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 media stream
22 * @see_also: #GstRTSPMedia
24 * The #GstRTSPStream object manages the data transport for one stream. It
25 * is created from a payloader element and a source pad that produce the RTP
26 * packets for the stream.
28 * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
29 * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
31 * The #GstRTSPStream will use the configured addresspool, as set with
32 * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
33 * stream. With gst_rtsp_stream_get_multicast_address() you can get the
36 * With gst_rtsp_stream_get_server_port () you can get the port that the server
37 * will use to receive RTCP. This is the part that the clients will use to send
40 * With gst_rtsp_stream_add_transport() destinations can be added where the
41 * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
42 * the destination again.
44 * Last reviewed on 2013-07-16 (1.0.0)
53 #include <gst/app/gstappsrc.h>
54 #include <gst/app/gstappsink.h>
56 #include "rtsp-stream.h"
58 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
59 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
61 struct _GstRTSPStreamPrivate
66 GstElement *payloader;
71 GstRTSPProfile profiles;
72 GstRTSPLowerTrans protocols;
74 /* pads on the rtpbin */
75 GstPad *send_rtp_sink;
79 /* the RTPSession object */
82 /* SRTP encoder/decoder */
87 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
89 GstElement *udpsrc_v4[2];
91 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
93 GstElement *udpsrc_v6[2];
95 GstElement *udpsink[2];
97 /* for TCP transport */
98 GstElement *appsrc[2];
99 GstElement *appqueue[2];
100 GstElement *appsink[2];
103 GstElement *funnel[2];
105 /* server ports for sending/receiving over ipv4 */
106 GstRTSPRange server_port_v4;
107 GstRTSPAddress *server_addr_v4;
110 /* server ports for sending/receiving over ipv6 */
111 GstRTSPRange server_port_v6;
112 GstRTSPAddress *server_addr_v6;
115 /* multicast addresses */
116 GstRTSPAddressPool *pool;
117 GstRTSPAddress *addr_v4;
118 GstRTSPAddress *addr_v6;
120 /* the caps of the stream */
124 /* transports we stream to */
132 /* stream blocking */
137 #define DEFAULT_CONTROL NULL
138 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
139 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
140 GST_RTSP_LOWER_TRANS_TCP
151 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
152 #define GST_CAT_DEFAULT rtsp_stream_debug
154 static GQuark ssrc_stream_map_key;
156 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
157 GValue * value, GParamSpec * pspec);
158 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
159 const GValue * value, GParamSpec * pspec);
161 static void gst_rtsp_stream_finalize (GObject * obj);
163 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
166 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
168 GObjectClass *gobject_class;
170 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
172 gobject_class = G_OBJECT_CLASS (klass);
174 gobject_class->get_property = gst_rtsp_stream_get_property;
175 gobject_class->set_property = gst_rtsp_stream_set_property;
176 gobject_class->finalize = gst_rtsp_stream_finalize;
178 g_object_class_install_property (gobject_class, PROP_CONTROL,
179 g_param_spec_string ("control", "Control",
180 "The control string for this stream", DEFAULT_CONTROL,
181 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
183 g_object_class_install_property (gobject_class, PROP_PROFILES,
184 g_param_spec_flags ("profiles", "Profiles",
185 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
186 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
188 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
189 g_param_spec_flags ("protocols", "Protocols",
190 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
191 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
193 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
195 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
199 gst_rtsp_stream_init (GstRTSPStream * stream)
201 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
203 GST_DEBUG ("new stream %p", stream);
208 priv->control = g_strdup (DEFAULT_CONTROL);
209 priv->profiles = DEFAULT_PROFILES;
210 priv->protocols = DEFAULT_PROTOCOLS;
212 g_mutex_init (&priv->lock);
214 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
215 NULL, (GDestroyNotify) gst_caps_unref);
219 gst_rtsp_stream_finalize (GObject * obj)
221 GstRTSPStream *stream;
222 GstRTSPStreamPrivate *priv;
224 stream = GST_RTSP_STREAM (obj);
227 GST_DEBUG ("finalize stream %p", stream);
229 /* we really need to be unjoined now */
230 g_return_if_fail (!priv->is_joined);
233 gst_rtsp_address_free (priv->addr_v4);
235 gst_rtsp_address_free (priv->addr_v6);
236 if (priv->server_addr_v4)
237 gst_rtsp_address_free (priv->server_addr_v4);
238 if (priv->server_addr_v6)
239 gst_rtsp_address_free (priv->server_addr_v6);
241 g_object_unref (priv->pool);
242 gst_object_unref (priv->payloader);
243 gst_object_unref (priv->srcpad);
244 g_free (priv->control);
245 g_mutex_clear (&priv->lock);
247 g_hash_table_unref (priv->keys);
249 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
253 gst_rtsp_stream_get_property (GObject * object, guint propid,
254 GValue * value, GParamSpec * pspec)
256 GstRTSPStream *stream = GST_RTSP_STREAM (object);
260 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
263 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
266 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
269 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
274 gst_rtsp_stream_set_property (GObject * object, guint propid,
275 const GValue * value, GParamSpec * pspec)
277 GstRTSPStream *stream = GST_RTSP_STREAM (object);
281 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
284 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
287 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
290 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
295 * gst_rtsp_stream_new:
298 * @payloader: a #GstElement
300 * Create a new media stream with index @idx that handles RTP data on
301 * @srcpad and has a payloader element @payloader.
303 * Returns: (transfer full): a new #GstRTSPStream
306 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
308 GstRTSPStreamPrivate *priv;
309 GstRTSPStream *stream;
311 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
312 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
313 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
315 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
318 priv->payloader = gst_object_ref (payloader);
319 priv->srcpad = gst_object_ref (srcpad);
325 * gst_rtsp_stream_get_index:
326 * @stream: a #GstRTSPStream
328 * Get the stream index.
330 * Return: the stream index.
333 gst_rtsp_stream_get_index (GstRTSPStream * stream)
335 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
337 return stream->priv->idx;
341 * gst_rtsp_stream_get_pt:
342 * @stream: a #GstRTSPStream
344 * Get the stream payload type.
346 * Return: the stream payload type.
349 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
351 GstRTSPStreamPrivate *priv;
354 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
358 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
364 * gst_rtsp_stream_get_srcpad:
365 * @stream: a #GstRTSPStream
367 * Get the srcpad associated with @stream.
369 * Returns: (transfer full): the srcpad. Unref after usage.
372 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
374 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
376 return gst_object_ref (stream->priv->srcpad);
380 * gst_rtsp_stream_get_control:
381 * @stream: a #GstRTSPStream
383 * Get the control string to identify this stream.
385 * Returns: (transfer full): the control string. g_free() after usage.
388 gst_rtsp_stream_get_control (GstRTSPStream * stream)
390 GstRTSPStreamPrivate *priv;
393 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
397 g_mutex_lock (&priv->lock);
398 if ((result = g_strdup (priv->control)) == NULL)
399 result = g_strdup_printf ("stream=%u", priv->idx);
400 g_mutex_unlock (&priv->lock);
406 * gst_rtsp_stream_set_control:
407 * @stream: a #GstRTSPStream
408 * @control: a control string
410 * Set the control string in @stream.
413 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
415 GstRTSPStreamPrivate *priv;
417 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
421 g_mutex_lock (&priv->lock);
422 g_free (priv->control);
423 priv->control = g_strdup (control);
424 g_mutex_unlock (&priv->lock);
428 * gst_rtsp_stream_has_control:
429 * @stream: a #GstRTSPStream
430 * @control: a control string
432 * Check if @stream has the control string @control.
434 * Returns: %TRUE is @stream has @control as the control string
437 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
439 GstRTSPStreamPrivate *priv;
442 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
446 g_mutex_lock (&priv->lock);
448 res = (g_strcmp0 (priv->control, control) == 0);
452 if (sscanf (control, "stream=%u", &streamid) > 0)
453 res = (streamid == priv->idx);
457 g_mutex_unlock (&priv->lock);
463 * gst_rtsp_stream_set_mtu:
464 * @stream: a #GstRTSPStream
467 * Configure the mtu in the payloader of @stream to @mtu.
470 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
472 GstRTSPStreamPrivate *priv;
474 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
478 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
480 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
484 * gst_rtsp_stream_get_mtu:
485 * @stream: a #GstRTSPStream
487 * Get the configured MTU in the payloader of @stream.
489 * Returns: the MTU of the payloader.
492 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
494 GstRTSPStreamPrivate *priv;
497 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
501 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
506 /* Update the dscp qos property on the udp sinks */
508 update_dscp_qos (GstRTSPStream * stream)
510 GstRTSPStreamPrivate *priv;
512 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
516 if (priv->udpsink[0]) {
517 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
521 if (priv->udpsink[1]) {
522 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
528 * gst_rtsp_stream_set_dscp_qos:
529 * @stream: a #GstRTSPStream
530 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
532 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
535 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
537 GstRTSPStreamPrivate *priv;
539 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
543 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
545 if (dscp_qos < -1 || dscp_qos > 63) {
546 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
550 priv->dscp_qos = dscp_qos;
552 update_dscp_qos (stream);
556 * gst_rtsp_stream_get_dscp_qos:
557 * @stream: a #GstRTSPStream
559 * Get the configured DSCP QoS in of the outgoing sockets.
561 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
564 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
566 GstRTSPStreamPrivate *priv;
568 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
572 return priv->dscp_qos;
576 * gst_rtsp_stream_is_transport_supported:
577 * @stream: a #GstRTSPStream
578 * @transport: (transfer none): a #GstRTSPTransport
580 * Check if @transport can be handled by stream
582 * Returns: %TRUE if @transport can be handled by @stream.
585 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
586 GstRTSPTransport * transport)
588 GstRTSPStreamPrivate *priv;
590 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
594 g_mutex_lock (&priv->lock);
595 if (transport->trans != GST_RTSP_TRANS_RTP)
596 goto unsupported_transmode;
598 if (!(transport->profile & priv->profiles))
599 goto unsupported_profile;
601 if (!(transport->lower_transport & priv->protocols))
602 goto unsupported_ltrans;
604 g_mutex_unlock (&priv->lock);
609 unsupported_transmode:
611 GST_DEBUG ("unsupported transport mode %d", transport->trans);
612 g_mutex_unlock (&priv->lock);
617 GST_DEBUG ("unsupported profile %d", transport->profile);
618 g_mutex_unlock (&priv->lock);
623 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
624 g_mutex_unlock (&priv->lock);
630 * gst_rtsp_stream_set_profiles:
631 * @stream: a #GstRTSPStream
632 * @profiles: the new profiles
634 * Configure the allowed profiles for @stream.
637 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
639 GstRTSPStreamPrivate *priv;
641 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
645 g_mutex_lock (&priv->lock);
646 priv->profiles = profiles;
647 g_mutex_unlock (&priv->lock);
651 * gst_rtsp_stream_get_profiles:
652 * @stream: a #GstRTSPStream
654 * Get the allowed profiles of @stream.
656 * Returns: a #GstRTSPProfile
659 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
661 GstRTSPStreamPrivate *priv;
664 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
668 g_mutex_lock (&priv->lock);
669 res = priv->profiles;
670 g_mutex_unlock (&priv->lock);
676 * gst_rtsp_stream_set_protocols:
677 * @stream: a #GstRTSPStream
678 * @protocols: the new flags
680 * Configure the allowed lower transport for @stream.
683 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
684 GstRTSPLowerTrans protocols)
686 GstRTSPStreamPrivate *priv;
688 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
692 g_mutex_lock (&priv->lock);
693 priv->protocols = protocols;
694 g_mutex_unlock (&priv->lock);
698 * gst_rtsp_stream_get_protocols:
699 * @stream: a #GstRTSPStream
701 * Get the allowed protocols of @stream.
703 * Returns: a #GstRTSPLowerTrans
706 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
708 GstRTSPStreamPrivate *priv;
709 GstRTSPLowerTrans res;
711 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
712 GST_RTSP_LOWER_TRANS_UNKNOWN);
716 g_mutex_lock (&priv->lock);
717 res = priv->protocols;
718 g_mutex_unlock (&priv->lock);
724 * gst_rtsp_stream_set_address_pool:
725 * @stream: a #GstRTSPStream
726 * @pool: (transfer none): a #GstRTSPAddressPool
728 * configure @pool to be used as the address pool of @stream.
731 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
732 GstRTSPAddressPool * pool)
734 GstRTSPStreamPrivate *priv;
735 GstRTSPAddressPool *old;
737 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
741 GST_LOG_OBJECT (stream, "set address pool %p", pool);
743 g_mutex_lock (&priv->lock);
744 if ((old = priv->pool) != pool)
745 priv->pool = pool ? g_object_ref (pool) : NULL;
748 g_mutex_unlock (&priv->lock);
751 g_object_unref (old);
755 * gst_rtsp_stream_get_address_pool:
756 * @stream: a #GstRTSPStream
758 * Get the #GstRTSPAddressPool used as the address pool of @stream.
760 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
764 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
766 GstRTSPStreamPrivate *priv;
767 GstRTSPAddressPool *result;
769 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
773 g_mutex_lock (&priv->lock);
774 if ((result = priv->pool))
775 g_object_ref (result);
776 g_mutex_unlock (&priv->lock);
782 * gst_rtsp_stream_get_multicast_address:
783 * @stream: a #GstRTSPStream
784 * @family: the #GSocketFamily
786 * Get the multicast address of @stream for @family.
788 * Returns: (transfer full): the #GstRTSPAddress of @stream or %NULL when no
789 * address could be allocated. gst_rtsp_address_free() after usage.
792 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
793 GSocketFamily family)
795 GstRTSPStreamPrivate *priv;
796 GstRTSPAddress *result;
797 GstRTSPAddress **addrp;
798 GstRTSPAddressFlags flags;
800 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
804 if (family == G_SOCKET_FAMILY_IPV6) {
805 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
806 addrp = &priv->addr_v6;
808 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
809 addrp = &priv->addr_v4;
812 g_mutex_lock (&priv->lock);
813 if (*addrp == NULL) {
814 if (priv->pool == NULL)
817 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
819 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
823 result = gst_rtsp_address_copy (*addrp);
824 g_mutex_unlock (&priv->lock);
831 GST_ERROR_OBJECT (stream, "no address pool specified");
832 g_mutex_unlock (&priv->lock);
837 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
838 g_mutex_unlock (&priv->lock);
844 * gst_rtsp_stream_reserve_address:
845 * @stream: a #GstRTSPStream
846 * @address: an address
851 * Reserve @address and @port as the address and port of @stream.
853 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
854 * reserved. gst_rtsp_address_free() after usage.
857 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
858 const gchar * address, guint port, guint n_ports, guint ttl)
860 GstRTSPStreamPrivate *priv;
861 GstRTSPAddress *result;
863 GSocketFamily family;
864 GstRTSPAddress **addrp;
866 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
867 g_return_val_if_fail (address != NULL, NULL);
868 g_return_val_if_fail (port > 0, NULL);
869 g_return_val_if_fail (n_ports > 0, NULL);
870 g_return_val_if_fail (ttl > 0, NULL);
874 addr = g_inet_address_new_from_string (address);
876 GST_ERROR ("failed to get inet addr from %s", address);
877 family = G_SOCKET_FAMILY_IPV4;
879 family = g_inet_address_get_family (addr);
880 g_object_unref (addr);
883 if (family == G_SOCKET_FAMILY_IPV6)
884 addrp = &priv->addr_v6;
886 addrp = &priv->addr_v4;
888 g_mutex_lock (&priv->lock);
889 if (*addrp == NULL) {
890 GstRTSPAddressPoolResult res;
892 if (priv->pool == NULL)
895 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
896 port, n_ports, ttl, addrp);
897 if (res != GST_RTSP_ADDRESS_POOL_OK)
900 if (strcmp ((*addrp)->address, address) ||
901 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
902 (*addrp)->ttl != ttl)
903 goto different_address;
905 result = gst_rtsp_address_copy (*addrp);
906 g_mutex_unlock (&priv->lock);
913 GST_ERROR_OBJECT (stream, "no address pool specified");
914 g_mutex_unlock (&priv->lock);
919 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
921 g_mutex_unlock (&priv->lock);
926 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
927 " reserved", address);
928 g_mutex_unlock (&priv->lock);
934 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
935 GSocketFamily family, GstElement * udpsrc_out[2],
936 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
937 GstRTSPAddress ** server_addr_out)
939 GstStateChangeReturn ret;
940 GstElement *udpsrc0, *udpsrc1;
941 GstElement *udpsink0, *udpsink1;
942 GSocket *rtp_socket = NULL;
943 GSocket *rtcp_socket;
944 gint tmp_rtp, tmp_rtcp;
946 gint rtpport, rtcpport;
947 GList *rejected_addresses = NULL;
948 GstRTSPAddress *addr = NULL;
949 GInetAddress *inetaddr = NULL;
950 GSocketAddress *rtp_sockaddr = NULL;
951 GSocketAddress *rtcp_sockaddr = NULL;
952 const gchar *multisink_socket;
954 if (family == G_SOCKET_FAMILY_IPV6)
955 multisink_socket = "socket-v6";
957 multisink_socket = "socket";
965 /* Start with random port */
968 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
969 G_SOCKET_PROTOCOL_UDP, NULL);
971 goto no_udp_protocol;
973 if (*server_addr_out)
974 gst_rtsp_address_free (*server_addr_out);
976 /* try to allocate 2 UDP ports, the RTP port should be an even
977 * number and the RTCP port should be the next (uneven) port */
980 if (rtp_socket == NULL) {
981 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
982 G_SOCKET_PROTOCOL_UDP, NULL);
984 goto no_udp_protocol;
987 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
988 GstRTSPAddressFlags flags;
991 rejected_addresses = g_list_prepend (rejected_addresses, addr);
993 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
994 if (family == G_SOCKET_FAMILY_IPV6)
995 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
997 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
999 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1004 tmp_rtp = addr->port;
1006 g_clear_object (&inetaddr);
1007 inetaddr = g_inet_address_new_from_string (addr->address);
1015 if (inetaddr == NULL)
1016 inetaddr = g_inet_address_new_any (family);
1019 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1020 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1021 g_object_unref (rtp_sockaddr);
1024 g_object_unref (rtp_sockaddr);
1026 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1027 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1028 g_clear_object (&rtp_sockaddr);
1033 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1034 g_object_unref (rtp_sockaddr);
1036 /* check if port is even */
1037 if ((tmp_rtp & 1) != 0) {
1038 /* port not even, close and allocate another */
1040 g_clear_object (&rtp_socket);
1045 tmp_rtcp = tmp_rtp + 1;
1047 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1048 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1049 g_object_unref (rtcp_sockaddr);
1050 g_clear_object (&rtp_socket);
1053 g_object_unref (rtcp_sockaddr);
1055 g_clear_object (&inetaddr);
1057 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1058 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1060 if (udpsrc0 == NULL || udpsrc1 == NULL)
1061 goto no_udp_protocol;
1063 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1064 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1066 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1067 if (ret == GST_STATE_CHANGE_FAILURE)
1069 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1070 if (ret == GST_STATE_CHANGE_FAILURE)
1073 /* all fine, do port check */
1074 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1075 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1077 /* this should not happen... */
1078 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1082 udpsink0 = udpsink_out[0];
1084 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1087 goto no_udp_protocol;
1089 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1090 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
1093 udpsink1 = udpsink_out[1];
1095 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1098 goto no_udp_protocol;
1100 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1101 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1102 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
1104 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1105 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
1106 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1107 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1108 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1109 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1110 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1111 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1113 /* we keep these elements, we will further configure them when the
1114 * client told us to really use the UDP ports. */
1115 udpsrc_out[0] = udpsrc0;
1116 udpsrc_out[1] = udpsrc1;
1117 udpsink_out[0] = udpsink0;
1118 udpsink_out[1] = udpsink1;
1119 server_port_out->min = rtpport;
1120 server_port_out->max = rtcpport;
1122 *server_addr_out = addr;
1123 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1125 g_object_unref (rtp_socket);
1126 g_object_unref (rtcp_socket);
1154 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1155 gst_object_unref (udpsrc0);
1158 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1159 gst_object_unref (udpsrc1);
1162 gst_element_set_state (udpsink0, GST_STATE_NULL);
1163 gst_object_unref (udpsink0);
1166 g_object_unref (inetaddr);
1167 g_list_free_full (rejected_addresses,
1168 (GDestroyNotify) gst_rtsp_address_free);
1170 gst_rtsp_address_free (addr);
1172 g_object_unref (rtp_socket);
1174 g_object_unref (rtcp_socket);
1179 /* must be called with lock */
1181 alloc_ports (GstRTSPStream * stream)
1183 GstRTSPStreamPrivate *priv = stream->priv;
1185 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1186 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1187 &priv->server_port_v4, &priv->server_addr_v4);
1189 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1190 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1191 &priv->server_port_v6, &priv->server_addr_v6);
1193 return priv->have_ipv4 || priv->have_ipv6;
1197 * gst_rtsp_stream_get_server_port:
1198 * @stream: a #GstRTSPStream
1199 * @server_port: (out): result server port
1200 * @family: the port family to get
1202 * Fill @server_port with the port pair used by the server. This function can
1203 * only be called when @stream has been joined.
1206 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1207 GstRTSPRange * server_port, GSocketFamily family)
1209 GstRTSPStreamPrivate *priv;
1211 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1212 priv = stream->priv;
1213 g_return_if_fail (priv->is_joined);
1215 g_mutex_lock (&priv->lock);
1216 if (family == G_SOCKET_FAMILY_IPV4) {
1218 *server_port = priv->server_port_v4;
1221 *server_port = priv->server_port_v6;
1223 g_mutex_unlock (&priv->lock);
1227 * gst_rtsp_stream_get_rtpsession:
1228 * @stream: a #GstRTSPStream
1230 * Get the RTP session of this stream.
1232 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1235 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1237 GstRTSPStreamPrivate *priv;
1240 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1242 priv = stream->priv;
1244 g_mutex_lock (&priv->lock);
1245 if ((session = priv->session))
1246 g_object_ref (session);
1247 g_mutex_unlock (&priv->lock);
1253 * gst_rtsp_stream_get_ssrc:
1254 * @stream: a #GstRTSPStream
1255 * @ssrc: (out): result ssrc
1257 * Get the SSRC used by the RTP session of this stream. This function can only
1258 * be called when @stream has been joined.
1261 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1263 GstRTSPStreamPrivate *priv;
1265 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1266 priv = stream->priv;
1267 g_return_if_fail (priv->is_joined);
1269 g_mutex_lock (&priv->lock);
1270 if (ssrc && priv->session)
1271 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1272 g_mutex_unlock (&priv->lock);
1275 /* executed from streaming thread */
1277 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1279 GstRTSPStreamPrivate *priv = stream->priv;
1280 GstCaps *newcaps, *oldcaps;
1282 newcaps = gst_pad_get_current_caps (pad);
1284 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1287 g_mutex_lock (&priv->lock);
1288 oldcaps = priv->caps;
1289 priv->caps = newcaps;
1290 g_mutex_unlock (&priv->lock);
1293 gst_caps_unref (oldcaps);
1297 dump_structure (const GstStructure * s)
1301 sstr = gst_structure_to_string (s);
1302 GST_INFO ("structure: %s", sstr);
1306 static GstRTSPStreamTransport *
1307 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1309 GstRTSPStreamPrivate *priv = stream->priv;
1311 GstRTSPStreamTransport *result = NULL;
1316 if (rtcp_from == NULL)
1319 tmp = g_strrstr (rtcp_from, ":");
1323 port = atoi (tmp + 1);
1324 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1326 g_mutex_lock (&priv->lock);
1327 GST_INFO ("finding %s:%d in %d transports", dest, port,
1328 g_list_length (priv->transports));
1330 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1331 GstRTSPStreamTransport *trans = walk->data;
1332 const GstRTSPTransport *tr;
1335 tr = gst_rtsp_stream_transport_get_transport (trans);
1337 min = tr->client_port.min;
1338 max = tr->client_port.max;
1340 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1346 g_object_ref (result);
1347 g_mutex_unlock (&priv->lock);
1354 static GstRTSPStreamTransport *
1355 check_transport (GObject * source, GstRTSPStream * stream)
1357 GstStructure *stats;
1358 GstRTSPStreamTransport *trans;
1360 /* see if we have a stream to match with the origin of the RTCP packet */
1361 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1362 if (trans == NULL) {
1363 g_object_get (source, "stats", &stats, NULL);
1365 const gchar *rtcp_from;
1367 dump_structure (stats);
1369 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1370 if ((trans = find_transport (stream, rtcp_from))) {
1371 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1373 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1376 gst_structure_free (stats);
1384 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1386 GstRTSPStreamTransport *trans;
1388 GST_INFO ("%p: new source %p", stream, source);
1390 trans = check_transport (source, stream);
1393 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1397 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1399 GST_INFO ("%p: new SDES %p", stream, source);
1403 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1405 GstRTSPStreamTransport *trans;
1407 trans = check_transport (source, stream);
1410 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1411 gst_rtsp_stream_transport_keep_alive (trans);
1415 GstStructure *stats;
1416 g_object_get (source, "stats", &stats, NULL);
1418 dump_structure (stats);
1419 gst_structure_free (stats);
1426 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1428 GST_INFO ("%p: source %p bye", stream, source);
1432 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1434 GstRTSPStreamTransport *trans;
1436 GST_INFO ("%p: source %p bye timeout", stream, source);
1438 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1439 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1440 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1445 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1447 GstRTSPStreamTransport *trans;
1449 GST_INFO ("%p: source %p timeout", stream, source);
1451 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1452 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1453 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1458 clear_tr_cache (GstRTSPStreamPrivate * priv)
1460 g_list_foreach (priv->tr_cache, (GFunc) g_object_unref, NULL);
1461 g_list_free (priv->tr_cache);
1462 priv->tr_cache = NULL;
1465 static GstFlowReturn
1466 handle_new_sample (GstAppSink * sink, gpointer user_data)
1468 GstRTSPStreamPrivate *priv;
1472 GstRTSPStream *stream;
1475 sample = gst_app_sink_pull_sample (sink);
1479 stream = (GstRTSPStream *) user_data;
1480 priv = stream->priv;
1481 buffer = gst_sample_get_buffer (sample);
1483 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1485 g_mutex_lock (&priv->lock);
1486 if (priv->tr_changed) {
1487 clear_tr_cache (priv);
1488 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1489 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1490 priv->tr_cache = g_list_prepend (priv->tr_cache, g_object_ref (tr));
1492 priv->tr_changed = FALSE;
1494 g_mutex_unlock (&priv->lock);
1496 for (walk = priv->tr_cache; walk; walk = g_list_next (walk)) {
1497 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1500 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1502 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1505 gst_sample_unref (sample);
1510 static GstAppSinkCallbacks sink_cb = {
1511 NULL, /* not interested in EOS */
1512 NULL, /* not interested in preroll samples */
1517 get_rtp_encoder (GstRTSPStream * stream, guint session)
1519 GstRTSPStreamPrivate *priv = stream->priv;
1521 if (priv->srtpenc == NULL) {
1524 name = g_strdup_printf ("srtpenc_%u", session);
1525 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1528 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1530 return gst_object_ref (priv->srtpenc);
1534 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1536 GstRTSPStreamPrivate *priv = stream->priv;
1541 if (priv->idx != session)
1544 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1546 enc = get_rtp_encoder (stream, session);
1547 name = g_strdup_printf ("rtp_sink_%d", session);
1548 pad = gst_element_get_request_pad (enc, name);
1550 gst_object_unref (pad);
1556 request_rtcp_encoder (GstElement * rtpbin, guint session,
1557 GstRTSPStream * stream)
1559 GstRTSPStreamPrivate *priv = stream->priv;
1564 if (priv->idx != session)
1567 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1569 enc = get_rtp_encoder (stream, session);
1570 name = g_strdup_printf ("rtcp_sink_%d", session);
1571 pad = gst_element_get_request_pad (enc, name);
1573 gst_object_unref (pad);
1579 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1581 GstRTSPStreamPrivate *priv = stream->priv;
1584 GST_DEBUG ("request key %08x", ssrc);
1586 g_mutex_lock (&priv->lock);
1587 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1588 gst_caps_ref (caps);
1589 g_mutex_unlock (&priv->lock);
1595 request_rtcp_decoder (GstElement * rtpbin, guint session,
1596 GstRTSPStream * stream)
1598 GstRTSPStreamPrivate *priv = stream->priv;
1600 if (priv->idx != session)
1603 if (priv->srtpdec == NULL) {
1606 name = g_strdup_printf ("srtpdec_%u", session);
1607 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1610 g_signal_connect (priv->srtpdec, "request-key",
1611 (GCallback) request_key, stream);
1613 return gst_object_ref (priv->srtpdec);
1617 * gst_rtsp_stream_join_bin:
1618 * @stream: a #GstRTSPStream
1619 * @bin: (transfer none): a #GstBin to join
1620 * @rtpbin: (transfer none): a rtpbin element in @bin
1621 * @state: the target state of the new elements
1623 * Join the #GstBin @bin that contains the element @rtpbin.
1625 * @stream will link to @rtpbin, which must be inside @bin. The elements
1626 * added to @bin will be set to the state given in @state.
1628 * Returns: %TRUE on success.
1631 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1632 GstElement * rtpbin, GstState state)
1634 GstRTSPStreamPrivate *priv;
1638 GstPad *pad, *sinkpad, *selpad;
1639 GstPadLinkReturn ret;
1641 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1642 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1643 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1645 priv = stream->priv;
1647 g_mutex_lock (&priv->lock);
1648 if (priv->is_joined)
1651 /* create a session with the same index as the stream */
1654 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1656 if (!alloc_ports (stream))
1659 /* update the dscp qos field in the sinks */
1660 update_dscp_qos (stream);
1662 if (priv->profiles & GST_RTSP_PROFILE_SAVP
1663 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
1665 g_signal_connect (rtpbin, "request-rtp-encoder",
1666 (GCallback) request_rtp_encoder, stream);
1667 g_signal_connect (rtpbin, "request-rtcp-encoder",
1668 (GCallback) request_rtcp_encoder, stream);
1669 g_signal_connect (rtpbin, "request-rtcp-decoder",
1670 (GCallback) request_rtcp_decoder, stream);
1673 /* get a pad for sending RTP */
1674 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1675 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1677 /* link the RTP pad to the session manager, it should not really fail unless
1678 * this is not really an RTP pad */
1679 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1680 if (ret != GST_PAD_LINK_OK)
1683 /* get pads from the RTP session element for sending and receiving
1685 name = g_strdup_printf ("send_rtp_src_%u", idx);
1686 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1688 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1689 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1691 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1692 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1694 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1695 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1698 /* get the session */
1699 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1701 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1703 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1705 g_signal_connect (priv->session, "on-ssrc-active",
1706 (GCallback) on_ssrc_active, stream);
1707 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1709 g_signal_connect (priv->session, "on-bye-timeout",
1710 (GCallback) on_bye_timeout, stream);
1711 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1714 for (i = 0; i < 2; i++) {
1715 GstPad *teepad, *queuepad;
1716 /* For the sender we create this bit of pipeline for both
1717 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1718 * we need to add a queue before appsink to make the pipeline
1719 * not block. For the TCP case, we want to pump data to the
1720 * client as fast as possible anyway.
1722 * .--------. .-----. .---------.
1723 * | rtpbin | | tee | | udpsink |
1724 * | send->sink src->sink |
1725 * '--------' | | '---------'
1726 * | | .---------. .---------.
1727 * | | | queue | | appsink |
1728 * | src->sink src->sink |
1729 * '-----' '---------' '---------'
1731 * When only UDP is allowed, we skip the tee, queue and appsink and link the
1732 * udpsink directly to the session.
1735 gst_bin_add (bin, priv->udpsink[i]);
1736 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1738 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1739 /* make tee for RTP/RTCP */
1740 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1741 gst_bin_add (bin, priv->tee[i]);
1743 /* and link to rtpbin send pad */
1744 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1745 gst_pad_link (priv->send_src[i], pad);
1746 gst_object_unref (pad);
1748 /* link tee to udpsink */
1749 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1750 gst_pad_link (teepad, sinkpad);
1751 gst_object_unref (teepad);
1754 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1755 gst_bin_add (bin, priv->appqueue[i]);
1756 /* and link to tee */
1757 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1758 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1759 gst_pad_link (teepad, pad);
1760 gst_object_unref (pad);
1761 gst_object_unref (teepad);
1764 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1765 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1766 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1767 gst_bin_add (bin, priv->appsink[i]);
1768 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1769 &sink_cb, stream, NULL);
1770 /* and link to queue */
1771 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1772 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1773 gst_pad_link (queuepad, pad);
1774 gst_object_unref (pad);
1775 gst_object_unref (queuepad);
1777 /* else only udpsink needed, link it to the session */
1778 gst_pad_link (priv->send_src[i], sinkpad);
1780 gst_object_unref (sinkpad);
1782 /* For the receiver we create this bit of pipeline for both
1783 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1784 * and it is all funneled into the rtpbin receive pad.
1786 * .--------. .--------. .--------.
1787 * | udpsrc | | funnel | | rtpbin |
1788 * | src->sink src->sink |
1789 * '--------' | | '--------'
1793 * '--------' '--------'
1795 /* make funnel for the RTP/RTCP receivers */
1796 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1797 gst_bin_add (bin, priv->funnel[i]);
1799 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1800 gst_pad_link (pad, priv->recv_sink[i]);
1801 gst_object_unref (pad);
1803 if (priv->udpsrc_v4[i]) {
1804 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1806 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1807 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1809 gst_bin_add (bin, priv->udpsrc_v4[i]);
1811 /* and link to the funnel v4 */
1812 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1813 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1814 gst_pad_link (pad, selpad);
1815 gst_object_unref (pad);
1816 gst_object_unref (selpad);
1819 if (priv->udpsrc_v6[i]) {
1820 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1821 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1822 gst_bin_add (bin, priv->udpsrc_v6[i]);
1824 /* and link to the funnel v6 */
1825 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1826 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1827 gst_pad_link (pad, selpad);
1828 gst_object_unref (pad);
1829 gst_object_unref (selpad);
1832 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1833 /* make and add appsrc */
1834 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1835 gst_bin_add (bin, priv->appsrc[i]);
1836 /* and link to the funnel */
1837 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1838 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1839 gst_pad_link (pad, selpad);
1840 gst_object_unref (pad);
1841 gst_object_unref (selpad);
1844 /* check if we need to set to a special state */
1845 if (state != GST_STATE_NULL) {
1846 if (priv->udpsink[i])
1847 gst_element_set_state (priv->udpsink[i], state);
1848 if (priv->appsink[i])
1849 gst_element_set_state (priv->appsink[i], state);
1850 if (priv->appqueue[i])
1851 gst_element_set_state (priv->appqueue[i], state);
1853 gst_element_set_state (priv->tee[i], state);
1854 if (priv->funnel[i])
1855 gst_element_set_state (priv->funnel[i], state);
1856 if (priv->appsrc[i])
1857 gst_element_set_state (priv->appsrc[i], state);
1861 /* be notified of caps changes */
1862 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
1863 (GCallback) caps_notify, stream);
1865 priv->is_joined = TRUE;
1866 g_mutex_unlock (&priv->lock);
1873 g_mutex_unlock (&priv->lock);
1878 g_mutex_unlock (&priv->lock);
1879 GST_WARNING ("failed to allocate ports %u", idx);
1884 GST_WARNING ("failed to link stream %u", idx);
1885 gst_object_unref (priv->send_rtp_sink);
1886 priv->send_rtp_sink = NULL;
1887 g_mutex_unlock (&priv->lock);
1893 * gst_rtsp_stream_leave_bin:
1894 * @stream: a #GstRTSPStream
1895 * @bin: (transfer none): a #GstBin
1896 * @rtpbin: (transfer none): a rtpbin #GstElement
1898 * Remove the elements of @stream from @bin.
1900 * Return: %TRUE on success.
1903 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1904 GstElement * rtpbin)
1906 GstRTSPStreamPrivate *priv;
1909 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1910 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1911 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1913 priv = stream->priv;
1915 g_mutex_lock (&priv->lock);
1916 if (!priv->is_joined)
1917 goto was_not_joined;
1919 /* all transports must be removed by now */
1920 g_return_val_if_fail (priv->transports == NULL, FALSE);
1922 clear_tr_cache (priv);
1924 GST_INFO ("stream %p leaving bin", stream);
1926 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1927 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
1928 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1929 gst_object_unref (priv->send_rtp_sink);
1930 priv->send_rtp_sink = NULL;
1932 for (i = 0; i < 2; i++) {
1933 if (priv->udpsink[i])
1934 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1935 if (priv->appsink[i])
1936 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1937 if (priv->appqueue[i])
1938 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1940 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1941 if (priv->funnel[i])
1942 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1943 if (priv->appsrc[i])
1944 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1945 if (priv->udpsrc_v4[i]) {
1946 /* and set udpsrc to NULL now before removing */
1947 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1948 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1949 /* removing them should also nicely release the request
1950 * pads when they finalize */
1951 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1953 if (priv->udpsrc_v6[i]) {
1954 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1955 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1956 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1958 if (priv->udpsink[i])
1959 gst_bin_remove (bin, priv->udpsink[i]);
1960 if (priv->appsrc[i])
1961 gst_bin_remove (bin, priv->appsrc[i]);
1962 if (priv->appsink[i])
1963 gst_bin_remove (bin, priv->appsink[i]);
1964 if (priv->appqueue[i])
1965 gst_bin_remove (bin, priv->appqueue[i]);
1967 gst_bin_remove (bin, priv->tee[i]);
1968 if (priv->funnel[i])
1969 gst_bin_remove (bin, priv->funnel[i]);
1971 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1972 gst_object_unref (priv->recv_sink[i]);
1973 priv->recv_sink[i] = NULL;
1975 priv->udpsrc_v4[i] = NULL;
1976 priv->udpsrc_v6[i] = NULL;
1977 priv->udpsink[i] = NULL;
1978 priv->appsrc[i] = NULL;
1979 priv->appsink[i] = NULL;
1980 priv->appqueue[i] = NULL;
1981 priv->tee[i] = NULL;
1982 priv->funnel[i] = NULL;
1984 gst_object_unref (priv->send_src[0]);
1985 priv->send_src[0] = NULL;
1987 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1988 gst_object_unref (priv->send_src[1]);
1989 priv->send_src[1] = NULL;
1991 g_object_unref (priv->session);
1992 priv->session = NULL;
1994 gst_caps_unref (priv->caps);
1998 gst_object_unref (priv->srtpenc);
2000 priv->is_joined = FALSE;
2001 g_mutex_unlock (&priv->lock);
2007 g_mutex_unlock (&priv->lock);
2013 * gst_rtsp_stream_get_rtpinfo:
2014 * @stream: a #GstRTSPStream
2015 * @rtptime: (allow-none): result RTP timestamp
2016 * @seq: (allow-none): result RTP seqnum
2017 * @clock_rate: (allow-none): the clock rate
2018 * @running_time: (allow-none): result running-time
2020 * Retrieve the current rtptime, seq and running-time. This is used to
2021 * construct a RTPInfo reply header.
2023 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2026 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2027 guint * rtptime, guint * seq, guint * clock_rate,
2028 GstClockTime * running_time)
2030 GstRTSPStreamPrivate *priv;
2031 GstStructure *stats;
2032 GObjectClass *payobjclass;
2034 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2036 priv = stream->priv;
2038 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2040 g_mutex_lock (&priv->lock);
2042 if (g_object_class_find_property (payobjclass, "stats")) {
2043 g_object_get (priv->payloader, "stats", &stats, NULL);
2048 gst_structure_get_uint (stats, "seqnum", seq);
2051 gst_structure_get_uint (stats, "timestamp", rtptime);
2054 gst_structure_get_clock_time (stats, "running-time", running_time);
2057 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2058 if (*clock_rate == 0 && running_time)
2059 *running_time = GST_CLOCK_TIME_NONE;
2061 gst_structure_free (stats);
2063 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2064 !g_object_class_find_property (payobjclass, "timestamp"))
2068 g_object_get (priv->payloader, "seqnum", seq, NULL);
2071 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2074 *running_time = GST_CLOCK_TIME_NONE;
2076 g_mutex_unlock (&priv->lock);
2083 GST_WARNING ("Could not get payloader stats");
2084 g_mutex_unlock (&priv->lock);
2090 * gst_rtsp_stream_get_caps:
2091 * @stream: a #GstRTSPStream
2093 * Retrieve the current caps of @stream.
2095 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2099 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2101 GstRTSPStreamPrivate *priv;
2104 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2106 priv = stream->priv;
2108 g_mutex_lock (&priv->lock);
2109 if ((result = priv->caps))
2110 gst_caps_ref (result);
2111 g_mutex_unlock (&priv->lock);
2117 * gst_rtsp_stream_recv_rtp:
2118 * @stream: a #GstRTSPStream
2119 * @buffer: (transfer full): a #GstBuffer
2121 * Handle an RTP buffer for the stream. This method is usually called when a
2122 * message has been received from a client using the TCP transport.
2124 * This function takes ownership of @buffer.
2126 * Returns: a GstFlowReturn.
2129 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2131 GstRTSPStreamPrivate *priv;
2133 GstElement *element;
2135 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2136 priv = stream->priv;
2137 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2138 g_return_val_if_fail (priv->is_joined, FALSE);
2140 g_mutex_lock (&priv->lock);
2141 if (priv->appsrc[0])
2142 element = gst_object_ref (priv->appsrc[0]);
2145 g_mutex_unlock (&priv->lock);
2148 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2149 gst_object_unref (element);
2157 * gst_rtsp_stream_recv_rtcp:
2158 * @stream: a #GstRTSPStream
2159 * @buffer: (transfer full): a #GstBuffer
2161 * Handle an RTCP buffer for the stream. This method is usually called when a
2162 * message has been received from a client using the TCP transport.
2164 * This function takes ownership of @buffer.
2166 * Returns: a GstFlowReturn.
2169 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2171 GstRTSPStreamPrivate *priv;
2173 GstElement *element;
2175 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2176 priv = stream->priv;
2177 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2178 g_return_val_if_fail (priv->is_joined, FALSE);
2180 g_mutex_lock (&priv->lock);
2181 if (priv->appsrc[1])
2182 element = gst_object_ref (priv->appsrc[1]);
2185 g_mutex_unlock (&priv->lock);
2188 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2189 gst_object_unref (element);
2192 gst_buffer_unref (buffer);
2197 /* must be called with lock */
2199 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
2202 GstRTSPStreamPrivate *priv = stream->priv;
2203 const GstRTSPTransport *tr;
2205 tr = gst_rtsp_stream_transport_get_transport (trans);
2207 switch (tr->lower_transport) {
2208 case GST_RTSP_LOWER_TRANS_UDP:
2209 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2215 dest = tr->destination;
2216 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2221 min = tr->client_port.min;
2222 max = tr->client_port.max;
2227 GST_INFO ("setting ttl-mc %d", ttl);
2228 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
2229 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
2231 GST_INFO ("adding %s:%d-%d", dest, min, max);
2232 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
2233 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
2234 priv->transports = g_list_prepend (priv->transports, trans);
2236 GST_INFO ("removing %s:%d-%d", dest, min, max);
2237 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
2238 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
2239 priv->transports = g_list_remove (priv->transports, trans);
2241 priv->tr_changed = TRUE;
2244 case GST_RTSP_LOWER_TRANS_TCP:
2246 GST_INFO ("adding TCP %s", tr->destination);
2247 priv->transports = g_list_prepend (priv->transports, trans);
2249 GST_INFO ("removing TCP %s", tr->destination);
2250 priv->transports = g_list_remove (priv->transports, trans);
2252 priv->tr_changed = TRUE;
2255 goto unknown_transport;
2262 GST_INFO ("Unknown transport %d", tr->lower_transport);
2269 * gst_rtsp_stream_add_transport:
2270 * @stream: a #GstRTSPStream
2271 * @trans: (transfer none): a #GstRTSPStreamTransport
2273 * Add the transport in @trans to @stream. The media of @stream will
2274 * then also be send to the values configured in @trans.
2276 * @stream must be joined to a bin.
2278 * @trans must contain a valid #GstRTSPTransport.
2280 * Returns: %TRUE if @trans was added
2283 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
2284 GstRTSPStreamTransport * trans)
2286 GstRTSPStreamPrivate *priv;
2289 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2290 priv = stream->priv;
2291 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2292 g_return_val_if_fail (priv->is_joined, FALSE);
2294 g_mutex_lock (&priv->lock);
2295 res = update_transport (stream, trans, TRUE);
2296 g_mutex_unlock (&priv->lock);
2302 * gst_rtsp_stream_remove_transport:
2303 * @stream: a #GstRTSPStream
2304 * @trans: (transfer none): a #GstRTSPStreamTransport
2306 * Remove the transport in @trans from @stream. The media of @stream will
2307 * not be sent to the values configured in @trans.
2309 * @stream must be joined to a bin.
2311 * @trans must contain a valid #GstRTSPTransport.
2313 * Returns: %TRUE if @trans was removed
2316 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2317 GstRTSPStreamTransport * trans)
2319 GstRTSPStreamPrivate *priv;
2322 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2323 priv = stream->priv;
2324 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2325 g_return_val_if_fail (priv->is_joined, FALSE);
2327 g_mutex_lock (&priv->lock);
2328 res = update_transport (stream, trans, FALSE);
2329 g_mutex_unlock (&priv->lock);
2335 * gst_rtsp_stream_update_crypto:
2336 * @stream: a #GstRTSPStream
2338 * @crypto: (transfer none) (allow none): a #GstCaps with crypto info
2340 * Update the new crypto information for @ssrc in @stream. If information
2341 * for @ssrc did not exist, it will be added. If information
2342 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
2343 * be removed from @stream.
2345 * Returns: %TRUE if @crypto could be updated
2348 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
2349 guint ssrc, GstCaps * crypto)
2351 GstRTSPStreamPrivate *priv;
2353 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2354 g_return_val_if_fail (GST_IS_CAPS (crypto), FALSE);
2356 priv = stream->priv;
2358 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
2360 g_mutex_lock (&priv->lock);
2362 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
2363 gst_caps_ref (crypto));
2365 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
2366 g_mutex_unlock (&priv->lock);
2372 * gst_rtsp_stream_get_rtp_socket:
2373 * @stream: a #GstRTSPStream
2374 * @family: the socket family
2376 * Get the RTP socket from @stream for a @family.
2378 * @stream must be joined to a bin.
2380 * Returns: (transfer full): the RTP socket or %NULL if no socket could be
2381 * allocated for @family. Unref after usage
2384 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
2386 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2390 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2391 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2392 family == G_SOCKET_FAMILY_IPV6, NULL);
2393 g_return_val_if_fail (priv->udpsink[0], NULL);
2395 if (family == G_SOCKET_FAMILY_IPV6)
2400 g_object_get (priv->udpsink[0], name, &socket, NULL);
2406 * gst_rtsp_stream_get_rtcp_socket:
2407 * @stream: a #GstRTSPStream
2408 * @family: the socket family
2410 * Get the RTCP socket from @stream for a @family.
2412 * @stream must be joined to a bin.
2414 * Returns: (transfer full): the RTCP socket or %NULL if no socket could be
2415 * allocated for @family. Unref after usage
2418 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
2420 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2424 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2425 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2426 family == G_SOCKET_FAMILY_IPV6, NULL);
2427 g_return_val_if_fail (priv->udpsink[1], NULL);
2429 if (family == G_SOCKET_FAMILY_IPV6)
2434 g_object_get (priv->udpsink[1], name, &socket, NULL);
2440 * gst_rtsp_stream_transport_filter:
2441 * @stream: a #GstRTSPStream
2442 * @func: (scope call) (allow-none): a callback
2443 * @user_data: (closure): user data passed to @func
2445 * Call @func for each transport managed by @stream. The result value of @func
2446 * determines what happens to the transport. @func will be called with @stream
2447 * locked so no further actions on @stream can be performed from @func.
2449 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
2452 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
2454 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
2455 * will also be added with an additional ref to the result #GList of this
2458 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
2460 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
2461 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
2462 * element in the #GList should be unreffed before the list is freed.
2465 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
2466 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
2468 GstRTSPStreamPrivate *priv;
2469 GList *result, *walk, *next;
2471 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2473 priv = stream->priv;
2477 g_mutex_lock (&priv->lock);
2478 for (walk = priv->transports; walk; walk = next) {
2479 GstRTSPStreamTransport *trans = walk->data;
2480 GstRTSPFilterResult res;
2482 next = g_list_next (walk);
2485 res = func (stream, trans, user_data);
2487 res = GST_RTSP_FILTER_REF;
2490 case GST_RTSP_FILTER_REMOVE:
2491 update_transport (stream, trans, FALSE);
2493 case GST_RTSP_FILTER_REF:
2494 result = g_list_prepend (result, g_object_ref (trans));
2496 case GST_RTSP_FILTER_KEEP:
2501 g_mutex_unlock (&priv->lock);
2506 static GstPadProbeReturn
2507 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2509 GstRTSPStreamPrivate *priv;
2510 GstRTSPStream *stream;
2513 priv = stream->priv;
2515 GST_DEBUG_OBJECT (pad, "now blocking");
2517 g_mutex_lock (&priv->lock);
2518 priv->blocking = TRUE;
2519 g_mutex_unlock (&priv->lock);
2521 gst_element_post_message (priv->payloader,
2522 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
2523 gst_structure_new_empty ("GstRTSPStreamBlocking")));
2525 return GST_PAD_PROBE_OK;
2529 * gst_rtsp_stream_set_blocked:
2530 * @stream: a #GstRTSPStream
2531 * @blocked: boolean indicating we should block or unblock
2533 * Blocks or unblocks the dataflow on @stream.
2535 * Returns: %TRUE on success
2538 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
2540 GstRTSPStreamPrivate *priv;
2542 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2544 priv = stream->priv;
2546 g_mutex_lock (&priv->lock);
2548 priv->blocking = FALSE;
2549 if (priv->blocked_id == 0) {
2550 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
2551 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
2552 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
2553 g_object_ref (stream), g_object_unref);
2556 if (priv->blocked_id != 0) {
2557 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
2558 priv->blocked_id = 0;
2559 priv->blocking = FALSE;
2562 g_mutex_unlock (&priv->lock);
2568 * gst_rtsp_stream_is_blocking:
2569 * @stream: a #GstRTSPStream
2571 * Check if @stream is blocking on a #GstBuffer.
2573 * Returns: %TRUE if @stream is blocking
2576 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
2578 GstRTSPStreamPrivate *priv;
2581 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2583 priv = stream->priv;
2585 g_mutex_lock (&priv->lock);
2586 result = priv->blocking;
2587 g_mutex_unlock (&priv->lock);