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 };
694 GstRTSPResult res = GST_RTSP_OK;
699 gst_rtsp_message_init_data (&message, channel);
701 /* FIXME, need some sort of iovec RTSPMessage here */
702 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
705 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
707 g_mutex_lock (&priv->send_lock);
709 res = priv->send_func (client, &message, FALSE, priv->send_data);
710 g_mutex_unlock (&priv->send_lock);
712 gst_rtsp_message_steal_body (&message, &data, &usize);
713 gst_buffer_unmap (buffer, &map_info);
715 gst_rtsp_message_unset (&message);
717 return res == GST_RTSP_OK;
721 * gst_rtsp_client_close:
722 * @client: a #GstRTSPClient
724 * Close the connection of @client and remove all media it was managing.
729 gst_rtsp_client_close (GstRTSPClient * client)
731 GstRTSPClientPrivate *priv = client->priv;
732 const gchar *tunnelid;
734 GST_DEBUG ("client %p: closing connection", client);
736 if (priv->connection) {
737 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
738 g_mutex_lock (&tunnels_lock);
739 /* remove from tunnelids */
740 g_hash_table_remove (tunnels, tunnelid);
741 g_mutex_unlock (&tunnels_lock);
743 gst_rtsp_connection_close (priv->connection);
746 /* connection is now closed, destroy the watch which will also cause the
747 * closed signal to be emitted */
749 GST_DEBUG ("client %p: destroying watch", client);
750 g_source_destroy ((GSource *) priv->watch);
752 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
757 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
762 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
764 path = g_strdup (uri->abspath);
770 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
772 GstRTSPClientPrivate *priv = client->priv;
773 GstRTSPClientClass *klass;
774 GstRTSPSession *session;
775 GstRTSPSessionMedia *sessmedia;
776 GstRTSPStatusCode code;
779 gboolean keep_session;
784 session = ctx->session;
789 klass = GST_RTSP_CLIENT_GET_CLASS (client);
790 path = klass->make_path_from_uri (client, ctx->uri);
792 /* get a handle to the configuration of the media in the session */
793 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
797 /* only aggregate control for now.. */
798 if (path[matched] != '\0')
803 ctx->sessmedia = sessmedia;
805 /* we emit the signal before closing the connection */
806 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
809 /* make sure we unblock the backlog and don't accept new messages
811 if (priv->watch != NULL)
812 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
814 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
816 /* allow messages again so that we can send the reply */
817 if (priv->watch != NULL)
818 gst_rtsp_watch_set_flushing (priv->watch, FALSE);
820 /* unmanage the media in the session, returns false if all media session
822 keep_session = gst_rtsp_session_release_media (session, sessmedia);
824 /* construct the response now */
825 code = GST_RTSP_STS_OK;
826 gst_rtsp_message_init_response (ctx->response, code,
827 gst_rtsp_status_as_text (code), ctx->request);
829 send_message (client, ctx, ctx->response, TRUE);
832 /* remove the session */
833 gst_rtsp_session_pool_remove (priv->session_pool, session);
841 GST_ERROR ("client %p: no session", client);
842 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
847 GST_ERROR ("client %p: no uri supplied", client);
848 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
853 GST_ERROR ("client %p: no media for uri", client);
854 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
860 GST_ERROR ("client %p: no aggregate path %s", client, path);
861 send_generic_response (client,
862 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
869 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
873 res = gst_rtsp_params_set (client, ctx);
879 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
883 res = gst_rtsp_params_get (client, ctx);
889 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
895 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
896 if (res != GST_RTSP_OK)
900 /* no body, keep-alive request */
901 send_generic_response (client, GST_RTSP_STS_OK, ctx);
903 /* there is a body, handle the params */
904 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
905 if (res != GST_RTSP_OK)
908 send_message (client, ctx, ctx->response, FALSE);
911 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
919 GST_ERROR ("client %p: bad request", client);
920 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
926 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
932 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
933 if (res != GST_RTSP_OK)
937 /* no body, keep-alive request */
938 send_generic_response (client, GST_RTSP_STS_OK, ctx);
940 /* there is a body, handle the params */
941 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
942 if (res != GST_RTSP_OK)
945 send_message (client, ctx, ctx->response, FALSE);
948 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
956 GST_ERROR ("client %p: bad request", client);
957 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
963 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
965 GstRTSPClientPrivate *priv = client->priv;
966 GstRTSPSession *session;
967 GstRTSPClientClass *klass;
968 GstRTSPSessionMedia *sessmedia;
969 GstRTSPStatusCode code;
970 GstRTSPState rtspstate;
974 if (!(session = ctx->session))
980 klass = GST_RTSP_CLIENT_GET_CLASS (client);
981 path = klass->make_path_from_uri (client, ctx->uri);
983 /* get a handle to the configuration of the media in the session */
984 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
988 if (path[matched] != '\0')
993 ctx->sessmedia = sessmedia;
995 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
996 /* the session state must be playing or recording */
997 if (rtspstate != GST_RTSP_STATE_PLAYING &&
998 rtspstate != GST_RTSP_STATE_RECORDING)
1001 /* No limit on watch queue because else we might be blocking in the appsink
1002 * render method and the PAUSE below will hang */
1003 if (priv->watch != NULL)
1004 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
1006 /* then pause sending */
1007 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1009 /* construct the response now */
1010 code = GST_RTSP_STS_OK;
1011 gst_rtsp_message_init_response (ctx->response, code,
1012 gst_rtsp_status_as_text (code), ctx->request);
1014 send_message (client, ctx, ctx->response, FALSE);
1016 if (priv->watch != NULL)
1017 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
1019 /* the state is now READY */
1020 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1022 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1029 GST_ERROR ("client %p: no seesion", client);
1030 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1035 GST_ERROR ("client %p: no uri supplied", client);
1036 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1041 GST_ERROR ("client %p: no media for uri", client);
1042 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1048 GST_ERROR ("client %p: no aggregate path %s", client, path);
1049 send_generic_response (client,
1050 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1056 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1057 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1063 /* convert @url and @path to a URL used as a content base for the factory
1064 * located at @path */
1066 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1072 /* check for trailing '/' and append one */
1073 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1078 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1080 result = gst_rtsp_url_get_request_uri (&tmp);
1081 g_free (tmp.abspath);
1087 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1089 GstRTSPSession *session;
1090 GstRTSPClientClass *klass;
1091 GstRTSPSessionMedia *sessmedia;
1092 GstRTSPMedia *media;
1093 GstRTSPStatusCode code;
1096 GstRTSPTimeRange *range;
1098 GstRTSPState rtspstate;
1099 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1100 gchar *path, *rtpinfo;
1103 if (!(session = ctx->session))
1106 if (!(uri = ctx->uri))
1109 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1110 path = klass->make_path_from_uri (client, uri);
1112 /* get a handle to the configuration of the media in the session */
1113 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1117 if (path[matched] != '\0')
1122 ctx->sessmedia = sessmedia;
1123 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1125 /* the session state must be playing or ready */
1126 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1127 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1130 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1131 if (!gst_rtsp_media_unsuspend (media))
1132 goto unsuspend_failed;
1134 /* parse the range header if we have one */
1135 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1136 if (res == GST_RTSP_OK) {
1137 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1138 /* we have a range, seek to the position */
1140 gst_rtsp_media_seek (media, range);
1141 gst_rtsp_range_free (range);
1145 /* grab RTPInfo from the media now */
1146 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1148 /* construct the response now */
1149 code = GST_RTSP_STS_OK;
1150 gst_rtsp_message_init_response (ctx->response, code,
1151 gst_rtsp_status_as_text (code), ctx->request);
1153 /* add the RTP-Info header */
1155 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1159 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1161 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1163 send_message (client, ctx, ctx->response, FALSE);
1165 /* start playing after sending the response */
1166 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1168 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1170 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1177 GST_ERROR ("client %p: no session", client);
1178 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1183 GST_ERROR ("client %p: no uri supplied", client);
1184 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1189 GST_ERROR ("client %p: media not found", client);
1190 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1195 GST_ERROR ("client %p: no aggregate path %s", client, path);
1196 send_generic_response (client,
1197 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1203 GST_ERROR ("client %p: not PLAYING or READY", client);
1204 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1210 GST_ERROR ("client %p: unsuspend failed", client);
1211 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1217 do_keepalive (GstRTSPSession * session)
1219 GST_INFO ("keep session %p alive", session);
1220 gst_rtsp_session_touch (session);
1223 /* parse @transport and return a valid transport in @tr. only transports
1224 * supported by @stream are returned. Returns FALSE if no valid transport
1227 parse_transport (const char *transport, GstRTSPStream * stream,
1228 GstRTSPTransport * tr)
1235 gst_rtsp_transport_init (tr);
1237 GST_DEBUG ("parsing transports %s", transport);
1239 transports = g_strsplit (transport, ",", 0);
1241 /* loop through the transports, try to parse */
1242 for (i = 0; transports[i]; i++) {
1243 res = gst_rtsp_transport_parse (transports[i], tr);
1244 if (res != GST_RTSP_OK) {
1245 /* no valid transport, search some more */
1246 GST_WARNING ("could not parse transport %s", transports[i]);
1250 /* we have a transport, see if it's supported */
1251 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1252 GST_WARNING ("unsupported transport %s", transports[i]);
1256 /* we have a valid transport */
1257 GST_INFO ("found valid transport %s", transports[i]);
1262 gst_rtsp_transport_init (tr);
1264 g_strfreev (transports);
1270 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1271 GstRTSPStream * stream, GstRTSPContext * ctx)
1273 GstRTSPMessage *request = ctx->request;
1274 gchar *blocksize_str;
1276 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1277 &blocksize_str, 0) == GST_RTSP_OK) {
1281 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1282 if (end == blocksize_str)
1285 /* we don't want to change the mtu when this media
1286 * can be shared because it impacts other clients */
1287 if (gst_rtsp_media_is_shared (media))
1290 if (blocksize > G_MAXUINT)
1291 blocksize = G_MAXUINT;
1293 gst_rtsp_stream_set_mtu (stream, blocksize);
1301 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1302 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1308 default_configure_client_transport (GstRTSPClient * client,
1309 GstRTSPContext * ctx, GstRTSPTransport * ct)
1311 GstRTSPClientPrivate *priv = client->priv;
1313 /* we have a valid transport now, set the destination of the client. */
1314 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1315 gboolean use_client_settings;
1317 use_client_settings =
1318 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1320 if (ct->destination && use_client_settings) {
1321 GstRTSPAddress *addr;
1323 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1324 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1329 gst_rtsp_address_free (addr);
1331 GstRTSPAddress *addr;
1332 GSocketFamily family;
1334 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1336 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1340 g_free (ct->destination);
1341 ct->destination = g_strdup (addr->address);
1342 ct->port.min = addr->port;
1343 ct->port.max = addr->port + addr->n_ports - 1;
1344 ct->ttl = addr->ttl;
1346 gst_rtsp_address_free (addr);
1351 url = gst_rtsp_connection_get_url (priv->connection);
1352 g_free (ct->destination);
1353 ct->destination = g_strdup (url->host);
1355 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1357 GSocketAddress *addr;
1359 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1360 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1361 /* our read port is the sender port of client */
1362 ct->client_port.min =
1363 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1364 g_object_unref (addr);
1366 if ((addr = g_socket_get_local_address (sock, NULL))) {
1367 ct->server_port.max =
1368 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1369 g_object_unref (addr);
1371 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1372 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1373 /* our write port is the receiver port of client */
1374 ct->client_port.max =
1375 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1376 g_object_unref (addr);
1378 if ((addr = g_socket_get_local_address (sock, NULL))) {
1379 ct->server_port.min =
1380 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1381 g_object_unref (addr);
1383 /* check if the client selected channels for TCP */
1384 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1385 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1395 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1400 static GstRTSPTransport *
1401 make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
1402 GstRTSPTransport * ct)
1404 GstRTSPTransport *st;
1406 GSocketFamily family;
1408 /* prepare the server transport */
1409 gst_rtsp_transport_new (&st);
1411 st->trans = ct->trans;
1412 st->profile = ct->profile;
1413 st->lower_transport = ct->lower_transport;
1415 addr = g_inet_address_new_from_string (ct->destination);
1418 GST_ERROR ("failed to get inet addr from client destination");
1419 family = G_SOCKET_FAMILY_IPV4;
1421 family = g_inet_address_get_family (addr);
1422 g_object_unref (addr);
1426 switch (st->lower_transport) {
1427 case GST_RTSP_LOWER_TRANS_UDP:
1428 st->client_port = ct->client_port;
1429 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1431 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1432 st->port = ct->port;
1433 st->destination = g_strdup (ct->destination);
1436 case GST_RTSP_LOWER_TRANS_TCP:
1437 st->interleaved = ct->interleaved;
1438 st->client_port = ct->client_port;
1439 st->server_port = ct->server_port;
1444 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1449 #define AES_128_KEY_LEN 16
1450 #define AES_256_KEY_LEN 32
1452 #define HMAC_32_KEY_LEN 4
1453 #define HMAC_80_KEY_LEN 10
1456 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1458 const gchar *srtp_cipher;
1459 const gchar *srtp_auth;
1460 const GstMIKEYPayload *sp;
1463 /* loop over Security policy until we find one containing policy */
1465 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1468 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1472 /* the default ciphers */
1473 srtp_cipher = "aes-128-icm";
1474 srtp_auth = "hmac-sha1-80";
1476 /* now override the defaults with what is in the Security Policy */
1480 /* collect all the params and go over them */
1481 len = gst_mikey_payload_sp_get_n_params (sp);
1482 for (i = 0; i < len; i++) {
1483 const GstMIKEYPayloadSPParam *param =
1484 gst_mikey_payload_sp_get_param (sp, i);
1486 switch (param->type) {
1487 case GST_MIKEY_SP_SRTP_ENC_ALG:
1488 switch (param->val[0]) {
1490 srtp_cipher = "null";
1494 srtp_cipher = "aes-128-icm";
1500 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1501 switch (param->val[0]) {
1502 case AES_128_KEY_LEN:
1503 srtp_cipher = "aes-128-icm";
1505 case AES_256_KEY_LEN:
1506 srtp_cipher = "aes-256-icm";
1512 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1513 switch (param->val[0]) {
1519 srtp_auth = "hmac-sha1-80";
1525 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1526 switch (param->val[0]) {
1527 case HMAC_32_KEY_LEN:
1528 srtp_auth = "hmac-sha1-32";
1530 case HMAC_80_KEY_LEN:
1531 srtp_auth = "hmac-sha1-80";
1537 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1539 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1546 /* now configure the SRTP parameters */
1547 gst_caps_set_simple (caps,
1548 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1549 "srtp-auth", G_TYPE_STRING, srtp_auth,
1550 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1551 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1557 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1558 guint8 * data, gsize size)
1560 GstMIKEYMessage *msg;
1562 GstCaps *caps = NULL;
1563 GstMIKEYPayloadKEMAC *kemac;
1564 const GstMIKEYPayloadKeyData *pkd;
1567 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1568 * set of Crypto Sessions protected with the same master key.
1569 * In the context of SRTP, an RTP and its RTCP stream is part of a
1571 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1574 /* we can only handle SRTP crypto sessions for now */
1575 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1576 goto invalid_map_type;
1578 /* get the number of crypto sessions. This maps SSRC to its
1579 * security parameters */
1580 n_cs = gst_mikey_message_get_n_cs (msg);
1582 goto no_crypto_sessions;
1584 /* we also need keys */
1585 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1586 (msg, GST_MIKEY_PT_KEMAC, 0)))
1589 /* we don't support encrypted keys */
1590 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1591 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1592 goto unsupported_encryption;
1594 /* get Key data sub-payload */
1595 pkd = (const GstMIKEYPayloadKeyData *)
1596 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1599 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1602 /* go over all crypto sessions and create the security policy for each
1604 for (i = 0; i < n_cs; i++) {
1605 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1607 caps = gst_caps_new_simple ("application/x-srtp",
1608 "ssrc", G_TYPE_UINT, map->ssrc,
1609 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1610 mikey_apply_policy (caps, msg, map->policy);
1612 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1613 gst_caps_unref (caps);
1615 gst_mikey_message_unref (msg);
1622 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1627 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1628 goto cleanup_message;
1632 GST_DEBUG_OBJECT (client, "no crypto sessions");
1633 goto cleanup_message;
1637 GST_DEBUG_OBJECT (client, "no keys found");
1638 goto cleanup_message;
1640 unsupported_encryption:
1642 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1643 goto cleanup_message;
1647 gst_mikey_message_unref (msg);
1652 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1655 strip_chars (gchar * str)
1662 if (!IS_STRIP_CHAR (str[len]))
1666 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1667 memmove (str, s, len + 1);
1670 /* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1671 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1674 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1679 specs = g_strsplit (keymgmt, ",", 0);
1680 for (i = 0; specs[i]; i++) {
1683 split = g_strsplit (specs[i], ";", 0);
1684 for (j = 0; split[j]; j++) {
1685 g_strstrip (split[j]);
1686 if (g_str_has_prefix (split[j], "prot=")) {
1687 g_strstrip (split[j] + 5);
1688 if (!g_str_equal (split[j] + 5, "mikey"))
1690 GST_DEBUG ("found mikey");
1691 } else if (g_str_has_prefix (split[j], "uri=")) {
1692 strip_chars (split[j] + 4);
1693 GST_DEBUG ("found uri '%s'", split[j] + 4);
1694 } else if (g_str_has_prefix (split[j], "data=")) {
1697 strip_chars (split[j] + 5);
1698 GST_DEBUG ("found data '%s'", split[j] + 5);
1699 data = g_base64_decode_inplace (split[j] + 5, &size);
1700 handle_mikey_data (client, ctx, data, size);
1708 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1710 GstRTSPClientPrivate *priv = client->priv;
1713 gchar *transport, *keymgmt;
1714 GstRTSPTransport *ct, *st;
1715 GstRTSPStatusCode code;
1716 GstRTSPSession *session;
1717 GstRTSPStreamTransport *trans;
1719 GstRTSPSessionMedia *sessmedia;
1720 GstRTSPMedia *media;
1721 GstRTSPStream *stream;
1722 GstRTSPState rtspstate;
1723 GstRTSPClientClass *klass;
1724 gchar *path, *control;
1726 gboolean new_session = FALSE;
1732 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1733 path = klass->make_path_from_uri (client, uri);
1735 /* parse the transport */
1737 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1739 if (res != GST_RTSP_OK)
1742 /* we create the session after parsing stuff so that we don't make
1743 * a session for malformed requests */
1744 if (priv->session_pool == NULL)
1747 session = ctx->session;
1750 g_object_ref (session);
1751 /* get a handle to the configuration of the media in the session, this can
1752 * return NULL if this is a new url to manage in this session. */
1753 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1755 /* we need a new media configuration in this session */
1759 /* we have no session media, find one and manage it */
1760 if (sessmedia == NULL) {
1761 /* get a handle to the configuration of the media in the session */
1762 media = find_media (client, ctx, path, &matched);
1764 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1765 g_object_ref (media);
1767 goto media_not_found;
1769 /* no media, not found then */
1771 goto media_not_found_no_reply;
1773 if (path[matched] == '\0')
1774 goto control_not_found;
1776 /* path is what matched. */
1777 path[matched] = '\0';
1778 /* control is remainder */
1779 control = &path[matched + 1];
1781 /* find the stream now using the control part */
1782 stream = gst_rtsp_media_find_stream (media, control);
1784 goto stream_not_found;
1786 /* now we have a uri identifying a valid media and stream */
1787 ctx->stream = stream;
1790 if (session == NULL) {
1791 /* create a session if this fails we probably reached our session limit or
1793 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1794 goto service_unavailable;
1796 /* make sure this client is closed when the session is closed */
1797 client_watch_session (client, session);
1800 /* signal new session */
1801 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1804 ctx->session = session;
1807 if (!klass->configure_client_media (client, media, stream, ctx))
1808 goto configure_media_failed_no_reply;
1810 gst_rtsp_transport_new (&ct);
1812 /* parse and find a usable supported transport */
1813 if (!parse_transport (transport, stream, ct))
1814 goto unsupported_transports;
1816 /* update the client transport */
1817 if (!klass->configure_client_transport (client, ctx, ct))
1818 goto unsupported_client_transport;
1820 /* parse the keymgmt */
1821 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1822 &keymgmt, 0) == GST_RTSP_OK) {
1823 if (!handle_keymgmt (client, ctx, keymgmt))
1827 if (sessmedia == NULL) {
1828 /* manage the media in our session now, if not done already */
1829 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1830 /* if we stil have no media, error */
1831 if (sessmedia == NULL)
1832 goto sessmedia_unavailable;
1834 g_object_unref (media);
1837 ctx->sessmedia = sessmedia;
1839 /* set in the session media transport */
1840 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1842 /* configure the url used to set this transport, this we will use when
1843 * generating the response for the PLAY request */
1844 gst_rtsp_stream_transport_set_url (trans, uri);
1845 /* configure keepalive for this transport */
1846 gst_rtsp_stream_transport_set_keepalive (trans,
1847 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1849 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1850 /* our callbacks to send data on this TCP connection */
1851 gst_rtsp_stream_transport_set_callbacks (trans,
1852 (GstRTSPSendFunc) do_send_data,
1853 (GstRTSPSendFunc) do_send_data, client, NULL);
1855 g_hash_table_insert (priv->transports,
1856 GINT_TO_POINTER (ct->interleaved.min), trans);
1857 g_hash_table_insert (priv->transports,
1858 GINT_TO_POINTER (ct->interleaved.max), trans);
1861 /* create and serialize the server transport */
1862 st = make_server_transport (client, ctx, ct);
1863 trans_str = gst_rtsp_transport_as_text (st);
1864 gst_rtsp_transport_free (st);
1866 /* construct the response now */
1867 code = GST_RTSP_STS_OK;
1868 gst_rtsp_message_init_response (ctx->response, code,
1869 gst_rtsp_status_as_text (code), ctx->request);
1871 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1875 send_message (client, ctx, ctx->response, FALSE);
1877 /* update the state */
1878 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1879 switch (rtspstate) {
1880 case GST_RTSP_STATE_PLAYING:
1881 case GST_RTSP_STATE_RECORDING:
1882 case GST_RTSP_STATE_READY:
1883 /* no state change */
1886 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1889 g_object_unref (session);
1892 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1899 GST_ERROR ("client %p: no uri", client);
1900 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1905 GST_ERROR ("client %p: no transport", client);
1906 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1911 GST_ERROR ("client %p: no session pool configured", client);
1912 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1915 media_not_found_no_reply:
1917 GST_ERROR ("client %p: media '%s' not found", client, path);
1918 /* error reply is already sent */
1923 GST_ERROR ("client %p: media '%s' not found", client, path);
1924 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1929 GST_ERROR ("client %p: no control in path '%s'", client, path);
1930 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1931 g_object_unref (media);
1936 GST_ERROR ("client %p: stream '%s' not found", client, control);
1937 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1938 g_object_unref (media);
1941 service_unavailable:
1943 GST_ERROR ("client %p: can't create session", client);
1944 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1945 g_object_unref (media);
1948 sessmedia_unavailable:
1950 GST_ERROR ("client %p: can't create session media", client);
1951 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1952 g_object_unref (media);
1953 goto cleanup_session;
1955 configure_media_failed_no_reply:
1957 GST_ERROR ("client %p: configure_media failed", client);
1958 /* error reply is already sent */
1959 goto cleanup_session;
1961 unsupported_transports:
1963 GST_ERROR ("client %p: unsupported transports", client);
1964 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1965 goto cleanup_transport;
1967 unsupported_client_transport:
1969 GST_ERROR ("client %p: unsupported client transport", client);
1970 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1971 goto cleanup_transport;
1975 GST_ERROR ("client %p: keymgmt error", client);
1976 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
1977 goto cleanup_transport;
1981 gst_rtsp_transport_free (ct);
1984 gst_rtsp_session_pool_remove (priv->session_pool, session);
1985 g_object_unref (session);
1992 static GstSDPMessage *
1993 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1995 GstRTSPClientPrivate *priv = client->priv;
2000 gst_sdp_message_new (&sdp);
2002 /* some standard things first */
2003 gst_sdp_message_set_version (sdp, "0");
2010 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
2013 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2014 gst_sdp_message_set_information (sdp, "rtsp-server");
2015 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2016 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2017 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2018 gst_sdp_message_add_attribute (sdp, "control", "*");
2020 info.is_ipv6 = priv->is_ipv6;
2021 info.server_ip = priv->server_ip;
2023 /* create an SDP for the media object */
2024 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2032 GST_ERROR ("client %p: could not create SDP", client);
2033 gst_sdp_message_free (sdp);
2038 /* for the describe we must generate an SDP */
2040 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2042 GstRTSPClientPrivate *priv = client->priv;
2047 GstRTSPMedia *media;
2048 GstRTSPClientClass *klass;
2050 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2055 /* check what kind of format is accepted, we don't really do anything with it
2056 * and always return SDP for now. */
2061 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2063 if (res == GST_RTSP_ENOTIMPL)
2066 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2070 if (!priv->mount_points)
2071 goto no_mount_points;
2073 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2076 /* find the media object for the uri */
2077 if (!(media = find_media (client, ctx, path, NULL)))
2080 /* create an SDP for the media object on this client */
2081 if (!(sdp = klass->create_sdp (client, media)))
2084 /* we suspend after the describe */
2085 gst_rtsp_media_suspend (media);
2086 g_object_unref (media);
2088 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2089 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2091 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2094 /* content base for some clients that might screw up creating the setup uri */
2095 str = make_base_url (client, ctx->uri, path);
2098 GST_INFO ("adding content-base: %s", str);
2099 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2101 /* add SDP to the response body */
2102 str = gst_sdp_message_as_text (sdp);
2103 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2104 gst_sdp_message_free (sdp);
2106 send_message (client, ctx, ctx->response, FALSE);
2108 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2116 GST_ERROR ("client %p: no uri", client);
2117 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2122 GST_ERROR ("client %p: no mount points configured", client);
2123 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2128 GST_ERROR ("client %p: can't find path for url", client);
2129 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2134 GST_ERROR ("client %p: no media", client);
2136 /* error reply is already sent */
2141 GST_ERROR ("client %p: can't create SDP", client);
2142 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2144 g_object_unref (media);
2150 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2152 GstRTSPMethod options;
2155 options = GST_RTSP_DESCRIBE |
2160 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2162 str = gst_rtsp_options_as_text (options);
2164 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2165 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2167 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2170 send_message (client, ctx, ctx->response, FALSE);
2172 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2178 /* remove duplicate and trailing '/' */
2180 sanitize_uri (GstRTSPUrl * uri)
2184 gboolean have_slash, prev_slash;
2186 s = d = uri->abspath;
2187 len = strlen (uri->abspath);
2191 for (i = 0; i < len; i++) {
2192 have_slash = s[i] == '/';
2194 if (!have_slash || !prev_slash)
2196 prev_slash = have_slash;
2198 len = d - uri->abspath;
2199 /* don't remove the first slash if that's the only thing left */
2200 if (len > 1 && *(d - 1) == '/')
2205 /* is called when the session is removed from its session pool. */
2207 client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
2208 GstRTSPClient * client)
2210 GstRTSPClientPrivate *priv = client->priv;
2212 GST_INFO ("client %p: session %p removed", client, session);
2214 g_mutex_lock (&priv->lock);
2215 if (priv->watch != NULL)
2216 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
2217 client_unwatch_session (client, session, NULL);
2218 if (priv->watch != NULL)
2219 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2220 g_mutex_unlock (&priv->lock);
2223 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2224 * and also returns a newly-allocated string of (comma-separated) unsupported
2225 * options in the unsupported_reqs variable .
2227 * There may be multiple Require headers, but we must send one single
2228 * Unsupported header with all the unsupported options as response. If
2229 * an incoming Require header contained a comma-separated list of options
2230 * GstRtspConnection will already have split that list up into multiple
2233 * TODO: allow the application to decide what features are supported
2236 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2239 GPtrArray *arr = NULL;
2245 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2247 if (res == GST_RTSP_ENOTIMPL)
2251 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2253 g_ptr_array_add (arr, g_strdup (reqs));
2257 /* if we don't have any Require headers at all, all is fine */
2261 /* otherwise we've now processed at all the Require headers */
2262 g_ptr_array_add (arr, NULL);
2264 /* for now we don't commit to supporting anything, so will just report
2265 * all of the required options as unsupported */
2266 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2268 g_ptr_array_unref (arr);
2273 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2275 GstRTSPClientPrivate *priv = client->priv;
2276 GstRTSPMethod method;
2277 const gchar *uristr;
2278 GstRTSPUrl *uri = NULL;
2279 GstRTSPVersion version;
2281 GstRTSPSession *session = NULL;
2282 GstRTSPContext sctx = { NULL }, *ctx;
2283 GstRTSPMessage response = { 0 };
2284 gchar *unsupported_reqs = NULL;
2287 if (!(ctx = gst_rtsp_context_get_current ())) {
2289 ctx->auth = priv->auth;
2290 gst_rtsp_context_push_current (ctx);
2293 ctx->conn = priv->connection;
2294 ctx->client = client;
2295 ctx->request = request;
2296 ctx->response = &response;
2298 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2299 gst_rtsp_message_dump (request);
2302 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2304 GST_INFO ("client %p: received a request %s %s %s", client,
2305 gst_rtsp_method_as_text (method), uristr,
2306 gst_rtsp_version_as_text (version));
2308 /* we can only handle 1.0 requests */
2309 if (version != GST_RTSP_VERSION_1_0)
2312 ctx->method = method;
2314 /* we always try to parse the url first */
2315 if (strcmp (uristr, "*") == 0) {
2316 /* special case where we have * as uri, keep uri = NULL */
2317 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2318 /* check if the uristr is an absolute path <=> scheme and host information
2322 scheme = g_uri_parse_scheme (uristr);
2323 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2324 gchar *absolute_uristr = NULL;
2326 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2327 if (priv->server_ip == NULL) {
2328 GST_WARNING_OBJECT (client, "host information missing");
2333 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2335 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2336 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2337 g_free (absolute_uristr);
2340 g_free (absolute_uristr);
2347 /* get the session if there is any */
2348 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2349 if (res == GST_RTSP_OK) {
2350 if (priv->session_pool == NULL)
2353 /* we had a session in the request, find it again */
2354 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2355 goto session_not_found;
2357 /* we add the session to the client list of watched sessions. When a session
2358 * disappears because it times out, we will be notified. If all sessions are
2359 * gone, we will close the connection */
2360 client_watch_session (client, session);
2363 /* sanitize the uri */
2367 ctx->session = session;
2369 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2370 goto not_authorized;
2372 /* handle any 'Require' headers */
2373 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2374 goto unsupported_requirement;
2376 /* now see what is asked and dispatch to a dedicated handler */
2378 case GST_RTSP_OPTIONS:
2379 handle_options_request (client, ctx);
2381 case GST_RTSP_DESCRIBE:
2382 handle_describe_request (client, ctx);
2384 case GST_RTSP_SETUP:
2385 handle_setup_request (client, ctx);
2388 handle_play_request (client, ctx);
2390 case GST_RTSP_PAUSE:
2391 handle_pause_request (client, ctx);
2393 case GST_RTSP_TEARDOWN:
2394 handle_teardown_request (client, ctx);
2396 case GST_RTSP_SET_PARAMETER:
2397 handle_set_param_request (client, ctx);
2399 case GST_RTSP_GET_PARAMETER:
2400 handle_get_param_request (client, ctx);
2402 case GST_RTSP_ANNOUNCE:
2403 case GST_RTSP_RECORD:
2404 case GST_RTSP_REDIRECT:
2405 goto not_implemented;
2406 case GST_RTSP_INVALID:
2413 gst_rtsp_context_pop_current (ctx);
2415 g_object_unref (session);
2417 gst_rtsp_url_free (uri);
2423 GST_ERROR ("client %p: version %d not supported", client, version);
2424 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2430 GST_ERROR ("client %p: bad request", client);
2431 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2436 GST_ERROR ("client %p: no pool configured", client);
2437 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2442 GST_ERROR ("client %p: session not found", client);
2443 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2448 GST_ERROR ("client %p: not allowed", client);
2449 /* error reply is already sent */
2452 unsupported_requirement:
2454 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2456 send_option_not_supported_response (client, ctx, unsupported_reqs);
2457 g_free (unsupported_reqs);
2462 GST_ERROR ("client %p: method %d not implemented", client, method);
2463 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2470 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2472 GstRTSPClientPrivate *priv = client->priv;
2474 GstRTSPSession *session = NULL;
2475 GstRTSPContext sctx = { NULL }, *ctx;
2478 if (!(ctx = gst_rtsp_context_get_current ())) {
2480 ctx->auth = priv->auth;
2481 gst_rtsp_context_push_current (ctx);
2484 ctx->conn = priv->connection;
2485 ctx->client = client;
2486 ctx->request = NULL;
2488 ctx->method = GST_RTSP_INVALID;
2489 ctx->response = response;
2491 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2492 gst_rtsp_message_dump (response);
2495 GST_INFO ("client %p: received a response", client);
2497 /* get the session if there is any */
2499 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2500 if (res == GST_RTSP_OK) {
2501 if (priv->session_pool == NULL)
2504 /* we had a session in the request, find it again */
2505 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2506 goto session_not_found;
2508 /* we add the session to the client list of watched sessions. When a session
2509 * disappears because it times out, we will be notified. If all sessions are
2510 * gone, we will close the connection */
2511 client_watch_session (client, session);
2514 ctx->session = session;
2516 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2521 gst_rtsp_context_pop_current (ctx);
2523 g_object_unref (session);
2528 GST_ERROR ("client %p: no pool configured", client);
2533 GST_ERROR ("client %p: session not found", client);
2539 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2541 GstRTSPClientPrivate *priv = client->priv;
2547 GstRTSPStreamTransport *trans;
2549 /* find the stream for this message */
2550 res = gst_rtsp_message_parse_data (message, &channel);
2551 if (res != GST_RTSP_OK)
2554 gst_rtsp_message_steal_body (message, &data, &size);
2556 buffer = gst_buffer_new_wrapped (data, size);
2559 g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
2561 /* dispatch to the stream based on the channel number */
2562 gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
2564 gst_buffer_unref (buffer);
2569 * gst_rtsp_client_set_session_pool:
2570 * @client: a #GstRTSPClient
2571 * @pool: (transfer none): a #GstRTSPSessionPool
2573 * Set @pool as the sessionpool for @client which it will use to find
2574 * or allocate sessions. the sessionpool is usually inherited from the server
2575 * that created the client but can be overridden later.
2578 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
2579 GstRTSPSessionPool * pool)
2581 GstRTSPSessionPool *old;
2582 GstRTSPClientPrivate *priv;
2584 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2586 priv = client->priv;
2589 g_object_ref (pool);
2591 g_mutex_lock (&priv->lock);
2592 old = priv->session_pool;
2593 priv->session_pool = pool;
2595 if (priv->session_removed_id) {
2596 g_signal_handler_disconnect (old, priv->session_removed_id);
2597 priv->session_removed_id = 0;
2599 g_mutex_unlock (&priv->lock);
2601 /* FIXME, should remove all sessions from the old pool for this client */
2603 g_object_unref (old);
2607 * gst_rtsp_client_get_session_pool:
2608 * @client: a #GstRTSPClient
2610 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
2612 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
2614 GstRTSPSessionPool *
2615 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2617 GstRTSPClientPrivate *priv;
2618 GstRTSPSessionPool *result;
2620 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2622 priv = client->priv;
2624 g_mutex_lock (&priv->lock);
2625 if ((result = priv->session_pool))
2626 g_object_ref (result);
2627 g_mutex_unlock (&priv->lock);
2633 * gst_rtsp_client_set_mount_points:
2634 * @client: a #GstRTSPClient
2635 * @mounts: (transfer none): a #GstRTSPMountPoints
2637 * Set @mounts as the mount points for @client which it will use to map urls
2638 * to media streams. These mount points are usually inherited from the server that
2639 * created the client but can be overriden later.
2642 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
2643 GstRTSPMountPoints * mounts)
2645 GstRTSPClientPrivate *priv;
2646 GstRTSPMountPoints *old;
2648 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2650 priv = client->priv;
2653 g_object_ref (mounts);
2655 g_mutex_lock (&priv->lock);
2656 old = priv->mount_points;
2657 priv->mount_points = mounts;
2658 g_mutex_unlock (&priv->lock);
2661 g_object_unref (old);
2665 * gst_rtsp_client_get_mount_points:
2666 * @client: a #GstRTSPClient
2668 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
2670 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
2672 GstRTSPMountPoints *
2673 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
2675 GstRTSPClientPrivate *priv;
2676 GstRTSPMountPoints *result;
2678 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2680 priv = client->priv;
2682 g_mutex_lock (&priv->lock);
2683 if ((result = priv->mount_points))
2684 g_object_ref (result);
2685 g_mutex_unlock (&priv->lock);
2691 * gst_rtsp_client_set_auth:
2692 * @client: a #GstRTSPClient
2693 * @auth: (transfer none): a #GstRTSPAuth
2695 * configure @auth to be used as the authentication manager of @client.
2698 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
2700 GstRTSPClientPrivate *priv;
2703 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2705 priv = client->priv;
2708 g_object_ref (auth);
2710 g_mutex_lock (&priv->lock);
2713 g_mutex_unlock (&priv->lock);
2716 g_object_unref (old);
2721 * gst_rtsp_client_get_auth:
2722 * @client: a #GstRTSPClient
2724 * Get the #GstRTSPAuth used as the authentication manager of @client.
2726 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2730 gst_rtsp_client_get_auth (GstRTSPClient * client)
2732 GstRTSPClientPrivate *priv;
2733 GstRTSPAuth *result;
2735 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2737 priv = client->priv;
2739 g_mutex_lock (&priv->lock);
2740 if ((result = priv->auth))
2741 g_object_ref (result);
2742 g_mutex_unlock (&priv->lock);
2748 * gst_rtsp_client_set_thread_pool:
2749 * @client: a #GstRTSPClient
2750 * @pool: (transfer none): a #GstRTSPThreadPool
2752 * configure @pool to be used as the thread pool of @client.
2755 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
2756 GstRTSPThreadPool * pool)
2758 GstRTSPClientPrivate *priv;
2759 GstRTSPThreadPool *old;
2761 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2763 priv = client->priv;
2766 g_object_ref (pool);
2768 g_mutex_lock (&priv->lock);
2769 old = priv->thread_pool;
2770 priv->thread_pool = pool;
2771 g_mutex_unlock (&priv->lock);
2774 g_object_unref (old);
2778 * gst_rtsp_client_get_thread_pool:
2779 * @client: a #GstRTSPClient
2781 * Get the #GstRTSPThreadPool used as the thread pool of @client.
2783 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
2787 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
2789 GstRTSPClientPrivate *priv;
2790 GstRTSPThreadPool *result;
2792 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2794 priv = client->priv;
2796 g_mutex_lock (&priv->lock);
2797 if ((result = priv->thread_pool))
2798 g_object_ref (result);
2799 g_mutex_unlock (&priv->lock);
2805 * gst_rtsp_client_set_connection:
2806 * @client: a #GstRTSPClient
2807 * @conn: (transfer full): a #GstRTSPConnection
2809 * Set the #GstRTSPConnection of @client. This function takes ownership of
2812 * Returns: %TRUE on success.
2815 gst_rtsp_client_set_connection (GstRTSPClient * client,
2816 GstRTSPConnection * conn)
2818 GstRTSPClientPrivate *priv;
2819 GSocket *read_socket;
2820 GSocketAddress *address;
2822 GError *error = NULL;
2824 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2825 g_return_val_if_fail (conn != NULL, FALSE);
2827 priv = client->priv;
2829 read_socket = gst_rtsp_connection_get_read_socket (conn);
2831 if (!(address = g_socket_get_local_address (read_socket, &error)))
2834 g_free (priv->server_ip);
2835 /* keep the original ip that the client connected to */
2836 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2837 GInetAddress *iaddr;
2839 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2841 /* socket might be ipv6 but adress still ipv4 */
2842 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
2843 priv->server_ip = g_inet_address_to_string (iaddr);
2844 g_object_unref (address);
2846 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
2847 priv->server_ip = g_strdup ("unknown");
2850 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2851 priv->server_ip, priv->is_ipv6);
2853 url = gst_rtsp_connection_get_url (conn);
2854 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2856 priv->connection = conn;
2863 GST_ERROR ("could not get local address %s", error->message);
2864 g_error_free (error);
2870 * gst_rtsp_client_get_connection:
2871 * @client: a #GstRTSPClient
2873 * Get the #GstRTSPConnection of @client.
2875 * Returns: (transfer none): the #GstRTSPConnection of @client.
2876 * The connection object returned remains valid until the client is freed.
2879 gst_rtsp_client_get_connection (GstRTSPClient * client)
2881 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2883 return client->priv->connection;
2887 * gst_rtsp_client_set_send_func:
2888 * @client: a #GstRTSPClient
2889 * @func: (scope notified): a #GstRTSPClientSendFunc
2890 * @user_data: (closure): user data passed to @func
2891 * @notify: (allow-none): called when @user_data is no longer in use
2893 * Set @func as the callback that will be called when a new message needs to be
2894 * sent to the client. @user_data is passed to @func and @notify is called when
2895 * @user_data is no longer in use.
2897 * By default, the client will send the messages on the #GstRTSPConnection that
2898 * was configured with gst_rtsp_client_attach() was called.
2901 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2902 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2904 GstRTSPClientPrivate *priv;
2905 GDestroyNotify old_notify;
2908 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2910 priv = client->priv;
2912 g_mutex_lock (&priv->send_lock);
2913 priv->send_func = func;
2914 old_notify = priv->send_notify;
2915 old_data = priv->send_data;
2916 priv->send_notify = notify;
2917 priv->send_data = user_data;
2918 g_mutex_unlock (&priv->send_lock);
2921 old_notify (old_data);
2925 * gst_rtsp_client_handle_message:
2926 * @client: a #GstRTSPClient
2927 * @message: (transfer none): an #GstRTSPMessage
2929 * Let the client handle @message.
2931 * Returns: a #GstRTSPResult.
2934 gst_rtsp_client_handle_message (GstRTSPClient * client,
2935 GstRTSPMessage * message)
2937 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2938 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2940 switch (message->type) {
2941 case GST_RTSP_MESSAGE_REQUEST:
2942 handle_request (client, message);
2944 case GST_RTSP_MESSAGE_RESPONSE:
2945 handle_response (client, message);
2947 case GST_RTSP_MESSAGE_DATA:
2948 handle_data (client, message);
2957 * gst_rtsp_client_send_message:
2958 * @client: a #GstRTSPClient
2959 * @session: (allow-none) (transfer none): a #GstRTSPSession to send
2960 * the message to or %NULL
2961 * @message: (transfer none): The #GstRTSPMessage to send
2963 * Send a message message to the remote end. @message must be a
2964 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
2967 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
2968 GstRTSPMessage * message)
2970 GstRTSPContext sctx = { NULL }
2972 GstRTSPClientPrivate *priv;
2974 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2975 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2976 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
2977 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
2979 priv = client->priv;
2981 if (!(ctx = gst_rtsp_context_get_current ())) {
2983 ctx->auth = priv->auth;
2984 gst_rtsp_context_push_current (ctx);
2987 ctx->conn = priv->connection;
2988 ctx->client = client;
2989 ctx->session = session;
2991 send_message (client, ctx, message, FALSE);
2994 gst_rtsp_context_pop_current (ctx);
2999 static GstRTSPResult
3000 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
3001 gboolean close, gpointer user_data)
3003 GstRTSPClientPrivate *priv = client->priv;
3011 /* send the response and store the seq number so we can wait until it's
3012 * written to the client to close the connection */
3014 gst_rtsp_watch_send_message (priv->watch, message,
3015 close ? &priv->close_seq : NULL);
3016 if (ret == GST_RTSP_OK)
3019 if (ret != GST_RTSP_ENOMEM)
3023 if (priv->drop_backlog)
3026 /* queue was full, wait for more space */
3027 GST_DEBUG_OBJECT (client, "waiting for backlog");
3028 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3029 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3030 } while (ret != GST_RTSP_EINTR);
3037 GST_DEBUG_OBJECT (client, "got error %d", ret);
3042 static GstRTSPResult
3043 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3046 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3049 static GstRTSPResult
3050 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3052 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3053 GstRTSPClientPrivate *priv = client->priv;
3055 if (priv->close_seq && priv->close_seq == cseq) {
3056 GST_INFO ("client %p: send close message", client);
3057 priv->close_seq = 0;
3058 gst_rtsp_client_close (client);
3064 static GstRTSPResult
3065 closed (GstRTSPWatch * watch, gpointer user_data)
3067 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3068 GstRTSPClientPrivate *priv = client->priv;
3069 const gchar *tunnelid;
3071 GST_INFO ("client %p: connection closed", client);
3073 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3074 g_mutex_lock (&tunnels_lock);
3075 /* remove from tunnelids */
3076 g_hash_table_remove (tunnels, tunnelid);
3077 g_mutex_unlock (&tunnels_lock);
3080 gst_rtsp_watch_set_flushing (watch, TRUE);
3081 g_mutex_lock (&priv->watch_lock);
3082 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3083 g_mutex_unlock (&priv->watch_lock);
3088 static GstRTSPResult
3089 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3091 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3094 str = gst_rtsp_strresult (result);
3095 GST_INFO ("client %p: received an error %s", client, str);
3101 static GstRTSPResult
3102 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3103 GstRTSPMessage * message, guint id, gpointer user_data)
3105 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3108 str = gst_rtsp_strresult (result);
3110 ("client %p: error when handling message %p with id %d: %s",
3111 client, message, id, str);
3118 remember_tunnel (GstRTSPClient * client)
3120 GstRTSPClientPrivate *priv = client->priv;
3121 const gchar *tunnelid;
3123 /* store client in the pending tunnels */
3124 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3125 if (tunnelid == NULL)
3128 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3130 /* we can't have two clients connecting with the same tunnelid */
3131 g_mutex_lock (&tunnels_lock);
3132 if (g_hash_table_lookup (tunnels, tunnelid))
3133 goto tunnel_existed;
3135 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3136 g_mutex_unlock (&tunnels_lock);
3143 GST_ERROR ("client %p: no tunnelid provided", client);
3148 g_mutex_unlock (&tunnels_lock);
3149 GST_ERROR ("client %p: tunnel session %s already existed", client,
3155 static GstRTSPResult
3156 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3158 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3159 GstRTSPClientPrivate *priv = client->priv;
3161 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3164 /* ignore error, it'll only be a problem when the client does a POST again */
3165 remember_tunnel (client);
3171 handle_tunnel (GstRTSPClient * client)
3173 GstRTSPClientPrivate *priv = client->priv;
3174 GstRTSPClient *oclient;
3175 GstRTSPClientPrivate *opriv;
3176 const gchar *tunnelid;
3178 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3179 if (tunnelid == NULL)
3182 /* check for previous tunnel */
3183 g_mutex_lock (&tunnels_lock);
3184 oclient = g_hash_table_lookup (tunnels, tunnelid);
3186 if (oclient == NULL) {
3187 /* no previous tunnel, remember tunnel */
3188 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3189 g_mutex_unlock (&tunnels_lock);
3191 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3192 client, priv->connection);
3194 /* merge both tunnels into the first client */
3195 /* remove the old client from the table. ref before because removing it will
3196 * remove the ref to it. */
3197 g_object_ref (oclient);
3198 g_hash_table_remove (tunnels, tunnelid);
3199 g_mutex_unlock (&tunnels_lock);
3201 opriv = oclient->priv;
3203 g_mutex_lock (&opriv->watch_lock);
3204 if (opriv->watch == NULL)
3207 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3208 oclient, opriv->connection, priv->connection);
3210 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3211 gst_rtsp_watch_reset (priv->watch);
3212 gst_rtsp_watch_reset (opriv->watch);
3213 g_mutex_unlock (&opriv->watch_lock);
3214 g_object_unref (oclient);
3216 /* the old client owns the tunnel now, the new one will be freed */
3217 g_source_destroy ((GSource *) priv->watch);
3219 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3227 GST_ERROR ("client %p: no tunnelid provided", client);
3232 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3233 g_mutex_unlock (&opriv->watch_lock);
3234 g_object_unref (oclient);
3239 static GstRTSPStatusCode
3240 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3242 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3244 GST_INFO ("client %p: tunnel get (connection %p)", client,
3245 client->priv->connection);
3247 if (!handle_tunnel (client)) {
3248 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3251 return GST_RTSP_STS_OK;
3254 static GstRTSPResult
3255 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3257 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3259 GST_INFO ("client %p: tunnel post (connection %p)", client,
3260 client->priv->connection);
3262 if (!handle_tunnel (client)) {
3263 return GST_RTSP_ERROR;
3269 static GstRTSPResult
3270 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3271 GstRTSPMessage * response, gpointer user_data)
3273 GstRTSPClientClass *klass;
3275 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3276 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3278 if (klass->tunnel_http_response) {
3279 klass->tunnel_http_response (client, request, response);
3285 static GstRTSPWatchFuncs watch_funcs = {
3294 tunnel_http_response
3298 client_watch_notify (GstRTSPClient * client)
3300 GstRTSPClientPrivate *priv = client->priv;
3302 GST_INFO ("client %p: watch destroyed", client);
3304 g_main_context_unref (priv->watch_context);
3305 priv->watch_context = NULL;
3306 /* remove all sessions and so drop the extra client ref */
3307 gst_rtsp_client_session_filter (client, cleanup_session, NULL);
3308 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3309 g_object_unref (client);
3313 * gst_rtsp_client_attach:
3314 * @client: a #GstRTSPClient
3315 * @context: (allow-none): a #GMainContext
3317 * Attaches @client to @context. When the mainloop for @context is run, the
3318 * client will be dispatched. When @context is %NULL, the default context will be
3321 * This function should be called when the client properties and urls are fully
3322 * configured and the client is ready to start.
3324 * Returns: the ID (greater than 0) for the source within the GMainContext.
3327 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3329 GstRTSPClientPrivate *priv;
3332 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3333 priv = client->priv;
3334 g_return_val_if_fail (priv->connection != NULL, 0);
3335 g_return_val_if_fail (priv->watch == NULL, 0);
3337 /* make sure noone will free the context before the watch is destroyed */
3338 priv->watch_context = g_main_context_ref (context);
3340 /* create watch for the connection and attach */
3341 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3342 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3343 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3344 (GDestroyNotify) gst_rtsp_watch_unref);
3346 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
3348 GST_INFO ("client %p: attaching to context %p", client, context);
3349 res = gst_rtsp_watch_attach (priv->watch, context);
3355 * gst_rtsp_client_session_filter:
3356 * @client: a #GstRTSPClient
3357 * @func: (scope call) (allow-none): a callback
3358 * @user_data: user data passed to @func
3360 * Call @func for each session managed by @client. The result value of @func
3361 * determines what happens to the session. @func will be called with @client
3362 * locked so no further actions on @client can be performed from @func.
3364 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3367 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3369 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3370 * will also be added with an additional ref to the result #GList of this
3373 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3375 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3376 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3377 * element in the #GList should be unreffed before the list is freed.
3380 gst_rtsp_client_session_filter (GstRTSPClient * client,
3381 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3383 GstRTSPClientPrivate *priv;
3384 GList *result, *walk, *next;
3385 GHashTable *visited;
3388 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3390 priv = client->priv;
3394 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3396 g_mutex_lock (&priv->lock);
3398 cookie = priv->sessions_cookie;
3399 for (walk = priv->sessions; walk; walk = next) {
3400 GstRTSPSession *sess = walk->data;
3401 GstRTSPFilterResult res;
3404 next = g_list_next (walk);
3407 /* only visit each session once */
3408 if (g_hash_table_contains (visited, sess))
3411 g_hash_table_add (visited, g_object_ref (sess));
3412 g_mutex_unlock (&priv->lock);
3414 res = func (client, sess, user_data);
3416 g_mutex_lock (&priv->lock);
3418 res = GST_RTSP_FILTER_REF;
3420 changed = (cookie != priv->sessions_cookie);
3423 case GST_RTSP_FILTER_REMOVE:
3424 /* stop watching the session and pretend it went away, if the list was
3425 * changed, we can't use the current list position, try to see if we
3426 * still have the session */
3427 client_unwatch_session (client, sess, changed ? NULL : walk);
3428 cookie = priv->sessions_cookie;
3430 case GST_RTSP_FILTER_REF:
3431 result = g_list_prepend (result, g_object_ref (sess));
3433 case GST_RTSP_FILTER_KEEP:
3440 g_mutex_unlock (&priv->lock);
3443 g_hash_table_unref (visited);