2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
21 * @short_description: A media stream
22 * @see_also: #GstRTSPMedia
24 * The #GstRTSPStream object manages the data transport for one stream. It
25 * is created from a payloader element and a source pad that produce the RTP
26 * packets for the stream.
28 * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
29 * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
31 * The #GstRTSPStream will use the configured addresspool, as set with
32 * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
33 * stream. With gst_rtsp_stream_get_multicast_address() you can get the
36 * With gst_rtsp_stream_get_server_port () you can get the port that the server
37 * will use to receive RTCP. This is the part that the clients will use to send
40 * With gst_rtsp_stream_add_transport() destinations can be added where the
41 * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
42 * the destination again.
44 * Last reviewed on 2013-07-16 (1.0.0)
53 #include <gst/app/gstappsrc.h>
54 #include <gst/app/gstappsink.h>
56 #include "rtsp-stream.h"
58 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
59 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
61 struct _GstRTSPStreamPrivate
66 GstElement *payloader;
71 GstRTSPProfile profiles;
72 GstRTSPLowerTrans protocols;
74 /* pads on the rtpbin */
75 GstPad *send_rtp_sink;
79 /* the RTPSession object */
82 /* SRTP encoder/decoder */
86 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
88 GstElement *udpsrc_v4[2];
90 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
92 GstElement *udpsrc_v6[2];
94 GstElement *udpsink[2];
96 /* for TCP transport */
97 GstElement *appsrc[2];
98 GstElement *appqueue[2];
99 GstElement *appsink[2];
102 GstElement *funnel[2];
104 /* server ports for sending/receiving over ipv4 */
105 GstRTSPRange server_port_v4;
106 GstRTSPAddress *server_addr_v4;
109 /* server ports for sending/receiving over ipv6 */
110 GstRTSPRange server_port_v6;
111 GstRTSPAddress *server_addr_v6;
114 /* multicast addresses */
115 GstRTSPAddressPool *pool;
116 GstRTSPAddress *addr_v4;
117 GstRTSPAddress *addr_v6;
119 /* the caps of the stream */
123 /* transports we stream to */
131 /* stream blocking */
136 #define DEFAULT_CONTROL NULL
137 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
138 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
139 GST_RTSP_LOWER_TRANS_TCP
150 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
151 #define GST_CAT_DEFAULT rtsp_stream_debug
153 static GQuark ssrc_stream_map_key;
155 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
156 GValue * value, GParamSpec * pspec);
157 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
158 const GValue * value, GParamSpec * pspec);
160 static void gst_rtsp_stream_finalize (GObject * obj);
162 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
165 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
167 GObjectClass *gobject_class;
169 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
171 gobject_class = G_OBJECT_CLASS (klass);
173 gobject_class->get_property = gst_rtsp_stream_get_property;
174 gobject_class->set_property = gst_rtsp_stream_set_property;
175 gobject_class->finalize = gst_rtsp_stream_finalize;
177 g_object_class_install_property (gobject_class, PROP_CONTROL,
178 g_param_spec_string ("control", "Control",
179 "The control string for this stream", DEFAULT_CONTROL,
180 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
182 g_object_class_install_property (gobject_class, PROP_PROFILES,
183 g_param_spec_flags ("profiles", "Profiles",
184 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
185 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
187 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
188 g_param_spec_flags ("protocols", "Protocols",
189 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
190 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
192 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
194 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
198 gst_rtsp_stream_init (GstRTSPStream * stream)
200 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
202 GST_DEBUG ("new stream %p", stream);
207 priv->control = g_strdup (DEFAULT_CONTROL);
208 priv->profiles = DEFAULT_PROFILES;
209 priv->protocols = DEFAULT_PROTOCOLS;
211 g_mutex_init (&priv->lock);
215 gst_rtsp_stream_finalize (GObject * obj)
217 GstRTSPStream *stream;
218 GstRTSPStreamPrivate *priv;
220 stream = GST_RTSP_STREAM (obj);
223 GST_DEBUG ("finalize stream %p", stream);
225 /* we really need to be unjoined now */
226 g_return_if_fail (!priv->is_joined);
229 gst_rtsp_address_free (priv->addr_v4);
231 gst_rtsp_address_free (priv->addr_v6);
232 if (priv->server_addr_v4)
233 gst_rtsp_address_free (priv->server_addr_v4);
234 if (priv->server_addr_v6)
235 gst_rtsp_address_free (priv->server_addr_v6);
237 g_object_unref (priv->pool);
238 gst_object_unref (priv->payloader);
239 gst_object_unref (priv->srcpad);
240 g_free (priv->control);
241 g_mutex_clear (&priv->lock);
243 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
247 gst_rtsp_stream_get_property (GObject * object, guint propid,
248 GValue * value, GParamSpec * pspec)
250 GstRTSPStream *stream = GST_RTSP_STREAM (object);
254 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
257 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
260 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
263 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
268 gst_rtsp_stream_set_property (GObject * object, guint propid,
269 const GValue * value, GParamSpec * pspec)
271 GstRTSPStream *stream = GST_RTSP_STREAM (object);
275 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
278 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
281 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
284 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
289 * gst_rtsp_stream_new:
292 * @payloader: a #GstElement
294 * Create a new media stream with index @idx that handles RTP data on
295 * @srcpad and has a payloader element @payloader.
297 * Returns: (transfer full): a new #GstRTSPStream
300 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
302 GstRTSPStreamPrivate *priv;
303 GstRTSPStream *stream;
305 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
306 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
307 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
309 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
312 priv->payloader = gst_object_ref (payloader);
313 priv->srcpad = gst_object_ref (srcpad);
319 * gst_rtsp_stream_get_index:
320 * @stream: a #GstRTSPStream
322 * Get the stream index.
324 * Return: the stream index.
327 gst_rtsp_stream_get_index (GstRTSPStream * stream)
329 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
331 return stream->priv->idx;
335 * gst_rtsp_stream_get_pt:
336 * @stream: a #GstRTSPStream
338 * Get the stream payload type.
340 * Return: the stream payload type.
343 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
345 GstRTSPStreamPrivate *priv;
348 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
352 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
358 * gst_rtsp_stream_get_srcpad:
359 * @stream: a #GstRTSPStream
361 * Get the srcpad associated with @stream.
363 * Returns: (transfer full): the srcpad. Unref after usage.
366 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
368 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
370 return gst_object_ref (stream->priv->srcpad);
374 * gst_rtsp_stream_get_control:
375 * @stream: a #GstRTSPStream
377 * Get the control string to identify this stream.
379 * Returns: (transfer full): the control string. g_free() after usage.
382 gst_rtsp_stream_get_control (GstRTSPStream * stream)
384 GstRTSPStreamPrivate *priv;
387 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
391 g_mutex_lock (&priv->lock);
392 if ((result = g_strdup (priv->control)) == NULL)
393 result = g_strdup_printf ("stream=%u", priv->idx);
394 g_mutex_unlock (&priv->lock);
400 * gst_rtsp_stream_set_control:
401 * @stream: a #GstRTSPStream
402 * @control: a control string
404 * Set the control string in @stream.
407 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
409 GstRTSPStreamPrivate *priv;
411 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
415 g_mutex_lock (&priv->lock);
416 g_free (priv->control);
417 priv->control = g_strdup (control);
418 g_mutex_unlock (&priv->lock);
422 * gst_rtsp_stream_has_control:
423 * @stream: a #GstRTSPStream
424 * @control: a control string
426 * Check if @stream has the control string @control.
428 * Returns: %TRUE is @stream has @control as the control string
431 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
433 GstRTSPStreamPrivate *priv;
436 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
440 g_mutex_lock (&priv->lock);
442 res = (g_strcmp0 (priv->control, control) == 0);
446 if (sscanf (control, "stream=%u", &streamid) > 0)
447 res = (streamid == priv->idx);
451 g_mutex_unlock (&priv->lock);
457 * gst_rtsp_stream_set_mtu:
458 * @stream: a #GstRTSPStream
461 * Configure the mtu in the payloader of @stream to @mtu.
464 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
466 GstRTSPStreamPrivate *priv;
468 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
472 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
474 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
478 * gst_rtsp_stream_get_mtu:
479 * @stream: a #GstRTSPStream
481 * Get the configured MTU in the payloader of @stream.
483 * Returns: the MTU of the payloader.
486 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
488 GstRTSPStreamPrivate *priv;
491 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
495 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
500 /* Update the dscp qos property on the udp sinks */
502 update_dscp_qos (GstRTSPStream * stream)
504 GstRTSPStreamPrivate *priv;
506 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
510 if (priv->udpsink[0]) {
511 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
515 if (priv->udpsink[1]) {
516 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
522 * gst_rtsp_stream_set_dscp_qos:
523 * @stream: a #GstRTSPStream
524 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
526 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
529 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
531 GstRTSPStreamPrivate *priv;
533 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
537 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
539 if (dscp_qos < -1 || dscp_qos > 63) {
540 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
544 priv->dscp_qos = dscp_qos;
546 update_dscp_qos (stream);
550 * gst_rtsp_stream_get_dscp_qos:
551 * @stream: a #GstRTSPStream
553 * Get the configured DSCP QoS in of the outgoing sockets.
555 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
558 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
560 GstRTSPStreamPrivate *priv;
562 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
566 return priv->dscp_qos;
570 * gst_rtsp_stream_is_transport_supported:
571 * @stream: a #GstRTSPStream
572 * @transport: (transfer none): a #GstRTSPTransport
574 * Check if @transport can be handled by stream
576 * Returns: %TRUE if @transport can be handled by @stream.
579 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
580 GstRTSPTransport * transport)
582 GstRTSPStreamPrivate *priv;
584 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
588 g_mutex_lock (&priv->lock);
589 if (transport->trans != GST_RTSP_TRANS_RTP)
590 goto unsupported_transmode;
592 if (!(transport->profile & priv->profiles))
593 goto unsupported_profile;
595 if (!(transport->lower_transport & priv->protocols))
596 goto unsupported_ltrans;
598 g_mutex_unlock (&priv->lock);
603 unsupported_transmode:
605 GST_DEBUG ("unsupported transport mode %d", transport->trans);
606 g_mutex_unlock (&priv->lock);
611 GST_DEBUG ("unsupported profile %d", transport->profile);
612 g_mutex_unlock (&priv->lock);
617 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
618 g_mutex_unlock (&priv->lock);
624 * gst_rtsp_stream_set_profiles:
625 * @stream: a #GstRTSPStream
626 * @profiles: the new profiles
628 * Configure the allowed profiles for @stream.
631 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
633 GstRTSPStreamPrivate *priv;
635 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
639 g_mutex_lock (&priv->lock);
640 priv->profiles = profiles;
641 g_mutex_unlock (&priv->lock);
645 * gst_rtsp_stream_get_profiles:
646 * @stream: a #GstRTSPStream
648 * Get the allowed profiles of @stream.
650 * Returns: a #GstRTSPProfile
653 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
655 GstRTSPStreamPrivate *priv;
658 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
662 g_mutex_lock (&priv->lock);
663 res = priv->profiles;
664 g_mutex_unlock (&priv->lock);
670 * gst_rtsp_stream_set_protocols:
671 * @stream: a #GstRTSPStream
672 * @protocols: the new flags
674 * Configure the allowed lower transport for @stream.
677 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
678 GstRTSPLowerTrans protocols)
680 GstRTSPStreamPrivate *priv;
682 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
686 g_mutex_lock (&priv->lock);
687 priv->protocols = protocols;
688 g_mutex_unlock (&priv->lock);
692 * gst_rtsp_stream_get_protocols:
693 * @stream: a #GstRTSPStream
695 * Get the allowed protocols of @stream.
697 * Returns: a #GstRTSPLowerTrans
700 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
702 GstRTSPStreamPrivate *priv;
703 GstRTSPLowerTrans res;
705 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
706 GST_RTSP_LOWER_TRANS_UNKNOWN);
710 g_mutex_lock (&priv->lock);
711 res = priv->protocols;
712 g_mutex_unlock (&priv->lock);
718 * gst_rtsp_stream_set_address_pool:
719 * @stream: a #GstRTSPStream
720 * @pool: (transfer none): a #GstRTSPAddressPool
722 * configure @pool to be used as the address pool of @stream.
725 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
726 GstRTSPAddressPool * pool)
728 GstRTSPStreamPrivate *priv;
729 GstRTSPAddressPool *old;
731 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
735 GST_LOG_OBJECT (stream, "set address pool %p", pool);
737 g_mutex_lock (&priv->lock);
738 if ((old = priv->pool) != pool)
739 priv->pool = pool ? g_object_ref (pool) : NULL;
742 g_mutex_unlock (&priv->lock);
745 g_object_unref (old);
749 * gst_rtsp_stream_get_address_pool:
750 * @stream: a #GstRTSPStream
752 * Get the #GstRTSPAddressPool used as the address pool of @stream.
754 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
758 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
760 GstRTSPStreamPrivate *priv;
761 GstRTSPAddressPool *result;
763 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
767 g_mutex_lock (&priv->lock);
768 if ((result = priv->pool))
769 g_object_ref (result);
770 g_mutex_unlock (&priv->lock);
776 * gst_rtsp_stream_get_multicast_address:
777 * @stream: a #GstRTSPStream
778 * @family: the #GSocketFamily
780 * Get the multicast address of @stream for @family.
782 * Returns: (transfer full): the #GstRTSPAddress of @stream or %NULL when no
783 * address could be allocated. gst_rtsp_address_free() after usage.
786 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
787 GSocketFamily family)
789 GstRTSPStreamPrivate *priv;
790 GstRTSPAddress *result;
791 GstRTSPAddress **addrp;
792 GstRTSPAddressFlags flags;
794 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
798 if (family == G_SOCKET_FAMILY_IPV6) {
799 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
800 addrp = &priv->addr_v6;
802 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
803 addrp = &priv->addr_v4;
806 g_mutex_lock (&priv->lock);
807 if (*addrp == NULL) {
808 if (priv->pool == NULL)
811 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
813 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
817 result = gst_rtsp_address_copy (*addrp);
818 g_mutex_unlock (&priv->lock);
825 GST_ERROR_OBJECT (stream, "no address pool specified");
826 g_mutex_unlock (&priv->lock);
831 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
832 g_mutex_unlock (&priv->lock);
838 * gst_rtsp_stream_reserve_address:
839 * @stream: a #GstRTSPStream
840 * @address: an address
845 * Reserve @address and @port as the address and port of @stream.
847 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
848 * reserved. gst_rtsp_address_free() after usage.
851 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
852 const gchar * address, guint port, guint n_ports, guint ttl)
854 GstRTSPStreamPrivate *priv;
855 GstRTSPAddress *result;
857 GSocketFamily family;
858 GstRTSPAddress **addrp;
860 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
861 g_return_val_if_fail (address != NULL, NULL);
862 g_return_val_if_fail (port > 0, NULL);
863 g_return_val_if_fail (n_ports > 0, NULL);
864 g_return_val_if_fail (ttl > 0, NULL);
868 addr = g_inet_address_new_from_string (address);
870 GST_ERROR ("failed to get inet addr from %s", address);
871 family = G_SOCKET_FAMILY_IPV4;
873 family = g_inet_address_get_family (addr);
874 g_object_unref (addr);
877 if (family == G_SOCKET_FAMILY_IPV6)
878 addrp = &priv->addr_v6;
880 addrp = &priv->addr_v4;
882 g_mutex_lock (&priv->lock);
883 if (*addrp == NULL) {
884 GstRTSPAddressPoolResult res;
886 if (priv->pool == NULL)
889 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
890 port, n_ports, ttl, addrp);
891 if (res != GST_RTSP_ADDRESS_POOL_OK)
894 if (strcmp ((*addrp)->address, address) ||
895 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
896 (*addrp)->ttl != ttl)
897 goto different_address;
899 result = gst_rtsp_address_copy (*addrp);
900 g_mutex_unlock (&priv->lock);
907 GST_ERROR_OBJECT (stream, "no address pool specified");
908 g_mutex_unlock (&priv->lock);
913 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
915 g_mutex_unlock (&priv->lock);
920 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
921 " reserved", address);
922 g_mutex_unlock (&priv->lock);
928 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
929 GSocketFamily family, GstElement * udpsrc_out[2],
930 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
931 GstRTSPAddress ** server_addr_out)
933 GstStateChangeReturn ret;
934 GstElement *udpsrc0, *udpsrc1;
935 GstElement *udpsink0, *udpsink1;
936 GSocket *rtp_socket = NULL;
937 GSocket *rtcp_socket;
938 gint tmp_rtp, tmp_rtcp;
940 gint rtpport, rtcpport;
941 GList *rejected_addresses = NULL;
942 GstRTSPAddress *addr = NULL;
943 GInetAddress *inetaddr = NULL;
944 GSocketAddress *rtp_sockaddr = NULL;
945 GSocketAddress *rtcp_sockaddr = NULL;
946 const gchar *multisink_socket;
948 if (family == G_SOCKET_FAMILY_IPV6)
949 multisink_socket = "socket-v6";
951 multisink_socket = "socket";
959 /* Start with random port */
962 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
963 G_SOCKET_PROTOCOL_UDP, NULL);
965 goto no_udp_protocol;
967 if (*server_addr_out)
968 gst_rtsp_address_free (*server_addr_out);
970 /* try to allocate 2 UDP ports, the RTP port should be an even
971 * number and the RTCP port should be the next (uneven) port */
974 if (rtp_socket == NULL) {
975 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
976 G_SOCKET_PROTOCOL_UDP, NULL);
978 goto no_udp_protocol;
981 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
982 GstRTSPAddressFlags flags;
985 rejected_addresses = g_list_prepend (rejected_addresses, addr);
987 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
988 if (family == G_SOCKET_FAMILY_IPV6)
989 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
991 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
993 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
998 tmp_rtp = addr->port;
1000 g_clear_object (&inetaddr);
1001 inetaddr = g_inet_address_new_from_string (addr->address);
1009 if (inetaddr == NULL)
1010 inetaddr = g_inet_address_new_any (family);
1013 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1014 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1015 g_object_unref (rtp_sockaddr);
1018 g_object_unref (rtp_sockaddr);
1020 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1021 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1022 g_clear_object (&rtp_sockaddr);
1027 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1028 g_object_unref (rtp_sockaddr);
1030 /* check if port is even */
1031 if ((tmp_rtp & 1) != 0) {
1032 /* port not even, close and allocate another */
1034 g_clear_object (&rtp_socket);
1039 tmp_rtcp = tmp_rtp + 1;
1041 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1042 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1043 g_object_unref (rtcp_sockaddr);
1044 g_clear_object (&rtp_socket);
1047 g_object_unref (rtcp_sockaddr);
1049 g_clear_object (&inetaddr);
1051 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1052 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1054 if (udpsrc0 == NULL || udpsrc1 == NULL)
1055 goto no_udp_protocol;
1057 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1058 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1060 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
1061 if (ret == GST_STATE_CHANGE_FAILURE)
1063 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
1064 if (ret == GST_STATE_CHANGE_FAILURE)
1067 /* all fine, do port check */
1068 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1069 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1071 /* this should not happen... */
1072 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1076 udpsink0 = udpsink_out[0];
1078 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1081 goto no_udp_protocol;
1083 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1084 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
1087 udpsink1 = udpsink_out[1];
1089 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1092 goto no_udp_protocol;
1094 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1095 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1096 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
1098 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1099 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
1100 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1101 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1102 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1103 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1104 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1105 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1107 /* we keep these elements, we will further configure them when the
1108 * client told us to really use the UDP ports. */
1109 udpsrc_out[0] = udpsrc0;
1110 udpsrc_out[1] = udpsrc1;
1111 udpsink_out[0] = udpsink0;
1112 udpsink_out[1] = udpsink1;
1113 server_port_out->min = rtpport;
1114 server_port_out->max = rtcpport;
1116 *server_addr_out = addr;
1117 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1119 g_object_unref (rtp_socket);
1120 g_object_unref (rtcp_socket);
1148 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1149 gst_object_unref (udpsrc0);
1152 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1153 gst_object_unref (udpsrc1);
1156 gst_element_set_state (udpsink0, GST_STATE_NULL);
1157 gst_object_unref (udpsink0);
1160 g_object_unref (inetaddr);
1161 g_list_free_full (rejected_addresses,
1162 (GDestroyNotify) gst_rtsp_address_free);
1164 gst_rtsp_address_free (addr);
1166 g_object_unref (rtp_socket);
1168 g_object_unref (rtcp_socket);
1173 /* must be called with lock */
1175 alloc_ports (GstRTSPStream * stream)
1177 GstRTSPStreamPrivate *priv = stream->priv;
1179 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1180 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1181 &priv->server_port_v4, &priv->server_addr_v4);
1183 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1184 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1185 &priv->server_port_v6, &priv->server_addr_v6);
1187 return priv->have_ipv4 || priv->have_ipv6;
1191 * gst_rtsp_stream_get_server_port:
1192 * @stream: a #GstRTSPStream
1193 * @server_port: (out): result server port
1194 * @family: the port family to get
1196 * Fill @server_port with the port pair used by the server. This function can
1197 * only be called when @stream has been joined.
1200 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1201 GstRTSPRange * server_port, GSocketFamily family)
1203 GstRTSPStreamPrivate *priv;
1205 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1206 priv = stream->priv;
1207 g_return_if_fail (priv->is_joined);
1209 g_mutex_lock (&priv->lock);
1210 if (family == G_SOCKET_FAMILY_IPV4) {
1212 *server_port = priv->server_port_v4;
1215 *server_port = priv->server_port_v6;
1217 g_mutex_unlock (&priv->lock);
1221 * gst_rtsp_stream_get_rtpsession:
1222 * @stream: a #GstRTSPStream
1224 * Get the RTP session of this stream.
1226 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1229 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1231 GstRTSPStreamPrivate *priv;
1234 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1236 priv = stream->priv;
1238 g_mutex_lock (&priv->lock);
1239 if ((session = priv->session))
1240 g_object_ref (session);
1241 g_mutex_unlock (&priv->lock);
1247 * gst_rtsp_stream_get_ssrc:
1248 * @stream: a #GstRTSPStream
1249 * @ssrc: (out): result ssrc
1251 * Get the SSRC used by the RTP session of this stream. This function can only
1252 * be called when @stream has been joined.
1255 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1257 GstRTSPStreamPrivate *priv;
1259 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1260 priv = stream->priv;
1261 g_return_if_fail (priv->is_joined);
1263 g_mutex_lock (&priv->lock);
1264 if (ssrc && priv->session)
1265 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1266 g_mutex_unlock (&priv->lock);
1269 /* executed from streaming thread */
1271 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1273 GstRTSPStreamPrivate *priv = stream->priv;
1274 GstCaps *newcaps, *oldcaps;
1276 newcaps = gst_pad_get_current_caps (pad);
1278 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1281 g_mutex_lock (&priv->lock);
1282 oldcaps = priv->caps;
1283 priv->caps = newcaps;
1284 g_mutex_unlock (&priv->lock);
1287 gst_caps_unref (oldcaps);
1291 dump_structure (const GstStructure * s)
1295 sstr = gst_structure_to_string (s);
1296 GST_INFO ("structure: %s", sstr);
1300 static GstRTSPStreamTransport *
1301 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1303 GstRTSPStreamPrivate *priv = stream->priv;
1305 GstRTSPStreamTransport *result = NULL;
1310 if (rtcp_from == NULL)
1313 tmp = g_strrstr (rtcp_from, ":");
1317 port = atoi (tmp + 1);
1318 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1320 g_mutex_lock (&priv->lock);
1321 GST_INFO ("finding %s:%d in %d transports", dest, port,
1322 g_list_length (priv->transports));
1324 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1325 GstRTSPStreamTransport *trans = walk->data;
1326 const GstRTSPTransport *tr;
1329 tr = gst_rtsp_stream_transport_get_transport (trans);
1331 min = tr->client_port.min;
1332 max = tr->client_port.max;
1334 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1340 g_object_ref (result);
1341 g_mutex_unlock (&priv->lock);
1348 static GstRTSPStreamTransport *
1349 check_transport (GObject * source, GstRTSPStream * stream)
1351 GstStructure *stats;
1352 GstRTSPStreamTransport *trans;
1354 /* see if we have a stream to match with the origin of the RTCP packet */
1355 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1356 if (trans == NULL) {
1357 g_object_get (source, "stats", &stats, NULL);
1359 const gchar *rtcp_from;
1361 dump_structure (stats);
1363 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1364 if ((trans = find_transport (stream, rtcp_from))) {
1365 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1367 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1370 gst_structure_free (stats);
1378 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1380 GstRTSPStreamTransport *trans;
1382 GST_INFO ("%p: new source %p", stream, source);
1384 trans = check_transport (source, stream);
1387 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1391 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1393 GST_INFO ("%p: new SDES %p", stream, source);
1397 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1399 GstRTSPStreamTransport *trans;
1401 trans = check_transport (source, stream);
1404 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1405 gst_rtsp_stream_transport_keep_alive (trans);
1409 GstStructure *stats;
1410 g_object_get (source, "stats", &stats, NULL);
1412 dump_structure (stats);
1413 gst_structure_free (stats);
1420 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1422 GST_INFO ("%p: source %p bye", stream, source);
1426 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1428 GstRTSPStreamTransport *trans;
1430 GST_INFO ("%p: source %p bye timeout", stream, source);
1432 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1433 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1434 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1439 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1441 GstRTSPStreamTransport *trans;
1443 GST_INFO ("%p: source %p timeout", stream, source);
1445 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1446 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1447 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1452 clear_tr_cache (GstRTSPStreamPrivate * priv)
1454 g_list_foreach (priv->tr_cache, (GFunc) g_object_unref, NULL);
1455 g_list_free (priv->tr_cache);
1456 priv->tr_cache = NULL;
1459 static GstFlowReturn
1460 handle_new_sample (GstAppSink * sink, gpointer user_data)
1462 GstRTSPStreamPrivate *priv;
1466 GstRTSPStream *stream;
1469 sample = gst_app_sink_pull_sample (sink);
1473 stream = (GstRTSPStream *) user_data;
1474 priv = stream->priv;
1475 buffer = gst_sample_get_buffer (sample);
1477 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1479 g_mutex_lock (&priv->lock);
1480 if (priv->tr_changed) {
1481 clear_tr_cache (priv);
1482 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1483 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1484 priv->tr_cache = g_list_prepend (priv->tr_cache, g_object_ref (tr));
1486 priv->tr_changed = FALSE;
1488 g_mutex_unlock (&priv->lock);
1490 for (walk = priv->tr_cache; walk; walk = g_list_next (walk)) {
1491 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1494 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1496 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1499 gst_sample_unref (sample);
1504 static GstAppSinkCallbacks sink_cb = {
1505 NULL, /* not interested in EOS */
1506 NULL, /* not interested in preroll samples */
1511 get_rtp_encoder (GstRTSPStream * stream, guint session)
1513 GstRTSPStreamPrivate *priv = stream->priv;
1515 if (priv->srtpenc == NULL) {
1518 name = g_strdup_printf ("srtpenc_%u", session);
1519 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1522 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1524 return gst_object_ref (priv->srtpenc);
1528 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1530 GstRTSPStreamPrivate *priv = stream->priv;
1535 if (priv->idx != session)
1538 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1540 enc = get_rtp_encoder (stream, session);
1541 name = g_strdup_printf ("rtp_sink_%d", session);
1542 pad = gst_element_get_request_pad (enc, name);
1544 gst_object_unref (pad);
1550 request_rtcp_encoder (GstElement * rtpbin, guint session,
1551 GstRTSPStream * stream)
1553 GstRTSPStreamPrivate *priv = stream->priv;
1558 if (priv->idx != session)
1561 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1563 enc = get_rtp_encoder (stream, session);
1564 name = g_strdup_printf ("rtcp_sink_%d", session);
1565 pad = gst_element_get_request_pad (enc, name);
1567 gst_object_unref (pad);
1573 request_rtcp_decoder (GstElement * rtpbin, guint session,
1574 GstRTSPStream * stream)
1576 GstRTSPStreamPrivate *priv = stream->priv;
1578 if (priv->idx != session)
1581 if (priv->srtpdec == NULL) {
1584 name = g_strdup_printf ("srtpdec_%u", session);
1585 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1588 return gst_object_ref (priv->srtpdec);
1592 * gst_rtsp_stream_join_bin:
1593 * @stream: a #GstRTSPStream
1594 * @bin: (transfer none): a #GstBin to join
1595 * @rtpbin: (transfer none): a rtpbin element in @bin
1596 * @state: the target state of the new elements
1598 * Join the #GstBin @bin that contains the element @rtpbin.
1600 * @stream will link to @rtpbin, which must be inside @bin. The elements
1601 * added to @bin will be set to the state given in @state.
1603 * Returns: %TRUE on success.
1606 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1607 GstElement * rtpbin, GstState state)
1609 GstRTSPStreamPrivate *priv;
1613 GstPad *pad, *sinkpad, *selpad;
1614 GstPadLinkReturn ret;
1616 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1617 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1618 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1620 priv = stream->priv;
1622 g_mutex_lock (&priv->lock);
1623 if (priv->is_joined)
1626 /* create a session with the same index as the stream */
1629 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1631 if (!alloc_ports (stream))
1634 /* update the dscp qos field in the sinks */
1635 update_dscp_qos (stream);
1637 if (priv->profiles & GST_RTSP_PROFILE_SAVP
1638 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
1640 g_signal_connect (rtpbin, "request-rtp-encoder",
1641 (GCallback) request_rtp_encoder, stream);
1642 g_signal_connect (rtpbin, "request-rtcp-encoder",
1643 (GCallback) request_rtcp_encoder, stream);
1645 g_signal_connect (rtpbin, "request-rtp-decoder",
1646 (GCallback) request_rtp_decoder, stream);
1648 g_signal_connect (rtpbin, "request-rtcp-decoder",
1649 (GCallback) request_rtcp_decoder, stream);
1652 /* get a pad for sending RTP */
1653 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1654 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1656 /* link the RTP pad to the session manager, it should not really fail unless
1657 * this is not really an RTP pad */
1658 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1659 if (ret != GST_PAD_LINK_OK)
1662 /* get pads from the RTP session element for sending and receiving
1664 name = g_strdup_printf ("send_rtp_src_%u", idx);
1665 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1667 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1668 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1670 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1671 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1673 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1674 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1677 /* get the session */
1678 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1680 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1682 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1684 g_signal_connect (priv->session, "on-ssrc-active",
1685 (GCallback) on_ssrc_active, stream);
1686 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1688 g_signal_connect (priv->session, "on-bye-timeout",
1689 (GCallback) on_bye_timeout, stream);
1690 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1693 for (i = 0; i < 2; i++) {
1694 GstPad *teepad, *queuepad;
1695 /* For the sender we create this bit of pipeline for both
1696 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1697 * we need to add a queue before appsink to make the pipeline
1698 * not block. For the TCP case, we want to pump data to the
1699 * client as fast as possible anyway.
1701 * .--------. .-----. .---------.
1702 * | rtpbin | | tee | | udpsink |
1703 * | send->sink src->sink |
1704 * '--------' | | '---------'
1705 * | | .---------. .---------.
1706 * | | | queue | | appsink |
1707 * | src->sink src->sink |
1708 * '-----' '---------' '---------'
1710 * When only UDP is allowed, we skip the tee, queue and appsink and link the
1711 * udpsink directly to the session.
1714 gst_bin_add (bin, priv->udpsink[i]);
1715 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1717 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1718 /* make tee for RTP/RTCP */
1719 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1720 gst_bin_add (bin, priv->tee[i]);
1722 /* and link to rtpbin send pad */
1723 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1724 gst_pad_link (priv->send_src[i], pad);
1725 gst_object_unref (pad);
1727 /* link tee to udpsink */
1728 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1729 gst_pad_link (teepad, sinkpad);
1730 gst_object_unref (teepad);
1733 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1734 gst_bin_add (bin, priv->appqueue[i]);
1735 /* and link to tee */
1736 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1737 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1738 gst_pad_link (teepad, pad);
1739 gst_object_unref (pad);
1740 gst_object_unref (teepad);
1743 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1744 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1745 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1746 gst_bin_add (bin, priv->appsink[i]);
1747 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1748 &sink_cb, stream, NULL);
1749 /* and link to queue */
1750 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1751 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1752 gst_pad_link (queuepad, pad);
1753 gst_object_unref (pad);
1754 gst_object_unref (queuepad);
1756 /* else only udpsink needed, link it to the session */
1757 gst_pad_link (priv->send_src[i], sinkpad);
1759 gst_object_unref (sinkpad);
1761 /* For the receiver we create this bit of pipeline for both
1762 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1763 * and it is all funneled into the rtpbin receive pad.
1765 * .--------. .--------. .--------.
1766 * | udpsrc | | funnel | | rtpbin |
1767 * | src->sink src->sink |
1768 * '--------' | | '--------'
1772 * '--------' '--------'
1774 /* make funnel for the RTP/RTCP receivers */
1775 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1776 gst_bin_add (bin, priv->funnel[i]);
1778 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1779 gst_pad_link (pad, priv->recv_sink[i]);
1780 gst_object_unref (pad);
1782 if (priv->udpsrc_v4[i]) {
1783 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1785 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1786 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1788 gst_bin_add (bin, priv->udpsrc_v4[i]);
1790 /* and link to the funnel v4 */
1791 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1792 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1793 gst_pad_link (pad, selpad);
1794 gst_object_unref (pad);
1795 gst_object_unref (selpad);
1798 if (priv->udpsrc_v6[i]) {
1799 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1800 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1801 gst_bin_add (bin, priv->udpsrc_v6[i]);
1803 /* and link to the funnel v6 */
1804 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1805 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1806 gst_pad_link (pad, selpad);
1807 gst_object_unref (pad);
1808 gst_object_unref (selpad);
1811 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1812 /* make and add appsrc */
1813 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1814 gst_bin_add (bin, priv->appsrc[i]);
1815 /* and link to the funnel */
1816 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1817 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1818 gst_pad_link (pad, selpad);
1819 gst_object_unref (pad);
1820 gst_object_unref (selpad);
1823 /* check if we need to set to a special state */
1824 if (state != GST_STATE_NULL) {
1825 if (priv->udpsink[i])
1826 gst_element_set_state (priv->udpsink[i], state);
1827 if (priv->appsink[i])
1828 gst_element_set_state (priv->appsink[i], state);
1829 if (priv->appqueue[i])
1830 gst_element_set_state (priv->appqueue[i], state);
1832 gst_element_set_state (priv->tee[i], state);
1833 if (priv->funnel[i])
1834 gst_element_set_state (priv->funnel[i], state);
1835 if (priv->appsrc[i])
1836 gst_element_set_state (priv->appsrc[i], state);
1840 /* be notified of caps changes */
1841 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
1842 (GCallback) caps_notify, stream);
1844 priv->is_joined = TRUE;
1845 g_mutex_unlock (&priv->lock);
1852 g_mutex_unlock (&priv->lock);
1857 g_mutex_unlock (&priv->lock);
1858 GST_WARNING ("failed to allocate ports %u", idx);
1863 GST_WARNING ("failed to link stream %u", idx);
1864 gst_object_unref (priv->send_rtp_sink);
1865 priv->send_rtp_sink = NULL;
1866 g_mutex_unlock (&priv->lock);
1872 * gst_rtsp_stream_leave_bin:
1873 * @stream: a #GstRTSPStream
1874 * @bin: (transfer none): a #GstBin
1875 * @rtpbin: (transfer none): a rtpbin #GstElement
1877 * Remove the elements of @stream from @bin.
1879 * Return: %TRUE on success.
1882 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1883 GstElement * rtpbin)
1885 GstRTSPStreamPrivate *priv;
1888 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1889 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1890 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1892 priv = stream->priv;
1894 g_mutex_lock (&priv->lock);
1895 if (!priv->is_joined)
1896 goto was_not_joined;
1898 /* all transports must be removed by now */
1899 g_return_val_if_fail (priv->transports == NULL, FALSE);
1901 clear_tr_cache (priv);
1903 GST_INFO ("stream %p leaving bin", stream);
1905 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1906 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
1907 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1908 gst_object_unref (priv->send_rtp_sink);
1909 priv->send_rtp_sink = NULL;
1911 for (i = 0; i < 2; i++) {
1912 if (priv->udpsink[i])
1913 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1914 if (priv->appsink[i])
1915 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1916 if (priv->appqueue[i])
1917 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1919 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1920 if (priv->funnel[i])
1921 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1922 if (priv->appsrc[i])
1923 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1924 if (priv->udpsrc_v4[i]) {
1925 /* and set udpsrc to NULL now before removing */
1926 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1927 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1928 /* removing them should also nicely release the request
1929 * pads when they finalize */
1930 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1932 if (priv->udpsrc_v6[i]) {
1933 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1934 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1935 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1937 if (priv->udpsink[i])
1938 gst_bin_remove (bin, priv->udpsink[i]);
1939 if (priv->appsrc[i])
1940 gst_bin_remove (bin, priv->appsrc[i]);
1941 if (priv->appsink[i])
1942 gst_bin_remove (bin, priv->appsink[i]);
1943 if (priv->appqueue[i])
1944 gst_bin_remove (bin, priv->appqueue[i]);
1946 gst_bin_remove (bin, priv->tee[i]);
1947 if (priv->funnel[i])
1948 gst_bin_remove (bin, priv->funnel[i]);
1950 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1951 gst_object_unref (priv->recv_sink[i]);
1952 priv->recv_sink[i] = NULL;
1954 priv->udpsrc_v4[i] = NULL;
1955 priv->udpsrc_v6[i] = NULL;
1956 priv->udpsink[i] = NULL;
1957 priv->appsrc[i] = NULL;
1958 priv->appsink[i] = NULL;
1959 priv->appqueue[i] = NULL;
1960 priv->tee[i] = NULL;
1961 priv->funnel[i] = NULL;
1963 gst_object_unref (priv->send_src[0]);
1964 priv->send_src[0] = NULL;
1966 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1967 gst_object_unref (priv->send_src[1]);
1968 priv->send_src[1] = NULL;
1970 g_object_unref (priv->session);
1971 priv->session = NULL;
1973 gst_caps_unref (priv->caps);
1977 gst_object_unref (priv->srtpenc);
1979 priv->is_joined = FALSE;
1980 g_mutex_unlock (&priv->lock);
1986 g_mutex_unlock (&priv->lock);
1992 * gst_rtsp_stream_get_rtpinfo:
1993 * @stream: a #GstRTSPStream
1994 * @rtptime: (allow-none): result RTP timestamp
1995 * @seq: (allow-none): result RTP seqnum
1996 * @clock_rate: (allow-none): the clock rate
1997 * @running_time: (allow-none): result running-time
1999 * Retrieve the current rtptime, seq and running-time. This is used to
2000 * construct a RTPInfo reply header.
2002 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2005 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2006 guint * rtptime, guint * seq, guint * clock_rate,
2007 GstClockTime * running_time)
2009 GstRTSPStreamPrivate *priv;
2010 GstStructure *stats;
2011 GObjectClass *payobjclass;
2013 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2015 priv = stream->priv;
2017 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2019 g_mutex_lock (&priv->lock);
2021 if (g_object_class_find_property (payobjclass, "stats")) {
2022 g_object_get (priv->payloader, "stats", &stats, NULL);
2027 gst_structure_get_uint (stats, "seqnum", seq);
2030 gst_structure_get_uint (stats, "timestamp", rtptime);
2033 gst_structure_get_clock_time (stats, "running-time", running_time);
2036 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2037 if (*clock_rate == 0 && running_time)
2038 *running_time = GST_CLOCK_TIME_NONE;
2040 gst_structure_free (stats);
2042 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2043 !g_object_class_find_property (payobjclass, "timestamp"))
2047 g_object_get (priv->payloader, "seqnum", seq, NULL);
2050 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2053 *running_time = GST_CLOCK_TIME_NONE;
2055 g_mutex_unlock (&priv->lock);
2062 GST_WARNING ("Could not get payloader stats");
2063 g_mutex_unlock (&priv->lock);
2069 * gst_rtsp_stream_get_caps:
2070 * @stream: a #GstRTSPStream
2072 * Retrieve the current caps of @stream.
2074 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2078 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2080 GstRTSPStreamPrivate *priv;
2083 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2085 priv = stream->priv;
2087 g_mutex_lock (&priv->lock);
2088 if ((result = priv->caps))
2089 gst_caps_ref (result);
2090 g_mutex_unlock (&priv->lock);
2096 * gst_rtsp_stream_recv_rtp:
2097 * @stream: a #GstRTSPStream
2098 * @buffer: (transfer full): a #GstBuffer
2100 * Handle an RTP buffer for the stream. This method is usually called when a
2101 * message has been received from a client using the TCP transport.
2103 * This function takes ownership of @buffer.
2105 * Returns: a GstFlowReturn.
2108 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2110 GstRTSPStreamPrivate *priv;
2112 GstElement *element;
2114 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2115 priv = stream->priv;
2116 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2117 g_return_val_if_fail (priv->is_joined, FALSE);
2119 g_mutex_lock (&priv->lock);
2120 if (priv->appsrc[0])
2121 element = gst_object_ref (priv->appsrc[0]);
2124 g_mutex_unlock (&priv->lock);
2127 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2128 gst_object_unref (element);
2136 * gst_rtsp_stream_recv_rtcp:
2137 * @stream: a #GstRTSPStream
2138 * @buffer: (transfer full): a #GstBuffer
2140 * Handle an RTCP buffer for the stream. This method is usually called when a
2141 * message has been received from a client using the TCP transport.
2143 * This function takes ownership of @buffer.
2145 * Returns: a GstFlowReturn.
2148 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2150 GstRTSPStreamPrivate *priv;
2152 GstElement *element;
2154 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2155 priv = stream->priv;
2156 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2157 g_return_val_if_fail (priv->is_joined, FALSE);
2159 g_mutex_lock (&priv->lock);
2160 if (priv->appsrc[1])
2161 element = gst_object_ref (priv->appsrc[1]);
2164 g_mutex_unlock (&priv->lock);
2167 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2168 gst_object_unref (element);
2171 gst_buffer_unref (buffer);
2176 /* must be called with lock */
2178 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
2181 GstRTSPStreamPrivate *priv = stream->priv;
2182 const GstRTSPTransport *tr;
2184 tr = gst_rtsp_stream_transport_get_transport (trans);
2186 switch (tr->lower_transport) {
2187 case GST_RTSP_LOWER_TRANS_UDP:
2188 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2194 dest = tr->destination;
2195 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2200 min = tr->client_port.min;
2201 max = tr->client_port.max;
2206 GST_INFO ("setting ttl-mc %d", ttl);
2207 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
2208 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
2210 GST_INFO ("adding %s:%d-%d", dest, min, max);
2211 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
2212 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
2213 priv->transports = g_list_prepend (priv->transports, trans);
2215 GST_INFO ("removing %s:%d-%d", dest, min, max);
2216 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
2217 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
2218 priv->transports = g_list_remove (priv->transports, trans);
2220 priv->tr_changed = TRUE;
2223 case GST_RTSP_LOWER_TRANS_TCP:
2225 GST_INFO ("adding TCP %s", tr->destination);
2226 priv->transports = g_list_prepend (priv->transports, trans);
2228 GST_INFO ("removing TCP %s", tr->destination);
2229 priv->transports = g_list_remove (priv->transports, trans);
2231 priv->tr_changed = TRUE;
2234 goto unknown_transport;
2241 GST_INFO ("Unknown transport %d", tr->lower_transport);
2248 * gst_rtsp_stream_add_transport:
2249 * @stream: a #GstRTSPStream
2250 * @trans: (transfer none): a #GstRTSPStreamTransport
2252 * Add the transport in @trans to @stream. The media of @stream will
2253 * then also be send to the values configured in @trans.
2255 * @stream must be joined to a bin.
2257 * @trans must contain a valid #GstRTSPTransport.
2259 * Returns: %TRUE if @trans was added
2262 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
2263 GstRTSPStreamTransport * trans)
2265 GstRTSPStreamPrivate *priv;
2268 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2269 priv = stream->priv;
2270 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2271 g_return_val_if_fail (priv->is_joined, FALSE);
2273 g_mutex_lock (&priv->lock);
2274 res = update_transport (stream, trans, TRUE);
2275 g_mutex_unlock (&priv->lock);
2281 * gst_rtsp_stream_remove_transport:
2282 * @stream: a #GstRTSPStream
2283 * @trans: (transfer none): a #GstRTSPStreamTransport
2285 * Remove the transport in @trans from @stream. The media of @stream will
2286 * not be sent to the values configured in @trans.
2288 * @stream must be joined to a bin.
2290 * @trans must contain a valid #GstRTSPTransport.
2292 * Returns: %TRUE if @trans was removed
2295 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2296 GstRTSPStreamTransport * trans)
2298 GstRTSPStreamPrivate *priv;
2301 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2302 priv = stream->priv;
2303 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2304 g_return_val_if_fail (priv->is_joined, FALSE);
2306 g_mutex_lock (&priv->lock);
2307 res = update_transport (stream, trans, FALSE);
2308 g_mutex_unlock (&priv->lock);
2314 * gst_rtsp_stream_get_rtp_socket:
2315 * @stream: a #GstRTSPStream
2316 * @family: the socket family
2318 * Get the RTP socket from @stream for a @family.
2320 * @stream must be joined to a bin.
2322 * Returns: (transfer full): the RTP socket or %NULL if no socket could be
2323 * allocated for @family. Unref after usage
2326 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
2328 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2332 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2333 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2334 family == G_SOCKET_FAMILY_IPV6, NULL);
2335 g_return_val_if_fail (priv->udpsink[0], NULL);
2337 if (family == G_SOCKET_FAMILY_IPV6)
2342 g_object_get (priv->udpsink[0], name, &socket, NULL);
2348 * gst_rtsp_stream_get_rtcp_socket:
2349 * @stream: a #GstRTSPStream
2350 * @family: the socket family
2352 * Get the RTCP socket from @stream for a @family.
2354 * @stream must be joined to a bin.
2356 * Returns: (transfer full): the RTCP socket or %NULL if no socket could be
2357 * allocated for @family. Unref after usage
2360 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
2362 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2366 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2367 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2368 family == G_SOCKET_FAMILY_IPV6, NULL);
2369 g_return_val_if_fail (priv->udpsink[1], NULL);
2371 if (family == G_SOCKET_FAMILY_IPV6)
2376 g_object_get (priv->udpsink[1], name, &socket, NULL);
2382 * gst_rtsp_stream_transport_filter:
2383 * @stream: a #GstRTSPStream
2384 * @func: (scope call) (allow-none): a callback
2385 * @user_data: (closure): user data passed to @func
2387 * Call @func for each transport managed by @stream. The result value of @func
2388 * determines what happens to the transport. @func will be called with @stream
2389 * locked so no further actions on @stream can be performed from @func.
2391 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
2394 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
2396 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
2397 * will also be added with an additional ref to the result #GList of this
2400 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
2402 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
2403 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
2404 * element in the #GList should be unreffed before the list is freed.
2407 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
2408 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
2410 GstRTSPStreamPrivate *priv;
2411 GList *result, *walk, *next;
2413 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2415 priv = stream->priv;
2419 g_mutex_lock (&priv->lock);
2420 for (walk = priv->transports; walk; walk = next) {
2421 GstRTSPStreamTransport *trans = walk->data;
2422 GstRTSPFilterResult res;
2424 next = g_list_next (walk);
2427 res = func (stream, trans, user_data);
2429 res = GST_RTSP_FILTER_REF;
2432 case GST_RTSP_FILTER_REMOVE:
2433 update_transport (stream, trans, FALSE);
2435 case GST_RTSP_FILTER_REF:
2436 result = g_list_prepend (result, g_object_ref (trans));
2438 case GST_RTSP_FILTER_KEEP:
2443 g_mutex_unlock (&priv->lock);
2448 static GstPadProbeReturn
2449 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2451 GstRTSPStreamPrivate *priv;
2452 GstRTSPStream *stream;
2455 priv = stream->priv;
2457 GST_DEBUG_OBJECT (pad, "now blocking");
2459 g_mutex_lock (&priv->lock);
2460 priv->blocking = TRUE;
2461 g_mutex_unlock (&priv->lock);
2463 gst_element_post_message (priv->payloader,
2464 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
2465 gst_structure_new_empty ("GstRTSPStreamBlocking")));
2467 return GST_PAD_PROBE_OK;
2471 * gst_rtsp_stream_set_blocked:
2472 * @stream: a #GstRTSPStream
2473 * @blocked: boolean indicating we should block or unblock
2475 * Blocks or unblocks the dataflow on @stream.
2477 * Returns: %TRUE on success
2480 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
2482 GstRTSPStreamPrivate *priv;
2484 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2486 priv = stream->priv;
2488 g_mutex_lock (&priv->lock);
2490 priv->blocking = FALSE;
2491 if (priv->blocked_id == 0) {
2492 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
2493 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
2494 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
2495 g_object_ref (stream), g_object_unref);
2498 if (priv->blocked_id != 0) {
2499 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
2500 priv->blocked_id = 0;
2501 priv->blocking = FALSE;
2504 g_mutex_unlock (&priv->lock);
2510 * gst_rtsp_stream_is_blocking:
2511 * @stream: a #GstRTSPStream
2513 * Check if @stream is blocking on a #GstBuffer.
2515 * Returns: %TRUE if @stream is blocking
2518 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
2520 GstRTSPStreamPrivate *priv;
2523 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2525 priv = stream->priv;
2527 g_mutex_lock (&priv->lock);
2528 result = priv->blocking;
2529 g_mutex_unlock (&priv->lock);