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.
26 #include <gst/app/gstappsrc.h>
27 #include <gst/app/gstappsink.h>
29 #include "rtsp-stream.h"
31 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
32 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
34 struct _GstRTSPStreamPrivate
39 GstElement *payloader;
44 /* pads on the rtpbin */
45 GstPad *send_rtp_sink;
49 /* the RTPSession object */
52 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
54 GstElement *udpsrc_v4[2];
56 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
58 GstElement *udpsrc_v6[2];
60 GstElement *udpsink[2];
62 /* for TCP transport */
63 GstElement *appsrc[2];
64 GstElement *appqueue[2];
65 GstElement *appsink[2];
68 GstElement *funnel[2];
70 /* server ports for sending/receiving over ipv4 */
71 GstRTSPRange server_port_v4;
72 GstRTSPAddress *server_addr_v4;
75 /* server ports for sending/receiving over ipv6 */
76 GstRTSPRange server_port_v6;
77 GstRTSPAddress *server_addr_v6;
80 /* multicast addresses */
81 GstRTSPAddressPool *pool;
82 GstRTSPAddress *addr_v4;
83 GstRTSPAddress *addr_v6;
85 /* the caps of the stream */
89 /* transports we stream to */
96 #define DEFAULT_CONTROL NULL
105 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
106 #define GST_CAT_DEFAULT rtsp_stream_debug
108 static GQuark ssrc_stream_map_key;
110 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
111 GValue * value, GParamSpec * pspec);
112 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
113 const GValue * value, GParamSpec * pspec);
115 static void gst_rtsp_stream_finalize (GObject * obj);
117 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
120 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
122 GObjectClass *gobject_class;
124 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
126 gobject_class = G_OBJECT_CLASS (klass);
128 gobject_class->get_property = gst_rtsp_stream_get_property;
129 gobject_class->set_property = gst_rtsp_stream_set_property;
130 gobject_class->finalize = gst_rtsp_stream_finalize;
132 g_object_class_install_property (gobject_class, PROP_CONTROL,
133 g_param_spec_string ("control", "Control",
134 "The control string for this stream", DEFAULT_CONTROL,
135 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
137 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
139 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
143 gst_rtsp_stream_init (GstRTSPStream * stream)
145 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
147 GST_DEBUG ("new stream %p", stream);
152 priv->control = g_strdup (DEFAULT_CONTROL);
154 g_mutex_init (&priv->lock);
158 gst_rtsp_stream_finalize (GObject * obj)
160 GstRTSPStream *stream;
161 GstRTSPStreamPrivate *priv;
163 stream = GST_RTSP_STREAM (obj);
166 GST_DEBUG ("finalize stream %p", stream);
168 /* we really need to be unjoined now */
169 g_return_if_fail (!priv->is_joined);
172 gst_rtsp_address_free (priv->addr_v4);
174 gst_rtsp_address_free (priv->addr_v6);
175 if (priv->server_addr_v4)
176 gst_rtsp_address_free (priv->server_addr_v4);
177 if (priv->server_addr_v6)
178 gst_rtsp_address_free (priv->server_addr_v6);
180 g_object_unref (priv->pool);
181 gst_object_unref (priv->payloader);
182 gst_object_unref (priv->srcpad);
183 g_free (priv->control);
184 g_mutex_clear (&priv->lock);
186 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
190 gst_rtsp_stream_get_property (GObject * object, guint propid,
191 GValue * value, GParamSpec * pspec)
193 GstRTSPStream *stream = GST_RTSP_STREAM (object);
197 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
200 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
205 gst_rtsp_stream_set_property (GObject * object, guint propid,
206 const GValue * value, GParamSpec * pspec)
208 GstRTSPStream *stream = GST_RTSP_STREAM (object);
212 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
215 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
220 * gst_rtsp_stream_new:
223 * @payloader: a #GstElement
225 * Create a new media stream with index @idx that handles RTP data on
226 * @srcpad and has a payloader element @payloader.
228 * Returns: a new #GstRTSPStream
231 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
233 GstRTSPStreamPrivate *priv;
234 GstRTSPStream *stream;
236 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
237 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
238 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
240 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
243 priv->payloader = gst_object_ref (payloader);
244 priv->srcpad = gst_object_ref (srcpad);
250 * gst_rtsp_stream_get_index:
251 * @stream: a #GstRTSPStream
253 * Get the stream index.
255 * Return: the stream index.
258 gst_rtsp_stream_get_index (GstRTSPStream * stream)
260 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
262 return stream->priv->idx;
266 * gst_rtsp_stream_get_srcpad:
267 * @stream: a #GstRTSPStream
269 * Get the srcpad associated with @stream.
271 * Return: the srcpad. Unref after usage.
274 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
276 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
278 return gst_object_ref (stream->priv->srcpad);
282 * gst_rtsp_stream_get_control:
283 * @stream: a #GstRTSPStream
285 * Get the control string to identify this stream.
287 * Return: the control string. free after usage.
290 gst_rtsp_stream_get_control (GstRTSPStream * stream)
292 GstRTSPStreamPrivate *priv;
295 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
299 g_mutex_lock (&priv->lock);
300 if ((result = g_strdup (priv->control)) == NULL)
301 result = g_strdup_printf ("stream=%u", priv->idx);
302 g_mutex_unlock (&priv->lock);
308 * gst_rtsp_stream_set_control:
309 * @stream: a #GstRTSPStream
310 * @control: a control string
312 * Set the control string in @stream.
315 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
317 GstRTSPStreamPrivate *priv;
319 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
323 g_mutex_lock (&priv->lock);
324 g_free (priv->control);
325 priv->control = g_strdup (control);
326 g_mutex_unlock (&priv->lock);
330 * gst_rtsp_stream_has_control:
331 * @stream: a #GstRTSPStream
332 * @control: a control string
334 * Check if @stream has the control string @control.
336 * Returns: %TRUE is @stream has @control as the control string
339 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
341 GstRTSPStreamPrivate *priv;
344 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
348 g_mutex_lock (&priv->lock);
350 res = g_strcmp0 (priv->control, control);
353 sscanf (control, "stream=%u", &streamid);
354 res = (streamid == priv->idx);
356 g_mutex_unlock (&priv->lock);
362 * gst_rtsp_stream_set_mtu:
363 * @stream: a #GstRTSPStream
366 * Configure the mtu in the payloader of @stream to @mtu.
369 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
371 GstRTSPStreamPrivate *priv;
373 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
377 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
379 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
383 * gst_rtsp_stream_get_mtu:
384 * @stream: a #GstRTSPStream
386 * Get the configured MTU in the payloader of @stream.
388 * Returns: the MTU of the payloader.
391 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
393 GstRTSPStreamPrivate *priv;
396 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
400 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
405 /* Update the dscp qos property on the udp sinks */
407 update_dscp_qos (GstRTSPStream * stream)
409 GstRTSPStreamPrivate *priv;
411 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
415 if (priv->udpsink[0]) {
416 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
420 if (priv->udpsink[1]) {
421 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
427 * gst_rtsp_stream_set_dscp_qos:
428 * @stream: a #GstRTSPStream
429 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
431 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
434 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
436 GstRTSPStreamPrivate *priv;
438 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
442 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
444 if (dscp_qos < -1 || dscp_qos > 63) {
445 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
449 priv->dscp_qos = dscp_qos;
451 update_dscp_qos (stream);
455 * gst_rtsp_stream_get_dscp_qos:
456 * @stream: a #GstRTSPStream
458 * Get the configured DSCP QoS in of the outgoing sockets.
460 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
463 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
465 GstRTSPStreamPrivate *priv;
467 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
471 return priv->dscp_qos;
476 * gst_rtsp_stream_set_address_pool:
477 * @stream: a #GstRTSPStream
478 * @pool: a #GstRTSPAddressPool
480 * configure @pool to be used as the address pool of @stream.
483 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
484 GstRTSPAddressPool * pool)
486 GstRTSPStreamPrivate *priv;
487 GstRTSPAddressPool *old;
489 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
493 GST_LOG_OBJECT (stream, "set address pool %p", pool);
495 g_mutex_lock (&priv->lock);
496 if ((old = priv->pool) != pool)
497 priv->pool = pool ? g_object_ref (pool) : NULL;
500 g_mutex_unlock (&priv->lock);
503 g_object_unref (old);
507 * gst_rtsp_stream_get_address_pool:
508 * @stream: a #GstRTSPStream
510 * Get the #GstRTSPAddressPool used as the address pool of @stream.
512 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
516 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
518 GstRTSPStreamPrivate *priv;
519 GstRTSPAddressPool *result;
521 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
525 g_mutex_lock (&priv->lock);
526 if ((result = priv->pool))
527 g_object_ref (result);
528 g_mutex_unlock (&priv->lock);
534 * gst_rtsp_stream_get_multicast_address:
535 * @stream: a #GstRTSPStream
536 * @family: the #GSocketFamily
538 * Get the multicast address of @stream for @family.
540 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
541 * allocated. gst_rtsp_address_free() after usage.
544 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
545 GSocketFamily family)
547 GstRTSPStreamPrivate *priv;
548 GstRTSPAddress *result;
549 GstRTSPAddress **addrp;
550 GstRTSPAddressFlags flags;
552 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
556 if (family == G_SOCKET_FAMILY_IPV6) {
557 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
558 addrp = &priv->addr_v4;
560 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
561 addrp = &priv->addr_v6;
564 g_mutex_lock (&priv->lock);
565 if (*addrp == NULL) {
566 if (priv->pool == NULL)
569 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
571 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
575 result = gst_rtsp_address_copy (*addrp);
576 g_mutex_unlock (&priv->lock);
583 GST_ERROR_OBJECT (stream, "no address pool specified");
584 g_mutex_unlock (&priv->lock);
589 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
590 g_mutex_unlock (&priv->lock);
596 * gst_rtsp_stream_reserve_address:
597 * @stream: a #GstRTSPStream
598 * @address: an address
603 * Reserve @address and @port as the address and port of @stream.
605 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
606 * reserved. gst_rtsp_address_free() after usage.
609 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
610 const gchar * address, guint port, guint n_ports, guint ttl)
612 GstRTSPStreamPrivate *priv;
613 GstRTSPAddress *result;
615 GSocketFamily family;
616 GstRTSPAddress **addrp;
618 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
619 g_return_val_if_fail (address != NULL, NULL);
620 g_return_val_if_fail (port > 0, NULL);
621 g_return_val_if_fail (n_ports > 0, NULL);
622 g_return_val_if_fail (ttl > 0, NULL);
626 addr = g_inet_address_new_from_string (address);
628 GST_ERROR ("failed to get inet addr from %s", address);
629 family = G_SOCKET_FAMILY_IPV4;
631 family = g_inet_address_get_family (addr);
632 g_object_unref (addr);
635 if (family == G_SOCKET_FAMILY_IPV6)
636 addrp = &priv->addr_v4;
638 addrp = &priv->addr_v6;
640 g_mutex_lock (&priv->lock);
641 if (*addrp == NULL) {
642 if (priv->pool == NULL)
645 *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
650 if (strcmp ((*addrp)->address, address) ||
651 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
652 (*addrp)->ttl != ttl)
653 goto different_address;
655 result = gst_rtsp_address_copy (*addrp);
656 g_mutex_unlock (&priv->lock);
663 GST_ERROR_OBJECT (stream, "no address pool specified");
664 g_mutex_unlock (&priv->lock);
669 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
671 g_mutex_unlock (&priv->lock);
676 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
677 " reserved", address);
678 g_mutex_unlock (&priv->lock);
684 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
685 GSocketFamily family, GstElement * udpsrc_out[2],
686 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
687 GstRTSPAddress ** server_addr_out)
689 GstStateChangeReturn ret;
690 GstElement *udpsrc0, *udpsrc1;
691 GstElement *udpsink0, *udpsink1;
692 GSocket *rtp_socket = NULL;
693 GSocket *rtcp_socket;
694 gint tmp_rtp, tmp_rtcp;
696 gint rtpport, rtcpport;
697 GList *rejected_addresses = NULL;
698 GstRTSPAddress *addr = NULL;
699 GInetAddress *inetaddr = NULL;
700 GSocketAddress *rtp_sockaddr = NULL;
701 GSocketAddress *rtcp_sockaddr = NULL;
702 const gchar *multisink_socket;
704 if (family == G_SOCKET_FAMILY_IPV6)
705 multisink_socket = "socket-v6";
707 multisink_socket = "socket";
715 /* Start with random port */
718 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
719 G_SOCKET_PROTOCOL_UDP, NULL);
721 goto no_udp_protocol;
723 if (*server_addr_out)
724 gst_rtsp_address_free (*server_addr_out);
726 /* try to allocate 2 UDP ports, the RTP port should be an even
727 * number and the RTCP port should be the next (uneven) port */
730 if (rtp_socket == NULL) {
731 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
732 G_SOCKET_PROTOCOL_UDP, NULL);
734 goto no_udp_protocol;
737 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
738 GstRTSPAddressFlags flags;
741 rejected_addresses = g_list_prepend (rejected_addresses, addr);
743 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
744 if (family == G_SOCKET_FAMILY_IPV6)
745 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
747 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
749 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
754 tmp_rtp = addr->port;
756 g_clear_object (&inetaddr);
757 inetaddr = g_inet_address_new_from_string (addr->address);
765 if (inetaddr == NULL)
766 inetaddr = g_inet_address_new_any (family);
769 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
770 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
771 g_object_unref (rtp_sockaddr);
774 g_object_unref (rtp_sockaddr);
776 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
777 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
778 g_clear_object (&rtp_sockaddr);
783 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
784 g_object_unref (rtp_sockaddr);
786 /* check if port is even */
787 if ((tmp_rtp & 1) != 0) {
788 /* port not even, close and allocate another */
790 g_clear_object (&rtp_socket);
795 tmp_rtcp = tmp_rtp + 1;
797 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
798 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
799 g_object_unref (rtcp_sockaddr);
800 g_clear_object (&rtp_socket);
803 g_object_unref (rtcp_sockaddr);
805 g_clear_object (&inetaddr);
807 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
808 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
810 if (udpsrc0 == NULL || udpsrc1 == NULL)
811 goto no_udp_protocol;
813 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
814 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
816 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
817 if (ret == GST_STATE_CHANGE_FAILURE)
819 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
820 if (ret == GST_STATE_CHANGE_FAILURE)
823 /* all fine, do port check */
824 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
825 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
827 /* this should not happen... */
828 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
832 udpsink0 = udpsink_out[0];
834 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
837 goto no_udp_protocol;
839 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
840 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
843 udpsink1 = udpsink_out[1];
845 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
848 goto no_udp_protocol;
850 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
851 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
852 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
854 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
855 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
856 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
857 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
858 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
859 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
860 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
861 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
863 /* we keep these elements, we will further configure them when the
864 * client told us to really use the UDP ports. */
865 udpsrc_out[0] = udpsrc0;
866 udpsrc_out[1] = udpsrc1;
867 udpsink_out[0] = udpsink0;
868 udpsink_out[1] = udpsink1;
869 server_port_out->min = rtpport;
870 server_port_out->max = rtcpport;
872 *server_addr_out = addr;
873 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
875 g_object_unref (rtp_socket);
876 g_object_unref (rtcp_socket);
904 gst_element_set_state (udpsrc0, GST_STATE_NULL);
905 gst_object_unref (udpsrc0);
908 gst_element_set_state (udpsrc1, GST_STATE_NULL);
909 gst_object_unref (udpsrc1);
912 gst_element_set_state (udpsink0, GST_STATE_NULL);
913 gst_object_unref (udpsink0);
916 gst_element_set_state (udpsink1, GST_STATE_NULL);
917 gst_object_unref (udpsink1);
920 g_object_unref (inetaddr);
921 g_list_free_full (rejected_addresses,
922 (GDestroyNotify) gst_rtsp_address_free);
924 gst_rtsp_address_free (addr);
926 g_object_unref (rtp_socket);
928 g_object_unref (rtcp_socket);
933 /* must be called with lock */
935 alloc_ports (GstRTSPStream * stream)
937 GstRTSPStreamPrivate *priv = stream->priv;
939 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
940 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
941 &priv->server_port_v4, &priv->server_addr_v4);
943 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
944 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
945 &priv->server_port_v6, &priv->server_addr_v6);
947 return priv->have_ipv4 || priv->have_ipv6;
951 * gst_rtsp_stream_get_server_port:
952 * @stream: a #GstRTSPStream
953 * @server_port: (out): result server port
955 * Fill @server_port with the port pair used by the server. This function can
956 * only be called when @stream has been joined.
959 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
960 GstRTSPRange * server_port, GSocketFamily family)
962 GstRTSPStreamPrivate *priv;
964 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
966 g_return_if_fail (priv->is_joined);
968 g_mutex_lock (&priv->lock);
969 if (family == G_SOCKET_FAMILY_IPV4) {
971 *server_port = priv->server_port_v4;
974 *server_port = priv->server_port_v6;
976 g_mutex_unlock (&priv->lock);
980 * gst_rtsp_stream_get_rtpsession:
981 * @stream: a #GstRTSPStream
983 * Get the RTP session of this stream.
985 * Returns: The RTP session of this stream. Unref after usage.
988 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
990 GstRTSPStreamPrivate *priv;
993 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
997 g_mutex_lock (&priv->lock);
998 if ((session = priv->session))
999 g_object_ref (session);
1000 g_mutex_unlock (&priv->lock);
1006 * gst_rtsp_stream_get_ssrc:
1007 * @stream: a #GstRTSPStream
1008 * @ssrc: (out): result ssrc
1010 * Get the SSRC used by the RTP session of this stream. This function can only
1011 * be called when @stream has been joined.
1014 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1016 GstRTSPStreamPrivate *priv;
1018 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1019 priv = stream->priv;
1020 g_return_if_fail (priv->is_joined);
1022 g_mutex_lock (&priv->lock);
1023 if (ssrc && priv->session)
1024 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1025 g_mutex_unlock (&priv->lock);
1028 /* executed from streaming thread */
1030 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1032 GstRTSPStreamPrivate *priv = stream->priv;
1033 GstCaps *newcaps, *oldcaps;
1035 newcaps = gst_pad_get_current_caps (pad);
1037 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1040 g_mutex_lock (&priv->lock);
1041 oldcaps = priv->caps;
1042 priv->caps = newcaps;
1043 g_mutex_unlock (&priv->lock);
1046 gst_caps_unref (oldcaps);
1050 dump_structure (const GstStructure * s)
1054 sstr = gst_structure_to_string (s);
1055 GST_INFO ("structure: %s", sstr);
1059 static GstRTSPStreamTransport *
1060 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1062 GstRTSPStreamPrivate *priv = stream->priv;
1064 GstRTSPStreamTransport *result = NULL;
1069 if (rtcp_from == NULL)
1072 tmp = g_strrstr (rtcp_from, ":");
1076 port = atoi (tmp + 1);
1077 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1079 g_mutex_lock (&priv->lock);
1080 GST_INFO ("finding %s:%d in %d transports", dest, port,
1081 g_list_length (priv->transports));
1083 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1084 GstRTSPStreamTransport *trans = walk->data;
1085 const GstRTSPTransport *tr;
1088 tr = gst_rtsp_stream_transport_get_transport (trans);
1090 min = tr->client_port.min;
1091 max = tr->client_port.max;
1093 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1099 g_object_ref (result);
1100 g_mutex_unlock (&priv->lock);
1107 static GstRTSPStreamTransport *
1108 check_transport (GObject * source, GstRTSPStream * stream)
1110 GstStructure *stats;
1111 GstRTSPStreamTransport *trans;
1113 /* see if we have a stream to match with the origin of the RTCP packet */
1114 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1115 if (trans == NULL) {
1116 g_object_get (source, "stats", &stats, NULL);
1118 const gchar *rtcp_from;
1120 dump_structure (stats);
1122 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1123 if ((trans = find_transport (stream, rtcp_from))) {
1124 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1126 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1129 gst_structure_free (stats);
1137 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1139 GstRTSPStreamTransport *trans;
1141 GST_INFO ("%p: new source %p", stream, source);
1143 trans = check_transport (source, stream);
1146 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1150 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1152 GST_INFO ("%p: new SDES %p", stream, source);
1156 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1158 GstRTSPStreamTransport *trans;
1160 trans = check_transport (source, stream);
1163 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1164 gst_rtsp_stream_transport_keep_alive (trans);
1168 GstStructure *stats;
1169 g_object_get (source, "stats", &stats, NULL);
1171 dump_structure (stats);
1172 gst_structure_free (stats);
1179 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1181 GST_INFO ("%p: source %p bye", stream, source);
1185 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1187 GstRTSPStreamTransport *trans;
1189 GST_INFO ("%p: source %p bye timeout", stream, source);
1191 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1192 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1193 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1198 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1200 GstRTSPStreamTransport *trans;
1202 GST_INFO ("%p: source %p timeout", stream, source);
1204 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1205 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1206 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1210 static GstFlowReturn
1211 handle_new_sample (GstAppSink * sink, gpointer user_data)
1213 GstRTSPStreamPrivate *priv;
1217 GstRTSPStream *stream;
1219 sample = gst_app_sink_pull_sample (sink);
1223 stream = (GstRTSPStream *) user_data;
1224 priv = stream->priv;
1225 buffer = gst_sample_get_buffer (sample);
1227 g_mutex_lock (&priv->lock);
1228 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1229 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1231 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1232 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1234 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1237 g_mutex_unlock (&priv->lock);
1239 gst_sample_unref (sample);
1244 static GstAppSinkCallbacks sink_cb = {
1245 NULL, /* not interested in EOS */
1246 NULL, /* not interested in preroll samples */
1251 * gst_rtsp_stream_join_bin:
1252 * @stream: a #GstRTSPStream
1253 * @bin: a #GstBin to join
1254 * @rtpbin: a rtpbin element in @bin
1255 * @state: the target state of the new elements
1257 * Join the #Gstbin @bin that contains the element @rtpbin.
1259 * @stream will link to @rtpbin, which must be inside @bin. The elements
1260 * added to @bin will be set to the state given in @state.
1262 * Returns: %TRUE on success.
1265 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1266 GstElement * rtpbin, GstState state)
1268 GstRTSPStreamPrivate *priv;
1272 GstPad *pad, *teepad, *queuepad, *selpad;
1273 GstPadLinkReturn ret;
1275 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1276 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1277 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1279 priv = stream->priv;
1281 g_mutex_lock (&priv->lock);
1282 if (priv->is_joined)
1285 /* create a session with the same index as the stream */
1288 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1290 if (!alloc_ports (stream))
1293 /* update the dscp qos field in the sinks */
1294 update_dscp_qos (stream);
1296 /* get a pad for sending RTP */
1297 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1298 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1300 /* link the RTP pad to the session manager, it should not really fail unless
1301 * this is not really an RTP pad */
1302 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1303 if (ret != GST_PAD_LINK_OK)
1306 /* get pads from the RTP session element for sending and receiving
1308 name = g_strdup_printf ("send_rtp_src_%u", idx);
1309 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1311 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1312 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1314 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1315 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1317 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1318 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1321 /* get the session */
1322 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1324 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1326 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1328 g_signal_connect (priv->session, "on-ssrc-active",
1329 (GCallback) on_ssrc_active, stream);
1330 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1332 g_signal_connect (priv->session, "on-bye-timeout",
1333 (GCallback) on_bye_timeout, stream);
1334 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1337 for (i = 0; i < 2; i++) {
1338 /* For the sender we create this bit of pipeline for both
1339 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1340 * we need to add a queue before appsink to make the pipeline
1341 * not block. For the TCP case, we want to pump data to the
1342 * client as fast as possible anyway.
1344 * .--------. .-----. .---------.
1345 * | rtpbin | | tee | | udpsink |
1346 * | send->sink src->sink |
1347 * '--------' | | '---------'
1348 * | | .---------. .---------.
1349 * | | | queue | | appsink |
1350 * | src->sink src->sink |
1351 * '-----' '---------' '---------'
1353 /* make tee for RTP/RTCP */
1354 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1355 gst_bin_add (bin, priv->tee[i]);
1357 /* and link to rtpbin send pad */
1358 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1359 gst_pad_link (priv->send_src[i], pad);
1360 gst_object_unref (pad);
1363 gst_bin_add (bin, priv->udpsink[i]);
1365 /* link tee to udpsink */
1366 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1367 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1368 gst_pad_link (teepad, pad);
1369 gst_object_unref (pad);
1370 gst_object_unref (teepad);
1373 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1374 gst_bin_add (bin, priv->appqueue[i]);
1375 /* and link to tee */
1376 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1377 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1378 gst_pad_link (teepad, pad);
1379 gst_object_unref (pad);
1380 gst_object_unref (teepad);
1383 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1384 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1385 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1386 gst_bin_add (bin, priv->appsink[i]);
1387 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1388 &sink_cb, stream, NULL);
1389 /* and link to queue */
1390 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1391 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1392 gst_pad_link (queuepad, pad);
1393 gst_object_unref (pad);
1394 gst_object_unref (queuepad);
1396 /* For the receiver we create this bit of pipeline for both
1397 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1398 * and it is all funneled into the rtpbin receive pad.
1400 * .--------. .--------. .--------.
1401 * | udpsrc | | funnel | | rtpbin |
1402 * | src->sink src->sink |
1403 * '--------' | | '--------'
1407 * '--------' '--------'
1409 /* make funnel for the RTP/RTCP receivers */
1410 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1411 gst_bin_add (bin, priv->funnel[i]);
1413 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1414 gst_pad_link (pad, priv->recv_sink[i]);
1415 gst_object_unref (pad);
1417 if (priv->udpsrc_v4[i]) {
1418 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1420 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1421 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1423 gst_bin_add (bin, priv->udpsrc_v4[i]);
1425 /* and link to the funnel v4 */
1426 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1427 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1428 gst_pad_link (pad, selpad);
1429 gst_object_unref (pad);
1430 gst_object_unref (selpad);
1433 if (priv->udpsrc_v6[i]) {
1434 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1435 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1436 gst_bin_add (bin, priv->udpsrc_v6[i]);
1438 /* and link to the funnel v6 */
1439 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1440 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1441 gst_pad_link (pad, selpad);
1442 gst_object_unref (pad);
1443 gst_object_unref (selpad);
1446 /* make and add appsrc */
1447 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1448 gst_bin_add (bin, priv->appsrc[i]);
1449 /* and link to the funnel */
1450 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1451 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1452 gst_pad_link (pad, selpad);
1453 gst_object_unref (pad);
1454 gst_object_unref (selpad);
1456 /* check if we need to set to a special state */
1457 if (state != GST_STATE_NULL) {
1458 gst_element_set_state (priv->udpsink[i], state);
1459 gst_element_set_state (priv->appsink[i], state);
1460 gst_element_set_state (priv->appqueue[i], state);
1461 gst_element_set_state (priv->tee[i], state);
1462 gst_element_set_state (priv->funnel[i], state);
1463 gst_element_set_state (priv->appsrc[i], state);
1467 /* be notified of caps changes */
1468 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1469 (GCallback) caps_notify, stream);
1471 priv->is_joined = TRUE;
1472 g_mutex_unlock (&priv->lock);
1479 g_mutex_unlock (&priv->lock);
1484 g_mutex_unlock (&priv->lock);
1485 GST_WARNING ("failed to allocate ports %u", idx);
1490 GST_WARNING ("failed to link stream %u", idx);
1491 gst_object_unref (priv->send_rtp_sink);
1492 priv->send_rtp_sink = NULL;
1493 g_mutex_unlock (&priv->lock);
1499 * gst_rtsp_stream_leave_bin:
1500 * @stream: a #GstRTSPStream
1502 * @rtpbin: a rtpbin #GstElement
1504 * Remove the elements of @stream from @bin.
1506 * Return: %TRUE on success.
1509 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1510 GstElement * rtpbin)
1512 GstRTSPStreamPrivate *priv;
1515 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1516 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1517 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1519 priv = stream->priv;
1521 g_mutex_lock (&priv->lock);
1522 if (!priv->is_joined)
1523 goto was_not_joined;
1525 /* all transports must be removed by now */
1526 g_return_val_if_fail (priv->transports == NULL, FALSE);
1528 GST_INFO ("stream %p leaving bin", stream);
1530 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1531 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1532 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1533 gst_object_unref (priv->send_rtp_sink);
1534 priv->send_rtp_sink = NULL;
1536 for (i = 0; i < 2; i++) {
1537 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1538 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1539 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1540 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1541 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1542 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1543 if (priv->udpsrc_v4[i]) {
1544 /* and set udpsrc to NULL now before removing */
1545 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1546 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1547 /* removing them should also nicely release the request
1548 * pads when they finalize */
1549 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1551 if (priv->udpsrc_v6[i]) {
1552 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1553 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1554 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1556 gst_bin_remove (bin, priv->udpsink[i]);
1557 gst_bin_remove (bin, priv->appsrc[i]);
1558 gst_bin_remove (bin, priv->appsink[i]);
1559 gst_bin_remove (bin, priv->appqueue[i]);
1560 gst_bin_remove (bin, priv->tee[i]);
1561 gst_bin_remove (bin, priv->funnel[i]);
1563 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1564 gst_object_unref (priv->recv_sink[i]);
1565 priv->recv_sink[i] = NULL;
1567 priv->udpsrc_v4[i] = NULL;
1568 priv->udpsrc_v6[i] = NULL;
1569 priv->udpsink[i] = NULL;
1570 priv->appsrc[i] = NULL;
1571 priv->appsink[i] = NULL;
1572 priv->appqueue[i] = NULL;
1573 priv->tee[i] = NULL;
1574 priv->funnel[i] = NULL;
1576 gst_object_unref (priv->send_src[0]);
1577 priv->send_src[0] = NULL;
1579 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1580 gst_object_unref (priv->send_src[1]);
1581 priv->send_src[1] = NULL;
1583 g_object_unref (priv->session);
1584 priv->session = NULL;
1586 gst_caps_unref (priv->caps);
1589 priv->is_joined = FALSE;
1590 g_mutex_unlock (&priv->lock);
1601 * gst_rtsp_stream_get_rtpinfo:
1602 * @stream: a #GstRTSPStream
1603 * @rtptime: result RTP timestamp
1604 * @seq: result RTP seqnum
1606 * Retrieve the current rtptime and seq. This is used to
1607 * construct a RTPInfo reply header.
1609 * Returns: %TRUE when rtptime and seq could be determined.
1612 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1613 guint * rtptime, guint * seq)
1615 GstRTSPStreamPrivate *priv;
1616 GObjectClass *payobjclass;
1618 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1619 g_return_val_if_fail (rtptime != NULL, FALSE);
1620 g_return_val_if_fail (seq != NULL, FALSE);
1622 priv = stream->priv;
1624 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1626 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1627 !g_object_class_find_property (payobjclass, "timestamp"))
1630 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1636 * gst_rtsp_stream_get_caps:
1637 * @stream: a #GstRTSPStream
1639 * Retrieve the current caps of @stream.
1641 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1645 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1647 GstRTSPStreamPrivate *priv;
1650 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1652 priv = stream->priv;
1654 g_mutex_lock (&priv->lock);
1655 if ((result = priv->caps))
1656 gst_caps_ref (result);
1657 g_mutex_unlock (&priv->lock);
1663 * gst_rtsp_stream_recv_rtp:
1664 * @stream: a #GstRTSPStream
1665 * @buffer: (transfer full): a #GstBuffer
1667 * Handle an RTP buffer for the stream. This method is usually called when a
1668 * message has been received from a client using the TCP transport.
1670 * This function takes ownership of @buffer.
1672 * Returns: a GstFlowReturn.
1675 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1677 GstRTSPStreamPrivate *priv;
1679 GstElement *element;
1681 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1682 priv = stream->priv;
1683 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1684 g_return_val_if_fail (priv->is_joined, FALSE);
1686 g_mutex_lock (&priv->lock);
1687 element = gst_object_ref (priv->appsrc[0]);
1688 g_mutex_unlock (&priv->lock);
1690 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1692 gst_object_unref (element);
1698 * gst_rtsp_stream_recv_rtcp:
1699 * @stream: a #GstRTSPStream
1700 * @buffer: (transfer full): a #GstBuffer
1702 * Handle an RTCP buffer for the stream. This method is usually called when a
1703 * message has been received from a client using the TCP transport.
1705 * This function takes ownership of @buffer.
1707 * Returns: a GstFlowReturn.
1710 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1712 GstRTSPStreamPrivate *priv;
1714 GstElement *element;
1716 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1717 priv = stream->priv;
1718 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1719 g_return_val_if_fail (priv->is_joined, FALSE);
1721 g_mutex_lock (&priv->lock);
1722 element = gst_object_ref (priv->appsrc[1]);
1723 g_mutex_unlock (&priv->lock);
1725 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1727 gst_object_unref (element);
1732 /* must be called with lock */
1734 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1737 GstRTSPStreamPrivate *priv = stream->priv;
1738 const GstRTSPTransport *tr;
1740 tr = gst_rtsp_stream_transport_get_transport (trans);
1742 switch (tr->lower_transport) {
1743 case GST_RTSP_LOWER_TRANS_UDP:
1744 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1750 dest = tr->destination;
1751 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1756 min = tr->client_port.min;
1757 max = tr->client_port.max;
1761 GST_INFO ("adding %s:%d-%d", dest, min, max);
1762 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1763 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1765 GST_INFO ("setting ttl-mc %d", ttl);
1766 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1767 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1769 priv->transports = g_list_prepend (priv->transports, trans);
1771 GST_INFO ("removing %s:%d-%d", dest, min, max);
1772 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1773 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1774 priv->transports = g_list_remove (priv->transports, trans);
1778 case GST_RTSP_LOWER_TRANS_TCP:
1780 GST_INFO ("adding TCP %s", tr->destination);
1781 priv->transports = g_list_prepend (priv->transports, trans);
1783 GST_INFO ("removing TCP %s", tr->destination);
1784 priv->transports = g_list_remove (priv->transports, trans);
1788 goto unknown_transport;
1795 GST_INFO ("Unknown transport %d", tr->lower_transport);
1802 * gst_rtsp_stream_add_transport:
1803 * @stream: a #GstRTSPStream
1804 * @trans: a #GstRTSPStreamTransport
1806 * Add the transport in @trans to @stream. The media of @stream will
1807 * then also be send to the values configured in @trans.
1809 * @stream must be joined to a bin.
1811 * @trans must contain a valid #GstRTSPTransport.
1813 * Returns: %TRUE if @trans was added
1816 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1817 GstRTSPStreamTransport * trans)
1819 GstRTSPStreamPrivate *priv;
1822 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1823 priv = stream->priv;
1824 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1825 g_return_val_if_fail (priv->is_joined, FALSE);
1827 g_mutex_lock (&priv->lock);
1828 res = update_transport (stream, trans, TRUE);
1829 g_mutex_unlock (&priv->lock);
1835 * gst_rtsp_stream_remove_transport:
1836 * @stream: a #GstRTSPStream
1837 * @trans: a #GstRTSPStreamTransport
1839 * Remove the transport in @trans from @stream. The media of @stream will
1840 * not be sent to the values configured in @trans.
1842 * @stream must be joined to a bin.
1844 * @trans must contain a valid #GstRTSPTransport.
1846 * Returns: %TRUE if @trans was removed
1849 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1850 GstRTSPStreamTransport * trans)
1852 GstRTSPStreamPrivate *priv;
1855 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1856 priv = stream->priv;
1857 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1858 g_return_val_if_fail (priv->is_joined, FALSE);
1860 g_mutex_lock (&priv->lock);
1861 res = update_transport (stream, trans, FALSE);
1862 g_mutex_unlock (&priv->lock);