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"
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 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>
***/
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),
};
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('y', OPTION_ANY_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 = '#';
struct ast_format rfmt;
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);
);
Multiple revisions 375993-375994 ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-07 19:15:26 +00:00
int ms;
struct timeval start;
ast_format_clear(&rfmt);
Multiple revisions 375993-375994 ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-07 19:15:26 +00:00
/* 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) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
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 {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
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) {
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");
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;
Multiple revisions 375993-375994 ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-07 19:15:26 +00:00
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) &&
((f->subclass.integer == terminator) ||
(ast_test_flag(&flags, OPTION_ANY_TERMINATE)))) {
ast_frfree(f);
pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "DTMF");
break;
}
ast_frfree(f);
}
Multiple revisions 375993-375994 ........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-07 19:15:26 +00:00
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) {
/* 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.id) {
res = ast_set_read_format(chan, &rfmt);
Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 14:43:44 +00:00
if (res) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(chan));
Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 14:43:44 +00:00
}
}
Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 14:43:44 +00:00
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");