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_TCP
136 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
137 #define GST_CAT_DEFAULT rtsp_stream_debug
139 static GQuark ssrc_stream_map_key;
141 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
142 GValue * value, GParamSpec * pspec);
143 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
144 const GValue * value, GParamSpec * pspec);
146 static void gst_rtsp_stream_finalize (GObject * obj);
148 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
151 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
153 GObjectClass *gobject_class;
155 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
157 gobject_class = G_OBJECT_CLASS (klass);
159 gobject_class->get_property = gst_rtsp_stream_get_property;
160 gobject_class->set_property = gst_rtsp_stream_set_property;
161 gobject_class->finalize = gst_rtsp_stream_finalize;
163 g_object_class_install_property (gobject_class, PROP_CONTROL,
164 g_param_spec_string ("control", "Control",
165 "The control string for this stream", DEFAULT_CONTROL,
166 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
168 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
169 g_param_spec_flags ("protocols", "Protocols",
170 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
171 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
173 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
175 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
179 gst_rtsp_stream_init (GstRTSPStream * stream)
181 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
183 GST_DEBUG ("new stream %p", stream);
188 priv->control = g_strdup (DEFAULT_CONTROL);
189 priv->protocols = DEFAULT_PROTOCOLS;
191 g_mutex_init (&priv->lock);
195 gst_rtsp_stream_finalize (GObject * obj)
197 GstRTSPStream *stream;
198 GstRTSPStreamPrivate *priv;
200 stream = GST_RTSP_STREAM (obj);
203 GST_DEBUG ("finalize stream %p", stream);
205 /* we really need to be unjoined now */
206 g_return_if_fail (!priv->is_joined);
209 gst_rtsp_address_free (priv->addr_v4);
211 gst_rtsp_address_free (priv->addr_v6);
212 if (priv->server_addr_v4)
213 gst_rtsp_address_free (priv->server_addr_v4);
214 if (priv->server_addr_v6)
215 gst_rtsp_address_free (priv->server_addr_v6);
217 g_object_unref (priv->pool);
218 gst_object_unref (priv->payloader);
219 gst_object_unref (priv->srcpad);
220 g_free (priv->control);
221 g_mutex_clear (&priv->lock);
223 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
227 gst_rtsp_stream_get_property (GObject * object, guint propid,
228 GValue * value, GParamSpec * pspec)
230 GstRTSPStream *stream = GST_RTSP_STREAM (object);
234 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
237 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
240 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
245 gst_rtsp_stream_set_property (GObject * object, guint propid,
246 const GValue * value, GParamSpec * pspec)
248 GstRTSPStream *stream = GST_RTSP_STREAM (object);
252 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
255 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
258 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
263 * gst_rtsp_stream_new:
266 * @payloader: a #GstElement
268 * Create a new media stream with index @idx that handles RTP data on
269 * @srcpad and has a payloader element @payloader.
271 * Returns: a new #GstRTSPStream
274 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
276 GstRTSPStreamPrivate *priv;
277 GstRTSPStream *stream;
279 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
280 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
281 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
283 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
286 priv->payloader = gst_object_ref (payloader);
287 priv->srcpad = gst_object_ref (srcpad);
293 * gst_rtsp_stream_get_index:
294 * @stream: a #GstRTSPStream
296 * Get the stream index.
298 * Return: the stream index.
301 gst_rtsp_stream_get_index (GstRTSPStream * stream)
303 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
305 return stream->priv->idx;
309 * gst_rtsp_stream_get_srcpad:
310 * @stream: a #GstRTSPStream
312 * Get the srcpad associated with @stream.
314 * Return: the srcpad. Unref after usage.
317 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
319 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
321 return gst_object_ref (stream->priv->srcpad);
325 * gst_rtsp_stream_get_control:
326 * @stream: a #GstRTSPStream
328 * Get the control string to identify this stream.
330 * Return: the control string. free after usage.
333 gst_rtsp_stream_get_control (GstRTSPStream * stream)
335 GstRTSPStreamPrivate *priv;
338 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
342 g_mutex_lock (&priv->lock);
343 if ((result = g_strdup (priv->control)) == NULL)
344 result = g_strdup_printf ("stream=%u", priv->idx);
345 g_mutex_unlock (&priv->lock);
351 * gst_rtsp_stream_set_control:
352 * @stream: a #GstRTSPStream
353 * @control: a control string
355 * Set the control string in @stream.
358 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
360 GstRTSPStreamPrivate *priv;
362 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
366 g_mutex_lock (&priv->lock);
367 g_free (priv->control);
368 priv->control = g_strdup (control);
369 g_mutex_unlock (&priv->lock);
373 * gst_rtsp_stream_has_control:
374 * @stream: a #GstRTSPStream
375 * @control: a control string
377 * Check if @stream has the control string @control.
379 * Returns: %TRUE is @stream has @control as the control string
382 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
384 GstRTSPStreamPrivate *priv;
387 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
391 g_mutex_lock (&priv->lock);
393 res = g_strcmp0 (priv->control, control);
396 sscanf (control, "stream=%u", &streamid);
397 res = (streamid == priv->idx);
399 g_mutex_unlock (&priv->lock);
405 * gst_rtsp_stream_set_mtu:
406 * @stream: a #GstRTSPStream
409 * Configure the mtu in the payloader of @stream to @mtu.
412 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
414 GstRTSPStreamPrivate *priv;
416 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
420 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
422 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
426 * gst_rtsp_stream_get_mtu:
427 * @stream: a #GstRTSPStream
429 * Get the configured MTU in the payloader of @stream.
431 * Returns: the MTU of the payloader.
434 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
436 GstRTSPStreamPrivate *priv;
439 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
443 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
448 /* Update the dscp qos property on the udp sinks */
450 update_dscp_qos (GstRTSPStream * stream)
452 GstRTSPStreamPrivate *priv;
454 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
458 if (priv->udpsink[0]) {
459 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
463 if (priv->udpsink[1]) {
464 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
470 * gst_rtsp_stream_set_dscp_qos:
471 * @stream: a #GstRTSPStream
472 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
474 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
477 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
479 GstRTSPStreamPrivate *priv;
481 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
485 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
487 if (dscp_qos < -1 || dscp_qos > 63) {
488 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
492 priv->dscp_qos = dscp_qos;
494 update_dscp_qos (stream);
498 * gst_rtsp_stream_get_dscp_qos:
499 * @stream: a #GstRTSPStream
501 * Get the configured DSCP QoS in of the outgoing sockets.
503 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
506 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
508 GstRTSPStreamPrivate *priv;
510 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
514 return priv->dscp_qos;
518 * gst_rtsp_stream_set_protocols:
519 * @stream: a #GstRTSPStream
520 * @protocols: the new flags
522 * Configure the allowed lower transport for @stream.
525 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
526 GstRTSPLowerTrans protocols)
528 GstRTSPStreamPrivate *priv;
530 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
534 g_mutex_lock (&priv->lock);
535 priv->protocols = protocols;
536 g_mutex_unlock (&priv->lock);
540 * gst_rtsp_stream_get_protocols:
541 * @stream: a #GstRTSPStream
543 * Get the allowed protocols of @stream.
545 * Returns: a #GstRTSPLowerTrans
548 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
550 GstRTSPStreamPrivate *priv;
551 GstRTSPLowerTrans res;
553 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
554 GST_RTSP_LOWER_TRANS_UNKNOWN);
558 g_mutex_lock (&priv->lock);
559 res = priv->protocols;
560 g_mutex_unlock (&priv->lock);
566 * gst_rtsp_stream_set_address_pool:
567 * @stream: a #GstRTSPStream
568 * @pool: a #GstRTSPAddressPool
570 * configure @pool to be used as the address pool of @stream.
573 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
574 GstRTSPAddressPool * pool)
576 GstRTSPStreamPrivate *priv;
577 GstRTSPAddressPool *old;
579 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
583 GST_LOG_OBJECT (stream, "set address pool %p", pool);
585 g_mutex_lock (&priv->lock);
586 if ((old = priv->pool) != pool)
587 priv->pool = pool ? g_object_ref (pool) : NULL;
590 g_mutex_unlock (&priv->lock);
593 g_object_unref (old);
597 * gst_rtsp_stream_get_address_pool:
598 * @stream: a #GstRTSPStream
600 * Get the #GstRTSPAddressPool used as the address pool of @stream.
602 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
606 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
608 GstRTSPStreamPrivate *priv;
609 GstRTSPAddressPool *result;
611 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
615 g_mutex_lock (&priv->lock);
616 if ((result = priv->pool))
617 g_object_ref (result);
618 g_mutex_unlock (&priv->lock);
624 * gst_rtsp_stream_get_multicast_address:
625 * @stream: a #GstRTSPStream
626 * @family: the #GSocketFamily
628 * Get the multicast address of @stream for @family.
630 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
631 * allocated. gst_rtsp_address_free() after usage.
634 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
635 GSocketFamily family)
637 GstRTSPStreamPrivate *priv;
638 GstRTSPAddress *result;
639 GstRTSPAddress **addrp;
640 GstRTSPAddressFlags flags;
642 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
646 if (family == G_SOCKET_FAMILY_IPV6) {
647 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
648 addrp = &priv->addr_v4;
650 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
651 addrp = &priv->addr_v6;
654 g_mutex_lock (&priv->lock);
655 if (*addrp == NULL) {
656 if (priv->pool == NULL)
659 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
661 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
665 result = gst_rtsp_address_copy (*addrp);
666 g_mutex_unlock (&priv->lock);
673 GST_ERROR_OBJECT (stream, "no address pool specified");
674 g_mutex_unlock (&priv->lock);
679 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
680 g_mutex_unlock (&priv->lock);
686 * gst_rtsp_stream_reserve_address:
687 * @stream: a #GstRTSPStream
688 * @address: an address
693 * Reserve @address and @port as the address and port of @stream.
695 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
696 * reserved. gst_rtsp_address_free() after usage.
699 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
700 const gchar * address, guint port, guint n_ports, guint ttl)
702 GstRTSPStreamPrivate *priv;
703 GstRTSPAddress *result;
705 GSocketFamily family;
706 GstRTSPAddress **addrp;
708 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
709 g_return_val_if_fail (address != NULL, NULL);
710 g_return_val_if_fail (port > 0, NULL);
711 g_return_val_if_fail (n_ports > 0, NULL);
712 g_return_val_if_fail (ttl > 0, NULL);
716 addr = g_inet_address_new_from_string (address);
718 GST_ERROR ("failed to get inet addr from %s", address);
719 family = G_SOCKET_FAMILY_IPV4;
721 family = g_inet_address_get_family (addr);
722 g_object_unref (addr);
725 if (family == G_SOCKET_FAMILY_IPV6)
726 addrp = &priv->addr_v4;
728 addrp = &priv->addr_v6;
730 g_mutex_lock (&priv->lock);
731 if (*addrp == NULL) {
732 if (priv->pool == NULL)
735 *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
740 if (strcmp ((*addrp)->address, address) ||
741 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
742 (*addrp)->ttl != ttl)
743 goto different_address;
745 result = gst_rtsp_address_copy (*addrp);
746 g_mutex_unlock (&priv->lock);
753 GST_ERROR_OBJECT (stream, "no address pool specified");
754 g_mutex_unlock (&priv->lock);
759 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
761 g_mutex_unlock (&priv->lock);
766 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
767 " reserved", address);
768 g_mutex_unlock (&priv->lock);
774 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
775 GSocketFamily family, GstElement * udpsrc_out[2],
776 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
777 GstRTSPAddress ** server_addr_out)
779 GstStateChangeReturn ret;
780 GstElement *udpsrc0, *udpsrc1;
781 GstElement *udpsink0, *udpsink1;
782 GSocket *rtp_socket = NULL;
783 GSocket *rtcp_socket;
784 gint tmp_rtp, tmp_rtcp;
786 gint rtpport, rtcpport;
787 GList *rejected_addresses = NULL;
788 GstRTSPAddress *addr = NULL;
789 GInetAddress *inetaddr = NULL;
790 GSocketAddress *rtp_sockaddr = NULL;
791 GSocketAddress *rtcp_sockaddr = NULL;
792 const gchar *multisink_socket;
794 if (family == G_SOCKET_FAMILY_IPV6)
795 multisink_socket = "socket-v6";
797 multisink_socket = "socket";
805 /* Start with random port */
808 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
809 G_SOCKET_PROTOCOL_UDP, NULL);
811 goto no_udp_protocol;
813 if (*server_addr_out)
814 gst_rtsp_address_free (*server_addr_out);
816 /* try to allocate 2 UDP ports, the RTP port should be an even
817 * number and the RTCP port should be the next (uneven) port */
820 if (rtp_socket == NULL) {
821 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
822 G_SOCKET_PROTOCOL_UDP, NULL);
824 goto no_udp_protocol;
827 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
828 GstRTSPAddressFlags flags;
831 rejected_addresses = g_list_prepend (rejected_addresses, addr);
833 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
834 if (family == G_SOCKET_FAMILY_IPV6)
835 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
837 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
839 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
844 tmp_rtp = addr->port;
846 g_clear_object (&inetaddr);
847 inetaddr = g_inet_address_new_from_string (addr->address);
855 if (inetaddr == NULL)
856 inetaddr = g_inet_address_new_any (family);
859 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
860 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
861 g_object_unref (rtp_sockaddr);
864 g_object_unref (rtp_sockaddr);
866 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
867 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
868 g_clear_object (&rtp_sockaddr);
873 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
874 g_object_unref (rtp_sockaddr);
876 /* check if port is even */
877 if ((tmp_rtp & 1) != 0) {
878 /* port not even, close and allocate another */
880 g_clear_object (&rtp_socket);
885 tmp_rtcp = tmp_rtp + 1;
887 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
888 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
889 g_object_unref (rtcp_sockaddr);
890 g_clear_object (&rtp_socket);
893 g_object_unref (rtcp_sockaddr);
895 g_clear_object (&inetaddr);
897 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
898 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
900 if (udpsrc0 == NULL || udpsrc1 == NULL)
901 goto no_udp_protocol;
903 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
904 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
906 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
907 if (ret == GST_STATE_CHANGE_FAILURE)
909 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
910 if (ret == GST_STATE_CHANGE_FAILURE)
913 /* all fine, do port check */
914 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
915 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
917 /* this should not happen... */
918 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
922 udpsink0 = udpsink_out[0];
924 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
927 goto no_udp_protocol;
929 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
930 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
933 udpsink1 = udpsink_out[1];
935 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
938 goto no_udp_protocol;
940 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
941 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
942 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
944 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
945 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
946 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
947 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
948 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
949 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
950 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
951 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
953 /* we keep these elements, we will further configure them when the
954 * client told us to really use the UDP ports. */
955 udpsrc_out[0] = udpsrc0;
956 udpsrc_out[1] = udpsrc1;
957 udpsink_out[0] = udpsink0;
958 udpsink_out[1] = udpsink1;
959 server_port_out->min = rtpport;
960 server_port_out->max = rtcpport;
962 *server_addr_out = addr;
963 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
965 g_object_unref (rtp_socket);
966 g_object_unref (rtcp_socket);
994 gst_element_set_state (udpsrc0, GST_STATE_NULL);
995 gst_object_unref (udpsrc0);
998 gst_element_set_state (udpsrc1, GST_STATE_NULL);
999 gst_object_unref (udpsrc1);
1002 gst_element_set_state (udpsink0, GST_STATE_NULL);
1003 gst_object_unref (udpsink0);
1006 gst_element_set_state (udpsink1, GST_STATE_NULL);
1007 gst_object_unref (udpsink1);
1010 g_object_unref (inetaddr);
1011 g_list_free_full (rejected_addresses,
1012 (GDestroyNotify) gst_rtsp_address_free);
1014 gst_rtsp_address_free (addr);
1016 g_object_unref (rtp_socket);
1018 g_object_unref (rtcp_socket);
1023 /* must be called with lock */
1025 alloc_ports (GstRTSPStream * stream)
1027 GstRTSPStreamPrivate *priv = stream->priv;
1029 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1030 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1031 &priv->server_port_v4, &priv->server_addr_v4);
1033 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1034 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1035 &priv->server_port_v6, &priv->server_addr_v6);
1037 return priv->have_ipv4 || priv->have_ipv6;
1041 * gst_rtsp_stream_get_server_port:
1042 * @stream: a #GstRTSPStream
1043 * @server_port: (out): result server port
1044 * @family: the port family to get
1046 * Fill @server_port with the port pair used by the server. This function can
1047 * only be called when @stream has been joined.
1050 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1051 GstRTSPRange * server_port, GSocketFamily family)
1053 GstRTSPStreamPrivate *priv;
1055 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1056 priv = stream->priv;
1057 g_return_if_fail (priv->is_joined);
1059 g_mutex_lock (&priv->lock);
1060 if (family == G_SOCKET_FAMILY_IPV4) {
1062 *server_port = priv->server_port_v4;
1065 *server_port = priv->server_port_v6;
1067 g_mutex_unlock (&priv->lock);
1071 * gst_rtsp_stream_get_rtpsession:
1072 * @stream: a #GstRTSPStream
1074 * Get the RTP session of this stream.
1076 * Returns: The RTP session of this stream. Unref after usage.
1079 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1081 GstRTSPStreamPrivate *priv;
1084 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1086 priv = stream->priv;
1088 g_mutex_lock (&priv->lock);
1089 if ((session = priv->session))
1090 g_object_ref (session);
1091 g_mutex_unlock (&priv->lock);
1097 * gst_rtsp_stream_get_ssrc:
1098 * @stream: a #GstRTSPStream
1099 * @ssrc: (out): result ssrc
1101 * Get the SSRC used by the RTP session of this stream. This function can only
1102 * be called when @stream has been joined.
1105 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1107 GstRTSPStreamPrivate *priv;
1109 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1110 priv = stream->priv;
1111 g_return_if_fail (priv->is_joined);
1113 g_mutex_lock (&priv->lock);
1114 if (ssrc && priv->session)
1115 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1116 g_mutex_unlock (&priv->lock);
1119 /* executed from streaming thread */
1121 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1123 GstRTSPStreamPrivate *priv = stream->priv;
1124 GstCaps *newcaps, *oldcaps;
1126 newcaps = gst_pad_get_current_caps (pad);
1128 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1131 g_mutex_lock (&priv->lock);
1132 oldcaps = priv->caps;
1133 priv->caps = newcaps;
1134 g_mutex_unlock (&priv->lock);
1137 gst_caps_unref (oldcaps);
1141 dump_structure (const GstStructure * s)
1145 sstr = gst_structure_to_string (s);
1146 GST_INFO ("structure: %s", sstr);
1150 static GstRTSPStreamTransport *
1151 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1153 GstRTSPStreamPrivate *priv = stream->priv;
1155 GstRTSPStreamTransport *result = NULL;
1160 if (rtcp_from == NULL)
1163 tmp = g_strrstr (rtcp_from, ":");
1167 port = atoi (tmp + 1);
1168 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1170 g_mutex_lock (&priv->lock);
1171 GST_INFO ("finding %s:%d in %d transports", dest, port,
1172 g_list_length (priv->transports));
1174 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1175 GstRTSPStreamTransport *trans = walk->data;
1176 const GstRTSPTransport *tr;
1179 tr = gst_rtsp_stream_transport_get_transport (trans);
1181 min = tr->client_port.min;
1182 max = tr->client_port.max;
1184 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1190 g_object_ref (result);
1191 g_mutex_unlock (&priv->lock);
1198 static GstRTSPStreamTransport *
1199 check_transport (GObject * source, GstRTSPStream * stream)
1201 GstStructure *stats;
1202 GstRTSPStreamTransport *trans;
1204 /* see if we have a stream to match with the origin of the RTCP packet */
1205 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1206 if (trans == NULL) {
1207 g_object_get (source, "stats", &stats, NULL);
1209 const gchar *rtcp_from;
1211 dump_structure (stats);
1213 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1214 if ((trans = find_transport (stream, rtcp_from))) {
1215 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1217 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1220 gst_structure_free (stats);
1228 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1230 GstRTSPStreamTransport *trans;
1232 GST_INFO ("%p: new source %p", stream, source);
1234 trans = check_transport (source, stream);
1237 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1241 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1243 GST_INFO ("%p: new SDES %p", stream, source);
1247 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1249 GstRTSPStreamTransport *trans;
1251 trans = check_transport (source, stream);
1254 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1255 gst_rtsp_stream_transport_keep_alive (trans);
1259 GstStructure *stats;
1260 g_object_get (source, "stats", &stats, NULL);
1262 dump_structure (stats);
1263 gst_structure_free (stats);
1270 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1272 GST_INFO ("%p: source %p bye", stream, source);
1276 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1278 GstRTSPStreamTransport *trans;
1280 GST_INFO ("%p: source %p bye timeout", stream, source);
1282 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1283 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1284 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1289 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1291 GstRTSPStreamTransport *trans;
1293 GST_INFO ("%p: source %p timeout", stream, source);
1295 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1296 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1297 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1301 static GstFlowReturn
1302 handle_new_sample (GstAppSink * sink, gpointer user_data)
1304 GstRTSPStreamPrivate *priv;
1308 GstRTSPStream *stream;
1310 sample = gst_app_sink_pull_sample (sink);
1314 stream = (GstRTSPStream *) user_data;
1315 priv = stream->priv;
1316 buffer = gst_sample_get_buffer (sample);
1318 g_mutex_lock (&priv->lock);
1319 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1320 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1322 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1323 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1325 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1328 g_mutex_unlock (&priv->lock);
1330 gst_sample_unref (sample);
1335 static GstAppSinkCallbacks sink_cb = {
1336 NULL, /* not interested in EOS */
1337 NULL, /* not interested in preroll samples */
1342 * gst_rtsp_stream_join_bin:
1343 * @stream: a #GstRTSPStream
1344 * @bin: a #GstBin to join
1345 * @rtpbin: a rtpbin element in @bin
1346 * @state: the target state of the new elements
1348 * Join the #Gstbin @bin that contains the element @rtpbin.
1350 * @stream will link to @rtpbin, which must be inside @bin. The elements
1351 * added to @bin will be set to the state given in @state.
1353 * Returns: %TRUE on success.
1356 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1357 GstElement * rtpbin, GstState state)
1359 GstRTSPStreamPrivate *priv;
1363 GstPad *pad, *teepad, *queuepad, *selpad;
1364 GstPadLinkReturn ret;
1366 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1367 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1368 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1370 priv = stream->priv;
1372 g_mutex_lock (&priv->lock);
1373 if (priv->is_joined)
1376 /* create a session with the same index as the stream */
1379 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1381 if (!alloc_ports (stream))
1384 /* update the dscp qos field in the sinks */
1385 update_dscp_qos (stream);
1387 /* get a pad for sending RTP */
1388 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1389 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1391 /* link the RTP pad to the session manager, it should not really fail unless
1392 * this is not really an RTP pad */
1393 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1394 if (ret != GST_PAD_LINK_OK)
1397 /* get pads from the RTP session element for sending and receiving
1399 name = g_strdup_printf ("send_rtp_src_%u", idx);
1400 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1402 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1403 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1405 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1406 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1408 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1409 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1412 /* get the session */
1413 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1415 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1417 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1419 g_signal_connect (priv->session, "on-ssrc-active",
1420 (GCallback) on_ssrc_active, stream);
1421 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1423 g_signal_connect (priv->session, "on-bye-timeout",
1424 (GCallback) on_bye_timeout, stream);
1425 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1428 for (i = 0; i < 2; i++) {
1429 /* For the sender we create this bit of pipeline for both
1430 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1431 * we need to add a queue before appsink to make the pipeline
1432 * not block. For the TCP case, we want to pump data to the
1433 * client as fast as possible anyway.
1435 * .--------. .-----. .---------.
1436 * | rtpbin | | tee | | udpsink |
1437 * | send->sink src->sink |
1438 * '--------' | | '---------'
1439 * | | .---------. .---------.
1440 * | | | queue | | appsink |
1441 * | src->sink src->sink |
1442 * '-----' '---------' '---------'
1444 /* make tee for RTP/RTCP */
1445 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1446 gst_bin_add (bin, priv->tee[i]);
1448 /* and link to rtpbin send pad */
1449 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1450 gst_pad_link (priv->send_src[i], pad);
1451 gst_object_unref (pad);
1454 gst_bin_add (bin, priv->udpsink[i]);
1456 /* link tee to udpsink */
1457 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1458 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1459 gst_pad_link (teepad, pad);
1460 gst_object_unref (pad);
1461 gst_object_unref (teepad);
1464 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1465 gst_bin_add (bin, priv->appqueue[i]);
1466 /* and link to tee */
1467 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1468 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1469 gst_pad_link (teepad, pad);
1470 gst_object_unref (pad);
1471 gst_object_unref (teepad);
1474 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1475 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1476 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1477 gst_bin_add (bin, priv->appsink[i]);
1478 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1479 &sink_cb, stream, NULL);
1480 /* and link to queue */
1481 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1482 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1483 gst_pad_link (queuepad, pad);
1484 gst_object_unref (pad);
1485 gst_object_unref (queuepad);
1487 /* For the receiver we create this bit of pipeline for both
1488 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1489 * and it is all funneled into the rtpbin receive pad.
1491 * .--------. .--------. .--------.
1492 * | udpsrc | | funnel | | rtpbin |
1493 * | src->sink src->sink |
1494 * '--------' | | '--------'
1498 * '--------' '--------'
1500 /* make funnel for the RTP/RTCP receivers */
1501 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1502 gst_bin_add (bin, priv->funnel[i]);
1504 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1505 gst_pad_link (pad, priv->recv_sink[i]);
1506 gst_object_unref (pad);
1508 if (priv->udpsrc_v4[i]) {
1509 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1511 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1512 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1514 gst_bin_add (bin, priv->udpsrc_v4[i]);
1516 /* and link to the funnel v4 */
1517 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1518 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1519 gst_pad_link (pad, selpad);
1520 gst_object_unref (pad);
1521 gst_object_unref (selpad);
1524 if (priv->udpsrc_v6[i]) {
1525 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1526 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1527 gst_bin_add (bin, priv->udpsrc_v6[i]);
1529 /* and link to the funnel v6 */
1530 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1531 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1532 gst_pad_link (pad, selpad);
1533 gst_object_unref (pad);
1534 gst_object_unref (selpad);
1537 /* make and add appsrc */
1538 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1539 gst_bin_add (bin, priv->appsrc[i]);
1540 /* and link to the funnel */
1541 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1542 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1543 gst_pad_link (pad, selpad);
1544 gst_object_unref (pad);
1545 gst_object_unref (selpad);
1547 /* check if we need to set to a special state */
1548 if (state != GST_STATE_NULL) {
1549 gst_element_set_state (priv->udpsink[i], state);
1550 gst_element_set_state (priv->appsink[i], state);
1551 gst_element_set_state (priv->appqueue[i], state);
1552 gst_element_set_state (priv->tee[i], state);
1553 gst_element_set_state (priv->funnel[i], state);
1554 gst_element_set_state (priv->appsrc[i], state);
1558 /* be notified of caps changes */
1559 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1560 (GCallback) caps_notify, stream);
1562 priv->is_joined = TRUE;
1563 g_mutex_unlock (&priv->lock);
1570 g_mutex_unlock (&priv->lock);
1575 g_mutex_unlock (&priv->lock);
1576 GST_WARNING ("failed to allocate ports %u", idx);
1581 GST_WARNING ("failed to link stream %u", idx);
1582 gst_object_unref (priv->send_rtp_sink);
1583 priv->send_rtp_sink = NULL;
1584 g_mutex_unlock (&priv->lock);
1590 * gst_rtsp_stream_leave_bin:
1591 * @stream: a #GstRTSPStream
1593 * @rtpbin: a rtpbin #GstElement
1595 * Remove the elements of @stream from @bin.
1597 * Return: %TRUE on success.
1600 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1601 GstElement * rtpbin)
1603 GstRTSPStreamPrivate *priv;
1606 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1607 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1608 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1610 priv = stream->priv;
1612 g_mutex_lock (&priv->lock);
1613 if (!priv->is_joined)
1614 goto was_not_joined;
1616 /* all transports must be removed by now */
1617 g_return_val_if_fail (priv->transports == NULL, FALSE);
1619 GST_INFO ("stream %p leaving bin", stream);
1621 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1622 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1623 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1624 gst_object_unref (priv->send_rtp_sink);
1625 priv->send_rtp_sink = NULL;
1627 for (i = 0; i < 2; i++) {
1628 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1629 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1630 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1631 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1632 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1633 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1634 if (priv->udpsrc_v4[i]) {
1635 /* and set udpsrc to NULL now before removing */
1636 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1637 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1638 /* removing them should also nicely release the request
1639 * pads when they finalize */
1640 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1642 if (priv->udpsrc_v6[i]) {
1643 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1644 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1645 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1647 gst_bin_remove (bin, priv->udpsink[i]);
1648 gst_bin_remove (bin, priv->appsrc[i]);
1649 gst_bin_remove (bin, priv->appsink[i]);
1650 gst_bin_remove (bin, priv->appqueue[i]);
1651 gst_bin_remove (bin, priv->tee[i]);
1652 gst_bin_remove (bin, priv->funnel[i]);
1654 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1655 gst_object_unref (priv->recv_sink[i]);
1656 priv->recv_sink[i] = NULL;
1658 priv->udpsrc_v4[i] = NULL;
1659 priv->udpsrc_v6[i] = NULL;
1660 priv->udpsink[i] = NULL;
1661 priv->appsrc[i] = NULL;
1662 priv->appsink[i] = NULL;
1663 priv->appqueue[i] = NULL;
1664 priv->tee[i] = NULL;
1665 priv->funnel[i] = NULL;
1667 gst_object_unref (priv->send_src[0]);
1668 priv->send_src[0] = NULL;
1670 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1671 gst_object_unref (priv->send_src[1]);
1672 priv->send_src[1] = NULL;
1674 g_object_unref (priv->session);
1675 priv->session = NULL;
1677 gst_caps_unref (priv->caps);
1680 priv->is_joined = FALSE;
1681 g_mutex_unlock (&priv->lock);
1692 * gst_rtsp_stream_get_rtpinfo:
1693 * @stream: a #GstRTSPStream
1694 * @rtptime: result RTP timestamp
1695 * @seq: result RTP seqnum
1697 * Retrieve the current rtptime and seq. This is used to
1698 * construct a RTPInfo reply header.
1700 * Returns: %TRUE when rtptime and seq could be determined.
1703 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1704 guint * rtptime, guint * seq)
1706 GstRTSPStreamPrivate *priv;
1707 GObjectClass *payobjclass;
1709 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1710 g_return_val_if_fail (rtptime != NULL, FALSE);
1711 g_return_val_if_fail (seq != NULL, FALSE);
1713 priv = stream->priv;
1715 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1717 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1718 !g_object_class_find_property (payobjclass, "timestamp"))
1721 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1727 * gst_rtsp_stream_get_caps:
1728 * @stream: a #GstRTSPStream
1730 * Retrieve the current caps of @stream.
1732 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1736 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1738 GstRTSPStreamPrivate *priv;
1741 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1743 priv = stream->priv;
1745 g_mutex_lock (&priv->lock);
1746 if ((result = priv->caps))
1747 gst_caps_ref (result);
1748 g_mutex_unlock (&priv->lock);
1754 * gst_rtsp_stream_recv_rtp:
1755 * @stream: a #GstRTSPStream
1756 * @buffer: (transfer full): a #GstBuffer
1758 * Handle an RTP buffer for the stream. This method is usually called when a
1759 * message has been received from a client using the TCP transport.
1761 * This function takes ownership of @buffer.
1763 * Returns: a GstFlowReturn.
1766 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1768 GstRTSPStreamPrivate *priv;
1770 GstElement *element;
1772 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1773 priv = stream->priv;
1774 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1775 g_return_val_if_fail (priv->is_joined, FALSE);
1777 g_mutex_lock (&priv->lock);
1778 element = gst_object_ref (priv->appsrc[0]);
1779 g_mutex_unlock (&priv->lock);
1781 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1783 gst_object_unref (element);
1789 * gst_rtsp_stream_recv_rtcp:
1790 * @stream: a #GstRTSPStream
1791 * @buffer: (transfer full): a #GstBuffer
1793 * Handle an RTCP buffer for the stream. This method is usually called when a
1794 * message has been received from a client using the TCP transport.
1796 * This function takes ownership of @buffer.
1798 * Returns: a GstFlowReturn.
1801 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1803 GstRTSPStreamPrivate *priv;
1805 GstElement *element;
1807 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1808 priv = stream->priv;
1809 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1810 g_return_val_if_fail (priv->is_joined, FALSE);
1812 g_mutex_lock (&priv->lock);
1813 element = gst_object_ref (priv->appsrc[1]);
1814 g_mutex_unlock (&priv->lock);
1816 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1818 gst_object_unref (element);
1823 /* must be called with lock */
1825 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1828 GstRTSPStreamPrivate *priv = stream->priv;
1829 const GstRTSPTransport *tr;
1831 tr = gst_rtsp_stream_transport_get_transport (trans);
1833 switch (tr->lower_transport) {
1834 case GST_RTSP_LOWER_TRANS_UDP:
1835 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1841 dest = tr->destination;
1842 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1847 min = tr->client_port.min;
1848 max = tr->client_port.max;
1852 GST_INFO ("adding %s:%d-%d", dest, min, max);
1853 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1854 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1856 GST_INFO ("setting ttl-mc %d", ttl);
1857 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1858 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1860 priv->transports = g_list_prepend (priv->transports, trans);
1862 GST_INFO ("removing %s:%d-%d", dest, min, max);
1863 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1864 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1865 priv->transports = g_list_remove (priv->transports, trans);
1869 case GST_RTSP_LOWER_TRANS_TCP:
1871 GST_INFO ("adding TCP %s", tr->destination);
1872 priv->transports = g_list_prepend (priv->transports, trans);
1874 GST_INFO ("removing TCP %s", tr->destination);
1875 priv->transports = g_list_remove (priv->transports, trans);
1879 goto unknown_transport;
1886 GST_INFO ("Unknown transport %d", tr->lower_transport);
1893 * gst_rtsp_stream_add_transport:
1894 * @stream: a #GstRTSPStream
1895 * @trans: a #GstRTSPStreamTransport
1897 * Add the transport in @trans to @stream. The media of @stream will
1898 * then also be send to the values configured in @trans.
1900 * @stream must be joined to a bin.
1902 * @trans must contain a valid #GstRTSPTransport.
1904 * Returns: %TRUE if @trans was added
1907 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1908 GstRTSPStreamTransport * trans)
1910 GstRTSPStreamPrivate *priv;
1913 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1914 priv = stream->priv;
1915 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1916 g_return_val_if_fail (priv->is_joined, FALSE);
1918 g_mutex_lock (&priv->lock);
1919 res = update_transport (stream, trans, TRUE);
1920 g_mutex_unlock (&priv->lock);
1926 * gst_rtsp_stream_remove_transport:
1927 * @stream: a #GstRTSPStream
1928 * @trans: a #GstRTSPStreamTransport
1930 * Remove the transport in @trans from @stream. The media of @stream will
1931 * not be sent to the values configured in @trans.
1933 * @stream must be joined to a bin.
1935 * @trans must contain a valid #GstRTSPTransport.
1937 * Returns: %TRUE if @trans was removed
1940 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1941 GstRTSPStreamTransport * trans)
1943 GstRTSPStreamPrivate *priv;
1946 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1947 priv = stream->priv;
1948 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1949 g_return_val_if_fail (priv->is_joined, FALSE);
1951 g_mutex_lock (&priv->lock);
1952 res = update_transport (stream, trans, FALSE);
1953 g_mutex_unlock (&priv->lock);