[Asterisk-code-review] chan rtp.c: Copy file from chan multicast rtp.c (asterisk[13])

[Richard Mudgett]

Richard Mudgett has uploaded a new change for review.

  https://gerrit.asterisk.org/3012

Change subject: chan_rtp.c: Copy file from chan_multicast_rtp.c
......................................................................

chan_rtp.c: Copy file from chan_multicast_rtp.c

Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef
---
A channels/chan_rtp.c
1 file changed, 223 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/12/3012/1

diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c
new file mode 100644
index 0000000..267baab
--- /dev/null
+++ b/channels/chan_rtp.c
@@ -0,0 +1,223 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at digium.com>
+ * Andreas 'MacBrody' Brodmann <andreas.brodmann at 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 at digium.com>
+ * \author Andreas 'MacBrody' Broadmann <andreas.brodmann at 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_assigned_ids *assignedids, 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_assigned_ids *assignedids, 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_cap *caps = NULL;
+	struct ast_format *fmt = NULL;
+
+	fmt = ast_format_cap_get_format(cap, 0);
+
+	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;
+	}
+
+	caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+	if (!caps) {
+		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, "", "", "", "", "", assignedids, 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_append(caps, fmt, 0);
+	ast_channel_nativeformats_set(chan, caps);
+	ast_channel_set_writeformat(chan, fmt);
+	ast_channel_set_rawwriteformat(chan, fmt);
+	ast_channel_set_readformat(chan, fmt);
+	ast_channel_set_rawreadformat(chan, fmt);
+
+	ast_channel_tech_pvt_set(chan, instance);
+
+	ast_channel_unlock(chan);
+
+	ao2_ref(fmt, -1);
+	ao2_ref(caps, -1);
+
+	return chan;
+
+failure:
+	ao2_cleanup(fmt);
+	ao2_cleanup(caps);
+	*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(AST_FORMAT_CAP_FLAG_DEFAULT))) {
+		return AST_MODULE_LOAD_DECLINE;
+	}
+	ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
+	if (ast_channel_register(&multicast_rtp_tech)) {
+		ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
+		ao2_ref(multicast_rtp_tech.capabilities, -1);
+		multicast_rtp_tech.capabilities = NULL;
+		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);
+	ao2_ref(multicast_rtp_tech.capabilities, -1);
+	multicast_rtp_tech.capabilities = NULL;
+
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
+	.support_level = AST_MODULE_SUPPORT_CORE,
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_CHANNEL_DRIVER,
+);

-- 
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>