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asterisk/apps/app_record.c
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro 
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00

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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"
ASTERISK_REGISTER_FILE()
#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"
/*** 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="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.</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),
FLAG_HAS_PERCENT = (1 << 7),
OPTION_ANY_TERMINATE = (1 << 8),
OPTION_OPERATOR_EXIT = (1 << 9),
};
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('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
*
* \retval 0 do not exit
* \retval -1 do exit
*/
static int 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))) {
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "OPERATOR");
return -1;
}
if ((dtmf_integer == terminator) ||
(ast_test_flag(flags, OPTION_ANY_TERMINATE))) {
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "DTMF");
return -1;
}
return 0;
}
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 = '#';
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;
/* 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, ast_channel_language(chan)) > 0);
pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
} else
ast_copy_string(tmp, args.filename, sizeof(tmp));
/* end of routine mentioned */
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));
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", 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));
}
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));
}
/* 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;
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);
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) {
if (record_dtmf_response(chan, &flags, f->subclass.integer, terminator)) {
ast_frfree(f);
break;
}
}
ast_frfree(f);
}
if (maxduration > 0 && !ms) {
gottimeout = 1;
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "TIMEOUT");
}
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 && 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);
}
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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