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);
753 g_main_context_unref (priv->watch_context);
754 priv->watch_context = NULL;
759 default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
764 path = g_strconcat (uri->abspath, "?", uri->query, NULL);
766 path = g_strdup (uri->abspath);
772 handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
774 GstRTSPClientPrivate *priv = client->priv;
775 GstRTSPClientClass *klass;
776 GstRTSPSession *session;
777 GstRTSPSessionMedia *sessmedia;
778 GstRTSPStatusCode code;
781 gboolean keep_session;
786 session = ctx->session;
791 klass = GST_RTSP_CLIENT_GET_CLASS (client);
792 path = klass->make_path_from_uri (client, ctx->uri);
794 /* get a handle to the configuration of the media in the session */
795 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
799 /* only aggregate control for now.. */
800 if (path[matched] != '\0')
805 ctx->sessmedia = sessmedia;
807 /* we emit the signal before closing the connection */
808 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
811 /* make sure we unblock the backlog and don't accept new messages
813 if (priv->watch != NULL)
814 gst_rtsp_watch_set_flushing (priv->watch, TRUE);
816 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
818 /* allow messages again so that we can send the reply */
819 if (priv->watch != NULL)
820 gst_rtsp_watch_set_flushing (priv->watch, FALSE);
822 /* unmanage the media in the session, returns false if all media session
824 keep_session = gst_rtsp_session_release_media (session, sessmedia);
826 /* construct the response now */
827 code = GST_RTSP_STS_OK;
828 gst_rtsp_message_init_response (ctx->response, code,
829 gst_rtsp_status_as_text (code), ctx->request);
831 send_message (client, ctx, ctx->response, TRUE);
834 /* remove the session */
835 gst_rtsp_session_pool_remove (priv->session_pool, session);
843 GST_ERROR ("client %p: no session", client);
844 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
849 GST_ERROR ("client %p: no uri supplied", client);
850 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
855 GST_ERROR ("client %p: no media for uri", client);
856 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
862 GST_ERROR ("client %p: no aggregate path %s", client, path);
863 send_generic_response (client,
864 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
871 default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
875 res = gst_rtsp_params_set (client, ctx);
881 default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
885 res = gst_rtsp_params_get (client, ctx);
891 handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
897 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
898 if (res != GST_RTSP_OK)
902 /* no body, keep-alive request */
903 send_generic_response (client, GST_RTSP_STS_OK, ctx);
905 /* there is a body, handle the params */
906 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
907 if (res != GST_RTSP_OK)
910 send_message (client, ctx, ctx->response, FALSE);
913 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
921 GST_ERROR ("client %p: bad request", client);
922 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
928 handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
934 res = gst_rtsp_message_get_body (ctx->request, &data, &size);
935 if (res != GST_RTSP_OK)
939 /* no body, keep-alive request */
940 send_generic_response (client, GST_RTSP_STS_OK, ctx);
942 /* there is a body, handle the params */
943 res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
944 if (res != GST_RTSP_OK)
947 send_message (client, ctx, ctx->response, FALSE);
950 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
958 GST_ERROR ("client %p: bad request", client);
959 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
965 handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
967 GstRTSPSession *session;
968 GstRTSPClientClass *klass;
969 GstRTSPSessionMedia *sessmedia;
970 GstRTSPStatusCode code;
971 GstRTSPState rtspstate;
975 if (!(session = ctx->session))
981 klass = GST_RTSP_CLIENT_GET_CLASS (client);
982 path = klass->make_path_from_uri (client, ctx->uri);
984 /* get a handle to the configuration of the media in the session */
985 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
989 if (path[matched] != '\0')
994 ctx->sessmedia = sessmedia;
996 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
997 /* the session state must be playing or recording */
998 if (rtspstate != GST_RTSP_STATE_PLAYING &&
999 rtspstate != GST_RTSP_STATE_RECORDING)
1002 /* then pause sending */
1003 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
1005 /* construct the response now */
1006 code = GST_RTSP_STS_OK;
1007 gst_rtsp_message_init_response (ctx->response, code,
1008 gst_rtsp_status_as_text (code), ctx->request);
1010 send_message (client, ctx, ctx->response, FALSE);
1012 /* the state is now READY */
1013 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1015 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
1022 GST_ERROR ("client %p: no seesion", client);
1023 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1028 GST_ERROR ("client %p: no uri supplied", client);
1029 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1034 GST_ERROR ("client %p: no media for uri", client);
1035 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1041 GST_ERROR ("client %p: no aggregate path %s", client, path);
1042 send_generic_response (client,
1043 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1049 GST_ERROR ("client %p: not PLAYING or RECORDING", client);
1050 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1056 /* convert @url and @path to a URL used as a content base for the factory
1057 * located at @path */
1059 make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
1065 /* check for trailing '/' and append one */
1066 trail = (path[strlen (path) - 1] != '/' ? "/" : "");
1071 tmp.abspath = g_strdup_printf ("%s%s", path, trail);
1073 result = gst_rtsp_url_get_request_uri (&tmp);
1074 g_free (tmp.abspath);
1080 handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
1082 GstRTSPSession *session;
1083 GstRTSPClientClass *klass;
1084 GstRTSPSessionMedia *sessmedia;
1085 GstRTSPMedia *media;
1086 GstRTSPStatusCode code;
1089 GstRTSPTimeRange *range;
1091 GstRTSPState rtspstate;
1092 GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
1093 gchar *path, *rtpinfo;
1096 if (!(session = ctx->session))
1099 if (!(uri = ctx->uri))
1102 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1103 path = klass->make_path_from_uri (client, uri);
1105 /* get a handle to the configuration of the media in the session */
1106 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1110 if (path[matched] != '\0')
1115 ctx->sessmedia = sessmedia;
1116 ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
1118 /* the session state must be playing or ready */
1119 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1120 if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
1123 /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
1124 if (!gst_rtsp_media_unsuspend (media))
1125 goto unsuspend_failed;
1127 /* parse the range header if we have one */
1128 res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
1129 if (res == GST_RTSP_OK) {
1130 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
1131 /* we have a range, seek to the position */
1133 gst_rtsp_media_seek (media, range);
1134 gst_rtsp_range_free (range);
1138 /* grab RTPInfo from the media now */
1139 rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
1141 /* construct the response now */
1142 code = GST_RTSP_STS_OK;
1143 gst_rtsp_message_init_response (ctx->response, code,
1144 gst_rtsp_status_as_text (code), ctx->request);
1146 /* add the RTP-Info header */
1148 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
1152 str = gst_rtsp_media_get_range_string (media, TRUE, unit);
1154 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
1156 send_message (client, ctx, ctx->response, FALSE);
1158 /* start playing after sending the response */
1159 gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
1161 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
1163 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
1170 GST_ERROR ("client %p: no session", client);
1171 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1176 GST_ERROR ("client %p: no uri supplied", client);
1177 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1182 GST_ERROR ("client %p: media not found", client);
1183 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1188 GST_ERROR ("client %p: no aggregate path %s", client, path);
1189 send_generic_response (client,
1190 GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
1196 GST_ERROR ("client %p: not PLAYING or READY", client);
1197 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
1203 GST_ERROR ("client %p: unsuspend failed", client);
1204 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1210 do_keepalive (GstRTSPSession * session)
1212 GST_INFO ("keep session %p alive", session);
1213 gst_rtsp_session_touch (session);
1216 /* parse @transport and return a valid transport in @tr. only transports
1217 * supported by @stream are returned. Returns FALSE if no valid transport
1220 parse_transport (const char *transport, GstRTSPStream * stream,
1221 GstRTSPTransport * tr)
1228 gst_rtsp_transport_init (tr);
1230 GST_DEBUG ("parsing transports %s", transport);
1232 transports = g_strsplit (transport, ",", 0);
1234 /* loop through the transports, try to parse */
1235 for (i = 0; transports[i]; i++) {
1236 res = gst_rtsp_transport_parse (transports[i], tr);
1237 if (res != GST_RTSP_OK) {
1238 /* no valid transport, search some more */
1239 GST_WARNING ("could not parse transport %s", transports[i]);
1243 /* we have a transport, see if it's supported */
1244 if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
1245 GST_WARNING ("unsupported transport %s", transports[i]);
1249 /* we have a valid transport */
1250 GST_INFO ("found valid transport %s", transports[i]);
1255 gst_rtsp_transport_init (tr);
1257 g_strfreev (transports);
1263 default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
1264 GstRTSPStream * stream, GstRTSPContext * ctx)
1266 GstRTSPMessage *request = ctx->request;
1267 gchar *blocksize_str;
1269 if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
1270 &blocksize_str, 0) == GST_RTSP_OK) {
1274 blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
1275 if (end == blocksize_str)
1278 /* we don't want to change the mtu when this media
1279 * can be shared because it impacts other clients */
1280 if (gst_rtsp_media_is_shared (media))
1283 if (blocksize > G_MAXUINT)
1284 blocksize = G_MAXUINT;
1286 gst_rtsp_stream_set_mtu (stream, blocksize);
1294 GST_ERROR_OBJECT (client, "failed to parse blocksize");
1295 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1301 default_configure_client_transport (GstRTSPClient * client,
1302 GstRTSPContext * ctx, GstRTSPTransport * ct)
1304 GstRTSPClientPrivate *priv = client->priv;
1306 /* we have a valid transport now, set the destination of the client. */
1307 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1308 gboolean use_client_settings;
1310 use_client_settings =
1311 gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
1313 if (ct->destination && use_client_settings) {
1314 GstRTSPAddress *addr;
1316 addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
1317 ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
1322 gst_rtsp_address_free (addr);
1324 GstRTSPAddress *addr;
1325 GSocketFamily family;
1327 family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
1329 addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
1333 g_free (ct->destination);
1334 ct->destination = g_strdup (addr->address);
1335 ct->port.min = addr->port;
1336 ct->port.max = addr->port + addr->n_ports - 1;
1337 ct->ttl = addr->ttl;
1339 gst_rtsp_address_free (addr);
1344 url = gst_rtsp_connection_get_url (priv->connection);
1345 g_free (ct->destination);
1346 ct->destination = g_strdup (url->host);
1348 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
1350 GSocketAddress *addr;
1352 sock = gst_rtsp_connection_get_read_socket (priv->connection);
1353 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1354 /* our read port is the sender port of client */
1355 ct->client_port.min =
1356 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1357 g_object_unref (addr);
1359 if ((addr = g_socket_get_local_address (sock, NULL))) {
1360 ct->server_port.max =
1361 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1362 g_object_unref (addr);
1364 sock = gst_rtsp_connection_get_write_socket (priv->connection);
1365 if ((addr = g_socket_get_remote_address (sock, NULL))) {
1366 /* our write port is the receiver port of client */
1367 ct->client_port.max =
1368 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1369 g_object_unref (addr);
1371 if ((addr = g_socket_get_local_address (sock, NULL))) {
1372 ct->server_port.min =
1373 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
1374 g_object_unref (addr);
1376 /* check if the client selected channels for TCP */
1377 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
1378 gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
1388 GST_ERROR_OBJECT (client, "failed to acquire address for stream");
1393 static GstRTSPTransport *
1394 make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
1395 GstRTSPTransport * ct)
1397 GstRTSPTransport *st;
1399 GSocketFamily family;
1401 /* prepare the server transport */
1402 gst_rtsp_transport_new (&st);
1404 st->trans = ct->trans;
1405 st->profile = ct->profile;
1406 st->lower_transport = ct->lower_transport;
1408 addr = g_inet_address_new_from_string (ct->destination);
1411 GST_ERROR ("failed to get inet addr from client destination");
1412 family = G_SOCKET_FAMILY_IPV4;
1414 family = g_inet_address_get_family (addr);
1415 g_object_unref (addr);
1419 switch (st->lower_transport) {
1420 case GST_RTSP_LOWER_TRANS_UDP:
1421 st->client_port = ct->client_port;
1422 gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
1424 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1425 st->port = ct->port;
1426 st->destination = g_strdup (ct->destination);
1429 case GST_RTSP_LOWER_TRANS_TCP:
1430 st->interleaved = ct->interleaved;
1431 st->client_port = ct->client_port;
1432 st->server_port = ct->server_port;
1437 gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
1442 #define AES_128_KEY_LEN 16
1443 #define AES_256_KEY_LEN 32
1445 #define HMAC_32_KEY_LEN 4
1446 #define HMAC_80_KEY_LEN 10
1449 mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
1451 const gchar *srtp_cipher;
1452 const gchar *srtp_auth;
1453 const GstMIKEYPayload *sp;
1456 /* loop over Security policy until we find one containing policy */
1458 if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
1461 if (((GstMIKEYPayloadSP *) sp)->policy == policy)
1465 /* the default ciphers */
1466 srtp_cipher = "aes-128-icm";
1467 srtp_auth = "hmac-sha1-80";
1469 /* now override the defaults with what is in the Security Policy */
1473 /* collect all the params and go over them */
1474 len = gst_mikey_payload_sp_get_n_params (sp);
1475 for (i = 0; i < len; i++) {
1476 const GstMIKEYPayloadSPParam *param =
1477 gst_mikey_payload_sp_get_param (sp, i);
1479 switch (param->type) {
1480 case GST_MIKEY_SP_SRTP_ENC_ALG:
1481 switch (param->val[0]) {
1483 srtp_cipher = "null";
1487 srtp_cipher = "aes-128-icm";
1493 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1494 switch (param->val[0]) {
1495 case AES_128_KEY_LEN:
1496 srtp_cipher = "aes-128-icm";
1498 case AES_256_KEY_LEN:
1499 srtp_cipher = "aes-256-icm";
1505 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1506 switch (param->val[0]) {
1512 srtp_auth = "hmac-sha1-80";
1518 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1519 switch (param->val[0]) {
1520 case HMAC_32_KEY_LEN:
1521 srtp_auth = "hmac-sha1-32";
1523 case HMAC_80_KEY_LEN:
1524 srtp_auth = "hmac-sha1-80";
1530 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1532 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1539 /* now configure the SRTP parameters */
1540 gst_caps_set_simple (caps,
1541 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1542 "srtp-auth", G_TYPE_STRING, srtp_auth,
1543 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1544 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1550 handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
1551 guint8 * data, gsize size)
1553 GstMIKEYMessage *msg;
1555 GstCaps *caps = NULL;
1556 GstMIKEYPayloadKEMAC *kemac;
1557 const GstMIKEYPayloadKeyData *pkd;
1560 /* the MIKEY message contains a CSB or crypto session bundle. It is a
1561 * set of Crypto Sessions protected with the same master key.
1562 * In the context of SRTP, an RTP and its RTCP stream is part of a
1564 if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
1567 /* we can only handle SRTP crypto sessions for now */
1568 if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
1569 goto invalid_map_type;
1571 /* get the number of crypto sessions. This maps SSRC to its
1572 * security parameters */
1573 n_cs = gst_mikey_message_get_n_cs (msg);
1575 goto no_crypto_sessions;
1577 /* we also need keys */
1578 if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
1579 (msg, GST_MIKEY_PT_KEMAC, 0)))
1582 /* we don't support encrypted keys */
1583 if (kemac->enc_alg != GST_MIKEY_ENC_NULL
1584 || kemac->mac_alg != GST_MIKEY_MAC_NULL)
1585 goto unsupported_encryption;
1587 /* get Key data sub-payload */
1588 pkd = (const GstMIKEYPayloadKeyData *)
1589 gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
1592 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1595 /* go over all crypto sessions and create the security policy for each
1597 for (i = 0; i < n_cs; i++) {
1598 const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
1600 caps = gst_caps_new_simple ("application/x-srtp",
1601 "ssrc", G_TYPE_UINT, map->ssrc,
1602 "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
1603 mikey_apply_policy (caps, msg, map->policy);
1605 gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
1606 gst_caps_unref (caps);
1608 gst_mikey_message_unref (msg);
1615 GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
1620 GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
1621 goto cleanup_message;
1625 GST_DEBUG_OBJECT (client, "no crypto sessions");
1626 goto cleanup_message;
1630 GST_DEBUG_OBJECT (client, "no keys found");
1631 goto cleanup_message;
1633 unsupported_encryption:
1635 GST_DEBUG_OBJECT (client, "unsupported key encryption");
1636 goto cleanup_message;
1640 gst_mikey_message_unref (msg);
1645 #define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
1648 strip_chars (gchar * str)
1655 if (!IS_STRIP_CHAR (str[len]))
1659 for (s = str; *s && IS_STRIP_CHAR (*s); s++);
1660 memmove (str, s, len + 1);
1663 /* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
1664 * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
1667 handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
1672 specs = g_strsplit (keymgmt, ",", 0);
1673 for (i = 0; specs[i]; i++) {
1676 split = g_strsplit (specs[i], ";", 0);
1677 for (j = 0; split[j]; j++) {
1678 g_strstrip (split[j]);
1679 if (g_str_has_prefix (split[j], "prot=")) {
1680 g_strstrip (split[j] + 5);
1681 if (!g_str_equal (split[j] + 5, "mikey"))
1683 GST_DEBUG ("found mikey");
1684 } else if (g_str_has_prefix (split[j], "uri=")) {
1685 strip_chars (split[j] + 4);
1686 GST_DEBUG ("found uri '%s'", split[j] + 4);
1687 } else if (g_str_has_prefix (split[j], "data=")) {
1690 strip_chars (split[j] + 5);
1691 GST_DEBUG ("found data '%s'", split[j] + 5);
1692 data = g_base64_decode_inplace (split[j] + 5, &size);
1693 handle_mikey_data (client, ctx, data, size);
1701 handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
1703 GstRTSPClientPrivate *priv = client->priv;
1706 gchar *transport, *keymgmt;
1707 GstRTSPTransport *ct, *st;
1708 GstRTSPStatusCode code;
1709 GstRTSPSession *session;
1710 GstRTSPStreamTransport *trans;
1712 GstRTSPSessionMedia *sessmedia;
1713 GstRTSPMedia *media;
1714 GstRTSPStream *stream;
1715 GstRTSPState rtspstate;
1716 GstRTSPClientClass *klass;
1717 gchar *path, *control;
1719 gboolean new_session = FALSE;
1725 klass = GST_RTSP_CLIENT_GET_CLASS (client);
1726 path = klass->make_path_from_uri (client, uri);
1728 /* parse the transport */
1730 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
1732 if (res != GST_RTSP_OK)
1735 /* we create the session after parsing stuff so that we don't make
1736 * a session for malformed requests */
1737 if (priv->session_pool == NULL)
1740 session = ctx->session;
1743 g_object_ref (session);
1744 /* get a handle to the configuration of the media in the session, this can
1745 * return NULL if this is a new url to manage in this session. */
1746 sessmedia = gst_rtsp_session_get_media (session, path, &matched);
1748 /* we need a new media configuration in this session */
1752 /* we have no session media, find one and manage it */
1753 if (sessmedia == NULL) {
1754 /* get a handle to the configuration of the media in the session */
1755 media = find_media (client, ctx, path, &matched);
1757 if ((media = gst_rtsp_session_media_get_media (sessmedia)))
1758 g_object_ref (media);
1760 goto media_not_found;
1762 /* no media, not found then */
1764 goto media_not_found_no_reply;
1766 if (path[matched] == '\0')
1767 goto control_not_found;
1769 /* path is what matched. */
1770 path[matched] = '\0';
1771 /* control is remainder */
1772 control = &path[matched + 1];
1774 /* find the stream now using the control part */
1775 stream = gst_rtsp_media_find_stream (media, control);
1777 goto stream_not_found;
1779 /* now we have a uri identifying a valid media and stream */
1780 ctx->stream = stream;
1783 if (session == NULL) {
1784 /* create a session if this fails we probably reached our session limit or
1786 if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
1787 goto service_unavailable;
1789 /* make sure this client is closed when the session is closed */
1790 client_watch_session (client, session);
1793 /* signal new session */
1794 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
1797 ctx->session = session;
1800 if (!klass->configure_client_media (client, media, stream, ctx))
1801 goto configure_media_failed_no_reply;
1803 gst_rtsp_transport_new (&ct);
1805 /* parse and find a usable supported transport */
1806 if (!parse_transport (transport, stream, ct))
1807 goto unsupported_transports;
1809 /* update the client transport */
1810 if (!klass->configure_client_transport (client, ctx, ct))
1811 goto unsupported_client_transport;
1813 /* parse the keymgmt */
1814 if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
1815 &keymgmt, 0) == GST_RTSP_OK) {
1816 if (!handle_keymgmt (client, ctx, keymgmt))
1820 if (sessmedia == NULL) {
1821 /* manage the media in our session now, if not done already */
1822 sessmedia = gst_rtsp_session_manage_media (session, path, media);
1823 /* if we stil have no media, error */
1824 if (sessmedia == NULL)
1825 goto sessmedia_unavailable;
1827 g_object_unref (media);
1830 ctx->sessmedia = sessmedia;
1832 /* set in the session media transport */
1833 trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
1835 /* configure the url used to set this transport, this we will use when
1836 * generating the response for the PLAY request */
1837 gst_rtsp_stream_transport_set_url (trans, uri);
1838 /* configure keepalive for this transport */
1839 gst_rtsp_stream_transport_set_keepalive (trans,
1840 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
1842 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1843 /* our callbacks to send data on this TCP connection */
1844 gst_rtsp_stream_transport_set_callbacks (trans,
1845 (GstRTSPSendFunc) do_send_data,
1846 (GstRTSPSendFunc) do_send_data, client, NULL);
1848 g_hash_table_insert (priv->transports,
1849 GINT_TO_POINTER (ct->interleaved.min), trans);
1850 g_hash_table_insert (priv->transports,
1851 GINT_TO_POINTER (ct->interleaved.max), trans);
1854 /* create and serialize the server transport */
1855 st = make_server_transport (client, ctx, ct);
1856 trans_str = gst_rtsp_transport_as_text (st);
1857 gst_rtsp_transport_free (st);
1859 /* construct the response now */
1860 code = GST_RTSP_STS_OK;
1861 gst_rtsp_message_init_response (ctx->response, code,
1862 gst_rtsp_status_as_text (code), ctx->request);
1864 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
1868 send_message (client, ctx, ctx->response, FALSE);
1870 /* update the state */
1871 rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
1872 switch (rtspstate) {
1873 case GST_RTSP_STATE_PLAYING:
1874 case GST_RTSP_STATE_RECORDING:
1875 case GST_RTSP_STATE_READY:
1876 /* no state change */
1879 gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
1882 g_object_unref (session);
1885 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
1892 GST_ERROR ("client %p: no uri", client);
1893 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
1898 GST_ERROR ("client %p: no transport", client);
1899 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1904 GST_ERROR ("client %p: no session pool configured", client);
1905 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
1908 media_not_found_no_reply:
1910 GST_ERROR ("client %p: media '%s' not found", client, path);
1911 /* error reply is already sent */
1916 GST_ERROR ("client %p: media '%s' not found", client, path);
1917 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1922 GST_ERROR ("client %p: no control in path '%s'", client, path);
1923 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1924 g_object_unref (media);
1929 GST_ERROR ("client %p: stream '%s' not found", client, control);
1930 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
1931 g_object_unref (media);
1934 service_unavailable:
1936 GST_ERROR ("client %p: can't create session", client);
1937 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1938 g_object_unref (media);
1941 sessmedia_unavailable:
1943 GST_ERROR ("client %p: can't create session media", client);
1944 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
1945 g_object_unref (media);
1946 goto cleanup_session;
1948 configure_media_failed_no_reply:
1950 GST_ERROR ("client %p: configure_media failed", client);
1951 /* error reply is already sent */
1952 goto cleanup_session;
1954 unsupported_transports:
1956 GST_ERROR ("client %p: unsupported transports", client);
1957 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1958 goto cleanup_transport;
1960 unsupported_client_transport:
1962 GST_ERROR ("client %p: unsupported client transport", client);
1963 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
1964 goto cleanup_transport;
1968 GST_ERROR ("client %p: keymgmt error", client);
1969 send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
1970 goto cleanup_transport;
1974 gst_rtsp_transport_free (ct);
1977 gst_rtsp_session_pool_remove (priv->session_pool, session);
1978 g_object_unref (session);
1985 static GstSDPMessage *
1986 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1988 GstRTSPClientPrivate *priv = client->priv;
1993 gst_sdp_message_new (&sdp);
1995 /* some standard things first */
1996 gst_sdp_message_set_version (sdp, "0");
2003 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
2006 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
2007 gst_sdp_message_set_information (sdp, "rtsp-server");
2008 gst_sdp_message_add_time (sdp, "0", "0", NULL);
2009 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
2010 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
2011 gst_sdp_message_add_attribute (sdp, "control", "*");
2013 info.is_ipv6 = priv->is_ipv6;
2014 info.server_ip = priv->server_ip;
2016 /* create an SDP for the media object */
2017 if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
2025 GST_ERROR ("client %p: could not create SDP", client);
2026 gst_sdp_message_free (sdp);
2031 /* for the describe we must generate an SDP */
2033 handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
2035 GstRTSPClientPrivate *priv = client->priv;
2040 GstRTSPMedia *media;
2041 GstRTSPClientClass *klass;
2043 klass = GST_RTSP_CLIENT_GET_CLASS (client);
2048 /* check what kind of format is accepted, we don't really do anything with it
2049 * and always return SDP for now. */
2054 gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
2056 if (res == GST_RTSP_ENOTIMPL)
2059 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
2063 if (!priv->mount_points)
2064 goto no_mount_points;
2066 if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
2069 /* find the media object for the uri */
2070 if (!(media = find_media (client, ctx, path, NULL)))
2073 /* create an SDP for the media object on this client */
2074 if (!(sdp = klass->create_sdp (client, media)))
2077 /* we suspend after the describe */
2078 gst_rtsp_media_suspend (media);
2079 g_object_unref (media);
2081 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2082 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2084 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
2087 /* content base for some clients that might screw up creating the setup uri */
2088 str = make_base_url (client, ctx->uri, path);
2091 GST_INFO ("adding content-base: %s", str);
2092 gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
2094 /* add SDP to the response body */
2095 str = gst_sdp_message_as_text (sdp);
2096 gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
2097 gst_sdp_message_free (sdp);
2099 send_message (client, ctx, ctx->response, FALSE);
2101 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
2109 GST_ERROR ("client %p: no uri", client);
2110 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2115 GST_ERROR ("client %p: no mount points configured", client);
2116 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2121 GST_ERROR ("client %p: can't find path for url", client);
2122 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
2127 GST_ERROR ("client %p: no media", client);
2129 /* error reply is already sent */
2134 GST_ERROR ("client %p: can't create SDP", client);
2135 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
2137 g_object_unref (media);
2143 handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
2145 GstRTSPMethod options;
2148 options = GST_RTSP_DESCRIBE |
2153 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
2155 str = gst_rtsp_options_as_text (options);
2157 gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
2158 gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
2160 gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
2163 send_message (client, ctx, ctx->response, FALSE);
2165 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
2171 /* remove duplicate and trailing '/' */
2173 sanitize_uri (GstRTSPUrl * uri)
2177 gboolean have_slash, prev_slash;
2179 s = d = uri->abspath;
2180 len = strlen (uri->abspath);
2184 for (i = 0; i < len; i++) {
2185 have_slash = s[i] == '/';
2187 if (!have_slash || !prev_slash)
2189 prev_slash = have_slash;
2191 len = d - uri->abspath;
2192 /* don't remove the first slash if that's the only thing left */
2193 if (len > 1 && *(d - 1) == '/')
2198 /* is called when the session is removed from its session pool. */
2200 client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
2201 GstRTSPClient * client)
2203 GstRTSPClientPrivate *priv = client->priv;
2205 GST_INFO ("client %p: session %p removed", client, session);
2207 g_mutex_lock (&priv->lock);
2208 if (priv->watch != NULL)
2209 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
2210 client_unwatch_session (client, session, NULL);
2211 if (priv->watch != NULL)
2212 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2213 g_mutex_unlock (&priv->lock);
2216 /* Returns TRUE if there are no Require headers, otherwise returns FALSE
2217 * and also returns a newly-allocated string of (comma-separated) unsupported
2218 * options in the unsupported_reqs variable .
2220 * There may be multiple Require headers, but we must send one single
2221 * Unsupported header with all the unsupported options as response. If
2222 * an incoming Require header contained a comma-separated list of options
2223 * GstRtspConnection will already have split that list up into multiple
2226 * TODO: allow the application to decide what features are supported
2229 check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
2232 GPtrArray *arr = NULL;
2238 res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
2240 if (res == GST_RTSP_ENOTIMPL)
2244 arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
2246 g_ptr_array_add (arr, g_strdup (reqs));
2250 /* if we don't have any Require headers at all, all is fine */
2254 /* otherwise we've now processed at all the Require headers */
2255 g_ptr_array_add (arr, NULL);
2257 /* for now we don't commit to supporting anything, so will just report
2258 * all of the required options as unsupported */
2259 *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
2261 g_ptr_array_unref (arr);
2266 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2268 GstRTSPClientPrivate *priv = client->priv;
2269 GstRTSPMethod method;
2270 const gchar *uristr;
2271 GstRTSPUrl *uri = NULL;
2272 GstRTSPVersion version;
2274 GstRTSPSession *session = NULL;
2275 GstRTSPContext sctx = { NULL }, *ctx;
2276 GstRTSPMessage response = { 0 };
2277 gchar *unsupported_reqs = NULL;
2280 if (!(ctx = gst_rtsp_context_get_current ())) {
2282 ctx->auth = priv->auth;
2283 gst_rtsp_context_push_current (ctx);
2286 ctx->conn = priv->connection;
2287 ctx->client = client;
2288 ctx->request = request;
2289 ctx->response = &response;
2291 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2292 gst_rtsp_message_dump (request);
2295 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
2297 GST_INFO ("client %p: received a request %s %s %s", client,
2298 gst_rtsp_method_as_text (method), uristr,
2299 gst_rtsp_version_as_text (version));
2301 /* we can only handle 1.0 requests */
2302 if (version != GST_RTSP_VERSION_1_0)
2305 ctx->method = method;
2307 /* we always try to parse the url first */
2308 if (strcmp (uristr, "*") == 0) {
2309 /* special case where we have * as uri, keep uri = NULL */
2310 } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
2311 /* check if the uristr is an absolute path <=> scheme and host information
2315 scheme = g_uri_parse_scheme (uristr);
2316 if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
2317 gchar *absolute_uristr = NULL;
2319 GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
2320 if (priv->server_ip == NULL) {
2321 GST_WARNING_OBJECT (client, "host information missing");
2326 g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
2328 GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
2329 if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
2330 g_free (absolute_uristr);
2333 g_free (absolute_uristr);
2340 /* get the session if there is any */
2341 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
2342 if (res == GST_RTSP_OK) {
2343 if (priv->session_pool == NULL)
2346 /* we had a session in the request, find it again */
2347 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2348 goto session_not_found;
2350 /* we add the session to the client list of watched sessions. When a session
2351 * disappears because it times out, we will be notified. If all sessions are
2352 * gone, we will close the connection */
2353 client_watch_session (client, session);
2356 /* sanitize the uri */
2360 ctx->session = session;
2362 if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
2363 goto not_authorized;
2365 /* handle any 'Require' headers */
2366 if (!check_request_requirements (ctx->request, &unsupported_reqs))
2367 goto unsupported_requirement;
2369 /* the backlog must be unlimited while processing requests.
2370 * the causes of this are two cases of deadlocks while streaming over TCP:
2372 * 1. consider the scenario where the media pipeline's streaming thread
2373 * is blocking in the appsink (taking the appsink's preroll lock) because
2374 * the backlog is full. when a PAUSE request is received by the RTSP
2375 * client thread then the the state of the session media ought to change
2376 * to PAUSED. while most elements in the pipeline can change state this
2377 * can never happen for the appsink since its preroll lock is taken by
2380 * 2. consider the scenario where the media pipeline's streaming thread
2381 * is blocking in the appsink new_sample callback (taking the send lock
2382 * in RTSP client) because the backlog is full. when e.g. a GET request
2383 * is received by the RTSP client thread then a response ought to be sent
2384 * but this can never happen since it requires taking the send lock
2385 * already taken by another thread.
2387 * the reason that the backlog is never emptied is that the source used
2388 * for dequeing messages from the backlog is never dispatched because it
2389 * is attached to the same mainloop as the source receving RTSP requests and
2390 * therefore run by the RTSP client thread which is alreayd blocking.
2392 * without significant changes the easiest way to cope with this is to
2393 * not block indefinitely when the backlog is full, but rather let the
2394 * backlog grow in size. this in effect means that there can not be any
2395 * upper boundary on its size.
2397 if (priv->watch != NULL)
2398 gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
2400 /* now see what is asked and dispatch to a dedicated handler */
2402 case GST_RTSP_OPTIONS:
2403 handle_options_request (client, ctx);
2405 case GST_RTSP_DESCRIBE:
2406 handle_describe_request (client, ctx);
2408 case GST_RTSP_SETUP:
2409 handle_setup_request (client, ctx);
2412 handle_play_request (client, ctx);
2414 case GST_RTSP_PAUSE:
2415 handle_pause_request (client, ctx);
2417 case GST_RTSP_TEARDOWN:
2418 handle_teardown_request (client, ctx);
2420 case GST_RTSP_SET_PARAMETER:
2421 handle_set_param_request (client, ctx);
2423 case GST_RTSP_GET_PARAMETER:
2424 handle_get_param_request (client, ctx);
2426 case GST_RTSP_ANNOUNCE:
2427 case GST_RTSP_RECORD:
2428 case GST_RTSP_REDIRECT:
2429 if (priv->watch != NULL)
2430 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2431 goto not_implemented;
2432 case GST_RTSP_INVALID:
2434 if (priv->watch != NULL)
2435 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2439 if (priv->watch != NULL)
2440 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
2444 gst_rtsp_context_pop_current (ctx);
2446 g_object_unref (session);
2448 gst_rtsp_url_free (uri);
2454 GST_ERROR ("client %p: version %d not supported", client, version);
2455 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
2461 GST_ERROR ("client %p: bad request", client);
2462 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
2467 GST_ERROR ("client %p: no pool configured", client);
2468 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2473 GST_ERROR ("client %p: session not found", client);
2474 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
2479 GST_ERROR ("client %p: not allowed", client);
2480 /* error reply is already sent */
2483 unsupported_requirement:
2485 GST_ERROR ("client %p: Required option is not supported (%s)", client,
2487 send_option_not_supported_response (client, ctx, unsupported_reqs);
2488 g_free (unsupported_reqs);
2493 GST_ERROR ("client %p: method %d not implemented", client, method);
2494 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
2501 handle_response (GstRTSPClient * client, GstRTSPMessage * response)
2503 GstRTSPClientPrivate *priv = client->priv;
2505 GstRTSPSession *session = NULL;
2506 GstRTSPContext sctx = { NULL }, *ctx;
2509 if (!(ctx = gst_rtsp_context_get_current ())) {
2511 ctx->auth = priv->auth;
2512 gst_rtsp_context_push_current (ctx);
2515 ctx->conn = priv->connection;
2516 ctx->client = client;
2517 ctx->request = NULL;
2519 ctx->method = GST_RTSP_INVALID;
2520 ctx->response = response;
2522 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
2523 gst_rtsp_message_dump (response);
2526 GST_INFO ("client %p: received a response", client);
2528 /* get the session if there is any */
2530 gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
2531 if (res == GST_RTSP_OK) {
2532 if (priv->session_pool == NULL)
2535 /* we had a session in the request, find it again */
2536 if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
2537 goto session_not_found;
2539 /* we add the session to the client list of watched sessions. When a session
2540 * disappears because it times out, we will be notified. If all sessions are
2541 * gone, we will close the connection */
2542 client_watch_session (client, session);
2545 ctx->session = session;
2547 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
2552 gst_rtsp_context_pop_current (ctx);
2554 g_object_unref (session);
2559 GST_ERROR ("client %p: no pool configured", client);
2564 GST_ERROR ("client %p: session not found", client);
2570 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
2572 GstRTSPClientPrivate *priv = client->priv;
2578 GstRTSPStreamTransport *trans;
2580 /* find the stream for this message */
2581 res = gst_rtsp_message_parse_data (message, &channel);
2582 if (res != GST_RTSP_OK)
2585 gst_rtsp_message_steal_body (message, &data, &size);
2587 buffer = gst_buffer_new_wrapped (data, size);
2590 g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
2592 /* dispatch to the stream based on the channel number */
2593 gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
2595 gst_buffer_unref (buffer);
2600 * gst_rtsp_client_set_session_pool:
2601 * @client: a #GstRTSPClient
2602 * @pool: (transfer none): a #GstRTSPSessionPool
2604 * Set @pool as the sessionpool for @client which it will use to find
2605 * or allocate sessions. the sessionpool is usually inherited from the server
2606 * that created the client but can be overridden later.
2609 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
2610 GstRTSPSessionPool * pool)
2612 GstRTSPSessionPool *old;
2613 GstRTSPClientPrivate *priv;
2615 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2617 priv = client->priv;
2620 g_object_ref (pool);
2622 g_mutex_lock (&priv->lock);
2623 old = priv->session_pool;
2624 priv->session_pool = pool;
2626 if (priv->session_removed_id) {
2627 g_signal_handler_disconnect (old, priv->session_removed_id);
2628 priv->session_removed_id = 0;
2630 g_mutex_unlock (&priv->lock);
2632 /* FIXME, should remove all sessions from the old pool for this client */
2634 g_object_unref (old);
2638 * gst_rtsp_client_get_session_pool:
2639 * @client: a #GstRTSPClient
2641 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
2643 * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
2645 GstRTSPSessionPool *
2646 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2648 GstRTSPClientPrivate *priv;
2649 GstRTSPSessionPool *result;
2651 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2653 priv = client->priv;
2655 g_mutex_lock (&priv->lock);
2656 if ((result = priv->session_pool))
2657 g_object_ref (result);
2658 g_mutex_unlock (&priv->lock);
2664 * gst_rtsp_client_set_mount_points:
2665 * @client: a #GstRTSPClient
2666 * @mounts: (transfer none): a #GstRTSPMountPoints
2668 * Set @mounts as the mount points for @client which it will use to map urls
2669 * to media streams. These mount points are usually inherited from the server that
2670 * created the client but can be overriden later.
2673 gst_rtsp_client_set_mount_points (GstRTSPClient * client,
2674 GstRTSPMountPoints * mounts)
2676 GstRTSPClientPrivate *priv;
2677 GstRTSPMountPoints *old;
2679 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2681 priv = client->priv;
2684 g_object_ref (mounts);
2686 g_mutex_lock (&priv->lock);
2687 old = priv->mount_points;
2688 priv->mount_points = mounts;
2689 g_mutex_unlock (&priv->lock);
2692 g_object_unref (old);
2696 * gst_rtsp_client_get_mount_points:
2697 * @client: a #GstRTSPClient
2699 * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
2701 * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
2703 GstRTSPMountPoints *
2704 gst_rtsp_client_get_mount_points (GstRTSPClient * client)
2706 GstRTSPClientPrivate *priv;
2707 GstRTSPMountPoints *result;
2709 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2711 priv = client->priv;
2713 g_mutex_lock (&priv->lock);
2714 if ((result = priv->mount_points))
2715 g_object_ref (result);
2716 g_mutex_unlock (&priv->lock);
2722 * gst_rtsp_client_set_auth:
2723 * @client: a #GstRTSPClient
2724 * @auth: (transfer none): a #GstRTSPAuth
2726 * configure @auth to be used as the authentication manager of @client.
2729 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
2731 GstRTSPClientPrivate *priv;
2734 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2736 priv = client->priv;
2739 g_object_ref (auth);
2741 g_mutex_lock (&priv->lock);
2744 g_mutex_unlock (&priv->lock);
2747 g_object_unref (old);
2752 * gst_rtsp_client_get_auth:
2753 * @client: a #GstRTSPClient
2755 * Get the #GstRTSPAuth used as the authentication manager of @client.
2757 * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
2761 gst_rtsp_client_get_auth (GstRTSPClient * client)
2763 GstRTSPClientPrivate *priv;
2764 GstRTSPAuth *result;
2766 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2768 priv = client->priv;
2770 g_mutex_lock (&priv->lock);
2771 if ((result = priv->auth))
2772 g_object_ref (result);
2773 g_mutex_unlock (&priv->lock);
2779 * gst_rtsp_client_set_thread_pool:
2780 * @client: a #GstRTSPClient
2781 * @pool: (transfer none): a #GstRTSPThreadPool
2783 * configure @pool to be used as the thread pool of @client.
2786 gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
2787 GstRTSPThreadPool * pool)
2789 GstRTSPClientPrivate *priv;
2790 GstRTSPThreadPool *old;
2792 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2794 priv = client->priv;
2797 g_object_ref (pool);
2799 g_mutex_lock (&priv->lock);
2800 old = priv->thread_pool;
2801 priv->thread_pool = pool;
2802 g_mutex_unlock (&priv->lock);
2805 g_object_unref (old);
2809 * gst_rtsp_client_get_thread_pool:
2810 * @client: a #GstRTSPClient
2812 * Get the #GstRTSPThreadPool used as the thread pool of @client.
2814 * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
2818 gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
2820 GstRTSPClientPrivate *priv;
2821 GstRTSPThreadPool *result;
2823 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2825 priv = client->priv;
2827 g_mutex_lock (&priv->lock);
2828 if ((result = priv->thread_pool))
2829 g_object_ref (result);
2830 g_mutex_unlock (&priv->lock);
2836 * gst_rtsp_client_set_connection:
2837 * @client: a #GstRTSPClient
2838 * @conn: (transfer full): a #GstRTSPConnection
2840 * Set the #GstRTSPConnection of @client. This function takes ownership of
2843 * Returns: %TRUE on success.
2846 gst_rtsp_client_set_connection (GstRTSPClient * client,
2847 GstRTSPConnection * conn)
2849 GstRTSPClientPrivate *priv;
2850 GSocket *read_socket;
2851 GSocketAddress *address;
2853 GError *error = NULL;
2855 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
2856 g_return_val_if_fail (conn != NULL, FALSE);
2858 priv = client->priv;
2860 read_socket = gst_rtsp_connection_get_read_socket (conn);
2862 if (!(address = g_socket_get_local_address (read_socket, &error)))
2865 g_free (priv->server_ip);
2866 /* keep the original ip that the client connected to */
2867 if (G_IS_INET_SOCKET_ADDRESS (address)) {
2868 GInetAddress *iaddr;
2870 iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
2872 /* socket might be ipv6 but adress still ipv4 */
2873 priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
2874 priv->server_ip = g_inet_address_to_string (iaddr);
2875 g_object_unref (address);
2877 priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
2878 priv->server_ip = g_strdup ("unknown");
2881 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
2882 priv->server_ip, priv->is_ipv6);
2884 url = gst_rtsp_connection_get_url (conn);
2885 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
2887 priv->connection = conn;
2894 GST_ERROR ("could not get local address %s", error->message);
2895 g_error_free (error);
2901 * gst_rtsp_client_get_connection:
2902 * @client: a #GstRTSPClient
2904 * Get the #GstRTSPConnection of @client.
2906 * Returns: (transfer none): the #GstRTSPConnection of @client.
2907 * The connection object returned remains valid until the client is freed.
2910 gst_rtsp_client_get_connection (GstRTSPClient * client)
2912 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
2914 return client->priv->connection;
2918 * gst_rtsp_client_set_send_func:
2919 * @client: a #GstRTSPClient
2920 * @func: (scope notified): a #GstRTSPClientSendFunc
2921 * @user_data: (closure): user data passed to @func
2922 * @notify: (allow-none): called when @user_data is no longer in use
2924 * Set @func as the callback that will be called when a new message needs to be
2925 * sent to the client. @user_data is passed to @func and @notify is called when
2926 * @user_data is no longer in use.
2928 * By default, the client will send the messages on the #GstRTSPConnection that
2929 * was configured with gst_rtsp_client_attach() was called.
2932 gst_rtsp_client_set_send_func (GstRTSPClient * client,
2933 GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
2935 GstRTSPClientPrivate *priv;
2936 GDestroyNotify old_notify;
2939 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
2941 priv = client->priv;
2943 g_mutex_lock (&priv->send_lock);
2944 priv->send_func = func;
2945 old_notify = priv->send_notify;
2946 old_data = priv->send_data;
2947 priv->send_notify = notify;
2948 priv->send_data = user_data;
2949 g_mutex_unlock (&priv->send_lock);
2952 old_notify (old_data);
2956 * gst_rtsp_client_handle_message:
2957 * @client: a #GstRTSPClient
2958 * @message: (transfer none): an #GstRTSPMessage
2960 * Let the client handle @message.
2962 * Returns: a #GstRTSPResult.
2965 gst_rtsp_client_handle_message (GstRTSPClient * client,
2966 GstRTSPMessage * message)
2968 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
2969 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
2971 switch (message->type) {
2972 case GST_RTSP_MESSAGE_REQUEST:
2973 handle_request (client, message);
2975 case GST_RTSP_MESSAGE_RESPONSE:
2976 handle_response (client, message);
2978 case GST_RTSP_MESSAGE_DATA:
2979 handle_data (client, message);
2988 * gst_rtsp_client_send_message:
2989 * @client: a #GstRTSPClient
2990 * @session: (allow-none) (transfer none): a #GstRTSPSession to send
2991 * the message to or %NULL
2992 * @message: (transfer none): The #GstRTSPMessage to send
2994 * Send a message message to the remote end. @message must be a
2995 * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
2998 gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
2999 GstRTSPMessage * message)
3001 GstRTSPContext sctx = { NULL }
3003 GstRTSPClientPrivate *priv;
3005 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
3006 g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
3007 g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
3008 message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
3010 priv = client->priv;
3012 if (!(ctx = gst_rtsp_context_get_current ())) {
3014 ctx->auth = priv->auth;
3015 gst_rtsp_context_push_current (ctx);
3018 ctx->conn = priv->connection;
3019 ctx->client = client;
3020 ctx->session = session;
3022 send_message (client, ctx, message, FALSE);
3025 gst_rtsp_context_pop_current (ctx);
3030 static GstRTSPResult
3031 do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
3032 gboolean close, gpointer user_data)
3034 GstRTSPClientPrivate *priv = client->priv;
3042 /* send the response and store the seq number so we can wait until it's
3043 * written to the client to close the connection */
3045 gst_rtsp_watch_send_message (priv->watch, message,
3046 close ? &priv->close_seq : NULL);
3047 if (ret == GST_RTSP_OK)
3050 if (ret != GST_RTSP_ENOMEM)
3054 if (priv->drop_backlog)
3057 /* queue was full, wait for more space */
3058 GST_DEBUG_OBJECT (client, "waiting for backlog");
3059 ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
3060 GST_DEBUG_OBJECT (client, "Resend due to backlog full");
3061 } while (ret != GST_RTSP_EINTR);
3068 GST_DEBUG_OBJECT (client, "got error %d", ret);
3073 static GstRTSPResult
3074 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
3077 return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
3080 static GstRTSPResult
3081 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
3083 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3084 GstRTSPClientPrivate *priv = client->priv;
3086 if (priv->close_seq && priv->close_seq == cseq) {
3087 GST_INFO ("client %p: send close message", client);
3088 priv->close_seq = 0;
3089 gst_rtsp_client_close (client);
3095 static GstRTSPResult
3096 closed (GstRTSPWatch * watch, gpointer user_data)
3098 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3099 GstRTSPClientPrivate *priv = client->priv;
3100 const gchar *tunnelid;
3102 GST_INFO ("client %p: connection closed", client);
3104 if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
3105 g_mutex_lock (&tunnels_lock);
3106 /* remove from tunnelids */
3107 g_hash_table_remove (tunnels, tunnelid);
3108 g_mutex_unlock (&tunnels_lock);
3111 gst_rtsp_watch_set_flushing (watch, TRUE);
3112 g_mutex_lock (&priv->watch_lock);
3113 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3114 g_mutex_unlock (&priv->watch_lock);
3119 static GstRTSPResult
3120 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
3122 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3125 str = gst_rtsp_strresult (result);
3126 GST_INFO ("client %p: received an error %s", client, str);
3132 static GstRTSPResult
3133 error_full (GstRTSPWatch * watch, GstRTSPResult result,
3134 GstRTSPMessage * message, guint id, gpointer user_data)
3136 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3139 str = gst_rtsp_strresult (result);
3141 ("client %p: error when handling message %p with id %d: %s",
3142 client, message, id, str);
3149 remember_tunnel (GstRTSPClient * client)
3151 GstRTSPClientPrivate *priv = client->priv;
3152 const gchar *tunnelid;
3154 /* store client in the pending tunnels */
3155 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3156 if (tunnelid == NULL)
3159 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
3161 /* we can't have two clients connecting with the same tunnelid */
3162 g_mutex_lock (&tunnels_lock);
3163 if (g_hash_table_lookup (tunnels, tunnelid))
3164 goto tunnel_existed;
3166 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3167 g_mutex_unlock (&tunnels_lock);
3174 GST_ERROR ("client %p: no tunnelid provided", client);
3179 g_mutex_unlock (&tunnels_lock);
3180 GST_ERROR ("client %p: tunnel session %s already existed", client,
3186 static GstRTSPResult
3187 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
3189 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3190 GstRTSPClientPrivate *priv = client->priv;
3192 GST_WARNING ("client %p: tunnel lost (connection %p)", client,
3195 /* ignore error, it'll only be a problem when the client does a POST again */
3196 remember_tunnel (client);
3202 handle_tunnel (GstRTSPClient * client)
3204 GstRTSPClientPrivate *priv = client->priv;
3205 GstRTSPClient *oclient;
3206 GstRTSPClientPrivate *opriv;
3207 const gchar *tunnelid;
3209 tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
3210 if (tunnelid == NULL)
3213 /* check for previous tunnel */
3214 g_mutex_lock (&tunnels_lock);
3215 oclient = g_hash_table_lookup (tunnels, tunnelid);
3217 if (oclient == NULL) {
3218 /* no previous tunnel, remember tunnel */
3219 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
3220 g_mutex_unlock (&tunnels_lock);
3222 GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
3223 client, priv->connection);
3225 /* merge both tunnels into the first client */
3226 /* remove the old client from the table. ref before because removing it will
3227 * remove the ref to it. */
3228 g_object_ref (oclient);
3229 g_hash_table_remove (tunnels, tunnelid);
3230 g_mutex_unlock (&tunnels_lock);
3232 opriv = oclient->priv;
3234 g_mutex_lock (&opriv->watch_lock);
3235 if (opriv->watch == NULL)
3238 GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
3239 oclient, opriv->connection, priv->connection);
3241 gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
3242 gst_rtsp_watch_reset (priv->watch);
3243 gst_rtsp_watch_reset (opriv->watch);
3244 g_mutex_unlock (&opriv->watch_lock);
3245 g_object_unref (oclient);
3247 /* the old client owns the tunnel now, the new one will be freed */
3248 g_source_destroy ((GSource *) priv->watch);
3250 gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
3258 GST_ERROR ("client %p: no tunnelid provided", client);
3263 GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
3264 g_mutex_unlock (&opriv->watch_lock);
3265 g_object_unref (oclient);
3270 static GstRTSPStatusCode
3271 tunnel_get (GstRTSPWatch * watch, gpointer user_data)
3273 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3275 GST_INFO ("client %p: tunnel get (connection %p)", client,
3276 client->priv->connection);
3278 if (!handle_tunnel (client)) {
3279 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
3282 return GST_RTSP_STS_OK;
3285 static GstRTSPResult
3286 tunnel_post (GstRTSPWatch * watch, gpointer user_data)
3288 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3290 GST_INFO ("client %p: tunnel post (connection %p)", client,
3291 client->priv->connection);
3293 if (!handle_tunnel (client)) {
3294 return GST_RTSP_ERROR;
3300 static GstRTSPResult
3301 tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
3302 GstRTSPMessage * response, gpointer user_data)
3304 GstRTSPClientClass *klass;
3306 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
3307 klass = GST_RTSP_CLIENT_GET_CLASS (client);
3309 if (klass->tunnel_http_response) {
3310 klass->tunnel_http_response (client, request, response);
3316 static GstRTSPWatchFuncs watch_funcs = {
3325 tunnel_http_response
3329 client_watch_notify (GstRTSPClient * client)
3331 GstRTSPClientPrivate *priv = client->priv;
3333 GST_INFO ("client %p: watch destroyed", client);
3335 /* remove all sessions and so drop the extra client ref */
3336 gst_rtsp_client_session_filter (client, cleanup_session, NULL);
3337 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
3338 g_object_unref (client);
3342 * gst_rtsp_client_attach:
3343 * @client: a #GstRTSPClient
3344 * @context: (allow-none): a #GMainContext
3346 * Attaches @client to @context. When the mainloop for @context is run, the
3347 * client will be dispatched. When @context is %NULL, the default context will be
3350 * This function should be called when the client properties and urls are fully
3351 * configured and the client is ready to start.
3353 * Returns: the ID (greater than 0) for the source within the GMainContext.
3356 gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
3358 GstRTSPClientPrivate *priv;
3361 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
3362 priv = client->priv;
3363 g_return_val_if_fail (priv->connection != NULL, 0);
3364 g_return_val_if_fail (priv->watch == NULL, 0);
3366 /* make sure noone will free the context before the watch is destroyed */
3367 priv->watch_context = g_main_context_ref (context);
3369 /* create watch for the connection and attach */
3370 priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
3371 g_object_ref (client), (GDestroyNotify) client_watch_notify);
3372 gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
3373 (GDestroyNotify) gst_rtsp_watch_unref);
3375 gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
3377 GST_INFO ("client %p: attaching to context %p", client, context);
3378 res = gst_rtsp_watch_attach (priv->watch, context);
3384 * gst_rtsp_client_session_filter:
3385 * @client: a #GstRTSPClient
3386 * @func: (scope call) (allow-none): a callback
3387 * @user_data: user data passed to @func
3389 * Call @func for each session managed by @client. The result value of @func
3390 * determines what happens to the session. @func will be called with @client
3391 * locked so no further actions on @client can be performed from @func.
3393 * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
3396 * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
3398 * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
3399 * will also be added with an additional ref to the result #GList of this
3402 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
3404 * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
3405 * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3406 * element in the #GList should be unreffed before the list is freed.
3409 gst_rtsp_client_session_filter (GstRTSPClient * client,
3410 GstRTSPClientSessionFilterFunc func, gpointer user_data)
3412 GstRTSPClientPrivate *priv;
3413 GList *result, *walk, *next;
3414 GHashTable *visited;
3417 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
3419 priv = client->priv;
3423 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3425 g_mutex_lock (&priv->lock);
3427 cookie = priv->sessions_cookie;
3428 for (walk = priv->sessions; walk; walk = next) {
3429 GstRTSPSession *sess = walk->data;
3430 GstRTSPFilterResult res;
3433 next = g_list_next (walk);
3436 /* only visit each session once */
3437 if (g_hash_table_contains (visited, sess))
3440 g_hash_table_add (visited, g_object_ref (sess));
3441 g_mutex_unlock (&priv->lock);
3443 res = func (client, sess, user_data);
3445 g_mutex_lock (&priv->lock);
3447 res = GST_RTSP_FILTER_REF;
3449 changed = (cookie != priv->sessions_cookie);
3452 case GST_RTSP_FILTER_REMOVE:
3453 /* stop watching the session and pretend it went away, if the list was
3454 * changed, we can't use the current list position, try to see if we
3455 * still have the session */
3456 client_unwatch_session (client, sess, changed ? NULL : walk);
3457 cookie = priv->sessions_cookie;
3459 case GST_RTSP_FILTER_REF:
3460 result = g_list_prepend (result, g_object_ref (sess));
3462 case GST_RTSP_FILTER_KEEP:
3469 g_mutex_unlock (&priv->lock);
3472 g_hash_table_unref (visited);