[svn-commits] oej: branch oej/mutestream-trunk r214993 - /team/oej/mutestream-trunk/res/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Aug 31 12:16:25 CDT 2009
Author: oej
Date: Mon Aug 31 12:16:22 2009
New Revision: 214993
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=214993
Log:
Adding the new module
Added:
team/oej/mutestream-trunk/res/res_mutestream.c (with props)
Added: team/oej/mutestream-trunk/res/res_mutestream.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/mutestream-trunk/res/res_mutestream.c?view=auto&rev=214993
==============================================================================
--- team/oej/mutestream-trunk/res/res_mutestream.c (added)
+++ team/oej/mutestream-trunk/res/res_mutestream.c Mon Aug 31 12:16:22 2009
@@ -1,0 +1,321 @@
+/*
+ * 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/options.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"
+#include "asterisk/manager.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;
+
+ /* Destroy the audiohook, and destroy ourselves */
+ ast_audiohook_destroy(&mute->audiohook);
+ free(mute);
+
+ 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) {
+ memset(frame->data.ptr, 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;
+
+
+ /* 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) {
+ return 0;
+ }
+
+ ast_channel_lock(chan);
+ /* Grab datastore which contains our mute information */
+ if (!(datastore = ast_channel_datastore_find(chan, &mute_datastore, NULL))) {
+ if (option_debug > 1)
+ 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) {
+ 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);
+ }
+ }
+ ast_channel_unlock(chan);
+
+ return 0;
+}
+
+/*! \brief Initialize mute hook on channel, but don't activate it */
+static struct ast_datastore *initialize_mutehook(struct ast_channel *chan)
+{
+ struct ast_datastore *datastore = NULL;
+ struct mute_information *mute = NULL;
+
+ 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_datastore_alloc(&mute_datastore, NULL))) {
+ return NULL;
+ }
+
+ if (!(mute = ast_calloc(1, sizeof(*mute)))) {
+ ast_datastore_free(datastore);
+ return NULL;
+ }
+ ast_audiohook_init(&mute->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Mute");
+ mute->audiohook.manipulate_callback = mute_callback;
+ datastore->data = mute;
+ return datastore;
+}
+
+/*! \brief Add or activate mute audiohook on channel */
+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;
+}
+
+/*! \brief Mute dialplan function */
+static int func_mute_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
+{
+ struct ast_datastore *datastore = NULL;
+ struct mute_information *mute = NULL;
+ int is_new = 0;
+
+ 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 (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 (option_debug > 1) {
+ ast_log(LOG_DEBUG, "%s channel - inbound *** \n", ast_true(value) ? "Muting" : "Unmuting");
+ }
+ } else if (!strcasecmp(data,"all")) {
+ mute->mute_write = mute->mute_read = ast_true(value);
+ }
+
+ if (is_new) {
+ mute_add_audiohook(chan, mute, datastore);
+ }
+
+ return 0;
+}
+
+/* Function for debugging - might be useful */
+static struct ast_custom_function mute_function = {
+ .name = "MUTESTREAM",
+ .write = func_mute_write,
+ .synopsis = "Muting streams in the channel",
+ .syntax = "MUTESTREAM(in|out|all) = true|false",
+ .desc = "The mute function mutes either inbound (to the PBX) or outbound"
+ "audio. \"all\" indicates both directions",
+};
+
+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_channel_get_by_name(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))) {
+ return 0;
+ }
+ is_new = 1;
+ }
+ mute = datastore->data;
+ turnon = ast_true(state);
+
+ if (!strcasecmp(direction, "in")) {
+ mute->mute_read = turnon;
+ } else if (!strcasecmp(direction, "out")) {
+ mute->mute_write = turnon;
+ } else if (!strcasecmp(direction, "all")) {
+ mute->mute_read = mute->mute_write = turnon;
+ }
+
+ if (is_new) {
+ mute_add_audiohook(c, mute, datastore);
+ }
+
+ 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 |all 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)
+{
+ return 0;
+}
+
+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;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS, "MUTE resource",
+ .load = load_module,
+ .unload = unload_module,
+ .reload = reload,
+ );
Propchange: team/oej/mutestream-trunk/res/res_mutestream.c
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Propchange: team/oej/mutestream-trunk/res/res_mutestream.c
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svn:keywords = "Author Date Id Revision"
Propchange: team/oej/mutestream-trunk/res/res_mutestream.c
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