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.
25 #include <gst/app/gstappsrc.h>
26 #include <gst/app/gstappsink.h>
28 #include "rtsp-stream.h"
30 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
31 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
33 struct _GstRTSPStreamPrivate
38 GstElement *payloader;
42 /* pads on the rtpbin */
43 GstPad *send_rtp_sink;
47 /* the RTPSession object */
50 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
52 GstElement *udpsrc_v4[2];
54 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
56 GstElement *udpsrc_v6[2];
58 GstElement *udpsink[2];
60 /* for TCP transport */
61 GstElement *appsrc[2];
62 GstElement *appqueue[2];
63 GstElement *appsink[2];
66 GstElement *funnel[2];
68 /* server ports for sending/receiving over ipv4 */
69 GstRTSPRange server_port_v4;
70 GstRTSPAddress *server_addr_v4;
73 /* server ports for sending/receiving over ipv6 */
74 GstRTSPRange server_port_v6;
75 GstRTSPAddress *server_addr_v6;
78 /* multicast addresses */
79 GstRTSPAddressPool *pool;
82 /* the caps of the stream */
86 /* transports we stream to */
100 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
101 #define GST_CAT_DEFAULT rtsp_stream_debug
103 static GQuark ssrc_stream_map_key;
105 static void gst_rtsp_stream_finalize (GObject * obj);
107 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
110 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
112 GObjectClass *gobject_class;
114 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
116 gobject_class = G_OBJECT_CLASS (klass);
118 gobject_class->finalize = gst_rtsp_stream_finalize;
120 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
122 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
126 gst_rtsp_stream_init (GstRTSPStream * stream)
128 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
130 GST_DEBUG ("new stream %p", stream);
134 stream->priv->dscp_qos = -1;
136 g_mutex_init (&priv->lock);
140 gst_rtsp_stream_finalize (GObject * obj)
142 GstRTSPStream *stream;
143 GstRTSPStreamPrivate *priv;
145 stream = GST_RTSP_STREAM (obj);
148 GST_DEBUG ("finalize stream %p", stream);
150 /* we really need to be unjoined now */
151 g_return_if_fail (!priv->is_joined);
154 gst_rtsp_address_free (priv->addr);
155 if (priv->server_addr_v4)
156 gst_rtsp_address_free (priv->server_addr_v4);
157 if (priv->server_addr_v6)
158 gst_rtsp_address_free (priv->server_addr_v6);
160 g_object_unref (priv->pool);
161 gst_object_unref (priv->payloader);
162 gst_object_unref (priv->srcpad);
163 g_mutex_clear (&priv->lock);
165 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
169 * gst_rtsp_stream_new:
172 * @payloader: a #GstElement
174 * Create a new media stream with index @idx that handles RTP data on
175 * @srcpad and has a payloader element @payloader.
177 * Returns: a new #GstRTSPStream
180 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
182 GstRTSPStreamPrivate *priv;
183 GstRTSPStream *stream;
185 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
186 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
187 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
189 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
192 priv->payloader = gst_object_ref (payloader);
193 priv->srcpad = gst_object_ref (srcpad);
199 * gst_rtsp_stream_get_index:
200 * @stream: a #GstRTSPStream
202 * Get the stream index.
204 * Return: the stream index.
207 gst_rtsp_stream_get_index (GstRTSPStream * stream)
209 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
211 return stream->priv->idx;
215 * gst_rtsp_stream_get_srcpad:
216 * @stream: a #GstRTSPStream
218 * Get the srcpad associated with @stream.
220 * Return: the srcpad. Unref after usage.
223 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
225 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
227 return gst_object_ref (stream->priv->srcpad);
231 * gst_rtsp_stream_set_mtu:
232 * @stream: a #GstRTSPStream
235 * Configure the mtu in the payloader of @stream to @mtu.
238 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
240 GstRTSPStreamPrivate *priv;
242 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
246 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
248 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
252 * gst_rtsp_stream_get_mtu:
253 * @stream: a #GstRTSPStream
255 * Get the configured MTU in the payloader of @stream.
257 * Returns: the MTU of the payloader.
260 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
262 GstRTSPStreamPrivate *priv;
265 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
269 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
274 /* Update the dscp qos property on the udp sinks */
276 update_dscp_qos (GstRTSPStream * stream)
278 GstRTSPStreamPrivate *priv;
280 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
284 if (priv->udpsink[0]) {
285 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
289 if (priv->udpsink[1]) {
290 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
296 * gst_rtsp_stream_set_dscp_qos:
297 * @stream: a #GstRTSPStream
298 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
300 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
303 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
305 GstRTSPStreamPrivate *priv;
307 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
311 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
313 if (dscp_qos < -1 || dscp_qos > 63) {
314 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
318 priv->dscp_qos = dscp_qos;
320 update_dscp_qos (stream);
324 * gst_rtsp_stream_get_dscp_qos:
325 * @stream: a #GstRTSPStream
327 * Get the configured DSCP QoS in of the outgoing sockets.
329 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
332 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
334 GstRTSPStreamPrivate *priv;
336 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
340 return priv->dscp_qos;
345 * gst_rtsp_stream_set_address_pool:
346 * @stream: a #GstRTSPStream
347 * @pool: a #GstRTSPAddressPool
349 * configure @pool to be used as the address pool of @stream.
352 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
353 GstRTSPAddressPool * pool)
355 GstRTSPStreamPrivate *priv;
356 GstRTSPAddressPool *old;
358 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
362 GST_LOG_OBJECT (stream, "set address pool %p", pool);
364 g_mutex_lock (&priv->lock);
365 if ((old = priv->pool) != pool)
366 priv->pool = pool ? g_object_ref (pool) : NULL;
369 g_mutex_unlock (&priv->lock);
372 g_object_unref (old);
376 * gst_rtsp_stream_get_address_pool:
377 * @stream: a #GstRTSPStream
379 * Get the #GstRTSPAddressPool used as the address pool of @stream.
381 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
385 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
387 GstRTSPStreamPrivate *priv;
388 GstRTSPAddressPool *result;
390 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
394 g_mutex_lock (&priv->lock);
395 if ((result = priv->pool))
396 g_object_ref (result);
397 g_mutex_unlock (&priv->lock);
403 * gst_rtsp_stream_get_address:
404 * @stream: a #GstRTSPStream
406 * Get the multicast address of @stream.
408 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
409 * allocated. gst_rtsp_address_free() after usage.
412 gst_rtsp_stream_get_address (GstRTSPStream * stream)
414 GstRTSPStreamPrivate *priv;
415 GstRTSPAddress *result;
417 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
421 g_mutex_lock (&priv->lock);
422 if (priv->addr == NULL) {
423 if (priv->pool == NULL)
426 priv->addr = gst_rtsp_address_pool_acquire_address (priv->pool,
427 GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
428 if (priv->addr == NULL)
431 result = gst_rtsp_address_copy (priv->addr);
432 g_mutex_unlock (&priv->lock);
439 GST_ERROR_OBJECT (stream, "no address pool specified");
440 g_mutex_unlock (&priv->lock);
445 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
446 g_mutex_unlock (&priv->lock);
452 * gst_rtsp_stream_reserve_address:
453 * @stream: a #GstRTSPStream
454 * @address: an address
459 * Reserve @address and @port as the address and port of @stream.
461 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
462 * reserved. gst_rtsp_address_free() after usage.
465 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
466 const gchar * address, guint port, guint n_ports, guint ttl)
468 GstRTSPStreamPrivate *priv;
469 GstRTSPAddress *result;
471 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
472 g_return_val_if_fail (address != NULL, NULL);
473 g_return_val_if_fail (port > 0, NULL);
474 g_return_val_if_fail (n_ports > 0, NULL);
475 g_return_val_if_fail (ttl > 0, NULL);
479 g_mutex_lock (&priv->lock);
480 if (priv->addr == NULL) {
481 if (priv->pool == NULL)
484 priv->addr = gst_rtsp_address_pool_reserve_address (priv->pool, address,
486 if (priv->addr == NULL)
489 if (strcmp (priv->addr->address, address) ||
490 priv->addr->port != port || priv->addr->n_ports != n_ports ||
491 priv->addr->ttl != ttl)
492 goto different_address;
494 result = gst_rtsp_address_copy (priv->addr);
495 g_mutex_unlock (&priv->lock);
502 GST_ERROR_OBJECT (stream, "no address pool specified");
503 g_mutex_unlock (&priv->lock);
508 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
510 g_mutex_unlock (&priv->lock);
515 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
516 " reserved", address);
517 g_mutex_unlock (&priv->lock);
523 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
524 GSocketFamily family, GstElement * udpsrc_out[2],
525 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
526 GstRTSPAddress ** server_addr_out)
528 GstStateChangeReturn ret;
529 GstElement *udpsrc0, *udpsrc1;
530 GstElement *udpsink0, *udpsink1;
531 GSocket *rtp_socket = NULL;
532 GSocket *rtcp_socket;
533 gint tmp_rtp, tmp_rtcp;
535 gint rtpport, rtcpport;
536 GList *rejected_addresses = NULL;
537 GstRTSPAddress *addr = NULL;
538 GInetAddress *inetaddr = NULL;
539 GSocketAddress *rtp_sockaddr = NULL;
540 GSocketAddress *rtcp_sockaddr = NULL;
541 const gchar *multisink_socket;
543 if (family == G_SOCKET_FAMILY_IPV6)
544 multisink_socket = "socket-v6";
546 multisink_socket = "socket";
554 /* Start with random port */
557 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
558 G_SOCKET_PROTOCOL_UDP, NULL);
560 goto no_udp_protocol;
562 if (*server_addr_out)
563 gst_rtsp_address_free (*server_addr_out);
565 /* try to allocate 2 UDP ports, the RTP port should be an even
566 * number and the RTCP port should be the next (uneven) port */
569 if (rtp_socket == NULL) {
570 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
571 G_SOCKET_PROTOCOL_UDP, NULL);
573 goto no_udp_protocol;
576 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
577 GstRTSPAddressFlags flags;
580 rejected_addresses = g_list_prepend (rejected_addresses, addr);
582 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
583 if (family == G_SOCKET_FAMILY_IPV6)
584 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
586 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
588 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
593 tmp_rtp = addr->port;
595 g_clear_object (&inetaddr);
596 inetaddr = g_inet_address_new_from_string (addr->address);
604 if (inetaddr == NULL)
605 inetaddr = g_inet_address_new_any (family);
608 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
609 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
610 g_object_unref (rtp_sockaddr);
613 g_object_unref (rtp_sockaddr);
615 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
616 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
617 g_clear_object (&rtp_sockaddr);
622 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
623 g_object_unref (rtp_sockaddr);
625 /* check if port is even */
626 if ((tmp_rtp & 1) != 0) {
627 /* port not even, close and allocate another */
629 g_clear_object (&rtp_socket);
634 tmp_rtcp = tmp_rtp + 1;
636 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
637 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
638 g_object_unref (rtcp_sockaddr);
639 g_clear_object (&rtp_socket);
642 g_object_unref (rtcp_sockaddr);
644 g_clear_object (&inetaddr);
646 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
647 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
649 if (udpsrc0 == NULL || udpsrc1 == NULL)
650 goto no_udp_protocol;
652 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
653 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
655 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
656 if (ret == GST_STATE_CHANGE_FAILURE)
658 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
659 if (ret == GST_STATE_CHANGE_FAILURE)
662 /* all fine, do port check */
663 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
664 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
666 /* this should not happen... */
667 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
671 udpsink0 = udpsink_out[0];
673 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
676 goto no_udp_protocol;
678 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
679 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
682 udpsink1 = udpsink_out[1];
684 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
687 goto no_udp_protocol;
689 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
690 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
691 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
693 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
694 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
695 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
696 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
697 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
698 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
699 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
700 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
702 /* we keep these elements, we will further configure them when the
703 * client told us to really use the UDP ports. */
704 udpsrc_out[0] = udpsrc0;
705 udpsrc_out[1] = udpsrc1;
706 udpsink_out[0] = udpsink0;
707 udpsink_out[1] = udpsink1;
708 server_port_out->min = rtpport;
709 server_port_out->max = rtcpport;
711 *server_addr_out = addr;
712 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
714 g_object_unref (rtp_socket);
715 g_object_unref (rtcp_socket);
743 gst_element_set_state (udpsrc0, GST_STATE_NULL);
744 gst_object_unref (udpsrc0);
747 gst_element_set_state (udpsrc1, GST_STATE_NULL);
748 gst_object_unref (udpsrc1);
751 gst_element_set_state (udpsink0, GST_STATE_NULL);
752 gst_object_unref (udpsink0);
755 gst_element_set_state (udpsink1, GST_STATE_NULL);
756 gst_object_unref (udpsink1);
759 g_object_unref (inetaddr);
760 g_list_free_full (rejected_addresses,
761 (GDestroyNotify) gst_rtsp_address_free);
763 gst_rtsp_address_free (addr);
765 g_object_unref (rtp_socket);
767 g_object_unref (rtcp_socket);
772 /* must be called with lock */
774 alloc_ports (GstRTSPStream * stream)
776 GstRTSPStreamPrivate *priv = stream->priv;
778 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
779 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
780 &priv->server_port_v4, &priv->server_addr_v4);
782 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
783 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
784 &priv->server_port_v6, &priv->server_addr_v6);
786 return priv->have_ipv4 || priv->have_ipv6;
790 * gst_rtsp_stream_get_server_port:
791 * @stream: a #GstRTSPStream
792 * @server_port: (out): result server port
794 * Fill @server_port with the port pair used by the server. This function can
795 * only be called when @stream has been joined.
798 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
799 GstRTSPRange * server_port, GSocketFamily family)
801 GstRTSPStreamPrivate *priv;
803 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
805 g_return_if_fail (priv->is_joined);
807 g_mutex_lock (&priv->lock);
808 if (family == G_SOCKET_FAMILY_IPV4) {
810 *server_port = priv->server_port_v4;
813 *server_port = priv->server_port_v6;
815 g_mutex_unlock (&priv->lock);
819 * gst_rtsp_stream_get_rtpsession:
820 * @stream: a #GstRTSPStream
822 * Get the RTP session of this stream.
824 * Returns: The RTP session of this stream. Unref after usage.
827 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
829 GstRTSPStreamPrivate *priv;
832 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
836 g_mutex_lock (&priv->lock);
837 if ((session = priv->session))
838 g_object_ref (session);
839 g_mutex_unlock (&priv->lock);
845 * gst_rtsp_stream_get_ssrc:
846 * @stream: a #GstRTSPStream
847 * @ssrc: (out): result ssrc
849 * Get the SSRC used by the RTP session of this stream. This function can only
850 * be called when @stream has been joined.
853 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
855 GstRTSPStreamPrivate *priv;
857 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
859 g_return_if_fail (priv->is_joined);
861 g_mutex_lock (&priv->lock);
862 if (ssrc && priv->session)
863 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
864 g_mutex_unlock (&priv->lock);
867 /* executed from streaming thread */
869 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
871 GstRTSPStreamPrivate *priv = stream->priv;
872 GstCaps *newcaps, *oldcaps;
874 newcaps = gst_pad_get_current_caps (pad);
876 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
879 g_mutex_lock (&priv->lock);
880 oldcaps = priv->caps;
881 priv->caps = newcaps;
882 g_mutex_unlock (&priv->lock);
885 gst_caps_unref (oldcaps);
889 dump_structure (const GstStructure * s)
893 sstr = gst_structure_to_string (s);
894 GST_INFO ("structure: %s", sstr);
898 static GstRTSPStreamTransport *
899 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
901 GstRTSPStreamPrivate *priv = stream->priv;
903 GstRTSPStreamTransport *result = NULL;
908 if (rtcp_from == NULL)
911 tmp = g_strrstr (rtcp_from, ":");
915 port = atoi (tmp + 1);
916 dest = g_strndup (rtcp_from, tmp - rtcp_from);
918 g_mutex_lock (&priv->lock);
919 GST_INFO ("finding %s:%d in %d transports", dest, port,
920 g_list_length (priv->transports));
922 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
923 GstRTSPStreamTransport *trans = walk->data;
924 const GstRTSPTransport *tr;
927 tr = gst_rtsp_stream_transport_get_transport (trans);
929 min = tr->client_port.min;
930 max = tr->client_port.max;
932 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
938 g_object_ref (result);
939 g_mutex_unlock (&priv->lock);
946 static GstRTSPStreamTransport *
947 check_transport (GObject * source, GstRTSPStream * stream)
950 GstRTSPStreamTransport *trans;
952 /* see if we have a stream to match with the origin of the RTCP packet */
953 trans = g_object_get_qdata (source, ssrc_stream_map_key);
955 g_object_get (source, "stats", &stats, NULL);
957 const gchar *rtcp_from;
959 dump_structure (stats);
961 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
962 if ((trans = find_transport (stream, rtcp_from))) {
963 GST_INFO ("%p: found transport %p for source %p", stream, trans,
965 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
968 gst_structure_free (stats);
976 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
978 GstRTSPStreamTransport *trans;
980 GST_INFO ("%p: new source %p", stream, source);
982 trans = check_transport (source, stream);
985 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
989 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
991 GST_INFO ("%p: new SDES %p", stream, source);
995 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
997 GstRTSPStreamTransport *trans;
999 trans = check_transport (source, stream);
1002 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1003 gst_rtsp_stream_transport_keep_alive (trans);
1007 GstStructure *stats;
1008 g_object_get (source, "stats", &stats, NULL);
1010 dump_structure (stats);
1011 gst_structure_free (stats);
1018 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1020 GST_INFO ("%p: source %p bye", stream, source);
1024 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1026 GstRTSPStreamTransport *trans;
1028 GST_INFO ("%p: source %p bye timeout", stream, source);
1030 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1031 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1032 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1037 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1039 GstRTSPStreamTransport *trans;
1041 GST_INFO ("%p: source %p timeout", stream, source);
1043 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1044 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1045 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1049 static GstFlowReturn
1050 handle_new_sample (GstAppSink * sink, gpointer user_data)
1052 GstRTSPStreamPrivate *priv;
1056 GstRTSPStream *stream;
1058 sample = gst_app_sink_pull_sample (sink);
1062 stream = (GstRTSPStream *) user_data;
1063 priv = stream->priv;
1064 buffer = gst_sample_get_buffer (sample);
1066 g_mutex_lock (&priv->lock);
1067 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1068 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1070 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1071 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1073 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1076 g_mutex_unlock (&priv->lock);
1078 gst_sample_unref (sample);
1083 static GstAppSinkCallbacks sink_cb = {
1084 NULL, /* not interested in EOS */
1085 NULL, /* not interested in preroll samples */
1090 * gst_rtsp_stream_join_bin:
1091 * @stream: a #GstRTSPStream
1092 * @bin: a #GstBin to join
1093 * @rtpbin: a rtpbin element in @bin
1094 * @state: the target state of the new elements
1096 * Join the #Gstbin @bin that contains the element @rtpbin.
1098 * @stream will link to @rtpbin, which must be inside @bin. The elements
1099 * added to @bin will be set to the state given in @state.
1101 * Returns: %TRUE on success.
1104 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1105 GstElement * rtpbin, GstState state)
1107 GstRTSPStreamPrivate *priv;
1110 GstPad *pad, *teepad, *queuepad, *selpad;
1111 GstPadLinkReturn ret;
1113 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1114 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1115 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1117 priv = stream->priv;
1119 g_mutex_lock (&priv->lock);
1120 if (priv->is_joined)
1123 /* create a session with the same index as the stream */
1126 GST_INFO ("stream %p joining bin as session %d", stream, idx);
1128 if (!alloc_ports (stream))
1131 /* update the dscp qos field in the sinks */
1132 update_dscp_qos (stream);
1134 /* get a pad for sending RTP */
1135 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1136 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1138 /* link the RTP pad to the session manager, it should not really fail unless
1139 * this is not really an RTP pad */
1140 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1141 if (ret != GST_PAD_LINK_OK)
1144 /* get pads from the RTP session element for sending and receiving
1146 name = g_strdup_printf ("send_rtp_src_%u", idx);
1147 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1149 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1150 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1152 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1153 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1155 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1156 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1159 /* get the session */
1160 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1162 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1164 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1166 g_signal_connect (priv->session, "on-ssrc-active",
1167 (GCallback) on_ssrc_active, stream);
1168 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1170 g_signal_connect (priv->session, "on-bye-timeout",
1171 (GCallback) on_bye_timeout, stream);
1172 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1175 for (i = 0; i < 2; i++) {
1176 /* For the sender we create this bit of pipeline for both
1177 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1178 * we need to add a queue before appsink to make the pipeline
1179 * not block. For the TCP case, we want to pump data to the
1180 * client as fast as possible anyway.
1182 * .--------. .-----. .---------.
1183 * | rtpbin | | tee | | udpsink |
1184 * | send->sink src->sink |
1185 * '--------' | | '---------'
1186 * | | .---------. .---------.
1187 * | | | queue | | appsink |
1188 * | src->sink src->sink |
1189 * '-----' '---------' '---------'
1191 /* make tee for RTP/RTCP */
1192 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1193 gst_bin_add (bin, priv->tee[i]);
1195 /* and link to rtpbin send pad */
1196 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1197 gst_pad_link (priv->send_src[i], pad);
1198 gst_object_unref (pad);
1201 gst_bin_add (bin, priv->udpsink[i]);
1203 /* link tee to udpsink */
1204 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1205 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1206 gst_pad_link (teepad, pad);
1207 gst_object_unref (pad);
1208 gst_object_unref (teepad);
1211 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1212 gst_bin_add (bin, priv->appqueue[i]);
1213 /* and link to tee */
1214 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1215 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1216 gst_pad_link (teepad, pad);
1217 gst_object_unref (pad);
1218 gst_object_unref (teepad);
1221 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1222 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1223 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1224 gst_bin_add (bin, priv->appsink[i]);
1225 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1226 &sink_cb, stream, NULL);
1227 /* and link to queue */
1228 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1229 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1230 gst_pad_link (queuepad, pad);
1231 gst_object_unref (pad);
1232 gst_object_unref (queuepad);
1234 /* For the receiver we create this bit of pipeline for both
1235 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1236 * and it is all funneled into the rtpbin receive pad.
1238 * .--------. .--------. .--------.
1239 * | udpsrc | | funnel | | rtpbin |
1240 * | src->sink src->sink |
1241 * '--------' | | '--------'
1245 * '--------' '--------'
1247 /* make funnel for the RTP/RTCP receivers */
1248 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1249 gst_bin_add (bin, priv->funnel[i]);
1251 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1252 gst_pad_link (pad, priv->recv_sink[i]);
1253 gst_object_unref (pad);
1255 if (priv->udpsrc_v4[i]) {
1256 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1258 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1259 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1261 gst_bin_add (bin, priv->udpsrc_v4[i]);
1263 /* and link to the funnel v4 */
1264 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1265 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1266 gst_pad_link (pad, selpad);
1267 gst_object_unref (pad);
1268 gst_object_unref (selpad);
1271 if (priv->udpsrc_v6[i]) {
1272 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1273 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1274 gst_bin_add (bin, priv->udpsrc_v6[i]);
1276 /* and link to the funnel v6 */
1277 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1278 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1279 gst_pad_link (pad, selpad);
1280 gst_object_unref (pad);
1281 gst_object_unref (selpad);
1284 /* make and add appsrc */
1285 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1286 gst_bin_add (bin, priv->appsrc[i]);
1287 /* and link to the funnel */
1288 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1289 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1290 gst_pad_link (pad, selpad);
1291 gst_object_unref (pad);
1292 gst_object_unref (selpad);
1294 /* check if we need to set to a special state */
1295 if (state != GST_STATE_NULL) {
1296 gst_element_set_state (priv->udpsink[i], state);
1297 gst_element_set_state (priv->appsink[i], state);
1298 gst_element_set_state (priv->appqueue[i], state);
1299 gst_element_set_state (priv->tee[i], state);
1300 gst_element_set_state (priv->funnel[i], state);
1301 gst_element_set_state (priv->appsrc[i], state);
1305 /* be notified of caps changes */
1306 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1307 (GCallback) caps_notify, stream);
1309 priv->is_joined = TRUE;
1310 g_mutex_unlock (&priv->lock);
1317 g_mutex_unlock (&priv->lock);
1322 g_mutex_unlock (&priv->lock);
1323 GST_WARNING ("failed to allocate ports %d", idx);
1328 GST_WARNING ("failed to link stream %d", idx);
1329 gst_object_unref (priv->send_rtp_sink);
1330 priv->send_rtp_sink = NULL;
1331 g_mutex_unlock (&priv->lock);
1337 * gst_rtsp_stream_leave_bin:
1338 * @stream: a #GstRTSPStream
1340 * @rtpbin: a rtpbin #GstElement
1342 * Remove the elements of @stream from @bin.
1344 * Return: %TRUE on success.
1347 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1348 GstElement * rtpbin)
1350 GstRTSPStreamPrivate *priv;
1353 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1354 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1355 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1357 priv = stream->priv;
1359 g_mutex_lock (&priv->lock);
1360 if (!priv->is_joined)
1361 goto was_not_joined;
1363 /* all transports must be removed by now */
1364 g_return_val_if_fail (priv->transports == NULL, FALSE);
1366 GST_INFO ("stream %p leaving bin", stream);
1368 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1369 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1370 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1371 gst_object_unref (priv->send_rtp_sink);
1372 priv->send_rtp_sink = NULL;
1374 for (i = 0; i < 2; i++) {
1375 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1376 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1377 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1378 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1379 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1380 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1381 if (priv->udpsrc_v4[i]) {
1382 /* and set udpsrc to NULL now before removing */
1383 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1384 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1385 /* removing them should also nicely release the request
1386 * pads when they finalize */
1387 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1389 if (priv->udpsrc_v6[i]) {
1390 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1391 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1392 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1394 gst_bin_remove (bin, priv->udpsink[i]);
1395 gst_bin_remove (bin, priv->appsrc[i]);
1396 gst_bin_remove (bin, priv->appsink[i]);
1397 gst_bin_remove (bin, priv->appqueue[i]);
1398 gst_bin_remove (bin, priv->tee[i]);
1399 gst_bin_remove (bin, priv->funnel[i]);
1401 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1402 gst_object_unref (priv->recv_sink[i]);
1403 priv->recv_sink[i] = NULL;
1405 priv->udpsrc_v4[i] = NULL;
1406 priv->udpsrc_v6[i] = NULL;
1407 priv->udpsink[i] = NULL;
1408 priv->appsrc[i] = NULL;
1409 priv->appsink[i] = NULL;
1410 priv->appqueue[i] = NULL;
1411 priv->tee[i] = NULL;
1412 priv->funnel[i] = NULL;
1414 gst_object_unref (priv->send_src[0]);
1415 priv->send_src[0] = NULL;
1417 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1418 gst_object_unref (priv->send_src[1]);
1419 priv->send_src[1] = NULL;
1421 g_object_unref (priv->session);
1422 priv->session = NULL;
1424 gst_caps_unref (priv->caps);
1427 priv->is_joined = FALSE;
1428 g_mutex_unlock (&priv->lock);
1439 * gst_rtsp_stream_get_rtpinfo:
1440 * @stream: a #GstRTSPStream
1441 * @rtptime: result RTP timestamp
1442 * @seq: result RTP seqnum
1444 * Retrieve the current rtptime and seq. This is used to
1445 * construct a RTPInfo reply header.
1447 * Returns: %TRUE when rtptime and seq could be determined.
1450 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1451 guint * rtptime, guint * seq)
1453 GstRTSPStreamPrivate *priv;
1454 GObjectClass *payobjclass;
1456 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1457 g_return_val_if_fail (rtptime != NULL, FALSE);
1458 g_return_val_if_fail (seq != NULL, FALSE);
1460 priv = stream->priv;
1462 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1464 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1465 !g_object_class_find_property (payobjclass, "timestamp"))
1468 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1474 * gst_rtsp_stream_get_caps:
1475 * @stream: a #GstRTSPStream
1477 * Retrieve the current caps of @stream.
1479 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1483 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1485 GstRTSPStreamPrivate *priv;
1488 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1490 priv = stream->priv;
1492 g_mutex_lock (&priv->lock);
1493 if ((result = priv->caps))
1494 gst_caps_ref (result);
1495 g_mutex_unlock (&priv->lock);
1501 * gst_rtsp_stream_recv_rtp:
1502 * @stream: a #GstRTSPStream
1503 * @buffer: (transfer full): a #GstBuffer
1505 * Handle an RTP buffer for the stream. This method is usually called when a
1506 * message has been received from a client using the TCP transport.
1508 * This function takes ownership of @buffer.
1510 * Returns: a GstFlowReturn.
1513 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1515 GstRTSPStreamPrivate *priv;
1517 GstElement *element;
1519 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1520 priv = stream->priv;
1521 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1522 g_return_val_if_fail (priv->is_joined, FALSE);
1524 g_mutex_lock (&priv->lock);
1525 element = gst_object_ref (priv->appsrc[0]);
1526 g_mutex_unlock (&priv->lock);
1528 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1530 gst_object_unref (element);
1536 * gst_rtsp_stream_recv_rtcp:
1537 * @stream: a #GstRTSPStream
1538 * @buffer: (transfer full): a #GstBuffer
1540 * Handle an RTCP buffer for the stream. This method is usually called when a
1541 * message has been received from a client using the TCP transport.
1543 * This function takes ownership of @buffer.
1545 * Returns: a GstFlowReturn.
1548 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1550 GstRTSPStreamPrivate *priv;
1552 GstElement *element;
1554 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1555 priv = stream->priv;
1556 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1557 g_return_val_if_fail (priv->is_joined, FALSE);
1559 g_mutex_lock (&priv->lock);
1560 element = gst_object_ref (priv->appsrc[1]);
1561 g_mutex_unlock (&priv->lock);
1563 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1565 gst_object_unref (element);
1570 /* must be called with lock */
1572 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1575 GstRTSPStreamPrivate *priv = stream->priv;
1576 const GstRTSPTransport *tr;
1578 tr = gst_rtsp_stream_transport_get_transport (trans);
1580 switch (tr->lower_transport) {
1581 case GST_RTSP_LOWER_TRANS_UDP:
1582 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1588 dest = tr->destination;
1589 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1594 min = tr->client_port.min;
1595 max = tr->client_port.max;
1599 GST_INFO ("adding %s:%d-%d", dest, min, max);
1600 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1601 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1603 GST_INFO ("setting ttl-mc %d", ttl);
1604 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1605 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1607 priv->transports = g_list_prepend (priv->transports, trans);
1609 GST_INFO ("removing %s:%d-%d", dest, min, max);
1610 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1611 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1612 priv->transports = g_list_remove (priv->transports, trans);
1616 case GST_RTSP_LOWER_TRANS_TCP:
1618 GST_INFO ("adding TCP %s", tr->destination);
1619 priv->transports = g_list_prepend (priv->transports, trans);
1621 GST_INFO ("removing TCP %s", tr->destination);
1622 priv->transports = g_list_remove (priv->transports, trans);
1626 goto unknown_transport;
1633 GST_INFO ("Unknown transport %d", tr->lower_transport);
1640 * gst_rtsp_stream_add_transport:
1641 * @stream: a #GstRTSPStream
1642 * @trans: a #GstRTSPStreamTransport
1644 * Add the transport in @trans to @stream. The media of @stream will
1645 * then also be send to the values configured in @trans.
1647 * @stream must be joined to a bin.
1649 * @trans must contain a valid #GstRTSPTransport.
1651 * Returns: %TRUE if @trans was added
1654 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1655 GstRTSPStreamTransport * trans)
1657 GstRTSPStreamPrivate *priv;
1660 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1661 priv = stream->priv;
1662 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1663 g_return_val_if_fail (priv->is_joined, FALSE);
1665 g_mutex_lock (&priv->lock);
1666 res = update_transport (stream, trans, TRUE);
1667 g_mutex_unlock (&priv->lock);
1673 * gst_rtsp_stream_remove_transport:
1674 * @stream: a #GstRTSPStream
1675 * @trans: a #GstRTSPStreamTransport
1677 * Remove the transport in @trans from @stream. The media of @stream will
1678 * not be sent to the values configured in @trans.
1680 * @stream must be joined to a bin.
1682 * @trans must contain a valid #GstRTSPTransport.
1684 * Returns: %TRUE if @trans was removed
1687 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1688 GstRTSPStreamTransport * trans)
1690 GstRTSPStreamPrivate *priv;
1693 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1694 priv = stream->priv;
1695 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1696 g_return_val_if_fail (priv->is_joined, FALSE);
1698 g_mutex_lock (&priv->lock);
1699 res = update_transport (stream, trans, FALSE);
1700 g_mutex_unlock (&priv->lock);