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 <gst/sdp/gstmikey.h>
47 #include "rtsp-client.h"
49 #include "rtsp-params.h"
51 #define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
52 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
55 * send_lock, lock, tunnels_lock
58 struct _GstRTSPClientPrivate
60 GMutex lock; /* protects everything else */
63 GstRTSPConnection *connection;
65 GMainContext *watch_context;
70 GstRTSPClientSendFunc send_func; /* protected by send_lock */
71 gpointer send_data; /* protected by send_lock */
72 GDestroyNotify send_notify; /* protected by send_lock */
74 GstRTSPSessionPool *session_pool;
75 gulong session_removed_id;
76 GstRTSPMountPoints *mount_points;
78 GstRTSPThreadPool *thread_pool;
80 /* used to cache the media in the last requested DESCRIBE so that
81 * we can pick it up in the next SETUP immediately */
85 GHashTable *transports;
87 guint sessions_cookie;
89 gboolean drop_backlog;
92 static GMutex tunnels_lock;
93 static GHashTable *tunnels; /* protected by tunnels_lock */
95 /* FIXME make this configurable. We don't want to do this yet because it will
96 * be superceeded by a cache object later */
97 #define WATCH_BACKLOG_SIZE 100
99 #define DEFAULT_SESSION_POOL NULL
100 #define DEFAULT_MOUNT_POINTS NULL
101 #define DEFAULT_DROP_BACKLOG TRUE
116 SIGNAL_OPTIONS_REQUEST,
117 SIGNAL_DESCRIBE_REQUEST,
118 SIGNAL_SETUP_REQUEST,
120 SIGNAL_PAUSE_REQUEST,
121 SIGNAL_TEARDOWN_REQUEST,
122 SIGNAL_SET_PARAMETER_REQUEST,
123 SIGNAL_GET_PARAMETER_REQUEST,
124 SIGNAL_HANDLE_RESPONSE,
129 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
130 #define GST_CAT_DEFAULT rtsp_client_debug
132 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
134 static void gst_rtsp_client_get_property (GObject * object, guint propid,
135 GValue * value, GParamSpec * pspec);
136 static void gst_rtsp_client_set_property (GObject * object, guint propid,
137 const GValue * value, GParamSpec * pspec);
138 static void gst_rtsp_client_finalize (GObject * obj);
140 static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
141 static gboolean default_configure_client_media (GstRTSPClient * client,
142 GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
143 static gboolean default_configure_client_transport (GstRTSPClient * client,
144 GstRTSPContext * ctx, GstRTSPTransport * ct);
145 static GstRTSPResult default_params_set (GstRTSPClient * client,
146 GstRTSPContext * ctx);
147 static GstRTSPResult default_params_get (GstRTSPClient * client,
148 GstRTSPContext * ctx);
149 static gchar *default_make_path_from_uri (GstRTSPClient * client,
150 const GstRTSPUrl * uri);
151 static void client_session_removed (GstRTSPSessionPool * pool,
152 GstRTSPSession * session, GstRTSPClient * client);
154 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
157 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
159 GObjectClass *gobject_class;
161 g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
163 gobject_class = G_OBJECT_CLASS (klass);
165 gobject_class->get_property = gst_rtsp_client_get_property;
166 gobject_class->set_property = gst_rtsp_client_set_property;
167 gobject_class->finalize = gst_rtsp_client_finalize;
169 klass->create_sdp = create_sdp;
170 klass->configure_client_media = default_configure_client_media;
171 klass->configure_client_transport = default_configure_client_transport;
172 klass->params_set = default_params_set;
173 klass->params_get = default_params_get;
174 klass->make_path_from_uri = default_make_path_from_uri;
176 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
177 g_param_spec_object ("session-pool", "Session Pool",
178 "The session pool to use for client session",
179 GST_TYPE_RTSP_SESSION_POOL,
180 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
182 g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
183 g_param_spec_object ("mount-points", "Mount Points",
184 "The mount points to use for client session",
185 GST_TYPE_RTSP_MOUNT_POINTS,
186 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
188 g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
189 g_param_spec_boolean ("drop-backlog", "Drop Backlog",
190 "Drop data when the backlog queue is full",
191 DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
193 gst_rtsp_client_signals[SIGNAL_CLOSED] =
194 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
195 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
196 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
198 gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
199 g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
200 G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
201 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
203 gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
204 g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
205 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
206 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
207 GST_TYPE_RTSP_CONTEXT);
209 gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
210 g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
212 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
213 GST_TYPE_RTSP_CONTEXT);
215 gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
216 g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
217 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
218 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
219 GST_TYPE_RTSP_CONTEXT);
221 gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
222 g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
224 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
225 GST_TYPE_RTSP_CONTEXT);
227 gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
228 g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
229 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
230 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
231 GST_TYPE_RTSP_CONTEXT);
233 gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
234 g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
235 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
236 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
237 GST_TYPE_RTSP_CONTEXT);
239 gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
240 g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
241 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
242 set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
243 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
245 gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
246 g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
248 get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
249 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
251 gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
252 g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
253 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
254 handle_response), NULL, NULL, g_cclosure_marshal_generic,
255 G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
258 * GstRTSPClient::send-message:
259 * @client: The RTSP client
260 * @session: (type GstRtspServer.RTSPSession): The session
261 * @message: (type GstRtsp.RTSPMessage): The message
263 gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
264 g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
265 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
266 G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
269 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
270 g_mutex_init (&tunnels_lock);
272 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
276 gst_rtsp_client_init (GstRTSPClient * client)
278 GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
282 g_mutex_init (&priv->lock);
283 g_mutex_init (&priv->send_lock);
284 g_mutex_init (&priv->watch_lock);
286 priv->drop_backlog = DEFAULT_DROP_BACKLOG;
287 priv->transports = g_hash_table_new (g_direct_hash, g_direct_equal);
290 static GstRTSPFilterResult
291 filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
294 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
296 return GST_RTSP_FILTER_REMOVE;
300 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
302 GstRTSPClientPrivate *priv = client->priv;
304 g_mutex_lock (&priv->lock);
305 /* check if we already know about this session */
306 if (g_list_find (priv->sessions, session) == NULL) {
307 GST_INFO ("watching session %p", session);
309 priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
310 priv->sessions_cookie++;
312 /* connect removed session handler, it will be disconnected when the last
313 * session gets removed */
314 if (priv->session_removed_id == 0)
315 priv->session_removed_id = g_signal_connect_data (priv->session_pool,
316 "session-removed", G_CALLBACK (client_session_removed),
317 g_object_ref (client), (GClosureNotify) g_object_unref, 0);
319 g_mutex_unlock (&priv->lock);
324 /* should be called with lock */
326 client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
329 GstRTSPClientPrivate *priv = client->priv;
331 GST_INFO ("client %p: unwatch session %p", client, session);
334 link = g_list_find (priv->sessions, session);
339 priv->sessions = g_list_delete_link (priv->sessions, link);
340 priv->sessions_cookie++;
342 /* if this was the last session, disconnect the handler.
343 * This will also drop the extra client ref */
344 if (!priv->sessions) {
345 g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
346 priv->session_removed_id = 0;
349 /* unlink all media managed in this session */
350 gst_rtsp_session_filter (session, filter_session_media, client);
352 /* remove the session */
353 g_object_unref (session);
356 static GstRTSPFilterResult
357 cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
360 return GST_RTSP_FILTER_REMOVE;
363 /* A client is finalized when the connection is broken */
365 gst_rtsp_client_finalize (GObject * obj)
367 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
368 GstRTSPClientPrivate *priv = client->priv;
370 GST_INFO ("finalize client %p", client);
373 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
374 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
377 g_source_destroy ((GSource *) priv->watch);
379 if (priv->watch_context)
380 g_main_context_unref (priv->watch_context);
382 /* all sessions should have been removed by now. We keep a ref to
383 * the client object for the session removed handler. The ref is
384 * dropped when the last session is removed from the list. */
385 g_assert (priv->sessions == NULL);
386 g_assert (priv->session_removed_id == 0);
388 g_hash_table_unref (priv->transports);
390 if (priv->connection)
391 gst_rtsp_connection_free (priv->connection);
392 if (priv->session_pool) {
393 g_object_unref (priv->session_pool);
395 if (priv->mount_points)
396 g_object_unref (priv->mount_points);
398 g_object_unref (priv->auth);
399 if (priv->thread_pool)
400 g_object_unref (priv->thread_pool);
405 gst_rtsp_media_unprepare (priv->media);
406 g_object_unref (priv->media);
409 g_free (priv->server_ip);
410 g_mutex_clear (&priv->lock);
411 g_mutex_clear (&priv->send_lock);
412 g_mutex_clear (&priv->watch_lock);
414 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
418 gst_rtsp_client_get_property (GObject * object, guint propid,
419 GValue * value, GParamSpec * pspec)
421 GstRTSPClient *client = GST_RTSP_CLIENT (object);
422 GstRTSPClientPrivate *priv = client->priv;
425 case PROP_SESSION_POOL:
426 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
428 case PROP_MOUNT_POINTS:
429 g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
431 case PROP_DROP_BACKLOG:
432 g_value_set_boolean (value, priv->drop_backlog);
435 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
440 gst_rtsp_client_set_property (GObject * object, guint propid,
441 const GValue * value, GParamSpec * pspec)
443 GstRTSPClient *client = GST_RTSP_CLIENT (object);
444 GstRTSPClientPrivate *priv = client->priv;
447 case PROP_SESSION_POOL:
448 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
450 case PROP_MOUNT_POINTS:
451 gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
453 case PROP_DROP_BACKLOG:
454 g_mutex_lock (&priv->lock);
455 priv->drop_backlog = g_value_get_boolean (value);
456 g_mutex_unlock (&priv->lock);
459 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
464 * gst_rtsp_client_new:
466 * Create a new #GstRTSPClient instance.
468 * Returns: (transfer full): a new #GstRTSPClient
471 gst_rtsp_client_new (void)
473 GstRTSPClient *result;
475 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
481 send_message (GstRTSPClient * client, GstRTSPContext * ctx,
482 GstRTSPMessage * message, gboolean close)
484 GstRTSPClientPrivate *priv = client->priv;
486 gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
487 "GStreamer RTSP server");
489 /* remove any previous header */
490 gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
492 /* add the new session header for new session ids */
494 gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
495 gst_rtsp_session_get_header (ctx->session));
498 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
499 gst_rtsp_message_dump (message);
503 gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
505 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
508 g_mutex_lock (&priv->send_lock);
510 priv->send_func (client, message, close, priv->send_data);
511 g_mutex_unlock (&priv->send_lock);
513 gst_rtsp_message_unset (message);
517 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
518 GstRTSPContext * ctx)
520 gst_rtsp_message_init_response (ctx->response, code,
521 gst_rtsp_status_as_text (code), ctx->request);
525 send_message (client, ctx, ctx->response, FALSE);
529 send_option_not_supported_response (GstRTSPClient * client,
530 GstRTSPContext * ctx, const gchar * unsupported_options)
532 GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
534 gst_rtsp_message_init_response (ctx->response, code,
535 gst_rtsp_status_as_text (code), ctx->request);
537 if (unsupported_options != NULL) {
538 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
539 unsupported_options);
544 send_message (client, ctx, ctx->response, FALSE);
548 paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
550 if (path1 == NULL || path2 == NULL)
553 if (strlen (path1) != len2)
556 if (strncmp (path1, path2, len2))
562 /* this function is called to initially find the media for the DESCRIBE request
563 * but is cached for when the same client (without breaking the connection) is
564 * doing a setup for the exact same url. */
565 static GstRTSPMedia *
566 find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
569 GstRTSPClientPrivate *priv = client->priv;
570 GstRTSPMediaFactory *factory;
574 /* find the longest matching factory for the uri first */
575 if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
579 ctx->factory = factory;
581 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
582 goto no_factory_access;
584 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
590 path_len = strlen (path);
592 if (!paths_are_equal (priv->path, path, path_len)) {
593 GstRTSPThread *thread;
595 /* remove any previously cached values before we try to construct a new
601 gst_rtsp_media_unprepare (priv->media);
602 g_object_unref (priv->media);
606 /* prepare the media and add it to the pipeline */
607 if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
612 thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
613 GST_RTSP_THREAD_TYPE_MEDIA, ctx);
617 /* prepare the media */
618 if (!(gst_rtsp_media_prepare (media, thread)))
621 /* now keep track of the uri and the media */
622 priv->path = g_strndup (path, path_len);
625 /* we have seen this path before, used cached media */
628 GST_INFO ("reusing cached media %p for path %s", media, priv->path);
631 g_object_unref (factory);
635 g_object_ref (media);
642 GST_ERROR ("client %p: no factory for path %s", client, path);
643 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
648 GST_ERROR ("client %p: not authorized to see factory path %s", client,
650 /* error reply is already sent */
655 GST_ERROR ("client %p: not authorized for factory path %s", client, path);
656 /* error reply is already sent */
661 GST_ERROR ("client %p: can't create media", client);
662 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
663 g_object_unref (factory);
669 GST_ERROR ("client %p: can't create thread", client);
670 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
671 g_object_unref (media);
673 g_object_unref (factory);
679 GST_ERROR ("client %p: can't prepare media", client);
680 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
681 g_object_unref (media);
683 g_object_unref (factory);
690 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
692 GstRTSPClientPrivate *priv = client->priv;
693 GstRTSPMessage message = { 0 };
698 gst_rtsp_message_init_data (&message, channel);
700 /* FIXME, need some sort of iovec RTSPMessage here */
701 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
704 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
706 g_mutex_lock (&priv->send_lock);
708 priv->send_func (client, &message, FALSE, priv->send_data);
709 g_mutex_unlock (&priv->send_lock);
711 gst_rtsp_message_steal_body (&message, &data, &usize);
712 gst_buffer_unmap (buffer, &map_info);
714 gst_rtsp_message_unset (&message);
720 * gst_rtsp_client_close:
721 * @client: a #GstRTSPClient
723 * Close the connection of @client and remove all media it was managing.
728 gst_rtsp_client_close (GstRTSPClient * client)
730 GstRTSPClientPrivate *priv = client->priv;
731 const gchar *tunnelid;
733 GST_DEBUG ("client %p: closing connection", client);
735 if (priv->connection) {
736 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
737 g_mutex_lock (&tunnels_lock);
738 /* remove from tunnelids */
739 g_hash_table_remove (tunnels, tunnelid);
740 g_mutex_unlock (&tunnels_lock);
742 gst_rtsp_connection_close (priv->connection);
745 /* connection is now closed, destroy the watch which will also cause the
746 * closed signal to be emitted */
748 GST_DEBUG ("client %p: destroying watch", client);
749 g_source_destroy ((GSource *) priv->watch);
751 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
756 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
761 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
763 path = g_strdup (uri->abspath);
769 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
771 GstRTSPClientPrivate *priv = client->priv;
772 GstRTSPClientClass *klass;
773 GstRTSPSession *session;
774 GstRTSPSessionMedia *sessmedia;
775 GstRTSPStatusCode code;
778 gboolean keep_session;
783 session = ctx->session;
788 klass = GST_RTSP_CLIENT_GET_CLASS (client);
789 path = klass->make_path_from_uri (client, ctx->uri);
791 /* get a handle to the configuration of the media in the session */
792 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
796 /* only aggregate control for now.. */
797 if (path[matched] != '\0')
802 ctx->sessmedia = sessmedia;
804 /* we emit the signal before closing the connection */
805 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
808 /* make sure we unblock the backlog and don't accept new messages
810 if (priv->watch != NULL)
811 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
813 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
815 /* allow messages again so that we can send the reply */
816 if (priv->watch != NULL)
817 gst_rtsp_watch_set_flushing (priv->watch, FALSE);
819 /* unmanage the media in the session, returns false if all media session
821 keep_session = gst_rtsp_session_release_media (session, sessmedia);
823 /* construct the response now */
824 code = GST_RTSP_STS_OK;
825 gst_rtsp_message_init_response (ctx->response, code,
826 gst_rtsp_status_as_text (code), ctx->request);
828 send_message (client, ctx, ctx->response, TRUE);
831 /* remove the session */
832 gst_rtsp_session_pool_remove (priv->session_pool, session);
840 GST_ERROR ("client %p: no session", client);
841 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
846 GST_ERROR ("client %p: no uri supplied", client);
847 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
852 GST_ERROR ("client %p: no media for uri", client);
853 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
859 GST_ERROR ("client %p: no aggregate path %s", client, path);
860 send_generic_response (client,
861 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
868 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
872 res = gst_rtsp_params_set (client, ctx);
878 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
882 res = gst_rtsp_params_get (client, ctx);
888 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
894 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
895 if (res != GST_RTSP_OK)
899 /* no body, keep-alive request */
900 send_generic_response (client, GST_RTSP_STS_OK, ctx);
902 /* there is a body, handle the params */
903 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
904 if (res != GST_RTSP_OK)
907 send_message (client, ctx, ctx->response, FALSE);
910 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
918 GST_ERROR ("client %p: bad request", client);
919 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
925 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
931 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
932 if (res != GST_RTSP_OK)
936 /* no body, keep-alive request */
937 send_generic_response (client, GST_RTSP_STS_OK, ctx);
939 /* there is a body, handle the params */
940 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
941 if (res != GST_RTSP_OK)
944 send_message (client, ctx, ctx->response, FALSE);
947 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
955 GST_ERROR ("client %p: bad request", client);
956 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
962 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
964 GstRTSPClientPrivate *priv = client->priv;
965 GstRTSPSession *session;
966 GstRTSPClientClass *klass;
967 GstRTSPSessionMedia *sessmedia;
968 GstRTSPStatusCode code;
969 GstRTSPState rtspstate;
973 if (!(session = ctx->session))
979 klass = GST_RTSP_CLIENT_GET_CLASS (client);
980 path = klass->make_path_from_uri (client, ctx->uri);
982 /* get a handle to the configuration of the media in the session */
983 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
987 if (path[matched] != '\0')
992 ctx->sessmedia = sessmedia;
994 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
995 /* the session state must be playing or recording */
996 if (rtspstate != GST_RTSP_STATE_PLAYING &&
997 rtspstate != GST_RTSP_STATE_RECORDING)
1000 /* No limit on watch queue because else we might be blocking in the appsink
1001 * render method and the PAUSE below will hang */
1002 if (priv->watch != NULL)
1003 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
1005 /* then pause sending */
1006 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1008 /* construct the response now */
1009 code = GST_RTSP_STS_OK;
1010 gst_rtsp_message_init_response (ctx->response, code,
1011 gst_rtsp_status_as_text (code), ctx->request);
1013 send_message (client, ctx, ctx->response, FALSE);
1015 if (priv->watch != NULL)
1016 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
1018 /* the state is now READY */
1019 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1021 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1028 GST_ERROR ("client %p: no seesion", client);
1029 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1034 GST_ERROR ("client %p: no uri supplied", client);
1035 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1040 GST_ERROR ("client %p: no media for uri", client);
1041 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1047 GST_ERROR ("client %p: no aggregate path %s", client, path);
1048 send_generic_response (client,
1049 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1055 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1056 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1062 /* convert @url and @path to a URL used as a content base for the factory
1063 * located at @path */
1065 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1071 /* check for trailing '/' and append one */
1072 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1077 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1079 result = gst_rtsp_url_get_request_uri (&tmp);
1080 g_free (tmp.abspath);
1086 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1088 GstRTSPSession *session;
1089 GstRTSPClientClass *klass;
1090 GstRTSPSessionMedia *sessmedia;
1091 GstRTSPMedia *media;
1092 GstRTSPStatusCode code;
1095 GstRTSPTimeRange *range;
1097 GstRTSPState rtspstate;
1098 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1099 gchar *path, *rtpinfo;
1102 if (!(session = ctx->session))
1105 if (!(uri = ctx->uri))
1108 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1109 path = klass->make_path_from_uri (client, uri);
1111 /* get a handle to the configuration of the media in the session */
1112 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1116 if (path[matched] != '\0')
1121 ctx->sessmedia = sessmedia;
1122 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1124 /* the session state must be playing or ready */
1125 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1126 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1129 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1130 if (!gst_rtsp_media_unsuspend (media))
1131 goto unsuspend_failed;
1133 /* parse the range header if we have one */
1134 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1135 if (res == GST_RTSP_OK) {
1136 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1137 /* we have a range, seek to the position */
1139 gst_rtsp_media_seek (media, range);
1140 gst_rtsp_range_free (range);
1144 /* grab RTPInfo from the media now */
1145 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1147 /* construct the response now */
1148 code = GST_RTSP_STS_OK;
1149 gst_rtsp_message_init_response (ctx->response, code,
1150 gst_rtsp_status_as_text (code), ctx->request);
1152 /* add the RTP-Info header */
1154 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1158 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1160 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1162 send_message (client, ctx, ctx->response, FALSE);
1164 /* start playing after sending the response */
1165 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1167 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1169 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1176 GST_ERROR ("client %p: no session", client);
1177 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1182 GST_ERROR ("client %p: no uri supplied", client);
1183 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1188 GST_ERROR ("client %p: media not found", client);
1189 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1194 GST_ERROR ("client %p: no aggregate path %s", client, path);
1195 send_generic_response (client,
1196 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1202 GST_ERROR ("client %p: not PLAYING or READY", client);
1203 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1209 GST_ERROR ("client %p: unsuspend failed", client);
1210 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1216 do_keepalive (GstRTSPSession * session)
1218 GST_INFO ("keep session %p alive", session);
1219 gst_rtsp_session_touch (session);
1222 /* parse @transport and return a valid transport in @tr. only transports
1223 * supported by @stream are returned. Returns FALSE if no valid transport
1226 parse_transport (const char *transport, GstRTSPStream * stream,
1227 GstRTSPTransport * tr)
1234 gst_rtsp_transport_init (tr);
1236 GST_DEBUG ("parsing transports %s", transport);
1238 transports = g_strsplit (transport, ",", 0);
1240 /* loop through the transports, try to parse */
1241 for (i = 0; transports[i]; i++) {
1242 res = gst_rtsp_transport_parse (transports[i], tr);
1243 if (res != GST_RTSP_OK) {
1244 /* no valid transport, search some more */
1245 GST_WARNING ("could not parse transport %s", transports[i]);
1249 /* we have a transport, see if it's supported */
1250 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1251 GST_WARNING ("unsupported transport %s", transports[i]);
1255 /* we have a valid transport */
1256 GST_INFO ("found valid transport %s", transports[i]);
1261 gst_rtsp_transport_init (tr);
1263 g_strfreev (transports);
1269 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1270 GstRTSPStream * stream, GstRTSPContext * ctx)
1272 GstRTSPMessage *request = ctx->request;
1273 gchar *blocksize_str;
1275 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1276 &blocksize_str, 0) == GST_RTSP_OK) {
1280 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1281 if (end == blocksize_str)
1284 /* we don't want to change the mtu when this media
1285 * can be shared because it impacts other clients */
1286 if (gst_rtsp_media_is_shared (media))
1289 if (blocksize > G_MAXUINT)
1290 blocksize = G_MAXUINT;
1292 gst_rtsp_stream_set_mtu (stream, blocksize);
1300 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1301 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1307 default_configure_client_transport (GstRTSPClient * client,
1308 GstRTSPContext * ctx, GstRTSPTransport * ct)
1310 GstRTSPClientPrivate *priv = client->priv;
1312 /* we have a valid transport now, set the destination of the client. */
1313 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1314 gboolean use_client_settings;
1316 use_client_settings =
1317 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1319 if (ct->destination && use_client_settings) {
1320 GstRTSPAddress *addr;
1322 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1323 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1328 gst_rtsp_address_free (addr);
1330 GstRTSPAddress *addr;
1331 GSocketFamily family;
1333 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1335 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1339 g_free (ct->destination);
1340 ct->destination = g_strdup (addr->address);
1341 ct->port.min = addr->port;
1342 ct->port.max = addr->port + addr->n_ports - 1;
1343 ct->ttl = addr->ttl;
1345 gst_rtsp_address_free (addr);
1350 url = gst_rtsp_connection_get_url (priv->connection);
1351 g_free (ct->destination);
1352 ct->destination = g_strdup (url->host);
1354 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1356 GSocketAddress *addr;
1358 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1359 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1360 /* our read port is the sender port of client */
1361 ct->client_port.min =
1362 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1363 g_object_unref (addr);
1365 if ((addr = g_socket_get_local_address (sock, NULL))) {
1366 ct->server_port.max =
1367 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1368 g_object_unref (addr);
1370 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1371 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1372 /* our write port is the receiver port of client */
1373 ct->client_port.max =
1374 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1375 g_object_unref (addr);
1377 if ((addr = g_socket_get_local_address (sock, NULL))) {
1378 ct->server_port.min =
1379 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1380 g_object_unref (addr);
1382 /* check if the client selected channels for TCP */
1383 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1384 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1394 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1399 static GstRTSPTransport *
1400 make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
1401 GstRTSPTransport * ct)
1403 GstRTSPTransport *st;
1405 GSocketFamily family;
1407 /* prepare the server transport */
1408 gst_rtsp_transport_new (&st);
1410 st->trans = ct->trans;
1411 st->profile = ct->profile;
1412 st->lower_transport = ct->lower_transport;
1414 addr = g_inet_address_new_from_string (ct->destination);
1417 GST_ERROR ("failed to get inet addr from client destination");
1418 family = G_SOCKET_FAMILY_IPV4;
1420 family = g_inet_address_get_family (addr);
1421 g_object_unref (addr);
1425 switch (st->lower_transport) {
1426 case GST_RTSP_LOWER_TRANS_UDP:
1427 st->client_port = ct->client_port;
1428 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1430 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1431 st->port = ct->port;
1432 st->destination = g_strdup (ct->destination);
1435 case GST_RTSP_LOWER_TRANS_TCP:
1436 st->interleaved = ct->interleaved;
1437 st->client_port = ct->client_port;
1438 st->server_port = ct->server_port;
1443 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1448 #define AES_128_KEY_LEN 16
1449 #define AES_256_KEY_LEN 32
1451 #define HMAC_32_KEY_LEN 4
1452 #define HMAC_80_KEY_LEN 10
1455 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1457 const gchar *srtp_cipher;
1458 const gchar *srtp_auth;
1459 const GstMIKEYPayload *sp;
1462 /* loop over Security policy until we find one containing policy */
1464 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1467 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1471 /* the default ciphers */
1472 srtp_cipher = "aes-128-icm";
1473 srtp_auth = "hmac-sha1-80";
1475 /* now override the defaults with what is in the Security Policy */
1479 /* collect all the params and go over them */
1480 len = gst_mikey_payload_sp_get_n_params (sp);
1481 for (i = 0; i < len; i++) {
1482 const GstMIKEYPayloadSPParam *param =
1483 gst_mikey_payload_sp_get_param (sp, i);
1485 switch (param->type) {
1486 case GST_MIKEY_SP_SRTP_ENC_ALG:
1487 switch (param->val[0]) {
1489 srtp_cipher = "null";
1493 srtp_cipher = "aes-128-icm";
1499 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1500 switch (param->val[0]) {
1501 case AES_128_KEY_LEN:
1502 srtp_cipher = "aes-128-icm";
1504 case AES_256_KEY_LEN:
1505 srtp_cipher = "aes-256-icm";
1511 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1512 switch (param->val[0]) {
1518 srtp_auth = "hmac-sha1-80";
1524 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1525 switch (param->val[0]) {
1526 case HMAC_32_KEY_LEN:
1527 srtp_auth = "hmac-sha1-32";
1529 case HMAC_80_KEY_LEN:
1530 srtp_auth = "hmac-sha1-80";
1536 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1538 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1545 /* now configure the SRTP parameters */
1546 gst_caps_set_simple (caps,
1547 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1548 "srtp-auth", G_TYPE_STRING, srtp_auth,
1549 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1550 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1556 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1557 guint8 * data, gsize size)
1559 GstMIKEYMessage *msg;
1561 GstCaps *caps = NULL;
1562 GstMIKEYPayloadKEMAC *kemac;
1563 const GstMIKEYPayloadKeyData *pkd;
1566 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1567 * set of Crypto Sessions protected with the same master key.
1568 * In the context of SRTP, an RTP and its RTCP stream is part of a
1570 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1573 /* we can only handle SRTP crypto sessions for now */
1574 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1575 goto invalid_map_type;
1577 /* get the number of crypto sessions. This maps SSRC to its
1578 * security parameters */
1579 n_cs = gst_mikey_message_get_n_cs (msg);
1581 goto no_crypto_sessions;
1583 /* we also need keys */
1584 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1585 (msg, GST_MIKEY_PT_KEMAC, 0)))
1588 /* we don't support encrypted keys */
1589 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1590 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1591 goto unsupported_encryption;
1593 /* get Key data sub-payload */
1594 pkd = (const GstMIKEYPayloadKeyData *)
1595 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1598 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1601 /* go over all crypto sessions and create the security policy for each
1603 for (i = 0; i < n_cs; i++) {
1604 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1606 caps = gst_caps_new_simple ("application/x-srtp",
1607 "ssrc", G_TYPE_UINT, map->ssrc,
1608 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1609 mikey_apply_policy (caps, msg, map->policy);
1611 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1612 gst_caps_unref (caps);
1614 gst_mikey_message_unref (msg);
1621 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1626 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1627 goto cleanup_message;
1631 GST_DEBUG_OBJECT (client, "no crypto sessions");
1632 goto cleanup_message;
1636 GST_DEBUG_OBJECT (client, "no keys found");
1637 goto cleanup_message;
1639 unsupported_encryption:
1641 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1642 goto cleanup_message;
1646 gst_mikey_message_unref (msg);
1651 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1654 strip_chars (gchar * str)
1661 if (!IS_STRIP_CHAR (str[len]))
1665 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1666 memmove (str, s, len + 1);
1669 /* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1670 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1673 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1678 specs = g_strsplit (keymgmt, ",", 0);
1679 for (i = 0; specs[i]; i++) {
1682 split = g_strsplit (specs[i], ";", 0);
1683 for (j = 0; split[j]; j++) {
1684 g_strstrip (split[j]);
1685 if (g_str_has_prefix (split[j], "prot=")) {
1686 g_strstrip (split[j] + 5);
1687 if (!g_str_equal (split[j] + 5, "mikey"))
1689 GST_DEBUG ("found mikey");
1690 } else if (g_str_has_prefix (split[j], "uri=")) {
1691 strip_chars (split[j] + 4);
1692 GST_DEBUG ("found uri '%s'", split[j] + 4);
1693 } else if (g_str_has_prefix (split[j], "data=")) {
1696 strip_chars (split[j] + 5);
1697 GST_DEBUG ("found data '%s'", split[j] + 5);
1698 data = g_base64_decode_inplace (split[j] + 5, &size);
1699 handle_mikey_data (client, ctx, data, size);
1707 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1709 GstRTSPClientPrivate *priv = client->priv;
1712 gchar *transport, *keymgmt;
1713 GstRTSPTransport *ct, *st;
1714 GstRTSPStatusCode code;
1715 GstRTSPSession *session;
1716 GstRTSPStreamTransport *trans;
1718 GstRTSPSessionMedia *sessmedia;
1719 GstRTSPMedia *media;
1720 GstRTSPStream *stream;
1721 GstRTSPState rtspstate;
1722 GstRTSPClientClass *klass;
1723 gchar *path, *control;
1725 gboolean new_session = FALSE;
1731 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1732 path = klass->make_path_from_uri (client, uri);
1734 /* parse the transport */
1736 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1738 if (res != GST_RTSP_OK)
1741 /* we create the session after parsing stuff so that we don't make
1742 * a session for malformed requests */
1743 if (priv->session_pool == NULL)
1746 session = ctx->session;
1749 g_object_ref (session);
1750 /* get a handle to the configuration of the media in the session, this can
1751 * return NULL if this is a new url to manage in this session. */
1752 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1754 /* we need a new media configuration in this session */
1758 /* we have no session media, find one and manage it */
1759 if (sessmedia == NULL) {
1760 /* get a handle to the configuration of the media in the session */
1761 media = find_media (client, ctx, path, &matched);
1763 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1764 g_object_ref (media);
1766 goto media_not_found;
1768 /* no media, not found then */
1770 goto media_not_found_no_reply;
1772 if (path[matched] == '\0')
1773 goto control_not_found;
1775 /* path is what matched. */
1776 path[matched] = '\0';
1777 /* control is remainder */
1778 control = &path[matched + 1];
1780 /* find the stream now using the control part */
1781 stream = gst_rtsp_media_find_stream (media, control);
1783 goto stream_not_found;
1785 /* now we have a uri identifying a valid media and stream */
1786 ctx->stream = stream;
1789 if (session == NULL) {
1790 /* create a session if this fails we probably reached our session limit or
1792 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1793 goto service_unavailable;
1795 /* make sure this client is closed when the session is closed */
1796 client_watch_session (client, session);
1799 /* signal new session */
1800 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1803 ctx->session = session;
1806 if (!klass->configure_client_media (client, media, stream, ctx))
1807 goto configure_media_failed_no_reply;
1809 gst_rtsp_transport_new (&ct);
1811 /* parse and find a usable supported transport */
1812 if (!parse_transport (transport, stream, ct))
1813 goto unsupported_transports;
1815 /* update the client transport */
1816 if (!klass->configure_client_transport (client, ctx, ct))
1817 goto unsupported_client_transport;
1819 /* parse the keymgmt */
1820 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1821 &keymgmt, 0) == GST_RTSP_OK) {
1822 if (!handle_keymgmt (client, ctx, keymgmt))
1826 if (sessmedia == NULL) {
1827 /* manage the media in our session now, if not done already */
1828 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1829 /* if we stil have no media, error */
1830 if (sessmedia == NULL)
1831 goto sessmedia_unavailable;
1833 g_object_unref (media);
1836 ctx->sessmedia = sessmedia;
1838 /* set in the session media transport */
1839 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1841 /* configure the url used to set this transport, this we will use when
1842 * generating the response for the PLAY request */
1843 gst_rtsp_stream_transport_set_url (trans, uri);
1844 /* configure keepalive for this transport */
1845 gst_rtsp_stream_transport_set_keepalive (trans,
1846 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1848 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1849 /* our callbacks to send data on this TCP connection */
1850 gst_rtsp_stream_transport_set_callbacks (trans,
1851 (GstRTSPSendFunc) do_send_data,
1852 (GstRTSPSendFunc) do_send_data, client, NULL);
1854 g_hash_table_insert (priv->transports,
1855 GINT_TO_POINTER (ct->interleaved.min), trans);
1856 g_hash_table_insert (priv->transports,
1857 GINT_TO_POINTER (ct->interleaved.max), trans);
1860 /* create and serialize the server transport */
1861 st = make_server_transport (client, ctx, ct);
1862 trans_str = gst_rtsp_transport_as_text (st);
1863 gst_rtsp_transport_free (st);
1865 /* construct the response now */
1866 code = GST_RTSP_STS_OK;
1867 gst_rtsp_message_init_response (ctx->response, code,
1868 gst_rtsp_status_as_text (code), ctx->request);
1870 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1874 send_message (client, ctx, ctx->response, FALSE);
1876 /* update the state */
1877 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1878 switch (rtspstate) {
1879 case GST_RTSP_STATE_PLAYING:
1880 case GST_RTSP_STATE_RECORDING:
1881 case GST_RTSP_STATE_READY:
1882 /* no state change */
1885 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1888 g_object_unref (session);
1891 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1898 GST_ERROR ("client %p: no uri", client);
1899 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1904 GST_ERROR ("client %p: no transport", client);
1905 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1910 GST_ERROR ("client %p: no session pool configured", client);
1911 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1914 media_not_found_no_reply:
1916 GST_ERROR ("client %p: media '%s' not found", client, path);
1917 /* error reply is already sent */
1922 GST_ERROR ("client %p: media '%s' not found", client, path);
1923 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1928 GST_ERROR ("client %p: no control in path '%s'", client, path);
1929 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1930 g_object_unref (media);
1935 GST_ERROR ("client %p: stream '%s' not found", client, control);
1936 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1937 g_object_unref (media);
1940 service_unavailable:
1942 GST_ERROR ("client %p: can't create session", client);
1943 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1944 g_object_unref (media);
1947 sessmedia_unavailable:
1949 GST_ERROR ("client %p: can't create session media", client);
1950 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1951 g_object_unref (media);
1952 goto cleanup_session;
1954 configure_media_failed_no_reply:
1956 GST_ERROR ("client %p: configure_media failed", client);
1957 /* error reply is already sent */
1958 goto cleanup_session;
1960 unsupported_transports:
1962 GST_ERROR ("client %p: unsupported transports", client);
1963 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1964 goto cleanup_transport;
1966 unsupported_client_transport:
1968 GST_ERROR ("client %p: unsupported client transport", client);
1969 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1970 goto cleanup_transport;
1974 GST_ERROR ("client %p: keymgmt error", client);
1975 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
1976 goto cleanup_transport;
1980 gst_rtsp_transport_free (ct);
1983 gst_rtsp_session_pool_remove (priv->session_pool, session);
1984 g_object_unref (session);
1991 static GstSDPMessage *
1992 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1994 GstRTSPClientPrivate *priv = client->priv;
1999 gst_sdp_message_new (&sdp);
2001 /* some standard things first */
2002 gst_sdp_message_set_version (sdp, "0");
2009 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
2012 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2013 gst_sdp_message_set_information (sdp, "rtsp-server");
2014 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2015 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2016 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2017 gst_sdp_message_add_attribute (sdp, "control", "*");
2019 info.is_ipv6 = priv->is_ipv6;
2020 info.server_ip = priv->server_ip;
2022 /* create an SDP for the media object */
2023 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2031 GST_ERROR ("client %p: could not create SDP", client);
2032 gst_sdp_message_free (sdp);
2037 /* for the describe we must generate an SDP */
2039 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2041 GstRTSPClientPrivate *priv = client->priv;
2046 GstRTSPMedia *media;
2047 GstRTSPClientClass *klass;
2049 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2054 /* check what kind of format is accepted, we don't really do anything with it
2055 * and always return SDP for now. */
2060 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2062 if (res == GST_RTSP_ENOTIMPL)
2065 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2069 if (!priv->mount_points)
2070 goto no_mount_points;
2072 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2075 /* find the media object for the uri */
2076 if (!(media = find_media (client, ctx, path, NULL)))
2079 /* create an SDP for the media object on this client */
2080 if (!(sdp = klass->create_sdp (client, media)))
2083 /* we suspend after the describe */
2084 gst_rtsp_media_suspend (media);
2085 g_object_unref (media);
2087 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2088 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2090 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2093 /* content base for some clients that might screw up creating the setup uri */
2094 str = make_base_url (client, ctx->uri, path);
2097 GST_INFO ("adding content-base: %s", str);
2098 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2100 /* add SDP to the response body */
2101 str = gst_sdp_message_as_text (sdp);
2102 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2103 gst_sdp_message_free (sdp);
2105 send_message (client, ctx, ctx->response, FALSE);
2107 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2115 GST_ERROR ("client %p: no uri", client);
2116 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2121 GST_ERROR ("client %p: no mount points configured", client);
2122 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2127 GST_ERROR ("client %p: can't find path for url", client);
2128 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2133 GST_ERROR ("client %p: no media", client);
2135 /* error reply is already sent */
2140 GST_ERROR ("client %p: can't create SDP", client);
2141 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2143 g_object_unref (media);
2149 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2151 GstRTSPMethod options;
2154 options = GST_RTSP_DESCRIBE |
2159 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2161 str = gst_rtsp_options_as_text (options);
2163 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2164 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2166 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2169 send_message (client, ctx, ctx->response, FALSE);
2171 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2177 /* remove duplicate and trailing '/' */
2179 sanitize_uri (GstRTSPUrl * uri)
2183 gboolean have_slash, prev_slash;
2185 s = d = uri->abspath;
2186 len = strlen (uri->abspath);
2190 for (i = 0; i < len; i++) {
2191 have_slash = s[i] == '/';
2193 if (!have_slash || !prev_slash)
2195 prev_slash = have_slash;
2197 len = d - uri->abspath;
2198 /* don't remove the first slash if that's the only thing left */
2199 if (len > 1 && *(d - 1) == '/')
2204 /* is called when the session is removed from its session pool. */
2206 client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
2207 GstRTSPClient * client)
2209 GstRTSPClientPrivate *priv = client->priv;
2211 GST_INFO ("client %p: session %p removed", client, session);
2213 g_mutex_lock (&priv->lock);
2214 client_unwatch_session (client, session, NULL);
2215 g_mutex_unlock (&priv->lock);
2218 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2219 * and also returns a newly-allocated string of (comma-separated) unsupported
2220 * options in the unsupported_reqs variable .
2222 * There may be multiple Require headers, but we must send one single
2223 * Unsupported header with all the unsupported options as response. If
2224 * an incoming Require header contained a comma-separated list of options
2225 * GstRtspConnection will already have split that list up into multiple
2228 * TODO: allow the application to decide what features are supported
2231 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2234 GPtrArray *arr = NULL;
2240 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2242 if (res == GST_RTSP_ENOTIMPL)
2246 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2248 g_ptr_array_add (arr, g_strdup (reqs));
2252 /* if we don't have any Require headers at all, all is fine */
2256 /* otherwise we've now processed at all the Require headers */
2257 g_ptr_array_add (arr, NULL);
2259 /* for now we don't commit to supporting anything, so will just report
2260 * all of the required options as unsupported */
2261 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2263 g_ptr_array_unref (arr);
2268 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2270 GstRTSPClientPrivate *priv = client->priv;
2271 GstRTSPMethod method;
2272 const gchar *uristr;
2273 GstRTSPUrl *uri = NULL;
2274 GstRTSPVersion version;
2276 GstRTSPSession *session = NULL;
2277 GstRTSPContext sctx = { NULL }, *ctx;
2278 GstRTSPMessage response = { 0 };
2279 gchar *unsupported_reqs = NULL;
2282 if (!(ctx = gst_rtsp_context_get_current ())) {
2284 ctx->auth = priv->auth;
2285 gst_rtsp_context_push_current (ctx);
2288 ctx->conn = priv->connection;
2289 ctx->client = client;
2290 ctx->request = request;
2291 ctx->response = &response;
2293 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2294 gst_rtsp_message_dump (request);
2297 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2299 GST_INFO ("client %p: received a request %s %s %s", client,
2300 gst_rtsp_method_as_text (method), uristr,
2301 gst_rtsp_version_as_text (version));
2303 /* we can only handle 1.0 requests */
2304 if (version != GST_RTSP_VERSION_1_0)
2307 ctx->method = method;
2309 /* we always try to parse the url first */
2310 if (strcmp (uristr, "*") == 0) {
2311 /* special case where we have * as uri, keep uri = NULL */
2312 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2313 /* check if the uristr is an absolute path <=> scheme and host information
2317 scheme = g_uri_parse_scheme (uristr);
2318 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2319 gchar *absolute_uristr = NULL;
2321 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2322 if (priv->server_ip == NULL) {
2323 GST_WARNING_OBJECT (client, "host information missing");
2328 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2330 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2331 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2332 g_free (absolute_uristr);
2335 g_free (absolute_uristr);
2342 /* get the session if there is any */
2343 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2344 if (res == GST_RTSP_OK) {
2345 if (priv->session_pool == NULL)
2348 /* we had a session in the request, find it again */
2349 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2350 goto session_not_found;
2352 /* we add the session to the client list of watched sessions. When a session
2353 * disappears because it times out, we will be notified. If all sessions are
2354 * gone, we will close the connection */
2355 client_watch_session (client, session);
2358 /* sanitize the uri */
2362 ctx->session = session;
2364 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2365 goto not_authorized;
2367 /* handle any 'Require' headers */
2368 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2369 goto unsupported_requirement;
2371 /* now see what is asked and dispatch to a dedicated handler */
2373 case GST_RTSP_OPTIONS:
2374 handle_options_request (client, ctx);
2376 case GST_RTSP_DESCRIBE:
2377 handle_describe_request (client, ctx);
2379 case GST_RTSP_SETUP:
2380 handle_setup_request (client, ctx);
2383 handle_play_request (client, ctx);
2385 case GST_RTSP_PAUSE:
2386 handle_pause_request (client, ctx);
2388 case GST_RTSP_TEARDOWN:
2389 handle_teardown_request (client, ctx);
2391 case GST_RTSP_SET_PARAMETER:
2392 handle_set_param_request (client, ctx);
2394 case GST_RTSP_GET_PARAMETER:
2395 handle_get_param_request (client, ctx);
2397 case GST_RTSP_ANNOUNCE:
2398 case GST_RTSP_RECORD:
2399 case GST_RTSP_REDIRECT:
2400 goto not_implemented;
2401 case GST_RTSP_INVALID:
2408 gst_rtsp_context_pop_current (ctx);
2410 g_object_unref (session);
2412 gst_rtsp_url_free (uri);
2418 GST_ERROR ("client %p: version %d not supported", client, version);
2419 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2425 GST_ERROR ("client %p: bad request", client);
2426 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2431 GST_ERROR ("client %p: no pool configured", client);
2432 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2437 GST_ERROR ("client %p: session not found", client);
2438 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2443 GST_ERROR ("client %p: not allowed", client);
2444 /* error reply is already sent */
2447 unsupported_requirement:
2449 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2451 send_option_not_supported_response (client, ctx, unsupported_reqs);
2452 g_free (unsupported_reqs);
2457 GST_ERROR ("client %p: method %d not implemented", client, method);
2458 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2465 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2467 GstRTSPClientPrivate *priv = client->priv;
2469 GstRTSPSession *session = NULL;
2470 GstRTSPContext sctx = { NULL }, *ctx;
2473 if (!(ctx = gst_rtsp_context_get_current ())) {
2475 ctx->auth = priv->auth;
2476 gst_rtsp_context_push_current (ctx);
2479 ctx->conn = priv->connection;
2480 ctx->client = client;
2481 ctx->request = NULL;
2483 ctx->method = GST_RTSP_INVALID;
2484 ctx->response = response;
2486 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2487 gst_rtsp_message_dump (response);
2490 GST_INFO ("client %p: received a response", client);
2492 /* get the session if there is any */
2494 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2495 if (res == GST_RTSP_OK) {
2496 if (priv->session_pool == NULL)
2499 /* we had a session in the request, find it again */
2500 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2501 goto session_not_found;
2503 /* we add the session to the client list of watched sessions. When a session
2504 * disappears because it times out, we will be notified. If all sessions are
2505 * gone, we will close the connection */
2506 client_watch_session (client, session);
2509 ctx->session = session;
2511 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2516 gst_rtsp_context_pop_current (ctx);
2518 g_object_unref (session);
2523 GST_ERROR ("client %p: no pool configured", client);
2528 GST_ERROR ("client %p: session not found", client);
2534 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2536 GstRTSPClientPrivate *priv = client->priv;
2542 GstRTSPStreamTransport *trans;
2544 /* find the stream for this message */
2545 res = gst_rtsp_message_parse_data (message, &channel);
2546 if (res != GST_RTSP_OK)
2549 gst_rtsp_message_steal_body (message, &data, &size);
2551 buffer = gst_buffer_new_wrapped (data, size);
2554 g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
2556 /* dispatch to the stream based on the channel number */
2557 gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
2559 gst_buffer_unref (buffer);
2564 * gst_rtsp_client_set_session_pool:
2565 * @client: a #GstRTSPClient
2566 * @pool: (transfer none): a #GstRTSPSessionPool
2568 * Set @pool as the sessionpool for @client which it will use to find
2569 * or allocate sessions. the sessionpool is usually inherited from the server
2570 * that created the client but can be overridden later.
2573 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
2574 GstRTSPSessionPool * pool)
2576 GstRTSPSessionPool *old;
2577 GstRTSPClientPrivate *priv;
2579 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2581 priv = client->priv;
2584 g_object_ref (pool);
2586 g_mutex_lock (&priv->lock);
2587 old = priv->session_pool;
2588 priv->session_pool = pool;
2590 if (priv->session_removed_id) {
2591 g_signal_handler_disconnect (old, priv->session_removed_id);
2592 priv->session_removed_id = 0;
2594 g_mutex_unlock (&priv->lock);
2596 /* FIXME, should remove all sessions from the old pool for this client */
2598 g_object_unref (old);
2602 * gst_rtsp_client_get_session_pool:
2603 * @client: a #GstRTSPClient
2605 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
2607 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
2609 GstRTSPSessionPool *
2610 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2612 GstRTSPClientPrivate *priv;
2613 GstRTSPSessionPool *result;
2615 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2617 priv = client->priv;
2619 g_mutex_lock (&priv->lock);
2620 if ((result = priv->session_pool))
2621 g_object_ref (result);
2622 g_mutex_unlock (&priv->lock);
2628 * gst_rtsp_client_set_mount_points:
2629 * @client: a #GstRTSPClient
2630 * @mounts: (transfer none): a #GstRTSPMountPoints
2632 * Set @mounts as the mount points for @client which it will use to map urls
2633 * to media streams. These mount points are usually inherited from the server that
2634 * created the client but can be overriden later.
2637 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
2638 GstRTSPMountPoints * mounts)
2640 GstRTSPClientPrivate *priv;
2641 GstRTSPMountPoints *old;
2643 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2645 priv = client->priv;
2648 g_object_ref (mounts);
2650 g_mutex_lock (&priv->lock);
2651 old = priv->mount_points;
2652 priv->mount_points = mounts;
2653 g_mutex_unlock (&priv->lock);
2656 g_object_unref (old);
2660 * gst_rtsp_client_get_mount_points:
2661 * @client: a #GstRTSPClient
2663 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
2665 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
2667 GstRTSPMountPoints *
2668 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
2670 GstRTSPClientPrivate *priv;
2671 GstRTSPMountPoints *result;
2673 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2675 priv = client->priv;
2677 g_mutex_lock (&priv->lock);
2678 if ((result = priv->mount_points))
2679 g_object_ref (result);
2680 g_mutex_unlock (&priv->lock);
2686 * gst_rtsp_client_set_auth:
2687 * @client: a #GstRTSPClient
2688 * @auth: (transfer none): a #GstRTSPAuth
2690 * configure @auth to be used as the authentication manager of @client.
2693 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
2695 GstRTSPClientPrivate *priv;
2698 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2700 priv = client->priv;
2703 g_object_ref (auth);
2705 g_mutex_lock (&priv->lock);
2708 g_mutex_unlock (&priv->lock);
2711 g_object_unref (old);
2716 * gst_rtsp_client_get_auth:
2717 * @client: a #GstRTSPClient
2719 * Get the #GstRTSPAuth used as the authentication manager of @client.
2721 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2725 gst_rtsp_client_get_auth (GstRTSPClient * client)
2727 GstRTSPClientPrivate *priv;
2728 GstRTSPAuth *result;
2730 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2732 priv = client->priv;
2734 g_mutex_lock (&priv->lock);
2735 if ((result = priv->auth))
2736 g_object_ref (result);
2737 g_mutex_unlock (&priv->lock);
2743 * gst_rtsp_client_set_thread_pool:
2744 * @client: a #GstRTSPClient
2745 * @pool: (transfer none): a #GstRTSPThreadPool
2747 * configure @pool to be used as the thread pool of @client.
2750 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
2751 GstRTSPThreadPool * pool)
2753 GstRTSPClientPrivate *priv;
2754 GstRTSPThreadPool *old;
2756 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2758 priv = client->priv;
2761 g_object_ref (pool);
2763 g_mutex_lock (&priv->lock);
2764 old = priv->thread_pool;
2765 priv->thread_pool = pool;
2766 g_mutex_unlock (&priv->lock);
2769 g_object_unref (old);
2773 * gst_rtsp_client_get_thread_pool:
2774 * @client: a #GstRTSPClient
2776 * Get the #GstRTSPThreadPool used as the thread pool of @client.
2778 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
2782 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
2784 GstRTSPClientPrivate *priv;
2785 GstRTSPThreadPool *result;
2787 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2789 priv = client->priv;
2791 g_mutex_lock (&priv->lock);
2792 if ((result = priv->thread_pool))
2793 g_object_ref (result);
2794 g_mutex_unlock (&priv->lock);
2800 * gst_rtsp_client_set_connection:
2801 * @client: a #GstRTSPClient
2802 * @conn: (transfer full): a #GstRTSPConnection
2804 * Set the #GstRTSPConnection of @client. This function takes ownership of
2807 * Returns: %TRUE on success.
2810 gst_rtsp_client_set_connection (GstRTSPClient * client,
2811 GstRTSPConnection * conn)
2813 GstRTSPClientPrivate *priv;
2814 GSocket *read_socket;
2815 GSocketAddress *address;
2817 GError *error = NULL;
2819 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2820 g_return_val_if_fail (conn != NULL, FALSE);
2822 priv = client->priv;
2824 read_socket = gst_rtsp_connection_get_read_socket (conn);
2826 if (!(address = g_socket_get_local_address (read_socket, &error)))
2829 g_free (priv->server_ip);
2830 /* keep the original ip that the client connected to */
2831 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2832 GInetAddress *iaddr;
2834 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2836 /* socket might be ipv6 but adress still ipv4 */
2837 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
2838 priv->server_ip = g_inet_address_to_string (iaddr);
2839 g_object_unref (address);
2841 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
2842 priv->server_ip = g_strdup ("unknown");
2845 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2846 priv->server_ip, priv->is_ipv6);
2848 url = gst_rtsp_connection_get_url (conn);
2849 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2851 priv->connection = conn;
2858 GST_ERROR ("could not get local address %s", error->message);
2859 g_error_free (error);
2865 * gst_rtsp_client_get_connection:
2866 * @client: a #GstRTSPClient
2868 * Get the #GstRTSPConnection of @client.
2870 * Returns: (transfer none): the #GstRTSPConnection of @client.
2871 * The connection object returned remains valid until the client is freed.
2874 gst_rtsp_client_get_connection (GstRTSPClient * client)
2876 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2878 return client->priv->connection;
2882 * gst_rtsp_client_set_send_func:
2883 * @client: a #GstRTSPClient
2884 * @func: (scope notified): a #GstRTSPClientSendFunc
2885 * @user_data: (closure): user data passed to @func
2886 * @notify: (allow-none): called when @user_data is no longer in use
2888 * Set @func as the callback that will be called when a new message needs to be
2889 * sent to the client. @user_data is passed to @func and @notify is called when
2890 * @user_data is no longer in use.
2892 * By default, the client will send the messages on the #GstRTSPConnection that
2893 * was configured with gst_rtsp_client_attach() was called.
2896 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2897 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2899 GstRTSPClientPrivate *priv;
2900 GDestroyNotify old_notify;
2903 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2905 priv = client->priv;
2907 g_mutex_lock (&priv->send_lock);
2908 priv->send_func = func;
2909 old_notify = priv->send_notify;
2910 old_data = priv->send_data;
2911 priv->send_notify = notify;
2912 priv->send_data = user_data;
2913 g_mutex_unlock (&priv->send_lock);
2916 old_notify (old_data);
2920 * gst_rtsp_client_handle_message:
2921 * @client: a #GstRTSPClient
2922 * @message: (transfer none): an #GstRTSPMessage
2924 * Let the client handle @message.
2926 * Returns: a #GstRTSPResult.
2929 gst_rtsp_client_handle_message (GstRTSPClient * client,
2930 GstRTSPMessage * message)
2932 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2933 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2935 switch (message->type) {
2936 case GST_RTSP_MESSAGE_REQUEST:
2937 handle_request (client, message);
2939 case GST_RTSP_MESSAGE_RESPONSE:
2940 handle_response (client, message);
2942 case GST_RTSP_MESSAGE_DATA:
2943 handle_data (client, message);
2952 * gst_rtsp_client_send_message:
2953 * @client: a #GstRTSPClient
2954 * @session: (allow-none) (transfer none): a #GstRTSPSession to send
2955 * the message to or %NULL
2956 * @message: (transfer none): The #GstRTSPMessage to send
2958 * Send a message message to the remote end. @message must be a
2959 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
2962 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
2963 GstRTSPMessage * message)
2965 GstRTSPContext sctx = { NULL }
2967 GstRTSPClientPrivate *priv;
2969 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2970 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2971 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
2972 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
2974 priv = client->priv;
2976 if (!(ctx = gst_rtsp_context_get_current ())) {
2978 ctx->auth = priv->auth;
2979 gst_rtsp_context_push_current (ctx);
2982 ctx->conn = priv->connection;
2983 ctx->client = client;
2984 ctx->session = session;
2986 send_message (client, ctx, message, FALSE);
2989 gst_rtsp_context_pop_current (ctx);
2994 static GstRTSPResult
2995 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
2996 gboolean close, gpointer user_data)
2998 GstRTSPClientPrivate *priv = client->priv;
3006 /* send the response and store the seq number so we can wait until it's
3007 * written to the client to close the connection */
3009 gst_rtsp_watch_send_message (priv->watch, message,
3010 close ? &priv->close_seq : NULL);
3011 if (ret == GST_RTSP_OK)
3014 if (ret != GST_RTSP_ENOMEM)
3018 if (priv->drop_backlog)
3021 /* queue was full, wait for more space */
3022 GST_DEBUG_OBJECT (client, "waiting for backlog");
3023 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3024 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3025 } while (ret != GST_RTSP_EINTR);
3032 GST_DEBUG_OBJECT (client, "got error %d", ret);
3037 static GstRTSPResult
3038 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3041 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3044 static GstRTSPResult
3045 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3047 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3048 GstRTSPClientPrivate *priv = client->priv;
3050 if (priv->close_seq && priv->close_seq == cseq) {
3051 GST_INFO ("client %p: send close message", client);
3052 priv->close_seq = 0;
3053 gst_rtsp_client_close (client);
3059 static GstRTSPResult
3060 closed (GstRTSPWatch * watch, gpointer user_data)
3062 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3063 GstRTSPClientPrivate *priv = client->priv;
3064 const gchar *tunnelid;
3066 GST_INFO ("client %p: connection closed", client);
3068 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3069 g_mutex_lock (&tunnels_lock);
3070 /* remove from tunnelids */
3071 g_hash_table_remove (tunnels, tunnelid);
3072 g_mutex_unlock (&tunnels_lock);
3075 gst_rtsp_watch_set_flushing (watch, TRUE);
3076 g_mutex_lock (&priv->watch_lock);
3077 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3078 g_mutex_unlock (&priv->watch_lock);
3083 static GstRTSPResult
3084 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3086 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3089 str = gst_rtsp_strresult (result);
3090 GST_INFO ("client %p: received an error %s", client, str);
3096 static GstRTSPResult
3097 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3098 GstRTSPMessage * message, guint id, gpointer user_data)
3100 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3103 str = gst_rtsp_strresult (result);
3105 ("client %p: error when handling message %p with id %d: %s",
3106 client, message, id, str);
3113 remember_tunnel (GstRTSPClient * client)
3115 GstRTSPClientPrivate *priv = client->priv;
3116 const gchar *tunnelid;
3118 /* store client in the pending tunnels */
3119 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3120 if (tunnelid == NULL)
3123 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3125 /* we can't have two clients connecting with the same tunnelid */
3126 g_mutex_lock (&tunnels_lock);
3127 if (g_hash_table_lookup (tunnels, tunnelid))
3128 goto tunnel_existed;
3130 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3131 g_mutex_unlock (&tunnels_lock);
3138 GST_ERROR ("client %p: no tunnelid provided", client);
3143 g_mutex_unlock (&tunnels_lock);
3144 GST_ERROR ("client %p: tunnel session %s already existed", client,
3150 static GstRTSPResult
3151 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3153 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3154 GstRTSPClientPrivate *priv = client->priv;
3156 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3159 /* ignore error, it'll only be a problem when the client does a POST again */
3160 remember_tunnel (client);
3166 handle_tunnel (GstRTSPClient * client)
3168 GstRTSPClientPrivate *priv = client->priv;
3169 GstRTSPClient *oclient;
3170 GstRTSPClientPrivate *opriv;
3171 const gchar *tunnelid;
3173 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3174 if (tunnelid == NULL)
3177 /* check for previous tunnel */
3178 g_mutex_lock (&tunnels_lock);
3179 oclient = g_hash_table_lookup (tunnels, tunnelid);
3181 if (oclient == NULL) {
3182 /* no previous tunnel, remember tunnel */
3183 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3184 g_mutex_unlock (&tunnels_lock);
3186 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3187 client, priv->connection);
3189 /* merge both tunnels into the first client */
3190 /* remove the old client from the table. ref before because removing it will
3191 * remove the ref to it. */
3192 g_object_ref (oclient);
3193 g_hash_table_remove (tunnels, tunnelid);
3194 g_mutex_unlock (&tunnels_lock);
3196 opriv = oclient->priv;
3198 g_mutex_lock (&opriv->watch_lock);
3199 if (opriv->watch == NULL)
3202 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3203 oclient, opriv->connection, priv->connection);
3205 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3206 gst_rtsp_watch_reset (priv->watch);
3207 gst_rtsp_watch_reset (opriv->watch);
3208 g_mutex_unlock (&opriv->watch_lock);
3209 g_object_unref (oclient);
3211 /* the old client owns the tunnel now, the new one will be freed */
3212 g_source_destroy ((GSource *) priv->watch);
3214 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3222 GST_ERROR ("client %p: no tunnelid provided", client);
3227 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3228 g_mutex_unlock (&opriv->watch_lock);
3229 g_object_unref (oclient);
3234 static GstRTSPStatusCode
3235 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3237 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3239 GST_INFO ("client %p: tunnel get (connection %p)", client,
3240 client->priv->connection);
3242 if (!handle_tunnel (client)) {
3243 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3246 return GST_RTSP_STS_OK;
3249 static GstRTSPResult
3250 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3252 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3254 GST_INFO ("client %p: tunnel post (connection %p)", client,
3255 client->priv->connection);
3257 if (!handle_tunnel (client)) {
3258 return GST_RTSP_ERROR;
3264 static GstRTSPResult
3265 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3266 GstRTSPMessage * response, gpointer user_data)
3268 GstRTSPClientClass *klass;
3270 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3271 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3273 if (klass->tunnel_http_response) {
3274 klass->tunnel_http_response (client, request, response);
3280 static GstRTSPWatchFuncs watch_funcs = {
3289 tunnel_http_response
3293 client_watch_notify (GstRTSPClient * client)
3295 GstRTSPClientPrivate *priv = client->priv;
3297 GST_INFO ("client %p: watch destroyed", client);
3299 g_main_context_unref (priv->watch_context);
3300 priv->watch_context = NULL;
3301 /* remove all sessions and so drop the extra client ref */
3302 gst_rtsp_client_session_filter (client, cleanup_session, NULL);
3303 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3304 g_object_unref (client);
3308 * gst_rtsp_client_attach:
3309 * @client: a #GstRTSPClient
3310 * @context: (allow-none): a #GMainContext
3312 * Attaches @client to @context. When the mainloop for @context is run, the
3313 * client will be dispatched. When @context is %NULL, the default context will be
3316 * This function should be called when the client properties and urls are fully
3317 * configured and the client is ready to start.
3319 * Returns: the ID (greater than 0) for the source within the GMainContext.
3322 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3324 GstRTSPClientPrivate *priv;
3327 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3328 priv = client->priv;
3329 g_return_val_if_fail (priv->connection != NULL, 0);
3330 g_return_val_if_fail (priv->watch == NULL, 0);
3332 /* make sure noone will free the context before the watch is destroyed */
3333 priv->watch_context = g_main_context_ref (context);
3335 /* create watch for the connection and attach */
3336 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3337 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3338 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3339 (GDestroyNotify) gst_rtsp_watch_unref);
3341 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
3343 GST_INFO ("client %p: attaching to context %p", client, context);
3344 res = gst_rtsp_watch_attach (priv->watch, context);
3350 * gst_rtsp_client_session_filter:
3351 * @client: a #GstRTSPClient
3352 * @func: (scope call) (allow-none): a callback
3353 * @user_data: user data passed to @func
3355 * Call @func for each session managed by @client. The result value of @func
3356 * determines what happens to the session. @func will be called with @client
3357 * locked so no further actions on @client can be performed from @func.
3359 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3362 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3364 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3365 * will also be added with an additional ref to the result #GList of this
3368 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3370 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3371 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3372 * element in the #GList should be unreffed before the list is freed.
3375 gst_rtsp_client_session_filter (GstRTSPClient * client,
3376 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3378 GstRTSPClientPrivate *priv;
3379 GList *result, *walk, *next;
3380 GHashTable *visited;
3383 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3385 priv = client->priv;
3389 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3391 g_mutex_lock (&priv->lock);
3393 cookie = priv->sessions_cookie;
3394 for (walk = priv->sessions; walk; walk = next) {
3395 GstRTSPSession *sess = walk->data;
3396 GstRTSPFilterResult res;
3399 next = g_list_next (walk);
3402 /* only visit each session once */
3403 if (g_hash_table_contains (visited, sess))
3406 g_hash_table_add (visited, g_object_ref (sess));
3407 g_mutex_unlock (&priv->lock);
3409 res = func (client, sess, user_data);
3411 g_mutex_lock (&priv->lock);
3413 res = GST_RTSP_FILTER_REF;
3415 changed = (cookie != priv->sessions_cookie);
3418 case GST_RTSP_FILTER_REMOVE:
3419 /* stop watching the session and pretend it went away, if the list was
3420 * changed, we can't use the current list position, try to see if we
3421 * still have the session */
3422 client_unwatch_session (client, sess, changed ? NULL : walk);
3423 cookie = priv->sessions_cookie;
3425 case GST_RTSP_FILTER_REF:
3426 result = g_list_prepend (result, g_object_ref (sess));
3428 case GST_RTSP_FILTER_KEEP:
3435 g_mutex_unlock (&priv->lock);
3438 g_hash_table_unref (visited);