2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
20 #include <sys/ioctl.h>
22 #include "rtsp-server.h"
23 #include "rtsp-client.h"
25 #define DEFAULT_BACKLOG 5
26 #define DEFAULT_PORT 8554
38 G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
40 static void gst_rtsp_server_get_property (GObject *object, guint propid,
41 GValue *value, GParamSpec *pspec);
42 static void gst_rtsp_server_set_property (GObject *object, guint propid,
43 const GValue *value, GParamSpec *pspec);
45 static GstRTSPClient * default_accept_client (GstRTSPServer *server,
49 gst_rtsp_server_class_init (GstRTSPServerClass * klass)
51 GObjectClass *gobject_class;
53 gobject_class = G_OBJECT_CLASS (klass);
55 gobject_class->get_property = gst_rtsp_server_get_property;
56 gobject_class->set_property = gst_rtsp_server_set_property;
59 * GstRTSPServer::backlog
61 * The backlog argument defines the maximum length to which the queue of
62 * pending connections for the server may grow. If a connection request arrives
63 * when the queue is full, the client may receive an error with an indication of
64 * ECONNREFUSED or, if the underlying protocol supports retransmission, the
65 * request may be ignored so that a later reattempt at connection succeeds.
67 g_object_class_install_property (gobject_class, PROP_BACKLOG,
68 g_param_spec_int ("backlog", "Backlog", "The maximum length to which the queue "
69 "of pending connections may grow",
70 0, G_MAXINT, DEFAULT_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
74 * The session port of the server. This is the port where the server will
77 g_object_class_install_property (gobject_class, PROP_PORT,
78 g_param_spec_int ("port", "Port", "The port the server uses to listen on",
79 1, 65535, DEFAULT_PORT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
81 * GstRTSPServer::session-pool
83 * The session pool of the server. By default each server has a separate
84 * session pool but sessions can be shared between servers by setting the same
85 * session pool on multiple servers.
87 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
88 g_param_spec_object ("session-pool", "Session Pool",
89 "The session pool to use for client session",
90 GST_TYPE_RTSP_SESSION_POOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
92 * GstRTSPServer::media-mapping
94 * The media mapping to use for this server. By default the server has no
95 * media mapping and thus cannot map urls to media streams.
97 g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
98 g_param_spec_object ("media-mapping", "Media Mapping",
99 "The media mapping to use for client session",
100 GST_TYPE_RTSP_MEDIA_MAPPING, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
102 klass->accept_client = default_accept_client;
106 gst_rtsp_server_init (GstRTSPServer * server)
108 server->port = DEFAULT_PORT;
109 server->backlog = DEFAULT_BACKLOG;
110 server->session_pool = gst_rtsp_session_pool_new ();
111 server->media_mapping = gst_rtsp_media_mapping_new ();
115 * gst_rtsp_server_new:
117 * Create a new #GstRTSPServer instance.
120 gst_rtsp_server_new (void)
122 GstRTSPServer *result;
124 result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
130 * gst_rtsp_server_set_port:
131 * @server: a #GstRTSPServer
134 * Configure @server to accept connections on the given port.
135 * @port should be a port number between 1 and 65535.
137 * This function must be called before the server is bound.
140 gst_rtsp_server_set_port (GstRTSPServer *server, gint port)
142 g_return_if_fail (GST_IS_RTSP_SERVER (server));
143 g_return_if_fail (port >= 1 && port <= 65535);
149 * gst_rtsp_server_get_port:
150 * @server: a #GstRTSPServer
152 * Get the port number on which the server will accept connections.
154 * Returns: the server port.
157 gst_rtsp_server_get_port (GstRTSPServer *server)
159 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
165 * gst_rtsp_server_set_backlog:
166 * @server: a #GstRTSPServer
167 * @backlog: the backlog
169 * configure the maximum amount of requests that may be queued for the
172 * This function must be called before the server is bound.
175 gst_rtsp_server_set_backlog (GstRTSPServer *server, gint backlog)
177 g_return_if_fail (GST_IS_RTSP_SERVER (server));
179 server->backlog = backlog;
183 * gst_rtsp_server_get_backlog:
184 * @server: a #GstRTSPServer
186 * The maximum amount of queued requests for the server.
188 * Returns: the server backlog.
191 gst_rtsp_server_get_backlog (GstRTSPServer *server)
193 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
195 return server->backlog;
199 * gst_rtsp_server_set_session_pool:
200 * @server: a #GstRTSPServer
201 * @pool: a #GstRTSPSessionPool
203 * configure @pool to be used as the session pool of @server.
206 gst_rtsp_server_set_session_pool (GstRTSPServer *server, GstRTSPSessionPool *pool)
208 GstRTSPSessionPool *old;
210 g_return_if_fail (GST_IS_RTSP_SERVER (server));
212 old = server->session_pool;
217 server->session_pool = pool;
219 g_object_unref (old);
224 * gst_rtsp_server_get_session_pool:
225 * @server: a #GstRTSPServer
227 * Get the #GstRTSPSessionPool used as the session pool of @server.
229 * Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
233 gst_rtsp_server_get_session_pool (GstRTSPServer *server)
235 GstRTSPSessionPool *result;
237 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
239 if ((result = server->session_pool))
240 g_object_ref (result);
246 * gst_rtsp_server_set_media_mapping:
247 * @server: a #GstRTSPServer
248 * @mapping: a #GstRTSPMediaMapping
250 * configure @mapping to be used as the media mapping of @server.
253 gst_rtsp_server_set_media_mapping (GstRTSPServer *server, GstRTSPMediaMapping *mapping)
255 GstRTSPMediaMapping *old;
257 g_return_if_fail (GST_IS_RTSP_SERVER (server));
259 old = server->media_mapping;
261 if (old != mapping) {
263 g_object_ref (mapping);
264 server->media_mapping = mapping;
266 g_object_unref (old);
272 * gst_rtsp_server_get_media_mapping:
273 * @server: a #GstRTSPServer
275 * Get the #GstRTSPMediaMapping used as the media mapping of @server.
277 * Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
280 GstRTSPMediaMapping *
281 gst_rtsp_server_get_media_mapping (GstRTSPServer *server)
283 GstRTSPMediaMapping *result;
285 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
287 if ((result = server->media_mapping))
288 g_object_ref (result);
294 gst_rtsp_server_get_property (GObject *object, guint propid,
295 GValue *value, GParamSpec *pspec)
297 GstRTSPServer *server = GST_RTSP_SERVER (object);
301 g_value_set_int (value, gst_rtsp_server_get_port (server));
304 g_value_set_int (value, gst_rtsp_server_get_backlog (server));
306 case PROP_SESSION_POOL:
307 g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
309 case PROP_MEDIA_MAPPING:
310 g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
313 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
318 gst_rtsp_server_set_property (GObject *object, guint propid,
319 const GValue *value, GParamSpec *pspec)
321 GstRTSPServer *server = GST_RTSP_SERVER (object);
325 gst_rtsp_server_set_port (server, g_value_get_int (value));
328 gst_rtsp_server_set_backlog (server, g_value_get_int (value));
330 case PROP_SESSION_POOL:
331 gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
333 case PROP_MEDIA_MAPPING:
334 gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
337 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
341 /* Prepare a server socket for @server and make it listen on the configured port */
343 gst_rtsp_server_sink_init_send (GstRTSPServer * server)
347 /* create server socket */
348 if ((server->server_sock.fd = socket (AF_INET, SOCK_STREAM, 0)) == -1)
351 GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
352 server->server_sock.fd);
354 /* make address reusable */
356 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_REUSEADDR,
357 (void *) &ret, sizeof (ret)) < 0)
360 /* keep connection alive; avoids SIGPIPE during write */
362 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
363 (void *) &ret, sizeof (ret)) < 0)
364 goto keepalive_failed;
366 /* name the socket */
367 memset (&server->server_sin, 0, sizeof (server->server_sin));
368 server->server_sin.sin_family = AF_INET; /* network socket */
369 server->server_sin.sin_port = htons (server->port); /* on port */
370 server->server_sin.sin_addr.s_addr = htonl (INADDR_ANY); /* for hosts */
373 GST_DEBUG_OBJECT (server, "binding server socket to address");
374 ret = bind (server->server_sock.fd, (struct sockaddr *) &server->server_sin,
375 sizeof (server->server_sin));
379 /* set the server socket to nonblocking */
380 fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
382 GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
383 server->server_sock.fd, server->backlog);
384 if (listen (server->server_sock.fd, server->backlog) == -1)
387 GST_DEBUG_OBJECT (server,
388 "listened on server socket %d, returning from connection setup",
389 server->server_sock.fd);
391 g_message ("listening on port %d", server->port);
398 GST_ERROR_OBJECT (server, "failed to create socket: %s", g_strerror (errno));
403 if (server->server_sock.fd >= 0) {
404 close (server->server_sock.fd);
405 server->server_sock.fd = -1;
407 GST_ERROR_OBJECT (server, "failed to reuse socket: %s", g_strerror (errno));
412 if (server->server_sock.fd >= 0) {
413 close (server->server_sock.fd);
414 server->server_sock.fd = -1;
416 GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s", g_strerror (errno));
421 if (server->server_sock.fd >= 0) {
422 close (server->server_sock.fd);
423 server->server_sock.fd = -1;
425 GST_ERROR_OBJECT (server, "failed to listen on socket: %s", g_strerror (errno));
430 if (server->server_sock.fd >= 0) {
431 close (server->server_sock.fd);
432 server->server_sock.fd = -1;
434 GST_ERROR_OBJECT (server, "failed to bind on socket: %s", g_strerror (errno));
439 /* default method for creating a new client object in the server to accept and
440 * handle a client connection on this server */
441 static GstRTSPClient *
442 default_accept_client (GstRTSPServer *server, GIOChannel *channel)
444 GstRTSPClient *client;
446 /* a new client connected, create a session to handle the client. */
447 client = gst_rtsp_client_new ();
449 /* set the session pool that this client should use */
450 gst_rtsp_client_set_session_pool (client, server->session_pool);
452 /* set the session pool that this client should use */
453 gst_rtsp_client_set_media_mapping (client, server->media_mapping);
455 /* accept connections for that client, this function returns after accepting
456 * the connection and will run the remainder of the communication with the
457 * client asyncronously. */
458 if (!gst_rtsp_client_accept (client, channel))
466 g_error ("Could not accept client on server socket %d: %s (%d)",
467 server->server_sock.fd, g_strerror (errno), errno);
468 gst_object_unref (client);
474 * gst_rtsp_server_io_func:
475 * @channel: a #GIOChannel
476 * @condition: the condition on @source
478 * A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
479 * new connection on @channel or @server.
481 * Returns: TRUE if the source could be connected, FALSE if an error occured.
484 gst_rtsp_server_io_func (GIOChannel *channel, GIOCondition condition, GstRTSPServer *server)
486 GstRTSPClient *client = NULL;
487 GstRTSPServerClass *klass;
489 if (condition & G_IO_IN) {
490 klass = GST_RTSP_SERVER_GET_CLASS (server);
492 /* a new client connected, create a client object to handle the client. */
493 if (klass->accept_client)
494 client = klass->accept_client (server, channel);
498 /* can unref the client now, when the request is finished, it will be
500 gst_object_unref (client);
503 g_print ("received unknown event %08x", condition);
510 GST_ERROR_OBJECT (server, "failed to create a client");
516 * gst_rtsp_server_get_io_channel:
517 * @server: a #GstRTSPServer
519 * Create a #GIOChannel for @server.
521 * Returns: the GIOChannel for @server or NULL when an error occured.
524 gst_rtsp_server_get_io_channel (GstRTSPServer *server)
526 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
528 if (server->io_channel == NULL) {
529 if (!gst_rtsp_server_sink_init_send (server))
532 /* create IO channel for the socket */
533 server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
535 return server->io_channel;
544 * gst_rtsp_server_create_watch:
545 * @server: a #GstRTSPServer
547 * Create a #GSource for @server. The new source will have a default
548 * #GIOFunc of gst_rtsp_server_io_func().
550 * Returns: the #GSource for @server or NULL when an error occured.
553 gst_rtsp_server_create_watch (GstRTSPServer *server)
555 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
557 if (server->io_watch == NULL) {
560 channel = gst_rtsp_server_get_io_channel (server);
564 /* create a watch for reads (new connections) and possible errors */
565 server->io_watch = g_io_create_watch (channel, G_IO_IN |
566 G_IO_ERR | G_IO_HUP | G_IO_NVAL);
568 /* configure the callback */
569 g_source_set_callback (server->io_watch, (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
571 return server->io_watch;
580 * gst_rtsp_server_attach:
581 * @server: a #GstRTSPServer
582 * @context: a #GMainContext
584 * Attaches @server to @context. When the mainloop for @context is run, the
585 * server will be dispatched.
587 * This function should be called when the server properties and urls are fully
588 * configured and the server is ready to start.
590 * Returns: the ID (greater than 0) for the source within the GMainContext.
593 gst_rtsp_server_attach (GstRTSPServer *server, GMainContext *context)
598 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
600 source = gst_rtsp_server_create_watch (server);
604 res = g_source_attach (source, context);