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asterisk/apps/app_mixmonitor.c

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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"
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
#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>
<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>
***/
#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;
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
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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);
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
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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);
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
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;
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
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);
Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173589 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05 18:34:06 +00:00
return;
}
mixmonitor->name = ast_strdup(chan->name);
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, chan->name);
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;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
static char *handle_cli_mixmonitor(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_channel *chan;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "mixmonitor {start|stop}";
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
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;
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
if (!(chan = ast_channel_get_by_name_prefix(a->argv[2], strlen(a->argv[2])))) {
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
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;
}
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
ast_channel_lock(chan);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
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);
}
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: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
chan = ast_channel_unref(chan);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
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 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");
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);
return res;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mixed Audio Monitoring Application");