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 GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
41 #define GST_CAT_DEFAULT rtsp_server_debug
43 static void gst_rtsp_server_get_property (GObject *object, guint propid,
44 GValue *value, GParamSpec *pspec);
45 static void gst_rtsp_server_set_property (GObject *object, guint propid,
46 const GValue *value, GParamSpec *pspec);
47 static void gst_rtsp_server_finalize (GObject *object);
49 static GstRTSPClient * default_accept_client (GstRTSPServer *server,
53 gst_rtsp_server_class_init (GstRTSPServerClass * klass)
55 GObjectClass *gobject_class;
57 gobject_class = G_OBJECT_CLASS (klass);
59 gobject_class->get_property = gst_rtsp_server_get_property;
60 gobject_class->set_property = gst_rtsp_server_set_property;
61 gobject_class->finalize = gst_rtsp_server_finalize;
64 * GstRTSPServer::backlog
66 * The backlog argument defines the maximum length to which the queue of
67 * pending connections for the server may grow. If a connection request arrives
68 * when the queue is full, the client may receive an error with an indication of
69 * ECONNREFUSED or, if the underlying protocol supports retransmission, the
70 * request may be ignored so that a later reattempt at connection succeeds.
72 g_object_class_install_property (gobject_class, PROP_BACKLOG,
73 g_param_spec_int ("backlog", "Backlog", "The maximum length to which the queue "
74 "of pending connections may grow",
75 0, G_MAXINT, DEFAULT_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
79 * The session port of the server. This is the port where the server will
82 g_object_class_install_property (gobject_class, PROP_PORT,
83 g_param_spec_int ("port", "Port", "The port the server uses to listen on",
84 1, 65535, DEFAULT_PORT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
86 * GstRTSPServer::session-pool
88 * The session pool of the server. By default each server has a separate
89 * session pool but sessions can be shared between servers by setting the same
90 * session pool on multiple servers.
92 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
93 g_param_spec_object ("session-pool", "Session Pool",
94 "The session pool to use for client session",
95 GST_TYPE_RTSP_SESSION_POOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
97 * GstRTSPServer::media-mapping
99 * The media mapping to use for this server. By default the server has no
100 * media mapping and thus cannot map urls to media streams.
102 g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
103 g_param_spec_object ("media-mapping", "Media Mapping",
104 "The media mapping to use for client session",
105 GST_TYPE_RTSP_MEDIA_MAPPING, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
107 klass->accept_client = default_accept_client;
109 GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
113 gst_rtsp_server_init (GstRTSPServer * server)
115 server->port = DEFAULT_PORT;
116 server->backlog = DEFAULT_BACKLOG;
117 server->session_pool = gst_rtsp_session_pool_new ();
118 server->media_mapping = gst_rtsp_media_mapping_new ();
122 gst_rtsp_server_finalize (GObject *object)
124 GstRTSPServer *server = GST_RTSP_SERVER (object);
126 g_object_unref (server->session_pool);
127 g_object_unref (server->media_mapping);
131 * gst_rtsp_server_new:
133 * Create a new #GstRTSPServer instance.
136 gst_rtsp_server_new (void)
138 GstRTSPServer *result;
140 result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
146 * gst_rtsp_server_set_port:
147 * @server: a #GstRTSPServer
150 * Configure @server to accept connections on the given port.
151 * @port should be a port number between 1 and 65535.
153 * This function must be called before the server is bound.
156 gst_rtsp_server_set_port (GstRTSPServer *server, gint port)
158 g_return_if_fail (GST_IS_RTSP_SERVER (server));
159 g_return_if_fail (port >= 1 && port <= 65535);
165 * gst_rtsp_server_get_port:
166 * @server: a #GstRTSPServer
168 * Get the port number on which the server will accept connections.
170 * Returns: the server port.
173 gst_rtsp_server_get_port (GstRTSPServer *server)
175 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
181 * gst_rtsp_server_set_backlog:
182 * @server: a #GstRTSPServer
183 * @backlog: the backlog
185 * configure the maximum amount of requests that may be queued for the
188 * This function must be called before the server is bound.
191 gst_rtsp_server_set_backlog (GstRTSPServer *server, gint backlog)
193 g_return_if_fail (GST_IS_RTSP_SERVER (server));
195 server->backlog = backlog;
199 * gst_rtsp_server_get_backlog:
200 * @server: a #GstRTSPServer
202 * The maximum amount of queued requests for the server.
204 * Returns: the server backlog.
207 gst_rtsp_server_get_backlog (GstRTSPServer *server)
209 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
211 return server->backlog;
215 * gst_rtsp_server_set_session_pool:
216 * @server: a #GstRTSPServer
217 * @pool: a #GstRTSPSessionPool
219 * configure @pool to be used as the session pool of @server.
222 gst_rtsp_server_set_session_pool (GstRTSPServer *server, GstRTSPSessionPool *pool)
224 GstRTSPSessionPool *old;
226 g_return_if_fail (GST_IS_RTSP_SERVER (server));
228 old = server->session_pool;
233 server->session_pool = pool;
235 g_object_unref (old);
240 * gst_rtsp_server_get_session_pool:
241 * @server: a #GstRTSPServer
243 * Get the #GstRTSPSessionPool used as the session pool of @server.
245 * Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
249 gst_rtsp_server_get_session_pool (GstRTSPServer *server)
251 GstRTSPSessionPool *result;
253 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
255 if ((result = server->session_pool))
256 g_object_ref (result);
262 * gst_rtsp_server_set_media_mapping:
263 * @server: a #GstRTSPServer
264 * @mapping: a #GstRTSPMediaMapping
266 * configure @mapping to be used as the media mapping of @server.
269 gst_rtsp_server_set_media_mapping (GstRTSPServer *server, GstRTSPMediaMapping *mapping)
271 GstRTSPMediaMapping *old;
273 g_return_if_fail (GST_IS_RTSP_SERVER (server));
275 old = server->media_mapping;
277 if (old != mapping) {
279 g_object_ref (mapping);
280 server->media_mapping = mapping;
282 g_object_unref (old);
288 * gst_rtsp_server_get_media_mapping:
289 * @server: a #GstRTSPServer
291 * Get the #GstRTSPMediaMapping used as the media mapping of @server.
293 * Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
296 GstRTSPMediaMapping *
297 gst_rtsp_server_get_media_mapping (GstRTSPServer *server)
299 GstRTSPMediaMapping *result;
301 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
303 if ((result = server->media_mapping))
304 g_object_ref (result);
310 gst_rtsp_server_get_property (GObject *object, guint propid,
311 GValue *value, GParamSpec *pspec)
313 GstRTSPServer *server = GST_RTSP_SERVER (object);
317 g_value_set_int (value, gst_rtsp_server_get_port (server));
320 g_value_set_int (value, gst_rtsp_server_get_backlog (server));
322 case PROP_SESSION_POOL:
323 g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
325 case PROP_MEDIA_MAPPING:
326 g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
329 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
334 gst_rtsp_server_set_property (GObject *object, guint propid,
335 const GValue *value, GParamSpec *pspec)
337 GstRTSPServer *server = GST_RTSP_SERVER (object);
341 gst_rtsp_server_set_port (server, g_value_get_int (value));
344 gst_rtsp_server_set_backlog (server, g_value_get_int (value));
346 case PROP_SESSION_POOL:
347 gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
349 case PROP_MEDIA_MAPPING:
350 gst_rtsp_server_set_media_mapping (server, g_value_get_object (value));
353 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
357 /* Prepare a server socket for @server and make it listen on the configured port */
359 gst_rtsp_server_sink_init_send (GstRTSPServer * server)
363 /* create server socket */
364 if ((server->server_sock.fd = socket (AF_INET, SOCK_STREAM, 0)) == -1)
367 GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
368 server->server_sock.fd);
370 /* make address reusable */
372 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_REUSEADDR,
373 (void *) &ret, sizeof (ret)) < 0)
376 /* keep connection alive; avoids SIGPIPE during write */
378 if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
379 (void *) &ret, sizeof (ret)) < 0)
380 goto keepalive_failed;
382 /* name the socket */
383 memset (&server->server_sin, 0, sizeof (server->server_sin));
384 server->server_sin.sin_family = AF_INET; /* network socket */
385 server->server_sin.sin_port = htons (server->port); /* on port */
386 server->server_sin.sin_addr.s_addr = htonl (INADDR_ANY); /* for hosts */
389 GST_DEBUG_OBJECT (server, "binding server socket to address");
390 ret = bind (server->server_sock.fd, (struct sockaddr *) &server->server_sin,
391 sizeof (server->server_sin));
395 /* set the server socket to nonblocking */
396 fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
398 GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
399 server->server_sock.fd, server->backlog);
400 if (listen (server->server_sock.fd, server->backlog) == -1)
403 GST_DEBUG_OBJECT (server,
404 "listened on server socket %d, returning from connection setup",
405 server->server_sock.fd);
407 GST_INFO_OBJECT (server, "listening on port %d", server->port);
414 GST_ERROR_OBJECT (server, "failed to create socket: %s", g_strerror (errno));
419 if (server->server_sock.fd >= 0) {
420 close (server->server_sock.fd);
421 server->server_sock.fd = -1;
423 GST_ERROR_OBJECT (server, "failed to reuse socket: %s", g_strerror (errno));
428 if (server->server_sock.fd >= 0) {
429 close (server->server_sock.fd);
430 server->server_sock.fd = -1;
432 GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s", g_strerror (errno));
437 if (server->server_sock.fd >= 0) {
438 close (server->server_sock.fd);
439 server->server_sock.fd = -1;
441 GST_ERROR_OBJECT (server, "failed to listen on socket: %s", g_strerror (errno));
446 if (server->server_sock.fd >= 0) {
447 close (server->server_sock.fd);
448 server->server_sock.fd = -1;
450 GST_ERROR_OBJECT (server, "failed to bind on socket: %s", g_strerror (errno));
455 /* default method for creating a new client object in the server to accept and
456 * handle a client connection on this server */
457 static GstRTSPClient *
458 default_accept_client (GstRTSPServer *server, GIOChannel *channel)
460 GstRTSPClient *client;
462 /* a new client connected, create a session to handle the client. */
463 client = gst_rtsp_client_new ();
465 /* set the session pool that this client should use */
466 gst_rtsp_client_set_session_pool (client, server->session_pool);
468 /* set the session pool that this client should use */
469 gst_rtsp_client_set_media_mapping (client, server->media_mapping);
471 /* accept connections for that client, this function returns after accepting
472 * the connection and will run the remainder of the communication with the
473 * client asyncronously. */
474 if (!gst_rtsp_client_accept (client, channel))
482 GST_ERROR_OBJECT (server, "Could not accept client on server socket %d: %s (%d)",
483 server->server_sock.fd, g_strerror (errno), errno);
484 gst_object_unref (client);
490 * gst_rtsp_server_io_func:
491 * @channel: a #GIOChannel
492 * @condition: the condition on @source
494 * A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
495 * new connection on @channel or @server.
497 * Returns: TRUE if the source could be connected, FALSE if an error occured.
500 gst_rtsp_server_io_func (GIOChannel *channel, GIOCondition condition, GstRTSPServer *server)
502 GstRTSPClient *client = NULL;
503 GstRTSPServerClass *klass;
505 if (condition & G_IO_IN) {
506 klass = GST_RTSP_SERVER_GET_CLASS (server);
508 /* a new client connected, create a client object to handle the client. */
509 if (klass->accept_client)
510 client = klass->accept_client (server, channel);
514 /* can unref the client now, when the request is finished, it will be
516 gst_object_unref (client);
519 GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
526 GST_ERROR_OBJECT (server, "failed to create a client");
532 * gst_rtsp_server_get_io_channel:
533 * @server: a #GstRTSPServer
535 * Create a #GIOChannel for @server.
537 * Returns: the GIOChannel for @server or NULL when an error occured.
540 gst_rtsp_server_get_io_channel (GstRTSPServer *server)
542 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
544 if (server->io_channel == NULL) {
545 if (!gst_rtsp_server_sink_init_send (server))
548 /* create IO channel for the socket */
549 server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
551 return server->io_channel;
560 * gst_rtsp_server_create_watch:
561 * @server: a #GstRTSPServer
563 * Create a #GSource for @server. The new source will have a default
564 * #GIOFunc of gst_rtsp_server_io_func().
566 * Returns: the #GSource for @server or NULL when an error occured.
569 gst_rtsp_server_create_watch (GstRTSPServer *server)
571 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
573 if (server->io_watch == NULL) {
576 channel = gst_rtsp_server_get_io_channel (server);
580 /* create a watch for reads (new connections) and possible errors */
581 server->io_watch = g_io_create_watch (channel, G_IO_IN |
582 G_IO_ERR | G_IO_HUP | G_IO_NVAL);
584 /* configure the callback */
585 g_source_set_callback (server->io_watch, (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
587 return server->io_watch;
596 * gst_rtsp_server_attach:
597 * @server: a #GstRTSPServer
598 * @context: a #GMainContext
600 * Attaches @server to @context. When the mainloop for @context is run, the
601 * server will be dispatched.
603 * This function should be called when the server properties and urls are fully
604 * configured and the server is ready to start.
606 * Returns: the ID (greater than 0) for the source within the GMainContext.
609 gst_rtsp_server_attach (GstRTSPServer *server, GMainContext *context)
614 g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
616 source = gst_rtsp_server_create_watch (server);
620 res = g_source_attach (source, context);