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 */
87 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
89 GstElement *udpsrc_v4[2];
91 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
93 GstElement *udpsrc_v6[2];
95 GstElement *udpsink[2];
97 /* for TCP transport */
98 GstElement *appsrc[2];
99 GstElement *appqueue[2];
100 GstElement *appsink[2];
103 GstElement *funnel[2];
105 /* server ports for sending/receiving over ipv4 */
106 GstRTSPRange server_port_v4;
107 GstRTSPAddress *server_addr_v4;
110 /* server ports for sending/receiving over ipv6 */
111 GstRTSPRange server_port_v6;
112 GstRTSPAddress *server_addr_v6;
115 /* multicast addresses */
116 GstRTSPAddressPool *pool;
117 GstRTSPAddress *addr_v4;
118 GstRTSPAddress *addr_v6;
120 /* the caps of the stream */
124 /* transports we stream to */
132 /* stream blocking */
137 #define DEFAULT_CONTROL NULL
138 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
139 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
140 GST_RTSP_LOWER_TRANS_TCP
153 SIGNAL_NEW_RTP_ENCODER,
154 SIGNAL_NEW_RTCP_ENCODER,
158 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
159 #define GST_CAT_DEFAULT rtsp_stream_debug
161 static GQuark ssrc_stream_map_key;
163 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
164 GValue * value, GParamSpec * pspec);
165 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
166 const GValue * value, GParamSpec * pspec);
168 static void gst_rtsp_stream_finalize (GObject * obj);
170 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
172 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
175 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
177 GObjectClass *gobject_class;
179 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
181 gobject_class = G_OBJECT_CLASS (klass);
183 gobject_class->get_property = gst_rtsp_stream_get_property;
184 gobject_class->set_property = gst_rtsp_stream_set_property;
185 gobject_class->finalize = gst_rtsp_stream_finalize;
187 g_object_class_install_property (gobject_class, PROP_CONTROL,
188 g_param_spec_string ("control", "Control",
189 "The control string for this stream", DEFAULT_CONTROL,
190 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
192 g_object_class_install_property (gobject_class, PROP_PROFILES,
193 g_param_spec_flags ("profiles", "Profiles",
194 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
195 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
197 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
198 g_param_spec_flags ("protocols", "Protocols",
199 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
200 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
202 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
203 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
204 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
205 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
207 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
208 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
209 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
210 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
212 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
214 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
218 gst_rtsp_stream_init (GstRTSPStream * stream)
220 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
222 GST_DEBUG ("new stream %p", stream);
227 priv->control = g_strdup (DEFAULT_CONTROL);
228 priv->profiles = DEFAULT_PROFILES;
229 priv->protocols = DEFAULT_PROTOCOLS;
231 g_mutex_init (&priv->lock);
233 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
234 NULL, (GDestroyNotify) gst_caps_unref);
238 gst_rtsp_stream_finalize (GObject * obj)
240 GstRTSPStream *stream;
241 GstRTSPStreamPrivate *priv;
243 stream = GST_RTSP_STREAM (obj);
246 GST_DEBUG ("finalize stream %p", stream);
248 /* we really need to be unjoined now */
249 g_return_if_fail (!priv->is_joined);
252 gst_rtsp_address_free (priv->addr_v4);
254 gst_rtsp_address_free (priv->addr_v6);
255 if (priv->server_addr_v4)
256 gst_rtsp_address_free (priv->server_addr_v4);
257 if (priv->server_addr_v6)
258 gst_rtsp_address_free (priv->server_addr_v6);
260 g_object_unref (priv->pool);
261 gst_object_unref (priv->payloader);
262 gst_object_unref (priv->srcpad);
263 g_free (priv->control);
264 g_mutex_clear (&priv->lock);
266 g_hash_table_unref (priv->keys);
268 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
272 gst_rtsp_stream_get_property (GObject * object, guint propid,
273 GValue * value, GParamSpec * pspec)
275 GstRTSPStream *stream = GST_RTSP_STREAM (object);
279 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
282 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
285 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
288 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
293 gst_rtsp_stream_set_property (GObject * object, guint propid,
294 const GValue * value, GParamSpec * pspec)
296 GstRTSPStream *stream = GST_RTSP_STREAM (object);
300 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
303 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
306 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
309 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
314 * gst_rtsp_stream_new:
317 * @payloader: a #GstElement
319 * Create a new media stream with index @idx that handles RTP data on
320 * @srcpad and has a payloader element @payloader.
322 * Returns: (transfer full): a new #GstRTSPStream
325 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
327 GstRTSPStreamPrivate *priv;
328 GstRTSPStream *stream;
330 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
331 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
332 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
334 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
337 priv->payloader = gst_object_ref (payloader);
338 priv->srcpad = gst_object_ref (srcpad);
344 * gst_rtsp_stream_get_index:
345 * @stream: a #GstRTSPStream
347 * Get the stream index.
349 * Return: the stream index.
352 gst_rtsp_stream_get_index (GstRTSPStream * stream)
354 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
356 return stream->priv->idx;
360 * gst_rtsp_stream_get_pt:
361 * @stream: a #GstRTSPStream
363 * Get the stream payload type.
365 * Return: the stream payload type.
368 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
370 GstRTSPStreamPrivate *priv;
373 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
377 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
383 * gst_rtsp_stream_get_srcpad:
384 * @stream: a #GstRTSPStream
386 * Get the srcpad associated with @stream.
388 * Returns: (transfer full): the srcpad. Unref after usage.
391 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
393 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
395 return gst_object_ref (stream->priv->srcpad);
399 * gst_rtsp_stream_get_control:
400 * @stream: a #GstRTSPStream
402 * Get the control string to identify this stream.
404 * Returns: (transfer full): the control string. g_free() after usage.
407 gst_rtsp_stream_get_control (GstRTSPStream * stream)
409 GstRTSPStreamPrivate *priv;
412 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
416 g_mutex_lock (&priv->lock);
417 if ((result = g_strdup (priv->control)) == NULL)
418 result = g_strdup_printf ("stream=%u", priv->idx);
419 g_mutex_unlock (&priv->lock);
425 * gst_rtsp_stream_set_control:
426 * @stream: a #GstRTSPStream
427 * @control: a control string
429 * Set the control string in @stream.
432 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
434 GstRTSPStreamPrivate *priv;
436 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
440 g_mutex_lock (&priv->lock);
441 g_free (priv->control);
442 priv->control = g_strdup (control);
443 g_mutex_unlock (&priv->lock);
447 * gst_rtsp_stream_has_control:
448 * @stream: a #GstRTSPStream
449 * @control: a control string
451 * Check if @stream has the control string @control.
453 * Returns: %TRUE is @stream has @control as the control string
456 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
458 GstRTSPStreamPrivate *priv;
461 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
465 g_mutex_lock (&priv->lock);
467 res = (g_strcmp0 (priv->control, control) == 0);
471 if (sscanf (control, "stream=%u", &streamid) > 0)
472 res = (streamid == priv->idx);
476 g_mutex_unlock (&priv->lock);
482 * gst_rtsp_stream_set_mtu:
483 * @stream: a #GstRTSPStream
486 * Configure the mtu in the payloader of @stream to @mtu.
489 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
491 GstRTSPStreamPrivate *priv;
493 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
497 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
499 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
503 * gst_rtsp_stream_get_mtu:
504 * @stream: a #GstRTSPStream
506 * Get the configured MTU in the payloader of @stream.
508 * Returns: the MTU of the payloader.
511 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
513 GstRTSPStreamPrivate *priv;
516 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
520 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
525 /* Update the dscp qos property on the udp sinks */
527 update_dscp_qos (GstRTSPStream * stream)
529 GstRTSPStreamPrivate *priv;
531 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
535 if (priv->udpsink[0]) {
536 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
540 if (priv->udpsink[1]) {
541 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
547 * gst_rtsp_stream_set_dscp_qos:
548 * @stream: a #GstRTSPStream
549 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
551 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
554 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
556 GstRTSPStreamPrivate *priv;
558 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
562 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
564 if (dscp_qos < -1 || dscp_qos > 63) {
565 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
569 priv->dscp_qos = dscp_qos;
571 update_dscp_qos (stream);
575 * gst_rtsp_stream_get_dscp_qos:
576 * @stream: a #GstRTSPStream
578 * Get the configured DSCP QoS in of the outgoing sockets.
580 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
583 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
585 GstRTSPStreamPrivate *priv;
587 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
591 return priv->dscp_qos;
595 * gst_rtsp_stream_is_transport_supported:
596 * @stream: a #GstRTSPStream
597 * @transport: (transfer none): a #GstRTSPTransport
599 * Check if @transport can be handled by stream
601 * Returns: %TRUE if @transport can be handled by @stream.
604 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
605 GstRTSPTransport * transport)
607 GstRTSPStreamPrivate *priv;
609 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
613 g_mutex_lock (&priv->lock);
614 if (transport->trans != GST_RTSP_TRANS_RTP)
615 goto unsupported_transmode;
617 if (!(transport->profile & priv->profiles))
618 goto unsupported_profile;
620 if (!(transport->lower_transport & priv->protocols))
621 goto unsupported_ltrans;
623 g_mutex_unlock (&priv->lock);
628 unsupported_transmode:
630 GST_DEBUG ("unsupported transport mode %d", transport->trans);
631 g_mutex_unlock (&priv->lock);
636 GST_DEBUG ("unsupported profile %d", transport->profile);
637 g_mutex_unlock (&priv->lock);
642 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
643 g_mutex_unlock (&priv->lock);
649 * gst_rtsp_stream_set_profiles:
650 * @stream: a #GstRTSPStream
651 * @profiles: the new profiles
653 * Configure the allowed profiles for @stream.
656 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
658 GstRTSPStreamPrivate *priv;
660 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
664 g_mutex_lock (&priv->lock);
665 priv->profiles = profiles;
666 g_mutex_unlock (&priv->lock);
670 * gst_rtsp_stream_get_profiles:
671 * @stream: a #GstRTSPStream
673 * Get the allowed profiles of @stream.
675 * Returns: a #GstRTSPProfile
678 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
680 GstRTSPStreamPrivate *priv;
683 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
687 g_mutex_lock (&priv->lock);
688 res = priv->profiles;
689 g_mutex_unlock (&priv->lock);
695 * gst_rtsp_stream_set_protocols:
696 * @stream: a #GstRTSPStream
697 * @protocols: the new flags
699 * Configure the allowed lower transport for @stream.
702 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
703 GstRTSPLowerTrans protocols)
705 GstRTSPStreamPrivate *priv;
707 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
711 g_mutex_lock (&priv->lock);
712 priv->protocols = protocols;
713 g_mutex_unlock (&priv->lock);
717 * gst_rtsp_stream_get_protocols:
718 * @stream: a #GstRTSPStream
720 * Get the allowed protocols of @stream.
722 * Returns: a #GstRTSPLowerTrans
725 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
727 GstRTSPStreamPrivate *priv;
728 GstRTSPLowerTrans res;
730 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
731 GST_RTSP_LOWER_TRANS_UNKNOWN);
735 g_mutex_lock (&priv->lock);
736 res = priv->protocols;
737 g_mutex_unlock (&priv->lock);
743 * gst_rtsp_stream_set_address_pool:
744 * @stream: a #GstRTSPStream
745 * @pool: (transfer none): a #GstRTSPAddressPool
747 * configure @pool to be used as the address pool of @stream.
750 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
751 GstRTSPAddressPool * pool)
753 GstRTSPStreamPrivate *priv;
754 GstRTSPAddressPool *old;
756 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
760 GST_LOG_OBJECT (stream, "set address pool %p", pool);
762 g_mutex_lock (&priv->lock);
763 if ((old = priv->pool) != pool)
764 priv->pool = pool ? g_object_ref (pool) : NULL;
767 g_mutex_unlock (&priv->lock);
770 g_object_unref (old);
774 * gst_rtsp_stream_get_address_pool:
775 * @stream: a #GstRTSPStream
777 * Get the #GstRTSPAddressPool used as the address pool of @stream.
779 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
783 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
785 GstRTSPStreamPrivate *priv;
786 GstRTSPAddressPool *result;
788 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
792 g_mutex_lock (&priv->lock);
793 if ((result = priv->pool))
794 g_object_ref (result);
795 g_mutex_unlock (&priv->lock);
801 * gst_rtsp_stream_get_multicast_address:
802 * @stream: a #GstRTSPStream
803 * @family: the #GSocketFamily
805 * Get the multicast address of @stream for @family.
807 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
808 * or %NULL when no address could be allocated. gst_rtsp_address_free()
812 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
813 GSocketFamily family)
815 GstRTSPStreamPrivate *priv;
816 GstRTSPAddress *result;
817 GstRTSPAddress **addrp;
818 GstRTSPAddressFlags flags;
820 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
824 if (family == G_SOCKET_FAMILY_IPV6) {
825 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
826 addrp = &priv->addr_v6;
828 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
829 addrp = &priv->addr_v4;
832 g_mutex_lock (&priv->lock);
833 if (*addrp == NULL) {
834 if (priv->pool == NULL)
837 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
839 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
843 result = gst_rtsp_address_copy (*addrp);
844 g_mutex_unlock (&priv->lock);
851 GST_ERROR_OBJECT (stream, "no address pool specified");
852 g_mutex_unlock (&priv->lock);
857 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
858 g_mutex_unlock (&priv->lock);
864 * gst_rtsp_stream_reserve_address:
865 * @stream: a #GstRTSPStream
866 * @address: an address
871 * Reserve @address and @port as the address and port of @stream.
873 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
874 * the address could be reserved. gst_rtsp_address_free() after usage.
877 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
878 const gchar * address, guint port, guint n_ports, guint ttl)
880 GstRTSPStreamPrivate *priv;
881 GstRTSPAddress *result;
883 GSocketFamily family;
884 GstRTSPAddress **addrp;
886 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
887 g_return_val_if_fail (address != NULL, NULL);
888 g_return_val_if_fail (port > 0, NULL);
889 g_return_val_if_fail (n_ports > 0, NULL);
890 g_return_val_if_fail (ttl > 0, NULL);
894 addr = g_inet_address_new_from_string (address);
896 GST_ERROR ("failed to get inet addr from %s", address);
897 family = G_SOCKET_FAMILY_IPV4;
899 family = g_inet_address_get_family (addr);
900 g_object_unref (addr);
903 if (family == G_SOCKET_FAMILY_IPV6)
904 addrp = &priv->addr_v6;
906 addrp = &priv->addr_v4;
908 g_mutex_lock (&priv->lock);
909 if (*addrp == NULL) {
910 GstRTSPAddressPoolResult res;
912 if (priv->pool == NULL)
915 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
916 port, n_ports, ttl, addrp);
917 if (res != GST_RTSP_ADDRESS_POOL_OK)
920 if (strcmp ((*addrp)->address, address) ||
921 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
922 (*addrp)->ttl != ttl)
923 goto different_address;
925 result = gst_rtsp_address_copy (*addrp);
926 g_mutex_unlock (&priv->lock);
933 GST_ERROR_OBJECT (stream, "no address pool specified");
934 g_mutex_unlock (&priv->lock);
939 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
941 g_mutex_unlock (&priv->lock);
946 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
947 " reserved", address);
948 g_mutex_unlock (&priv->lock);
954 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
955 GSocketFamily family, GstElement * udpsrc_out[2],
956 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
957 GstRTSPAddress ** server_addr_out)
959 GstStateChangeReturn ret;
960 GstElement *udpsrc0, *udpsrc1;
961 GstElement *udpsink0, *udpsink1;
962 GSocket *rtp_socket = NULL;
963 GSocket *rtcp_socket;
964 gint tmp_rtp, tmp_rtcp;
966 gint rtpport, rtcpport;
967 GList *rejected_addresses = NULL;
968 GstRTSPAddress *addr = NULL;
969 GInetAddress *inetaddr = NULL;
970 GSocketAddress *rtp_sockaddr = NULL;
971 GSocketAddress *rtcp_sockaddr = NULL;
972 const gchar *multisink_socket;
974 if (family == G_SOCKET_FAMILY_IPV6)
975 multisink_socket = "socket-v6";
977 multisink_socket = "socket";
985 /* Start with random port */
988 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
989 G_SOCKET_PROTOCOL_UDP, NULL);
991 goto no_udp_protocol;
993 if (*server_addr_out)
994 gst_rtsp_address_free (*server_addr_out);
996 /* try to allocate 2 UDP ports, the RTP port should be an even
997 * number and the RTCP port should be the next (uneven) port */
1000 if (rtp_socket == NULL) {
1001 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1002 G_SOCKET_PROTOCOL_UDP, NULL);
1004 goto no_udp_protocol;
1007 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
1008 GstRTSPAddressFlags flags;
1011 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1013 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1014 if (family == G_SOCKET_FAMILY_IPV6)
1015 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1017 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1019 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1024 tmp_rtp = addr->port;
1026 g_clear_object (&inetaddr);
1027 inetaddr = g_inet_address_new_from_string (addr->address);
1035 if (inetaddr == NULL)
1036 inetaddr = g_inet_address_new_any (family);
1039 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1040 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1041 g_object_unref (rtp_sockaddr);
1044 g_object_unref (rtp_sockaddr);
1046 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1047 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1048 g_clear_object (&rtp_sockaddr);
1053 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1054 g_object_unref (rtp_sockaddr);
1056 /* check if port is even */
1057 if ((tmp_rtp & 1) != 0) {
1058 /* port not even, close and allocate another */
1060 g_clear_object (&rtp_socket);
1065 tmp_rtcp = tmp_rtp + 1;
1067 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1068 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1069 g_object_unref (rtcp_sockaddr);
1070 g_clear_object (&rtp_socket);
1073 g_object_unref (rtcp_sockaddr);
1075 g_clear_object (&inetaddr);
1077 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1078 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1080 if (udpsrc0 == NULL || udpsrc1 == NULL)
1081 goto no_udp_protocol;
1083 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1084 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1086 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1087 if (ret == GST_STATE_CHANGE_FAILURE)
1089 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1090 if (ret == GST_STATE_CHANGE_FAILURE)
1093 /* all fine, do port check */
1094 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1095 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1097 /* this should not happen... */
1098 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1102 udpsink0 = udpsink_out[0];
1104 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1107 goto no_udp_protocol;
1109 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1110 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
1113 udpsink1 = udpsink_out[1];
1115 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1118 goto no_udp_protocol;
1120 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1121 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1122 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
1124 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1125 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
1126 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1127 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1128 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1129 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1130 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1131 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1133 /* we keep these elements, we will further configure them when the
1134 * client told us to really use the UDP ports. */
1135 udpsrc_out[0] = udpsrc0;
1136 udpsrc_out[1] = udpsrc1;
1137 udpsink_out[0] = udpsink0;
1138 udpsink_out[1] = udpsink1;
1139 server_port_out->min = rtpport;
1140 server_port_out->max = rtcpport;
1142 *server_addr_out = addr;
1143 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1145 g_object_unref (rtp_socket);
1146 g_object_unref (rtcp_socket);
1174 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1175 gst_object_unref (udpsrc0);
1178 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1179 gst_object_unref (udpsrc1);
1182 gst_element_set_state (udpsink0, GST_STATE_NULL);
1183 gst_object_unref (udpsink0);
1186 g_object_unref (inetaddr);
1187 g_list_free_full (rejected_addresses,
1188 (GDestroyNotify) gst_rtsp_address_free);
1190 gst_rtsp_address_free (addr);
1192 g_object_unref (rtp_socket);
1194 g_object_unref (rtcp_socket);
1199 /* must be called with lock */
1201 alloc_ports (GstRTSPStream * stream)
1203 GstRTSPStreamPrivate *priv = stream->priv;
1205 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1206 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1207 &priv->server_port_v4, &priv->server_addr_v4);
1209 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1210 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1211 &priv->server_port_v6, &priv->server_addr_v6);
1213 return priv->have_ipv4 || priv->have_ipv6;
1217 * gst_rtsp_stream_get_server_port:
1218 * @stream: a #GstRTSPStream
1219 * @server_port: (out): result server port
1220 * @family: the port family to get
1222 * Fill @server_port with the port pair used by the server. This function can
1223 * only be called when @stream has been joined.
1226 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1227 GstRTSPRange * server_port, GSocketFamily family)
1229 GstRTSPStreamPrivate *priv;
1231 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1232 priv = stream->priv;
1233 g_return_if_fail (priv->is_joined);
1235 g_mutex_lock (&priv->lock);
1236 if (family == G_SOCKET_FAMILY_IPV4) {
1238 *server_port = priv->server_port_v4;
1241 *server_port = priv->server_port_v6;
1243 g_mutex_unlock (&priv->lock);
1247 * gst_rtsp_stream_get_rtpsession:
1248 * @stream: a #GstRTSPStream
1250 * Get the RTP session of this stream.
1252 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1255 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1257 GstRTSPStreamPrivate *priv;
1260 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1262 priv = stream->priv;
1264 g_mutex_lock (&priv->lock);
1265 if ((session = priv->session))
1266 g_object_ref (session);
1267 g_mutex_unlock (&priv->lock);
1273 * gst_rtsp_stream_get_ssrc:
1274 * @stream: a #GstRTSPStream
1275 * @ssrc: (out): result ssrc
1277 * Get the SSRC used by the RTP session of this stream. This function can only
1278 * be called when @stream has been joined.
1281 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1283 GstRTSPStreamPrivate *priv;
1285 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1286 priv = stream->priv;
1287 g_return_if_fail (priv->is_joined);
1289 g_mutex_lock (&priv->lock);
1290 if (ssrc && priv->session)
1291 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1292 g_mutex_unlock (&priv->lock);
1295 /* executed from streaming thread */
1297 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1299 GstRTSPStreamPrivate *priv = stream->priv;
1300 GstCaps *newcaps, *oldcaps;
1302 newcaps = gst_pad_get_current_caps (pad);
1304 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1307 g_mutex_lock (&priv->lock);
1308 oldcaps = priv->caps;
1309 priv->caps = newcaps;
1310 g_mutex_unlock (&priv->lock);
1313 gst_caps_unref (oldcaps);
1317 dump_structure (const GstStructure * s)
1321 sstr = gst_structure_to_string (s);
1322 GST_INFO ("structure: %s", sstr);
1326 static GstRTSPStreamTransport *
1327 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1329 GstRTSPStreamPrivate *priv = stream->priv;
1331 GstRTSPStreamTransport *result = NULL;
1336 if (rtcp_from == NULL)
1339 tmp = g_strrstr (rtcp_from, ":");
1343 port = atoi (tmp + 1);
1344 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1346 g_mutex_lock (&priv->lock);
1347 GST_INFO ("finding %s:%d in %d transports", dest, port,
1348 g_list_length (priv->transports));
1350 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1351 GstRTSPStreamTransport *trans = walk->data;
1352 const GstRTSPTransport *tr;
1355 tr = gst_rtsp_stream_transport_get_transport (trans);
1357 min = tr->client_port.min;
1358 max = tr->client_port.max;
1360 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1366 g_object_ref (result);
1367 g_mutex_unlock (&priv->lock);
1374 static GstRTSPStreamTransport *
1375 check_transport (GObject * source, GstRTSPStream * stream)
1377 GstStructure *stats;
1378 GstRTSPStreamTransport *trans;
1380 /* see if we have a stream to match with the origin of the RTCP packet */
1381 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1382 if (trans == NULL) {
1383 g_object_get (source, "stats", &stats, NULL);
1385 const gchar *rtcp_from;
1387 dump_structure (stats);
1389 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1390 if ((trans = find_transport (stream, rtcp_from))) {
1391 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1393 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1396 gst_structure_free (stats);
1404 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1406 GstRTSPStreamTransport *trans;
1408 GST_INFO ("%p: new source %p", stream, source);
1410 trans = check_transport (source, stream);
1413 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1417 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1419 GST_INFO ("%p: new SDES %p", stream, source);
1423 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1425 GstRTSPStreamTransport *trans;
1427 trans = check_transport (source, stream);
1430 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1431 gst_rtsp_stream_transport_keep_alive (trans);
1435 GstStructure *stats;
1436 g_object_get (source, "stats", &stats, NULL);
1438 dump_structure (stats);
1439 gst_structure_free (stats);
1446 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1448 GST_INFO ("%p: source %p bye", stream, source);
1452 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1454 GstRTSPStreamTransport *trans;
1456 GST_INFO ("%p: source %p bye timeout", stream, source);
1458 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1459 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1460 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1465 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1467 GstRTSPStreamTransport *trans;
1469 GST_INFO ("%p: source %p timeout", stream, source);
1471 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1472 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1473 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1478 clear_tr_cache (GstRTSPStreamPrivate * priv)
1480 g_list_foreach (priv->tr_cache, (GFunc) g_object_unref, NULL);
1481 g_list_free (priv->tr_cache);
1482 priv->tr_cache = NULL;
1485 static GstFlowReturn
1486 handle_new_sample (GstAppSink * sink, gpointer user_data)
1488 GstRTSPStreamPrivate *priv;
1492 GstRTSPStream *stream;
1495 sample = gst_app_sink_pull_sample (sink);
1499 stream = (GstRTSPStream *) user_data;
1500 priv = stream->priv;
1501 buffer = gst_sample_get_buffer (sample);
1503 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1505 g_mutex_lock (&priv->lock);
1506 if (priv->tr_changed) {
1507 clear_tr_cache (priv);
1508 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1509 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1510 priv->tr_cache = g_list_prepend (priv->tr_cache, g_object_ref (tr));
1512 priv->tr_changed = FALSE;
1514 g_mutex_unlock (&priv->lock);
1516 for (walk = priv->tr_cache; walk; walk = g_list_next (walk)) {
1517 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1520 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1522 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1525 gst_sample_unref (sample);
1530 static GstAppSinkCallbacks sink_cb = {
1531 NULL, /* not interested in EOS */
1532 NULL, /* not interested in preroll samples */
1537 get_rtp_encoder (GstRTSPStream * stream, guint session)
1539 GstRTSPStreamPrivate *priv = stream->priv;
1541 if (priv->srtpenc == NULL) {
1544 name = g_strdup_printf ("srtpenc_%u", session);
1545 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1548 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1550 return gst_object_ref (priv->srtpenc);
1554 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1556 GstRTSPStreamPrivate *priv = stream->priv;
1557 GstElement *oldenc, *enc;
1561 if (priv->idx != session)
1564 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1566 oldenc = priv->srtpenc;
1567 enc = get_rtp_encoder (stream, session);
1568 name = g_strdup_printf ("rtp_sink_%d", session);
1569 pad = gst_element_get_request_pad (enc, name);
1571 gst_object_unref (pad);
1574 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
1581 request_rtcp_encoder (GstElement * rtpbin, guint session,
1582 GstRTSPStream * stream)
1584 GstRTSPStreamPrivate *priv = stream->priv;
1585 GstElement *oldenc, *enc;
1589 if (priv->idx != session)
1592 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1594 oldenc = priv->srtpenc;
1595 enc = get_rtp_encoder (stream, session);
1596 name = g_strdup_printf ("rtcp_sink_%d", session);
1597 pad = gst_element_get_request_pad (enc, name);
1599 gst_object_unref (pad);
1602 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
1609 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1611 GstRTSPStreamPrivate *priv = stream->priv;
1614 GST_DEBUG ("request key %08x", ssrc);
1616 g_mutex_lock (&priv->lock);
1617 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1618 gst_caps_ref (caps);
1619 g_mutex_unlock (&priv->lock);
1625 request_rtcp_decoder (GstElement * rtpbin, guint session,
1626 GstRTSPStream * stream)
1628 GstRTSPStreamPrivate *priv = stream->priv;
1630 if (priv->idx != session)
1633 if (priv->srtpdec == NULL) {
1636 name = g_strdup_printf ("srtpdec_%u", session);
1637 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1640 g_signal_connect (priv->srtpdec, "request-key",
1641 (GCallback) request_key, stream);
1643 return gst_object_ref (priv->srtpdec);
1647 * gst_rtsp_stream_join_bin:
1648 * @stream: a #GstRTSPStream
1649 * @bin: (transfer none): a #GstBin to join
1650 * @rtpbin: (transfer none): a rtpbin element in @bin
1651 * @state: the target state of the new elements
1653 * Join the #GstBin @bin that contains the element @rtpbin.
1655 * @stream will link to @rtpbin, which must be inside @bin. The elements
1656 * added to @bin will be set to the state given in @state.
1658 * Returns: %TRUE on success.
1661 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1662 GstElement * rtpbin, GstState state)
1664 GstRTSPStreamPrivate *priv;
1668 GstPad *pad, *sinkpad, *selpad;
1669 GstPadLinkReturn ret;
1671 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1672 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1673 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1675 priv = stream->priv;
1677 g_mutex_lock (&priv->lock);
1678 if (priv->is_joined)
1681 /* create a session with the same index as the stream */
1684 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1686 if (!alloc_ports (stream))
1689 /* update the dscp qos field in the sinks */
1690 update_dscp_qos (stream);
1692 if (priv->profiles & GST_RTSP_PROFILE_SAVP
1693 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
1695 g_signal_connect (rtpbin, "request-rtp-encoder",
1696 (GCallback) request_rtp_encoder, stream);
1697 g_signal_connect (rtpbin, "request-rtcp-encoder",
1698 (GCallback) request_rtcp_encoder, stream);
1699 g_signal_connect (rtpbin, "request-rtcp-decoder",
1700 (GCallback) request_rtcp_decoder, stream);
1703 /* get a pad for sending RTP */
1704 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1705 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1707 /* link the RTP pad to the session manager, it should not really fail unless
1708 * this is not really an RTP pad */
1709 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1710 if (ret != GST_PAD_LINK_OK)
1713 /* get pads from the RTP session element for sending and receiving
1715 name = g_strdup_printf ("send_rtp_src_%u", idx);
1716 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1718 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1719 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1721 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1722 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1724 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1725 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1728 /* get the session */
1729 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1731 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1733 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1735 g_signal_connect (priv->session, "on-ssrc-active",
1736 (GCallback) on_ssrc_active, stream);
1737 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1739 g_signal_connect (priv->session, "on-bye-timeout",
1740 (GCallback) on_bye_timeout, stream);
1741 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1744 for (i = 0; i < 2; i++) {
1745 GstPad *teepad, *queuepad;
1746 /* For the sender we create this bit of pipeline for both
1747 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1748 * we need to add a queue before appsink to make the pipeline
1749 * not block. For the TCP case, we want to pump data to the
1750 * client as fast as possible anyway.
1752 * .--------. .-----. .---------.
1753 * | rtpbin | | tee | | udpsink |
1754 * | send->sink src->sink |
1755 * '--------' | | '---------'
1756 * | | .---------. .---------.
1757 * | | | queue | | appsink |
1758 * | src->sink src->sink |
1759 * '-----' '---------' '---------'
1761 * When only UDP is allowed, we skip the tee, queue and appsink and link the
1762 * udpsink directly to the session.
1765 gst_bin_add (bin, priv->udpsink[i]);
1766 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1768 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1769 /* make tee for RTP/RTCP */
1770 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1771 gst_bin_add (bin, priv->tee[i]);
1773 /* and link to rtpbin send pad */
1774 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1775 gst_pad_link (priv->send_src[i], pad);
1776 gst_object_unref (pad);
1778 /* link tee to udpsink */
1779 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1780 gst_pad_link (teepad, sinkpad);
1781 gst_object_unref (teepad);
1784 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1785 gst_bin_add (bin, priv->appqueue[i]);
1786 /* and link to tee */
1787 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1788 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1789 gst_pad_link (teepad, pad);
1790 gst_object_unref (pad);
1791 gst_object_unref (teepad);
1794 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1795 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1796 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1797 gst_bin_add (bin, priv->appsink[i]);
1798 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1799 &sink_cb, stream, NULL);
1800 /* and link to queue */
1801 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1802 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1803 gst_pad_link (queuepad, pad);
1804 gst_object_unref (pad);
1805 gst_object_unref (queuepad);
1807 /* else only udpsink needed, link it to the session */
1808 gst_pad_link (priv->send_src[i], sinkpad);
1810 gst_object_unref (sinkpad);
1812 /* For the receiver we create this bit of pipeline for both
1813 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1814 * and it is all funneled into the rtpbin receive pad.
1816 * .--------. .--------. .--------.
1817 * | udpsrc | | funnel | | rtpbin |
1818 * | src->sink src->sink |
1819 * '--------' | | '--------'
1823 * '--------' '--------'
1825 /* make funnel for the RTP/RTCP receivers */
1826 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1827 gst_bin_add (bin, priv->funnel[i]);
1829 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1830 gst_pad_link (pad, priv->recv_sink[i]);
1831 gst_object_unref (pad);
1833 if (priv->udpsrc_v4[i]) {
1834 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1836 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1837 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1839 gst_bin_add (bin, priv->udpsrc_v4[i]);
1841 /* and link to the funnel v4 */
1842 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1843 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1844 gst_pad_link (pad, selpad);
1845 gst_object_unref (pad);
1846 gst_object_unref (selpad);
1849 if (priv->udpsrc_v6[i]) {
1850 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1851 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1852 gst_bin_add (bin, priv->udpsrc_v6[i]);
1854 /* and link to the funnel v6 */
1855 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1856 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1857 gst_pad_link (pad, selpad);
1858 gst_object_unref (pad);
1859 gst_object_unref (selpad);
1862 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1863 /* make and add appsrc */
1864 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1865 gst_bin_add (bin, priv->appsrc[i]);
1866 /* and link to the funnel */
1867 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1868 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1869 gst_pad_link (pad, selpad);
1870 gst_object_unref (pad);
1871 gst_object_unref (selpad);
1874 /* check if we need to set to a special state */
1875 if (state != GST_STATE_NULL) {
1876 if (priv->udpsink[i])
1877 gst_element_set_state (priv->udpsink[i], state);
1878 if (priv->appsink[i])
1879 gst_element_set_state (priv->appsink[i], state);
1880 if (priv->appqueue[i])
1881 gst_element_set_state (priv->appqueue[i], state);
1883 gst_element_set_state (priv->tee[i], state);
1884 if (priv->funnel[i])
1885 gst_element_set_state (priv->funnel[i], state);
1886 if (priv->appsrc[i])
1887 gst_element_set_state (priv->appsrc[i], state);
1891 /* be notified of caps changes */
1892 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
1893 (GCallback) caps_notify, stream);
1895 priv->is_joined = TRUE;
1896 g_mutex_unlock (&priv->lock);
1903 g_mutex_unlock (&priv->lock);
1908 g_mutex_unlock (&priv->lock);
1909 GST_WARNING ("failed to allocate ports %u", idx);
1914 GST_WARNING ("failed to link stream %u", idx);
1915 gst_object_unref (priv->send_rtp_sink);
1916 priv->send_rtp_sink = NULL;
1917 g_mutex_unlock (&priv->lock);
1923 * gst_rtsp_stream_leave_bin:
1924 * @stream: a #GstRTSPStream
1925 * @bin: (transfer none): a #GstBin
1926 * @rtpbin: (transfer none): a rtpbin #GstElement
1928 * Remove the elements of @stream from @bin.
1930 * Return: %TRUE on success.
1933 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1934 GstElement * rtpbin)
1936 GstRTSPStreamPrivate *priv;
1939 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1940 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1941 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1943 priv = stream->priv;
1945 g_mutex_lock (&priv->lock);
1946 if (!priv->is_joined)
1947 goto was_not_joined;
1949 /* all transports must be removed by now */
1950 g_return_val_if_fail (priv->transports == NULL, FALSE);
1952 clear_tr_cache (priv);
1954 GST_INFO ("stream %p leaving bin", stream);
1956 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1957 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
1958 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1959 gst_object_unref (priv->send_rtp_sink);
1960 priv->send_rtp_sink = NULL;
1962 for (i = 0; i < 2; i++) {
1963 if (priv->udpsink[i])
1964 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1965 if (priv->appsink[i])
1966 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1967 if (priv->appqueue[i])
1968 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1970 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1971 if (priv->funnel[i])
1972 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1973 if (priv->appsrc[i])
1974 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1975 if (priv->udpsrc_v4[i]) {
1976 /* and set udpsrc to NULL now before removing */
1977 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1978 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1979 /* removing them should also nicely release the request
1980 * pads when they finalize */
1981 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1983 if (priv->udpsrc_v6[i]) {
1984 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1985 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1986 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1988 if (priv->udpsink[i])
1989 gst_bin_remove (bin, priv->udpsink[i]);
1990 if (priv->appsrc[i])
1991 gst_bin_remove (bin, priv->appsrc[i]);
1992 if (priv->appsink[i])
1993 gst_bin_remove (bin, priv->appsink[i]);
1994 if (priv->appqueue[i])
1995 gst_bin_remove (bin, priv->appqueue[i]);
1997 gst_bin_remove (bin, priv->tee[i]);
1998 if (priv->funnel[i])
1999 gst_bin_remove (bin, priv->funnel[i]);
2001 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2002 gst_object_unref (priv->recv_sink[i]);
2003 priv->recv_sink[i] = NULL;
2005 priv->udpsrc_v4[i] = NULL;
2006 priv->udpsrc_v6[i] = NULL;
2007 priv->udpsink[i] = NULL;
2008 priv->appsrc[i] = NULL;
2009 priv->appsink[i] = NULL;
2010 priv->appqueue[i] = NULL;
2011 priv->tee[i] = NULL;
2012 priv->funnel[i] = NULL;
2014 gst_object_unref (priv->send_src[0]);
2015 priv->send_src[0] = NULL;
2017 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2018 gst_object_unref (priv->send_src[1]);
2019 priv->send_src[1] = NULL;
2021 g_object_unref (priv->session);
2022 priv->session = NULL;
2024 gst_caps_unref (priv->caps);
2028 gst_object_unref (priv->srtpenc);
2030 priv->is_joined = FALSE;
2031 g_mutex_unlock (&priv->lock);
2037 g_mutex_unlock (&priv->lock);
2043 * gst_rtsp_stream_get_rtpinfo:
2044 * @stream: a #GstRTSPStream
2045 * @rtptime: (allow-none): result RTP timestamp
2046 * @seq: (allow-none): result RTP seqnum
2047 * @clock_rate: (allow-none): the clock rate
2048 * @running_time: (allow-none): result running-time
2050 * Retrieve the current rtptime, seq and running-time. This is used to
2051 * construct a RTPInfo reply header.
2053 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2056 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2057 guint * rtptime, guint * seq, guint * clock_rate,
2058 GstClockTime * running_time)
2060 GstRTSPStreamPrivate *priv;
2061 GstStructure *stats;
2062 GObjectClass *payobjclass;
2064 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2066 priv = stream->priv;
2068 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2070 g_mutex_lock (&priv->lock);
2072 if (g_object_class_find_property (payobjclass, "stats")) {
2073 g_object_get (priv->payloader, "stats", &stats, NULL);
2078 gst_structure_get_uint (stats, "seqnum", seq);
2081 gst_structure_get_uint (stats, "timestamp", rtptime);
2084 gst_structure_get_clock_time (stats, "running-time", running_time);
2087 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2088 if (*clock_rate == 0 && running_time)
2089 *running_time = GST_CLOCK_TIME_NONE;
2091 gst_structure_free (stats);
2093 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2094 !g_object_class_find_property (payobjclass, "timestamp"))
2098 g_object_get (priv->payloader, "seqnum", seq, NULL);
2101 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2104 *running_time = GST_CLOCK_TIME_NONE;
2106 g_mutex_unlock (&priv->lock);
2113 GST_WARNING ("Could not get payloader stats");
2114 g_mutex_unlock (&priv->lock);
2120 * gst_rtsp_stream_get_caps:
2121 * @stream: a #GstRTSPStream
2123 * Retrieve the current caps of @stream.
2125 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2129 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2131 GstRTSPStreamPrivate *priv;
2134 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2136 priv = stream->priv;
2138 g_mutex_lock (&priv->lock);
2139 if ((result = priv->caps))
2140 gst_caps_ref (result);
2141 g_mutex_unlock (&priv->lock);
2147 * gst_rtsp_stream_recv_rtp:
2148 * @stream: a #GstRTSPStream
2149 * @buffer: (transfer full): a #GstBuffer
2151 * Handle an RTP buffer for the stream. This method is usually called when a
2152 * message has been received from a client using the TCP transport.
2154 * This function takes ownership of @buffer.
2156 * Returns: a GstFlowReturn.
2159 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2161 GstRTSPStreamPrivate *priv;
2163 GstElement *element;
2165 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2166 priv = stream->priv;
2167 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2168 g_return_val_if_fail (priv->is_joined, FALSE);
2170 g_mutex_lock (&priv->lock);
2171 if (priv->appsrc[0])
2172 element = gst_object_ref (priv->appsrc[0]);
2175 g_mutex_unlock (&priv->lock);
2178 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2179 gst_object_unref (element);
2187 * gst_rtsp_stream_recv_rtcp:
2188 * @stream: a #GstRTSPStream
2189 * @buffer: (transfer full): a #GstBuffer
2191 * Handle an RTCP buffer for the stream. This method is usually called when a
2192 * message has been received from a client using the TCP transport.
2194 * This function takes ownership of @buffer.
2196 * Returns: a GstFlowReturn.
2199 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2201 GstRTSPStreamPrivate *priv;
2203 GstElement *element;
2205 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2206 priv = stream->priv;
2207 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2208 g_return_val_if_fail (priv->is_joined, FALSE);
2210 g_mutex_lock (&priv->lock);
2211 if (priv->appsrc[1])
2212 element = gst_object_ref (priv->appsrc[1]);
2215 g_mutex_unlock (&priv->lock);
2218 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2219 gst_object_unref (element);
2222 gst_buffer_unref (buffer);
2227 /* must be called with lock */
2229 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
2232 GstRTSPStreamPrivate *priv = stream->priv;
2233 const GstRTSPTransport *tr;
2235 tr = gst_rtsp_stream_transport_get_transport (trans);
2237 switch (tr->lower_transport) {
2238 case GST_RTSP_LOWER_TRANS_UDP:
2239 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2245 dest = tr->destination;
2246 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2251 min = tr->client_port.min;
2252 max = tr->client_port.max;
2257 GST_INFO ("setting ttl-mc %d", ttl);
2258 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
2259 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
2261 GST_INFO ("adding %s:%d-%d", dest, min, max);
2262 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
2263 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
2264 priv->transports = g_list_prepend (priv->transports, trans);
2266 GST_INFO ("removing %s:%d-%d", dest, min, max);
2267 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
2268 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
2269 priv->transports = g_list_remove (priv->transports, trans);
2271 priv->tr_changed = TRUE;
2274 case GST_RTSP_LOWER_TRANS_TCP:
2276 GST_INFO ("adding TCP %s", tr->destination);
2277 priv->transports = g_list_prepend (priv->transports, trans);
2279 GST_INFO ("removing TCP %s", tr->destination);
2280 priv->transports = g_list_remove (priv->transports, trans);
2282 priv->tr_changed = TRUE;
2285 goto unknown_transport;
2292 GST_INFO ("Unknown transport %d", tr->lower_transport);
2299 * gst_rtsp_stream_add_transport:
2300 * @stream: a #GstRTSPStream
2301 * @trans: (transfer none): a #GstRTSPStreamTransport
2303 * Add the transport in @trans to @stream. The media of @stream will
2304 * then also be send to the values configured in @trans.
2306 * @stream must be joined to a bin.
2308 * @trans must contain a valid #GstRTSPTransport.
2310 * Returns: %TRUE if @trans was added
2313 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
2314 GstRTSPStreamTransport * trans)
2316 GstRTSPStreamPrivate *priv;
2319 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2320 priv = stream->priv;
2321 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2322 g_return_val_if_fail (priv->is_joined, FALSE);
2324 g_mutex_lock (&priv->lock);
2325 res = update_transport (stream, trans, TRUE);
2326 g_mutex_unlock (&priv->lock);
2332 * gst_rtsp_stream_remove_transport:
2333 * @stream: a #GstRTSPStream
2334 * @trans: (transfer none): a #GstRTSPStreamTransport
2336 * Remove the transport in @trans from @stream. The media of @stream will
2337 * not be sent to the values configured in @trans.
2339 * @stream must be joined to a bin.
2341 * @trans must contain a valid #GstRTSPTransport.
2343 * Returns: %TRUE if @trans was removed
2346 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2347 GstRTSPStreamTransport * trans)
2349 GstRTSPStreamPrivate *priv;
2352 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2353 priv = stream->priv;
2354 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2355 g_return_val_if_fail (priv->is_joined, FALSE);
2357 g_mutex_lock (&priv->lock);
2358 res = update_transport (stream, trans, FALSE);
2359 g_mutex_unlock (&priv->lock);
2365 * gst_rtsp_stream_update_crypto:
2366 * @stream: a #GstRTSPStream
2368 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
2370 * Update the new crypto information for @ssrc in @stream. If information
2371 * for @ssrc did not exist, it will be added. If information
2372 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
2373 * be removed from @stream.
2375 * Returns: %TRUE if @crypto could be updated
2378 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
2379 guint ssrc, GstCaps * crypto)
2381 GstRTSPStreamPrivate *priv;
2383 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2384 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
2386 priv = stream->priv;
2388 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
2390 g_mutex_lock (&priv->lock);
2392 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
2393 gst_caps_ref (crypto));
2395 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
2396 g_mutex_unlock (&priv->lock);
2402 * gst_rtsp_stream_get_rtp_socket:
2403 * @stream: a #GstRTSPStream
2404 * @family: the socket family
2406 * Get the RTP socket from @stream for a @family.
2408 * @stream must be joined to a bin.
2410 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
2411 * socket could be allocated for @family. Unref after usage
2414 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
2416 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2420 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2421 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2422 family == G_SOCKET_FAMILY_IPV6, NULL);
2423 g_return_val_if_fail (priv->udpsink[0], NULL);
2425 if (family == G_SOCKET_FAMILY_IPV6)
2430 g_object_get (priv->udpsink[0], name, &socket, NULL);
2436 * gst_rtsp_stream_get_rtcp_socket:
2437 * @stream: a #GstRTSPStream
2438 * @family: the socket family
2440 * Get the RTCP socket from @stream for a @family.
2442 * @stream must be joined to a bin.
2444 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
2445 * socket could be allocated for @family. Unref after usage
2448 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
2450 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2454 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2455 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2456 family == G_SOCKET_FAMILY_IPV6, NULL);
2457 g_return_val_if_fail (priv->udpsink[1], NULL);
2459 if (family == G_SOCKET_FAMILY_IPV6)
2464 g_object_get (priv->udpsink[1], name, &socket, NULL);
2470 * gst_rtsp_stream_transport_filter:
2471 * @stream: a #GstRTSPStream
2472 * @func: (scope call) (allow-none): a callback
2473 * @user_data: (closure): user data passed to @func
2475 * Call @func for each transport managed by @stream. The result value of @func
2476 * determines what happens to the transport. @func will be called with @stream
2477 * locked so no further actions on @stream can be performed from @func.
2479 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
2482 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
2484 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
2485 * will also be added with an additional ref to the result #GList of this
2488 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
2490 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
2491 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
2492 * element in the #GList should be unreffed before the list is freed.
2495 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
2496 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
2498 GstRTSPStreamPrivate *priv;
2499 GList *result, *walk, *next;
2501 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2503 priv = stream->priv;
2507 g_mutex_lock (&priv->lock);
2508 for (walk = priv->transports; walk; walk = next) {
2509 GstRTSPStreamTransport *trans = walk->data;
2510 GstRTSPFilterResult res;
2512 next = g_list_next (walk);
2515 res = func (stream, trans, user_data);
2517 res = GST_RTSP_FILTER_REF;
2520 case GST_RTSP_FILTER_REMOVE:
2521 update_transport (stream, trans, FALSE);
2523 case GST_RTSP_FILTER_REF:
2524 result = g_list_prepend (result, g_object_ref (trans));
2526 case GST_RTSP_FILTER_KEEP:
2531 g_mutex_unlock (&priv->lock);
2536 static GstPadProbeReturn
2537 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2539 GstRTSPStreamPrivate *priv;
2540 GstRTSPStream *stream;
2543 priv = stream->priv;
2545 GST_DEBUG_OBJECT (pad, "now blocking");
2547 g_mutex_lock (&priv->lock);
2548 priv->blocking = TRUE;
2549 g_mutex_unlock (&priv->lock);
2551 gst_element_post_message (priv->payloader,
2552 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
2553 gst_structure_new_empty ("GstRTSPStreamBlocking")));
2555 return GST_PAD_PROBE_OK;
2559 * gst_rtsp_stream_set_blocked:
2560 * @stream: a #GstRTSPStream
2561 * @blocked: boolean indicating we should block or unblock
2563 * Blocks or unblocks the dataflow on @stream.
2565 * Returns: %TRUE on success
2568 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
2570 GstRTSPStreamPrivate *priv;
2572 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2574 priv = stream->priv;
2576 g_mutex_lock (&priv->lock);
2578 priv->blocking = FALSE;
2579 if (priv->blocked_id == 0) {
2580 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
2581 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
2582 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
2583 g_object_ref (stream), g_object_unref);
2586 if (priv->blocked_id != 0) {
2587 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
2588 priv->blocked_id = 0;
2589 priv->blocking = FALSE;
2592 g_mutex_unlock (&priv->lock);
2598 * gst_rtsp_stream_is_blocking:
2599 * @stream: a #GstRTSPStream
2601 * Check if @stream is blocking on a #GstBuffer.
2603 * Returns: %TRUE if @stream is blocking
2606 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
2608 GstRTSPStreamPrivate *priv;
2611 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2613 priv = stream->priv;
2615 g_mutex_lock (&priv->lock);
2616 result = priv->blocking;
2617 g_mutex_unlock (&priv->lock);