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;
72 /* server ports for sending/receiving over ipv6 */
73 GstRTSPRange server_port_v6;
74 GstRTSPAddress *server_addr_v6;
76 /* multicast addresses */
77 GstRTSPAddressPool *pool;
80 /* the caps of the stream */
84 /* transports we stream to */
98 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
99 #define GST_CAT_DEFAULT rtsp_stream_debug
101 static GQuark ssrc_stream_map_key;
103 static void gst_rtsp_stream_finalize (GObject * obj);
105 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
108 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
110 GObjectClass *gobject_class;
112 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
114 gobject_class = G_OBJECT_CLASS (klass);
116 gobject_class->finalize = gst_rtsp_stream_finalize;
118 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
120 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
124 gst_rtsp_stream_init (GstRTSPStream * stream)
126 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
128 GST_DEBUG ("new stream %p", stream);
132 stream->priv->dscp_qos = -1;
134 g_mutex_init (&priv->lock);
138 gst_rtsp_stream_finalize (GObject * obj)
140 GstRTSPStream *stream;
141 GstRTSPStreamPrivate *priv;
143 stream = GST_RTSP_STREAM (obj);
146 GST_DEBUG ("finalize stream %p", stream);
148 /* we really need to be unjoined now */
149 g_return_if_fail (!priv->is_joined);
152 gst_rtsp_address_free (priv->addr);
153 if (priv->server_addr_v4)
154 gst_rtsp_address_free (priv->server_addr_v4);
155 if (priv->server_addr_v6)
156 gst_rtsp_address_free (priv->server_addr_v6);
158 g_object_unref (priv->pool);
159 gst_object_unref (priv->payloader);
160 gst_object_unref (priv->srcpad);
161 g_mutex_clear (&priv->lock);
163 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
167 * gst_rtsp_stream_new:
170 * @payloader: a #GstElement
172 * Create a new media stream with index @idx that handles RTP data on
173 * @srcpad and has a payloader element @payloader.
175 * Returns: a new #GstRTSPStream
178 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
180 GstRTSPStreamPrivate *priv;
181 GstRTSPStream *stream;
183 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
184 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
185 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
187 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
190 priv->payloader = gst_object_ref (payloader);
191 priv->srcpad = gst_object_ref (srcpad);
197 * gst_rtsp_stream_get_index:
198 * @stream: a #GstRTSPStream
200 * Get the stream index.
202 * Return: the stream index.
205 gst_rtsp_stream_get_index (GstRTSPStream * stream)
207 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
209 return stream->priv->idx;
213 * gst_rtsp_stream_get_srcpad:
214 * @stream: a #GstRTSPStream
216 * Get the srcpad associated with @stream.
218 * Return: the srcpad. Unref after usage.
221 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
223 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
225 return gst_object_ref (stream->priv->srcpad);
229 * gst_rtsp_stream_set_mtu:
230 * @stream: a #GstRTSPStream
233 * Configure the mtu in the payloader of @stream to @mtu.
236 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
238 GstRTSPStreamPrivate *priv;
240 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
244 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
246 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
250 * gst_rtsp_stream_get_mtu:
251 * @stream: a #GstRTSPStream
253 * Get the configured MTU in the payloader of @stream.
255 * Returns: the MTU of the payloader.
258 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
260 GstRTSPStreamPrivate *priv;
263 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
267 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
272 /* Update the dscp qos property on the udp sinks */
274 update_dscp_qos (GstRTSPStream * stream)
276 GstRTSPStreamPrivate *priv;
278 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
282 if (priv->udpsink[0]) {
283 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
287 if (priv->udpsink[1]) {
288 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
294 * gst_rtsp_stream_set_dscp_qos:
295 * @stream: a #GstRTSPStream
296 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
298 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
301 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
303 GstRTSPStreamPrivate *priv;
305 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
309 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
311 if (dscp_qos < -1 || dscp_qos > 63) {
312 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
316 priv->dscp_qos = dscp_qos;
318 update_dscp_qos (stream);
322 * gst_rtsp_stream_get_dscp_qos:
323 * @stream: a #GstRTSPStream
325 * Get the configured DSCP QoS in of the outgoing sockets.
327 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
330 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
332 GstRTSPStreamPrivate *priv;
334 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
338 return priv->dscp_qos;
343 * gst_rtsp_stream_set_address_pool:
344 * @stream: a #GstRTSPStream
345 * @pool: a #GstRTSPAddressPool
347 * configure @pool to be used as the address pool of @stream.
350 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
351 GstRTSPAddressPool * pool)
353 GstRTSPStreamPrivate *priv;
354 GstRTSPAddressPool *old;
356 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
360 GST_LOG_OBJECT (stream, "set address pool %p", pool);
362 g_mutex_lock (&priv->lock);
363 if ((old = priv->pool) != pool)
364 priv->pool = pool ? g_object_ref (pool) : NULL;
367 g_mutex_unlock (&priv->lock);
370 g_object_unref (old);
374 * gst_rtsp_stream_get_address_pool:
375 * @stream: a #GstRTSPStream
377 * Get the #GstRTSPAddressPool used as the address pool of @stream.
379 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
383 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
385 GstRTSPStreamPrivate *priv;
386 GstRTSPAddressPool *result;
388 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
392 g_mutex_lock (&priv->lock);
393 if ((result = priv->pool))
394 g_object_ref (result);
395 g_mutex_unlock (&priv->lock);
401 * gst_rtsp_stream_get_address:
402 * @stream: a #GstRTSPStream
404 * Get the multicast address of @stream.
406 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
407 * allocated. gst_rtsp_address_free() after usage.
410 gst_rtsp_stream_get_address (GstRTSPStream * stream)
412 GstRTSPStreamPrivate *priv;
413 GstRTSPAddress *result;
415 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
419 g_mutex_lock (&priv->lock);
420 if (priv->addr == NULL) {
421 if (priv->pool == NULL)
424 priv->addr = gst_rtsp_address_pool_acquire_address (priv->pool,
425 GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
426 if (priv->addr == NULL)
429 result = gst_rtsp_address_copy (priv->addr);
430 g_mutex_unlock (&priv->lock);
437 GST_ERROR_OBJECT (stream, "no address pool specified");
438 g_mutex_unlock (&priv->lock);
443 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
444 g_mutex_unlock (&priv->lock);
450 * gst_rtsp_stream_reserve_address:
451 * @stream: a #GstRTSPStream
452 * @address: an address
457 * Reserve @address and @port as the address and port of @stream.
459 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
460 * reserved. gst_rtsp_address_free() after usage.
463 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
464 const gchar * address, guint port, guint n_ports, guint ttl)
466 GstRTSPStreamPrivate *priv;
467 GstRTSPAddress *result;
469 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
470 g_return_val_if_fail (address != NULL, NULL);
471 g_return_val_if_fail (port > 0, NULL);
472 g_return_val_if_fail (n_ports > 0, NULL);
473 g_return_val_if_fail (ttl > 0, NULL);
477 g_mutex_lock (&priv->lock);
478 if (priv->addr == NULL) {
479 if (priv->pool == NULL)
482 priv->addr = gst_rtsp_address_pool_reserve_address (priv->pool, address,
484 if (priv->addr == NULL)
487 if (strcmp (priv->addr->address, address) ||
488 priv->addr->port != port || priv->addr->n_ports != n_ports ||
489 priv->addr->ttl != ttl)
490 goto different_address;
492 result = gst_rtsp_address_copy (priv->addr);
493 g_mutex_unlock (&priv->lock);
500 GST_ERROR_OBJECT (stream, "no address pool specified");
501 g_mutex_unlock (&priv->lock);
506 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
508 g_mutex_unlock (&priv->lock);
513 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
514 " reserved", address);
515 g_mutex_unlock (&priv->lock);
521 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
522 GSocketFamily family, GstElement * udpsrc_out[2],
523 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
524 GstRTSPAddress ** server_addr_out)
526 GstStateChangeReturn ret;
527 GstElement *udpsrc0, *udpsrc1;
528 GstElement *udpsink0, *udpsink1;
529 GSocket *rtp_socket = NULL;
530 GSocket *rtcp_socket;
531 gint tmp_rtp, tmp_rtcp;
533 gint rtpport, rtcpport;
534 GList *rejected_addresses = NULL;
535 GstRTSPAddress *addr = NULL;
536 GInetAddress *inetaddr = NULL;
537 GSocketAddress *rtp_sockaddr = NULL;
538 GSocketAddress *rtcp_sockaddr = NULL;
539 const gchar *multisink_socket = "socket";
541 if (family == G_SOCKET_FAMILY_IPV6) {
542 multisink_socket = "socket-v6";
551 /* Start with random port */
554 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
555 G_SOCKET_PROTOCOL_UDP, NULL);
557 goto no_udp_protocol;
559 if (*server_addr_out)
560 gst_rtsp_address_free (*server_addr_out);
562 /* try to allocate 2 UDP ports, the RTP port should be an even
563 * number and the RTCP port should be the next (uneven) port */
566 if (rtp_socket == NULL) {
567 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
568 G_SOCKET_PROTOCOL_UDP, NULL);
570 goto no_udp_protocol;
573 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
574 GstRTSPAddressFlags flags;
577 rejected_addresses = g_list_prepend (rejected_addresses, addr);
579 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
580 if (family == G_SOCKET_FAMILY_IPV6)
581 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
583 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
585 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
590 tmp_rtp = addr->port;
592 g_clear_object (&inetaddr);
593 inetaddr = g_inet_address_new_from_string (addr->address);
601 if (inetaddr == NULL)
602 inetaddr = g_inet_address_new_any (family);
605 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
606 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
607 g_object_unref (rtp_sockaddr);
610 g_object_unref (rtp_sockaddr);
612 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
613 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
614 g_clear_object (&rtp_sockaddr);
619 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
620 g_object_unref (rtp_sockaddr);
622 /* check if port is even */
623 if ((tmp_rtp & 1) != 0) {
624 /* port not even, close and allocate another */
626 g_clear_object (&rtp_socket);
631 tmp_rtcp = tmp_rtp + 1;
633 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
634 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
635 g_object_unref (rtcp_sockaddr);
636 g_clear_object (&rtp_socket);
639 g_object_unref (rtcp_sockaddr);
641 g_clear_object (&inetaddr);
643 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
644 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
646 if (udpsrc0 == NULL || udpsrc1 == NULL)
647 goto no_udp_protocol;
649 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
650 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
652 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
653 if (ret == GST_STATE_CHANGE_FAILURE)
655 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
656 if (ret == GST_STATE_CHANGE_FAILURE)
659 /* all fine, do port check */
660 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
661 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
663 /* this should not happen... */
664 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
668 udpsink0 = udpsink_out[0];
670 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
673 goto no_udp_protocol;
675 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
676 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
679 udpsink1 = udpsink_out[1];
681 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
684 goto no_udp_protocol;
686 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
687 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
688 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
690 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
691 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
692 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
693 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
694 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
695 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
696 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
697 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
699 /* we keep these elements, we will further configure them when the
700 * client told us to really use the UDP ports. */
701 udpsrc_out[0] = udpsrc0;
702 udpsrc_out[1] = udpsrc1;
703 udpsink_out[0] = udpsink0;
704 udpsink_out[1] = udpsink1;
705 server_port_out->min = rtpport;
706 server_port_out->max = rtcpport;
708 *server_addr_out = addr;
709 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
711 g_object_unref (rtp_socket);
712 g_object_unref (rtcp_socket);
740 gst_element_set_state (udpsrc0, GST_STATE_NULL);
741 gst_object_unref (udpsrc0);
744 gst_element_set_state (udpsrc1, GST_STATE_NULL);
745 gst_object_unref (udpsrc1);
748 gst_element_set_state (udpsink0, GST_STATE_NULL);
749 gst_object_unref (udpsink0);
752 gst_element_set_state (udpsink1, GST_STATE_NULL);
753 gst_object_unref (udpsink1);
756 g_object_unref (inetaddr);
757 g_list_free_full (rejected_addresses,
758 (GDestroyNotify) gst_rtsp_address_free);
760 gst_rtsp_address_free (addr);
762 g_object_unref (rtp_socket);
764 g_object_unref (rtcp_socket);
769 /* must be called with lock */
771 alloc_ports (GstRTSPStream * stream)
773 GstRTSPStreamPrivate *priv = stream->priv;
775 return alloc_ports_one_family (priv->pool, priv->buffer_size,
776 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
777 &priv->server_port_v4, &priv->server_addr_v4) &&
778 alloc_ports_one_family (priv->pool, priv->buffer_size,
779 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
780 &priv->server_port_v6, &priv->server_addr_v6);
784 * gst_rtsp_stream_get_server_port:
785 * @stream: a #GstRTSPStream
786 * @server_port: (out): result server port
788 * Fill @server_port with the port pair used by the server. This function can
789 * only be called when @stream has been joined.
792 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
793 GstRTSPRange * server_port, GSocketFamily family)
795 GstRTSPStreamPrivate *priv;
797 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
799 g_return_if_fail (priv->is_joined);
801 g_mutex_lock (&priv->lock);
802 if (family == G_SOCKET_FAMILY_IPV4) {
804 *server_port = priv->server_port_v4;
807 *server_port = priv->server_port_v6;
809 g_mutex_unlock (&priv->lock);
813 * gst_rtsp_stream_get_rtpsession:
814 * @stream: a #GstRTSPStream
816 * Get the RTP session of this stream.
818 * Returns: The RTP session of this stream. Unref after usage.
821 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
823 GstRTSPStreamPrivate *priv;
826 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
830 g_mutex_lock (&priv->lock);
831 if ((session = priv->session))
832 g_object_ref (session);
833 g_mutex_unlock (&priv->lock);
839 * gst_rtsp_stream_get_ssrc:
840 * @stream: a #GstRTSPStream
841 * @ssrc: (out): result ssrc
843 * Get the SSRC used by the RTP session of this stream. This function can only
844 * be called when @stream has been joined.
847 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
849 GstRTSPStreamPrivate *priv;
851 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
853 g_return_if_fail (priv->is_joined);
855 g_mutex_lock (&priv->lock);
856 if (ssrc && priv->session)
857 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
858 g_mutex_unlock (&priv->lock);
861 /* executed from streaming thread */
863 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
865 GstRTSPStreamPrivate *priv = stream->priv;
866 GstCaps *newcaps, *oldcaps;
868 newcaps = gst_pad_get_current_caps (pad);
870 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
873 g_mutex_lock (&priv->lock);
874 oldcaps = priv->caps;
875 priv->caps = newcaps;
876 g_mutex_unlock (&priv->lock);
879 gst_caps_unref (oldcaps);
883 dump_structure (const GstStructure * s)
887 sstr = gst_structure_to_string (s);
888 GST_INFO ("structure: %s", sstr);
892 static GstRTSPStreamTransport *
893 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
895 GstRTSPStreamPrivate *priv = stream->priv;
897 GstRTSPStreamTransport *result = NULL;
902 if (rtcp_from == NULL)
905 tmp = g_strrstr (rtcp_from, ":");
909 port = atoi (tmp + 1);
910 dest = g_strndup (rtcp_from, tmp - rtcp_from);
912 g_mutex_lock (&priv->lock);
913 GST_INFO ("finding %s:%d in %d transports", dest, port,
914 g_list_length (priv->transports));
916 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
917 GstRTSPStreamTransport *trans = walk->data;
918 const GstRTSPTransport *tr;
921 tr = gst_rtsp_stream_transport_get_transport (trans);
923 min = tr->client_port.min;
924 max = tr->client_port.max;
926 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
932 g_object_ref (result);
933 g_mutex_unlock (&priv->lock);
940 static GstRTSPStreamTransport *
941 check_transport (GObject * source, GstRTSPStream * stream)
944 GstRTSPStreamTransport *trans;
946 /* see if we have a stream to match with the origin of the RTCP packet */
947 trans = g_object_get_qdata (source, ssrc_stream_map_key);
949 g_object_get (source, "stats", &stats, NULL);
951 const gchar *rtcp_from;
953 dump_structure (stats);
955 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
956 if ((trans = find_transport (stream, rtcp_from))) {
957 GST_INFO ("%p: found transport %p for source %p", stream, trans,
959 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
962 gst_structure_free (stats);
970 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
972 GstRTSPStreamTransport *trans;
974 GST_INFO ("%p: new source %p", stream, source);
976 trans = check_transport (source, stream);
979 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
983 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
985 GST_INFO ("%p: new SDES %p", stream, source);
989 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
991 GstRTSPStreamTransport *trans;
993 trans = check_transport (source, stream);
996 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
997 gst_rtsp_stream_transport_keep_alive (trans);
1001 GstStructure *stats;
1002 g_object_get (source, "stats", &stats, NULL);
1004 dump_structure (stats);
1005 gst_structure_free (stats);
1012 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1014 GST_INFO ("%p: source %p bye", stream, source);
1018 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1020 GstRTSPStreamTransport *trans;
1022 GST_INFO ("%p: source %p bye timeout", stream, source);
1024 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1025 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1026 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1031 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1033 GstRTSPStreamTransport *trans;
1035 GST_INFO ("%p: source %p timeout", stream, source);
1037 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1038 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1039 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1043 static GstFlowReturn
1044 handle_new_sample (GstAppSink * sink, gpointer user_data)
1046 GstRTSPStreamPrivate *priv;
1050 GstRTSPStream *stream;
1052 sample = gst_app_sink_pull_sample (sink);
1056 stream = (GstRTSPStream *) user_data;
1057 priv = stream->priv;
1058 buffer = gst_sample_get_buffer (sample);
1060 g_mutex_lock (&priv->lock);
1061 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1062 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1064 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1065 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1067 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1070 g_mutex_unlock (&priv->lock);
1072 gst_sample_unref (sample);
1077 static GstAppSinkCallbacks sink_cb = {
1078 NULL, /* not interested in EOS */
1079 NULL, /* not interested in preroll samples */
1084 * gst_rtsp_stream_join_bin:
1085 * @stream: a #GstRTSPStream
1086 * @bin: a #GstBin to join
1087 * @rtpbin: a rtpbin element in @bin
1088 * @state: the target state of the new elements
1090 * Join the #Gstbin @bin that contains the element @rtpbin.
1092 * @stream will link to @rtpbin, which must be inside @bin. The elements
1093 * added to @bin will be set to the state given in @state.
1095 * Returns: %TRUE on success.
1098 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1099 GstElement * rtpbin, GstState state)
1101 GstRTSPStreamPrivate *priv;
1104 GstPad *pad, *teepad, *queuepad, *selpad;
1105 GstPadLinkReturn ret;
1107 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1108 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1109 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1111 priv = stream->priv;
1113 g_mutex_lock (&priv->lock);
1114 if (priv->is_joined)
1117 /* create a session with the same index as the stream */
1120 GST_INFO ("stream %p joining bin as session %d", stream, idx);
1122 if (!alloc_ports (stream))
1125 /* update the dscp qos field in the sinks */
1126 update_dscp_qos (stream);
1128 /* get a pad for sending RTP */
1129 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1130 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1132 /* link the RTP pad to the session manager, it should not really fail unless
1133 * this is not really an RTP pad */
1134 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1135 if (ret != GST_PAD_LINK_OK)
1138 /* get pads from the RTP session element for sending and receiving
1140 name = g_strdup_printf ("send_rtp_src_%u", idx);
1141 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1143 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1144 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1146 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1147 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1149 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1150 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1153 /* get the session */
1154 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1156 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1158 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1160 g_signal_connect (priv->session, "on-ssrc-active",
1161 (GCallback) on_ssrc_active, stream);
1162 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1164 g_signal_connect (priv->session, "on-bye-timeout",
1165 (GCallback) on_bye_timeout, stream);
1166 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1169 for (i = 0; i < 2; i++) {
1170 /* For the sender we create this bit of pipeline for both
1171 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1172 * we need to add a queue before appsink to make the pipeline
1173 * not block. For the TCP case, we want to pump data to the
1174 * client as fast as possible anyway.
1176 * .--------. .-----. .---------.
1177 * | rtpbin | | tee | | udpsink |
1178 * | send->sink src->sink |
1179 * '--------' | | '---------'
1180 * | | .---------. .---------.
1181 * | | | queue | | appsink |
1182 * | src->sink src->sink |
1183 * '-----' '---------' '---------'
1185 /* make tee for RTP/RTCP */
1186 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1187 gst_bin_add (bin, priv->tee[i]);
1189 /* and link to rtpbin send pad */
1190 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1191 gst_pad_link (priv->send_src[i], pad);
1192 gst_object_unref (pad);
1195 gst_bin_add (bin, priv->udpsink[i]);
1197 /* link tee to udpsink */
1198 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1199 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1200 gst_pad_link (teepad, pad);
1201 gst_object_unref (pad);
1202 gst_object_unref (teepad);
1205 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1206 gst_bin_add (bin, priv->appqueue[i]);
1207 /* and link to tee */
1208 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1209 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1210 gst_pad_link (teepad, pad);
1211 gst_object_unref (pad);
1212 gst_object_unref (teepad);
1215 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1216 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1217 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1218 gst_bin_add (bin, priv->appsink[i]);
1219 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1220 &sink_cb, stream, NULL);
1221 /* and link to queue */
1222 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1223 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1224 gst_pad_link (queuepad, pad);
1225 gst_object_unref (pad);
1226 gst_object_unref (queuepad);
1228 /* For the receiver we create this bit of pipeline for both
1229 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1230 * and it is all funneled into the rtpbin receive pad.
1232 * .--------. .--------. .--------.
1233 * | udpsrc | | funnel | | rtpbin |
1234 * | src->sink src->sink |
1235 * '--------' | | '--------'
1239 * '--------' '--------'
1241 /* make funnel for the RTP/RTCP receivers */
1242 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1243 gst_bin_add (bin, priv->funnel[i]);
1245 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1246 gst_pad_link (pad, priv->recv_sink[i]);
1247 gst_object_unref (pad);
1249 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1251 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1252 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1253 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1254 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1256 gst_bin_add (bin, priv->udpsrc_v4[i]);
1257 gst_bin_add (bin, priv->udpsrc_v6[i]);
1258 /* and link to the funnel v4 */
1259 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1260 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1261 gst_pad_link (pad, selpad);
1262 gst_object_unref (pad);
1263 gst_object_unref (selpad);
1265 /* and link to the funnel v6 */
1266 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1267 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1268 gst_pad_link (pad, selpad);
1269 gst_object_unref (pad);
1270 gst_object_unref (selpad);
1272 /* make and add appsrc */
1273 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1274 gst_bin_add (bin, priv->appsrc[i]);
1275 /* and link to the funnel */
1276 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1277 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1278 gst_pad_link (pad, selpad);
1279 gst_object_unref (pad);
1280 gst_object_unref (selpad);
1282 /* check if we need to set to a special state */
1283 if (state != GST_STATE_NULL) {
1284 gst_element_set_state (priv->udpsink[i], state);
1285 gst_element_set_state (priv->appsink[i], state);
1286 gst_element_set_state (priv->appqueue[i], state);
1287 gst_element_set_state (priv->tee[i], state);
1288 gst_element_set_state (priv->funnel[i], state);
1289 gst_element_set_state (priv->appsrc[i], state);
1293 /* be notified of caps changes */
1294 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1295 (GCallback) caps_notify, stream);
1297 priv->is_joined = TRUE;
1298 g_mutex_unlock (&priv->lock);
1305 g_mutex_unlock (&priv->lock);
1310 g_mutex_unlock (&priv->lock);
1311 GST_WARNING ("failed to allocate ports %d", idx);
1316 GST_WARNING ("failed to link stream %d", idx);
1317 gst_object_unref (priv->send_rtp_sink);
1318 priv->send_rtp_sink = NULL;
1319 g_mutex_unlock (&priv->lock);
1325 * gst_rtsp_stream_leave_bin:
1326 * @stream: a #GstRTSPStream
1328 * @rtpbin: a rtpbin #GstElement
1330 * Remove the elements of @stream from @bin.
1332 * Return: %TRUE on success.
1335 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1336 GstElement * rtpbin)
1338 GstRTSPStreamPrivate *priv;
1341 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1342 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1343 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1345 priv = stream->priv;
1347 g_mutex_lock (&priv->lock);
1348 if (!priv->is_joined)
1349 goto was_not_joined;
1351 /* all transports must be removed by now */
1352 g_return_val_if_fail (priv->transports == NULL, FALSE);
1354 GST_INFO ("stream %p leaving bin", stream);
1356 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1357 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1358 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1359 gst_object_unref (priv->send_rtp_sink);
1360 priv->send_rtp_sink = NULL;
1362 for (i = 0; i < 2; i++) {
1363 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1364 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1365 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1366 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1367 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1368 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1369 /* and set udpsrc to NULL now before removing */
1370 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1371 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1372 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1373 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1375 /* removing them should also nicely release the request
1376 * pads when they finalize */
1377 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1378 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1379 gst_bin_remove (bin, priv->udpsink[i]);
1380 gst_bin_remove (bin, priv->appsrc[i]);
1381 gst_bin_remove (bin, priv->appsink[i]);
1382 gst_bin_remove (bin, priv->appqueue[i]);
1383 gst_bin_remove (bin, priv->tee[i]);
1384 gst_bin_remove (bin, priv->funnel[i]);
1386 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1387 gst_object_unref (priv->recv_sink[i]);
1388 priv->recv_sink[i] = NULL;
1390 priv->udpsrc_v4[i] = NULL;
1391 priv->udpsrc_v6[i] = NULL;
1392 priv->udpsink[i] = NULL;
1393 priv->appsrc[i] = NULL;
1394 priv->appsink[i] = NULL;
1395 priv->appqueue[i] = NULL;
1396 priv->tee[i] = NULL;
1397 priv->funnel[i] = NULL;
1399 gst_object_unref (priv->send_src[0]);
1400 priv->send_src[0] = NULL;
1402 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1403 gst_object_unref (priv->send_src[1]);
1404 priv->send_src[1] = NULL;
1406 g_object_unref (priv->session);
1407 priv->session = NULL;
1409 gst_caps_unref (priv->caps);
1412 priv->is_joined = FALSE;
1413 g_mutex_unlock (&priv->lock);
1424 * gst_rtsp_stream_get_rtpinfo:
1425 * @stream: a #GstRTSPStream
1426 * @rtptime: result RTP timestamp
1427 * @seq: result RTP seqnum
1429 * Retrieve the current rtptime and seq. This is used to
1430 * construct a RTPInfo reply header.
1432 * Returns: %TRUE when rtptime and seq could be determined.
1435 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1436 guint * rtptime, guint * seq)
1438 GstRTSPStreamPrivate *priv;
1439 GObjectClass *payobjclass;
1441 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1442 g_return_val_if_fail (rtptime != NULL, FALSE);
1443 g_return_val_if_fail (seq != NULL, FALSE);
1445 priv = stream->priv;
1447 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1449 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1450 !g_object_class_find_property (payobjclass, "timestamp"))
1453 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1459 * gst_rtsp_stream_get_caps:
1460 * @stream: a #GstRTSPStream
1462 * Retrieve the current caps of @stream.
1464 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1468 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1470 GstRTSPStreamPrivate *priv;
1473 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1475 priv = stream->priv;
1477 g_mutex_lock (&priv->lock);
1478 if ((result = priv->caps))
1479 gst_caps_ref (result);
1480 g_mutex_unlock (&priv->lock);
1486 * gst_rtsp_stream_recv_rtp:
1487 * @stream: a #GstRTSPStream
1488 * @buffer: (transfer full): a #GstBuffer
1490 * Handle an RTP buffer for the stream. This method is usually called when a
1491 * message has been received from a client using the TCP transport.
1493 * This function takes ownership of @buffer.
1495 * Returns: a GstFlowReturn.
1498 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1500 GstRTSPStreamPrivate *priv;
1502 GstElement *element;
1504 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1505 priv = stream->priv;
1506 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1507 g_return_val_if_fail (priv->is_joined, FALSE);
1509 g_mutex_lock (&priv->lock);
1510 element = gst_object_ref (priv->appsrc[0]);
1511 g_mutex_unlock (&priv->lock);
1513 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1515 gst_object_unref (element);
1521 * gst_rtsp_stream_recv_rtcp:
1522 * @stream: a #GstRTSPStream
1523 * @buffer: (transfer full): a #GstBuffer
1525 * Handle an RTCP buffer for the stream. This method is usually called when a
1526 * message has been received from a client using the TCP transport.
1528 * This function takes ownership of @buffer.
1530 * Returns: a GstFlowReturn.
1533 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1535 GstRTSPStreamPrivate *priv;
1537 GstElement *element;
1539 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1540 priv = stream->priv;
1541 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1542 g_return_val_if_fail (priv->is_joined, FALSE);
1544 g_mutex_lock (&priv->lock);
1545 element = gst_object_ref (priv->appsrc[1]);
1546 g_mutex_unlock (&priv->lock);
1548 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1550 gst_object_unref (element);
1555 /* must be called with lock */
1557 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1560 GstRTSPStreamPrivate *priv = stream->priv;
1561 const GstRTSPTransport *tr;
1563 tr = gst_rtsp_stream_transport_get_transport (trans);
1565 switch (tr->lower_transport) {
1566 case GST_RTSP_LOWER_TRANS_UDP:
1567 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1573 dest = tr->destination;
1574 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1579 min = tr->client_port.min;
1580 max = tr->client_port.max;
1584 GST_INFO ("adding %s:%d-%d", dest, min, max);
1585 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1586 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1588 GST_INFO ("setting ttl-mc %d", ttl);
1589 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1590 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1592 priv->transports = g_list_prepend (priv->transports, trans);
1594 GST_INFO ("removing %s:%d-%d", dest, min, max);
1595 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1596 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1597 priv->transports = g_list_remove (priv->transports, trans);
1601 case GST_RTSP_LOWER_TRANS_TCP:
1603 GST_INFO ("adding TCP %s", tr->destination);
1604 priv->transports = g_list_prepend (priv->transports, trans);
1606 GST_INFO ("removing TCP %s", tr->destination);
1607 priv->transports = g_list_remove (priv->transports, trans);
1611 goto unknown_transport;
1618 GST_INFO ("Unknown transport %d", tr->lower_transport);
1625 * gst_rtsp_stream_add_transport:
1626 * @stream: a #GstRTSPStream
1627 * @trans: a #GstRTSPStreamTransport
1629 * Add the transport in @trans to @stream. The media of @stream will
1630 * then also be send to the values configured in @trans.
1632 * @stream must be joined to a bin.
1634 * @trans must contain a valid #GstRTSPTransport.
1636 * Returns: %TRUE if @trans was added
1639 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1640 GstRTSPStreamTransport * trans)
1642 GstRTSPStreamPrivate *priv;
1645 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1646 priv = stream->priv;
1647 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1648 g_return_val_if_fail (priv->is_joined, FALSE);
1650 g_mutex_lock (&priv->lock);
1651 res = update_transport (stream, trans, TRUE);
1652 g_mutex_unlock (&priv->lock);
1658 * gst_rtsp_stream_remove_transport:
1659 * @stream: a #GstRTSPStream
1660 * @trans: a #GstRTSPStreamTransport
1662 * Remove the transport in @trans from @stream. The media of @stream will
1663 * not be sent to the values configured in @trans.
1665 * @stream must be joined to a bin.
1667 * @trans must contain a valid #GstRTSPTransport.
1669 * Returns: %TRUE if @trans was removed
1672 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1673 GstRTSPStreamTransport * trans)
1675 GstRTSPStreamPrivate *priv;
1678 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1679 priv = stream->priv;
1680 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1681 g_return_val_if_fail (priv->is_joined, FALSE);
1683 g_mutex_lock (&priv->lock);
1684 res = update_transport (stream, trans, FALSE);
1685 g_mutex_unlock (&priv->lock);