Let's also include aclocal.m4
[asterisk-bristuff.git] / channels / chan_nbs.c
blobb231127ce54dee1c442ead35e92f53d0eeeb117d
1 /*
2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2006, Digium, Inc.
6 * Mark Spencer <markster@digium.com>
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
19 /*! \file
21 * \brief Network broadcast sound support channel driver
23 * \author Mark Spencer <markster@digium.com>
25 * \ingroup channel_drivers
28 /*** MODULEINFO
29 <depend>nbs</depend>
30 ***/
32 #include "asterisk.h"
34 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
36 #include <stdio.h>
37 #include <string.h>
38 #include <sys/socket.h>
39 #include <sys/time.h>
40 #include <errno.h>
41 #include <unistd.h>
42 #include <stdlib.h>
43 #include <arpa/inet.h>
44 #include <fcntl.h>
45 #include <sys/ioctl.h>
46 #include <nbs.h>
48 #include "asterisk/lock.h"
49 #include "asterisk/channel.h"
50 #include "asterisk/config.h"
51 #include "asterisk/logger.h"
52 #include "asterisk/module.h"
53 #include "asterisk/pbx.h"
54 #include "asterisk/options.h"
55 #include "asterisk/utils.h"
57 static const char tdesc[] = "Network Broadcast Sound Driver";
59 /* Only linear is allowed */
60 static int prefformat = AST_FORMAT_SLINEAR;
62 static char context[AST_MAX_EXTENSION] = "default";
63 static char type[] = "NBS";
65 /* NBS creates private structures on demand */
67 struct nbs_pvt {
68 NBS *nbs;
69 struct ast_channel *owner; /* Channel we belong to, possibly NULL */
70 char app[16]; /* Our app */
71 char stream[80]; /* Our stream */
72 struct ast_frame fr; /* "null" frame */
73 struct ast_module_user *u; /*! for holding a reference to this module */
76 static struct ast_channel *nbs_request(const char *type, int format, void *data, int *cause);
77 static int nbs_call(struct ast_channel *ast, char *dest, int timeout);
78 static int nbs_hangup(struct ast_channel *ast);
79 static struct ast_frame *nbs_xread(struct ast_channel *ast);
80 static int nbs_xwrite(struct ast_channel *ast, struct ast_frame *frame);
82 static const struct ast_channel_tech nbs_tech = {
83 .type = type,
84 .description = tdesc,
85 .capabilities = AST_FORMAT_SLINEAR,
86 .requester = nbs_request,
87 .call = nbs_call,
88 .hangup = nbs_hangup,
89 .read = nbs_xread,
90 .write = nbs_xwrite,
93 static int nbs_call(struct ast_channel *ast, char *dest, int timeout)
95 struct nbs_pvt *p;
97 p = ast->tech_pvt;
99 if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
100 ast_log(LOG_WARNING, "nbs_call called on %s, neither down nor reserved\n", ast->name);
101 return -1;
103 /* When we call, it just works, really, there's no destination... Just
104 ring the phone and wait for someone to answer */
105 if (option_debug)
106 ast_log(LOG_DEBUG, "Calling %s on %s\n", dest, ast->name);
108 /* If we can't connect, return congestion */
109 if (nbs_connect(p->nbs)) {
110 ast_log(LOG_WARNING, "NBS Connection failed on %s\n", ast->name);
111 ast_queue_control(ast, AST_CONTROL_CONGESTION);
112 } else {
113 ast_setstate(ast, AST_STATE_RINGING);
114 ast_queue_control(ast, AST_CONTROL_ANSWER);
117 return 0;
120 static void nbs_destroy(struct nbs_pvt *p)
122 if (p->nbs)
123 nbs_delstream(p->nbs);
124 ast_module_user_remove(p->u);
125 free(p);
128 static struct nbs_pvt *nbs_alloc(void *data)
130 struct nbs_pvt *p;
131 int flags = 0;
132 char stream[256];
133 char *opts;
135 ast_copy_string(stream, data, sizeof(stream));
136 if ((opts = strchr(stream, ':'))) {
137 *opts = '\0';
138 opts++;
139 } else
140 opts = "";
141 p = malloc(sizeof(struct nbs_pvt));
142 if (p) {
143 memset(p, 0, sizeof(struct nbs_pvt));
144 if (!ast_strlen_zero(opts)) {
145 if (strchr(opts, 'm'))
146 flags |= NBS_FLAG_MUTE;
147 if (strchr(opts, 'o'))
148 flags |= NBS_FLAG_OVERSPEAK;
149 if (strchr(opts, 'e'))
150 flags |= NBS_FLAG_EMERGENCY;
151 if (strchr(opts, 'O'))
152 flags |= NBS_FLAG_OVERRIDE;
153 } else
154 flags = NBS_FLAG_OVERSPEAK;
156 ast_copy_string(p->stream, stream, sizeof(p->stream));
157 p->nbs = nbs_newstream("asterisk", stream, flags);
158 if (!p->nbs) {
159 ast_log(LOG_WARNING, "Unable to allocate new NBS stream '%s' with flags %d\n", stream, flags);
160 free(p);
161 p = NULL;
162 } else {
163 /* Set for 8000 hz mono, 640 samples */
164 nbs_setbitrate(p->nbs, 8000);
165 nbs_setchannels(p->nbs, 1);
166 nbs_setblocksize(p->nbs, 640);
167 nbs_setblocking(p->nbs, 0);
170 return p;
173 static int nbs_hangup(struct ast_channel *ast)
175 struct nbs_pvt *p;
176 p = ast->tech_pvt;
177 if (option_debug)
178 ast_log(LOG_DEBUG, "nbs_hangup(%s)\n", ast->name);
179 if (!ast->tech_pvt) {
180 ast_log(LOG_WARNING, "Asked to hangup channel not connected\n");
181 return 0;
183 nbs_destroy(p);
184 ast->tech_pvt = NULL;
185 ast_setstate(ast, AST_STATE_DOWN);
186 return 0;
189 static struct ast_frame *nbs_xread(struct ast_channel *ast)
191 struct nbs_pvt *p = ast->tech_pvt;
194 /* Some nice norms */
195 p->fr.datalen = 0;
196 p->fr.samples = 0;
197 p->fr.data = NULL;
198 p->fr.src = type;
199 p->fr.offset = 0;
200 p->fr.mallocd=0;
201 p->fr.delivery.tv_sec = 0;
202 p->fr.delivery.tv_usec = 0;
204 ast_log(LOG_DEBUG, "Returning null frame on %s\n", ast->name);
206 return &p->fr;
209 static int nbs_xwrite(struct ast_channel *ast, struct ast_frame *frame)
211 struct nbs_pvt *p = ast->tech_pvt;
212 /* Write a frame of (presumably voice) data */
213 if (frame->frametype != AST_FRAME_VOICE) {
214 if (frame->frametype != AST_FRAME_IMAGE)
215 ast_log(LOG_WARNING, "Don't know what to do with frame type '%d'\n", frame->frametype);
216 return 0;
218 if (!(frame->subclass &
219 (AST_FORMAT_SLINEAR))) {
220 ast_log(LOG_WARNING, "Cannot handle frames in %d format\n", frame->subclass);
221 return 0;
223 if (ast->_state != AST_STATE_UP) {
224 /* Don't try tos end audio on-hook */
225 return 0;
227 if (nbs_write(p->nbs, frame->data, frame->datalen / 2) < 0)
228 return -1;
229 return 0;
232 static struct ast_channel *nbs_new(struct nbs_pvt *i, int state)
234 struct ast_channel *tmp;
235 tmp = ast_channel_alloc(1, state, 0, 0, "", "s", context, 0, "NBS/%s", i->stream);
236 if (tmp) {
237 tmp->tech = &nbs_tech;
238 tmp->fds[0] = nbs_fd(i->nbs);
239 tmp->nativeformats = prefformat;
240 tmp->rawreadformat = prefformat;
241 tmp->rawwriteformat = prefformat;
242 tmp->writeformat = prefformat;
243 tmp->readformat = prefformat;
244 if (state == AST_STATE_RING)
245 tmp->rings = 1;
246 tmp->tech_pvt = i;
247 ast_copy_string(tmp->context, context, sizeof(tmp->context));
248 ast_copy_string(tmp->exten, "s", sizeof(tmp->exten));
249 ast_string_field_set(tmp, language, "");
250 i->owner = tmp;
251 i->u = ast_module_user_add(tmp);
252 if (state != AST_STATE_DOWN) {
253 if (ast_pbx_start(tmp)) {
254 ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
255 ast_hangup(tmp);
258 } else
259 ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
260 return tmp;
264 static struct ast_channel *nbs_request(const char *type, int format, void *data, int *cause)
266 int oldformat;
267 struct nbs_pvt *p;
268 struct ast_channel *tmp = NULL;
270 oldformat = format;
271 format &= (AST_FORMAT_SLINEAR);
272 if (!format) {
273 ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", oldformat);
274 return NULL;
276 p = nbs_alloc(data);
277 if (p) {
278 tmp = nbs_new(p, AST_STATE_DOWN);
279 if (!tmp)
280 nbs_destroy(p);
282 return tmp;
285 static int unload_module(void)
287 /* First, take us out of the channel loop */
288 ast_channel_unregister(&nbs_tech);
289 return 0;
292 static int load_module(void)
294 /* Make sure we can register our channel type */
295 if (ast_channel_register(&nbs_tech)) {
296 ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
297 return -1;
299 return 0;
302 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Network Broadcast Sound Support");