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 "rtsp-stream.h"
60 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
61 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
65 GstRTSPStreamTransport *transport;
67 /* RTP and RTCP source */
68 GstElement *udpsrc[2];
70 } GstRTSPMulticastTransportSource;
72 struct _GstRTSPStreamPrivate
76 /* Only one pad is ever set */
77 GstPad *srcpad, *sinkpad;
78 GstElement *payloader;
83 GstRTSPProfile profiles;
84 GstRTSPLowerTrans protocols;
86 /* pads on the rtpbin */
87 GstPad *send_rtp_sink;
92 /* the RTPSession object */
95 /* SRTP encoder/decoder */
100 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
102 GstElement *udpsrc_v4[2];
104 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
106 GstElement *udpsrc_v6[2];
108 GstElement *udpqueue[2];
109 GstElement *udpsink[2];
111 /* for TCP transport */
112 GstElement *appsrc[2];
113 GstClockTime appsrc_base_time[2];
114 GstElement *appqueue[2];
115 GstElement *appsink[2];
118 GstElement *funnel[2];
123 GstClockTime rtx_time;
125 /* server ports for sending/receiving over ipv4 */
126 GstRTSPRange server_port_v4;
127 GstRTSPAddress *server_addr_v4;
130 /* server ports for sending/receiving over ipv6 */
131 GstRTSPRange server_port_v6;
132 GstRTSPAddress *server_addr_v6;
135 /* multicast addresses */
136 GstRTSPAddressPool *pool;
137 GstRTSPAddress *addr_v4;
138 GstRTSPAddress *addr_v6;
140 /* the caps of the stream */
144 /* transports we stream to */
147 guint transports_cookie;
149 GList *tr_cache_rtcp;
150 guint tr_cache_cookie_rtp;
151 guint tr_cache_cookie_rtcp;
154 /* UDP sources for UDP multicast transports */
155 GList *transport_sources;
159 /* stream blocking */
163 /* pt->caps map for RECORD streams */
167 #define DEFAULT_CONTROL NULL
168 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
169 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
170 GST_RTSP_LOWER_TRANS_TCP
183 SIGNAL_NEW_RTP_ENCODER,
184 SIGNAL_NEW_RTCP_ENCODER,
188 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
189 #define GST_CAT_DEFAULT rtsp_stream_debug
191 static GQuark ssrc_stream_map_key;
193 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
194 GValue * value, GParamSpec * pspec);
195 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
196 const GValue * value, GParamSpec * pspec);
198 static void gst_rtsp_stream_finalize (GObject * obj);
200 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
202 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
205 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
207 GObjectClass *gobject_class;
209 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
211 gobject_class = G_OBJECT_CLASS (klass);
213 gobject_class->get_property = gst_rtsp_stream_get_property;
214 gobject_class->set_property = gst_rtsp_stream_set_property;
215 gobject_class->finalize = gst_rtsp_stream_finalize;
217 g_object_class_install_property (gobject_class, PROP_CONTROL,
218 g_param_spec_string ("control", "Control",
219 "The control string for this stream", DEFAULT_CONTROL,
220 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
222 g_object_class_install_property (gobject_class, PROP_PROFILES,
223 g_param_spec_flags ("profiles", "Profiles",
224 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
225 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
227 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
228 g_param_spec_flags ("protocols", "Protocols",
229 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
230 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
232 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
233 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
234 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
235 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
237 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
238 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
239 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
240 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
242 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
244 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
248 gst_rtsp_stream_init (GstRTSPStream * stream)
250 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
252 GST_DEBUG ("new stream %p", stream);
257 priv->control = g_strdup (DEFAULT_CONTROL);
258 priv->profiles = DEFAULT_PROFILES;
259 priv->protocols = DEFAULT_PROTOCOLS;
261 g_mutex_init (&priv->lock);
263 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
264 NULL, (GDestroyNotify) gst_caps_unref);
265 priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
266 (GDestroyNotify) gst_caps_unref);
270 gst_rtsp_stream_finalize (GObject * obj)
272 GstRTSPStream *stream;
273 GstRTSPStreamPrivate *priv;
275 stream = GST_RTSP_STREAM (obj);
278 GST_DEBUG ("finalize stream %p", stream);
280 /* we really need to be unjoined now */
281 g_return_if_fail (!priv->is_joined);
284 gst_rtsp_address_free (priv->addr_v4);
286 gst_rtsp_address_free (priv->addr_v6);
287 if (priv->server_addr_v4)
288 gst_rtsp_address_free (priv->server_addr_v4);
289 if (priv->server_addr_v6)
290 gst_rtsp_address_free (priv->server_addr_v6);
292 g_object_unref (priv->pool);
294 g_object_unref (priv->rtxsend);
296 gst_object_unref (priv->payloader);
298 gst_object_unref (priv->srcpad);
300 gst_object_unref (priv->sinkpad);
301 g_free (priv->control);
302 g_mutex_clear (&priv->lock);
304 g_hash_table_unref (priv->keys);
305 g_hash_table_destroy (priv->ptmap);
307 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
311 gst_rtsp_stream_get_property (GObject * object, guint propid,
312 GValue * value, GParamSpec * pspec)
314 GstRTSPStream *stream = GST_RTSP_STREAM (object);
318 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
321 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
324 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
327 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
332 gst_rtsp_stream_set_property (GObject * object, guint propid,
333 const GValue * value, GParamSpec * pspec)
335 GstRTSPStream *stream = GST_RTSP_STREAM (object);
339 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
342 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
345 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
348 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
353 * gst_rtsp_stream_new:
356 * @payloader: a #GstElement
358 * Create a new media stream with index @idx that handles RTP data on
359 * @pad and has a payloader element @payloader if @pad is a source pad
360 * or a depayloader element @payloader if @pad is a sink pad.
362 * Returns: (transfer full): a new #GstRTSPStream
365 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
367 GstRTSPStreamPrivate *priv;
368 GstRTSPStream *stream;
370 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
371 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
373 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
376 priv->payloader = gst_object_ref (payloader);
377 if (GST_PAD_IS_SRC (pad))
378 priv->srcpad = gst_object_ref (pad);
380 priv->sinkpad = gst_object_ref (pad);
386 * gst_rtsp_stream_get_index:
387 * @stream: a #GstRTSPStream
389 * Get the stream index.
391 * Return: the stream index.
394 gst_rtsp_stream_get_index (GstRTSPStream * stream)
396 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
398 return stream->priv->idx;
402 * gst_rtsp_stream_get_pt:
403 * @stream: a #GstRTSPStream
405 * Get the stream payload type.
407 * Return: the stream payload type.
410 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
412 GstRTSPStreamPrivate *priv;
415 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
419 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
425 * gst_rtsp_stream_get_srcpad:
426 * @stream: a #GstRTSPStream
428 * Get the srcpad associated with @stream.
430 * Returns: (transfer full): the srcpad. Unref after usage.
433 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
435 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
437 if (!stream->priv->srcpad)
440 return gst_object_ref (stream->priv->srcpad);
444 * gst_rtsp_stream_get_sinkpad:
445 * @stream: a #GstRTSPStream
447 * Get the sinkpad associated with @stream.
449 * Returns: (transfer full): the sinkpad. Unref after usage.
452 gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
454 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
456 if (!stream->priv->sinkpad)
459 return gst_object_ref (stream->priv->sinkpad);
463 * gst_rtsp_stream_get_control:
464 * @stream: a #GstRTSPStream
466 * Get the control string to identify this stream.
468 * Returns: (transfer full): the control string. g_free() after usage.
471 gst_rtsp_stream_get_control (GstRTSPStream * stream)
473 GstRTSPStreamPrivate *priv;
476 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
480 g_mutex_lock (&priv->lock);
481 if ((result = g_strdup (priv->control)) == NULL)
482 result = g_strdup_printf ("stream=%u", priv->idx);
483 g_mutex_unlock (&priv->lock);
489 * gst_rtsp_stream_set_control:
490 * @stream: a #GstRTSPStream
491 * @control: a control string
493 * Set the control string in @stream.
496 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
498 GstRTSPStreamPrivate *priv;
500 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
504 g_mutex_lock (&priv->lock);
505 g_free (priv->control);
506 priv->control = g_strdup (control);
507 g_mutex_unlock (&priv->lock);
511 * gst_rtsp_stream_has_control:
512 * @stream: a #GstRTSPStream
513 * @control: a control string
515 * Check if @stream has the control string @control.
517 * Returns: %TRUE is @stream has @control as the control string
520 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
522 GstRTSPStreamPrivate *priv;
525 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
529 g_mutex_lock (&priv->lock);
531 res = (g_strcmp0 (priv->control, control) == 0);
535 if (sscanf (control, "stream=%u", &streamid) > 0)
536 res = (streamid == priv->idx);
540 g_mutex_unlock (&priv->lock);
546 * gst_rtsp_stream_set_mtu:
547 * @stream: a #GstRTSPStream
550 * Configure the mtu in the payloader of @stream to @mtu.
553 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
555 GstRTSPStreamPrivate *priv;
557 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
561 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
563 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
567 * gst_rtsp_stream_get_mtu:
568 * @stream: a #GstRTSPStream
570 * Get the configured MTU in the payloader of @stream.
572 * Returns: the MTU of the payloader.
575 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
577 GstRTSPStreamPrivate *priv;
580 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
584 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
589 /* Update the dscp qos property on the udp sinks */
591 update_dscp_qos (GstRTSPStream * stream)
593 GstRTSPStreamPrivate *priv;
595 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
599 if (priv->udpsink[0]) {
600 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
604 if (priv->udpsink[1]) {
605 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
611 * gst_rtsp_stream_set_dscp_qos:
612 * @stream: a #GstRTSPStream
613 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
615 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
618 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
620 GstRTSPStreamPrivate *priv;
622 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
626 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
628 if (dscp_qos < -1 || dscp_qos > 63) {
629 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
633 priv->dscp_qos = dscp_qos;
635 update_dscp_qos (stream);
639 * gst_rtsp_stream_get_dscp_qos:
640 * @stream: a #GstRTSPStream
642 * Get the configured DSCP QoS in of the outgoing sockets.
644 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
647 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
649 GstRTSPStreamPrivate *priv;
651 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
655 return priv->dscp_qos;
659 * gst_rtsp_stream_is_transport_supported:
660 * @stream: a #GstRTSPStream
661 * @transport: (transfer none): a #GstRTSPTransport
663 * Check if @transport can be handled by stream
665 * Returns: %TRUE if @transport can be handled by @stream.
668 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
669 GstRTSPTransport * transport)
671 GstRTSPStreamPrivate *priv;
673 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
677 g_mutex_lock (&priv->lock);
678 if (transport->trans != GST_RTSP_TRANS_RTP)
679 goto unsupported_transmode;
681 if (!(transport->profile & priv->profiles))
682 goto unsupported_profile;
684 if (!(transport->lower_transport & priv->protocols))
685 goto unsupported_ltrans;
687 g_mutex_unlock (&priv->lock);
692 unsupported_transmode:
694 GST_DEBUG ("unsupported transport mode %d", transport->trans);
695 g_mutex_unlock (&priv->lock);
700 GST_DEBUG ("unsupported profile %d", transport->profile);
701 g_mutex_unlock (&priv->lock);
706 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
707 g_mutex_unlock (&priv->lock);
713 * gst_rtsp_stream_set_profiles:
714 * @stream: a #GstRTSPStream
715 * @profiles: the new profiles
717 * Configure the allowed profiles for @stream.
720 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
722 GstRTSPStreamPrivate *priv;
724 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
728 g_mutex_lock (&priv->lock);
729 priv->profiles = profiles;
730 g_mutex_unlock (&priv->lock);
734 * gst_rtsp_stream_get_profiles:
735 * @stream: a #GstRTSPStream
737 * Get the allowed profiles of @stream.
739 * Returns: a #GstRTSPProfile
742 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
744 GstRTSPStreamPrivate *priv;
747 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
751 g_mutex_lock (&priv->lock);
752 res = priv->profiles;
753 g_mutex_unlock (&priv->lock);
759 * gst_rtsp_stream_set_protocols:
760 * @stream: a #GstRTSPStream
761 * @protocols: the new flags
763 * Configure the allowed lower transport for @stream.
766 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
767 GstRTSPLowerTrans protocols)
769 GstRTSPStreamPrivate *priv;
771 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
775 g_mutex_lock (&priv->lock);
776 priv->protocols = protocols;
777 g_mutex_unlock (&priv->lock);
781 * gst_rtsp_stream_get_protocols:
782 * @stream: a #GstRTSPStream
784 * Get the allowed protocols of @stream.
786 * Returns: a #GstRTSPLowerTrans
789 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
791 GstRTSPStreamPrivate *priv;
792 GstRTSPLowerTrans res;
794 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
795 GST_RTSP_LOWER_TRANS_UNKNOWN);
799 g_mutex_lock (&priv->lock);
800 res = priv->protocols;
801 g_mutex_unlock (&priv->lock);
807 * gst_rtsp_stream_set_address_pool:
808 * @stream: a #GstRTSPStream
809 * @pool: (transfer none): a #GstRTSPAddressPool
811 * configure @pool to be used as the address pool of @stream.
814 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
815 GstRTSPAddressPool * pool)
817 GstRTSPStreamPrivate *priv;
818 GstRTSPAddressPool *old;
820 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
824 GST_LOG_OBJECT (stream, "set address pool %p", pool);
826 g_mutex_lock (&priv->lock);
827 if ((old = priv->pool) != pool)
828 priv->pool = pool ? g_object_ref (pool) : NULL;
831 g_mutex_unlock (&priv->lock);
834 g_object_unref (old);
838 * gst_rtsp_stream_get_address_pool:
839 * @stream: a #GstRTSPStream
841 * Get the #GstRTSPAddressPool used as the address pool of @stream.
843 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
847 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
849 GstRTSPStreamPrivate *priv;
850 GstRTSPAddressPool *result;
852 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
856 g_mutex_lock (&priv->lock);
857 if ((result = priv->pool))
858 g_object_ref (result);
859 g_mutex_unlock (&priv->lock);
865 * gst_rtsp_stream_get_multicast_address:
866 * @stream: a #GstRTSPStream
867 * @family: the #GSocketFamily
869 * Get the multicast address of @stream for @family.
871 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
872 * or %NULL when no address could be allocated. gst_rtsp_address_free()
876 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
877 GSocketFamily family)
879 GstRTSPStreamPrivate *priv;
880 GstRTSPAddress *result;
881 GstRTSPAddress **addrp;
882 GstRTSPAddressFlags flags;
884 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
888 if (family == G_SOCKET_FAMILY_IPV6) {
889 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
890 addrp = &priv->addr_v6;
892 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
893 addrp = &priv->addr_v4;
896 g_mutex_lock (&priv->lock);
897 if (*addrp == NULL) {
898 if (priv->pool == NULL)
901 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
903 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
907 result = gst_rtsp_address_copy (*addrp);
908 g_mutex_unlock (&priv->lock);
915 GST_ERROR_OBJECT (stream, "no address pool specified");
916 g_mutex_unlock (&priv->lock);
921 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
922 g_mutex_unlock (&priv->lock);
928 * gst_rtsp_stream_reserve_address:
929 * @stream: a #GstRTSPStream
930 * @address: an address
935 * Reserve @address and @port as the address and port of @stream.
937 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
938 * the address could be reserved. gst_rtsp_address_free() after usage.
941 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
942 const gchar * address, guint port, guint n_ports, guint ttl)
944 GstRTSPStreamPrivate *priv;
945 GstRTSPAddress *result;
947 GSocketFamily family;
948 GstRTSPAddress **addrp;
950 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
951 g_return_val_if_fail (address != NULL, NULL);
952 g_return_val_if_fail (port > 0, NULL);
953 g_return_val_if_fail (n_ports > 0, NULL);
954 g_return_val_if_fail (ttl > 0, NULL);
958 addr = g_inet_address_new_from_string (address);
960 GST_ERROR ("failed to get inet addr from %s", address);
961 family = G_SOCKET_FAMILY_IPV4;
963 family = g_inet_address_get_family (addr);
964 g_object_unref (addr);
967 if (family == G_SOCKET_FAMILY_IPV6)
968 addrp = &priv->addr_v6;
970 addrp = &priv->addr_v4;
972 g_mutex_lock (&priv->lock);
973 if (*addrp == NULL) {
974 GstRTSPAddressPoolResult res;
976 if (priv->pool == NULL)
979 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
980 port, n_ports, ttl, addrp);
981 if (res != GST_RTSP_ADDRESS_POOL_OK)
984 if (strcmp ((*addrp)->address, address) ||
985 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
986 (*addrp)->ttl != ttl)
987 goto different_address;
989 result = gst_rtsp_address_copy (*addrp);
990 g_mutex_unlock (&priv->lock);
997 GST_ERROR_OBJECT (stream, "no address pool specified");
998 g_mutex_unlock (&priv->lock);
1003 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
1005 g_mutex_unlock (&priv->lock);
1010 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
1011 " reserved", address);
1012 g_mutex_unlock (&priv->lock);
1018 alloc_ports_one_family (GstRTSPStream * stream, GstRTSPAddressPool * pool,
1019 gint buffer_size, GSocketFamily family, GstElement * udpsrc_out[2],
1020 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
1021 GstRTSPAddress ** server_addr_out)
1023 GstRTSPStreamPrivate *priv = stream->priv;
1024 GstStateChangeReturn ret;
1025 GstElement *udpsrc0, *udpsrc1;
1026 GstElement *udpsink0, *udpsink1;
1027 GSocket *rtp_socket = NULL;
1028 GSocket *rtcp_socket;
1029 gint tmp_rtp, tmp_rtcp;
1031 gint rtpport, rtcpport;
1032 GList *rejected_addresses = NULL;
1033 GstRTSPAddress *addr = NULL;
1034 GInetAddress *inetaddr = NULL;
1035 GSocketAddress *rtp_sockaddr = NULL;
1036 GSocketAddress *rtcp_sockaddr = NULL;
1037 const gchar *multisink_socket;
1039 if (family == G_SOCKET_FAMILY_IPV6)
1040 multisink_socket = "socket-v6";
1042 multisink_socket = "socket";
1050 /* Start with random port */
1053 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1054 G_SOCKET_PROTOCOL_UDP, NULL);
1056 goto no_udp_protocol;
1058 if (*server_addr_out)
1059 gst_rtsp_address_free (*server_addr_out);
1061 /* try to allocate 2 UDP ports, the RTP port should be an even
1062 * number and the RTCP port should be the next (uneven) port */
1065 if (rtp_socket == NULL) {
1066 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1067 G_SOCKET_PROTOCOL_UDP, NULL);
1069 goto no_udp_protocol;
1072 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
1073 GstRTSPAddressFlags flags;
1076 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1078 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1079 if (family == G_SOCKET_FAMILY_IPV6)
1080 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1082 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1084 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1089 tmp_rtp = addr->port;
1091 g_clear_object (&inetaddr);
1092 inetaddr = g_inet_address_new_from_string (addr->address);
1100 if (inetaddr == NULL)
1101 inetaddr = g_inet_address_new_any (family);
1104 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1105 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1106 g_object_unref (rtp_sockaddr);
1109 g_object_unref (rtp_sockaddr);
1111 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1112 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1113 g_clear_object (&rtp_sockaddr);
1118 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1119 g_object_unref (rtp_sockaddr);
1121 /* check if port is even */
1122 if ((tmp_rtp & 1) != 0) {
1123 /* port not even, close and allocate another */
1125 g_clear_object (&rtp_socket);
1130 tmp_rtcp = tmp_rtp + 1;
1132 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1133 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1134 g_object_unref (rtcp_sockaddr);
1135 g_clear_object (&rtp_socket);
1138 g_object_unref (rtcp_sockaddr);
1140 g_clear_object (&inetaddr);
1142 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1143 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1145 if (udpsrc0 == NULL || udpsrc1 == NULL)
1146 goto no_udp_protocol;
1148 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1149 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1151 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1152 if (ret == GST_STATE_CHANGE_FAILURE)
1154 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1155 if (ret == GST_STATE_CHANGE_FAILURE)
1158 /* all fine, do port check */
1159 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1160 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1162 /* this should not happen... */
1163 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1167 udpsink0 = udpsink_out[0];
1169 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1172 goto no_udp_protocol;
1174 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1175 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
1178 udpsink1 = udpsink_out[1];
1180 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1183 goto no_udp_protocol;
1185 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1186 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1187 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
1189 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1190 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
1191 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1192 /* Needs to be async for RECORD streams, otherwise we will never go to
1193 * PLAYING because the sinks will wait for data while the udpsrc can't
1194 * provide data with timestamps in PAUSED. */
1196 g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
1197 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1198 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1199 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1200 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1201 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1203 /* we keep these elements, we will further configure them when the
1204 * client told us to really use the UDP ports. */
1205 udpsrc_out[0] = udpsrc0;
1206 udpsrc_out[1] = udpsrc1;
1207 udpsink_out[0] = udpsink0;
1208 udpsink_out[1] = udpsink1;
1210 server_port_out->min = rtpport;
1211 server_port_out->max = rtcpport;
1213 *server_addr_out = addr;
1214 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1216 g_object_unref (rtp_socket);
1217 g_object_unref (rtcp_socket);
1245 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1246 gst_object_unref (udpsrc0);
1249 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1250 gst_object_unref (udpsrc1);
1253 gst_element_set_state (udpsink0, GST_STATE_NULL);
1254 gst_object_unref (udpsink0);
1257 g_object_unref (inetaddr);
1258 g_list_free_full (rejected_addresses,
1259 (GDestroyNotify) gst_rtsp_address_free);
1261 gst_rtsp_address_free (addr);
1263 g_object_unref (rtp_socket);
1265 g_object_unref (rtcp_socket);
1270 /* must be called with lock */
1272 alloc_ports (GstRTSPStream * stream)
1274 GstRTSPStreamPrivate *priv = stream->priv;
1277 alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
1278 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1279 &priv->server_port_v4, &priv->server_addr_v4);
1282 alloc_ports_one_family (stream, priv->pool, priv->buffer_size,
1283 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1284 &priv->server_port_v6, &priv->server_addr_v6);
1286 return priv->have_ipv4 || priv->have_ipv6;
1290 * gst_rtsp_stream_get_server_port:
1291 * @stream: a #GstRTSPStream
1292 * @server_port: (out): result server port
1293 * @family: the port family to get
1295 * Fill @server_port with the port pair used by the server. This function can
1296 * only be called when @stream has been joined.
1299 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1300 GstRTSPRange * server_port, GSocketFamily family)
1302 GstRTSPStreamPrivate *priv;
1304 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1305 priv = stream->priv;
1306 g_return_if_fail (priv->is_joined);
1308 g_mutex_lock (&priv->lock);
1309 if (family == G_SOCKET_FAMILY_IPV4) {
1311 *server_port = priv->server_port_v4;
1314 *server_port = priv->server_port_v6;
1316 g_mutex_unlock (&priv->lock);
1320 * gst_rtsp_stream_get_rtpsession:
1321 * @stream: a #GstRTSPStream
1323 * Get the RTP session of this stream.
1325 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1328 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1330 GstRTSPStreamPrivate *priv;
1333 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1335 priv = stream->priv;
1337 g_mutex_lock (&priv->lock);
1338 if ((session = priv->session))
1339 g_object_ref (session);
1340 g_mutex_unlock (&priv->lock);
1346 * gst_rtsp_stream_get_ssrc:
1347 * @stream: a #GstRTSPStream
1348 * @ssrc: (out): result ssrc
1350 * Get the SSRC used by the RTP session of this stream. This function can only
1351 * be called when @stream has been joined.
1354 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1356 GstRTSPStreamPrivate *priv;
1358 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1359 priv = stream->priv;
1360 g_return_if_fail (priv->is_joined);
1362 g_mutex_lock (&priv->lock);
1363 if (ssrc && priv->session)
1364 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1365 g_mutex_unlock (&priv->lock);
1369 * gst_rtsp_stream_set_retransmission_time:
1370 * @stream: a #GstRTSPStream
1371 * @time: a #GstClockTime
1373 * Set the amount of time to store retransmission packets.
1376 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1379 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1381 g_mutex_lock (&stream->priv->lock);
1382 stream->priv->rtx_time = time;
1383 if (stream->priv->rtxsend)
1384 g_object_set (stream->priv->rtxsend, "max-size-time",
1385 GST_TIME_AS_MSECONDS (time), NULL);
1386 g_mutex_unlock (&stream->priv->lock);
1390 * gst_rtsp_media_get_retransmission_time:
1391 * @media: a #GstRTSPMedia
1393 * Get the amount of time to store retransmission data.
1395 * Returns: the amount of time to store retransmission data.
1398 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1402 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1404 g_mutex_lock (&stream->priv->lock);
1405 ret = stream->priv->rtx_time;
1406 g_mutex_unlock (&stream->priv->lock);
1412 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1414 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1416 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1418 g_mutex_lock (&stream->priv->lock);
1419 stream->priv->rtx_pt = rtx_pt;
1420 if (stream->priv->rtxsend) {
1421 guint pt = gst_rtsp_stream_get_pt (stream);
1422 gchar *pt_s = g_strdup_printf ("%d", pt);
1423 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1424 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1425 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1427 gst_structure_free (rtx_pt_map);
1429 g_mutex_unlock (&stream->priv->lock);
1433 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1437 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1439 g_mutex_lock (&stream->priv->lock);
1440 rtx_pt = stream->priv->rtx_pt;
1441 g_mutex_unlock (&stream->priv->lock);
1446 /* executed from streaming thread */
1448 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1450 GstRTSPStreamPrivate *priv = stream->priv;
1451 GstCaps *newcaps, *oldcaps;
1453 newcaps = gst_pad_get_current_caps (pad);
1455 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1458 g_mutex_lock (&priv->lock);
1459 oldcaps = priv->caps;
1460 priv->caps = newcaps;
1461 g_mutex_unlock (&priv->lock);
1464 gst_caps_unref (oldcaps);
1468 dump_structure (const GstStructure * s)
1472 sstr = gst_structure_to_string (s);
1473 GST_INFO ("structure: %s", sstr);
1477 static GstRTSPStreamTransport *
1478 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1480 GstRTSPStreamPrivate *priv = stream->priv;
1482 GstRTSPStreamTransport *result = NULL;
1487 if (rtcp_from == NULL)
1490 tmp = g_strrstr (rtcp_from, ":");
1494 port = atoi (tmp + 1);
1495 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1497 g_mutex_lock (&priv->lock);
1498 GST_INFO ("finding %s:%d in %d transports", dest, port,
1499 g_list_length (priv->transports));
1501 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1502 GstRTSPStreamTransport *trans = walk->data;
1503 const GstRTSPTransport *tr;
1506 tr = gst_rtsp_stream_transport_get_transport (trans);
1508 min = tr->client_port.min;
1509 max = tr->client_port.max;
1511 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1517 g_object_ref (result);
1518 g_mutex_unlock (&priv->lock);
1525 static GstRTSPStreamTransport *
1526 check_transport (GObject * source, GstRTSPStream * stream)
1528 GstStructure *stats;
1529 GstRTSPStreamTransport *trans;
1531 /* see if we have a stream to match with the origin of the RTCP packet */
1532 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1533 if (trans == NULL) {
1534 g_object_get (source, "stats", &stats, NULL);
1536 const gchar *rtcp_from;
1538 dump_structure (stats);
1540 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1541 if ((trans = find_transport (stream, rtcp_from))) {
1542 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1544 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1547 gst_structure_free (stats);
1555 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1557 GstRTSPStreamTransport *trans;
1559 GST_INFO ("%p: new source %p", stream, source);
1561 trans = check_transport (source, stream);
1564 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1568 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1570 GST_INFO ("%p: new SDES %p", stream, source);
1574 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1576 GstRTSPStreamTransport *trans;
1578 trans = check_transport (source, stream);
1581 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1582 gst_rtsp_stream_transport_keep_alive (trans);
1586 GstStructure *stats;
1587 g_object_get (source, "stats", &stats, NULL);
1589 dump_structure (stats);
1590 gst_structure_free (stats);
1597 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1599 GST_INFO ("%p: source %p bye", stream, source);
1603 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1605 GstRTSPStreamTransport *trans;
1607 GST_INFO ("%p: source %p bye timeout", stream, source);
1609 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1610 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1611 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1616 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1618 GstRTSPStreamTransport *trans;
1620 GST_INFO ("%p: source %p timeout", stream, source);
1622 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1623 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1624 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1629 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1632 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1633 g_list_free (priv->tr_cache_rtp);
1634 priv->tr_cache_rtp = NULL;
1636 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1637 g_list_free (priv->tr_cache_rtcp);
1638 priv->tr_cache_rtcp = NULL;
1642 static GstFlowReturn
1643 handle_new_sample (GstAppSink * sink, gpointer user_data)
1645 GstRTSPStreamPrivate *priv;
1649 GstRTSPStream *stream;
1652 sample = gst_app_sink_pull_sample (sink);
1656 stream = (GstRTSPStream *) user_data;
1657 priv = stream->priv;
1658 buffer = gst_sample_get_buffer (sample);
1660 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1662 g_mutex_lock (&priv->lock);
1664 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1665 clear_tr_cache (priv, is_rtp);
1666 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1667 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1668 priv->tr_cache_rtp =
1669 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1671 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1674 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1675 clear_tr_cache (priv, is_rtp);
1676 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1677 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1678 priv->tr_cache_rtcp =
1679 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
1681 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
1684 g_mutex_unlock (&priv->lock);
1687 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
1688 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1689 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1692 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
1693 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1694 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1697 gst_sample_unref (sample);
1702 static GstAppSinkCallbacks sink_cb = {
1703 NULL, /* not interested in EOS */
1704 NULL, /* not interested in preroll samples */
1709 get_rtp_encoder (GstRTSPStream * stream, guint session)
1711 GstRTSPStreamPrivate *priv = stream->priv;
1713 if (priv->srtpenc == NULL) {
1716 name = g_strdup_printf ("srtpenc_%u", session);
1717 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1720 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1722 return gst_object_ref (priv->srtpenc);
1726 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1728 GstRTSPStreamPrivate *priv = stream->priv;
1729 GstElement *oldenc, *enc;
1733 if (priv->idx != session)
1736 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1738 oldenc = priv->srtpenc;
1739 enc = get_rtp_encoder (stream, session);
1740 name = g_strdup_printf ("rtp_sink_%d", session);
1741 pad = gst_element_get_request_pad (enc, name);
1743 gst_object_unref (pad);
1746 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
1753 request_rtcp_encoder (GstElement * rtpbin, guint session,
1754 GstRTSPStream * stream)
1756 GstRTSPStreamPrivate *priv = stream->priv;
1757 GstElement *oldenc, *enc;
1761 if (priv->idx != session)
1764 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1766 oldenc = priv->srtpenc;
1767 enc = get_rtp_encoder (stream, session);
1768 name = g_strdup_printf ("rtcp_sink_%d", session);
1769 pad = gst_element_get_request_pad (enc, name);
1771 gst_object_unref (pad);
1774 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
1781 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1783 GstRTSPStreamPrivate *priv = stream->priv;
1786 GST_DEBUG ("request key %08x", ssrc);
1788 g_mutex_lock (&priv->lock);
1789 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1790 gst_caps_ref (caps);
1791 g_mutex_unlock (&priv->lock);
1797 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
1798 GstRTSPStream * stream)
1800 GstRTSPStreamPrivate *priv = stream->priv;
1802 if (priv->idx != session)
1805 if (priv->srtpdec == NULL) {
1808 name = g_strdup_printf ("srtpdec_%u", session);
1809 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1812 g_signal_connect (priv->srtpdec, "request-key",
1813 (GCallback) request_key, stream);
1815 return gst_object_ref (priv->srtpdec);
1819 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPStream * stream)
1823 GstStructure *pt_map;
1828 pt = gst_rtsp_stream_get_pt (stream);
1829 pt_s = g_strdup_printf ("%u", pt);
1830 rtx_pt = stream->priv->rtx_pt;
1832 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
1834 bin = gst_bin_new (NULL);
1835 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
1836 pt_map = gst_structure_new ("application/x-rtp-pt-map",
1837 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1838 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
1839 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
1841 gst_structure_free (pt_map);
1842 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
1844 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
1845 name = g_strdup_printf ("src_%u", sessid);
1846 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
1848 gst_object_unref (pad);
1850 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
1851 name = g_strdup_printf ("sink_%u", sessid);
1852 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
1854 gst_object_unref (pad);
1860 * gst_rtsp_stream_set_pt_map:
1861 * @stream: a #GstRTSPStream
1865 * Configure a pt map between @pt and @caps.
1868 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
1870 GstRTSPStreamPrivate *priv = stream->priv;
1872 g_mutex_lock (&priv->lock);
1873 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
1874 g_mutex_unlock (&priv->lock);
1878 request_pt_map (GstElement * rtpbin, guint session, guint pt,
1879 GstRTSPStream * stream)
1881 GstRTSPStreamPrivate *priv = stream->priv;
1882 GstCaps *caps = NULL;
1884 g_mutex_lock (&priv->lock);
1886 if (priv->idx == session) {
1887 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
1889 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
1890 gst_caps_ref (caps);
1892 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
1896 g_mutex_unlock (&priv->lock);
1902 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
1904 GstRTSPStreamPrivate *priv = stream->priv;
1906 GstPadLinkReturn ret;
1909 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
1910 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
1912 name = gst_pad_get_name (pad);
1913 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
1919 if (priv->idx != sessid)
1922 if (gst_pad_is_linked (priv->sinkpad)) {
1923 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
1924 GST_DEBUG_PAD_NAME (priv->sinkpad));
1928 /* link the RTP pad to the session manager, it should not really fail unless
1929 * this is not really an RTP pad */
1930 ret = gst_pad_link (pad, priv->sinkpad);
1931 if (ret != GST_PAD_LINK_OK)
1933 priv->recv_rtp_src = gst_object_ref (pad);
1940 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
1941 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
1946 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
1947 GstRTSPStream * stream)
1949 /* TODO: What to do here other than this? */
1950 GST_DEBUG ("Stream %p: Got EOS", stream);
1951 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
1955 * gst_rtsp_stream_join_bin:
1956 * @stream: a #GstRTSPStream
1957 * @bin: (transfer none): a #GstBin to join
1958 * @rtpbin: (transfer none): a rtpbin element in @bin
1959 * @state: the target state of the new elements
1961 * Join the #GstBin @bin that contains the element @rtpbin.
1963 * @stream will link to @rtpbin, which must be inside @bin. The elements
1964 * added to @bin will be set to the state given in @state.
1966 * Returns: %TRUE on success.
1969 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1970 GstElement * rtpbin, GstState state)
1972 GstRTSPStreamPrivate *priv;
1976 GstPad *pad, *sinkpad, *selpad;
1977 GstPadLinkReturn ret;
1979 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1980 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1981 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1983 priv = stream->priv;
1985 g_mutex_lock (&priv->lock);
1986 if (priv->is_joined)
1989 /* create a session with the same index as the stream */
1992 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1994 if (!alloc_ports (stream))
1997 /* update the dscp qos field in the sinks */
1998 update_dscp_qos (stream);
2000 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2001 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2003 g_signal_connect (rtpbin, "request-rtp-encoder",
2004 (GCallback) request_rtp_encoder, stream);
2005 g_signal_connect (rtpbin, "request-rtcp-encoder",
2006 (GCallback) request_rtcp_encoder, stream);
2007 g_signal_connect (rtpbin, "request-rtp-decoder",
2008 (GCallback) request_rtp_rtcp_decoder, stream);
2009 g_signal_connect (rtpbin, "request-rtcp-decoder",
2010 (GCallback) request_rtp_rtcp_decoder, stream);
2013 if (priv->rtx_time > 0 && priv->srcpad) {
2014 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
2015 g_signal_connect (rtpbin, "request-aux-sender",
2016 (GCallback) request_aux_sender, stream);
2018 if (priv->sinkpad) {
2019 g_signal_connect (rtpbin, "request-pt-map",
2020 (GCallback) request_pt_map, stream);
2023 /* get a pad for sending RTP */
2024 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2025 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2029 /* link the RTP pad to the session manager, it should not really fail unless
2030 * this is not really an RTP pad */
2031 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2032 if (ret != GST_PAD_LINK_OK)
2035 /* Need to connect our sinkpad from here */
2036 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2038 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2041 /* get pads from the RTP session element for sending and receiving
2043 name = g_strdup_printf ("send_rtp_src_%u", idx);
2044 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2046 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2047 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2050 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2051 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2053 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2054 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2057 /* get the session */
2058 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2060 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2062 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2064 g_signal_connect (priv->session, "on-ssrc-active",
2065 (GCallback) on_ssrc_active, stream);
2066 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2068 g_signal_connect (priv->session, "on-bye-timeout",
2069 (GCallback) on_bye_timeout, stream);
2070 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2073 for (i = 0; i < 2; i++) {
2074 GstPad *teepad, *queuepad;
2075 /* For the sender we create this bit of pipeline for both
2076 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2077 * we need to add a queue before appsink to make the pipeline
2078 * not block. For the TCP case, we want to pump data to the
2079 * client as fast as possible anyway.
2081 * .--------. .-----. .---------.
2082 * | rtpbin | | tee | | udpsink |
2083 * | send->sink src->sink |
2084 * '--------' | | '---------'
2085 * | | .---------. .---------.
2086 * | | | queue | | appsink |
2087 * | src->sink src->sink |
2088 * '-----' '---------' '---------'
2090 * When only UDP is allowed, we skip the tee, queue and appsink and link the
2091 * udpsink directly to the session.
2094 gst_bin_add (bin, priv->udpsink[i]);
2095 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2097 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
2098 /* make tee for RTP/RTCP */
2099 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2100 gst_bin_add (bin, priv->tee[i]);
2102 /* and link to rtpbin send pad */
2103 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2104 gst_pad_link (priv->send_src[i], pad);
2105 gst_object_unref (pad);
2107 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2108 gst_bin_add (bin, priv->udpqueue[i]);
2109 /* link tee to udpqueue */
2110 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2111 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2112 gst_pad_link (teepad, pad);
2113 gst_object_unref (pad);
2114 gst_object_unref (teepad);
2116 /* link udpqueue to udpsink */
2117 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2118 gst_pad_link (queuepad, sinkpad);
2119 gst_object_unref (queuepad);
2122 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2123 gst_bin_add (bin, priv->appqueue[i]);
2124 /* and link to tee */
2125 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2126 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2127 gst_pad_link (teepad, pad);
2128 gst_object_unref (pad);
2129 gst_object_unref (teepad);
2132 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2133 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2134 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2135 gst_bin_add (bin, priv->appsink[i]);
2136 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2137 &sink_cb, stream, NULL);
2138 /* and link to queue */
2139 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2140 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2141 gst_pad_link (queuepad, pad);
2142 gst_object_unref (pad);
2143 gst_object_unref (queuepad);
2145 /* else only udpsink needed, link it to the session */
2146 gst_pad_link (priv->send_src[i], sinkpad);
2148 gst_object_unref (sinkpad);
2150 /* For the receiver we create this bit of pipeline for both
2151 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2152 * and it is all funneled into the rtpbin receive pad.
2154 * .--------. .--------. .--------.
2155 * | udpsrc | | funnel | | rtpbin |
2156 * | src->sink src->sink |
2157 * '--------' | | '--------'
2161 * '--------' '--------'
2163 /* make funnel for the RTP/RTCP receivers */
2164 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2165 gst_bin_add (bin, priv->funnel[i]);
2167 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2168 gst_pad_link (pad, priv->recv_sink[i]);
2169 gst_object_unref (pad);
2171 if (priv->udpsrc_v4[i]) {
2173 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2174 * values. This is only relevant for PLAY pipelines */
2175 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2176 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2179 gst_bin_add (bin, priv->udpsrc_v4[i]);
2181 /* and link to the funnel v4 */
2182 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2183 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2184 gst_pad_link (pad, selpad);
2185 gst_object_unref (pad);
2186 gst_object_unref (selpad);
2189 if (priv->udpsrc_v6[i]) {
2191 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2192 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2194 gst_bin_add (bin, priv->udpsrc_v6[i]);
2196 /* and link to the funnel v6 */
2197 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2198 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2199 gst_pad_link (pad, selpad);
2200 gst_object_unref (pad);
2201 gst_object_unref (selpad);
2204 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
2205 /* make and add appsrc */
2206 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2207 priv->appsrc_base_time[i] = -1;
2208 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2209 gst_bin_add (bin, priv->appsrc[i]);
2210 /* and link to the funnel */
2211 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2212 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2213 gst_pad_link (pad, selpad);
2214 gst_object_unref (pad);
2215 gst_object_unref (selpad);
2218 /* check if we need to set to a special state */
2219 if (state != GST_STATE_NULL) {
2220 if (priv->udpsink[i])
2221 gst_element_set_state (priv->udpsink[i], state);
2222 if (priv->appsink[i])
2223 gst_element_set_state (priv->appsink[i], state);
2224 if (priv->appqueue[i])
2225 gst_element_set_state (priv->appqueue[i], state);
2226 if (priv->udpqueue[i])
2227 gst_element_set_state (priv->udpqueue[i], state);
2229 gst_element_set_state (priv->tee[i], state);
2230 if (priv->funnel[i])
2231 gst_element_set_state (priv->funnel[i], state);
2232 if (priv->appsrc[i])
2233 gst_element_set_state (priv->appsrc[i], state);
2237 /* be notified of caps changes */
2238 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2239 (GCallback) caps_notify, stream);
2241 priv->is_joined = TRUE;
2242 g_mutex_unlock (&priv->lock);
2249 g_mutex_unlock (&priv->lock);
2254 g_mutex_unlock (&priv->lock);
2255 GST_WARNING ("failed to allocate ports %u", idx);
2260 GST_WARNING ("failed to link stream %u", idx);
2261 gst_object_unref (priv->send_rtp_sink);
2262 priv->send_rtp_sink = NULL;
2263 g_mutex_unlock (&priv->lock);
2269 * gst_rtsp_stream_leave_bin:
2270 * @stream: a #GstRTSPStream
2271 * @bin: (transfer none): a #GstBin
2272 * @rtpbin: (transfer none): a rtpbin #GstElement
2274 * Remove the elements of @stream from @bin.
2276 * Return: %TRUE on success.
2279 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2280 GstElement * rtpbin)
2282 GstRTSPStreamPrivate *priv;
2286 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2287 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2288 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2290 priv = stream->priv;
2292 g_mutex_lock (&priv->lock);
2293 if (!priv->is_joined)
2294 goto was_not_joined;
2296 /* all transports must be removed by now */
2297 if (priv->transports != NULL)
2298 goto transports_not_removed;
2300 clear_tr_cache (priv, TRUE);
2301 clear_tr_cache (priv, FALSE);
2303 GST_INFO ("stream %p leaving bin", stream);
2306 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2307 } else if (priv->recv_rtp_src) {
2308 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2309 gst_object_unref (priv->recv_rtp_src);
2310 priv->recv_rtp_src = NULL;
2312 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2313 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2314 gst_object_unref (priv->send_rtp_sink);
2315 priv->send_rtp_sink = NULL;
2317 for (i = 0; i < 2; i++) {
2318 if (priv->udpsink[i])
2319 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2320 if (priv->appsink[i])
2321 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2322 if (priv->appqueue[i])
2323 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2324 if (priv->udpqueue[i])
2325 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2327 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2328 if (priv->funnel[i])
2329 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2330 if (priv->appsrc[i])
2331 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2332 if (priv->udpsrc_v4[i]) {
2333 /* and set udpsrc to NULL now before removing */
2334 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2335 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2336 /* removing them should also nicely release the request
2337 * pads when they finalize */
2338 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2340 if (priv->udpsrc_v6[i]) {
2341 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2342 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2343 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2346 for (l = priv->transport_sources; l; l = l->next) {
2347 GstRTSPMulticastTransportSource *s = l->data;
2352 gst_element_set_locked_state (s->udpsrc[i], FALSE);
2353 gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
2354 gst_bin_remove (bin, s->udpsrc[i]);
2357 if (priv->udpsink[i])
2358 gst_bin_remove (bin, priv->udpsink[i]);
2359 if (priv->appsrc[i])
2360 gst_bin_remove (bin, priv->appsrc[i]);
2361 if (priv->appsink[i])
2362 gst_bin_remove (bin, priv->appsink[i]);
2363 if (priv->appqueue[i])
2364 gst_bin_remove (bin, priv->appqueue[i]);
2365 if (priv->udpqueue[i])
2366 gst_bin_remove (bin, priv->udpqueue[i]);
2368 gst_bin_remove (bin, priv->tee[i]);
2369 if (priv->funnel[i])
2370 gst_bin_remove (bin, priv->funnel[i]);
2372 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2373 gst_object_unref (priv->recv_sink[i]);
2374 priv->recv_sink[i] = NULL;
2376 priv->udpsrc_v4[i] = NULL;
2377 priv->udpsrc_v6[i] = NULL;
2378 priv->udpsink[i] = NULL;
2379 priv->appsrc[i] = NULL;
2380 priv->appsink[i] = NULL;
2381 priv->appqueue[i] = NULL;
2382 priv->udpqueue[i] = NULL;
2383 priv->tee[i] = NULL;
2384 priv->funnel[i] = NULL;
2387 for (l = priv->transport_sources; l; l = l->next) {
2388 GstRTSPMulticastTransportSource *s = l->data;
2389 g_slice_free (GstRTSPMulticastTransportSource, s);
2391 g_list_free (priv->transport_sources);
2392 priv->transport_sources = NULL;
2394 gst_object_unref (priv->send_src[0]);
2395 priv->send_src[0] = NULL;
2397 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2398 gst_object_unref (priv->send_src[1]);
2399 priv->send_src[1] = NULL;
2401 g_object_unref (priv->session);
2402 priv->session = NULL;
2404 gst_caps_unref (priv->caps);
2408 gst_object_unref (priv->srtpenc);
2410 gst_object_unref (priv->srtpdec);
2412 priv->is_joined = FALSE;
2413 g_mutex_unlock (&priv->lock);
2419 g_mutex_unlock (&priv->lock);
2422 transports_not_removed:
2424 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2425 g_mutex_unlock (&priv->lock);
2431 * gst_rtsp_stream_get_rtpinfo:
2432 * @stream: a #GstRTSPStream
2433 * @rtptime: (allow-none): result RTP timestamp
2434 * @seq: (allow-none): result RTP seqnum
2435 * @clock_rate: (allow-none): the clock rate
2436 * @running_time: (allow-none): result running-time
2438 * Retrieve the current rtptime, seq and running-time. This is used to
2439 * construct a RTPInfo reply header.
2441 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2444 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2445 guint * rtptime, guint * seq, guint * clock_rate,
2446 GstClockTime * running_time)
2448 GstRTSPStreamPrivate *priv;
2449 GstStructure *stats;
2450 GObjectClass *payobjclass;
2452 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2454 priv = stream->priv;
2456 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2458 g_mutex_lock (&priv->lock);
2460 if (g_object_class_find_property (payobjclass, "stats")) {
2461 g_object_get (priv->payloader, "stats", &stats, NULL);
2466 gst_structure_get_uint (stats, "seqnum", seq);
2469 gst_structure_get_uint (stats, "timestamp", rtptime);
2472 gst_structure_get_clock_time (stats, "running-time", running_time);
2475 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2476 if (*clock_rate == 0 && running_time)
2477 *running_time = GST_CLOCK_TIME_NONE;
2479 gst_structure_free (stats);
2481 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2482 !g_object_class_find_property (payobjclass, "timestamp"))
2486 g_object_get (priv->payloader, "seqnum", seq, NULL);
2489 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2492 *running_time = GST_CLOCK_TIME_NONE;
2494 g_mutex_unlock (&priv->lock);
2501 GST_WARNING ("Could not get payloader stats");
2502 g_mutex_unlock (&priv->lock);
2508 * gst_rtsp_stream_get_caps:
2509 * @stream: a #GstRTSPStream
2511 * Retrieve the current caps of @stream.
2513 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2517 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2519 GstRTSPStreamPrivate *priv;
2522 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2524 priv = stream->priv;
2526 g_mutex_lock (&priv->lock);
2527 if ((result = priv->caps))
2528 gst_caps_ref (result);
2529 g_mutex_unlock (&priv->lock);
2535 * gst_rtsp_stream_recv_rtp:
2536 * @stream: a #GstRTSPStream
2537 * @buffer: (transfer full): a #GstBuffer
2539 * Handle an RTP buffer for the stream. This method is usually called when a
2540 * message has been received from a client using the TCP transport.
2542 * This function takes ownership of @buffer.
2544 * Returns: a GstFlowReturn.
2547 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2549 GstRTSPStreamPrivate *priv;
2551 GstElement *element;
2553 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2554 priv = stream->priv;
2555 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2556 g_return_val_if_fail (priv->is_joined, FALSE);
2558 g_mutex_lock (&priv->lock);
2559 if (priv->appsrc[0])
2560 element = gst_object_ref (priv->appsrc[0]);
2563 g_mutex_unlock (&priv->lock);
2566 if (priv->appsrc_base_time[0] == -1) {
2567 /* Take current running_time. This timestamp will be put on
2568 * the first buffer of each stream because we are a live source and so we
2569 * timestamp with the running_time. When we are dealing with TCP, we also
2570 * only timestamp the first buffer (using the DISCONT flag) because a server
2571 * typically bursts data, for which we don't want to compensate by speeding
2572 * up the media. The other timestamps will be interpollated from this one
2573 * using the RTP timestamps. */
2574 GST_OBJECT_LOCK (element);
2575 if (GST_ELEMENT_CLOCK (element)) {
2577 GstClockTime base_time;
2579 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
2580 base_time = GST_ELEMENT_CAST (element)->base_time;
2582 priv->appsrc_base_time[0] = now - base_time;
2583 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
2584 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
2585 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
2586 GST_TIME_ARGS (base_time));
2588 GST_OBJECT_UNLOCK (element);
2591 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2592 gst_object_unref (element);
2600 * gst_rtsp_stream_recv_rtcp:
2601 * @stream: a #GstRTSPStream
2602 * @buffer: (transfer full): a #GstBuffer
2604 * Handle an RTCP buffer for the stream. This method is usually called when a
2605 * message has been received from a client using the TCP transport.
2607 * This function takes ownership of @buffer.
2609 * Returns: a GstFlowReturn.
2612 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2614 GstRTSPStreamPrivate *priv;
2616 GstElement *element;
2618 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2619 priv = stream->priv;
2620 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2622 if (!priv->is_joined) {
2623 gst_buffer_unref (buffer);
2624 return GST_FLOW_NOT_LINKED;
2626 g_mutex_lock (&priv->lock);
2627 if (priv->appsrc[1])
2628 element = gst_object_ref (priv->appsrc[1]);
2631 g_mutex_unlock (&priv->lock);
2634 if (priv->appsrc_base_time[1] == -1) {
2635 /* Take current running_time. This timestamp will be put on
2636 * the first buffer of each stream because we are a live source and so we
2637 * timestamp with the running_time. When we are dealing with TCP, we also
2638 * only timestamp the first buffer (using the DISCONT flag) because a server
2639 * typically bursts data, for which we don't want to compensate by speeding
2640 * up the media. The other timestamps will be interpollated from this one
2641 * using the RTP timestamps. */
2642 GST_OBJECT_LOCK (element);
2643 if (GST_ELEMENT_CLOCK (element)) {
2645 GstClockTime base_time;
2647 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
2648 base_time = GST_ELEMENT_CAST (element)->base_time;
2650 priv->appsrc_base_time[1] = now - base_time;
2651 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
2652 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
2653 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
2654 GST_TIME_ARGS (base_time));
2656 GST_OBJECT_UNLOCK (element);
2659 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2660 gst_object_unref (element);
2663 gst_buffer_unref (buffer);
2668 /* must be called with lock */
2670 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
2673 GstRTSPStreamPrivate *priv = stream->priv;
2674 const GstRTSPTransport *tr;
2676 tr = gst_rtsp_stream_transport_get_transport (trans);
2678 switch (tr->lower_transport) {
2679 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2681 GstRTSPMulticastTransportSource *source;
2684 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[0])));
2689 GstPad *selpad, *pad;
2691 source = g_slice_new0 (GstRTSPMulticastTransportSource);
2692 source->transport = trans;
2694 for (i = 0; i < 2; i++) {
2696 g_strdup_printf ("udp://%s:%d", tr->destination,
2697 (i == 0) ? tr->port.min : tr->port.max);
2699 gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2703 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2704 * values. This is only relevant for PLAY pipelines */
2705 gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
2706 gst_element_set_locked_state (source->udpsrc[i], TRUE);
2709 gst_bin_add (bin, source->udpsrc[i]);
2711 /* and link to the funnel v4 */
2712 source->selpad[i] = selpad =
2713 gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2714 pad = gst_element_get_static_pad (source->udpsrc[i], "src");
2715 gst_pad_link (pad, selpad);
2716 gst_object_unref (pad);
2717 gst_object_unref (selpad);
2719 gst_object_unref (bin);
2721 priv->transport_sources =
2722 g_list_prepend (priv->transport_sources, source);
2726 for (l = priv->transport_sources; l; l = l->next) {
2729 if (source->transport == trans) {
2730 priv->transport_sources =
2731 g_list_delete_link (priv->transport_sources, l);
2739 for (i = 0; i < 2; i++) {
2740 /* Will automatically unlink everything */
2741 gst_bin_remove (bin,
2742 GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
2744 gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
2745 gst_object_unref (source->udpsrc[i]);
2747 gst_element_release_request_pad (priv->funnel[i],
2751 g_slice_free (GstRTSPMulticastTransportSource, source);
2755 /* fall through for the generic case */
2757 case GST_RTSP_LOWER_TRANS_UDP:
2763 dest = tr->destination;
2764 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2769 min = tr->client_port.min;
2770 max = tr->client_port.max;
2775 GST_INFO ("setting ttl-mc %d", ttl);
2776 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
2777 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
2779 GST_INFO ("adding %s:%d-%d", dest, min, max);
2780 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
2781 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
2782 priv->transports = g_list_prepend (priv->transports, trans);
2784 GST_INFO ("removing %s:%d-%d", dest, min, max);
2785 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
2786 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
2787 priv->transports = g_list_remove (priv->transports, trans);
2789 priv->transports_cookie++;
2792 case GST_RTSP_LOWER_TRANS_TCP:
2794 GST_INFO ("adding TCP %s", tr->destination);
2795 priv->transports = g_list_prepend (priv->transports, trans);
2797 GST_INFO ("removing TCP %s", tr->destination);
2798 priv->transports = g_list_remove (priv->transports, trans);
2800 priv->transports_cookie++;
2803 goto unknown_transport;
2810 GST_INFO ("Unknown transport %d", tr->lower_transport);
2817 * gst_rtsp_stream_add_transport:
2818 * @stream: a #GstRTSPStream
2819 * @trans: (transfer none): a #GstRTSPStreamTransport
2821 * Add the transport in @trans to @stream. The media of @stream will
2822 * then also be send to the values configured in @trans.
2824 * @stream must be joined to a bin.
2826 * @trans must contain a valid #GstRTSPTransport.
2828 * Returns: %TRUE if @trans was added
2831 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
2832 GstRTSPStreamTransport * trans)
2834 GstRTSPStreamPrivate *priv;
2837 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2838 priv = stream->priv;
2839 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2840 g_return_val_if_fail (priv->is_joined, FALSE);
2842 g_mutex_lock (&priv->lock);
2843 res = update_transport (stream, trans, TRUE);
2844 g_mutex_unlock (&priv->lock);
2850 * gst_rtsp_stream_remove_transport:
2851 * @stream: a #GstRTSPStream
2852 * @trans: (transfer none): a #GstRTSPStreamTransport
2854 * Remove the transport in @trans from @stream. The media of @stream will
2855 * not be sent to the values configured in @trans.
2857 * @stream must be joined to a bin.
2859 * @trans must contain a valid #GstRTSPTransport.
2861 * Returns: %TRUE if @trans was removed
2864 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
2865 GstRTSPStreamTransport * trans)
2867 GstRTSPStreamPrivate *priv;
2870 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2871 priv = stream->priv;
2872 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
2873 g_return_val_if_fail (priv->is_joined, FALSE);
2875 g_mutex_lock (&priv->lock);
2876 res = update_transport (stream, trans, FALSE);
2877 g_mutex_unlock (&priv->lock);
2883 * gst_rtsp_stream_update_crypto:
2884 * @stream: a #GstRTSPStream
2886 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
2888 * Update the new crypto information for @ssrc in @stream. If information
2889 * for @ssrc did not exist, it will be added. If information
2890 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
2891 * be removed from @stream.
2893 * Returns: %TRUE if @crypto could be updated
2896 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
2897 guint ssrc, GstCaps * crypto)
2899 GstRTSPStreamPrivate *priv;
2901 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2902 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
2904 priv = stream->priv;
2906 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
2908 g_mutex_lock (&priv->lock);
2910 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
2911 gst_caps_ref (crypto));
2913 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
2914 g_mutex_unlock (&priv->lock);
2920 * gst_rtsp_stream_get_rtp_socket:
2921 * @stream: a #GstRTSPStream
2922 * @family: the socket family
2924 * Get the RTP socket from @stream for a @family.
2926 * @stream must be joined to a bin.
2928 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
2929 * socket could be allocated for @family. Unref after usage
2932 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
2934 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2938 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2939 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2940 family == G_SOCKET_FAMILY_IPV6, NULL);
2941 g_return_val_if_fail (priv->udpsink[0], NULL);
2943 if (family == G_SOCKET_FAMILY_IPV6)
2948 g_object_get (priv->udpsink[0], name, &socket, NULL);
2954 * gst_rtsp_stream_get_rtcp_socket:
2955 * @stream: a #GstRTSPStream
2956 * @family: the socket family
2958 * Get the RTCP socket from @stream for a @family.
2960 * @stream must be joined to a bin.
2962 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
2963 * socket could be allocated for @family. Unref after usage
2966 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
2968 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
2972 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2973 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
2974 family == G_SOCKET_FAMILY_IPV6, NULL);
2975 g_return_val_if_fail (priv->udpsink[1], NULL);
2977 if (family == G_SOCKET_FAMILY_IPV6)
2982 g_object_get (priv->udpsink[1], name, &socket, NULL);
2988 * gst_rtsp_stream_set_seqnum:
2989 * @stream: a #GstRTSPStream
2990 * @seqnum: a new sequence number
2992 * Configure the sequence number in the payloader of @stream to @seqnum.
2995 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
2997 GstRTSPStreamPrivate *priv;
2999 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3001 priv = stream->priv;
3003 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3007 * gst_rtsp_stream_get_seqnum:
3008 * @stream: a #GstRTSPStream
3010 * Get the configured sequence number in the payloader of @stream.
3012 * Returns: the sequence number of the payloader.
3015 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3017 GstRTSPStreamPrivate *priv;
3020 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3022 priv = stream->priv;
3024 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3030 * gst_rtsp_stream_transport_filter:
3031 * @stream: a #GstRTSPStream
3032 * @func: (scope call) (allow-none): a callback
3033 * @user_data: (closure): user data passed to @func
3035 * Call @func for each transport managed by @stream. The result value of @func
3036 * determines what happens to the transport. @func will be called with @stream
3037 * locked so no further actions on @stream can be performed from @func.
3039 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3042 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3044 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3045 * will also be added with an additional ref to the result #GList of this
3048 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3050 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3051 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3052 * element in the #GList should be unreffed before the list is freed.
3055 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3056 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3058 GstRTSPStreamPrivate *priv;
3059 GList *result, *walk, *next;
3060 GHashTable *visited = NULL;
3063 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3065 priv = stream->priv;
3069 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3071 g_mutex_lock (&priv->lock);
3073 cookie = priv->transports_cookie;
3074 for (walk = priv->transports; walk; walk = next) {
3075 GstRTSPStreamTransport *trans = walk->data;
3076 GstRTSPFilterResult res;
3079 next = g_list_next (walk);
3082 /* only visit each transport once */
3083 if (g_hash_table_contains (visited, trans))
3086 g_hash_table_add (visited, g_object_ref (trans));
3087 g_mutex_unlock (&priv->lock);
3089 res = func (stream, trans, user_data);
3091 g_mutex_lock (&priv->lock);
3093 res = GST_RTSP_FILTER_REF;
3095 changed = (cookie != priv->transports_cookie);
3098 case GST_RTSP_FILTER_REMOVE:
3099 update_transport (stream, trans, FALSE);
3101 case GST_RTSP_FILTER_REF:
3102 result = g_list_prepend (result, g_object_ref (trans));
3104 case GST_RTSP_FILTER_KEEP:
3111 g_mutex_unlock (&priv->lock);
3114 g_hash_table_unref (visited);
3119 static GstPadProbeReturn
3120 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3122 GstRTSPStreamPrivate *priv;
3123 GstRTSPStream *stream;
3126 priv = stream->priv;
3128 GST_DEBUG_OBJECT (pad, "now blocking");
3130 g_mutex_lock (&priv->lock);
3131 priv->blocking = TRUE;
3132 g_mutex_unlock (&priv->lock);
3134 gst_element_post_message (priv->payloader,
3135 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3136 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3138 return GST_PAD_PROBE_OK;
3142 * gst_rtsp_stream_set_blocked:
3143 * @stream: a #GstRTSPStream
3144 * @blocked: boolean indicating we should block or unblock
3146 * Blocks or unblocks the dataflow on @stream.
3148 * Returns: %TRUE on success
3151 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3153 GstRTSPStreamPrivate *priv;
3155 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3157 priv = stream->priv;
3159 g_mutex_lock (&priv->lock);
3161 priv->blocking = FALSE;
3162 if (priv->blocked_id == 0) {
3163 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3164 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3165 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3166 g_object_ref (stream), g_object_unref);
3169 if (priv->blocked_id != 0) {
3170 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3171 priv->blocked_id = 0;
3172 priv->blocking = FALSE;
3175 g_mutex_unlock (&priv->lock);
3181 * gst_rtsp_stream_is_blocking:
3182 * @stream: a #GstRTSPStream
3184 * Check if @stream is blocking on a #GstBuffer.
3186 * Returns: %TRUE if @stream is blocking
3189 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3191 GstRTSPStreamPrivate *priv;
3194 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3196 priv = stream->priv;
3198 g_mutex_lock (&priv->lock);
3199 result = priv->blocking;
3200 g_mutex_unlock (&priv->lock);
3206 * gst_rtsp_stream_query_position:
3207 * @stream: a #GstRTSPStream
3209 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3210 * the RTP parts of the pipeline and not the RTCP parts.
3212 * Returns: %TRUE if the position could be queried
3215 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3217 GstRTSPStreamPrivate *priv;
3221 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3223 priv = stream->priv;
3225 g_mutex_lock (&priv->lock);
3226 if ((sink = priv->udpsink[0]))
3227 gst_object_ref (sink);
3228 g_mutex_unlock (&priv->lock);
3233 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3234 gst_object_unref (sink);
3240 * gst_rtsp_stream_query_stop:
3241 * @stream: a #GstRTSPStream
3243 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3244 * the RTP parts of the pipeline and not the RTCP parts.
3246 * Returns: %TRUE if the stop could be queried
3249 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3251 GstRTSPStreamPrivate *priv;
3256 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3258 priv = stream->priv;
3260 g_mutex_lock (&priv->lock);
3261 if ((sink = priv->udpsink[0]))
3262 gst_object_ref (sink);
3263 g_mutex_unlock (&priv->lock);
3268 query = gst_query_new_segment (GST_FORMAT_TIME);
3269 if ((ret = gst_element_query (sink, query))) {
3272 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3273 if (format != GST_FORMAT_TIME)
3276 gst_query_unref (query);
3277 gst_object_unref (sink);