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asterisk/apps/app_record.c

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/*
* 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
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#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 */
#include "asterisk/format_cache.h"
#include "asterisk/paths.h"
/*** 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="o">
<para>Exit when 0 is pressed, setting the variable <variable>RECORD_STATUS</variable>
to <literal>OPERATOR</literal> instead of <literal>DTMF</literal></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="u">
<para>Don't truncate recorded silence.</para>
</option>
<option name="x">
<para>Ignore all terminator keys (DTMF) and keep recording until hangup.</para>
</option>
<option name="k">
<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.
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, without an extension.</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>
***/
#define OPERATOR_KEY '0'
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),
OPTION_ANY_TERMINATE = (1 << 7),
OPTION_OPERATOR_EXIT = (1 << 8),
OPTION_NO_TRUNCATE = (1 << 9),
};
enum dtmf_response {
RESPONSE_NO_MATCH = 0,
RESPONSE_OPERATOR,
RESPONSE_DTMF,
};
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('o', OPTION_OPERATOR_EXIT),
AST_APP_OPTION('q', OPTION_QUIET),
AST_APP_OPTION('s', OPTION_SKIP),
AST_APP_OPTION('t', OPTION_STAR_TERMINATE),
AST_APP_OPTION('u', OPTION_NO_TRUNCATE),
AST_APP_OPTION('y', OPTION_ANY_TERMINATE),
AST_APP_OPTION('x', OPTION_IGNORE_TERMINATE),
});
/*!
* \internal
* \brief Used to determine what action to take when DTMF is received while recording
* \since 13.0.0
*
* \param chan channel being recorded
* \param flags option flags in use by the record application
* \param dtmf_integer the integer value of the DTMF key received
* \param terminator key currently set to be pressed for normal termination
*
* \returns One of enum dtmf_response
*/
static enum dtmf_response record_dtmf_response(struct ast_channel *chan,
struct ast_flags *flags, int dtmf_integer, int terminator)
{
if ((dtmf_integer == OPERATOR_KEY) &&
(ast_test_flag(flags, OPTION_OPERATOR_EXIT))) {
return RESPONSE_OPERATOR;
}
if ((dtmf_integer == terminator) ||
(ast_test_flag(flags, OPTION_ANY_TERMINATE))) {
return RESPONSE_DTMF;
}
return RESPONSE_NO_MATCH;
}
static int create_destination_directory(const char *path)
{
int res;
char directory[PATH_MAX], *file_sep;
if (!(file_sep = strrchr(path, '/'))) {
/* No directory to create */
return 0;
}
/* Overwrite temporarily */
*file_sep = '\0';
/* Absolute path? */
if (path[0] == '/') {
res = ast_mkdir(path, 0777);
*file_sep = '/';
return res;
}
/* Relative path */
res = snprintf(directory, sizeof(directory), "%s/sounds/%s",
ast_config_AST_DATA_DIR, path);
*file_sep = '/';
if (res >= sizeof(directory)) {
/* We truncated, so we fail */
return -1;
}
return ast_mkdir(directory, 0777);
}
static int record_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
char *ext = NULL, *opts[0];
char *parse;
int i = 0;
char tmp[PATH_MAX];
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 truncate_silence = 1; /* truncate on complete silence recording */
int maxduration = 0; /* max duration of recording in milliseconds */
int gottimeout = 0; /* did we timeout for maxduration exceeded? */
int terminator = '#';
RAII_VAR(struct ast_format *, rfmt, NULL, ao2_cleanup);
int ioflags;
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);
);
int ms;
struct timeval start;
const char *status_response = "ERROR";
/* 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)) {
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 (ast_test_flag(&flags, OPTION_NO_TRUNCATE))
truncate_silence = 0;
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';
/*
If a '%d' is specified as part of the filename, we replace that token with
sequentially incrementing numbers until we find a unique filename.
*/
if (strchr(args.filename, '%')) {
size_t src, dst, count = 0;
size_t src_len = strlen(args.filename);
size_t dst_len = sizeof(tmp) - 1;
do {
for (src = 0, dst = 0; src < src_len && dst < dst_len; src++) {
if (!strncmp(&args.filename[src], "%d", 2)) {
int s = snprintf(&tmp[dst], PATH_MAX - dst, "%zu", count);
if (s >= PATH_MAX - dst) {
/* We truncated, so we need to bail */
ast_log(LOG_WARNING, "Failed to create unique filename from template: %s\n", args.filename);
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
return -1;
}
dst += s;
src++;
} else {
tmp[dst] = args.filename[src];
tmp[++dst] = '\0';
}
}
count++;
} while (ast_fileexists(tmp, ext, ast_channel_language(chan)) > 0);
} else
ast_copy_string(tmp, args.filename, sizeof(tmp));
pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
if (ast_channel_state(chan) != 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", ast_channel_name(chan));
status_response = "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", ast_channel_language(chan));
if (!res) {
res = ast_waitstream(chan, "");
} else {
ast_log(LOG_WARNING, "ast_streamfile(beep) failed on %s\n", ast_channel_name(chan));
res = 0;
}
ast_stopstream(chan);
}
/* The end of beep code. Now the recording starts */
if (silence > 0) {
rfmt = ao2_bump(ast_channel_readformat(chan));
res = ast_set_read_format(chan, ast_format_slin);
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));
}
if (create_destination_directory(tmp)) {
ast_log(LOG_WARNING, "Could not create directory for file %s\n", args.filename);
status_response = "ERROR";
goto out;
}
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);
status_response = "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;
start = ast_tvnow();
while ((ms = ast_remaining_ms(start, maxduration))) {
ms = ast_waitfor(chan, ms);
if (ms < 0) {
break;
}
if (maxduration > 0 && ms == 0) {
break;
}
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);
status_response = "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;
status_response = "SILENCE";
break;
}
}
} else if (f->frametype == AST_FRAME_VIDEO) {
res = ast_writestream(s, f);
if (res) {
ast_log(LOG_WARNING, "Problem writing frame\n");
status_response = "ERROR";
ast_frfree(f);
break;
}
} else if (f->frametype == AST_FRAME_DTMF) {
enum dtmf_response rc =
record_dtmf_response(chan, &flags, f->subclass.integer, terminator);
switch(rc) {
case RESPONSE_NO_MATCH:
break;
case RESPONSE_OPERATOR:
status_response = "OPERATOR";
ast_debug(1, "Got OPERATOR\n");
break;
case RESPONSE_DTMF:
status_response = "DTMF";
ast_debug(1, "Got DTMF\n");
break;
}
if (rc != RESPONSE_NO_MATCH) {
ast_frfree(f);
break;
}
}
ast_frfree(f);
}
if (maxduration > 0 && !ms) {
gottimeout = 1;
status_response = "TIMEOUT";
}
if (!f) {
ast_debug(1, "Got hangup\n");
res = -1;
status_response = "HANGUP";
if (!ast_test_flag(&flags, OPTION_KEEP)) {
ast_filedelete(args.filename, NULL);
}
}
if (gotsilence && truncate_silence) {
ast_stream_rewind(s, silence - 1000);
ast_truncstream(s);
} else if (!gottimeout && f) {
/*
* Strip off the last 1/4 second of it, if we didn't end because of a timeout,
* or a hangup. This must mean we ended because of a DTMF tone and while this
* 1/4 second stripping is very old code the most likely explanation is that it
* relates to stripping a partial DTMF tone.
*/
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", ast_channel_name(chan));
}
}
if (sildet) {
ast_dsp_free(sildet);
}
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", status_response);
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");
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