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];
117 GstClockTime rtx_time;
119 /* server ports for sending/receiving over ipv4 */
120 GstRTSPRange server_port_v4;
121 GstRTSPAddress *server_addr_v4;
124 /* server ports for sending/receiving over ipv6 */
125 GstRTSPRange server_port_v6;
126 GstRTSPAddress *server_addr_v6;
129 /* multicast addresses */
130 GstRTSPAddressPool *pool;
131 GstRTSPAddress *addr_v4;
132 GstRTSPAddress *addr_v6;
134 /* the caps of the stream */
138 /* transports we stream to */
141 guint transports_cookie;
143 GList *tr_cache_rtcp;
144 guint tr_cache_cookie_rtp;
145 guint tr_cache_cookie_rtcp;
148 /* UDP sources for UDP multicast transports */
149 GList *transport_sources;
153 /* stream blocking */
158 #define DEFAULT_CONTROL NULL
159 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
160 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
161 GST_RTSP_LOWER_TRANS_TCP
174 SIGNAL_NEW_RTP_ENCODER,
175 SIGNAL_NEW_RTCP_ENCODER,
179 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
180 #define GST_CAT_DEFAULT rtsp_stream_debug
182 static GQuark ssrc_stream_map_key;
184 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
185 GValue * value, GParamSpec * pspec);
186 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
187 const GValue * value, GParamSpec * pspec);
189 static void gst_rtsp_stream_finalize (GObject * obj);
191 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
193 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
196 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
198 GObjectClass *gobject_class;
200 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
202 gobject_class = G_OBJECT_CLASS (klass);
204 gobject_class->get_property = gst_rtsp_stream_get_property;
205 gobject_class->set_property = gst_rtsp_stream_set_property;
206 gobject_class->finalize = gst_rtsp_stream_finalize;
208 g_object_class_install_property (gobject_class, PROP_CONTROL,
209 g_param_spec_string ("control", "Control",
210 "The control string for this stream", DEFAULT_CONTROL,
211 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
213 g_object_class_install_property (gobject_class, PROP_PROFILES,
214 g_param_spec_flags ("profiles", "Profiles",
215 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
216 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
218 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
219 g_param_spec_flags ("protocols", "Protocols",
220 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
221 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
223 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
224 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
225 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
226 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
228 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
229 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
230 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
231 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
233 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
235 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
239 gst_rtsp_stream_init (GstRTSPStream * stream)
241 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
243 GST_DEBUG ("new stream %p", stream);
248 priv->control = g_strdup (DEFAULT_CONTROL);
249 priv->profiles = DEFAULT_PROFILES;
250 priv->protocols = DEFAULT_PROTOCOLS;
252 g_mutex_init (&priv->lock);
254 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
255 NULL, (GDestroyNotify) gst_caps_unref);
259 gst_rtsp_stream_finalize (GObject * obj)
261 GstRTSPStream *stream;
262 GstRTSPStreamPrivate *priv;
264 stream = GST_RTSP_STREAM (obj);
267 GST_DEBUG ("finalize stream %p", stream);
269 /* we really need to be unjoined now */
270 g_return_if_fail (!priv->is_joined);
273 gst_rtsp_address_free (priv->addr_v4);
275 gst_rtsp_address_free (priv->addr_v6);
276 if (priv->server_addr_v4)
277 gst_rtsp_address_free (priv->server_addr_v4);
278 if (priv->server_addr_v6)
279 gst_rtsp_address_free (priv->server_addr_v6);
281 g_object_unref (priv->pool);
283 g_object_unref (priv->rtxsend);
285 gst_object_unref (priv->payloader);
286 gst_object_unref (priv->srcpad);
287 g_free (priv->control);
288 g_mutex_clear (&priv->lock);
290 g_hash_table_unref (priv->keys);
292 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
296 gst_rtsp_stream_get_property (GObject * object, guint propid,
297 GValue * value, GParamSpec * pspec)
299 GstRTSPStream *stream = GST_RTSP_STREAM (object);
303 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
306 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
309 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
312 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
317 gst_rtsp_stream_set_property (GObject * object, guint propid,
318 const GValue * value, GParamSpec * pspec)
320 GstRTSPStream *stream = GST_RTSP_STREAM (object);
324 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
327 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
330 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
333 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
338 * gst_rtsp_stream_new:
341 * @payloader: a #GstElement
343 * Create a new media stream with index @idx that handles RTP data on
344 * @srcpad and has a payloader element @payloader.
346 * Returns: (transfer full): a new #GstRTSPStream
349 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
351 GstRTSPStreamPrivate *priv;
352 GstRTSPStream *stream;
354 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
355 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
356 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
358 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
361 priv->payloader = gst_object_ref (payloader);
362 priv->srcpad = gst_object_ref (srcpad);
368 * gst_rtsp_stream_get_index:
369 * @stream: a #GstRTSPStream
371 * Get the stream index.
373 * Return: the stream index.
376 gst_rtsp_stream_get_index (GstRTSPStream * stream)
378 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
380 return stream->priv->idx;
384 * gst_rtsp_stream_get_pt:
385 * @stream: a #GstRTSPStream
387 * Get the stream payload type.
389 * Return: the stream payload type.
392 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
394 GstRTSPStreamPrivate *priv;
397 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
401 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
407 * gst_rtsp_stream_get_srcpad:
408 * @stream: a #GstRTSPStream
410 * Get the srcpad associated with @stream.
412 * Returns: (transfer full): the srcpad. Unref after usage.
415 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
417 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
419 return gst_object_ref (stream->priv->srcpad);
423 * gst_rtsp_stream_get_control:
424 * @stream: a #GstRTSPStream
426 * Get the control string to identify this stream.
428 * Returns: (transfer full): the control string. g_free() after usage.
431 gst_rtsp_stream_get_control (GstRTSPStream * stream)
433 GstRTSPStreamPrivate *priv;
436 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
440 g_mutex_lock (&priv->lock);
441 if ((result = g_strdup (priv->control)) == NULL)
442 result = g_strdup_printf ("stream=%u", priv->idx);
443 g_mutex_unlock (&priv->lock);
449 * gst_rtsp_stream_set_control:
450 * @stream: a #GstRTSPStream
451 * @control: a control string
453 * Set the control string in @stream.
456 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
458 GstRTSPStreamPrivate *priv;
460 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
464 g_mutex_lock (&priv->lock);
465 g_free (priv->control);
466 priv->control = g_strdup (control);
467 g_mutex_unlock (&priv->lock);
471 * gst_rtsp_stream_has_control:
472 * @stream: a #GstRTSPStream
473 * @control: a control string
475 * Check if @stream has the control string @control.
477 * Returns: %TRUE is @stream has @control as the control string
480 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
482 GstRTSPStreamPrivate *priv;
485 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
489 g_mutex_lock (&priv->lock);
491 res = (g_strcmp0 (priv->control, control) == 0);
495 if (sscanf (control, "stream=%u", &streamid) > 0)
496 res = (streamid == priv->idx);
500 g_mutex_unlock (&priv->lock);
506 * gst_rtsp_stream_set_mtu:
507 * @stream: a #GstRTSPStream
510 * Configure the mtu in the payloader of @stream to @mtu.
513 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
515 GstRTSPStreamPrivate *priv;
517 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
521 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
523 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
527 * gst_rtsp_stream_get_mtu:
528 * @stream: a #GstRTSPStream
530 * Get the configured MTU in the payloader of @stream.
532 * Returns: the MTU of the payloader.
535 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
537 GstRTSPStreamPrivate *priv;
540 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
544 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
549 /* Update the dscp qos property on the udp sinks */
551 update_dscp_qos (GstRTSPStream * stream)
553 GstRTSPStreamPrivate *priv;
555 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
559 if (priv->udpsink[0]) {
560 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
564 if (priv->udpsink[1]) {
565 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
571 * gst_rtsp_stream_set_dscp_qos:
572 * @stream: a #GstRTSPStream
573 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
575 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
578 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
580 GstRTSPStreamPrivate *priv;
582 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
586 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
588 if (dscp_qos < -1 || dscp_qos > 63) {
589 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
593 priv->dscp_qos = dscp_qos;
595 update_dscp_qos (stream);
599 * gst_rtsp_stream_get_dscp_qos:
600 * @stream: a #GstRTSPStream
602 * Get the configured DSCP QoS in of the outgoing sockets.
604 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
607 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
609 GstRTSPStreamPrivate *priv;
611 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
615 return priv->dscp_qos;
619 * gst_rtsp_stream_is_transport_supported:
620 * @stream: a #GstRTSPStream
621 * @transport: (transfer none): a #GstRTSPTransport
623 * Check if @transport can be handled by stream
625 * Returns: %TRUE if @transport can be handled by @stream.
628 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
629 GstRTSPTransport * transport)
631 GstRTSPStreamPrivate *priv;
633 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
637 g_mutex_lock (&priv->lock);
638 if (transport->trans != GST_RTSP_TRANS_RTP)
639 goto unsupported_transmode;
641 if (!(transport->profile & priv->profiles))
642 goto unsupported_profile;
644 if (!(transport->lower_transport & priv->protocols))
645 goto unsupported_ltrans;
647 g_mutex_unlock (&priv->lock);
652 unsupported_transmode:
654 GST_DEBUG ("unsupported transport mode %d", transport->trans);
655 g_mutex_unlock (&priv->lock);
660 GST_DEBUG ("unsupported profile %d", transport->profile);
661 g_mutex_unlock (&priv->lock);
666 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
667 g_mutex_unlock (&priv->lock);
673 * gst_rtsp_stream_set_profiles:
674 * @stream: a #GstRTSPStream
675 * @profiles: the new profiles
677 * Configure the allowed profiles for @stream.
680 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
682 GstRTSPStreamPrivate *priv;
684 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
688 g_mutex_lock (&priv->lock);
689 priv->profiles = profiles;
690 g_mutex_unlock (&priv->lock);
694 * gst_rtsp_stream_get_profiles:
695 * @stream: a #GstRTSPStream
697 * Get the allowed profiles of @stream.
699 * Returns: a #GstRTSPProfile
702 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
704 GstRTSPStreamPrivate *priv;
707 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
711 g_mutex_lock (&priv->lock);
712 res = priv->profiles;
713 g_mutex_unlock (&priv->lock);
719 * gst_rtsp_stream_set_protocols:
720 * @stream: a #GstRTSPStream
721 * @protocols: the new flags
723 * Configure the allowed lower transport for @stream.
726 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
727 GstRTSPLowerTrans protocols)
729 GstRTSPStreamPrivate *priv;
731 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
735 g_mutex_lock (&priv->lock);
736 priv->protocols = protocols;
737 g_mutex_unlock (&priv->lock);
741 * gst_rtsp_stream_get_protocols:
742 * @stream: a #GstRTSPStream
744 * Get the allowed protocols of @stream.
746 * Returns: a #GstRTSPLowerTrans
749 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
751 GstRTSPStreamPrivate *priv;
752 GstRTSPLowerTrans res;
754 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
755 GST_RTSP_LOWER_TRANS_UNKNOWN);
759 g_mutex_lock (&priv->lock);
760 res = priv->protocols;
761 g_mutex_unlock (&priv->lock);
767 * gst_rtsp_stream_set_address_pool:
768 * @stream: a #GstRTSPStream
769 * @pool: (transfer none): a #GstRTSPAddressPool
771 * configure @pool to be used as the address pool of @stream.
774 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
775 GstRTSPAddressPool * pool)
777 GstRTSPStreamPrivate *priv;
778 GstRTSPAddressPool *old;
780 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
784 GST_LOG_OBJECT (stream, "set address pool %p", pool);
786 g_mutex_lock (&priv->lock);
787 if ((old = priv->pool) != pool)
788 priv->pool = pool ? g_object_ref (pool) : NULL;
791 g_mutex_unlock (&priv->lock);
794 g_object_unref (old);
798 * gst_rtsp_stream_get_address_pool:
799 * @stream: a #GstRTSPStream
801 * Get the #GstRTSPAddressPool used as the address pool of @stream.
803 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
807 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
809 GstRTSPStreamPrivate *priv;
810 GstRTSPAddressPool *result;
812 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
816 g_mutex_lock (&priv->lock);
817 if ((result = priv->pool))
818 g_object_ref (result);
819 g_mutex_unlock (&priv->lock);
825 * gst_rtsp_stream_get_multicast_address:
826 * @stream: a #GstRTSPStream
827 * @family: the #GSocketFamily
829 * Get the multicast address of @stream for @family.
831 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
832 * or %NULL when no address could be allocated. gst_rtsp_address_free()
836 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
837 GSocketFamily family)
839 GstRTSPStreamPrivate *priv;
840 GstRTSPAddress *result;
841 GstRTSPAddress **addrp;
842 GstRTSPAddressFlags flags;
844 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
848 if (family == G_SOCKET_FAMILY_IPV6) {
849 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
850 addrp = &priv->addr_v6;
852 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
853 addrp = &priv->addr_v4;
856 g_mutex_lock (&priv->lock);
857 if (*addrp == NULL) {
858 if (priv->pool == NULL)
861 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
863 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
867 result = gst_rtsp_address_copy (*addrp);
868 g_mutex_unlock (&priv->lock);
875 GST_ERROR_OBJECT (stream, "no address pool specified");
876 g_mutex_unlock (&priv->lock);
881 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
882 g_mutex_unlock (&priv->lock);
888 * gst_rtsp_stream_reserve_address:
889 * @stream: a #GstRTSPStream
890 * @address: an address
895 * Reserve @address and @port as the address and port of @stream.
897 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
898 * the address could be reserved. gst_rtsp_address_free() after usage.
901 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
902 const gchar * address, guint port, guint n_ports, guint ttl)
904 GstRTSPStreamPrivate *priv;
905 GstRTSPAddress *result;
907 GSocketFamily family;
908 GstRTSPAddress **addrp;
910 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
911 g_return_val_if_fail (address != NULL, NULL);
912 g_return_val_if_fail (port > 0, NULL);
913 g_return_val_if_fail (n_ports > 0, NULL);
914 g_return_val_if_fail (ttl > 0, NULL);
918 addr = g_inet_address_new_from_string (address);
920 GST_ERROR ("failed to get inet addr from %s", address);
921 family = G_SOCKET_FAMILY_IPV4;
923 family = g_inet_address_get_family (addr);
924 g_object_unref (addr);
927 if (family == G_SOCKET_FAMILY_IPV6)
928 addrp = &priv->addr_v6;
930 addrp = &priv->addr_v4;
932 g_mutex_lock (&priv->lock);
933 if (*addrp == NULL) {
934 GstRTSPAddressPoolResult res;
936 if (priv->pool == NULL)
939 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
940 port, n_ports, ttl, addrp);
941 if (res != GST_RTSP_ADDRESS_POOL_OK)
944 if (strcmp ((*addrp)->address, address) ||
945 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
946 (*addrp)->ttl != ttl)
947 goto different_address;
949 result = gst_rtsp_address_copy (*addrp);
950 g_mutex_unlock (&priv->lock);
957 GST_ERROR_OBJECT (stream, "no address pool specified");
958 g_mutex_unlock (&priv->lock);
963 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
965 g_mutex_unlock (&priv->lock);
970 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
971 " reserved", address);
972 g_mutex_unlock (&priv->lock);
978 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
979 GSocketFamily family, GstElement * udpsrc_out[2],
980 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
981 GstRTSPAddress ** server_addr_out)
983 GstStateChangeReturn ret;
984 GstElement *udpsrc0, *udpsrc1;
985 GstElement *udpsink0, *udpsink1;
986 GSocket *rtp_socket = NULL;
987 GSocket *rtcp_socket;
988 gint tmp_rtp, tmp_rtcp;
990 gint rtpport, rtcpport;
991 GList *rejected_addresses = NULL;
992 GstRTSPAddress *addr = NULL;
993 GInetAddress *inetaddr = NULL;
994 GSocketAddress *rtp_sockaddr = NULL;
995 GSocketAddress *rtcp_sockaddr = NULL;
996 const gchar *multisink_socket;
998 if (family == G_SOCKET_FAMILY_IPV6)
999 multisink_socket = "socket-v6";
1001 multisink_socket = "socket";
1009 /* Start with random port */
1012 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1013 G_SOCKET_PROTOCOL_UDP, NULL);
1015 goto no_udp_protocol;
1017 if (*server_addr_out)
1018 gst_rtsp_address_free (*server_addr_out);
1020 /* try to allocate 2 UDP ports, the RTP port should be an even
1021 * number and the RTCP port should be the next (uneven) port */
1024 if (rtp_socket == NULL) {
1025 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1026 G_SOCKET_PROTOCOL_UDP, NULL);
1028 goto no_udp_protocol;
1031 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
1032 GstRTSPAddressFlags flags;
1035 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1037 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1038 if (family == G_SOCKET_FAMILY_IPV6)
1039 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1041 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1043 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1048 tmp_rtp = addr->port;
1050 g_clear_object (&inetaddr);
1051 inetaddr = g_inet_address_new_from_string (addr->address);
1059 if (inetaddr == NULL)
1060 inetaddr = g_inet_address_new_any (family);
1063 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1064 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1065 g_object_unref (rtp_sockaddr);
1068 g_object_unref (rtp_sockaddr);
1070 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1071 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1072 g_clear_object (&rtp_sockaddr);
1077 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1078 g_object_unref (rtp_sockaddr);
1080 /* check if port is even */
1081 if ((tmp_rtp & 1) != 0) {
1082 /* port not even, close and allocate another */
1084 g_clear_object (&rtp_socket);
1089 tmp_rtcp = tmp_rtp + 1;
1091 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1092 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1093 g_object_unref (rtcp_sockaddr);
1094 g_clear_object (&rtp_socket);
1097 g_object_unref (rtcp_sockaddr);
1099 g_clear_object (&inetaddr);
1101 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1102 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1104 if (udpsrc0 == NULL || udpsrc1 == NULL)
1105 goto no_udp_protocol;
1107 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1108 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1110 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1111 if (ret == GST_STATE_CHANGE_FAILURE)
1113 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1114 if (ret == GST_STATE_CHANGE_FAILURE)
1117 /* all fine, do port check */
1118 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1119 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1121 /* this should not happen... */
1122 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1126 udpsink0 = udpsink_out[0];
1128 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1131 goto no_udp_protocol;
1133 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1134 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
1137 udpsink1 = udpsink_out[1];
1139 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1142 goto no_udp_protocol;
1144 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1145 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1146 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
1148 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1149 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
1150 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1151 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1152 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1153 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1154 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1155 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1157 /* we keep these elements, we will further configure them when the
1158 * client told us to really use the UDP ports. */
1159 udpsrc_out[0] = udpsrc0;
1160 udpsrc_out[1] = udpsrc1;
1161 udpsink_out[0] = udpsink0;
1162 udpsink_out[1] = udpsink1;
1164 server_port_out->min = rtpport;
1165 server_port_out->max = rtcpport;
1167 *server_addr_out = addr;
1168 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1170 g_object_unref (rtp_socket);
1171 g_object_unref (rtcp_socket);
1199 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1200 gst_object_unref (udpsrc0);
1203 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1204 gst_object_unref (udpsrc1);
1207 gst_element_set_state (udpsink0, GST_STATE_NULL);
1208 gst_object_unref (udpsink0);
1211 g_object_unref (inetaddr);
1212 g_list_free_full (rejected_addresses,
1213 (GDestroyNotify) gst_rtsp_address_free);
1215 gst_rtsp_address_free (addr);
1217 g_object_unref (rtp_socket);
1219 g_object_unref (rtcp_socket);
1224 /* must be called with lock */
1226 alloc_ports (GstRTSPStream * stream)
1228 GstRTSPStreamPrivate *priv = stream->priv;
1230 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1231 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1232 &priv->server_port_v4, &priv->server_addr_v4);
1234 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1235 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1236 &priv->server_port_v6, &priv->server_addr_v6);
1238 return priv->have_ipv4 || priv->have_ipv6;
1242 * gst_rtsp_stream_get_server_port:
1243 * @stream: a #GstRTSPStream
1244 * @server_port: (out): result server port
1245 * @family: the port family to get
1247 * Fill @server_port with the port pair used by the server. This function can
1248 * only be called when @stream has been joined.
1251 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1252 GstRTSPRange * server_port, GSocketFamily family)
1254 GstRTSPStreamPrivate *priv;
1256 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1257 priv = stream->priv;
1258 g_return_if_fail (priv->is_joined);
1260 g_mutex_lock (&priv->lock);
1261 if (family == G_SOCKET_FAMILY_IPV4) {
1263 *server_port = priv->server_port_v4;
1266 *server_port = priv->server_port_v6;
1268 g_mutex_unlock (&priv->lock);
1272 * gst_rtsp_stream_get_rtpsession:
1273 * @stream: a #GstRTSPStream
1275 * Get the RTP session of this stream.
1277 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1280 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1282 GstRTSPStreamPrivate *priv;
1285 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1287 priv = stream->priv;
1289 g_mutex_lock (&priv->lock);
1290 if ((session = priv->session))
1291 g_object_ref (session);
1292 g_mutex_unlock (&priv->lock);
1298 * gst_rtsp_stream_get_ssrc:
1299 * @stream: a #GstRTSPStream
1300 * @ssrc: (out): result ssrc
1302 * Get the SSRC used by the RTP session of this stream. This function can only
1303 * be called when @stream has been joined.
1306 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1308 GstRTSPStreamPrivate *priv;
1310 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1311 priv = stream->priv;
1312 g_return_if_fail (priv->is_joined);
1314 g_mutex_lock (&priv->lock);
1315 if (ssrc && priv->session)
1316 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1317 g_mutex_unlock (&priv->lock);
1321 * gst_rtsp_stream_set_retransmission_time:
1322 * @stream: a #GstRTSPStream
1323 * @time: a #GstClockTime
1325 * Set the amount of time to store retransmission packets.
1328 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1331 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1333 g_mutex_lock (&stream->priv->lock);
1334 stream->priv->rtx_time = time;
1335 if (stream->priv->rtxsend)
1336 g_object_set (stream->priv->rtxsend, "max-size-time",
1337 GST_TIME_AS_MSECONDS (time), NULL);
1338 g_mutex_unlock (&stream->priv->lock);
1342 * gst_rtsp_media_get_retransmission_time:
1343 * @media: a #GstRTSPMedia
1345 * Get the amount of time to store retransmission data.
1347 * Returns: the amount of time to store retransmission data.
1350 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1354 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1356 g_mutex_lock (&stream->priv->lock);
1357 ret = stream->priv->rtx_time;
1358 g_mutex_unlock (&stream->priv->lock);
1364 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1366 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1368 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1370 g_mutex_lock (&stream->priv->lock);
1371 stream->priv->rtx_pt = rtx_pt;
1372 if (stream->priv->rtxsend) {
1373 guint pt = gst_rtsp_stream_get_pt (stream);
1374 gchar *pt_s = g_strdup_printf ("%d", pt);
1375 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1376 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1377 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1379 gst_structure_free (rtx_pt_map);
1381 g_mutex_unlock (&stream->priv->lock);
1385 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1389 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1391 g_mutex_lock (&stream->priv->lock);
1392 rtx_pt = stream->priv->rtx_pt;
1393 g_mutex_unlock (&stream->priv->lock);
1398 /* executed from streaming thread */
1400 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1402 GstRTSPStreamPrivate *priv = stream->priv;
1403 GstCaps *newcaps, *oldcaps;
1405 newcaps = gst_pad_get_current_caps (pad);
1407 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1410 g_mutex_lock (&priv->lock);
1411 oldcaps = priv->caps;
1412 priv->caps = newcaps;
1413 g_mutex_unlock (&priv->lock);
1416 gst_caps_unref (oldcaps);
1420 dump_structure (const GstStructure * s)
1424 sstr = gst_structure_to_string (s);
1425 GST_INFO ("structure: %s", sstr);
1429 static GstRTSPStreamTransport *
1430 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1432 GstRTSPStreamPrivate *priv = stream->priv;
1434 GstRTSPStreamTransport *result = NULL;
1439 if (rtcp_from == NULL)
1442 tmp = g_strrstr (rtcp_from, ":");
1446 port = atoi (tmp + 1);
1447 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1449 g_mutex_lock (&priv->lock);
1450 GST_INFO ("finding %s:%d in %d transports", dest, port,
1451 g_list_length (priv->transports));
1453 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1454 GstRTSPStreamTransport *trans = walk->data;
1455 const GstRTSPTransport *tr;
1458 tr = gst_rtsp_stream_transport_get_transport (trans);
1460 min = tr->client_port.min;
1461 max = tr->client_port.max;
1463 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1469 g_object_ref (result);
1470 g_mutex_unlock (&priv->lock);
1477 static GstRTSPStreamTransport *
1478 check_transport (GObject * source, GstRTSPStream * stream)
1480 GstStructure *stats;
1481 GstRTSPStreamTransport *trans;
1483 /* see if we have a stream to match with the origin of the RTCP packet */
1484 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1485 if (trans == NULL) {
1486 g_object_get (source, "stats", &stats, NULL);
1488 const gchar *rtcp_from;
1490 dump_structure (stats);
1492 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1493 if ((trans = find_transport (stream, rtcp_from))) {
1494 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1496 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1499 gst_structure_free (stats);
1507 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1509 GstRTSPStreamTransport *trans;
1511 GST_INFO ("%p: new source %p", stream, source);
1513 trans = check_transport (source, stream);
1516 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1520 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1522 GST_INFO ("%p: new SDES %p", stream, source);
1526 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1528 GstRTSPStreamTransport *trans;
1530 trans = check_transport (source, stream);
1533 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1534 gst_rtsp_stream_transport_keep_alive (trans);
1538 GstStructure *stats;
1539 g_object_get (source, "stats", &stats, NULL);
1541 dump_structure (stats);
1542 gst_structure_free (stats);
1549 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1551 GST_INFO ("%p: source %p bye", stream, source);
1555 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1557 GstRTSPStreamTransport *trans;
1559 GST_INFO ("%p: source %p bye timeout", stream, source);
1561 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1562 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1563 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1568 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1570 GstRTSPStreamTransport *trans;
1572 GST_INFO ("%p: source %p timeout", stream, source);
1574 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1575 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1576 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1581 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1584 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1585 g_list_free (priv->tr_cache_rtp);
1586 priv->tr_cache_rtp = NULL;
1588 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1589 g_list_free (priv->tr_cache_rtcp);
1590 priv->tr_cache_rtcp = NULL;
1594 static GstFlowReturn
1595 handle_new_sample (GstAppSink * sink, gpointer user_data)
1597 GstRTSPStreamPrivate *priv;
1601 GstRTSPStream *stream;
1604 sample = gst_app_sink_pull_sample (sink);
1608 stream = (GstRTSPStream *) user_data;
1609 priv = stream->priv;
1610 buffer = gst_sample_get_buffer (sample);
1612 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1614 g_mutex_lock (&priv->lock);
1616 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1617 clear_tr_cache (priv, is_rtp);
1618 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1619 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1620 priv->tr_cache_rtp =
1621 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1623 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1626 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1627 clear_tr_cache (priv, is_rtp);
1628 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1629 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1630 priv->tr_cache_rtcp =
1631 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
1633 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
1636 g_mutex_unlock (&priv->lock);
1639 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
1640 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1641 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1644 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
1645 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1646 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1649 gst_sample_unref (sample);
1654 static GstAppSinkCallbacks sink_cb = {
1655 NULL, /* not interested in EOS */
1656 NULL, /* not interested in preroll samples */
1661 get_rtp_encoder (GstRTSPStream * stream, guint session)
1663 GstRTSPStreamPrivate *priv = stream->priv;
1665 if (priv->srtpenc == NULL) {
1668 name = g_strdup_printf ("srtpenc_%u", session);
1669 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1672 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1674 return gst_object_ref (priv->srtpenc);
1678 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1680 GstRTSPStreamPrivate *priv = stream->priv;
1681 GstElement *oldenc, *enc;
1685 if (priv->idx != session)
1688 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1690 oldenc = priv->srtpenc;
1691 enc = get_rtp_encoder (stream, session);
1692 name = g_strdup_printf ("rtp_sink_%d", session);
1693 pad = gst_element_get_request_pad (enc, name);
1695 gst_object_unref (pad);
1698 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
1705 request_rtcp_encoder (GstElement * rtpbin, guint session,
1706 GstRTSPStream * stream)
1708 GstRTSPStreamPrivate *priv = stream->priv;
1709 GstElement *oldenc, *enc;
1713 if (priv->idx != session)
1716 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1718 oldenc = priv->srtpenc;
1719 enc = get_rtp_encoder (stream, session);
1720 name = g_strdup_printf ("rtcp_sink_%d", session);
1721 pad = gst_element_get_request_pad (enc, name);
1723 gst_object_unref (pad);
1726 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
1733 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1735 GstRTSPStreamPrivate *priv = stream->priv;
1738 GST_DEBUG ("request key %08x", ssrc);
1740 g_mutex_lock (&priv->lock);
1741 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1742 gst_caps_ref (caps);
1743 g_mutex_unlock (&priv->lock);
1749 request_rtcp_decoder (GstElement * rtpbin, guint session,
1750 GstRTSPStream * stream)
1752 GstRTSPStreamPrivate *priv = stream->priv;
1754 if (priv->idx != session)
1757 if (priv->srtpdec == NULL) {
1760 name = g_strdup_printf ("srtpdec_%u", session);
1761 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1764 g_signal_connect (priv->srtpdec, "request-key",
1765 (GCallback) request_key, stream);
1767 return gst_object_ref (priv->srtpdec);
1771 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPStream * stream)
1775 GstStructure *pt_map;
1780 pt = gst_rtsp_stream_get_pt (stream);
1781 pt_s = g_strdup_printf ("%u", pt);
1782 rtx_pt = stream->priv->rtx_pt;
1784 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
1786 bin = gst_bin_new (NULL);
1787 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
1788 pt_map = gst_structure_new ("application/x-rtp-pt-map",
1789 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1790 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
1791 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
1793 gst_structure_free (pt_map);
1794 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
1796 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
1797 name = g_strdup_printf ("src_%u", sessid);
1798 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
1800 gst_object_unref (pad);
1802 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
1803 name = g_strdup_printf ("sink_%u", sessid);
1804 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
1806 gst_object_unref (pad);
1812 * gst_rtsp_stream_join_bin:
1813 * @stream: a #GstRTSPStream
1814 * @bin: (transfer none): a #GstBin to join
1815 * @rtpbin: (transfer none): a rtpbin element in @bin
1816 * @state: the target state of the new elements
1818 * Join the #GstBin @bin that contains the element @rtpbin.
1820 * @stream will link to @rtpbin, which must be inside @bin. The elements
1821 * added to @bin will be set to the state given in @state.
1823 * Returns: %TRUE on success.
1826 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1827 GstElement * rtpbin, GstState state)
1829 GstRTSPStreamPrivate *priv;
1833 GstPad *pad, *sinkpad, *selpad;
1834 GstPadLinkReturn ret;
1836 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1837 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1838 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1840 priv = stream->priv;
1842 g_mutex_lock (&priv->lock);
1843 if (priv->is_joined)
1846 /* create a session with the same index as the stream */
1849 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1851 if (!alloc_ports (stream))
1854 /* update the dscp qos field in the sinks */
1855 update_dscp_qos (stream);
1857 if (priv->profiles & GST_RTSP_PROFILE_SAVP
1858 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
1860 g_signal_connect (rtpbin, "request-rtp-encoder",
1861 (GCallback) request_rtp_encoder, stream);
1862 g_signal_connect (rtpbin, "request-rtcp-encoder",
1863 (GCallback) request_rtcp_encoder, stream);
1864 g_signal_connect (rtpbin, "request-rtcp-decoder",
1865 (GCallback) request_rtcp_decoder, stream);
1868 if (priv->rtx_time > 0) {
1869 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
1870 g_signal_connect (rtpbin, "request-aux-sender",
1871 (GCallback) request_aux_sender, stream);
1874 /* get a pad for sending RTP */
1875 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1876 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1878 /* link the RTP pad to the session manager, it should not really fail unless
1879 * this is not really an RTP pad */
1880 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1881 if (ret != GST_PAD_LINK_OK)
1884 /* get pads from the RTP session element for sending and receiving
1886 name = g_strdup_printf ("send_rtp_src_%u", idx);
1887 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1889 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1890 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1892 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1893 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1895 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1896 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1899 /* get the session */
1900 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1902 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1904 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1906 g_signal_connect (priv->session, "on-ssrc-active",
1907 (GCallback) on_ssrc_active, stream);
1908 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1910 g_signal_connect (priv->session, "on-bye-timeout",
1911 (GCallback) on_bye_timeout, stream);
1912 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1915 for (i = 0; i < 2; i++) {
1916 GstPad *teepad, *queuepad;
1917 /* For the sender we create this bit of pipeline for both
1918 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1919 * we need to add a queue before appsink to make the pipeline
1920 * not block. For the TCP case, we want to pump data to the
1921 * client as fast as possible anyway.
1923 * .--------. .-----. .---------.
1924 * | rtpbin | | tee | | udpsink |
1925 * | send->sink src->sink |
1926 * '--------' | | '---------'
1927 * | | .---------. .---------.
1928 * | | | queue | | appsink |
1929 * | src->sink src->sink |
1930 * '-----' '---------' '---------'
1932 * When only UDP is allowed, we skip the tee, queue and appsink and link the
1933 * udpsink directly to the session.
1936 gst_bin_add (bin, priv->udpsink[i]);
1937 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1939 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1940 /* make tee for RTP/RTCP */
1941 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1942 gst_bin_add (bin, priv->tee[i]);
1944 /* and link to rtpbin send pad */
1945 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1946 gst_pad_link (priv->send_src[i], pad);
1947 gst_object_unref (pad);
1949 /* link tee to udpsink */
1950 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1951 gst_pad_link (teepad, sinkpad);
1952 gst_object_unref (teepad);
1955 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1956 gst_bin_add (bin, priv->appqueue[i]);
1957 /* and link to tee */
1958 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1959 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1960 gst_pad_link (teepad, pad);
1961 gst_object_unref (pad);
1962 gst_object_unref (teepad);
1965 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1966 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1967 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1968 gst_bin_add (bin, priv->appsink[i]);
1969 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1970 &sink_cb, stream, NULL);
1971 /* and link to queue */
1972 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1973 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1974 gst_pad_link (queuepad, pad);
1975 gst_object_unref (pad);
1976 gst_object_unref (queuepad);
1978 /* else only udpsink needed, link it to the session */
1979 gst_pad_link (priv->send_src[i], sinkpad);
1981 gst_object_unref (sinkpad);
1983 /* For the receiver we create this bit of pipeline for both
1984 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1985 * and it is all funneled into the rtpbin receive pad.
1987 * .--------. .--------. .--------.
1988 * | udpsrc | | funnel | | rtpbin |
1989 * | src->sink src->sink |
1990 * '--------' | | '--------'
1994 * '--------' '--------'
1996 /* make funnel for the RTP/RTCP receivers */
1997 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1998 gst_bin_add (bin, priv->funnel[i]);
2000 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2001 gst_pad_link (pad, priv->recv_sink[i]);
2002 gst_object_unref (pad);
2004 if (priv->udpsrc_v4[i]) {
2005 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2007 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2008 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2010 gst_bin_add (bin, priv->udpsrc_v4[i]);
2012 /* and link to the funnel v4 */
2013 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2014 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2015 gst_pad_link (pad, selpad);
2016 gst_object_unref (pad);
2017 gst_object_unref (selpad);
2020 if (priv->udpsrc_v6[i]) {
2021 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2022 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2023 gst_bin_add (bin, priv->udpsrc_v6[i]);
2025 /* and link to the funnel v6 */
2026 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2027 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2028 gst_pad_link (pad, selpad);
2029 gst_object_unref (pad);
2030 gst_object_unref (selpad);
2033 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
2034 /* make and add appsrc */
2035 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2036 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2037 gst_bin_add (bin, priv->appsrc[i]);
2038 /* and link to the funnel */
2039 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2040 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2041 gst_pad_link (pad, selpad);
2042 gst_object_unref (pad);
2043 gst_object_unref (selpad);
2046 /* check if we need to set to a special state */
2047 if (state != GST_STATE_NULL) {
2048 if (priv->udpsink[i])
2049 gst_element_set_state (priv->udpsink[i], state);
2050 if (priv->appsink[i])
2051 gst_element_set_state (priv->appsink[i], state);
2052 if (priv->appqueue[i])
2053 gst_element_set_state (priv->appqueue[i], state);
2055 gst_element_set_state (priv->tee[i], state);
2056 if (priv->funnel[i])
2057 gst_element_set_state (priv->funnel[i], state);
2058 if (priv->appsrc[i])
2059 gst_element_set_state (priv->appsrc[i], state);
2063 /* be notified of caps changes */
2064 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2065 (GCallback) caps_notify, stream);
2067 priv->is_joined = TRUE;
2068 g_mutex_unlock (&priv->lock);
2075 g_mutex_unlock (&priv->lock);
2080 g_mutex_unlock (&priv->lock);
2081 GST_WARNING ("failed to allocate ports %u", idx);
2086 GST_WARNING ("failed to link stream %u", idx);
2087 gst_object_unref (priv->send_rtp_sink);
2088 priv->send_rtp_sink = NULL;
2089 g_mutex_unlock (&priv->lock);
2095 * gst_rtsp_stream_leave_bin:
2096 * @stream: a #GstRTSPStream
2097 * @bin: (transfer none): a #GstBin
2098 * @rtpbin: (transfer none): a rtpbin #GstElement
2100 * Remove the elements of @stream from @bin.
2102 * Return: %TRUE on success.
2105 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2106 GstElement * rtpbin)
2108 GstRTSPStreamPrivate *priv;
2112 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2113 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2114 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2116 priv = stream->priv;
2118 g_mutex_lock (&priv->lock);
2119 if (!priv->is_joined)
2120 goto was_not_joined;
2122 /* all transports must be removed by now */
2123 if (priv->transports != NULL)
2124 goto transports_not_removed;
2126 clear_tr_cache (priv, TRUE);
2127 clear_tr_cache (priv, FALSE);
2129 GST_INFO ("stream %p leaving bin", stream);
2131 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2132 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2133 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2134 gst_object_unref (priv->send_rtp_sink);
2135 priv->send_rtp_sink = NULL;
2137 for (i = 0; i < 2; i++) {
2138 if (priv->udpsink[i])
2139 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2140 if (priv->appsink[i])
2141 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2142 if (priv->appqueue[i])
2143 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2145 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2146 if (priv->funnel[i])
2147 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2148 if (priv->appsrc[i])
2149 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2150 if (priv->udpsrc_v4[i]) {
2151 /* and set udpsrc to NULL now before removing */
2152 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2153 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2154 /* removing them should also nicely release the request
2155 * pads when they finalize */
2156 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2158 if (priv->udpsrc_v6[i]) {
2159 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2160 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2161 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2164 for (l = priv->transport_sources; l; l = l->next) {
2165 GstRTSPMulticastTransportSource *s = l->data;
2170 gst_element_set_locked_state (s->udpsrc[i], FALSE);
2171 gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
2172 gst_bin_remove (bin, s->udpsrc[i]);
2175 if (priv->udpsink[i])
2176 gst_bin_remove (bin, priv->udpsink[i]);
2177 if (priv->appsrc[i])
2178 gst_bin_remove (bin, priv->appsrc[i]);
2179 if (priv->appsink[i])
2180 gst_bin_remove (bin, priv->appsink[i]);
2181 if (priv->appqueue[i])
2182 gst_bin_remove (bin, priv->appqueue[i]);
2184 gst_bin_remove (bin, priv->tee[i]);
2185 if (priv->funnel[i])
2186 gst_bin_remove (bin, priv->funnel[i]);
2188 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2189 gst_object_unref (priv->recv_sink[i]);
2190 priv->recv_sink[i] = NULL;
2192 priv->udpsrc_v4[i] = NULL;
2193 priv->udpsrc_v6[i] = NULL;
2194 priv->udpsink[i] = NULL;
2195 priv->appsrc[i] = NULL;
2196 priv->appsink[i] = NULL;
2197 priv->appqueue[i] = NULL;
2198 priv->tee[i] = NULL;
2199 priv->funnel[i] = NULL;
2202 for (l = priv->transport_sources; l; l = l->next) {
2203 GstRTSPMulticastTransportSource *s = l->data;
2204 g_slice_free (GstRTSPMulticastTransportSource, s);
2206 g_list_free (priv->transport_sources);
2207 priv->transport_sources = NULL;
2209 gst_object_unref (priv->send_src[0]);
2210 priv->send_src[0] = NULL;
2212 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2213 gst_object_unref (priv->send_src[1]);
2214 priv->send_src[1] = NULL;
2216 g_object_unref (priv->session);
2217 priv->session = NULL;
2219 gst_caps_unref (priv->caps);
2223 gst_object_unref (priv->srtpenc);
2225 gst_object_unref (priv->srtpdec);
2227 priv->is_joined = FALSE;
2228 g_mutex_unlock (&priv->lock);
2234 g_mutex_unlock (&priv->lock);
2237 transports_not_removed:
2239 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2240 g_mutex_unlock (&priv->lock);
2246 * gst_rtsp_stream_get_rtpinfo:
2247 * @stream: a #GstRTSPStream
2248 * @rtptime: (allow-none): result RTP timestamp
2249 * @seq: (allow-none): result RTP seqnum
2250 * @clock_rate: (allow-none): the clock rate
2251 * @running_time: (allow-none): result running-time
2253 * Retrieve the current rtptime, seq and running-time. This is used to
2254 * construct a RTPInfo reply header.
2256 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2259 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2260 guint * rtptime, guint * seq, guint * clock_rate,
2261 GstClockTime * running_time)
2263 GstRTSPStreamPrivate *priv;
2264 GstStructure *stats;
2265 GObjectClass *payobjclass;
2267 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2269 priv = stream->priv;
2271 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2273 g_mutex_lock (&priv->lock);
2275 if (g_object_class_find_property (payobjclass, "stats")) {
2276 g_object_get (priv->payloader, "stats", &stats, NULL);
2281 gst_structure_get_uint (stats, "seqnum", seq);
2284 gst_structure_get_uint (stats, "timestamp", rtptime);
2287 gst_structure_get_clock_time (stats, "running-time", running_time);
2290 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2291 if (*clock_rate == 0 && running_time)
2292 *running_time = GST_CLOCK_TIME_NONE;
2294 gst_structure_free (stats);
2296 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2297 !g_object_class_find_property (payobjclass, "timestamp"))
2301 g_object_get (priv->payloader, "seqnum", seq, NULL);
2304 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2307 *running_time = GST_CLOCK_TIME_NONE;
2309 g_mutex_unlock (&priv->lock);
2316 GST_WARNING ("Could not get payloader stats");
2317 g_mutex_unlock (&priv->lock);
2323 * gst_rtsp_stream_get_caps:
2324 * @stream: a #GstRTSPStream
2326 * Retrieve the current caps of @stream.
2328 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2332 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2334 GstRTSPStreamPrivate *priv;
2337 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2339 priv = stream->priv;
2341 g_mutex_lock (&priv->lock);
2342 if ((result = priv->caps))
2343 gst_caps_ref (result);
2344 g_mutex_unlock (&priv->lock);
2350 * gst_rtsp_stream_recv_rtp:
2351 * @stream: a #GstRTSPStream
2352 * @buffer: (transfer full): a #GstBuffer
2354 * Handle an RTP buffer for the stream. This method is usually called when a
2355 * message has been received from a client using the TCP transport.
2357 * This function takes ownership of @buffer.
2359 * Returns: a GstFlowReturn.
2362 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2364 GstRTSPStreamPrivate *priv;
2366 GstElement *element;
2368 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2369 priv = stream->priv;
2370 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2371 g_return_val_if_fail (priv->is_joined, FALSE);
2373 g_mutex_lock (&priv->lock);
2374 if (priv->appsrc[0])
2375 element = gst_object_ref (priv->appsrc[0]);
2378 g_mutex_unlock (&priv->lock);
2381 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2382 gst_object_unref (element);
2390 * gst_rtsp_stream_recv_rtcp:
2391 * @stream: a #GstRTSPStream
2392 * @buffer: (transfer full): a #GstBuffer
2394 * Handle an RTCP buffer for the stream. This method is usually called when a
2395 * message has been received from a client using the TCP transport.
2397 * This function takes ownership of @buffer.
2399 * Returns: a GstFlowReturn.
2402 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2404 GstRTSPStreamPrivate *priv;
2406 GstElement *element;
2408 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2409 priv = stream->priv;
2410 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2412 if (!priv->is_joined) {
2413 gst_buffer_unref (buffer);
2414 return GST_FLOW_NOT_LINKED;
2416 g_mutex_lock (&priv->lock);
2417 if (priv->appsrc[1])
2418 element = gst_object_ref (priv->appsrc[1]);
2421 g_mutex_unlock (&priv->lock);
2424 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2425 gst_object_unref (element);
2428 gst_buffer_unref (buffer);
2433 /* must be called with lock */
2435 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
2438 GstRTSPStreamPrivate *priv = stream->priv;
2439 const GstRTSPTransport *tr;
2441 tr = gst_rtsp_stream_transport_get_transport (trans);
2443 switch (tr->lower_transport) {
2444 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2446 GstRTSPMulticastTransportSource *source;
2449 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[0])));
2454 GstPad *selpad, *pad;
2456 source = g_slice_new0 (GstRTSPMulticastTransportSource);
2457 source->transport = trans;
2459 for (i = 0; i < 2; i++) {
2461 g_strdup_printf ("udp://%s:%d", tr->destination,
2462 (i == 0) ? tr->port.min : tr->port.max);
2464 gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2467 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2469 gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
2470 gst_element_set_locked_state (source->udpsrc[i], TRUE);
2472 gst_bin_add (bin, source->udpsrc[i]);
2474 /* and link to the funnel v4 */
2475 source->selpad[i] = selpad =
2476 gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2477 pad = gst_element_get_static_pad (source->udpsrc[i], "src");
2478 gst_pad_link (pad, selpad);
2479 gst_object_unref (pad);
2480 gst_object_unref (selpad);
2482 gst_object_unref (bin);
2484 priv->transport_sources =
2485 g_list_prepend (priv->transport_sources, source);
2489 for (l = priv->transport_sources; l; l = l->next) {
2492 if (source->transport == trans) {
2493 priv->transport_sources =
2494 g_list_delete_link (priv->transport_sources, l);
2502 for (i = 0; i < 2; i++) {
2503 /* Will automatically unlink everything */
2504 gst_bin_remove (bin,
2505 GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
2507 gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
2508 gst_object_unref (source->udpsrc[i]);
2510 gst_element_release_request_pad (priv->funnel[i],
2514 g_slice_free (GstRTSPMulticastTransportSource, source);
2518 /* fall through for the generic case */
2520 case GST_RTSP_LOWER_TRANS_UDP:
2526 dest = tr->destination;
2527 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2532 min = tr->client_port.min;
2533 max = tr->client_port.max;
2538 GST_INFO ("setting ttl-mc %d", ttl);
2539 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
2540 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
2542 GST_INFO ("adding %s:%d-%d", dest, min, max);
2543 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
2544 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
2545 priv->transports = g_list_prepend (priv->transports, trans);
2547 GST_INFO ("removing %s:%d-%d", dest, min, max);
2548 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
2549 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
2550 priv->transports = g_list_remove (priv->transports, trans);
2552 priv->transports_cookie++;
2555 case GST_RTSP_LOWER_TRANS_TCP:
2557 GST_INFO ("adding TCP %s", tr->destination);
2558 priv->transports = g_list_prepend (priv->transports, trans);
2560 GST_INFO ("removing TCP %s", tr->destination);
2561 priv->transports = g_list_remove (priv->transports, trans);
2563 priv->transports_cookie++;
2566 goto unknown_transport;
2573 GST_INFO ("Unknown transport %d", tr->lower_transport);
2580 * gst_rtsp_stream_add_transport:
2581 * @stream: a #GstRTSPStream
2582 * @trans: (transfer none): a #GstRTSPStreamTransport
2584 * Add the transport in @trans to @stream. The media of @stream will
2585 * then also be send to the values configured in @trans.
2587 * @stream must be joined to a bin.
2589 * @trans must contain a valid #GstRTSPTransport.
2591 * Returns: %TRUE if @trans was added
2594 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
2595 GstRTSPStreamTransport * trans)
2597 GstRTSPStreamPrivate *priv;
2600 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2601 priv = stream->priv;
2602 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2603 g_return_val_if_fail (priv->is_joined, FALSE);
2605 g_mutex_lock (&priv->lock);
2606 res = update_transport (stream, trans, TRUE);
2607 g_mutex_unlock (&priv->lock);
2613 * gst_rtsp_stream_remove_transport:
2614 * @stream: a #GstRTSPStream
2615 * @trans: (transfer none): a #GstRTSPStreamTransport
2617 * Remove the transport in @trans from @stream. The media of @stream will
2618 * not be sent to the values configured in @trans.
2620 * @stream must be joined to a bin.
2622 * @trans must contain a valid #GstRTSPTransport.
2624 * Returns: %TRUE if @trans was removed
2627 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2628 GstRTSPStreamTransport * trans)
2630 GstRTSPStreamPrivate *priv;
2633 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2634 priv = stream->priv;
2635 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2636 g_return_val_if_fail (priv->is_joined, FALSE);
2638 g_mutex_lock (&priv->lock);
2639 res = update_transport (stream, trans, FALSE);
2640 g_mutex_unlock (&priv->lock);
2646 * gst_rtsp_stream_update_crypto:
2647 * @stream: a #GstRTSPStream
2649 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
2651 * Update the new crypto information for @ssrc in @stream. If information
2652 * for @ssrc did not exist, it will be added. If information
2653 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
2654 * be removed from @stream.
2656 * Returns: %TRUE if @crypto could be updated
2659 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
2660 guint ssrc, GstCaps * crypto)
2662 GstRTSPStreamPrivate *priv;
2664 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2665 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
2667 priv = stream->priv;
2669 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
2671 g_mutex_lock (&priv->lock);
2673 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
2674 gst_caps_ref (crypto));
2676 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
2677 g_mutex_unlock (&priv->lock);
2683 * gst_rtsp_stream_get_rtp_socket:
2684 * @stream: a #GstRTSPStream
2685 * @family: the socket family
2687 * Get the RTP socket from @stream for a @family.
2689 * @stream must be joined to a bin.
2691 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
2692 * socket could be allocated for @family. Unref after usage
2695 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
2697 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2701 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2702 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2703 family == G_SOCKET_FAMILY_IPV6, NULL);
2704 g_return_val_if_fail (priv->udpsink[0], NULL);
2706 if (family == G_SOCKET_FAMILY_IPV6)
2711 g_object_get (priv->udpsink[0], name, &socket, NULL);
2717 * gst_rtsp_stream_get_rtcp_socket:
2718 * @stream: a #GstRTSPStream
2719 * @family: the socket family
2721 * Get the RTCP socket from @stream for a @family.
2723 * @stream must be joined to a bin.
2725 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
2726 * socket could be allocated for @family. Unref after usage
2729 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
2731 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2735 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2736 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2737 family == G_SOCKET_FAMILY_IPV6, NULL);
2738 g_return_val_if_fail (priv->udpsink[1], NULL);
2740 if (family == G_SOCKET_FAMILY_IPV6)
2745 g_object_get (priv->udpsink[1], name, &socket, NULL);
2751 * gst_rtsp_stream_set_seqnum:
2752 * @stream: a #GstRTSPStream
2753 * @seqnum: a new sequence number
2755 * Configure the sequence number in the payloader of @stream to @seqnum.
2758 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
2760 GstRTSPStreamPrivate *priv;
2762 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2764 priv = stream->priv;
2766 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
2770 * gst_rtsp_stream_get_seqnum:
2771 * @stream: a #GstRTSPStream
2773 * Get the configured sequence number in the payloader of @stream.
2775 * Returns: the sequence number of the payloader.
2778 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
2780 GstRTSPStreamPrivate *priv;
2783 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
2785 priv = stream->priv;
2787 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
2793 * gst_rtsp_stream_transport_filter:
2794 * @stream: a #GstRTSPStream
2795 * @func: (scope call) (allow-none): a callback
2796 * @user_data: (closure): user data passed to @func
2798 * Call @func for each transport managed by @stream. The result value of @func
2799 * determines what happens to the transport. @func will be called with @stream
2800 * locked so no further actions on @stream can be performed from @func.
2802 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
2805 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
2807 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
2808 * will also be added with an additional ref to the result #GList of this
2811 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
2813 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
2814 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
2815 * element in the #GList should be unreffed before the list is freed.
2818 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
2819 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
2821 GstRTSPStreamPrivate *priv;
2822 GList *result, *walk, *next;
2823 GHashTable *visited = NULL;
2826 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2828 priv = stream->priv;
2832 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
2834 g_mutex_lock (&priv->lock);
2836 cookie = priv->transports_cookie;
2837 for (walk = priv->transports; walk; walk = next) {
2838 GstRTSPStreamTransport *trans = walk->data;
2839 GstRTSPFilterResult res;
2842 next = g_list_next (walk);
2845 /* only visit each transport once */
2846 if (g_hash_table_contains (visited, trans))
2849 g_hash_table_add (visited, g_object_ref (trans));
2850 g_mutex_unlock (&priv->lock);
2852 res = func (stream, trans, user_data);
2854 g_mutex_lock (&priv->lock);
2856 res = GST_RTSP_FILTER_REF;
2858 changed = (cookie != priv->transports_cookie);
2861 case GST_RTSP_FILTER_REMOVE:
2862 update_transport (stream, trans, FALSE);
2864 case GST_RTSP_FILTER_REF:
2865 result = g_list_prepend (result, g_object_ref (trans));
2867 case GST_RTSP_FILTER_KEEP:
2874 g_mutex_unlock (&priv->lock);
2877 g_hash_table_unref (visited);
2882 static GstPadProbeReturn
2883 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2885 GstRTSPStreamPrivate *priv;
2886 GstRTSPStream *stream;
2889 priv = stream->priv;
2891 GST_DEBUG_OBJECT (pad, "now blocking");
2893 g_mutex_lock (&priv->lock);
2894 priv->blocking = TRUE;
2895 g_mutex_unlock (&priv->lock);
2897 gst_element_post_message (priv->payloader,
2898 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
2899 gst_structure_new_empty ("GstRTSPStreamBlocking")));
2901 return GST_PAD_PROBE_OK;
2905 * gst_rtsp_stream_set_blocked:
2906 * @stream: a #GstRTSPStream
2907 * @blocked: boolean indicating we should block or unblock
2909 * Blocks or unblocks the dataflow on @stream.
2911 * Returns: %TRUE on success
2914 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
2916 GstRTSPStreamPrivate *priv;
2918 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2920 priv = stream->priv;
2922 g_mutex_lock (&priv->lock);
2924 priv->blocking = FALSE;
2925 if (priv->blocked_id == 0) {
2926 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
2927 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
2928 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
2929 g_object_ref (stream), g_object_unref);
2932 if (priv->blocked_id != 0) {
2933 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
2934 priv->blocked_id = 0;
2935 priv->blocking = FALSE;
2938 g_mutex_unlock (&priv->lock);
2944 * gst_rtsp_stream_is_blocking:
2945 * @stream: a #GstRTSPStream
2947 * Check if @stream is blocking on a #GstBuffer.
2949 * Returns: %TRUE if @stream is blocking
2952 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
2954 GstRTSPStreamPrivate *priv;
2957 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2959 priv = stream->priv;
2961 g_mutex_lock (&priv->lock);
2962 result = priv->blocking;
2963 g_mutex_unlock (&priv->lock);
2969 * gst_rtsp_stream_query_position:
2970 * @stream: a #GstRTSPStream
2972 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
2973 * the RTP parts of the pipeline and not the RTCP parts.
2975 * Returns: %TRUE if the position could be queried
2978 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
2980 GstRTSPStreamPrivate *priv;
2984 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2986 priv = stream->priv;
2988 g_mutex_lock (&priv->lock);
2989 if ((sink = priv->udpsink[0]))
2990 gst_object_ref (sink);
2991 g_mutex_unlock (&priv->lock);
2996 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
2997 gst_object_unref (sink);
3003 * gst_rtsp_stream_query_stop:
3004 * @stream: a #GstRTSPStream
3006 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3007 * the RTP parts of the pipeline and not the RTCP parts.
3009 * Returns: %TRUE if the stop could be queried
3012 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3014 GstRTSPStreamPrivate *priv;
3019 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3021 priv = stream->priv;
3023 g_mutex_lock (&priv->lock);
3024 if ((sink = priv->udpsink[0]))
3025 gst_object_ref (sink);
3026 g_mutex_unlock (&priv->lock);
3031 query = gst_query_new_segment (GST_FORMAT_TIME);
3032 if ((ret = gst_element_query (sink, query))) {
3035 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3036 if (format != GST_FORMAT_TIME)
3039 gst_query_unref (query);
3040 gst_object_unref (sink);