[asterisk-commits] oej: trunk r215382 - in /trunk: CHANGES res/res_mutestream.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 2 01:23:10 CDT 2009
Author: oej
Date: Wed Sep 2 01:23:05 2009
New Revision: 215382
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=215382
Log:
Adding MUTEAUDIO() dialplan function and MuteAudio AMI action (pinepeach)
Reviewboard #345
Added:
trunk/res/res_mutestream.c (with props)
Modified:
trunk/CHANGES
Modified: trunk/CHANGES
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/CHANGES?view=diff&rev=215382&r1=215381&r2=215382
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Wed Sep 2 01:23:05 2009
@@ -103,6 +103,8 @@
construct (which all could set variables on the master channel). Usage
of the HASH() dialplan function, with the key set to the name of the slave
channel, is one approach that will avoid conflicts.
+ * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
+ audio in a channel.
Dialplan Variables
------------------
@@ -201,6 +203,8 @@
across all .conf files. All affected sample.conf files have been modified to
reflect this change. Previous options such as 'sslenable' still work,
but options with the 'tls' prefix are preferred.
+ * Added a MuteAudio AMI action for muting inbound and/or outbound audio
+ in a channel. (res_mutestream.so)
Channel Event Logging
---------------------
Added: trunk/res/res_mutestream.c
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/res/res_mutestream.c?view=auto&rev=215382
==============================================================================
--- trunk/res/res_mutestream.c (added)
+++ trunk/res/res_mutestream.c Wed Sep 2 01:23:05 2009
@@ -1,0 +1,360 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009, Olle E. Johansson
+ *
+ * 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 MUTESTREAM audiohooks
+ *
+ * \author Olle E. Johansson <oej at edvina.net>
+ *
+ * \ingroup functions
+ *
+ * \note This module only handles audio streams today, but can easily be appended to also
+ * zero out text streams if there's an application for it.
+ * When we know and understands what happens if we zero out video, we can do that too.
+ */
+
+#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 <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"
+
+/*** DOCUMENTATION
+ <function name="MUTEAUDIO" language="en_US">
+ <synopsis>
+ Muting audio streams in the channel
+ </synopsis>
+ <syntax>
+ <parameter name="direction" required="true">
+ <para>Must be one of </para>
+ <enumlist>
+ <enum name="in">
+ <para>Inbound stream (to the PBX)</para>
+ </enum>
+ <enum name="out">
+ <para>Outbound stream (from the PBX)</para>
+ </enum>
+ <enum name="all">
+ <para>Both streams</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>The MUTEAUDIO function can be used to mute inbound (to the PBX) or outbound audio in a call.
+ Example:
+ </para>
+ <para>
+ MUTEAUDIO(in)=on
+ MUTEAUDIO(in)=off
+ </para>
+ </description>
+ </function>
+ ***/
+
+
+/*! 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);
+ ast_free(mute);
+ ast_module_unref(ast_module_info->self);
+
+ 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))) {
+ ast_channel_unlock(chan);
+ ast_debug(2, "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_debug(2, "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
+ \pre Assumes that the channel is locked
+*/
+static struct ast_datastore *initialize_mutehook(struct ast_channel *chan)
+{
+ struct ast_datastore *datastore = NULL;
+ struct mute_information *mute = NULL;
+
+ ast_debug(2, "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
+ Assumes channel is locked
+*/
+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;
+ }
+ ast_module_ref(ast_module_info->self);
+ ast_debug(2, "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;
+
+ ast_channel_lock(chan);
+ if (!(datastore = ast_channel_datastore_find(chan, &mute_datastore, NULL))) {
+ if (!(datastore = initialize_mutehook(chan))) {
+ ast_channel_unlock(chan);
+ return 0;
+ }
+ is_new = 1;
+ }
+
+ mute = datastore->data;
+
+ if (!strcasecmp(data, "out")) {
+ mute->mute_write = ast_true(value);
+ ast_debug(1, "%s channel - outbound \n", ast_true(value) ? "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");
+ } else if (!strcasecmp(data,"all")) {
+ mute->mute_write = mute->mute_read = ast_true(value);
+ }
+
+ if (is_new) {
+ if (mute_add_audiohook(chan, mute, datastore)) {
+ /* Can't add audiohook - already printed error message */
+ ast_datastore_free(datastore);
+ ast_free(mute);
+ }
+ }
+ ast_channel_unlock(chan);
+
+ return 0;
+}
+
+/* Function for debugging - might be useful */
+static struct ast_custom_function mute_function = {
+ .name = "MUTEAUDIO",
+ .write = func_mute_write,
+};
+
+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 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;
+
+ 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(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
+ }
+
+ c = ast_channel_get_by_name(channel);
+ if (!c) {
+ astman_send_error(s, m, "No such channel");
+ return 0;
+ }
+
+ ast_channel_lock(c);
+
+ if (!(datastore = ast_channel_datastore_find(c, &mute_datastore, NULL))) {
+ if (!(datastore = initialize_mutehook(c))) {
+ ast_channel_unlock(c);
+ ast_channel_unref(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) {
+ if (mute_add_audiohook(c, mute, datastore)) {
+ /* Can't add audiohook - already printed error message */
+ ast_datastore_free(datastore);
+ ast_free(mute);
+ }
+ }
+ ast_channel_unlock(c);
+ ast_channel_unref(c);
+
+ astman_append(s, "Response: Success\r\n"
+ "%s"
+ "\r\n\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);
+
+ return (res ? AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS);
+}
+
+static int unload_module(void)
+{
+ ast_custom_function_unregister(&mute_function);
+ /* Unregister AMI actions */
+ ast_manager_unregister("MuteAudio");
+
+ return 0;
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Mute audio stream resources");
Propchange: trunk/res/res_mutestream.c
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Propchange: trunk/res/res_mutestream.c
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