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., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
21 * @short_description: A client connection state
22 * @see_also: #GstRTSPServer, #GstRTSPThreadPool
24 * The client object handles the connection with a client for as long as a TCP
27 * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
28 * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
29 * #GstRTSPAuth and #GstRTSPThreadPool from the server.
31 * The client connection should be configured with the #GstRTSPConnection using
32 * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
33 * using gst_rtsp_client_attach(). From then on the client will handle requests
36 * Use gst_rtsp_client_session_filter() to iterate or modify all the
37 * #GstRTSPSession objects managed by the client object.
39 * Last reviewed on 2013-07-11 (1.0.0)
45 #include "rtsp-client.h"
47 #include "rtsp-params.h"
49 #define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
50 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
53 * send_lock, lock, tunnels_lock
56 struct _GstRTSPClientPrivate
58 GMutex lock; /* protects everything else */
60 GstRTSPConnection *connection;
66 GstRTSPClientSendFunc send_func; /* protected by send_lock */
67 gpointer send_data; /* protected by send_lock */
68 GDestroyNotify send_notify; /* protected by send_lock */
70 GstRTSPSessionPool *session_pool;
71 GstRTSPMountPoints *mount_points;
73 GstRTSPThreadPool *thread_pool;
75 /* used to cache the media in the last requested DESCRIBE so that
76 * we can pick it up in the next SETUP immediately */
84 static GMutex tunnels_lock;
85 static GHashTable *tunnels; /* protected by tunnels_lock */
87 #define DEFAULT_SESSION_POOL NULL
88 #define DEFAULT_MOUNT_POINTS NULL
102 SIGNAL_OPTIONS_REQUEST,
103 SIGNAL_DESCRIBE_REQUEST,
104 SIGNAL_SETUP_REQUEST,
106 SIGNAL_PAUSE_REQUEST,
107 SIGNAL_TEARDOWN_REQUEST,
108 SIGNAL_SET_PARAMETER_REQUEST,
109 SIGNAL_GET_PARAMETER_REQUEST,
110 SIGNAL_HANDLE_RESPONSE,
114 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
115 #define GST_CAT_DEFAULT rtsp_client_debug
117 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
119 static void gst_rtsp_client_get_property (GObject * object, guint propid,
120 GValue * value, GParamSpec * pspec);
121 static void gst_rtsp_client_set_property (GObject * object, guint propid,
122 const GValue * value, GParamSpec * pspec);
123 static void gst_rtsp_client_finalize (GObject * obj);
125 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
126 static void client_session_finalized (GstRTSPClient * client,
127 GstRTSPSession * session);
128 static void unlink_session_transports (GstRTSPClient * client,
129 GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
130 static gboolean default_configure_client_transport (GstRTSPClient * client,
131 GstRTSPContext * ctx, GstRTSPTransport * ct);
132 static GstRTSPResult default_params_set (GstRTSPClient * client,
133 GstRTSPContext * ctx);
134 static GstRTSPResult default_params_get (GstRTSPClient * client,
135 GstRTSPContext * ctx);
137 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
140 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
142 GObjectClass *gobject_class;
144 g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
146 gobject_class = G_OBJECT_CLASS (klass);
148 gobject_class->get_property = gst_rtsp_client_get_property;
149 gobject_class->set_property = gst_rtsp_client_set_property;
150 gobject_class->finalize = gst_rtsp_client_finalize;
152 klass->create_sdp = create_sdp;
153 klass->configure_client_transport = default_configure_client_transport;
154 klass->params_set = default_params_set;
155 klass->params_get = default_params_get;
157 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
158 g_param_spec_object ("session-pool", "Session Pool",
159 "The session pool to use for client session",
160 GST_TYPE_RTSP_SESSION_POOL,
161 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
163 g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
164 g_param_spec_object ("mount-points", "Mount Points",
165 "The mount points to use for client session",
166 GST_TYPE_RTSP_MOUNT_POINTS,
167 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
169 gst_rtsp_client_signals[SIGNAL_CLOSED] =
170 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
171 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
172 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
174 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
175 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
176 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
177 g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
179 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
180 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
181 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
182 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
185 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
186 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
187 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
188 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
191 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
192 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
193 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
194 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
197 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
198 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
199 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
200 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
203 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
204 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
205 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
206 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
209 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
210 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
212 NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
215 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
216 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
217 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
218 set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
219 G_TYPE_NONE, 1, G_TYPE_POINTER);
221 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
222 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
224 get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
225 G_TYPE_NONE, 1, G_TYPE_POINTER);
227 gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
228 g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
229 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
230 handle_response), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
231 G_TYPE_NONE, 1, G_TYPE_POINTER);
234 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
235 g_mutex_init (&tunnels_lock);
237 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
241 gst_rtsp_client_init (GstRTSPClient * client)
243 GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
247 g_mutex_init (&priv->lock);
248 g_mutex_init (&priv->send_lock);
252 static GstRTSPFilterResult
253 filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
256 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
258 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
259 unlink_session_transports (client, sess, sessmedia);
261 /* unmanage the media in the session */
262 return GST_RTSP_FILTER_REMOVE;
266 client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
268 /* unlink all media managed in this session */
269 gst_rtsp_session_filter (session, filter_session, client);
273 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
275 GstRTSPClientPrivate *priv = client->priv;
278 for (walk = priv->sessions; walk; walk = g_list_next (walk)) {
279 GstRTSPSession *msession = (GstRTSPSession *) walk->data;
281 /* we already know about this session */
282 if (msession == session)
286 GST_INFO ("watching session %p", session);
288 g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
290 priv->sessions = g_list_prepend (priv->sessions, session);
294 client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session)
296 GstRTSPClientPrivate *priv = client->priv;
298 GST_INFO ("unwatching session %p", session);
300 g_object_weak_unref (G_OBJECT (session),
301 (GWeakNotify) client_session_finalized, client);
302 priv->sessions = g_list_remove (priv->sessions, session);
306 client_cleanup_session (GstRTSPClient * client, GstRTSPSession * session)
308 g_object_weak_unref (G_OBJECT (session),
309 (GWeakNotify) client_session_finalized, client);
310 client_unlink_session (client, session);
314 client_cleanup_sessions (GstRTSPClient * client)
316 GstRTSPClientPrivate *priv = client->priv;
319 /* remove weak-ref from sessions */
320 for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) {
321 client_cleanup_session (client, (GstRTSPSession *) sessions->data);
323 g_list_free (priv->sessions);
324 priv->sessions = NULL;
327 /* A client is finalized when the connection is broken */
329 gst_rtsp_client_finalize (GObject * obj)
331 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
332 GstRTSPClientPrivate *priv = client->priv;
334 GST_INFO ("finalize client %p", client);
336 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
339 g_source_destroy ((GSource *) priv->watch);
341 client_cleanup_sessions (client);
343 if (priv->connection)
344 gst_rtsp_connection_free (priv->connection);
345 if (priv->session_pool)
346 g_object_unref (priv->session_pool);
347 if (priv->mount_points)
348 g_object_unref (priv->mount_points);
350 g_object_unref (priv->auth);
351 if (priv->thread_pool)
352 g_object_unref (priv->thread_pool);
357 gst_rtsp_media_unprepare (priv->media);
358 g_object_unref (priv->media);
361 g_free (priv->server_ip);
362 g_mutex_clear (&priv->lock);
363 g_mutex_clear (&priv->send_lock);
365 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
369 gst_rtsp_client_get_property (GObject * object, guint propid,
370 GValue * value, GParamSpec * pspec)
372 GstRTSPClient *client = GST_RTSP_CLIENT (object);
375 case PROP_SESSION_POOL:
376 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
378 case PROP_MOUNT_POINTS:
379 g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
382 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
387 gst_rtsp_client_set_property (GObject * object, guint propid,
388 const GValue * value, GParamSpec * pspec)
390 GstRTSPClient *client = GST_RTSP_CLIENT (object);
393 case PROP_SESSION_POOL:
394 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
396 case PROP_MOUNT_POINTS:
397 gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
400 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
405 * gst_rtsp_client_new:
407 * Create a new #GstRTSPClient instance.
409 * Returns: a new #GstRTSPClient
412 gst_rtsp_client_new (void)
414 GstRTSPClient *result;
416 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
422 send_message (GstRTSPClient * client, GstRTSPSession * session,
423 GstRTSPMessage * message, gboolean close)
425 GstRTSPClientPrivate *priv = client->priv;
427 gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
428 "GStreamer RTSP server");
430 /* remove any previous header */
431 gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
433 /* add the new session header for new session ids */
435 gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
436 gst_rtsp_session_get_header (session));
439 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
440 gst_rtsp_message_dump (message);
444 gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
446 g_mutex_lock (&priv->send_lock);
448 priv->send_func (client, message, close, priv->send_data);
449 g_mutex_unlock (&priv->send_lock);
451 gst_rtsp_message_unset (message);
455 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
456 GstRTSPContext * ctx)
458 gst_rtsp_message_init_response (ctx->response, code,
459 gst_rtsp_status_as_text (code), ctx->request);
461 send_message (client, NULL, ctx->response, FALSE);
465 paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
467 if (path1 == NULL || path2 == NULL)
470 if (strlen (path1) != len2)
473 if (strncmp (path1, path2, len2))
479 /* this function is called to initially find the media for the DESCRIBE request
480 * but is cached for when the same client (without breaking the connection) is
481 * doing a setup for the exact same url. */
482 static GstRTSPMedia *
483 find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
486 GstRTSPClientPrivate *priv = client->priv;
487 GstRTSPMediaFactory *factory;
491 /* find the longest matching factory for the uri first */
492 if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
496 ctx->factory = factory;
498 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
499 goto no_factory_access;
501 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
507 path_len = strlen (path);
509 if (!paths_are_equal (priv->path, path, path_len)) {
510 GstRTSPThread *thread;
512 /* remove any previously cached values before we try to construct a new
518 gst_rtsp_media_unprepare (priv->media);
519 g_object_unref (priv->media);
523 /* prepare the media and add it to the pipeline */
524 if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
529 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
530 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
534 /* prepare the media */
535 if (!(gst_rtsp_media_prepare (media, thread)))
538 /* now keep track of the uri and the media */
539 priv->path = g_strndup (path, path_len);
542 /* we have seen this path before, used cached media */
545 GST_INFO ("reusing cached media %p for path %s", media, priv->path);
548 g_object_unref (factory);
552 g_object_ref (media);
559 GST_ERROR ("client %p: no factory for path %s", client, path);
560 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
565 GST_ERROR ("client %p: not authorized to see factory path %s", client,
567 /* error reply is already sent */
572 GST_ERROR ("client %p: not authorized for factory path %s", client, path);
573 /* error reply is already sent */
578 GST_ERROR ("client %p: can't create media", client);
579 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
580 g_object_unref (factory);
586 GST_ERROR ("client %p: can't create thread", client);
587 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
588 g_object_unref (media);
590 g_object_unref (factory);
596 GST_ERROR ("client %p: can't prepare media", client);
597 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
598 g_object_unref (media);
600 g_object_unref (factory);
607 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
609 GstRTSPClientPrivate *priv = client->priv;
610 GstRTSPMessage message = { 0 };
615 gst_rtsp_message_init_data (&message, channel);
617 /* FIXME, need some sort of iovec RTSPMessage here */
618 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
621 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
623 g_mutex_lock (&priv->send_lock);
625 priv->send_func (client, &message, FALSE, priv->send_data);
626 g_mutex_unlock (&priv->send_lock);
628 gst_rtsp_message_steal_body (&message, &data, &usize);
629 gst_buffer_unmap (buffer, &map_info);
631 gst_rtsp_message_unset (&message);
637 link_transport (GstRTSPClient * client, GstRTSPSession * session,
638 GstRTSPStreamTransport * trans)
640 GstRTSPClientPrivate *priv = client->priv;
642 GST_DEBUG ("client %p: linking transport %p", client, trans);
644 gst_rtsp_stream_transport_set_callbacks (trans,
645 (GstRTSPSendFunc) do_send_data,
646 (GstRTSPSendFunc) do_send_data, client, NULL);
648 priv->transports = g_list_prepend (priv->transports, trans);
650 /* make sure our session can't expire */
651 gst_rtsp_session_prevent_expire (session);
655 unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
656 GstRTSPStreamTransport * trans)
658 GstRTSPClientPrivate *priv = client->priv;
660 GST_DEBUG ("client %p: unlinking transport %p", client, trans);
662 gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
664 priv->transports = g_list_remove (priv->transports, trans);
666 /* our session can now expire */
667 gst_rtsp_session_allow_expire (session);
671 unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
672 GstRTSPSessionMedia * sessmedia)
677 gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
678 for (i = 0; i < n_streams; i++) {
679 GstRTSPStreamTransport *trans;
680 const GstRTSPTransport *tr;
682 /* get the transport, if there is no transport configured, skip this stream */
683 trans = gst_rtsp_session_media_get_transport (sessmedia, i);
687 tr = gst_rtsp_stream_transport_get_transport (trans);
689 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
690 /* for TCP, unlink the stream from the TCP connection of the client */
691 unlink_transport (client, session, trans);
697 close_connection (GstRTSPClient * client)
699 GstRTSPClientPrivate *priv = client->priv;
700 const gchar *tunnelid;
702 GST_DEBUG ("client %p: closing connection", client);
704 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
705 g_mutex_lock (&tunnels_lock);
706 /* remove from tunnelids */
707 g_hash_table_remove (tunnels, tunnelid);
708 g_mutex_unlock (&tunnels_lock);
711 gst_rtsp_connection_close (priv->connection);
715 make_path_from_uri (GstRTSPClient * client, GstRTSPUrl * uri)
720 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
722 path = g_strdup (uri->abspath);
728 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
730 GstRTSPClientPrivate *priv = client->priv;
731 GstRTSPSession *session;
732 GstRTSPSessionMedia *sessmedia;
733 GstRTSPStatusCode code;
740 session = ctx->session;
745 path = make_path_from_uri (client, ctx->uri);
747 /* get a handle to the configuration of the media in the session */
748 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
752 /* only aggregate control for now.. */
753 if (path[matched] != '\0')
758 ctx->sessmedia = sessmedia;
760 /* we emit the signal before closing the connection */
761 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
764 /* unlink the all TCP callbacks */
765 unlink_session_transports (client, session, sessmedia);
767 /* remove the session from the watched sessions */
768 client_unwatch_session (client, session);
770 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
772 /* unmanage the media in the session, returns false if all media session
774 if (!gst_rtsp_session_release_media (session, sessmedia)) {
775 /* remove the session */
776 gst_rtsp_session_pool_remove (priv->session_pool, session);
778 /* construct the response now */
779 code = GST_RTSP_STS_OK;
780 gst_rtsp_message_init_response (ctx->response, code,
781 gst_rtsp_status_as_text (code), ctx->request);
783 send_message (client, session, ctx->response, TRUE);
790 GST_ERROR ("client %p: no session", client);
791 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
796 GST_ERROR ("client %p: no uri supplied", client);
797 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
802 GST_ERROR ("client %p: no media for uri", client);
803 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
809 GST_ERROR ("client %p: no aggregate path %s", client, path);
810 send_generic_response (client,
811 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
818 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
822 res = gst_rtsp_params_set (client, ctx);
828 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
832 res = gst_rtsp_params_get (client, ctx);
838 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
844 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
845 if (res != GST_RTSP_OK)
849 /* no body, keep-alive request */
850 send_generic_response (client, GST_RTSP_STS_OK, ctx);
852 /* there is a body, handle the params */
853 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
854 if (res != GST_RTSP_OK)
857 send_message (client, ctx->session, ctx->response, FALSE);
860 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
868 GST_ERROR ("client %p: bad request", client);
869 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
875 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
881 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
882 if (res != GST_RTSP_OK)
886 /* no body, keep-alive request */
887 send_generic_response (client, GST_RTSP_STS_OK, ctx);
889 /* there is a body, handle the params */
890 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
891 if (res != GST_RTSP_OK)
894 send_message (client, ctx->session, ctx->response, FALSE);
897 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
905 GST_ERROR ("client %p: bad request", client);
906 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
912 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
914 GstRTSPSession *session;
915 GstRTSPSessionMedia *sessmedia;
916 GstRTSPStatusCode code;
917 GstRTSPState rtspstate;
921 if (!(session = ctx->session))
927 path = make_path_from_uri (client, ctx->uri);
929 /* get a handle to the configuration of the media in the session */
930 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
934 if (path[matched] != '\0')
939 ctx->sessmedia = sessmedia;
941 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
942 /* the session state must be playing or recording */
943 if (rtspstate != GST_RTSP_STATE_PLAYING &&
944 rtspstate != GST_RTSP_STATE_RECORDING)
947 /* unlink the all TCP callbacks */
948 unlink_session_transports (client, session, sessmedia);
950 /* then pause sending */
951 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
953 /* construct the response now */
954 code = GST_RTSP_STS_OK;
955 gst_rtsp_message_init_response (ctx->response, code,
956 gst_rtsp_status_as_text (code), ctx->request);
958 send_message (client, session, ctx->response, FALSE);
960 /* the state is now READY */
961 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
963 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
970 GST_ERROR ("client %p: no seesion", client);
971 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
976 GST_ERROR ("client %p: no uri supplied", client);
977 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
982 GST_ERROR ("client %p: no media for uri", client);
983 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
989 GST_ERROR ("client %p: no aggregate path %s", client, path);
990 send_generic_response (client,
991 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
997 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
998 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1004 /* convert @url and @path to a URL used as a content base for the factory
1005 * located at @path */
1007 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, gchar * path)
1010 gchar *result, *trail;
1012 /* check for trailing '/' and append one */
1013 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1018 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1020 result = gst_rtsp_url_get_request_uri (&tmp);
1021 g_free (tmp.abspath);
1027 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1029 GstRTSPSession *session;
1030 GstRTSPSessionMedia *sessmedia;
1031 GstRTSPMedia *media;
1032 GstRTSPStatusCode code;
1035 guint n_streams, i, infocount;
1036 gchar *str, *base_url;
1037 GstRTSPTimeRange *range;
1039 GstRTSPState rtspstate;
1040 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1044 if (!(session = ctx->session))
1047 if (!(uri = ctx->uri))
1050 path = make_path_from_uri (client, uri);
1052 /* get a handle to the configuration of the media in the session */
1053 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1057 if (path[matched] != '\0')
1060 ctx->sessmedia = sessmedia;
1061 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1063 /* the session state must be playing or ready */
1064 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1065 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1068 /* parse the range header if we have one */
1069 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1070 if (res == GST_RTSP_OK) {
1071 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1072 /* we have a range, seek to the position */
1074 gst_rtsp_media_seek (media, range);
1075 gst_rtsp_range_free (range);
1079 /* grab RTPInfo from the payloaders now */
1080 rtpinfo = g_string_new ("");
1082 base_url = make_base_url (client, uri, path);
1084 n_streams = gst_rtsp_media_n_streams (media);
1085 for (i = 0, infocount = 0; i < n_streams; i++) {
1086 GstRTSPStreamTransport *trans;
1087 GstRTSPStream *stream;
1088 const GstRTSPTransport *tr;
1091 /* get the transport, if there is no transport configured, skip this stream */
1092 trans = gst_rtsp_session_media_get_transport (sessmedia, i);
1093 if (trans == NULL) {
1094 GST_INFO ("stream %d is not configured", i);
1097 tr = gst_rtsp_stream_transport_get_transport (trans);
1099 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1100 /* for TCP, link the stream to the TCP connection of the client */
1101 link_transport (client, session, trans);
1104 stream = gst_rtsp_stream_transport_get_stream (trans);
1105 if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
1109 g_string_append (rtpinfo, ", ");
1111 control = gst_rtsp_stream_get_control (stream);
1112 g_string_append_printf (rtpinfo, "url=%s%s;seq=%u;rtptime=%u",
1113 base_url, control, seq, rtptime);
1118 GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
1124 /* construct the response now */
1125 code = GST_RTSP_STS_OK;
1126 gst_rtsp_message_init_response (ctx->response, code,
1127 gst_rtsp_status_as_text (code), ctx->request);
1129 /* add the RTP-Info header */
1130 if (infocount > 0) {
1131 str = g_string_free (rtpinfo, FALSE);
1132 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO, str);
1134 g_string_free (rtpinfo, TRUE);
1138 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1140 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1142 send_message (client, session, ctx->response, FALSE);
1144 /* start playing after sending the request */
1145 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1147 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1149 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1156 GST_ERROR ("client %p: no session", client);
1157 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1162 GST_ERROR ("client %p: no uri supplied", client);
1163 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1168 GST_ERROR ("client %p: media not found", client);
1169 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1174 GST_ERROR ("client %p: no aggregate path %s", client, path);
1175 send_generic_response (client,
1176 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1182 GST_ERROR ("client %p: not PLAYING or READY", client);
1183 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1191 do_keepalive (GstRTSPSession * session)
1193 GST_INFO ("keep session %p alive", session);
1194 gst_rtsp_session_touch (session);
1197 /* parse @transport and return a valid transport in @tr. only transports
1198 * from @supported are returned. Returns FALSE if no valid transport
1201 parse_transport (const char *transport, GstRTSPLowerTrans supported,
1202 GstRTSPTransport * tr)
1209 gst_rtsp_transport_init (tr);
1211 GST_DEBUG ("parsing transports %s", transport);
1213 transports = g_strsplit (transport, ",", 0);
1215 /* loop through the transports, try to parse */
1216 for (i = 0; transports[i]; i++) {
1217 res = gst_rtsp_transport_parse (transports[i], tr);
1218 if (res != GST_RTSP_OK) {
1219 /* no valid transport, search some more */
1220 GST_WARNING ("could not parse transport %s", transports[i]);
1224 /* we have a transport, see if it's RTP/AVP */
1225 if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) {
1226 GST_WARNING ("invalid transport %s", transports[i]);
1230 if (!(tr->lower_transport & supported)) {
1231 GST_WARNING ("unsupported transport %s", transports[i]);
1235 /* we have a valid transport */
1236 GST_INFO ("found valid transport %s", transports[i]);
1241 gst_rtsp_transport_init (tr);
1243 g_strfreev (transports);
1249 handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
1250 GstRTSPMessage * request)
1252 gchar *blocksize_str;
1253 gboolean ret = TRUE;
1255 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1256 &blocksize_str, 0) == GST_RTSP_OK) {
1260 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1261 if (end == blocksize_str) {
1262 GST_ERROR ("failed to parse blocksize");
1265 /* we don't want to change the mtu when this media
1266 * can be shared because it impacts other clients */
1267 if (gst_rtsp_media_is_shared (media))
1270 if (blocksize > G_MAXUINT)
1271 blocksize = G_MAXUINT;
1272 gst_rtsp_stream_set_mtu (stream, blocksize);
1279 default_configure_client_transport (GstRTSPClient * client,
1280 GstRTSPContext * ctx, GstRTSPTransport * ct)
1282 GstRTSPClientPrivate *priv = client->priv;
1284 /* we have a valid transport now, set the destination of the client. */
1285 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1286 gboolean use_client_settings;
1288 use_client_settings =
1289 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1291 if (ct->destination && use_client_settings) {
1292 GstRTSPAddress *addr;
1294 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1295 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1300 gst_rtsp_address_free (addr);
1302 GstRTSPAddress *addr;
1303 GSocketFamily family;
1305 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1307 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1311 g_free (ct->destination);
1312 ct->destination = g_strdup (addr->address);
1313 ct->port.min = addr->port;
1314 ct->port.max = addr->port + addr->n_ports - 1;
1315 ct->ttl = addr->ttl;
1317 gst_rtsp_address_free (addr);
1322 url = gst_rtsp_connection_get_url (priv->connection);
1323 g_free (ct->destination);
1324 ct->destination = g_strdup (url->host);
1326 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1327 /* check if the client selected channels for TCP */
1328 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1329 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1339 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1344 static GstRTSPTransport *
1345 make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
1346 GstRTSPTransport * ct)
1348 GstRTSPTransport *st;
1350 GSocketFamily family;
1352 /* prepare the server transport */
1353 gst_rtsp_transport_new (&st);
1355 st->trans = ct->trans;
1356 st->profile = ct->profile;
1357 st->lower_transport = ct->lower_transport;
1359 addr = g_inet_address_new_from_string (ct->destination);
1362 GST_ERROR ("failed to get inet addr from client destination");
1363 family = G_SOCKET_FAMILY_IPV4;
1365 family = g_inet_address_get_family (addr);
1366 g_object_unref (addr);
1370 switch (st->lower_transport) {
1371 case GST_RTSP_LOWER_TRANS_UDP:
1372 st->client_port = ct->client_port;
1373 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1375 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1376 st->port = ct->port;
1377 st->destination = g_strdup (ct->destination);
1380 case GST_RTSP_LOWER_TRANS_TCP:
1381 st->interleaved = ct->interleaved;
1386 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1392 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1394 GstRTSPClientPrivate *priv = client->priv;
1398 GstRTSPTransport *ct, *st;
1399 GstRTSPLowerTrans supported;
1400 GstRTSPStatusCode code;
1401 GstRTSPSession *session;
1402 GstRTSPStreamTransport *trans;
1404 GstRTSPSessionMedia *sessmedia;
1405 GstRTSPMedia *media;
1406 GstRTSPStream *stream;
1407 GstRTSPState rtspstate;
1408 GstRTSPClientClass *klass;
1409 gchar *path, *control;
1416 path = make_path_from_uri (client, uri);
1418 /* parse the transport */
1420 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1422 if (res != GST_RTSP_OK)
1425 /* we create the session after parsing stuff so that we don't make
1426 * a session for malformed requests */
1427 if (priv->session_pool == NULL)
1430 session = ctx->session;
1433 g_object_ref (session);
1434 /* get a handle to the configuration of the media in the session, this can
1435 * return NULL if this is a new url to manage in this session. */
1436 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1438 /* we need a new media configuration in this session */
1442 /* we have no session media, find one and manage it */
1443 if (sessmedia == NULL) {
1444 /* get a handle to the configuration of the media in the session */
1445 media = find_media (client, ctx, path, &matched);
1447 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1448 g_object_ref (media);
1450 goto media_not_found;
1452 /* no media, not found then */
1454 goto media_not_found_no_reply;
1456 if (path[matched] == '\0')
1457 goto control_not_found;
1459 /* path is what matched. We can modify the parsed uri in place */
1460 path[matched] = '\0';
1461 /* control is remainder */
1462 control = &path[matched + 1];
1464 /* find the stream now using the control part */
1465 stream = gst_rtsp_media_find_stream (media, control);
1467 goto stream_not_found;
1469 /* now we have a uri identifying a valid media and stream */
1470 ctx->stream = stream;
1473 if (session == NULL) {
1474 /* create a session if this fails we probably reached our session limit or
1476 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1477 goto service_unavailable;
1479 /* make sure this client is closed when the session is closed */
1480 client_watch_session (client, session);
1482 /* signal new session */
1483 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1486 ctx->session = session;
1489 if (sessmedia == NULL) {
1490 /* manage the media in our session now, if not done already */
1491 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1492 /* if we stil have no media, error */
1493 if (sessmedia == NULL)
1494 goto sessmedia_unavailable;
1496 g_object_unref (media);
1499 ctx->sessmedia = sessmedia;
1501 /* set blocksize on this stream */
1502 if (!handle_blocksize (media, stream, ctx->request))
1503 goto invalid_blocksize;
1505 gst_rtsp_transport_new (&ct);
1507 /* our supported transports */
1508 supported = gst_rtsp_stream_get_protocols (stream);
1510 /* parse and find a usable supported transport */
1511 if (!parse_transport (transport, supported, ct))
1512 goto unsupported_transports;
1514 /* update the client transport */
1515 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1516 if (!klass->configure_client_transport (client, ctx, ct))
1517 goto unsupported_client_transport;
1519 /* set in the session media transport */
1520 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1522 /* configure keepalive for this transport */
1523 gst_rtsp_stream_transport_set_keepalive (trans,
1524 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1526 /* create and serialize the server transport */
1527 st = make_server_transport (client, ctx, ct);
1528 trans_str = gst_rtsp_transport_as_text (st);
1529 gst_rtsp_transport_free (st);
1531 /* construct the response now */
1532 code = GST_RTSP_STS_OK;
1533 gst_rtsp_message_init_response (ctx->response, code,
1534 gst_rtsp_status_as_text (code), ctx->request);
1536 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1540 send_message (client, session, ctx->response, FALSE);
1542 /* update the state */
1543 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1544 switch (rtspstate) {
1545 case GST_RTSP_STATE_PLAYING:
1546 case GST_RTSP_STATE_RECORDING:
1547 case GST_RTSP_STATE_READY:
1548 /* no state change */
1551 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1554 g_object_unref (session);
1557 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1564 GST_ERROR ("client %p: no uri", client);
1565 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1570 GST_ERROR ("client %p: no transport", client);
1571 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1577 GST_ERROR ("client %p: no session pool configured", client);
1578 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1582 media_not_found_no_reply:
1584 GST_ERROR ("client %p: media '%s' not found", client, path);
1586 /* error reply is already sent */
1591 GST_ERROR ("client %p: media '%s' not found", client, path);
1592 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1598 GST_ERROR ("client %p: no control in path '%s'", client, path);
1599 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1600 g_object_unref (media);
1606 GST_ERROR ("client %p: stream '%s' not found", client, control);
1607 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1608 g_object_unref (media);
1612 service_unavailable:
1614 GST_ERROR ("client %p: can't create session", client);
1615 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1616 g_object_unref (media);
1620 sessmedia_unavailable:
1622 GST_ERROR ("client %p: can't create session media", client);
1623 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1624 g_object_unref (media);
1625 g_object_unref (session);
1631 GST_ERROR ("client %p: invalid blocksize", client);
1632 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1633 g_object_unref (session);
1637 unsupported_transports:
1639 GST_ERROR ("client %p: unsupported transports", client);
1640 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1641 gst_rtsp_transport_free (ct);
1642 g_object_unref (session);
1646 unsupported_client_transport:
1648 GST_ERROR ("client %p: unsupported client transport", client);
1649 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1650 gst_rtsp_transport_free (ct);
1651 g_object_unref (session);
1657 static GstSDPMessage *
1658 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1660 GstRTSPClientPrivate *priv = client->priv;
1665 gst_sdp_message_new (&sdp);
1667 /* some standard things first */
1668 gst_sdp_message_set_version (sdp, "0");
1675 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
1678 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
1679 gst_sdp_message_set_information (sdp, "rtsp-server");
1680 gst_sdp_message_add_time (sdp, "0", "0", NULL);
1681 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
1682 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
1683 gst_sdp_message_add_attribute (sdp, "control", "*");
1685 info.is_ipv6 = priv->is_ipv6;
1686 info.server_ip = priv->server_ip;
1688 /* create an SDP for the media object */
1689 if (!gst_rtsp_sdp_from_media (sdp, &info, media))
1697 GST_ERROR ("client %p: could not create SDP", client);
1698 gst_sdp_message_free (sdp);
1703 /* for the describe we must generate an SDP */
1705 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
1707 GstRTSPClientPrivate *priv = client->priv;
1712 GstRTSPMedia *media;
1713 GstRTSPClientClass *klass;
1715 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1720 /* check what kind of format is accepted, we don't really do anything with it
1721 * and always return SDP for now. */
1726 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
1728 if (res == GST_RTSP_ENOTIMPL)
1731 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
1735 if (!priv->mount_points)
1736 goto no_mount_points;
1738 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
1741 /* find the media object for the uri */
1742 if (!(media = find_media (client, ctx, path, NULL)))
1745 /* create an SDP for the media object on this client */
1746 if (!(sdp = klass->create_sdp (client, media)))
1749 g_object_unref (media);
1751 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
1752 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
1754 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
1757 /* content base for some clients that might screw up creating the setup uri */
1758 str = make_base_url (client, ctx->uri, path);
1761 GST_INFO ("adding content-base: %s", str);
1762 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
1764 /* add SDP to the response body */
1765 str = gst_sdp_message_as_text (sdp);
1766 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
1767 gst_sdp_message_free (sdp);
1769 send_message (client, ctx->session, ctx->response, FALSE);
1771 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
1779 GST_ERROR ("client %p: no uri", client);
1780 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1785 GST_ERROR ("client %p: no mount points configured", client);
1786 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1791 GST_ERROR ("client %p: can't find path for url", client);
1792 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1797 GST_ERROR ("client %p: no media", client);
1799 /* error reply is already sent */
1804 GST_ERROR ("client %p: can't create SDP", client);
1805 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1807 g_object_unref (media);
1813 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
1815 GstRTSPMethod options;
1818 options = GST_RTSP_DESCRIBE |
1823 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
1825 str = gst_rtsp_options_as_text (options);
1827 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
1828 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
1830 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
1833 send_message (client, ctx->session, ctx->response, FALSE);
1835 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
1841 /* remove duplicate and trailing '/' */
1843 sanitize_uri (GstRTSPUrl * uri)
1847 gboolean have_slash, prev_slash;
1849 s = d = uri->abspath;
1850 len = strlen (uri->abspath);
1854 for (i = 0; i < len; i++) {
1855 have_slash = s[i] == '/';
1857 if (!have_slash || !prev_slash)
1859 prev_slash = have_slash;
1861 len = d - uri->abspath;
1862 /* don't remove the first slash if that's the only thing left */
1863 if (len > 1 && *(d - 1) == '/')
1869 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
1871 GstRTSPClientPrivate *priv = client->priv;
1873 GST_INFO ("client %p: session %p finished", client, session);
1875 /* unlink all media managed in this session */
1876 client_unlink_session (client, session);
1878 /* remove the session */
1879 if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
1880 GST_INFO ("client %p: all sessions finalized, close the connection",
1882 close_connection (client);
1887 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
1889 GstRTSPClientPrivate *priv = client->priv;
1890 GstRTSPMethod method;
1891 const gchar *uristr;
1892 GstRTSPUrl *uri = NULL;
1893 GstRTSPVersion version;
1895 GstRTSPSession *session = NULL;
1896 GstRTSPContext sctx = { NULL }, *ctx;
1897 GstRTSPMessage response = { 0 };
1900 if (!(ctx = gst_rtsp_context_get_current ())) {
1902 ctx->auth = priv->auth;
1903 gst_rtsp_context_push_current (ctx);
1906 ctx->conn = priv->connection;
1907 ctx->client = client;
1908 ctx->request = request;
1909 ctx->response = &response;
1911 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
1912 gst_rtsp_message_dump (request);
1915 GST_INFO ("client %p: received a request", client);
1917 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
1919 /* we can only handle 1.0 requests */
1920 if (version != GST_RTSP_VERSION_1_0)
1923 ctx->method = method;
1925 /* we always try to parse the url first */
1926 if (strcmp (uristr, "*") == 0) {
1927 /* special case where we have * as uri, keep uri = NULL */
1928 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK)
1931 /* get the session if there is any */
1932 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
1933 if (res == GST_RTSP_OK) {
1934 if (priv->session_pool == NULL)
1937 /* we had a session in the request, find it again */
1938 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
1939 goto session_not_found;
1941 /* we add the session to the client list of watched sessions. When a session
1942 * disappears because it times out, we will be notified. If all sessions are
1943 * gone, we will close the connection */
1944 client_watch_session (client, session);
1947 /* sanitize the uri */
1951 ctx->session = session;
1953 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
1954 goto not_authorized;
1956 /* now see what is asked and dispatch to a dedicated handler */
1958 case GST_RTSP_OPTIONS:
1959 handle_options_request (client, ctx);
1961 case GST_RTSP_DESCRIBE:
1962 handle_describe_request (client, ctx);
1964 case GST_RTSP_SETUP:
1965 handle_setup_request (client, ctx);
1968 handle_play_request (client, ctx);
1970 case GST_RTSP_PAUSE:
1971 handle_pause_request (client, ctx);
1973 case GST_RTSP_TEARDOWN:
1974 handle_teardown_request (client, ctx);
1976 case GST_RTSP_SET_PARAMETER:
1977 handle_set_param_request (client, ctx);
1979 case GST_RTSP_GET_PARAMETER:
1980 handle_get_param_request (client, ctx);
1982 case GST_RTSP_ANNOUNCE:
1983 case GST_RTSP_RECORD:
1984 case GST_RTSP_REDIRECT:
1985 goto not_implemented;
1986 case GST_RTSP_INVALID:
1993 gst_rtsp_context_pop_current (ctx);
1995 g_object_unref (session);
1997 gst_rtsp_url_free (uri);
2003 GST_ERROR ("client %p: version %d not supported", client, version);
2004 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2010 GST_ERROR ("client %p: bad request", client);
2011 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2016 GST_ERROR ("client %p: no pool configured", client);
2017 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2022 GST_ERROR ("client %p: session not found", client);
2023 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2028 GST_ERROR ("client %p: not allowed", client);
2029 /* error reply is already sent */
2034 GST_ERROR ("client %p: method %d not implemented", client, method);
2035 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2042 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2044 GstRTSPClientPrivate *priv = client->priv;
2046 GstRTSPSession *session = NULL;
2047 GstRTSPContext sctx = { NULL }, *ctx;
2050 if (!(ctx = gst_rtsp_context_get_current ())) {
2052 ctx->auth = priv->auth;
2053 gst_rtsp_context_push_current (ctx);
2056 ctx->conn = priv->connection;
2057 ctx->client = client;
2058 ctx->request = NULL;
2060 ctx->method = GST_RTSP_INVALID;
2061 ctx->response = response;
2063 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2064 gst_rtsp_message_dump (response);
2067 GST_INFO ("client %p: received a response", client);
2069 /* get the session if there is any */
2071 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2072 if (res == GST_RTSP_OK) {
2073 if (priv->session_pool == NULL)
2076 /* we had a session in the request, find it again */
2077 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2078 goto session_not_found;
2080 /* we add the session to the client list of watched sessions. When a session
2081 * disappears because it times out, we will be notified. If all sessions are
2082 * gone, we will close the connection */
2083 client_watch_session (client, session);
2086 ctx->session = session;
2088 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2093 gst_rtsp_context_pop_current (ctx);
2095 g_object_unref (session);
2100 GST_ERROR ("client %p: no pool configured", client);
2105 GST_ERROR ("client %p: session not found", client);
2111 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2113 GstRTSPClientPrivate *priv = client->priv;
2122 /* find the stream for this message */
2123 res = gst_rtsp_message_parse_data (message, &channel);
2124 if (res != GST_RTSP_OK)
2127 gst_rtsp_message_steal_body (message, &data, &size);
2129 buffer = gst_buffer_new_wrapped (data, size);
2132 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
2133 GstRTSPStreamTransport *trans;
2134 GstRTSPStream *stream;
2135 const GstRTSPTransport *tr;
2139 tr = gst_rtsp_stream_transport_get_transport (trans);
2140 stream = gst_rtsp_stream_transport_get_stream (trans);
2142 /* check for TCP transport */
2143 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
2144 /* dispatch to the stream based on the channel number */
2145 if (tr->interleaved.min == channel) {
2146 gst_rtsp_stream_recv_rtp (stream, buffer);
2149 } else if (tr->interleaved.max == channel) {
2150 gst_rtsp_stream_recv_rtcp (stream, buffer);
2157 gst_buffer_unref (buffer);
2161 * gst_rtsp_client_set_session_pool:
2162 * @client: a #GstRTSPClient
2163 * @pool: a #GstRTSPSessionPool
2165 * Set @pool as the sessionpool for @client which it will use to find
2166 * or allocate sessions. the sessionpool is usually inherited from the server
2167 * that created the client but can be overridden later.
2170 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
2171 GstRTSPSessionPool * pool)
2173 GstRTSPSessionPool *old;
2174 GstRTSPClientPrivate *priv;
2176 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2178 priv = client->priv;
2181 g_object_ref (pool);
2183 g_mutex_lock (&priv->lock);
2184 old = priv->session_pool;
2185 priv->session_pool = pool;
2186 g_mutex_unlock (&priv->lock);
2189 g_object_unref (old);
2193 * gst_rtsp_client_get_session_pool:
2194 * @client: a #GstRTSPClient
2196 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
2198 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
2200 GstRTSPSessionPool *
2201 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2203 GstRTSPClientPrivate *priv;
2204 GstRTSPSessionPool *result;
2206 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2208 priv = client->priv;
2210 g_mutex_lock (&priv->lock);
2211 if ((result = priv->session_pool))
2212 g_object_ref (result);
2213 g_mutex_unlock (&priv->lock);
2219 * gst_rtsp_client_set_mount_points:
2220 * @client: a #GstRTSPClient
2221 * @mounts: a #GstRTSPMountPoints
2223 * Set @mounts as the mount points for @client which it will use to map urls
2224 * to media streams. These mount points are usually inherited from the server that
2225 * created the client but can be overriden later.
2228 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
2229 GstRTSPMountPoints * mounts)
2231 GstRTSPClientPrivate *priv;
2232 GstRTSPMountPoints *old;
2234 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2236 priv = client->priv;
2239 g_object_ref (mounts);
2241 g_mutex_lock (&priv->lock);
2242 old = priv->mount_points;
2243 priv->mount_points = mounts;
2244 g_mutex_unlock (&priv->lock);
2247 g_object_unref (old);
2251 * gst_rtsp_client_get_mount_points:
2252 * @client: a #GstRTSPClient
2254 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
2256 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
2258 GstRTSPMountPoints *
2259 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
2261 GstRTSPClientPrivate *priv;
2262 GstRTSPMountPoints *result;
2264 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2266 priv = client->priv;
2268 g_mutex_lock (&priv->lock);
2269 if ((result = priv->mount_points))
2270 g_object_ref (result);
2271 g_mutex_unlock (&priv->lock);
2277 * gst_rtsp_client_set_auth:
2278 * @client: a #GstRTSPClient
2279 * @auth: a #GstRTSPAuth
2281 * configure @auth to be used as the authentication manager of @client.
2284 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
2286 GstRTSPClientPrivate *priv;
2289 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2291 priv = client->priv;
2294 g_object_ref (auth);
2296 g_mutex_lock (&priv->lock);
2299 g_mutex_unlock (&priv->lock);
2302 g_object_unref (old);
2307 * gst_rtsp_client_get_auth:
2308 * @client: a #GstRTSPClient
2310 * Get the #GstRTSPAuth used as the authentication manager of @client.
2312 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2316 gst_rtsp_client_get_auth (GstRTSPClient * client)
2318 GstRTSPClientPrivate *priv;
2319 GstRTSPAuth *result;
2321 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2323 priv = client->priv;
2325 g_mutex_lock (&priv->lock);
2326 if ((result = priv->auth))
2327 g_object_ref (result);
2328 g_mutex_unlock (&priv->lock);
2334 * gst_rtsp_client_set_thread_pool:
2335 * @client: a #GstRTSPClient
2336 * @pool: a #GstRTSPThreadPool
2338 * configure @pool to be used as the thread pool of @client.
2341 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
2342 GstRTSPThreadPool * pool)
2344 GstRTSPClientPrivate *priv;
2345 GstRTSPThreadPool *old;
2347 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2349 priv = client->priv;
2352 g_object_ref (pool);
2354 g_mutex_lock (&priv->lock);
2355 old = priv->thread_pool;
2356 priv->thread_pool = pool;
2357 g_mutex_unlock (&priv->lock);
2360 g_object_unref (old);
2364 * gst_rtsp_client_get_thread_pool:
2365 * @client: a #GstRTSPClient
2367 * Get the #GstRTSPThreadPool used as the thread pool of @client.
2369 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
2373 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
2375 GstRTSPClientPrivate *priv;
2376 GstRTSPThreadPool *result;
2378 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2380 priv = client->priv;
2382 g_mutex_lock (&priv->lock);
2383 if ((result = priv->thread_pool))
2384 g_object_ref (result);
2385 g_mutex_unlock (&priv->lock);
2391 * gst_rtsp_client_set_connection:
2392 * @client: a #GstRTSPClient
2393 * @conn: (transfer full): a #GstRTSPConnection
2395 * Set the #GstRTSPConnection of @client. This function takes ownership of
2398 * Returns: %TRUE on success.
2401 gst_rtsp_client_set_connection (GstRTSPClient * client,
2402 GstRTSPConnection * conn)
2404 GstRTSPClientPrivate *priv;
2405 GSocket *read_socket;
2406 GSocketAddress *address;
2408 GError *error = NULL;
2410 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2411 g_return_val_if_fail (conn != NULL, FALSE);
2413 priv = client->priv;
2415 read_socket = gst_rtsp_connection_get_read_socket (conn);
2417 if (!(address = g_socket_get_local_address (read_socket, &error)))
2420 g_free (priv->server_ip);
2421 /* keep the original ip that the client connected to */
2422 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2423 GInetAddress *iaddr;
2425 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2427 /* socket might be ipv6 but adress still ipv4 */
2428 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
2429 priv->server_ip = g_inet_address_to_string (iaddr);
2430 g_object_unref (address);
2432 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
2433 priv->server_ip = g_strdup ("unknown");
2436 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2437 priv->server_ip, priv->is_ipv6);
2439 url = gst_rtsp_connection_get_url (conn);
2440 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2442 priv->connection = conn;
2449 GST_ERROR ("could not get local address %s", error->message);
2450 g_error_free (error);
2456 * gst_rtsp_client_get_connection:
2457 * @client: a #GstRTSPClient
2459 * Get the #GstRTSPConnection of @client.
2461 * Returns: (transfer none): the #GstRTSPConnection of @client.
2462 * The connection object returned remains valid until the client is freed.
2465 gst_rtsp_client_get_connection (GstRTSPClient * client)
2467 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2469 return client->priv->connection;
2473 * gst_rtsp_client_set_send_func:
2474 * @client: a #GstRTSPClient
2475 * @func: a #GstRTSPClientSendFunc
2476 * @user_data: user data passed to @func
2477 * @notify: called when @user_data is no longer in use
2479 * Set @func as the callback that will be called when a new message needs to be
2480 * sent to the client. @user_data is passed to @func and @notify is called when
2481 * @user_data is no longer in use.
2483 * By default, the client will send the messages on the #GstRTSPConnection that
2484 * was configured with gst_rtsp_client_attach() was called.
2487 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2488 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2490 GstRTSPClientPrivate *priv;
2491 GDestroyNotify old_notify;
2494 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2496 priv = client->priv;
2498 g_mutex_lock (&priv->send_lock);
2499 priv->send_func = func;
2500 old_notify = priv->send_notify;
2501 old_data = priv->send_data;
2502 priv->send_notify = notify;
2503 priv->send_data = user_data;
2504 g_mutex_unlock (&priv->send_lock);
2507 old_notify (old_data);
2511 * gst_rtsp_client_handle_message:
2512 * @client: a #GstRTSPClient
2513 * @message: an #GstRTSPMessage
2515 * Let the client handle @message.
2517 * Returns: a #GstRTSPResult.
2520 gst_rtsp_client_handle_message (GstRTSPClient * client,
2521 GstRTSPMessage * message)
2523 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2524 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2526 switch (message->type) {
2527 case GST_RTSP_MESSAGE_REQUEST:
2528 handle_request (client, message);
2530 case GST_RTSP_MESSAGE_RESPONSE:
2531 handle_response (client, message);
2533 case GST_RTSP_MESSAGE_DATA:
2534 handle_data (client, message);
2543 * gst_rtsp_client_send_message:
2544 * @client: a #GstRTSPClient
2545 * @session: a #GstRTSPSession to send the message to or %NULL
2546 * @message: The #GstRTSPMessage to send
2548 * Send a message message to the remote end. @message must be a
2549 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
2552 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
2553 GstRTSPMessage * message)
2555 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2556 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2557 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
2558 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
2560 send_message (client, session, message, FALSE);
2565 static GstRTSPResult
2566 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
2567 gboolean close, gpointer user_data)
2569 GstRTSPClientPrivate *priv = client->priv;
2571 /* send the response and store the seq number so we can wait until it's
2572 * written to the client to close the connection */
2573 return gst_rtsp_watch_send_message (priv->watch, message, close ?
2574 &priv->close_seq : NULL);
2577 static GstRTSPResult
2578 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
2581 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
2584 static GstRTSPResult
2585 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
2587 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2588 GstRTSPClientPrivate *priv = client->priv;
2590 if (priv->close_seq && priv->close_seq == cseq) {
2591 priv->close_seq = 0;
2592 close_connection (client);
2598 static GstRTSPResult
2599 closed (GstRTSPWatch * watch, gpointer user_data)
2601 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2602 GstRTSPClientPrivate *priv = client->priv;
2603 const gchar *tunnelid;
2605 GST_INFO ("client %p: connection closed", client);
2607 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
2608 g_mutex_lock (&tunnels_lock);
2609 /* remove from tunnelids */
2610 g_hash_table_remove (tunnels, tunnelid);
2611 g_mutex_unlock (&tunnels_lock);
2614 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
2619 static GstRTSPResult
2620 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
2622 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2625 str = gst_rtsp_strresult (result);
2626 GST_INFO ("client %p: received an error %s", client, str);
2632 static GstRTSPResult
2633 error_full (GstRTSPWatch * watch, GstRTSPResult result,
2634 GstRTSPMessage * message, guint id, gpointer user_data)
2636 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2639 str = gst_rtsp_strresult (result);
2641 ("client %p: error when handling message %p with id %d: %s",
2642 client, message, id, str);
2649 remember_tunnel (GstRTSPClient * client)
2651 GstRTSPClientPrivate *priv = client->priv;
2652 const gchar *tunnelid;
2654 /* store client in the pending tunnels */
2655 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
2656 if (tunnelid == NULL)
2659 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
2661 /* we can't have two clients connecting with the same tunnelid */
2662 g_mutex_lock (&tunnels_lock);
2663 if (g_hash_table_lookup (tunnels, tunnelid))
2664 goto tunnel_existed;
2666 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
2667 g_mutex_unlock (&tunnels_lock);
2674 GST_ERROR ("client %p: no tunnelid provided", client);
2679 g_mutex_unlock (&tunnels_lock);
2680 GST_ERROR ("client %p: tunnel session %s already existed", client,
2686 static GstRTSPStatusCode
2687 tunnel_start (GstRTSPWatch * watch, gpointer user_data)
2689 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2690 GstRTSPClientPrivate *priv = client->priv;
2692 GST_INFO ("client %p: tunnel start (connection %p)", client,
2695 if (!remember_tunnel (client))
2698 return GST_RTSP_STS_OK;
2703 GST_ERROR ("client %p: error starting tunnel", client);
2704 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
2708 static GstRTSPResult
2709 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
2711 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2712 GstRTSPClientPrivate *priv = client->priv;
2714 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
2717 /* ignore error, it'll only be a problem when the client does a POST again */
2718 remember_tunnel (client);
2723 static GstRTSPResult
2724 tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
2726 const gchar *tunnelid;
2727 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2728 GstRTSPClientPrivate *priv = client->priv;
2729 GstRTSPClient *oclient;
2730 GstRTSPClientPrivate *opriv;
2732 GST_INFO ("client %p: tunnel complete", client);
2734 /* find previous tunnel */
2735 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
2736 if (tunnelid == NULL)
2739 g_mutex_lock (&tunnels_lock);
2740 if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
2743 /* remove the old client from the table. ref before because removing it will
2744 * remove the ref to it. */
2745 g_object_ref (oclient);
2746 g_hash_table_remove (tunnels, tunnelid);
2748 opriv = oclient->priv;
2750 if (opriv->watch == NULL)
2752 g_mutex_unlock (&tunnels_lock);
2754 GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
2755 opriv->connection, priv->connection);
2757 /* merge the tunnels into the first client */
2758 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
2759 gst_rtsp_watch_reset (opriv->watch);
2760 g_object_unref (oclient);
2767 GST_ERROR ("client %p: no tunnelid provided", client);
2768 return GST_RTSP_ERROR;
2772 g_mutex_unlock (&tunnels_lock);
2773 GST_ERROR ("client %p: tunnel session %s not found", client, tunnelid);
2774 return GST_RTSP_ERROR;
2778 g_mutex_unlock (&tunnels_lock);
2779 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
2780 g_object_unref (oclient);
2781 return GST_RTSP_ERROR;
2785 static GstRTSPWatchFuncs watch_funcs = {
2797 client_watch_notify (GstRTSPClient * client)
2799 GstRTSPClientPrivate *priv = client->priv;
2801 GST_INFO ("client %p: watch destroyed", client);
2803 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
2804 g_object_unref (client);
2808 * gst_rtsp_client_attach:
2809 * @client: a #GstRTSPClient
2810 * @context: (allow-none): a #GMainContext
2812 * Attaches @client to @context. When the mainloop for @context is run, the
2813 * client will be dispatched. When @context is NULL, the default context will be
2816 * This function should be called when the client properties and urls are fully
2817 * configured and the client is ready to start.
2819 * Returns: the ID (greater than 0) for the source within the GMainContext.
2822 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
2824 GstRTSPClientPrivate *priv;
2827 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
2828 priv = client->priv;
2829 g_return_val_if_fail (priv->connection != NULL, 0);
2830 g_return_val_if_fail (priv->watch == NULL, 0);
2832 /* create watch for the connection and attach */
2833 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
2834 g_object_ref (client), (GDestroyNotify) client_watch_notify);
2835 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
2836 (GDestroyNotify) gst_rtsp_watch_unref);
2838 /* FIXME make this configurable. We don't want to do this yet because it will
2839 * be superceeded by a cache object later */
2840 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
2842 GST_INFO ("attaching to context %p", context);
2843 res = gst_rtsp_watch_attach (priv->watch, context);
2849 * gst_rtsp_client_session_filter:
2850 * @client: a #GstRTSPClient
2851 * @func: (scope call): a callback
2852 * @user_data: user data passed to @func
2854 * Call @func for each session managed by @client. The result value of @func
2855 * determines what happens to the session. @func will be called with @client
2856 * locked so no further actions on @client can be performed from @func.
2858 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
2861 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
2863 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
2864 * will also be added with an additional ref to the result #GList of this
2867 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
2868 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
2869 * element in the #GList should be unreffed before the list is freed.
2872 gst_rtsp_client_session_filter (GstRTSPClient * client,
2873 GstRTSPClientSessionFilterFunc func, gpointer user_data)
2875 GstRTSPClientPrivate *priv;
2876 GList *result, *walk, *next;
2878 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2879 g_return_val_if_fail (func != NULL, NULL);
2881 priv = client->priv;
2885 g_mutex_lock (&priv->lock);
2886 for (walk = priv->sessions; walk; walk = next) {
2887 GstRTSPSession *sess = walk->data;
2889 next = g_list_next (walk);
2891 switch (func (client, sess, user_data)) {
2892 case GST_RTSP_FILTER_REMOVE:
2893 /* stop watching the session and pretent it went away */
2894 client_cleanup_session (client, sess);
2896 case GST_RTSP_FILTER_REF:
2897 result = g_list_prepend (result, g_object_ref (sess));
2899 case GST_RTSP_FILTER_KEEP:
2904 g_mutex_unlock (&priv->lock);