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asterisk/apps/app_mixmonitor.c
Terry Wilson 04da92c379 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

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/*
* 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>
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#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"
#include "asterisk/manager.h"
#include "asterisk/mod_format.h"
/*** DOCUMENTATION
<application name="MixMonitor" language="en_US">
<synopsis>
Record a call and mix the audio during the recording. Use of StopMixMonitor is required
to guarantee the audio file is available for processing during dialplan execution.
</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>
<note><para>If you utilize this option inside a Local channel, you must make sure the Local
channel is not optimized away. To do this, be sure to call your Local channel with the
<literal>/n</literal> option. For example: Dial(Local/start@mycontext/n)</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>
<option name="r">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>receive</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</option>
<option name="t">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>transmit</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</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>
<para>This application does not automatically answer and should be preceeded by
an application such as Answer or Progress().</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, and free the recording's file handle.
</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>
<manager name="MixMonitorMute" language="en_US">
<synopsis>
Mute / unMute a Mixmonitor recording.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to mute.</para>
</parameter>
<parameter name="Direction">
<para>Which part of the recording to mute: read, write or both (from channel, to channel or both channels).</para>
</parameter>
<parameter name="State">
<para>Turn mute on or off : 1 to turn on, 0 to turn off.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to mute a MixMonitor recording.</para>
</description>
</manager>
<manager name="MixMonitor" language="en_US">
<synopsis>
Record a call and mix the audio during the recording. Use of StopMixMonitor is required
to guarantee the audio file is available for processing during dialplan execution.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to record.</para>
</parameter>
<parameter name="File">
<para>Is the name of the file created in the monitor spool directory.
Defaults to the same name as the channel (with slashes replaced with dashes).
This argument is optional if you specify to record unidirectional audio with
either the r(filename) or t(filename) options in the options field. If
neither MIXMONITOR_FILENAME or this parameter is set, the mixed stream won't
be recorded.</para>
</parameter>
<parameter name="options">
<para>Options that apply to the MixMonitor in the same way as they
would apply if invoked from the MixMonitor application. For a list of
available options, see the documentation for the mixmonitor application. </para>
</parameter>
</syntax>
<description>
<para>This action 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 the mixed stream.</para>
</variable>
</variablelist>
</description>
</manager>
<manager name="StopMixMonitor" language="en_US">
<synopsis>
Stop recording a call through MixMonitor, and free the recording's file handle.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The name of the channel monitored.</para>
</parameter>
</syntax>
<description>
<para>This action stops the audio recording that was started with the <literal>MixMonitor</literal>
action on the current channel.</para>
</description>
</manager>
***/
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
static const char * const app = "MixMonitor";
static const char * const stop_app = "StopMixMonitor";
static const char * const mixmonitor_spy_type = "MixMonitor";
struct mixmonitor {
struct ast_audiohook audiohook;
char *filename;
char *filename_read;
char *filename_write;
char *post_process;
char *name;
unsigned int flags;
struct ast_autochan *autochan;
struct mixmonitor_ds *mixmonitor_ds;
};
enum mixmonitor_flags {
MUXFLAG_APPEND = (1 << 1),
MUXFLAG_BRIDGED = (1 << 2),
MUXFLAG_VOLUME = (1 << 3),
MUXFLAG_READVOLUME = (1 << 4),
MUXFLAG_WRITEVOLUME = (1 << 5),
MUXFLAG_READ = (1 << 6),
MUXFLAG_WRITE = (1 << 7),
MUXFLAG_COMBINED = (1 << 8),
};
enum mixmonitor_args {
OPT_ARG_READVOLUME = 0,
OPT_ARG_WRITEVOLUME,
OPT_ARG_VOLUME,
OPT_ARG_WRITENAME,
OPT_ARG_READNAME,
OPT_ARG_ARRAY_SIZE, /* Always last element of the enum */
};
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),
AST_APP_OPTION_ARG('r', MUXFLAG_READ, OPT_ARG_READNAME),
AST_APP_OPTION_ARG('t', MUXFLAG_WRITE, OPT_ARG_WRITENAME),
});
struct mixmonitor_ds {
unsigned int destruction_ok;
ast_cond_t destruction_condition;
ast_mutex_t lock;
/* The filestream is held in the datastore so it can be stopped
* immediately during stop_mixmonitor or channel destruction. */
int fs_quit;
struct ast_filestream *fs;
struct ast_filestream *fs_read;
struct ast_filestream *fs_write;
struct ast_audiohook *audiohook;
unsigned int samp_rate;
};
/*!
* \internal
* \pre mixmonitor_ds must be locked before calling this function
*/
static void mixmonitor_ds_close_fs(struct mixmonitor_ds *mixmonitor_ds)
{
unsigned char quitting = 0;
if (mixmonitor_ds->fs) {
quitting = 1;
ast_closestream(mixmonitor_ds->fs);
mixmonitor_ds->fs = NULL;
ast_verb(2, "MixMonitor close filestream (mixed)\n");
}
if (mixmonitor_ds->fs_read) {
quitting = 1;
ast_closestream(mixmonitor_ds->fs_read);
mixmonitor_ds->fs_read = NULL;
ast_verb(2, "MixMonitor close filestream (read)\n");
}
if (mixmonitor_ds->fs_write) {
quitting = 1;
ast_closestream(mixmonitor_ds->fs_write);
mixmonitor_ds->fs_write = NULL;
ast_verb(2, "MixMonitor close filestream (write)\n");
}
if (quitting) {
mixmonitor_ds->fs_quit = 1;
}
}
static void mixmonitor_ds_destroy(void *data)
{
struct mixmonitor_ds *mixmonitor_ds = data;
ast_mutex_lock(&mixmonitor_ds->lock);
mixmonitor_ds->audiohook = NULL;
mixmonitor_ds->destruction_ok = 1;
ast_cond_signal(&mixmonitor_ds->destruction_condition);
ast_mutex_unlock(&mixmonitor_ds->lock);
}
static struct ast_datastore_info mixmonitor_ds_info = {
.type = "mixmonitor",
.destroy = mixmonitor_ds_destroy,
};
static void destroy_monitor_audiohook(struct mixmonitor *mixmonitor)
{
if (mixmonitor->mixmonitor_ds) {
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
mixmonitor->mixmonitor_ds->audiohook = NULL;
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
}
/* kill the audiohook.*/
ast_audiohook_lock(&mixmonitor->audiohook);
ast_audiohook_detach(&mixmonitor->audiohook);
ast_audiohook_unlock(&mixmonitor->audiohook);
ast_audiohook_destroy(&mixmonitor->audiohook);
}
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_free(struct mixmonitor *mixmonitor)
{
if (mixmonitor) {
if (mixmonitor->mixmonitor_ds) {
ast_mutex_destroy(&mixmonitor->mixmonitor_ds->lock);
ast_cond_destroy(&mixmonitor->mixmonitor_ds->destruction_condition);
ast_free(mixmonitor->filename_write);
ast_free(mixmonitor->filename_read);
ast_free(mixmonitor->mixmonitor_ds);
ast_free(mixmonitor->name);
ast_free(mixmonitor->post_process);
}
ast_free(mixmonitor);
}
}
static void mixmonitor_save_prep(struct mixmonitor *mixmonitor, char *filename, struct ast_filestream **fs, unsigned int *oflags, int *errflag)
{
/* Initialize the file if not already done so */
char *ext = NULL;
char *last_slash = NULL;
if (!ast_strlen_zero(filename)) {
if (!*fs && !*errflag && !mixmonitor->mixmonitor_ds->fs_quit) {
*oflags = O_CREAT | O_WRONLY;
*oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
last_slash = strrchr(filename, '/');
if ((ext = strrchr(filename, '.')) && (ext > last_slash)) {
*(ext++) = '\0';
} else {
ext = "raw";
}
if (!(*fs = ast_writefile(filename, ext, NULL, *oflags, 0, 0666))) {
ast_log(LOG_ERROR, "Cannot open %s.%s\n", filename, ext);
*errflag = 1;
} else {
struct ast_filestream *tmp = *fs;
mixmonitor->mixmonitor_ds->samp_rate = MAX(mixmonitor->mixmonitor_ds->samp_rate, ast_format_rate(&tmp->fmt->format));
}
}
}
}
static void *mixmonitor_thread(void *obj)
{
struct mixmonitor *mixmonitor = obj;
struct ast_filestream **fs = NULL;
struct ast_filestream **fs_read = NULL;
struct ast_filestream **fs_write = NULL;
unsigned int oflags;
int errflag = 0;
struct ast_format format_slin;
ast_verb(2, "Begin MixMonitor Recording %s\n", mixmonitor->name);
fs = &mixmonitor->mixmonitor_ds->fs;
fs_read = &mixmonitor->mixmonitor_ds->fs_read;
fs_write = &mixmonitor->mixmonitor_ds->fs_write;
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
mixmonitor_save_prep(mixmonitor, mixmonitor->filename, fs, &oflags, &errflag);
mixmonitor_save_prep(mixmonitor, mixmonitor->filename_read, fs_read, &oflags, &errflag);
mixmonitor_save_prep(mixmonitor, mixmonitor->filename_write, fs_write, &oflags, &errflag);
ast_format_set(&format_slin, ast_format_slin_by_rate(mixmonitor->mixmonitor_ds->samp_rate), 0);
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
/* The audiohook must enter and exit the loop locked */
ast_audiohook_lock(&mixmonitor->audiohook);
while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING && !mixmonitor->mixmonitor_ds->fs_quit) {
struct ast_frame *fr = NULL;
struct ast_frame *fr_read = NULL;
struct ast_frame *fr_write = NULL;
if (!(fr = ast_audiohook_read_frame_all(&mixmonitor->audiohook, SAMPLES_PER_FRAME, &format_slin,
&fr_read, &fr_write))) {
ast_audiohook_trigger_wait(&mixmonitor->audiohook);
if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
break;
}
continue;
}
/* audiohook lock is not required for the next block.
* Unlock it, but remember to lock it before looping or exiting */
ast_audiohook_unlock(&mixmonitor->audiohook);
if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || (mixmonitor->autochan->chan && ast_bridged_channel(mixmonitor->autochan->chan))) {
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
/* Write out the frame(s) */
if ((*fs_read) && (fr_read)) {
struct ast_frame *cur;
for (cur = fr_read; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
ast_writestream(*fs_read, cur);
}
}
if ((*fs_write) && (fr_write)) {
struct ast_frame *cur;
for (cur = fr_write; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
ast_writestream(*fs_write, cur);
}
}
if ((*fs) && (fr)) {
struct ast_frame *cur;
for (cur = fr; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
ast_writestream(*fs, cur);
}
}
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
}
/* All done! free it. */
if (fr) {
ast_frame_free(fr, 0);
}
if (fr_read) {
ast_frame_free(fr_read, 0);
}
if (fr_write) {
ast_frame_free(fr_write, 0);
}
fr = NULL;
fr_write = NULL;
fr_read = NULL;
ast_audiohook_lock(&mixmonitor->audiohook);
}
ast_audiohook_unlock(&mixmonitor->audiohook);
ast_autochan_destroy(mixmonitor->autochan);
/* Datastore cleanup. close the filestream and wait for ds destruction */
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
mixmonitor_ds_close_fs(mixmonitor->mixmonitor_ds);
if (!mixmonitor->mixmonitor_ds->destruction_ok) {
ast_cond_wait(&mixmonitor->mixmonitor_ds->destruction_condition, &mixmonitor->mixmonitor_ds->lock);
}
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
/* kill the audiohook */
destroy_monitor_audiohook(mixmonitor);
if (mixmonitor->post_process) {
ast_verb(2, "Executing [%s]\n", mixmonitor->post_process);
ast_safe_system(mixmonitor->post_process);
}
ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
mixmonitor_free(mixmonitor);
return NULL;
}
static int setup_mixmonitor_ds(struct mixmonitor *mixmonitor, struct ast_channel *chan)
{
struct ast_datastore *datastore = NULL;
struct mixmonitor_ds *mixmonitor_ds;
if (!(mixmonitor_ds = ast_calloc(1, sizeof(*mixmonitor_ds)))) {
return -1;
}
ast_mutex_init(&mixmonitor_ds->lock);
ast_cond_init(&mixmonitor_ds->destruction_condition, NULL);
if (!(datastore = ast_datastore_alloc(&mixmonitor_ds_info, NULL))) {
ast_mutex_destroy(&mixmonitor_ds->lock);
ast_cond_destroy(&mixmonitor_ds->destruction_condition);
ast_free(mixmonitor_ds);
return -1;
}
mixmonitor_ds->samp_rate = 8000;
mixmonitor_ds->audiohook = &mixmonitor->audiohook;
datastore->data = mixmonitor_ds;
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
mixmonitor->mixmonitor_ds = mixmonitor_ds;
return 0;
}
static void launch_monitor_thread(struct ast_channel *chan, const char *filename,
unsigned int flags, int readvol, int writevol,
const char *post_process, const char *filename_write,
const char *filename_read)
{
pthread_t thread;
struct mixmonitor *mixmonitor;
char postprocess2[1024] = "";
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);
}
/* Pre-allocate mixmonitor structure and spy */
if (!(mixmonitor = ast_calloc(1, sizeof(*mixmonitor)))) {
return;
}
/* Setup the actual spy before creating our thread */
if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type, 0)) {
mixmonitor_free(mixmonitor);
return;
}
/* Copy over flags and channel name */
mixmonitor->flags = flags;
if (!(mixmonitor->autochan = ast_autochan_setup(chan))) {
mixmonitor_free(mixmonitor);
return;
}
if (setup_mixmonitor_ds(mixmonitor, chan)) {
ast_autochan_destroy(mixmonitor->autochan);
mixmonitor_free(mixmonitor);
return;
}
mixmonitor->name = ast_strdup(ast_channel_name(chan));
if (!ast_strlen_zero(postprocess2)) {
mixmonitor->post_process = ast_strdup(postprocess2);
}
if (!ast_strlen_zero(filename)) {
mixmonitor->filename = ast_strdup(filename);
}
if (!ast_strlen_zero(filename_write)) {
mixmonitor->filename_write = ast_strdup(filename_write);
}
if (!ast_strlen_zero(filename_read)) {
mixmonitor->filename_read = ast_strdup(filename_read);
}
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, ast_channel_name(chan));
ast_audiohook_destroy(&mixmonitor->audiohook);
mixmonitor_free(mixmonitor);
return;
}
ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
}
/* a note on filename_parse: creates directory structure and assigns absolute path from relative paths for filenames */
/* requires immediate copying of string from return to retain data since otherwise it will immediately lose scope */
static char *filename_parse(char *filename, char *buffer, size_t len)
{
char *slash;
if (ast_strlen_zero(filename)) {
ast_log(LOG_WARNING, "No file name was provided for a file save option.\n");
} else if (filename[0] != '/') {
char *build;
build = alloca(strlen(ast_config_AST_MONITOR_DIR) + strlen(filename) + 3);
sprintf(build, "%s/%s", ast_config_AST_MONITOR_DIR, filename);
filename = build;
}
ast_copy_string(buffer, filename, len);
if ((slash = strrchr(filename, '/'))) {
*slash = '\0';
}
ast_mkdir(filename, 0777);
return buffer;
}
static int mixmonitor_exec(struct ast_channel *chan, const char *data)
{
int x, readvol = 0, writevol = 0;
char *filename_read = NULL;
char *filename_write = NULL;
char filename_buffer[1024] = "";
struct ast_flags flags = { 0 };
char *parse;
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 or ,t(filename) and/or r(filename)\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
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], "%2d", &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], "%2d", &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], "%2d", &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 (ast_test_flag(&flags, MUXFLAG_WRITE)) {
filename_write = ast_strdupa(filename_parse(opts[OPT_ARG_WRITENAME], filename_buffer, sizeof(filename_buffer)));
}
if (ast_test_flag(&flags, MUXFLAG_READ)) {
filename_read = ast_strdupa(filename_parse(opts[OPT_ARG_READNAME], filename_buffer, sizeof(filename_buffer)));
}
}
/* If there are no file writing arguments/options for the mix monitor, send a warning message and return -1 */
if (!ast_test_flag(&flags, MUXFLAG_WRITE) && !ast_test_flag(&flags, MUXFLAG_READ) && ast_strlen_zero(args.filename)) {
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
return -1;
}
/* If filename exists, try to create directories for it */
if (!(ast_strlen_zero(args.filename))) {
args.filename = ast_strdupa(filename_parse(args.filename, filename_buffer, sizeof(filename_buffer)));
}
pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
launch_monitor_thread(chan, args.filename, flags.flags, readvol, writevol, args.post_process, filename_write, filename_read);
return 0;
}
static int stop_mixmonitor_exec(struct ast_channel *chan, const char *data)
{
struct ast_datastore *datastore = NULL;
ast_channel_lock(chan);
ast_audiohook_detach_source(chan, mixmonitor_spy_type);
if ((datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, NULL))) {
struct mixmonitor_ds *mixmonitor_ds = datastore->data;
ast_mutex_lock(&mixmonitor_ds->lock);
/* closing the filestream here guarantees the file is avaliable to the dialplan
* after calling StopMixMonitor */
mixmonitor_ds_close_fs(mixmonitor_ds);
/* The mixmonitor thread may be waiting on the audiohook trigger.
* In order to exit from the mixmonitor loop before waiting on channel
* destruction, poke the audiohook trigger. */
if (mixmonitor_ds->audiohook) {
ast_audiohook_lock(mixmonitor_ds->audiohook);
ast_cond_signal(&mixmonitor_ds->audiohook->trigger);
ast_audiohook_unlock(mixmonitor_ds->audiohook);
mixmonitor_ds->audiohook = NULL;
}
ast_mutex_unlock(&mixmonitor_ds->lock);
/* Remove the datastore so the monitor thread can exit */
if (!ast_channel_datastore_remove(chan, datastore)) {
ast_datastore_free(datastore);
}
}
ast_channel_unlock(chan);
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}";
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;
}
/*! \brief Mute / unmute a MixMonitor channel */
static int manager_mute_mixmonitor(struct mansession *s, const struct message *m)
{
struct ast_channel *c = NULL;
const char *name = astman_get_header(m, "Channel");
const char *id = astman_get_header(m, "ActionID");
const char *state = astman_get_header(m, "State");
const char *direction = astman_get_header(m,"Direction");
int clearmute = 1;
enum ast_audiohook_flags flag;
if (ast_strlen_zero(direction)) {
astman_send_error(s, m, "No direction specified. Must be read, write or both");
return AMI_SUCCESS;
}
if (!strcasecmp(direction, "read")) {
flag = AST_AUDIOHOOK_MUTE_READ;
} else if (!strcasecmp(direction, "write")) {
flag = AST_AUDIOHOOK_MUTE_WRITE;
} else if (!strcasecmp(direction, "both")) {
flag = AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE;
} else {
astman_send_error(s, m, "Invalid direction specified. Must be read, write or both");
return AMI_SUCCESS;
}
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return AMI_SUCCESS;
}
if (ast_strlen_zero(state)) {
astman_send_error(s, m, "No state specified");
return AMI_SUCCESS;
}
clearmute = ast_false(state);
c = ast_channel_get_by_name(name);
if (!c) {
astman_send_error(s, m, "No such channel");
return AMI_SUCCESS;
}
if (ast_audiohook_set_mute(c, mixmonitor_spy_type, flag, clearmute)) {
c = ast_channel_unref(c);
astman_send_error(s, m, "Cannot set mute flag");
return AMI_SUCCESS;
}
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id)) {
astman_append(s, "ActionID: %s\r\n", id);
}
astman_append(s, "\r\n");
c = ast_channel_unref(c);
return AMI_SUCCESS;
}
static int manager_mixmonitor(struct mansession *s, const struct message *m)
{
struct ast_channel *c = NULL;
const char *name = astman_get_header(m, "Channel");
const char *id = astman_get_header(m, "ActionID");
const char *file = astman_get_header(m, "File");
const char *options = astman_get_header(m, "Options");
int res;
char args[PATH_MAX] = "";
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return AMI_SUCCESS;
}
c = ast_channel_get_by_name(name);
if (!c) {
astman_send_error(s, m, "No such channel");
return AMI_SUCCESS;
}
strcpy(args, file);
strcat(args, ",");
strcat(args, options);
ast_channel_lock(c);
res = mixmonitor_exec(c, args);
ast_channel_unlock(c);
if (res) {
astman_send_error(s, m, "Could not start monitoring channel");
return AMI_SUCCESS;
}
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id)) {
astman_append(s, "ActionID: %s\r\n", id);
}
astman_append(s, "\r\n");
c = ast_channel_unref(c);
return AMI_SUCCESS;
}
static int manager_stop_mixmonitor(struct mansession *s, const struct message *m)
{
struct ast_channel *c = NULL;
const char *name = astman_get_header(m, "Channel");
const char *id = astman_get_header(m, "ActionID");
int res;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return AMI_SUCCESS;
}
c = ast_channel_get_by_name(name);
if (!c) {
astman_send_error(s, m, "No such channel");
return AMI_SUCCESS;
}
res = stop_mixmonitor_exec(c, NULL);
if (res) {
astman_send_error(s, m, "Could not stop monitoring channel");
return AMI_SUCCESS;
}
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id)) {
astman_append(s, "ActionID: %s\r\n", id);
}
astman_append(s, "\r\n");
c = ast_channel_unref(c);
return AMI_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);
res |= ast_manager_unregister("MixMonitorMute");
res |= ast_manager_unregister("MixMonitor");
res |= ast_manager_unregister("StopMixMonitor");
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);
res |= ast_manager_register_xml("MixMonitorMute", 0, manager_mute_mixmonitor);
res |= ast_manager_register_xml("MixMonitor", 0, manager_mixmonitor);
res |= ast_manager_register_xml("StopMixMonitor", 0, manager_stop_mixmonitor);
return res;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mixed Audio Monitoring Application");
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