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;
84 /* TRUE if this stream is running on
85 * the client side of an RTSP link (for RECORD) */
89 GstRTSPProfile profiles;
90 GstRTSPLowerTrans protocols;
92 /* pads on the rtpbin */
93 GstPad *send_rtp_sink;
98 /* the RTPSession object */
101 /* SRTP encoder/decoder */
106 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
108 GstElement *udpsrc_v4[2];
110 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
112 GstElement *udpsrc_v6[2];
114 GstElement *udpqueue[2];
115 GstElement *udpsink[2];
117 /* for TCP transport */
118 GstElement *appsrc[2];
119 GstClockTime appsrc_base_time[2];
120 GstElement *appqueue[2];
121 GstElement *appsink[2];
124 GstElement *funnel[2];
129 GstClockTime rtx_time;
131 /* server ports for sending/receiving over ipv4 */
132 GstRTSPRange server_port_v4;
133 GstRTSPAddress *server_addr_v4;
136 /* server ports for sending/receiving over ipv6 */
137 GstRTSPRange server_port_v6;
138 GstRTSPAddress *server_addr_v6;
141 /* multicast addresses */
142 GstRTSPAddressPool *pool;
143 GstRTSPAddress *addr_v4;
144 GstRTSPAddress *addr_v6;
146 /* the caps of the stream */
150 /* transports we stream to */
153 guint transports_cookie;
155 GList *tr_cache_rtcp;
156 guint tr_cache_cookie_rtp;
157 guint tr_cache_cookie_rtcp;
160 /* UDP sources for UDP multicast transports */
161 GList *transport_sources;
165 /* stream blocking */
169 /* pt->caps map for RECORD streams */
173 #define DEFAULT_CONTROL NULL
174 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
175 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
176 GST_RTSP_LOWER_TRANS_TCP
189 SIGNAL_NEW_RTP_ENCODER,
190 SIGNAL_NEW_RTCP_ENCODER,
194 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
195 #define GST_CAT_DEFAULT rtsp_stream_debug
197 static GQuark ssrc_stream_map_key;
199 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
200 GValue * value, GParamSpec * pspec);
201 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
202 const GValue * value, GParamSpec * pspec);
204 static void gst_rtsp_stream_finalize (GObject * obj);
206 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
208 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
211 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
213 GObjectClass *gobject_class;
215 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
217 gobject_class = G_OBJECT_CLASS (klass);
219 gobject_class->get_property = gst_rtsp_stream_get_property;
220 gobject_class->set_property = gst_rtsp_stream_set_property;
221 gobject_class->finalize = gst_rtsp_stream_finalize;
223 g_object_class_install_property (gobject_class, PROP_CONTROL,
224 g_param_spec_string ("control", "Control",
225 "The control string for this stream", DEFAULT_CONTROL,
226 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
228 g_object_class_install_property (gobject_class, PROP_PROFILES,
229 g_param_spec_flags ("profiles", "Profiles",
230 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
231 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
233 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
234 g_param_spec_flags ("protocols", "Protocols",
235 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
236 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
238 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
239 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
240 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
241 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
243 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
244 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
245 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
246 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
248 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
250 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
254 gst_rtsp_stream_init (GstRTSPStream * stream)
256 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
258 GST_DEBUG ("new stream %p", stream);
263 priv->control = g_strdup (DEFAULT_CONTROL);
264 priv->profiles = DEFAULT_PROFILES;
265 priv->protocols = DEFAULT_PROTOCOLS;
267 g_mutex_init (&priv->lock);
269 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
270 NULL, (GDestroyNotify) gst_caps_unref);
271 priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
272 (GDestroyNotify) gst_caps_unref);
276 gst_rtsp_stream_finalize (GObject * obj)
278 GstRTSPStream *stream;
279 GstRTSPStreamPrivate *priv;
281 stream = GST_RTSP_STREAM (obj);
284 GST_DEBUG ("finalize stream %p", stream);
286 /* we really need to be unjoined now */
287 g_return_if_fail (!priv->is_joined);
290 gst_rtsp_address_free (priv->addr_v4);
292 gst_rtsp_address_free (priv->addr_v6);
293 if (priv->server_addr_v4)
294 gst_rtsp_address_free (priv->server_addr_v4);
295 if (priv->server_addr_v6)
296 gst_rtsp_address_free (priv->server_addr_v6);
298 g_object_unref (priv->pool);
300 g_object_unref (priv->rtxsend);
302 gst_object_unref (priv->payloader);
304 gst_object_unref (priv->srcpad);
306 gst_object_unref (priv->sinkpad);
307 g_free (priv->control);
308 g_mutex_clear (&priv->lock);
310 g_hash_table_unref (priv->keys);
311 g_hash_table_destroy (priv->ptmap);
313 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
317 gst_rtsp_stream_get_property (GObject * object, guint propid,
318 GValue * value, GParamSpec * pspec)
320 GstRTSPStream *stream = GST_RTSP_STREAM (object);
324 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
327 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
330 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
333 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
338 gst_rtsp_stream_set_property (GObject * object, guint propid,
339 const GValue * value, GParamSpec * pspec)
341 GstRTSPStream *stream = GST_RTSP_STREAM (object);
345 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
348 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
351 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
354 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
359 * gst_rtsp_stream_new:
362 * @payloader: a #GstElement
364 * Create a new media stream with index @idx that handles RTP data on
365 * @pad and has a payloader element @payloader if @pad is a source pad
366 * or a depayloader element @payloader if @pad is a sink pad.
368 * Returns: (transfer full): a new #GstRTSPStream
371 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
373 GstRTSPStreamPrivate *priv;
374 GstRTSPStream *stream;
376 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
377 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
379 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
382 priv->payloader = gst_object_ref (payloader);
383 if (GST_PAD_IS_SRC (pad))
384 priv->srcpad = gst_object_ref (pad);
386 priv->sinkpad = gst_object_ref (pad);
392 * gst_rtsp_stream_get_index:
393 * @stream: a #GstRTSPStream
395 * Get the stream index.
397 * Return: the stream index.
400 gst_rtsp_stream_get_index (GstRTSPStream * stream)
402 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
404 return stream->priv->idx;
408 * gst_rtsp_stream_get_pt:
409 * @stream: a #GstRTSPStream
411 * Get the stream payload type.
413 * Return: the stream payload type.
416 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
418 GstRTSPStreamPrivate *priv;
421 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
425 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
431 * gst_rtsp_stream_get_srcpad:
432 * @stream: a #GstRTSPStream
434 * Get the srcpad associated with @stream.
436 * Returns: (transfer full): the srcpad. Unref after usage.
439 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
441 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
443 if (!stream->priv->srcpad)
446 return gst_object_ref (stream->priv->srcpad);
450 * gst_rtsp_stream_get_sinkpad:
451 * @stream: a #GstRTSPStream
453 * Get the sinkpad associated with @stream.
455 * Returns: (transfer full): the sinkpad. Unref after usage.
458 gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
460 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
462 if (!stream->priv->sinkpad)
465 return gst_object_ref (stream->priv->sinkpad);
469 * gst_rtsp_stream_get_control:
470 * @stream: a #GstRTSPStream
472 * Get the control string to identify this stream.
474 * Returns: (transfer full): the control string. g_free() after usage.
477 gst_rtsp_stream_get_control (GstRTSPStream * stream)
479 GstRTSPStreamPrivate *priv;
482 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
486 g_mutex_lock (&priv->lock);
487 if ((result = g_strdup (priv->control)) == NULL)
488 result = g_strdup_printf ("stream=%u", priv->idx);
489 g_mutex_unlock (&priv->lock);
495 * gst_rtsp_stream_set_control:
496 * @stream: a #GstRTSPStream
497 * @control: a control string
499 * Set the control string in @stream.
502 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
504 GstRTSPStreamPrivate *priv;
506 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
510 g_mutex_lock (&priv->lock);
511 g_free (priv->control);
512 priv->control = g_strdup (control);
513 g_mutex_unlock (&priv->lock);
517 * gst_rtsp_stream_has_control:
518 * @stream: a #GstRTSPStream
519 * @control: a control string
521 * Check if @stream has the control string @control.
523 * Returns: %TRUE is @stream has @control as the control string
526 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
528 GstRTSPStreamPrivate *priv;
531 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
535 g_mutex_lock (&priv->lock);
537 res = (g_strcmp0 (priv->control, control) == 0);
541 if (sscanf (control, "stream=%u", &streamid) > 0)
542 res = (streamid == priv->idx);
546 g_mutex_unlock (&priv->lock);
552 * gst_rtsp_stream_set_mtu:
553 * @stream: a #GstRTSPStream
556 * Configure the mtu in the payloader of @stream to @mtu.
559 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
561 GstRTSPStreamPrivate *priv;
563 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
567 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
569 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
573 * gst_rtsp_stream_get_mtu:
574 * @stream: a #GstRTSPStream
576 * Get the configured MTU in the payloader of @stream.
578 * Returns: the MTU of the payloader.
581 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
583 GstRTSPStreamPrivate *priv;
586 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
590 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
595 /* Update the dscp qos property on the udp sinks */
597 update_dscp_qos (GstRTSPStream * stream)
599 GstRTSPStreamPrivate *priv;
601 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
605 if (priv->udpsink[0]) {
606 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
610 if (priv->udpsink[1]) {
611 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
617 * gst_rtsp_stream_set_dscp_qos:
618 * @stream: a #GstRTSPStream
619 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
621 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
624 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
626 GstRTSPStreamPrivate *priv;
628 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
632 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
634 if (dscp_qos < -1 || dscp_qos > 63) {
635 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
639 priv->dscp_qos = dscp_qos;
641 update_dscp_qos (stream);
645 * gst_rtsp_stream_get_dscp_qos:
646 * @stream: a #GstRTSPStream
648 * Get the configured DSCP QoS in of the outgoing sockets.
650 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
653 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
655 GstRTSPStreamPrivate *priv;
657 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
661 return priv->dscp_qos;
665 * gst_rtsp_stream_is_transport_supported:
666 * @stream: a #GstRTSPStream
667 * @transport: (transfer none): a #GstRTSPTransport
669 * Check if @transport can be handled by stream
671 * Returns: %TRUE if @transport can be handled by @stream.
674 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
675 GstRTSPTransport * transport)
677 GstRTSPStreamPrivate *priv;
679 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
683 g_mutex_lock (&priv->lock);
684 if (transport->trans != GST_RTSP_TRANS_RTP)
685 goto unsupported_transmode;
687 if (!(transport->profile & priv->profiles))
688 goto unsupported_profile;
690 if (!(transport->lower_transport & priv->protocols))
691 goto unsupported_ltrans;
693 g_mutex_unlock (&priv->lock);
698 unsupported_transmode:
700 GST_DEBUG ("unsupported transport mode %d", transport->trans);
701 g_mutex_unlock (&priv->lock);
706 GST_DEBUG ("unsupported profile %d", transport->profile);
707 g_mutex_unlock (&priv->lock);
712 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
713 g_mutex_unlock (&priv->lock);
719 * gst_rtsp_stream_set_profiles:
720 * @stream: a #GstRTSPStream
721 * @profiles: the new profiles
723 * Configure the allowed profiles for @stream.
726 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
728 GstRTSPStreamPrivate *priv;
730 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
734 g_mutex_lock (&priv->lock);
735 priv->profiles = profiles;
736 g_mutex_unlock (&priv->lock);
740 * gst_rtsp_stream_get_profiles:
741 * @stream: a #GstRTSPStream
743 * Get the allowed profiles of @stream.
745 * Returns: a #GstRTSPProfile
748 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
750 GstRTSPStreamPrivate *priv;
753 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
757 g_mutex_lock (&priv->lock);
758 res = priv->profiles;
759 g_mutex_unlock (&priv->lock);
765 * gst_rtsp_stream_set_protocols:
766 * @stream: a #GstRTSPStream
767 * @protocols: the new flags
769 * Configure the allowed lower transport for @stream.
772 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
773 GstRTSPLowerTrans protocols)
775 GstRTSPStreamPrivate *priv;
777 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
781 g_mutex_lock (&priv->lock);
782 priv->protocols = protocols;
783 g_mutex_unlock (&priv->lock);
787 * gst_rtsp_stream_get_protocols:
788 * @stream: a #GstRTSPStream
790 * Get the allowed protocols of @stream.
792 * Returns: a #GstRTSPLowerTrans
795 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
797 GstRTSPStreamPrivate *priv;
798 GstRTSPLowerTrans res;
800 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
801 GST_RTSP_LOWER_TRANS_UNKNOWN);
805 g_mutex_lock (&priv->lock);
806 res = priv->protocols;
807 g_mutex_unlock (&priv->lock);
813 * gst_rtsp_stream_set_address_pool:
814 * @stream: a #GstRTSPStream
815 * @pool: (transfer none): a #GstRTSPAddressPool
817 * configure @pool to be used as the address pool of @stream.
820 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
821 GstRTSPAddressPool * pool)
823 GstRTSPStreamPrivate *priv;
824 GstRTSPAddressPool *old;
826 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
830 GST_LOG_OBJECT (stream, "set address pool %p", pool);
832 g_mutex_lock (&priv->lock);
833 if ((old = priv->pool) != pool)
834 priv->pool = pool ? g_object_ref (pool) : NULL;
837 g_mutex_unlock (&priv->lock);
840 g_object_unref (old);
844 * gst_rtsp_stream_get_address_pool:
845 * @stream: a #GstRTSPStream
847 * Get the #GstRTSPAddressPool used as the address pool of @stream.
849 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
853 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
855 GstRTSPStreamPrivate *priv;
856 GstRTSPAddressPool *result;
858 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
862 g_mutex_lock (&priv->lock);
863 if ((result = priv->pool))
864 g_object_ref (result);
865 g_mutex_unlock (&priv->lock);
871 * gst_rtsp_stream_get_multicast_address:
872 * @stream: a #GstRTSPStream
873 * @family: the #GSocketFamily
875 * Get the multicast address of @stream for @family.
877 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
878 * or %NULL when no address could be allocated. gst_rtsp_address_free()
882 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
883 GSocketFamily family)
885 GstRTSPStreamPrivate *priv;
886 GstRTSPAddress *result;
887 GstRTSPAddress **addrp;
888 GstRTSPAddressFlags flags;
890 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
894 if (family == G_SOCKET_FAMILY_IPV6) {
895 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
896 addrp = &priv->addr_v6;
898 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
899 addrp = &priv->addr_v4;
902 g_mutex_lock (&priv->lock);
903 if (*addrp == NULL) {
904 if (priv->pool == NULL)
907 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
909 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
913 result = gst_rtsp_address_copy (*addrp);
914 g_mutex_unlock (&priv->lock);
921 GST_ERROR_OBJECT (stream, "no address pool specified");
922 g_mutex_unlock (&priv->lock);
927 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
928 g_mutex_unlock (&priv->lock);
934 * gst_rtsp_stream_reserve_address:
935 * @stream: a #GstRTSPStream
936 * @address: an address
941 * Reserve @address and @port as the address and port of @stream.
943 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
944 * the address could be reserved. gst_rtsp_address_free() after usage.
947 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
948 const gchar * address, guint port, guint n_ports, guint ttl)
950 GstRTSPStreamPrivate *priv;
951 GstRTSPAddress *result;
953 GSocketFamily family;
954 GstRTSPAddress **addrp;
956 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
957 g_return_val_if_fail (address != NULL, NULL);
958 g_return_val_if_fail (port > 0, NULL);
959 g_return_val_if_fail (n_ports > 0, NULL);
960 g_return_val_if_fail (ttl > 0, NULL);
964 addr = g_inet_address_new_from_string (address);
966 GST_ERROR ("failed to get inet addr from %s", address);
967 family = G_SOCKET_FAMILY_IPV4;
969 family = g_inet_address_get_family (addr);
970 g_object_unref (addr);
973 if (family == G_SOCKET_FAMILY_IPV6)
974 addrp = &priv->addr_v6;
976 addrp = &priv->addr_v4;
978 g_mutex_lock (&priv->lock);
979 if (*addrp == NULL) {
980 GstRTSPAddressPoolResult res;
982 if (priv->pool == NULL)
985 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
986 port, n_ports, ttl, addrp);
987 if (res != GST_RTSP_ADDRESS_POOL_OK)
990 if (strcmp ((*addrp)->address, address) ||
991 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
992 (*addrp)->ttl != ttl)
993 goto different_address;
995 result = gst_rtsp_address_copy (*addrp);
996 g_mutex_unlock (&priv->lock);
1003 GST_ERROR_OBJECT (stream, "no address pool specified");
1004 g_mutex_unlock (&priv->lock);
1009 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
1011 g_mutex_unlock (&priv->lock);
1016 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
1017 " reserved", address);
1018 g_mutex_unlock (&priv->lock);
1023 /* must be called with lock */
1025 set_sockets_for_udpsinks (GstRTSPStream * stream, GSocket * rtp_socket,
1026 GSocket * rtcp_socket, GSocketFamily family)
1028 GstRTSPStreamPrivate *priv = stream->priv;
1029 const gchar *multisink_socket;
1031 if (family == G_SOCKET_FAMILY_IPV6)
1032 multisink_socket = "socket-v6";
1034 multisink_socket = "socket";
1036 g_object_set (G_OBJECT (priv->udpsink[0]), multisink_socket, rtp_socket,
1038 g_object_set (G_OBJECT (priv->udpsink[1]), multisink_socket, rtcp_socket,
1042 /* must be called with lock */
1044 create_and_configure_udpsinks (GstRTSPStream * stream)
1046 GstRTSPStreamPrivate *priv = stream->priv;
1047 GstElement *udpsink0, *udpsink1;
1052 if (priv->udpsink[0])
1053 udpsink0 = priv->udpsink[0];
1055 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1058 goto no_udp_protocol;
1060 if (priv->udpsink[1])
1061 udpsink1 = priv->udpsink[1];
1063 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1066 goto no_udp_protocol;
1068 /* configure sinks */
1070 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1071 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1073 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1074 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1076 g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
1078 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1079 /* Needs to be async for RECORD streams, otherwise we will never go to
1080 * PLAYING because the sinks will wait for data while the udpsrc can't
1081 * provide data with timestamps in PAUSED. */
1083 g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
1084 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1086 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1087 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1089 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1090 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1092 /* update the dscp qos field in the sinks */
1093 update_dscp_qos (stream);
1095 priv->udpsink[0] = udpsink0;
1096 priv->udpsink[1] = udpsink1;
1107 /* must be called with lock */
1109 create_and_configure_udpsources_one_family (GstElement * udpsrc_out[2],
1110 GSocket * rtp_socket, GSocket * rtcp_socket, GSocketFamily family)
1112 GstStateChangeReturn ret;
1114 /* we keep these elements, we will further configure them when the
1115 * client told us to really use the UDP ports. */
1116 udpsrc_out[0] = gst_element_factory_make ("udpsrc", NULL);
1117 udpsrc_out[1] = gst_element_factory_make ("udpsrc", NULL);
1119 if (udpsrc_out[0] == NULL || udpsrc_out[1] == NULL)
1122 g_object_set (G_OBJECT (udpsrc_out[0]), "socket", rtp_socket, NULL);
1123 g_object_set (G_OBJECT (udpsrc_out[1]), "socket", rtcp_socket, NULL);
1125 ret = gst_element_set_state (udpsrc_out[0], GST_STATE_READY);
1126 if (ret == GST_STATE_CHANGE_FAILURE)
1128 ret = gst_element_set_state (udpsrc_out[1], GST_STATE_READY);
1129 if (ret == GST_STATE_CHANGE_FAILURE)
1139 gst_object_unref (udpsrc_out[0]);
1141 gst_object_unref (udpsrc_out[1]);
1147 alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
1148 GstElement * udpsrc_out[2], GstRTSPRange * server_port_out,
1149 GstRTSPAddress ** server_addr_out)
1151 GstRTSPStreamPrivate *priv = stream->priv;
1152 GSocket *rtp_socket = NULL;
1153 GSocket *rtcp_socket;
1154 gint tmp_rtp, tmp_rtcp;
1156 gint rtpport, rtcpport;
1157 GList *rejected_addresses = NULL;
1158 GstRTSPAddress *addr = NULL;
1159 GInetAddress *inetaddr = NULL;
1160 GSocketAddress *rtp_sockaddr = NULL;
1161 GSocketAddress *rtcp_sockaddr = NULL;
1162 GstRTSPAddressPool * pool;
1167 /* Start with random port */
1170 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1171 G_SOCKET_PROTOCOL_UDP, NULL);
1173 goto no_udp_protocol;
1175 if (*server_addr_out)
1176 gst_rtsp_address_free (*server_addr_out);
1178 /* try to allocate 2 UDP ports, the RTP port should be an even
1179 * number and the RTCP port should be the next (uneven) port */
1182 if (rtp_socket == NULL) {
1183 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1184 G_SOCKET_PROTOCOL_UDP, NULL);
1186 goto no_udp_protocol;
1189 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
1190 GstRTSPAddressFlags flags;
1193 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1195 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1196 if (family == G_SOCKET_FAMILY_IPV6)
1197 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1199 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1201 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1206 tmp_rtp = addr->port;
1208 g_clear_object (&inetaddr);
1209 inetaddr = g_inet_address_new_from_string (addr->address);
1217 if (inetaddr == NULL)
1218 inetaddr = g_inet_address_new_any (family);
1221 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1222 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1223 g_object_unref (rtp_sockaddr);
1226 g_object_unref (rtp_sockaddr);
1228 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1229 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1230 g_clear_object (&rtp_sockaddr);
1235 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1236 g_object_unref (rtp_sockaddr);
1238 /* check if port is even */
1239 if ((tmp_rtp & 1) != 0) {
1240 /* port not even, close and allocate another */
1242 g_clear_object (&rtp_socket);
1247 tmp_rtcp = tmp_rtp + 1;
1249 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1250 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1251 g_object_unref (rtcp_sockaddr);
1252 g_clear_object (&rtp_socket);
1255 g_object_unref (rtcp_sockaddr);
1257 g_clear_object (&inetaddr);
1259 if (!create_and_configure_udpsources_one_family (udpsrc_out, rtp_socket,
1260 rtcp_socket, family))
1261 goto no_udp_protocol;
1263 g_object_get (G_OBJECT (udpsrc_out[0]), "port", &rtpport, NULL);
1264 g_object_get (G_OBJECT (udpsrc_out[1]), "port", &rtcpport, NULL);
1266 /* this should not happen... */
1267 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1270 if (!create_and_configure_udpsinks (stream))
1271 goto no_udp_protocol;
1273 /* set RTP and RTCP sockets */
1274 set_sockets_for_udpsinks (stream, rtp_socket, rtcp_socket, family);
1276 server_port_out->min = rtpport;
1277 server_port_out->max = rtcpport;
1279 *server_addr_out = addr;
1280 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1282 g_object_unref (rtp_socket);
1283 g_object_unref (rtcp_socket);
1307 g_object_unref (inetaddr);
1308 g_list_free_full (rejected_addresses,
1309 (GDestroyNotify) gst_rtsp_address_free);
1311 gst_rtsp_address_free (addr);
1313 g_object_unref (rtp_socket);
1315 g_object_unref (rtcp_socket);
1320 /* must be called with lock */
1322 alloc_ports (GstRTSPStream * stream)
1324 GstRTSPStreamPrivate *priv = stream->priv;
1327 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4,
1328 &priv->server_port_v4, &priv->server_addr_v4);
1331 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6,
1332 &priv->server_port_v6, &priv->server_addr_v6);
1334 return priv->have_ipv4 || priv->have_ipv6;
1338 * gst_rtsp_stream_set_client_side:
1339 * @stream: a #GstRTSPStream
1340 * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
1341 * an RTSP connection.
1343 * Sets the #GstRTSPStream as a 'client side' stream - used for sending
1344 * streams to an RTSP server via RECORD. This has the practical effect
1345 * of changing which UDP port numbers are used when setting up the local
1346 * side of the stream sending to be either the 'server' or 'client' pair
1347 * of a configured UDP transport.
1350 gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
1352 GstRTSPStreamPrivate *priv;
1354 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1355 priv = stream->priv;
1356 g_mutex_lock (&priv->lock);
1357 priv->client_side = client_side;
1358 g_mutex_unlock (&priv->lock);
1362 * gst_rtsp_stream_set_client_side:
1363 * @stream: a #GstRTSPStream
1365 * See gst_rtsp_stream_set_client_side()
1367 * Returns: TRUE if this #GstRTSPStream is client-side.
1370 gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
1372 GstRTSPStreamPrivate *priv;
1375 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1377 priv = stream->priv;
1378 g_mutex_lock (&priv->lock);
1379 ret = priv->client_side;
1380 g_mutex_unlock (&priv->lock);
1386 * gst_rtsp_stream_get_server_port:
1387 * @stream: a #GstRTSPStream
1388 * @server_port: (out): result server port
1389 * @family: the port family to get
1391 * Fill @server_port with the port pair used by the server. This function can
1392 * only be called when @stream has been joined.
1395 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1396 GstRTSPRange * server_port, GSocketFamily family)
1398 GstRTSPStreamPrivate *priv;
1400 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1401 priv = stream->priv;
1402 g_return_if_fail (priv->is_joined);
1404 g_mutex_lock (&priv->lock);
1405 if (family == G_SOCKET_FAMILY_IPV4) {
1407 *server_port = priv->server_port_v4;
1410 *server_port = priv->server_port_v6;
1412 g_mutex_unlock (&priv->lock);
1416 * gst_rtsp_stream_get_rtpsession:
1417 * @stream: a #GstRTSPStream
1419 * Get the RTP session of this stream.
1421 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1424 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1426 GstRTSPStreamPrivate *priv;
1429 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1431 priv = stream->priv;
1433 g_mutex_lock (&priv->lock);
1434 if ((session = priv->session))
1435 g_object_ref (session);
1436 g_mutex_unlock (&priv->lock);
1442 * gst_rtsp_stream_get_ssrc:
1443 * @stream: a #GstRTSPStream
1444 * @ssrc: (out): result ssrc
1446 * Get the SSRC used by the RTP session of this stream. This function can only
1447 * be called when @stream has been joined.
1450 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1452 GstRTSPStreamPrivate *priv;
1454 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1455 priv = stream->priv;
1456 g_return_if_fail (priv->is_joined);
1458 g_mutex_lock (&priv->lock);
1459 if (ssrc && priv->session)
1460 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1461 g_mutex_unlock (&priv->lock);
1465 * gst_rtsp_stream_set_retransmission_time:
1466 * @stream: a #GstRTSPStream
1467 * @time: a #GstClockTime
1469 * Set the amount of time to store retransmission packets.
1472 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1475 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1477 g_mutex_lock (&stream->priv->lock);
1478 stream->priv->rtx_time = time;
1479 if (stream->priv->rtxsend)
1480 g_object_set (stream->priv->rtxsend, "max-size-time",
1481 GST_TIME_AS_MSECONDS (time), NULL);
1482 g_mutex_unlock (&stream->priv->lock);
1486 * gst_rtsp_stream_get_retransmission_time:
1487 * @stream: a #GstRTSPStream
1489 * Get the amount of time to store retransmission data.
1491 * Returns: the amount of time to store retransmission data.
1494 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1498 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1500 g_mutex_lock (&stream->priv->lock);
1501 ret = stream->priv->rtx_time;
1502 g_mutex_unlock (&stream->priv->lock);
1508 * gst_rtsp_stream_set_retransmission_pt:
1509 * @stream: a #GstRTSPStream
1512 * Set the payload type (pt) for retransmission of this stream.
1515 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1517 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1519 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1521 g_mutex_lock (&stream->priv->lock);
1522 stream->priv->rtx_pt = rtx_pt;
1523 if (stream->priv->rtxsend) {
1524 guint pt = gst_rtsp_stream_get_pt (stream);
1525 gchar *pt_s = g_strdup_printf ("%d", pt);
1526 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1527 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1528 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1530 gst_structure_free (rtx_pt_map);
1532 g_mutex_unlock (&stream->priv->lock);
1536 * gst_rtsp_stream_get_retransmission_pt:
1537 * @stream: a #GstRTSPStream
1539 * Get the payload-type used for retransmission of this stream
1541 * Returns: The retransmission PT.
1544 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1548 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1550 g_mutex_lock (&stream->priv->lock);
1551 rtx_pt = stream->priv->rtx_pt;
1552 g_mutex_unlock (&stream->priv->lock);
1558 * gst_rtsp_stream_set_buffer_size:
1559 * @stream: a #GstRTSPStream
1560 * @size: the buffer size
1562 * Set the size of the UDP transmission buffer (in bytes)
1563 * Needs to be set before the stream is joined to a bin.
1568 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1570 g_mutex_lock (&stream->priv->lock);
1571 stream->priv->buffer_size = size;
1572 g_mutex_unlock (&stream->priv->lock);
1576 * gst_rtsp_stream_get_buffer_size:
1577 * @stream: a #GstRTSPStream
1579 * Get the size of the UDP transmission buffer (in bytes)
1581 * Returns: the size of the UDP TX buffer
1586 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1590 g_mutex_lock (&stream->priv->lock);
1591 buffer_size = stream->priv->buffer_size;
1592 g_mutex_unlock (&stream->priv->lock);
1597 /* executed from streaming thread */
1599 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1601 GstRTSPStreamPrivate *priv = stream->priv;
1602 GstCaps *newcaps, *oldcaps;
1604 newcaps = gst_pad_get_current_caps (pad);
1606 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1609 g_mutex_lock (&priv->lock);
1610 oldcaps = priv->caps;
1611 priv->caps = newcaps;
1612 g_mutex_unlock (&priv->lock);
1615 gst_caps_unref (oldcaps);
1619 dump_structure (const GstStructure * s)
1623 sstr = gst_structure_to_string (s);
1624 GST_INFO ("structure: %s", sstr);
1628 static GstRTSPStreamTransport *
1629 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1631 GstRTSPStreamPrivate *priv = stream->priv;
1633 GstRTSPStreamTransport *result = NULL;
1638 if (rtcp_from == NULL)
1641 tmp = g_strrstr (rtcp_from, ":");
1645 port = atoi (tmp + 1);
1646 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1648 g_mutex_lock (&priv->lock);
1649 GST_INFO ("finding %s:%d in %d transports", dest, port,
1650 g_list_length (priv->transports));
1652 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1653 GstRTSPStreamTransport *trans = walk->data;
1654 const GstRTSPTransport *tr;
1657 tr = gst_rtsp_stream_transport_get_transport (trans);
1659 if (priv->client_side) {
1660 /* In client side mode the 'destination' is the RTSP server, so send
1662 min = tr->server_port.min;
1663 max = tr->server_port.max;
1665 min = tr->client_port.min;
1666 max = tr->client_port.max;
1669 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1675 g_object_ref (result);
1676 g_mutex_unlock (&priv->lock);
1683 static GstRTSPStreamTransport *
1684 check_transport (GObject * source, GstRTSPStream * stream)
1686 GstStructure *stats;
1687 GstRTSPStreamTransport *trans;
1689 /* see if we have a stream to match with the origin of the RTCP packet */
1690 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1691 if (trans == NULL) {
1692 g_object_get (source, "stats", &stats, NULL);
1694 const gchar *rtcp_from;
1696 dump_structure (stats);
1698 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1699 if ((trans = find_transport (stream, rtcp_from))) {
1700 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1702 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1705 gst_structure_free (stats);
1713 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1715 GstRTSPStreamTransport *trans;
1717 GST_INFO ("%p: new source %p", stream, source);
1719 trans = check_transport (source, stream);
1722 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1726 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1728 GST_INFO ("%p: new SDES %p", stream, source);
1732 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1734 GstRTSPStreamTransport *trans;
1736 trans = check_transport (source, stream);
1739 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1740 gst_rtsp_stream_transport_keep_alive (trans);
1744 GstStructure *stats;
1745 g_object_get (source, "stats", &stats, NULL);
1747 dump_structure (stats);
1748 gst_structure_free (stats);
1755 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1757 GST_INFO ("%p: source %p bye", stream, source);
1761 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1763 GstRTSPStreamTransport *trans;
1765 GST_INFO ("%p: source %p bye timeout", stream, source);
1767 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1768 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1769 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1774 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1776 GstRTSPStreamTransport *trans;
1778 GST_INFO ("%p: source %p timeout", stream, source);
1780 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1781 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1782 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1787 on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1789 GST_INFO ("%p: new sender source %p", stream, source);
1792 GstStructure *stats;
1793 g_object_get (source, "stats", &stats, NULL);
1795 dump_structure (stats);
1796 gst_structure_free (stats);
1803 on_sender_ssrc_active (GObject * session, GObject * source,
1804 GstRTSPStream * stream)
1808 GstStructure *stats;
1809 g_object_get (source, "stats", &stats, NULL);
1811 dump_structure (stats);
1812 gst_structure_free (stats);
1819 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1822 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1823 g_list_free (priv->tr_cache_rtp);
1824 priv->tr_cache_rtp = NULL;
1826 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1827 g_list_free (priv->tr_cache_rtcp);
1828 priv->tr_cache_rtcp = NULL;
1832 static GstFlowReturn
1833 handle_new_sample (GstAppSink * sink, gpointer user_data)
1835 GstRTSPStreamPrivate *priv;
1839 GstRTSPStream *stream;
1842 sample = gst_app_sink_pull_sample (sink);
1846 stream = (GstRTSPStream *) user_data;
1847 priv = stream->priv;
1848 buffer = gst_sample_get_buffer (sample);
1850 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1852 g_mutex_lock (&priv->lock);
1854 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1855 clear_tr_cache (priv, is_rtp);
1856 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1857 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1858 priv->tr_cache_rtp =
1859 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1861 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1864 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1865 clear_tr_cache (priv, is_rtp);
1866 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1867 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1868 priv->tr_cache_rtcp =
1869 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
1871 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
1874 g_mutex_unlock (&priv->lock);
1877 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
1878 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1879 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1882 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
1883 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1884 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1887 gst_sample_unref (sample);
1892 static GstAppSinkCallbacks sink_cb = {
1893 NULL, /* not interested in EOS */
1894 NULL, /* not interested in preroll samples */
1899 get_rtp_encoder (GstRTSPStream * stream, guint session)
1901 GstRTSPStreamPrivate *priv = stream->priv;
1903 if (priv->srtpenc == NULL) {
1906 name = g_strdup_printf ("srtpenc_%u", session);
1907 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1910 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1912 return gst_object_ref (priv->srtpenc);
1916 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1918 GstRTSPStreamPrivate *priv = stream->priv;
1919 GstElement *oldenc, *enc;
1923 if (priv->idx != session)
1926 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1928 oldenc = priv->srtpenc;
1929 enc = get_rtp_encoder (stream, session);
1930 name = g_strdup_printf ("rtp_sink_%d", session);
1931 pad = gst_element_get_request_pad (enc, name);
1933 gst_object_unref (pad);
1936 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
1943 request_rtcp_encoder (GstElement * rtpbin, guint session,
1944 GstRTSPStream * stream)
1946 GstRTSPStreamPrivate *priv = stream->priv;
1947 GstElement *oldenc, *enc;
1951 if (priv->idx != session)
1954 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1956 oldenc = priv->srtpenc;
1957 enc = get_rtp_encoder (stream, session);
1958 name = g_strdup_printf ("rtcp_sink_%d", session);
1959 pad = gst_element_get_request_pad (enc, name);
1961 gst_object_unref (pad);
1964 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
1971 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1973 GstRTSPStreamPrivate *priv = stream->priv;
1976 GST_DEBUG ("request key %08x", ssrc);
1978 g_mutex_lock (&priv->lock);
1979 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1980 gst_caps_ref (caps);
1981 g_mutex_unlock (&priv->lock);
1987 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
1988 GstRTSPStream * stream)
1990 GstRTSPStreamPrivate *priv = stream->priv;
1992 if (priv->idx != session)
1995 if (priv->srtpdec == NULL) {
1998 name = g_strdup_printf ("srtpdec_%u", session);
1999 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
2002 g_signal_connect (priv->srtpdec, "request-key",
2003 (GCallback) request_key, stream);
2005 return gst_object_ref (priv->srtpdec);
2009 * gst_rtsp_stream_request_aux_sender:
2010 * @stream: a #GstRTSPStream
2011 * @sessid: the session id
2013 * Creating a rtxsend bin
2015 * Returns: (transfer full): a #GstElement.
2020 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
2024 GstStructure *pt_map;
2029 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2031 pt = gst_rtsp_stream_get_pt (stream);
2032 pt_s = g_strdup_printf ("%u", pt);
2033 rtx_pt = stream->priv->rtx_pt;
2035 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
2037 bin = gst_bin_new (NULL);
2038 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
2039 pt_map = gst_structure_new ("application/x-rtp-pt-map",
2040 pt_s, G_TYPE_UINT, rtx_pt, NULL);
2041 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
2042 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
2044 gst_structure_free (pt_map);
2045 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
2047 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
2048 name = g_strdup_printf ("src_%u", sessid);
2049 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2051 gst_object_unref (pad);
2053 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
2054 name = g_strdup_printf ("sink_%u", sessid);
2055 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2057 gst_object_unref (pad);
2063 * gst_rtsp_stream_set_pt_map:
2064 * @stream: a #GstRTSPStream
2068 * Configure a pt map between @pt and @caps.
2071 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
2073 GstRTSPStreamPrivate *priv = stream->priv;
2075 g_mutex_lock (&priv->lock);
2076 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
2077 g_mutex_unlock (&priv->lock);
2081 request_pt_map (GstElement * rtpbin, guint session, guint pt,
2082 GstRTSPStream * stream)
2084 GstRTSPStreamPrivate *priv = stream->priv;
2085 GstCaps *caps = NULL;
2087 g_mutex_lock (&priv->lock);
2089 if (priv->idx == session) {
2090 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
2092 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
2093 gst_caps_ref (caps);
2095 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
2099 g_mutex_unlock (&priv->lock);
2105 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
2107 GstRTSPStreamPrivate *priv = stream->priv;
2109 GstPadLinkReturn ret;
2112 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
2113 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2115 name = gst_pad_get_name (pad);
2116 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
2122 if (priv->idx != sessid)
2125 if (gst_pad_is_linked (priv->sinkpad)) {
2126 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
2127 GST_DEBUG_PAD_NAME (priv->sinkpad));
2131 /* link the RTP pad to the session manager, it should not really fail unless
2132 * this is not really an RTP pad */
2133 ret = gst_pad_link (pad, priv->sinkpad);
2134 if (ret != GST_PAD_LINK_OK)
2136 priv->recv_rtp_src = gst_object_ref (pad);
2143 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2144 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2149 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2150 GstRTSPStream * stream)
2152 /* TODO: What to do here other than this? */
2153 GST_DEBUG ("Stream %p: Got EOS", stream);
2154 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2157 /* must be called with lock */
2159 create_sender_part (GstRTSPStream * stream, GstBin * bin,
2162 GstRTSPStreamPrivate *priv;
2163 GstPad *pad, *sinkpad = NULL;
2164 gboolean is_tcp = FALSE, is_udp = FALSE;
2167 priv = stream->priv;
2169 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2170 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2171 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2173 for (i = 0; i < 2; i++) {
2174 GstPad *teepad, *queuepad;
2175 /* For the sender we create this bit of pipeline for both
2176 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2177 * we need to add a queue before appsink and udpsink to make
2178 * the pipeline not block. For the TCP case, we want to pump
2179 * client as fast as possible anyway. This pipeline is used
2180 * when both TCP and UDP are present.
2182 * .--------. .-----. .---------. .---------.
2183 * | rtpbin | | tee | | queue | | udpsink |
2184 * | send->sink src->sink src->sink |
2185 * '--------' | | '---------' '---------'
2186 * | | .---------. .---------.
2187 * | | | queue | | appsink |
2188 * | src->sink src->sink |
2189 * '-----' '---------' '---------'
2191 * When only UDP or only TCP is allowed, we skip the tee and queue
2192 * and link the udpsink (for UDP) or appsink (for TCP) directly to
2195 /* Only link the RTP send src if we're going to send RTP, link
2196 * the RTCP send src always */
2197 if (priv->srcpad || i == 1) {
2200 gst_bin_add (bin, priv->udpsink[i]);
2201 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2206 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2207 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2208 gst_bin_add (bin, priv->appsink[i]);
2209 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2210 &sink_cb, stream, NULL);
2213 if (is_udp && is_tcp) {
2214 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2216 /* make tee for RTP/RTCP */
2217 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2218 gst_bin_add (bin, priv->tee[i]);
2220 /* and link to rtpbin send pad */
2221 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2222 gst_pad_link (priv->send_src[i], pad);
2223 gst_object_unref (pad);
2225 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2226 g_object_set (priv->udpqueue[i], "max-size-buffers",
2227 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2229 gst_bin_add (bin, priv->udpqueue[i]);
2230 /* link tee to udpqueue */
2231 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2232 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2233 gst_pad_link (teepad, pad);
2234 gst_object_unref (pad);
2235 gst_object_unref (teepad);
2237 /* link udpqueue to udpsink */
2238 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2239 gst_pad_link (queuepad, sinkpad);
2240 gst_object_unref (queuepad);
2241 gst_object_unref (sinkpad);
2244 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2245 g_object_set (priv->appqueue[i], "max-size-buffers",
2246 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2248 gst_bin_add (bin, priv->appqueue[i]);
2249 /* and link tee to appqueue */
2250 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2251 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2252 gst_pad_link (teepad, pad);
2253 gst_object_unref (pad);
2254 gst_object_unref (teepad);
2256 /* and link appqueue to appsink */
2257 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2258 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2259 gst_pad_link (queuepad, pad);
2260 gst_object_unref (pad);
2261 gst_object_unref (queuepad);
2262 } else if (is_tcp) {
2263 /* only appsink needed, link it to the session */
2264 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2265 gst_pad_link (priv->send_src[i], pad);
2266 gst_object_unref (pad);
2268 /* when its only TCP, we need to set sync and preroll to FALSE
2269 * for the sink to avoid deadlock. And this is only needed for
2270 * sink used for RTCP data, not the RTP data. */
2272 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2274 /* else only udpsink needed, link it to the session */
2275 gst_pad_link (priv->send_src[i], sinkpad);
2276 gst_object_unref (sinkpad);
2280 /* check if we need to set to a special state */
2281 if (state != GST_STATE_NULL) {
2282 if (priv->udpsink[i] && (priv->srcpad || i == 1))
2283 gst_element_set_state (priv->udpsink[i], state);
2284 if (priv->appsink[i] && (priv->srcpad || i == 1))
2285 gst_element_set_state (priv->appsink[i], state);
2286 if (priv->appqueue[i] && (priv->srcpad || i == 1))
2287 gst_element_set_state (priv->appqueue[i], state);
2288 if (priv->udpqueue[i] && (priv->srcpad || i == 1))
2289 gst_element_set_state (priv->udpqueue[i], state);
2290 if (priv->tee[i] && (priv->srcpad || i == 1))
2291 gst_element_set_state (priv->tee[i], state);
2296 /* must be called with lock */
2298 create_receiver_part (GstRTSPStream * stream, GstBin * bin,
2301 GstRTSPStreamPrivate *priv;
2302 GstPad *pad, *selpad;
2306 priv = stream->priv;
2308 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2310 for (i = 0; i < 2; i++) {
2311 /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
2312 * RTCP sink always */
2313 if (priv->sinkpad || i == 1) {
2314 /* For the receiver we create this bit of pipeline for both
2315 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2316 * and it is all funneled into the rtpbin receive pad.
2318 * .--------. .--------. .--------.
2319 * | udpsrc | | funnel | | rtpbin |
2320 * | src->sink src->sink |
2321 * '--------' | | '--------'
2325 * '--------' '--------'
2327 /* make funnel for the RTP/RTCP receivers */
2328 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2329 gst_bin_add (bin, priv->funnel[i]);
2331 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2332 gst_pad_link (pad, priv->recv_sink[i]);
2333 gst_object_unref (pad);
2335 if (priv->udpsrc_v4[i]) {
2337 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2338 * values. This is only relevant for PLAY pipelines */
2339 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2340 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2343 gst_bin_add (bin, priv->udpsrc_v4[i]);
2345 /* and link to the funnel v4 */
2346 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2347 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2348 gst_pad_link (pad, selpad);
2349 gst_object_unref (pad);
2350 gst_object_unref (selpad);
2353 if (priv->udpsrc_v6[i]) {
2355 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2356 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2358 gst_bin_add (bin, priv->udpsrc_v6[i]);
2360 /* and link to the funnel v6 */
2361 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2362 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2363 gst_pad_link (pad, selpad);
2364 gst_object_unref (pad);
2365 gst_object_unref (selpad);
2369 /* make and add appsrc */
2370 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2371 priv->appsrc_base_time[i] = -1;
2372 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2373 gst_bin_add (bin, priv->appsrc[i]);
2374 /* and link to the funnel */
2375 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2376 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2377 gst_pad_link (pad, selpad);
2378 gst_object_unref (pad);
2379 gst_object_unref (selpad);
2383 /* check if we need to set to a special state */
2384 if (state != GST_STATE_NULL) {
2385 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2386 gst_element_set_state (priv->funnel[i], state);
2387 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2388 gst_element_set_state (priv->appsrc[i], state);
2394 * gst_rtsp_stream_join_bin:
2395 * @stream: a #GstRTSPStream
2396 * @bin: (transfer none): a #GstBin to join
2397 * @rtpbin: (transfer none): a rtpbin element in @bin
2398 * @state: the target state of the new elements
2400 * Join the #GstBin @bin that contains the element @rtpbin.
2402 * @stream will link to @rtpbin, which must be inside @bin. The elements
2403 * added to @bin will be set to the state given in @state.
2405 * Returns: %TRUE on success.
2408 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2409 GstElement * rtpbin, GstState state)
2411 GstRTSPStreamPrivate *priv;
2414 GstPadLinkReturn ret;
2417 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2418 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2419 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2421 priv = stream->priv;
2423 g_mutex_lock (&priv->lock);
2424 if (priv->is_joined)
2427 /* create a session with the same index as the stream */
2430 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2432 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2433 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2435 if (is_udp && !alloc_ports (stream))
2438 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2439 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2441 g_signal_connect (rtpbin, "request-rtp-encoder",
2442 (GCallback) request_rtp_encoder, stream);
2443 g_signal_connect (rtpbin, "request-rtcp-encoder",
2444 (GCallback) request_rtcp_encoder, stream);
2445 g_signal_connect (rtpbin, "request-rtp-decoder",
2446 (GCallback) request_rtp_rtcp_decoder, stream);
2447 g_signal_connect (rtpbin, "request-rtcp-decoder",
2448 (GCallback) request_rtp_rtcp_decoder, stream);
2451 if (priv->sinkpad) {
2452 g_signal_connect (rtpbin, "request-pt-map",
2453 (GCallback) request_pt_map, stream);
2456 /* get pads from the RTP session element for sending and receiving
2459 /* get a pad for sending RTP */
2460 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2461 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2464 /* link the RTP pad to the session manager, it should not really fail unless
2465 * this is not really an RTP pad */
2466 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2467 if (ret != GST_PAD_LINK_OK)
2470 name = g_strdup_printf ("send_rtp_src_%u", idx);
2471 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2474 /* Need to connect our sinkpad from here */
2475 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2477 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2479 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2480 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2484 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2485 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2487 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2488 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2491 /* get the session */
2492 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2494 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2496 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2498 g_signal_connect (priv->session, "on-ssrc-active",
2499 (GCallback) on_ssrc_active, stream);
2500 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2502 g_signal_connect (priv->session, "on-bye-timeout",
2503 (GCallback) on_bye_timeout, stream);
2504 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2507 /* signal for sender ssrc */
2508 g_signal_connect (priv->session, "on-new-sender-ssrc",
2509 (GCallback) on_new_sender_ssrc, stream);
2510 g_signal_connect (priv->session, "on-sender-ssrc-active",
2511 (GCallback) on_sender_ssrc_active, stream);
2513 create_sender_part (stream, bin, state);
2515 create_receiver_part (stream, bin, state);
2518 /* be notified of caps changes */
2519 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2520 (GCallback) caps_notify, stream);
2523 priv->is_joined = TRUE;
2524 g_mutex_unlock (&priv->lock);
2531 g_mutex_unlock (&priv->lock);
2536 g_mutex_unlock (&priv->lock);
2537 GST_WARNING ("failed to allocate ports %u", idx);
2542 GST_WARNING ("failed to link stream %u", idx);
2543 gst_object_unref (priv->send_rtp_sink);
2544 priv->send_rtp_sink = NULL;
2545 g_mutex_unlock (&priv->lock);
2551 * gst_rtsp_stream_leave_bin:
2552 * @stream: a #GstRTSPStream
2553 * @bin: (transfer none): a #GstBin
2554 * @rtpbin: (transfer none): a rtpbin #GstElement
2556 * Remove the elements of @stream from @bin.
2558 * Return: %TRUE on success.
2561 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2562 GstElement * rtpbin)
2564 GstRTSPStreamPrivate *priv;
2567 gboolean is_tcp, is_udp;
2569 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2570 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2571 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2573 priv = stream->priv;
2575 g_mutex_lock (&priv->lock);
2576 if (!priv->is_joined)
2577 goto was_not_joined;
2579 /* all transports must be removed by now */
2580 if (priv->transports != NULL)
2581 goto transports_not_removed;
2583 clear_tr_cache (priv, TRUE);
2584 clear_tr_cache (priv, FALSE);
2586 GST_INFO ("stream %p leaving bin", stream);
2589 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2591 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2592 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2593 gst_object_unref (priv->send_rtp_sink);
2594 priv->send_rtp_sink = NULL;
2595 } else if (priv->recv_rtp_src) {
2596 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2597 gst_object_unref (priv->recv_rtp_src);
2598 priv->recv_rtp_src = NULL;
2601 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2603 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2604 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2607 for (i = 0; i < 2; i++) {
2608 if (priv->udpsink[i])
2609 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2610 if (priv->appsink[i])
2611 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2612 if (priv->appqueue[i])
2613 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2614 if (priv->udpqueue[i])
2615 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2617 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2618 if (priv->funnel[i])
2619 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2620 if (priv->appsrc[i])
2621 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2623 if (priv->udpsrc_v4[i]) {
2624 if (priv->sinkpad || i == 1) {
2625 /* and set udpsrc to NULL now before removing */
2626 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2627 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2628 /* removing them should also nicely release the request
2629 * pads when they finalize */
2630 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2632 /* we need to set the state to NULL before unref */
2633 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2634 gst_object_unref (priv->udpsrc_v4[i]);
2638 if (priv->udpsrc_v6[i]) {
2639 if (priv->sinkpad || i == 1) {
2640 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2641 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2642 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2644 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2645 gst_object_unref (priv->udpsrc_v6[i]);
2649 for (l = priv->transport_sources; l; l = l->next) {
2650 GstRTSPMulticastTransportSource *s = l->data;
2655 gst_element_set_locked_state (s->udpsrc[i], FALSE);
2656 gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
2657 gst_bin_remove (bin, s->udpsrc[i]);
2660 if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
2661 gst_bin_remove (bin, priv->udpsink[i]);
2662 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2663 gst_bin_remove (bin, priv->appsrc[i]);
2664 if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
2665 gst_bin_remove (bin, priv->appsink[i]);
2666 if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2667 gst_bin_remove (bin, priv->appqueue[i]);
2668 if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2669 gst_bin_remove (bin, priv->udpqueue[i]);
2670 if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2671 gst_bin_remove (bin, priv->tee[i]);
2672 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2673 gst_bin_remove (bin, priv->funnel[i]);
2675 if (priv->sinkpad || i == 1) {
2676 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2677 gst_object_unref (priv->recv_sink[i]);
2678 priv->recv_sink[i] = NULL;
2681 priv->udpsrc_v4[i] = NULL;
2682 priv->udpsrc_v6[i] = NULL;
2683 priv->udpsink[i] = NULL;
2684 priv->appsrc[i] = NULL;
2685 priv->appsink[i] = NULL;
2686 priv->appqueue[i] = NULL;
2687 priv->udpqueue[i] = NULL;
2688 priv->tee[i] = NULL;
2689 priv->funnel[i] = NULL;
2692 for (l = priv->transport_sources; l; l = l->next) {
2693 GstRTSPMulticastTransportSource *s = l->data;
2694 g_slice_free (GstRTSPMulticastTransportSource, s);
2696 g_list_free (priv->transport_sources);
2697 priv->transport_sources = NULL;
2700 gst_object_unref (priv->send_src[0]);
2701 priv->send_src[0] = NULL;
2704 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2705 gst_object_unref (priv->send_src[1]);
2706 priv->send_src[1] = NULL;
2708 g_object_unref (priv->session);
2709 priv->session = NULL;
2711 gst_caps_unref (priv->caps);
2715 gst_object_unref (priv->srtpenc);
2717 gst_object_unref (priv->srtpdec);
2719 priv->is_joined = FALSE;
2720 g_mutex_unlock (&priv->lock);
2726 g_mutex_unlock (&priv->lock);
2729 transports_not_removed:
2731 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2732 g_mutex_unlock (&priv->lock);
2738 * gst_rtsp_stream_get_rtpinfo:
2739 * @stream: a #GstRTSPStream
2740 * @rtptime: (allow-none): result RTP timestamp
2741 * @seq: (allow-none): result RTP seqnum
2742 * @clock_rate: (allow-none): the clock rate
2743 * @running_time: (allow-none): result running-time
2745 * Retrieve the current rtptime, seq and running-time. This is used to
2746 * construct a RTPInfo reply header.
2748 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2751 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2752 guint * rtptime, guint * seq, guint * clock_rate,
2753 GstClockTime * running_time)
2755 GstRTSPStreamPrivate *priv;
2756 GstStructure *stats;
2757 GObjectClass *payobjclass;
2759 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2761 priv = stream->priv;
2763 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2765 g_mutex_lock (&priv->lock);
2767 /* First try to extract the information from the last buffer on the sinks.
2768 * This will have a more accurate sequence number and timestamp, as between
2769 * the payloader and the sink there can be some queues
2771 if (priv->udpsink[0] || priv->appsink[0]) {
2772 GstSample *last_sample;
2774 if (priv->udpsink[0])
2775 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
2777 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
2782 GstSegment *segment;
2783 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
2785 caps = gst_sample_get_caps (last_sample);
2786 buffer = gst_sample_get_buffer (last_sample);
2787 segment = gst_sample_get_segment (last_sample);
2789 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
2791 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
2795 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
2798 gst_rtp_buffer_unmap (&rtp_buffer);
2802 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
2803 GST_BUFFER_TIMESTAMP (buffer));
2807 GstStructure *s = gst_caps_get_structure (caps, 0);
2809 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
2811 if (*clock_rate == 0 && running_time)
2812 *running_time = GST_CLOCK_TIME_NONE;
2814 gst_sample_unref (last_sample);
2818 gst_sample_unref (last_sample);
2823 if (g_object_class_find_property (payobjclass, "stats")) {
2824 g_object_get (priv->payloader, "stats", &stats, NULL);
2829 gst_structure_get_uint (stats, "seqnum", seq);
2832 gst_structure_get_uint (stats, "timestamp", rtptime);
2835 gst_structure_get_clock_time (stats, "running-time", running_time);
2838 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2839 if (*clock_rate == 0 && running_time)
2840 *running_time = GST_CLOCK_TIME_NONE;
2842 gst_structure_free (stats);
2844 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2845 !g_object_class_find_property (payobjclass, "timestamp"))
2849 g_object_get (priv->payloader, "seqnum", seq, NULL);
2852 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2855 *running_time = GST_CLOCK_TIME_NONE;
2859 g_mutex_unlock (&priv->lock);
2866 GST_WARNING ("Could not get payloader stats");
2867 g_mutex_unlock (&priv->lock);
2873 * gst_rtsp_stream_get_caps:
2874 * @stream: a #GstRTSPStream
2876 * Retrieve the current caps of @stream.
2878 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2882 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2884 GstRTSPStreamPrivate *priv;
2887 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2889 priv = stream->priv;
2891 g_mutex_lock (&priv->lock);
2892 if ((result = priv->caps))
2893 gst_caps_ref (result);
2894 g_mutex_unlock (&priv->lock);
2900 * gst_rtsp_stream_recv_rtp:
2901 * @stream: a #GstRTSPStream
2902 * @buffer: (transfer full): a #GstBuffer
2904 * Handle an RTP buffer for the stream. This method is usually called when a
2905 * message has been received from a client using the TCP transport.
2907 * This function takes ownership of @buffer.
2909 * Returns: a GstFlowReturn.
2912 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2914 GstRTSPStreamPrivate *priv;
2916 GstElement *element;
2918 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2919 priv = stream->priv;
2920 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2921 g_return_val_if_fail (priv->is_joined, FALSE);
2923 g_mutex_lock (&priv->lock);
2924 if (priv->appsrc[0])
2925 element = gst_object_ref (priv->appsrc[0]);
2928 g_mutex_unlock (&priv->lock);
2931 if (priv->appsrc_base_time[0] == -1) {
2932 /* Take current running_time. This timestamp will be put on
2933 * the first buffer of each stream because we are a live source and so we
2934 * timestamp with the running_time. When we are dealing with TCP, we also
2935 * only timestamp the first buffer (using the DISCONT flag) because a server
2936 * typically bursts data, for which we don't want to compensate by speeding
2937 * up the media. The other timestamps will be interpollated from this one
2938 * using the RTP timestamps. */
2939 GST_OBJECT_LOCK (element);
2940 if (GST_ELEMENT_CLOCK (element)) {
2942 GstClockTime base_time;
2944 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
2945 base_time = GST_ELEMENT_CAST (element)->base_time;
2947 priv->appsrc_base_time[0] = now - base_time;
2948 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
2949 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
2950 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
2951 GST_TIME_ARGS (base_time));
2953 GST_OBJECT_UNLOCK (element);
2956 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2957 gst_object_unref (element);
2965 * gst_rtsp_stream_recv_rtcp:
2966 * @stream: a #GstRTSPStream
2967 * @buffer: (transfer full): a #GstBuffer
2969 * Handle an RTCP buffer for the stream. This method is usually called when a
2970 * message has been received from a client using the TCP transport.
2972 * This function takes ownership of @buffer.
2974 * Returns: a GstFlowReturn.
2977 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2979 GstRTSPStreamPrivate *priv;
2981 GstElement *element;
2983 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2984 priv = stream->priv;
2985 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2987 if (!priv->is_joined) {
2988 gst_buffer_unref (buffer);
2989 return GST_FLOW_NOT_LINKED;
2991 g_mutex_lock (&priv->lock);
2992 if (priv->appsrc[1])
2993 element = gst_object_ref (priv->appsrc[1]);
2996 g_mutex_unlock (&priv->lock);
2999 if (priv->appsrc_base_time[1] == -1) {
3000 /* Take current running_time. This timestamp will be put on
3001 * the first buffer of each stream because we are a live source and so we
3002 * timestamp with the running_time. When we are dealing with TCP, we also
3003 * only timestamp the first buffer (using the DISCONT flag) because a server
3004 * typically bursts data, for which we don't want to compensate by speeding
3005 * up the media. The other timestamps will be interpollated from this one
3006 * using the RTP timestamps. */
3007 GST_OBJECT_LOCK (element);
3008 if (GST_ELEMENT_CLOCK (element)) {
3010 GstClockTime base_time;
3012 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3013 base_time = GST_ELEMENT_CAST (element)->base_time;
3015 priv->appsrc_base_time[1] = now - base_time;
3016 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
3017 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3018 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3019 GST_TIME_ARGS (base_time));
3021 GST_OBJECT_UNLOCK (element);
3024 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3025 gst_object_unref (element);
3028 gst_buffer_unref (buffer);
3033 /* must be called with lock */
3035 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
3038 GstRTSPStreamPrivate *priv = stream->priv;
3039 const GstRTSPTransport *tr;
3041 tr = gst_rtsp_stream_transport_get_transport (trans);
3043 switch (tr->lower_transport) {
3044 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3046 GstRTSPMulticastTransportSource *source;
3049 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[1])));
3054 GstPad *selpad, *pad;
3056 source = g_slice_new0 (GstRTSPMulticastTransportSource);
3057 source->transport = trans;
3059 for (i = 0; i < 2; i++) {
3061 g_strdup_printf ("udp://%s:%d", tr->destination,
3062 (i == 0) ? tr->port.min : tr->port.max);
3064 gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
3066 g_object_set (source->udpsrc[i], "loop", FALSE, NULL);
3069 /* we set and keep these to playing so that they don't cause NO_PREROLL return
3070 * values. This is only relevant for PLAY pipelines */
3071 gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
3072 gst_element_set_locked_state (source->udpsrc[i], TRUE);
3075 gst_bin_add (bin, source->udpsrc[i]);
3077 /* and link to the funnel v4 */
3078 if (priv->sinkpad || i == 1) {
3079 source->selpad[i] = selpad =
3080 gst_element_get_request_pad (priv->funnel[i], "sink_%u");
3081 pad = gst_element_get_static_pad (source->udpsrc[i], "src");
3082 gst_pad_link (pad, selpad);
3083 gst_object_unref (pad);
3084 gst_object_unref (selpad);
3088 priv->transport_sources =
3089 g_list_prepend (priv->transport_sources, source);
3093 for (l = priv->transport_sources; l; l = l->next) {
3096 if (source->transport == trans) {
3097 priv->transport_sources =
3098 g_list_delete_link (priv->transport_sources, l);
3106 for (i = 0; i < 2; i++) {
3107 /* Will automatically unlink everything */
3108 gst_bin_remove (bin,
3109 GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
3111 gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
3112 gst_object_unref (source->udpsrc[i]);
3114 if (priv->sinkpad || i == 1) {
3115 gst_element_release_request_pad (priv->funnel[i],
3120 g_slice_free (GstRTSPMulticastTransportSource, source);
3124 gst_object_unref (bin);
3126 /* fall through for the generic case */
3128 case GST_RTSP_LOWER_TRANS_UDP:
3134 dest = tr->destination;
3135 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3139 } else if (priv->client_side) {
3140 /* In client side mode the 'destination' is the RTSP server, so send
3142 min = tr->server_port.min;
3143 max = tr->server_port.max;
3145 min = tr->client_port.min;
3146 max = tr->client_port.max;
3151 GST_INFO ("setting ttl-mc %d", ttl);
3152 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
3153 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
3155 GST_INFO ("adding %s:%d-%d", dest, min, max);
3156 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
3157 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
3158 priv->transports = g_list_prepend (priv->transports, trans);
3160 GST_INFO ("removing %s:%d-%d", dest, min, max);
3161 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
3162 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
3163 priv->transports = g_list_remove (priv->transports, trans);
3165 priv->transports_cookie++;
3168 case GST_RTSP_LOWER_TRANS_TCP:
3170 GST_INFO ("adding TCP %s", tr->destination);
3171 priv->transports = g_list_prepend (priv->transports, trans);
3173 GST_INFO ("removing TCP %s", tr->destination);
3174 priv->transports = g_list_remove (priv->transports, trans);
3176 priv->transports_cookie++;
3179 goto unknown_transport;
3186 GST_INFO ("Unknown transport %d", tr->lower_transport);
3193 * gst_rtsp_stream_add_transport:
3194 * @stream: a #GstRTSPStream
3195 * @trans: (transfer none): a #GstRTSPStreamTransport
3197 * Add the transport in @trans to @stream. The media of @stream will
3198 * then also be send to the values configured in @trans.
3200 * @stream must be joined to a bin.
3202 * @trans must contain a valid #GstRTSPTransport.
3204 * Returns: %TRUE if @trans was added
3207 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
3208 GstRTSPStreamTransport * trans)
3210 GstRTSPStreamPrivate *priv;
3213 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3214 priv = stream->priv;
3215 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3216 g_return_val_if_fail (priv->is_joined, FALSE);
3218 g_mutex_lock (&priv->lock);
3219 res = update_transport (stream, trans, TRUE);
3220 g_mutex_unlock (&priv->lock);
3226 * gst_rtsp_stream_remove_transport:
3227 * @stream: a #GstRTSPStream
3228 * @trans: (transfer none): a #GstRTSPStreamTransport
3230 * Remove the transport in @trans from @stream. The media of @stream will
3231 * not be sent to the values configured in @trans.
3233 * @stream must be joined to a bin.
3235 * @trans must contain a valid #GstRTSPTransport.
3237 * Returns: %TRUE if @trans was removed
3240 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
3241 GstRTSPStreamTransport * trans)
3243 GstRTSPStreamPrivate *priv;
3246 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3247 priv = stream->priv;
3248 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3249 g_return_val_if_fail (priv->is_joined, FALSE);
3251 g_mutex_lock (&priv->lock);
3252 res = update_transport (stream, trans, FALSE);
3253 g_mutex_unlock (&priv->lock);
3259 * gst_rtsp_stream_update_crypto:
3260 * @stream: a #GstRTSPStream
3262 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3264 * Update the new crypto information for @ssrc in @stream. If information
3265 * for @ssrc did not exist, it will be added. If information
3266 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3267 * be removed from @stream.
3269 * Returns: %TRUE if @crypto could be updated
3272 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3273 guint ssrc, GstCaps * crypto)
3275 GstRTSPStreamPrivate *priv;
3277 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3278 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3280 priv = stream->priv;
3282 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3284 g_mutex_lock (&priv->lock);
3286 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3287 gst_caps_ref (crypto));
3289 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3290 g_mutex_unlock (&priv->lock);
3296 * gst_rtsp_stream_get_rtp_socket:
3297 * @stream: a #GstRTSPStream
3298 * @family: the socket family
3300 * Get the RTP socket from @stream for a @family.
3302 * @stream must be joined to a bin.
3304 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3305 * socket could be allocated for @family. Unref after usage
3308 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3310 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3314 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3315 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3316 family == G_SOCKET_FAMILY_IPV6, NULL);
3317 g_return_val_if_fail (priv->udpsink[0], NULL);
3319 if (family == G_SOCKET_FAMILY_IPV6)
3324 g_object_get (priv->udpsink[0], name, &socket, NULL);
3330 * gst_rtsp_stream_get_rtcp_socket:
3331 * @stream: a #GstRTSPStream
3332 * @family: the socket family
3334 * Get the RTCP socket from @stream for a @family.
3336 * @stream must be joined to a bin.
3338 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3339 * socket could be allocated for @family. Unref after usage
3342 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3344 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3348 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3349 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3350 family == G_SOCKET_FAMILY_IPV6, NULL);
3351 g_return_val_if_fail (priv->udpsink[1], NULL);
3353 if (family == G_SOCKET_FAMILY_IPV6)
3358 g_object_get (priv->udpsink[1], name, &socket, NULL);
3364 * gst_rtsp_stream_set_seqnum:
3365 * @stream: a #GstRTSPStream
3366 * @seqnum: a new sequence number
3368 * Configure the sequence number in the payloader of @stream to @seqnum.
3371 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3373 GstRTSPStreamPrivate *priv;
3375 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3377 priv = stream->priv;
3379 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3383 * gst_rtsp_stream_get_seqnum:
3384 * @stream: a #GstRTSPStream
3386 * Get the configured sequence number in the payloader of @stream.
3388 * Returns: the sequence number of the payloader.
3391 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3393 GstRTSPStreamPrivate *priv;
3396 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3398 priv = stream->priv;
3400 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3406 * gst_rtsp_stream_transport_filter:
3407 * @stream: a #GstRTSPStream
3408 * @func: (scope call) (allow-none): a callback
3409 * @user_data: (closure): user data passed to @func
3411 * Call @func for each transport managed by @stream. The result value of @func
3412 * determines what happens to the transport. @func will be called with @stream
3413 * locked so no further actions on @stream can be performed from @func.
3415 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3418 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3420 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3421 * will also be added with an additional ref to the result #GList of this
3424 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3426 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3427 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3428 * element in the #GList should be unreffed before the list is freed.
3431 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3432 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3434 GstRTSPStreamPrivate *priv;
3435 GList *result, *walk, *next;
3436 GHashTable *visited = NULL;
3439 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3441 priv = stream->priv;
3445 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3447 g_mutex_lock (&priv->lock);
3449 cookie = priv->transports_cookie;
3450 for (walk = priv->transports; walk; walk = next) {
3451 GstRTSPStreamTransport *trans = walk->data;
3452 GstRTSPFilterResult res;
3455 next = g_list_next (walk);
3458 /* only visit each transport once */
3459 if (g_hash_table_contains (visited, trans))
3462 g_hash_table_add (visited, g_object_ref (trans));
3463 g_mutex_unlock (&priv->lock);
3465 res = func (stream, trans, user_data);
3467 g_mutex_lock (&priv->lock);
3469 res = GST_RTSP_FILTER_REF;
3471 changed = (cookie != priv->transports_cookie);
3474 case GST_RTSP_FILTER_REMOVE:
3475 update_transport (stream, trans, FALSE);
3477 case GST_RTSP_FILTER_REF:
3478 result = g_list_prepend (result, g_object_ref (trans));
3480 case GST_RTSP_FILTER_KEEP:
3487 g_mutex_unlock (&priv->lock);
3490 g_hash_table_unref (visited);
3495 static GstPadProbeReturn
3496 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3498 GstRTSPStreamPrivate *priv;
3499 GstRTSPStream *stream;
3502 priv = stream->priv;
3504 GST_DEBUG_OBJECT (pad, "now blocking");
3506 g_mutex_lock (&priv->lock);
3507 priv->blocking = TRUE;
3508 g_mutex_unlock (&priv->lock);
3510 gst_element_post_message (priv->payloader,
3511 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3512 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3514 return GST_PAD_PROBE_OK;
3518 * gst_rtsp_stream_set_blocked:
3519 * @stream: a #GstRTSPStream
3520 * @blocked: boolean indicating we should block or unblock
3522 * Blocks or unblocks the dataflow on @stream.
3524 * Returns: %TRUE on success
3527 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3529 GstRTSPStreamPrivate *priv;
3531 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3533 priv = stream->priv;
3535 g_mutex_lock (&priv->lock);
3537 priv->blocking = FALSE;
3538 if (priv->blocked_id == 0) {
3539 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3540 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3541 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3542 g_object_ref (stream), g_object_unref);
3545 if (priv->blocked_id != 0) {
3546 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3547 priv->blocked_id = 0;
3548 priv->blocking = FALSE;
3551 g_mutex_unlock (&priv->lock);
3557 * gst_rtsp_stream_is_blocking:
3558 * @stream: a #GstRTSPStream
3560 * Check if @stream is blocking on a #GstBuffer.
3562 * Returns: %TRUE if @stream is blocking
3565 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3567 GstRTSPStreamPrivate *priv;
3570 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3572 priv = stream->priv;
3574 g_mutex_lock (&priv->lock);
3575 result = priv->blocking;
3576 g_mutex_unlock (&priv->lock);
3582 * gst_rtsp_stream_query_position:
3583 * @stream: a #GstRTSPStream
3585 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3586 * the RTP parts of the pipeline and not the RTCP parts.
3588 * Returns: %TRUE if the position could be queried
3591 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3593 GstRTSPStreamPrivate *priv;
3597 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3599 priv = stream->priv;
3601 g_mutex_lock (&priv->lock);
3602 /* depending on the transport type, it should query corresponding sink */
3603 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3604 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3605 sink = priv->udpsink[0];
3607 sink = priv->appsink[0];
3610 gst_object_ref (sink);
3611 g_mutex_unlock (&priv->lock);
3616 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3617 gst_object_unref (sink);
3623 * gst_rtsp_stream_query_stop:
3624 * @stream: a #GstRTSPStream
3626 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3627 * the RTP parts of the pipeline and not the RTCP parts.
3629 * Returns: %TRUE if the stop could be queried
3632 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3634 GstRTSPStreamPrivate *priv;
3639 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3641 priv = stream->priv;
3643 g_mutex_lock (&priv->lock);
3644 /* depending on the transport type, it should query corresponding sink */
3645 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3646 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3647 sink = priv->udpsink[0];
3649 sink = priv->appsink[0];
3652 gst_object_ref (sink);
3653 g_mutex_unlock (&priv->lock);
3658 query = gst_query_new_segment (GST_FORMAT_TIME);
3659 if ((ret = gst_element_query (sink, query))) {
3662 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3663 if (format != GST_FORMAT_TIME)
3666 gst_query_unref (query);
3667 gst_object_unref (sink);