2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
25 #include <gst/app/gstappsrc.h>
26 #include <gst/app/gstappsink.h>
28 #include "rtsp-stream.h"
30 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
31 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
33 struct _GstRTSPStreamPrivate
38 GstElement *payloader;
43 /* pads on the rtpbin */
44 GstPad *send_rtp_sink;
48 /* the RTPSession object */
51 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
53 GstElement *udpsrc_v4[2];
55 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
57 GstElement *udpsrc_v6[2];
59 GstElement *udpsink[2];
61 /* for TCP transport */
62 GstElement *appsrc[2];
63 GstElement *appqueue[2];
64 GstElement *appsink[2];
67 GstElement *funnel[2];
69 /* server ports for sending/receiving over ipv4 */
70 GstRTSPRange server_port_v4;
71 GstRTSPAddress *server_addr_v4;
74 /* server ports for sending/receiving over ipv6 */
75 GstRTSPRange server_port_v6;
76 GstRTSPAddress *server_addr_v6;
79 /* multicast addresses */
80 GstRTSPAddressPool *pool;
81 GstRTSPAddress *addr_v4;
82 GstRTSPAddress *addr_v6;
84 /* the caps of the stream */
88 /* transports we stream to */
95 #define DEFAULT_CONTROL NULL
104 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
105 #define GST_CAT_DEFAULT rtsp_stream_debug
107 static GQuark ssrc_stream_map_key;
109 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
110 GValue * value, GParamSpec * pspec);
111 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
112 const GValue * value, GParamSpec * pspec);
114 static void gst_rtsp_stream_finalize (GObject * obj);
116 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
119 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
121 GObjectClass *gobject_class;
123 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
125 gobject_class = G_OBJECT_CLASS (klass);
127 gobject_class->get_property = gst_rtsp_stream_get_property;
128 gobject_class->set_property = gst_rtsp_stream_set_property;
129 gobject_class->finalize = gst_rtsp_stream_finalize;
131 g_object_class_install_property (gobject_class, PROP_CONTROL,
132 g_param_spec_string ("control", "Control",
133 "The control string for this stream", DEFAULT_CONTROL,
134 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
136 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
138 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
142 gst_rtsp_stream_init (GstRTSPStream * stream)
144 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
146 GST_DEBUG ("new stream %p", stream);
151 priv->control = g_strdup (DEFAULT_CONTROL);
153 g_mutex_init (&priv->lock);
157 gst_rtsp_stream_finalize (GObject * obj)
159 GstRTSPStream *stream;
160 GstRTSPStreamPrivate *priv;
162 stream = GST_RTSP_STREAM (obj);
165 GST_DEBUG ("finalize stream %p", stream);
167 /* we really need to be unjoined now */
168 g_return_if_fail (!priv->is_joined);
171 gst_rtsp_address_free (priv->addr_v4);
173 gst_rtsp_address_free (priv->addr_v6);
174 if (priv->server_addr_v4)
175 gst_rtsp_address_free (priv->server_addr_v4);
176 if (priv->server_addr_v6)
177 gst_rtsp_address_free (priv->server_addr_v6);
179 g_object_unref (priv->pool);
180 gst_object_unref (priv->payloader);
181 gst_object_unref (priv->srcpad);
182 g_free (priv->control);
183 g_mutex_clear (&priv->lock);
185 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
189 gst_rtsp_stream_get_property (GObject * object, guint propid,
190 GValue * value, GParamSpec * pspec)
192 GstRTSPStream *stream = GST_RTSP_STREAM (object);
196 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
199 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
204 gst_rtsp_stream_set_property (GObject * object, guint propid,
205 const GValue * value, GParamSpec * pspec)
207 GstRTSPStream *stream = GST_RTSP_STREAM (object);
211 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
214 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
219 * gst_rtsp_stream_new:
222 * @payloader: a #GstElement
224 * Create a new media stream with index @idx that handles RTP data on
225 * @srcpad and has a payloader element @payloader.
227 * Returns: a new #GstRTSPStream
230 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
232 GstRTSPStreamPrivate *priv;
233 GstRTSPStream *stream;
235 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
236 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
237 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
239 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
242 priv->payloader = gst_object_ref (payloader);
243 priv->srcpad = gst_object_ref (srcpad);
249 * gst_rtsp_stream_get_index:
250 * @stream: a #GstRTSPStream
252 * Get the stream index.
254 * Return: the stream index.
257 gst_rtsp_stream_get_index (GstRTSPStream * stream)
259 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
261 return stream->priv->idx;
265 * gst_rtsp_stream_get_srcpad:
266 * @stream: a #GstRTSPStream
268 * Get the srcpad associated with @stream.
270 * Return: the srcpad. Unref after usage.
273 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
275 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
277 return gst_object_ref (stream->priv->srcpad);
281 * gst_rtsp_stream_get_control:
282 * @stream: a #GstRTSPStream
284 * Get the control string to identify this stream.
286 * Return: the control string. free after usage.
289 gst_rtsp_stream_get_control (GstRTSPStream * stream)
291 GstRTSPStreamPrivate *priv;
294 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
298 g_mutex_lock (&priv->lock);
299 if ((result = g_strdup (priv->control)) == NULL)
300 result = g_strdup_printf ("stream=%d", priv->idx);
301 g_mutex_unlock (&priv->lock);
307 * gst_rtsp_stream_set_control:
308 * @stream: a #GstRTSPStream
309 * @control: a control string
311 * Set the control string in @stream.
314 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
316 GstRTSPStreamPrivate *priv;
318 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
322 g_mutex_lock (&priv->lock);
323 g_free (priv->control);
324 priv->control = g_strdup (control);
325 g_mutex_unlock (&priv->lock);
329 * gst_rtsp_stream_set_mtu:
330 * @stream: a #GstRTSPStream
333 * Configure the mtu in the payloader of @stream to @mtu.
336 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
338 GstRTSPStreamPrivate *priv;
340 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
344 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
346 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
350 * gst_rtsp_stream_get_mtu:
351 * @stream: a #GstRTSPStream
353 * Get the configured MTU in the payloader of @stream.
355 * Returns: the MTU of the payloader.
358 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
360 GstRTSPStreamPrivate *priv;
363 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
367 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
372 /* Update the dscp qos property on the udp sinks */
374 update_dscp_qos (GstRTSPStream * stream)
376 GstRTSPStreamPrivate *priv;
378 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
382 if (priv->udpsink[0]) {
383 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
387 if (priv->udpsink[1]) {
388 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
394 * gst_rtsp_stream_set_dscp_qos:
395 * @stream: a #GstRTSPStream
396 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
398 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
401 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
403 GstRTSPStreamPrivate *priv;
405 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
409 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
411 if (dscp_qos < -1 || dscp_qos > 63) {
412 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
416 priv->dscp_qos = dscp_qos;
418 update_dscp_qos (stream);
422 * gst_rtsp_stream_get_dscp_qos:
423 * @stream: a #GstRTSPStream
425 * Get the configured DSCP QoS in of the outgoing sockets.
427 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
430 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
432 GstRTSPStreamPrivate *priv;
434 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
438 return priv->dscp_qos;
443 * gst_rtsp_stream_set_address_pool:
444 * @stream: a #GstRTSPStream
445 * @pool: a #GstRTSPAddressPool
447 * configure @pool to be used as the address pool of @stream.
450 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
451 GstRTSPAddressPool * pool)
453 GstRTSPStreamPrivate *priv;
454 GstRTSPAddressPool *old;
456 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
460 GST_LOG_OBJECT (stream, "set address pool %p", pool);
462 g_mutex_lock (&priv->lock);
463 if ((old = priv->pool) != pool)
464 priv->pool = pool ? g_object_ref (pool) : NULL;
467 g_mutex_unlock (&priv->lock);
470 g_object_unref (old);
474 * gst_rtsp_stream_get_address_pool:
475 * @stream: a #GstRTSPStream
477 * Get the #GstRTSPAddressPool used as the address pool of @stream.
479 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
483 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
485 GstRTSPStreamPrivate *priv;
486 GstRTSPAddressPool *result;
488 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
492 g_mutex_lock (&priv->lock);
493 if ((result = priv->pool))
494 g_object_ref (result);
495 g_mutex_unlock (&priv->lock);
501 * gst_rtsp_stream_get_multicast_address:
502 * @stream: a #GstRTSPStream
503 * @family: the #GSocketFamily
505 * Get the multicast address of @stream for @family.
507 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
508 * allocated. gst_rtsp_address_free() after usage.
511 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
512 GSocketFamily family)
514 GstRTSPStreamPrivate *priv;
515 GstRTSPAddress *result;
516 GstRTSPAddress **addrp;
517 GstRTSPAddressFlags flags;
519 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
523 if (family == G_SOCKET_FAMILY_IPV6) {
524 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
525 addrp = &priv->addr_v4;
527 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
528 addrp = &priv->addr_v6;
531 g_mutex_lock (&priv->lock);
532 if (*addrp == NULL) {
533 if (priv->pool == NULL)
536 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
538 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
542 result = gst_rtsp_address_copy (*addrp);
543 g_mutex_unlock (&priv->lock);
550 GST_ERROR_OBJECT (stream, "no address pool specified");
551 g_mutex_unlock (&priv->lock);
556 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
557 g_mutex_unlock (&priv->lock);
563 * gst_rtsp_stream_reserve_address:
564 * @stream: a #GstRTSPStream
565 * @address: an address
570 * Reserve @address and @port as the address and port of @stream.
572 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
573 * reserved. gst_rtsp_address_free() after usage.
576 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
577 const gchar * address, guint port, guint n_ports, guint ttl)
579 GstRTSPStreamPrivate *priv;
580 GstRTSPAddress *result;
582 GSocketFamily family;
583 GstRTSPAddress **addrp;
585 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
586 g_return_val_if_fail (address != NULL, NULL);
587 g_return_val_if_fail (port > 0, NULL);
588 g_return_val_if_fail (n_ports > 0, NULL);
589 g_return_val_if_fail (ttl > 0, NULL);
593 addr = g_inet_address_new_from_string (address);
595 GST_ERROR ("failed to get inet addr from %s", address);
596 family = G_SOCKET_FAMILY_IPV4;
598 family = g_inet_address_get_family (addr);
599 g_object_unref (addr);
602 if (family == G_SOCKET_FAMILY_IPV6)
603 addrp = &priv->addr_v4;
605 addrp = &priv->addr_v6;
607 g_mutex_lock (&priv->lock);
608 if (*addrp == NULL) {
609 if (priv->pool == NULL)
612 *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
617 if (strcmp ((*addrp)->address, address) ||
618 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
619 (*addrp)->ttl != ttl)
620 goto different_address;
622 result = gst_rtsp_address_copy (*addrp);
623 g_mutex_unlock (&priv->lock);
630 GST_ERROR_OBJECT (stream, "no address pool specified");
631 g_mutex_unlock (&priv->lock);
636 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
638 g_mutex_unlock (&priv->lock);
643 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
644 " reserved", address);
645 g_mutex_unlock (&priv->lock);
651 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
652 GSocketFamily family, GstElement * udpsrc_out[2],
653 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
654 GstRTSPAddress ** server_addr_out)
656 GstStateChangeReturn ret;
657 GstElement *udpsrc0, *udpsrc1;
658 GstElement *udpsink0, *udpsink1;
659 GSocket *rtp_socket = NULL;
660 GSocket *rtcp_socket;
661 gint tmp_rtp, tmp_rtcp;
663 gint rtpport, rtcpport;
664 GList *rejected_addresses = NULL;
665 GstRTSPAddress *addr = NULL;
666 GInetAddress *inetaddr = NULL;
667 GSocketAddress *rtp_sockaddr = NULL;
668 GSocketAddress *rtcp_sockaddr = NULL;
669 const gchar *multisink_socket;
671 if (family == G_SOCKET_FAMILY_IPV6)
672 multisink_socket = "socket-v6";
674 multisink_socket = "socket";
682 /* Start with random port */
685 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
686 G_SOCKET_PROTOCOL_UDP, NULL);
688 goto no_udp_protocol;
690 if (*server_addr_out)
691 gst_rtsp_address_free (*server_addr_out);
693 /* try to allocate 2 UDP ports, the RTP port should be an even
694 * number and the RTCP port should be the next (uneven) port */
697 if (rtp_socket == NULL) {
698 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
699 G_SOCKET_PROTOCOL_UDP, NULL);
701 goto no_udp_protocol;
704 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
705 GstRTSPAddressFlags flags;
708 rejected_addresses = g_list_prepend (rejected_addresses, addr);
710 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
711 if (family == G_SOCKET_FAMILY_IPV6)
712 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
714 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
716 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
721 tmp_rtp = addr->port;
723 g_clear_object (&inetaddr);
724 inetaddr = g_inet_address_new_from_string (addr->address);
732 if (inetaddr == NULL)
733 inetaddr = g_inet_address_new_any (family);
736 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
737 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
738 g_object_unref (rtp_sockaddr);
741 g_object_unref (rtp_sockaddr);
743 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
744 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
745 g_clear_object (&rtp_sockaddr);
750 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
751 g_object_unref (rtp_sockaddr);
753 /* check if port is even */
754 if ((tmp_rtp & 1) != 0) {
755 /* port not even, close and allocate another */
757 g_clear_object (&rtp_socket);
762 tmp_rtcp = tmp_rtp + 1;
764 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
765 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
766 g_object_unref (rtcp_sockaddr);
767 g_clear_object (&rtp_socket);
770 g_object_unref (rtcp_sockaddr);
772 g_clear_object (&inetaddr);
774 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
775 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
777 if (udpsrc0 == NULL || udpsrc1 == NULL)
778 goto no_udp_protocol;
780 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
781 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
783 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
784 if (ret == GST_STATE_CHANGE_FAILURE)
786 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
787 if (ret == GST_STATE_CHANGE_FAILURE)
790 /* all fine, do port check */
791 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
792 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
794 /* this should not happen... */
795 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
799 udpsink0 = udpsink_out[0];
801 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
804 goto no_udp_protocol;
806 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
807 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
810 udpsink1 = udpsink_out[1];
812 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
815 goto no_udp_protocol;
817 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
818 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
819 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
821 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
822 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
823 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
824 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
825 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
826 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
827 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
828 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
830 /* we keep these elements, we will further configure them when the
831 * client told us to really use the UDP ports. */
832 udpsrc_out[0] = udpsrc0;
833 udpsrc_out[1] = udpsrc1;
834 udpsink_out[0] = udpsink0;
835 udpsink_out[1] = udpsink1;
836 server_port_out->min = rtpport;
837 server_port_out->max = rtcpport;
839 *server_addr_out = addr;
840 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
842 g_object_unref (rtp_socket);
843 g_object_unref (rtcp_socket);
871 gst_element_set_state (udpsrc0, GST_STATE_NULL);
872 gst_object_unref (udpsrc0);
875 gst_element_set_state (udpsrc1, GST_STATE_NULL);
876 gst_object_unref (udpsrc1);
879 gst_element_set_state (udpsink0, GST_STATE_NULL);
880 gst_object_unref (udpsink0);
883 gst_element_set_state (udpsink1, GST_STATE_NULL);
884 gst_object_unref (udpsink1);
887 g_object_unref (inetaddr);
888 g_list_free_full (rejected_addresses,
889 (GDestroyNotify) gst_rtsp_address_free);
891 gst_rtsp_address_free (addr);
893 g_object_unref (rtp_socket);
895 g_object_unref (rtcp_socket);
900 /* must be called with lock */
902 alloc_ports (GstRTSPStream * stream)
904 GstRTSPStreamPrivate *priv = stream->priv;
906 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
907 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
908 &priv->server_port_v4, &priv->server_addr_v4);
910 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
911 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
912 &priv->server_port_v6, &priv->server_addr_v6);
914 return priv->have_ipv4 || priv->have_ipv6;
918 * gst_rtsp_stream_get_server_port:
919 * @stream: a #GstRTSPStream
920 * @server_port: (out): result server port
922 * Fill @server_port with the port pair used by the server. This function can
923 * only be called when @stream has been joined.
926 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
927 GstRTSPRange * server_port, GSocketFamily family)
929 GstRTSPStreamPrivate *priv;
931 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
933 g_return_if_fail (priv->is_joined);
935 g_mutex_lock (&priv->lock);
936 if (family == G_SOCKET_FAMILY_IPV4) {
938 *server_port = priv->server_port_v4;
941 *server_port = priv->server_port_v6;
943 g_mutex_unlock (&priv->lock);
947 * gst_rtsp_stream_get_rtpsession:
948 * @stream: a #GstRTSPStream
950 * Get the RTP session of this stream.
952 * Returns: The RTP session of this stream. Unref after usage.
955 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
957 GstRTSPStreamPrivate *priv;
960 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
964 g_mutex_lock (&priv->lock);
965 if ((session = priv->session))
966 g_object_ref (session);
967 g_mutex_unlock (&priv->lock);
973 * gst_rtsp_stream_get_ssrc:
974 * @stream: a #GstRTSPStream
975 * @ssrc: (out): result ssrc
977 * Get the SSRC used by the RTP session of this stream. This function can only
978 * be called when @stream has been joined.
981 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
983 GstRTSPStreamPrivate *priv;
985 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
987 g_return_if_fail (priv->is_joined);
989 g_mutex_lock (&priv->lock);
990 if (ssrc && priv->session)
991 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
992 g_mutex_unlock (&priv->lock);
995 /* executed from streaming thread */
997 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
999 GstRTSPStreamPrivate *priv = stream->priv;
1000 GstCaps *newcaps, *oldcaps;
1002 newcaps = gst_pad_get_current_caps (pad);
1004 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1007 g_mutex_lock (&priv->lock);
1008 oldcaps = priv->caps;
1009 priv->caps = newcaps;
1010 g_mutex_unlock (&priv->lock);
1013 gst_caps_unref (oldcaps);
1017 dump_structure (const GstStructure * s)
1021 sstr = gst_structure_to_string (s);
1022 GST_INFO ("structure: %s", sstr);
1026 static GstRTSPStreamTransport *
1027 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1029 GstRTSPStreamPrivate *priv = stream->priv;
1031 GstRTSPStreamTransport *result = NULL;
1036 if (rtcp_from == NULL)
1039 tmp = g_strrstr (rtcp_from, ":");
1043 port = atoi (tmp + 1);
1044 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1046 g_mutex_lock (&priv->lock);
1047 GST_INFO ("finding %s:%d in %d transports", dest, port,
1048 g_list_length (priv->transports));
1050 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1051 GstRTSPStreamTransport *trans = walk->data;
1052 const GstRTSPTransport *tr;
1055 tr = gst_rtsp_stream_transport_get_transport (trans);
1057 min = tr->client_port.min;
1058 max = tr->client_port.max;
1060 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1066 g_object_ref (result);
1067 g_mutex_unlock (&priv->lock);
1074 static GstRTSPStreamTransport *
1075 check_transport (GObject * source, GstRTSPStream * stream)
1077 GstStructure *stats;
1078 GstRTSPStreamTransport *trans;
1080 /* see if we have a stream to match with the origin of the RTCP packet */
1081 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1082 if (trans == NULL) {
1083 g_object_get (source, "stats", &stats, NULL);
1085 const gchar *rtcp_from;
1087 dump_structure (stats);
1089 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1090 if ((trans = find_transport (stream, rtcp_from))) {
1091 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1093 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1096 gst_structure_free (stats);
1104 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1106 GstRTSPStreamTransport *trans;
1108 GST_INFO ("%p: new source %p", stream, source);
1110 trans = check_transport (source, stream);
1113 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1117 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1119 GST_INFO ("%p: new SDES %p", stream, source);
1123 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1125 GstRTSPStreamTransport *trans;
1127 trans = check_transport (source, stream);
1130 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1131 gst_rtsp_stream_transport_keep_alive (trans);
1135 GstStructure *stats;
1136 g_object_get (source, "stats", &stats, NULL);
1138 dump_structure (stats);
1139 gst_structure_free (stats);
1146 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1148 GST_INFO ("%p: source %p bye", stream, source);
1152 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1154 GstRTSPStreamTransport *trans;
1156 GST_INFO ("%p: source %p bye timeout", stream, source);
1158 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1159 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1160 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1165 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1167 GstRTSPStreamTransport *trans;
1169 GST_INFO ("%p: source %p timeout", stream, source);
1171 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1172 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1173 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1177 static GstFlowReturn
1178 handle_new_sample (GstAppSink * sink, gpointer user_data)
1180 GstRTSPStreamPrivate *priv;
1184 GstRTSPStream *stream;
1186 sample = gst_app_sink_pull_sample (sink);
1190 stream = (GstRTSPStream *) user_data;
1191 priv = stream->priv;
1192 buffer = gst_sample_get_buffer (sample);
1194 g_mutex_lock (&priv->lock);
1195 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1196 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1198 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1199 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1201 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1204 g_mutex_unlock (&priv->lock);
1206 gst_sample_unref (sample);
1211 static GstAppSinkCallbacks sink_cb = {
1212 NULL, /* not interested in EOS */
1213 NULL, /* not interested in preroll samples */
1218 * gst_rtsp_stream_join_bin:
1219 * @stream: a #GstRTSPStream
1220 * @bin: a #GstBin to join
1221 * @rtpbin: a rtpbin element in @bin
1222 * @state: the target state of the new elements
1224 * Join the #Gstbin @bin that contains the element @rtpbin.
1226 * @stream will link to @rtpbin, which must be inside @bin. The elements
1227 * added to @bin will be set to the state given in @state.
1229 * Returns: %TRUE on success.
1232 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1233 GstElement * rtpbin, GstState state)
1235 GstRTSPStreamPrivate *priv;
1238 GstPad *pad, *teepad, *queuepad, *selpad;
1239 GstPadLinkReturn ret;
1241 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1242 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1243 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1245 priv = stream->priv;
1247 g_mutex_lock (&priv->lock);
1248 if (priv->is_joined)
1251 /* create a session with the same index as the stream */
1254 GST_INFO ("stream %p joining bin as session %d", stream, idx);
1256 if (!alloc_ports (stream))
1259 /* update the dscp qos field in the sinks */
1260 update_dscp_qos (stream);
1262 /* get a pad for sending RTP */
1263 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1264 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1266 /* link the RTP pad to the session manager, it should not really fail unless
1267 * this is not really an RTP pad */
1268 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1269 if (ret != GST_PAD_LINK_OK)
1272 /* get pads from the RTP session element for sending and receiving
1274 name = g_strdup_printf ("send_rtp_src_%u", idx);
1275 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1277 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1278 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1280 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1281 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1283 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1284 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1287 /* get the session */
1288 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1290 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1292 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1294 g_signal_connect (priv->session, "on-ssrc-active",
1295 (GCallback) on_ssrc_active, stream);
1296 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1298 g_signal_connect (priv->session, "on-bye-timeout",
1299 (GCallback) on_bye_timeout, stream);
1300 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1303 for (i = 0; i < 2; i++) {
1304 /* For the sender we create this bit of pipeline for both
1305 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1306 * we need to add a queue before appsink to make the pipeline
1307 * not block. For the TCP case, we want to pump data to the
1308 * client as fast as possible anyway.
1310 * .--------. .-----. .---------.
1311 * | rtpbin | | tee | | udpsink |
1312 * | send->sink src->sink |
1313 * '--------' | | '---------'
1314 * | | .---------. .---------.
1315 * | | | queue | | appsink |
1316 * | src->sink src->sink |
1317 * '-----' '---------' '---------'
1319 /* make tee for RTP/RTCP */
1320 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1321 gst_bin_add (bin, priv->tee[i]);
1323 /* and link to rtpbin send pad */
1324 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1325 gst_pad_link (priv->send_src[i], pad);
1326 gst_object_unref (pad);
1329 gst_bin_add (bin, priv->udpsink[i]);
1331 /* link tee to udpsink */
1332 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1333 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1334 gst_pad_link (teepad, pad);
1335 gst_object_unref (pad);
1336 gst_object_unref (teepad);
1339 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1340 gst_bin_add (bin, priv->appqueue[i]);
1341 /* and link to tee */
1342 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1343 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1344 gst_pad_link (teepad, pad);
1345 gst_object_unref (pad);
1346 gst_object_unref (teepad);
1349 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1350 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1351 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1352 gst_bin_add (bin, priv->appsink[i]);
1353 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1354 &sink_cb, stream, NULL);
1355 /* and link to queue */
1356 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1357 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1358 gst_pad_link (queuepad, pad);
1359 gst_object_unref (pad);
1360 gst_object_unref (queuepad);
1362 /* For the receiver we create this bit of pipeline for both
1363 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1364 * and it is all funneled into the rtpbin receive pad.
1366 * .--------. .--------. .--------.
1367 * | udpsrc | | funnel | | rtpbin |
1368 * | src->sink src->sink |
1369 * '--------' | | '--------'
1373 * '--------' '--------'
1375 /* make funnel for the RTP/RTCP receivers */
1376 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1377 gst_bin_add (bin, priv->funnel[i]);
1379 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1380 gst_pad_link (pad, priv->recv_sink[i]);
1381 gst_object_unref (pad);
1383 if (priv->udpsrc_v4[i]) {
1384 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1386 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1387 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1389 gst_bin_add (bin, priv->udpsrc_v4[i]);
1391 /* and link to the funnel v4 */
1392 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1393 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1394 gst_pad_link (pad, selpad);
1395 gst_object_unref (pad);
1396 gst_object_unref (selpad);
1399 if (priv->udpsrc_v6[i]) {
1400 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1401 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1402 gst_bin_add (bin, priv->udpsrc_v6[i]);
1404 /* and link to the funnel v6 */
1405 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1406 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1407 gst_pad_link (pad, selpad);
1408 gst_object_unref (pad);
1409 gst_object_unref (selpad);
1412 /* make and add appsrc */
1413 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1414 gst_bin_add (bin, priv->appsrc[i]);
1415 /* and link to the funnel */
1416 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1417 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1418 gst_pad_link (pad, selpad);
1419 gst_object_unref (pad);
1420 gst_object_unref (selpad);
1422 /* check if we need to set to a special state */
1423 if (state != GST_STATE_NULL) {
1424 gst_element_set_state (priv->udpsink[i], state);
1425 gst_element_set_state (priv->appsink[i], state);
1426 gst_element_set_state (priv->appqueue[i], state);
1427 gst_element_set_state (priv->tee[i], state);
1428 gst_element_set_state (priv->funnel[i], state);
1429 gst_element_set_state (priv->appsrc[i], state);
1433 /* be notified of caps changes */
1434 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1435 (GCallback) caps_notify, stream);
1437 priv->is_joined = TRUE;
1438 g_mutex_unlock (&priv->lock);
1445 g_mutex_unlock (&priv->lock);
1450 g_mutex_unlock (&priv->lock);
1451 GST_WARNING ("failed to allocate ports %d", idx);
1456 GST_WARNING ("failed to link stream %d", idx);
1457 gst_object_unref (priv->send_rtp_sink);
1458 priv->send_rtp_sink = NULL;
1459 g_mutex_unlock (&priv->lock);
1465 * gst_rtsp_stream_leave_bin:
1466 * @stream: a #GstRTSPStream
1468 * @rtpbin: a rtpbin #GstElement
1470 * Remove the elements of @stream from @bin.
1472 * Return: %TRUE on success.
1475 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1476 GstElement * rtpbin)
1478 GstRTSPStreamPrivate *priv;
1481 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1482 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1483 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1485 priv = stream->priv;
1487 g_mutex_lock (&priv->lock);
1488 if (!priv->is_joined)
1489 goto was_not_joined;
1491 /* all transports must be removed by now */
1492 g_return_val_if_fail (priv->transports == NULL, FALSE);
1494 GST_INFO ("stream %p leaving bin", stream);
1496 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1497 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1498 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1499 gst_object_unref (priv->send_rtp_sink);
1500 priv->send_rtp_sink = NULL;
1502 for (i = 0; i < 2; i++) {
1503 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1504 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1505 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1506 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1507 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1508 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1509 if (priv->udpsrc_v4[i]) {
1510 /* and set udpsrc to NULL now before removing */
1511 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1512 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1513 /* removing them should also nicely release the request
1514 * pads when they finalize */
1515 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1517 if (priv->udpsrc_v6[i]) {
1518 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1519 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1520 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1522 gst_bin_remove (bin, priv->udpsink[i]);
1523 gst_bin_remove (bin, priv->appsrc[i]);
1524 gst_bin_remove (bin, priv->appsink[i]);
1525 gst_bin_remove (bin, priv->appqueue[i]);
1526 gst_bin_remove (bin, priv->tee[i]);
1527 gst_bin_remove (bin, priv->funnel[i]);
1529 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1530 gst_object_unref (priv->recv_sink[i]);
1531 priv->recv_sink[i] = NULL;
1533 priv->udpsrc_v4[i] = NULL;
1534 priv->udpsrc_v6[i] = NULL;
1535 priv->udpsink[i] = NULL;
1536 priv->appsrc[i] = NULL;
1537 priv->appsink[i] = NULL;
1538 priv->appqueue[i] = NULL;
1539 priv->tee[i] = NULL;
1540 priv->funnel[i] = NULL;
1542 gst_object_unref (priv->send_src[0]);
1543 priv->send_src[0] = NULL;
1545 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1546 gst_object_unref (priv->send_src[1]);
1547 priv->send_src[1] = NULL;
1549 g_object_unref (priv->session);
1550 priv->session = NULL;
1552 gst_caps_unref (priv->caps);
1555 priv->is_joined = FALSE;
1556 g_mutex_unlock (&priv->lock);
1567 * gst_rtsp_stream_get_rtpinfo:
1568 * @stream: a #GstRTSPStream
1569 * @rtptime: result RTP timestamp
1570 * @seq: result RTP seqnum
1572 * Retrieve the current rtptime and seq. This is used to
1573 * construct a RTPInfo reply header.
1575 * Returns: %TRUE when rtptime and seq could be determined.
1578 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1579 guint * rtptime, guint * seq)
1581 GstRTSPStreamPrivate *priv;
1582 GObjectClass *payobjclass;
1584 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1585 g_return_val_if_fail (rtptime != NULL, FALSE);
1586 g_return_val_if_fail (seq != NULL, FALSE);
1588 priv = stream->priv;
1590 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1592 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1593 !g_object_class_find_property (payobjclass, "timestamp"))
1596 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1602 * gst_rtsp_stream_get_caps:
1603 * @stream: a #GstRTSPStream
1605 * Retrieve the current caps of @stream.
1607 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1611 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1613 GstRTSPStreamPrivate *priv;
1616 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1618 priv = stream->priv;
1620 g_mutex_lock (&priv->lock);
1621 if ((result = priv->caps))
1622 gst_caps_ref (result);
1623 g_mutex_unlock (&priv->lock);
1629 * gst_rtsp_stream_recv_rtp:
1630 * @stream: a #GstRTSPStream
1631 * @buffer: (transfer full): a #GstBuffer
1633 * Handle an RTP buffer for the stream. This method is usually called when a
1634 * message has been received from a client using the TCP transport.
1636 * This function takes ownership of @buffer.
1638 * Returns: a GstFlowReturn.
1641 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1643 GstRTSPStreamPrivate *priv;
1645 GstElement *element;
1647 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1648 priv = stream->priv;
1649 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1650 g_return_val_if_fail (priv->is_joined, FALSE);
1652 g_mutex_lock (&priv->lock);
1653 element = gst_object_ref (priv->appsrc[0]);
1654 g_mutex_unlock (&priv->lock);
1656 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1658 gst_object_unref (element);
1664 * gst_rtsp_stream_recv_rtcp:
1665 * @stream: a #GstRTSPStream
1666 * @buffer: (transfer full): a #GstBuffer
1668 * Handle an RTCP buffer for the stream. This method is usually called when a
1669 * message has been received from a client using the TCP transport.
1671 * This function takes ownership of @buffer.
1673 * Returns: a GstFlowReturn.
1676 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1678 GstRTSPStreamPrivate *priv;
1680 GstElement *element;
1682 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1683 priv = stream->priv;
1684 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1685 g_return_val_if_fail (priv->is_joined, FALSE);
1687 g_mutex_lock (&priv->lock);
1688 element = gst_object_ref (priv->appsrc[1]);
1689 g_mutex_unlock (&priv->lock);
1691 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1693 gst_object_unref (element);
1698 /* must be called with lock */
1700 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1703 GstRTSPStreamPrivate *priv = stream->priv;
1704 const GstRTSPTransport *tr;
1706 tr = gst_rtsp_stream_transport_get_transport (trans);
1708 switch (tr->lower_transport) {
1709 case GST_RTSP_LOWER_TRANS_UDP:
1710 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1716 dest = tr->destination;
1717 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1722 min = tr->client_port.min;
1723 max = tr->client_port.max;
1727 GST_INFO ("adding %s:%d-%d", dest, min, max);
1728 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1729 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1731 GST_INFO ("setting ttl-mc %d", ttl);
1732 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1733 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1735 priv->transports = g_list_prepend (priv->transports, trans);
1737 GST_INFO ("removing %s:%d-%d", dest, min, max);
1738 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1739 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1740 priv->transports = g_list_remove (priv->transports, trans);
1744 case GST_RTSP_LOWER_TRANS_TCP:
1746 GST_INFO ("adding TCP %s", tr->destination);
1747 priv->transports = g_list_prepend (priv->transports, trans);
1749 GST_INFO ("removing TCP %s", tr->destination);
1750 priv->transports = g_list_remove (priv->transports, trans);
1754 goto unknown_transport;
1761 GST_INFO ("Unknown transport %d", tr->lower_transport);
1768 * gst_rtsp_stream_add_transport:
1769 * @stream: a #GstRTSPStream
1770 * @trans: a #GstRTSPStreamTransport
1772 * Add the transport in @trans to @stream. The media of @stream will
1773 * then also be send to the values configured in @trans.
1775 * @stream must be joined to a bin.
1777 * @trans must contain a valid #GstRTSPTransport.
1779 * Returns: %TRUE if @trans was added
1782 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1783 GstRTSPStreamTransport * trans)
1785 GstRTSPStreamPrivate *priv;
1788 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1789 priv = stream->priv;
1790 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1791 g_return_val_if_fail (priv->is_joined, FALSE);
1793 g_mutex_lock (&priv->lock);
1794 res = update_transport (stream, trans, TRUE);
1795 g_mutex_unlock (&priv->lock);
1801 * gst_rtsp_stream_remove_transport:
1802 * @stream: a #GstRTSPStream
1803 * @trans: a #GstRTSPStreamTransport
1805 * Remove the transport in @trans from @stream. The media of @stream will
1806 * not be sent to the values configured in @trans.
1808 * @stream must be joined to a bin.
1810 * @trans must contain a valid #GstRTSPTransport.
1812 * Returns: %TRUE if @trans was removed
1815 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1816 GstRTSPStreamTransport * trans)
1818 GstRTSPStreamPrivate *priv;
1821 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1822 priv = stream->priv;
1823 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1824 g_return_val_if_fail (priv->is_joined, FALSE);
1826 g_mutex_lock (&priv->lock);
1827 res = update_transport (stream, trans, FALSE);
1828 g_mutex_unlock (&priv->lock);