2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: A media stream
24 * @see_also: #GstRTSPMedia
26 * The #GstRTSPStream object manages the data transport for one stream. It
27 * is created from a payloader element and a source pad that produce the RTP
28 * packets for the stream.
30 * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
31 * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
33 * The #GstRTSPStream will use the configured addresspool, as set with
34 * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
35 * stream. With gst_rtsp_stream_get_multicast_address() you can get the
38 * With gst_rtsp_stream_get_server_port () you can get the port that the server
39 * will use to receive RTCP. This is the part that the clients will use to send
42 * With gst_rtsp_stream_add_transport() destinations can be added where the
43 * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
44 * the destination again.
46 * Last reviewed on 2013-07-16 (1.0.0)
55 #include <gst/app/gstappsrc.h>
56 #include <gst/app/gstappsink.h>
58 #include <gst/rtp/gstrtpbuffer.h>
60 #include "rtsp-stream.h"
62 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
63 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
67 GstRTSPStreamTransport *transport;
69 /* RTP and RTCP source */
70 GstElement *udpsrc[2];
72 } GstRTSPMulticastTransportSource;
74 struct _GstRTSPStreamPrivate
78 /* Only one pad is ever set */
79 GstPad *srcpad, *sinkpad;
80 GstElement *payloader;
85 GstRTSPProfile profiles;
86 GstRTSPLowerTrans protocols;
88 /* pads on the rtpbin */
89 GstPad *send_rtp_sink;
94 /* the RTPSession object */
97 /* SRTP encoder/decoder */
102 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
104 GstElement *udpsrc_v4[2];
106 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
108 GstElement *udpsrc_v6[2];
110 GstElement *udpqueue[2];
111 GstElement *udpsink[2];
113 /* for TCP transport */
114 GstElement *appsrc[2];
115 GstClockTime appsrc_base_time[2];
116 GstElement *appqueue[2];
117 GstElement *appsink[2];
120 GstElement *funnel[2];
125 GstClockTime rtx_time;
127 /* server ports for sending/receiving over ipv4 */
128 GstRTSPRange server_port_v4;
129 GstRTSPAddress *server_addr_v4;
132 /* server ports for sending/receiving over ipv6 */
133 GstRTSPRange server_port_v6;
134 GstRTSPAddress *server_addr_v6;
137 /* multicast addresses */
138 GstRTSPAddressPool *pool;
139 GstRTSPAddress *addr_v4;
140 GstRTSPAddress *addr_v6;
142 /* the caps of the stream */
146 /* transports we stream to */
149 guint transports_cookie;
151 GList *tr_cache_rtcp;
152 guint tr_cache_cookie_rtp;
153 guint tr_cache_cookie_rtcp;
156 /* UDP sources for UDP multicast transports */
157 GList *transport_sources;
161 /* stream blocking */
165 /* pt->caps map for RECORD streams */
169 #define DEFAULT_CONTROL NULL
170 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
171 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
172 GST_RTSP_LOWER_TRANS_TCP
185 SIGNAL_NEW_RTP_ENCODER,
186 SIGNAL_NEW_RTCP_ENCODER,
190 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
191 #define GST_CAT_DEFAULT rtsp_stream_debug
193 static GQuark ssrc_stream_map_key;
195 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
196 GValue * value, GParamSpec * pspec);
197 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
198 const GValue * value, GParamSpec * pspec);
200 static void gst_rtsp_stream_finalize (GObject * obj);
202 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
204 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
207 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
209 GObjectClass *gobject_class;
211 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
213 gobject_class = G_OBJECT_CLASS (klass);
215 gobject_class->get_property = gst_rtsp_stream_get_property;
216 gobject_class->set_property = gst_rtsp_stream_set_property;
217 gobject_class->finalize = gst_rtsp_stream_finalize;
219 g_object_class_install_property (gobject_class, PROP_CONTROL,
220 g_param_spec_string ("control", "Control",
221 "The control string for this stream", DEFAULT_CONTROL,
222 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
224 g_object_class_install_property (gobject_class, PROP_PROFILES,
225 g_param_spec_flags ("profiles", "Profiles",
226 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
227 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
229 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
230 g_param_spec_flags ("protocols", "Protocols",
231 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
232 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
234 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
235 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
236 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
237 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
239 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
240 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
241 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
242 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
244 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
246 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
250 gst_rtsp_stream_init (GstRTSPStream * stream)
252 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
254 GST_DEBUG ("new stream %p", stream);
259 priv->control = g_strdup (DEFAULT_CONTROL);
260 priv->profiles = DEFAULT_PROFILES;
261 priv->protocols = DEFAULT_PROTOCOLS;
263 g_mutex_init (&priv->lock);
265 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
266 NULL, (GDestroyNotify) gst_caps_unref);
267 priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
268 (GDestroyNotify) gst_caps_unref);
272 gst_rtsp_stream_finalize (GObject * obj)
274 GstRTSPStream *stream;
275 GstRTSPStreamPrivate *priv;
277 stream = GST_RTSP_STREAM (obj);
280 GST_DEBUG ("finalize stream %p", stream);
282 /* we really need to be unjoined now */
283 g_return_if_fail (!priv->is_joined);
286 gst_rtsp_address_free (priv->addr_v4);
288 gst_rtsp_address_free (priv->addr_v6);
289 if (priv->server_addr_v4)
290 gst_rtsp_address_free (priv->server_addr_v4);
291 if (priv->server_addr_v6)
292 gst_rtsp_address_free (priv->server_addr_v6);
294 g_object_unref (priv->pool);
296 g_object_unref (priv->rtxsend);
298 gst_object_unref (priv->payloader);
300 gst_object_unref (priv->srcpad);
302 gst_object_unref (priv->sinkpad);
303 g_free (priv->control);
304 g_mutex_clear (&priv->lock);
306 g_hash_table_unref (priv->keys);
307 g_hash_table_destroy (priv->ptmap);
309 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
313 gst_rtsp_stream_get_property (GObject * object, guint propid,
314 GValue * value, GParamSpec * pspec)
316 GstRTSPStream *stream = GST_RTSP_STREAM (object);
320 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
323 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
326 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
329 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
334 gst_rtsp_stream_set_property (GObject * object, guint propid,
335 const GValue * value, GParamSpec * pspec)
337 GstRTSPStream *stream = GST_RTSP_STREAM (object);
341 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
344 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
347 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
350 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
355 * gst_rtsp_stream_new:
358 * @payloader: a #GstElement
360 * Create a new media stream with index @idx that handles RTP data on
361 * @pad and has a payloader element @payloader if @pad is a source pad
362 * or a depayloader element @payloader if @pad is a sink pad.
364 * Returns: (transfer full): a new #GstRTSPStream
367 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
369 GstRTSPStreamPrivate *priv;
370 GstRTSPStream *stream;
372 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
373 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
375 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
378 priv->payloader = gst_object_ref (payloader);
379 if (GST_PAD_IS_SRC (pad))
380 priv->srcpad = gst_object_ref (pad);
382 priv->sinkpad = gst_object_ref (pad);
388 * gst_rtsp_stream_get_index:
389 * @stream: a #GstRTSPStream
391 * Get the stream index.
393 * Return: the stream index.
396 gst_rtsp_stream_get_index (GstRTSPStream * stream)
398 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
400 return stream->priv->idx;
404 * gst_rtsp_stream_get_pt:
405 * @stream: a #GstRTSPStream
407 * Get the stream payload type.
409 * Return: the stream payload type.
412 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
414 GstRTSPStreamPrivate *priv;
417 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
421 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
427 * gst_rtsp_stream_get_srcpad:
428 * @stream: a #GstRTSPStream
430 * Get the srcpad associated with @stream.
432 * Returns: (transfer full): the srcpad. Unref after usage.
435 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
437 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
439 if (!stream->priv->srcpad)
442 return gst_object_ref (stream->priv->srcpad);
446 * gst_rtsp_stream_get_sinkpad:
447 * @stream: a #GstRTSPStream
449 * Get the sinkpad associated with @stream.
451 * Returns: (transfer full): the sinkpad. Unref after usage.
454 gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
456 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
458 if (!stream->priv->sinkpad)
461 return gst_object_ref (stream->priv->sinkpad);
465 * gst_rtsp_stream_get_control:
466 * @stream: a #GstRTSPStream
468 * Get the control string to identify this stream.
470 * Returns: (transfer full): the control string. g_free() after usage.
473 gst_rtsp_stream_get_control (GstRTSPStream * stream)
475 GstRTSPStreamPrivate *priv;
478 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
482 g_mutex_lock (&priv->lock);
483 if ((result = g_strdup (priv->control)) == NULL)
484 result = g_strdup_printf ("stream=%u", priv->idx);
485 g_mutex_unlock (&priv->lock);
491 * gst_rtsp_stream_set_control:
492 * @stream: a #GstRTSPStream
493 * @control: a control string
495 * Set the control string in @stream.
498 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
500 GstRTSPStreamPrivate *priv;
502 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
506 g_mutex_lock (&priv->lock);
507 g_free (priv->control);
508 priv->control = g_strdup (control);
509 g_mutex_unlock (&priv->lock);
513 * gst_rtsp_stream_has_control:
514 * @stream: a #GstRTSPStream
515 * @control: a control string
517 * Check if @stream has the control string @control.
519 * Returns: %TRUE is @stream has @control as the control string
522 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
524 GstRTSPStreamPrivate *priv;
527 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
531 g_mutex_lock (&priv->lock);
533 res = (g_strcmp0 (priv->control, control) == 0);
537 if (sscanf (control, "stream=%u", &streamid) > 0)
538 res = (streamid == priv->idx);
542 g_mutex_unlock (&priv->lock);
548 * gst_rtsp_stream_set_mtu:
549 * @stream: a #GstRTSPStream
552 * Configure the mtu in the payloader of @stream to @mtu.
555 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
557 GstRTSPStreamPrivate *priv;
559 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
563 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
565 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
569 * gst_rtsp_stream_get_mtu:
570 * @stream: a #GstRTSPStream
572 * Get the configured MTU in the payloader of @stream.
574 * Returns: the MTU of the payloader.
577 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
579 GstRTSPStreamPrivate *priv;
582 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
586 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
591 /* Update the dscp qos property on the udp sinks */
593 update_dscp_qos (GstRTSPStream * stream)
595 GstRTSPStreamPrivate *priv;
597 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
601 if (priv->udpsink[0]) {
602 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
606 if (priv->udpsink[1]) {
607 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
613 * gst_rtsp_stream_set_dscp_qos:
614 * @stream: a #GstRTSPStream
615 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
617 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
620 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
622 GstRTSPStreamPrivate *priv;
624 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
628 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
630 if (dscp_qos < -1 || dscp_qos > 63) {
631 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
635 priv->dscp_qos = dscp_qos;
637 update_dscp_qos (stream);
641 * gst_rtsp_stream_get_dscp_qos:
642 * @stream: a #GstRTSPStream
644 * Get the configured DSCP QoS in of the outgoing sockets.
646 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
649 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
651 GstRTSPStreamPrivate *priv;
653 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
657 return priv->dscp_qos;
661 * gst_rtsp_stream_is_transport_supported:
662 * @stream: a #GstRTSPStream
663 * @transport: (transfer none): a #GstRTSPTransport
665 * Check if @transport can be handled by stream
667 * Returns: %TRUE if @transport can be handled by @stream.
670 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
671 GstRTSPTransport * transport)
673 GstRTSPStreamPrivate *priv;
675 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
679 g_mutex_lock (&priv->lock);
680 if (transport->trans != GST_RTSP_TRANS_RTP)
681 goto unsupported_transmode;
683 if (!(transport->profile & priv->profiles))
684 goto unsupported_profile;
686 if (!(transport->lower_transport & priv->protocols))
687 goto unsupported_ltrans;
689 g_mutex_unlock (&priv->lock);
694 unsupported_transmode:
696 GST_DEBUG ("unsupported transport mode %d", transport->trans);
697 g_mutex_unlock (&priv->lock);
702 GST_DEBUG ("unsupported profile %d", transport->profile);
703 g_mutex_unlock (&priv->lock);
708 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
709 g_mutex_unlock (&priv->lock);
715 * gst_rtsp_stream_set_profiles:
716 * @stream: a #GstRTSPStream
717 * @profiles: the new profiles
719 * Configure the allowed profiles for @stream.
722 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
724 GstRTSPStreamPrivate *priv;
726 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
730 g_mutex_lock (&priv->lock);
731 priv->profiles = profiles;
732 g_mutex_unlock (&priv->lock);
736 * gst_rtsp_stream_get_profiles:
737 * @stream: a #GstRTSPStream
739 * Get the allowed profiles of @stream.
741 * Returns: a #GstRTSPProfile
744 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
746 GstRTSPStreamPrivate *priv;
749 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
753 g_mutex_lock (&priv->lock);
754 res = priv->profiles;
755 g_mutex_unlock (&priv->lock);
761 * gst_rtsp_stream_set_protocols:
762 * @stream: a #GstRTSPStream
763 * @protocols: the new flags
765 * Configure the allowed lower transport for @stream.
768 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
769 GstRTSPLowerTrans protocols)
771 GstRTSPStreamPrivate *priv;
773 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
777 g_mutex_lock (&priv->lock);
778 priv->protocols = protocols;
779 g_mutex_unlock (&priv->lock);
783 * gst_rtsp_stream_get_protocols:
784 * @stream: a #GstRTSPStream
786 * Get the allowed protocols of @stream.
788 * Returns: a #GstRTSPLowerTrans
791 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
793 GstRTSPStreamPrivate *priv;
794 GstRTSPLowerTrans res;
796 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
797 GST_RTSP_LOWER_TRANS_UNKNOWN);
801 g_mutex_lock (&priv->lock);
802 res = priv->protocols;
803 g_mutex_unlock (&priv->lock);
809 * gst_rtsp_stream_set_address_pool:
810 * @stream: a #GstRTSPStream
811 * @pool: (transfer none): a #GstRTSPAddressPool
813 * configure @pool to be used as the address pool of @stream.
816 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
817 GstRTSPAddressPool * pool)
819 GstRTSPStreamPrivate *priv;
820 GstRTSPAddressPool *old;
822 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
826 GST_LOG_OBJECT (stream, "set address pool %p", pool);
828 g_mutex_lock (&priv->lock);
829 if ((old = priv->pool) != pool)
830 priv->pool = pool ? g_object_ref (pool) : NULL;
833 g_mutex_unlock (&priv->lock);
836 g_object_unref (old);
840 * gst_rtsp_stream_get_address_pool:
841 * @stream: a #GstRTSPStream
843 * Get the #GstRTSPAddressPool used as the address pool of @stream.
845 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
849 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
851 GstRTSPStreamPrivate *priv;
852 GstRTSPAddressPool *result;
854 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
858 g_mutex_lock (&priv->lock);
859 if ((result = priv->pool))
860 g_object_ref (result);
861 g_mutex_unlock (&priv->lock);
867 * gst_rtsp_stream_get_multicast_address:
868 * @stream: a #GstRTSPStream
869 * @family: the #GSocketFamily
871 * Get the multicast address of @stream for @family.
873 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
874 * or %NULL when no address could be allocated. gst_rtsp_address_free()
878 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
879 GSocketFamily family)
881 GstRTSPStreamPrivate *priv;
882 GstRTSPAddress *result;
883 GstRTSPAddress **addrp;
884 GstRTSPAddressFlags flags;
886 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
890 if (family == G_SOCKET_FAMILY_IPV6) {
891 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
892 addrp = &priv->addr_v6;
894 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
895 addrp = &priv->addr_v4;
898 g_mutex_lock (&priv->lock);
899 if (*addrp == NULL) {
900 if (priv->pool == NULL)
903 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
905 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
909 result = gst_rtsp_address_copy (*addrp);
910 g_mutex_unlock (&priv->lock);
917 GST_ERROR_OBJECT (stream, "no address pool specified");
918 g_mutex_unlock (&priv->lock);
923 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
924 g_mutex_unlock (&priv->lock);
930 * gst_rtsp_stream_reserve_address:
931 * @stream: a #GstRTSPStream
932 * @address: an address
937 * Reserve @address and @port as the address and port of @stream.
939 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
940 * the address could be reserved. gst_rtsp_address_free() after usage.
943 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
944 const gchar * address, guint port, guint n_ports, guint ttl)
946 GstRTSPStreamPrivate *priv;
947 GstRTSPAddress *result;
949 GSocketFamily family;
950 GstRTSPAddress **addrp;
952 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
953 g_return_val_if_fail (address != NULL, NULL);
954 g_return_val_if_fail (port > 0, NULL);
955 g_return_val_if_fail (n_ports > 0, NULL);
956 g_return_val_if_fail (ttl > 0, NULL);
960 addr = g_inet_address_new_from_string (address);
962 GST_ERROR ("failed to get inet addr from %s", address);
963 family = G_SOCKET_FAMILY_IPV4;
965 family = g_inet_address_get_family (addr);
966 g_object_unref (addr);
969 if (family == G_SOCKET_FAMILY_IPV6)
970 addrp = &priv->addr_v6;
972 addrp = &priv->addr_v4;
974 g_mutex_lock (&priv->lock);
975 if (*addrp == NULL) {
976 GstRTSPAddressPoolResult res;
978 if (priv->pool == NULL)
981 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
982 port, n_ports, ttl, addrp);
983 if (res != GST_RTSP_ADDRESS_POOL_OK)
986 if (strcmp ((*addrp)->address, address) ||
987 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
988 (*addrp)->ttl != ttl)
989 goto different_address;
991 result = gst_rtsp_address_copy (*addrp);
992 g_mutex_unlock (&priv->lock);
999 GST_ERROR_OBJECT (stream, "no address pool specified");
1000 g_mutex_unlock (&priv->lock);
1005 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
1007 g_mutex_unlock (&priv->lock);
1012 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
1013 " reserved", address);
1014 g_mutex_unlock (&priv->lock);
1020 alloc_ports_one_family (GstRTSPStream * stream, GstRTSPAddressPool * pool,
1021 gint buffer_size, GSocketFamily family, GstElement * udpsrc_out[2],
1022 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
1023 GstRTSPAddress ** server_addr_out)
1025 GstRTSPStreamPrivate *priv = stream->priv;
1026 GstStateChangeReturn ret;
1027 GstElement *udpsrc0, *udpsrc1;
1028 GstElement *udpsink0, *udpsink1;
1029 GSocket *rtp_socket = NULL;
1030 GSocket *rtcp_socket;
1031 gint tmp_rtp, tmp_rtcp;
1033 gint rtpport, rtcpport;
1034 GList *rejected_addresses = NULL;
1035 GstRTSPAddress *addr = NULL;
1036 GInetAddress *inetaddr = NULL;
1037 GSocketAddress *rtp_sockaddr = NULL;
1038 GSocketAddress *rtcp_sockaddr = NULL;
1039 const gchar *multisink_socket;
1041 if (family == G_SOCKET_FAMILY_IPV6)
1042 multisink_socket = "socket-v6";
1044 multisink_socket = "socket";
1052 /* Start with random port */
1055 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1056 G_SOCKET_PROTOCOL_UDP, NULL);
1058 goto no_udp_protocol;
1060 if (*server_addr_out)
1061 gst_rtsp_address_free (*server_addr_out);
1063 /* try to allocate 2 UDP ports, the RTP port should be an even
1064 * number and the RTCP port should be the next (uneven) port */
1067 if (rtp_socket == NULL) {
1068 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1069 G_SOCKET_PROTOCOL_UDP, NULL);
1071 goto no_udp_protocol;
1074 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
1075 GstRTSPAddressFlags flags;
1078 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1080 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1081 if (family == G_SOCKET_FAMILY_IPV6)
1082 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1084 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1086 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1091 tmp_rtp = addr->port;
1093 g_clear_object (&inetaddr);
1094 inetaddr = g_inet_address_new_from_string (addr->address);
1102 if (inetaddr == NULL)
1103 inetaddr = g_inet_address_new_any (family);
1106 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1107 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1108 g_object_unref (rtp_sockaddr);
1111 g_object_unref (rtp_sockaddr);
1113 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1114 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1115 g_clear_object (&rtp_sockaddr);
1120 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1121 g_object_unref (rtp_sockaddr);
1123 /* check if port is even */
1124 if ((tmp_rtp & 1) != 0) {
1125 /* port not even, close and allocate another */
1127 g_clear_object (&rtp_socket);
1132 tmp_rtcp = tmp_rtp + 1;
1134 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1135 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1136 g_object_unref (rtcp_sockaddr);
1137 g_clear_object (&rtp_socket);
1140 g_object_unref (rtcp_sockaddr);
1142 g_clear_object (&inetaddr);
1144 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1145 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1147 if (udpsrc0 == NULL || udpsrc1 == NULL)
1148 goto no_udp_protocol;
1150 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1151 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1153 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1154 if (ret == GST_STATE_CHANGE_FAILURE)
1156 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1157 if (ret == GST_STATE_CHANGE_FAILURE)
1160 /* all fine, do port check */
1161 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1162 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1164 /* this should not happen... */
1165 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1169 udpsink0 = udpsink_out[0];
1171 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1174 goto no_udp_protocol;
1176 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1177 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
1180 udpsink1 = udpsink_out[1];
1182 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1185 goto no_udp_protocol;
1187 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1188 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1189 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
1191 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1192 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
1193 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1194 /* Needs to be async for RECORD streams, otherwise we will never go to
1195 * PLAYING because the sinks will wait for data while the udpsrc can't
1196 * provide data with timestamps in PAUSED. */
1198 g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
1199 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1200 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1201 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1202 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1203 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1205 /* we keep these elements, we will further configure them when the
1206 * client told us to really use the UDP ports. */
1207 udpsrc_out[0] = udpsrc0;
1208 udpsrc_out[1] = udpsrc1;
1209 udpsink_out[0] = udpsink0;
1210 udpsink_out[1] = udpsink1;
1212 server_port_out->min = rtpport;
1213 server_port_out->max = rtcpport;
1215 *server_addr_out = addr;
1216 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1218 g_object_unref (rtp_socket);
1219 g_object_unref (rtcp_socket);
1247 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1248 gst_object_unref (udpsrc0);
1251 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1252 gst_object_unref (udpsrc1);
1255 gst_element_set_state (udpsink0, GST_STATE_NULL);
1256 gst_object_unref (udpsink0);
1259 g_object_unref (inetaddr);
1260 g_list_free_full (rejected_addresses,
1261 (GDestroyNotify) gst_rtsp_address_free);
1263 gst_rtsp_address_free (addr);
1265 g_object_unref (rtp_socket);
1267 g_object_unref (rtcp_socket);
1272 /* must be called with lock */
1274 alloc_ports (GstRTSPStream * stream)
1276 GstRTSPStreamPrivate *priv = stream->priv;
1279 alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
1280 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1281 &priv->server_port_v4, &priv->server_addr_v4);
1284 alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
1285 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1286 &priv->server_port_v6, &priv->server_addr_v6);
1288 return priv->have_ipv4 || priv->have_ipv6;
1292 * gst_rtsp_stream_get_server_port:
1293 * @stream: a #GstRTSPStream
1294 * @server_port: (out): result server port
1295 * @family: the port family to get
1297 * Fill @server_port with the port pair used by the server. This function can
1298 * only be called when @stream has been joined.
1301 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1302 GstRTSPRange * server_port, GSocketFamily family)
1304 GstRTSPStreamPrivate *priv;
1306 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1307 priv = stream->priv;
1308 g_return_if_fail (priv->is_joined);
1310 g_mutex_lock (&priv->lock);
1311 if (family == G_SOCKET_FAMILY_IPV4) {
1313 *server_port = priv->server_port_v4;
1316 *server_port = priv->server_port_v6;
1318 g_mutex_unlock (&priv->lock);
1322 * gst_rtsp_stream_get_rtpsession:
1323 * @stream: a #GstRTSPStream
1325 * Get the RTP session of this stream.
1327 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1330 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1332 GstRTSPStreamPrivate *priv;
1335 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1337 priv = stream->priv;
1339 g_mutex_lock (&priv->lock);
1340 if ((session = priv->session))
1341 g_object_ref (session);
1342 g_mutex_unlock (&priv->lock);
1348 * gst_rtsp_stream_get_ssrc:
1349 * @stream: a #GstRTSPStream
1350 * @ssrc: (out): result ssrc
1352 * Get the SSRC used by the RTP session of this stream. This function can only
1353 * be called when @stream has been joined.
1356 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1358 GstRTSPStreamPrivate *priv;
1360 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1361 priv = stream->priv;
1362 g_return_if_fail (priv->is_joined);
1364 g_mutex_lock (&priv->lock);
1365 if (ssrc && priv->session)
1366 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1367 g_mutex_unlock (&priv->lock);
1371 * gst_rtsp_stream_set_retransmission_time:
1372 * @stream: a #GstRTSPStream
1373 * @time: a #GstClockTime
1375 * Set the amount of time to store retransmission packets.
1378 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1381 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1383 g_mutex_lock (&stream->priv->lock);
1384 stream->priv->rtx_time = time;
1385 if (stream->priv->rtxsend)
1386 g_object_set (stream->priv->rtxsend, "max-size-time",
1387 GST_TIME_AS_MSECONDS (time), NULL);
1388 g_mutex_unlock (&stream->priv->lock);
1392 * gst_rtsp_stream_get_retransmission_time:
1393 * @stream: a #GstRTSPStream
1395 * Get the amount of time to store retransmission data.
1397 * Returns: the amount of time to store retransmission data.
1400 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1404 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1406 g_mutex_lock (&stream->priv->lock);
1407 ret = stream->priv->rtx_time;
1408 g_mutex_unlock (&stream->priv->lock);
1414 * gst_rtsp_stream_set_retransmission_pt:
1415 * @stream: a #GstRTSPStream
1418 * Set the payload type (pt) for retransmission of this stream.
1421 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1423 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1425 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1427 g_mutex_lock (&stream->priv->lock);
1428 stream->priv->rtx_pt = rtx_pt;
1429 if (stream->priv->rtxsend) {
1430 guint pt = gst_rtsp_stream_get_pt (stream);
1431 gchar *pt_s = g_strdup_printf ("%d", pt);
1432 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1433 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1434 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1436 gst_structure_free (rtx_pt_map);
1438 g_mutex_unlock (&stream->priv->lock);
1442 * gst_rtsp_stream_get_retransmission_pt:
1443 * @stream: a #GstRTSPStream
1445 * Get the payload-type used for retransmission of this stream
1447 * Returns: The retransmission PT.
1450 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1454 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1456 g_mutex_lock (&stream->priv->lock);
1457 rtx_pt = stream->priv->rtx_pt;
1458 g_mutex_unlock (&stream->priv->lock);
1464 * gst_rtsp_stream_set_buffer_size:
1465 * @stream: a #GstRTSPStream
1466 * @size: the buffer size
1468 * Set the size of the UDP transmission buffer (in bytes)
1469 * Needs to be set before the stream is joined to a bin.
1474 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1476 g_mutex_lock (&stream->priv->lock);
1477 stream->priv->buffer_size = size;
1478 g_mutex_unlock (&stream->priv->lock);
1482 * gst_rtsp_stream_get_buffer_size:
1483 * @stream: a #GstRTSPStream
1485 * Get the size of the UDP transmission buffer (in bytes)
1487 * Returns: the size of the UDP TX buffer
1492 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1496 g_mutex_lock (&stream->priv->lock);
1497 buffer_size = stream->priv->buffer_size;
1498 g_mutex_unlock (&stream->priv->lock);
1503 /* executed from streaming thread */
1505 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1507 GstRTSPStreamPrivate *priv = stream->priv;
1508 GstCaps *newcaps, *oldcaps;
1510 newcaps = gst_pad_get_current_caps (pad);
1512 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1515 g_mutex_lock (&priv->lock);
1516 oldcaps = priv->caps;
1517 priv->caps = newcaps;
1518 g_mutex_unlock (&priv->lock);
1521 gst_caps_unref (oldcaps);
1525 dump_structure (const GstStructure * s)
1529 sstr = gst_structure_to_string (s);
1530 GST_INFO ("structure: %s", sstr);
1534 static GstRTSPStreamTransport *
1535 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1537 GstRTSPStreamPrivate *priv = stream->priv;
1539 GstRTSPStreamTransport *result = NULL;
1544 if (rtcp_from == NULL)
1547 tmp = g_strrstr (rtcp_from, ":");
1551 port = atoi (tmp + 1);
1552 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1554 g_mutex_lock (&priv->lock);
1555 GST_INFO ("finding %s:%d in %d transports", dest, port,
1556 g_list_length (priv->transports));
1558 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1559 GstRTSPStreamTransport *trans = walk->data;
1560 const GstRTSPTransport *tr;
1563 tr = gst_rtsp_stream_transport_get_transport (trans);
1565 min = tr->client_port.min;
1566 max = tr->client_port.max;
1568 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1574 g_object_ref (result);
1575 g_mutex_unlock (&priv->lock);
1582 static GstRTSPStreamTransport *
1583 check_transport (GObject * source, GstRTSPStream * stream)
1585 GstStructure *stats;
1586 GstRTSPStreamTransport *trans;
1588 /* see if we have a stream to match with the origin of the RTCP packet */
1589 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1590 if (trans == NULL) {
1591 g_object_get (source, "stats", &stats, NULL);
1593 const gchar *rtcp_from;
1595 dump_structure (stats);
1597 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1598 if ((trans = find_transport (stream, rtcp_from))) {
1599 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1601 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1604 gst_structure_free (stats);
1612 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1614 GstRTSPStreamTransport *trans;
1616 GST_INFO ("%p: new source %p", stream, source);
1618 trans = check_transport (source, stream);
1621 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1625 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1627 GST_INFO ("%p: new SDES %p", stream, source);
1631 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1633 GstRTSPStreamTransport *trans;
1635 trans = check_transport (source, stream);
1638 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1639 gst_rtsp_stream_transport_keep_alive (trans);
1643 GstStructure *stats;
1644 g_object_get (source, "stats", &stats, NULL);
1646 dump_structure (stats);
1647 gst_structure_free (stats);
1654 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1656 GST_INFO ("%p: source %p bye", stream, source);
1660 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1662 GstRTSPStreamTransport *trans;
1664 GST_INFO ("%p: source %p bye timeout", stream, source);
1666 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1667 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1668 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1673 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1675 GstRTSPStreamTransport *trans;
1677 GST_INFO ("%p: source %p timeout", stream, source);
1679 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1680 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1681 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1686 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1689 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1690 g_list_free (priv->tr_cache_rtp);
1691 priv->tr_cache_rtp = NULL;
1693 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1694 g_list_free (priv->tr_cache_rtcp);
1695 priv->tr_cache_rtcp = NULL;
1699 static GstFlowReturn
1700 handle_new_sample (GstAppSink * sink, gpointer user_data)
1702 GstRTSPStreamPrivate *priv;
1706 GstRTSPStream *stream;
1709 sample = gst_app_sink_pull_sample (sink);
1713 stream = (GstRTSPStream *) user_data;
1714 priv = stream->priv;
1715 buffer = gst_sample_get_buffer (sample);
1717 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1719 g_mutex_lock (&priv->lock);
1721 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1722 clear_tr_cache (priv, is_rtp);
1723 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1724 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1725 priv->tr_cache_rtp =
1726 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1728 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1731 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1732 clear_tr_cache (priv, is_rtp);
1733 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1734 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1735 priv->tr_cache_rtcp =
1736 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
1738 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
1741 g_mutex_unlock (&priv->lock);
1744 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
1745 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1746 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1749 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
1750 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1751 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1754 gst_sample_unref (sample);
1759 static GstAppSinkCallbacks sink_cb = {
1760 NULL, /* not interested in EOS */
1761 NULL, /* not interested in preroll samples */
1766 get_rtp_encoder (GstRTSPStream * stream, guint session)
1768 GstRTSPStreamPrivate *priv = stream->priv;
1770 if (priv->srtpenc == NULL) {
1773 name = g_strdup_printf ("srtpenc_%u", session);
1774 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1777 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1779 return gst_object_ref (priv->srtpenc);
1783 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1785 GstRTSPStreamPrivate *priv = stream->priv;
1786 GstElement *oldenc, *enc;
1790 if (priv->idx != session)
1793 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1795 oldenc = priv->srtpenc;
1796 enc = get_rtp_encoder (stream, session);
1797 name = g_strdup_printf ("rtp_sink_%d", session);
1798 pad = gst_element_get_request_pad (enc, name);
1800 gst_object_unref (pad);
1803 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
1810 request_rtcp_encoder (GstElement * rtpbin, guint session,
1811 GstRTSPStream * stream)
1813 GstRTSPStreamPrivate *priv = stream->priv;
1814 GstElement *oldenc, *enc;
1818 if (priv->idx != session)
1821 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1823 oldenc = priv->srtpenc;
1824 enc = get_rtp_encoder (stream, session);
1825 name = g_strdup_printf ("rtcp_sink_%d", session);
1826 pad = gst_element_get_request_pad (enc, name);
1828 gst_object_unref (pad);
1831 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
1838 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1840 GstRTSPStreamPrivate *priv = stream->priv;
1843 GST_DEBUG ("request key %08x", ssrc);
1845 g_mutex_lock (&priv->lock);
1846 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1847 gst_caps_ref (caps);
1848 g_mutex_unlock (&priv->lock);
1854 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
1855 GstRTSPStream * stream)
1857 GstRTSPStreamPrivate *priv = stream->priv;
1859 if (priv->idx != session)
1862 if (priv->srtpdec == NULL) {
1865 name = g_strdup_printf ("srtpdec_%u", session);
1866 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1869 g_signal_connect (priv->srtpdec, "request-key",
1870 (GCallback) request_key, stream);
1872 return gst_object_ref (priv->srtpdec);
1876 * gst_rtsp_stream_request_aux_sender:
1877 * @stream: a #GstRTSPStream
1878 * @sessid: the session id
1880 * Creating a rtxsend bin
1882 * Returns: (transfer full): a #GstElement.
1887 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
1891 GstStructure *pt_map;
1896 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1898 pt = gst_rtsp_stream_get_pt (stream);
1899 pt_s = g_strdup_printf ("%u", pt);
1900 rtx_pt = stream->priv->rtx_pt;
1902 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
1904 bin = gst_bin_new (NULL);
1905 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
1906 pt_map = gst_structure_new ("application/x-rtp-pt-map",
1907 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1908 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
1909 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
1911 gst_structure_free (pt_map);
1912 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
1914 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
1915 name = g_strdup_printf ("src_%u", sessid);
1916 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
1918 gst_object_unref (pad);
1920 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
1921 name = g_strdup_printf ("sink_%u", sessid);
1922 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
1924 gst_object_unref (pad);
1930 * gst_rtsp_stream_set_pt_map:
1931 * @stream: a #GstRTSPStream
1935 * Configure a pt map between @pt and @caps.
1938 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
1940 GstRTSPStreamPrivate *priv = stream->priv;
1942 g_mutex_lock (&priv->lock);
1943 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
1944 g_mutex_unlock (&priv->lock);
1948 request_pt_map (GstElement * rtpbin, guint session, guint pt,
1949 GstRTSPStream * stream)
1951 GstRTSPStreamPrivate *priv = stream->priv;
1952 GstCaps *caps = NULL;
1954 g_mutex_lock (&priv->lock);
1956 if (priv->idx == session) {
1957 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
1959 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
1960 gst_caps_ref (caps);
1962 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
1966 g_mutex_unlock (&priv->lock);
1972 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
1974 GstRTSPStreamPrivate *priv = stream->priv;
1976 GstPadLinkReturn ret;
1979 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
1980 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
1982 name = gst_pad_get_name (pad);
1983 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
1989 if (priv->idx != sessid)
1992 if (gst_pad_is_linked (priv->sinkpad)) {
1993 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
1994 GST_DEBUG_PAD_NAME (priv->sinkpad));
1998 /* link the RTP pad to the session manager, it should not really fail unless
1999 * this is not really an RTP pad */
2000 ret = gst_pad_link (pad, priv->sinkpad);
2001 if (ret != GST_PAD_LINK_OK)
2003 priv->recv_rtp_src = gst_object_ref (pad);
2010 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2011 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2016 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2017 GstRTSPStream * stream)
2019 /* TODO: What to do here other than this? */
2020 GST_DEBUG ("Stream %p: Got EOS", stream);
2021 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2025 * gst_rtsp_stream_join_bin:
2026 * @stream: a #GstRTSPStream
2027 * @bin: (transfer none): a #GstBin to join
2028 * @rtpbin: (transfer none): a rtpbin element in @bin
2029 * @state: the target state of the new elements
2031 * Join the #GstBin @bin that contains the element @rtpbin.
2033 * @stream will link to @rtpbin, which must be inside @bin. The elements
2034 * added to @bin will be set to the state given in @state.
2036 * Returns: %TRUE on success.
2039 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2040 GstElement * rtpbin, GstState state)
2042 GstRTSPStreamPrivate *priv;
2046 GstPad *pad, *sinkpad, *selpad;
2047 GstPadLinkReturn ret;
2049 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2050 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2051 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2053 priv = stream->priv;
2055 g_mutex_lock (&priv->lock);
2056 if (priv->is_joined)
2059 /* create a session with the same index as the stream */
2062 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2064 if (!alloc_ports (stream))
2067 /* update the dscp qos field in the sinks */
2068 update_dscp_qos (stream);
2070 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2071 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2073 g_signal_connect (rtpbin, "request-rtp-encoder",
2074 (GCallback) request_rtp_encoder, stream);
2075 g_signal_connect (rtpbin, "request-rtcp-encoder",
2076 (GCallback) request_rtcp_encoder, stream);
2077 g_signal_connect (rtpbin, "request-rtp-decoder",
2078 (GCallback) request_rtp_rtcp_decoder, stream);
2079 g_signal_connect (rtpbin, "request-rtcp-decoder",
2080 (GCallback) request_rtp_rtcp_decoder, stream);
2083 if (priv->sinkpad) {
2084 g_signal_connect (rtpbin, "request-pt-map",
2085 (GCallback) request_pt_map, stream);
2088 /* get a pad for sending RTP */
2089 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2090 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2094 /* link the RTP pad to the session manager, it should not really fail unless
2095 * this is not really an RTP pad */
2096 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2097 if (ret != GST_PAD_LINK_OK)
2100 /* Need to connect our sinkpad from here */
2101 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2103 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2106 /* get pads from the RTP session element for sending and receiving
2108 name = g_strdup_printf ("send_rtp_src_%u", idx);
2109 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2111 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2112 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2115 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2116 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2118 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2119 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2122 /* get the session */
2123 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2125 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2127 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2129 g_signal_connect (priv->session, "on-ssrc-active",
2130 (GCallback) on_ssrc_active, stream);
2131 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2133 g_signal_connect (priv->session, "on-bye-timeout",
2134 (GCallback) on_bye_timeout, stream);
2135 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2138 for (i = 0; i < 2; i++) {
2139 GstPad *teepad, *queuepad;
2140 /* For the sender we create this bit of pipeline for both
2141 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2142 * we need to add a queue before appsink and udpsink to make
2143 * the pipeline not block. For the TCP case, we want to pump
2144 * data to the client as fast as possible.
2146 * .--------. .-----. .---------. .---------.
2147 * | rtpbin | | tee | | queue | | udpsink |
2148 * | send->sink src->sink src->sink |
2149 * '--------' | | '---------' '---------'
2150 * | | .---------. .---------.
2151 * | | | queue | | appsink |
2152 * | src->sink src->sink |
2153 * '-----' '---------' '---------'
2155 * When only UDP is allowed, we skip the tee, queue and appsink and link the
2156 * udpsink directly to the session.
2159 gst_bin_add (bin, priv->udpsink[i]);
2160 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2162 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
2163 /* make tee for RTP/RTCP */
2164 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2165 gst_bin_add (bin, priv->tee[i]);
2167 /* and link to rtpbin send pad */
2168 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2169 gst_pad_link (priv->send_src[i], pad);
2170 gst_object_unref (pad);
2172 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2173 g_object_set (priv->udpqueue[i], "max-size-buffers",
2174 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0), NULL);
2175 gst_bin_add (bin, priv->udpqueue[i]);
2176 /* link tee to udpqueue */
2177 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2178 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2179 gst_pad_link (teepad, pad);
2180 gst_object_unref (pad);
2181 gst_object_unref (teepad);
2183 /* link udpqueue to udpsink */
2184 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2185 gst_pad_link (queuepad, sinkpad);
2186 gst_object_unref (queuepad);
2189 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2190 g_object_set (priv->appqueue[i], "max-size-buffers",
2191 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0), NULL);
2192 gst_bin_add (bin, priv->appqueue[i]);
2193 /* and link to tee */
2194 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2195 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2196 gst_pad_link (teepad, pad);
2197 gst_object_unref (pad);
2198 gst_object_unref (teepad);
2201 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2202 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2203 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2204 gst_bin_add (bin, priv->appsink[i]);
2205 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2206 &sink_cb, stream, NULL);
2207 /* and link to queue */
2208 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2209 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2210 gst_pad_link (queuepad, pad);
2211 gst_object_unref (pad);
2212 gst_object_unref (queuepad);
2214 /* else only udpsink needed, link it to the session */
2215 gst_pad_link (priv->send_src[i], sinkpad);
2217 gst_object_unref (sinkpad);
2219 /* For the receiver we create this bit of pipeline for both
2220 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2221 * and it is all funneled into the rtpbin receive pad.
2223 * .--------. .--------. .--------.
2224 * | udpsrc | | funnel | | rtpbin |
2225 * | src->sink src->sink |
2226 * '--------' | | '--------'
2230 * '--------' '--------'
2232 /* make funnel for the RTP/RTCP receivers */
2233 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2234 gst_bin_add (bin, priv->funnel[i]);
2236 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2237 gst_pad_link (pad, priv->recv_sink[i]);
2238 gst_object_unref (pad);
2240 if (priv->udpsrc_v4[i]) {
2242 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2243 * values. This is only relevant for PLAY pipelines */
2244 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2245 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2248 gst_bin_add (bin, priv->udpsrc_v4[i]);
2250 /* and link to the funnel v4 */
2251 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2252 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2253 gst_pad_link (pad, selpad);
2254 gst_object_unref (pad);
2255 gst_object_unref (selpad);
2258 if (priv->udpsrc_v6[i]) {
2260 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2261 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2263 gst_bin_add (bin, priv->udpsrc_v6[i]);
2265 /* and link to the funnel v6 */
2266 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2267 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2268 gst_pad_link (pad, selpad);
2269 gst_object_unref (pad);
2270 gst_object_unref (selpad);
2273 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
2274 /* make and add appsrc */
2275 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2276 priv->appsrc_base_time[i] = -1;
2277 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2278 gst_bin_add (bin, priv->appsrc[i]);
2279 /* and link to the funnel */
2280 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2281 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2282 gst_pad_link (pad, selpad);
2283 gst_object_unref (pad);
2284 gst_object_unref (selpad);
2287 /* check if we need to set to a special state */
2288 if (state != GST_STATE_NULL) {
2289 if (priv->udpsink[i])
2290 gst_element_set_state (priv->udpsink[i], state);
2291 if (priv->appsink[i])
2292 gst_element_set_state (priv->appsink[i], state);
2293 if (priv->appqueue[i])
2294 gst_element_set_state (priv->appqueue[i], state);
2295 if (priv->udpqueue[i])
2296 gst_element_set_state (priv->udpqueue[i], state);
2298 gst_element_set_state (priv->tee[i], state);
2299 if (priv->funnel[i])
2300 gst_element_set_state (priv->funnel[i], state);
2301 if (priv->appsrc[i])
2302 gst_element_set_state (priv->appsrc[i], state);
2306 /* be notified of caps changes */
2307 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2308 (GCallback) caps_notify, stream);
2310 priv->is_joined = TRUE;
2311 g_mutex_unlock (&priv->lock);
2318 g_mutex_unlock (&priv->lock);
2323 g_mutex_unlock (&priv->lock);
2324 GST_WARNING ("failed to allocate ports %u", idx);
2329 GST_WARNING ("failed to link stream %u", idx);
2330 gst_object_unref (priv->send_rtp_sink);
2331 priv->send_rtp_sink = NULL;
2332 g_mutex_unlock (&priv->lock);
2338 * gst_rtsp_stream_leave_bin:
2339 * @stream: a #GstRTSPStream
2340 * @bin: (transfer none): a #GstBin
2341 * @rtpbin: (transfer none): a rtpbin #GstElement
2343 * Remove the elements of @stream from @bin.
2345 * Return: %TRUE on success.
2348 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2349 GstElement * rtpbin)
2351 GstRTSPStreamPrivate *priv;
2355 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2356 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2357 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2359 priv = stream->priv;
2361 g_mutex_lock (&priv->lock);
2362 if (!priv->is_joined)
2363 goto was_not_joined;
2365 /* all transports must be removed by now */
2366 if (priv->transports != NULL)
2367 goto transports_not_removed;
2369 clear_tr_cache (priv, TRUE);
2370 clear_tr_cache (priv, FALSE);
2372 GST_INFO ("stream %p leaving bin", stream);
2375 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2376 } else if (priv->recv_rtp_src) {
2377 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2378 gst_object_unref (priv->recv_rtp_src);
2379 priv->recv_rtp_src = NULL;
2381 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2382 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2383 gst_object_unref (priv->send_rtp_sink);
2384 priv->send_rtp_sink = NULL;
2386 for (i = 0; i < 2; i++) {
2387 if (priv->udpsink[i])
2388 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2389 if (priv->appsink[i])
2390 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2391 if (priv->appqueue[i])
2392 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2393 if (priv->udpqueue[i])
2394 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2396 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2397 if (priv->funnel[i])
2398 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2399 if (priv->appsrc[i])
2400 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2401 if (priv->udpsrc_v4[i]) {
2402 /* and set udpsrc to NULL now before removing */
2403 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2404 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2405 /* removing them should also nicely release the request
2406 * pads when they finalize */
2407 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2409 if (priv->udpsrc_v6[i]) {
2410 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2411 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2412 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2415 for (l = priv->transport_sources; l; l = l->next) {
2416 GstRTSPMulticastTransportSource *s = l->data;
2421 gst_element_set_locked_state (s->udpsrc[i], FALSE);
2422 gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
2423 gst_bin_remove (bin, s->udpsrc[i]);
2426 if (priv->udpsink[i])
2427 gst_bin_remove (bin, priv->udpsink[i]);
2428 if (priv->appsrc[i])
2429 gst_bin_remove (bin, priv->appsrc[i]);
2430 if (priv->appsink[i])
2431 gst_bin_remove (bin, priv->appsink[i]);
2432 if (priv->appqueue[i])
2433 gst_bin_remove (bin, priv->appqueue[i]);
2434 if (priv->udpqueue[i])
2435 gst_bin_remove (bin, priv->udpqueue[i]);
2437 gst_bin_remove (bin, priv->tee[i]);
2438 if (priv->funnel[i])
2439 gst_bin_remove (bin, priv->funnel[i]);
2441 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2442 gst_object_unref (priv->recv_sink[i]);
2443 priv->recv_sink[i] = NULL;
2445 priv->udpsrc_v4[i] = NULL;
2446 priv->udpsrc_v6[i] = NULL;
2447 priv->udpsink[i] = NULL;
2448 priv->appsrc[i] = NULL;
2449 priv->appsink[i] = NULL;
2450 priv->appqueue[i] = NULL;
2451 priv->udpqueue[i] = NULL;
2452 priv->tee[i] = NULL;
2453 priv->funnel[i] = NULL;
2456 for (l = priv->transport_sources; l; l = l->next) {
2457 GstRTSPMulticastTransportSource *s = l->data;
2458 g_slice_free (GstRTSPMulticastTransportSource, s);
2460 g_list_free (priv->transport_sources);
2461 priv->transport_sources = NULL;
2463 gst_object_unref (priv->send_src[0]);
2464 priv->send_src[0] = NULL;
2466 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2467 gst_object_unref (priv->send_src[1]);
2468 priv->send_src[1] = NULL;
2470 g_object_unref (priv->session);
2471 priv->session = NULL;
2473 gst_caps_unref (priv->caps);
2477 gst_object_unref (priv->srtpenc);
2479 gst_object_unref (priv->srtpdec);
2481 priv->is_joined = FALSE;
2482 g_mutex_unlock (&priv->lock);
2488 g_mutex_unlock (&priv->lock);
2491 transports_not_removed:
2493 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2494 g_mutex_unlock (&priv->lock);
2500 * gst_rtsp_stream_get_rtpinfo:
2501 * @stream: a #GstRTSPStream
2502 * @rtptime: (allow-none): result RTP timestamp
2503 * @seq: (allow-none): result RTP seqnum
2504 * @clock_rate: (allow-none): the clock rate
2505 * @running_time: (allow-none): result running-time
2507 * Retrieve the current rtptime, seq and running-time. This is used to
2508 * construct a RTPInfo reply header.
2510 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2513 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2514 guint * rtptime, guint * seq, guint * clock_rate,
2515 GstClockTime * running_time)
2517 GstRTSPStreamPrivate *priv;
2518 GstStructure *stats;
2519 GObjectClass *payobjclass;
2521 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2523 priv = stream->priv;
2525 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2527 g_mutex_lock (&priv->lock);
2529 /* First try to extract the information from the last buffer on the sinks.
2530 * This will have a more accurate sequence number and timestamp, as between
2531 * the payloader and the sink there can be some queues
2533 if (priv->udpsink[0] || priv->appsink[0]) {
2534 GstSample *last_sample;
2536 if (priv->udpsink[0])
2537 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
2539 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
2544 GstSegment *segment;
2545 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
2547 caps = gst_sample_get_caps (last_sample);
2548 buffer = gst_sample_get_buffer (last_sample);
2549 segment = gst_sample_get_segment (last_sample);
2551 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
2553 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
2557 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
2560 gst_rtp_buffer_unmap (&rtp_buffer);
2564 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
2565 GST_BUFFER_TIMESTAMP (buffer));
2569 GstStructure *s = gst_caps_get_structure (caps, 0);
2571 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
2573 if (*clock_rate == 0 && running_time)
2574 *running_time = GST_CLOCK_TIME_NONE;
2576 gst_sample_unref (last_sample);
2580 gst_sample_unref (last_sample);
2585 if (g_object_class_find_property (payobjclass, "stats")) {
2586 g_object_get (priv->payloader, "stats", &stats, NULL);
2591 gst_structure_get_uint (stats, "seqnum", seq);
2594 gst_structure_get_uint (stats, "timestamp", rtptime);
2597 gst_structure_get_clock_time (stats, "running-time", running_time);
2600 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2601 if (*clock_rate == 0 && running_time)
2602 *running_time = GST_CLOCK_TIME_NONE;
2604 gst_structure_free (stats);
2606 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2607 !g_object_class_find_property (payobjclass, "timestamp"))
2611 g_object_get (priv->payloader, "seqnum", seq, NULL);
2614 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2617 *running_time = GST_CLOCK_TIME_NONE;
2621 g_mutex_unlock (&priv->lock);
2628 GST_WARNING ("Could not get payloader stats");
2629 g_mutex_unlock (&priv->lock);
2635 * gst_rtsp_stream_get_caps:
2636 * @stream: a #GstRTSPStream
2638 * Retrieve the current caps of @stream.
2640 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2644 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2646 GstRTSPStreamPrivate *priv;
2649 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2651 priv = stream->priv;
2653 g_mutex_lock (&priv->lock);
2654 if ((result = priv->caps))
2655 gst_caps_ref (result);
2656 g_mutex_unlock (&priv->lock);
2662 * gst_rtsp_stream_recv_rtp:
2663 * @stream: a #GstRTSPStream
2664 * @buffer: (transfer full): a #GstBuffer
2666 * Handle an RTP buffer for the stream. This method is usually called when a
2667 * message has been received from a client using the TCP transport.
2669 * This function takes ownership of @buffer.
2671 * Returns: a GstFlowReturn.
2674 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2676 GstRTSPStreamPrivate *priv;
2678 GstElement *element;
2680 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2681 priv = stream->priv;
2682 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2683 g_return_val_if_fail (priv->is_joined, FALSE);
2685 g_mutex_lock (&priv->lock);
2686 if (priv->appsrc[0])
2687 element = gst_object_ref (priv->appsrc[0]);
2690 g_mutex_unlock (&priv->lock);
2693 if (priv->appsrc_base_time[0] == -1) {
2694 /* Take current running_time. This timestamp will be put on
2695 * the first buffer of each stream because we are a live source and so we
2696 * timestamp with the running_time. When we are dealing with TCP, we also
2697 * only timestamp the first buffer (using the DISCONT flag) because a server
2698 * typically bursts data, for which we don't want to compensate by speeding
2699 * up the media. The other timestamps will be interpollated from this one
2700 * using the RTP timestamps. */
2701 GST_OBJECT_LOCK (element);
2702 if (GST_ELEMENT_CLOCK (element)) {
2704 GstClockTime base_time;
2706 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
2707 base_time = GST_ELEMENT_CAST (element)->base_time;
2709 priv->appsrc_base_time[0] = now - base_time;
2710 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
2711 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
2712 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
2713 GST_TIME_ARGS (base_time));
2715 GST_OBJECT_UNLOCK (element);
2718 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2719 gst_object_unref (element);
2727 * gst_rtsp_stream_recv_rtcp:
2728 * @stream: a #GstRTSPStream
2729 * @buffer: (transfer full): a #GstBuffer
2731 * Handle an RTCP buffer for the stream. This method is usually called when a
2732 * message has been received from a client using the TCP transport.
2734 * This function takes ownership of @buffer.
2736 * Returns: a GstFlowReturn.
2739 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2741 GstRTSPStreamPrivate *priv;
2743 GstElement *element;
2745 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2746 priv = stream->priv;
2747 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2749 if (!priv->is_joined) {
2750 gst_buffer_unref (buffer);
2751 return GST_FLOW_NOT_LINKED;
2753 g_mutex_lock (&priv->lock);
2754 if (priv->appsrc[1])
2755 element = gst_object_ref (priv->appsrc[1]);
2758 g_mutex_unlock (&priv->lock);
2761 if (priv->appsrc_base_time[1] == -1) {
2762 /* Take current running_time. This timestamp will be put on
2763 * the first buffer of each stream because we are a live source and so we
2764 * timestamp with the running_time. When we are dealing with TCP, we also
2765 * only timestamp the first buffer (using the DISCONT flag) because a server
2766 * typically bursts data, for which we don't want to compensate by speeding
2767 * up the media. The other timestamps will be interpollated from this one
2768 * using the RTP timestamps. */
2769 GST_OBJECT_LOCK (element);
2770 if (GST_ELEMENT_CLOCK (element)) {
2772 GstClockTime base_time;
2774 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
2775 base_time = GST_ELEMENT_CAST (element)->base_time;
2777 priv->appsrc_base_time[1] = now - base_time;
2778 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
2779 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
2780 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
2781 GST_TIME_ARGS (base_time));
2783 GST_OBJECT_UNLOCK (element);
2786 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2787 gst_object_unref (element);
2790 gst_buffer_unref (buffer);
2795 /* must be called with lock */
2797 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
2800 GstRTSPStreamPrivate *priv = stream->priv;
2801 const GstRTSPTransport *tr;
2803 tr = gst_rtsp_stream_transport_get_transport (trans);
2805 switch (tr->lower_transport) {
2806 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2808 GstRTSPMulticastTransportSource *source;
2811 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[0])));
2816 GstPad *selpad, *pad;
2818 source = g_slice_new0 (GstRTSPMulticastTransportSource);
2819 source->transport = trans;
2821 for (i = 0; i < 2; i++) {
2823 g_strdup_printf ("udp://%s:%d", tr->destination,
2824 (i == 0) ? tr->port.min : tr->port.max);
2826 gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2830 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2831 * values. This is only relevant for PLAY pipelines */
2832 gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
2833 gst_element_set_locked_state (source->udpsrc[i], TRUE);
2836 gst_bin_add (bin, source->udpsrc[i]);
2838 /* and link to the funnel v4 */
2839 source->selpad[i] = selpad =
2840 gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2841 pad = gst_element_get_static_pad (source->udpsrc[i], "src");
2842 gst_pad_link (pad, selpad);
2843 gst_object_unref (pad);
2844 gst_object_unref (selpad);
2846 gst_object_unref (bin);
2848 priv->transport_sources =
2849 g_list_prepend (priv->transport_sources, source);
2853 for (l = priv->transport_sources; l; l = l->next) {
2856 if (source->transport == trans) {
2857 priv->transport_sources =
2858 g_list_delete_link (priv->transport_sources, l);
2866 for (i = 0; i < 2; i++) {
2867 /* Will automatically unlink everything */
2868 gst_bin_remove (bin,
2869 GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
2871 gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
2872 gst_object_unref (source->udpsrc[i]);
2874 gst_element_release_request_pad (priv->funnel[i],
2878 g_slice_free (GstRTSPMulticastTransportSource, source);
2882 /* fall through for the generic case */
2884 case GST_RTSP_LOWER_TRANS_UDP:
2890 dest = tr->destination;
2891 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2896 min = tr->client_port.min;
2897 max = tr->client_port.max;
2902 GST_INFO ("setting ttl-mc %d", ttl);
2903 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
2904 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
2906 GST_INFO ("adding %s:%d-%d", dest, min, max);
2907 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
2908 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
2909 priv->transports = g_list_prepend (priv->transports, trans);
2911 GST_INFO ("removing %s:%d-%d", dest, min, max);
2912 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
2913 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
2914 priv->transports = g_list_remove (priv->transports, trans);
2916 priv->transports_cookie++;
2919 case GST_RTSP_LOWER_TRANS_TCP:
2921 GST_INFO ("adding TCP %s", tr->destination);
2922 priv->transports = g_list_prepend (priv->transports, trans);
2924 GST_INFO ("removing TCP %s", tr->destination);
2925 priv->transports = g_list_remove (priv->transports, trans);
2927 priv->transports_cookie++;
2930 goto unknown_transport;
2937 GST_INFO ("Unknown transport %d", tr->lower_transport);
2944 * gst_rtsp_stream_add_transport:
2945 * @stream: a #GstRTSPStream
2946 * @trans: (transfer none): a #GstRTSPStreamTransport
2948 * Add the transport in @trans to @stream. The media of @stream will
2949 * then also be send to the values configured in @trans.
2951 * @stream must be joined to a bin.
2953 * @trans must contain a valid #GstRTSPTransport.
2955 * Returns: %TRUE if @trans was added
2958 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
2959 GstRTSPStreamTransport * trans)
2961 GstRTSPStreamPrivate *priv;
2964 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2965 priv = stream->priv;
2966 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2967 g_return_val_if_fail (priv->is_joined, FALSE);
2969 g_mutex_lock (&priv->lock);
2970 res = update_transport (stream, trans, TRUE);
2971 g_mutex_unlock (&priv->lock);
2977 * gst_rtsp_stream_remove_transport:
2978 * @stream: a #GstRTSPStream
2979 * @trans: (transfer none): a #GstRTSPStreamTransport
2981 * Remove the transport in @trans from @stream. The media of @stream will
2982 * not be sent to the values configured in @trans.
2984 * @stream must be joined to a bin.
2986 * @trans must contain a valid #GstRTSPTransport.
2988 * Returns: %TRUE if @trans was removed
2991 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2992 GstRTSPStreamTransport * trans)
2994 GstRTSPStreamPrivate *priv;
2997 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2998 priv = stream->priv;
2999 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3000 g_return_val_if_fail (priv->is_joined, FALSE);
3002 g_mutex_lock (&priv->lock);
3003 res = update_transport (stream, trans, FALSE);
3004 g_mutex_unlock (&priv->lock);
3010 * gst_rtsp_stream_update_crypto:
3011 * @stream: a #GstRTSPStream
3013 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3015 * Update the new crypto information for @ssrc in @stream. If information
3016 * for @ssrc did not exist, it will be added. If information
3017 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3018 * be removed from @stream.
3020 * Returns: %TRUE if @crypto could be updated
3023 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3024 guint ssrc, GstCaps * crypto)
3026 GstRTSPStreamPrivate *priv;
3028 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3029 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3031 priv = stream->priv;
3033 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3035 g_mutex_lock (&priv->lock);
3037 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3038 gst_caps_ref (crypto));
3040 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3041 g_mutex_unlock (&priv->lock);
3047 * gst_rtsp_stream_get_rtp_socket:
3048 * @stream: a #GstRTSPStream
3049 * @family: the socket family
3051 * Get the RTP socket from @stream for a @family.
3053 * @stream must be joined to a bin.
3055 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3056 * socket could be allocated for @family. Unref after usage
3059 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3061 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3065 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3066 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3067 family == G_SOCKET_FAMILY_IPV6, NULL);
3068 g_return_val_if_fail (priv->udpsink[0], NULL);
3070 if (family == G_SOCKET_FAMILY_IPV6)
3075 g_object_get (priv->udpsink[0], name, &socket, NULL);
3081 * gst_rtsp_stream_get_rtcp_socket:
3082 * @stream: a #GstRTSPStream
3083 * @family: the socket family
3085 * Get the RTCP socket from @stream for a @family.
3087 * @stream must be joined to a bin.
3089 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3090 * socket could be allocated for @family. Unref after usage
3093 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3095 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3099 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3100 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3101 family == G_SOCKET_FAMILY_IPV6, NULL);
3102 g_return_val_if_fail (priv->udpsink[1], NULL);
3104 if (family == G_SOCKET_FAMILY_IPV6)
3109 g_object_get (priv->udpsink[1], name, &socket, NULL);
3115 * gst_rtsp_stream_set_seqnum:
3116 * @stream: a #GstRTSPStream
3117 * @seqnum: a new sequence number
3119 * Configure the sequence number in the payloader of @stream to @seqnum.
3122 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3124 GstRTSPStreamPrivate *priv;
3126 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3128 priv = stream->priv;
3130 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3134 * gst_rtsp_stream_get_seqnum:
3135 * @stream: a #GstRTSPStream
3137 * Get the configured sequence number in the payloader of @stream.
3139 * Returns: the sequence number of the payloader.
3142 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3144 GstRTSPStreamPrivate *priv;
3147 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3149 priv = stream->priv;
3151 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3157 * gst_rtsp_stream_transport_filter:
3158 * @stream: a #GstRTSPStream
3159 * @func: (scope call) (allow-none): a callback
3160 * @user_data: (closure): user data passed to @func
3162 * Call @func for each transport managed by @stream. The result value of @func
3163 * determines what happens to the transport. @func will be called with @stream
3164 * locked so no further actions on @stream can be performed from @func.
3166 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3169 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3171 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3172 * will also be added with an additional ref to the result #GList of this
3175 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3177 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3178 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3179 * element in the #GList should be unreffed before the list is freed.
3182 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3183 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3185 GstRTSPStreamPrivate *priv;
3186 GList *result, *walk, *next;
3187 GHashTable *visited = NULL;
3190 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3192 priv = stream->priv;
3196 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3198 g_mutex_lock (&priv->lock);
3200 cookie = priv->transports_cookie;
3201 for (walk = priv->transports; walk; walk = next) {
3202 GstRTSPStreamTransport *trans = walk->data;
3203 GstRTSPFilterResult res;
3206 next = g_list_next (walk);
3209 /* only visit each transport once */
3210 if (g_hash_table_contains (visited, trans))
3213 g_hash_table_add (visited, g_object_ref (trans));
3214 g_mutex_unlock (&priv->lock);
3216 res = func (stream, trans, user_data);
3218 g_mutex_lock (&priv->lock);
3220 res = GST_RTSP_FILTER_REF;
3222 changed = (cookie != priv->transports_cookie);
3225 case GST_RTSP_FILTER_REMOVE:
3226 update_transport (stream, trans, FALSE);
3228 case GST_RTSP_FILTER_REF:
3229 result = g_list_prepend (result, g_object_ref (trans));
3231 case GST_RTSP_FILTER_KEEP:
3238 g_mutex_unlock (&priv->lock);
3241 g_hash_table_unref (visited);
3246 static GstPadProbeReturn
3247 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3249 GstRTSPStreamPrivate *priv;
3250 GstRTSPStream *stream;
3253 priv = stream->priv;
3255 GST_DEBUG_OBJECT (pad, "now blocking");
3257 g_mutex_lock (&priv->lock);
3258 priv->blocking = TRUE;
3259 g_mutex_unlock (&priv->lock);
3261 gst_element_post_message (priv->payloader,
3262 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3263 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3265 return GST_PAD_PROBE_OK;
3269 * gst_rtsp_stream_set_blocked:
3270 * @stream: a #GstRTSPStream
3271 * @blocked: boolean indicating we should block or unblock
3273 * Blocks or unblocks the dataflow on @stream.
3275 * Returns: %TRUE on success
3278 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3280 GstRTSPStreamPrivate *priv;
3282 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3284 priv = stream->priv;
3286 g_mutex_lock (&priv->lock);
3288 priv->blocking = FALSE;
3289 if (priv->blocked_id == 0) {
3290 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3291 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3292 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3293 g_object_ref (stream), g_object_unref);
3296 if (priv->blocked_id != 0) {
3297 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3298 priv->blocked_id = 0;
3299 priv->blocking = FALSE;
3302 g_mutex_unlock (&priv->lock);
3308 * gst_rtsp_stream_is_blocking:
3309 * @stream: a #GstRTSPStream
3311 * Check if @stream is blocking on a #GstBuffer.
3313 * Returns: %TRUE if @stream is blocking
3316 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3318 GstRTSPStreamPrivate *priv;
3321 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3323 priv = stream->priv;
3325 g_mutex_lock (&priv->lock);
3326 result = priv->blocking;
3327 g_mutex_unlock (&priv->lock);
3333 * gst_rtsp_stream_query_position:
3334 * @stream: a #GstRTSPStream
3336 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3337 * the RTP parts of the pipeline and not the RTCP parts.
3339 * Returns: %TRUE if the position could be queried
3342 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3344 GstRTSPStreamPrivate *priv;
3348 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3350 priv = stream->priv;
3352 g_mutex_lock (&priv->lock);
3353 if ((sink = priv->udpsink[0]))
3354 gst_object_ref (sink);
3355 g_mutex_unlock (&priv->lock);
3360 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3361 gst_object_unref (sink);
3367 * gst_rtsp_stream_query_stop:
3368 * @stream: a #GstRTSPStream
3370 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3371 * the RTP parts of the pipeline and not the RTCP parts.
3373 * Returns: %TRUE if the stop could be queried
3376 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3378 GstRTSPStreamPrivate *priv;
3383 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3385 priv = stream->priv;
3387 g_mutex_lock (&priv->lock);
3388 if ((sink = priv->udpsink[0]))
3389 gst_object_ref (sink);
3390 g_mutex_unlock (&priv->lock);
3395 query = gst_query_new_segment (GST_FORMAT_TIME);
3396 if ((ret = gst_element_query (sink, query))) {
3399 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3400 if (format != GST_FORMAT_TIME)
3403 gst_query_unref (query);
3404 gst_object_unref (sink);