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;
1108 alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
1109 GstElement * udpsrc_out[2], GstRTSPRange * server_port_out,
1110 GstRTSPAddress ** server_addr_out)
1112 GstRTSPStreamPrivate *priv = stream->priv;
1113 GstStateChangeReturn ret;
1114 GstElement *udpsrc0, *udpsrc1;
1115 GSocket *rtp_socket = NULL;
1116 GSocket *rtcp_socket;
1117 gint tmp_rtp, tmp_rtcp;
1119 gint rtpport, rtcpport;
1120 GList *rejected_addresses = NULL;
1121 GstRTSPAddress *addr = NULL;
1122 GInetAddress *inetaddr = NULL;
1123 GSocketAddress *rtp_sockaddr = NULL;
1124 GSocketAddress *rtcp_sockaddr = NULL;
1125 GstRTSPAddressPool * pool;
1132 /* Start with random port */
1135 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1136 G_SOCKET_PROTOCOL_UDP, NULL);
1138 goto no_udp_protocol;
1140 if (*server_addr_out)
1141 gst_rtsp_address_free (*server_addr_out);
1143 /* try to allocate 2 UDP ports, the RTP port should be an even
1144 * number and the RTCP port should be the next (uneven) port */
1147 if (rtp_socket == NULL) {
1148 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1149 G_SOCKET_PROTOCOL_UDP, NULL);
1151 goto no_udp_protocol;
1154 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
1155 GstRTSPAddressFlags flags;
1158 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1160 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1161 if (family == G_SOCKET_FAMILY_IPV6)
1162 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1164 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1166 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1171 tmp_rtp = addr->port;
1173 g_clear_object (&inetaddr);
1174 inetaddr = g_inet_address_new_from_string (addr->address);
1182 if (inetaddr == NULL)
1183 inetaddr = g_inet_address_new_any (family);
1186 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1187 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1188 g_object_unref (rtp_sockaddr);
1191 g_object_unref (rtp_sockaddr);
1193 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1194 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1195 g_clear_object (&rtp_sockaddr);
1200 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1201 g_object_unref (rtp_sockaddr);
1203 /* check if port is even */
1204 if ((tmp_rtp & 1) != 0) {
1205 /* port not even, close and allocate another */
1207 g_clear_object (&rtp_socket);
1212 tmp_rtcp = tmp_rtp + 1;
1214 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1215 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1216 g_object_unref (rtcp_sockaddr);
1217 g_clear_object (&rtp_socket);
1220 g_object_unref (rtcp_sockaddr);
1222 g_clear_object (&inetaddr);
1224 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
1225 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
1227 if (udpsrc0 == NULL || udpsrc1 == NULL)
1228 goto no_udp_protocol;
1230 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
1231 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
1233 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1234 if (ret == GST_STATE_CHANGE_FAILURE)
1236 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1237 if (ret == GST_STATE_CHANGE_FAILURE)
1240 /* all fine, do port check */
1241 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
1242 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
1244 /* this should not happen... */
1245 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1248 if (!create_and_configure_udpsinks (stream))
1249 goto no_udp_protocol;
1251 /* set RTP and RTCP sockets */
1252 set_sockets_for_udpsinks (stream, rtp_socket, rtcp_socket, family);
1254 /* we keep these elements, we will further configure them when the
1255 * client told us to really use the UDP ports. */
1256 udpsrc_out[0] = udpsrc0;
1257 udpsrc_out[1] = udpsrc1;
1259 server_port_out->min = rtpport;
1260 server_port_out->max = rtcpport;
1262 *server_addr_out = addr;
1263 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1265 g_object_unref (rtp_socket);
1266 g_object_unref (rtcp_socket);
1294 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1295 gst_object_unref (udpsrc0);
1298 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1299 gst_object_unref (udpsrc1);
1302 g_object_unref (inetaddr);
1303 g_list_free_full (rejected_addresses,
1304 (GDestroyNotify) gst_rtsp_address_free);
1306 gst_rtsp_address_free (addr);
1308 g_object_unref (rtp_socket);
1310 g_object_unref (rtcp_socket);
1315 /* must be called with lock */
1317 alloc_ports (GstRTSPStream * stream)
1319 GstRTSPStreamPrivate *priv = stream->priv;
1322 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4,
1323 &priv->server_port_v4, &priv->server_addr_v4);
1326 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6,
1327 &priv->server_port_v6, &priv->server_addr_v6);
1329 return priv->have_ipv4 || priv->have_ipv6;
1333 * gst_rtsp_stream_set_client_side:
1334 * @stream: a #GstRTSPStream
1335 * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
1336 * an RTSP connection.
1338 * Sets the #GstRTSPStream as a 'client side' stream - used for sending
1339 * streams to an RTSP server via RECORD. This has the practical effect
1340 * of changing which UDP port numbers are used when setting up the local
1341 * side of the stream sending to be either the 'server' or 'client' pair
1342 * of a configured UDP transport.
1345 gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
1347 GstRTSPStreamPrivate *priv;
1349 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1350 priv = stream->priv;
1351 g_mutex_lock (&priv->lock);
1352 priv->client_side = client_side;
1353 g_mutex_unlock (&priv->lock);
1357 * gst_rtsp_stream_set_client_side:
1358 * @stream: a #GstRTSPStream
1360 * See gst_rtsp_stream_set_client_side()
1362 * Returns: TRUE if this #GstRTSPStream is client-side.
1365 gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
1367 GstRTSPStreamPrivate *priv;
1370 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1372 priv = stream->priv;
1373 g_mutex_lock (&priv->lock);
1374 ret = priv->client_side;
1375 g_mutex_unlock (&priv->lock);
1381 * gst_rtsp_stream_get_server_port:
1382 * @stream: a #GstRTSPStream
1383 * @server_port: (out): result server port
1384 * @family: the port family to get
1386 * Fill @server_port with the port pair used by the server. This function can
1387 * only be called when @stream has been joined.
1390 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1391 GstRTSPRange * server_port, GSocketFamily family)
1393 GstRTSPStreamPrivate *priv;
1395 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1396 priv = stream->priv;
1397 g_return_if_fail (priv->is_joined);
1399 g_mutex_lock (&priv->lock);
1400 if (family == G_SOCKET_FAMILY_IPV4) {
1402 *server_port = priv->server_port_v4;
1405 *server_port = priv->server_port_v6;
1407 g_mutex_unlock (&priv->lock);
1411 * gst_rtsp_stream_get_rtpsession:
1412 * @stream: a #GstRTSPStream
1414 * Get the RTP session of this stream.
1416 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1419 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1421 GstRTSPStreamPrivate *priv;
1424 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1426 priv = stream->priv;
1428 g_mutex_lock (&priv->lock);
1429 if ((session = priv->session))
1430 g_object_ref (session);
1431 g_mutex_unlock (&priv->lock);
1437 * gst_rtsp_stream_get_ssrc:
1438 * @stream: a #GstRTSPStream
1439 * @ssrc: (out): result ssrc
1441 * Get the SSRC used by the RTP session of this stream. This function can only
1442 * be called when @stream has been joined.
1445 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1447 GstRTSPStreamPrivate *priv;
1449 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1450 priv = stream->priv;
1451 g_return_if_fail (priv->is_joined);
1453 g_mutex_lock (&priv->lock);
1454 if (ssrc && priv->session)
1455 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1456 g_mutex_unlock (&priv->lock);
1460 * gst_rtsp_stream_set_retransmission_time:
1461 * @stream: a #GstRTSPStream
1462 * @time: a #GstClockTime
1464 * Set the amount of time to store retransmission packets.
1467 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1470 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1472 g_mutex_lock (&stream->priv->lock);
1473 stream->priv->rtx_time = time;
1474 if (stream->priv->rtxsend)
1475 g_object_set (stream->priv->rtxsend, "max-size-time",
1476 GST_TIME_AS_MSECONDS (time), NULL);
1477 g_mutex_unlock (&stream->priv->lock);
1481 * gst_rtsp_stream_get_retransmission_time:
1482 * @stream: a #GstRTSPStream
1484 * Get the amount of time to store retransmission data.
1486 * Returns: the amount of time to store retransmission data.
1489 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1493 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1495 g_mutex_lock (&stream->priv->lock);
1496 ret = stream->priv->rtx_time;
1497 g_mutex_unlock (&stream->priv->lock);
1503 * gst_rtsp_stream_set_retransmission_pt:
1504 * @stream: a #GstRTSPStream
1507 * Set the payload type (pt) for retransmission of this stream.
1510 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1512 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1514 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1516 g_mutex_lock (&stream->priv->lock);
1517 stream->priv->rtx_pt = rtx_pt;
1518 if (stream->priv->rtxsend) {
1519 guint pt = gst_rtsp_stream_get_pt (stream);
1520 gchar *pt_s = g_strdup_printf ("%d", pt);
1521 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1522 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1523 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1525 gst_structure_free (rtx_pt_map);
1527 g_mutex_unlock (&stream->priv->lock);
1531 * gst_rtsp_stream_get_retransmission_pt:
1532 * @stream: a #GstRTSPStream
1534 * Get the payload-type used for retransmission of this stream
1536 * Returns: The retransmission PT.
1539 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1543 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1545 g_mutex_lock (&stream->priv->lock);
1546 rtx_pt = stream->priv->rtx_pt;
1547 g_mutex_unlock (&stream->priv->lock);
1553 * gst_rtsp_stream_set_buffer_size:
1554 * @stream: a #GstRTSPStream
1555 * @size: the buffer size
1557 * Set the size of the UDP transmission buffer (in bytes)
1558 * Needs to be set before the stream is joined to a bin.
1563 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1565 g_mutex_lock (&stream->priv->lock);
1566 stream->priv->buffer_size = size;
1567 g_mutex_unlock (&stream->priv->lock);
1571 * gst_rtsp_stream_get_buffer_size:
1572 * @stream: a #GstRTSPStream
1574 * Get the size of the UDP transmission buffer (in bytes)
1576 * Returns: the size of the UDP TX buffer
1581 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1585 g_mutex_lock (&stream->priv->lock);
1586 buffer_size = stream->priv->buffer_size;
1587 g_mutex_unlock (&stream->priv->lock);
1592 /* executed from streaming thread */
1594 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1596 GstRTSPStreamPrivate *priv = stream->priv;
1597 GstCaps *newcaps, *oldcaps;
1599 newcaps = gst_pad_get_current_caps (pad);
1601 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1604 g_mutex_lock (&priv->lock);
1605 oldcaps = priv->caps;
1606 priv->caps = newcaps;
1607 g_mutex_unlock (&priv->lock);
1610 gst_caps_unref (oldcaps);
1614 dump_structure (const GstStructure * s)
1618 sstr = gst_structure_to_string (s);
1619 GST_INFO ("structure: %s", sstr);
1623 static GstRTSPStreamTransport *
1624 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1626 GstRTSPStreamPrivate *priv = stream->priv;
1628 GstRTSPStreamTransport *result = NULL;
1633 if (rtcp_from == NULL)
1636 tmp = g_strrstr (rtcp_from, ":");
1640 port = atoi (tmp + 1);
1641 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1643 g_mutex_lock (&priv->lock);
1644 GST_INFO ("finding %s:%d in %d transports", dest, port,
1645 g_list_length (priv->transports));
1647 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1648 GstRTSPStreamTransport *trans = walk->data;
1649 const GstRTSPTransport *tr;
1652 tr = gst_rtsp_stream_transport_get_transport (trans);
1654 if (priv->client_side) {
1655 /* In client side mode the 'destination' is the RTSP server, so send
1657 min = tr->server_port.min;
1658 max = tr->server_port.max;
1660 min = tr->client_port.min;
1661 max = tr->client_port.max;
1664 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1670 g_object_ref (result);
1671 g_mutex_unlock (&priv->lock);
1678 static GstRTSPStreamTransport *
1679 check_transport (GObject * source, GstRTSPStream * stream)
1681 GstStructure *stats;
1682 GstRTSPStreamTransport *trans;
1684 /* see if we have a stream to match with the origin of the RTCP packet */
1685 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1686 if (trans == NULL) {
1687 g_object_get (source, "stats", &stats, NULL);
1689 const gchar *rtcp_from;
1691 dump_structure (stats);
1693 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1694 if ((trans = find_transport (stream, rtcp_from))) {
1695 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1697 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1700 gst_structure_free (stats);
1708 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1710 GstRTSPStreamTransport *trans;
1712 GST_INFO ("%p: new source %p", stream, source);
1714 trans = check_transport (source, stream);
1717 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1721 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1723 GST_INFO ("%p: new SDES %p", stream, source);
1727 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1729 GstRTSPStreamTransport *trans;
1731 trans = check_transport (source, stream);
1734 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1735 gst_rtsp_stream_transport_keep_alive (trans);
1739 GstStructure *stats;
1740 g_object_get (source, "stats", &stats, NULL);
1742 dump_structure (stats);
1743 gst_structure_free (stats);
1750 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1752 GST_INFO ("%p: source %p bye", stream, source);
1756 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1758 GstRTSPStreamTransport *trans;
1760 GST_INFO ("%p: source %p bye timeout", stream, source);
1762 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1763 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1764 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1769 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1771 GstRTSPStreamTransport *trans;
1773 GST_INFO ("%p: source %p timeout", stream, source);
1775 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1776 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1777 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1782 on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1784 GST_INFO ("%p: new sender source %p", stream, source);
1787 GstStructure *stats;
1788 g_object_get (source, "stats", &stats, NULL);
1790 dump_structure (stats);
1791 gst_structure_free (stats);
1798 on_sender_ssrc_active (GObject * session, GObject * source,
1799 GstRTSPStream * stream)
1803 GstStructure *stats;
1804 g_object_get (source, "stats", &stats, NULL);
1806 dump_structure (stats);
1807 gst_structure_free (stats);
1814 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1817 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1818 g_list_free (priv->tr_cache_rtp);
1819 priv->tr_cache_rtp = NULL;
1821 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1822 g_list_free (priv->tr_cache_rtcp);
1823 priv->tr_cache_rtcp = NULL;
1827 static GstFlowReturn
1828 handle_new_sample (GstAppSink * sink, gpointer user_data)
1830 GstRTSPStreamPrivate *priv;
1834 GstRTSPStream *stream;
1837 sample = gst_app_sink_pull_sample (sink);
1841 stream = (GstRTSPStream *) user_data;
1842 priv = stream->priv;
1843 buffer = gst_sample_get_buffer (sample);
1845 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1847 g_mutex_lock (&priv->lock);
1849 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1850 clear_tr_cache (priv, is_rtp);
1851 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1852 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1853 priv->tr_cache_rtp =
1854 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1856 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1859 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1860 clear_tr_cache (priv, is_rtp);
1861 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1862 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1863 priv->tr_cache_rtcp =
1864 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
1866 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
1869 g_mutex_unlock (&priv->lock);
1872 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
1873 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1874 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1877 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
1878 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1879 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1882 gst_sample_unref (sample);
1887 static GstAppSinkCallbacks sink_cb = {
1888 NULL, /* not interested in EOS */
1889 NULL, /* not interested in preroll samples */
1894 get_rtp_encoder (GstRTSPStream * stream, guint session)
1896 GstRTSPStreamPrivate *priv = stream->priv;
1898 if (priv->srtpenc == NULL) {
1901 name = g_strdup_printf ("srtpenc_%u", session);
1902 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
1905 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
1907 return gst_object_ref (priv->srtpenc);
1911 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
1913 GstRTSPStreamPrivate *priv = stream->priv;
1914 GstElement *oldenc, *enc;
1918 if (priv->idx != session)
1921 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
1923 oldenc = priv->srtpenc;
1924 enc = get_rtp_encoder (stream, session);
1925 name = g_strdup_printf ("rtp_sink_%d", session);
1926 pad = gst_element_get_request_pad (enc, name);
1928 gst_object_unref (pad);
1931 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
1938 request_rtcp_encoder (GstElement * rtpbin, guint session,
1939 GstRTSPStream * stream)
1941 GstRTSPStreamPrivate *priv = stream->priv;
1942 GstElement *oldenc, *enc;
1946 if (priv->idx != session)
1949 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
1951 oldenc = priv->srtpenc;
1952 enc = get_rtp_encoder (stream, session);
1953 name = g_strdup_printf ("rtcp_sink_%d", session);
1954 pad = gst_element_get_request_pad (enc, name);
1956 gst_object_unref (pad);
1959 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
1966 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
1968 GstRTSPStreamPrivate *priv = stream->priv;
1971 GST_DEBUG ("request key %08x", ssrc);
1973 g_mutex_lock (&priv->lock);
1974 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
1975 gst_caps_ref (caps);
1976 g_mutex_unlock (&priv->lock);
1982 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
1983 GstRTSPStream * stream)
1985 GstRTSPStreamPrivate *priv = stream->priv;
1987 if (priv->idx != session)
1990 if (priv->srtpdec == NULL) {
1993 name = g_strdup_printf ("srtpdec_%u", session);
1994 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
1997 g_signal_connect (priv->srtpdec, "request-key",
1998 (GCallback) request_key, stream);
2000 return gst_object_ref (priv->srtpdec);
2004 * gst_rtsp_stream_request_aux_sender:
2005 * @stream: a #GstRTSPStream
2006 * @sessid: the session id
2008 * Creating a rtxsend bin
2010 * Returns: (transfer full): a #GstElement.
2015 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
2019 GstStructure *pt_map;
2024 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2026 pt = gst_rtsp_stream_get_pt (stream);
2027 pt_s = g_strdup_printf ("%u", pt);
2028 rtx_pt = stream->priv->rtx_pt;
2030 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
2032 bin = gst_bin_new (NULL);
2033 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
2034 pt_map = gst_structure_new ("application/x-rtp-pt-map",
2035 pt_s, G_TYPE_UINT, rtx_pt, NULL);
2036 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
2037 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
2039 gst_structure_free (pt_map);
2040 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
2042 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
2043 name = g_strdup_printf ("src_%u", sessid);
2044 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2046 gst_object_unref (pad);
2048 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
2049 name = g_strdup_printf ("sink_%u", sessid);
2050 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2052 gst_object_unref (pad);
2058 * gst_rtsp_stream_set_pt_map:
2059 * @stream: a #GstRTSPStream
2063 * Configure a pt map between @pt and @caps.
2066 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
2068 GstRTSPStreamPrivate *priv = stream->priv;
2070 g_mutex_lock (&priv->lock);
2071 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
2072 g_mutex_unlock (&priv->lock);
2076 request_pt_map (GstElement * rtpbin, guint session, guint pt,
2077 GstRTSPStream * stream)
2079 GstRTSPStreamPrivate *priv = stream->priv;
2080 GstCaps *caps = NULL;
2082 g_mutex_lock (&priv->lock);
2084 if (priv->idx == session) {
2085 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
2087 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
2088 gst_caps_ref (caps);
2090 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
2094 g_mutex_unlock (&priv->lock);
2100 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
2102 GstRTSPStreamPrivate *priv = stream->priv;
2104 GstPadLinkReturn ret;
2107 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
2108 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2110 name = gst_pad_get_name (pad);
2111 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
2117 if (priv->idx != sessid)
2120 if (gst_pad_is_linked (priv->sinkpad)) {
2121 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
2122 GST_DEBUG_PAD_NAME (priv->sinkpad));
2126 /* link the RTP pad to the session manager, it should not really fail unless
2127 * this is not really an RTP pad */
2128 ret = gst_pad_link (pad, priv->sinkpad);
2129 if (ret != GST_PAD_LINK_OK)
2131 priv->recv_rtp_src = gst_object_ref (pad);
2138 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2139 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2144 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2145 GstRTSPStream * stream)
2147 /* TODO: What to do here other than this? */
2148 GST_DEBUG ("Stream %p: Got EOS", stream);
2149 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2152 /* must be called with lock */
2154 create_sender_part (GstRTSPStream * stream, GstBin * bin,
2157 GstRTSPStreamPrivate *priv;
2158 GstPad *pad, *sinkpad = NULL;
2159 gboolean is_tcp = FALSE, is_udp = FALSE;
2162 priv = stream->priv;
2164 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2165 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2166 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2168 for (i = 0; i < 2; i++) {
2169 GstPad *teepad, *queuepad;
2170 /* For the sender we create this bit of pipeline for both
2171 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2172 * we need to add a queue before appsink and udpsink to make
2173 * the pipeline not block. For the TCP case, we want to pump
2174 * client as fast as possible anyway. This pipeline is used
2175 * when both TCP and UDP are present.
2177 * .--------. .-----. .---------. .---------.
2178 * | rtpbin | | tee | | queue | | udpsink |
2179 * | send->sink src->sink src->sink |
2180 * '--------' | | '---------' '---------'
2181 * | | .---------. .---------.
2182 * | | | queue | | appsink |
2183 * | src->sink src->sink |
2184 * '-----' '---------' '---------'
2186 * When only UDP or only TCP is allowed, we skip the tee and queue
2187 * and link the udpsink (for UDP) or appsink (for TCP) directly to
2190 /* Only link the RTP send src if we're going to send RTP, link
2191 * the RTCP send src always */
2192 if (priv->srcpad || i == 1) {
2195 gst_bin_add (bin, priv->udpsink[i]);
2196 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2201 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2202 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2203 gst_bin_add (bin, priv->appsink[i]);
2204 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2205 &sink_cb, stream, NULL);
2208 if (is_udp && is_tcp) {
2209 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2211 /* make tee for RTP/RTCP */
2212 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2213 gst_bin_add (bin, priv->tee[i]);
2215 /* and link to rtpbin send pad */
2216 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2217 gst_pad_link (priv->send_src[i], pad);
2218 gst_object_unref (pad);
2220 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2221 g_object_set (priv->udpqueue[i], "max-size-buffers",
2222 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2224 gst_bin_add (bin, priv->udpqueue[i]);
2225 /* link tee to udpqueue */
2226 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2227 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2228 gst_pad_link (teepad, pad);
2229 gst_object_unref (pad);
2230 gst_object_unref (teepad);
2232 /* link udpqueue to udpsink */
2233 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2234 gst_pad_link (queuepad, sinkpad);
2235 gst_object_unref (queuepad);
2236 gst_object_unref (sinkpad);
2239 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2240 g_object_set (priv->appqueue[i], "max-size-buffers",
2241 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2243 gst_bin_add (bin, priv->appqueue[i]);
2244 /* and link tee to appqueue */
2245 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2246 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2247 gst_pad_link (teepad, pad);
2248 gst_object_unref (pad);
2249 gst_object_unref (teepad);
2251 /* and link appqueue to appsink */
2252 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2253 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2254 gst_pad_link (queuepad, pad);
2255 gst_object_unref (pad);
2256 gst_object_unref (queuepad);
2257 } else if (is_tcp) {
2258 /* only appsink needed, link it to the session */
2259 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2260 gst_pad_link (priv->send_src[i], pad);
2261 gst_object_unref (pad);
2263 /* when its only TCP, we need to set sync and preroll to FALSE
2264 * for the sink to avoid deadlock. And this is only needed for
2265 * sink used for RTCP data, not the RTP data. */
2267 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2269 /* else only udpsink needed, link it to the session */
2270 gst_pad_link (priv->send_src[i], sinkpad);
2271 gst_object_unref (sinkpad);
2275 /* check if we need to set to a special state */
2276 if (state != GST_STATE_NULL) {
2277 if (priv->udpsink[i] && (priv->srcpad || i == 1))
2278 gst_element_set_state (priv->udpsink[i], state);
2279 if (priv->appsink[i] && (priv->srcpad || i == 1))
2280 gst_element_set_state (priv->appsink[i], state);
2281 if (priv->appqueue[i] && (priv->srcpad || i == 1))
2282 gst_element_set_state (priv->appqueue[i], state);
2283 if (priv->udpqueue[i] && (priv->srcpad || i == 1))
2284 gst_element_set_state (priv->udpqueue[i], state);
2285 if (priv->tee[i] && (priv->srcpad || i == 1))
2286 gst_element_set_state (priv->tee[i], state);
2291 /* must be called with lock */
2293 create_receiver_part (GstRTSPStream * stream, GstBin * bin,
2296 GstRTSPStreamPrivate *priv;
2297 GstPad *pad, *selpad;
2301 priv = stream->priv;
2303 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2305 for (i = 0; i < 2; i++) {
2306 /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
2307 * RTCP sink always */
2308 if (priv->sinkpad || i == 1) {
2309 /* For the receiver we create this bit of pipeline for both
2310 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2311 * and it is all funneled into the rtpbin receive pad.
2313 * .--------. .--------. .--------.
2314 * | udpsrc | | funnel | | rtpbin |
2315 * | src->sink src->sink |
2316 * '--------' | | '--------'
2320 * '--------' '--------'
2322 /* make funnel for the RTP/RTCP receivers */
2323 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2324 gst_bin_add (bin, priv->funnel[i]);
2326 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2327 gst_pad_link (pad, priv->recv_sink[i]);
2328 gst_object_unref (pad);
2330 if (priv->udpsrc_v4[i]) {
2332 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2333 * values. This is only relevant for PLAY pipelines */
2334 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2335 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2338 gst_bin_add (bin, priv->udpsrc_v4[i]);
2340 /* and link to the funnel v4 */
2341 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2342 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2343 gst_pad_link (pad, selpad);
2344 gst_object_unref (pad);
2345 gst_object_unref (selpad);
2348 if (priv->udpsrc_v6[i]) {
2350 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2351 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2353 gst_bin_add (bin, priv->udpsrc_v6[i]);
2355 /* and link to the funnel v6 */
2356 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2357 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2358 gst_pad_link (pad, selpad);
2359 gst_object_unref (pad);
2360 gst_object_unref (selpad);
2364 /* make and add appsrc */
2365 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2366 priv->appsrc_base_time[i] = -1;
2367 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2368 gst_bin_add (bin, priv->appsrc[i]);
2369 /* and link to the funnel */
2370 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2371 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2372 gst_pad_link (pad, selpad);
2373 gst_object_unref (pad);
2374 gst_object_unref (selpad);
2378 /* check if we need to set to a special state */
2379 if (state != GST_STATE_NULL) {
2380 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2381 gst_element_set_state (priv->funnel[i], state);
2382 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2383 gst_element_set_state (priv->appsrc[i], state);
2389 * gst_rtsp_stream_join_bin:
2390 * @stream: a #GstRTSPStream
2391 * @bin: (transfer none): a #GstBin to join
2392 * @rtpbin: (transfer none): a rtpbin element in @bin
2393 * @state: the target state of the new elements
2395 * Join the #GstBin @bin that contains the element @rtpbin.
2397 * @stream will link to @rtpbin, which must be inside @bin. The elements
2398 * added to @bin will be set to the state given in @state.
2400 * Returns: %TRUE on success.
2403 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2404 GstElement * rtpbin, GstState state)
2406 GstRTSPStreamPrivate *priv;
2409 GstPadLinkReturn ret;
2412 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2413 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2414 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2416 priv = stream->priv;
2418 g_mutex_lock (&priv->lock);
2419 if (priv->is_joined)
2422 /* create a session with the same index as the stream */
2425 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2427 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2428 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2430 if (is_udp && !alloc_ports (stream))
2433 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2434 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2436 g_signal_connect (rtpbin, "request-rtp-encoder",
2437 (GCallback) request_rtp_encoder, stream);
2438 g_signal_connect (rtpbin, "request-rtcp-encoder",
2439 (GCallback) request_rtcp_encoder, stream);
2440 g_signal_connect (rtpbin, "request-rtp-decoder",
2441 (GCallback) request_rtp_rtcp_decoder, stream);
2442 g_signal_connect (rtpbin, "request-rtcp-decoder",
2443 (GCallback) request_rtp_rtcp_decoder, stream);
2446 if (priv->sinkpad) {
2447 g_signal_connect (rtpbin, "request-pt-map",
2448 (GCallback) request_pt_map, stream);
2451 /* get pads from the RTP session element for sending and receiving
2454 /* get a pad for sending RTP */
2455 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2456 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2459 /* link the RTP pad to the session manager, it should not really fail unless
2460 * this is not really an RTP pad */
2461 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2462 if (ret != GST_PAD_LINK_OK)
2465 name = g_strdup_printf ("send_rtp_src_%u", idx);
2466 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2469 /* Need to connect our sinkpad from here */
2470 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2472 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2474 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2475 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2479 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2480 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2482 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2483 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2486 /* get the session */
2487 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2489 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2491 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2493 g_signal_connect (priv->session, "on-ssrc-active",
2494 (GCallback) on_ssrc_active, stream);
2495 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2497 g_signal_connect (priv->session, "on-bye-timeout",
2498 (GCallback) on_bye_timeout, stream);
2499 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2502 /* signal for sender ssrc */
2503 g_signal_connect (priv->session, "on-new-sender-ssrc",
2504 (GCallback) on_new_sender_ssrc, stream);
2505 g_signal_connect (priv->session, "on-sender-ssrc-active",
2506 (GCallback) on_sender_ssrc_active, stream);
2508 create_sender_part (stream, bin, state);
2510 create_receiver_part (stream, bin, state);
2513 /* be notified of caps changes */
2514 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2515 (GCallback) caps_notify, stream);
2518 priv->is_joined = TRUE;
2519 g_mutex_unlock (&priv->lock);
2526 g_mutex_unlock (&priv->lock);
2531 g_mutex_unlock (&priv->lock);
2532 GST_WARNING ("failed to allocate ports %u", idx);
2537 GST_WARNING ("failed to link stream %u", idx);
2538 gst_object_unref (priv->send_rtp_sink);
2539 priv->send_rtp_sink = NULL;
2540 g_mutex_unlock (&priv->lock);
2546 * gst_rtsp_stream_leave_bin:
2547 * @stream: a #GstRTSPStream
2548 * @bin: (transfer none): a #GstBin
2549 * @rtpbin: (transfer none): a rtpbin #GstElement
2551 * Remove the elements of @stream from @bin.
2553 * Return: %TRUE on success.
2556 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2557 GstElement * rtpbin)
2559 GstRTSPStreamPrivate *priv;
2562 gboolean is_tcp, is_udp;
2564 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2565 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2566 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2568 priv = stream->priv;
2570 g_mutex_lock (&priv->lock);
2571 if (!priv->is_joined)
2572 goto was_not_joined;
2574 /* all transports must be removed by now */
2575 if (priv->transports != NULL)
2576 goto transports_not_removed;
2578 clear_tr_cache (priv, TRUE);
2579 clear_tr_cache (priv, FALSE);
2581 GST_INFO ("stream %p leaving bin", stream);
2584 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2586 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2587 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2588 gst_object_unref (priv->send_rtp_sink);
2589 priv->send_rtp_sink = NULL;
2590 } else if (priv->recv_rtp_src) {
2591 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2592 gst_object_unref (priv->recv_rtp_src);
2593 priv->recv_rtp_src = NULL;
2596 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2598 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2599 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2602 for (i = 0; i < 2; i++) {
2603 if (priv->udpsink[i])
2604 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2605 if (priv->appsink[i])
2606 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2607 if (priv->appqueue[i])
2608 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2609 if (priv->udpqueue[i])
2610 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2612 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2613 if (priv->funnel[i])
2614 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2615 if (priv->appsrc[i])
2616 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2618 if (priv->udpsrc_v4[i]) {
2619 if (priv->sinkpad || i == 1) {
2620 /* and set udpsrc to NULL now before removing */
2621 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2622 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2623 /* removing them should also nicely release the request
2624 * pads when they finalize */
2625 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2627 /* we need to set the state to NULL before unref */
2628 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2629 gst_object_unref (priv->udpsrc_v4[i]);
2633 if (priv->udpsrc_v6[i]) {
2634 if (priv->sinkpad || i == 1) {
2635 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2636 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2637 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2639 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2640 gst_object_unref (priv->udpsrc_v6[i]);
2644 for (l = priv->transport_sources; l; l = l->next) {
2645 GstRTSPMulticastTransportSource *s = l->data;
2650 gst_element_set_locked_state (s->udpsrc[i], FALSE);
2651 gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
2652 gst_bin_remove (bin, s->udpsrc[i]);
2655 if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
2656 gst_bin_remove (bin, priv->udpsink[i]);
2657 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2658 gst_bin_remove (bin, priv->appsrc[i]);
2659 if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
2660 gst_bin_remove (bin, priv->appsink[i]);
2661 if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2662 gst_bin_remove (bin, priv->appqueue[i]);
2663 if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2664 gst_bin_remove (bin, priv->udpqueue[i]);
2665 if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2666 gst_bin_remove (bin, priv->tee[i]);
2667 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2668 gst_bin_remove (bin, priv->funnel[i]);
2670 if (priv->sinkpad || i == 1) {
2671 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2672 gst_object_unref (priv->recv_sink[i]);
2673 priv->recv_sink[i] = NULL;
2676 priv->udpsrc_v4[i] = NULL;
2677 priv->udpsrc_v6[i] = NULL;
2678 priv->udpsink[i] = NULL;
2679 priv->appsrc[i] = NULL;
2680 priv->appsink[i] = NULL;
2681 priv->appqueue[i] = NULL;
2682 priv->udpqueue[i] = NULL;
2683 priv->tee[i] = NULL;
2684 priv->funnel[i] = NULL;
2687 for (l = priv->transport_sources; l; l = l->next) {
2688 GstRTSPMulticastTransportSource *s = l->data;
2689 g_slice_free (GstRTSPMulticastTransportSource, s);
2691 g_list_free (priv->transport_sources);
2692 priv->transport_sources = NULL;
2695 gst_object_unref (priv->send_src[0]);
2696 priv->send_src[0] = NULL;
2699 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2700 gst_object_unref (priv->send_src[1]);
2701 priv->send_src[1] = NULL;
2703 g_object_unref (priv->session);
2704 priv->session = NULL;
2706 gst_caps_unref (priv->caps);
2710 gst_object_unref (priv->srtpenc);
2712 gst_object_unref (priv->srtpdec);
2714 priv->is_joined = FALSE;
2715 g_mutex_unlock (&priv->lock);
2721 g_mutex_unlock (&priv->lock);
2724 transports_not_removed:
2726 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2727 g_mutex_unlock (&priv->lock);
2733 * gst_rtsp_stream_get_rtpinfo:
2734 * @stream: a #GstRTSPStream
2735 * @rtptime: (allow-none): result RTP timestamp
2736 * @seq: (allow-none): result RTP seqnum
2737 * @clock_rate: (allow-none): the clock rate
2738 * @running_time: (allow-none): result running-time
2740 * Retrieve the current rtptime, seq and running-time. This is used to
2741 * construct a RTPInfo reply header.
2743 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2746 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2747 guint * rtptime, guint * seq, guint * clock_rate,
2748 GstClockTime * running_time)
2750 GstRTSPStreamPrivate *priv;
2751 GstStructure *stats;
2752 GObjectClass *payobjclass;
2754 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2756 priv = stream->priv;
2758 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2760 g_mutex_lock (&priv->lock);
2762 /* First try to extract the information from the last buffer on the sinks.
2763 * This will have a more accurate sequence number and timestamp, as between
2764 * the payloader and the sink there can be some queues
2766 if (priv->udpsink[0] || priv->appsink[0]) {
2767 GstSample *last_sample;
2769 if (priv->udpsink[0])
2770 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
2772 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
2777 GstSegment *segment;
2778 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
2780 caps = gst_sample_get_caps (last_sample);
2781 buffer = gst_sample_get_buffer (last_sample);
2782 segment = gst_sample_get_segment (last_sample);
2784 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
2786 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
2790 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
2793 gst_rtp_buffer_unmap (&rtp_buffer);
2797 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
2798 GST_BUFFER_TIMESTAMP (buffer));
2802 GstStructure *s = gst_caps_get_structure (caps, 0);
2804 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
2806 if (*clock_rate == 0 && running_time)
2807 *running_time = GST_CLOCK_TIME_NONE;
2809 gst_sample_unref (last_sample);
2813 gst_sample_unref (last_sample);
2818 if (g_object_class_find_property (payobjclass, "stats")) {
2819 g_object_get (priv->payloader, "stats", &stats, NULL);
2824 gst_structure_get_uint (stats, "seqnum", seq);
2827 gst_structure_get_uint (stats, "timestamp", rtptime);
2830 gst_structure_get_clock_time (stats, "running-time", running_time);
2833 gst_structure_get_uint (stats, "clock-rate", clock_rate);
2834 if (*clock_rate == 0 && running_time)
2835 *running_time = GST_CLOCK_TIME_NONE;
2837 gst_structure_free (stats);
2839 if (!g_object_class_find_property (payobjclass, "seqnum") ||
2840 !g_object_class_find_property (payobjclass, "timestamp"))
2844 g_object_get (priv->payloader, "seqnum", seq, NULL);
2847 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
2850 *running_time = GST_CLOCK_TIME_NONE;
2854 g_mutex_unlock (&priv->lock);
2861 GST_WARNING ("Could not get payloader stats");
2862 g_mutex_unlock (&priv->lock);
2868 * gst_rtsp_stream_get_caps:
2869 * @stream: a #GstRTSPStream
2871 * Retrieve the current caps of @stream.
2873 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
2877 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
2879 GstRTSPStreamPrivate *priv;
2882 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2884 priv = stream->priv;
2886 g_mutex_lock (&priv->lock);
2887 if ((result = priv->caps))
2888 gst_caps_ref (result);
2889 g_mutex_unlock (&priv->lock);
2895 * gst_rtsp_stream_recv_rtp:
2896 * @stream: a #GstRTSPStream
2897 * @buffer: (transfer full): a #GstBuffer
2899 * Handle an RTP buffer for the stream. This method is usually called when a
2900 * message has been received from a client using the TCP transport.
2902 * This function takes ownership of @buffer.
2904 * Returns: a GstFlowReturn.
2907 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
2909 GstRTSPStreamPrivate *priv;
2911 GstElement *element;
2913 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2914 priv = stream->priv;
2915 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2916 g_return_val_if_fail (priv->is_joined, FALSE);
2918 g_mutex_lock (&priv->lock);
2919 if (priv->appsrc[0])
2920 element = gst_object_ref (priv->appsrc[0]);
2923 g_mutex_unlock (&priv->lock);
2926 if (priv->appsrc_base_time[0] == -1) {
2927 /* Take current running_time. This timestamp will be put on
2928 * the first buffer of each stream because we are a live source and so we
2929 * timestamp with the running_time. When we are dealing with TCP, we also
2930 * only timestamp the first buffer (using the DISCONT flag) because a server
2931 * typically bursts data, for which we don't want to compensate by speeding
2932 * up the media. The other timestamps will be interpollated from this one
2933 * using the RTP timestamps. */
2934 GST_OBJECT_LOCK (element);
2935 if (GST_ELEMENT_CLOCK (element)) {
2937 GstClockTime base_time;
2939 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
2940 base_time = GST_ELEMENT_CAST (element)->base_time;
2942 priv->appsrc_base_time[0] = now - base_time;
2943 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
2944 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
2945 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
2946 GST_TIME_ARGS (base_time));
2948 GST_OBJECT_UNLOCK (element);
2951 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
2952 gst_object_unref (element);
2960 * gst_rtsp_stream_recv_rtcp:
2961 * @stream: a #GstRTSPStream
2962 * @buffer: (transfer full): a #GstBuffer
2964 * Handle an RTCP buffer for the stream. This method is usually called when a
2965 * message has been received from a client using the TCP transport.
2967 * This function takes ownership of @buffer.
2969 * Returns: a GstFlowReturn.
2972 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
2974 GstRTSPStreamPrivate *priv;
2976 GstElement *element;
2978 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
2979 priv = stream->priv;
2980 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2982 if (!priv->is_joined) {
2983 gst_buffer_unref (buffer);
2984 return GST_FLOW_NOT_LINKED;
2986 g_mutex_lock (&priv->lock);
2987 if (priv->appsrc[1])
2988 element = gst_object_ref (priv->appsrc[1]);
2991 g_mutex_unlock (&priv->lock);
2994 if (priv->appsrc_base_time[1] == -1) {
2995 /* Take current running_time. This timestamp will be put on
2996 * the first buffer of each stream because we are a live source and so we
2997 * timestamp with the running_time. When we are dealing with TCP, we also
2998 * only timestamp the first buffer (using the DISCONT flag) because a server
2999 * typically bursts data, for which we don't want to compensate by speeding
3000 * up the media. The other timestamps will be interpollated from this one
3001 * using the RTP timestamps. */
3002 GST_OBJECT_LOCK (element);
3003 if (GST_ELEMENT_CLOCK (element)) {
3005 GstClockTime base_time;
3007 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3008 base_time = GST_ELEMENT_CAST (element)->base_time;
3010 priv->appsrc_base_time[1] = now - base_time;
3011 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
3012 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3013 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3014 GST_TIME_ARGS (base_time));
3016 GST_OBJECT_UNLOCK (element);
3019 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3020 gst_object_unref (element);
3023 gst_buffer_unref (buffer);
3028 /* must be called with lock */
3030 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
3033 GstRTSPStreamPrivate *priv = stream->priv;
3034 const GstRTSPTransport *tr;
3036 tr = gst_rtsp_stream_transport_get_transport (trans);
3038 switch (tr->lower_transport) {
3039 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3041 GstRTSPMulticastTransportSource *source;
3044 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[1])));
3049 GstPad *selpad, *pad;
3051 source = g_slice_new0 (GstRTSPMulticastTransportSource);
3052 source->transport = trans;
3054 for (i = 0; i < 2; i++) {
3056 g_strdup_printf ("udp://%s:%d", tr->destination,
3057 (i == 0) ? tr->port.min : tr->port.max);
3059 gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
3061 g_object_set (source->udpsrc[i], "loop", FALSE, NULL);
3064 /* we set and keep these to playing so that they don't cause NO_PREROLL return
3065 * values. This is only relevant for PLAY pipelines */
3066 gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
3067 gst_element_set_locked_state (source->udpsrc[i], TRUE);
3070 gst_bin_add (bin, source->udpsrc[i]);
3072 /* and link to the funnel v4 */
3073 if (priv->sinkpad || i == 1) {
3074 source->selpad[i] = selpad =
3075 gst_element_get_request_pad (priv->funnel[i], "sink_%u");
3076 pad = gst_element_get_static_pad (source->udpsrc[i], "src");
3077 gst_pad_link (pad, selpad);
3078 gst_object_unref (pad);
3079 gst_object_unref (selpad);
3083 priv->transport_sources =
3084 g_list_prepend (priv->transport_sources, source);
3088 for (l = priv->transport_sources; l; l = l->next) {
3091 if (source->transport == trans) {
3092 priv->transport_sources =
3093 g_list_delete_link (priv->transport_sources, l);
3101 for (i = 0; i < 2; i++) {
3102 /* Will automatically unlink everything */
3103 gst_bin_remove (bin,
3104 GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
3106 gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
3107 gst_object_unref (source->udpsrc[i]);
3109 if (priv->sinkpad || i == 1) {
3110 gst_element_release_request_pad (priv->funnel[i],
3115 g_slice_free (GstRTSPMulticastTransportSource, source);
3119 gst_object_unref (bin);
3121 /* fall through for the generic case */
3123 case GST_RTSP_LOWER_TRANS_UDP:
3129 dest = tr->destination;
3130 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3134 } else if (priv->client_side) {
3135 /* In client side mode the 'destination' is the RTSP server, so send
3137 min = tr->server_port.min;
3138 max = tr->server_port.max;
3140 min = tr->client_port.min;
3141 max = tr->client_port.max;
3146 GST_INFO ("setting ttl-mc %d", ttl);
3147 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
3148 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
3150 GST_INFO ("adding %s:%d-%d", dest, min, max);
3151 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
3152 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
3153 priv->transports = g_list_prepend (priv->transports, trans);
3155 GST_INFO ("removing %s:%d-%d", dest, min, max);
3156 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
3157 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
3158 priv->transports = g_list_remove (priv->transports, trans);
3160 priv->transports_cookie++;
3163 case GST_RTSP_LOWER_TRANS_TCP:
3165 GST_INFO ("adding TCP %s", tr->destination);
3166 priv->transports = g_list_prepend (priv->transports, trans);
3168 GST_INFO ("removing TCP %s", tr->destination);
3169 priv->transports = g_list_remove (priv->transports, trans);
3171 priv->transports_cookie++;
3174 goto unknown_transport;
3181 GST_INFO ("Unknown transport %d", tr->lower_transport);
3188 * gst_rtsp_stream_add_transport:
3189 * @stream: a #GstRTSPStream
3190 * @trans: (transfer none): a #GstRTSPStreamTransport
3192 * Add the transport in @trans to @stream. The media of @stream will
3193 * then also be send to the values configured in @trans.
3195 * @stream must be joined to a bin.
3197 * @trans must contain a valid #GstRTSPTransport.
3199 * Returns: %TRUE if @trans was added
3202 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
3203 GstRTSPStreamTransport * trans)
3205 GstRTSPStreamPrivate *priv;
3208 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3209 priv = stream->priv;
3210 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3211 g_return_val_if_fail (priv->is_joined, FALSE);
3213 g_mutex_lock (&priv->lock);
3214 res = update_transport (stream, trans, TRUE);
3215 g_mutex_unlock (&priv->lock);
3221 * gst_rtsp_stream_remove_transport:
3222 * @stream: a #GstRTSPStream
3223 * @trans: (transfer none): a #GstRTSPStreamTransport
3225 * Remove the transport in @trans from @stream. The media of @stream will
3226 * not be sent to the values configured in @trans.
3228 * @stream must be joined to a bin.
3230 * @trans must contain a valid #GstRTSPTransport.
3232 * Returns: %TRUE if @trans was removed
3235 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
3236 GstRTSPStreamTransport * trans)
3238 GstRTSPStreamPrivate *priv;
3241 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3242 priv = stream->priv;
3243 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3244 g_return_val_if_fail (priv->is_joined, FALSE);
3246 g_mutex_lock (&priv->lock);
3247 res = update_transport (stream, trans, FALSE);
3248 g_mutex_unlock (&priv->lock);
3254 * gst_rtsp_stream_update_crypto:
3255 * @stream: a #GstRTSPStream
3257 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3259 * Update the new crypto information for @ssrc in @stream. If information
3260 * for @ssrc did not exist, it will be added. If information
3261 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3262 * be removed from @stream.
3264 * Returns: %TRUE if @crypto could be updated
3267 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3268 guint ssrc, GstCaps * crypto)
3270 GstRTSPStreamPrivate *priv;
3272 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3273 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3275 priv = stream->priv;
3277 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3279 g_mutex_lock (&priv->lock);
3281 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3282 gst_caps_ref (crypto));
3284 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3285 g_mutex_unlock (&priv->lock);
3291 * gst_rtsp_stream_get_rtp_socket:
3292 * @stream: a #GstRTSPStream
3293 * @family: the socket family
3295 * Get the RTP socket from @stream for a @family.
3297 * @stream must be joined to a bin.
3299 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3300 * socket could be allocated for @family. Unref after usage
3303 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3305 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3309 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3310 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3311 family == G_SOCKET_FAMILY_IPV6, NULL);
3312 g_return_val_if_fail (priv->udpsink[0], NULL);
3314 if (family == G_SOCKET_FAMILY_IPV6)
3319 g_object_get (priv->udpsink[0], name, &socket, NULL);
3325 * gst_rtsp_stream_get_rtcp_socket:
3326 * @stream: a #GstRTSPStream
3327 * @family: the socket family
3329 * Get the RTCP socket from @stream for a @family.
3331 * @stream must be joined to a bin.
3333 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3334 * socket could be allocated for @family. Unref after usage
3337 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3339 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3343 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3344 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3345 family == G_SOCKET_FAMILY_IPV6, NULL);
3346 g_return_val_if_fail (priv->udpsink[1], NULL);
3348 if (family == G_SOCKET_FAMILY_IPV6)
3353 g_object_get (priv->udpsink[1], name, &socket, NULL);
3359 * gst_rtsp_stream_set_seqnum:
3360 * @stream: a #GstRTSPStream
3361 * @seqnum: a new sequence number
3363 * Configure the sequence number in the payloader of @stream to @seqnum.
3366 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3368 GstRTSPStreamPrivate *priv;
3370 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3372 priv = stream->priv;
3374 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3378 * gst_rtsp_stream_get_seqnum:
3379 * @stream: a #GstRTSPStream
3381 * Get the configured sequence number in the payloader of @stream.
3383 * Returns: the sequence number of the payloader.
3386 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3388 GstRTSPStreamPrivate *priv;
3391 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3393 priv = stream->priv;
3395 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3401 * gst_rtsp_stream_transport_filter:
3402 * @stream: a #GstRTSPStream
3403 * @func: (scope call) (allow-none): a callback
3404 * @user_data: (closure): user data passed to @func
3406 * Call @func for each transport managed by @stream. The result value of @func
3407 * determines what happens to the transport. @func will be called with @stream
3408 * locked so no further actions on @stream can be performed from @func.
3410 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3413 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3415 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3416 * will also be added with an additional ref to the result #GList of this
3419 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3421 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3422 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3423 * element in the #GList should be unreffed before the list is freed.
3426 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3427 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3429 GstRTSPStreamPrivate *priv;
3430 GList *result, *walk, *next;
3431 GHashTable *visited = NULL;
3434 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3436 priv = stream->priv;
3440 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3442 g_mutex_lock (&priv->lock);
3444 cookie = priv->transports_cookie;
3445 for (walk = priv->transports; walk; walk = next) {
3446 GstRTSPStreamTransport *trans = walk->data;
3447 GstRTSPFilterResult res;
3450 next = g_list_next (walk);
3453 /* only visit each transport once */
3454 if (g_hash_table_contains (visited, trans))
3457 g_hash_table_add (visited, g_object_ref (trans));
3458 g_mutex_unlock (&priv->lock);
3460 res = func (stream, trans, user_data);
3462 g_mutex_lock (&priv->lock);
3464 res = GST_RTSP_FILTER_REF;
3466 changed = (cookie != priv->transports_cookie);
3469 case GST_RTSP_FILTER_REMOVE:
3470 update_transport (stream, trans, FALSE);
3472 case GST_RTSP_FILTER_REF:
3473 result = g_list_prepend (result, g_object_ref (trans));
3475 case GST_RTSP_FILTER_KEEP:
3482 g_mutex_unlock (&priv->lock);
3485 g_hash_table_unref (visited);
3490 static GstPadProbeReturn
3491 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3493 GstRTSPStreamPrivate *priv;
3494 GstRTSPStream *stream;
3497 priv = stream->priv;
3499 GST_DEBUG_OBJECT (pad, "now blocking");
3501 g_mutex_lock (&priv->lock);
3502 priv->blocking = TRUE;
3503 g_mutex_unlock (&priv->lock);
3505 gst_element_post_message (priv->payloader,
3506 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3507 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3509 return GST_PAD_PROBE_OK;
3513 * gst_rtsp_stream_set_blocked:
3514 * @stream: a #GstRTSPStream
3515 * @blocked: boolean indicating we should block or unblock
3517 * Blocks or unblocks the dataflow on @stream.
3519 * Returns: %TRUE on success
3522 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3524 GstRTSPStreamPrivate *priv;
3526 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3528 priv = stream->priv;
3530 g_mutex_lock (&priv->lock);
3532 priv->blocking = FALSE;
3533 if (priv->blocked_id == 0) {
3534 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3535 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3536 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3537 g_object_ref (stream), g_object_unref);
3540 if (priv->blocked_id != 0) {
3541 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3542 priv->blocked_id = 0;
3543 priv->blocking = FALSE;
3546 g_mutex_unlock (&priv->lock);
3552 * gst_rtsp_stream_is_blocking:
3553 * @stream: a #GstRTSPStream
3555 * Check if @stream is blocking on a #GstBuffer.
3557 * Returns: %TRUE if @stream is blocking
3560 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3562 GstRTSPStreamPrivate *priv;
3565 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3567 priv = stream->priv;
3569 g_mutex_lock (&priv->lock);
3570 result = priv->blocking;
3571 g_mutex_unlock (&priv->lock);
3577 * gst_rtsp_stream_query_position:
3578 * @stream: a #GstRTSPStream
3580 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3581 * the RTP parts of the pipeline and not the RTCP parts.
3583 * Returns: %TRUE if the position could be queried
3586 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3588 GstRTSPStreamPrivate *priv;
3592 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3594 priv = stream->priv;
3596 g_mutex_lock (&priv->lock);
3597 /* depending on the transport type, it should query corresponding sink */
3598 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3599 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3600 sink = priv->udpsink[0];
3602 sink = priv->appsink[0];
3605 gst_object_ref (sink);
3606 g_mutex_unlock (&priv->lock);
3611 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3612 gst_object_unref (sink);
3618 * gst_rtsp_stream_query_stop:
3619 * @stream: a #GstRTSPStream
3621 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3622 * the RTP parts of the pipeline and not the RTCP parts.
3624 * Returns: %TRUE if the stop could be queried
3627 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3629 GstRTSPStreamPrivate *priv;
3634 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3636 priv = stream->priv;
3638 g_mutex_lock (&priv->lock);
3639 /* depending on the transport type, it should query corresponding sink */
3640 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3641 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3642 sink = priv->udpsink[0];
3644 sink = priv->appsink[0];
3647 gst_object_ref (sink);
3648 g_mutex_unlock (&priv->lock);
3653 query = gst_query_new_segment (GST_FORMAT_TIME);
3654 if ((ret = gst_element_query (sink, query))) {
3657 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3658 if (format != GST_FORMAT_TIME)
3661 gst_query_unref (query);
3662 gst_object_unref (sink);