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))
63 GstRTSPStreamTransport *transport;
65 /* RTP and RTCP source */
66 GstElement *udpsrc[2];
68 } GstRTSPMulticastTransportSource;
70 struct _GstRTSPStreamPrivate
75 GstElement *payloader;
80 GstRTSPProfile profiles;
81 GstRTSPLowerTrans protocols;
83 /* pads on the rtpbin */
84 GstPad *send_rtp_sink;
88 /* the RTPSession object */
91 /* SRTP encoder/decoder */
96 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
98 GstElement *udpsrc_v4[2];
100 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
102 GstElement *udpsrc_v6[2];
104 GstElement *udpsink[2];
106 /* for TCP transport */
107 GstElement *appsrc[2];
108 GstElement *appqueue[2];
109 GstElement *appsink[2];
112 GstElement *funnel[2];
114 /* server ports for sending/receiving over ipv4 */
115 GstRTSPRange server_port_v4;
116 GstRTSPAddress *server_addr_v4;
119 /* server ports for sending/receiving over ipv6 */
120 GstRTSPRange server_port_v6;
121 GstRTSPAddress *server_addr_v6;
124 /* multicast addresses */
125 GstRTSPAddressPool *pool;
126 GstRTSPAddress *addr_v4;
127 GstRTSPAddress *addr_v6;
129 /* the caps of the stream */
133 /* transports we stream to */
136 guint transports_cookie;
138 guint tr_cache_cookie;
140 /* UDP sources for UDP multicast transports */
141 GList *transport_sources;
145 /* stream blocking */
150 #define DEFAULT_CONTROL NULL
151 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
152 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
153 GST_RTSP_LOWER_TRANS_TCP
166 SIGNAL_NEW_RTP_ENCODER,
167 SIGNAL_NEW_RTCP_ENCODER,
171 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
172 #define GST_CAT_DEFAULT rtsp_stream_debug
174 static GQuark ssrc_stream_map_key;
176 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
177 GValue * value, GParamSpec * pspec);
178 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
179 const GValue * value, GParamSpec * pspec);
181 static void gst_rtsp_stream_finalize (GObject * obj);
183 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
185 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
188 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
190 GObjectClass *gobject_class;
192 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
194 gobject_class = G_OBJECT_CLASS (klass);
196 gobject_class->get_property = gst_rtsp_stream_get_property;
197 gobject_class->set_property = gst_rtsp_stream_set_property;
198 gobject_class->finalize = gst_rtsp_stream_finalize;
200 g_object_class_install_property (gobject_class, PROP_CONTROL,
201 g_param_spec_string ("control", "Control",
202 "The control string for this stream", DEFAULT_CONTROL,
203 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
205 g_object_class_install_property (gobject_class, PROP_PROFILES,
206 g_param_spec_flags ("profiles", "Profiles",
207 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
208 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
210 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
211 g_param_spec_flags ("protocols", "Protocols",
212 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
213 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
215 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
216 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
217 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
218 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
220 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
221 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
222 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
223 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
225 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
227 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
231 gst_rtsp_stream_init (GstRTSPStream * stream)
233 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
235 GST_DEBUG ("new stream %p", stream);
240 priv->control = g_strdup (DEFAULT_CONTROL);
241 priv->profiles = DEFAULT_PROFILES;
242 priv->protocols = DEFAULT_PROTOCOLS;
244 g_mutex_init (&priv->lock);
246 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
247 NULL, (GDestroyNotify) gst_caps_unref);
251 gst_rtsp_stream_finalize (GObject * obj)
253 GstRTSPStream *stream;
254 GstRTSPStreamPrivate *priv;
256 stream = GST_RTSP_STREAM (obj);
259 GST_DEBUG ("finalize stream %p", stream);
261 /* we really need to be unjoined now */
262 g_return_if_fail (!priv->is_joined);
265 gst_rtsp_address_free (priv->addr_v4);
267 gst_rtsp_address_free (priv->addr_v6);
268 if (priv->server_addr_v4)
269 gst_rtsp_address_free (priv->server_addr_v4);
270 if (priv->server_addr_v6)
271 gst_rtsp_address_free (priv->server_addr_v6);
273 g_object_unref (priv->pool);
274 gst_object_unref (priv->payloader);
275 gst_object_unref (priv->srcpad);
276 g_free (priv->control);
277 g_mutex_clear (&priv->lock);
279 g_hash_table_unref (priv->keys);
281 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
285 gst_rtsp_stream_get_property (GObject * object, guint propid,
286 GValue * value, GParamSpec * pspec)
288 GstRTSPStream *stream = GST_RTSP_STREAM (object);
292 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
295 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
298 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
301 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
306 gst_rtsp_stream_set_property (GObject * object, guint propid,
307 const GValue * value, GParamSpec * pspec)
309 GstRTSPStream *stream = GST_RTSP_STREAM (object);
313 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
316 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
319 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
322 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
327 * gst_rtsp_stream_new:
330 * @payloader: a #GstElement
332 * Create a new media stream with index @idx that handles RTP data on
333 * @srcpad and has a payloader element @payloader.
335 * Returns: (transfer full): a new #GstRTSPStream
338 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
340 GstRTSPStreamPrivate *priv;
341 GstRTSPStream *stream;
343 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
344 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
345 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
347 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
350 priv->payloader = gst_object_ref (payloader);
351 priv->srcpad = gst_object_ref (srcpad);
357 * gst_rtsp_stream_get_index:
358 * @stream: a #GstRTSPStream
360 * Get the stream index.
362 * Return: the stream index.
365 gst_rtsp_stream_get_index (GstRTSPStream * stream)
367 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
369 return stream->priv->idx;
373 * gst_rtsp_stream_get_pt:
374 * @stream: a #GstRTSPStream
376 * Get the stream payload type.
378 * Return: the stream payload type.
381 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
383 GstRTSPStreamPrivate *priv;
386 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
390 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
396 * gst_rtsp_stream_get_srcpad:
397 * @stream: a #GstRTSPStream
399 * Get the srcpad associated with @stream.
401 * Returns: (transfer full): the srcpad. Unref after usage.
404 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
406 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
408 return gst_object_ref (stream->priv->srcpad);
412 * gst_rtsp_stream_get_control:
413 * @stream: a #GstRTSPStream
415 * Get the control string to identify this stream.
417 * Returns: (transfer full): the control string. g_free() after usage.
420 gst_rtsp_stream_get_control (GstRTSPStream * stream)
422 GstRTSPStreamPrivate *priv;
425 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
429 g_mutex_lock (&priv->lock);
430 if ((result = g_strdup (priv->control)) == NULL)
431 result = g_strdup_printf ("stream=%u", priv->idx);
432 g_mutex_unlock (&priv->lock);
438 * gst_rtsp_stream_set_control:
439 * @stream: a #GstRTSPStream
440 * @control: a control string
442 * Set the control string in @stream.
445 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
447 GstRTSPStreamPrivate *priv;
449 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
453 g_mutex_lock (&priv->lock);
454 g_free (priv->control);
455 priv->control = g_strdup (control);
456 g_mutex_unlock (&priv->lock);
460 * gst_rtsp_stream_has_control:
461 * @stream: a #GstRTSPStream
462 * @control: a control string
464 * Check if @stream has the control string @control.
466 * Returns: %TRUE is @stream has @control as the control string
469 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
471 GstRTSPStreamPrivate *priv;
474 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
478 g_mutex_lock (&priv->lock);
480 res = (g_strcmp0 (priv->control, control) == 0);
484 if (sscanf (control, "stream=%u", &streamid) > 0)
485 res = (streamid == priv->idx);
489 g_mutex_unlock (&priv->lock);
495 * gst_rtsp_stream_set_mtu:
496 * @stream: a #GstRTSPStream
499 * Configure the mtu in the payloader of @stream to @mtu.
502 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
504 GstRTSPStreamPrivate *priv;
506 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
510 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
512 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
516 * gst_rtsp_stream_get_mtu:
517 * @stream: a #GstRTSPStream
519 * Get the configured MTU in the payloader of @stream.
521 * Returns: the MTU of the payloader.
524 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
526 GstRTSPStreamPrivate *priv;
529 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
533 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
538 /* Update the dscp qos property on the udp sinks */
540 update_dscp_qos (GstRTSPStream * stream)
542 GstRTSPStreamPrivate *priv;
544 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
548 if (priv->udpsink[0]) {
549 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
553 if (priv->udpsink[1]) {
554 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
560 * gst_rtsp_stream_set_dscp_qos:
561 * @stream: a #GstRTSPStream
562 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
564 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
567 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
569 GstRTSPStreamPrivate *priv;
571 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
575 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
577 if (dscp_qos < -1 || dscp_qos > 63) {
578 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
582 priv->dscp_qos = dscp_qos;
584 update_dscp_qos (stream);
588 * gst_rtsp_stream_get_dscp_qos:
589 * @stream: a #GstRTSPStream
591 * Get the configured DSCP QoS in of the outgoing sockets.
593 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
596 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
598 GstRTSPStreamPrivate *priv;
600 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
604 return priv->dscp_qos;
608 * gst_rtsp_stream_is_transport_supported:
609 * @stream: a #GstRTSPStream
610 * @transport: (transfer none): a #GstRTSPTransport
612 * Check if @transport can be handled by stream
614 * Returns: %TRUE if @transport can be handled by @stream.
617 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
618 GstRTSPTransport * transport)
620 GstRTSPStreamPrivate *priv;
622 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
626 g_mutex_lock (&priv->lock);
627 if (transport->trans != GST_RTSP_TRANS_RTP)
628 goto unsupported_transmode;
630 if (!(transport->profile & priv->profiles))
631 goto unsupported_profile;
633 if (!(transport->lower_transport & priv->protocols))
634 goto unsupported_ltrans;
636 g_mutex_unlock (&priv->lock);
641 unsupported_transmode:
643 GST_DEBUG ("unsupported transport mode %d", transport->trans);
644 g_mutex_unlock (&priv->lock);
649 GST_DEBUG ("unsupported profile %d", transport->profile);
650 g_mutex_unlock (&priv->lock);
655 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
656 g_mutex_unlock (&priv->lock);
662 * gst_rtsp_stream_set_profiles:
663 * @stream: a #GstRTSPStream
664 * @profiles: the new profiles
666 * Configure the allowed profiles for @stream.
669 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
671 GstRTSPStreamPrivate *priv;
673 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
677 g_mutex_lock (&priv->lock);
678 priv->profiles = profiles;
679 g_mutex_unlock (&priv->lock);
683 * gst_rtsp_stream_get_profiles:
684 * @stream: a #GstRTSPStream
686 * Get the allowed profiles of @stream.
688 * Returns: a #GstRTSPProfile
691 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
693 GstRTSPStreamPrivate *priv;
696 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
700 g_mutex_lock (&priv->lock);
701 res = priv->profiles;
702 g_mutex_unlock (&priv->lock);
708 * gst_rtsp_stream_set_protocols:
709 * @stream: a #GstRTSPStream
710 * @protocols: the new flags
712 * Configure the allowed lower transport for @stream.
715 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
716 GstRTSPLowerTrans protocols)
718 GstRTSPStreamPrivate *priv;
720 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
724 g_mutex_lock (&priv->lock);
725 priv->protocols = protocols;
726 g_mutex_unlock (&priv->lock);
730 * gst_rtsp_stream_get_protocols:
731 * @stream: a #GstRTSPStream
733 * Get the allowed protocols of @stream.
735 * Returns: a #GstRTSPLowerTrans
738 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
740 GstRTSPStreamPrivate *priv;
741 GstRTSPLowerTrans res;
743 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
744 GST_RTSP_LOWER_TRANS_UNKNOWN);
748 g_mutex_lock (&priv->lock);
749 res = priv->protocols;
750 g_mutex_unlock (&priv->lock);
756 * gst_rtsp_stream_set_address_pool:
757 * @stream: a #GstRTSPStream
758 * @pool: (transfer none): a #GstRTSPAddressPool
760 * configure @pool to be used as the address pool of @stream.
763 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
764 GstRTSPAddressPool * pool)
766 GstRTSPStreamPrivate *priv;
767 GstRTSPAddressPool *old;
769 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
773 GST_LOG_OBJECT (stream, "set address pool %p", pool);
775 g_mutex_lock (&priv->lock);
776 if ((old = priv->pool) != pool)
777 priv->pool = pool ? g_object_ref (pool) : NULL;
780 g_mutex_unlock (&priv->lock);
783 g_object_unref (old);
787 * gst_rtsp_stream_get_address_pool:
788 * @stream: a #GstRTSPStream
790 * Get the #GstRTSPAddressPool used as the address pool of @stream.
792 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
796 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
798 GstRTSPStreamPrivate *priv;
799 GstRTSPAddressPool *result;
801 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
805 g_mutex_lock (&priv->lock);
806 if ((result = priv->pool))
807 g_object_ref (result);
808 g_mutex_unlock (&priv->lock);
814 * gst_rtsp_stream_get_multicast_address:
815 * @stream: a #GstRTSPStream
816 * @family: the #GSocketFamily
818 * Get the multicast address of @stream for @family.
820 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
821 * or %NULL when no address could be allocated. gst_rtsp_address_free()
825 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
826 GSocketFamily family)
828 GstRTSPStreamPrivate *priv;
829 GstRTSPAddress *result;
830 GstRTSPAddress **addrp;
831 GstRTSPAddressFlags flags;
833 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
837 if (family == G_SOCKET_FAMILY_IPV6) {
838 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
839 addrp = &priv->addr_v6;
841 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
842 addrp = &priv->addr_v4;
845 g_mutex_lock (&priv->lock);
846 if (*addrp == NULL) {
847 if (priv->pool == NULL)
850 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
852 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
856 result = gst_rtsp_address_copy (*addrp);
857 g_mutex_unlock (&priv->lock);
864 GST_ERROR_OBJECT (stream, "no address pool specified");
865 g_mutex_unlock (&priv->lock);
870 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
871 g_mutex_unlock (&priv->lock);
877 * gst_rtsp_stream_reserve_address:
878 * @stream: a #GstRTSPStream
879 * @address: an address
884 * Reserve @address and @port as the address and port of @stream.
886 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
887 * the address could be reserved. gst_rtsp_address_free() after usage.
890 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
891 const gchar * address, guint port, guint n_ports, guint ttl)
893 GstRTSPStreamPrivate *priv;
894 GstRTSPAddress *result;
896 GSocketFamily family;
897 GstRTSPAddress **addrp;
899 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
900 g_return_val_if_fail (address != NULL, NULL);
901 g_return_val_if_fail (port > 0, NULL);
902 g_return_val_if_fail (n_ports > 0, NULL);
903 g_return_val_if_fail (ttl > 0, NULL);
907 addr = g_inet_address_new_from_string (address);
909 GST_ERROR ("failed to get inet addr from %s", address);
910 family = G_SOCKET_FAMILY_IPV4;
912 family = g_inet_address_get_family (addr);
913 g_object_unref (addr);
916 if (family == G_SOCKET_FAMILY_IPV6)
917 addrp = &priv->addr_v6;
919 addrp = &priv->addr_v4;
921 g_mutex_lock (&priv->lock);
922 if (*addrp == NULL) {
923 GstRTSPAddressPoolResult res;
925 if (priv->pool == NULL)
928 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
929 port, n_ports, ttl, addrp);
930 if (res != GST_RTSP_ADDRESS_POOL_OK)
933 if (strcmp ((*addrp)->address, address) ||
934 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
935 (*addrp)->ttl != ttl)
936 goto different_address;
938 result = gst_rtsp_address_copy (*addrp);
939 g_mutex_unlock (&priv->lock);
946 GST_ERROR_OBJECT (stream, "no address pool specified");
947 g_mutex_unlock (&priv->lock);
952 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
954 g_mutex_unlock (&priv->lock);
959 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
960 " reserved", address);
961 g_mutex_unlock (&priv->lock);
967 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
968 GSocketFamily family, GstElement * udpsrc_out[2],
969 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
970 GstRTSPAddress ** server_addr_out)
972 GstStateChangeReturn ret;
973 GstElement *udpsrc0, *udpsrc1;
974 GstElement *udpsink0, *udpsink1;
975 GSocket *rtp_socket = NULL;
976 GSocket *rtcp_socket;
977 gint tmp_rtp, tmp_rtcp;
979 gint rtpport, rtcpport;
980 GList *rejected_addresses = NULL;
981 GstRTSPAddress *addr = NULL;
982 GInetAddress *inetaddr = NULL;
983 GSocketAddress *rtp_sockaddr = NULL;
984 GSocketAddress *rtcp_sockaddr = NULL;
985 const gchar *multisink_socket;
987 if (family == G_SOCKET_FAMILY_IPV6)
988 multisink_socket = "socket-v6";
990 multisink_socket = "socket";
998 /* Start with random port */
1001 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1002 G_SOCKET_PROTOCOL_UDP, NULL);
1004 goto no_udp_protocol;
1006 if (*server_addr_out)
1007 gst_rtsp_address_free (*server_addr_out);
1009 /* try to allocate 2 UDP ports, the RTP port should be an even
1010 * number and the RTCP port should be the next (uneven) port */
1013 if (rtp_socket == NULL) {
1014 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1015 G_SOCKET_PROTOCOL_UDP, NULL);
1017 goto no_udp_protocol;
1020 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
1021 GstRTSPAddressFlags flags;
1024 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1026 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1027 if (family == G_SOCKET_FAMILY_IPV6)
1028 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1030 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1032 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1037 tmp_rtp = addr->port;
1039 g_clear_object (&inetaddr);
1040 inetaddr = g_inet_address_new_from_string (addr->address);
1048 if (inetaddr == NULL)
1049 inetaddr = g_inet_address_new_any (family);
1052 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1053 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1054 g_object_unref (rtp_sockaddr);
1057 g_object_unref (rtp_sockaddr);
1059 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1060 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1061 g_clear_object (&rtp_sockaddr);
1066 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1067 g_object_unref (rtp_sockaddr);
1069 /* check if port is even */
1070 if ((tmp_rtp & 1) != 0) {
1071 /* port not even, close and allocate another */
1073 g_clear_object (&rtp_socket);
1078 tmp_rtcp = tmp_rtp + 1;
1080 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1081 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1082 g_object_unref (rtcp_sockaddr);
1083 g_clear_object (&rtp_socket);
1086 g_object_unref (rtcp_sockaddr);
1088 g_clear_object (&inetaddr);
1090 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1091 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1093 if (udpsrc0 == NULL || udpsrc1 == NULL)
1094 goto no_udp_protocol;
1096 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1097 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1099 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1100 if (ret == GST_STATE_CHANGE_FAILURE)
1102 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1103 if (ret == GST_STATE_CHANGE_FAILURE)
1106 /* all fine, do port check */
1107 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1108 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1110 /* this should not happen... */
1111 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1115 udpsink0 = udpsink_out[0];
1117 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1120 goto no_udp_protocol;
1122 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1123 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
1126 udpsink1 = udpsink_out[1];
1128 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1131 goto no_udp_protocol;
1133 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1134 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1135 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
1137 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1138 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
1139 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1140 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1141 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1142 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1143 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1144 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1146 /* we keep these elements, we will further configure them when the
1147 * client told us to really use the UDP ports. */
1148 udpsrc_out[0] = udpsrc0;
1149 udpsrc_out[1] = udpsrc1;
1150 udpsink_out[0] = udpsink0;
1151 udpsink_out[1] = udpsink1;
1153 server_port_out->min = rtpport;
1154 server_port_out->max = rtcpport;
1156 *server_addr_out = addr;
1157 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1159 g_object_unref (rtp_socket);
1160 g_object_unref (rtcp_socket);
1188 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1189 gst_object_unref (udpsrc0);
1192 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1193 gst_object_unref (udpsrc1);
1196 gst_element_set_state (udpsink0, GST_STATE_NULL);
1197 gst_object_unref (udpsink0);
1200 g_object_unref (inetaddr);
1201 g_list_free_full (rejected_addresses,
1202 (GDestroyNotify) gst_rtsp_address_free);
1204 gst_rtsp_address_free (addr);
1206 g_object_unref (rtp_socket);
1208 g_object_unref (rtcp_socket);
1213 /* must be called with lock */
1215 alloc_ports (GstRTSPStream * stream)
1217 GstRTSPStreamPrivate *priv = stream->priv;
1219 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1220 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1221 &priv->server_port_v4, &priv->server_addr_v4);
1223 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1224 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1225 &priv->server_port_v6, &priv->server_addr_v6);
1227 return priv->have_ipv4 || priv->have_ipv6;
1231 * gst_rtsp_stream_get_server_port:
1232 * @stream: a #GstRTSPStream
1233 * @server_port: (out): result server port
1234 * @family: the port family to get
1236 * Fill @server_port with the port pair used by the server. This function can
1237 * only be called when @stream has been joined.
1240 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1241 GstRTSPRange * server_port, GSocketFamily family)
1243 GstRTSPStreamPrivate *priv;
1245 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1246 priv = stream->priv;
1247 g_return_if_fail (priv->is_joined);
1249 g_mutex_lock (&priv->lock);
1250 if (family == G_SOCKET_FAMILY_IPV4) {
1252 *server_port = priv->server_port_v4;
1255 *server_port = priv->server_port_v6;
1257 g_mutex_unlock (&priv->lock);
1261 * gst_rtsp_stream_get_rtpsession:
1262 * @stream: a #GstRTSPStream
1264 * Get the RTP session of this stream.
1266 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1269 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1271 GstRTSPStreamPrivate *priv;
1274 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1276 priv = stream->priv;
1278 g_mutex_lock (&priv->lock);
1279 if ((session = priv->session))
1280 g_object_ref (session);
1281 g_mutex_unlock (&priv->lock);
1287 * gst_rtsp_stream_get_ssrc:
1288 * @stream: a #GstRTSPStream
1289 * @ssrc: (out): result ssrc
1291 * Get the SSRC used by the RTP session of this stream. This function can only
1292 * be called when @stream has been joined.
1295 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1297 GstRTSPStreamPrivate *priv;
1299 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1300 priv = stream->priv;
1301 g_return_if_fail (priv->is_joined);
1303 g_mutex_lock (&priv->lock);
1304 if (ssrc && priv->session)
1305 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1306 g_mutex_unlock (&priv->lock);
1309 /* executed from streaming thread */
1311 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1313 GstRTSPStreamPrivate *priv = stream->priv;
1314 GstCaps *newcaps, *oldcaps;
1316 newcaps = gst_pad_get_current_caps (pad);
1318 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1321 g_mutex_lock (&priv->lock);
1322 oldcaps = priv->caps;
1323 priv->caps = newcaps;
1324 g_mutex_unlock (&priv->lock);
1327 gst_caps_unref (oldcaps);
1331 dump_structure (const GstStructure * s)
1335 sstr = gst_structure_to_string (s);
1336 GST_INFO ("structure: %s", sstr);
1340 static GstRTSPStreamTransport *
1341 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1343 GstRTSPStreamPrivate *priv = stream->priv;
1345 GstRTSPStreamTransport *result = NULL;
1350 if (rtcp_from == NULL)
1353 tmp = g_strrstr (rtcp_from, ":");
1357 port = atoi (tmp + 1);
1358 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1360 g_mutex_lock (&priv->lock);
1361 GST_INFO ("finding %s:%d in %d transports", dest, port,
1362 g_list_length (priv->transports));
1364 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1365 GstRTSPStreamTransport *trans = walk->data;
1366 const GstRTSPTransport *tr;
1369 tr = gst_rtsp_stream_transport_get_transport (trans);
1371 min = tr->client_port.min;
1372 max = tr->client_port.max;
1374 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1380 g_object_ref (result);
1381 g_mutex_unlock (&priv->lock);
1388 static GstRTSPStreamTransport *
1389 check_transport (GObject * source, GstRTSPStream * stream)
1391 GstStructure *stats;
1392 GstRTSPStreamTransport *trans;
1394 /* see if we have a stream to match with the origin of the RTCP packet */
1395 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1396 if (trans == NULL) {
1397 g_object_get (source, "stats", &stats, NULL);
1399 const gchar *rtcp_from;
1401 dump_structure (stats);
1403 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1404 if ((trans = find_transport (stream, rtcp_from))) {
1405 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1407 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1410 gst_structure_free (stats);
1418 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1420 GstRTSPStreamTransport *trans;
1422 GST_INFO ("%p: new source %p", stream, source);
1424 trans = check_transport (source, stream);
1427 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1431 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1433 GST_INFO ("%p: new SDES %p", stream, source);
1437 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1439 GstRTSPStreamTransport *trans;
1441 trans = check_transport (source, stream);
1444 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1445 gst_rtsp_stream_transport_keep_alive (trans);
1449 GstStructure *stats;
1450 g_object_get (source, "stats", &stats, NULL);
1452 dump_structure (stats);
1453 gst_structure_free (stats);
1460 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1462 GST_INFO ("%p: source %p bye", stream, source);
1466 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1468 GstRTSPStreamTransport *trans;
1470 GST_INFO ("%p: source %p bye timeout", stream, source);
1472 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1473 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1474 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1479 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1481 GstRTSPStreamTransport *trans;
1483 GST_INFO ("%p: source %p timeout", stream, source);
1485 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1486 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1487 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1492 clear_tr_cache (GstRTSPStreamPrivate * priv)
1494 g_list_foreach (priv->tr_cache, (GFunc) g_object_unref, NULL);
1495 g_list_free (priv->tr_cache);
1496 priv->tr_cache = NULL;
1499 static GstFlowReturn
1500 handle_new_sample (GstAppSink * sink, gpointer user_data)
1502 GstRTSPStreamPrivate *priv;
1506 GstRTSPStream *stream;
1509 sample = gst_app_sink_pull_sample (sink);
1513 stream = (GstRTSPStream *) user_data;
1514 priv = stream->priv;
1515 buffer = gst_sample_get_buffer (sample);
1517 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1519 g_mutex_lock (&priv->lock);
1520 if (priv->tr_cache_cookie != priv->transports_cookie) {
1521 clear_tr_cache (priv);
1522 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1523 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1524 priv->tr_cache = g_list_prepend (priv->tr_cache, g_object_ref (tr));
1526 priv->tr_cache_cookie = priv->transports_cookie;
1528 g_mutex_unlock (&priv->lock);
1530 for (walk = priv->tr_cache; walk; walk = g_list_next (walk)) {
1531 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1534 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1536 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1539 gst_sample_unref (sample);
1544 static GstAppSinkCallbacks sink_cb = {
1545 NULL, /* not interested in EOS */
1546 NULL, /* not interested in preroll samples */
1551 get_rtp_encoder (GstRTSPStream * stream, guint session)
1553 GstRTSPStreamPrivate *priv = stream->priv;
1555 if (priv->srtpenc == NULL) {
1558 name = g_strdup_printf ("srtpenc_%u", session);
1559 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1562 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1564 return gst_object_ref (priv->srtpenc);
1568 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1570 GstRTSPStreamPrivate *priv = stream->priv;
1571 GstElement *oldenc, *enc;
1575 if (priv->idx != session)
1578 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1580 oldenc = priv->srtpenc;
1581 enc = get_rtp_encoder (stream, session);
1582 name = g_strdup_printf ("rtp_sink_%d", session);
1583 pad = gst_element_get_request_pad (enc, name);
1585 gst_object_unref (pad);
1588 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
1595 request_rtcp_encoder (GstElement * rtpbin, guint session,
1596 GstRTSPStream * stream)
1598 GstRTSPStreamPrivate *priv = stream->priv;
1599 GstElement *oldenc, *enc;
1603 if (priv->idx != session)
1606 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1608 oldenc = priv->srtpenc;
1609 enc = get_rtp_encoder (stream, session);
1610 name = g_strdup_printf ("rtcp_sink_%d", session);
1611 pad = gst_element_get_request_pad (enc, name);
1613 gst_object_unref (pad);
1616 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
1623 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1625 GstRTSPStreamPrivate *priv = stream->priv;
1628 GST_DEBUG ("request key %08x", ssrc);
1630 g_mutex_lock (&priv->lock);
1631 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1632 gst_caps_ref (caps);
1633 g_mutex_unlock (&priv->lock);
1639 request_rtcp_decoder (GstElement * rtpbin, guint session,
1640 GstRTSPStream * stream)
1642 GstRTSPStreamPrivate *priv = stream->priv;
1644 if (priv->idx != session)
1647 if (priv->srtpdec == NULL) {
1650 name = g_strdup_printf ("srtpdec_%u", session);
1651 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1654 g_signal_connect (priv->srtpdec, "request-key",
1655 (GCallback) request_key, stream);
1657 return gst_object_ref (priv->srtpdec);
1661 * gst_rtsp_stream_join_bin:
1662 * @stream: a #GstRTSPStream
1663 * @bin: (transfer none): a #GstBin to join
1664 * @rtpbin: (transfer none): a rtpbin element in @bin
1665 * @state: the target state of the new elements
1667 * Join the #GstBin @bin that contains the element @rtpbin.
1669 * @stream will link to @rtpbin, which must be inside @bin. The elements
1670 * added to @bin will be set to the state given in @state.
1672 * Returns: %TRUE on success.
1675 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1676 GstElement * rtpbin, GstState state)
1678 GstRTSPStreamPrivate *priv;
1682 GstPad *pad, *sinkpad, *selpad;
1683 GstPadLinkReturn ret;
1685 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1686 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1687 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1689 priv = stream->priv;
1691 g_mutex_lock (&priv->lock);
1692 if (priv->is_joined)
1695 /* create a session with the same index as the stream */
1698 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1700 if (!alloc_ports (stream))
1703 /* update the dscp qos field in the sinks */
1704 update_dscp_qos (stream);
1706 if (priv->profiles & GST_RTSP_PROFILE_SAVP
1707 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
1709 g_signal_connect (rtpbin, "request-rtp-encoder",
1710 (GCallback) request_rtp_encoder, stream);
1711 g_signal_connect (rtpbin, "request-rtcp-encoder",
1712 (GCallback) request_rtcp_encoder, stream);
1713 g_signal_connect (rtpbin, "request-rtcp-decoder",
1714 (GCallback) request_rtcp_decoder, stream);
1717 /* get a pad for sending RTP */
1718 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1719 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1721 /* link the RTP pad to the session manager, it should not really fail unless
1722 * this is not really an RTP pad */
1723 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1724 if (ret != GST_PAD_LINK_OK)
1727 /* get pads from the RTP session element for sending and receiving
1729 name = g_strdup_printf ("send_rtp_src_%u", idx);
1730 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1732 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1733 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1735 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1736 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1738 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1739 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1742 /* get the session */
1743 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1745 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1747 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1749 g_signal_connect (priv->session, "on-ssrc-active",
1750 (GCallback) on_ssrc_active, stream);
1751 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1753 g_signal_connect (priv->session, "on-bye-timeout",
1754 (GCallback) on_bye_timeout, stream);
1755 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1758 for (i = 0; i < 2; i++) {
1759 GstPad *teepad, *queuepad;
1760 /* For the sender we create this bit of pipeline for both
1761 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1762 * we need to add a queue before appsink to make the pipeline
1763 * not block. For the TCP case, we want to pump data to the
1764 * client as fast as possible anyway.
1766 * .--------. .-----. .---------.
1767 * | rtpbin | | tee | | udpsink |
1768 * | send->sink src->sink |
1769 * '--------' | | '---------'
1770 * | | .---------. .---------.
1771 * | | | queue | | appsink |
1772 * | src->sink src->sink |
1773 * '-----' '---------' '---------'
1775 * When only UDP is allowed, we skip the tee, queue and appsink and link the
1776 * udpsink directly to the session.
1779 gst_bin_add (bin, priv->udpsink[i]);
1780 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1782 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1783 /* make tee for RTP/RTCP */
1784 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1785 gst_bin_add (bin, priv->tee[i]);
1787 /* and link to rtpbin send pad */
1788 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1789 gst_pad_link (priv->send_src[i], pad);
1790 gst_object_unref (pad);
1792 /* link tee to udpsink */
1793 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1794 gst_pad_link (teepad, sinkpad);
1795 gst_object_unref (teepad);
1798 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1799 gst_bin_add (bin, priv->appqueue[i]);
1800 /* and link to tee */
1801 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1802 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1803 gst_pad_link (teepad, pad);
1804 gst_object_unref (pad);
1805 gst_object_unref (teepad);
1808 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1809 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1810 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1811 gst_bin_add (bin, priv->appsink[i]);
1812 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1813 &sink_cb, stream, NULL);
1814 /* and link to queue */
1815 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1816 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1817 gst_pad_link (queuepad, pad);
1818 gst_object_unref (pad);
1819 gst_object_unref (queuepad);
1821 /* else only udpsink needed, link it to the session */
1822 gst_pad_link (priv->send_src[i], sinkpad);
1824 gst_object_unref (sinkpad);
1826 /* For the receiver we create this bit of pipeline for both
1827 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1828 * and it is all funneled into the rtpbin receive pad.
1830 * .--------. .--------. .--------.
1831 * | udpsrc | | funnel | | rtpbin |
1832 * | src->sink src->sink |
1833 * '--------' | | '--------'
1837 * '--------' '--------'
1839 /* make funnel for the RTP/RTCP receivers */
1840 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1841 gst_bin_add (bin, priv->funnel[i]);
1843 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1844 gst_pad_link (pad, priv->recv_sink[i]);
1845 gst_object_unref (pad);
1847 if (priv->udpsrc_v4[i]) {
1848 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1850 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1851 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1853 gst_bin_add (bin, priv->udpsrc_v4[i]);
1855 /* and link to the funnel v4 */
1856 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1857 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1858 gst_pad_link (pad, selpad);
1859 gst_object_unref (pad);
1860 gst_object_unref (selpad);
1863 if (priv->udpsrc_v6[i]) {
1864 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1865 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1866 gst_bin_add (bin, priv->udpsrc_v6[i]);
1868 /* and link to the funnel v6 */
1869 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1870 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1871 gst_pad_link (pad, selpad);
1872 gst_object_unref (pad);
1873 gst_object_unref (selpad);
1876 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1877 /* make and add appsrc */
1878 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1879 gst_bin_add (bin, priv->appsrc[i]);
1880 /* and link to the funnel */
1881 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1882 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1883 gst_pad_link (pad, selpad);
1884 gst_object_unref (pad);
1885 gst_object_unref (selpad);
1888 /* check if we need to set to a special state */
1889 if (state != GST_STATE_NULL) {
1890 if (priv->udpsink[i])
1891 gst_element_set_state (priv->udpsink[i], state);
1892 if (priv->appsink[i])
1893 gst_element_set_state (priv->appsink[i], state);
1894 if (priv->appqueue[i])
1895 gst_element_set_state (priv->appqueue[i], state);
1897 gst_element_set_state (priv->tee[i], state);
1898 if (priv->funnel[i])
1899 gst_element_set_state (priv->funnel[i], state);
1900 if (priv->appsrc[i])
1901 gst_element_set_state (priv->appsrc[i], state);
1905 /* be notified of caps changes */
1906 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
1907 (GCallback) caps_notify, stream);
1909 priv->is_joined = TRUE;
1910 g_mutex_unlock (&priv->lock);
1917 g_mutex_unlock (&priv->lock);
1922 g_mutex_unlock (&priv->lock);
1923 GST_WARNING ("failed to allocate ports %u", idx);
1928 GST_WARNING ("failed to link stream %u", idx);
1929 gst_object_unref (priv->send_rtp_sink);
1930 priv->send_rtp_sink = NULL;
1931 g_mutex_unlock (&priv->lock);
1937 * gst_rtsp_stream_leave_bin:
1938 * @stream: a #GstRTSPStream
1939 * @bin: (transfer none): a #GstBin
1940 * @rtpbin: (transfer none): a rtpbin #GstElement
1942 * Remove the elements of @stream from @bin.
1944 * Return: %TRUE on success.
1947 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1948 GstElement * rtpbin)
1950 GstRTSPStreamPrivate *priv;
1954 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1955 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1956 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1958 priv = stream->priv;
1960 g_mutex_lock (&priv->lock);
1961 if (!priv->is_joined)
1962 goto was_not_joined;
1964 /* all transports must be removed by now */
1965 if (priv->transports != NULL)
1966 goto transports_not_removed;
1968 clear_tr_cache (priv);
1970 GST_INFO ("stream %p leaving bin", stream);
1972 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1973 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
1974 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1975 gst_object_unref (priv->send_rtp_sink);
1976 priv->send_rtp_sink = NULL;
1978 for (i = 0; i < 2; i++) {
1979 if (priv->udpsink[i])
1980 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1981 if (priv->appsink[i])
1982 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1983 if (priv->appqueue[i])
1984 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1986 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1987 if (priv->funnel[i])
1988 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1989 if (priv->appsrc[i])
1990 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1991 if (priv->udpsrc_v4[i]) {
1992 /* and set udpsrc to NULL now before removing */
1993 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1994 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1995 /* removing them should also nicely release the request
1996 * pads when they finalize */
1997 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1999 if (priv->udpsrc_v6[i]) {
2000 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2001 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2002 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2005 for (l = priv->transport_sources; l; l = l->next) {
2006 GstRTSPMulticastTransportSource *s = l->data;
2011 gst_element_set_locked_state (s->udpsrc[i], FALSE);
2012 gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
2013 gst_bin_remove (bin, s->udpsrc[i]);
2016 if (priv->udpsink[i])
2017 gst_bin_remove (bin, priv->udpsink[i]);
2018 if (priv->appsrc[i])
2019 gst_bin_remove (bin, priv->appsrc[i]);
2020 if (priv->appsink[i])
2021 gst_bin_remove (bin, priv->appsink[i]);
2022 if (priv->appqueue[i])
2023 gst_bin_remove (bin, priv->appqueue[i]);
2025 gst_bin_remove (bin, priv->tee[i]);
2026 if (priv->funnel[i])
2027 gst_bin_remove (bin, priv->funnel[i]);
2029 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2030 gst_object_unref (priv->recv_sink[i]);
2031 priv->recv_sink[i] = NULL;
2033 priv->udpsrc_v4[i] = NULL;
2034 priv->udpsrc_v6[i] = NULL;
2035 priv->udpsink[i] = NULL;
2036 priv->appsrc[i] = NULL;
2037 priv->appsink[i] = NULL;
2038 priv->appqueue[i] = NULL;
2039 priv->tee[i] = NULL;
2040 priv->funnel[i] = NULL;
2043 for (l = priv->transport_sources; l; l = l->next) {
2044 GstRTSPMulticastTransportSource *s = l->data;
2045 g_slice_free (GstRTSPMulticastTransportSource, s);
2047 g_list_free (priv->transport_sources);
2048 priv->transport_sources = NULL;
2050 gst_object_unref (priv->send_src[0]);
2051 priv->send_src[0] = NULL;
2053 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2054 gst_object_unref (priv->send_src[1]);
2055 priv->send_src[1] = NULL;
2057 g_object_unref (priv->session);
2058 priv->session = NULL;
2060 gst_caps_unref (priv->caps);
2064 gst_object_unref (priv->srtpenc);
2066 gst_object_unref (priv->srtpdec);
2068 priv->is_joined = FALSE;
2069 g_mutex_unlock (&priv->lock);
2075 g_mutex_unlock (&priv->lock);
2078 transports_not_removed:
2080 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2081 g_mutex_unlock (&priv->lock);
2087 * gst_rtsp_stream_get_rtpinfo:
2088 * @stream: a #GstRTSPStream
2089 * @rtptime: (allow-none): result RTP timestamp
2090 * @seq: (allow-none): result RTP seqnum
2091 * @clock_rate: (allow-none): the clock rate
2092 * @running_time: (allow-none): result running-time
2094 * Retrieve the current rtptime, seq and running-time. This is used to
2095 * construct a RTPInfo reply header.
2097 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2100 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2101 guint * rtptime, guint * seq, guint * clock_rate,
2102 GstClockTime * running_time)
2104 GstRTSPStreamPrivate *priv;
2105 GstStructure *stats;
2106 GObjectClass *payobjclass;
2108 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2110 priv = stream->priv;
2112 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2114 g_mutex_lock (&priv->lock);
2116 if (g_object_class_find_property (payobjclass, "stats")) {
2117 g_object_get (priv->payloader, "stats", &stats, NULL);
2122 gst_structure_get_uint (stats, "seqnum", seq);
2125 gst_structure_get_uint (stats, "timestamp", rtptime);
2128 gst_structure_get_clock_time (stats, "running-time", running_time);
2131 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2132 if (*clock_rate == 0 && running_time)
2133 *running_time = GST_CLOCK_TIME_NONE;
2135 gst_structure_free (stats);
2137 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2138 !g_object_class_find_property (payobjclass, "timestamp"))
2142 g_object_get (priv->payloader, "seqnum", seq, NULL);
2145 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2148 *running_time = GST_CLOCK_TIME_NONE;
2150 g_mutex_unlock (&priv->lock);
2157 GST_WARNING ("Could not get payloader stats");
2158 g_mutex_unlock (&priv->lock);
2164 * gst_rtsp_stream_get_caps:
2165 * @stream: a #GstRTSPStream
2167 * Retrieve the current caps of @stream.
2169 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2173 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2175 GstRTSPStreamPrivate *priv;
2178 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2180 priv = stream->priv;
2182 g_mutex_lock (&priv->lock);
2183 if ((result = priv->caps))
2184 gst_caps_ref (result);
2185 g_mutex_unlock (&priv->lock);
2191 * gst_rtsp_stream_recv_rtp:
2192 * @stream: a #GstRTSPStream
2193 * @buffer: (transfer full): a #GstBuffer
2195 * Handle an RTP buffer for the stream. This method is usually called when a
2196 * message has been received from a client using the TCP transport.
2198 * This function takes ownership of @buffer.
2200 * Returns: a GstFlowReturn.
2203 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2205 GstRTSPStreamPrivate *priv;
2207 GstElement *element;
2209 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2210 priv = stream->priv;
2211 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2212 g_return_val_if_fail (priv->is_joined, FALSE);
2214 g_mutex_lock (&priv->lock);
2215 if (priv->appsrc[0])
2216 element = gst_object_ref (priv->appsrc[0]);
2219 g_mutex_unlock (&priv->lock);
2222 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2223 gst_object_unref (element);
2231 * gst_rtsp_stream_recv_rtcp:
2232 * @stream: a #GstRTSPStream
2233 * @buffer: (transfer full): a #GstBuffer
2235 * Handle an RTCP buffer for the stream. This method is usually called when a
2236 * message has been received from a client using the TCP transport.
2238 * This function takes ownership of @buffer.
2240 * Returns: a GstFlowReturn.
2243 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2245 GstRTSPStreamPrivate *priv;
2247 GstElement *element;
2249 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2250 priv = stream->priv;
2251 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2253 if (!priv->is_joined) {
2254 gst_buffer_unref (buffer);
2255 return GST_FLOW_NOT_LINKED;
2257 g_mutex_lock (&priv->lock);
2258 if (priv->appsrc[1])
2259 element = gst_object_ref (priv->appsrc[1]);
2262 g_mutex_unlock (&priv->lock);
2265 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2266 gst_object_unref (element);
2269 gst_buffer_unref (buffer);
2274 /* must be called with lock */
2276 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
2279 GstRTSPStreamPrivate *priv = stream->priv;
2280 const GstRTSPTransport *tr;
2282 tr = gst_rtsp_stream_transport_get_transport (trans);
2284 switch (tr->lower_transport) {
2285 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2287 GstRTSPMulticastTransportSource *source;
2290 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[0])));
2295 GstPad *selpad, *pad;
2297 source = g_slice_new0 (GstRTSPMulticastTransportSource);
2298 source->transport = trans;
2300 for (i = 0; i < 2; i++) {
2302 g_strdup_printf ("udp://%s:%d", tr->destination,
2303 (i == 0) ? tr->port.min : tr->port.max);
2305 gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2308 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2310 gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
2311 gst_element_set_locked_state (source->udpsrc[i], TRUE);
2313 gst_bin_add (bin, source->udpsrc[i]);
2315 /* and link to the funnel v4 */
2316 source->selpad[i] = selpad =
2317 gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2318 pad = gst_element_get_static_pad (source->udpsrc[i], "src");
2319 gst_pad_link (pad, selpad);
2320 gst_object_unref (pad);
2321 gst_object_unref (selpad);
2323 gst_object_unref (bin);
2325 priv->transport_sources =
2326 g_list_prepend (priv->transport_sources, source);
2330 for (l = priv->transport_sources; l; l = l->next) {
2333 if (source->transport == trans) {
2334 priv->transport_sources =
2335 g_list_delete_link (priv->transport_sources, l);
2343 for (i = 0; i < 2; i++) {
2344 /* Will automatically unlink everything */
2345 gst_bin_remove (bin,
2346 GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
2348 gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
2349 gst_object_unref (source->udpsrc[i]);
2351 gst_element_release_request_pad (priv->funnel[i],
2355 g_slice_free (GstRTSPMulticastTransportSource, source);
2359 /* fall through for the generic case */
2361 case GST_RTSP_LOWER_TRANS_UDP:
2367 dest = tr->destination;
2368 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2373 min = tr->client_port.min;
2374 max = tr->client_port.max;
2379 GST_INFO ("setting ttl-mc %d", ttl);
2380 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
2381 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
2383 GST_INFO ("adding %s:%d-%d", dest, min, max);
2384 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
2385 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
2386 priv->transports = g_list_prepend (priv->transports, trans);
2388 GST_INFO ("removing %s:%d-%d", dest, min, max);
2389 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
2390 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
2391 priv->transports = g_list_remove (priv->transports, trans);
2393 priv->transports_cookie++;
2396 case GST_RTSP_LOWER_TRANS_TCP:
2398 GST_INFO ("adding TCP %s", tr->destination);
2399 priv->transports = g_list_prepend (priv->transports, trans);
2401 GST_INFO ("removing TCP %s", tr->destination);
2402 priv->transports = g_list_remove (priv->transports, trans);
2404 priv->transports_cookie++;
2407 goto unknown_transport;
2414 GST_INFO ("Unknown transport %d", tr->lower_transport);
2421 * gst_rtsp_stream_add_transport:
2422 * @stream: a #GstRTSPStream
2423 * @trans: (transfer none): a #GstRTSPStreamTransport
2425 * Add the transport in @trans to @stream. The media of @stream will
2426 * then also be send to the values configured in @trans.
2428 * @stream must be joined to a bin.
2430 * @trans must contain a valid #GstRTSPTransport.
2432 * Returns: %TRUE if @trans was added
2435 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
2436 GstRTSPStreamTransport * trans)
2438 GstRTSPStreamPrivate *priv;
2441 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2442 priv = stream->priv;
2443 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2444 g_return_val_if_fail (priv->is_joined, FALSE);
2446 g_mutex_lock (&priv->lock);
2447 res = update_transport (stream, trans, TRUE);
2448 g_mutex_unlock (&priv->lock);
2454 * gst_rtsp_stream_remove_transport:
2455 * @stream: a #GstRTSPStream
2456 * @trans: (transfer none): a #GstRTSPStreamTransport
2458 * Remove the transport in @trans from @stream. The media of @stream will
2459 * not be sent to the values configured in @trans.
2461 * @stream must be joined to a bin.
2463 * @trans must contain a valid #GstRTSPTransport.
2465 * Returns: %TRUE if @trans was removed
2468 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2469 GstRTSPStreamTransport * trans)
2471 GstRTSPStreamPrivate *priv;
2474 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2475 priv = stream->priv;
2476 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2477 g_return_val_if_fail (priv->is_joined, FALSE);
2479 g_mutex_lock (&priv->lock);
2480 res = update_transport (stream, trans, FALSE);
2481 g_mutex_unlock (&priv->lock);
2487 * gst_rtsp_stream_update_crypto:
2488 * @stream: a #GstRTSPStream
2490 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
2492 * Update the new crypto information for @ssrc in @stream. If information
2493 * for @ssrc did not exist, it will be added. If information
2494 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
2495 * be removed from @stream.
2497 * Returns: %TRUE if @crypto could be updated
2500 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
2501 guint ssrc, GstCaps * crypto)
2503 GstRTSPStreamPrivate *priv;
2505 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2506 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
2508 priv = stream->priv;
2510 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
2512 g_mutex_lock (&priv->lock);
2514 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
2515 gst_caps_ref (crypto));
2517 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
2518 g_mutex_unlock (&priv->lock);
2524 * gst_rtsp_stream_get_rtp_socket:
2525 * @stream: a #GstRTSPStream
2526 * @family: the socket family
2528 * Get the RTP socket from @stream for a @family.
2530 * @stream must be joined to a bin.
2532 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
2533 * socket could be allocated for @family. Unref after usage
2536 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
2538 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2542 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2543 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2544 family == G_SOCKET_FAMILY_IPV6, NULL);
2545 g_return_val_if_fail (priv->udpsink[0], NULL);
2547 if (family == G_SOCKET_FAMILY_IPV6)
2552 g_object_get (priv->udpsink[0], name, &socket, NULL);
2558 * gst_rtsp_stream_get_rtcp_socket:
2559 * @stream: a #GstRTSPStream
2560 * @family: the socket family
2562 * Get the RTCP socket from @stream for a @family.
2564 * @stream must be joined to a bin.
2566 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
2567 * socket could be allocated for @family. Unref after usage
2570 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
2572 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2576 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2577 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2578 family == G_SOCKET_FAMILY_IPV6, NULL);
2579 g_return_val_if_fail (priv->udpsink[1], NULL);
2581 if (family == G_SOCKET_FAMILY_IPV6)
2586 g_object_get (priv->udpsink[1], name, &socket, NULL);
2592 * gst_rtsp_stream_set_seqnum:
2593 * @stream: a #GstRTSPStream
2594 * @seqnum: a new sequence number
2596 * Configure the sequence number in the payloader of @stream to @seqnum.
2599 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
2601 GstRTSPStreamPrivate *priv;
2603 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2605 priv = stream->priv;
2607 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
2611 * gst_rtsp_stream_get_seqnum:
2612 * @stream: a #GstRTSPStream
2614 * Get the configured sequence number in the payloader of @stream.
2616 * Returns: the sequence number of the payloader.
2619 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
2621 GstRTSPStreamPrivate *priv;
2624 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
2626 priv = stream->priv;
2628 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
2634 * gst_rtsp_stream_transport_filter:
2635 * @stream: a #GstRTSPStream
2636 * @func: (scope call) (allow-none): a callback
2637 * @user_data: (closure): user data passed to @func
2639 * Call @func for each transport managed by @stream. The result value of @func
2640 * determines what happens to the transport. @func will be called with @stream
2641 * locked so no further actions on @stream can be performed from @func.
2643 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
2646 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
2648 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
2649 * will also be added with an additional ref to the result #GList of this
2652 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
2654 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
2655 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
2656 * element in the #GList should be unreffed before the list is freed.
2659 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
2660 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
2662 GstRTSPStreamPrivate *priv;
2663 GList *result, *walk, *next;
2664 GHashTable *visited;
2667 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2669 priv = stream->priv;
2673 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
2675 g_mutex_lock (&priv->lock);
2677 cookie = priv->transports_cookie;
2678 for (walk = priv->transports; walk; walk = next) {
2679 GstRTSPStreamTransport *trans = walk->data;
2680 GstRTSPFilterResult res;
2683 next = g_list_next (walk);
2686 /* only visit each transport once */
2687 if (g_hash_table_contains (visited, trans))
2690 g_hash_table_add (visited, g_object_ref (trans));
2691 g_mutex_unlock (&priv->lock);
2693 res = func (stream, trans, user_data);
2695 g_mutex_lock (&priv->lock);
2697 res = GST_RTSP_FILTER_REF;
2699 changed = (cookie != priv->transports_cookie);
2702 case GST_RTSP_FILTER_REMOVE:
2703 update_transport (stream, trans, FALSE);
2705 case GST_RTSP_FILTER_REF:
2706 result = g_list_prepend (result, g_object_ref (trans));
2708 case GST_RTSP_FILTER_KEEP:
2715 g_mutex_unlock (&priv->lock);
2718 g_hash_table_unref (visited);
2723 static GstPadProbeReturn
2724 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2726 GstRTSPStreamPrivate *priv;
2727 GstRTSPStream *stream;
2730 priv = stream->priv;
2732 GST_DEBUG_OBJECT (pad, "now blocking");
2734 g_mutex_lock (&priv->lock);
2735 priv->blocking = TRUE;
2736 g_mutex_unlock (&priv->lock);
2738 gst_element_post_message (priv->payloader,
2739 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
2740 gst_structure_new_empty ("GstRTSPStreamBlocking")));
2742 return GST_PAD_PROBE_OK;
2746 * gst_rtsp_stream_set_blocked:
2747 * @stream: a #GstRTSPStream
2748 * @blocked: boolean indicating we should block or unblock
2750 * Blocks or unblocks the dataflow on @stream.
2752 * Returns: %TRUE on success
2755 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
2757 GstRTSPStreamPrivate *priv;
2759 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2761 priv = stream->priv;
2763 g_mutex_lock (&priv->lock);
2765 priv->blocking = FALSE;
2766 if (priv->blocked_id == 0) {
2767 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
2768 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
2769 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
2770 g_object_ref (stream), g_object_unref);
2773 if (priv->blocked_id != 0) {
2774 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
2775 priv->blocked_id = 0;
2776 priv->blocking = FALSE;
2779 g_mutex_unlock (&priv->lock);
2785 * gst_rtsp_stream_is_blocking:
2786 * @stream: a #GstRTSPStream
2788 * Check if @stream is blocking on a #GstBuffer.
2790 * Returns: %TRUE if @stream is blocking
2793 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
2795 GstRTSPStreamPrivate *priv;
2798 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2800 priv = stream->priv;
2802 g_mutex_lock (&priv->lock);
2803 result = priv->blocking;
2804 g_mutex_unlock (&priv->lock);
2810 * gst_rtsp_stream_query_position:
2811 * @stream: a #GstRTSPStream
2813 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
2814 * the RTP parts of the pipeline and not the RTCP parts.
2816 * Returns: %TRUE if the position could be queried
2819 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
2821 GstRTSPStreamPrivate *priv;
2825 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2827 priv = stream->priv;
2829 g_mutex_lock (&priv->lock);
2830 if ((sink = priv->udpsink[0]))
2831 gst_object_ref (sink);
2832 g_mutex_unlock (&priv->lock);
2837 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
2838 gst_object_unref (sink);
2844 * gst_rtsp_stream_query_stop:
2845 * @stream: a #GstRTSPStream
2847 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
2848 * the RTP parts of the pipeline and not the RTCP parts.
2850 * Returns: %TRUE if the stop could be queried
2853 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
2855 GstRTSPStreamPrivate *priv;
2860 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2862 priv = stream->priv;
2864 g_mutex_lock (&priv->lock);
2865 if ((sink = priv->udpsink[0]))
2866 gst_object_ref (sink);
2867 g_mutex_unlock (&priv->lock);
2872 query = gst_query_new_segment (GST_FORMAT_TIME);
2873 if ((ret = gst_element_query (sink, query))) {
2876 gst_query_parse_segment (query, NULL, &format, NULL, stop);
2877 if (format != GST_FORMAT_TIME)
2880 gst_query_unref (query);
2881 gst_object_unref (sink);