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authorRichard Mudgett <rmudgett@digium.com>2012-03-19 20:26:51 +0000
committerRichard Mudgett <rmudgett@digium.com>2012-03-19 20:26:51 +0000
commit3714e8b1e528fd3d19ac169ef689160f4bfad2eb (patch)
treec7ab5b8bb78c8ee23872d585f665a89c05ba1e85 /res
parent0399daaa2e789a187ebb4a25720814c32fed2fc2 (diff)
Convert MuteAudio documentation to XML.
* Added missing error exits with cause in manager_mutestream(). * Cleaned up manager_mutestream() and func_mute_write(). * Some whitespace and comment cleanup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res')
-rw-r--r--res/res_mutestream.c112
1 files changed, 69 insertions, 43 deletions
diff --git a/res/res_mutestream.c b/res/res_mutestream.c
index 04106656d..ce1d167de 100644
--- a/res/res_mutestream.c
+++ b/res/res_mutestream.c
@@ -37,13 +37,6 @@
ASTERISK_FILE_VERSION(__FILE__, "$Revision: 89545 $")
-//#include <time.h>
-//#include <string.h>
-//#include <stdio.h>
-//#include <stdlib.h>
-//#include <unistd.h>
-//#include <errno.h>
-
#include "asterisk/options.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
@@ -79,14 +72,54 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision: 89545 $")
</syntax>
<description>
<para>The MUTEAUDIO function can be used to mute inbound (to the PBX) or outbound audio in a call.
- Example:
+ </para>
+ <para>Examples:
</para>
<para>
MUTEAUDIO(in)=on
+ </para>
+ <para>
MUTEAUDIO(in)=off
</para>
</description>
</function>
+ <manager name="MuteAudio" language="en_US">
+ <synopsis>
+ Mute an audio stream.
+ </synopsis>
+ <syntax>
+ <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+ <parameter name="Channel" required="true">
+ <para>The channel you want to mute.</para>
+ </parameter>
+ <parameter name="Direction" required="true">
+ <enumlist>
+ <enum name="in">
+ <para>Set muting on inbound audio stream. (to the PBX)</para>
+ </enum>
+ <enum name="out">
+ <para>Set muting on outbound audio stream. (from the PBX)</para>
+ </enum>
+ <enum name="all">
+ <para>Set muting on inbound and outbound audio streams.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ <parameter name="State" required="true">
+ <enumlist>
+ <enum name="on">
+ <para>Turn muting on.</para>
+ </enum>
+ <enum name="off">
+ <para>Turn muting off.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Mute an incoming or outgoing audio stream on a channel.</para>
+ </description>
+ </manager>
***/
@@ -98,9 +131,6 @@ struct mute_information {
};
-#define TRUE 1
-#define FALSE 0
-
/*! Datastore destroy audiohook callback */
static void destroy_callback(void *data)
{
@@ -110,8 +140,6 @@ static void destroy_callback(void *data)
ast_audiohook_destroy(&mute->audiohook);
ast_free(mute);
ast_module_unref(ast_module_info->self);
-
- return;
}
/*! \brief Static structure for datastore information */
@@ -205,6 +233,7 @@ static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data
struct ast_datastore *datastore = NULL;
struct mute_information *mute = NULL;
int is_new = 0;
+ int turnon;
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &mute_datastore, NULL))) {
@@ -214,17 +243,17 @@ static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data
}
is_new = 1;
}
-
mute = datastore->data;
+ turnon = ast_true(value);
if (!strcasecmp(data, "out")) {
- mute->mute_write = ast_true(value);
- ast_debug(1, "%s channel - outbound \n", ast_true(value) ? "Muting" : "Unmuting");
+ mute->mute_write = turnon;
+ ast_debug(1, "%s channel - outbound \n", turnon ? "Muting" : "Unmuting");
} else if (!strcasecmp(data, "in")) {
- mute->mute_read = ast_true(value);
- ast_debug(1, "%s channel - inbound \n", ast_true(value) ? "Muting" : "Unmuting");
+ mute->mute_read = turnon;
+ ast_debug(1, "%s channel - inbound \n", turnon ? "Muting" : "Unmuting");
} else if (!strcasecmp(data,"all")) {
- mute->mute_write = mute->mute_read = ast_true(value);
+ mute->mute_write = mute->mute_read = turnon;
}
if (is_new) {
@@ -241,8 +270,8 @@ static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data
/* Function for debugging - might be useful */
static struct ast_custom_function mute_function = {
- .name = "MUTEAUDIO",
- .write = func_mute_write,
+ .name = "MUTEAUDIO",
+ .write = func_mute_write,
};
static int manager_mutestream(struct mansession *s, const struct message *m)
@@ -251,12 +280,12 @@ static int manager_mutestream(struct mansession *s, const struct message *m)
const char *id = astman_get_header(m,"ActionID");
const char *state = astman_get_header(m,"State");
const char *direction = astman_get_header(m,"Direction");
- char id_text[256] = "";
+ char id_text[256];
struct ast_channel *c = NULL;
struct ast_datastore *datastore = NULL;
struct mute_information *mute = NULL;
int is_new = 0;
- int turnon = TRUE;
+ int turnon;
if (ast_strlen_zero(channel)) {
astman_send_error(s, m, "Channel not specified");
@@ -271,9 +300,6 @@ static int manager_mutestream(struct mansession *s, const struct message *m)
return 0;
}
/* Ok, we have everything */
- if (!ast_strlen_zero(id)) {
- snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
- }
c = ast_channel_get_by_name(channel);
if (!c) {
@@ -287,13 +313,14 @@ static int manager_mutestream(struct mansession *s, const struct message *m)
if (!(datastore = initialize_mutehook(c))) {
ast_channel_unlock(c);
ast_channel_unref(c);
+ astman_send_error(s, m, "Memory allocation failure");
return 0;
}
is_new = 1;
}
mute = datastore->data;
- turnon = ast_true(state);
+ turnon = ast_true(state);
if (!strcasecmp(direction, "in")) {
mute->mute_read = turnon;
} else if (!strcasecmp(direction, "out")) {
@@ -304,37 +331,36 @@ static int manager_mutestream(struct mansession *s, const struct message *m)
if (is_new) {
if (mute_add_audiohook(c, mute, datastore)) {
- /* Can't add audiohook - already printed error message */
+ /* Can't add audiohook */
ast_datastore_free(datastore);
ast_free(mute);
+ ast_channel_unlock(c);
+ ast_channel_unref(c);
+ astman_send_error(s, m, "Couldn't add mute audiohook");
+ return 0;
}
}
ast_channel_unlock(c);
ast_channel_unref(c);
+ if (!ast_strlen_zero(id)) {
+ snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
+ } else {
+ id_text[0] = '\0';
+ }
astman_append(s, "Response: Success\r\n"
- "%s"
- "\r\n\r\n", id_text);
+ "%s"
+ "\r\n", id_text);
return 0;
}
-static const char mandescr_mutestream[] =
-"Description: Mute an incoming or outbound audio stream in a channel.\n"
-"Variables: \n"
-" Channel: <name> The channel you want to mute.\n"
-" Direction: in | out |all The stream you want to mute.\n"
-" State: on | off Whether to turn mute on or off.\n"
-" ActionID: <id> Optional action ID for this AMI transaction.\n";
-
-
static int load_module(void)
{
int res;
- res = ast_custom_function_register(&mute_function);
- res |= ast_manager_register2("MuteAudio", EVENT_FLAG_SYSTEM, manager_mutestream,
- "Mute an audio stream", mandescr_mutestream);
+ res = ast_custom_function_register(&mute_function);
+ res |= ast_manager_register_xml("MuteAudio", EVENT_FLAG_SYSTEM, manager_mutestream);
return (res ? AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS);
}
@@ -343,7 +369,7 @@ static int unload_module(void)
{
ast_custom_function_unregister(&mute_function);
/* Unregister AMI actions */
- ast_manager_unregister("MuteAudio");
+ ast_manager_unregister("MuteAudio");
return 0;
}