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 GstRTSPLowerTrans protocols;
73 /* pads on the rtpbin */
74 GstPad *send_rtp_sink;
78 /* the RTPSession object */
81 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
83 GstElement *udpsrc_v4[2];
85 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
87 GstElement *udpsrc_v6[2];
89 GstElement *udpsink[2];
91 /* for TCP transport */
92 GstElement *appsrc[2];
93 GstElement *appqueue[2];
94 GstElement *appsink[2];
97 GstElement *funnel[2];
99 /* server ports for sending/receiving over ipv4 */
100 GstRTSPRange server_port_v4;
101 GstRTSPAddress *server_addr_v4;
104 /* server ports for sending/receiving over ipv6 */
105 GstRTSPRange server_port_v6;
106 GstRTSPAddress *server_addr_v6;
109 /* multicast addresses */
110 GstRTSPAddressPool *pool;
111 GstRTSPAddress *addr_v4;
112 GstRTSPAddress *addr_v6;
114 /* the caps of the stream */
118 /* transports we stream to */
125 #define DEFAULT_CONTROL NULL
126 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
127 GST_RTSP_LOWER_TRANS_TCP
137 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
138 #define GST_CAT_DEFAULT rtsp_stream_debug
140 static GQuark ssrc_stream_map_key;
142 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
143 GValue * value, GParamSpec * pspec);
144 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
145 const GValue * value, GParamSpec * pspec);
147 static void gst_rtsp_stream_finalize (GObject * obj);
149 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
152 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
154 GObjectClass *gobject_class;
156 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
158 gobject_class = G_OBJECT_CLASS (klass);
160 gobject_class->get_property = gst_rtsp_stream_get_property;
161 gobject_class->set_property = gst_rtsp_stream_set_property;
162 gobject_class->finalize = gst_rtsp_stream_finalize;
164 g_object_class_install_property (gobject_class, PROP_CONTROL,
165 g_param_spec_string ("control", "Control",
166 "The control string for this stream", DEFAULT_CONTROL,
167 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
169 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
170 g_param_spec_flags ("protocols", "Protocols",
171 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
172 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
174 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
176 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
180 gst_rtsp_stream_init (GstRTSPStream * stream)
182 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
184 GST_DEBUG ("new stream %p", stream);
189 priv->control = g_strdup (DEFAULT_CONTROL);
190 priv->protocols = DEFAULT_PROTOCOLS;
192 g_mutex_init (&priv->lock);
196 gst_rtsp_stream_finalize (GObject * obj)
198 GstRTSPStream *stream;
199 GstRTSPStreamPrivate *priv;
201 stream = GST_RTSP_STREAM (obj);
204 GST_DEBUG ("finalize stream %p", stream);
206 /* we really need to be unjoined now */
207 g_return_if_fail (!priv->is_joined);
210 gst_rtsp_address_free (priv->addr_v4);
212 gst_rtsp_address_free (priv->addr_v6);
213 if (priv->server_addr_v4)
214 gst_rtsp_address_free (priv->server_addr_v4);
215 if (priv->server_addr_v6)
216 gst_rtsp_address_free (priv->server_addr_v6);
218 g_object_unref (priv->pool);
219 gst_object_unref (priv->payloader);
220 gst_object_unref (priv->srcpad);
221 g_free (priv->control);
222 g_mutex_clear (&priv->lock);
224 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
228 gst_rtsp_stream_get_property (GObject * object, guint propid,
229 GValue * value, GParamSpec * pspec)
231 GstRTSPStream *stream = GST_RTSP_STREAM (object);
235 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
238 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
241 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
246 gst_rtsp_stream_set_property (GObject * object, guint propid,
247 const GValue * value, GParamSpec * pspec)
249 GstRTSPStream *stream = GST_RTSP_STREAM (object);
253 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
256 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
259 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
264 * gst_rtsp_stream_new:
267 * @payloader: a #GstElement
269 * Create a new media stream with index @idx that handles RTP data on
270 * @srcpad and has a payloader element @payloader.
272 * Returns: a new #GstRTSPStream
275 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
277 GstRTSPStreamPrivate *priv;
278 GstRTSPStream *stream;
280 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
281 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
282 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
284 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
287 priv->payloader = gst_object_ref (payloader);
288 priv->srcpad = gst_object_ref (srcpad);
294 * gst_rtsp_stream_get_index:
295 * @stream: a #GstRTSPStream
297 * Get the stream index.
299 * Return: the stream index.
302 gst_rtsp_stream_get_index (GstRTSPStream * stream)
304 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
306 return stream->priv->idx;
310 * gst_rtsp_stream_get_pt:
311 * @stream: a #GstRTSPStream
313 * Get the stream payload type.
315 * Return: the stream payload type.
318 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
320 GstRTSPStreamPrivate *priv;
323 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
327 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
333 * gst_rtsp_stream_get_srcpad:
334 * @stream: a #GstRTSPStream
336 * Get the srcpad associated with @stream.
338 * Returns: (transfer full): the srcpad. Unref after usage.
341 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
343 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
345 return gst_object_ref (stream->priv->srcpad);
349 * gst_rtsp_stream_get_control:
350 * @stream: a #GstRTSPStream
352 * Get the control string to identify this stream.
354 * Returns: (transfer full): the control string. free after usage.
357 gst_rtsp_stream_get_control (GstRTSPStream * stream)
359 GstRTSPStreamPrivate *priv;
362 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
366 g_mutex_lock (&priv->lock);
367 if ((result = g_strdup (priv->control)) == NULL)
368 result = g_strdup_printf ("stream=%u", priv->idx);
369 g_mutex_unlock (&priv->lock);
375 * gst_rtsp_stream_set_control:
376 * @stream: a #GstRTSPStream
377 * @control: a control string
379 * Set the control string in @stream.
382 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
384 GstRTSPStreamPrivate *priv;
386 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
390 g_mutex_lock (&priv->lock);
391 g_free (priv->control);
392 priv->control = g_strdup (control);
393 g_mutex_unlock (&priv->lock);
397 * gst_rtsp_stream_has_control:
398 * @stream: a #GstRTSPStream
399 * @control: a control string
401 * Check if @stream has the control string @control.
403 * Returns: %TRUE is @stream has @control as the control string
406 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
408 GstRTSPStreamPrivate *priv;
411 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
415 g_mutex_lock (&priv->lock);
417 res = (g_strcmp0 (priv->control, control) == 0);
420 sscanf (control, "stream=%u", &streamid);
421 res = (streamid == priv->idx);
423 g_mutex_unlock (&priv->lock);
429 * gst_rtsp_stream_set_mtu:
430 * @stream: a #GstRTSPStream
433 * Configure the mtu in the payloader of @stream to @mtu.
436 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
438 GstRTSPStreamPrivate *priv;
440 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
444 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
446 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
450 * gst_rtsp_stream_get_mtu:
451 * @stream: a #GstRTSPStream
453 * Get the configured MTU in the payloader of @stream.
455 * Returns: the MTU of the payloader.
458 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
460 GstRTSPStreamPrivate *priv;
463 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
467 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
472 /* Update the dscp qos property on the udp sinks */
474 update_dscp_qos (GstRTSPStream * stream)
476 GstRTSPStreamPrivate *priv;
478 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
482 if (priv->udpsink[0]) {
483 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
487 if (priv->udpsink[1]) {
488 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
494 * gst_rtsp_stream_set_dscp_qos:
495 * @stream: a #GstRTSPStream
496 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
498 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
501 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
503 GstRTSPStreamPrivate *priv;
505 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
509 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
511 if (dscp_qos < -1 || dscp_qos > 63) {
512 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
516 priv->dscp_qos = dscp_qos;
518 update_dscp_qos (stream);
522 * gst_rtsp_stream_get_dscp_qos:
523 * @stream: a #GstRTSPStream
525 * Get the configured DSCP QoS in of the outgoing sockets.
527 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
530 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
532 GstRTSPStreamPrivate *priv;
534 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
538 return priv->dscp_qos;
542 * gst_rtsp_stream_set_protocols:
543 * @stream: a #GstRTSPStream
544 * @protocols: the new flags
546 * Configure the allowed lower transport for @stream.
549 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
550 GstRTSPLowerTrans protocols)
552 GstRTSPStreamPrivate *priv;
554 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
558 g_mutex_lock (&priv->lock);
559 priv->protocols = protocols;
560 g_mutex_unlock (&priv->lock);
564 * gst_rtsp_stream_get_protocols:
565 * @stream: a #GstRTSPStream
567 * Get the allowed protocols of @stream.
569 * Returns: a #GstRTSPLowerTrans
572 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
574 GstRTSPStreamPrivate *priv;
575 GstRTSPLowerTrans res;
577 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
578 GST_RTSP_LOWER_TRANS_UNKNOWN);
582 g_mutex_lock (&priv->lock);
583 res = priv->protocols;
584 g_mutex_unlock (&priv->lock);
590 * gst_rtsp_stream_set_address_pool:
591 * @stream: a #GstRTSPStream
592 * @pool: a #GstRTSPAddressPool
594 * configure @pool to be used as the address pool of @stream.
597 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
598 GstRTSPAddressPool * pool)
600 GstRTSPStreamPrivate *priv;
601 GstRTSPAddressPool *old;
603 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
607 GST_LOG_OBJECT (stream, "set address pool %p", pool);
609 g_mutex_lock (&priv->lock);
610 if ((old = priv->pool) != pool)
611 priv->pool = pool ? g_object_ref (pool) : NULL;
614 g_mutex_unlock (&priv->lock);
617 g_object_unref (old);
621 * gst_rtsp_stream_get_address_pool:
622 * @stream: a #GstRTSPStream
624 * Get the #GstRTSPAddressPool used as the address pool of @stream.
626 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
630 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
632 GstRTSPStreamPrivate *priv;
633 GstRTSPAddressPool *result;
635 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
639 g_mutex_lock (&priv->lock);
640 if ((result = priv->pool))
641 g_object_ref (result);
642 g_mutex_unlock (&priv->lock);
648 * gst_rtsp_stream_get_multicast_address:
649 * @stream: a #GstRTSPStream
650 * @family: the #GSocketFamily
652 * Get the multicast address of @stream for @family.
654 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
655 * allocated. gst_rtsp_address_free() after usage.
658 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
659 GSocketFamily family)
661 GstRTSPStreamPrivate *priv;
662 GstRTSPAddress *result;
663 GstRTSPAddress **addrp;
664 GstRTSPAddressFlags flags;
666 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
670 if (family == G_SOCKET_FAMILY_IPV6) {
671 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
672 addrp = &priv->addr_v4;
674 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
675 addrp = &priv->addr_v6;
678 g_mutex_lock (&priv->lock);
679 if (*addrp == NULL) {
680 if (priv->pool == NULL)
683 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
685 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
689 result = gst_rtsp_address_copy (*addrp);
690 g_mutex_unlock (&priv->lock);
697 GST_ERROR_OBJECT (stream, "no address pool specified");
698 g_mutex_unlock (&priv->lock);
703 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
704 g_mutex_unlock (&priv->lock);
710 * gst_rtsp_stream_reserve_address:
711 * @stream: a #GstRTSPStream
712 * @address: an address
717 * Reserve @address and @port as the address and port of @stream.
719 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
720 * reserved. gst_rtsp_address_free() after usage.
723 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
724 const gchar * address, guint port, guint n_ports, guint ttl)
726 GstRTSPStreamPrivate *priv;
727 GstRTSPAddress *result;
729 GSocketFamily family;
730 GstRTSPAddress **addrp;
732 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
733 g_return_val_if_fail (address != NULL, NULL);
734 g_return_val_if_fail (port > 0, NULL);
735 g_return_val_if_fail (n_ports > 0, NULL);
736 g_return_val_if_fail (ttl > 0, NULL);
740 addr = g_inet_address_new_from_string (address);
742 GST_ERROR ("failed to get inet addr from %s", address);
743 family = G_SOCKET_FAMILY_IPV4;
745 family = g_inet_address_get_family (addr);
746 g_object_unref (addr);
749 if (family == G_SOCKET_FAMILY_IPV6)
750 addrp = &priv->addr_v4;
752 addrp = &priv->addr_v6;
754 g_mutex_lock (&priv->lock);
755 if (*addrp == NULL) {
756 GstRTSPAddressPoolResult res;
758 if (priv->pool == NULL)
761 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
762 port, n_ports, ttl, addrp);
763 if (res != GST_RTSP_ADDRESS_POOL_OK)
766 if (strcmp ((*addrp)->address, address) ||
767 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
768 (*addrp)->ttl != ttl)
769 goto different_address;
771 result = gst_rtsp_address_copy (*addrp);
772 g_mutex_unlock (&priv->lock);
779 GST_ERROR_OBJECT (stream, "no address pool specified");
780 g_mutex_unlock (&priv->lock);
785 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
787 g_mutex_unlock (&priv->lock);
792 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
793 " reserved", address);
794 g_mutex_unlock (&priv->lock);
800 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
801 GSocketFamily family, GstElement * udpsrc_out[2],
802 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
803 GstRTSPAddress ** server_addr_out)
805 GstStateChangeReturn ret;
806 GstElement *udpsrc0, *udpsrc1;
807 GstElement *udpsink0, *udpsink1;
808 GSocket *rtp_socket = NULL;
809 GSocket *rtcp_socket;
810 gint tmp_rtp, tmp_rtcp;
812 gint rtpport, rtcpport;
813 GList *rejected_addresses = NULL;
814 GstRTSPAddress *addr = NULL;
815 GInetAddress *inetaddr = NULL;
816 GSocketAddress *rtp_sockaddr = NULL;
817 GSocketAddress *rtcp_sockaddr = NULL;
818 const gchar *multisink_socket;
820 if (family == G_SOCKET_FAMILY_IPV6)
821 multisink_socket = "socket-v6";
823 multisink_socket = "socket";
831 /* Start with random port */
834 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
835 G_SOCKET_PROTOCOL_UDP, NULL);
837 goto no_udp_protocol;
839 if (*server_addr_out)
840 gst_rtsp_address_free (*server_addr_out);
842 /* try to allocate 2 UDP ports, the RTP port should be an even
843 * number and the RTCP port should be the next (uneven) port */
846 if (rtp_socket == NULL) {
847 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
848 G_SOCKET_PROTOCOL_UDP, NULL);
850 goto no_udp_protocol;
853 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
854 GstRTSPAddressFlags flags;
857 rejected_addresses = g_list_prepend (rejected_addresses, addr);
859 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
860 if (family == G_SOCKET_FAMILY_IPV6)
861 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
863 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
865 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
870 tmp_rtp = addr->port;
872 g_clear_object (&inetaddr);
873 inetaddr = g_inet_address_new_from_string (addr->address);
881 if (inetaddr == NULL)
882 inetaddr = g_inet_address_new_any (family);
885 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
886 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
887 g_object_unref (rtp_sockaddr);
890 g_object_unref (rtp_sockaddr);
892 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
893 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
894 g_clear_object (&rtp_sockaddr);
899 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
900 g_object_unref (rtp_sockaddr);
902 /* check if port is even */
903 if ((tmp_rtp & 1) != 0) {
904 /* port not even, close and allocate another */
906 g_clear_object (&rtp_socket);
911 tmp_rtcp = tmp_rtp + 1;
913 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
914 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
915 g_object_unref (rtcp_sockaddr);
916 g_clear_object (&rtp_socket);
919 g_object_unref (rtcp_sockaddr);
921 g_clear_object (&inetaddr);
923 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
924 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
926 if (udpsrc0 == NULL || udpsrc1 == NULL)
927 goto no_udp_protocol;
929 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
930 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
932 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
933 if (ret == GST_STATE_CHANGE_FAILURE)
935 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
936 if (ret == GST_STATE_CHANGE_FAILURE)
939 /* all fine, do port check */
940 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
941 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
943 /* this should not happen... */
944 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
948 udpsink0 = udpsink_out[0];
950 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
953 goto no_udp_protocol;
955 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
956 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
959 udpsink1 = udpsink_out[1];
961 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
964 goto no_udp_protocol;
966 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
967 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
968 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
970 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
971 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
972 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
973 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
974 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
975 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
976 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
977 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
979 /* we keep these elements, we will further configure them when the
980 * client told us to really use the UDP ports. */
981 udpsrc_out[0] = udpsrc0;
982 udpsrc_out[1] = udpsrc1;
983 udpsink_out[0] = udpsink0;
984 udpsink_out[1] = udpsink1;
985 server_port_out->min = rtpport;
986 server_port_out->max = rtcpport;
988 *server_addr_out = addr;
989 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
991 g_object_unref (rtp_socket);
992 g_object_unref (rtcp_socket);
1020 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1021 gst_object_unref (udpsrc0);
1024 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1025 gst_object_unref (udpsrc1);
1028 gst_element_set_state (udpsink0, GST_STATE_NULL);
1029 gst_object_unref (udpsink0);
1032 gst_element_set_state (udpsink1, GST_STATE_NULL);
1033 gst_object_unref (udpsink1);
1036 g_object_unref (inetaddr);
1037 g_list_free_full (rejected_addresses,
1038 (GDestroyNotify) gst_rtsp_address_free);
1040 gst_rtsp_address_free (addr);
1042 g_object_unref (rtp_socket);
1044 g_object_unref (rtcp_socket);
1049 /* must be called with lock */
1051 alloc_ports (GstRTSPStream * stream)
1053 GstRTSPStreamPrivate *priv = stream->priv;
1055 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1056 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1057 &priv->server_port_v4, &priv->server_addr_v4);
1059 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1060 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1061 &priv->server_port_v6, &priv->server_addr_v6);
1063 return priv->have_ipv4 || priv->have_ipv6;
1067 * gst_rtsp_stream_get_server_port:
1068 * @stream: a #GstRTSPStream
1069 * @server_port: (out): result server port
1070 * @family: the port family to get
1072 * Fill @server_port with the port pair used by the server. This function can
1073 * only be called when @stream has been joined.
1076 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1077 GstRTSPRange * server_port, GSocketFamily family)
1079 GstRTSPStreamPrivate *priv;
1081 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1082 priv = stream->priv;
1083 g_return_if_fail (priv->is_joined);
1085 g_mutex_lock (&priv->lock);
1086 if (family == G_SOCKET_FAMILY_IPV4) {
1088 *server_port = priv->server_port_v4;
1091 *server_port = priv->server_port_v6;
1093 g_mutex_unlock (&priv->lock);
1097 * gst_rtsp_stream_get_rtpsession:
1098 * @stream: a #GstRTSPStream
1100 * Get the RTP session of this stream.
1102 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1105 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1107 GstRTSPStreamPrivate *priv;
1110 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1112 priv = stream->priv;
1114 g_mutex_lock (&priv->lock);
1115 if ((session = priv->session))
1116 g_object_ref (session);
1117 g_mutex_unlock (&priv->lock);
1123 * gst_rtsp_stream_get_ssrc:
1124 * @stream: a #GstRTSPStream
1125 * @ssrc: (out): result ssrc
1127 * Get the SSRC used by the RTP session of this stream. This function can only
1128 * be called when @stream has been joined.
1131 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1133 GstRTSPStreamPrivate *priv;
1135 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1136 priv = stream->priv;
1137 g_return_if_fail (priv->is_joined);
1139 g_mutex_lock (&priv->lock);
1140 if (ssrc && priv->session)
1141 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1142 g_mutex_unlock (&priv->lock);
1145 /* executed from streaming thread */
1147 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1149 GstRTSPStreamPrivate *priv = stream->priv;
1150 GstCaps *newcaps, *oldcaps;
1152 newcaps = gst_pad_get_current_caps (pad);
1154 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1157 g_mutex_lock (&priv->lock);
1158 oldcaps = priv->caps;
1159 priv->caps = newcaps;
1160 g_mutex_unlock (&priv->lock);
1163 gst_caps_unref (oldcaps);
1167 dump_structure (const GstStructure * s)
1171 sstr = gst_structure_to_string (s);
1172 GST_INFO ("structure: %s", sstr);
1176 static GstRTSPStreamTransport *
1177 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1179 GstRTSPStreamPrivate *priv = stream->priv;
1181 GstRTSPStreamTransport *result = NULL;
1186 if (rtcp_from == NULL)
1189 tmp = g_strrstr (rtcp_from, ":");
1193 port = atoi (tmp + 1);
1194 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1196 g_mutex_lock (&priv->lock);
1197 GST_INFO ("finding %s:%d in %d transports", dest, port,
1198 g_list_length (priv->transports));
1200 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1201 GstRTSPStreamTransport *trans = walk->data;
1202 const GstRTSPTransport *tr;
1205 tr = gst_rtsp_stream_transport_get_transport (trans);
1207 min = tr->client_port.min;
1208 max = tr->client_port.max;
1210 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1216 g_object_ref (result);
1217 g_mutex_unlock (&priv->lock);
1224 static GstRTSPStreamTransport *
1225 check_transport (GObject * source, GstRTSPStream * stream)
1227 GstStructure *stats;
1228 GstRTSPStreamTransport *trans;
1230 /* see if we have a stream to match with the origin of the RTCP packet */
1231 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1232 if (trans == NULL) {
1233 g_object_get (source, "stats", &stats, NULL);
1235 const gchar *rtcp_from;
1237 dump_structure (stats);
1239 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1240 if ((trans = find_transport (stream, rtcp_from))) {
1241 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1243 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1246 gst_structure_free (stats);
1254 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1256 GstRTSPStreamTransport *trans;
1258 GST_INFO ("%p: new source %p", stream, source);
1260 trans = check_transport (source, stream);
1263 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1267 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1269 GST_INFO ("%p: new SDES %p", stream, source);
1273 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1275 GstRTSPStreamTransport *trans;
1277 trans = check_transport (source, stream);
1280 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1281 gst_rtsp_stream_transport_keep_alive (trans);
1285 GstStructure *stats;
1286 g_object_get (source, "stats", &stats, NULL);
1288 dump_structure (stats);
1289 gst_structure_free (stats);
1296 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1298 GST_INFO ("%p: source %p bye", stream, source);
1302 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1304 GstRTSPStreamTransport *trans;
1306 GST_INFO ("%p: source %p bye timeout", stream, source);
1308 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1309 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1310 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1315 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1317 GstRTSPStreamTransport *trans;
1319 GST_INFO ("%p: source %p timeout", stream, source);
1321 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1322 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1323 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1327 static GstFlowReturn
1328 handle_new_sample (GstAppSink * sink, gpointer user_data)
1330 GstRTSPStreamPrivate *priv;
1334 GstRTSPStream *stream;
1336 sample = gst_app_sink_pull_sample (sink);
1340 stream = (GstRTSPStream *) user_data;
1341 priv = stream->priv;
1342 buffer = gst_sample_get_buffer (sample);
1344 g_mutex_lock (&priv->lock);
1345 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1346 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1348 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1349 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1351 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1354 g_mutex_unlock (&priv->lock);
1356 gst_sample_unref (sample);
1361 static GstAppSinkCallbacks sink_cb = {
1362 NULL, /* not interested in EOS */
1363 NULL, /* not interested in preroll samples */
1368 * gst_rtsp_stream_join_bin:
1369 * @stream: a #GstRTSPStream
1370 * @bin: a #GstBin to join
1371 * @rtpbin: a rtpbin element in @bin
1372 * @state: the target state of the new elements
1374 * Join the #GstBin @bin that contains the element @rtpbin.
1376 * @stream will link to @rtpbin, which must be inside @bin. The elements
1377 * added to @bin will be set to the state given in @state.
1379 * Returns: %TRUE on success.
1382 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1383 GstElement * rtpbin, GstState state)
1385 GstRTSPStreamPrivate *priv;
1389 GstPad *pad, *sinkpad, *selpad;
1390 GstPadLinkReturn ret;
1392 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1393 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1394 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1396 priv = stream->priv;
1398 g_mutex_lock (&priv->lock);
1399 if (priv->is_joined)
1402 /* create a session with the same index as the stream */
1405 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1407 if (!alloc_ports (stream))
1410 /* update the dscp qos field in the sinks */
1411 update_dscp_qos (stream);
1413 /* get a pad for sending RTP */
1414 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1415 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1417 /* link the RTP pad to the session manager, it should not really fail unless
1418 * this is not really an RTP pad */
1419 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1420 if (ret != GST_PAD_LINK_OK)
1423 /* get pads from the RTP session element for sending and receiving
1425 name = g_strdup_printf ("send_rtp_src_%u", idx);
1426 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1428 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1429 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1431 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1432 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1434 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1435 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1438 /* get the session */
1439 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1441 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1443 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1445 g_signal_connect (priv->session, "on-ssrc-active",
1446 (GCallback) on_ssrc_active, stream);
1447 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1449 g_signal_connect (priv->session, "on-bye-timeout",
1450 (GCallback) on_bye_timeout, stream);
1451 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1454 for (i = 0; i < 2; i++) {
1455 GstPad *teepad, *queuepad;
1456 /* For the sender we create this bit of pipeline for both
1457 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1458 * we need to add a queue before appsink to make the pipeline
1459 * not block. For the TCP case, we want to pump data to the
1460 * client as fast as possible anyway.
1462 * .--------. .-----. .---------.
1463 * | rtpbin | | tee | | udpsink |
1464 * | send->sink src->sink |
1465 * '--------' | | '---------'
1466 * | | .---------. .---------.
1467 * | | | queue | | appsink |
1468 * | src->sink src->sink |
1469 * '-----' '---------' '---------'
1471 * When only UDP is allowed, we skip the tee, queue and appsink and link the
1472 * udpsink directly to the session.
1475 gst_bin_add (bin, priv->udpsink[i]);
1476 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1478 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1479 /* make tee for RTP/RTCP */
1480 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1481 gst_bin_add (bin, priv->tee[i]);
1483 /* and link to rtpbin send pad */
1484 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1485 gst_pad_link (priv->send_src[i], pad);
1486 gst_object_unref (pad);
1488 /* link tee to udpsink */
1489 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1490 gst_pad_link (teepad, sinkpad);
1491 gst_object_unref (teepad);
1494 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1495 gst_bin_add (bin, priv->appqueue[i]);
1496 /* and link to tee */
1497 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1498 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1499 gst_pad_link (teepad, pad);
1500 gst_object_unref (pad);
1501 gst_object_unref (teepad);
1504 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1505 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1506 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1507 gst_bin_add (bin, priv->appsink[i]);
1508 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1509 &sink_cb, stream, NULL);
1510 /* and link to queue */
1511 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1512 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1513 gst_pad_link (queuepad, pad);
1514 gst_object_unref (pad);
1515 gst_object_unref (queuepad);
1517 /* else only udpsink needed, link it to the session */
1518 gst_pad_link (priv->send_src[i], sinkpad);
1520 gst_object_unref (sinkpad);
1522 /* For the receiver we create this bit of pipeline for both
1523 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1524 * and it is all funneled into the rtpbin receive pad.
1526 * .--------. .--------. .--------.
1527 * | udpsrc | | funnel | | rtpbin |
1528 * | src->sink src->sink |
1529 * '--------' | | '--------'
1533 * '--------' '--------'
1535 /* make funnel for the RTP/RTCP receivers */
1536 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1537 gst_bin_add (bin, priv->funnel[i]);
1539 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1540 gst_pad_link (pad, priv->recv_sink[i]);
1541 gst_object_unref (pad);
1543 if (priv->udpsrc_v4[i]) {
1544 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1546 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1547 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1549 gst_bin_add (bin, priv->udpsrc_v4[i]);
1551 /* and link to the funnel v4 */
1552 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1553 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1554 gst_pad_link (pad, selpad);
1555 gst_object_unref (pad);
1556 gst_object_unref (selpad);
1559 if (priv->udpsrc_v6[i]) {
1560 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1561 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1562 gst_bin_add (bin, priv->udpsrc_v6[i]);
1564 /* and link to the funnel v6 */
1565 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1566 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1567 gst_pad_link (pad, selpad);
1568 gst_object_unref (pad);
1569 gst_object_unref (selpad);
1572 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1573 /* make and add appsrc */
1574 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1575 gst_bin_add (bin, priv->appsrc[i]);
1576 /* and link to the funnel */
1577 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1578 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1579 gst_pad_link (pad, selpad);
1580 gst_object_unref (pad);
1581 gst_object_unref (selpad);
1584 /* check if we need to set to a special state */
1585 if (state != GST_STATE_NULL) {
1586 if (priv->udpsink[i])
1587 gst_element_set_state (priv->udpsink[i], state);
1588 if (priv->appsink[i])
1589 gst_element_set_state (priv->appsink[i], state);
1590 if (priv->appqueue[i])
1591 gst_element_set_state (priv->appqueue[i], state);
1593 gst_element_set_state (priv->tee[i], state);
1594 if (priv->funnel[i])
1595 gst_element_set_state (priv->funnel[i], state);
1596 if (priv->appsrc[i])
1597 gst_element_set_state (priv->appsrc[i], state);
1601 /* be notified of caps changes */
1602 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1603 (GCallback) caps_notify, stream);
1605 priv->is_joined = TRUE;
1606 g_mutex_unlock (&priv->lock);
1613 g_mutex_unlock (&priv->lock);
1618 g_mutex_unlock (&priv->lock);
1619 GST_WARNING ("failed to allocate ports %u", idx);
1624 GST_WARNING ("failed to link stream %u", idx);
1625 gst_object_unref (priv->send_rtp_sink);
1626 priv->send_rtp_sink = NULL;
1627 g_mutex_unlock (&priv->lock);
1633 * gst_rtsp_stream_leave_bin:
1634 * @stream: a #GstRTSPStream
1636 * @rtpbin: a rtpbin #GstElement
1638 * Remove the elements of @stream from @bin.
1640 * Return: %TRUE on success.
1643 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1644 GstElement * rtpbin)
1646 GstRTSPStreamPrivate *priv;
1649 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1650 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1651 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1653 priv = stream->priv;
1655 g_mutex_lock (&priv->lock);
1656 if (!priv->is_joined)
1657 goto was_not_joined;
1659 /* all transports must be removed by now */
1660 g_return_val_if_fail (priv->transports == NULL, FALSE);
1662 GST_INFO ("stream %p leaving bin", stream);
1664 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1665 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1666 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1667 gst_object_unref (priv->send_rtp_sink);
1668 priv->send_rtp_sink = NULL;
1670 for (i = 0; i < 2; i++) {
1671 if (priv->udpsink[i])
1672 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1673 if (priv->appsink[i])
1674 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1675 if (priv->appqueue[i])
1676 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1678 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1679 if (priv->funnel[i])
1680 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1681 if (priv->appsrc[i])
1682 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1683 if (priv->udpsrc_v4[i]) {
1684 /* and set udpsrc to NULL now before removing */
1685 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1686 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1687 /* removing them should also nicely release the request
1688 * pads when they finalize */
1689 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1691 if (priv->udpsrc_v6[i]) {
1692 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1693 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1694 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1696 if (priv->udpsink[i])
1697 gst_bin_remove (bin, priv->udpsink[i]);
1698 if (priv->appsrc[i])
1699 gst_bin_remove (bin, priv->appsrc[i]);
1700 if (priv->appsink[i])
1701 gst_bin_remove (bin, priv->appsink[i]);
1702 if (priv->appqueue[i])
1703 gst_bin_remove (bin, priv->appqueue[i]);
1705 gst_bin_remove (bin, priv->tee[i]);
1706 if (priv->funnel[i])
1707 gst_bin_remove (bin, priv->funnel[i]);
1709 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1710 gst_object_unref (priv->recv_sink[i]);
1711 priv->recv_sink[i] = NULL;
1713 priv->udpsrc_v4[i] = NULL;
1714 priv->udpsrc_v6[i] = NULL;
1715 priv->udpsink[i] = NULL;
1716 priv->appsrc[i] = NULL;
1717 priv->appsink[i] = NULL;
1718 priv->appqueue[i] = NULL;
1719 priv->tee[i] = NULL;
1720 priv->funnel[i] = NULL;
1722 gst_object_unref (priv->send_src[0]);
1723 priv->send_src[0] = NULL;
1725 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1726 gst_object_unref (priv->send_src[1]);
1727 priv->send_src[1] = NULL;
1729 g_object_unref (priv->session);
1730 priv->session = NULL;
1732 gst_caps_unref (priv->caps);
1735 priv->is_joined = FALSE;
1736 g_mutex_unlock (&priv->lock);
1747 * gst_rtsp_stream_get_rtpinfo:
1748 * @stream: a #GstRTSPStream
1749 * @rtptime: result RTP timestamp
1750 * @seq: result RTP seqnum
1752 * Retrieve the current rtptime and seq. This is used to
1753 * construct a RTPInfo reply header.
1755 * Returns: %TRUE when rtptime and seq could be determined.
1758 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1759 guint * rtptime, guint * seq)
1761 GstRTSPStreamPrivate *priv;
1762 GObjectClass *payobjclass;
1764 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1765 g_return_val_if_fail (rtptime != NULL, FALSE);
1766 g_return_val_if_fail (seq != NULL, FALSE);
1768 priv = stream->priv;
1770 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1772 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1773 !g_object_class_find_property (payobjclass, "timestamp"))
1776 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1782 * gst_rtsp_stream_get_caps:
1783 * @stream: a #GstRTSPStream
1785 * Retrieve the current caps of @stream.
1787 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1791 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1793 GstRTSPStreamPrivate *priv;
1796 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1798 priv = stream->priv;
1800 g_mutex_lock (&priv->lock);
1801 if ((result = priv->caps))
1802 gst_caps_ref (result);
1803 g_mutex_unlock (&priv->lock);
1809 * gst_rtsp_stream_recv_rtp:
1810 * @stream: a #GstRTSPStream
1811 * @buffer: (transfer full): a #GstBuffer
1813 * Handle an RTP buffer for the stream. This method is usually called when a
1814 * message has been received from a client using the TCP transport.
1816 * This function takes ownership of @buffer.
1818 * Returns: a GstFlowReturn.
1821 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1823 GstRTSPStreamPrivate *priv;
1825 GstElement *element;
1827 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1828 priv = stream->priv;
1829 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1830 g_return_val_if_fail (priv->is_joined, FALSE);
1832 g_mutex_lock (&priv->lock);
1833 if (priv->appsrc[0])
1834 element = gst_object_ref (priv->appsrc[0]);
1837 g_mutex_unlock (&priv->lock);
1840 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1841 gst_object_unref (element);
1849 * gst_rtsp_stream_recv_rtcp:
1850 * @stream: a #GstRTSPStream
1851 * @buffer: (transfer full): a #GstBuffer
1853 * Handle an RTCP buffer for the stream. This method is usually called when a
1854 * message has been received from a client using the TCP transport.
1856 * This function takes ownership of @buffer.
1858 * Returns: a GstFlowReturn.
1861 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1863 GstRTSPStreamPrivate *priv;
1865 GstElement *element;
1867 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1868 priv = stream->priv;
1869 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1870 g_return_val_if_fail (priv->is_joined, FALSE);
1872 g_mutex_lock (&priv->lock);
1873 if (priv->appsrc[1])
1874 element = gst_object_ref (priv->appsrc[1]);
1877 g_mutex_unlock (&priv->lock);
1880 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1881 gst_object_unref (element);
1888 /* must be called with lock */
1890 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1893 GstRTSPStreamPrivate *priv = stream->priv;
1894 const GstRTSPTransport *tr;
1896 tr = gst_rtsp_stream_transport_get_transport (trans);
1898 switch (tr->lower_transport) {
1899 case GST_RTSP_LOWER_TRANS_UDP:
1900 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1906 dest = tr->destination;
1907 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1912 min = tr->client_port.min;
1913 max = tr->client_port.max;
1917 GST_INFO ("adding %s:%d-%d", dest, min, max);
1918 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1919 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1921 GST_INFO ("setting ttl-mc %d", ttl);
1922 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1923 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1925 priv->transports = g_list_prepend (priv->transports, trans);
1927 GST_INFO ("removing %s:%d-%d", dest, min, max);
1928 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1929 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1930 priv->transports = g_list_remove (priv->transports, trans);
1934 case GST_RTSP_LOWER_TRANS_TCP:
1936 GST_INFO ("adding TCP %s", tr->destination);
1937 priv->transports = g_list_prepend (priv->transports, trans);
1939 GST_INFO ("removing TCP %s", tr->destination);
1940 priv->transports = g_list_remove (priv->transports, trans);
1944 goto unknown_transport;
1951 GST_INFO ("Unknown transport %d", tr->lower_transport);
1958 * gst_rtsp_stream_add_transport:
1959 * @stream: a #GstRTSPStream
1960 * @trans: a #GstRTSPStreamTransport
1962 * Add the transport in @trans to @stream. The media of @stream will
1963 * then also be send to the values configured in @trans.
1965 * @stream must be joined to a bin.
1967 * @trans must contain a valid #GstRTSPTransport.
1969 * Returns: %TRUE if @trans was added
1972 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1973 GstRTSPStreamTransport * trans)
1975 GstRTSPStreamPrivate *priv;
1978 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1979 priv = stream->priv;
1980 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1981 g_return_val_if_fail (priv->is_joined, FALSE);
1983 g_mutex_lock (&priv->lock);
1984 res = update_transport (stream, trans, TRUE);
1985 g_mutex_unlock (&priv->lock);
1991 * gst_rtsp_stream_remove_transport:
1992 * @stream: a #GstRTSPStream
1993 * @trans: a #GstRTSPStreamTransport
1995 * Remove the transport in @trans from @stream. The media of @stream will
1996 * not be sent to the values configured in @trans.
1998 * @stream must be joined to a bin.
2000 * @trans must contain a valid #GstRTSPTransport.
2002 * Returns: %TRUE if @trans was removed
2005 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2006 GstRTSPStreamTransport * trans)
2008 GstRTSPStreamPrivate *priv;
2011 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2012 priv = stream->priv;
2013 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2014 g_return_val_if_fail (priv->is_joined, FALSE);
2016 g_mutex_lock (&priv->lock);
2017 res = update_transport (stream, trans, FALSE);
2018 g_mutex_unlock (&priv->lock);
2024 * gst_rtsp_stream_get_rtp_socket:
2025 * @stream: a #GstRTSPStream
2026 * @family: the socket family
2028 * Get the RTP socket from @stream for a @family.
2030 * @stream must be joined to a bin.
2032 * Returns: the RTP socket or %NULL if no socket could be allocated for @family.
2036 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
2038 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2042 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2043 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2044 family == G_SOCKET_FAMILY_IPV6, NULL);
2045 g_return_val_if_fail (priv->udpsink[0], NULL);
2047 if (family == G_SOCKET_FAMILY_IPV6)
2052 g_object_get (priv->udpsink[0], name, &socket, NULL);
2058 * gst_rtsp_stream_get_rtcp_socket:
2059 * @stream: a #GstRTSPStream
2060 * @family: the socket family
2062 * Get the RTCP socket from @stream for a @family.
2064 * @stream must be joined to a bin.
2066 * Returns: the RTCP socket or %NULL if no socket could be allocated for
2067 * @family. Unref after usage
2070 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
2072 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2076 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2077 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2078 family == G_SOCKET_FAMILY_IPV6, NULL);
2079 g_return_val_if_fail (priv->udpsink[1], NULL);
2081 if (family == G_SOCKET_FAMILY_IPV6)
2086 g_object_get (priv->udpsink[1], name, &socket, NULL);
2092 * gst_rtsp_stream_transport_filter:
2093 * @stream: a #GstRTSPStream
2094 * @func: (scope call) (allow-none): a callback
2095 * @user_data: user data passed to @func
2097 * Call @func for each transport managed by @stream. The result value of @func
2098 * determines what happens to the transport. @func will be called with @stream
2099 * locked so no further actions on @stream can be performed from @func.
2101 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
2104 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
2106 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
2107 * will also be added with an additional ref to the result #GList of this
2110 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
2112 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
2113 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
2114 * element in the #GList should be unreffed before the list is freed.
2117 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
2118 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
2120 GstRTSPStreamPrivate *priv;
2121 GList *result, *walk, *next;
2123 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2125 priv = stream->priv;
2129 g_mutex_lock (&priv->lock);
2130 for (walk = priv->transports; walk; walk = next) {
2131 GstRTSPStreamTransport *trans = walk->data;
2132 GstRTSPFilterResult res;
2134 next = g_list_next (walk);
2137 res = func (stream, trans, user_data);
2139 res = GST_RTSP_FILTER_REF;
2142 case GST_RTSP_FILTER_REMOVE:
2143 update_transport (stream, trans, FALSE);
2145 case GST_RTSP_FILTER_REF:
2146 result = g_list_prepend (result, g_object_ref (trans));
2148 case GST_RTSP_FILTER_KEEP:
2153 g_mutex_unlock (&priv->lock);