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 * Last reviewed on 2013-07-11 (1.0.0)
33 #include <gst/app/gstappsrc.h>
34 #include <gst/app/gstappsink.h>
36 #include "rtsp-stream.h"
38 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
39 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
41 struct _GstRTSPStreamPrivate
46 GstElement *payloader;
51 /* pads on the rtpbin */
52 GstPad *send_rtp_sink;
56 /* the RTPSession object */
59 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
61 GstElement *udpsrc_v4[2];
63 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
65 GstElement *udpsrc_v6[2];
67 GstElement *udpsink[2];
69 /* for TCP transport */
70 GstElement *appsrc[2];
71 GstElement *appqueue[2];
72 GstElement *appsink[2];
75 GstElement *funnel[2];
77 /* server ports for sending/receiving over ipv4 */
78 GstRTSPRange server_port_v4;
79 GstRTSPAddress *server_addr_v4;
82 /* server ports for sending/receiving over ipv6 */
83 GstRTSPRange server_port_v6;
84 GstRTSPAddress *server_addr_v6;
87 /* multicast addresses */
88 GstRTSPAddressPool *pool;
89 GstRTSPAddress *addr_v4;
90 GstRTSPAddress *addr_v6;
92 /* the caps of the stream */
96 /* transports we stream to */
103 #define DEFAULT_CONTROL NULL
112 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
113 #define GST_CAT_DEFAULT rtsp_stream_debug
115 static GQuark ssrc_stream_map_key;
117 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
118 GValue * value, GParamSpec * pspec);
119 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
120 const GValue * value, GParamSpec * pspec);
122 static void gst_rtsp_stream_finalize (GObject * obj);
124 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
127 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
129 GObjectClass *gobject_class;
131 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
133 gobject_class = G_OBJECT_CLASS (klass);
135 gobject_class->get_property = gst_rtsp_stream_get_property;
136 gobject_class->set_property = gst_rtsp_stream_set_property;
137 gobject_class->finalize = gst_rtsp_stream_finalize;
139 g_object_class_install_property (gobject_class, PROP_CONTROL,
140 g_param_spec_string ("control", "Control",
141 "The control string for this stream", DEFAULT_CONTROL,
142 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
144 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
146 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
150 gst_rtsp_stream_init (GstRTSPStream * stream)
152 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
154 GST_DEBUG ("new stream %p", stream);
159 priv->control = g_strdup (DEFAULT_CONTROL);
161 g_mutex_init (&priv->lock);
165 gst_rtsp_stream_finalize (GObject * obj)
167 GstRTSPStream *stream;
168 GstRTSPStreamPrivate *priv;
170 stream = GST_RTSP_STREAM (obj);
173 GST_DEBUG ("finalize stream %p", stream);
175 /* we really need to be unjoined now */
176 g_return_if_fail (!priv->is_joined);
179 gst_rtsp_address_free (priv->addr_v4);
181 gst_rtsp_address_free (priv->addr_v6);
182 if (priv->server_addr_v4)
183 gst_rtsp_address_free (priv->server_addr_v4);
184 if (priv->server_addr_v6)
185 gst_rtsp_address_free (priv->server_addr_v6);
187 g_object_unref (priv->pool);
188 gst_object_unref (priv->payloader);
189 gst_object_unref (priv->srcpad);
190 g_free (priv->control);
191 g_mutex_clear (&priv->lock);
193 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
197 gst_rtsp_stream_get_property (GObject * object, guint propid,
198 GValue * value, GParamSpec * pspec)
200 GstRTSPStream *stream = GST_RTSP_STREAM (object);
204 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
207 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
212 gst_rtsp_stream_set_property (GObject * object, guint propid,
213 const GValue * value, GParamSpec * pspec)
215 GstRTSPStream *stream = GST_RTSP_STREAM (object);
219 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
222 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
227 * gst_rtsp_stream_new:
230 * @payloader: a #GstElement
232 * Create a new media stream with index @idx that handles RTP data on
233 * @srcpad and has a payloader element @payloader.
235 * Returns: a new #GstRTSPStream
238 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
240 GstRTSPStreamPrivate *priv;
241 GstRTSPStream *stream;
243 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
244 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
245 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
247 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
250 priv->payloader = gst_object_ref (payloader);
251 priv->srcpad = gst_object_ref (srcpad);
257 * gst_rtsp_stream_get_index:
258 * @stream: a #GstRTSPStream
260 * Get the stream index.
262 * Return: the stream index.
265 gst_rtsp_stream_get_index (GstRTSPStream * stream)
267 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
269 return stream->priv->idx;
273 * gst_rtsp_stream_get_srcpad:
274 * @stream: a #GstRTSPStream
276 * Get the srcpad associated with @stream.
278 * Return: the srcpad. Unref after usage.
281 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
283 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
285 return gst_object_ref (stream->priv->srcpad);
289 * gst_rtsp_stream_get_control:
290 * @stream: a #GstRTSPStream
292 * Get the control string to identify this stream.
294 * Return: the control string. free after usage.
297 gst_rtsp_stream_get_control (GstRTSPStream * stream)
299 GstRTSPStreamPrivate *priv;
302 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
306 g_mutex_lock (&priv->lock);
307 if ((result = g_strdup (priv->control)) == NULL)
308 result = g_strdup_printf ("stream=%u", priv->idx);
309 g_mutex_unlock (&priv->lock);
315 * gst_rtsp_stream_set_control:
316 * @stream: a #GstRTSPStream
317 * @control: a control string
319 * Set the control string in @stream.
322 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
324 GstRTSPStreamPrivate *priv;
326 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
330 g_mutex_lock (&priv->lock);
331 g_free (priv->control);
332 priv->control = g_strdup (control);
333 g_mutex_unlock (&priv->lock);
337 * gst_rtsp_stream_has_control:
338 * @stream: a #GstRTSPStream
339 * @control: a control string
341 * Check if @stream has the control string @control.
343 * Returns: %TRUE is @stream has @control as the control string
346 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
348 GstRTSPStreamPrivate *priv;
351 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
355 g_mutex_lock (&priv->lock);
357 res = g_strcmp0 (priv->control, control);
360 sscanf (control, "stream=%u", &streamid);
361 res = (streamid == priv->idx);
363 g_mutex_unlock (&priv->lock);
369 * gst_rtsp_stream_set_mtu:
370 * @stream: a #GstRTSPStream
373 * Configure the mtu in the payloader of @stream to @mtu.
376 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
378 GstRTSPStreamPrivate *priv;
380 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
384 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
386 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
390 * gst_rtsp_stream_get_mtu:
391 * @stream: a #GstRTSPStream
393 * Get the configured MTU in the payloader of @stream.
395 * Returns: the MTU of the payloader.
398 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
400 GstRTSPStreamPrivate *priv;
403 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
407 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
412 /* Update the dscp qos property on the udp sinks */
414 update_dscp_qos (GstRTSPStream * stream)
416 GstRTSPStreamPrivate *priv;
418 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
422 if (priv->udpsink[0]) {
423 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
427 if (priv->udpsink[1]) {
428 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
434 * gst_rtsp_stream_set_dscp_qos:
435 * @stream: a #GstRTSPStream
436 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
438 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
441 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
443 GstRTSPStreamPrivate *priv;
445 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
449 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
451 if (dscp_qos < -1 || dscp_qos > 63) {
452 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
456 priv->dscp_qos = dscp_qos;
458 update_dscp_qos (stream);
462 * gst_rtsp_stream_get_dscp_qos:
463 * @stream: a #GstRTSPStream
465 * Get the configured DSCP QoS in of the outgoing sockets.
467 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
470 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
472 GstRTSPStreamPrivate *priv;
474 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
478 return priv->dscp_qos;
483 * gst_rtsp_stream_set_address_pool:
484 * @stream: a #GstRTSPStream
485 * @pool: a #GstRTSPAddressPool
487 * configure @pool to be used as the address pool of @stream.
490 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
491 GstRTSPAddressPool * pool)
493 GstRTSPStreamPrivate *priv;
494 GstRTSPAddressPool *old;
496 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
500 GST_LOG_OBJECT (stream, "set address pool %p", pool);
502 g_mutex_lock (&priv->lock);
503 if ((old = priv->pool) != pool)
504 priv->pool = pool ? g_object_ref (pool) : NULL;
507 g_mutex_unlock (&priv->lock);
510 g_object_unref (old);
514 * gst_rtsp_stream_get_address_pool:
515 * @stream: a #GstRTSPStream
517 * Get the #GstRTSPAddressPool used as the address pool of @stream.
519 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
523 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
525 GstRTSPStreamPrivate *priv;
526 GstRTSPAddressPool *result;
528 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
532 g_mutex_lock (&priv->lock);
533 if ((result = priv->pool))
534 g_object_ref (result);
535 g_mutex_unlock (&priv->lock);
541 * gst_rtsp_stream_get_multicast_address:
542 * @stream: a #GstRTSPStream
543 * @family: the #GSocketFamily
545 * Get the multicast address of @stream for @family.
547 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
548 * allocated. gst_rtsp_address_free() after usage.
551 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
552 GSocketFamily family)
554 GstRTSPStreamPrivate *priv;
555 GstRTSPAddress *result;
556 GstRTSPAddress **addrp;
557 GstRTSPAddressFlags flags;
559 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
563 if (family == G_SOCKET_FAMILY_IPV6) {
564 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
565 addrp = &priv->addr_v4;
567 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
568 addrp = &priv->addr_v6;
571 g_mutex_lock (&priv->lock);
572 if (*addrp == NULL) {
573 if (priv->pool == NULL)
576 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
578 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
582 result = gst_rtsp_address_copy (*addrp);
583 g_mutex_unlock (&priv->lock);
590 GST_ERROR_OBJECT (stream, "no address pool specified");
591 g_mutex_unlock (&priv->lock);
596 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
597 g_mutex_unlock (&priv->lock);
603 * gst_rtsp_stream_reserve_address:
604 * @stream: a #GstRTSPStream
605 * @address: an address
610 * Reserve @address and @port as the address and port of @stream.
612 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
613 * reserved. gst_rtsp_address_free() after usage.
616 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
617 const gchar * address, guint port, guint n_ports, guint ttl)
619 GstRTSPStreamPrivate *priv;
620 GstRTSPAddress *result;
622 GSocketFamily family;
623 GstRTSPAddress **addrp;
625 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
626 g_return_val_if_fail (address != NULL, NULL);
627 g_return_val_if_fail (port > 0, NULL);
628 g_return_val_if_fail (n_ports > 0, NULL);
629 g_return_val_if_fail (ttl > 0, NULL);
633 addr = g_inet_address_new_from_string (address);
635 GST_ERROR ("failed to get inet addr from %s", address);
636 family = G_SOCKET_FAMILY_IPV4;
638 family = g_inet_address_get_family (addr);
639 g_object_unref (addr);
642 if (family == G_SOCKET_FAMILY_IPV6)
643 addrp = &priv->addr_v4;
645 addrp = &priv->addr_v6;
647 g_mutex_lock (&priv->lock);
648 if (*addrp == NULL) {
649 if (priv->pool == NULL)
652 *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
657 if (strcmp ((*addrp)->address, address) ||
658 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
659 (*addrp)->ttl != ttl)
660 goto different_address;
662 result = gst_rtsp_address_copy (*addrp);
663 g_mutex_unlock (&priv->lock);
670 GST_ERROR_OBJECT (stream, "no address pool specified");
671 g_mutex_unlock (&priv->lock);
676 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
678 g_mutex_unlock (&priv->lock);
683 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
684 " reserved", address);
685 g_mutex_unlock (&priv->lock);
691 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
692 GSocketFamily family, GstElement * udpsrc_out[2],
693 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
694 GstRTSPAddress ** server_addr_out)
696 GstStateChangeReturn ret;
697 GstElement *udpsrc0, *udpsrc1;
698 GstElement *udpsink0, *udpsink1;
699 GSocket *rtp_socket = NULL;
700 GSocket *rtcp_socket;
701 gint tmp_rtp, tmp_rtcp;
703 gint rtpport, rtcpport;
704 GList *rejected_addresses = NULL;
705 GstRTSPAddress *addr = NULL;
706 GInetAddress *inetaddr = NULL;
707 GSocketAddress *rtp_sockaddr = NULL;
708 GSocketAddress *rtcp_sockaddr = NULL;
709 const gchar *multisink_socket;
711 if (family == G_SOCKET_FAMILY_IPV6)
712 multisink_socket = "socket-v6";
714 multisink_socket = "socket";
722 /* Start with random port */
725 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
726 G_SOCKET_PROTOCOL_UDP, NULL);
728 goto no_udp_protocol;
730 if (*server_addr_out)
731 gst_rtsp_address_free (*server_addr_out);
733 /* try to allocate 2 UDP ports, the RTP port should be an even
734 * number and the RTCP port should be the next (uneven) port */
737 if (rtp_socket == NULL) {
738 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
739 G_SOCKET_PROTOCOL_UDP, NULL);
741 goto no_udp_protocol;
744 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
745 GstRTSPAddressFlags flags;
748 rejected_addresses = g_list_prepend (rejected_addresses, addr);
750 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
751 if (family == G_SOCKET_FAMILY_IPV6)
752 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
754 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
756 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
761 tmp_rtp = addr->port;
763 g_clear_object (&inetaddr);
764 inetaddr = g_inet_address_new_from_string (addr->address);
772 if (inetaddr == NULL)
773 inetaddr = g_inet_address_new_any (family);
776 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
777 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
778 g_object_unref (rtp_sockaddr);
781 g_object_unref (rtp_sockaddr);
783 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
784 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
785 g_clear_object (&rtp_sockaddr);
790 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
791 g_object_unref (rtp_sockaddr);
793 /* check if port is even */
794 if ((tmp_rtp & 1) != 0) {
795 /* port not even, close and allocate another */
797 g_clear_object (&rtp_socket);
802 tmp_rtcp = tmp_rtp + 1;
804 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
805 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
806 g_object_unref (rtcp_sockaddr);
807 g_clear_object (&rtp_socket);
810 g_object_unref (rtcp_sockaddr);
812 g_clear_object (&inetaddr);
814 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
815 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
817 if (udpsrc0 == NULL || udpsrc1 == NULL)
818 goto no_udp_protocol;
820 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
821 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
823 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
824 if (ret == GST_STATE_CHANGE_FAILURE)
826 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
827 if (ret == GST_STATE_CHANGE_FAILURE)
830 /* all fine, do port check */
831 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
832 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
834 /* this should not happen... */
835 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
839 udpsink0 = udpsink_out[0];
841 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
844 goto no_udp_protocol;
846 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
847 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
850 udpsink1 = udpsink_out[1];
852 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
855 goto no_udp_protocol;
857 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
858 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
859 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
861 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
862 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
863 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
864 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
865 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
866 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
867 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
868 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
870 /* we keep these elements, we will further configure them when the
871 * client told us to really use the UDP ports. */
872 udpsrc_out[0] = udpsrc0;
873 udpsrc_out[1] = udpsrc1;
874 udpsink_out[0] = udpsink0;
875 udpsink_out[1] = udpsink1;
876 server_port_out->min = rtpport;
877 server_port_out->max = rtcpport;
879 *server_addr_out = addr;
880 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
882 g_object_unref (rtp_socket);
883 g_object_unref (rtcp_socket);
911 gst_element_set_state (udpsrc0, GST_STATE_NULL);
912 gst_object_unref (udpsrc0);
915 gst_element_set_state (udpsrc1, GST_STATE_NULL);
916 gst_object_unref (udpsrc1);
919 gst_element_set_state (udpsink0, GST_STATE_NULL);
920 gst_object_unref (udpsink0);
923 gst_element_set_state (udpsink1, GST_STATE_NULL);
924 gst_object_unref (udpsink1);
927 g_object_unref (inetaddr);
928 g_list_free_full (rejected_addresses,
929 (GDestroyNotify) gst_rtsp_address_free);
931 gst_rtsp_address_free (addr);
933 g_object_unref (rtp_socket);
935 g_object_unref (rtcp_socket);
940 /* must be called with lock */
942 alloc_ports (GstRTSPStream * stream)
944 GstRTSPStreamPrivate *priv = stream->priv;
946 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
947 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
948 &priv->server_port_v4, &priv->server_addr_v4);
950 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
951 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
952 &priv->server_port_v6, &priv->server_addr_v6);
954 return priv->have_ipv4 || priv->have_ipv6;
958 * gst_rtsp_stream_get_server_port:
959 * @stream: a #GstRTSPStream
960 * @server_port: (out): result server port
961 * @family: the port family to get
963 * Fill @server_port with the port pair used by the server. This function can
964 * only be called when @stream has been joined.
967 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
968 GstRTSPRange * server_port, GSocketFamily family)
970 GstRTSPStreamPrivate *priv;
972 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
974 g_return_if_fail (priv->is_joined);
976 g_mutex_lock (&priv->lock);
977 if (family == G_SOCKET_FAMILY_IPV4) {
979 *server_port = priv->server_port_v4;
982 *server_port = priv->server_port_v6;
984 g_mutex_unlock (&priv->lock);
988 * gst_rtsp_stream_get_rtpsession:
989 * @stream: a #GstRTSPStream
991 * Get the RTP session of this stream.
993 * Returns: The RTP session of this stream. Unref after usage.
996 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
998 GstRTSPStreamPrivate *priv;
1001 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1003 priv = stream->priv;
1005 g_mutex_lock (&priv->lock);
1006 if ((session = priv->session))
1007 g_object_ref (session);
1008 g_mutex_unlock (&priv->lock);
1014 * gst_rtsp_stream_get_ssrc:
1015 * @stream: a #GstRTSPStream
1016 * @ssrc: (out): result ssrc
1018 * Get the SSRC used by the RTP session of this stream. This function can only
1019 * be called when @stream has been joined.
1022 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1024 GstRTSPStreamPrivate *priv;
1026 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1027 priv = stream->priv;
1028 g_return_if_fail (priv->is_joined);
1030 g_mutex_lock (&priv->lock);
1031 if (ssrc && priv->session)
1032 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1033 g_mutex_unlock (&priv->lock);
1036 /* executed from streaming thread */
1038 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1040 GstRTSPStreamPrivate *priv = stream->priv;
1041 GstCaps *newcaps, *oldcaps;
1043 newcaps = gst_pad_get_current_caps (pad);
1045 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1048 g_mutex_lock (&priv->lock);
1049 oldcaps = priv->caps;
1050 priv->caps = newcaps;
1051 g_mutex_unlock (&priv->lock);
1054 gst_caps_unref (oldcaps);
1058 dump_structure (const GstStructure * s)
1062 sstr = gst_structure_to_string (s);
1063 GST_INFO ("structure: %s", sstr);
1067 static GstRTSPStreamTransport *
1068 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1070 GstRTSPStreamPrivate *priv = stream->priv;
1072 GstRTSPStreamTransport *result = NULL;
1077 if (rtcp_from == NULL)
1080 tmp = g_strrstr (rtcp_from, ":");
1084 port = atoi (tmp + 1);
1085 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1087 g_mutex_lock (&priv->lock);
1088 GST_INFO ("finding %s:%d in %d transports", dest, port,
1089 g_list_length (priv->transports));
1091 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1092 GstRTSPStreamTransport *trans = walk->data;
1093 const GstRTSPTransport *tr;
1096 tr = gst_rtsp_stream_transport_get_transport (trans);
1098 min = tr->client_port.min;
1099 max = tr->client_port.max;
1101 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1107 g_object_ref (result);
1108 g_mutex_unlock (&priv->lock);
1115 static GstRTSPStreamTransport *
1116 check_transport (GObject * source, GstRTSPStream * stream)
1118 GstStructure *stats;
1119 GstRTSPStreamTransport *trans;
1121 /* see if we have a stream to match with the origin of the RTCP packet */
1122 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1123 if (trans == NULL) {
1124 g_object_get (source, "stats", &stats, NULL);
1126 const gchar *rtcp_from;
1128 dump_structure (stats);
1130 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1131 if ((trans = find_transport (stream, rtcp_from))) {
1132 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1134 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1137 gst_structure_free (stats);
1145 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1147 GstRTSPStreamTransport *trans;
1149 GST_INFO ("%p: new source %p", stream, source);
1151 trans = check_transport (source, stream);
1154 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1158 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1160 GST_INFO ("%p: new SDES %p", stream, source);
1164 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1166 GstRTSPStreamTransport *trans;
1168 trans = check_transport (source, stream);
1171 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1172 gst_rtsp_stream_transport_keep_alive (trans);
1176 GstStructure *stats;
1177 g_object_get (source, "stats", &stats, NULL);
1179 dump_structure (stats);
1180 gst_structure_free (stats);
1187 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1189 GST_INFO ("%p: source %p bye", stream, source);
1193 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1195 GstRTSPStreamTransport *trans;
1197 GST_INFO ("%p: source %p bye timeout", stream, source);
1199 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1200 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1201 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1206 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1208 GstRTSPStreamTransport *trans;
1210 GST_INFO ("%p: source %p timeout", stream, source);
1212 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1213 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1214 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1218 static GstFlowReturn
1219 handle_new_sample (GstAppSink * sink, gpointer user_data)
1221 GstRTSPStreamPrivate *priv;
1225 GstRTSPStream *stream;
1227 sample = gst_app_sink_pull_sample (sink);
1231 stream = (GstRTSPStream *) user_data;
1232 priv = stream->priv;
1233 buffer = gst_sample_get_buffer (sample);
1235 g_mutex_lock (&priv->lock);
1236 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1237 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1239 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1240 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1242 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1245 g_mutex_unlock (&priv->lock);
1247 gst_sample_unref (sample);
1252 static GstAppSinkCallbacks sink_cb = {
1253 NULL, /* not interested in EOS */
1254 NULL, /* not interested in preroll samples */
1259 * gst_rtsp_stream_join_bin:
1260 * @stream: a #GstRTSPStream
1261 * @bin: a #GstBin to join
1262 * @rtpbin: a rtpbin element in @bin
1263 * @state: the target state of the new elements
1265 * Join the #Gstbin @bin that contains the element @rtpbin.
1267 * @stream will link to @rtpbin, which must be inside @bin. The elements
1268 * added to @bin will be set to the state given in @state.
1270 * Returns: %TRUE on success.
1273 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1274 GstElement * rtpbin, GstState state)
1276 GstRTSPStreamPrivate *priv;
1280 GstPad *pad, *teepad, *queuepad, *selpad;
1281 GstPadLinkReturn ret;
1283 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1284 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1285 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1287 priv = stream->priv;
1289 g_mutex_lock (&priv->lock);
1290 if (priv->is_joined)
1293 /* create a session with the same index as the stream */
1296 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1298 if (!alloc_ports (stream))
1301 /* update the dscp qos field in the sinks */
1302 update_dscp_qos (stream);
1304 /* get a pad for sending RTP */
1305 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1306 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1308 /* link the RTP pad to the session manager, it should not really fail unless
1309 * this is not really an RTP pad */
1310 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1311 if (ret != GST_PAD_LINK_OK)
1314 /* get pads from the RTP session element for sending and receiving
1316 name = g_strdup_printf ("send_rtp_src_%u", idx);
1317 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1319 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1320 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1322 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1323 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1325 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1326 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1329 /* get the session */
1330 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1332 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1334 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1336 g_signal_connect (priv->session, "on-ssrc-active",
1337 (GCallback) on_ssrc_active, stream);
1338 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1340 g_signal_connect (priv->session, "on-bye-timeout",
1341 (GCallback) on_bye_timeout, stream);
1342 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1345 for (i = 0; i < 2; i++) {
1346 /* For the sender we create this bit of pipeline for both
1347 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1348 * we need to add a queue before appsink to make the pipeline
1349 * not block. For the TCP case, we want to pump data to the
1350 * client as fast as possible anyway.
1352 * .--------. .-----. .---------.
1353 * | rtpbin | | tee | | udpsink |
1354 * | send->sink src->sink |
1355 * '--------' | | '---------'
1356 * | | .---------. .---------.
1357 * | | | queue | | appsink |
1358 * | src->sink src->sink |
1359 * '-----' '---------' '---------'
1361 /* make tee for RTP/RTCP */
1362 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1363 gst_bin_add (bin, priv->tee[i]);
1365 /* and link to rtpbin send pad */
1366 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1367 gst_pad_link (priv->send_src[i], pad);
1368 gst_object_unref (pad);
1371 gst_bin_add (bin, priv->udpsink[i]);
1373 /* link tee to udpsink */
1374 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1375 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1376 gst_pad_link (teepad, pad);
1377 gst_object_unref (pad);
1378 gst_object_unref (teepad);
1381 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1382 gst_bin_add (bin, priv->appqueue[i]);
1383 /* and link to tee */
1384 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1385 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1386 gst_pad_link (teepad, pad);
1387 gst_object_unref (pad);
1388 gst_object_unref (teepad);
1391 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1392 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1393 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1394 gst_bin_add (bin, priv->appsink[i]);
1395 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1396 &sink_cb, stream, NULL);
1397 /* and link to queue */
1398 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1399 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1400 gst_pad_link (queuepad, pad);
1401 gst_object_unref (pad);
1402 gst_object_unref (queuepad);
1404 /* For the receiver we create this bit of pipeline for both
1405 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1406 * and it is all funneled into the rtpbin receive pad.
1408 * .--------. .--------. .--------.
1409 * | udpsrc | | funnel | | rtpbin |
1410 * | src->sink src->sink |
1411 * '--------' | | '--------'
1415 * '--------' '--------'
1417 /* make funnel for the RTP/RTCP receivers */
1418 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1419 gst_bin_add (bin, priv->funnel[i]);
1421 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1422 gst_pad_link (pad, priv->recv_sink[i]);
1423 gst_object_unref (pad);
1425 if (priv->udpsrc_v4[i]) {
1426 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1428 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1429 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1431 gst_bin_add (bin, priv->udpsrc_v4[i]);
1433 /* and link to the funnel v4 */
1434 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1435 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1436 gst_pad_link (pad, selpad);
1437 gst_object_unref (pad);
1438 gst_object_unref (selpad);
1441 if (priv->udpsrc_v6[i]) {
1442 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1443 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1444 gst_bin_add (bin, priv->udpsrc_v6[i]);
1446 /* and link to the funnel v6 */
1447 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1448 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1449 gst_pad_link (pad, selpad);
1450 gst_object_unref (pad);
1451 gst_object_unref (selpad);
1454 /* make and add appsrc */
1455 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1456 gst_bin_add (bin, priv->appsrc[i]);
1457 /* and link to the funnel */
1458 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1459 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1460 gst_pad_link (pad, selpad);
1461 gst_object_unref (pad);
1462 gst_object_unref (selpad);
1464 /* check if we need to set to a special state */
1465 if (state != GST_STATE_NULL) {
1466 gst_element_set_state (priv->udpsink[i], state);
1467 gst_element_set_state (priv->appsink[i], state);
1468 gst_element_set_state (priv->appqueue[i], state);
1469 gst_element_set_state (priv->tee[i], state);
1470 gst_element_set_state (priv->funnel[i], state);
1471 gst_element_set_state (priv->appsrc[i], state);
1475 /* be notified of caps changes */
1476 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1477 (GCallback) caps_notify, stream);
1479 priv->is_joined = TRUE;
1480 g_mutex_unlock (&priv->lock);
1487 g_mutex_unlock (&priv->lock);
1492 g_mutex_unlock (&priv->lock);
1493 GST_WARNING ("failed to allocate ports %u", idx);
1498 GST_WARNING ("failed to link stream %u", idx);
1499 gst_object_unref (priv->send_rtp_sink);
1500 priv->send_rtp_sink = NULL;
1501 g_mutex_unlock (&priv->lock);
1507 * gst_rtsp_stream_leave_bin:
1508 * @stream: a #GstRTSPStream
1510 * @rtpbin: a rtpbin #GstElement
1512 * Remove the elements of @stream from @bin.
1514 * Return: %TRUE on success.
1517 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1518 GstElement * rtpbin)
1520 GstRTSPStreamPrivate *priv;
1523 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1524 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1525 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1527 priv = stream->priv;
1529 g_mutex_lock (&priv->lock);
1530 if (!priv->is_joined)
1531 goto was_not_joined;
1533 /* all transports must be removed by now */
1534 g_return_val_if_fail (priv->transports == NULL, FALSE);
1536 GST_INFO ("stream %p leaving bin", stream);
1538 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1539 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1540 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1541 gst_object_unref (priv->send_rtp_sink);
1542 priv->send_rtp_sink = NULL;
1544 for (i = 0; i < 2; i++) {
1545 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1546 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1547 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1548 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1549 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1550 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1551 if (priv->udpsrc_v4[i]) {
1552 /* and set udpsrc to NULL now before removing */
1553 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1554 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1555 /* removing them should also nicely release the request
1556 * pads when they finalize */
1557 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1559 if (priv->udpsrc_v6[i]) {
1560 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1561 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1562 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1564 gst_bin_remove (bin, priv->udpsink[i]);
1565 gst_bin_remove (bin, priv->appsrc[i]);
1566 gst_bin_remove (bin, priv->appsink[i]);
1567 gst_bin_remove (bin, priv->appqueue[i]);
1568 gst_bin_remove (bin, priv->tee[i]);
1569 gst_bin_remove (bin, priv->funnel[i]);
1571 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1572 gst_object_unref (priv->recv_sink[i]);
1573 priv->recv_sink[i] = NULL;
1575 priv->udpsrc_v4[i] = NULL;
1576 priv->udpsrc_v6[i] = NULL;
1577 priv->udpsink[i] = NULL;
1578 priv->appsrc[i] = NULL;
1579 priv->appsink[i] = NULL;
1580 priv->appqueue[i] = NULL;
1581 priv->tee[i] = NULL;
1582 priv->funnel[i] = NULL;
1584 gst_object_unref (priv->send_src[0]);
1585 priv->send_src[0] = NULL;
1587 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1588 gst_object_unref (priv->send_src[1]);
1589 priv->send_src[1] = NULL;
1591 g_object_unref (priv->session);
1592 priv->session = NULL;
1594 gst_caps_unref (priv->caps);
1597 priv->is_joined = FALSE;
1598 g_mutex_unlock (&priv->lock);
1609 * gst_rtsp_stream_get_rtpinfo:
1610 * @stream: a #GstRTSPStream
1611 * @rtptime: result RTP timestamp
1612 * @seq: result RTP seqnum
1614 * Retrieve the current rtptime and seq. This is used to
1615 * construct a RTPInfo reply header.
1617 * Returns: %TRUE when rtptime and seq could be determined.
1620 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1621 guint * rtptime, guint * seq)
1623 GstRTSPStreamPrivate *priv;
1624 GObjectClass *payobjclass;
1626 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1627 g_return_val_if_fail (rtptime != NULL, FALSE);
1628 g_return_val_if_fail (seq != NULL, FALSE);
1630 priv = stream->priv;
1632 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1634 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1635 !g_object_class_find_property (payobjclass, "timestamp"))
1638 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1644 * gst_rtsp_stream_get_caps:
1645 * @stream: a #GstRTSPStream
1647 * Retrieve the current caps of @stream.
1649 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1653 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1655 GstRTSPStreamPrivate *priv;
1658 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1660 priv = stream->priv;
1662 g_mutex_lock (&priv->lock);
1663 if ((result = priv->caps))
1664 gst_caps_ref (result);
1665 g_mutex_unlock (&priv->lock);
1671 * gst_rtsp_stream_recv_rtp:
1672 * @stream: a #GstRTSPStream
1673 * @buffer: (transfer full): a #GstBuffer
1675 * Handle an RTP buffer for the stream. This method is usually called when a
1676 * message has been received from a client using the TCP transport.
1678 * This function takes ownership of @buffer.
1680 * Returns: a GstFlowReturn.
1683 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1685 GstRTSPStreamPrivate *priv;
1687 GstElement *element;
1689 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1690 priv = stream->priv;
1691 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1692 g_return_val_if_fail (priv->is_joined, FALSE);
1694 g_mutex_lock (&priv->lock);
1695 element = gst_object_ref (priv->appsrc[0]);
1696 g_mutex_unlock (&priv->lock);
1698 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1700 gst_object_unref (element);
1706 * gst_rtsp_stream_recv_rtcp:
1707 * @stream: a #GstRTSPStream
1708 * @buffer: (transfer full): a #GstBuffer
1710 * Handle an RTCP buffer for the stream. This method is usually called when a
1711 * message has been received from a client using the TCP transport.
1713 * This function takes ownership of @buffer.
1715 * Returns: a GstFlowReturn.
1718 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1720 GstRTSPStreamPrivate *priv;
1722 GstElement *element;
1724 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1725 priv = stream->priv;
1726 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1727 g_return_val_if_fail (priv->is_joined, FALSE);
1729 g_mutex_lock (&priv->lock);
1730 element = gst_object_ref (priv->appsrc[1]);
1731 g_mutex_unlock (&priv->lock);
1733 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1735 gst_object_unref (element);
1740 /* must be called with lock */
1742 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1745 GstRTSPStreamPrivate *priv = stream->priv;
1746 const GstRTSPTransport *tr;
1748 tr = gst_rtsp_stream_transport_get_transport (trans);
1750 switch (tr->lower_transport) {
1751 case GST_RTSP_LOWER_TRANS_UDP:
1752 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1758 dest = tr->destination;
1759 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1764 min = tr->client_port.min;
1765 max = tr->client_port.max;
1769 GST_INFO ("adding %s:%d-%d", dest, min, max);
1770 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1771 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1773 GST_INFO ("setting ttl-mc %d", ttl);
1774 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1775 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1777 priv->transports = g_list_prepend (priv->transports, trans);
1779 GST_INFO ("removing %s:%d-%d", dest, min, max);
1780 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1781 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1782 priv->transports = g_list_remove (priv->transports, trans);
1786 case GST_RTSP_LOWER_TRANS_TCP:
1788 GST_INFO ("adding TCP %s", tr->destination);
1789 priv->transports = g_list_prepend (priv->transports, trans);
1791 GST_INFO ("removing TCP %s", tr->destination);
1792 priv->transports = g_list_remove (priv->transports, trans);
1796 goto unknown_transport;
1803 GST_INFO ("Unknown transport %d", tr->lower_transport);
1810 * gst_rtsp_stream_add_transport:
1811 * @stream: a #GstRTSPStream
1812 * @trans: a #GstRTSPStreamTransport
1814 * Add the transport in @trans to @stream. The media of @stream will
1815 * then also be send to the values configured in @trans.
1817 * @stream must be joined to a bin.
1819 * @trans must contain a valid #GstRTSPTransport.
1821 * Returns: %TRUE if @trans was added
1824 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1825 GstRTSPStreamTransport * trans)
1827 GstRTSPStreamPrivate *priv;
1830 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1831 priv = stream->priv;
1832 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1833 g_return_val_if_fail (priv->is_joined, FALSE);
1835 g_mutex_lock (&priv->lock);
1836 res = update_transport (stream, trans, TRUE);
1837 g_mutex_unlock (&priv->lock);
1843 * gst_rtsp_stream_remove_transport:
1844 * @stream: a #GstRTSPStream
1845 * @trans: a #GstRTSPStreamTransport
1847 * Remove the transport in @trans from @stream. The media of @stream will
1848 * not be sent to the values configured in @trans.
1850 * @stream must be joined to a bin.
1852 * @trans must contain a valid #GstRTSPTransport.
1854 * Returns: %TRUE if @trans was removed
1857 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1858 GstRTSPStreamTransport * trans)
1860 GstRTSPStreamPrivate *priv;
1863 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1864 priv = stream->priv;
1865 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1866 g_return_val_if_fail (priv->is_joined, FALSE);
1868 g_mutex_lock (&priv->lock);
1869 res = update_transport (stream, trans, FALSE);
1870 g_mutex_unlock (&priv->lock);