[svn-commits] oej: branch oej/pinepeach r214989 - /team/oej/pinepeach/res/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Aug 31 11:40:31 CDT 2009
Author: oej
Date: Mon Aug 31 11:40:28 2009
New Revision: 214989
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=214989
Log:
Thanks to Josh Colp, I'm now actually muting audio frames... :-)
Adding manager command, cleaning up a tiny bit. Renaming module
Added:
team/oej/pinepeach/res/res_mutestream.c (contents, props changed)
- copied, changed from r214938, team/oej/pinepeach/res/res_hookmute.c
Removed:
team/oej/pinepeach/res/res_hookmute.c
Copied: team/oej/pinepeach/res/res_mutestream.c (from r214938, team/oej/pinepeach/res/res_hookmute.c)
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/pinepeach/res/res_mutestream.c?view=diff&rev=214989&p1=team/oej/pinepeach/res/res_hookmute.c&r1=214938&p2=team/oej/pinepeach/res/res_mutestream.c&r2=214989
==============================================================================
--- team/oej/pinepeach/res/res_hookmute.c (original)
+++ team/oej/pinepeach/res/res_mutestream.c Mon Aug 31 11:40:28 2009
@@ -36,7 +36,7 @@
#include <math.h>
#include <errno.h>
-#include "asterisk/callerid.h"
+#include "asterisk/options.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
@@ -46,6 +46,7 @@
#include "asterisk/frame.h"
#include "asterisk/utils.h"
#include "asterisk/audiohook.h"
+#include "asterisk/manager.h"
@@ -88,30 +89,28 @@
for (next = AST_LIST_NEXT(frame, frame_list);
frame;
frame = next, next = frame ? AST_LIST_NEXT(frame, frame_list) : NULL) {
-
- ast_log(LOG_DEBUG, " ---- CLEANING FRAME ---- Datalen %d\n", frame->datalen);
- memset(frame->data, frame->datalen, 0);
+ memset(frame->data, 0, frame->datalen);
}
}
+/*! \brief The callback from the audiohook subsystem. We basically get a frame to have fun with */
static int mute_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
{
struct ast_datastore *datastore = NULL;
struct mute_information *mute = NULL;
- ast_log(LOG_DEBUG, "''' Mute callback on %s \n", chan ? chan->name : "No channel");
/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
- ast_log(LOG_DEBUG, " *** We're done here. Good bye.\n");
return 0;
}
ast_channel_lock(chan);
/* Grab datastore which contains our mute information */
if (!(datastore = ast_channel_datastore_find(chan, &mute_datastore, NULL))) {
- ast_log(LOG_DEBUG, " *** Can't find any datastore to use. Bad. \n");
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, " *** Can't find any datastore to use. Bad. \n");
return 0;
}
@@ -120,27 +119,15 @@
/* If this is audio then allow them to increase/decrease the gains */
if (frame->frametype == AST_FRAME_VOICE) {
- ast_log(LOG_DEBUG, "''' Audio frame - direction %s mute READ %s WRITE %s\n", direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write", mute->mute_read ? "on" : "off", mute->mute_write ? "on" : "off");
+ if (option_debug > 3)
+ ast_log(LOG_DEBUG, "Audio frame - direction %s mute READ %s WRITE %s\n", direction == AST_AUDIOHOOK_DIRECTION_READ ? "read" : "write", mute->mute_read ? "on" : "off", mute->mute_write ? "on" : "off");
/* Based on direction of frame grab the gain, and confirm it is applicable */
if ((direction == AST_AUDIOHOOK_DIRECTION_READ && mute->mute_read) || (direction == AST_AUDIOHOOK_DIRECTION_WRITE && mute->mute_write)) {
/* Ok, we just want to reset all audio in this frame. Keep NOTHING, thanks. */
ast_frame_clear(frame);
}
- /* DTMF Just for debugging - kind of stupid */
- } else if (frame->frametype == AST_FRAME_DTMF) {
- ast_log(LOG_DEBUG, "*** Frame is a DTMF frame\n");
- if (frame->subclass == '1') {
- mute->mute_read = TRUE;
- mute->mute_write = TRUE;
- } else if (frame->subclass == '0') {
- mute->mute_read = FALSE;
- mute->mute_write = FALSE;
- ast_log(LOG_DEBUG, "*** Turning off mute \n");
- }
- } else {
- ast_log(LOG_DEBUG, "*** Frame is not a voice or DTMF frame. What is it? -- %d\n", frame->frametype);
- }
+ }
ast_channel_unlock(chan);
return 0;
@@ -151,7 +138,9 @@
struct ast_datastore *datastore = NULL;
struct mute_information *mute = NULL;
- ast_log(LOG_DEBUG, "**** Initializing new Mute Audiohook \n");
+ if (option_debug > 2 )
+ ast_log(LOG_DEBUG, "Initializing new Mute Audiohook \n");
+
/* Allocate a new datastore to hold the reference to this mute_datastore and audiohook information */
if (!(datastore = ast_channel_datastore_alloc(&mute_datastore, NULL))) {
return NULL;
@@ -163,10 +152,22 @@
}
ast_audiohook_init(&mute->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Mute");
mute->audiohook.manipulate_callback = mute_callback;
- /* For debugging control, listen to DTMF */
- ast_set_flag(&mute->audiohook, AST_AUDIOHOOK_WANTS_DTMF);
datastore->data = mute;
return datastore;
+}
+
+static int mute_add_audiohook(struct ast_channel *chan, struct mute_information *mute, struct ast_datastore *datastore)
+{
+ /* Activate the settings */
+ ast_channel_datastore_add(chan, datastore);
+ if(ast_audiohook_attach(chan, &mute->audiohook)) {
+ ast_log(LOG_ERROR, "Failed to attach audiohook for muting channel %s\n", chan->name);
+ return -1;
+ }
+ if (option_debug) {
+ ast_log(LOG_DEBUG, "*** Initialized audiohook on channel %s\n", chan->name);
+ }
+ return 0;
}
static int func_mute_write(struct ast_channel *chan, char *cmd, char *data, const char *value)
@@ -175,8 +176,6 @@
struct mute_information *mute = NULL;
int is_new = 0;
- ast_log(LOG_DEBUG, "**** Mute write - data %s value %s \n", data, value);
-
if (!(datastore = ast_channel_datastore_find(chan, &mute_datastore, NULL))) {
if (!(datastore = initialize_mutehook(chan))) {
return 0;
@@ -188,29 +187,20 @@
if (!strcasecmp(data, "out")) {
mute->mute_write = ast_true(value);
- if (ast_true(value))
- ast_log(LOG_DEBUG, "*** Muting channel - outbound *** \n");
- else
- ast_log(LOG_DEBUG, "*** UN-Muting channel - outbound *** \n");
- }
-
- else if (!strcasecmp(data, "in")){
+ if (option_debug > 1) {
+ ast_log(LOG_DEBUG, "%s channel - outbound *** \n", ast_true(value) ? "Muting" : "Unmuting");
+ }
+ }
+
+ else if (!strcasecmp(data, "in")) {
mute->mute_read = ast_true(value);
- if (ast_true(value))
- ast_log(LOG_DEBUG, "*** Muting channel - inbound *** \n");
- else
- ast_log(LOG_DEBUG, "*** UN-Muting channel - inbound *** \n");
- }
- /* DEBUG */
- mute->mute_read = TRUE;
- mute->mute_write = TRUE;
+ if (option_debug > 1) {
+ ast_log(LOG_DEBUG, "%s channel - inbound *** \n", ast_true(value) ? "Muting" : "Unmuting");
+ }
+ }
if (is_new) {
- /* Activate the settings */
- ast_channel_datastore_add(chan, datastore);
- if(ast_audiohook_attach(chan, &mute->audiohook))
- ast_log(LOG_DEBUG, "*** Failed to attach audiohook for muting!\n");
- ast_log(LOG_DEBUG, "*** Initialized audiohook on channel %s\n", chan->name);
+ mute_add_audiohook(chan, mute, datastore);
}
return 0;
@@ -222,9 +212,83 @@
.write = func_mute_write,
.synopsis = "Muting the channel, totally and utterly",
.syntax = "MUTE(in|out) = true|false",
- .desc = "Use this function instead of shouting SHUT UP.",
+ .desc = "The mute function mutes either inbound (to the PBX) or outbound"
+ "audio.",
};
+static int manager_mutestream(struct mansession *s, const struct message *m)
+{
+ const char *channel = 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");
+ char idText[256] = "";
+ struct ast_channel *c = NULL;
+ struct ast_datastore *datastore = NULL;
+ struct mute_information *mute = NULL;
+ int is_new = 0;
+ int turnon = TRUE;
+
+ if (ast_strlen_zero(channel)) {
+ astman_send_error(s, m, "Channel not specified");
+ return 0;
+ }
+ if (ast_strlen_zero(state)) {
+ astman_send_error(s, m, "State not specified");
+ return 0;
+ }
+ if (ast_strlen_zero(direction)) {
+ astman_send_error(s, m, "Direction not specified");
+ return 0;
+ }
+ /* Ok, we have everything */
+ if (!ast_strlen_zero(id)) {
+ snprintf(idText, sizeof(idText), "ActionID: %s\r\n", id);
+ }
+
+ c = ast_get_channel_by_name_locked(channel);
+ if (!c) {
+ astman_send_error(s, m, "No such channel");
+ return 0;
+ }
+
+ if (!(datastore = ast_channel_datastore_find(c, &mute_datastore, NULL))) {
+ if (!(datastore = initialize_mutehook(c))) {
+ ast_channel_unlock(c);
+ return 0;
+ }
+ is_new = 1;
+ }
+ mute = datastore->data;
+ turnon = ast_true(state);
+
+ if (!strcasecmp(direction, "in")) {
+ mute->mute_read = turnon;
+ } else {
+ mute->mute_write = turnon;
+ }
+
+ if (is_new) {
+ mute_add_audiohook(c, mute, datastore);
+ }
+ ast_channel_unlock(c);
+
+ astman_append(s, "Response: Success\r\n"
+ "%s"
+ "\r\n\r\n", idText);
+ return 0;
+}
+
+
+static 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 The stream you wan 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 reload(void)
@@ -235,12 +299,18 @@
static int load_module(void)
{
ast_custom_function_register(&mute_function);
+
+ ast_manager_register2("MuteStream", EVENT_FLAG_SYSTEM, manager_mutestream,
+ "Mute an audio stream", mandescr_mutestream);
return 0;
}
static int unload_module(void)
{
ast_custom_function_unregister(&mute_function);
+ /* Unregister AMI actions */
+ ast_manager_unregister("MuteStream");
+
/* Can't unload this once we're loaded */
return -1;
}
Propchange: team/oej/pinepeach/res/res_mutestream.c
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Propchange: team/oej/pinepeach/res/res_mutestream.c
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Propchange: team/oej/pinepeach/res/res_mutestream.c
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svn:mergeinfo =
Propchange: team/oej/pinepeach/res/res_mutestream.c
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