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 /* pads on the rtpbin */
72 GstPad *send_rtp_sink;
76 /* the RTPSession object */
79 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
81 GstElement *udpsrc_v4[2];
83 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
85 GstElement *udpsrc_v6[2];
87 GstElement *udpsink[2];
89 /* for TCP transport */
90 GstElement *appsrc[2];
91 GstElement *appqueue[2];
92 GstElement *appsink[2];
95 GstElement *funnel[2];
97 /* server ports for sending/receiving over ipv4 */
98 GstRTSPRange server_port_v4;
99 GstRTSPAddress *server_addr_v4;
102 /* server ports for sending/receiving over ipv6 */
103 GstRTSPRange server_port_v6;
104 GstRTSPAddress *server_addr_v6;
107 /* multicast addresses */
108 GstRTSPAddressPool *pool;
109 GstRTSPAddress *addr_v4;
110 GstRTSPAddress *addr_v6;
112 /* the caps of the stream */
116 /* transports we stream to */
123 #define DEFAULT_CONTROL NULL
132 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
133 #define GST_CAT_DEFAULT rtsp_stream_debug
135 static GQuark ssrc_stream_map_key;
137 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
138 GValue * value, GParamSpec * pspec);
139 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
140 const GValue * value, GParamSpec * pspec);
142 static void gst_rtsp_stream_finalize (GObject * obj);
144 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
147 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
149 GObjectClass *gobject_class;
151 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
153 gobject_class = G_OBJECT_CLASS (klass);
155 gobject_class->get_property = gst_rtsp_stream_get_property;
156 gobject_class->set_property = gst_rtsp_stream_set_property;
157 gobject_class->finalize = gst_rtsp_stream_finalize;
159 g_object_class_install_property (gobject_class, PROP_CONTROL,
160 g_param_spec_string ("control", "Control",
161 "The control string for this stream", DEFAULT_CONTROL,
162 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
164 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
166 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
170 gst_rtsp_stream_init (GstRTSPStream * stream)
172 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
174 GST_DEBUG ("new stream %p", stream);
179 priv->control = g_strdup (DEFAULT_CONTROL);
181 g_mutex_init (&priv->lock);
185 gst_rtsp_stream_finalize (GObject * obj)
187 GstRTSPStream *stream;
188 GstRTSPStreamPrivate *priv;
190 stream = GST_RTSP_STREAM (obj);
193 GST_DEBUG ("finalize stream %p", stream);
195 /* we really need to be unjoined now */
196 g_return_if_fail (!priv->is_joined);
199 gst_rtsp_address_free (priv->addr_v4);
201 gst_rtsp_address_free (priv->addr_v6);
202 if (priv->server_addr_v4)
203 gst_rtsp_address_free (priv->server_addr_v4);
204 if (priv->server_addr_v6)
205 gst_rtsp_address_free (priv->server_addr_v6);
207 g_object_unref (priv->pool);
208 gst_object_unref (priv->payloader);
209 gst_object_unref (priv->srcpad);
210 g_free (priv->control);
211 g_mutex_clear (&priv->lock);
213 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
217 gst_rtsp_stream_get_property (GObject * object, guint propid,
218 GValue * value, GParamSpec * pspec)
220 GstRTSPStream *stream = GST_RTSP_STREAM (object);
224 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
227 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
232 gst_rtsp_stream_set_property (GObject * object, guint propid,
233 const GValue * value, GParamSpec * pspec)
235 GstRTSPStream *stream = GST_RTSP_STREAM (object);
239 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
242 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
247 * gst_rtsp_stream_new:
250 * @payloader: a #GstElement
252 * Create a new media stream with index @idx that handles RTP data on
253 * @srcpad and has a payloader element @payloader.
255 * Returns: a new #GstRTSPStream
258 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
260 GstRTSPStreamPrivate *priv;
261 GstRTSPStream *stream;
263 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
264 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
265 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
267 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
270 priv->payloader = gst_object_ref (payloader);
271 priv->srcpad = gst_object_ref (srcpad);
277 * gst_rtsp_stream_get_index:
278 * @stream: a #GstRTSPStream
280 * Get the stream index.
282 * Return: the stream index.
285 gst_rtsp_stream_get_index (GstRTSPStream * stream)
287 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
289 return stream->priv->idx;
293 * gst_rtsp_stream_get_srcpad:
294 * @stream: a #GstRTSPStream
296 * Get the srcpad associated with @stream.
298 * Return: the srcpad. Unref after usage.
301 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
303 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
305 return gst_object_ref (stream->priv->srcpad);
309 * gst_rtsp_stream_get_control:
310 * @stream: a #GstRTSPStream
312 * Get the control string to identify this stream.
314 * Return: the control string. free after usage.
317 gst_rtsp_stream_get_control (GstRTSPStream * stream)
319 GstRTSPStreamPrivate *priv;
322 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
326 g_mutex_lock (&priv->lock);
327 if ((result = g_strdup (priv->control)) == NULL)
328 result = g_strdup_printf ("stream=%u", priv->idx);
329 g_mutex_unlock (&priv->lock);
335 * gst_rtsp_stream_set_control:
336 * @stream: a #GstRTSPStream
337 * @control: a control string
339 * Set the control string in @stream.
342 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
344 GstRTSPStreamPrivate *priv;
346 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
350 g_mutex_lock (&priv->lock);
351 g_free (priv->control);
352 priv->control = g_strdup (control);
353 g_mutex_unlock (&priv->lock);
357 * gst_rtsp_stream_has_control:
358 * @stream: a #GstRTSPStream
359 * @control: a control string
361 * Check if @stream has the control string @control.
363 * Returns: %TRUE is @stream has @control as the control string
366 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
368 GstRTSPStreamPrivate *priv;
371 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
375 g_mutex_lock (&priv->lock);
377 res = g_strcmp0 (priv->control, control);
380 sscanf (control, "stream=%u", &streamid);
381 res = (streamid == priv->idx);
383 g_mutex_unlock (&priv->lock);
389 * gst_rtsp_stream_set_mtu:
390 * @stream: a #GstRTSPStream
393 * Configure the mtu in the payloader of @stream to @mtu.
396 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
398 GstRTSPStreamPrivate *priv;
400 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
404 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
406 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
410 * gst_rtsp_stream_get_mtu:
411 * @stream: a #GstRTSPStream
413 * Get the configured MTU in the payloader of @stream.
415 * Returns: the MTU of the payloader.
418 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
420 GstRTSPStreamPrivate *priv;
423 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
427 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
432 /* Update the dscp qos property on the udp sinks */
434 update_dscp_qos (GstRTSPStream * stream)
436 GstRTSPStreamPrivate *priv;
438 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
442 if (priv->udpsink[0]) {
443 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
447 if (priv->udpsink[1]) {
448 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
454 * gst_rtsp_stream_set_dscp_qos:
455 * @stream: a #GstRTSPStream
456 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
458 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
461 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
463 GstRTSPStreamPrivate *priv;
465 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
469 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
471 if (dscp_qos < -1 || dscp_qos > 63) {
472 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
476 priv->dscp_qos = dscp_qos;
478 update_dscp_qos (stream);
482 * gst_rtsp_stream_get_dscp_qos:
483 * @stream: a #GstRTSPStream
485 * Get the configured DSCP QoS in of the outgoing sockets.
487 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
490 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
492 GstRTSPStreamPrivate *priv;
494 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
498 return priv->dscp_qos;
503 * gst_rtsp_stream_set_address_pool:
504 * @stream: a #GstRTSPStream
505 * @pool: a #GstRTSPAddressPool
507 * configure @pool to be used as the address pool of @stream.
510 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
511 GstRTSPAddressPool * pool)
513 GstRTSPStreamPrivate *priv;
514 GstRTSPAddressPool *old;
516 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
520 GST_LOG_OBJECT (stream, "set address pool %p", pool);
522 g_mutex_lock (&priv->lock);
523 if ((old = priv->pool) != pool)
524 priv->pool = pool ? g_object_ref (pool) : NULL;
527 g_mutex_unlock (&priv->lock);
530 g_object_unref (old);
534 * gst_rtsp_stream_get_address_pool:
535 * @stream: a #GstRTSPStream
537 * Get the #GstRTSPAddressPool used as the address pool of @stream.
539 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
543 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
545 GstRTSPStreamPrivate *priv;
546 GstRTSPAddressPool *result;
548 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
552 g_mutex_lock (&priv->lock);
553 if ((result = priv->pool))
554 g_object_ref (result);
555 g_mutex_unlock (&priv->lock);
561 * gst_rtsp_stream_get_multicast_address:
562 * @stream: a #GstRTSPStream
563 * @family: the #GSocketFamily
565 * Get the multicast address of @stream for @family.
567 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
568 * allocated. gst_rtsp_address_free() after usage.
571 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
572 GSocketFamily family)
574 GstRTSPStreamPrivate *priv;
575 GstRTSPAddress *result;
576 GstRTSPAddress **addrp;
577 GstRTSPAddressFlags flags;
579 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
583 if (family == G_SOCKET_FAMILY_IPV6) {
584 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
585 addrp = &priv->addr_v4;
587 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
588 addrp = &priv->addr_v6;
591 g_mutex_lock (&priv->lock);
592 if (*addrp == NULL) {
593 if (priv->pool == NULL)
596 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
598 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
602 result = gst_rtsp_address_copy (*addrp);
603 g_mutex_unlock (&priv->lock);
610 GST_ERROR_OBJECT (stream, "no address pool specified");
611 g_mutex_unlock (&priv->lock);
616 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
617 g_mutex_unlock (&priv->lock);
623 * gst_rtsp_stream_reserve_address:
624 * @stream: a #GstRTSPStream
625 * @address: an address
630 * Reserve @address and @port as the address and port of @stream.
632 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
633 * reserved. gst_rtsp_address_free() after usage.
636 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
637 const gchar * address, guint port, guint n_ports, guint ttl)
639 GstRTSPStreamPrivate *priv;
640 GstRTSPAddress *result;
642 GSocketFamily family;
643 GstRTSPAddress **addrp;
645 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
646 g_return_val_if_fail (address != NULL, NULL);
647 g_return_val_if_fail (port > 0, NULL);
648 g_return_val_if_fail (n_ports > 0, NULL);
649 g_return_val_if_fail (ttl > 0, NULL);
653 addr = g_inet_address_new_from_string (address);
655 GST_ERROR ("failed to get inet addr from %s", address);
656 family = G_SOCKET_FAMILY_IPV4;
658 family = g_inet_address_get_family (addr);
659 g_object_unref (addr);
662 if (family == G_SOCKET_FAMILY_IPV6)
663 addrp = &priv->addr_v4;
665 addrp = &priv->addr_v6;
667 g_mutex_lock (&priv->lock);
668 if (*addrp == NULL) {
669 if (priv->pool == NULL)
672 *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
677 if (strcmp ((*addrp)->address, address) ||
678 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
679 (*addrp)->ttl != ttl)
680 goto different_address;
682 result = gst_rtsp_address_copy (*addrp);
683 g_mutex_unlock (&priv->lock);
690 GST_ERROR_OBJECT (stream, "no address pool specified");
691 g_mutex_unlock (&priv->lock);
696 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
698 g_mutex_unlock (&priv->lock);
703 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
704 " reserved", address);
705 g_mutex_unlock (&priv->lock);
711 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
712 GSocketFamily family, GstElement * udpsrc_out[2],
713 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
714 GstRTSPAddress ** server_addr_out)
716 GstStateChangeReturn ret;
717 GstElement *udpsrc0, *udpsrc1;
718 GstElement *udpsink0, *udpsink1;
719 GSocket *rtp_socket = NULL;
720 GSocket *rtcp_socket;
721 gint tmp_rtp, tmp_rtcp;
723 gint rtpport, rtcpport;
724 GList *rejected_addresses = NULL;
725 GstRTSPAddress *addr = NULL;
726 GInetAddress *inetaddr = NULL;
727 GSocketAddress *rtp_sockaddr = NULL;
728 GSocketAddress *rtcp_sockaddr = NULL;
729 const gchar *multisink_socket;
731 if (family == G_SOCKET_FAMILY_IPV6)
732 multisink_socket = "socket-v6";
734 multisink_socket = "socket";
742 /* Start with random port */
745 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
746 G_SOCKET_PROTOCOL_UDP, NULL);
748 goto no_udp_protocol;
750 if (*server_addr_out)
751 gst_rtsp_address_free (*server_addr_out);
753 /* try to allocate 2 UDP ports, the RTP port should be an even
754 * number and the RTCP port should be the next (uneven) port */
757 if (rtp_socket == NULL) {
758 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
759 G_SOCKET_PROTOCOL_UDP, NULL);
761 goto no_udp_protocol;
764 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
765 GstRTSPAddressFlags flags;
768 rejected_addresses = g_list_prepend (rejected_addresses, addr);
770 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
771 if (family == G_SOCKET_FAMILY_IPV6)
772 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
774 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
776 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
781 tmp_rtp = addr->port;
783 g_clear_object (&inetaddr);
784 inetaddr = g_inet_address_new_from_string (addr->address);
792 if (inetaddr == NULL)
793 inetaddr = g_inet_address_new_any (family);
796 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
797 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
798 g_object_unref (rtp_sockaddr);
801 g_object_unref (rtp_sockaddr);
803 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
804 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
805 g_clear_object (&rtp_sockaddr);
810 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
811 g_object_unref (rtp_sockaddr);
813 /* check if port is even */
814 if ((tmp_rtp & 1) != 0) {
815 /* port not even, close and allocate another */
817 g_clear_object (&rtp_socket);
822 tmp_rtcp = tmp_rtp + 1;
824 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
825 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
826 g_object_unref (rtcp_sockaddr);
827 g_clear_object (&rtp_socket);
830 g_object_unref (rtcp_sockaddr);
832 g_clear_object (&inetaddr);
834 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
835 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
837 if (udpsrc0 == NULL || udpsrc1 == NULL)
838 goto no_udp_protocol;
840 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
841 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
843 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
844 if (ret == GST_STATE_CHANGE_FAILURE)
846 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
847 if (ret == GST_STATE_CHANGE_FAILURE)
850 /* all fine, do port check */
851 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
852 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
854 /* this should not happen... */
855 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
859 udpsink0 = udpsink_out[0];
861 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
864 goto no_udp_protocol;
866 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
867 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
870 udpsink1 = udpsink_out[1];
872 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
875 goto no_udp_protocol;
877 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
878 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
879 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
881 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
882 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
883 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
884 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
885 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
886 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
887 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
888 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
890 /* we keep these elements, we will further configure them when the
891 * client told us to really use the UDP ports. */
892 udpsrc_out[0] = udpsrc0;
893 udpsrc_out[1] = udpsrc1;
894 udpsink_out[0] = udpsink0;
895 udpsink_out[1] = udpsink1;
896 server_port_out->min = rtpport;
897 server_port_out->max = rtcpport;
899 *server_addr_out = addr;
900 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
902 g_object_unref (rtp_socket);
903 g_object_unref (rtcp_socket);
931 gst_element_set_state (udpsrc0, GST_STATE_NULL);
932 gst_object_unref (udpsrc0);
935 gst_element_set_state (udpsrc1, GST_STATE_NULL);
936 gst_object_unref (udpsrc1);
939 gst_element_set_state (udpsink0, GST_STATE_NULL);
940 gst_object_unref (udpsink0);
943 gst_element_set_state (udpsink1, GST_STATE_NULL);
944 gst_object_unref (udpsink1);
947 g_object_unref (inetaddr);
948 g_list_free_full (rejected_addresses,
949 (GDestroyNotify) gst_rtsp_address_free);
951 gst_rtsp_address_free (addr);
953 g_object_unref (rtp_socket);
955 g_object_unref (rtcp_socket);
960 /* must be called with lock */
962 alloc_ports (GstRTSPStream * stream)
964 GstRTSPStreamPrivate *priv = stream->priv;
966 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
967 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
968 &priv->server_port_v4, &priv->server_addr_v4);
970 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
971 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
972 &priv->server_port_v6, &priv->server_addr_v6);
974 return priv->have_ipv4 || priv->have_ipv6;
978 * gst_rtsp_stream_get_server_port:
979 * @stream: a #GstRTSPStream
980 * @server_port: (out): result server port
981 * @family: the port family to get
983 * Fill @server_port with the port pair used by the server. This function can
984 * only be called when @stream has been joined.
987 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
988 GstRTSPRange * server_port, GSocketFamily family)
990 GstRTSPStreamPrivate *priv;
992 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
994 g_return_if_fail (priv->is_joined);
996 g_mutex_lock (&priv->lock);
997 if (family == G_SOCKET_FAMILY_IPV4) {
999 *server_port = priv->server_port_v4;
1002 *server_port = priv->server_port_v6;
1004 g_mutex_unlock (&priv->lock);
1008 * gst_rtsp_stream_get_rtpsession:
1009 * @stream: a #GstRTSPStream
1011 * Get the RTP session of this stream.
1013 * Returns: The RTP session of this stream. Unref after usage.
1016 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1018 GstRTSPStreamPrivate *priv;
1021 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1023 priv = stream->priv;
1025 g_mutex_lock (&priv->lock);
1026 if ((session = priv->session))
1027 g_object_ref (session);
1028 g_mutex_unlock (&priv->lock);
1034 * gst_rtsp_stream_get_ssrc:
1035 * @stream: a #GstRTSPStream
1036 * @ssrc: (out): result ssrc
1038 * Get the SSRC used by the RTP session of this stream. This function can only
1039 * be called when @stream has been joined.
1042 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1044 GstRTSPStreamPrivate *priv;
1046 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1047 priv = stream->priv;
1048 g_return_if_fail (priv->is_joined);
1050 g_mutex_lock (&priv->lock);
1051 if (ssrc && priv->session)
1052 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1053 g_mutex_unlock (&priv->lock);
1056 /* executed from streaming thread */
1058 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1060 GstRTSPStreamPrivate *priv = stream->priv;
1061 GstCaps *newcaps, *oldcaps;
1063 newcaps = gst_pad_get_current_caps (pad);
1065 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1068 g_mutex_lock (&priv->lock);
1069 oldcaps = priv->caps;
1070 priv->caps = newcaps;
1071 g_mutex_unlock (&priv->lock);
1074 gst_caps_unref (oldcaps);
1078 dump_structure (const GstStructure * s)
1082 sstr = gst_structure_to_string (s);
1083 GST_INFO ("structure: %s", sstr);
1087 static GstRTSPStreamTransport *
1088 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1090 GstRTSPStreamPrivate *priv = stream->priv;
1092 GstRTSPStreamTransport *result = NULL;
1097 if (rtcp_from == NULL)
1100 tmp = g_strrstr (rtcp_from, ":");
1104 port = atoi (tmp + 1);
1105 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1107 g_mutex_lock (&priv->lock);
1108 GST_INFO ("finding %s:%d in %d transports", dest, port,
1109 g_list_length (priv->transports));
1111 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1112 GstRTSPStreamTransport *trans = walk->data;
1113 const GstRTSPTransport *tr;
1116 tr = gst_rtsp_stream_transport_get_transport (trans);
1118 min = tr->client_port.min;
1119 max = tr->client_port.max;
1121 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1127 g_object_ref (result);
1128 g_mutex_unlock (&priv->lock);
1135 static GstRTSPStreamTransport *
1136 check_transport (GObject * source, GstRTSPStream * stream)
1138 GstStructure *stats;
1139 GstRTSPStreamTransport *trans;
1141 /* see if we have a stream to match with the origin of the RTCP packet */
1142 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1143 if (trans == NULL) {
1144 g_object_get (source, "stats", &stats, NULL);
1146 const gchar *rtcp_from;
1148 dump_structure (stats);
1150 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1151 if ((trans = find_transport (stream, rtcp_from))) {
1152 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1154 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1157 gst_structure_free (stats);
1165 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1167 GstRTSPStreamTransport *trans;
1169 GST_INFO ("%p: new source %p", stream, source);
1171 trans = check_transport (source, stream);
1174 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1178 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1180 GST_INFO ("%p: new SDES %p", stream, source);
1184 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1186 GstRTSPStreamTransport *trans;
1188 trans = check_transport (source, stream);
1191 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1192 gst_rtsp_stream_transport_keep_alive (trans);
1196 GstStructure *stats;
1197 g_object_get (source, "stats", &stats, NULL);
1199 dump_structure (stats);
1200 gst_structure_free (stats);
1207 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1209 GST_INFO ("%p: source %p bye", stream, source);
1213 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1215 GstRTSPStreamTransport *trans;
1217 GST_INFO ("%p: source %p bye timeout", stream, source);
1219 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1220 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1221 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1226 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1228 GstRTSPStreamTransport *trans;
1230 GST_INFO ("%p: source %p timeout", stream, source);
1232 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1233 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1234 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1238 static GstFlowReturn
1239 handle_new_sample (GstAppSink * sink, gpointer user_data)
1241 GstRTSPStreamPrivate *priv;
1245 GstRTSPStream *stream;
1247 sample = gst_app_sink_pull_sample (sink);
1251 stream = (GstRTSPStream *) user_data;
1252 priv = stream->priv;
1253 buffer = gst_sample_get_buffer (sample);
1255 g_mutex_lock (&priv->lock);
1256 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1257 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1259 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1260 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1262 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1265 g_mutex_unlock (&priv->lock);
1267 gst_sample_unref (sample);
1272 static GstAppSinkCallbacks sink_cb = {
1273 NULL, /* not interested in EOS */
1274 NULL, /* not interested in preroll samples */
1279 * gst_rtsp_stream_join_bin:
1280 * @stream: a #GstRTSPStream
1281 * @bin: a #GstBin to join
1282 * @rtpbin: a rtpbin element in @bin
1283 * @state: the target state of the new elements
1285 * Join the #Gstbin @bin that contains the element @rtpbin.
1287 * @stream will link to @rtpbin, which must be inside @bin. The elements
1288 * added to @bin will be set to the state given in @state.
1290 * Returns: %TRUE on success.
1293 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1294 GstElement * rtpbin, GstState state)
1296 GstRTSPStreamPrivate *priv;
1300 GstPad *pad, *teepad, *queuepad, *selpad;
1301 GstPadLinkReturn ret;
1303 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1304 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1305 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1307 priv = stream->priv;
1309 g_mutex_lock (&priv->lock);
1310 if (priv->is_joined)
1313 /* create a session with the same index as the stream */
1316 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1318 if (!alloc_ports (stream))
1321 /* update the dscp qos field in the sinks */
1322 update_dscp_qos (stream);
1324 /* get a pad for sending RTP */
1325 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1326 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1328 /* link the RTP pad to the session manager, it should not really fail unless
1329 * this is not really an RTP pad */
1330 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1331 if (ret != GST_PAD_LINK_OK)
1334 /* get pads from the RTP session element for sending and receiving
1336 name = g_strdup_printf ("send_rtp_src_%u", idx);
1337 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1339 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1340 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1342 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1343 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1345 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1346 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1349 /* get the session */
1350 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1352 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1354 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1356 g_signal_connect (priv->session, "on-ssrc-active",
1357 (GCallback) on_ssrc_active, stream);
1358 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1360 g_signal_connect (priv->session, "on-bye-timeout",
1361 (GCallback) on_bye_timeout, stream);
1362 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1365 for (i = 0; i < 2; i++) {
1366 /* For the sender we create this bit of pipeline for both
1367 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1368 * we need to add a queue before appsink to make the pipeline
1369 * not block. For the TCP case, we want to pump data to the
1370 * client as fast as possible anyway.
1372 * .--------. .-----. .---------.
1373 * | rtpbin | | tee | | udpsink |
1374 * | send->sink src->sink |
1375 * '--------' | | '---------'
1376 * | | .---------. .---------.
1377 * | | | queue | | appsink |
1378 * | src->sink src->sink |
1379 * '-----' '---------' '---------'
1381 /* make tee for RTP/RTCP */
1382 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1383 gst_bin_add (bin, priv->tee[i]);
1385 /* and link to rtpbin send pad */
1386 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1387 gst_pad_link (priv->send_src[i], pad);
1388 gst_object_unref (pad);
1391 gst_bin_add (bin, priv->udpsink[i]);
1393 /* link tee to udpsink */
1394 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1395 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1396 gst_pad_link (teepad, pad);
1397 gst_object_unref (pad);
1398 gst_object_unref (teepad);
1401 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1402 gst_bin_add (bin, priv->appqueue[i]);
1403 /* and link to tee */
1404 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1405 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1406 gst_pad_link (teepad, pad);
1407 gst_object_unref (pad);
1408 gst_object_unref (teepad);
1411 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1412 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1413 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1414 gst_bin_add (bin, priv->appsink[i]);
1415 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1416 &sink_cb, stream, NULL);
1417 /* and link to queue */
1418 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1419 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1420 gst_pad_link (queuepad, pad);
1421 gst_object_unref (pad);
1422 gst_object_unref (queuepad);
1424 /* For the receiver we create this bit of pipeline for both
1425 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1426 * and it is all funneled into the rtpbin receive pad.
1428 * .--------. .--------. .--------.
1429 * | udpsrc | | funnel | | rtpbin |
1430 * | src->sink src->sink |
1431 * '--------' | | '--------'
1435 * '--------' '--------'
1437 /* make funnel for the RTP/RTCP receivers */
1438 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1439 gst_bin_add (bin, priv->funnel[i]);
1441 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1442 gst_pad_link (pad, priv->recv_sink[i]);
1443 gst_object_unref (pad);
1445 if (priv->udpsrc_v4[i]) {
1446 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1448 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1449 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1451 gst_bin_add (bin, priv->udpsrc_v4[i]);
1453 /* and link to the funnel v4 */
1454 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1455 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1456 gst_pad_link (pad, selpad);
1457 gst_object_unref (pad);
1458 gst_object_unref (selpad);
1461 if (priv->udpsrc_v6[i]) {
1462 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1463 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1464 gst_bin_add (bin, priv->udpsrc_v6[i]);
1466 /* and link to the funnel v6 */
1467 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1468 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1469 gst_pad_link (pad, selpad);
1470 gst_object_unref (pad);
1471 gst_object_unref (selpad);
1474 /* make and add appsrc */
1475 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1476 gst_bin_add (bin, priv->appsrc[i]);
1477 /* and link to the funnel */
1478 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1479 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1480 gst_pad_link (pad, selpad);
1481 gst_object_unref (pad);
1482 gst_object_unref (selpad);
1484 /* check if we need to set to a special state */
1485 if (state != GST_STATE_NULL) {
1486 gst_element_set_state (priv->udpsink[i], state);
1487 gst_element_set_state (priv->appsink[i], state);
1488 gst_element_set_state (priv->appqueue[i], state);
1489 gst_element_set_state (priv->tee[i], state);
1490 gst_element_set_state (priv->funnel[i], state);
1491 gst_element_set_state (priv->appsrc[i], state);
1495 /* be notified of caps changes */
1496 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1497 (GCallback) caps_notify, stream);
1499 priv->is_joined = TRUE;
1500 g_mutex_unlock (&priv->lock);
1507 g_mutex_unlock (&priv->lock);
1512 g_mutex_unlock (&priv->lock);
1513 GST_WARNING ("failed to allocate ports %u", idx);
1518 GST_WARNING ("failed to link stream %u", idx);
1519 gst_object_unref (priv->send_rtp_sink);
1520 priv->send_rtp_sink = NULL;
1521 g_mutex_unlock (&priv->lock);
1527 * gst_rtsp_stream_leave_bin:
1528 * @stream: a #GstRTSPStream
1530 * @rtpbin: a rtpbin #GstElement
1532 * Remove the elements of @stream from @bin.
1534 * Return: %TRUE on success.
1537 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1538 GstElement * rtpbin)
1540 GstRTSPStreamPrivate *priv;
1543 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1544 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1545 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1547 priv = stream->priv;
1549 g_mutex_lock (&priv->lock);
1550 if (!priv->is_joined)
1551 goto was_not_joined;
1553 /* all transports must be removed by now */
1554 g_return_val_if_fail (priv->transports == NULL, FALSE);
1556 GST_INFO ("stream %p leaving bin", stream);
1558 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1559 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1560 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1561 gst_object_unref (priv->send_rtp_sink);
1562 priv->send_rtp_sink = NULL;
1564 for (i = 0; i < 2; i++) {
1565 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1566 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1567 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1568 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1569 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1570 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1571 if (priv->udpsrc_v4[i]) {
1572 /* and set udpsrc to NULL now before removing */
1573 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1574 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1575 /* removing them should also nicely release the request
1576 * pads when they finalize */
1577 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1579 if (priv->udpsrc_v6[i]) {
1580 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1581 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1582 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1584 gst_bin_remove (bin, priv->udpsink[i]);
1585 gst_bin_remove (bin, priv->appsrc[i]);
1586 gst_bin_remove (bin, priv->appsink[i]);
1587 gst_bin_remove (bin, priv->appqueue[i]);
1588 gst_bin_remove (bin, priv->tee[i]);
1589 gst_bin_remove (bin, priv->funnel[i]);
1591 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1592 gst_object_unref (priv->recv_sink[i]);
1593 priv->recv_sink[i] = NULL;
1595 priv->udpsrc_v4[i] = NULL;
1596 priv->udpsrc_v6[i] = NULL;
1597 priv->udpsink[i] = NULL;
1598 priv->appsrc[i] = NULL;
1599 priv->appsink[i] = NULL;
1600 priv->appqueue[i] = NULL;
1601 priv->tee[i] = NULL;
1602 priv->funnel[i] = NULL;
1604 gst_object_unref (priv->send_src[0]);
1605 priv->send_src[0] = NULL;
1607 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1608 gst_object_unref (priv->send_src[1]);
1609 priv->send_src[1] = NULL;
1611 g_object_unref (priv->session);
1612 priv->session = NULL;
1614 gst_caps_unref (priv->caps);
1617 priv->is_joined = FALSE;
1618 g_mutex_unlock (&priv->lock);
1629 * gst_rtsp_stream_get_rtpinfo:
1630 * @stream: a #GstRTSPStream
1631 * @rtptime: result RTP timestamp
1632 * @seq: result RTP seqnum
1634 * Retrieve the current rtptime and seq. This is used to
1635 * construct a RTPInfo reply header.
1637 * Returns: %TRUE when rtptime and seq could be determined.
1640 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1641 guint * rtptime, guint * seq)
1643 GstRTSPStreamPrivate *priv;
1644 GObjectClass *payobjclass;
1646 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1647 g_return_val_if_fail (rtptime != NULL, FALSE);
1648 g_return_val_if_fail (seq != NULL, FALSE);
1650 priv = stream->priv;
1652 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1654 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1655 !g_object_class_find_property (payobjclass, "timestamp"))
1658 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1664 * gst_rtsp_stream_get_caps:
1665 * @stream: a #GstRTSPStream
1667 * Retrieve the current caps of @stream.
1669 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1673 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1675 GstRTSPStreamPrivate *priv;
1678 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1680 priv = stream->priv;
1682 g_mutex_lock (&priv->lock);
1683 if ((result = priv->caps))
1684 gst_caps_ref (result);
1685 g_mutex_unlock (&priv->lock);
1691 * gst_rtsp_stream_recv_rtp:
1692 * @stream: a #GstRTSPStream
1693 * @buffer: (transfer full): a #GstBuffer
1695 * Handle an RTP buffer for the stream. This method is usually called when a
1696 * message has been received from a client using the TCP transport.
1698 * This function takes ownership of @buffer.
1700 * Returns: a GstFlowReturn.
1703 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1705 GstRTSPStreamPrivate *priv;
1707 GstElement *element;
1709 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1710 priv = stream->priv;
1711 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1712 g_return_val_if_fail (priv->is_joined, FALSE);
1714 g_mutex_lock (&priv->lock);
1715 element = gst_object_ref (priv->appsrc[0]);
1716 g_mutex_unlock (&priv->lock);
1718 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1720 gst_object_unref (element);
1726 * gst_rtsp_stream_recv_rtcp:
1727 * @stream: a #GstRTSPStream
1728 * @buffer: (transfer full): a #GstBuffer
1730 * Handle an RTCP buffer for the stream. This method is usually called when a
1731 * message has been received from a client using the TCP transport.
1733 * This function takes ownership of @buffer.
1735 * Returns: a GstFlowReturn.
1738 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1740 GstRTSPStreamPrivate *priv;
1742 GstElement *element;
1744 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1745 priv = stream->priv;
1746 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1747 g_return_val_if_fail (priv->is_joined, FALSE);
1749 g_mutex_lock (&priv->lock);
1750 element = gst_object_ref (priv->appsrc[1]);
1751 g_mutex_unlock (&priv->lock);
1753 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1755 gst_object_unref (element);
1760 /* must be called with lock */
1762 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1765 GstRTSPStreamPrivate *priv = stream->priv;
1766 const GstRTSPTransport *tr;
1768 tr = gst_rtsp_stream_transport_get_transport (trans);
1770 switch (tr->lower_transport) {
1771 case GST_RTSP_LOWER_TRANS_UDP:
1772 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1778 dest = tr->destination;
1779 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1784 min = tr->client_port.min;
1785 max = tr->client_port.max;
1789 GST_INFO ("adding %s:%d-%d", dest, min, max);
1790 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1791 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1793 GST_INFO ("setting ttl-mc %d", ttl);
1794 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1795 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1797 priv->transports = g_list_prepend (priv->transports, trans);
1799 GST_INFO ("removing %s:%d-%d", dest, min, max);
1800 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1801 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1802 priv->transports = g_list_remove (priv->transports, trans);
1806 case GST_RTSP_LOWER_TRANS_TCP:
1808 GST_INFO ("adding TCP %s", tr->destination);
1809 priv->transports = g_list_prepend (priv->transports, trans);
1811 GST_INFO ("removing TCP %s", tr->destination);
1812 priv->transports = g_list_remove (priv->transports, trans);
1816 goto unknown_transport;
1823 GST_INFO ("Unknown transport %d", tr->lower_transport);
1830 * gst_rtsp_stream_add_transport:
1831 * @stream: a #GstRTSPStream
1832 * @trans: a #GstRTSPStreamTransport
1834 * Add the transport in @trans to @stream. The media of @stream will
1835 * then also be send to the values configured in @trans.
1837 * @stream must be joined to a bin.
1839 * @trans must contain a valid #GstRTSPTransport.
1841 * Returns: %TRUE if @trans was added
1844 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1845 GstRTSPStreamTransport * trans)
1847 GstRTSPStreamPrivate *priv;
1850 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1851 priv = stream->priv;
1852 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1853 g_return_val_if_fail (priv->is_joined, FALSE);
1855 g_mutex_lock (&priv->lock);
1856 res = update_transport (stream, trans, TRUE);
1857 g_mutex_unlock (&priv->lock);
1863 * gst_rtsp_stream_remove_transport:
1864 * @stream: a #GstRTSPStream
1865 * @trans: a #GstRTSPStreamTransport
1867 * Remove the transport in @trans from @stream. The media of @stream will
1868 * not be sent to the values configured in @trans.
1870 * @stream must be joined to a bin.
1872 * @trans must contain a valid #GstRTSPTransport.
1874 * Returns: %TRUE if @trans was removed
1877 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1878 GstRTSPStreamTransport * trans)
1880 GstRTSPStreamPrivate *priv;
1883 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1884 priv = stream->priv;
1885 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1886 g_return_val_if_fail (priv->is_joined, FALSE);
1888 g_mutex_lock (&priv->lock);
1889 res = update_transport (stream, trans, FALSE);
1890 g_mutex_unlock (&priv->lock);