Files
asterisk/channels/chan_multicast_rtp.c
Mark Michelson ee21eee7e0 Cache string values of formats on ast_format_cap() to save processing.
Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.

Review: https://reviewboard.asterisk.org/r/2879
........

Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03 14:58:16 +00:00

205 lines
6.0 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2009, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
* Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
*
* 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
*
* \author Joshua Colp <jcolp@digium.com>
* \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
*
* \brief Multicast RTP Paging Channel
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <fcntl.h>
#include <sys/signal.h>
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/causes.h"
static const char tdesc[] = "Multicast RTP Paging Channel Driver";
/* Forward declarations */
static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout);
static int multicast_rtp_hangup(struct ast_channel *ast);
static struct ast_frame *multicast_rtp_read(struct ast_channel *ast);
static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f);
/* Channel driver declaration */
static struct ast_channel_tech multicast_rtp_tech = {
.type = "MulticastRTP",
.description = tdesc,
.requester = multicast_rtp_request,
.call = multicast_rtp_call,
.hangup = multicast_rtp_hangup,
.read = multicast_rtp_read,
.write = multicast_rtp_write,
};
/*! \brief Function called when we should read a frame from the channel */
static struct ast_frame *multicast_rtp_read(struct ast_channel *ast)
{
return &ast_null_frame;
}
/*! \brief Function called when we should write a frame to the channel */
static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f)
{
struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
return ast_rtp_instance_write(instance, f);
}
/*! \brief Function called when we should actually call the destination */
static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout)
{
struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
ast_queue_control(ast, AST_CONTROL_ANSWER);
return ast_rtp_instance_activate(instance);
}
/*! \brief Function called when we should hang the channel up */
static int multicast_rtp_hangup(struct ast_channel *ast)
{
struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
ast_rtp_instance_destroy(instance);
ast_channel_tech_pvt_set(ast, NULL);
return 0;
}
/*! \brief Function called when we should prepare to call the destination */
static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
{
char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
struct ast_rtp_instance *instance;
struct ast_sockaddr control_address;
struct ast_sockaddr destination_address;
struct ast_channel *chan;
struct ast_format fmt;
ast_best_codec(cap, &fmt);
ast_sockaddr_setnull(&control_address);
/* If no type was given we can't do anything */
if (ast_strlen_zero(multicast_type)) {
goto failure;
}
if (!(destination = strchr(tmp, '/'))) {
goto failure;
}
*destination++ = '\0';
if ((control = strchr(destination, '/'))) {
*control++ = '\0';
if (!ast_sockaddr_parse(&control_address, control,
PARSE_PORT_REQUIRE)) {
goto failure;
}
}
if (!ast_sockaddr_parse(&destination_address, destination,
PARSE_PORT_REQUIRE)) {
goto failure;
}
if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
goto failure;
}
if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", requestor ? ast_channel_linkedid(requestor) : "", 0, "MulticastRTP/%p", instance))) {
ast_rtp_instance_destroy(instance);
goto failure;
}
ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
ast_rtp_instance_set_remote_address(instance, &destination_address);
ast_channel_tech_set(chan, &multicast_rtp_tech);
ast_format_cap_add(ast_channel_nativeformats(chan), &fmt);
ast_format_copy(ast_channel_writeformat(chan), &fmt);
ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
ast_format_copy(ast_channel_readformat(chan), &fmt);
ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
ast_channel_tech_pvt_set(chan, instance);
return chan;
failure:
*cause = AST_CAUSE_FAILURE;
return NULL;
}
/*! \brief Function called when our module is loaded */
static int load_module(void)
{
if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(0))) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_add_all(multicast_rtp_tech.capabilities);
if (ast_channel_register(&multicast_rtp_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
/*! \brief Function called when our module is unloaded */
static int unload_module(void)
{
ast_channel_unregister(&multicast_rtp_tech);
multicast_rtp_tech.capabilities = ast_format_cap_destroy(multicast_rtp_tech.capabilities);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);