[svn-commits] oej: branch oej/pinepeach r214938 - in /team/oej/pinepeach: apps/ res/ utils/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Aug 31 09:19:12 CDT 2009
Author: oej
Date: Mon Aug 31 09:19:08 2009
New Revision: 214938
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=214938
Log:
Adding work on mute audiohook for meetme and others
Kind of stupid dialplan function right now that mutes everything
(debugging mode - but who wants to hear you? :-) )
Added:
team/oej/pinepeach/res/res_hookmute.c (with props)
Modified:
team/oej/pinepeach/apps/app_chanspy.c
team/oej/pinepeach/apps/app_meetme.c
team/oej/pinepeach/utils/Makefile
Modified: team/oej/pinepeach/apps/app_chanspy.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/pinepeach/apps/app_chanspy.c?view=diff&rev=214938&r1=214937&r2=214938
==============================================================================
--- team/oej/pinepeach/apps/app_chanspy.c (original)
+++ team/oej/pinepeach/apps/app_chanspy.c Mon Aug 31 09:19:08 2009
@@ -44,6 +44,7 @@
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/audiohook.h"
+#include "asterisk/manager.h"
#include "asterisk/features.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
@@ -216,6 +217,12 @@
struct ast_channel *peer;
ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan_name, chan->name);
+ manager_event(EVENT_FLAG_CALL, "SpyHook",
+ "Channel: %s\r\n"
+ "SpyChannel: %s\r\n"
+ "Action: %s\r\n",
+ chan->name, spychan_name, "start");
+
ast_set_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC | AST_AUDIOHOOK_SMALL_QUEUE);
@@ -509,6 +516,14 @@
return setup_chanspy_ds(this, chanspy_ds);
}
+static void manager_stop(const char *chan_name)
+{
+ manager_event(EVENT_FLAG_CALL, "SpyHook",
+ "Channel: %s\r\n"
+ "Action: stop\r\n",
+ chan_name);
+}
+
static int common_exec(struct ast_channel *chan, const struct ast_flags *flags,
int volfactor, const int fd, const char *mygroup, const char *spec,
const char *exten, const char *context)
@@ -551,6 +566,7 @@
res = ast_waitfordigit(chan, waitms);
if (res < 0) {
ast_clear_flag(chan, AST_FLAG_SPYING);
+ manager_stop(chan->name);
break;
}
@@ -681,6 +697,7 @@
if (res == -1 || ast_check_hangup(chan))
break;
}
+ manager_stop(chan->name);
ast_clear_flag(chan, AST_FLAG_SPYING);
Modified: team/oej/pinepeach/apps/app_meetme.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/pinepeach/apps/app_meetme.c?view=diff&rev=214938&r1=214937&r2=214938
==============================================================================
--- team/oej/pinepeach/apps/app_meetme.c (original)
+++ team/oej/pinepeach/apps/app_meetme.c Mon Aug 31 09:19:08 2009
@@ -167,6 +167,8 @@
CONFFLAG_SLA_TRUNK = (1 << 27),
/*! Do not write any audio to this channel until the state is up. */
CONFFLAG_NO_AUDIO_UNTIL_UP = (1 << 28),
+ /*! Do not *send* audio to user, a.k.a. reverse mute */
+ CONFFLAG_REVERSE_MUTE = (1 << 29),
};
enum {
@@ -281,6 +283,7 @@
" 'M' -- Mute one user\n"
" 'n' -- Unmute all users in the conference\n"
" 'N' -- Mute all non-admin users in the conference\n"
+" 'Q' -- Reverse mute one user\n"
" 'r' -- Reset one user's volume settings\n"
" 'R' -- Reset all users volume settings\n"
" 's' -- Lower entire conference speaking volume\n"
Added: team/oej/pinepeach/res/res_hookmute.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/pinepeach/res/res_hookmute.c?view=auto&rev=214938
==============================================================================
--- team/oej/pinepeach/res/res_hookmute.c (added)
+++ team/oej/pinepeach/res/res_hookmute.c Mon Aug 31 09:19:08 2009
@@ -1,0 +1,252 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Olle E. Johansson <oej at edvina.net>
+ *
+ * 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 MUTE audiohooks
+ *
+ * \author Olle E. Johansson <oej at edvina.net>
+ *
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 89545 $")
+
+#include <time.h>
+#include <string.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <math.h>
+#include <errno.h>
+
+#include "asterisk/callerid.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/module.h"
+#include "asterisk/config.h"
+#include "asterisk/file.h"
+#include "asterisk/pbx.h"
+#include "asterisk/frame.h"
+#include "asterisk/utils.h"
+#include "asterisk/audiohook.h"
+
+
+
+/* Our own datastore */
+struct mute_information {
+ struct ast_audiohook audiohook;
+ int mute_write;
+ int mute_read;
+};
+
+
+#define TRUE 1
+#define FALSE 0
+
+/*! Datastore destroy audiohook callback */
+static void destroy_callback(void *data)
+{
+ struct mute_information *mute = data;
+ ast_log(LOG_DEBUG, "***** About to destroy mute audiohook for this channel\n");
+
+ /* Destroy the audiohook, and destroy ourselves */
+ ast_audiohook_destroy(&mute->audiohook);
+ free(mute);
+ ast_log(LOG_DEBUG, "***** Destroying mute audiohook for this channel\n");
+
+ return;
+}
+
+/*! \brief Static structure for datastore information */
+static const struct ast_datastore_info mute_datastore = {
+ .type = "mute",
+ .destroy = destroy_callback
+};
+
+/*! \brief Wipe out all audio samples from an ast_frame. Clean it. */
+static void ast_frame_clear(struct ast_frame *frame)
+{
+ struct ast_frame *next;
+
+ 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);
+ }
+}
+
+
+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");
+ return 0;
+ }
+
+ mute = datastore->data;
+
+
+ /* 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");
+
+ /* 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;
+}
+
+static struct ast_datastore *initialize_mutehook(struct ast_channel *chan)
+{
+ struct ast_datastore *datastore = NULL;
+ struct mute_information *mute = NULL;
+
+ 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;
+ }
+
+ if (!(mute = ast_calloc(1, sizeof(*mute)))) {
+ ast_channel_datastore_free(datastore);
+ return NULL;
+ }
+ 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 func_mute_write(struct ast_channel *chan, char *cmd, char *data, const char *value)
+{
+ struct ast_datastore *datastore = NULL;
+ 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;
+ }
+ is_new = 1;
+ }
+
+ mute = datastore->data;
+
+ 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")){
+ 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 (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);
+ }
+
+ return 0;
+}
+
+/* Function for debugging - might be useful */
+static struct ast_custom_function mute_function = {
+ .name = "MUTE",
+ .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.",
+};
+
+
+
+static int reload(void)
+{
+ return 0;
+}
+
+static int load_module(void)
+{
+ ast_custom_function_register(&mute_function);
+ return 0;
+}
+
+static int unload_module(void)
+{
+ ast_custom_function_unregister(&mute_function);
+ /* Can't unload this once we're loaded */
+ return -1;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS, "MUTE resource",
+ .load = load_module,
+ .unload = unload_module,
+ .reload = reload,
+ );
Propchange: team/oej/pinepeach/res/res_hookmute.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/oej/pinepeach/res/res_hookmute.c
------------------------------------------------------------------------------
svn:keywords = "Author Date Id Revision"
Propchange: team/oej/pinepeach/res/res_hookmute.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/oej/pinepeach/utils/Makefile
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/pinepeach/utils/Makefile?view=diff&rev=214938&r1=214937&r2=214938
==============================================================================
--- team/oej/pinepeach/utils/Makefile (original)
+++ team/oej/pinepeach/utils/Makefile Mon Aug 31 09:19:08 2009
@@ -26,7 +26,7 @@
# changes are made to ast_expr2.y or ast_expr2.fl (or the corresponding .c files),
# as a regression test. Others (mere mortals?) need not bother, but they are
# more than welcome to play! The regression test itself is in expr2.testinput.
-ALL_UTILS:=astman smsq stereorize streamplayer aelparse muted
+ALL_UTILS:=astman smsq stereorize streamplayer aelparse
UTILS:=$(ALL_UTILS)
include $(ASTTOPDIR)/Makefile.rules
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