Files
asterisk/apps/app_mixmonitor.c
Russell Bryant cba19c8a67 Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big 
improvement for performance, stability, code maintainability, 
and ease of future code development.

The channel list is no longer an unsorted linked list.  The main container 
for channels is an astobj2 hash table.  All of the code related to searching 
for channels or iterating active channels has been rewritten.  Let n be 
the number of active channels.  Iterating the channel list has gone from 
O(n^2) to O(n).  Searching for a channel by name went from O(n) to O(1).  
Searching for a channel by extension is still O(n), but uses a new method 
for doing so, which is more efficient.

The ast_channel object is now a reference counted object.  The benefits 
here are plentiful.  Some benefits directly related to issues in the 
previous code include:

1) When threads other than the channel thread owning a channel wanted 
   access to a channel, it had to hold the lock on it to ensure that it didn't 
   go away.  This is no longer a requirement.  Holding a reference is 
   sufficient.

2) There are places that now require less dealing with channel locks.

3) There are places where channel locks are held for much shorter periods 
   of time.

4) There are places where dealing with more than one channel at a time becomes 
   _MUCH_ easier.  ChanSpy is a great example of this.  Writing code in the 
   future that deals with multiple channels will be much easier.

Some additional information regarding channel locking and reference count 
handling can be found in channel.h, where a new section has been added that 
discusses some of the rules associated with it.

Mark Michelson also assisted with the development of this patch.  He did the 
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it 
much easier to deal with holding on to a channel pointer for an extended period 
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.

Thanks to David Vossel for his assistance with this branch, as well.  David 
did the conversion of the DAHDIScan application by making it become a wrapper 
for ChanSpy internally.

The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.

Review: http://reviewboard.digium.com/r/203/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00

479 lines
14 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2005, Anthony Minessale II
* Copyright (C) 2005 - 2006, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
* Kevin P. Fleming <kpfleming@digium.com>
*
* Based on app_muxmon.c provided by
* Anthony Minessale II <anthmct@yahoo.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 MixMonitor() - Record a call and mix the audio during the recording
* \ingroup applications
*
* \author Mark Spencer <markster@digium.com>
* \author Kevin P. Fleming <kpfleming@digium.com>
*
* \note Based on app_muxmon.c provided by
* Anthony Minessale II <anthmct@yahoo.com>
*/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/paths.h" /* use ast_config_AST_MONITOR_DIR */
#include "asterisk/file.h"
#include "asterisk/audiohook.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/autochan.h"
/*** DOCUMENTATION
<application name="MixMonitor" language="en_US">
<synopsis>
Record a call and mix the audio during the recording.
</synopsis>
<syntax>
<parameter name="file" required="true" argsep=".">
<argument name="filename" required="true">
<para>If <replaceable>filename</replaceable> is an absolute path, uses that path, otherwise
creates the file in the configured monitoring directory from <filename>asterisk.conf.</filename></para>
</argument>
<argument name="extension" required="true" />
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Append to the file instead of overwriting it.</para>
</option>
<option name="b">
<para>Only save audio to the file while the channel is bridged.</para>
<note><para>Does not include conferences or sounds played to each bridged party</para></note>
</option>
<option name="v">
<para>Adjust the <emphasis>heard</emphasis> volume by a factor of <replaceable>x</replaceable>
(range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="V">
<para>Adjust the <emphasis>spoken</emphasis> volume by a factor
of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="W">
<para>Adjust both, <emphasis>heard and spoken</emphasis> volumes by a factor
of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
</optionlist>
</parameter>
<parameter name="command">
<para>Will be executed when the recording is over.</para>
<para>Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.</para>
<para>All variables will be evaluated at the time MixMonitor is called.</para>
</parameter>
</syntax>
<description>
<para>Records the audio on the current channel to the specified file.</para>
<variablelist>
<variable name="MIXMONITOR_FILENAME">
<para>Will contain the filename used to record.</para>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">Monitor</ref>
<ref type="application">StopMixMonitor</ref>
<ref type="application">PauseMonitor</ref>
<ref type="application">UnpauseMonitor</ref>
</see-also>
</application>
<application name="StopMixMonitor" language="en_US">
<synopsis>
Stop recording a call through MixMonitor.
</synopsis>
<syntax />
<description>
<para>Stops the audio recording that was started with a call to <literal>MixMonitor()</literal>
on the current channel.</para>
</description>
<see-also>
<ref type="application">MixMonitor</ref>
</see-also>
</application>
***/
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
static const char *app = "MixMonitor";
static const char *stop_app = "StopMixMonitor";
struct module_symbols *me;
static const char *mixmonitor_spy_type = "MixMonitor";
struct mixmonitor {
struct ast_audiohook audiohook;
char *filename;
char *post_process;
char *name;
unsigned int flags;
struct ast_autochan *autochan;
};
enum {
MUXFLAG_APPEND = (1 << 1),
MUXFLAG_BRIDGED = (1 << 2),
MUXFLAG_VOLUME = (1 << 3),
MUXFLAG_READVOLUME = (1 << 4),
MUXFLAG_WRITEVOLUME = (1 << 5),
} mixmonitor_flags;
enum {
OPT_ARG_READVOLUME = 0,
OPT_ARG_WRITEVOLUME,
OPT_ARG_VOLUME,
OPT_ARG_ARRAY_SIZE,
} mixmonitor_args;
AST_APP_OPTIONS(mixmonitor_opts, {
AST_APP_OPTION('a', MUXFLAG_APPEND),
AST_APP_OPTION('b', MUXFLAG_BRIDGED),
AST_APP_OPTION_ARG('v', MUXFLAG_READVOLUME, OPT_ARG_READVOLUME),
AST_APP_OPTION_ARG('V', MUXFLAG_WRITEVOLUME, OPT_ARG_WRITEVOLUME),
AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
});
static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
struct ast_channel *peer = NULL;
int res = 0;
if (!chan)
return -1;
ast_audiohook_attach(chan, audiohook);
if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
return res;
}
#define SAMPLES_PER_FRAME 160
static void *mixmonitor_thread(void *obj)
{
struct mixmonitor *mixmonitor = obj;
struct ast_filestream *fs = NULL;
unsigned int oflags;
char *ext;
int errflag = 0;
ast_verb(2, "Begin MixMonitor Recording %s\n", mixmonitor->name);
ast_audiohook_lock(&mixmonitor->audiohook);
while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) {
struct ast_frame *fr = NULL;
ast_audiohook_trigger_wait(&mixmonitor->audiohook);
if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING)
break;
if (!(fr = ast_audiohook_read_frame(&mixmonitor->audiohook, SAMPLES_PER_FRAME, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR)))
continue;
if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || (mixmonitor->autochan->chan && ast_bridged_channel(mixmonitor->autochan->chan))) {
/* Initialize the file if not already done so */
if (!fs && !errflag) {
oflags = O_CREAT | O_WRONLY;
oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
if ((ext = strrchr(mixmonitor->filename, '.')))
*(ext++) = '\0';
else
ext = "raw";
if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0666))) {
ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
errflag = 1;
}
}
/* Write out frame */
if (fs)
ast_writestream(fs, fr);
}
/* All done! free it. */
ast_frame_free(fr, 0);
}
ast_audiohook_detach(&mixmonitor->audiohook);
ast_audiohook_unlock(&mixmonitor->audiohook);
ast_audiohook_destroy(&mixmonitor->audiohook);
ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
if (fs)
ast_closestream(fs);
if (mixmonitor->post_process) {
ast_verb(2, "Executing [%s]\n", mixmonitor->post_process);
ast_safe_system(mixmonitor->post_process);
}
ast_autochan_destroy(mixmonitor->autochan);
ast_free(mixmonitor);
return NULL;
}
static void launch_monitor_thread(struct ast_channel *chan, const char *filename, unsigned int flags,
int readvol, int writevol, const char *post_process)
{
pthread_t thread;
struct mixmonitor *mixmonitor;
char postprocess2[1024] = "";
size_t len;
len = sizeof(*mixmonitor) + strlen(chan->name) + strlen(filename) + 2;
postprocess2[0] = 0;
/* If a post process system command is given attach it to the structure */
if (!ast_strlen_zero(post_process)) {
char *p1, *p2;
p1 = ast_strdupa(post_process);
for (p2 = p1; *p2 ; p2++) {
if (*p2 == '^' && *(p2+1) == '{') {
*p2 = '$';
}
}
pbx_substitute_variables_helper(chan, p1, postprocess2, sizeof(postprocess2) - 1);
if (!ast_strlen_zero(postprocess2))
len += strlen(postprocess2) + 1;
}
/* Pre-allocate mixmonitor structure and spy */
if (!(mixmonitor = ast_calloc(1, len))) {
return;
}
/* Copy over flags and channel name */
mixmonitor->flags = flags;
if (!(mixmonitor->autochan = ast_autochan_setup(chan))) {
return;
}
mixmonitor->name = (char *) mixmonitor + sizeof(*mixmonitor);
strcpy(mixmonitor->name, chan->name);
if (!ast_strlen_zero(postprocess2)) {
mixmonitor->post_process = mixmonitor->name + strlen(mixmonitor->name) + strlen(filename) + 2;
strcpy(mixmonitor->post_process, postprocess2);
}
mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor) + strlen(chan->name) + 1;
strcpy(mixmonitor->filename, filename);
/* Setup the actual spy before creating our thread */
if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) {
ast_free(mixmonitor);
return;
}
ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
if (readvol)
mixmonitor->audiohook.options.read_volume = readvol;
if (writevol)
mixmonitor->audiohook.options.write_volume = writevol;
if (startmon(chan, &mixmonitor->audiohook)) {
ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
mixmonitor_spy_type, chan->name);
ast_audiohook_destroy(&mixmonitor->audiohook);
ast_free(mixmonitor);
return;
}
ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
}
static int mixmonitor_exec(struct ast_channel *chan, void *data)
{
int x, readvol = 0, writevol = 0;
struct ast_flags flags = {0};
char *parse, *tmp, *slash;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
AST_APP_ARG(options);
AST_APP_ARG(post_process);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (ast_strlen_zero(args.filename)) {
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
return -1;
}
if (args.options) {
char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
ast_app_parse_options(mixmonitor_opts, &flags, opts, args.options);
if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
if (ast_strlen_zero(opts[OPT_ARG_READVOLUME])) {
ast_log(LOG_WARNING, "No volume level was provided for the heard volume ('v') option.\n");
} else if ((sscanf(opts[OPT_ARG_READVOLUME], "%d", &x) != 1) || (x < -4) || (x > 4)) {
ast_log(LOG_NOTICE, "Heard volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_READVOLUME]);
} else {
readvol = get_volfactor(x);
}
}
if (ast_test_flag(&flags, MUXFLAG_WRITEVOLUME)) {
if (ast_strlen_zero(opts[OPT_ARG_WRITEVOLUME])) {
ast_log(LOG_WARNING, "No volume level was provided for the spoken volume ('V') option.\n");
} else if ((sscanf(opts[OPT_ARG_WRITEVOLUME], "%d", &x) != 1) || (x < -4) || (x > 4)) {
ast_log(LOG_NOTICE, "Spoken volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_WRITEVOLUME]);
} else {
writevol = get_volfactor(x);
}
}
if (ast_test_flag(&flags, MUXFLAG_VOLUME)) {
if (ast_strlen_zero(opts[OPT_ARG_VOLUME])) {
ast_log(LOG_WARNING, "No volume level was provided for the combined volume ('W') option.\n");
} else if ((sscanf(opts[OPT_ARG_VOLUME], "%d", &x) != 1) || (x < -4) || (x > 4)) {
ast_log(LOG_NOTICE, "Combined volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_VOLUME]);
} else {
readvol = writevol = get_volfactor(x);
}
}
}
/* if not provided an absolute path, use the system-configured monitoring directory */
if (args.filename[0] != '/') {
char *build;
build = alloca(strlen(ast_config_AST_MONITOR_DIR) + strlen(args.filename) + 3);
sprintf(build, "%s/%s", ast_config_AST_MONITOR_DIR, args.filename);
args.filename = build;
}
tmp = ast_strdupa(args.filename);
if ((slash = strrchr(tmp, '/')))
*slash = '\0';
ast_mkdir(tmp, 0777);
pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process);
return 0;
}
static int stop_mixmonitor_exec(struct ast_channel *chan, void *data)
{
ast_audiohook_detach_source(chan, mixmonitor_spy_type);
return 0;
}
static char *handle_cli_mixmonitor(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_channel *chan;
switch (cmd) {
case CLI_INIT:
e->command = "mixmonitor {start|stop} {<chan_name>} [args]";
e->usage =
"Usage: mixmonitor <start|stop> <chan_name> [args]\n"
" The optional arguments are passed to the MixMonitor\n"
" application when the 'start' command is used.\n";
return NULL;
case CLI_GENERATE:
return ast_complete_channels(a->line, a->word, a->pos, a->n, 2);
}
if (a->argc < 3)
return CLI_SHOWUSAGE;
if (!(chan = ast_channel_get_by_name_prefix(a->argv[2], strlen(a->argv[2])))) {
ast_cli(a->fd, "No channel matching '%s' found.\n", a->argv[2]);
/* Technically this is a failure, but we don't want 2 errors printing out */
return CLI_SUCCESS;
}
ast_channel_lock(chan);
if (!strcasecmp(a->argv[1], "start")) {
mixmonitor_exec(chan, a->argv[3]);
ast_channel_unlock(chan);
} else {
ast_channel_unlock(chan);
ast_audiohook_detach_source(chan, mixmonitor_spy_type);
}
chan = ast_channel_unref(chan);
return CLI_SUCCESS;
}
static struct ast_cli_entry cli_mixmonitor[] = {
AST_CLI_DEFINE(handle_cli_mixmonitor, "Execute a MixMonitor command")
};
static int unload_module(void)
{
int res;
ast_cli_unregister_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
res = ast_unregister_application(stop_app);
res |= ast_unregister_application(app);
return res;
}
static int load_module(void)
{
int res;
ast_cli_register_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
res = ast_register_application_xml(app, mixmonitor_exec);
res |= ast_register_application_xml(stop_app, stop_mixmonitor_exec);
return res;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mixed Audio Monitoring Application");