
|
ext++;
|
|
}
|
|
}
|
|
if (!ext) {
|
|
ast_log(LOG_WARNING, "No extension specified to filename!\n");
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
return -1;
|
|
}
|
|
if (args.silence) {
|
|
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);
|
|
}
|
|
}
|
|
|
|
if (args.maxduration) {
|
|
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';
|
|
|
|
/* done parsing */
|
|
|
|
/* 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 *tmp2 = ast_strdupa(args.filename);
|
|
char countstring[15];
|
|
int idx;
|
|
|
|
/* Separate each piece out by the format specifier */
|
|
AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%');
|
|
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);
|
|
}
|
|
count++;
|
|
} while (ast_fileexists(tmp, ext, chan->language) > 0);
|
|
pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
|
|
} else
|
|
ast_copy_string(tmp, args.filename, sizeof(tmp));
|
|
/* end of routine mentioned */
|
|
|
|
if (chan->_state != AST_STATE_UP) {
|
|
if (ast_test_flag(&flags, OPTION_SKIP)) {
|
|
/* 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);
|
|
}
|
|
}
|
|
|
|
if (res) {
|
|
ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
goto out;
|
|
}
|
|
|
|
if (!ast_test_flag(&flags, OPTION_QUIET)) {
|
|
/* Some code to play a nice little beep to signify the start of the record operation */
|
|
res = ast_streamfile(chan, "beep", chan->language);
|
|
if (!res) {
|
|
res = ast_waitstream(chan, "");
|
|
} else {
|
|
ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", chan->name);
|
|
}
|
|
ast_stopstream(chan);
|
|
}
|
|
|
|
/* The end of beep code. Now the recording starts */
|
|
|
|
if (silence > 0) {
|
|
rfmt = chan->readformat;
|
|
res = ast_set_read_format(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");
|
|
return -1;
|
|
}
|
|
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);
|
|
|
|
if (!s) {
|
|
ast_log(LOG_WARNING, "Could not create file %s\n", args.filename);
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
goto out;
|
|
}
|
|
|
|
if (ast_opt_transmit_silence)
|
|
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");
|
|
break;
|
|
}
|
|
maxduration = waitres;
|
|
}
|
|
|
|
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");
|
|
ast_frfree(f);
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
|
|
break;
|
|
}
|
|
|
|
if (silence > 0) {
|
|
dspsilence = 0;
|
|
ast_dsp_silence(sildet, f, &dspsilence);
|
|
if (dspsilence) {
|
|
totalsilence = dspsilence;
|
|
} else {
|
|
totalsilence = 0;
|
|
}
|
|
if (totalsilence > silence) {
|
|
/* Ended happily with silence */
|
|
ast_frfree(f);
|
|
gotsilence = 1;
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SILENCE");
|
|
break;
|
|
}
|
|
}
|
|
} 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");
|
|
ast_frfree(f);
|
|
break;
|
|
}
|
|
} else if ((f->frametype == AST_FRAME_DTMF) &&
|
|
(f->subclass == terminator)) {
|
|
ast_frfree(f);
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "DTMF");
|
|
break;
|
|
}
|
|
ast_frfree(f);
|
|
}
|
|
if (!f) {
|
|
ast_debug(1, "Got hangup\n");
|
|
res = -1;
|
|
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "HANGUP");
|
|
if (!ast_test_flag(&flags, OPTION_KEEP)) {
|
|
ast_filedelete(args.filename, NULL);
|
|
}
|
|
}
|
|
|
|
if (gotsilence) {
|
|
ast_stream_rewind(s, silence - 1000);
|
|
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);
|
|
|
|
out:
|
|
if ((silence > 0) && rfmt) {
|
|
res = ast_set_read_format(chan, rfmt);
|
|
if (res)
|
|
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name);
|
|
if (sildet)
|
|
ast_dsp_free(sildet);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
return ast_unregister_application(app);
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
return ast_register_application_xml(app, record_exec);
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Trivial Record Application");
|